Migrate absl::optional to std::optional
Bug: webrtc:342905193
No-Try: True
Change-Id: Icc968be43b8830038ea9a1f5f604307220457807
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361021
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42911}
diff --git a/api/BUILD.gn b/api/BUILD.gn
index 53e4caa..bf258c5 100644
--- a/api/BUILD.gn
+++ b/api/BUILD.gn
@@ -127,7 +127,6 @@
"../rtc_base/system:rtc_export",
"units:timestamp",
"video:video_rtp_headers",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -147,14 +146,12 @@
"../rtc_base/system:rtc_export",
"units:time_delta",
"units:timestamp",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
rtc_source_set("video_track_source_constraints") {
visibility = [ "*" ]
sources = [ "video_track_source_constraints.h" ]
- deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
}
rtc_library("media_stream_interface") {
@@ -181,7 +178,6 @@
"video:recordable_encoded_frame",
"video:video_frame",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -243,7 +239,6 @@
"../rtc_base:ssl",
"../rtc_base/system:rtc_export",
"//third_party/abseil-cpp/absl/base:core_headers",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -391,7 +386,6 @@
"//third_party/abseil-cpp/absl/functional:any_invocable",
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
# Basically, don't add stuff here. You might break sensitive downstream
# targets like pnacl. API should not depend on anything outside of this
@@ -421,7 +415,6 @@
"units:timestamp",
"video:encoded_frame",
"video:video_frame_metadata",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -437,7 +430,6 @@
"../rtc_base:macromagic",
"../rtc_base/system:rtc_export",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -498,7 +490,6 @@
"video:video_rtp_headers",
"video_codecs:video_codecs_api",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -542,7 +533,6 @@
"video_codecs:scalability_mode",
"//third_party/abseil-cpp/absl/container:inlined_vector",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -634,7 +624,6 @@
"//third_party/abseil-cpp/absl/base:core_headers",
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -649,7 +638,6 @@
":scoped_refptr",
"../rtc_base:checks",
"video:video_frame",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -688,7 +676,6 @@
"transport:bitrate_settings",
"transport:network_control",
"video_codecs:video_codecs_api",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -758,7 +745,6 @@
"environment:environment_factory",
"//third_party/abseil-cpp/absl/base:nullability",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -777,7 +763,6 @@
"../test:fileutils",
"test/pclf:media_configuration",
"units:time_delta",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -824,7 +809,6 @@
"../rtc_base:refcount",
"../rtc_base/system:rtc_export",
"units:timestamp",
- "//third_party/abseil-cpp/absl/types:optional",
"//third_party/abseil-cpp/absl/types:variant",
]
}
@@ -840,7 +824,6 @@
":array_view",
"../rtc_base:stringutils",
"../rtc_base/system:rtc_export",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -875,7 +858,6 @@
"../rtc_base:random",
"units:data_rate",
"//third_party/abseil-cpp/absl/functional:any_invocable",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -905,7 +887,6 @@
"units:data_size",
"units:timestamp",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -1052,7 +1033,6 @@
"//third_party/abseil-cpp/absl/flags:flag",
"//third_party/abseil-cpp/absl/flags:parse",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
}
@@ -1089,7 +1069,6 @@
"../api/units:frequency",
"../rtc_base:stringutils",
"video:video_frame_type",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -1104,7 +1083,6 @@
"video:encoded_image",
"video:video_frame",
"video_codecs:video_codecs_api",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -1143,7 +1121,6 @@
deps = [
"../api:media_stream_interface",
"../test:test_support",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -1160,7 +1137,6 @@
"../rtc_base:refcount",
"../test:test_support",
"//third_party/abseil-cpp/absl/functional:any_invocable",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -1238,7 +1214,6 @@
"../test:test_support",
"units:data_rate",
"video:render_resolution",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -1329,7 +1304,6 @@
"transport:bandwidth_estimation_settings",
"transport:bitrate_settings",
"transport:network_control",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -1414,7 +1388,6 @@
"crypto:frame_encryptor_interface",
"transport/rtp:rtp_source",
"video_codecs:video_codecs_api",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -1428,7 +1401,6 @@
":frame_transformer_interface",
"../api/units:timestamp",
"../test:test_support",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -1443,7 +1415,6 @@
"../test:test_support",
"units:timestamp",
"video:video_frame_metadata",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -1498,7 +1469,6 @@
"../test:test_support",
"video:encoded_image",
"video:video_frame",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -1630,7 +1600,6 @@
"//third_party/abseil-cpp/absl/functional:any_invocable",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
diff --git a/api/audio/BUILD.gn b/api/audio/BUILD.gn
index 1f9e43d..22c3296 100644
--- a/api/audio/BUILD.gn
+++ b/api/audio/BUILD.gn
@@ -20,7 +20,6 @@
"../../rtc_base:checks",
"../../rtc_base:stringutils",
"../task_queue",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -41,7 +40,6 @@
"../../rtc_base:logging",
"../../rtc_base:macromagic",
"../../rtc_base:timeutils",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -83,7 +81,6 @@
"../task_queue",
"//third_party/abseil-cpp/absl/base:nullability",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -93,10 +90,7 @@
"audio_processing_statistics.cc",
"audio_processing_statistics.h",
]
- deps = [
- "../../rtc_base/system:rtc_export",
- "//third_party/abseil-cpp/absl/types:optional",
- ]
+ deps = [ "../../rtc_base/system:rtc_export" ]
}
rtc_library("aec3_config") {
@@ -125,7 +119,6 @@
":echo_control",
"../../modules/audio_processing/aec3",
"../../rtc_base/system:rtc_export",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
diff --git a/api/audio/audio_device.h b/api/audio/audio_device.h
index eb8b7ec..5a2c2d2 100644
--- a/api/audio/audio_device.h
+++ b/api/audio/audio_device.h
@@ -11,7 +11,7 @@
#ifndef API_AUDIO_AUDIO_DEVICE_H_
#define API_AUDIO_AUDIO_DEVICE_H_
-#include "absl/types/optional.h"
+#include <optional>
#include "api/audio/audio_device_defines.h"
#include "api/ref_count.h"
#include "api/scoped_refptr.h"
@@ -169,7 +169,7 @@
// Used to generate RTC stats. If not implemented, RTCAudioPlayoutStats will
// not be present in the stats.
- virtual absl::optional<Stats> GetStats() const { return absl::nullopt; }
+ virtual std::optional<Stats> GetStats() const { return std::nullopt; }
// Only supported on iOS.
#if defined(WEBRTC_IOS)
diff --git a/api/audio/audio_device_defines.h b/api/audio/audio_device_defines.h
index 304b876..d27f732 100644
--- a/api/audio/audio_device_defines.h
+++ b/api/audio/audio_device_defines.h
@@ -14,9 +14,9 @@
#include <stddef.h>
#include <cstdint>
+#include <optional>
#include <string>
-#include "absl/types/optional.h"
#include "rtc_base/strings/string_builder.h"
namespace webrtc {
@@ -57,7 +57,7 @@
uint32_t currentMicLevel,
bool keyPressed,
uint32_t& newMicLevel,
- absl::optional<int64_t> estimatedCaptureTimeNS) { // NOLINT
+ std::optional<int64_t> estimatedCaptureTimeNS) { // NOLINT
// TODO(webrtc:13620) Make the default behaver of the new API to behave as
// the old API. This can be pure virtual if all uses of the old API is
// removed.
diff --git a/api/audio/audio_frame.cc b/api/audio/audio_frame.cc
index a2aa774..ae4380e 100644
--- a/api/audio/audio_frame.cc
+++ b/api/audio/audio_frame.cc
@@ -13,8 +13,8 @@
#include <string.h>
#include <cstdint>
+#include <optional>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/audio/audio_view.h"
#include "api/audio/channel_layout.h"
@@ -62,7 +62,7 @@
vad_activity_ = kVadUnknown;
profile_timestamp_ms_ = 0;
packet_infos_ = RtpPacketInfos();
- absolute_capture_timestamp_ms_ = absl::nullopt;
+ absolute_capture_timestamp_ms_ = std::nullopt;
}
void AudioFrame::UpdateFrame(uint32_t timestamp,
diff --git a/api/audio/audio_frame.h b/api/audio/audio_frame.h
index 5683e8b..9afd516 100644
--- a/api/audio/audio_frame.h
+++ b/api/audio/audio_frame.h
@@ -15,8 +15,8 @@
#include <stdint.h>
#include <array>
+#include <optional>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/audio/audio_view.h"
#include "api/audio/channel_layout.h"
@@ -164,7 +164,7 @@
absolute_capture_timestamp_ms_ = absolute_capture_time_stamp_ms;
}
- absl::optional<int64_t> absolute_capture_timestamp_ms() const {
+ std::optional<int64_t> absolute_capture_timestamp_ms() const {
return absolute_capture_timestamp_ms_;
}
@@ -189,7 +189,7 @@
// Typically used for measuring elapsed time between two different points in
// the audio path. No lock is used to save resources and we are thread safe
// by design.
- // TODO(nisse@webrtc.org): consider using absl::optional.
+ // TODO(nisse@webrtc.org): consider using std::optional.
int64_t profile_timestamp_ms_ = 0;
// Information about packets used to assemble this audio frame. This is needed
@@ -222,7 +222,7 @@
// This is only valid for audio frames captured on this machine. The absolute
// capture timestamp of a received frame is found in `packet_infos_`.
// This timestamp MUST be based on the same clock as rtc::TimeMillis().
- absl::optional<int64_t> absolute_capture_timestamp_ms_;
+ std::optional<int64_t> absolute_capture_timestamp_ms_;
};
} // namespace webrtc
diff --git a/api/audio/audio_processing.h b/api/audio/audio_processing.h
index 2448c9f..dca75f2 100644
--- a/api/audio/audio_processing.h
+++ b/api/audio/audio_processing.h
@@ -24,12 +24,12 @@
#include <array>
#include <cstdint>
#include <memory>
+#include <optional>
#include <string>
#include <utility>
#include "absl/base/nullability.h"
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/audio/audio_processing_statistics.h"
#include "api/audio/echo_control.h"
@@ -931,8 +931,8 @@
rtc::ArrayView<const float> capture_audio) = 0;
struct Metrics {
- absl::optional<double> echo_likelihood;
- absl::optional<double> echo_likelihood_recent_max;
+ std::optional<double> echo_likelihood;
+ std::optional<double> echo_likelihood_recent_max;
};
// Collect current metrics from the echo detector.
diff --git a/api/audio/audio_processing_statistics.h b/api/audio/audio_processing_statistics.h
index 6f77d07..d6f8b6e 100644
--- a/api/audio/audio_processing_statistics.h
+++ b/api/audio/audio_processing_statistics.h
@@ -13,7 +13,8 @@
#include <stdint.h>
-#include "absl/types/optional.h"
+#include <optional>
+
#include "rtc_base/system/rtc_export.h"
namespace webrtc {
@@ -30,16 +31,16 @@
// It is conservative in flagging audio as speech, with low likelihood of
// incorrectly flagging a frame as voice.
// Only reported if voice detection is enabled in AudioProcessing::Config.
- absl::optional<bool> voice_detected;
+ std::optional<bool> voice_detected;
// AEC Statistics.
// ERL = 10log_10(P_far / P_echo)
- absl::optional<double> echo_return_loss;
+ std::optional<double> echo_return_loss;
// ERLE = 10log_10(P_echo / P_out)
- absl::optional<double> echo_return_loss_enhancement;
+ std::optional<double> echo_return_loss_enhancement;
// Fraction of time that the AEC linear filter is divergent, in a 1-second
// non-overlapped aggregation window.
- absl::optional<double> divergent_filter_fraction;
+ std::optional<double> divergent_filter_fraction;
// The delay metrics consists of the delay median and standard deviation. It
// also consists of the fraction of delay estimates that can make the echo
@@ -48,18 +49,18 @@
// second. Note that if there are several clients pulling metrics from
// `GetStatistics()` during a session the first call from any of them will
// change to one second aggregation window for all.
- absl::optional<int32_t> delay_median_ms;
- absl::optional<int32_t> delay_standard_deviation_ms;
+ std::optional<int32_t> delay_median_ms;
+ std::optional<int32_t> delay_standard_deviation_ms;
// Residual echo detector likelihood.
- absl::optional<double> residual_echo_likelihood;
+ std::optional<double> residual_echo_likelihood;
// Maximum residual echo likelihood from the last time period.
- absl::optional<double> residual_echo_likelihood_recent_max;
+ std::optional<double> residual_echo_likelihood_recent_max;
// The instantaneous delay estimate produced in the AEC. The unit is in
// milliseconds and the value is the instantaneous value at the time of the
// call to `GetStatistics()`.
- absl::optional<int32_t> delay_ms;
+ std::optional<int32_t> delay_ms;
};
} // namespace webrtc
diff --git a/api/audio/echo_canceller3_factory.cc b/api/audio/echo_canceller3_factory.cc
index 8422a3e..9b1273d 100644
--- a/api/audio/echo_canceller3_factory.cc
+++ b/api/audio/echo_canceller3_factory.cc
@@ -10,8 +10,8 @@
#include "api/audio/echo_canceller3_factory.h"
#include <memory>
+#include <optional>
-#include "absl/types/optional.h"
#include "api/audio/echo_canceller3_config.h"
#include "api/audio/echo_control.h"
#include "modules/audio_processing/aec3/echo_canceller3.h"
@@ -28,7 +28,7 @@
int num_render_channels,
int num_capture_channels) {
return std::make_unique<EchoCanceller3>(
- config_, /*multichannel_config=*/absl::nullopt, sample_rate_hz,
+ config_, /*multichannel_config=*/std::nullopt, sample_rate_hz,
num_render_channels, num_capture_channels);
}
diff --git a/api/audio_codecs/BUILD.gn b/api/audio_codecs/BUILD.gn
index d91c501..1995c6d 100644
--- a/api/audio_codecs/BUILD.gn
+++ b/api/audio_codecs/BUILD.gn
@@ -48,7 +48,6 @@
"//third_party/abseil-cpp/absl/base:nullability",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -65,7 +64,6 @@
"L16:audio_decoder_L16",
"g711:audio_decoder_g711",
"g722:audio_decoder_g722",
- "//third_party/abseil-cpp/absl/types:optional",
]
defines = []
if (rtc_include_ilbc) {
@@ -99,14 +97,12 @@
"L16:audio_encoder_L16",
"g711:audio_encoder_g711",
"g722:audio_encoder_g722",
- "//third_party/abseil-cpp/absl/types:optional",
]
defines = []
if (rtc_include_ilbc) {
deps += [
"..:field_trials_view",
"ilbc:audio_encoder_ilbc",
- "//third_party/abseil-cpp/absl/types:optional",
]
defines += [ "WEBRTC_USE_BUILTIN_ILBC=1" ]
} else {
@@ -136,7 +132,6 @@
"..:scoped_refptr",
"opus:audio_decoder_multiopus",
"opus:audio_decoder_opus",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -153,6 +148,5 @@
"..:scoped_refptr",
"opus:audio_encoder_multiopus",
"opus:audio_encoder_opus",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
diff --git a/api/audio_codecs/L16/BUILD.gn b/api/audio_codecs/L16/BUILD.gn
index f2cefbd..8ce7122 100644
--- a/api/audio_codecs/L16/BUILD.gn
+++ b/api/audio_codecs/L16/BUILD.gn
@@ -29,7 +29,6 @@
"../../../rtc_base:stringutils",
"../../../rtc_base/system:rtc_export",
"//third_party/abseil-cpp/absl/strings",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -47,6 +46,5 @@
"../../../rtc_base:safe_conversions",
"../../../rtc_base/system:rtc_export",
"//third_party/abseil-cpp/absl/strings",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
diff --git a/api/audio_codecs/L16/audio_decoder_L16.cc b/api/audio_codecs/L16/audio_decoder_L16.cc
index b392523..a84aa1c 100644
--- a/api/audio_codecs/L16/audio_decoder_L16.cc
+++ b/api/audio_codecs/L16/audio_decoder_L16.cc
@@ -11,10 +11,10 @@
#include "api/audio_codecs/L16/audio_decoder_L16.h"
#include <memory>
+#include <optional>
#include <vector>
#include "absl/strings/match.h"
-#include "absl/types/optional.h"
#include "api/audio_codecs/audio_codec_pair_id.h"
#include "api/audio_codecs/audio_decoder.h"
#include "api/audio_codecs/audio_format.h"
@@ -25,7 +25,7 @@
namespace webrtc {
-absl::optional<AudioDecoderL16::Config> AudioDecoderL16::SdpToConfig(
+std::optional<AudioDecoderL16::Config> AudioDecoderL16::SdpToConfig(
const SdpAudioFormat& format) {
Config config;
config.sample_rate_hz = format.clockrate_hz;
@@ -33,7 +33,7 @@
if (absl::EqualsIgnoreCase(format.name, "L16") && config.IsOk()) {
return config;
}
- return absl::nullopt;
+ return std::nullopt;
}
void AudioDecoderL16::AppendSupportedDecoders(
@@ -43,7 +43,7 @@
std::unique_ptr<AudioDecoder> AudioDecoderL16::MakeAudioDecoder(
const Config& config,
- absl::optional<AudioCodecPairId> /*codec_pair_id*/,
+ std::optional<AudioCodecPairId> /*codec_pair_id*/,
const FieldTrialsView* field_trials) {
if (!config.IsOk()) {
return nullptr;
diff --git a/api/audio_codecs/L16/audio_decoder_L16.h b/api/audio_codecs/L16/audio_decoder_L16.h
index 5a01b7d..339527f 100644
--- a/api/audio_codecs/L16/audio_decoder_L16.h
+++ b/api/audio_codecs/L16/audio_decoder_L16.h
@@ -12,9 +12,9 @@
#define API_AUDIO_CODECS_L16_AUDIO_DECODER_L16_H_
#include <memory>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/audio_codecs/audio_codec_pair_id.h"
#include "api/audio_codecs/audio_decoder.h"
#include "api/audio_codecs/audio_format.h"
@@ -36,11 +36,11 @@
int sample_rate_hz = 8000;
int num_channels = 1;
};
- static absl::optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
+ static std::optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
static void AppendSupportedDecoders(std::vector<AudioCodecSpec>* specs);
static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
const Config& config,
- absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
+ std::optional<AudioCodecPairId> codec_pair_id = std::nullopt,
const FieldTrialsView* field_trials = nullptr);
};
diff --git a/api/audio_codecs/L16/audio_encoder_L16.cc b/api/audio_codecs/L16/audio_encoder_L16.cc
index 6435cd0..3bc3241 100644
--- a/api/audio_codecs/L16/audio_encoder_L16.cc
+++ b/api/audio_codecs/L16/audio_encoder_L16.cc
@@ -14,11 +14,11 @@
#include <map>
#include <memory>
+#include <optional>
#include <utility>
#include <vector>
#include "absl/strings/match.h"
-#include "absl/types/optional.h"
#include "api/audio_codecs/audio_codec_pair_id.h"
#include "api/audio_codecs/audio_encoder.h"
#include "api/audio_codecs/audio_format.h"
@@ -32,11 +32,11 @@
namespace webrtc {
-absl::optional<AudioEncoderL16::Config> AudioEncoderL16::SdpToConfig(
+std::optional<AudioEncoderL16::Config> AudioEncoderL16::SdpToConfig(
const SdpAudioFormat& format) {
if (!rtc::IsValueInRangeForNumericType<int>(format.num_channels)) {
RTC_DCHECK_NOTREACHED();
- return absl::nullopt;
+ return std::nullopt;
}
Config config;
config.sample_rate_hz = format.clockrate_hz;
@@ -51,7 +51,7 @@
if (absl::EqualsIgnoreCase(format.name, "L16") && config.IsOk()) {
return config;
}
- return absl::nullopt;
+ return std::nullopt;
}
void AudioEncoderL16::AppendSupportedEncoders(
@@ -70,7 +70,7 @@
std::unique_ptr<AudioEncoder> AudioEncoderL16::MakeAudioEncoder(
const AudioEncoderL16::Config& config,
int payload_type,
- absl::optional<AudioCodecPairId> /*codec_pair_id*/,
+ std::optional<AudioCodecPairId> /*codec_pair_id*/,
const FieldTrialsView* field_trials) {
AudioEncoderPcm16B::Config c;
c.sample_rate_hz = config.sample_rate_hz;
diff --git a/api/audio_codecs/L16/audio_encoder_L16.h b/api/audio_codecs/L16/audio_encoder_L16.h
index 4750984..a104a63 100644
--- a/api/audio_codecs/L16/audio_encoder_L16.h
+++ b/api/audio_codecs/L16/audio_encoder_L16.h
@@ -12,9 +12,9 @@
#define API_AUDIO_CODECS_L16_AUDIO_ENCODER_L16_H_
#include <memory>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/audio_codecs/audio_codec_pair_id.h"
#include "api/audio_codecs/audio_encoder.h"
#include "api/audio_codecs/audio_format.h"
@@ -39,13 +39,13 @@
int num_channels = 1;
int frame_size_ms = 10;
};
- static absl::optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
+ static std::optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs);
static AudioCodecInfo QueryAudioEncoder(const Config& config);
static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
const Config& config,
int payload_type,
- absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
+ std::optional<AudioCodecPairId> codec_pair_id = std::nullopt,
const FieldTrialsView* field_trials = nullptr);
};
diff --git a/api/audio_codecs/audio_decoder.cc b/api/audio_codecs/audio_decoder.cc
index 84a3eb2..23ef2c8 100644
--- a/api/audio_codecs/audio_decoder.cc
+++ b/api/audio_codecs/audio_decoder.cc
@@ -13,10 +13,10 @@
#include <cstddef>
#include <cstdint>
#include <memory>
+#include <optional>
#include <utility>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "rtc_base/buffer.h"
#include "rtc_base/checks.h"
@@ -37,14 +37,14 @@
return ret < 0 ? 0 : static_cast<size_t>(ret);
}
- absl::optional<DecodeResult> Decode(
+ std::optional<DecodeResult> Decode(
rtc::ArrayView<int16_t> decoded) const override {
auto speech_type = AudioDecoder::kSpeech;
const int ret = decoder_->Decode(
payload_.data(), payload_.size(), decoder_->SampleRateHz(),
decoded.size() * sizeof(int16_t), decoded.data(), &speech_type);
- return ret < 0 ? absl::nullopt
- : absl::optional<DecodeResult>(
+ return ret < 0 ? std::nullopt
+ : std::optional<DecodeResult>(
{static_cast<size_t>(ret), speech_type});
}
diff --git a/api/audio_codecs/audio_decoder.h b/api/audio_codecs/audio_decoder.h
index 4113874..21d6056 100644
--- a/api/audio_codecs/audio_decoder.h
+++ b/api/audio_codecs/audio_decoder.h
@@ -15,9 +15,9 @@
#include <stdint.h>
#include <memory>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "rtc_base/buffer.h"
@@ -57,11 +57,11 @@
// Decodes this frame of audio and writes the result in `decoded`.
// `decoded` must be large enough to store as many samples as indicated by a
- // call to Duration() . On success, returns an absl::optional containing the
+ // call to Duration() . On success, returns an std::optional containing the
// total number of samples across all channels, as well as whether the
// decoder produced comfort noise or speech. On failure, returns an empty
- // absl::optional. Decode may be called at most once per frame object.
- virtual absl::optional<DecodeResult> Decode(
+ // std::optional. Decode may be called at most once per frame object.
+ virtual std::optional<DecodeResult> Decode(
rtc::ArrayView<int16_t> decoded) const = 0;
};
diff --git a/api/audio_codecs/audio_decoder_factory.h b/api/audio_codecs/audio_decoder_factory.h
index 282d4d9..a1ef2ca 100644
--- a/api/audio_codecs/audio_decoder_factory.h
+++ b/api/audio_codecs/audio_decoder_factory.h
@@ -12,10 +12,10 @@
#define API_AUDIO_CODECS_AUDIO_DECODER_FACTORY_H_
#include <memory>
+#include <optional>
#include <vector>
#include "absl/base/nullability.h"
-#include "absl/types/optional.h"
#include "api/audio_codecs/audio_codec_pair_id.h"
#include "api/audio_codecs/audio_decoder.h"
#include "api/audio_codecs/audio_format.h"
@@ -49,7 +49,7 @@
[[deprecated("bugs.webrtc.org/356878416 - Use `Create` instead")]] //
virtual std::unique_ptr<AudioDecoder>
MakeAudioDecoder(const SdpAudioFormat& format,
- absl::optional<AudioCodecPairId> codec_pair_id) {
+ std::optional<AudioCodecPairId> codec_pair_id) {
RTC_DCHECK_NOTREACHED();
return nullptr;
}
@@ -59,7 +59,7 @@
virtual absl::Nullable<std::unique_ptr<AudioDecoder>> Create(
const Environment& env,
const SdpAudioFormat& format,
- absl::optional<AudioCodecPairId> codec_pair_id) {
+ std::optional<AudioCodecPairId> codec_pair_id) {
#pragma clang diagnostic push
#pragma clang diagnostic ignored "-Wdeprecated-declarations"
return MakeAudioDecoder(format, codec_pair_id);
diff --git a/api/audio_codecs/audio_decoder_factory_template.h b/api/audio_codecs/audio_decoder_factory_template.h
index f72531c..257a609 100644
--- a/api/audio_codecs/audio_decoder_factory_template.h
+++ b/api/audio_codecs/audio_decoder_factory_template.h
@@ -12,10 +12,10 @@
#define API_AUDIO_CODECS_AUDIO_DECODER_FACTORY_TEMPLATE_H_
#include <memory>
+#include <optional>
#include <type_traits>
#include <vector>
-#include "absl/types/optional.h"
#include "api/audio_codecs/audio_codec_pair_id.h"
#include "api/audio_codecs/audio_decoder.h"
#include "api/audio_codecs/audio_decoder_factory.h"
@@ -40,7 +40,7 @@
static absl::Nullable<std::unique_ptr<AudioDecoder>> MakeAudioDecoder(
const Environment& env,
const SdpAudioFormat& format,
- absl::optional<AudioCodecPairId> codec_pair_id) {
+ std::optional<AudioCodecPairId> codec_pair_id) {
return nullptr;
}
};
@@ -54,13 +54,13 @@
decltype(Trait::MakeAudioDecoder(
std::declval<Environment>(),
std::declval<typename Trait::Config>(),
- std::declval<absl::optional<AudioCodecPairId>>())),
+ std::declval<std::optional<AudioCodecPairId>>())),
std::unique_ptr<AudioDecoder>>>>
absl::Nullable<std::unique_ptr<AudioDecoder>> CreateDecoder(
Rank1,
const Environment& env,
const typename Trait::Config& config,
- absl::optional<AudioCodecPairId> codec_pair_id) {
+ std::optional<AudioCodecPairId> codec_pair_id) {
return Trait::MakeAudioDecoder(env, config, codec_pair_id);
}
@@ -68,13 +68,13 @@
typename = std::enable_if_t<std::is_convertible_v<
decltype(Trait::MakeAudioDecoder(
std::declval<typename Trait::Config>(),
- std::declval<absl::optional<AudioCodecPairId>>())),
+ std::declval<std::optional<AudioCodecPairId>>())),
std::unique_ptr<AudioDecoder>>>>
absl::Nullable<std::unique_ptr<AudioDecoder>> CreateDecoder(
Rank0,
const Environment& env,
const typename Trait::Config& config,
- absl::optional<AudioCodecPairId> codec_pair_id) {
+ std::optional<AudioCodecPairId> codec_pair_id) {
return Trait::MakeAudioDecoder(config, codec_pair_id);
}
@@ -89,16 +89,16 @@
static bool IsSupportedDecoder(const SdpAudioFormat& format) {
auto opt_config = T::SdpToConfig(format);
static_assert(std::is_same<decltype(opt_config),
- absl::optional<typename T::Config>>::value,
+ std::optional<typename T::Config>>::value,
"T::SdpToConfig() must return a value of type "
- "absl::optional<T::Config>");
+ "std::optional<T::Config>");
return opt_config ? true : Helper<Ts...>::IsSupportedDecoder(format);
}
static absl::Nullable<std::unique_ptr<AudioDecoder>> MakeAudioDecoder(
const Environment& env,
const SdpAudioFormat& format,
- absl::optional<AudioCodecPairId> codec_pair_id) {
+ std::optional<AudioCodecPairId> codec_pair_id) {
auto opt_config = T::SdpToConfig(format);
return opt_config.has_value()
? CreateDecoder<T>(Rank1{}, env, *opt_config, codec_pair_id)
@@ -122,7 +122,7 @@
absl::Nullable<std::unique_ptr<AudioDecoder>> Create(
const Environment& env,
const SdpAudioFormat& format,
- absl::optional<AudioCodecPairId> codec_pair_id) override {
+ std::optional<AudioCodecPairId> codec_pair_id) override {
return Helper<Ts...>::MakeAudioDecoder(env, format, codec_pair_id);
}
};
@@ -137,7 +137,7 @@
// // Converts `audio_format` to a ConfigType instance. Returns an empty
// // optional if `audio_format` doesn't correctly specify a decoder of our
// // type.
-// absl::optional<ConfigType> SdpToConfig(const SdpAudioFormat& audio_format);
+// std::optional<ConfigType> SdpToConfig(const SdpAudioFormat& audio_format);
//
// // Appends zero or more AudioCodecSpecs to the list that will be returned
// // by AudioDecoderFactory::GetSupportedDecoders().
@@ -148,11 +148,11 @@
// std::unique_ptr<AudioEncoder> MakeAudioDecoder(
// const Environment& env,
// const ConfigType& config,
-// absl::optional<AudioCodecPairId> codec_pair_id);
+// std::optional<AudioCodecPairId> codec_pair_id);
// or
// std::unique_ptr<AudioDecoder> MakeAudioDecoder(
// const ConfigType& config,
-// absl::optional<AudioCodecPairId> codec_pair_id);
+// std::optional<AudioCodecPairId> codec_pair_id);
//
// ConfigType should be a type that encapsulates all the settings needed to
// create an AudioDecoder. T::Config (where T is the decoder struct) should
diff --git a/api/audio_codecs/audio_encoder.cc b/api/audio_codecs/audio_encoder.cc
index bb558be..de64550 100644
--- a/api/audio_codecs/audio_encoder.cc
+++ b/api/audio_codecs/audio_encoder.cc
@@ -13,9 +13,9 @@
#include <cstddef>
#include <cstdint>
#include <memory>
+#include <optional>
#include <string>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/call/bitrate_allocation.h"
#include "rtc_base/buffer.h"
@@ -96,12 +96,12 @@
}
void AudioEncoder::OnReceivedTargetAudioBitrate(int target_audio_bitrate_bps) {
- OnReceivedUplinkBandwidth(target_audio_bitrate_bps, absl::nullopt);
+ OnReceivedUplinkBandwidth(target_audio_bitrate_bps, std::nullopt);
}
void AudioEncoder::OnReceivedUplinkBandwidth(
int target_audio_bitrate_bps,
- absl::optional<int64_t> bwe_period_ms) {}
+ std::optional<int64_t> bwe_period_ms) {}
void AudioEncoder::OnReceivedUplinkAllocation(BitrateAllocationUpdate update) {
OnReceivedUplinkBandwidth(update.target_bitrate.bps(),
diff --git a/api/audio_codecs/audio_encoder.h b/api/audio_codecs/audio_encoder.h
index 552319b..d84d2b5 100644
--- a/api/audio_codecs/audio_encoder.h
+++ b/api/audio_codecs/audio_encoder.h
@@ -15,12 +15,12 @@
#include <stdint.h>
#include <memory>
+#include <optional>
#include <string>
#include <utility>
#include <vector>
#include "absl/base/attributes.h"
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/call/bitrate_allocation.h"
#include "api/units/data_rate.h"
@@ -39,30 +39,30 @@
// Number of actions taken by the ANA bitrate controller since the start of
// the call. If this value is not set, it indicates that the bitrate
// controller is disabled.
- absl::optional<uint32_t> bitrate_action_counter;
+ std::optional<uint32_t> bitrate_action_counter;
// Number of actions taken by the ANA channel controller since the start of
// the call. If this value is not set, it indicates that the channel
// controller is disabled.
- absl::optional<uint32_t> channel_action_counter;
+ std::optional<uint32_t> channel_action_counter;
// Number of actions taken by the ANA DTX controller since the start of the
// call. If this value is not set, it indicates that the DTX controller is
// disabled.
- absl::optional<uint32_t> dtx_action_counter;
+ std::optional<uint32_t> dtx_action_counter;
// Number of actions taken by the ANA FEC controller since the start of the
// call. If this value is not set, it indicates that the FEC controller is
// disabled.
- absl::optional<uint32_t> fec_action_counter;
+ std::optional<uint32_t> fec_action_counter;
// Number of times the ANA frame length controller decided to increase the
// frame length since the start of the call. If this value is not set, it
// indicates that the frame length controller is disabled.
- absl::optional<uint32_t> frame_length_increase_counter;
+ std::optional<uint32_t> frame_length_increase_counter;
// Number of times the ANA frame length controller decided to decrease the
// frame length since the start of the call. If this value is not set, it
// indicates that the frame length controller is disabled.
- absl::optional<uint32_t> frame_length_decrease_counter;
+ std::optional<uint32_t> frame_length_decrease_counter;
// The uplink packet loss fractions as set by the ANA FEC controller. If this
// value is not set, it indicates that the ANA FEC controller is not active.
- absl::optional<float> uplink_packet_loss_fraction;
+ std::optional<float> uplink_packet_loss_fraction;
};
// This is the interface class for encoders in AudioCoding module. Each codec
@@ -223,7 +223,7 @@
// Provides target audio bitrate and corresponding probing interval of
// the bandwidth estimator to this encoder to allow it to adapt.
virtual void OnReceivedUplinkBandwidth(int target_audio_bitrate_bps,
- absl::optional<int64_t> bwe_period_ms);
+ std::optional<int64_t> bwe_period_ms);
// Provides target audio bitrate and corresponding probing interval of
// the bandwidth estimator to this encoder to allow it to adapt.
@@ -247,15 +247,14 @@
// The range of frame lengths that are supported or nullopt if there's no such
// information. This is used together with the bitrate range to calculate the
// full bitrate range, including overhead.
- virtual absl::optional<std::pair<TimeDelta, TimeDelta>> GetFrameLengthRange()
+ virtual std::optional<std::pair<TimeDelta, TimeDelta>> GetFrameLengthRange()
const = 0;
// The range of payload bitrates that are supported. This is used together
// with the frame length range to calculate the full bitrate range, including
// overhead.
- virtual absl::optional<std::pair<DataRate, DataRate>> GetBitrateRange()
- const {
- return absl::nullopt;
+ virtual std::optional<std::pair<DataRate, DataRate>> GetBitrateRange() const {
+ return std::nullopt;
}
// The maximum number of audio channels supported by WebRTC encoders.
diff --git a/api/audio_codecs/audio_encoder_factory.h b/api/audio_codecs/audio_encoder_factory.h
index 8fe09f9..c73fe22 100644
--- a/api/audio_codecs/audio_encoder_factory.h
+++ b/api/audio_codecs/audio_encoder_factory.h
@@ -12,9 +12,9 @@
#define API_AUDIO_CODECS_AUDIO_ENCODER_FACTORY_H_
#include <memory>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/audio_codecs/audio_codec_pair_id.h"
#include "api/audio_codecs/audio_encoder.h"
#include "api/audio_codecs/audio_format.h"
@@ -44,7 +44,7 @@
// Note: Implementations need to be robust against combinations other than
// one encoder, one decoder getting the same ID; such encoders must still
// work.
- absl::optional<AudioCodecPairId> codec_pair_id;
+ std::optional<AudioCodecPairId> codec_pair_id;
};
// Returns a prioritized list of audio codecs, to use for signaling etc.
@@ -53,7 +53,7 @@
// Returns information about how this format would be encoded, provided it's
// supported. More format and format variations may be supported than those
// returned by GetSupportedEncoders().
- virtual absl::optional<AudioCodecInfo> QueryAudioEncoder(
+ virtual std::optional<AudioCodecInfo> QueryAudioEncoder(
const SdpAudioFormat& format) = 0;
// Creates an AudioEncoder for the specified format.
diff --git a/api/audio_codecs/audio_encoder_factory_template.h b/api/audio_codecs/audio_encoder_factory_template.h
index 6a70c57..8b26330 100644
--- a/api/audio_codecs/audio_encoder_factory_template.h
+++ b/api/audio_codecs/audio_encoder_factory_template.h
@@ -12,11 +12,11 @@
#define API_AUDIO_CODECS_AUDIO_ENCODER_FACTORY_TEMPLATE_H_
#include <memory>
+#include <optional>
#include <type_traits>
#include <vector>
#include "absl/base/nullability.h"
-#include "absl/types/optional.h"
#include "api/audio_codecs/audio_codec_pair_id.h"
#include "api/audio_codecs/audio_encoder.h"
#include "api/audio_codecs/audio_encoder_factory.h"
@@ -36,9 +36,9 @@
template <>
struct Helper<> {
static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs) {}
- static absl::optional<AudioCodecInfo> QueryAudioEncoder(
+ static std::optional<AudioCodecInfo> QueryAudioEncoder(
const SdpAudioFormat& format) {
- return absl::nullopt;
+ return std::nullopt;
}
static absl::Nullable<std::unique_ptr<AudioEncoder>> CreateAudioEncoder(
const Environment& env,
@@ -72,7 +72,7 @@
decltype(Trait::MakeAudioEncoder(
std::declval<typename Trait::Config>(),
int{},
- std::declval<absl::optional<AudioCodecPairId>>())),
+ std::declval<std::optional<AudioCodecPairId>>())),
std::unique_ptr<AudioEncoder>>>>
absl::Nullable<std::unique_ptr<AudioEncoder>> CreateEncoder(
Rank0,
@@ -91,14 +91,14 @@
T::AppendSupportedEncoders(specs);
Helper<Ts...>::AppendSupportedEncoders(specs);
}
- static absl::optional<AudioCodecInfo> QueryAudioEncoder(
+ static std::optional<AudioCodecInfo> QueryAudioEncoder(
const SdpAudioFormat& format) {
auto opt_config = T::SdpToConfig(format);
static_assert(std::is_same<decltype(opt_config),
- absl::optional<typename T::Config>>::value,
+ std::optional<typename T::Config>>::value,
"T::SdpToConfig() must return a value of type "
- "absl::optional<T::Config>");
- return opt_config ? absl::optional<AudioCodecInfo>(
+ "std::optional<T::Config>");
+ return opt_config ? std::optional<AudioCodecInfo>(
T::QueryAudioEncoder(*opt_config))
: Helper<Ts...>::QueryAudioEncoder(format);
}
@@ -123,7 +123,7 @@
return specs;
}
- absl::optional<AudioCodecInfo> QueryAudioEncoder(
+ std::optional<AudioCodecInfo> QueryAudioEncoder(
const SdpAudioFormat& format) override {
return Helper<Ts...>::QueryAudioEncoder(format);
}
@@ -146,7 +146,7 @@
// // Converts `audio_format` to a ConfigType instance. Returns an empty
// // optional if `audio_format` doesn't correctly specify an encoder of our
// // type.
-// absl::optional<ConfigType> SdpToConfig(const SdpAudioFormat& audio_format);
+// std::optional<ConfigType> SdpToConfig(const SdpAudioFormat& audio_format);
//
// // Appends zero or more AudioCodecSpecs to the list that will be returned
// // by AudioEncoderFactory::GetSupportedEncoders().
@@ -166,7 +166,7 @@
// std::unique_ptr<AudioEncoder> MakeAudioEncoder(
// const ConfigType& config,
// int payload_type,
-// absl::optional<AudioCodecPairId> codec_pair_id);
+// std::optional<AudioCodecPairId> codec_pair_id);
//
// ConfigType should be a type that encapsulates all the settings needed to
// create an AudioEncoder. T::Config (where T is the encoder struct) should
diff --git a/api/audio_codecs/builtin_audio_decoder_factory.cc b/api/audio_codecs/builtin_audio_decoder_factory.cc
index 25be26e..ce3b534 100644
--- a/api/audio_codecs/builtin_audio_decoder_factory.cc
+++ b/api/audio_codecs/builtin_audio_decoder_factory.cc
@@ -11,9 +11,9 @@
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include <memory>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/audio_codecs/L16/audio_decoder_L16.h"
#include "api/audio_codecs/audio_codec_pair_id.h"
#include "api/audio_codecs/audio_decoder.h"
@@ -39,8 +39,7 @@
template <typename T>
struct NotAdvertised {
using Config = typename T::Config;
- static absl::optional<Config> SdpToConfig(
- const SdpAudioFormat& audio_format) {
+ static std::optional<Config> SdpToConfig(const SdpAudioFormat& audio_format) {
return T::SdpToConfig(audio_format);
}
static void AppendSupportedDecoders(std::vector<AudioCodecSpec>* specs) {
@@ -48,7 +47,7 @@
}
static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
const Config& config,
- absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt) {
+ std::optional<AudioCodecPairId> codec_pair_id = std::nullopt) {
return T::MakeAudioDecoder(config, codec_pair_id);
}
};
diff --git a/api/audio_codecs/builtin_audio_encoder_factory.cc b/api/audio_codecs/builtin_audio_encoder_factory.cc
index 864d1a6..8d9c62c 100644
--- a/api/audio_codecs/builtin_audio_encoder_factory.cc
+++ b/api/audio_codecs/builtin_audio_encoder_factory.cc
@@ -11,9 +11,9 @@
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
#include <memory>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/audio_codecs/L16/audio_encoder_L16.h"
#include "api/audio_codecs/audio_codec_pair_id.h"
#include "api/audio_codecs/audio_encoder.h"
@@ -40,8 +40,7 @@
template <typename T>
struct NotAdvertised {
using Config = typename T::Config;
- static absl::optional<Config> SdpToConfig(
- const SdpAudioFormat& audio_format) {
+ static std::optional<Config> SdpToConfig(const SdpAudioFormat& audio_format) {
return T::SdpToConfig(audio_format);
}
static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs) {
@@ -53,7 +52,7 @@
static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
const Config& config,
int payload_type,
- absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
+ std::optional<AudioCodecPairId> codec_pair_id = std::nullopt,
const FieldTrialsView* field_trials = nullptr) {
return T::MakeAudioEncoder(config, payload_type, codec_pair_id,
field_trials);
diff --git a/api/audio_codecs/g711/BUILD.gn b/api/audio_codecs/g711/BUILD.gn
index df377e2..6c00e04 100644
--- a/api/audio_codecs/g711/BUILD.gn
+++ b/api/audio_codecs/g711/BUILD.gn
@@ -29,7 +29,6 @@
"../../../rtc_base:stringutils",
"../../../rtc_base/system:rtc_export",
"//third_party/abseil-cpp/absl/strings",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -48,6 +47,5 @@
"../../../rtc_base:safe_conversions",
"../../../rtc_base/system:rtc_export",
"//third_party/abseil-cpp/absl/strings",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
diff --git a/api/audio_codecs/g711/audio_decoder_g711.cc b/api/audio_codecs/g711/audio_decoder_g711.cc
index 28963c5..724e09f 100644
--- a/api/audio_codecs/g711/audio_decoder_g711.cc
+++ b/api/audio_codecs/g711/audio_decoder_g711.cc
@@ -12,10 +12,10 @@
#include <initializer_list>
#include <memory>
+#include <optional>
#include <vector>
#include "absl/strings/match.h"
-#include "absl/types/optional.h"
#include "api/audio_codecs/audio_codec_pair_id.h"
#include "api/audio_codecs/audio_decoder.h"
#include "api/audio_codecs/audio_format.h"
@@ -26,7 +26,7 @@
namespace webrtc {
-absl::optional<AudioDecoderG711::Config> AudioDecoderG711::SdpToConfig(
+std::optional<AudioDecoderG711::Config> AudioDecoderG711::SdpToConfig(
const SdpAudioFormat& format) {
const bool is_pcmu = absl::EqualsIgnoreCase(format.name, "PCMU");
const bool is_pcma = absl::EqualsIgnoreCase(format.name, "PCMA");
@@ -37,11 +37,11 @@
config.num_channels = rtc::dchecked_cast<int>(format.num_channels);
if (!config.IsOk()) {
RTC_DCHECK_NOTREACHED();
- return absl::nullopt;
+ return std::nullopt;
}
return config;
} else {
- return absl::nullopt;
+ return std::nullopt;
}
}
@@ -54,7 +54,7 @@
std::unique_ptr<AudioDecoder> AudioDecoderG711::MakeAudioDecoder(
const Config& config,
- absl::optional<AudioCodecPairId> /*codec_pair_id*/,
+ std::optional<AudioCodecPairId> /*codec_pair_id*/,
const FieldTrialsView* field_trials) {
if (!config.IsOk()) {
RTC_DCHECK_NOTREACHED();
diff --git a/api/audio_codecs/g711/audio_decoder_g711.h b/api/audio_codecs/g711/audio_decoder_g711.h
index 0f7a98d..155c13d 100644
--- a/api/audio_codecs/g711/audio_decoder_g711.h
+++ b/api/audio_codecs/g711/audio_decoder_g711.h
@@ -12,9 +12,9 @@
#define API_AUDIO_CODECS_G711_AUDIO_DECODER_G711_H_
#include <memory>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/audio_codecs/audio_codec_pair_id.h"
#include "api/audio_codecs/audio_decoder.h"
#include "api/audio_codecs/audio_format.h"
@@ -36,11 +36,11 @@
Type type;
int num_channels;
};
- static absl::optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
+ static std::optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
static void AppendSupportedDecoders(std::vector<AudioCodecSpec>* specs);
static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
const Config& config,
- absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
+ std::optional<AudioCodecPairId> codec_pair_id = std::nullopt,
const FieldTrialsView* field_trials = nullptr);
};
diff --git a/api/audio_codecs/g711/audio_encoder_g711.cc b/api/audio_codecs/g711/audio_encoder_g711.cc
index e32aa28..aa00318 100644
--- a/api/audio_codecs/g711/audio_encoder_g711.cc
+++ b/api/audio_codecs/g711/audio_encoder_g711.cc
@@ -15,11 +15,11 @@
#include <initializer_list>
#include <map>
#include <memory>
+#include <optional>
#include <utility>
#include <vector>
#include "absl/strings/match.h"
-#include "absl/types/optional.h"
#include "api/audio_codecs/audio_codec_pair_id.h"
#include "api/audio_codecs/audio_encoder.h"
#include "api/audio_codecs/audio_format.h"
@@ -32,7 +32,7 @@
namespace webrtc {
-absl::optional<AudioEncoderG711::Config> AudioEncoderG711::SdpToConfig(
+std::optional<AudioEncoderG711::Config> AudioEncoderG711::SdpToConfig(
const SdpAudioFormat& format) {
const bool is_pcmu = absl::EqualsIgnoreCase(format.name, "PCMU");
const bool is_pcma = absl::EqualsIgnoreCase(format.name, "PCMA");
@@ -51,11 +51,11 @@
}
if (!config.IsOk()) {
RTC_DCHECK_NOTREACHED();
- return absl::nullopt;
+ return std::nullopt;
}
return config;
} else {
- return absl::nullopt;
+ return std::nullopt;
}
}
@@ -75,7 +75,7 @@
std::unique_ptr<AudioEncoder> AudioEncoderG711::MakeAudioEncoder(
const Config& config,
int payload_type,
- absl::optional<AudioCodecPairId> /*codec_pair_id*/,
+ std::optional<AudioCodecPairId> /*codec_pair_id*/,
const FieldTrialsView* field_trials) {
if (!config.IsOk()) {
RTC_DCHECK_NOTREACHED();
diff --git a/api/audio_codecs/g711/audio_encoder_g711.h b/api/audio_codecs/g711/audio_encoder_g711.h
index 4b3eb84..db39a98 100644
--- a/api/audio_codecs/g711/audio_encoder_g711.h
+++ b/api/audio_codecs/g711/audio_encoder_g711.h
@@ -12,9 +12,9 @@
#define API_AUDIO_CODECS_G711_AUDIO_ENCODER_G711_H_
#include <memory>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/audio_codecs/audio_codec_pair_id.h"
#include "api/audio_codecs/audio_encoder.h"
#include "api/audio_codecs/audio_format.h"
@@ -38,14 +38,14 @@
int num_channels = 1;
int frame_size_ms = 20;
};
- static absl::optional<AudioEncoderG711::Config> SdpToConfig(
+ static std::optional<AudioEncoderG711::Config> SdpToConfig(
const SdpAudioFormat& audio_format);
static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs);
static AudioCodecInfo QueryAudioEncoder(const Config& config);
static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
const Config& config,
int payload_type,
- absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
+ std::optional<AudioCodecPairId> codec_pair_id = std::nullopt,
const FieldTrialsView* field_trials = nullptr);
};
diff --git a/api/audio_codecs/g722/BUILD.gn b/api/audio_codecs/g722/BUILD.gn
index 58b6e24..a3ee0d4 100644
--- a/api/audio_codecs/g722/BUILD.gn
+++ b/api/audio_codecs/g722/BUILD.gn
@@ -36,7 +36,6 @@
"../../../rtc_base:stringutils",
"../../../rtc_base/system:rtc_export",
"//third_party/abseil-cpp/absl/strings",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -55,6 +54,5 @@
"../../../rtc_base:safe_conversions",
"../../../rtc_base/system:rtc_export",
"//third_party/abseil-cpp/absl/strings",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
diff --git a/api/audio_codecs/g722/audio_decoder_g722.cc b/api/audio_codecs/g722/audio_decoder_g722.cc
index b5e5f42..2b8c4f7 100644
--- a/api/audio_codecs/g722/audio_decoder_g722.cc
+++ b/api/audio_codecs/g722/audio_decoder_g722.cc
@@ -11,10 +11,10 @@
#include "api/audio_codecs/g722/audio_decoder_g722.h"
#include <memory>
+#include <optional>
#include <vector>
#include "absl/strings/match.h"
-#include "absl/types/optional.h"
#include "api/audio_codecs/audio_codec_pair_id.h"
#include "api/audio_codecs/audio_decoder.h"
#include "api/audio_codecs/audio_format.h"
@@ -25,14 +25,14 @@
namespace webrtc {
-absl::optional<AudioDecoderG722::Config> AudioDecoderG722::SdpToConfig(
+std::optional<AudioDecoderG722::Config> AudioDecoderG722::SdpToConfig(
const SdpAudioFormat& format) {
if (absl::EqualsIgnoreCase(format.name, "G722") &&
format.clockrate_hz == 8000 &&
(format.num_channels == 1 || format.num_channels == 2)) {
return Config{rtc::dchecked_cast<int>(format.num_channels)};
}
- return absl::nullopt;
+ return std::nullopt;
}
void AudioDecoderG722::AppendSupportedDecoders(
@@ -42,7 +42,7 @@
std::unique_ptr<AudioDecoder> AudioDecoderG722::MakeAudioDecoder(
Config config,
- absl::optional<AudioCodecPairId> /*codec_pair_id*/,
+ std::optional<AudioCodecPairId> /*codec_pair_id*/,
const FieldTrialsView* field_trials) {
if (!config.IsOk()) {
RTC_DCHECK_NOTREACHED();
diff --git a/api/audio_codecs/g722/audio_decoder_g722.h b/api/audio_codecs/g722/audio_decoder_g722.h
index 6f7b253..1859866 100644
--- a/api/audio_codecs/g722/audio_decoder_g722.h
+++ b/api/audio_codecs/g722/audio_decoder_g722.h
@@ -12,9 +12,9 @@
#define API_AUDIO_CODECS_G722_AUDIO_DECODER_G722_H_
#include <memory>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/audio_codecs/audio_codec_pair_id.h"
#include "api/audio_codecs/audio_decoder.h"
#include "api/audio_codecs/audio_format.h"
@@ -30,11 +30,11 @@
bool IsOk() const { return num_channels == 1 || num_channels == 2; }
int num_channels;
};
- static absl::optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
+ static std::optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
static void AppendSupportedDecoders(std::vector<AudioCodecSpec>* specs);
static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
Config config,
- absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
+ std::optional<AudioCodecPairId> codec_pair_id = std::nullopt,
const FieldTrialsView* field_trials = nullptr);
};
diff --git a/api/audio_codecs/g722/audio_encoder_g722.cc b/api/audio_codecs/g722/audio_encoder_g722.cc
index d7a1c89..5ba9598 100644
--- a/api/audio_codecs/g722/audio_encoder_g722.cc
+++ b/api/audio_codecs/g722/audio_encoder_g722.cc
@@ -14,11 +14,11 @@
#include <map>
#include <memory>
+#include <optional>
#include <utility>
#include <vector>
#include "absl/strings/match.h"
-#include "absl/types/optional.h"
#include "api/audio_codecs/audio_codec_pair_id.h"
#include "api/audio_codecs/audio_encoder.h"
#include "api/audio_codecs/audio_format.h"
@@ -32,11 +32,11 @@
namespace webrtc {
-absl::optional<AudioEncoderG722Config> AudioEncoderG722::SdpToConfig(
+std::optional<AudioEncoderG722Config> AudioEncoderG722::SdpToConfig(
const SdpAudioFormat& format) {
if (!absl::EqualsIgnoreCase(format.name, "g722") ||
format.clockrate_hz != 8000) {
- return absl::nullopt;
+ return std::nullopt;
}
AudioEncoderG722Config config;
@@ -51,7 +51,7 @@
}
if (!config.IsOk()) {
RTC_DCHECK_NOTREACHED();
- return absl::nullopt;
+ return std::nullopt;
}
return config;
}
@@ -73,7 +73,7 @@
std::unique_ptr<AudioEncoder> AudioEncoderG722::MakeAudioEncoder(
const AudioEncoderG722Config& config,
int payload_type,
- absl::optional<AudioCodecPairId> /*codec_pair_id*/,
+ std::optional<AudioCodecPairId> /*codec_pair_id*/,
const FieldTrialsView* field_trials) {
if (!config.IsOk()) {
RTC_DCHECK_NOTREACHED();
diff --git a/api/audio_codecs/g722/audio_encoder_g722.h b/api/audio_codecs/g722/audio_encoder_g722.h
index 78ceddd..0997d8b 100644
--- a/api/audio_codecs/g722/audio_encoder_g722.h
+++ b/api/audio_codecs/g722/audio_encoder_g722.h
@@ -12,9 +12,9 @@
#define API_AUDIO_CODECS_G722_AUDIO_ENCODER_G722_H_
#include <memory>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/audio_codecs/audio_codec_pair_id.h"
#include "api/audio_codecs/audio_encoder.h"
#include "api/audio_codecs/audio_format.h"
@@ -28,14 +28,14 @@
// CreateAudioEncoderFactory<...>().
struct RTC_EXPORT AudioEncoderG722 {
using Config = AudioEncoderG722Config;
- static absl::optional<AudioEncoderG722Config> SdpToConfig(
+ static std::optional<AudioEncoderG722Config> SdpToConfig(
const SdpAudioFormat& audio_format);
static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs);
static AudioCodecInfo QueryAudioEncoder(const AudioEncoderG722Config& config);
static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
const AudioEncoderG722Config& config,
int payload_type,
- absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
+ std::optional<AudioCodecPairId> codec_pair_id = std::nullopt,
const FieldTrialsView* field_trials = nullptr);
};
diff --git a/api/audio_codecs/ilbc/BUILD.gn b/api/audio_codecs/ilbc/BUILD.gn
index a71867a..8ab719f 100644
--- a/api/audio_codecs/ilbc/BUILD.gn
+++ b/api/audio_codecs/ilbc/BUILD.gn
@@ -34,7 +34,6 @@
"../../../rtc_base:safe_minmax",
"../../../rtc_base:stringutils",
"//third_party/abseil-cpp/absl/strings",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -50,6 +49,5 @@
"../../../api:field_trials_view",
"../../../modules/audio_coding:ilbc",
"//third_party/abseil-cpp/absl/strings",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
diff --git a/api/audio_codecs/ilbc/audio_decoder_ilbc.cc b/api/audio_codecs/ilbc/audio_decoder_ilbc.cc
index 6b3c0cb..4b1c7de 100644
--- a/api/audio_codecs/ilbc/audio_decoder_ilbc.cc
+++ b/api/audio_codecs/ilbc/audio_decoder_ilbc.cc
@@ -11,10 +11,10 @@
#include "api/audio_codecs/ilbc/audio_decoder_ilbc.h"
#include <memory>
+#include <optional>
#include <vector>
#include "absl/strings/match.h"
-#include "absl/types/optional.h"
#include "api/audio_codecs/audio_codec_pair_id.h"
#include "api/audio_codecs/audio_decoder.h"
#include "api/audio_codecs/audio_format.h"
@@ -23,13 +23,13 @@
namespace webrtc {
-absl::optional<AudioDecoderIlbc::Config> AudioDecoderIlbc::SdpToConfig(
+std::optional<AudioDecoderIlbc::Config> AudioDecoderIlbc::SdpToConfig(
const SdpAudioFormat& format) {
if (absl::EqualsIgnoreCase(format.name, "ILBC") &&
format.clockrate_hz == 8000 && format.num_channels == 1) {
return Config();
}
- return absl::nullopt;
+ return std::nullopt;
}
void AudioDecoderIlbc::AppendSupportedDecoders(
@@ -39,7 +39,7 @@
std::unique_ptr<AudioDecoder> AudioDecoderIlbc::MakeAudioDecoder(
Config config,
- absl::optional<AudioCodecPairId> /*codec_pair_id*/,
+ std::optional<AudioCodecPairId> /*codec_pair_id*/,
const FieldTrialsView* field_trials) {
return std::make_unique<AudioDecoderIlbcImpl>();
}
diff --git a/api/audio_codecs/ilbc/audio_decoder_ilbc.h b/api/audio_codecs/ilbc/audio_decoder_ilbc.h
index 60566c8..f1a89b9 100644
--- a/api/audio_codecs/ilbc/audio_decoder_ilbc.h
+++ b/api/audio_codecs/ilbc/audio_decoder_ilbc.h
@@ -12,9 +12,9 @@
#define API_AUDIO_CODECS_ILBC_AUDIO_DECODER_ILBC_H_
#include <memory>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/audio_codecs/audio_codec_pair_id.h"
#include "api/audio_codecs/audio_decoder.h"
#include "api/audio_codecs/audio_format.h"
@@ -26,11 +26,11 @@
// CreateAudioDecoderFactory<...>().
struct AudioDecoderIlbc {
struct Config {}; // Empty---no config values needed!
- static absl::optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
+ static std::optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
static void AppendSupportedDecoders(std::vector<AudioCodecSpec>* specs);
static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
Config config,
- absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
+ std::optional<AudioCodecPairId> codec_pair_id = std::nullopt,
const FieldTrialsView* field_trials = nullptr);
};
diff --git a/api/audio_codecs/ilbc/audio_encoder_ilbc.cc b/api/audio_codecs/ilbc/audio_encoder_ilbc.cc
index 7345591..5fac14b 100644
--- a/api/audio_codecs/ilbc/audio_encoder_ilbc.cc
+++ b/api/audio_codecs/ilbc/audio_encoder_ilbc.cc
@@ -12,12 +12,12 @@
#include <map>
#include <memory>
+#include <optional>
#include <string>
#include <utility>
#include <vector>
#include "absl/strings/match.h"
-#include "absl/types/optional.h"
#include "api/audio_codecs/audio_codec_pair_id.h"
#include "api/audio_codecs/audio_encoder.h"
#include "api/audio_codecs/audio_format.h"
@@ -46,11 +46,11 @@
}
} // namespace
-absl::optional<AudioEncoderIlbcConfig> AudioEncoderIlbc::SdpToConfig(
+std::optional<AudioEncoderIlbcConfig> AudioEncoderIlbc::SdpToConfig(
const SdpAudioFormat& format) {
if (!absl::EqualsIgnoreCase(format.name.c_str(), "ILBC") ||
format.clockrate_hz != 8000 || format.num_channels != 1) {
- return absl::nullopt;
+ return std::nullopt;
}
AudioEncoderIlbcConfig config;
@@ -64,7 +64,7 @@
}
if (!config.IsOk()) {
RTC_DCHECK_NOTREACHED();
- return absl::nullopt;
+ return std::nullopt;
}
return config;
}
@@ -85,7 +85,7 @@
std::unique_ptr<AudioEncoder> AudioEncoderIlbc::MakeAudioEncoder(
const AudioEncoderIlbcConfig& config,
int payload_type,
- absl::optional<AudioCodecPairId> /*codec_pair_id*/,
+ std::optional<AudioCodecPairId> /*codec_pair_id*/,
const FieldTrialsView* field_trials) {
if (!config.IsOk()) {
RTC_DCHECK_NOTREACHED();
diff --git a/api/audio_codecs/ilbc/audio_encoder_ilbc.h b/api/audio_codecs/ilbc/audio_encoder_ilbc.h
index a530684..74c75fd 100644
--- a/api/audio_codecs/ilbc/audio_encoder_ilbc.h
+++ b/api/audio_codecs/ilbc/audio_encoder_ilbc.h
@@ -12,9 +12,9 @@
#define API_AUDIO_CODECS_ILBC_AUDIO_ENCODER_ILBC_H_
#include <memory>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/audio_codecs/audio_codec_pair_id.h"
#include "api/audio_codecs/audio_encoder.h"
#include "api/audio_codecs/audio_format.h"
@@ -27,14 +27,14 @@
// CreateAudioEncoderFactory<...>().
struct AudioEncoderIlbc {
using Config = AudioEncoderIlbcConfig;
- static absl::optional<AudioEncoderIlbcConfig> SdpToConfig(
+ static std::optional<AudioEncoderIlbcConfig> SdpToConfig(
const SdpAudioFormat& audio_format);
static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs);
static AudioCodecInfo QueryAudioEncoder(const AudioEncoderIlbcConfig& config);
static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
const AudioEncoderIlbcConfig& config,
int payload_type,
- absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
+ std::optional<AudioCodecPairId> codec_pair_id = std::nullopt,
const FieldTrialsView* field_trials = nullptr);
};
diff --git a/api/audio_codecs/opus/BUILD.gn b/api/audio_codecs/opus/BUILD.gn
index f96256e..29b2999 100644
--- a/api/audio_codecs/opus/BUILD.gn
+++ b/api/audio_codecs/opus/BUILD.gn
@@ -20,10 +20,7 @@
"audio_encoder_opus_config.cc",
"audio_encoder_opus_config.h",
]
- deps = [
- "../../../rtc_base/system:rtc_export",
- "//third_party/abseil-cpp/absl/types:optional",
- ]
+ deps = [ "../../../rtc_base/system:rtc_export" ]
defines = []
if (rtc_opus_variable_complexity) {
defines += [ "WEBRTC_OPUS_VARIABLE_COMPLEXITY=1" ]
@@ -50,7 +47,6 @@
"../../../rtc_base:checks",
"../../../rtc_base/system:rtc_export",
"../../environment",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -68,7 +64,6 @@
"../../../rtc_base/system:rtc_export",
"../../environment",
"//third_party/abseil-cpp/absl/strings",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -83,7 +78,6 @@
"../../../modules/audio_coding:webrtc_multiopus",
"../../../rtc_base/system:rtc_export",
"../opus:audio_encoder_opus_config",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -102,6 +96,5 @@
"../../../rtc_base/system:rtc_export",
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/strings",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
diff --git a/api/audio_codecs/opus/audio_decoder_multi_channel_opus.cc b/api/audio_codecs/opus/audio_decoder_multi_channel_opus.cc
index 497e212..a9381c0 100644
--- a/api/audio_codecs/opus/audio_decoder_multi_channel_opus.cc
+++ b/api/audio_codecs/opus/audio_decoder_multi_channel_opus.cc
@@ -11,10 +11,10 @@
#include "api/audio_codecs/opus/audio_decoder_multi_channel_opus.h"
#include <memory>
+#include <optional>
#include <utility>
#include <vector>
-#include "absl/types/optional.h"
#include "api/audio_codecs/audio_codec_pair_id.h"
#include "api/audio_codecs/audio_decoder.h"
#include "api/audio_codecs/audio_format.h"
@@ -24,7 +24,7 @@
namespace webrtc {
-absl::optional<AudioDecoderMultiChannelOpusConfig>
+std::optional<AudioDecoderMultiChannelOpusConfig>
AudioDecoderMultiChannelOpus::SdpToConfig(const SdpAudioFormat& format) {
return AudioDecoderMultiChannelOpusImpl::SdpToConfig(format);
}
@@ -68,7 +68,7 @@
std::unique_ptr<AudioDecoder> AudioDecoderMultiChannelOpus::MakeAudioDecoder(
AudioDecoderMultiChannelOpusConfig config,
- absl::optional<AudioCodecPairId> /*codec_pair_id*/,
+ std::optional<AudioCodecPairId> /*codec_pair_id*/,
const FieldTrialsView* field_trials) {
return AudioDecoderMultiChannelOpusImpl::MakeAudioDecoder(config);
}
diff --git a/api/audio_codecs/opus/audio_decoder_multi_channel_opus.h b/api/audio_codecs/opus/audio_decoder_multi_channel_opus.h
index eafd6c6..d9fc693 100644
--- a/api/audio_codecs/opus/audio_decoder_multi_channel_opus.h
+++ b/api/audio_codecs/opus/audio_decoder_multi_channel_opus.h
@@ -12,9 +12,9 @@
#define API_AUDIO_CODECS_OPUS_AUDIO_DECODER_MULTI_CHANNEL_OPUS_H_
#include <memory>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/audio_codecs/audio_codec_pair_id.h"
#include "api/audio_codecs/audio_decoder.h"
#include "api/audio_codecs/audio_format.h"
@@ -28,12 +28,12 @@
// CreateAudioDecoderFactory<...>().
struct RTC_EXPORT AudioDecoderMultiChannelOpus {
using Config = AudioDecoderMultiChannelOpusConfig;
- static absl::optional<AudioDecoderMultiChannelOpusConfig> SdpToConfig(
+ static std::optional<AudioDecoderMultiChannelOpusConfig> SdpToConfig(
const SdpAudioFormat& audio_format);
static void AppendSupportedDecoders(std::vector<AudioCodecSpec>* specs);
static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
AudioDecoderMultiChannelOpusConfig config,
- absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
+ std::optional<AudioCodecPairId> codec_pair_id = std::nullopt,
const FieldTrialsView* field_trials = nullptr);
};
diff --git a/api/audio_codecs/opus/audio_decoder_opus.cc b/api/audio_codecs/opus/audio_decoder_opus.cc
index c19cffc..4db27a3 100644
--- a/api/audio_codecs/opus/audio_decoder_opus.cc
+++ b/api/audio_codecs/opus/audio_decoder_opus.cc
@@ -12,12 +12,12 @@
#include <map>
#include <memory>
+#include <optional>
#include <string>
#include <utility>
#include <vector>
#include "absl/strings/match.h"
-#include "absl/types/optional.h"
#include "api/audio_codecs/audio_codec_pair_id.h"
#include "api/audio_codecs/audio_decoder.h"
#include "api/audio_codecs/audio_format.h"
@@ -38,9 +38,9 @@
return true;
}
-absl::optional<AudioDecoderOpus::Config> AudioDecoderOpus::SdpToConfig(
+std::optional<AudioDecoderOpus::Config> AudioDecoderOpus::SdpToConfig(
const SdpAudioFormat& format) {
- const auto num_channels = [&]() -> absl::optional<int> {
+ const auto num_channels = [&]() -> std::optional<int> {
auto stereo = format.parameters.find("stereo");
if (stereo != format.parameters.end()) {
if (stereo->second == "0") {
@@ -48,7 +48,7 @@
} else if (stereo->second == "1") {
return 2;
} else {
- return absl::nullopt; // Bad stereo parameter.
+ return std::nullopt; // Bad stereo parameter.
}
}
return 1; // Default to mono.
@@ -60,11 +60,11 @@
config.num_channels = *num_channels;
if (!config.IsOk()) {
RTC_DCHECK_NOTREACHED();
- return absl::nullopt;
+ return std::nullopt;
}
return config;
} else {
- return absl::nullopt;
+ return std::nullopt;
}
}
diff --git a/api/audio_codecs/opus/audio_decoder_opus.h b/api/audio_codecs/opus/audio_decoder_opus.h
index c2d5fda..802de98 100644
--- a/api/audio_codecs/opus/audio_decoder_opus.h
+++ b/api/audio_codecs/opus/audio_decoder_opus.h
@@ -12,9 +12,9 @@
#define API_AUDIO_CODECS_OPUS_AUDIO_DECODER_OPUS_H_
#include <memory>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/audio_codecs/audio_codec_pair_id.h"
#include "api/audio_codecs/audio_decoder.h"
#include "api/audio_codecs/audio_format.h"
@@ -31,7 +31,7 @@
int sample_rate_hz = 48000;
int num_channels = 1;
};
- static absl::optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
+ static std::optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
static void AppendSupportedDecoders(std::vector<AudioCodecSpec>* specs);
static std::unique_ptr<AudioDecoder> MakeAudioDecoder(const Environment& env,
@@ -39,7 +39,7 @@
static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
const Environment& env,
Config config,
- absl::optional<AudioCodecPairId> /*codec_pair_id*/) {
+ std::optional<AudioCodecPairId> /*codec_pair_id*/) {
return MakeAudioDecoder(env, config);
}
};
diff --git a/api/audio_codecs/opus/audio_encoder_multi_channel_opus.cc b/api/audio_codecs/opus/audio_encoder_multi_channel_opus.cc
index 14f480b..ef0699b 100644
--- a/api/audio_codecs/opus/audio_encoder_multi_channel_opus.cc
+++ b/api/audio_codecs/opus/audio_encoder_multi_channel_opus.cc
@@ -16,7 +16,7 @@
namespace webrtc {
-absl::optional<AudioEncoderMultiChannelOpusConfig>
+std::optional<AudioEncoderMultiChannelOpusConfig>
AudioEncoderMultiChannelOpus::SdpToConfig(const SdpAudioFormat& format) {
return AudioEncoderMultiChannelOpusImpl::SdpToConfig(format);
}
@@ -66,7 +66,7 @@
std::unique_ptr<AudioEncoder> AudioEncoderMultiChannelOpus::MakeAudioEncoder(
const AudioEncoderMultiChannelOpusConfig& config,
int payload_type,
- absl::optional<AudioCodecPairId> /*codec_pair_id*/,
+ std::optional<AudioCodecPairId> /*codec_pair_id*/,
const FieldTrialsView* field_trials) {
return AudioEncoderMultiChannelOpusImpl::MakeAudioEncoder(config,
payload_type);
diff --git a/api/audio_codecs/opus/audio_encoder_multi_channel_opus.h b/api/audio_codecs/opus/audio_encoder_multi_channel_opus.h
index c1c4db3..923e5c0 100644
--- a/api/audio_codecs/opus/audio_encoder_multi_channel_opus.h
+++ b/api/audio_codecs/opus/audio_encoder_multi_channel_opus.h
@@ -12,9 +12,9 @@
#define API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_MULTI_CHANNEL_OPUS_H_
#include <memory>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/audio_codecs/audio_codec_pair_id.h"
#include "api/audio_codecs/audio_encoder.h"
#include "api/audio_codecs/audio_format.h"
@@ -28,13 +28,13 @@
// CreateAudioEncoderFactory<...>().
struct RTC_EXPORT AudioEncoderMultiChannelOpus {
using Config = AudioEncoderMultiChannelOpusConfig;
- static absl::optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
+ static std::optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs);
static AudioCodecInfo QueryAudioEncoder(const Config& config);
static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
const Config& config,
int payload_type,
- absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
+ std::optional<AudioCodecPairId> codec_pair_id = std::nullopt,
const FieldTrialsView* field_trials = nullptr);
};
diff --git a/api/audio_codecs/opus/audio_encoder_opus.cc b/api/audio_codecs/opus/audio_encoder_opus.cc
index f099aeb..89033ea 100644
--- a/api/audio_codecs/opus/audio_encoder_opus.cc
+++ b/api/audio_codecs/opus/audio_encoder_opus.cc
@@ -11,9 +11,9 @@
#include "api/audio_codecs/opus/audio_encoder_opus.h"
#include <memory>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/audio_codecs/audio_codec_pair_id.h"
#include "api/audio_codecs/audio_encoder.h"
#include "api/audio_codecs/audio_encoder_factory.h"
@@ -24,7 +24,7 @@
namespace webrtc {
-absl::optional<AudioEncoderOpusConfig> AudioEncoderOpus::SdpToConfig(
+std::optional<AudioEncoderOpusConfig> AudioEncoderOpus::SdpToConfig(
const SdpAudioFormat& format) {
return AudioEncoderOpusImpl::SdpToConfig(format);
}
diff --git a/api/audio_codecs/opus/audio_encoder_opus.h b/api/audio_codecs/opus/audio_encoder_opus.h
index d3e769e..1a7a4ca 100644
--- a/api/audio_codecs/opus/audio_encoder_opus.h
+++ b/api/audio_codecs/opus/audio_encoder_opus.h
@@ -12,9 +12,9 @@
#define API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_
#include <memory>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/audio_codecs/audio_encoder.h"
#include "api/audio_codecs/audio_encoder_factory.h"
#include "api/audio_codecs/audio_format.h"
@@ -28,7 +28,7 @@
// CreateAudioEncoderFactory<...>().
struct RTC_EXPORT AudioEncoderOpus {
using Config = AudioEncoderOpusConfig;
- static absl::optional<AudioEncoderOpusConfig> SdpToConfig(
+ static std::optional<AudioEncoderOpusConfig> SdpToConfig(
const SdpAudioFormat& audio_format);
static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs);
static AudioCodecInfo QueryAudioEncoder(const AudioEncoderOpusConfig& config);
diff --git a/api/audio_codecs/opus/audio_encoder_opus_config.h b/api/audio_codecs/opus/audio_encoder_opus_config.h
index d5d7256..1fe5c18 100644
--- a/api/audio_codecs/opus/audio_encoder_opus_config.h
+++ b/api/audio_codecs/opus/audio_encoder_opus_config.h
@@ -13,9 +13,9 @@
#include <stddef.h>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "rtc_base/system/rtc_export.h"
namespace webrtc {
@@ -43,7 +43,7 @@
// NOTE: This member must always be set.
// TODO(kwiberg): Turn it into just an int.
- absl::optional<int> bitrate_bps;
+ std::optional<int> bitrate_bps;
bool fec_enabled;
bool cbr_enabled;
diff --git a/api/audio_codecs/opus_audio_decoder_factory.cc b/api/audio_codecs/opus_audio_decoder_factory.cc
index cf7bda2..c3d5df4 100644
--- a/api/audio_codecs/opus_audio_decoder_factory.cc
+++ b/api/audio_codecs/opus_audio_decoder_factory.cc
@@ -11,9 +11,9 @@
#include "api/audio_codecs/opus_audio_decoder_factory.h"
#include <memory>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/audio_codecs/audio_codec_pair_id.h"
#include "api/audio_codecs/audio_decoder.h"
#include "api/audio_codecs/audio_decoder_factory.h"
@@ -31,8 +31,7 @@
template <typename T>
struct NotAdvertised {
using Config = typename T::Config;
- static absl::optional<Config> SdpToConfig(
- const SdpAudioFormat& audio_format) {
+ static std::optional<Config> SdpToConfig(const SdpAudioFormat& audio_format) {
return T::SdpToConfig(audio_format);
}
static void AppendSupportedDecoders(std::vector<AudioCodecSpec>* specs) {
@@ -40,7 +39,7 @@
}
static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
const Config& config,
- absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt) {
+ std::optional<AudioCodecPairId> codec_pair_id = std::nullopt) {
return T::MakeAudioDecoder(config, codec_pair_id);
}
};
diff --git a/api/audio_codecs/opus_audio_encoder_factory.cc b/api/audio_codecs/opus_audio_encoder_factory.cc
index 8d1d150..164b7a1 100644
--- a/api/audio_codecs/opus_audio_encoder_factory.cc
+++ b/api/audio_codecs/opus_audio_encoder_factory.cc
@@ -11,9 +11,9 @@
#include "api/audio_codecs/opus_audio_encoder_factory.h"
#include <memory>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/audio_codecs/audio_codec_pair_id.h"
#include "api/audio_codecs/audio_encoder.h"
#include "api/audio_codecs/audio_encoder_factory.h"
@@ -31,8 +31,7 @@
template <typename T>
struct NotAdvertised {
using Config = typename T::Config;
- static absl::optional<Config> SdpToConfig(
- const SdpAudioFormat& audio_format) {
+ static std::optional<Config> SdpToConfig(const SdpAudioFormat& audio_format) {
return T::SdpToConfig(audio_format);
}
static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs) {
@@ -44,7 +43,7 @@
static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
const Config& config,
int payload_type,
- absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
+ std::optional<AudioCodecPairId> codec_pair_id = std::nullopt,
const FieldTrialsView* field_trials = nullptr) {
return T::MakeAudioEncoder(config, payload_type, codec_pair_id,
field_trials);
diff --git a/api/audio_codecs/test/BUILD.gn b/api/audio_codecs/test/BUILD.gn
index 0ed352c..e895884 100644
--- a/api/audio_codecs/test/BUILD.gn
+++ b/api/audio_codecs/test/BUILD.gn
@@ -37,7 +37,6 @@
"../ilbc:audio_encoder_ilbc",
"../opus:audio_decoder_opus",
"../opus:audio_encoder_opus",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
}
diff --git a/api/audio_codecs/test/audio_decoder_factory_template_unittest.cc b/api/audio_codecs/test/audio_decoder_factory_template_unittest.cc
index 134e0f5..7994836 100644
--- a/api/audio_codecs/test/audio_decoder_factory_template_unittest.cc
+++ b/api/audio_codecs/test/audio_decoder_factory_template_unittest.cc
@@ -11,10 +11,10 @@
#include "api/audio_codecs/audio_decoder_factory_template.h"
#include <memory>
+#include <optional>
#include <utility>
#include <vector>
-#include "absl/types/optional.h"
#include "api/audio_codecs/L16/audio_decoder_L16.h"
#include "api/audio_codecs/audio_codec_pair_id.h"
#include "api/audio_codecs/audio_decoder.h"
@@ -59,13 +59,12 @@
SdpAudioFormat audio_format;
};
- static absl::optional<Config> SdpToConfig(
- const SdpAudioFormat& audio_format) {
+ static std::optional<Config> SdpToConfig(const SdpAudioFormat& audio_format) {
if (Params::AudioFormat() == audio_format) {
Config config = {audio_format};
return config;
} else {
- return absl::nullopt;
+ return std::nullopt;
}
}
@@ -79,7 +78,7 @@
static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
const Config&,
- absl::optional<AudioCodecPairId> /*codec_pair_id*/ = absl::nullopt) {
+ std::optional<AudioCodecPairId> /*codec_pair_id*/ = std::nullopt) {
auto dec = std::make_unique<testing::StrictMock<MockAudioDecoder>>();
EXPECT_CALL(*dec, SampleRateHz())
.WillOnce(::testing::Return(Params::CodecInfo().sample_rate_hz));
@@ -95,8 +94,7 @@
static SdpAudioFormat AudioFormat() { return {"fake", 16'000, 2, {}}; }
- static absl::optional<Config> SdpToConfig(
- const SdpAudioFormat& audio_format) {
+ static std::optional<Config> SdpToConfig(const SdpAudioFormat& audio_format) {
return Config();
}
@@ -114,7 +112,7 @@
static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
const Config& config,
- absl::optional<AudioCodecPairId> codec_pair_id) {
+ std::optional<AudioCodecPairId> codec_pair_id) {
auto decoder = std::make_unique<NiceMock<MockAudioDecoder>>();
ON_CALL(*decoder, SampleRateHz).WillByDefault(Return(kRateWithoutEnv));
return decoder;
@@ -123,7 +121,7 @@
static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
const Environment& env,
const Config& config,
- absl::optional<AudioCodecPairId> codec_pair_id) {
+ std::optional<AudioCodecPairId> codec_pair_id) {
auto decoder = std::make_unique<NiceMock<MockAudioDecoder>>();
ON_CALL(*decoder, SampleRateHz).WillByDefault(Return(kRateWithEnv));
return decoder;
@@ -143,7 +141,7 @@
struct AudioDecoderApiWithV1Make : BaseAudioDecoderApi {
static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
const Config& config,
- absl::optional<AudioCodecPairId> codec_pair_id) {
+ std::optional<AudioCodecPairId> codec_pair_id) {
return std::make_unique<NiceMock<MockAudioDecoder>>();
}
};
@@ -163,7 +161,7 @@
audio_decoder_factory_template_impl::AudioDecoderFactoryT<>>());
EXPECT_THAT(factory->GetSupportedDecoders(), ::testing::IsEmpty());
EXPECT_FALSE(factory->IsSupportedDecoder({"foo", 8000, 1}));
- EXPECT_EQ(nullptr, factory->Create(env, {"bar", 16000, 1}, absl::nullopt));
+ EXPECT_EQ(nullptr, factory->Create(env, {"bar", 16000, 1}, std::nullopt));
}
TEST(AudioDecoderFactoryTemplateTest, OneDecoderType) {
@@ -174,8 +172,8 @@
AudioCodecSpec{{"bogus", 8000, 1}, {8000, 1, 12345}}));
EXPECT_FALSE(factory->IsSupportedDecoder({"foo", 8000, 1}));
EXPECT_TRUE(factory->IsSupportedDecoder({"bogus", 8000, 1}));
- EXPECT_EQ(nullptr, factory->Create(env, {"bar", 16000, 1}, absl::nullopt));
- auto dec = factory->Create(env, {"bogus", 8000, 1}, absl::nullopt);
+ EXPECT_EQ(nullptr, factory->Create(env, {"bar", 16000, 1}, std::nullopt));
+ auto dec = factory->Create(env, {"bogus", 8000, 1}, std::nullopt);
ASSERT_NE(nullptr, dec);
EXPECT_EQ(8000, dec->SampleRateHz());
}
@@ -193,13 +191,13 @@
EXPECT_TRUE(factory->IsSupportedDecoder({"bogus", 8000, 1}));
EXPECT_TRUE(
factory->IsSupportedDecoder({"sham", 16000, 2, {{"param", "value"}}}));
- EXPECT_EQ(nullptr, factory->Create(env, {"bar", 16000, 1}, absl::nullopt));
- auto dec1 = factory->Create(env, {"bogus", 8000, 1}, absl::nullopt);
+ EXPECT_EQ(nullptr, factory->Create(env, {"bar", 16000, 1}, std::nullopt));
+ auto dec1 = factory->Create(env, {"bogus", 8000, 1}, std::nullopt);
ASSERT_NE(nullptr, dec1);
EXPECT_EQ(8000, dec1->SampleRateHz());
- EXPECT_EQ(nullptr, factory->Create(env, {"sham", 16000, 2}, absl::nullopt));
+ EXPECT_EQ(nullptr, factory->Create(env, {"sham", 16000, 2}, std::nullopt));
auto dec2 = factory->Create(env, {"sham", 16000, 2, {{"param", "value"}}},
- absl::nullopt);
+ std::nullopt);
ASSERT_NE(nullptr, dec2);
EXPECT_EQ(16000, dec2->SampleRateHz());
}
@@ -214,11 +212,11 @@
EXPECT_FALSE(factory->IsSupportedDecoder({"G711", 8000, 1}));
EXPECT_TRUE(factory->IsSupportedDecoder({"PCMU", 8000, 1}));
EXPECT_TRUE(factory->IsSupportedDecoder({"pcma", 8000, 1}));
- EXPECT_EQ(nullptr, factory->Create(env, {"pcmu", 16000, 1}, absl::nullopt));
- auto dec1 = factory->Create(env, {"pcmu", 8000, 1}, absl::nullopt);
+ EXPECT_EQ(nullptr, factory->Create(env, {"pcmu", 16000, 1}, std::nullopt));
+ auto dec1 = factory->Create(env, {"pcmu", 8000, 1}, std::nullopt);
ASSERT_NE(nullptr, dec1);
EXPECT_EQ(8000, dec1->SampleRateHz());
- auto dec2 = factory->Create(env, {"PCMA", 8000, 1}, absl::nullopt);
+ auto dec2 = factory->Create(env, {"PCMA", 8000, 1}, std::nullopt);
ASSERT_NE(nullptr, dec2);
EXPECT_EQ(8000, dec2->SampleRateHz());
}
@@ -231,16 +229,16 @@
AudioCodecSpec{{"G722", 8000, 1}, {16000, 1, 64000}}));
EXPECT_FALSE(factory->IsSupportedDecoder({"foo", 8000, 1}));
EXPECT_TRUE(factory->IsSupportedDecoder({"G722", 8000, 1}));
- EXPECT_EQ(nullptr, factory->Create(env, {"bar", 16000, 1}, absl::nullopt));
- auto dec1 = factory->Create(env, {"G722", 8000, 1}, absl::nullopt);
+ EXPECT_EQ(nullptr, factory->Create(env, {"bar", 16000, 1}, std::nullopt));
+ auto dec1 = factory->Create(env, {"G722", 8000, 1}, std::nullopt);
ASSERT_NE(nullptr, dec1);
EXPECT_EQ(16000, dec1->SampleRateHz());
EXPECT_EQ(1u, dec1->Channels());
- auto dec2 = factory->Create(env, {"G722", 8000, 2}, absl::nullopt);
+ auto dec2 = factory->Create(env, {"G722", 8000, 2}, std::nullopt);
ASSERT_NE(nullptr, dec2);
EXPECT_EQ(16000, dec2->SampleRateHz());
EXPECT_EQ(2u, dec2->Channels());
- auto dec3 = factory->Create(env, {"G722", 8000, 3}, absl::nullopt);
+ auto dec3 = factory->Create(env, {"G722", 8000, 3}, std::nullopt);
ASSERT_EQ(nullptr, dec3);
}
@@ -252,8 +250,8 @@
AudioCodecSpec{{"ILBC", 8000, 1}, {8000, 1, 13300}}));
EXPECT_FALSE(factory->IsSupportedDecoder({"foo", 8000, 1}));
EXPECT_TRUE(factory->IsSupportedDecoder({"ilbc", 8000, 1}));
- EXPECT_EQ(nullptr, factory->Create(env, {"bar", 8000, 1}, absl::nullopt));
- auto dec = factory->Create(env, {"ilbc", 8000, 1}, absl::nullopt);
+ EXPECT_EQ(nullptr, factory->Create(env, {"bar", 8000, 1}, std::nullopt));
+ auto dec = factory->Create(env, {"ilbc", 8000, 1}, std::nullopt);
ASSERT_NE(nullptr, dec);
EXPECT_EQ(8000, dec->SampleRateHz());
}
@@ -273,8 +271,8 @@
EXPECT_FALSE(factory->IsSupportedDecoder({"foo", 8000, 1}));
EXPECT_TRUE(factory->IsSupportedDecoder({"L16", 48000, 1}));
EXPECT_FALSE(factory->IsSupportedDecoder({"L16", 96000, 1}));
- EXPECT_EQ(nullptr, factory->Create(env, {"L16", 8000, 0}, absl::nullopt));
- auto dec = factory->Create(env, {"L16", 48000, 2}, absl::nullopt);
+ EXPECT_EQ(nullptr, factory->Create(env, {"L16", 8000, 0}, std::nullopt));
+ auto dec = factory->Create(env, {"L16", 48000, 2}, std::nullopt);
ASSERT_NE(nullptr, dec);
EXPECT_EQ(48000, dec->SampleRateHz());
}
@@ -291,8 +289,8 @@
::testing::ElementsAre(AudioCodecSpec{opus_format, opus_info}));
EXPECT_FALSE(factory->IsSupportedDecoder({"opus", 48000, 1}));
EXPECT_TRUE(factory->IsSupportedDecoder({"opus", 48000, 2}));
- EXPECT_EQ(nullptr, factory->Create(env, {"bar", 16000, 1}, absl::nullopt));
- auto dec = factory->Create(env, {"opus", 48000, 2}, absl::nullopt);
+ EXPECT_EQ(nullptr, factory->Create(env, {"bar", 16000, 1}, std::nullopt));
+ auto dec = factory->Create(env, {"opus", 48000, 2}, std::nullopt);
ASSERT_NE(nullptr, dec);
EXPECT_EQ(48000, dec->SampleRateHz());
}
diff --git a/api/audio_codecs/test/audio_encoder_factory_template_unittest.cc b/api/audio_codecs/test/audio_encoder_factory_template_unittest.cc
index 92277f5..ec60d91 100644
--- a/api/audio_codecs/test/audio_encoder_factory_template_unittest.cc
+++ b/api/audio_codecs/test/audio_encoder_factory_template_unittest.cc
@@ -11,10 +11,10 @@
#include "api/audio_codecs/audio_encoder_factory_template.h"
#include <memory>
+#include <optional>
#include <utility>
#include <vector>
-#include "absl/types/optional.h"
#include "api/audio_codecs/L16/audio_encoder_L16.h"
#include "api/audio_codecs/audio_codec_pair_id.h"
#include "api/audio_codecs/audio_encoder.h"
@@ -59,13 +59,12 @@
SdpAudioFormat audio_format;
};
- static absl::optional<Config> SdpToConfig(
- const SdpAudioFormat& audio_format) {
+ static std::optional<Config> SdpToConfig(const SdpAudioFormat& audio_format) {
if (Params::AudioFormat() == audio_format) {
Config config = {audio_format};
return config;
} else {
- return absl::nullopt;
+ return std::nullopt;
}
}
@@ -80,7 +79,7 @@
static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
const Config&,
int payload_type,
- absl::optional<AudioCodecPairId> /*codec_pair_id*/ = absl::nullopt) {
+ std::optional<AudioCodecPairId> /*codec_pair_id*/ = std::nullopt) {
auto enc = std::make_unique<testing::StrictMock<MockAudioEncoder>>();
EXPECT_CALL(*enc, SampleRateHz())
.WillOnce(::testing::Return(Params::CodecInfo().sample_rate_hz));
@@ -101,8 +100,7 @@
static SdpAudioFormat AudioFormat() { return {"fake", 16'000, 2, {}}; }
static AudioCodecInfo CodecInfo() { return {16'000, 2, 23456}; }
- static absl::optional<Config> SdpToConfig(
- const SdpAudioFormat& audio_format) {
+ static std::optional<Config> SdpToConfig(const SdpAudioFormat& audio_format) {
return Config();
}
@@ -117,7 +115,7 @@
static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
const Config&,
int payload_type,
- absl::optional<AudioCodecPairId> codec_pair_id) {
+ std::optional<AudioCodecPairId> codec_pair_id) {
auto encoder = std::make_unique<NiceMock<MockAudioEncoder>>();
ON_CALL(*encoder, SampleRateHz).WillByDefault(Return(kV1SameRate));
return encoder;
@@ -139,7 +137,7 @@
static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
const Config&,
int payload_type,
- absl::optional<AudioCodecPairId> codec_pair_id) {
+ std::optional<AudioCodecPairId> codec_pair_id) {
auto encoder = std::make_unique<NiceMock<MockAudioEncoder>>();
ON_CALL(*encoder, SampleRateHz).WillByDefault(Return(kV1SameRate));
return encoder;
@@ -190,7 +188,7 @@
rtc::make_ref_counted<
audio_encoder_factory_template_impl::AudioEncoderFactoryT<>>());
EXPECT_THAT(factory->GetSupportedEncoders(), ::testing::IsEmpty());
- EXPECT_EQ(absl::nullopt, factory->QueryAudioEncoder({"foo", 8000, 1}));
+ EXPECT_EQ(std::nullopt, factory->QueryAudioEncoder({"foo", 8000, 1}));
EXPECT_THAT(factory->Create(env, {"bar", 16000, 1}, {}), IsNull());
}
@@ -201,7 +199,7 @@
EXPECT_THAT(factory->GetSupportedEncoders(),
::testing::ElementsAre(
AudioCodecSpec{{"bogus", 8000, 1}, {8000, 1, 12345}}));
- EXPECT_EQ(absl::nullopt, factory->QueryAudioEncoder({"foo", 8000, 1}));
+ EXPECT_EQ(std::nullopt, factory->QueryAudioEncoder({"foo", 8000, 1}));
EXPECT_EQ(AudioCodecInfo(8000, 1, 12345),
factory->QueryAudioEncoder({"bogus", 8000, 1}));
@@ -219,7 +217,7 @@
AudioCodecSpec{{"bogus", 8000, 1}, {8000, 1, 12345}},
AudioCodecSpec{{"sham", 16000, 2, {{"param", "value"}}},
{16000, 2, 23456}}));
- EXPECT_EQ(absl::nullopt, factory->QueryAudioEncoder({"foo", 8000, 1}));
+ EXPECT_EQ(std::nullopt, factory->QueryAudioEncoder({"foo", 8000, 1}));
EXPECT_EQ(AudioCodecInfo(8000, 1, 12345),
factory->QueryAudioEncoder({"bogus", 8000, 1}));
EXPECT_EQ(
@@ -242,7 +240,7 @@
::testing::ElementsAre(
AudioCodecSpec{{"PCMU", 8000, 1}, {8000, 1, 64000}},
AudioCodecSpec{{"PCMA", 8000, 1}, {8000, 1, 64000}}));
- EXPECT_EQ(absl::nullopt, factory->QueryAudioEncoder({"PCMA", 16000, 1}));
+ EXPECT_EQ(std::nullopt, factory->QueryAudioEncoder({"PCMA", 16000, 1}));
EXPECT_EQ(AudioCodecInfo(8000, 1, 64000),
factory->QueryAudioEncoder({"PCMA", 8000, 1}));
@@ -259,7 +257,7 @@
EXPECT_THAT(factory->GetSupportedEncoders(),
::testing::ElementsAre(
AudioCodecSpec{{"G722", 8000, 1}, {16000, 1, 64000}}));
- EXPECT_EQ(absl::nullopt, factory->QueryAudioEncoder({"foo", 8000, 1}));
+ EXPECT_EQ(std::nullopt, factory->QueryAudioEncoder({"foo", 8000, 1}));
EXPECT_EQ(AudioCodecInfo(16000, 1, 64000),
factory->QueryAudioEncoder({"G722", 8000, 1}));
@@ -274,7 +272,7 @@
EXPECT_THAT(factory->GetSupportedEncoders(),
::testing::ElementsAre(
AudioCodecSpec{{"ILBC", 8000, 1}, {8000, 1, 13333}}));
- EXPECT_EQ(absl::nullopt, factory->QueryAudioEncoder({"foo", 8000, 1}));
+ EXPECT_EQ(std::nullopt, factory->QueryAudioEncoder({"foo", 8000, 1}));
EXPECT_EQ(AudioCodecInfo(8000, 1, 13333),
factory->QueryAudioEncoder({"ilbc", 8000, 1}));
@@ -295,7 +293,7 @@
AudioCodecSpec{{"L16", 8000, 2}, {8000, 2, 8000 * 16 * 2}},
AudioCodecSpec{{"L16", 16000, 2}, {16000, 2, 16000 * 16 * 2}},
AudioCodecSpec{{"L16", 32000, 2}, {32000, 2, 32000 * 16 * 2}}));
- EXPECT_EQ(absl::nullopt, factory->QueryAudioEncoder({"L16", 8000, 0}));
+ EXPECT_EQ(std::nullopt, factory->QueryAudioEncoder({"L16", 8000, 0}));
EXPECT_EQ(AudioCodecInfo(48000, 1, 48000 * 16),
factory->QueryAudioEncoder({"L16", 48000, 1}));
@@ -315,7 +313,7 @@
::testing::ElementsAre(AudioCodecSpec{
{"opus", 48000, 2, {{"minptime", "10"}, {"useinbandfec", "1"}}},
info}));
- EXPECT_EQ(absl::nullopt, factory->QueryAudioEncoder({"foo", 8000, 1}));
+ EXPECT_EQ(std::nullopt, factory->QueryAudioEncoder({"foo", 8000, 1}));
EXPECT_EQ(
info,
factory->QueryAudioEncoder(
diff --git a/api/audio_options.cc b/api/audio_options.cc
index a3f2b6e..f5baf06 100644
--- a/api/audio_options.cc
+++ b/api/audio_options.cc
@@ -19,14 +19,14 @@
template <class T>
void ToStringIfSet(rtc::SimpleStringBuilder* result,
const char* key,
- const absl::optional<T>& val) {
+ const std::optional<T>& val) {
if (val) {
(*result) << key << ": " << *val << ", ";
}
}
template <typename T>
-void SetFrom(absl::optional<T>* s, const absl::optional<T>& o) {
+void SetFrom(std::optional<T>* s, const std::optional<T>& o) {
if (o) {
*s = o;
}
diff --git a/api/audio_options.h b/api/audio_options.h
index 2162471..b16808e 100644
--- a/api/audio_options.h
+++ b/api/audio_options.h
@@ -11,9 +11,9 @@
#ifndef API_AUDIO_OPTIONS_H_
#define API_AUDIO_OPTIONS_H_
+#include <optional>
#include <string>
-#include "absl/types/optional.h"
#include "rtc_base/system/rtc_export.h"
namespace cricket {
@@ -34,38 +34,38 @@
// Audio processing that attempts to filter away the output signal from
// later inbound pickup.
- absl::optional<bool> echo_cancellation;
+ std::optional<bool> echo_cancellation;
#if defined(WEBRTC_IOS)
// Forces software echo cancellation on iOS. This is a temporary workaround
// (until Apple fixes the bug) for a device with non-functioning AEC. May
// improve performance on that particular device, but will cause unpredictable
// behavior in all other cases. See http://bugs.webrtc.org/8682.
- absl::optional<bool> ios_force_software_aec_HACK;
+ std::optional<bool> ios_force_software_aec_HACK;
#endif
// Audio processing to adjust the sensitivity of the local mic dynamically.
- absl::optional<bool> auto_gain_control;
+ std::optional<bool> auto_gain_control;
// Audio processing to filter out background noise.
- absl::optional<bool> noise_suppression;
+ std::optional<bool> noise_suppression;
// Audio processing to remove background noise of lower frequencies.
- absl::optional<bool> highpass_filter;
+ std::optional<bool> highpass_filter;
// Audio processing to swap the left and right channels.
- absl::optional<bool> stereo_swapping;
+ std::optional<bool> stereo_swapping;
// Audio receiver jitter buffer (NetEq) max capacity in number of packets.
- absl::optional<int> audio_jitter_buffer_max_packets;
+ std::optional<int> audio_jitter_buffer_max_packets;
// Audio receiver jitter buffer (NetEq) fast accelerate mode.
- absl::optional<bool> audio_jitter_buffer_fast_accelerate;
+ std::optional<bool> audio_jitter_buffer_fast_accelerate;
// Audio receiver jitter buffer (NetEq) minimum target delay in milliseconds.
- absl::optional<int> audio_jitter_buffer_min_delay_ms;
+ std::optional<int> audio_jitter_buffer_min_delay_ms;
// Enable audio network adaptor.
// TODO(webrtc:11717): Remove this API in favor of adaptivePtime in
// RtpEncodingParameters.
- absl::optional<bool> audio_network_adaptor;
+ std::optional<bool> audio_network_adaptor;
// Config string for audio network adaptor.
- absl::optional<std::string> audio_network_adaptor_config;
+ std::optional<std::string> audio_network_adaptor_config;
// Pre-initialize the ADM for recording when starting to send. Default to
// true.
// TODO(webrtc:13566): Remove this option. See issue for details.
- absl::optional<bool> init_recording_on_send;
+ std::optional<bool> init_recording_on_send;
};
} // namespace cricket
diff --git a/api/data_channel_interface.cc b/api/data_channel_interface.cc
index 8e04009..966faf24 100644
--- a/api/data_channel_interface.cc
+++ b/api/data_channel_interface.cc
@@ -27,12 +27,12 @@
return 0;
}
-absl::optional<int> DataChannelInterface::maxRetransmitsOpt() const {
- return absl::nullopt;
+std::optional<int> DataChannelInterface::maxRetransmitsOpt() const {
+ return std::nullopt;
}
-absl::optional<int> DataChannelInterface::maxPacketLifeTime() const {
- return absl::nullopt;
+std::optional<int> DataChannelInterface::maxPacketLifeTime() const {
+ return std::nullopt;
}
std::string DataChannelInterface::protocol() const {
diff --git a/api/data_channel_interface.h b/api/data_channel_interface.h
index 300012e..23936b1 100644
--- a/api/data_channel_interface.h
+++ b/api/data_channel_interface.h
@@ -17,10 +17,10 @@
#include <stddef.h>
#include <stdint.h>
+#include <optional>
#include <string>
#include "absl/functional/any_invocable.h"
-#include "absl/types/optional.h"
#include "api/priority.h"
#include "api/ref_count.h"
#include "api/rtc_error.h"
@@ -31,7 +31,7 @@
namespace webrtc {
// C++ version of: https://www.w3.org/TR/webrtc/#idl-def-rtcdatachannelinit
-// TODO(deadbeef): Use absl::optional for the "-1 if unset" things.
+// TODO(deadbeef): Use std::optional for the "-1 if unset" things.
struct DataChannelInit {
// Deprecated. Reliability is assumed, and channel will be unreliable if
// maxRetransmitTime or MaxRetransmits is set.
@@ -46,13 +46,13 @@
// Cannot be set along with `maxRetransmits`.
// This is called `maxPacketLifeTime` in the WebRTC JS API.
// Negative values are ignored, and positive values are clamped to [0-65535]
- absl::optional<int> maxRetransmitTime;
+ std::optional<int> maxRetransmitTime;
// The max number of retransmissions.
//
// Cannot be set along with `maxRetransmitTime`.
// Negative values are ignored, and positive values are clamped to [0-65535]
- absl::optional<int> maxRetransmits;
+ std::optional<int> maxRetransmits;
// This is set by the application and opaque to the WebRTC implementation.
std::string protocol;
@@ -67,7 +67,7 @@
int id = -1;
// https://w3c.github.io/webrtc-priority/#new-rtcdatachannelinit-member
- absl::optional<PriorityValue> priority;
+ std::optional<PriorityValue> priority;
};
// At the JavaScript level, data can be passed in as a string or a blob, so
@@ -163,8 +163,8 @@
// TODO(hta): Deprecate and remove the following two functions.
virtual uint16_t maxRetransmitTime() const;
virtual uint16_t maxRetransmits() const;
- virtual absl::optional<int> maxRetransmitsOpt() const;
- virtual absl::optional<int> maxPacketLifeTime() const;
+ virtual std::optional<int> maxRetransmitsOpt() const;
+ virtual std::optional<int> maxPacketLifeTime() const;
virtual std::string protocol() const;
virtual bool negotiated() const;
diff --git a/api/dtls_transport_interface.cc b/api/dtls_transport_interface.cc
index 8133525..7a560cf 100644
--- a/api/dtls_transport_interface.cc
+++ b/api/dtls_transport_interface.cc
@@ -11,9 +11,9 @@
#include "api/dtls_transport_interface.h"
#include <memory>
+#include <optional>
#include <utility>
-#include "absl/types/optional.h"
#include "rtc_base/ssl_certificate.h"
namespace webrtc {
@@ -26,10 +26,10 @@
DtlsTransportInformation::DtlsTransportInformation(
DtlsTransportState state,
- absl::optional<DtlsTransportTlsRole> role,
- absl::optional<int> tls_version,
- absl::optional<int> ssl_cipher_suite,
- absl::optional<int> srtp_cipher_suite,
+ std::optional<DtlsTransportTlsRole> role,
+ std::optional<int> tls_version,
+ std::optional<int> ssl_cipher_suite,
+ std::optional<int> srtp_cipher_suite,
std::unique_ptr<rtc::SSLCertChain> remote_ssl_certificates)
: state_(state),
role_(role),
@@ -41,12 +41,12 @@
// Deprecated version
DtlsTransportInformation::DtlsTransportInformation(
DtlsTransportState state,
- absl::optional<int> tls_version,
- absl::optional<int> ssl_cipher_suite,
- absl::optional<int> srtp_cipher_suite,
+ std::optional<int> tls_version,
+ std::optional<int> ssl_cipher_suite,
+ std::optional<int> srtp_cipher_suite,
std::unique_ptr<rtc::SSLCertChain> remote_ssl_certificates)
: state_(state),
- role_(absl::nullopt),
+ role_(std::nullopt),
tls_version_(tls_version),
ssl_cipher_suite_(ssl_cipher_suite),
srtp_cipher_suite_(srtp_cipher_suite),
diff --git a/api/dtls_transport_interface.h b/api/dtls_transport_interface.h
index 66ae4d4..0f78221 100644
--- a/api/dtls_transport_interface.h
+++ b/api/dtls_transport_interface.h
@@ -12,9 +12,9 @@
#define API_DTLS_TRANSPORT_INTERFACE_H_
#include <memory>
+#include <optional>
#include "absl/base/attributes.h"
-#include "absl/types/optional.h"
#include "api/ice_transport_interface.h"
#include "api/ref_count.h"
#include "api/rtc_error.h"
@@ -49,17 +49,17 @@
explicit DtlsTransportInformation(DtlsTransportState state);
DtlsTransportInformation(
DtlsTransportState state,
- absl::optional<DtlsTransportTlsRole> role,
- absl::optional<int> tls_version,
- absl::optional<int> ssl_cipher_suite,
- absl::optional<int> srtp_cipher_suite,
+ std::optional<DtlsTransportTlsRole> role,
+ std::optional<int> tls_version,
+ std::optional<int> ssl_cipher_suite,
+ std::optional<int> srtp_cipher_suite,
std::unique_ptr<rtc::SSLCertChain> remote_ssl_certificates);
ABSL_DEPRECATED("Use version with role parameter")
DtlsTransportInformation(
DtlsTransportState state,
- absl::optional<int> tls_version,
- absl::optional<int> ssl_cipher_suite,
- absl::optional<int> srtp_cipher_suite,
+ std::optional<int> tls_version,
+ std::optional<int> ssl_cipher_suite,
+ std::optional<int> srtp_cipher_suite,
std::unique_ptr<rtc::SSLCertChain> remote_ssl_certificates);
// Copy and assign
@@ -71,10 +71,10 @@
default;
DtlsTransportState state() const { return state_; }
- absl::optional<DtlsTransportTlsRole> role() const { return role_; }
- absl::optional<int> tls_version() const { return tls_version_; }
- absl::optional<int> ssl_cipher_suite() const { return ssl_cipher_suite_; }
- absl::optional<int> srtp_cipher_suite() const { return srtp_cipher_suite_; }
+ std::optional<DtlsTransportTlsRole> role() const { return role_; }
+ std::optional<int> tls_version() const { return tls_version_; }
+ std::optional<int> ssl_cipher_suite() const { return ssl_cipher_suite_; }
+ std::optional<int> srtp_cipher_suite() const { return srtp_cipher_suite_; }
// The accessor returns a temporary pointer, it does not release ownership.
const rtc::SSLCertChain* remote_ssl_certificates() const {
return remote_ssl_certificates_.get();
@@ -82,10 +82,10 @@
private:
DtlsTransportState state_;
- absl::optional<DtlsTransportTlsRole> role_;
- absl::optional<int> tls_version_;
- absl::optional<int> ssl_cipher_suite_;
- absl::optional<int> srtp_cipher_suite_;
+ std::optional<DtlsTransportTlsRole> role_;
+ std::optional<int> tls_version_;
+ std::optional<int> ssl_cipher_suite_;
+ std::optional<int> srtp_cipher_suite_;
std::unique_ptr<rtc::SSLCertChain> remote_ssl_certificates_;
};
diff --git a/api/environment/BUILD.gn b/api/environment/BUILD.gn
index 10d008f..4db7c46 100644
--- a/api/environment/BUILD.gn
+++ b/api/environment/BUILD.gn
@@ -58,7 +58,6 @@
"../units:timestamp",
"//third_party/abseil-cpp/absl/functional:any_invocable",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
}
diff --git a/api/environment/environment_unittest.cc b/api/environment/environment_unittest.cc
index 17bc668..bcadb49 100644
--- a/api/environment/environment_unittest.cc
+++ b/api/environment/environment_unittest.cc
@@ -11,13 +11,13 @@
#include "api/environment/environment.h"
#include <memory>
+#include <optional>
#include <string>
#include <utility>
#include <vector>
#include "absl/functional/any_invocable.h"
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/environment/environment_factory.h"
#include "api/field_trials_view.h"
#include "api/rtc_event_log/rtc_event.h"
@@ -168,10 +168,10 @@
/*on_destroyed=*/[&] { utility_destroyed = true; });
// Wrap Environment into optional to have explicit control when it is deleted.
- absl::optional<Environment> env = CreateEnvironment(std::move(field_trials));
+ std::optional<Environment> env = CreateEnvironment(std::move(field_trials));
EXPECT_FALSE(utility_destroyed);
- env = absl::nullopt;
+ env = std::nullopt;
EXPECT_TRUE(utility_destroyed);
}
@@ -180,13 +180,13 @@
auto field_trials = std::make_unique<FakeFieldTrials>(
/*on_destroyed=*/[&] { utility_destroyed = true; });
- absl::optional<Environment> env1 = CreateEnvironment(std::move(field_trials));
- absl::optional<Environment> env2 = env1;
+ std::optional<Environment> env1 = CreateEnvironment(std::move(field_trials));
+ std::optional<Environment> env2 = env1;
EXPECT_FALSE(utility_destroyed);
- env1 = absl::nullopt;
+ env1 = std::nullopt;
EXPECT_FALSE(utility_destroyed);
- env2 = absl::nullopt;
+ env2 = std::nullopt;
EXPECT_TRUE(utility_destroyed);
}
@@ -234,27 +234,27 @@
TEST(EnvironmentTest, KeepsOwnershipsWhenCreateNewEnvironmentFromExistingOne) {
bool utility1_destroyed = false;
bool utility2_destroyed = false;
- absl::optional<Environment> env1 =
+ std::optional<Environment> env1 =
CreateEnvironment(std::make_unique<FakeTaskQueueFactory>(
/*on_destroyed=*/[&] { utility1_destroyed = true; }));
- absl::optional<EnvironmentFactory> factory = EnvironmentFactory(*env1);
+ std::optional<EnvironmentFactory> factory = EnvironmentFactory(*env1);
// Destroy env1, check utility1 it was using is still alive.
- env1 = absl::nullopt;
+ env1 = std::nullopt;
EXPECT_FALSE(utility1_destroyed);
factory->Set(std::make_unique<FakeFieldTrials>(
/*on_destroyed=*/[&] { utility2_destroyed = true; }));
- absl::optional<Environment> env2 = factory->Create();
+ std::optional<Environment> env2 = factory->Create();
// Destroy the factory, check all utilities used by env2 are alive.
- factory = absl::nullopt;
+ factory = std::nullopt;
EXPECT_FALSE(utility1_destroyed);
EXPECT_FALSE(utility2_destroyed);
// Once last Environment object is deleted, utilties should be deleted too.
- env2 = absl::nullopt;
+ env2 = std::nullopt;
EXPECT_TRUE(utility1_destroyed);
EXPECT_TRUE(utility2_destroyed);
}
@@ -266,11 +266,11 @@
auto task_queue_factory = std::make_unique<FakeTaskQueueFactory>(
/*on_destroyed=*/[&] { destroyed.push_back("task_queue_factory"); });
- absl::optional<Environment> env =
+ std::optional<Environment> env =
CreateEnvironment(std::move(field_trials), std::move(task_queue_factory));
ASSERT_THAT(destroyed, IsEmpty());
- env = absl::nullopt;
+ env = std::nullopt;
EXPECT_THAT(destroyed, ElementsAre("task_queue_factory", "field_trials"));
}
diff --git a/api/frame_transformer_interface.h b/api/frame_transformer_interface.h
index b1f5426..6959445 100644
--- a/api/frame_transformer_interface.h
+++ b/api/frame_transformer_interface.h
@@ -13,9 +13,9 @@
#include <cstdint>
#include <memory>
+#include <optional>
#include <string>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/ref_count.h"
#include "api/scoped_refptr.h"
@@ -56,8 +56,8 @@
// TODO(https://bugs.webrtc.org/14878): Change this to pure virtual after it
// is implemented everywhere.
- virtual absl::optional<Timestamp> GetCaptureTimeIdentifier() const {
- return absl::nullopt;
+ virtual std::optional<Timestamp> GetCaptureTimeIdentifier() const {
+ return std::nullopt;
}
enum class Direction {
@@ -91,9 +91,9 @@
virtual rtc::ArrayView<const uint32_t> GetContributingSources() const = 0;
- virtual const absl::optional<uint16_t> SequenceNumber() const = 0;
+ virtual const std::optional<uint16_t> SequenceNumber() const = 0;
- virtual absl::optional<uint64_t> AbsoluteCaptureTimestamp() const = 0;
+ virtual std::optional<uint64_t> AbsoluteCaptureTimestamp() const = 0;
enum class FrameType { kEmptyFrame, kAudioFrameSpeech, kAudioFrameCN };
@@ -104,11 +104,11 @@
// Audio level in -dBov. Values range from 0 to 127, representing 0 to -127
// dBov. 127 represents digital silence. Only present on remote frames if
// the audio level header extension was included.
- virtual absl::optional<uint8_t> AudioLevel() const = 0;
+ virtual std::optional<uint8_t> AudioLevel() const = 0;
// Timestamp at which the packet has been first seen on the network interface.
// Only defined for received audio packet.
- virtual absl::optional<Timestamp> ReceiveTime() const = 0;
+ virtual std::optional<Timestamp> ReceiveTime() const = 0;
};
// Objects implement this interface to be notified with the transformed frame.
diff --git a/api/jsep.cc b/api/jsep.cc
index 5fdc890..21781d9 100644
--- a/api/jsep.cc
+++ b/api/jsep.cc
@@ -40,7 +40,7 @@
return "";
}
-absl::optional<SdpType> SdpTypeFromString(const std::string& type_str) {
+std::optional<SdpType> SdpTypeFromString(const std::string& type_str) {
if (type_str == SessionDescriptionInterface::kOffer) {
return SdpType::kOffer;
} else if (type_str == SessionDescriptionInterface::kPrAnswer) {
@@ -50,7 +50,7 @@
} else if (type_str == SessionDescriptionInterface::kRollback) {
return SdpType::kRollback;
} else {
- return absl::nullopt;
+ return std::nullopt;
}
}
diff --git a/api/jsep.h b/api/jsep.h
index 2578a28..d8fb27c 100644
--- a/api/jsep.h
+++ b/api/jsep.h
@@ -23,10 +23,10 @@
#include <stddef.h>
#include <memory>
+#include <optional>
#include <string>
#include <vector>
-#include "absl/types/optional.h"
#include "api/ref_count.h"
#include "api/rtc_error.h"
#include "rtc_base/system/rtc_export.h"
@@ -117,7 +117,7 @@
// Returns the SdpType from its string form. The string form can be one of the
// constants defined in SessionDescriptionInterface. Passing in any other string
// results in nullopt.
-RTC_EXPORT absl::optional<SdpType> SdpTypeFromString(
+RTC_EXPORT std::optional<SdpType> SdpTypeFromString(
const std::string& type_str);
// Class representation of an SDP session description.
diff --git a/api/media_stream_interface.h b/api/media_stream_interface.h
index 1597b8d..9547d4a 100644
--- a/api/media_stream_interface.h
+++ b/api/media_stream_interface.h
@@ -19,10 +19,10 @@
#include <stddef.h>
#include <stdint.h>
+#include <optional>
#include <string>
#include <vector>
-#include "absl/types/optional.h"
#include "api/audio/audio_processing_statistics.h"
#include "api/audio_options.h"
#include "api/ref_count.h"
@@ -131,7 +131,7 @@
// depending on video codec.
// TODO(perkj): Remove this once denoising is done by the source, and not by
// the encoder.
- virtual absl::optional<bool> needs_denoising() const = 0;
+ virtual std::optional<bool> needs_denoising() const = 0;
// Returns false if no stats are available, e.g, for a remote source, or a
// source which has not seen its first frame yet.
@@ -217,7 +217,7 @@
int sample_rate,
size_t number_of_channels,
size_t number_of_frames,
- absl::optional<int64_t> absolute_capture_timestamp_ms) {
+ std::optional<int64_t> absolute_capture_timestamp_ms) {
// TODO(bugs.webrtc.org/10739): Deprecate the old OnData and make this one
// pure virtual.
return OnData(audio_data, bits_per_sample, sample_rate, number_of_channels,
diff --git a/api/neteq/BUILD.gn b/api/neteq/BUILD.gn
index 7cda234..cdc44fc 100644
--- a/api/neteq/BUILD.gn
+++ b/api/neteq/BUILD.gn
@@ -25,7 +25,6 @@
"../audio_codecs:audio_codecs_api",
"../environment",
"../units:timestamp",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -57,7 +56,6 @@
":neteq_api",
":tick_timer",
"../environment",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
diff --git a/api/neteq/neteq.h b/api/neteq/neteq.h
index f9ee3b6..faa5624 100644
--- a/api/neteq/neteq.h
+++ b/api/neteq/neteq.h
@@ -15,10 +15,10 @@
#include <stdint.h>
#include <map>
+#include <optional>
#include <string>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/audio_codecs/audio_codec_pair_id.h"
#include "api/audio_codecs/audio_format.h"
@@ -136,7 +136,7 @@
bool enable_fast_accelerate = false;
bool enable_muted_state = false;
bool enable_rtx_handling = false;
- absl::optional<AudioCodecPairId> codec_pair_id;
+ std::optional<AudioCodecPairId> codec_pair_id;
bool for_test_no_time_stretching = false; // Use only for testing.
};
@@ -223,7 +223,7 @@
AudioFrame* audio_frame,
bool* muted = nullptr,
int* current_sample_rate_hz = nullptr,
- absl::optional<Operation> action_override = absl::nullopt) = 0;
+ std::optional<Operation> action_override = std::nullopt) = 0;
// Replaces the current set of decoders with the given one.
virtual void SetCodecs(const std::map<int, SdpAudioFormat>& codecs) = 0;
@@ -289,7 +289,7 @@
// Returns the RTP timestamp for the last sample delivered by GetAudio().
// The return value will be empty if no valid timestamp is available.
- virtual absl::optional<uint32_t> GetPlayoutTimestamp() const = 0;
+ virtual std::optional<uint32_t> GetPlayoutTimestamp() const = 0;
// Returns the sample rate in Hz of the audio produced in the last GetAudio
// call. If GetAudio has not been called yet, the configured sample rate
@@ -299,14 +299,14 @@
// Returns the decoder info for the given payload type. Returns empty if no
// such payload type was registered.
[[deprecated(
- "Use GetCurrentDecoderFormat")]] virtual absl::optional<DecoderFormat>
+ "Use GetCurrentDecoderFormat")]] virtual std::optional<DecoderFormat>
GetDecoderFormat(int payload_type) const {
- return absl::nullopt;
+ return std::nullopt;
}
// Returns info for the most recently used decoder.
- virtual absl::optional<DecoderFormat> GetCurrentDecoderFormat() const {
- return absl::nullopt;
+ virtual std::optional<DecoderFormat> GetCurrentDecoderFormat() const {
+ return std::nullopt;
}
// Flushes both the packet buffer and the sync buffer.
diff --git a/api/neteq/neteq_controller.h b/api/neteq/neteq_controller.h
index 27ec218..f764417 100644
--- a/api/neteq/neteq_controller.h
+++ b/api/neteq/neteq_controller.h
@@ -15,8 +15,8 @@
#include <cstdint>
#include <functional>
#include <memory>
+#include <optional>
-#include "absl/types/optional.h"
#include "api/neteq/neteq.h"
#include "api/neteq/tick_timer.h"
@@ -86,7 +86,7 @@
uint32_t target_timestamp;
int16_t expand_mutefactor;
size_t last_packet_samples;
- absl::optional<PacketInfo> next_packet;
+ std::optional<PacketInfo> next_packet;
NetEq::Mode last_mode;
bool play_dtmf;
size_t generated_noise_samples;
@@ -161,9 +161,9 @@
// Notify the NetEqController that a packet has arrived. Returns the relative
// arrival delay, if it can be computed.
- virtual absl::optional<int> PacketArrived(int fs_hz,
- bool should_update_stats,
- const PacketArrivedInfo& info) = 0;
+ virtual std::optional<int> PacketArrived(int fs_hz,
+ bool should_update_stats,
+ const PacketArrivedInfo& info) = 0;
// Notify the NetEqController that we are currently in muted state.
// TODO(bugs.webrtc.org/14270): Make pure virtual when downstream is updated.
diff --git a/api/peer_connection_interface.h b/api/peer_connection_interface.h
index 46b9133..8e2893a 100644
--- a/api/peer_connection_interface.h
+++ b/api/peer_connection_interface.h
@@ -73,12 +73,12 @@
#include <functional>
#include <memory>
+#include <optional>
#include <string>
#include <vector>
#include "absl/base/attributes.h"
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/adaptation/resource.h"
#include "api/async_dns_resolver.h"
#include "api/audio/audio_device.h"
@@ -433,7 +433,7 @@
// The below fields correspond to constraints from the deprecated
// constraints interface for constructing a PeerConnection.
//
- // absl::optional fields can be "missing", in which case the implementation
+ // std::optional fields can be "missing", in which case the implementation
// default will be used.
//////////////////////////////////////////////////////////////////////////
@@ -457,7 +457,7 @@
// Minimum bitrate at which screencast video tracks will be encoded at.
// This means adding padding bits up to this bitrate, which can help
// when switching from a static scene to one with motion.
- absl::optional<int> screencast_min_bitrate;
+ std::optional<int> screencast_min_bitrate;
/////////////////////////////////////////////////
// The below fields are not part of the standard.
@@ -581,28 +581,28 @@
// 3) ice_check_min_interval defines the minimal interval (equivalently the
// maximum rate) that overrides the above two intervals when either of them
// is less.
- absl::optional<int> ice_check_interval_strong_connectivity;
- absl::optional<int> ice_check_interval_weak_connectivity;
- absl::optional<int> ice_check_min_interval;
+ std::optional<int> ice_check_interval_strong_connectivity;
+ std::optional<int> ice_check_interval_weak_connectivity;
+ std::optional<int> ice_check_min_interval;
// The min time period for which a candidate pair must wait for response to
// connectivity checks before it becomes unwritable. This parameter
// overrides the default value in the ICE implementation if set.
- absl::optional<int> ice_unwritable_timeout;
+ std::optional<int> ice_unwritable_timeout;
// The min number of connectivity checks that a candidate pair must sent
// without receiving response before it becomes unwritable. This parameter
// overrides the default value in the ICE implementation if set.
- absl::optional<int> ice_unwritable_min_checks;
+ std::optional<int> ice_unwritable_min_checks;
// The min time period for which a candidate pair must wait for response to
// connectivity checks it becomes inactive. This parameter overrides the
// default value in the ICE implementation if set.
- absl::optional<int> ice_inactive_timeout;
+ std::optional<int> ice_inactive_timeout;
// The interval in milliseconds at which STUN candidates will resend STUN
// binding requests to keep NAT bindings open.
- absl::optional<int> stun_candidate_keepalive_interval;
+ std::optional<int> stun_candidate_keepalive_interval;
// Optional TurnCustomizer.
// With this class one can modify outgoing TURN messages.
@@ -614,7 +614,7 @@
// A candidate pair on a preferred network has a higher precedence in ICE
// than one on an un-preferred network, regardless of priority or network
// cost.
- absl::optional<rtc::AdapterType> network_preference;
+ std::optional<rtc::AdapterType> network_preference;
// Configure the SDP semantics used by this PeerConnection. By default, this
// is Unified Plan which is compliant to the WebRTC 1.0 specification. It is
@@ -648,7 +648,7 @@
// Defines advanced optional cryptographic settings related to SRTP and
// frame encryption for native WebRTC. Setting this will overwrite any
// settings set in PeerConnectionFactory (which is deprecated).
- absl::optional<CryptoOptions> crypto_options;
+ std::optional<CryptoOptions> crypto_options;
// Configure if we should include the SDP attribute extmap-allow-mixed in
// our offer on session level.
@@ -665,11 +665,11 @@
// The delay before doing a usage histogram report for long-lived
// PeerConnections. Used for testing only.
- absl::optional<int> report_usage_pattern_delay_ms;
+ std::optional<int> report_usage_pattern_delay_ms;
// The ping interval (ms) when the connection is stable and writable. This
// parameter overrides the default value in the ICE implementation if set.
- absl::optional<int> stable_writable_connection_ping_interval_ms;
+ std::optional<int> stable_writable_connection_ping_interval_ms;
// Whether this PeerConnection will avoid VPNs (kAvoidVpn), prefer VPNs
// (kPreferVpn), only work over VPN (kOnlyUseVpn) or only work over non-VPN
@@ -685,7 +685,7 @@
PortAllocatorConfig port_allocator_config;
// The burst interval of the pacer, see TaskQueuePacedSender constructor.
- absl::optional<TimeDelta> pacer_burst_interval;
+ std::optional<TimeDelta> pacer_burst_interval;
//
// Don't forget to update operator== if adding something.
@@ -1187,7 +1187,7 @@
// Returns the current state of canTrickleIceCandidates per
// https://w3c.github.io/webrtc-pc/#attributes-1
- virtual absl::optional<bool> can_trickle_ice_candidates() = 0;
+ virtual std::optional<bool> can_trickle_ice_candidates() = 0;
// When a resource is overused, the PeerConnection will try to reduce the load
// on the sysem, for example by reducing the resolution or frame rate of
diff --git a/api/rtc_error.h b/api/rtc_error.h
index 8579ff6..9de3b1c 100644
--- a/api/rtc_error.h
+++ b/api/rtc_error.h
@@ -13,11 +13,11 @@
#include <stdint.h>
+#include <optional>
#include <string>
#include <utility> // For std::move.
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/system/rtc_export.h"
@@ -137,7 +137,7 @@
RTCErrorDetailType error_detail() const { return error_detail_; }
void set_error_detail(RTCErrorDetailType detail) { error_detail_ = detail; }
- absl::optional<uint16_t> sctp_cause_code() const { return sctp_cause_code_; }
+ std::optional<uint16_t> sctp_cause_code() const { return sctp_cause_code_; }
void set_sctp_cause_code(uint16_t cause_code) {
sctp_cause_code_ = cause_code;
}
@@ -150,7 +150,7 @@
RTCErrorType type_ = RTCErrorType::NONE;
std::string message_;
RTCErrorDetailType error_detail_ = RTCErrorDetailType::NONE;
- absl::optional<uint16_t> sctp_cause_code_;
+ std::optional<uint16_t> sctp_cause_code_;
};
// Outputs the error as a friendly string. Update this method when adding a new
@@ -309,7 +309,7 @@
private:
RTCError error_;
- absl::optional<T> value_;
+ std::optional<T> value_;
};
} // namespace webrtc
diff --git a/api/rtp_headers.h b/api/rtp_headers.h
index 129ab5f..6ba3292 100644
--- a/api/rtp_headers.h
+++ b/api/rtp_headers.h
@@ -14,9 +14,9 @@
#include <stddef.h>
#include <stdint.h>
+#include <optional>
#include <string>
-#include "absl/types/optional.h"
#include "api/units/timestamp.h"
#include "api/video/color_space.h"
#include "api/video/video_content_type.h"
@@ -75,7 +75,7 @@
// system’s NTP clock:
//
// Capture NTP Clock = Sender NTP Clock + Capture Clock Offset
- absl::optional<int64_t> estimated_capture_clock_offset;
+ std::optional<int64_t> estimated_capture_clock_offset;
};
// The audio level extension is used to indicate the voice activity and the
@@ -131,16 +131,16 @@
int32_t transmissionTimeOffset;
bool hasAbsoluteSendTime;
uint32_t absoluteSendTime;
- absl::optional<AbsoluteCaptureTime> absolute_capture_time;
+ std::optional<AbsoluteCaptureTime> absolute_capture_time;
bool hasTransportSequenceNumber;
uint16_t transportSequenceNumber;
- absl::optional<FeedbackRequest> feedback_request;
+ std::optional<FeedbackRequest> feedback_request;
// Audio Level includes both level in dBov and voiced/unvoiced bit. See:
// https://tools.ietf.org/html/rfc6464#section-3
- absl::optional<AudioLevel> audio_level() const { return audio_level_; }
+ std::optional<AudioLevel> audio_level() const { return audio_level_; }
- void set_audio_level(absl::optional<AudioLevel> audio_level) {
+ void set_audio_level(std::optional<AudioLevel> audio_level) {
audio_level_ = audio_level;
}
@@ -150,7 +150,7 @@
bool hasVideoRotation;
VideoRotation videoRotation;
- // TODO(ilnik): Refactor this and one above to be absl::optional() and remove
+ // TODO(ilnik): Refactor this and one above to be std::optional() and remove
// a corresponding bool flag.
bool hasVideoContentType;
VideoContentType videoContentType;
@@ -169,10 +169,10 @@
// https://tools.ietf.org/html/rfc8843
std::string mid;
- absl::optional<ColorSpace> color_space;
+ std::optional<ColorSpace> color_space;
private:
- absl::optional<AudioLevel> audio_level_;
+ std::optional<AudioLevel> audio_level_;
};
enum { kRtpCsrcSize = 15 }; // RFC 3550 page 13
diff --git a/api/rtp_packet_info.h b/api/rtp_packet_info.h
index 8df12a3..8e8236d 100644
--- a/api/rtp_packet_info.h
+++ b/api/rtp_packet_info.h
@@ -12,10 +12,10 @@
#define API_RTP_PACKET_INFO_H_
#include <cstdint>
+#include <optional>
#include <utility>
#include <vector>
-#include "absl/types/optional.h"
#include "api/rtp_headers.h"
#include "api/units/time_delta.h"
#include "api/units/timestamp.h"
@@ -56,26 +56,26 @@
Timestamp receive_time() const { return receive_time_; }
void set_receive_time(Timestamp value) { receive_time_ = value; }
- absl::optional<uint8_t> audio_level() const { return audio_level_; }
- RtpPacketInfo& set_audio_level(absl::optional<uint8_t> value) {
+ std::optional<uint8_t> audio_level() const { return audio_level_; }
+ RtpPacketInfo& set_audio_level(std::optional<uint8_t> value) {
audio_level_ = value;
return *this;
}
- const absl::optional<AbsoluteCaptureTime>& absolute_capture_time() const {
+ const std::optional<AbsoluteCaptureTime>& absolute_capture_time() const {
return absolute_capture_time_;
}
RtpPacketInfo& set_absolute_capture_time(
- const absl::optional<AbsoluteCaptureTime>& value) {
+ const std::optional<AbsoluteCaptureTime>& value) {
absolute_capture_time_ = value;
return *this;
}
- const absl::optional<TimeDelta>& local_capture_clock_offset() const {
+ const std::optional<TimeDelta>& local_capture_clock_offset() const {
return local_capture_clock_offset_;
}
RtpPacketInfo& set_local_capture_clock_offset(
- absl::optional<TimeDelta> value) {
+ std::optional<TimeDelta> value) {
local_capture_clock_offset_ = value;
return *this;
}
@@ -92,18 +92,18 @@
// Fields from the Audio Level header extension:
// https://tools.ietf.org/html/rfc6464#section-3
- absl::optional<uint8_t> audio_level_;
+ std::optional<uint8_t> audio_level_;
// Fields from the Absolute Capture Time header extension:
// http://www.webrtc.org/experiments/rtp-hdrext/abs-capture-time
- absl::optional<AbsoluteCaptureTime> absolute_capture_time_;
+ std::optional<AbsoluteCaptureTime> absolute_capture_time_;
// Clock offset between the local clock and the capturer's clock.
// Do not confuse with `AbsoluteCaptureTime::estimated_capture_clock_offset`
// which instead represents the clock offset between a remote sender and the
// capturer. The following holds:
// Capture's NTP Clock = Local NTP Clock + Local-Capture Clock Offset
- absl::optional<TimeDelta> local_capture_clock_offset_;
+ std::optional<TimeDelta> local_capture_clock_offset_;
};
bool operator==(const RtpPacketInfo& lhs, const RtpPacketInfo& rhs);
diff --git a/api/rtp_packet_info_unittest.cc b/api/rtp_packet_info_unittest.cc
index 49ed337..9aea11d 100644
--- a/api/rtp_packet_info_unittest.cc
+++ b/api/rtp_packet_info_unittest.cc
@@ -11,9 +11,9 @@
#include "api/rtp_packet_info.h"
#include <cstdint>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/rtp_headers.h"
#include "api/units/time_delta.h"
#include "api/units/timestamp.h"
@@ -134,7 +134,7 @@
}
TEST(RtpPacketInfoTest, AudioLevel) {
- constexpr absl::optional<uint8_t> kValue = 31;
+ constexpr std::optional<uint8_t> kValue = 31;
RtpPacketInfo lhs;
RtpPacketInfo rhs;
@@ -163,7 +163,7 @@
}
TEST(RtpPacketInfoTest, AbsoluteCaptureTime) {
- constexpr absl::optional<AbsoluteCaptureTime> kValue = AbsoluteCaptureTime{
+ constexpr std::optional<AbsoluteCaptureTime> kValue = AbsoluteCaptureTime{
.absolute_capture_timestamp = 12, .estimated_capture_clock_offset = 34};
RtpPacketInfo lhs;
@@ -213,7 +213,7 @@
EXPECT_FALSE(lhs != rhs);
rhs = RtpPacketInfo();
- EXPECT_EQ(rhs.local_capture_clock_offset(), absl::nullopt);
+ EXPECT_EQ(rhs.local_capture_clock_offset(), std::nullopt);
rhs = RtpPacketInfo(/*ssrc=*/{}, /*csrcs=*/{}, /*rtp_timestamp=*/{},
/*receive_time=*/Timestamp::Zero());
diff --git a/api/rtp_parameters.h b/api/rtp_parameters.h
index 70d7686..b393c3c 100644
--- a/api/rtp_parameters.h
+++ b/api/rtp_parameters.h
@@ -14,12 +14,12 @@
#include <stdint.h>
#include <map>
+#include <optional>
#include <string>
#include <vector>
#include "absl/container/inlined_vector.h"
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/media_types.h"
#include "api/priority.h"
#include "api/rtp_transceiver_direction.h"
@@ -109,7 +109,7 @@
// 1. It's an enum instead of a string.
// 2. Generic NACK feedback is represented by a GENERIC_NACK message type,
// rather than an unset "parameter" value.
- absl::optional<RtcpFeedbackMessageType> message_type;
+ std::optional<RtcpFeedbackMessageType> message_type;
// Constructors for convenience.
RtcpFeedback();
@@ -139,14 +139,14 @@
cricket::MediaType kind = cricket::MEDIA_TYPE_AUDIO;
// If unset, the implementation default is used.
- absl::optional<int> clock_rate;
+ std::optional<int> clock_rate;
// The number of audio channels used. Unset for video codecs. If unset for
// audio, the implementation default is used.
// TODO(deadbeef): The "implementation default" part isn't fully implemented.
// Only defaults to 1, even though some codecs (such as opus) should really
// default to 2.
- absl::optional<int> num_channels;
+ std::optional<int> num_channels;
// Feedback mechanisms to be used for this codec.
// TODO(deadbeef): Not implemented with PeerConnection senders/receivers.
@@ -180,7 +180,7 @@
// Default payload type for this codec. Mainly needed for codecs that have
// statically assigned payload types.
- absl::optional<int> preferred_payload_type;
+ std::optional<int> preferred_payload_type;
// List of scalability modes supported by the video codec.
absl::InlinedVector<ScalabilityMode, kScalabilityModeCount> scalability_modes;
@@ -209,7 +209,7 @@
std::string uri;
// Preferred value of ID that goes in the packet.
- absl::optional<int> preferred_id;
+ std::optional<int> preferred_id;
// If true, it's preferred that the value in the header is encrypted.
// TODO(deadbeef): Not implemented.
@@ -397,7 +397,7 @@
struct RTC_EXPORT RtpFecParameters {
// If unset, a value is chosen by the implementation.
// Works just like RtpEncodingParameters::ssrc.
- absl::optional<uint32_t> ssrc;
+ std::optional<uint32_t> ssrc;
FecMechanism mechanism = FecMechanism::RED;
@@ -417,7 +417,7 @@
struct RTC_EXPORT RtpRtxParameters {
// If unset, a value is chosen by the implementation.
// Works just like RtpEncodingParameters::ssrc.
- absl::optional<uint32_t> ssrc;
+ std::optional<uint32_t> ssrc;
// Constructors for convenience.
RtpRtxParameters();
@@ -440,7 +440,7 @@
// may change due to an SSRC conflict, in which case the conflict is handled
// internally without any event. Another way of looking at this is that an
// unset SSRC acts as a "wildcard" SSRC.
- absl::optional<uint32_t> ssrc;
+ std::optional<uint32_t> ssrc;
// The relative bitrate priority of this encoding. Currently this is
// implemented for the entire rtp sender by using the value of the first
@@ -475,24 +475,24 @@
// bandwidth for the entire bandwidth estimator (audio and video). This is
// just always how "b=AS" was handled, but it's not correct and should be
// fixed.
- absl::optional<int> max_bitrate_bps;
+ std::optional<int> max_bitrate_bps;
// Specifies the minimum bitrate in bps for video.
- absl::optional<int> min_bitrate_bps;
+ std::optional<int> min_bitrate_bps;
// Specifies the maximum framerate in fps for video.
- absl::optional<double> max_framerate;
+ std::optional<double> max_framerate;
// Specifies the number of temporal layers for video (if the feature is
// supported by the codec implementation).
// Screencast support is experimental.
- absl::optional<int> num_temporal_layers;
+ std::optional<int> num_temporal_layers;
// For video, scale the resolution down by this factor.
- absl::optional<double> scale_resolution_down_by;
+ std::optional<double> scale_resolution_down_by;
// https://w3c.github.io/webrtc-svc/#rtcrtpencodingparameters
- absl::optional<std::string> scalability_mode;
+ std::optional<std::string> scalability_mode;
// Requested encode resolution.
//
@@ -510,7 +510,7 @@
//
// It is an error to set both `requested_resolution` and
// `scale_resolution_down_by`.
- absl::optional<Resolution> requested_resolution;
+ std::optional<Resolution> requested_resolution;
// For an RtpSender, set to true to cause this encoding to be encoded and
// sent, and false for it not to be encoded and sent. This allows control
@@ -530,7 +530,7 @@
bool adaptive_ptime = false;
// Allow changing the used codec for this encoding.
- absl::optional<RtpCodec> codec;
+ std::optional<RtpCodec> codec;
bool operator==(const RtpEncodingParameters& o) const {
return ssrc == o.ssrc && bitrate_priority == o.bitrate_priority &&
@@ -597,7 +597,7 @@
// The SSRC to be used in the "SSRC of packet sender" field. If not set, one
// will be chosen by the implementation.
// TODO(deadbeef): Not implemented.
- absl::optional<uint32_t> ssrc;
+ std::optional<uint32_t> ssrc;
// The Canonical Name (CNAME) used by RTCP (e.g. in SDES messages).
//
@@ -653,7 +653,7 @@
// When bandwidth is constrained and the RtpSender needs to choose between
// degrading resolution or degrading framerate, degradationPreference
// indicates which is preferred. Only for video tracks.
- absl::optional<DegradationPreference> degradation_preference;
+ std::optional<DegradationPreference> degradation_preference;
bool operator==(const RtpParameters& o) const {
return mid == o.mid && codecs == o.codecs &&
diff --git a/api/rtp_receiver_interface.h b/api/rtp_receiver_interface.h
index b1b36b5..f97ed53 100644
--- a/api/rtp_receiver_interface.h
+++ b/api/rtp_receiver_interface.h
@@ -14,11 +14,11 @@
#ifndef API_RTP_RECEIVER_INTERFACE_H_
#define API_RTP_RECEIVER_INTERFACE_H_
+#include <optional>
#include <string>
#include <utility>
#include <vector>
-#include "absl/types/optional.h"
#include "api/crypto/frame_decryptor_interface.h"
#include "api/dtls_transport_interface.h"
#include "api/frame_transformer_interface.h"
@@ -91,7 +91,7 @@
// positive value including 0.0 measured in seconds. `nullopt` means default
// value must be used.
virtual void SetJitterBufferMinimumDelay(
- absl::optional<double> delay_seconds) = 0;
+ std::optional<double> delay_seconds) = 0;
// TODO(zhihuang): Remove the default implementation once the subclasses
// implement this. Currently, the only relevant subclass is the
diff --git a/api/rtp_sender_interface.h b/api/rtp_sender_interface.h
index 7d92c42..aa2eb6f 100644
--- a/api/rtp_sender_interface.h
+++ b/api/rtp_sender_interface.h
@@ -53,7 +53,7 @@
// Returns primary SSRC used by this sender for sending media.
// Returns 0 if not yet determined.
- // TODO(deadbeef): Change to absl::optional.
+ // TODO(deadbeef): Change to std::optional.
// TODO(deadbeef): Remove? With GetParameters this should be redundant.
virtual uint32_t ssrc() const = 0;
diff --git a/api/rtp_transceiver_interface.cc b/api/rtp_transceiver_interface.cc
index 7267b28..5f16767 100644
--- a/api/rtp_transceiver_interface.cc
+++ b/api/rtp_transceiver_interface.cc
@@ -20,9 +20,9 @@
RtpTransceiverInit::~RtpTransceiverInit() = default;
-absl::optional<RtpTransceiverDirection>
+std::optional<RtpTransceiverDirection>
RtpTransceiverInterface::fired_direction() const {
- return absl::nullopt;
+ return std::nullopt;
}
bool RtpTransceiverInterface::stopping() const {
diff --git a/api/rtp_transceiver_interface.h b/api/rtp_transceiver_interface.h
index 555566d..2d0bf29 100644
--- a/api/rtp_transceiver_interface.h
+++ b/api/rtp_transceiver_interface.h
@@ -11,11 +11,11 @@
#ifndef API_RTP_TRANSCEIVER_INTERFACE_H_
#define API_RTP_TRANSCEIVER_INTERFACE_H_
+#include <optional>
#include <string>
#include <vector>
#include "absl/base/attributes.h"
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/media_types.h"
#include "api/ref_count.h"
@@ -69,7 +69,7 @@
// remote descriptions. Before negotiation is complete, the mid value may be
// null. After rollbacks, the value may change from a non-null value to null.
// https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-mid
- virtual absl::optional<std::string> mid() const = 0;
+ virtual std::optional<std::string> mid() const = 0;
// The sender attribute exposes the RtpSender corresponding to the RTP media
// that may be sent with the transceiver's mid. The sender is always present,
@@ -119,14 +119,14 @@
// for this transceiver. If this transceiver has never been represented in an
// offer/answer exchange, or if the transceiver is stopped, the value is null.
// https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-currentdirection
- virtual absl::optional<RtpTransceiverDirection> current_direction() const = 0;
+ virtual std::optional<RtpTransceiverDirection> current_direction() const = 0;
// An internal slot designating for which direction the relevant
// PeerConnection events have been fired. This is to ensure that events like
// OnAddTrack only get fired once even if the same session description is
// applied again.
// Exposed in the public interface for use by Chromium.
- virtual absl::optional<RtpTransceiverDirection> fired_direction() const;
+ virtual std::optional<RtpTransceiverDirection> fired_direction() const;
// Initiates a stop of the transceiver.
// The stop is complete when stopped() returns true.
diff --git a/api/sctp_transport_interface.cc b/api/sctp_transport_interface.cc
index 8a0a866..8bd279c 100644
--- a/api/sctp_transport_interface.cc
+++ b/api/sctp_transport_interface.cc
@@ -20,8 +20,8 @@
SctpTransportInformation::SctpTransportInformation(
SctpTransportState state,
rtc::scoped_refptr<DtlsTransportInterface> dtls_transport,
- absl::optional<double> max_message_size,
- absl::optional<int> max_channels)
+ std::optional<double> max_message_size,
+ std::optional<int> max_channels)
: state_(state),
dtls_transport_(std::move(dtls_transport)),
max_message_size_(max_message_size),
diff --git a/api/sctp_transport_interface.h b/api/sctp_transport_interface.h
index 25c2ba0..173f3ef 100644
--- a/api/sctp_transport_interface.h
+++ b/api/sctp_transport_interface.h
@@ -11,7 +11,8 @@
#ifndef API_SCTP_TRANSPORT_INTERFACE_H_
#define API_SCTP_TRANSPORT_INTERFACE_H_
-#include "absl/types/optional.h"
+#include <optional>
+
#include "api/dtls_transport_interface.h"
#include "api/ref_count.h"
#include "api/scoped_refptr.h"
@@ -41,22 +42,22 @@
SctpTransportInformation(
SctpTransportState state,
rtc::scoped_refptr<DtlsTransportInterface> dtls_transport,
- absl::optional<double> max_message_size,
- absl::optional<int> max_channels);
+ std::optional<double> max_message_size,
+ std::optional<int> max_channels);
~SctpTransportInformation();
// The DTLS transport that supports this SCTP transport.
rtc::scoped_refptr<DtlsTransportInterface> dtls_transport() const {
return dtls_transport_;
}
SctpTransportState state() const { return state_; }
- absl::optional<double> MaxMessageSize() const { return max_message_size_; }
- absl::optional<int> MaxChannels() const { return max_channels_; }
+ std::optional<double> MaxMessageSize() const { return max_message_size_; }
+ std::optional<int> MaxChannels() const { return max_channels_; }
private:
SctpTransportState state_ = SctpTransportState::kNew;
rtc::scoped_refptr<DtlsTransportInterface> dtls_transport_;
- absl::optional<double> max_message_size_;
- absl::optional<int> max_channels_;
+ std::optional<double> max_message_size_;
+ std::optional<int> max_channels_;
};
class SctpTransportObserverInterface {
diff --git a/api/stats/attribute.h b/api/stats/attribute.h
index 36500db..b5dd8d5 100644
--- a/api/stats/attribute.h
+++ b/api/stats/attribute.h
@@ -13,10 +13,10 @@
#include <map>
#include <memory>
+#include <optional>
#include <string>
#include <vector>
-#include "absl/types/optional.h"
#include "absl/types/variant.h"
#include "rtc_base/checks.h"
#include "rtc_base/system/rtc_export.h"
@@ -24,30 +24,30 @@
namespace webrtc {
// A light-weight wrapper of an RTCStats attribute, i.e. an individual metric of
-// type absl::optional<T>.
+// type std::optional<T>.
class RTC_EXPORT Attribute {
public:
// All supported attribute types.
- typedef absl::variant<const absl::optional<bool>*,
- const absl::optional<int32_t>*,
- const absl::optional<uint32_t>*,
- const absl::optional<int64_t>*,
- const absl::optional<uint64_t>*,
- const absl::optional<double>*,
- const absl::optional<std::string>*,
- const absl::optional<std::vector<bool>>*,
- const absl::optional<std::vector<int32_t>>*,
- const absl::optional<std::vector<uint32_t>>*,
- const absl::optional<std::vector<int64_t>>*,
- const absl::optional<std::vector<uint64_t>>*,
- const absl::optional<std::vector<double>>*,
- const absl::optional<std::vector<std::string>>*,
- const absl::optional<std::map<std::string, uint64_t>>*,
- const absl::optional<std::map<std::string, double>>*>
+ typedef absl::variant<const std::optional<bool>*,
+ const std::optional<int32_t>*,
+ const std::optional<uint32_t>*,
+ const std::optional<int64_t>*,
+ const std::optional<uint64_t>*,
+ const std::optional<double>*,
+ const std::optional<std::string>*,
+ const std::optional<std::vector<bool>>*,
+ const std::optional<std::vector<int32_t>>*,
+ const std::optional<std::vector<uint32_t>>*,
+ const std::optional<std::vector<int64_t>>*,
+ const std::optional<std::vector<uint64_t>>*,
+ const std::optional<std::vector<double>>*,
+ const std::optional<std::vector<std::string>>*,
+ const std::optional<std::map<std::string, uint64_t>>*,
+ const std::optional<std::map<std::string, double>>*>
StatVariant;
template <typename T>
- Attribute(const char* name, const absl::optional<T>* attribute)
+ Attribute(const char* name, const std::optional<T>* attribute)
: name_(name), attribute_(attribute) {}
const char* name() const;
@@ -56,18 +56,18 @@
bool has_value() const;
template <typename T>
bool holds_alternative() const {
- return absl::holds_alternative<const absl::optional<T>*>(attribute_);
+ return absl::holds_alternative<const std::optional<T>*>(attribute_);
}
template <typename T>
- const absl::optional<T>& as_optional() const {
+ const std::optional<T>& as_optional() const {
RTC_CHECK(holds_alternative<T>());
- return *absl::get<const absl::optional<T>*>(attribute_);
+ return *absl::get<const std::optional<T>*>(attribute_);
}
template <typename T>
const T& get() const {
RTC_CHECK(holds_alternative<T>());
RTC_CHECK(has_value());
- return absl::get<const absl::optional<T>*>(attribute_)->value();
+ return absl::get<const std::optional<T>*>(attribute_)->value();
}
bool is_sequence() const;
diff --git a/api/stats/rtc_stats.h b/api/stats/rtc_stats.h
index 74e7fc4..e1f3bb1 100644
--- a/api/stats/rtc_stats.h
+++ b/api/stats/rtc_stats.h
@@ -16,11 +16,11 @@
#include <map>
#include <memory>
+#include <optional>
#include <string>
#include <utility>
#include <vector>
-#include "absl/types/optional.h"
#include "api/stats/attribute.h"
#include "api/units/timestamp.h"
#include "rtc_base/checks.h"
@@ -39,7 +39,7 @@
// Use the `WEBRTC_RTCSTATS_IMPL` macro when implementing subclasses, see macro
// for details.
//
-// Derived classes list their dictionary attributes, absl::optional<T>, as
+// Derived classes list their dictionary attributes, std::optional<T>, as
// public fields, allowing the following:
//
// RTCFooStats foo("fooId", Timestamp::Micros(GetCurrentTime()));
@@ -73,13 +73,12 @@
// metrics as viewed via the Attribute wrapper.
std::vector<Attribute> Attributes() const;
template <typename T>
- Attribute GetAttribute(const absl::optional<T>& stat) const {
+ Attribute GetAttribute(const std::optional<T>& stat) const {
for (const auto& attribute : Attributes()) {
if (!attribute.holds_alternative<T>()) {
continue;
}
- if (absl::get<const absl::optional<T>*>(attribute.as_variant()) ==
- &stat) {
+ if (absl::get<const std::optional<T>*>(attribute.as_variant()) == &stat) {
return attribute;
}
}
@@ -136,8 +135,8 @@
//
// RTCFooStats(const std::string& id, Timestamp timestamp);
//
-// absl::optional<int32_t> foo;
-// absl::optional<int32_t> bar;
+// std::optional<int32_t> foo;
+// std::optional<int32_t> bar;
// };
//
// rtcfoostats.cc:
diff --git a/api/stats/rtcstats_objects.h b/api/stats/rtcstats_objects.h
index 9f51f56..0645ee6 100644
--- a/api/stats/rtcstats_objects.h
+++ b/api/stats/rtcstats_objects.h
@@ -15,10 +15,10 @@
#include <map>
#include <memory>
+#include <optional>
#include <string>
#include <vector>
-#include "absl/types/optional.h"
#include "api/stats/rtc_stats.h"
#include "rtc_base/system/rtc_export.h"
@@ -31,10 +31,10 @@
RTCCertificateStats(std::string id, Timestamp timestamp);
~RTCCertificateStats() override;
- absl::optional<std::string> fingerprint;
- absl::optional<std::string> fingerprint_algorithm;
- absl::optional<std::string> base64_certificate;
- absl::optional<std::string> issuer_certificate_id;
+ std::optional<std::string> fingerprint;
+ std::optional<std::string> fingerprint_algorithm;
+ std::optional<std::string> base64_certificate;
+ std::optional<std::string> issuer_certificate_id;
};
// https://w3c.github.io/webrtc-stats/#codec-dict*
@@ -44,12 +44,12 @@
RTCCodecStats(std::string id, Timestamp timestamp);
~RTCCodecStats() override;
- absl::optional<std::string> transport_id;
- absl::optional<uint32_t> payload_type;
- absl::optional<std::string> mime_type;
- absl::optional<uint32_t> clock_rate;
- absl::optional<uint32_t> channels;
- absl::optional<std::string> sdp_fmtp_line;
+ std::optional<std::string> transport_id;
+ std::optional<uint32_t> payload_type;
+ std::optional<std::string> mime_type;
+ std::optional<uint32_t> clock_rate;
+ std::optional<uint32_t> channels;
+ std::optional<std::string> sdp_fmtp_line;
};
// https://w3c.github.io/webrtc-stats/#dcstats-dict*
@@ -59,14 +59,14 @@
RTCDataChannelStats(std::string id, Timestamp timestamp);
~RTCDataChannelStats() override;
- absl::optional<std::string> label;
- absl::optional<std::string> protocol;
- absl::optional<int32_t> data_channel_identifier;
- absl::optional<std::string> state;
- absl::optional<uint32_t> messages_sent;
- absl::optional<uint64_t> bytes_sent;
- absl::optional<uint32_t> messages_received;
- absl::optional<uint64_t> bytes_received;
+ std::optional<std::string> label;
+ std::optional<std::string> protocol;
+ std::optional<int32_t> data_channel_identifier;
+ std::optional<std::string> state;
+ std::optional<uint32_t> messages_sent;
+ std::optional<uint64_t> bytes_sent;
+ std::optional<uint32_t> messages_received;
+ std::optional<uint64_t> bytes_received;
};
// https://w3c.github.io/webrtc-stats/#candidatepair-dict*
@@ -76,35 +76,35 @@
RTCIceCandidatePairStats(std::string id, Timestamp timestamp);
~RTCIceCandidatePairStats() override;
- absl::optional<std::string> transport_id;
- absl::optional<std::string> local_candidate_id;
- absl::optional<std::string> remote_candidate_id;
- absl::optional<std::string> state;
+ std::optional<std::string> transport_id;
+ std::optional<std::string> local_candidate_id;
+ std::optional<std::string> remote_candidate_id;
+ std::optional<std::string> state;
// Obsolete: priority
- absl::optional<uint64_t> priority;
- absl::optional<bool> nominated;
+ std::optional<uint64_t> priority;
+ std::optional<bool> nominated;
// `writable` does not exist in the spec and old comments suggest it used to
// exist but was incorrectly implemented.
// TODO(https://crbug.com/webrtc/14171): Standardize and/or modify
// implementation.
- absl::optional<bool> writable;
- absl::optional<uint64_t> packets_sent;
- absl::optional<uint64_t> packets_received;
- absl::optional<uint64_t> bytes_sent;
- absl::optional<uint64_t> bytes_received;
- absl::optional<double> total_round_trip_time;
- absl::optional<double> current_round_trip_time;
- absl::optional<double> available_outgoing_bitrate;
- absl::optional<double> available_incoming_bitrate;
- absl::optional<uint64_t> requests_received;
- absl::optional<uint64_t> requests_sent;
- absl::optional<uint64_t> responses_received;
- absl::optional<uint64_t> responses_sent;
- absl::optional<uint64_t> consent_requests_sent;
- absl::optional<uint64_t> packets_discarded_on_send;
- absl::optional<uint64_t> bytes_discarded_on_send;
- absl::optional<double> last_packet_received_timestamp;
- absl::optional<double> last_packet_sent_timestamp;
+ std::optional<bool> writable;
+ std::optional<uint64_t> packets_sent;
+ std::optional<uint64_t> packets_received;
+ std::optional<uint64_t> bytes_sent;
+ std::optional<uint64_t> bytes_received;
+ std::optional<double> total_round_trip_time;
+ std::optional<double> current_round_trip_time;
+ std::optional<double> available_outgoing_bitrate;
+ std::optional<double> available_incoming_bitrate;
+ std::optional<uint64_t> requests_received;
+ std::optional<uint64_t> requests_sent;
+ std::optional<uint64_t> responses_received;
+ std::optional<uint64_t> responses_sent;
+ std::optional<uint64_t> consent_requests_sent;
+ std::optional<uint64_t> packets_discarded_on_send;
+ std::optional<uint64_t> bytes_discarded_on_send;
+ std::optional<double> last_packet_received_timestamp;
+ std::optional<double> last_packet_sent_timestamp;
};
// https://w3c.github.io/webrtc-stats/#icecandidate-dict*
@@ -113,28 +113,28 @@
WEBRTC_RTCSTATS_DECL();
~RTCIceCandidateStats() override;
- absl::optional<std::string> transport_id;
+ std::optional<std::string> transport_id;
// Obsolete: is_remote
- absl::optional<bool> is_remote;
- absl::optional<std::string> network_type;
- absl::optional<std::string> ip;
- absl::optional<std::string> address;
- absl::optional<int32_t> port;
- absl::optional<std::string> protocol;
- absl::optional<std::string> relay_protocol;
- absl::optional<std::string> candidate_type;
- absl::optional<int32_t> priority;
- absl::optional<std::string> url;
- absl::optional<std::string> foundation;
- absl::optional<std::string> related_address;
- absl::optional<int32_t> related_port;
- absl::optional<std::string> username_fragment;
- absl::optional<std::string> tcp_type;
+ std::optional<bool> is_remote;
+ std::optional<std::string> network_type;
+ std::optional<std::string> ip;
+ std::optional<std::string> address;
+ std::optional<int32_t> port;
+ std::optional<std::string> protocol;
+ std::optional<std::string> relay_protocol;
+ std::optional<std::string> candidate_type;
+ std::optional<int32_t> priority;
+ std::optional<std::string> url;
+ std::optional<std::string> foundation;
+ std::optional<std::string> related_address;
+ std::optional<int32_t> related_port;
+ std::optional<std::string> username_fragment;
+ std::optional<std::string> tcp_type;
// The following metrics are NOT exposed to JavaScript. We should consider
// standardizing or removing them.
- absl::optional<bool> vpn;
- absl::optional<std::string> network_adapter_type;
+ std::optional<bool> vpn;
+ std::optional<std::string> network_adapter_type;
protected:
RTCIceCandidateStats(std::string id, Timestamp timestamp, bool is_remote);
@@ -169,8 +169,8 @@
RTCPeerConnectionStats(std::string id, Timestamp timestamp);
~RTCPeerConnectionStats() override;
- absl::optional<uint32_t> data_channels_opened;
- absl::optional<uint32_t> data_channels_closed;
+ std::optional<uint32_t> data_channels_opened;
+ std::optional<uint32_t> data_channels_closed;
};
// https://w3c.github.io/webrtc-stats/#streamstats-dict*
@@ -179,10 +179,10 @@
WEBRTC_RTCSTATS_DECL();
~RTCRtpStreamStats() override;
- absl::optional<uint32_t> ssrc;
- absl::optional<std::string> kind;
- absl::optional<std::string> transport_id;
- absl::optional<std::string> codec_id;
+ std::optional<uint32_t> ssrc;
+ std::optional<std::string> kind;
+ std::optional<std::string> transport_id;
+ std::optional<std::string> codec_id;
protected:
RTCRtpStreamStats(std::string id, Timestamp timestamp);
@@ -194,8 +194,8 @@
WEBRTC_RTCSTATS_DECL();
~RTCReceivedRtpStreamStats() override;
- absl::optional<double> jitter;
- absl::optional<int32_t> packets_lost; // Signed per RFC 3550
+ std::optional<double> jitter;
+ std::optional<int32_t> packets_lost; // Signed per RFC 3550
protected:
RTCReceivedRtpStreamStats(std::string id, Timestamp timestamp);
@@ -207,8 +207,8 @@
WEBRTC_RTCSTATS_DECL();
~RTCSentRtpStreamStats() override;
- absl::optional<uint64_t> packets_sent;
- absl::optional<uint64_t> bytes_sent;
+ std::optional<uint64_t> packets_sent;
+ std::optional<uint64_t> bytes_sent;
protected:
RTCSentRtpStreamStats(std::string id, Timestamp timestamp);
@@ -222,51 +222,51 @@
RTCInboundRtpStreamStats(std::string id, Timestamp timestamp);
~RTCInboundRtpStreamStats() override;
- absl::optional<std::string> playout_id;
- absl::optional<std::string> track_identifier;
- absl::optional<std::string> mid;
- absl::optional<std::string> remote_id;
- absl::optional<uint32_t> packets_received;
- absl::optional<uint64_t> packets_discarded;
- absl::optional<uint64_t> fec_packets_received;
- absl::optional<uint64_t> fec_bytes_received;
- absl::optional<uint64_t> fec_packets_discarded;
+ std::optional<std::string> playout_id;
+ std::optional<std::string> track_identifier;
+ std::optional<std::string> mid;
+ std::optional<std::string> remote_id;
+ std::optional<uint32_t> packets_received;
+ std::optional<uint64_t> packets_discarded;
+ std::optional<uint64_t> fec_packets_received;
+ std::optional<uint64_t> fec_bytes_received;
+ std::optional<uint64_t> fec_packets_discarded;
// Inbound FEC SSRC. Only present if a mechanism like FlexFEC is negotiated.
- absl::optional<uint32_t> fec_ssrc;
- absl::optional<uint64_t> bytes_received;
- absl::optional<uint64_t> header_bytes_received;
+ std::optional<uint32_t> fec_ssrc;
+ std::optional<uint64_t> bytes_received;
+ std::optional<uint64_t> header_bytes_received;
// Inbound RTX stats. Only defined when RTX is used and it is therefore
// possible to distinguish retransmissions.
- absl::optional<uint64_t> retransmitted_packets_received;
- absl::optional<uint64_t> retransmitted_bytes_received;
- absl::optional<uint32_t> rtx_ssrc;
+ std::optional<uint64_t> retransmitted_packets_received;
+ std::optional<uint64_t> retransmitted_bytes_received;
+ std::optional<uint32_t> rtx_ssrc;
- absl::optional<double> last_packet_received_timestamp;
- absl::optional<double> jitter_buffer_delay;
- absl::optional<double> jitter_buffer_target_delay;
- absl::optional<double> jitter_buffer_minimum_delay;
- absl::optional<uint64_t> jitter_buffer_emitted_count;
- absl::optional<uint64_t> total_samples_received;
- absl::optional<uint64_t> concealed_samples;
- absl::optional<uint64_t> silent_concealed_samples;
- absl::optional<uint64_t> concealment_events;
- absl::optional<uint64_t> inserted_samples_for_deceleration;
- absl::optional<uint64_t> removed_samples_for_acceleration;
- absl::optional<double> audio_level;
- absl::optional<double> total_audio_energy;
- absl::optional<double> total_samples_duration;
+ std::optional<double> last_packet_received_timestamp;
+ std::optional<double> jitter_buffer_delay;
+ std::optional<double> jitter_buffer_target_delay;
+ std::optional<double> jitter_buffer_minimum_delay;
+ std::optional<uint64_t> jitter_buffer_emitted_count;
+ std::optional<uint64_t> total_samples_received;
+ std::optional<uint64_t> concealed_samples;
+ std::optional<uint64_t> silent_concealed_samples;
+ std::optional<uint64_t> concealment_events;
+ std::optional<uint64_t> inserted_samples_for_deceleration;
+ std::optional<uint64_t> removed_samples_for_acceleration;
+ std::optional<double> audio_level;
+ std::optional<double> total_audio_energy;
+ std::optional<double> total_samples_duration;
// Stats below are only implemented or defined for video.
- absl::optional<uint32_t> frames_received;
- absl::optional<uint32_t> frame_width;
- absl::optional<uint32_t> frame_height;
- absl::optional<double> frames_per_second;
- absl::optional<uint32_t> frames_decoded;
- absl::optional<uint32_t> key_frames_decoded;
- absl::optional<uint32_t> frames_dropped;
- absl::optional<double> total_decode_time;
- absl::optional<double> total_processing_delay;
- absl::optional<double> total_assembly_time;
- absl::optional<uint32_t> frames_assembled_from_multiple_packets;
+ std::optional<uint32_t> frames_received;
+ std::optional<uint32_t> frame_width;
+ std::optional<uint32_t> frame_height;
+ std::optional<double> frames_per_second;
+ std::optional<uint32_t> frames_decoded;
+ std::optional<uint32_t> key_frames_decoded;
+ std::optional<uint32_t> frames_dropped;
+ std::optional<double> total_decode_time;
+ std::optional<double> total_processing_delay;
+ std::optional<double> total_assembly_time;
+ std::optional<uint32_t> frames_assembled_from_multiple_packets;
// TODO(https://crbug.com/webrtc/15600): Implement framesRendered, which is
// incremented at the same time that totalInterFrameDelay and
// totalSquaredInterFrameDelay is incremented. (Dividing inter-frame delay by
@@ -278,43 +278,43 @@
// at delivery to sink, not at actual render time. When we have an actual
// frame rendered callback, move the calculating of these metrics to there in
// order to make them more accurate.
- absl::optional<double> total_inter_frame_delay;
- absl::optional<double> total_squared_inter_frame_delay;
- absl::optional<uint32_t> pause_count;
- absl::optional<double> total_pauses_duration;
- absl::optional<uint32_t> freeze_count;
- absl::optional<double> total_freezes_duration;
+ std::optional<double> total_inter_frame_delay;
+ std::optional<double> total_squared_inter_frame_delay;
+ std::optional<uint32_t> pause_count;
+ std::optional<double> total_pauses_duration;
+ std::optional<uint32_t> freeze_count;
+ std::optional<double> total_freezes_duration;
// https://w3c.github.io/webrtc-provisional-stats/#dom-rtcinboundrtpstreamstats-contenttype
- absl::optional<std::string> content_type;
+ std::optional<std::string> content_type;
// Only populated if audio/video sync is enabled.
// TODO(https://crbug.com/webrtc/14177): Expose even if A/V sync is off?
- absl::optional<double> estimated_playout_timestamp;
+ std::optional<double> estimated_playout_timestamp;
// Only defined for video.
// In JavaScript, this is only exposed if HW exposure is allowed.
- absl::optional<std::string> decoder_implementation;
+ std::optional<std::string> decoder_implementation;
// FIR and PLI counts are only defined for |kind == "video"|.
- absl::optional<uint32_t> fir_count;
- absl::optional<uint32_t> pli_count;
- absl::optional<uint32_t> nack_count;
- absl::optional<uint64_t> qp_sum;
+ std::optional<uint32_t> fir_count;
+ std::optional<uint32_t> pli_count;
+ std::optional<uint32_t> nack_count;
+ std::optional<uint64_t> qp_sum;
// This is a remnant of the legacy getStats() API. When the "video-timing"
// header extension is used,
// https://webrtc.github.io/webrtc-org/experiments/rtp-hdrext/video-timing/,
// `googTimingFrameInfo` is exposed with the value of
// TimingFrameInfo::ToString().
// TODO(https://crbug.com/webrtc/14586): Unship or standardize this metric.
- absl::optional<std::string> goog_timing_frame_info;
+ std::optional<std::string> goog_timing_frame_info;
// In JavaScript, this is only exposed if HW exposure is allowed.
- absl::optional<bool> power_efficient_decoder;
+ std::optional<bool> power_efficient_decoder;
// The following metrics are NOT exposed to JavaScript. We should consider
// standardizing or removing them.
- absl::optional<uint64_t> jitter_buffer_flushes;
- absl::optional<uint64_t> delayed_packet_outage_samples;
- absl::optional<double> relative_packet_arrival_delay;
- absl::optional<uint32_t> interruption_count;
- absl::optional<double> total_interruption_duration;
- absl::optional<double> min_playout_delay;
+ std::optional<uint64_t> jitter_buffer_flushes;
+ std::optional<uint64_t> delayed_packet_outage_samples;
+ std::optional<double> relative_packet_arrival_delay;
+ std::optional<uint32_t> interruption_count;
+ std::optional<double> total_interruption_duration;
+ std::optional<double> min_playout_delay;
};
// https://w3c.github.io/webrtc-stats/#outboundrtpstats-dict*
@@ -325,46 +325,46 @@
RTCOutboundRtpStreamStats(std::string id, Timestamp timestamp);
~RTCOutboundRtpStreamStats() override;
- absl::optional<std::string> media_source_id;
- absl::optional<std::string> remote_id;
- absl::optional<std::string> mid;
- absl::optional<std::string> rid;
- absl::optional<uint64_t> retransmitted_packets_sent;
- absl::optional<uint64_t> header_bytes_sent;
- absl::optional<uint64_t> retransmitted_bytes_sent;
- absl::optional<double> target_bitrate;
- absl::optional<uint32_t> frames_encoded;
- absl::optional<uint32_t> key_frames_encoded;
- absl::optional<double> total_encode_time;
- absl::optional<uint64_t> total_encoded_bytes_target;
- absl::optional<uint32_t> frame_width;
- absl::optional<uint32_t> frame_height;
- absl::optional<double> frames_per_second;
- absl::optional<uint32_t> frames_sent;
- absl::optional<uint32_t> huge_frames_sent;
- absl::optional<double> total_packet_send_delay;
- absl::optional<std::string> quality_limitation_reason;
- absl::optional<std::map<std::string, double>> quality_limitation_durations;
+ std::optional<std::string> media_source_id;
+ std::optional<std::string> remote_id;
+ std::optional<std::string> mid;
+ std::optional<std::string> rid;
+ std::optional<uint64_t> retransmitted_packets_sent;
+ std::optional<uint64_t> header_bytes_sent;
+ std::optional<uint64_t> retransmitted_bytes_sent;
+ std::optional<double> target_bitrate;
+ std::optional<uint32_t> frames_encoded;
+ std::optional<uint32_t> key_frames_encoded;
+ std::optional<double> total_encode_time;
+ std::optional<uint64_t> total_encoded_bytes_target;
+ std::optional<uint32_t> frame_width;
+ std::optional<uint32_t> frame_height;
+ std::optional<double> frames_per_second;
+ std::optional<uint32_t> frames_sent;
+ std::optional<uint32_t> huge_frames_sent;
+ std::optional<double> total_packet_send_delay;
+ std::optional<std::string> quality_limitation_reason;
+ std::optional<std::map<std::string, double>> quality_limitation_durations;
// https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-qualitylimitationresolutionchanges
- absl::optional<uint32_t> quality_limitation_resolution_changes;
+ std::optional<uint32_t> quality_limitation_resolution_changes;
// https://w3c.github.io/webrtc-provisional-stats/#dom-rtcoutboundrtpstreamstats-contenttype
- absl::optional<std::string> content_type;
+ std::optional<std::string> content_type;
// In JavaScript, this is only exposed if HW exposure is allowed.
// Only implemented for video.
// TODO(https://crbug.com/webrtc/14178): Implement for audio as well.
- absl::optional<std::string> encoder_implementation;
+ std::optional<std::string> encoder_implementation;
// FIR and PLI counts are only defined for |kind == "video"|.
- absl::optional<uint32_t> fir_count;
- absl::optional<uint32_t> pli_count;
- absl::optional<uint32_t> nack_count;
- absl::optional<uint64_t> qp_sum;
- absl::optional<bool> active;
+ std::optional<uint32_t> fir_count;
+ std::optional<uint32_t> pli_count;
+ std::optional<uint32_t> nack_count;
+ std::optional<uint64_t> qp_sum;
+ std::optional<bool> active;
// In JavaScript, this is only exposed if HW exposure is allowed.
- absl::optional<bool> power_efficient_encoder;
- absl::optional<std::string> scalability_mode;
+ std::optional<bool> power_efficient_encoder;
+ std::optional<std::string> scalability_mode;
// RTX ssrc. Only present if RTX is negotiated.
- absl::optional<uint32_t> rtx_ssrc;
+ std::optional<uint32_t> rtx_ssrc;
};
// https://w3c.github.io/webrtc-stats/#remoteinboundrtpstats-dict*
@@ -375,11 +375,11 @@
RTCRemoteInboundRtpStreamStats(std::string id, Timestamp timestamp);
~RTCRemoteInboundRtpStreamStats() override;
- absl::optional<std::string> local_id;
- absl::optional<double> round_trip_time;
- absl::optional<double> fraction_lost;
- absl::optional<double> total_round_trip_time;
- absl::optional<int32_t> round_trip_time_measurements;
+ std::optional<std::string> local_id;
+ std::optional<double> round_trip_time;
+ std::optional<double> fraction_lost;
+ std::optional<double> total_round_trip_time;
+ std::optional<int32_t> round_trip_time_measurements;
};
// https://w3c.github.io/webrtc-stats/#remoteoutboundrtpstats-dict*
@@ -390,12 +390,12 @@
RTCRemoteOutboundRtpStreamStats(std::string id, Timestamp timestamp);
~RTCRemoteOutboundRtpStreamStats() override;
- absl::optional<std::string> local_id;
- absl::optional<double> remote_timestamp;
- absl::optional<uint64_t> reports_sent;
- absl::optional<double> round_trip_time;
- absl::optional<uint64_t> round_trip_time_measurements;
- absl::optional<double> total_round_trip_time;
+ std::optional<std::string> local_id;
+ std::optional<double> remote_timestamp;
+ std::optional<uint64_t> reports_sent;
+ std::optional<double> round_trip_time;
+ std::optional<uint64_t> round_trip_time_measurements;
+ std::optional<double> total_round_trip_time;
};
// https://w3c.github.io/webrtc-stats/#dom-rtcmediasourcestats
@@ -404,8 +404,8 @@
WEBRTC_RTCSTATS_DECL();
~RTCMediaSourceStats() override;
- absl::optional<std::string> track_identifier;
- absl::optional<std::string> kind;
+ std::optional<std::string> track_identifier;
+ std::optional<std::string> kind;
protected:
RTCMediaSourceStats(std::string id, Timestamp timestamp);
@@ -418,11 +418,11 @@
RTCAudioSourceStats(std::string id, Timestamp timestamp);
~RTCAudioSourceStats() override;
- absl::optional<double> audio_level;
- absl::optional<double> total_audio_energy;
- absl::optional<double> total_samples_duration;
- absl::optional<double> echo_return_loss;
- absl::optional<double> echo_return_loss_enhancement;
+ std::optional<double> audio_level;
+ std::optional<double> total_audio_energy;
+ std::optional<double> total_samples_duration;
+ std::optional<double> echo_return_loss;
+ std::optional<double> echo_return_loss_enhancement;
};
// https://w3c.github.io/webrtc-stats/#dom-rtcvideosourcestats
@@ -432,10 +432,10 @@
RTCVideoSourceStats(std::string id, Timestamp timestamp);
~RTCVideoSourceStats() override;
- absl::optional<uint32_t> width;
- absl::optional<uint32_t> height;
- absl::optional<uint32_t> frames;
- absl::optional<double> frames_per_second;
+ std::optional<uint32_t> width;
+ std::optional<uint32_t> height;
+ std::optional<uint32_t> frames;
+ std::optional<double> frames_per_second;
};
// https://w3c.github.io/webrtc-stats/#transportstats-dict*
@@ -445,23 +445,23 @@
RTCTransportStats(std::string id, Timestamp timestamp);
~RTCTransportStats() override;
- absl::optional<uint64_t> bytes_sent;
- absl::optional<uint64_t> packets_sent;
- absl::optional<uint64_t> bytes_received;
- absl::optional<uint64_t> packets_received;
- absl::optional<std::string> rtcp_transport_stats_id;
- absl::optional<std::string> dtls_state;
- absl::optional<std::string> selected_candidate_pair_id;
- absl::optional<std::string> local_certificate_id;
- absl::optional<std::string> remote_certificate_id;
- absl::optional<std::string> tls_version;
- absl::optional<std::string> dtls_cipher;
- absl::optional<std::string> dtls_role;
- absl::optional<std::string> srtp_cipher;
- absl::optional<uint32_t> selected_candidate_pair_changes;
- absl::optional<std::string> ice_role;
- absl::optional<std::string> ice_local_username_fragment;
- absl::optional<std::string> ice_state;
+ std::optional<uint64_t> bytes_sent;
+ std::optional<uint64_t> packets_sent;
+ std::optional<uint64_t> bytes_received;
+ std::optional<uint64_t> packets_received;
+ std::optional<std::string> rtcp_transport_stats_id;
+ std::optional<std::string> dtls_state;
+ std::optional<std::string> selected_candidate_pair_id;
+ std::optional<std::string> local_certificate_id;
+ std::optional<std::string> remote_certificate_id;
+ std::optional<std::string> tls_version;
+ std::optional<std::string> dtls_cipher;
+ std::optional<std::string> dtls_role;
+ std::optional<std::string> srtp_cipher;
+ std::optional<uint32_t> selected_candidate_pair_changes;
+ std::optional<std::string> ice_role;
+ std::optional<std::string> ice_local_username_fragment;
+ std::optional<std::string> ice_state;
};
// https://w3c.github.io/webrtc-stats/#playoutstats-dict*
@@ -471,12 +471,12 @@
RTCAudioPlayoutStats(const std::string& id, Timestamp timestamp);
~RTCAudioPlayoutStats() override;
- absl::optional<std::string> kind;
- absl::optional<double> synthesized_samples_duration;
- absl::optional<uint64_t> synthesized_samples_events;
- absl::optional<double> total_samples_duration;
- absl::optional<double> total_playout_delay;
- absl::optional<uint64_t> total_samples_count;
+ std::optional<std::string> kind;
+ std::optional<double> synthesized_samples_duration;
+ std::optional<uint64_t> synthesized_samples_events;
+ std::optional<double> total_samples_duration;
+ std::optional<double> total_playout_delay;
+ std::optional<uint64_t> total_samples_count;
};
} // namespace webrtc
diff --git a/api/test/create_frame_generator.cc b/api/test/create_frame_generator.cc
index f0d8e05..5750835 100644
--- a/api/test/create_frame_generator.cc
+++ b/api/test/create_frame_generator.cc
@@ -13,12 +13,12 @@
#include <cstdint>
#include <cstdio>
#include <memory>
+#include <optional>
#include <string>
#include <vector>
#include "absl/base/nullability.h"
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/environment/environment.h"
#include "api/environment/environment_factory.h"
#include "api/test/frame_generator_interface.h"
@@ -32,8 +32,8 @@
std::unique_ptr<FrameGeneratorInterface> CreateSquareFrameGenerator(
int width,
int height,
- absl::optional<FrameGeneratorInterface::OutputType> type,
- absl::optional<int> num_squares) {
+ std::optional<FrameGeneratorInterface::OutputType> type,
+ std::optional<int> num_squares) {
return std::make_unique<SquareGenerator>(
width, height, type.value_or(FrameGeneratorInterface::OutputType::kI420),
num_squares.value_or(10));
diff --git a/api/test/create_frame_generator.h b/api/test/create_frame_generator.h
index 14c88d3..6e9a882 100644
--- a/api/test/create_frame_generator.h
+++ b/api/test/create_frame_generator.h
@@ -14,12 +14,12 @@
#include <cstddef>
#include <cstdint>
#include <memory>
+#include <optional>
#include <string>
#include <vector>
#include "absl/base/nullability.h"
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/environment/environment.h"
#include "api/test/frame_generator_interface.h"
#include "system_wrappers/include/clock.h"
@@ -34,8 +34,8 @@
std::unique_ptr<FrameGeneratorInterface> CreateSquareFrameGenerator(
int width,
int height,
- absl::optional<FrameGeneratorInterface::OutputType> type,
- absl::optional<int> num_squares);
+ std::optional<FrameGeneratorInterface::OutputType> type,
+ std::optional<int> num_squares);
// Creates a frame generator that repeatedly plays a set of yuv files.
// The frame_repeat_count determines how many times each frame is shown,
diff --git a/api/test/create_peer_connection_quality_test_frame_generator.cc b/api/test/create_peer_connection_quality_test_frame_generator.cc
index d4aa46d..b99f1bd 100644
--- a/api/test/create_peer_connection_quality_test_frame_generator.cc
+++ b/api/test/create_peer_connection_quality_test_frame_generator.cc
@@ -11,11 +11,11 @@
#include "api/test/create_peer_connection_quality_test_frame_generator.h"
#include <memory>
+#include <optional>
#include <string>
#include <utility>
#include <vector>
-#include "absl/types/optional.h"
#include "api/test/create_frame_generator.h"
#include "api/test/frame_generator_interface.h"
#include "api/test/pclf/media_configuration.h"
@@ -55,9 +55,9 @@
std::unique_ptr<test::FrameGeneratorInterface> CreateSquareFrameGenerator(
const VideoConfig& video_config,
- absl::optional<test::FrameGeneratorInterface::OutputType> type) {
+ std::optional<test::FrameGeneratorInterface::OutputType> type) {
return test::CreateSquareFrameGenerator(
- video_config.width, video_config.height, std::move(type), absl::nullopt);
+ video_config.width, video_config.height, std::move(type), std::nullopt);
}
std::unique_ptr<test::FrameGeneratorInterface> CreateFromYuvFileFrameGenerator(
diff --git a/api/test/create_peer_connection_quality_test_frame_generator.h b/api/test/create_peer_connection_quality_test_frame_generator.h
index 62043d1..a851507 100644
--- a/api/test/create_peer_connection_quality_test_frame_generator.h
+++ b/api/test/create_peer_connection_quality_test_frame_generator.h
@@ -11,9 +11,9 @@
#define API_TEST_CREATE_PEER_CONNECTION_QUALITY_TEST_FRAME_GENERATOR_H_
#include <memory>
+#include <optional>
#include <string>
-#include "absl/types/optional.h"
#include "api/test/frame_generator_interface.h"
#include "api/test/pclf/media_configuration.h"
@@ -26,7 +26,7 @@
// weight and height.
std::unique_ptr<test::FrameGeneratorInterface> CreateSquareFrameGenerator(
const VideoConfig& video_config,
- absl::optional<test::FrameGeneratorInterface::OutputType> type);
+ std::optional<test::FrameGeneratorInterface::OutputType> type);
// Creates a frame generator that plays frames from the yuv file.
std::unique_ptr<test::FrameGeneratorInterface> CreateFromYuvFileFrameGenerator(
diff --git a/api/test/frame_generator_interface.h b/api/test/frame_generator_interface.h
index 169ead8..ed03bce 100644
--- a/api/test/frame_generator_interface.h
+++ b/api/test/frame_generator_interface.h
@@ -12,9 +12,9 @@
#define API_TEST_FRAME_GENERATOR_INTERFACE_H_
#include <cstddef>
+#include <optional>
#include <utility>
-#include "absl/types/optional.h"
#include "api/scoped_refptr.h"
#include "api/video/video_frame.h"
#include "api/video/video_frame_buffer.h"
@@ -30,11 +30,11 @@
};
struct VideoFrameData {
VideoFrameData(rtc::scoped_refptr<VideoFrameBuffer> buffer,
- absl::optional<VideoFrame::UpdateRect> update_rect)
+ std::optional<VideoFrame::UpdateRect> update_rect)
: buffer(std::move(buffer)), update_rect(update_rect) {}
rtc::scoped_refptr<VideoFrameBuffer> buffer;
- absl::optional<VideoFrame::UpdateRect> update_rect;
+ std::optional<VideoFrame::UpdateRect> update_rect;
};
enum class OutputType { kI420, kI420A, kI010, kNV12 };
@@ -56,8 +56,8 @@
// Returns the frames per second this generator is supposed to provide
// according to its data source. Not all frame generators know the frames per
- // second of the data source, in such case this method returns absl::nullopt.
- virtual absl::optional<int> fps() const = 0;
+ // second of the data source, in such case this method returns std::nullopt.
+ virtual std::optional<int> fps() const = 0;
};
} // namespace test
diff --git a/api/test/metrics/BUILD.gn b/api/test/metrics/BUILD.gn
index dbbb38a..04a28f4 100644
--- a/api/test/metrics/BUILD.gn
+++ b/api/test/metrics/BUILD.gn
@@ -49,10 +49,7 @@
"metric.cc",
"metric.h",
]
- deps = [
- "../../../api/units:timestamp",
- "//third_party/abseil-cpp/absl/types:optional",
- ]
+ deps = [ "../../../api/units:timestamp" ]
}
rtc_library("metrics_logger") {
@@ -71,7 +68,6 @@
"../../numerics",
"../../units:timestamp",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -111,7 +107,6 @@
":metrics_exporter",
"../..:array_view",
"../../../rtc_base:stringutils",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -203,7 +198,6 @@
"../../../test:test_support",
"../../numerics",
"../../units:timestamp",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -215,7 +209,6 @@
":metrics_accumulator",
"../../../test:test_support",
"../../units:timestamp",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -251,7 +244,6 @@
":metrics_logger",
"../../../system_wrappers",
"../../../test:test_support",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
diff --git a/api/test/metrics/global_metrics_logger_and_exporter_test.cc b/api/test/metrics/global_metrics_logger_and_exporter_test.cc
index 567b3da..175fa6e 100644
--- a/api/test/metrics/global_metrics_logger_and_exporter_test.cc
+++ b/api/test/metrics/global_metrics_logger_and_exporter_test.cc
@@ -11,11 +11,11 @@
#include <map>
#include <memory>
+#include <optional>
#include <string>
#include <utility>
#include <vector>
-#include "absl/types/optional.h"
#include "api/test/metrics/metric.h"
#include "api/test/metrics/metrics_exporter.h"
#include "api/test/metrics/metrics_logger.h"
@@ -91,10 +91,10 @@
EXPECT_THAT(metric.time_series.samples[0].value, Eq(10.0));
EXPECT_THAT(metric.time_series.samples[0].sample_metadata,
Eq(std::map<std::string, std::string>{}));
- ASSERT_THAT(metric.stats.mean, absl::optional<double>(10.0));
- ASSERT_THAT(metric.stats.stddev, absl::nullopt);
- ASSERT_THAT(metric.stats.min, absl::optional<double>(10.0));
- ASSERT_THAT(metric.stats.max, absl::optional<double>(10.0));
+ ASSERT_THAT(metric.stats.mean, std::optional<double>(10.0));
+ ASSERT_THAT(metric.stats.stddev, std::nullopt);
+ ASSERT_THAT(metric.stats.min, std::optional<double>(10.0));
+ ASSERT_THAT(metric.stats.max, std::optional<double>(10.0));
}
TEST(ExportPerfMetricTest, OneFailedExporterDoesNotPreventExportToOthers) {
diff --git a/api/test/metrics/metric.h b/api/test/metrics/metric.h
index 17c1755..ee5680e 100644
--- a/api/test/metrics/metric.h
+++ b/api/test/metrics/metric.h
@@ -12,10 +12,10 @@
#define API_TEST_METRICS_METRIC_H_
#include <map>
+#include <optional>
#include <string>
#include <vector>
-#include "absl/types/optional.h"
#include "api/units/timestamp.h"
namespace webrtc {
@@ -66,12 +66,12 @@
struct Stats {
// Sample mean of the metric
// (https://en.wikipedia.org/wiki/Sample_mean_and_covariance).
- absl::optional<double> mean;
+ std::optional<double> mean;
// Standard deviation (https://en.wikipedia.org/wiki/Standard_deviation).
// Is undefined if `time_series` contains only a single value.
- absl::optional<double> stddev;
- absl::optional<double> min;
- absl::optional<double> max;
+ std::optional<double> stddev;
+ std::optional<double> min;
+ std::optional<double> max;
};
// Metric name, for example PSNR, SSIM, decode_time, etc.
diff --git a/api/test/metrics/metrics_accumulator_test.cc b/api/test/metrics/metrics_accumulator_test.cc
index 6b3efea..9f88395 100644
--- a/api/test/metrics/metrics_accumulator_test.cc
+++ b/api/test/metrics/metrics_accumulator_test.cc
@@ -10,10 +10,10 @@
#include "api/test/metrics/metrics_accumulator.h"
#include <map>
+#include <optional>
#include <string>
#include <vector>
-#include "absl/types/optional.h"
#include "api/test/metrics/metric.h"
#include "api/units/timestamp.h"
#include "test/gmock.h"
@@ -49,10 +49,10 @@
Eq(Timestamp::Seconds(1)));
EXPECT_THAT(metric.time_series.samples[0].sample_metadata,
Eq(std::map<std::string, std::string>{{"key", "value"}}));
- ASSERT_THAT(metric.stats.mean, absl::optional<double>(10.0));
- ASSERT_THAT(metric.stats.stddev, absl::optional<double>(0.0));
- ASSERT_THAT(metric.stats.min, absl::optional<double>(10.0));
- ASSERT_THAT(metric.stats.max, absl::optional<double>(10.0));
+ ASSERT_THAT(metric.stats.mean, std::optional<double>(10.0));
+ ASSERT_THAT(metric.stats.stddev, std::optional<double>(0.0));
+ ASSERT_THAT(metric.stats.min, std::optional<double>(10.0));
+ ASSERT_THAT(metric.stats.max, std::optional<double>(10.0));
}
TEST(MetricsAccumulatorTest, AddSamplesToExistingMetricWontCreateNewOne) {
@@ -88,10 +88,10 @@
Eq(Timestamp::Seconds(2)));
EXPECT_THAT(metric.time_series.samples[1].sample_metadata,
Eq(std::map<std::string, std::string>{{"key2", "value2"}}));
- ASSERT_THAT(metric.stats.mean, absl::optional<double>(15.0));
- ASSERT_THAT(metric.stats.stddev, absl::optional<double>(5.0));
- ASSERT_THAT(metric.stats.min, absl::optional<double>(10.0));
- ASSERT_THAT(metric.stats.max, absl::optional<double>(20.0));
+ ASSERT_THAT(metric.stats.mean, std::optional<double>(15.0));
+ ASSERT_THAT(metric.stats.stddev, std::optional<double>(5.0));
+ ASSERT_THAT(metric.stats.min, std::optional<double>(10.0));
+ ASSERT_THAT(metric.stats.max, std::optional<double>(20.0));
}
TEST(MetricsAccumulatorTest, AddSampleToDifferentMetricsWillCreateBoth) {
@@ -121,10 +121,10 @@
Eq(Timestamp::Seconds(1)));
EXPECT_THAT(metrics[0].time_series.samples[0].sample_metadata,
Eq(std::map<std::string, std::string>{{"key1", "value1"}}));
- ASSERT_THAT(metrics[0].stats.mean, absl::optional<double>(10.0));
- ASSERT_THAT(metrics[0].stats.stddev, absl::optional<double>(0.0));
- ASSERT_THAT(metrics[0].stats.min, absl::optional<double>(10.0));
- ASSERT_THAT(metrics[0].stats.max, absl::optional<double>(10.0));
+ ASSERT_THAT(metrics[0].stats.mean, std::optional<double>(10.0));
+ ASSERT_THAT(metrics[0].stats.stddev, std::optional<double>(0.0));
+ ASSERT_THAT(metrics[0].stats.min, std::optional<double>(10.0));
+ ASSERT_THAT(metrics[0].stats.max, std::optional<double>(10.0));
EXPECT_THAT(metrics[1].name, Eq("metric_name2"));
EXPECT_THAT(metrics[1].test_case, Eq("test_case_name2"));
EXPECT_THAT(metrics[1].unit, Eq(Unit::kUnitless));
@@ -137,10 +137,10 @@
Eq(Timestamp::Seconds(2)));
EXPECT_THAT(metrics[1].time_series.samples[0].sample_metadata,
Eq(std::map<std::string, std::string>{{"key2", "value2"}}));
- ASSERT_THAT(metrics[1].stats.mean, absl::optional<double>(20.0));
- ASSERT_THAT(metrics[1].stats.stddev, absl::optional<double>(0.0));
- ASSERT_THAT(metrics[1].stats.min, absl::optional<double>(20.0));
- ASSERT_THAT(metrics[1].stats.max, absl::optional<double>(20.0));
+ ASSERT_THAT(metrics[1].stats.mean, std::optional<double>(20.0));
+ ASSERT_THAT(metrics[1].stats.stddev, std::optional<double>(0.0));
+ ASSERT_THAT(metrics[1].stats.min, std::optional<double>(20.0));
+ ASSERT_THAT(metrics[1].stats.max, std::optional<double>(20.0));
}
TEST(MetricsAccumulatorTest, AddMetadataToTheNewMetricWillCreateOne) {
@@ -162,10 +162,10 @@
EXPECT_THAT(metric.metric_metadata,
Eq(std::map<std::string, std::string>{{"key", "value"}}));
ASSERT_THAT(metric.time_series.samples, IsEmpty());
- ASSERT_THAT(metric.stats.mean, absl::nullopt);
- ASSERT_THAT(metric.stats.stddev, absl::nullopt);
- ASSERT_THAT(metric.stats.min, absl::nullopt);
- ASSERT_THAT(metric.stats.max, absl::nullopt);
+ ASSERT_THAT(metric.stats.mean, std::nullopt);
+ ASSERT_THAT(metric.stats.stddev, std::nullopt);
+ ASSERT_THAT(metric.stats.min, std::nullopt);
+ ASSERT_THAT(metric.stats.max, std::nullopt);
}
TEST(MetricsAccumulatorTest,
@@ -194,10 +194,10 @@
EXPECT_THAT(metric.metric_metadata,
Eq(std::map<std::string, std::string>{{"key2", "value2"}}));
ASSERT_THAT(metric.time_series.samples, IsEmpty());
- ASSERT_THAT(metric.stats.mean, absl::nullopt);
- ASSERT_THAT(metric.stats.stddev, absl::nullopt);
- ASSERT_THAT(metric.stats.min, absl::nullopt);
- ASSERT_THAT(metric.stats.max, absl::nullopt);
+ ASSERT_THAT(metric.stats.mean, std::nullopt);
+ ASSERT_THAT(metric.stats.stddev, std::nullopt);
+ ASSERT_THAT(metric.stats.min, std::nullopt);
+ ASSERT_THAT(metric.stats.max, std::nullopt);
}
TEST(MetricsAccumulatorTest, AddMetadataToDifferentMetricsWillCreateBoth) {
@@ -224,10 +224,10 @@
EXPECT_THAT(metrics[0].metric_metadata,
Eq(std::map<std::string, std::string>{{"key1", "value1"}}));
ASSERT_THAT(metrics[0].time_series.samples, IsEmpty());
- ASSERT_THAT(metrics[0].stats.mean, absl::nullopt);
- ASSERT_THAT(metrics[0].stats.stddev, absl::nullopt);
- ASSERT_THAT(metrics[0].stats.min, absl::nullopt);
- ASSERT_THAT(metrics[0].stats.max, absl::nullopt);
+ ASSERT_THAT(metrics[0].stats.mean, std::nullopt);
+ ASSERT_THAT(metrics[0].stats.stddev, std::nullopt);
+ ASSERT_THAT(metrics[0].stats.min, std::nullopt);
+ ASSERT_THAT(metrics[0].stats.max, std::nullopt);
EXPECT_THAT(metrics[1].name, Eq("metric_name2"));
EXPECT_THAT(metrics[1].test_case, Eq("test_case_name2"));
EXPECT_THAT(metrics[1].unit, Eq(Unit::kBytes));
@@ -236,10 +236,10 @@
EXPECT_THAT(metrics[1].metric_metadata,
Eq(std::map<std::string, std::string>{{"key2", "value2"}}));
ASSERT_THAT(metrics[1].time_series.samples, IsEmpty());
- ASSERT_THAT(metrics[1].stats.mean, absl::nullopt);
- ASSERT_THAT(metrics[1].stats.stddev, absl::nullopt);
- ASSERT_THAT(metrics[1].stats.min, absl::nullopt);
- ASSERT_THAT(metrics[1].stats.max, absl::nullopt);
+ ASSERT_THAT(metrics[1].stats.mean, std::nullopt);
+ ASSERT_THAT(metrics[1].stats.stddev, std::nullopt);
+ ASSERT_THAT(metrics[1].stats.min, std::nullopt);
+ ASSERT_THAT(metrics[1].stats.max, std::nullopt);
}
TEST(MetricsAccumulatorTest, AddMetadataAfterAddingSampleWontCreateNewMetric) {
@@ -271,10 +271,10 @@
Eq(Timestamp::Seconds(1)));
EXPECT_THAT(metric.time_series.samples[0].sample_metadata,
Eq(std::map<std::string, std::string>{{"key_s", "value_s"}}));
- ASSERT_THAT(metric.stats.mean, absl::optional<double>(10.0));
- ASSERT_THAT(metric.stats.stddev, absl::optional<double>(0.0));
- ASSERT_THAT(metric.stats.min, absl::optional<double>(10.0));
- ASSERT_THAT(metric.stats.max, absl::optional<double>(10.0));
+ ASSERT_THAT(metric.stats.mean, std::optional<double>(10.0));
+ ASSERT_THAT(metric.stats.stddev, std::optional<double>(0.0));
+ ASSERT_THAT(metric.stats.min, std::optional<double>(10.0));
+ ASSERT_THAT(metric.stats.max, std::optional<double>(10.0));
}
TEST(MetricsAccumulatorTest, AddSampleAfterAddingMetadataWontCreateNewMetric) {
@@ -306,10 +306,10 @@
Eq(Timestamp::Seconds(1)));
EXPECT_THAT(metric.time_series.samples[0].sample_metadata,
Eq(std::map<std::string, std::string>{{"key_s", "value_s"}}));
- ASSERT_THAT(metric.stats.mean, absl::optional<double>(10.0));
- ASSERT_THAT(metric.stats.stddev, absl::optional<double>(0.0));
- ASSERT_THAT(metric.stats.min, absl::optional<double>(10.0));
- ASSERT_THAT(metric.stats.max, absl::optional<double>(10.0));
+ ASSERT_THAT(metric.stats.mean, std::optional<double>(10.0));
+ ASSERT_THAT(metric.stats.stddev, std::optional<double>(0.0));
+ ASSERT_THAT(metric.stats.min, std::optional<double>(10.0));
+ ASSERT_THAT(metric.stats.max, std::optional<double>(10.0));
}
} // namespace
diff --git a/api/test/metrics/metrics_logger.cc b/api/test/metrics/metrics_logger.cc
index 0695d73..ac927425 100644
--- a/api/test/metrics/metrics_logger.cc
+++ b/api/test/metrics/metrics_logger.cc
@@ -10,12 +10,12 @@
#include "api/test/metrics/metrics_logger.h"
#include <map>
+#include <optional>
#include <string>
#include <utility>
#include <vector>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/numerics/samples_stats_counter.h"
#include "api/test/metrics/metric.h"
#include "api/units/timestamp.h"
@@ -55,7 +55,7 @@
Metric::TimeSeries{.samples = std::vector{Metric::TimeSeries::Sample{
.timestamp = Now(), .value = value}}},
.stats = Metric::Stats{
- .mean = value, .stddev = absl::nullopt, .min = value, .max = value}});
+ .mean = value, .stddev = std::nullopt, .min = value, .max = value}});
}
void DefaultMetricsLogger::LogMetric(
diff --git a/api/test/metrics/metrics_logger_test.cc b/api/test/metrics/metrics_logger_test.cc
index 324a758..a75fdd6 100644
--- a/api/test/metrics/metrics_logger_test.cc
+++ b/api/test/metrics/metrics_logger_test.cc
@@ -10,10 +10,10 @@
#include "api/test/metrics/metrics_logger.h"
#include <map>
+#include <optional>
#include <string>
#include <vector>
-#include "absl/types/optional.h"
#include "api/numerics/samples_stats_counter.h"
#include "api/test/metrics/metric.h"
#include "api/units/timestamp.h"
@@ -54,10 +54,10 @@
EXPECT_THAT(metric.time_series.samples[0].value, Eq(10.0));
EXPECT_THAT(metric.time_series.samples[0].sample_metadata,
Eq(std::map<std::string, std::string>{}));
- ASSERT_THAT(metric.stats.mean, absl::optional<double>(10.0));
- ASSERT_THAT(metric.stats.stddev, absl::nullopt);
- ASSERT_THAT(metric.stats.min, absl::optional<double>(10.0));
- ASSERT_THAT(metric.stats.max, absl::optional<double>(10.0));
+ ASSERT_THAT(metric.stats.mean, std::optional<double>(10.0));
+ ASSERT_THAT(metric.stats.stddev, std::nullopt);
+ ASSERT_THAT(metric.stats.min, std::optional<double>(10.0));
+ ASSERT_THAT(metric.stats.max, std::optional<double>(10.0));
}
TEST(DefaultMetricsLoggerTest, LogMetricWithSamplesStatsCounterRecordsMetric) {
@@ -95,10 +95,10 @@
EXPECT_THAT(metric.time_series.samples[1].value, Eq(20.0));
EXPECT_THAT(metric.time_series.samples[1].sample_metadata,
Eq(std::map<std::string, std::string>{{"point_key2", "value2"}}));
- ASSERT_THAT(metric.stats.mean, absl::optional<double>(15.0));
- ASSERT_THAT(metric.stats.stddev, absl::optional<double>(5.0));
- ASSERT_THAT(metric.stats.min, absl::optional<double>(10.0));
- ASSERT_THAT(metric.stats.max, absl::optional<double>(20.0));
+ ASSERT_THAT(metric.stats.mean, std::optional<double>(15.0));
+ ASSERT_THAT(metric.stats.stddev, std::optional<double>(5.0));
+ ASSERT_THAT(metric.stats.min, std::optional<double>(10.0));
+ ASSERT_THAT(metric.stats.max, std::optional<double>(20.0));
}
TEST(DefaultMetricsLoggerTest,
@@ -113,10 +113,10 @@
EXPECT_THAT(metrics[0].name, Eq("metric_name"));
EXPECT_THAT(metrics[0].test_case, Eq("test_case_name"));
EXPECT_THAT(metrics[0].time_series.samples, IsEmpty());
- ASSERT_THAT(metrics[0].stats.mean, Eq(absl::nullopt));
- ASSERT_THAT(metrics[0].stats.stddev, Eq(absl::nullopt));
- ASSERT_THAT(metrics[0].stats.min, Eq(absl::nullopt));
- ASSERT_THAT(metrics[0].stats.max, Eq(absl::nullopt));
+ ASSERT_THAT(metrics[0].stats.mean, Eq(std::nullopt));
+ ASSERT_THAT(metrics[0].stats.stddev, Eq(std::nullopt));
+ ASSERT_THAT(metrics[0].stats.min, Eq(std::nullopt));
+ ASSERT_THAT(metrics[0].stats.max, Eq(std::nullopt));
}
TEST(DefaultMetricsLoggerTest, LogMetricWithStatsRecordsMetric) {
@@ -137,10 +137,10 @@
EXPECT_THAT(metric.metric_metadata,
Eq(std::map<std::string, std::string>{{"key", "value"}}));
ASSERT_THAT(metric.time_series.samples, IsEmpty());
- ASSERT_THAT(metric.stats.mean, absl::optional<double>(15.0));
- ASSERT_THAT(metric.stats.stddev, absl::optional<double>(5.0));
- ASSERT_THAT(metric.stats.min, absl::optional<double>(10.0));
- ASSERT_THAT(metric.stats.max, absl::optional<double>(20.0));
+ ASSERT_THAT(metric.stats.mean, std::optional<double>(15.0));
+ ASSERT_THAT(metric.stats.stddev, std::optional<double>(5.0));
+ ASSERT_THAT(metric.stats.min, std::optional<double>(10.0));
+ ASSERT_THAT(metric.stats.max, std::optional<double>(20.0));
}
TEST(DefaultMetricsLoggerTest, LogSingleValueMetricRecordsMultipleMetrics) {
@@ -267,10 +267,10 @@
Eq(Timestamp::Seconds(1)));
EXPECT_THAT(metric.time_series.samples[0].sample_metadata,
Eq(std::map<std::string, std::string>{{"key", "value"}}));
- ASSERT_THAT(metric.stats.mean, absl::optional<double>(10.0));
- ASSERT_THAT(metric.stats.stddev, absl::optional<double>(0.0));
- ASSERT_THAT(metric.stats.min, absl::optional<double>(10.0));
- ASSERT_THAT(metric.stats.max, absl::optional<double>(10.0));
+ ASSERT_THAT(metric.stats.mean, std::optional<double>(10.0));
+ ASSERT_THAT(metric.stats.stddev, std::optional<double>(0.0));
+ ASSERT_THAT(metric.stats.min, std::optional<double>(10.0));
+ ASSERT_THAT(metric.stats.max, std::optional<double>(10.0));
}
TEST(DefaultMetricsLoggerTest,
@@ -300,10 +300,10 @@
Eq(Timestamp::Seconds(1)));
EXPECT_THAT(metrics[0].time_series.samples[0].sample_metadata,
Eq(std::map<std::string, std::string>{{"key_s", "value_s"}}));
- ASSERT_THAT(metrics[0].stats.mean, absl::optional<double>(10.0));
- ASSERT_THAT(metrics[0].stats.stddev, absl::optional<double>(0.0));
- ASSERT_THAT(metrics[0].stats.min, absl::optional<double>(10.0));
- ASSERT_THAT(metrics[0].stats.max, absl::optional<double>(10.0));
+ ASSERT_THAT(metrics[0].stats.mean, std::optional<double>(10.0));
+ ASSERT_THAT(metrics[0].stats.stddev, std::optional<double>(0.0));
+ ASSERT_THAT(metrics[0].stats.min, std::optional<double>(10.0));
+ ASSERT_THAT(metrics[0].stats.max, std::optional<double>(10.0));
EXPECT_THAT(metrics[1].name, Eq("metric_name1"));
EXPECT_THAT(metrics[1].test_case, Eq("test_case_name1"));
EXPECT_THAT(metrics[1].unit, Eq(Unit::kMilliseconds));
@@ -315,10 +315,10 @@
EXPECT_THAT(metrics[1].time_series.samples[0].value, Eq(10.0));
EXPECT_THAT(metrics[1].time_series.samples[0].sample_metadata,
Eq(std::map<std::string, std::string>{}));
- ASSERT_THAT(metrics[1].stats.mean, absl::optional<double>(10.0));
- ASSERT_THAT(metrics[1].stats.stddev, absl::nullopt);
- ASSERT_THAT(metrics[1].stats.min, absl::optional<double>(10.0));
- ASSERT_THAT(metrics[1].stats.max, absl::optional<double>(10.0));
+ ASSERT_THAT(metrics[1].stats.mean, std::optional<double>(10.0));
+ ASSERT_THAT(metrics[1].stats.stddev, std::nullopt);
+ ASSERT_THAT(metrics[1].stats.min, std::optional<double>(10.0));
+ ASSERT_THAT(metrics[1].stats.max, std::optional<double>(10.0));
}
} // namespace
diff --git a/api/test/metrics/stdout_metrics_exporter.cc b/api/test/metrics/stdout_metrics_exporter.cc
index 22243e7..16ea7d1 100644
--- a/api/test/metrics/stdout_metrics_exporter.cc
+++ b/api/test/metrics/stdout_metrics_exporter.cc
@@ -12,9 +12,9 @@
#include <stdio.h>
#include <cmath>
+#include <optional>
#include <string>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/test/metrics/metric.h"
#include "rtc_base/strings/string_builder.h"
diff --git a/api/test/mock_audio_sink.h b/api/test/mock_audio_sink.h
index 6218b86..dcb01cc 100644
--- a/api/test/mock_audio_sink.h
+++ b/api/test/mock_audio_sink.h
@@ -13,8 +13,8 @@
#include <cstddef>
#include <cstdint>
+#include <optional>
-#include "absl/types/optional.h"
#include "api/media_stream_interface.h"
#include "test/gmock.h"
@@ -38,7 +38,7 @@
int sample_rate,
size_t number_of_channels,
size_t number_of_frames,
- absl::optional<int64_t> absolute_capture_timestamp_ms),
+ std::optional<int64_t> absolute_capture_timestamp_ms),
(override));
};
diff --git a/api/test/mock_data_channel.h b/api/test/mock_data_channel.h
index 8b8d56a..f30fa74 100644
--- a/api/test/mock_data_channel.h
+++ b/api/test/mock_data_channel.h
@@ -12,10 +12,10 @@
#define API_TEST_MOCK_DATA_CHANNEL_H_
#include <cstdint>
+#include <optional>
#include <string>
#include "absl/functional/any_invocable.h"
-#include "absl/types/optional.h"
#include "api/data_channel_interface.h"
#include "api/priority.h"
#include "api/rtc_error.h"
@@ -43,8 +43,8 @@
MOCK_METHOD(bool, ordered, (), (const, override));
MOCK_METHOD(uint16_t, maxRetransmitTime, (), (const, override));
MOCK_METHOD(uint16_t, maxRetransmits, (), (const, override));
- MOCK_METHOD(absl::optional<int>, maxRetransmitsOpt, (), (const, override));
- MOCK_METHOD(absl::optional<int>, maxPacketLifeTime, (), (const, override));
+ MOCK_METHOD(std::optional<int>, maxRetransmitsOpt, (), (const, override));
+ MOCK_METHOD(std::optional<int>, maxPacketLifeTime, (), (const, override));
MOCK_METHOD(std::string, protocol, (), (const, override));
MOCK_METHOD(bool, negotiated, (), (const, override));
MOCK_METHOD(int, id, (), (const, override));
diff --git a/api/test/mock_encoder_selector.h b/api/test/mock_encoder_selector.h
index a54c4c1..bb5c3a0 100644
--- a/api/test/mock_encoder_selector.h
+++ b/api/test/mock_encoder_selector.h
@@ -11,7 +11,8 @@
#ifndef API_TEST_MOCK_ENCODER_SELECTOR_H_
#define API_TEST_MOCK_ENCODER_SELECTOR_H_
-#include "absl/types/optional.h"
+#include <optional>
+
#include "api/units/data_rate.h"
#include "api/video/render_resolution.h"
#include "api/video_codecs/sdp_video_format.h"
@@ -28,17 +29,17 @@
(const SdpVideoFormat& format),
(override));
- MOCK_METHOD(absl::optional<SdpVideoFormat>,
+ MOCK_METHOD(std::optional<SdpVideoFormat>,
OnAvailableBitrate,
(const DataRate& rate),
(override));
- MOCK_METHOD(absl::optional<SdpVideoFormat>,
+ MOCK_METHOD(std::optional<SdpVideoFormat>,
OnResolutionChange,
(const RenderResolution& resolution),
(override));
- MOCK_METHOD(absl::optional<SdpVideoFormat>, OnEncoderBroken, (), (override));
+ MOCK_METHOD(std::optional<SdpVideoFormat>, OnEncoderBroken, (), (override));
};
} // namespace webrtc
diff --git a/api/test/mock_peerconnectioninterface.h b/api/test/mock_peerconnectioninterface.h
index 782e45e..2b985f4 100644
--- a/api/test/mock_peerconnectioninterface.h
+++ b/api/test/mock_peerconnectioninterface.h
@@ -13,11 +13,11 @@
#include <cstdint>
#include <memory>
+#include <optional>
#include <string>
#include <type_traits>
#include <vector>
-#include "absl/types/optional.h"
#include "api/adaptation/resource.h"
#include "api/candidate.h"
#include "api/data_channel_interface.h"
@@ -223,7 +223,7 @@
AddAdaptationResource,
(rtc::scoped_refptr<Resource>),
(override));
- MOCK_METHOD(absl::optional<bool>, can_trickle_ice_candidates, (), (override));
+ MOCK_METHOD(std::optional<bool>, can_trickle_ice_candidates, (), (override));
MOCK_METHOD(bool,
StartRtcEventLog,
(std::unique_ptr<RtcEventLogOutput>, int64_t),
diff --git a/api/test/mock_rtp_transceiver.h b/api/test/mock_rtp_transceiver.h
index f3dbec6..8bc2a6f 100644
--- a/api/test/mock_rtp_transceiver.h
+++ b/api/test/mock_rtp_transceiver.h
@@ -11,10 +11,10 @@
#ifndef API_TEST_MOCK_RTP_TRANSCEIVER_H_
#define API_TEST_MOCK_RTP_TRANSCEIVER_H_
+#include <optional>
#include <string>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/make_ref_counted.h"
#include "api/media_types.h"
@@ -38,7 +38,7 @@
}
MOCK_METHOD(cricket::MediaType, media_type, (), (const, override));
- MOCK_METHOD(absl::optional<std::string>, mid, (), (const, override));
+ MOCK_METHOD(std::optional<std::string>, mid, (), (const, override));
MOCK_METHOD(rtc::scoped_refptr<RtpSenderInterface>,
sender,
(),
@@ -58,11 +58,11 @@
SetDirectionWithError,
(RtpTransceiverDirection new_direction),
(override));
- MOCK_METHOD(absl::optional<RtpTransceiverDirection>,
+ MOCK_METHOD(std::optional<RtpTransceiverDirection>,
current_direction,
(),
(const, override));
- MOCK_METHOD(absl::optional<RtpTransceiverDirection>,
+ MOCK_METHOD(std::optional<RtpTransceiverDirection>,
fired_direction,
(),
(const, override));
diff --git a/api/test/mock_rtpreceiver.h b/api/test/mock_rtpreceiver.h
index b591db4..8f8c3b4 100644
--- a/api/test/mock_rtpreceiver.h
+++ b/api/test/mock_rtpreceiver.h
@@ -11,10 +11,10 @@
#ifndef API_TEST_MOCK_RTPRECEIVER_H_
#define API_TEST_MOCK_RTPRECEIVER_H_
+#include <optional>
#include <string>
#include <vector>
-#include "absl/types/optional.h"
#include "api/crypto/frame_decryptor_interface.h"
#include "api/media_stream_interface.h"
#include "api/media_types.h"
@@ -47,7 +47,7 @@
MOCK_METHOD(void, SetObserver, (RtpReceiverObserverInterface*), (override));
MOCK_METHOD(void,
SetJitterBufferMinimumDelay,
- (absl::optional<double>),
+ (std::optional<double>),
(override));
MOCK_METHOD(std::vector<RtpSource>, GetSources, (), (const, override));
MOCK_METHOD(void,
diff --git a/api/test/mock_transformable_audio_frame.h b/api/test/mock_transformable_audio_frame.h
index 0bed626..899624b 100644
--- a/api/test/mock_transformable_audio_frame.h
+++ b/api/test/mock_transformable_audio_frame.h
@@ -12,9 +12,9 @@
#define API_TEST_MOCK_TRANSFORMABLE_AUDIO_FRAME_H_
#include <cstdint>
+#include <optional>
#include <string>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/frame_transformer_interface.h"
#include "api/units/timestamp.h"
@@ -37,7 +37,7 @@
GetContributingSources,
(),
(const override));
- MOCK_METHOD(const absl::optional<uint16_t>,
+ MOCK_METHOD(const std::optional<uint16_t>,
SequenceNumber,
(),
(const, override));
@@ -45,7 +45,7 @@
GetDirection,
(),
(const, override));
- MOCK_METHOD(absl::optional<uint64_t>,
+ MOCK_METHOD(std::optional<uint64_t>,
AbsoluteCaptureTimestamp,
(),
(const, override));
@@ -53,9 +53,9 @@
Type,
(),
(const, override));
- MOCK_METHOD(absl::optional<uint8_t>, AudioLevel, (), (const, override));
+ MOCK_METHOD(std::optional<uint8_t>, AudioLevel, (), (const, override));
- MOCK_METHOD(absl::optional<Timestamp>, ReceiveTime, (), (const, override));
+ MOCK_METHOD(std::optional<Timestamp>, ReceiveTime, (), (const, override));
};
} // namespace webrtc
diff --git a/api/test/mock_transformable_video_frame.h b/api/test/mock_transformable_video_frame.h
index 2ec3579..27a063e 100644
--- a/api/test/mock_transformable_video_frame.h
+++ b/api/test/mock_transformable_video_frame.h
@@ -12,10 +12,10 @@
#define API_TEST_MOCK_TRANSFORMABLE_VIDEO_FRAME_H_
#include <cstdint>
+#include <optional>
#include <string>
#include <type_traits>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/frame_transformer_interface.h"
#include "api/units/timestamp.h"
@@ -45,7 +45,7 @@
(const, override));
MOCK_METHOD(std::string, GetMimeType, (), (const, override));
MOCK_METHOD(VideoFrameMetadata, Metadata, (), (const, override));
- MOCK_METHOD(absl::optional<Timestamp>,
+ MOCK_METHOD(std::optional<Timestamp>,
GetCaptureTimeIdentifier,
(),
(const, override));
diff --git a/api/test/mock_video_decoder.h b/api/test/mock_video_decoder.h
index c252c24..8a391a4 100644
--- a/api/test/mock_video_decoder.h
+++ b/api/test/mock_video_decoder.h
@@ -12,8 +12,8 @@
#define API_TEST_MOCK_VIDEO_DECODER_H_
#include <cstdint>
+#include <optional>
-#include "absl/types/optional.h"
#include "api/video/encoded_image.h"
#include "api/video/video_frame.h"
#include "api/video_codecs/video_decoder.h"
@@ -38,8 +38,8 @@
MOCK_METHOD(void,
Decoded,
(VideoFrame & decoded_image, // NOLINT
- absl::optional<int32_t> decode_time_ms,
- absl::optional<uint8_t> qp),
+ std::optional<int32_t> decode_time_ms,
+ std::optional<uint8_t> qp),
(override));
};
diff --git a/api/test/neteq_simulator_factory.h b/api/test/neteq_simulator_factory.h
index 79e515f..220daa5 100644
--- a/api/test/neteq_simulator_factory.h
+++ b/api/test/neteq_simulator_factory.h
@@ -13,10 +13,10 @@
#include <cstdint>
#include <memory>
+#include <optional>
#include <string>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/neteq/neteq_factory.h"
#include "api/test/neteq_simulator.h"
@@ -44,15 +44,15 @@
// A WebRTC field trial string to be used during the simulation.
std::string field_trial_string;
// A filename for the generated output audio file.
- absl::optional<std::string> output_audio_filename;
+ std::optional<std::string> output_audio_filename;
// A filename for the python plot.
- absl::optional<std::string> python_plot_filename;
+ std::optional<std::string> python_plot_filename;
// A filename for the text log.
- absl::optional<std::string> text_log_filename;
+ std::optional<std::string> text_log_filename;
// A custom NetEqFactory can be used.
NetEqFactory* neteq_factory = nullptr;
// The SSRC to use for the simulation.
- absl::optional<uint32_t> ssrc_filter;
+ std::optional<uint32_t> ssrc_filter;
};
std::unique_ptr<NetEqSimulator> CreateSimulatorFromFile(
absl::string_view event_log_filename,
diff --git a/api/test/network_emulation/BUILD.gn b/api/test/network_emulation/BUILD.gn
index 809b1b3..30e88a0 100644
--- a/api/test/network_emulation/BUILD.gn
+++ b/api/test/network_emulation/BUILD.gn
@@ -55,7 +55,6 @@
"../../units:data_size",
"../../units:time_delta",
"../../units:timestamp",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
diff --git a/api/test/network_emulation/network_emulation_interfaces.h b/api/test/network_emulation/network_emulation_interfaces.h
index c5831f1..1789e04 100644
--- a/api/test/network_emulation/network_emulation_interfaces.h
+++ b/api/test/network_emulation/network_emulation_interfaces.h
@@ -14,9 +14,9 @@
#include <cstdint>
#include <functional>
#include <map>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/numerics/samples_stats_counter.h"
#include "api/units/data_rate.h"
#include "api/units/data_size.h"
@@ -263,12 +263,12 @@
//
// Returns the port, that should be used (it will be equals to desired, if
// `desired_port` != 0 and is free or will be the one, selected by endpoint)
- // or absl::nullopt if desired_port in used. Also fails if there are no more
+ // or std::nullopt if desired_port in used. Also fails if there are no more
// free ports to bind to.
//
// The Bind- and Unbind-methods must not be called from within a bound
// receiver's OnPacketReceived method.
- virtual absl::optional<uint16_t> BindReceiver(
+ virtual std::optional<uint16_t> BindReceiver(
uint16_t desired_port,
EmulatedNetworkReceiverInterface* receiver) = 0;
// Unbinds receiver from the specified port. Do nothing if no receiver was
diff --git a/api/test/network_emulation_manager.h b/api/test/network_emulation_manager.h
index aeb32e7..5e20be5 100644
--- a/api/test/network_emulation_manager.h
+++ b/api/test/network_emulation_manager.h
@@ -14,12 +14,12 @@
#include <cstdint>
#include <functional>
#include <memory>
+#include <optional>
#include <string>
#include <utility>
#include <vector>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/field_trials_view.h"
#include "api/packet_socket_factory.h"
@@ -69,11 +69,11 @@
enum class IpAddressFamily { kIpv4, kIpv6 };
// If specified will be used to name endpoint for logging purposes.
- absl::optional<std::string> name = absl::nullopt;
+ std::optional<std::string> name = std::nullopt;
IpAddressFamily generated_ip_family = IpAddressFamily::kIpv4;
// If specified will be used as IP address for endpoint node. Must be unique
// among all created nodes.
- absl::optional<rtc::IPAddress> ip;
+ std::optional<rtc::IPAddress> ip;
// Should endpoint be enabled or not, when it will be created.
// Enabled endpoints will be available for webrtc to send packets.
bool start_as_enabled = true;
diff --git a/api/test/pclf/BUILD.gn b/api/test/pclf/BUILD.gn
index 047bca8..2f3c2dd 100644
--- a/api/test/pclf/BUILD.gn
+++ b/api/test/pclf/BUILD.gn
@@ -51,7 +51,6 @@
"../video:video_frame_writer",
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -86,7 +85,6 @@
"../../neteq:neteq_api",
"../../transport:bitrate_settings",
"../../units:time_delta",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -121,7 +119,6 @@
"../../../rtc_base:ssl",
"../../audio:audio_processing",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
"//third_party/abseil-cpp/absl/types:variant",
]
}
diff --git a/api/test/pclf/media_configuration.cc b/api/test/pclf/media_configuration.cc
index bf60837..126e474 100644
--- a/api/test/pclf/media_configuration.cc
+++ b/api/test/pclf/media_configuration.cc
@@ -15,12 +15,12 @@
#include <functional>
#include <map>
#include <memory>
+#include <optional>
#include <string>
#include <utility>
#include <vector>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/test/video/video_frame_writer.h"
#include "api/units/time_delta.h"
@@ -64,8 +64,8 @@
RTC_CHECK_GE(target_layer_index, 0);
}
-EmulatedSFUConfig::EmulatedSFUConfig(absl::optional<int> target_layer_index,
- absl::optional<int> target_temporal_index)
+EmulatedSFUConfig::EmulatedSFUConfig(std::optional<int> target_layer_index,
+ std::optional<int> target_temporal_index)
: target_layer_index(target_layer_index),
target_temporal_index(target_temporal_index) {
RTC_CHECK_GE(target_temporal_index.value_or(0), 0);
@@ -127,7 +127,7 @@
const VideoResolution& resolution) const {
std::unique_ptr<test::VideoFrameWriter> writer = video_frame_writer_factory_(
GetInputDumpFileName(stream_label, resolution), resolution);
- absl::optional<std::string> frame_ids_file =
+ std::optional<std::string> frame_ids_file =
GetInputFrameIdsDumpFileName(stream_label, resolution);
if (frame_ids_file.has_value()) {
writer = CreateVideoFrameWithIdsWriter(std::move(writer), *frame_ids_file);
@@ -142,7 +142,7 @@
const VideoResolution& resolution) const {
std::unique_ptr<test::VideoFrameWriter> writer = video_frame_writer_factory_(
GetOutputDumpFileName(stream_label, receiver, resolution), resolution);
- absl::optional<std::string> frame_ids_file =
+ std::optional<std::string> frame_ids_file =
GetOutputFrameIdsDumpFileName(stream_label, receiver, resolution);
if (frame_ids_file.has_value()) {
writer = CreateVideoFrameWithIdsWriter(std::move(writer), *frame_ids_file);
@@ -168,11 +168,11 @@
return test::JoinFilename(output_directory_, file_name.Release());
}
-absl::optional<std::string> VideoDumpOptions::GetInputFrameIdsDumpFileName(
+std::optional<std::string> VideoDumpOptions::GetInputFrameIdsDumpFileName(
absl::string_view stream_label,
const VideoResolution& resolution) const {
if (!export_frame_ids_) {
- return absl::nullopt;
+ return std::nullopt;
}
return GetInputDumpFileName(stream_label, resolution) + ".frame_ids.txt";
}
@@ -187,12 +187,12 @@
return test::JoinFilename(output_directory_, file_name.Release());
}
-absl::optional<std::string> VideoDumpOptions::GetOutputFrameIdsDumpFileName(
+std::optional<std::string> VideoDumpOptions::GetOutputFrameIdsDumpFileName(
absl::string_view stream_label,
absl::string_view receiver,
const VideoResolution& resolution) const {
if (!export_frame_ids_) {
- return absl::nullopt;
+ return std::nullopt;
}
return GetOutputDumpFileName(stream_label, receiver, resolution) +
".frame_ids.txt";
@@ -231,7 +231,7 @@
std::map<std::string, std::string> required_params)
: name(name), required_params(std::move(required_params)) {}
-absl::optional<VideoResolution> VideoSubscription::GetMaxResolution(
+std::optional<VideoResolution> VideoSubscription::GetMaxResolution(
rtc::ArrayView<const VideoConfig> video_configs) {
std::vector<VideoResolution> resolutions;
for (const auto& video_config : video_configs) {
@@ -240,10 +240,10 @@
return GetMaxResolution(resolutions);
}
-absl::optional<VideoResolution> VideoSubscription::GetMaxResolution(
+std::optional<VideoResolution> VideoSubscription::GetMaxResolution(
rtc::ArrayView<const VideoResolution> resolutions) {
if (resolutions.empty()) {
- return absl::nullopt;
+ return std::nullopt;
}
VideoResolution max_resolution;
@@ -282,7 +282,7 @@
return *this;
}
-absl::optional<VideoResolution> VideoSubscription::GetResolutionForPeer(
+std::optional<VideoResolution> VideoSubscription::GetResolutionForPeer(
absl::string_view peer_name) const {
auto it = peers_resolution_.find(std::string(peer_name));
if (it == peers_resolution_.end()) {
diff --git a/api/test/pclf/media_configuration.h b/api/test/pclf/media_configuration.h
index aba8040..32b0d21 100644
--- a/api/test/pclf/media_configuration.h
+++ b/api/test/pclf/media_configuration.h
@@ -16,11 +16,11 @@
#include <functional>
#include <map>
#include <memory>
+#include <optional>
#include <string>
#include <vector>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/audio_options.h"
#include "api/media_stream_interface.h"
@@ -79,7 +79,7 @@
bool generate_slides = false;
// If present scrolling will be applied. Please read extra requirement on
// `slides_yuv_file_names` for scrolling.
- absl::optional<ScrollingParams> scrolling_params;
+ std::optional<ScrollingParams> scrolling_params;
// Contains list of yuv files with slides.
//
// If empty, default set of slides will be used. In such case
@@ -130,8 +130,8 @@
struct EmulatedSFUConfig {
EmulatedSFUConfig() = default;
explicit EmulatedSFUConfig(int target_layer_index);
- EmulatedSFUConfig(absl::optional<int> target_layer_index,
- absl::optional<int> target_temporal_index);
+ EmulatedSFUConfig(std::optional<int> target_layer_index,
+ std::optional<int> target_temporal_index);
// Specifies simulcast or spatial index of the video stream to analyze.
// There are 2 cases:
@@ -146,12 +146,12 @@
// If not specified then all streams will be received and analyzed.
// When set, it instructs the framework to create an emulated Selective
// Forwarding Unit (SFU) that will propagate only the requested layers.
- absl::optional<int> target_layer_index;
+ std::optional<int> target_layer_index;
// Specifies the index of the maximum temporal unit to keep.
// If not specified then all temporal layers will be received and analyzed.
// When set, it instructs the framework to create an emulated Selective
// Forwarding Unit (SFU) that will propagate only up to the requested layer.
- absl::optional<int> target_temporal_index;
+ std::optional<int> target_temporal_index;
};
class VideoResolution {
@@ -252,16 +252,16 @@
std::string GetInputDumpFileName(absl::string_view stream_label,
const VideoResolution& resolution) const;
// Returns file name for input frame ids dump if `export_frame_ids()` is
- // true, absl::nullopt otherwise.
- absl::optional<std::string> GetInputFrameIdsDumpFileName(
+ // true, std::nullopt otherwise.
+ std::optional<std::string> GetInputFrameIdsDumpFileName(
absl::string_view stream_label,
const VideoResolution& resolution) const;
std::string GetOutputDumpFileName(absl::string_view stream_label,
absl::string_view receiver,
const VideoResolution& resolution) const;
// Returns file name for output frame ids dump if `export_frame_ids()` is
- // true, absl::nullopt otherwise.
- absl::optional<std::string> GetOutputFrameIdsDumpFileName(
+ // true, std::nullopt otherwise.
+ std::optional<std::string> GetOutputFrameIdsDumpFileName(
absl::string_view stream_label,
absl::string_view receiver,
const VideoResolution& resolution) const;
@@ -295,10 +295,10 @@
// Have to be unique among all specified configs for all peers in the call.
// Will be auto generated if omitted.
- absl::optional<std::string> stream_label;
+ std::optional<std::string> stream_label;
// Will be set for current video track. If equals to kText or kDetailed -
// screencast in on.
- absl::optional<VideoTrackInterface::ContentHint> content_hint;
+ std::optional<VideoTrackInterface::ContentHint> content_hint;
// If presented video will be transfered in simulcast/SVC mode depending on
// which encoder is used.
//
@@ -307,9 +307,9 @@
// simulcast tracks. For VP9 simulcast enables VP9 SVC mode and support RTX,
// but only on non-lossy networks. See more in documentation to
// VideoSimulcastConfig.
- absl::optional<VideoSimulcastConfig> simulcast_config;
+ std::optional<VideoSimulcastConfig> simulcast_config;
// Configuration for the emulated Selective Forward Unit (SFU).
- absl::optional<EmulatedSFUConfig> emulated_sfu_config;
+ std::optional<EmulatedSFUConfig> emulated_sfu_config;
// Encoding parameters for both singlecast and per simulcast layer.
// If singlecast is used, if not empty, a single value can be provided.
// If simulcast is used, if not empty, `encoding_params` size have to be
@@ -321,16 +321,16 @@
// Count of temporal layers for video stream. This value will be set into
// each RtpEncodingParameters of RtpParameters of corresponding
// RtpSenderInterface for this video stream.
- absl::optional<int> temporal_layers_count;
+ std::optional<int> temporal_layers_count;
// If specified defines how input should be dumped. It is actually one of
// the test's output file, which contains copy of what was captured during
// the test for this video stream on sender side. It is useful when
// generator is used as input.
- absl::optional<VideoDumpOptions> input_dump_options;
+ std::optional<VideoDumpOptions> input_dump_options;
// If specified defines how output should be dumped on the receiver side for
// this stream. The produced files contain what was rendered for this video
// stream on receiver side per each receiver.
- absl::optional<VideoDumpOptions> output_dump_options;
+ std::optional<VideoDumpOptions> output_dump_options;
// If set to true uses fixed frame rate while dumping output video to the
// file. Requested `VideoSubscription::fps()` will be used as frame rate.
bool output_dump_use_fixed_framerate = false;
@@ -339,11 +339,11 @@
// If specified, determines a sync group to which this video stream belongs.
// According to bugs.webrtc.org/4762 WebRTC supports synchronization only
// for pair of single audio and single video stream.
- absl::optional<std::string> sync_group;
+ std::optional<std::string> sync_group;
// If specified, it will be set into RtpParameters of corresponding
// RtpSenderInterface for this video stream.
// Note that this setting takes precedence over `content_hint`.
- absl::optional<DegradationPreference> degradation_preference;
+ std::optional<DegradationPreference> degradation_preference;
};
// Contains properties for audio in the call.
@@ -353,13 +353,13 @@
// Have to be unique among all specified configs for all peers in the call.
// Will be auto generated if omitted.
- absl::optional<std::string> stream_label;
+ std::optional<std::string> stream_label;
// If no file is specified an audio will be generated.
- absl::optional<std::string> input_file_name;
+ std::optional<std::string> input_file_name;
// If specified the input stream will be also copied to specified file.
- absl::optional<std::string> input_dump_file_name;
+ std::optional<std::string> input_dump_file_name;
// If specified the output stream will be copied to specified file.
- absl::optional<std::string> output_dump_file_name;
+ std::optional<std::string> output_dump_file_name;
// Audio options to use.
cricket::AudioOptions audio_options;
@@ -368,7 +368,7 @@
// If specified, determines a sync group to which this audio stream belongs.
// According to bugs.webrtc.org/4762 WebRTC supports synchronization only
// for pair of single audio and single video stream.
- absl::optional<std::string> sync_group;
+ std::optional<std::string> sync_group;
};
struct VideoCodecConfig {
@@ -396,9 +396,9 @@
public:
// Returns the resolution constructed as maximum from all resolution
// dimensions: width, height and fps.
- static absl::optional<VideoResolution> GetMaxResolution(
+ static std::optional<VideoResolution> GetMaxResolution(
rtc::ArrayView<const VideoConfig> video_configs);
- static absl::optional<VideoResolution> GetMaxResolution(
+ static std::optional<VideoResolution> GetMaxResolution(
rtc::ArrayView<const VideoResolution> resolutions);
bool operator==(const VideoSubscription& other) const;
@@ -422,9 +422,9 @@
// Returns resolution for specific sender. If no specific resolution was
// set for this sender, then will return resolution used for all streams.
- // If subscription doesn't subscribe to all streams, `absl::nullopt` will be
+ // If subscription doesn't subscribe to all streams, `std::nullopt` will be
// returned.
- absl::optional<VideoResolution> GetResolutionForPeer(
+ std::optional<VideoResolution> GetResolutionForPeer(
absl::string_view peer_name) const;
// Returns a maybe empty list of senders for which peer explicitly
@@ -434,7 +434,7 @@
std::string ToString() const;
private:
- absl::optional<VideoResolution> default_resolution_ = absl::nullopt;
+ std::optional<VideoResolution> default_resolution_ = std::nullopt;
std::map<std::string, VideoResolution> peers_resolution_;
};
diff --git a/api/test/pclf/media_quality_test_params.h b/api/test/pclf/media_quality_test_params.h
index cf0a51e..56c4c1b 100644
--- a/api/test/pclf/media_quality_test_params.h
+++ b/api/test/pclf/media_quality_test_params.h
@@ -13,10 +13,10 @@
#include <cstddef>
#include <cstdint>
#include <memory>
+#include <optional>
#include <string>
#include <vector>
-#include "absl/types/optional.h"
#include "api/async_dns_resolver.h"
#include "api/audio/audio_mixer.h"
#include "api/audio/audio_processing.h"
@@ -125,18 +125,18 @@
// to set up peer connection.
struct Params {
// Peer name. If empty - default one will be set by the fixture.
- absl::optional<std::string> name;
+ std::optional<std::string> name;
// If `audio_config` is set audio stream will be configured
- absl::optional<AudioConfig> audio_config;
+ std::optional<AudioConfig> audio_config;
// Flags to set on `cricket::PortAllocator`. These flags will be added
// to the default ones that are presented on the port allocator.
uint32_t port_allocator_extra_flags = cricket::kDefaultPortAllocatorFlags;
// If `rtc_event_log_path` is set, an RTCEventLog will be saved in that
// location and it will be available for further analysis.
- absl::optional<std::string> rtc_event_log_path;
+ std::optional<std::string> rtc_event_log_path;
// If `aec_dump_path` is set, an AEC dump will be saved in that location and
// it will be available for further analysis.
- absl::optional<std::string> aec_dump_path;
+ std::optional<std::string> aec_dump_path;
bool use_ulp_fec = false;
bool use_flex_fec = false;
@@ -190,7 +190,7 @@
// If specified echo emulation will be done, by mixing the render audio into
// the capture signal. In such case input signal will be reduced by half to
// avoid saturation or compression in the echo path simulation.
- absl::optional<EchoEmulationConfig> echo_emulation_config;
+ std::optional<EchoEmulationConfig> echo_emulation_config;
};
} // namespace webrtc_pc_e2e
diff --git a/api/test/pclf/peer_configurer.cc b/api/test/pclf/peer_configurer.cc
index 765ec2d..a19e250 100644
--- a/api/test/pclf/peer_configurer.cc
+++ b/api/test/pclf/peer_configurer.cc
@@ -12,12 +12,12 @@
#include <cstdint>
#include <memory>
+#include <optional>
#include <string>
#include <utility>
#include <vector>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/async_dns_resolver.h"
#include "api/audio/audio_mixer.h"
#include "api/audio/audio_processing.h"
@@ -122,7 +122,7 @@
PeerConfigurer* PeerConfigurer::AddVideoConfig(VideoConfig config) {
video_sources_.push_back(
- CreateSquareFrameGenerator(config, /*type=*/absl::nullopt));
+ CreateSquareFrameGenerator(config, /*type=*/std::nullopt));
configurable_params_->video_configs.push_back(std::move(config));
return this;
}
diff --git a/api/test/peerconnection_quality_test_fixture_unittest.cc b/api/test/peerconnection_quality_test_fixture_unittest.cc
index cd3dbb2..970fbf6 100644
--- a/api/test/peerconnection_quality_test_fixture_unittest.cc
+++ b/api/test/peerconnection_quality_test_fixture_unittest.cc
@@ -10,11 +10,11 @@
#include <cstddef>
#include <memory>
+#include <optional>
#include <string>
#include <vector>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/test/pclf/media_configuration.h"
#include "api/test/video/video_frame_writer.h"
#include "api/video/video_frame.h"
@@ -60,7 +60,7 @@
}
TEST(PclfVideoSubscriptionTest, GetMaxResolutionForEmptyReturnsNullopt) {
- absl::optional<VideoResolution> resolution =
+ std::optional<VideoResolution> resolution =
VideoSubscription::GetMaxResolution(std::vector<VideoConfig>{});
ASSERT_FALSE(resolution.has_value());
}
@@ -70,7 +70,7 @@
VideoConfig max_height(/*width=*/1, /*height=*/100, /*fps=*/1);
VideoConfig max_fps(/*width=*/1, /*height=*/1, /*fps=*/10);
- absl::optional<VideoResolution> resolution =
+ std::optional<VideoResolution> resolution =
VideoSubscription::GetMaxResolution(
std::vector<VideoConfig>{max_width, max_height, max_fps});
ASSERT_TRUE(resolution.has_value());
diff --git a/api/test/simulated_network.h b/api/test/simulated_network.h
index 618fe71..245b8f0 100644
--- a/api/test/simulated_network.h
+++ b/api/test/simulated_network.h
@@ -15,10 +15,10 @@
#include <stdint.h>
#include <functional>
+#include <optional>
#include <vector>
#include "absl/functional/any_invocable.h"
-#include "absl/types/optional.h"
#include "api/units/data_rate.h"
namespace webrtc {
@@ -116,7 +116,7 @@
// possible that no packet will be delivered by that time (e.g. in case of
// random extra delay), in such case this method should be called again to get
// the updated estimated delivery time.
- virtual absl::optional<int64_t> NextDeliveryTimeUs() const = 0;
+ virtual std::optional<int64_t> NextDeliveryTimeUs() const = 0;
// Registers a callback that should be triggered by an implementation if the
// next NextDeliveryTimeUs() has changed between a call to NextDeliveryTimeUs
// and DequeueDeliverablePackets.
diff --git a/api/test/video/BUILD.gn b/api/test/video/BUILD.gn
index 61af6b0..f1f8db6 100644
--- a/api/test/video/BUILD.gn
+++ b/api/test/video/BUILD.gn
@@ -46,6 +46,5 @@
"../../../rtc_base/system:no_unique_address",
"../../video:recordable_encoded_frame",
"../../video:video_frame",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
diff --git a/api/test/video/test_video_track_source.cc b/api/test/video/test_video_track_source.cc
index fd7f18b..6eaf0b9 100644
--- a/api/test/video/test_video_track_source.cc
+++ b/api/test/video/test_video_track_source.cc
@@ -9,10 +9,10 @@
*/
#include "api/test/video/test_video_track_source.h"
+#include <optional>
#include <string>
#include <utility>
-#include "absl/types/optional.h"
#include "api/media_stream_interface.h"
#include "api/sequence_checker.h"
#include "api/video/video_frame.h"
@@ -25,7 +25,7 @@
TestVideoTrackSource::TestVideoTrackSource(
bool remote,
- absl::optional<std::string> stream_label)
+ std::optional<std::string> stream_label)
: stream_label_(std::move(stream_label)),
state_(kInitializing),
remote_(remote) {
diff --git a/api/test/video/test_video_track_source.h b/api/test/video/test_video_track_source.h
index 173bb64..95daccf 100644
--- a/api/test/video/test_video_track_source.h
+++ b/api/test/video/test_video_track_source.h
@@ -11,9 +11,9 @@
#ifndef API_TEST_VIDEO_TEST_VIDEO_TRACK_SOURCE_H_
#define API_TEST_VIDEO_TEST_VIDEO_TRACK_SOURCE_H_
+#include <optional>
#include <string>
-#include "absl/types/optional.h"
#include "api/media_stream_interface.h"
#include "api/notifier.h"
#include "api/sequence_checker.h"
@@ -32,7 +32,7 @@
public:
explicit TestVideoTrackSource(
bool remote,
- absl::optional<std::string> stream_label = absl::nullopt);
+ std::optional<std::string> stream_label = std::nullopt);
~TestVideoTrackSource() override = default;
void SetState(SourceState new_state);
@@ -41,9 +41,7 @@
bool remote() const override { return remote_; }
bool is_screencast() const override { return false; }
- absl::optional<bool> needs_denoising() const override {
- return absl::nullopt;
- }
+ std::optional<bool> needs_denoising() const override { return std::nullopt; }
bool GetStats(Stats* stats) override { return false; }
@@ -74,17 +72,17 @@
virtual void OnOutputFormatRequest(int width,
int height,
- const absl::optional<int>& max_fps) {}
+ const std::optional<int>& max_fps) {}
// Returns stream label for this video source if present. Implementations
// may override this method to increase debugability and testability.
- virtual absl::optional<std::string> GetStreamLabel() { return stream_label_; }
+ virtual std::optional<std::string> GetStreamLabel() { return stream_label_; }
protected:
virtual rtc::VideoSourceInterface<VideoFrame>* source() = 0;
private:
- const absl::optional<std::string> stream_label_;
+ const std::optional<std::string> stream_label_;
RTC_NO_UNIQUE_ADDRESS SequenceChecker worker_thread_checker_;
RTC_NO_UNIQUE_ADDRESS SequenceChecker signaling_thread_checker_;
SourceState state_ RTC_GUARDED_BY(&signaling_thread_checker_);
diff --git a/api/test/video_quality_analyzer_interface.h b/api/test/video_quality_analyzer_interface.h
index 6e69ede..8c9498f 100644
--- a/api/test/video_quality_analyzer_interface.h
+++ b/api/test/video_quality_analyzer_interface.h
@@ -12,10 +12,10 @@
#define API_TEST_VIDEO_QUALITY_ANALYZER_INTERFACE_H_
#include <cstdint>
+#include <optional>
#include <string>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/scoped_refptr.h"
#include "api/stats/rtc_stats_report.h"
@@ -73,9 +73,9 @@
std::string decoder_name = "unknown";
// Decode time provided by decoder itself. If decoder doesn’t produce such
// information can be omitted.
- absl::optional<int32_t> decode_time_ms = absl::nullopt;
+ std::optional<int32_t> decode_time_ms = std::nullopt;
// Decoder quantizer value.
- absl::optional<uint8_t> qp = absl::nullopt;
+ std::optional<uint8_t> qp = std::nullopt;
};
~VideoQualityAnalyzerInterface() override = default;
diff --git a/api/test/video_quality_test_fixture.h b/api/test/video_quality_test_fixture.h
index 0302e89..763dd7d 100644
--- a/api/test/video_quality_test_fixture.h
+++ b/api/test/video_quality_test_fixture.h
@@ -15,10 +15,10 @@
#include <cstdint>
#include <map>
#include <memory>
+#include <optional>
#include <string>
#include <vector>
-#include "absl/types/optional.h"
#include "api/fec_controller.h"
#include "api/media_types.h"
#include "api/network_state_predictor.h"
@@ -74,7 +74,7 @@
bool sync_video = false;
bool dtx = false;
bool use_real_adm = false;
- absl::optional<std::string> ana_config;
+ std::optional<std::string> ana_config;
} audio;
struct Screenshare {
bool enabled = false;
@@ -95,7 +95,7 @@
// `sender_network` and `receiver_network` in InjectionComponents are
// non-null. May be nullopt even if `sender_network` and `receiver_network`
// are null; in that case, a default config will be used.
- absl::optional<BuiltInNetworkBehaviorConfig> config;
+ std::optional<BuiltInNetworkBehaviorConfig> config;
struct SS { // Spatial scalability.
std::vector<VideoStream> streams; // If empty, one stream is assumed.
size_t selected_stream = 0;
diff --git a/api/test/videocodec_test_fixture.h b/api/test/videocodec_test_fixture.h
index dd39c4f..42c3303 100644
--- a/api/test/videocodec_test_fixture.h
+++ b/api/test/videocodec_test_fixture.h
@@ -12,10 +12,10 @@
#define API_TEST_VIDEOCODEC_TEST_FIXTURE_H_
#include <cstddef>
+#include <optional>
#include <string>
#include <vector>
-#include "absl/types/optional.h"
#include "api/test/videocodec_test_stats.h"
#include "api/video/encoded_image.h"
#include "api/video/video_codec_type.h"
@@ -95,15 +95,15 @@
std::string filename;
// Dimensions of test clip. Falls back to (codec_settings.width/height) if
// not set.
- absl::optional<int> clip_width;
- absl::optional<int> clip_height;
+ std::optional<int> clip_width;
+ std::optional<int> clip_height;
// Framerate of input clip. Defaults to 30fps if not set.
- absl::optional<int> clip_fps;
+ std::optional<int> clip_fps;
// The resolution at which psnr/ssim comparisons should be made. Frames
// will be scaled to this size if different.
- absl::optional<int> reference_width;
- absl::optional<int> reference_height;
+ std::optional<int> reference_width;
+ std::optional<int> reference_height;
// File to process. This must be a video file in the YUV format.
std::string filepath;
@@ -138,8 +138,8 @@
// default `SdpVideoFormat` based on `codec_name`.
// Encoder and decoder name (`SdpVideoFormat::name`) should be the same as
// `codec_name`.
- absl::optional<SdpVideoFormat> encoder_format;
- absl::optional<SdpVideoFormat> decoder_format;
+ std::optional<SdpVideoFormat> encoder_format;
+ std::optional<SdpVideoFormat> decoder_format;
// H.264 specific settings.
struct H264CodecSettings {
diff --git a/api/transport/BUILD.gn b/api/transport/BUILD.gn
index 991a0d3..7d19c60 100644
--- a/api/transport/BUILD.gn
+++ b/api/transport/BUILD.gn
@@ -14,10 +14,7 @@
"bitrate_settings.cc",
"bitrate_settings.h",
]
- deps = [
- "../../rtc_base/system:rtc_export",
- "//third_party/abseil-cpp/absl/types:optional",
- ]
+ deps = [ "../../rtc_base/system:rtc_export" ]
}
rtc_source_set("bandwidth_usage") {
@@ -54,7 +51,6 @@
"../units:time_delta",
"../units:timestamp",
"//third_party/abseil-cpp/absl/base:core_headers",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -80,7 +76,6 @@
"..:priority",
"..:rtc_error",
"../../rtc_base:copy_on_write_buffer",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
diff --git a/api/transport/bitrate_settings.h b/api/transport/bitrate_settings.h
index f0dcbb8..9bcd694 100644
--- a/api/transport/bitrate_settings.h
+++ b/api/transport/bitrate_settings.h
@@ -11,8 +11,8 @@
#ifndef API_TRANSPORT_BITRATE_SETTINGS_H_
#define API_TRANSPORT_BITRATE_SETTINGS_H_
+#include <optional>
-#include "absl/types/optional.h"
#include "rtc_base/system/rtc_export.h"
namespace webrtc {
@@ -27,9 +27,9 @@
~BitrateSettings();
BitrateSettings(const BitrateSettings&);
// 0 <= min <= start <= max should hold for set parameters.
- absl::optional<int> min_bitrate_bps;
- absl::optional<int> start_bitrate_bps;
- absl::optional<int> max_bitrate_bps;
+ std::optional<int> min_bitrate_bps;
+ std::optional<int> start_bitrate_bps;
+ std::optional<int> max_bitrate_bps;
};
// TODO(srte): BitrateConstraints and BitrateSettings should be merged.
diff --git a/api/transport/data_channel_transport_interface.h b/api/transport/data_channel_transport_interface.h
index ad83544..ddf6147 100644
--- a/api/transport/data_channel_transport_interface.h
+++ b/api/transport/data_channel_transport_interface.h
@@ -13,8 +13,8 @@
#define API_TRANSPORT_DATA_CHANNEL_TRANSPORT_INTERFACE_H_
#include <cstddef>
+#include <optional>
-#include "absl/types/optional.h"
#include "api/priority.h"
#include "api/rtc_error.h"
#include "rtc_base/copy_on_write_buffer.h"
@@ -49,14 +49,14 @@
// Setting this value to zero disables retransmission.
// Valid values are in the range [0-UINT16_MAX].
// `max_rtx_count` and `max_rtx_ms` may not be set simultaneously.
- absl::optional<int> max_rtx_count;
+ std::optional<int> max_rtx_count;
// If set, the maximum number of milliseconds for which the transport
// may retransmit this message before it is dropped.
// Setting this value to zero disables retransmission.
// Valid values are in the range [0-UINT16_MAX].
// `max_rtx_count` and `max_rtx_ms` may not be set simultaneously.
- absl::optional<int> max_rtx_ms;
+ std::optional<int> max_rtx_ms;
};
// Sink for callbacks related to a data channel.
diff --git a/api/transport/network_control.h b/api/transport/network_control.h
index de408a5..90cdebe 100644
--- a/api/transport/network_control.h
+++ b/api/transport/network_control.h
@@ -12,9 +12,9 @@
#define API_TRANSPORT_NETWORK_CONTROL_H_
#include <memory>
+#include <optional>
#include "absl/base/attributes.h"
-#include "absl/types/optional.h"
#include "api/environment/environment.h"
#include "api/field_trials_view.h"
#include "api/transport/network_types.h"
@@ -118,7 +118,7 @@
class NetworkStateEstimator {
public:
// Gets the current best estimate according to the estimator.
- virtual absl::optional<NetworkStateEstimate> GetCurrentEstimate() = 0;
+ virtual std::optional<NetworkStateEstimate> GetCurrentEstimate() = 0;
// Called with per packet feedback regarding receive time.
// Used when the NetworkStateEstimator runs in the sending endpoint.
virtual void OnTransportPacketsFeedback(const TransportPacketsFeedback&) = 0;
diff --git a/api/transport/network_types.h b/api/transport/network_types.h
index ebf09ef..57fa657 100644
--- a/api/transport/network_types.h
+++ b/api/transport/network_types.h
@@ -13,9 +13,9 @@
#include <stdint.h>
#include <cmath>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/units/data_rate.h"
#include "api/units/data_size.h"
#include "api/units/time_delta.h"
@@ -47,17 +47,17 @@
StreamsConfig(const StreamsConfig&);
~StreamsConfig();
Timestamp at_time = Timestamp::PlusInfinity();
- absl::optional<bool> requests_alr_probing;
+ std::optional<bool> requests_alr_probing;
// If `enable_repeated_initial_probing` is set to true, Probes are sent
// periodically every 1s during the first 5s after the network becomes
// available. The probes ignores max_total_allocated_bitrate.
- absl::optional<bool> enable_repeated_initial_probing;
- absl::optional<double> pacing_factor;
+ std::optional<bool> enable_repeated_initial_probing;
+ std::optional<double> pacing_factor;
// TODO(srte): Use BitrateAllocationLimits here.
- absl::optional<DataRate> min_total_allocated_bitrate;
- absl::optional<DataRate> max_padding_rate;
- absl::optional<DataRate> max_total_allocated_bitrate;
+ std::optional<DataRate> min_total_allocated_bitrate;
+ std::optional<DataRate> max_padding_rate;
+ std::optional<DataRate> max_total_allocated_bitrate;
};
struct RTC_EXPORT TargetRateConstraints {
@@ -65,11 +65,11 @@
TargetRateConstraints(const TargetRateConstraints&);
~TargetRateConstraints();
Timestamp at_time = Timestamp::PlusInfinity();
- absl::optional<DataRate> min_data_rate;
- absl::optional<DataRate> max_data_rate;
+ std::optional<DataRate> min_data_rate;
+ std::optional<DataRate> max_data_rate;
// The initial bandwidth estimate to base target rate on. This should be used
// as the basis for initial OnTargetTransferRate and OnPacerConfig callbacks.
- absl::optional<DataRate> starting_rate;
+ std::optional<DataRate> starting_rate;
};
// Send side information
@@ -246,16 +246,16 @@
!probe_cluster_configs.empty() || target_rate.has_value();
}
- absl::optional<DataSize> congestion_window;
- absl::optional<PacerConfig> pacer_config;
+ std::optional<DataSize> congestion_window;
+ std::optional<PacerConfig> pacer_config;
std::vector<ProbeClusterConfig> probe_cluster_configs;
- absl::optional<TargetTransferRate> target_rate;
+ std::optional<TargetTransferRate> target_rate;
};
// Process control
struct RTC_EXPORT ProcessInterval {
Timestamp at_time = Timestamp::PlusInfinity();
- absl::optional<DataSize> pacer_queue;
+ std::optional<DataSize> pacer_queue;
};
// Under development, subject to change without notice.
diff --git a/api/transport/rtp/BUILD.gn b/api/transport/rtp/BUILD.gn
index 3bcee82..c4c43fa 100644
--- a/api/transport/rtp/BUILD.gn
+++ b/api/transport/rtp/BUILD.gn
@@ -16,7 +16,6 @@
"../../../api/units:time_delta",
"../../../api/units:timestamp",
"../../../rtc_base:checks",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -31,6 +30,5 @@
"../../video:render_resolution",
"//third_party/abseil-cpp/absl/container:inlined_vector",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
diff --git a/api/transport/rtp/dependency_descriptor.h b/api/transport/rtp/dependency_descriptor.h
index f546a0a..2ff5ed8 100644
--- a/api/transport/rtp/dependency_descriptor.h
+++ b/api/transport/rtp/dependency_descriptor.h
@@ -15,11 +15,11 @@
#include <initializer_list>
#include <memory>
+#include <optional>
#include <vector>
#include "absl/container/inlined_vector.h"
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/video/render_resolution.h"
namespace webrtc {
@@ -109,8 +109,8 @@
bool last_packet_in_frame = true;
int frame_number = 0;
FrameDependencyTemplate frame_dependencies;
- absl::optional<RenderResolution> resolution;
- absl::optional<uint32_t> active_decode_targets_bitmask;
+ std::optional<RenderResolution> resolution;
+ std::optional<uint32_t> active_decode_targets_bitmask;
std::unique_ptr<FrameDependencyStructure> attached_structure;
};
diff --git a/api/transport/rtp/rtp_source.h b/api/transport/rtp/rtp_source.h
index 9618973..4732044 100644
--- a/api/transport/rtp/rtp_source.h
+++ b/api/transport/rtp/rtp_source.h
@@ -13,7 +13,8 @@
#include <stdint.h>
-#include "absl/types/optional.h"
+#include <optional>
+
#include "api/rtp_headers.h"
#include "api/units/time_delta.h"
#include "api/units/timestamp.h"
@@ -28,18 +29,18 @@
class RtpSource {
public:
struct Extensions {
- absl::optional<uint8_t> audio_level;
+ std::optional<uint8_t> audio_level;
// Fields from the Absolute Capture Time header extension:
// http://www.webrtc.org/experiments/rtp-hdrext/abs-capture-time
- absl::optional<AbsoluteCaptureTime> absolute_capture_time;
+ std::optional<AbsoluteCaptureTime> absolute_capture_time;
// Clock offset between the local clock and the capturer's clock.
// Do not confuse with `AbsoluteCaptureTime::estimated_capture_clock_offset`
// which instead represents the clock offset between a remote sender and the
// capturer. The following holds:
// Capture's NTP Clock = Local NTP Clock + Local-Capture Clock Offset
- absl::optional<TimeDelta> local_capture_clock_offset;
+ std::optional<TimeDelta> local_capture_clock_offset;
};
RtpSource() = delete;
@@ -67,21 +68,19 @@
// The source can be either a contributing source or a synchronization source.
RtpSourceType source_type() const { return source_type_; }
- absl::optional<uint8_t> audio_level() const {
- return extensions_.audio_level;
- }
+ std::optional<uint8_t> audio_level() const { return extensions_.audio_level; }
- void set_audio_level(const absl::optional<uint8_t>& level) {
+ void set_audio_level(const std::optional<uint8_t>& level) {
extensions_.audio_level = level;
}
uint32_t rtp_timestamp() const { return rtp_timestamp_; }
- absl::optional<AbsoluteCaptureTime> absolute_capture_time() const {
+ std::optional<AbsoluteCaptureTime> absolute_capture_time() const {
return extensions_.absolute_capture_time;
}
- absl::optional<TimeDelta> local_capture_clock_offset() const {
+ std::optional<TimeDelta> local_capture_clock_offset() const {
return extensions_.local_capture_clock_offset;
}
diff --git a/api/transport/test/mock_network_control.h b/api/transport/test/mock_network_control.h
index e3a15b8..e672d2e 100644
--- a/api/transport/test/mock_network_control.h
+++ b/api/transport/test/mock_network_control.h
@@ -67,7 +67,7 @@
class MockNetworkStateEstimator : public NetworkStateEstimator {
public:
- MOCK_METHOD(absl::optional<NetworkStateEstimate>,
+ MOCK_METHOD(std::optional<NetworkStateEstimate>,
GetCurrentEstimate,
(),
(override));
diff --git a/api/video/BUILD.gn b/api/video/BUILD.gn
index 9351dea..866757e 100644
--- a/api/video/BUILD.gn
+++ b/api/video/BUILD.gn
@@ -35,7 +35,6 @@
"../../rtc_base/system:rtc_export",
"../units:data_rate",
"../units:time_delta",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -75,7 +74,6 @@
"../../rtc_base/system:rtc_export",
"../units:time_delta",
"../units:timestamp",
- "//third_party/abseil-cpp/absl/types:optional",
"//third_party/libyuv",
]
}
@@ -121,7 +119,6 @@
"..:make_ref_counted",
"..:scoped_refptr",
"../units:timestamp",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -165,7 +162,6 @@
"../../rtc_base:refcount",
"../../rtc_base/system:rtc_export",
"../units:timestamp",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -183,7 +179,6 @@
"../../modules/video_coding:codec_globals_headers",
"../../modules/video_coding:video_codec_interface",
"../units:timestamp",
- "//third_party/abseil-cpp/absl/types:optional",
"//third_party/abseil-cpp/absl/types:variant",
]
}
@@ -213,7 +208,6 @@
"../../rtc_base:rtc_numerics",
"../transport/rtp:dependency_descriptor",
"//third_party/abseil-cpp/absl/container:inlined_vector",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -235,7 +229,6 @@
"../../rtc_base:checks",
"../../test:test_support",
"../transport/rtp:dependency_descriptor",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -257,7 +250,6 @@
"../../rtc_base:safe_conversions",
"../../rtc_base:stringutils",
"../../rtc_base/system:rtc_export",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -343,7 +335,6 @@
"../../rtc_base/system:rtc_export",
"../transport/rtp:dependency_descriptor",
"//third_party/abseil-cpp/absl/container:inlined_vector",
- "//third_party/abseil-cpp/absl/types:optional",
"//third_party/abseil-cpp/absl/types:variant",
]
}
@@ -386,7 +377,6 @@
"../../rtc_base:rtc_numerics",
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/container:inlined_vector",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -401,7 +391,6 @@
"../../test:field_trial",
"../../test:scoped_key_value_config",
"../../test:test_support",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
diff --git a/api/video/color_space.cc b/api/video/color_space.cc
index 05c0135..99648a3 100644
--- a/api/video/color_space.cc
+++ b/api/video/color_space.cc
@@ -12,9 +12,9 @@
#include <cstddef>
#include <cstdint>
+#include <optional>
#include <string>
-#include "absl/types/optional.h"
#include "api/video/hdr_metadata.h"
#include "rtc_base/strings/string_builder.h"
@@ -101,8 +101,8 @@
range_(range),
chroma_siting_horizontal_(chroma_siting_horz),
chroma_siting_vertical_(chroma_siting_vert),
- hdr_metadata_(hdr_metadata ? absl::make_optional(*hdr_metadata)
- : absl::nullopt) {}
+ hdr_metadata_(hdr_metadata ? std::make_optional(*hdr_metadata)
+ : std::nullopt) {}
ColorSpace::PrimaryID ColorSpace::primaries() const {
return primaries_;
@@ -263,7 +263,7 @@
void ColorSpace::set_hdr_metadata(const HdrMetadata* hdr_metadata) {
hdr_metadata_ =
- hdr_metadata ? absl::make_optional(*hdr_metadata) : absl::nullopt;
+ hdr_metadata ? std::make_optional(*hdr_metadata) : std::nullopt;
}
} // namespace webrtc
diff --git a/api/video/color_space.h b/api/video/color_space.h
index 31963a1..00f5cb9 100644
--- a/api/video/color_space.h
+++ b/api/video/color_space.h
@@ -13,9 +13,9 @@
#include <stdint.h>
+#include <optional>
#include <string>
-#include "absl/types/optional.h"
#include "api/video/hdr_metadata.h"
#include "rtc_base/system/rtc_export.h"
@@ -174,7 +174,7 @@
RangeID range_ = RangeID::kInvalid;
ChromaSiting chroma_siting_horizontal_ = ChromaSiting::kUnspecified;
ChromaSiting chroma_siting_vertical_ = ChromaSiting::kUnspecified;
- absl::optional<HdrMetadata> hdr_metadata_;
+ std::optional<HdrMetadata> hdr_metadata_;
};
} // namespace webrtc
diff --git a/api/video/encoded_frame.cc b/api/video/encoded_frame.cc
index 023e233..8e90fa3 100644
--- a/api/video/encoded_frame.cc
+++ b/api/video/encoded_frame.cc
@@ -12,8 +12,8 @@
#include <cstddef>
#include <cstdint>
+#include <optional>
-#include "absl/types/optional.h"
#include "absl/types/variant.h"
#include "api/units/timestamp.h"
#include "api/video/video_codec_type.h"
@@ -24,16 +24,16 @@
namespace webrtc {
-absl::optional<Timestamp> EncodedFrame::ReceivedTimestamp() const {
+std::optional<Timestamp> EncodedFrame::ReceivedTimestamp() const {
return ReceivedTime() >= 0
- ? absl::make_optional(Timestamp::Millis(ReceivedTime()))
- : absl::nullopt;
+ ? std::make_optional(Timestamp::Millis(ReceivedTime()))
+ : std::nullopt;
}
-absl::optional<Timestamp> EncodedFrame::RenderTimestamp() const {
+std::optional<Timestamp> EncodedFrame::RenderTimestamp() const {
return RenderTimeMs() >= 0
- ? absl::make_optional(Timestamp::Millis(RenderTimeMs()))
- : absl::nullopt;
+ ? std::make_optional(Timestamp::Millis(RenderTimeMs()))
+ : std::nullopt;
}
bool EncodedFrame::delayed_by_retransmission() const {
diff --git a/api/video/encoded_frame.h b/api/video/encoded_frame.h
index 8905c36..0bb65cf 100644
--- a/api/video/encoded_frame.h
+++ b/api/video/encoded_frame.h
@@ -14,7 +14,8 @@
#include <stddef.h>
#include <stdint.h>
-#include "absl/types/optional.h"
+#include <optional>
+
#include "api/units/timestamp.h"
#include "api/video/encoded_image.h"
#include "api/video/video_codec_type.h"
@@ -38,7 +39,7 @@
virtual int64_t ReceivedTime() const { return -1; }
// Returns a Timestamp from `ReceivedTime`, or nullopt if there is no receive
// time.
- absl::optional<webrtc::Timestamp> ReceivedTimestamp() const;
+ std::optional<webrtc::Timestamp> ReceivedTimestamp() const;
// When this frame should be rendered.
// TODO(bugs.webrtc.org/13756): Use Timestamp instead of int.
@@ -47,7 +48,7 @@
int64_t RenderTimeMs() const { return _renderTimeMs; }
// Returns a Timestamp from `RenderTime`, or nullopt if there is no
// render time.
- absl::optional<webrtc::Timestamp> RenderTimestamp() const;
+ std::optional<webrtc::Timestamp> RenderTimestamp() const;
// This information is currently needed by the timing calculation class.
// TODO(philipel): Remove this function when a new timing class has
diff --git a/api/video/encoded_image.cc b/api/video/encoded_image.cc
index a859250..3e6fa54 100644
--- a/api/video/encoded_image.cc
+++ b/api/video/encoded_image.cc
@@ -14,8 +14,8 @@
#include <algorithm>
#include <cstdint>
+#include <optional>
-#include "absl/types/optional.h"
#include "api/make_ref_counted.h"
#include "api/scoped_refptr.h"
#include "api/units/timestamp.h"
@@ -88,14 +88,14 @@
: Timestamp::MinusInfinity();
}
-absl::optional<size_t> EncodedImage::SpatialLayerFrameSize(
+std::optional<size_t> EncodedImage::SpatialLayerFrameSize(
int spatial_index) const {
RTC_DCHECK_GE(spatial_index, 0);
RTC_DCHECK_LE(spatial_index, spatial_index_.value_or(0));
auto it = spatial_layer_frame_size_bytes_.find(spatial_index);
if (it == spatial_layer_frame_size_bytes_.end()) {
- return absl::nullopt;
+ return std::nullopt;
}
return it->second;
diff --git a/api/video/encoded_image.h b/api/video/encoded_image.h
index ec4252e..a9147e1 100644
--- a/api/video/encoded_image.h
+++ b/api/video/encoded_image.h
@@ -15,9 +15,9 @@
#include <cstddef>
#include <map>
+#include <optional>
#include <utility>
-#include "absl/types/optional.h"
#include "api/ref_count.h"
#include "api/rtp_packet_infos.h"
#include "api/scoped_refptr.h"
@@ -97,33 +97,33 @@
// Every simulcast layer (= encoding) has its own encoder and RTP stream.
// There can be no dependencies between different simulcast layers.
- absl::optional<int> SimulcastIndex() const { return simulcast_index_; }
- void SetSimulcastIndex(absl::optional<int> simulcast_index) {
+ std::optional<int> SimulcastIndex() const { return simulcast_index_; }
+ void SetSimulcastIndex(std::optional<int> simulcast_index) {
RTC_DCHECK_GE(simulcast_index.value_or(0), 0);
RTC_DCHECK_LT(simulcast_index.value_or(0), kMaxSimulcastStreams);
simulcast_index_ = simulcast_index;
}
- const absl::optional<Timestamp>& CaptureTimeIdentifier() const {
+ const std::optional<Timestamp>& CaptureTimeIdentifier() const {
return capture_time_identifier_;
}
void SetCaptureTimeIdentifier(
- const absl::optional<Timestamp>& capture_time_identifier) {
+ const std::optional<Timestamp>& capture_time_identifier) {
capture_time_identifier_ = capture_time_identifier;
}
// Encoded images can have dependencies between spatial and/or temporal
// layers, depending on the scalability mode used by the encoder. See diagrams
// at https://w3c.github.io/webrtc-svc/#dependencydiagrams*.
- absl::optional<int> SpatialIndex() const { return spatial_index_; }
- void SetSpatialIndex(absl::optional<int> spatial_index) {
+ std::optional<int> SpatialIndex() const { return spatial_index_; }
+ void SetSpatialIndex(std::optional<int> spatial_index) {
RTC_DCHECK_GE(spatial_index.value_or(0), 0);
RTC_DCHECK_LT(spatial_index.value_or(0), kMaxSpatialLayers);
spatial_index_ = spatial_index;
}
- absl::optional<int> TemporalIndex() const { return temporal_index_; }
- void SetTemporalIndex(absl::optional<int> temporal_index) {
+ std::optional<int> TemporalIndex() const { return temporal_index_; }
+ void SetTemporalIndex(std::optional<int> temporal_index) {
RTC_DCHECK_GE(temporal_index_.value_or(0), 0);
RTC_DCHECK_LT(temporal_index_.value_or(0), kMaxTemporalStreams);
temporal_index_ = temporal_index;
@@ -131,30 +131,30 @@
// These methods can be used to set/get size of subframe with spatial index
// `spatial_index` on encoded frames that consist of multiple spatial layers.
- absl::optional<size_t> SpatialLayerFrameSize(int spatial_index) const;
+ std::optional<size_t> SpatialLayerFrameSize(int spatial_index) const;
void SetSpatialLayerFrameSize(int spatial_index, size_t size_bytes);
const webrtc::ColorSpace* ColorSpace() const {
return color_space_ ? &*color_space_ : nullptr;
}
- void SetColorSpace(const absl::optional<webrtc::ColorSpace>& color_space) {
+ void SetColorSpace(const std::optional<webrtc::ColorSpace>& color_space) {
color_space_ = color_space;
}
- absl::optional<VideoPlayoutDelay> PlayoutDelay() const {
+ std::optional<VideoPlayoutDelay> PlayoutDelay() const {
return playout_delay_;
}
- void SetPlayoutDelay(absl::optional<VideoPlayoutDelay> playout_delay) {
+ void SetPlayoutDelay(std::optional<VideoPlayoutDelay> playout_delay) {
playout_delay_ = playout_delay;
}
// These methods along with the private member video_frame_tracking_id_ are
// meant for media quality testing purpose only.
- absl::optional<uint16_t> VideoFrameTrackingId() const {
+ std::optional<uint16_t> VideoFrameTrackingId() const {
return video_frame_tracking_id_;
}
- void SetVideoFrameTrackingId(absl::optional<uint16_t> tracking_id) {
+ void SetVideoFrameTrackingId(std::optional<uint16_t> tracking_id) {
video_frame_tracking_id_ = tracking_id;
}
@@ -255,20 +255,20 @@
// When set, indicates that all future frames will be constrained with those
// limits until the application indicates a change again.
- absl::optional<VideoPlayoutDelay> playout_delay_;
+ std::optional<VideoPlayoutDelay> playout_delay_;
rtc::scoped_refptr<EncodedImageBufferInterface> encoded_data_;
size_t size_ = 0; // Size of encoded frame data.
uint32_t timestamp_rtp_ = 0;
- absl::optional<int> simulcast_index_;
- absl::optional<Timestamp> capture_time_identifier_;
- absl::optional<int> spatial_index_;
- absl::optional<int> temporal_index_;
+ std::optional<int> simulcast_index_;
+ std::optional<Timestamp> capture_time_identifier_;
+ std::optional<int> spatial_index_;
+ std::optional<int> temporal_index_;
std::map<int, size_t> spatial_layer_frame_size_bytes_;
- absl::optional<webrtc::ColorSpace> color_space_;
+ std::optional<webrtc::ColorSpace> color_space_;
// This field is meant for media quality testing purpose only. When enabled it
// carries the webrtc::VideoFrame id field from the sender to the receiver.
- absl::optional<uint16_t> video_frame_tracking_id_;
+ std::optional<uint16_t> video_frame_tracking_id_;
// Information about packets used to assemble this video frame. This is needed
// by `SourceTracker` when the frame is delivered to the RTCRtpReceiver's
// MediaStreamTrack, in order to implement getContributingSources(). See:
diff --git a/api/video/frame_buffer.cc b/api/video/frame_buffer.cc
index b99ab0c..fe9d4c3 100644
--- a/api/video/frame_buffer.cc
+++ b/api/video/frame_buffer.cc
@@ -15,11 +15,11 @@
#include <cstdint>
#include <iterator>
#include <memory>
+#include <optional>
#include <utility>
#include "absl/algorithm/container.h"
#include "absl/container/inlined_vector.h"
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/field_trials_view.h"
#include "api/video/encoded_frame.h"
@@ -157,15 +157,15 @@
FindNextAndLastDecodableTemporalUnit();
}
-absl::optional<int64_t> FrameBuffer::LastContinuousFrameId() const {
+std::optional<int64_t> FrameBuffer::LastContinuousFrameId() const {
return last_continuous_frame_id_;
}
-absl::optional<int64_t> FrameBuffer::LastContinuousTemporalUnitFrameId() const {
+std::optional<int64_t> FrameBuffer::LastContinuousTemporalUnitFrameId() const {
return last_continuous_temporal_unit_frame_id_;
}
-absl::optional<FrameBuffer::DecodabilityInfo>
+std::optional<FrameBuffer::DecodabilityInfo>
FrameBuffer::DecodableTemporalUnitsInfo() const {
return decodable_temporal_units_info_;
}
diff --git a/api/video/frame_buffer.h b/api/video/frame_buffer.h
index 7791770..e607a18 100644
--- a/api/video/frame_buffer.h
+++ b/api/video/frame_buffer.h
@@ -15,9 +15,9 @@
#include <cstdint>
#include <map>
#include <memory>
+#include <optional>
#include "absl/container/inlined_vector.h"
-#include "absl/types/optional.h"
#include "api/field_trials_view.h"
#include "api/video/encoded_frame.h"
#include "modules/video_coding/utility/decoded_frames_history.h"
@@ -61,9 +61,9 @@
// Drop all frames in the next decodable unit.
void DropNextDecodableTemporalUnit();
- absl::optional<int64_t> LastContinuousFrameId() const;
- absl::optional<int64_t> LastContinuousTemporalUnitFrameId() const;
- absl::optional<DecodabilityInfo> DecodableTemporalUnitsInfo() const;
+ std::optional<int64_t> LastContinuousFrameId() const;
+ std::optional<int64_t> LastContinuousTemporalUnitFrameId() const;
+ std::optional<DecodabilityInfo> DecodableTemporalUnitsInfo() const;
int GetTotalNumberOfContinuousTemporalUnits() const;
int GetTotalNumberOfDroppedFrames() const;
@@ -92,10 +92,10 @@
const bool legacy_frame_id_jump_behavior_;
const size_t max_size_;
FrameMap frames_;
- absl::optional<TemporalUnit> next_decodable_temporal_unit_;
- absl::optional<DecodabilityInfo> decodable_temporal_units_info_;
- absl::optional<int64_t> last_continuous_frame_id_;
- absl::optional<int64_t> last_continuous_temporal_unit_frame_id_;
+ std::optional<TemporalUnit> next_decodable_temporal_unit_;
+ std::optional<DecodabilityInfo> decodable_temporal_units_info_;
+ std::optional<int64_t> last_continuous_frame_id_;
+ std::optional<int64_t> last_continuous_temporal_unit_frame_id_;
video_coding::DecodedFramesHistory decoded_frame_history_;
int num_continuous_temporal_units_ = 0;
diff --git a/api/video/frame_buffer_unittest.cc b/api/video/frame_buffer_unittest.cc
index 5d3341c..d97c7ec 100644
--- a/api/video/frame_buffer_unittest.cc
+++ b/api/video/frame_buffer_unittest.cc
@@ -9,9 +9,9 @@
*/
#include "api/video/frame_buffer.h"
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "test/fake_encoded_frame.h"
#include "test/gmock.h"
#include "test/gtest.h"
@@ -36,7 +36,7 @@
// Ref must be less than the id of this frame.
EXPECT_FALSE(buffer.InsertFrame(
test::FakeFrameBuilder().Time(0).Id(0).Refs({0}).AsLast().Build()));
- EXPECT_THAT(buffer.LastContinuousFrameId(), Eq(absl::nullopt));
+ EXPECT_THAT(buffer.LastContinuousFrameId(), Eq(std::nullopt));
// Duplicate ids are also invalid.
EXPECT_TRUE(buffer.InsertFrame(
@@ -50,13 +50,13 @@
test::ScopedKeyValueConfig field_trials;
FrameBuffer buffer(/*max_frame_slots=*/10, /*max_decode_history=*/100,
field_trials);
- EXPECT_THAT(buffer.LastContinuousFrameId(), Eq(absl::nullopt));
- EXPECT_THAT(buffer.LastContinuousTemporalUnitFrameId(), Eq(absl::nullopt));
+ EXPECT_THAT(buffer.LastContinuousFrameId(), Eq(std::nullopt));
+ EXPECT_THAT(buffer.LastContinuousTemporalUnitFrameId(), Eq(std::nullopt));
EXPECT_TRUE(
buffer.InsertFrame(test::FakeFrameBuilder().Time(10).Id(1).Build()));
EXPECT_THAT(buffer.LastContinuousFrameId(), Eq(1));
- EXPECT_THAT(buffer.LastContinuousTemporalUnitFrameId(), Eq(absl::nullopt));
+ EXPECT_THAT(buffer.LastContinuousTemporalUnitFrameId(), Eq(std::nullopt));
EXPECT_TRUE(buffer.InsertFrame(
test::FakeFrameBuilder().Time(10).Id(2).Refs({1}).AsLast().Build()));
@@ -87,7 +87,7 @@
EXPECT_TRUE(
buffer.InsertFrame(test::FakeFrameBuilder().Time(10).Id(1).Build()));
- EXPECT_THAT(buffer.LastContinuousTemporalUnitFrameId(), Eq(absl::nullopt));
+ EXPECT_THAT(buffer.LastContinuousTemporalUnitFrameId(), Eq(std::nullopt));
EXPECT_TRUE(buffer.InsertFrame(
test::FakeFrameBuilder().Time(10).Id(2).Refs({1}).AsLast().Build()));
EXPECT_THAT(buffer.LastContinuousTemporalUnitFrameId(), Eq(2));
@@ -104,7 +104,7 @@
test::FakeFrameBuilder().Time(20).Id(3).Refs({1}).Build()));
EXPECT_TRUE(buffer.InsertFrame(
test::FakeFrameBuilder().Time(20).Id(4).Refs({2, 3}).AsLast().Build()));
- EXPECT_THAT(buffer.LastContinuousTemporalUnitFrameId(), Eq(absl::nullopt));
+ EXPECT_THAT(buffer.LastContinuousTemporalUnitFrameId(), Eq(std::nullopt));
EXPECT_TRUE(buffer.InsertFrame(
test::FakeFrameBuilder().Time(10).Id(2).Refs({1}).AsLast().Build()));
@@ -116,7 +116,7 @@
FrameBuffer buffer(/*max_frame_slots=*/10, /*max_decode_history=*/100,
field_trials);
- EXPECT_THAT(buffer.DecodableTemporalUnitsInfo(), Eq(absl::nullopt));
+ EXPECT_THAT(buffer.DecodableTemporalUnitsInfo(), Eq(std::nullopt));
EXPECT_TRUE(buffer.InsertFrame(
test::FakeFrameBuilder().Time(10).Id(1).AsLast().Build()));
EXPECT_THAT(buffer.DecodableTemporalUnitsInfo()->next_rtp_timestamp, Eq(10U));
diff --git a/api/video/recordable_encoded_frame.h b/api/video/recordable_encoded_frame.h
index 5b56561..a561d09 100644
--- a/api/video/recordable_encoded_frame.h
+++ b/api/video/recordable_encoded_frame.h
@@ -11,7 +11,8 @@
#ifndef API_VIDEO_RECORDABLE_ENCODED_FRAME_H_
#define API_VIDEO_RECORDABLE_ENCODED_FRAME_H_
-#include "absl/types/optional.h"
+#include <optional>
+
#include "api/scoped_refptr.h"
#include "api/units/timestamp.h"
#include "api/video/color_space.h"
@@ -40,7 +41,7 @@
// Optionally returns the colorspace of the encoded frame. This can differ
// from the eventually decoded frame's colorspace.
- virtual absl::optional<webrtc::ColorSpace> color_space() const = 0;
+ virtual std::optional<webrtc::ColorSpace> color_space() const = 0;
// Returns the codec of the encoded frame
virtual VideoCodecType codec() const = 0;
diff --git a/api/video/rtp_video_frame_assembler.cc b/api/video/rtp_video_frame_assembler.cc
index 4631824..5dcb8f3 100644
--- a/api/video/rtp_video_frame_assembler.cc
+++ b/api/video/rtp_video_frame_assembler.cc
@@ -12,11 +12,11 @@
#include <cstdint>
#include <memory>
+#include <optional>
#include <utility>
#include <vector>
#include "absl/container/inlined_vector.h"
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/rtp_packet_infos.h"
#include "api/scoped_refptr.h"
@@ -94,7 +94,7 @@
std::unique_ptr<FrameDependencyStructure> video_structure_;
SeqNumUnwrapper<uint16_t> rtp_sequence_number_unwrapper_;
SeqNumUnwrapper<uint16_t> frame_id_unwrapper_;
- absl::optional<int64_t> video_structure_frame_id_;
+ std::optional<int64_t> video_structure_frame_id_;
std::unique_ptr<VideoRtpDepacketizer> depacketizer_;
video_coding::PacketBuffer packet_buffer_;
RtpFrameReferenceFinder reference_finder_;
@@ -112,10 +112,10 @@
return UpdateWithPadding(rtp_packet.SequenceNumber());
}
- absl::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed_payload =
+ std::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed_payload =
depacketizer_->Parse(rtp_packet.PayloadBuffer());
- if (parsed_payload == absl::nullopt) {
+ if (parsed_payload == std::nullopt) {
return {};
}
diff --git a/api/video/rtp_video_frame_assembler_unittests.cc b/api/video/rtp_video_frame_assembler_unittests.cc
index 0becb9f..1143f46 100644
--- a/api/video/rtp_video_frame_assembler_unittests.cc
+++ b/api/video/rtp_video_frame_assembler_unittests.cc
@@ -11,9 +11,9 @@
#include <cstdint>
#include <iterator>
#include <memory>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/transport/rtp/dependency_descriptor.h"
#include "api/video/encoded_frame.h"
@@ -86,10 +86,10 @@
}
private:
- absl::optional<VideoCodecType> GetVideoCodecType() {
+ std::optional<VideoCodecType> GetVideoCodecType() {
switch (format_) {
case PayloadFormat::kRaw: {
- return absl::nullopt;
+ return std::nullopt;
}
case PayloadFormat::kH264: {
return kVideoCodecH264;
@@ -111,7 +111,7 @@
}
}
RTC_DCHECK_NOTREACHED();
- return absl::nullopt;
+ return std::nullopt;
}
const RtpVideoFrameAssembler::PayloadFormat format_;
diff --git a/api/video/test/BUILD.gn b/api/video/test/BUILD.gn
index 4450a0a..3fbd7d3 100644
--- a/api/video/test/BUILD.gn
+++ b/api/video/test/BUILD.gn
@@ -29,7 +29,6 @@
"../..:scoped_refptr",
"../../../test:frame_utils",
"../../../test:test_support",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
diff --git a/api/video/test/mock_recordable_encoded_frame.h b/api/video/test/mock_recordable_encoded_frame.h
index 2178932..086b17b 100644
--- a/api/video/test/mock_recordable_encoded_frame.h
+++ b/api/video/test/mock_recordable_encoded_frame.h
@@ -21,7 +21,7 @@
encoded_buffer,
(),
(const, override));
- MOCK_METHOD(absl::optional<webrtc::ColorSpace>,
+ MOCK_METHOD(std::optional<webrtc::ColorSpace>,
color_space,
(),
(const, override));
diff --git a/api/video/test/video_bitrate_allocation_unittest.cc b/api/video/test/video_bitrate_allocation_unittest.cc
index 8e66d4b..f84fa69 100644
--- a/api/video/test/video_bitrate_allocation_unittest.cc
+++ b/api/video/test/video_bitrate_allocation_unittest.cc
@@ -10,9 +10,9 @@
#include "api/video/video_bitrate_allocation.h"
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "test/gtest.h"
namespace webrtc {
@@ -31,7 +31,7 @@
layer1_bitrate.SetBitrate(0, 0, 40000);
layer1_bitrate.SetBitrate(0, 1, 80000);
- std::vector<absl::optional<VideoBitrateAllocation>> layer_allocations =
+ std::vector<std::optional<VideoBitrateAllocation>> layer_allocations =
bitrate.GetSimulcastAllocations();
EXPECT_EQ(layer0_bitrate, layer_allocations[0]);
@@ -54,7 +54,7 @@
layer2_bitrate.SetBitrate(0, 0, 40000);
layer2_bitrate.SetBitrate(0, 1, 80000);
- std::vector<absl::optional<VideoBitrateAllocation>> layer_allocations =
+ std::vector<std::optional<VideoBitrateAllocation>> layer_allocations =
bitrate.GetSimulcastAllocations();
EXPECT_EQ(layer0_bitrate, layer_allocations[0]);
diff --git a/api/video/video_bitrate_allocation.cc b/api/video/video_bitrate_allocation.cc
index af5ce82..889dbd0 100644
--- a/api/video/video_bitrate_allocation.cc
+++ b/api/video/video_bitrate_allocation.cc
@@ -12,10 +12,10 @@
#include <cstddef>
#include <cstdint>
+#include <optional>
#include <string>
#include <vector>
-#include "absl/types/optional.h"
#include "api/video/video_codec_constants.h"
#include "rtc_base/checks.h"
#include "rtc_base/numerics/safe_conversions.h"
@@ -32,7 +32,7 @@
RTC_CHECK_LT(spatial_index, kMaxSpatialLayers);
RTC_CHECK_LT(temporal_index, kMaxTemporalStreams);
int64_t new_bitrate_sum_bps = sum_;
- absl::optional<uint32_t>& layer_bitrate =
+ std::optional<uint32_t>& layer_bitrate =
bitrates_[spatial_index][temporal_index];
if (layer_bitrate) {
RTC_DCHECK_LE(*layer_bitrate, sum_);
@@ -112,11 +112,11 @@
return temporal_rates;
}
-std::vector<absl::optional<VideoBitrateAllocation>>
+std::vector<std::optional<VideoBitrateAllocation>>
VideoBitrateAllocation::GetSimulcastAllocations() const {
- std::vector<absl::optional<VideoBitrateAllocation>> bitrates;
+ std::vector<std::optional<VideoBitrateAllocation>> bitrates;
for (size_t si = 0; si < kMaxSpatialLayers; ++si) {
- absl::optional<VideoBitrateAllocation> layer_bitrate;
+ std::optional<VideoBitrateAllocation> layer_bitrate;
if (IsSpatialLayerUsed(si)) {
layer_bitrate = VideoBitrateAllocation();
for (int tl = 0; tl < kMaxTemporalStreams; ++tl) {
diff --git a/api/video/video_bitrate_allocation.h b/api/video/video_bitrate_allocation.h
index 4feffa2..63ac26a 100644
--- a/api/video/video_bitrate_allocation.h
+++ b/api/video/video_bitrate_allocation.h
@@ -15,10 +15,10 @@
#include <stdint.h>
#include <limits>
+#include <optional>
#include <string>
#include <vector>
-#include "absl/types/optional.h"
#include "api/video/video_codec_constants.h"
#include "rtc_base/system/rtc_export.h"
@@ -64,7 +64,7 @@
// Returns one VideoBitrateAllocation for each spatial layer. This is used to
// configure simulcast streams. Note that the length of the returned vector is
// always kMaxSpatialLayers, the optional is unset for unused layers.
- std::vector<absl::optional<VideoBitrateAllocation>> GetSimulcastAllocations()
+ std::vector<std::optional<VideoBitrateAllocation>> GetSimulcastAllocations()
const;
uint32_t get_sum_bps() const { return sum_; } // Sum of all bitrates.
@@ -87,7 +87,7 @@
private:
uint32_t sum_;
- absl::optional<uint32_t> bitrates_[kMaxSpatialLayers][kMaxTemporalStreams];
+ std::optional<uint32_t> bitrates_[kMaxSpatialLayers][kMaxTemporalStreams];
bool is_bw_limited_;
};
diff --git a/api/video/video_frame.cc b/api/video/video_frame.cc
index cb40c89b..9e226f8 100644
--- a/api/video/video_frame.cc
+++ b/api/video/video_frame.cc
@@ -12,9 +12,9 @@
#include <algorithm>
#include <cstdint>
+#include <optional>
#include <utility>
-#include "absl/types/optional.h"
#include "api/rtp_packet_infos.h"
#include "api/scoped_refptr.h"
#include "api/units/timestamp.h"
@@ -196,13 +196,13 @@
}
VideoFrame::Builder& VideoFrame::Builder::set_capture_time_identifier(
- const absl::optional<Timestamp>& capture_time_identifier) {
+ const std::optional<Timestamp>& capture_time_identifier) {
capture_time_identifier_ = capture_time_identifier;
return *this;
}
VideoFrame::Builder& VideoFrame::Builder::set_reference_time(
- const absl::optional<Timestamp>& reference_time) {
+ const std::optional<Timestamp>& reference_time) {
reference_time_ = reference_time;
return *this;
}
@@ -230,15 +230,14 @@
}
VideoFrame::Builder& VideoFrame::Builder::set_color_space(
- const absl::optional<ColorSpace>& color_space) {
+ const std::optional<ColorSpace>& color_space) {
color_space_ = color_space;
return *this;
}
VideoFrame::Builder& VideoFrame::Builder::set_color_space(
const ColorSpace* color_space) {
- color_space_ =
- color_space ? absl::make_optional(*color_space) : absl::nullopt;
+ color_space_ = color_space ? std::make_optional(*color_space) : std::nullopt;
return *this;
}
@@ -248,7 +247,7 @@
}
VideoFrame::Builder& VideoFrame::Builder::set_update_rect(
- const absl::optional<VideoFrame::UpdateRect>& update_rect) {
+ const std::optional<VideoFrame::UpdateRect>& update_rect) {
update_rect_ = update_rect;
return *this;
}
@@ -283,14 +282,14 @@
VideoFrame::VideoFrame(uint16_t id,
const rtc::scoped_refptr<VideoFrameBuffer>& buffer,
int64_t timestamp_us,
- const absl::optional<Timestamp>& capture_time_identifier,
- const absl::optional<Timestamp>& reference_time,
+ const std::optional<Timestamp>& capture_time_identifier,
+ const std::optional<Timestamp>& reference_time,
uint32_t timestamp_rtp,
int64_t ntp_time_ms,
VideoRotation rotation,
- const absl::optional<ColorSpace>& color_space,
+ const std::optional<ColorSpace>& color_space,
const RenderParameters& render_parameters,
- const absl::optional<UpdateRect>& update_rect,
+ const std::optional<UpdateRect>& update_rect,
RtpPacketInfos packet_infos)
: id_(id),
video_frame_buffer_(buffer),
diff --git a/api/video/video_frame.h b/api/video/video_frame.h
index ef01e5f..b1aa751 100644
--- a/api/video/video_frame.h
+++ b/api/video/video_frame.h
@@ -13,9 +13,9 @@
#include <stdint.h>
+#include <optional>
#include <utility>
-#include "absl/types/optional.h"
#include "api/rtp_packet_infos.h"
#include "api/scoped_refptr.h"
#include "api/units/time_delta.h"
@@ -84,7 +84,7 @@
struct RTC_EXPORT RenderParameters {
bool use_low_latency_rendering = false;
- absl::optional<int32_t> max_composition_delay_in_frames;
+ std::optional<int32_t> max_composition_delay_in_frames;
bool operator==(const RenderParameters& other) const {
return other.use_low_latency_rendering == use_low_latency_rendering &&
@@ -109,32 +109,31 @@
Builder& set_timestamp_ms(int64_t timestamp_ms);
Builder& set_timestamp_us(int64_t timestamp_us);
Builder& set_capture_time_identifier(
- const absl::optional<Timestamp>& capture_time_identifier);
- Builder& set_reference_time(
- const absl::optional<Timestamp>& reference_time);
+ const std::optional<Timestamp>& capture_time_identifier);
+ Builder& set_reference_time(const std::optional<Timestamp>& reference_time);
Builder& set_rtp_timestamp(uint32_t rtp_timestamp);
// TODO(https://bugs.webrtc.org/13756): Deprecate and use set_rtp_timestamp.
Builder& set_timestamp_rtp(uint32_t timestamp_rtp);
Builder& set_ntp_time_ms(int64_t ntp_time_ms);
Builder& set_rotation(VideoRotation rotation);
- Builder& set_color_space(const absl::optional<ColorSpace>& color_space);
+ Builder& set_color_space(const std::optional<ColorSpace>& color_space);
Builder& set_color_space(const ColorSpace* color_space);
Builder& set_id(uint16_t id);
- Builder& set_update_rect(const absl::optional<UpdateRect>& update_rect);
+ Builder& set_update_rect(const std::optional<UpdateRect>& update_rect);
Builder& set_packet_infos(RtpPacketInfos packet_infos);
private:
uint16_t id_ = kNotSetId;
rtc::scoped_refptr<webrtc::VideoFrameBuffer> video_frame_buffer_;
int64_t timestamp_us_ = 0;
- absl::optional<Timestamp> capture_time_identifier_;
- absl::optional<Timestamp> reference_time_;
+ std::optional<Timestamp> capture_time_identifier_;
+ std::optional<Timestamp> reference_time_;
uint32_t timestamp_rtp_ = 0;
int64_t ntp_time_ms_ = 0;
VideoRotation rotation_ = kVideoRotation_0;
- absl::optional<ColorSpace> color_space_;
+ std::optional<ColorSpace> color_space_;
RenderParameters render_parameters_;
- absl::optional<UpdateRect> update_rect_;
+ std::optional<UpdateRect> update_rect_;
RtpPacketInfos packet_infos_;
};
@@ -175,18 +174,18 @@
int64_t timestamp_us() const { return timestamp_us_; }
void set_timestamp_us(int64_t timestamp_us) { timestamp_us_ = timestamp_us; }
- const absl::optional<Timestamp>& capture_time_identifier() const {
+ const std::optional<Timestamp>& capture_time_identifier() const {
return capture_time_identifier_;
}
void set_capture_time_identifier(
- const absl::optional<Timestamp>& capture_time_identifier) {
+ const std::optional<Timestamp>& capture_time_identifier) {
capture_time_identifier_ = capture_time_identifier;
}
- const absl::optional<Timestamp>& reference_time() const {
+ const std::optional<Timestamp>& reference_time() const {
return reference_time_;
}
- void set_reference_time(const absl::optional<Timestamp>& reference_time) {
+ void set_reference_time(const std::optional<Timestamp>& reference_time) {
reference_time_ = reference_time;
}
@@ -218,8 +217,8 @@
void set_rotation(VideoRotation rotation) { rotation_ = rotation; }
// Get color space when available.
- const absl::optional<ColorSpace>& color_space() const { return color_space_; }
- void set_color_space(const absl::optional<ColorSpace>& color_space) {
+ const std::optional<ColorSpace>& color_space() const { return color_space_; }
+ void set_color_space(const std::optional<ColorSpace>& color_space) {
color_space_ = color_space;
}
@@ -260,7 +259,7 @@
update_rect_ = update_rect;
}
- void clear_update_rect() { update_rect_ = absl::nullopt; }
+ void clear_update_rect() { update_rect_ = std::nullopt; }
// Get information about packets used to assemble this video frame. Might be
// empty if the information isn't available.
@@ -269,7 +268,7 @@
packet_infos_ = std::move(value);
}
- const absl::optional<ProcessingTime> processing_time() const {
+ const std::optional<ProcessingTime> processing_time() const {
return processing_time_;
}
void set_processing_time(const ProcessingTime& processing_time) {
@@ -280,14 +279,14 @@
VideoFrame(uint16_t id,
const rtc::scoped_refptr<VideoFrameBuffer>& buffer,
int64_t timestamp_us,
- const absl::optional<Timestamp>& capture_time_identifier,
- const absl::optional<Timestamp>& reference_time,
+ const std::optional<Timestamp>& capture_time_identifier,
+ const std::optional<Timestamp>& reference_time,
uint32_t timestamp_rtp,
int64_t ntp_time_ms,
VideoRotation rotation,
- const absl::optional<ColorSpace>& color_space,
+ const std::optional<ColorSpace>& color_space,
const RenderParameters& render_parameters,
- const absl::optional<UpdateRect>& update_rect,
+ const std::optional<UpdateRect>& update_rect,
RtpPacketInfos packet_infos);
uint16_t id_;
@@ -296,21 +295,21 @@
uint32_t timestamp_rtp_;
int64_t ntp_time_ms_;
int64_t timestamp_us_;
- absl::optional<Timestamp> capture_time_identifier_;
+ std::optional<Timestamp> capture_time_identifier_;
// Contains a monotonically increasing clock time and represents the time
// when the frame was captured. Not all platforms provide the "true" sample
// capture time in |reference_time| but might instead use a somewhat delayed
// (by the time it took to capture the frame) version of it.
- absl::optional<Timestamp> reference_time_;
+ std::optional<Timestamp> reference_time_;
VideoRotation rotation_;
- absl::optional<ColorSpace> color_space_;
+ std::optional<ColorSpace> color_space_;
// Contains parameters that affect have the frame should be rendered.
RenderParameters render_parameters_;
// Updated since the last frame area. If present it means that the bounding
// box of all the changes is within the rectangular area and is close to it.
// If absent, it means that there's no information about the change at all and
// update_rect() will return a rectangle corresponding to the entire frame.
- absl::optional<UpdateRect> update_rect_;
+ std::optional<UpdateRect> update_rect_;
// Information about packets used to assemble this video frame. This is needed
// by `SourceTracker` when the frame is delivered to the RTCRtpReceiver's
// MediaStreamTrack, in order to implement getContributingSources(). See:
@@ -320,7 +319,7 @@
// timestamps when the frame is sent to the decoder and the decoded image
// returned from the decoder.
// Currently, not set for locally captured video frames.
- absl::optional<ProcessingTime> processing_time_;
+ std::optional<ProcessingTime> processing_time_;
};
} // namespace webrtc
diff --git a/api/video/video_frame_metadata.cc b/api/video/video_frame_metadata.cc
index ca14fae..dc92cdb 100644
--- a/api/video/video_frame_metadata.cc
+++ b/api/video/video_frame_metadata.cc
@@ -11,10 +11,10 @@
#include "api/video/video_frame_metadata.h"
#include <cstdint>
+#include <optional>
#include <utility>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/transport/rtp/dependency_descriptor.h"
#include "api/video/video_codec_type.h"
@@ -66,11 +66,11 @@
content_type_ = content_type;
}
-absl::optional<int64_t> VideoFrameMetadata::GetFrameId() const {
+std::optional<int64_t> VideoFrameMetadata::GetFrameId() const {
return frame_id_;
}
-void VideoFrameMetadata::SetFrameId(absl::optional<int64_t> frame_id) {
+void VideoFrameMetadata::SetFrameId(std::optional<int64_t> frame_id) {
frame_id_ = frame_id;
}
diff --git a/api/video/video_frame_metadata.h b/api/video/video_frame_metadata.h
index bf46387..4ec0e98 100644
--- a/api/video/video_frame_metadata.h
+++ b/api/video/video_frame_metadata.h
@@ -12,10 +12,10 @@
#define API_VIDEO_VIDEO_FRAME_METADATA_H_
#include <cstdint>
+#include <optional>
#include <vector>
#include "absl/container/inlined_vector.h"
-#include "absl/types/optional.h"
#include "absl/types/variant.h"
#include "api/array_view.h"
#include "api/transport/rtp/dependency_descriptor.h"
@@ -58,8 +58,8 @@
VideoContentType GetContentType() const;
void SetContentType(VideoContentType content_type);
- absl::optional<int64_t> GetFrameId() const;
- void SetFrameId(absl::optional<int64_t> frame_id);
+ std::optional<int64_t> GetFrameId() const;
+ void SetFrameId(std::optional<int64_t> frame_id);
int GetSpatialIndex() const;
void SetSpatialIndex(int spatial_index);
@@ -107,7 +107,7 @@
VideoContentType content_type_ = VideoContentType::UNSPECIFIED;
// Corresponding to GenericDescriptorInfo.
- absl::optional<int64_t> frame_id_;
+ std::optional<int64_t> frame_id_;
int spatial_index_ = 0;
int temporal_index_ = 0;
absl::InlinedVector<int64_t, 5> frame_dependencies_;
diff --git a/api/video/video_source_interface.h b/api/video/video_source_interface.h
index c636c2f..d85e56b 100644
--- a/api/video/video_source_interface.h
+++ b/api/video/video_source_interface.h
@@ -12,9 +12,9 @@
#define API_VIDEO_VIDEO_SOURCE_INTERFACE_H_
#include <limits>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/video/video_sink_interface.h"
#include "rtc_base/system/rtc_export.h"
@@ -50,7 +50,7 @@
// have improved after an earlier downgrade. The source should select the
// closest resolution to this pixel count, but if max_pixel_count is set, it
// still sets the absolute upper bound.
- absl::optional<int> target_pixel_count;
+ std::optional<int> target_pixel_count;
// Tells the source the maximum framerate the sink wants.
int max_framerate_fps = std::numeric_limits<int>::max();
@@ -83,7 +83,7 @@
std::vector<FrameSize> resolutions;
// This is the resolution requested by the user using RtpEncodingParameters.
- absl::optional<FrameSize> requested_resolution;
+ std::optional<FrameSize> requested_resolution;
// `is_active` : Is this VideoSinkWants from an encoder that is encoding any
// layer. IF YES, it will affect how the VideoAdapter will choose to
@@ -102,7 +102,7 @@
// OnOutputFormatRequest to handle encode resolution.
bool any_active_without_requested_resolution = false;
};
- absl::optional<Aggregates> aggregates;
+ std::optional<Aggregates> aggregates;
};
inline bool operator==(const VideoSinkWants::FrameSize& a,
diff --git a/api/video_codecs/BUILD.gn b/api/video_codecs/BUILD.gn
index 6990894..108e11c 100644
--- a/api/video_codecs/BUILD.gn
+++ b/api/video_codecs/BUILD.gn
@@ -35,7 +35,6 @@
":scalability_mode",
"../../modules/video_coding/svc:scalability_mode_util",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -104,17 +103,13 @@
"//third_party/abseil-cpp/absl/container:inlined_vector",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
rtc_source_set("bitstream_parser_api") {
visibility = [ "*" ]
sources = [ "bitstream_parser.h" ]
- deps = [
- "..:array_view",
- "//third_party/abseil-cpp/absl/types:optional",
- ]
+ deps = [ "..:array_view" ]
}
rtc_library("builtin_video_decoder_factory") {
@@ -150,7 +145,6 @@
"../../media:rtc_simulcast_encoder_adapter",
"../../rtc_base/system:rtc_export",
"../environment",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -166,7 +160,6 @@
"../../modules/video_coding/svc:scalability_mode_util",
"../environment",
"//third_party/abseil-cpp/absl/algorithm:container",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -306,7 +299,6 @@
"../../api/video_codecs:video_codecs_api",
"../../rtc_base:rtc_numerics",
"//third_party/abseil-cpp/absl/functional:any_invocable",
- "//third_party/abseil-cpp/absl/types:optional",
"//third_party/abseil-cpp/absl/types:variant",
]
}
@@ -321,7 +313,6 @@
"../../api/video:resolution",
"../../rtc_base:rtc_numerics",
"../video:video_frame",
- "//third_party/abseil-cpp/absl/types:optional",
"//third_party/abseil-cpp/absl/types:variant",
]
}
@@ -354,7 +345,6 @@
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/functional:any_invocable",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
"//third_party/abseil-cpp/absl/types:variant",
]
}
@@ -374,7 +364,6 @@
"../../test:fileutils",
"../../test:test_support",
"../../test:video_test_support",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -402,7 +391,6 @@
"../video:video_frame",
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/cleanup",
- "//third_party/abseil-cpp/absl/types:optional",
"//third_party/abseil-cpp/absl/types:variant",
"//third_party/libaom",
]
@@ -435,7 +423,6 @@
"../units:time_delta",
"../units:timestamp",
"../video:encoded_image",
- "//third_party/abseil-cpp/absl/types:optional",
"//third_party/abseil-cpp/absl/types:variant",
]
}
@@ -489,6 +476,5 @@
"../video:video_frame_type",
"../video:video_rtp_headers",
"//third_party/abseil-cpp/absl/strings",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
diff --git a/api/video_codecs/av1_profile.cc b/api/video_codecs/av1_profile.cc
index 5181fe1..177ebf0 100644
--- a/api/video_codecs/av1_profile.cc
+++ b/api/video_codecs/av1_profile.cc
@@ -11,11 +11,11 @@
#include "api/video_codecs/av1_profile.h"
#include <map>
+#include <optional>
#include <string>
#include <utility>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/rtp_parameters.h"
#include "media/base/media_constants.h"
#include "rtc_base/string_to_number.h"
@@ -34,10 +34,10 @@
return "0";
}
-absl::optional<AV1Profile> StringToAV1Profile(absl::string_view str) {
- const absl::optional<int> i = rtc::StringToNumber<int>(str);
+std::optional<AV1Profile> StringToAV1Profile(absl::string_view str) {
+ const std::optional<int> i = rtc::StringToNumber<int>(str);
if (!i.has_value())
- return absl::nullopt;
+ return std::nullopt;
switch (i.value()) {
case 0:
@@ -47,11 +47,11 @@
case 2:
return AV1Profile::kProfile2;
default:
- return absl::nullopt;
+ return std::nullopt;
}
}
-absl::optional<AV1Profile> ParseSdpForAV1Profile(
+std::optional<AV1Profile> ParseSdpForAV1Profile(
const CodecParameterMap& params) {
const auto profile_it = params.find(cricket::kAv1FmtpProfile);
if (profile_it == params.end())
@@ -62,8 +62,8 @@
bool AV1IsSameProfile(const CodecParameterMap& params1,
const CodecParameterMap& params2) {
- const absl::optional<AV1Profile> profile = ParseSdpForAV1Profile(params1);
- const absl::optional<AV1Profile> other_profile =
+ const std::optional<AV1Profile> profile = ParseSdpForAV1Profile(params1);
+ const std::optional<AV1Profile> other_profile =
ParseSdpForAV1Profile(params2);
return profile && other_profile && profile == other_profile;
}
diff --git a/api/video_codecs/av1_profile.h b/api/video_codecs/av1_profile.h
index fde4633..a96285d 100644
--- a/api/video_codecs/av1_profile.h
+++ b/api/video_codecs/av1_profile.h
@@ -11,9 +11,9 @@
#ifndef API_VIDEO_CODECS_AV1_PROFILE_H_
#define API_VIDEO_CODECS_AV1_PROFILE_H_
+#include <optional>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/rtp_parameters.h"
#include "rtc_base/system/rtc_export.h"
@@ -34,13 +34,13 @@
// Helper function which converts a std::string to AV1Profile. Returns null if
// |profile| is not a valid profile string.
-absl::optional<AV1Profile> StringToAV1Profile(absl::string_view profile);
+std::optional<AV1Profile> StringToAV1Profile(absl::string_view profile);
// Parses an SDP key-value map of format parameters to retrive an AV1 profile.
// Returns an AV1Profile if one has been specified, `kProfile0` if no profile is
// specified and an empty value if the profile key is present but contains an
// invalid value.
-RTC_EXPORT absl::optional<AV1Profile> ParseSdpForAV1Profile(
+RTC_EXPORT std::optional<AV1Profile> ParseSdpForAV1Profile(
const CodecParameterMap& params);
// Returns true if the parameters have the same AV1 profile or neither contains
diff --git a/api/video_codecs/bitstream_parser.h b/api/video_codecs/bitstream_parser.h
index 86ce192..282b105 100644
--- a/api/video_codecs/bitstream_parser.h
+++ b/api/video_codecs/bitstream_parser.h
@@ -14,7 +14,8 @@
#include <stddef.h>
#include <stdint.h>
-#include "absl/types/optional.h"
+#include <optional>
+
#include "api/array_view.h"
namespace webrtc {
@@ -28,8 +29,8 @@
virtual void ParseBitstream(rtc::ArrayView<const uint8_t> bitstream) = 0;
// Get the last extracted QP value from the parsed bitstream. If no QP
- // value could be parsed, returns absl::nullopt.
- virtual absl::optional<int> GetLastSliceQp() const = 0;
+ // value could be parsed, returns std::nullopt.
+ virtual std::optional<int> GetLastSliceQp() const = 0;
};
} // namespace webrtc
diff --git a/api/video_codecs/builtin_video_encoder_factory.cc b/api/video_codecs/builtin_video_encoder_factory.cc
index bbd5222..bd4bad6 100644
--- a/api/video_codecs/builtin_video_encoder_factory.cc
+++ b/api/video_codecs/builtin_video_encoder_factory.cc
@@ -11,10 +11,10 @@
#include "api/video_codecs/builtin_video_encoder_factory.h"
#include <memory>
+#include <optional>
#include <string>
#include <vector>
-#include "absl/types/optional.h"
#include "api/environment/environment.h"
#include "api/video_codecs/sdp_video_format.h"
#include "api/video_codecs/video_encoder.h"
@@ -54,7 +54,7 @@
CodecSupport QueryCodecSupport(
const SdpVideoFormat& format,
- absl::optional<std::string> scalability_mode) const override {
+ std::optional<std::string> scalability_mode) const override {
return internal_encoder_factory_->QueryCodecSupport(format,
scalability_mode);
}
diff --git a/api/video_codecs/h264_profile_level_id.cc b/api/video_codecs/h264_profile_level_id.cc
index 09f7fa5..9a2dca9 100644
--- a/api/video_codecs/h264_profile_level_id.cc
+++ b/api/video_codecs/h264_profile_level_id.cc
@@ -14,9 +14,9 @@
#include <cstdio>
#include <cstdlib>
#include <cstring>
+#include <optional>
#include <string>
-#include "absl/types/optional.h"
#include "api/rtp_parameters.h"
#include "rtc_base/arraysize.h"
@@ -103,13 +103,13 @@
} // anonymous namespace
-absl::optional<H264ProfileLevelId> ParseH264ProfileLevelId(const char* str) {
+std::optional<H264ProfileLevelId> ParseH264ProfileLevelId(const char* str) {
// The string should consist of 3 bytes in hexadecimal format.
if (strlen(str) != 6u)
- return absl::nullopt;
+ return std::nullopt;
const uint32_t profile_level_id_numeric = strtol(str, nullptr, 16);
if (profile_level_id_numeric == 0)
- return absl::nullopt;
+ return std::nullopt;
// Separate into three bytes.
const uint8_t level_idc =
@@ -147,7 +147,7 @@
break;
default:
// Unrecognized level_idc.
- return absl::nullopt;
+ return std::nullopt;
}
// Parse profile_idc/profile_iop into a Profile enum.
@@ -159,11 +159,11 @@
}
// Unrecognized profile_idc/profile_iop combination.
- return absl::nullopt;
+ return std::nullopt;
}
-absl::optional<H264Level> H264SupportedLevel(int max_frame_pixel_count,
- float max_fps) {
+std::optional<H264Level> H264SupportedLevel(int max_frame_pixel_count,
+ float max_fps) {
static const int kPixelsPerMacroblock = 16 * 16;
for (int i = arraysize(kLevelConstraints) - 1; i >= 0; --i) {
@@ -177,10 +177,10 @@
}
// No level supported.
- return absl::nullopt;
+ return std::nullopt;
}
-absl::optional<H264ProfileLevelId> ParseSdpForH264ProfileLevelId(
+std::optional<H264ProfileLevelId> ParseSdpForH264ProfileLevelId(
const CodecParameterMap& params) {
// TODO(magjed): The default should really be kProfileBaseline and kLevel1
// according to the spec: https://tools.ietf.org/html/rfc6184#section-8.1. In
@@ -198,7 +198,7 @@
: ParseH264ProfileLevelId(profile_level_id_it->second.c_str());
}
-absl::optional<std::string> H264ProfileLevelIdToString(
+std::optional<std::string> H264ProfileLevelIdToString(
const H264ProfileLevelId& profile_level_id) {
// Handle special case level == 1b.
if (profile_level_id.level == H264Level::kLevel1_b) {
@@ -211,7 +211,7 @@
return {"4d100b"};
// Level 1b is not allowed for other profiles.
default:
- return absl::nullopt;
+ return std::nullopt;
}
}
@@ -237,7 +237,7 @@
break;
// Unrecognized profile.
default:
- return absl::nullopt;
+ return std::nullopt;
}
char str[7];
@@ -248,9 +248,9 @@
bool H264IsSameProfile(const CodecParameterMap& params1,
const CodecParameterMap& params2) {
- const absl::optional<H264ProfileLevelId> profile_level_id =
+ const std::optional<H264ProfileLevelId> profile_level_id =
ParseSdpForH264ProfileLevelId(params1);
- const absl::optional<H264ProfileLevelId> other_profile_level_id =
+ const std::optional<H264ProfileLevelId> other_profile_level_id =
ParseSdpForH264ProfileLevelId(params2);
// Compare H264 profiles, but not levels.
return profile_level_id && other_profile_level_id &&
diff --git a/api/video_codecs/h264_profile_level_id.h b/api/video_codecs/h264_profile_level_id.h
index ca8c1d5..593b8e9 100644
--- a/api/video_codecs/h264_profile_level_id.h
+++ b/api/video_codecs/h264_profile_level_id.h
@@ -11,9 +11,9 @@
#ifndef API_VIDEO_CODECS_H264_PROFILE_LEVEL_ID_H_
#define API_VIDEO_CODECS_H264_PROFILE_LEVEL_ID_H_
+#include <optional>
#include <string>
-#include "absl/types/optional.h"
#include "api/rtp_parameters.h"
#include "rtc_base/system/rtc_export.h"
@@ -60,26 +60,26 @@
// Parse profile level id that is represented as a string of 3 hex bytes.
// Nothing will be returned if the string is not a recognized H264
// profile level id.
-absl::optional<H264ProfileLevelId> ParseH264ProfileLevelId(const char* str);
+std::optional<H264ProfileLevelId> ParseH264ProfileLevelId(const char* str);
// Parse profile level id that is represented as a string of 3 hex bytes
// contained in an SDP key-value map. A default profile level id will be
// returned if the profile-level-id key is missing. Nothing will be returned if
// the key is present but the string is invalid.
-RTC_EXPORT absl::optional<H264ProfileLevelId> ParseSdpForH264ProfileLevelId(
+RTC_EXPORT std::optional<H264ProfileLevelId> ParseSdpForH264ProfileLevelId(
const CodecParameterMap& params);
// Given that a decoder supports up to a given frame size (in pixels) at up to a
// given number of frames per second, return the highest H.264 level where it
// can guarantee that it will be able to support all valid encoded streams that
// are within that level.
-RTC_EXPORT absl::optional<H264Level> H264SupportedLevel(
+RTC_EXPORT std::optional<H264Level> H264SupportedLevel(
int max_frame_pixel_count,
float max_fps);
// Returns canonical string representation as three hex bytes of the profile
// level id, or returns nothing for invalid profile level ids.
-RTC_EXPORT absl::optional<std::string> H264ProfileLevelIdToString(
+RTC_EXPORT std::optional<std::string> H264ProfileLevelIdToString(
const H264ProfileLevelId& profile_level_id);
// Returns true if the parameters have the same H264 profile (Baseline, High,
diff --git a/api/video_codecs/h265_profile_tier_level.cc b/api/video_codecs/h265_profile_tier_level.cc
index 3a25843..89eb553 100644
--- a/api/video_codecs/h265_profile_tier_level.cc
+++ b/api/video_codecs/h265_profile_tier_level.cc
@@ -10,9 +10,9 @@
#include "api/video_codecs/h265_profile_tier_level.h"
+#include <optional>
#include <string>
-#include "absl/types/optional.h"
#include "api/rtp_parameters.h"
#include "api/video/resolution.h"
#include "rtc_base/arraysize.h"
@@ -65,10 +65,10 @@
} // anonymous namespace
// Annex A of https://www.itu.int/rec/T-REC-H.265 (08/21), section A.3.
-absl::optional<H265Profile> StringToH265Profile(const std::string& profile) {
- absl::optional<int> i = rtc::StringToNumber<int>(profile);
+std::optional<H265Profile> StringToH265Profile(const std::string& profile) {
+ std::optional<int> i = rtc::StringToNumber<int>(profile);
if (!i.has_value()) {
- return absl::nullopt;
+ return std::nullopt;
}
switch (i.value()) {
@@ -95,16 +95,16 @@
case 11:
return H265Profile::kProfileHighThroughputScreenContentCoding;
default:
- return absl::nullopt;
+ return std::nullopt;
}
}
// Annex A of https://www.itu.int/rec/T-REC-H.265 (08/21), section A.4,
// tiers and levels.
-absl::optional<H265Tier> StringToH265Tier(const std::string& tier) {
- absl::optional<int> i = rtc::StringToNumber<int>(tier);
+std::optional<H265Tier> StringToH265Tier(const std::string& tier) {
+ std::optional<int> i = rtc::StringToNumber<int>(tier);
if (!i.has_value()) {
- return absl::nullopt;
+ return std::nullopt;
}
switch (i.value()) {
@@ -113,14 +113,14 @@
case 1:
return H265Tier::kTier1;
default:
- return absl::nullopt;
+ return std::nullopt;
}
}
-absl::optional<H265Level> StringToH265Level(const std::string& level) {
- const absl::optional<int> i = rtc::StringToNumber<int>(level);
+std::optional<H265Level> StringToH265Level(const std::string& level) {
+ const std::optional<int> i = rtc::StringToNumber<int>(level);
if (!i.has_value())
- return absl::nullopt;
+ return std::nullopt;
switch (i.value()) {
case 30:
@@ -150,7 +150,7 @@
case 186:
return H265Level::kLevel6_2;
default:
- return absl::nullopt;
+ return std::nullopt;
}
}
@@ -221,44 +221,44 @@
}
}
-absl::optional<H265ProfileTierLevel> ParseSdpForH265ProfileTierLevel(
+std::optional<H265ProfileTierLevel> ParseSdpForH265ProfileTierLevel(
const CodecParameterMap& params) {
static const H265ProfileTierLevel kDefaultProfileTierLevel(
H265Profile::kProfileMain, H265Tier::kTier0, H265Level::kLevel3_1);
bool profile_tier_level_specified = false;
- absl::optional<H265Profile> profile;
+ std::optional<H265Profile> profile;
const auto profile_it = params.find(kH265FmtpProfile);
if (profile_it != params.end()) {
profile_tier_level_specified = true;
const std::string& profile_str = profile_it->second;
profile = StringToH265Profile(profile_str);
if (!profile) {
- return absl::nullopt;
+ return std::nullopt;
}
} else {
profile = H265Profile::kProfileMain;
}
- absl::optional<H265Tier> tier;
+ std::optional<H265Tier> tier;
const auto tier_it = params.find(kH265FmtpTier);
if (tier_it != params.end()) {
profile_tier_level_specified = true;
const std::string& tier_str = tier_it->second;
tier = StringToH265Tier(tier_str);
if (!tier) {
- return absl::nullopt;
+ return std::nullopt;
}
} else {
tier = H265Tier::kTier0;
}
- absl::optional<H265Level> level;
+ std::optional<H265Level> level;
const auto level_it = params.find(kH265FmtpLevel);
if (level_it != params.end()) {
profile_tier_level_specified = true;
const std::string& level_str = level_it->second;
level = StringToH265Level(level_str);
if (!level) {
- return absl::nullopt;
+ return std::nullopt;
}
} else {
level = H265Level::kLevel3_1;
@@ -266,7 +266,7 @@
// Spec Table A.9, level 1 to level 3.1 does not allow high tiers.
if (level <= H265Level::kLevel3_1 && tier == H265Tier::kTier1) {
- return absl::nullopt;
+ return std::nullopt;
}
return !profile_tier_level_specified
@@ -277,16 +277,16 @@
bool H265IsSameProfileTierLevel(const CodecParameterMap& params1,
const CodecParameterMap& params2) {
- const absl::optional<H265ProfileTierLevel> ptl1 =
+ const std::optional<H265ProfileTierLevel> ptl1 =
ParseSdpForH265ProfileTierLevel(params1);
- const absl::optional<H265ProfileTierLevel> ptl2 =
+ const std::optional<H265ProfileTierLevel> ptl2 =
ParseSdpForH265ProfileTierLevel(params2);
return ptl1 && ptl2 && ptl1->profile == ptl2->profile &&
ptl1->tier == ptl2->tier && ptl1->level == ptl2->level;
}
-absl::optional<H265Level> GetSupportedH265Level(const Resolution& resolution,
- float max_fps) {
+std::optional<H265Level> GetSupportedH265Level(const Resolution& resolution,
+ float max_fps) {
int aligned_width =
(resolution.width + kMinCbSizeYMax - 1) & ~(kMinCbSizeYMax - 1);
int aligned_height =
@@ -303,7 +303,7 @@
return level_constraint.level;
}
}
- return absl::nullopt;
+ return std::nullopt;
}
} // namespace webrtc
diff --git a/api/video_codecs/h265_profile_tier_level.h b/api/video_codecs/h265_profile_tier_level.h
index 91e64b7..d954055 100644
--- a/api/video_codecs/h265_profile_tier_level.h
+++ b/api/video_codecs/h265_profile_tier_level.h
@@ -11,9 +11,9 @@
#ifndef API_VIDEO_CODECS_H265_PROFILE_TIER_LEVEL_H_
#define API_VIDEO_CODECS_H265_PROFILE_TIER_LEVEL_H_
+#include <optional>
#include <string>
-#include "absl/types/optional.h"
#include "api/rtp_parameters.h"
#include "api/video/resolution.h"
#include "rtc_base/system/rtc_export.h"
@@ -80,21 +80,20 @@
RTC_EXPORT std::string H265LevelToString(H265Level level);
// Helper function to get H265Profile from profile string.
-RTC_EXPORT absl::optional<H265Profile> StringToH265Profile(
+RTC_EXPORT std::optional<H265Profile> StringToH265Profile(
const std::string& profile);
// Helper function to get H265Tier from tier string.
-RTC_EXPORT absl::optional<H265Tier> StringToH265Tier(const std::string& tier);
+RTC_EXPORT std::optional<H265Tier> StringToH265Tier(const std::string& tier);
// Helper function to get H265Level from level string.
-RTC_EXPORT absl::optional<H265Level> StringToH265Level(
- const std::string& level);
+RTC_EXPORT std::optional<H265Level> StringToH265Level(const std::string& level);
// Given that a decoder supports up to a give frame size(in pixels) at up to a
// given number of frames per second, return the highest H.265 level where it
// can guranatee that it will be able to support all valid encoded streams that
// are within that level.
-RTC_EXPORT absl::optional<H265Level> GetSupportedH265Level(
+RTC_EXPORT std::optional<H265Level> GetSupportedH265Level(
const Resolution& resolution,
float max_fps);
@@ -105,7 +104,7 @@
// level defaults to "kLevel3_1" if no level-id is specified.
// Returns empty value if any of the profile/tier/level key is present but
// contains an invalid value.
-RTC_EXPORT absl::optional<H265ProfileTierLevel> ParseSdpForH265ProfileTierLevel(
+RTC_EXPORT std::optional<H265ProfileTierLevel> ParseSdpForH265ProfileTierLevel(
const CodecParameterMap& params);
// Returns true if the parameters have the same H265 profile or neither contains
diff --git a/api/video_codecs/libaom_av1_encoder_factory.cc b/api/video_codecs/libaom_av1_encoder_factory.cc
index 16555c5..968120e 100644
--- a/api/video_codecs/libaom_av1_encoder_factory.cc
+++ b/api/video_codecs/libaom_av1_encoder_factory.cc
@@ -16,13 +16,13 @@
#include <cstring>
#include <map>
#include <memory>
+#include <optional>
#include <string>
#include <type_traits>
#include <vector>
#include "absl/algorithm/container.h"
#include "absl/cleanup/cleanup.h"
-#include "absl/types/optional.h"
#include "absl/types/variant.h"
#include "api/array_view.h"
#include "api/scoped_refptr.h"
@@ -82,8 +82,8 @@
constexpr std::array<Rational, 7> kSupportedScalingFactors = {
{{8, 1}, {4, 1}, {2, 1}, {1, 1}, {1, 2}, {1, 4}, {1, 8}}};
-absl::optional<Rational> GetScalingFactor(const Resolution& from,
- const Resolution& to) {
+std::optional<Rational> GetScalingFactor(const Resolution& from,
+ const Resolution& to) {
auto it = absl::c_find_if(kSupportedScalingFactors, [&](const Rational& r) {
return (from.width * r.numerator / r.denominator) == to.width &&
(from.height * r.numerator / r.denominator) == to.height;
@@ -114,10 +114,10 @@
aom_codec_ctx_t ctx_;
aom_codec_enc_cfg_t cfg_;
- absl::optional<VideoCodecMode> current_content_type_;
- std::array<absl::optional<int>, kMaxSpatialLayersWtf> current_effort_level_;
+ std::optional<VideoCodecMode> current_content_type_;
+ std::array<std::optional<int>, kMaxSpatialLayersWtf> current_effort_level_;
int max_number_of_threads_;
- std::array<absl::optional<Resolution>, 8> last_resolution_in_buffer_;
+ std::array<std::optional<Resolution>, 8> last_resolution_in_buffer_;
};
template <typename T>
@@ -273,7 +273,7 @@
const VideoEncoderInterface::TemporalUnitSettings& tu_settings,
const std::vector<VideoEncoderInterface::FrameEncodeSettings>&
frame_settings,
- const std::array<absl::optional<Resolution>, 8>& last_resolution_in_buffer,
+ const std::array<std::optional<Resolution>, 8>& last_resolution_in_buffer,
aom_rc_mode rc_mode) {
if (frame_settings.empty()) {
RTC_LOG(LS_ERROR) << "No frame settings provided.";
@@ -343,7 +343,7 @@
// Figure out which frame resolution a certain buffer will hold when the
// frame described by `settings` is encoded.
- absl::optional<Resolution> referenced_resolution;
+ std::optional<Resolution> referenced_resolution;
bool keyframe_on_previous_layer = false;
// Will some other frame in this temporal unit update the buffer?
@@ -560,7 +560,7 @@
for (const VideoEncoderInterface::FrameEncodeSettings& settings :
frame_settings) {
- absl::optional<Rational> scaling_factor = GetScalingFactor(
+ std::optional<Rational> scaling_factor = GetScalingFactor(
{frame_buffer.width(), frame_buffer.height()}, settings.resolution);
RTC_CHECK(scaling_factor);
svc_params.scaling_factor_num[settings.spatial_id] =
diff --git a/api/video_codecs/libaom_av1_encoder_factory_test.cc b/api/video_codecs/libaom_av1_encoder_factory_test.cc
index ba4623c..0d87a8c 100644
--- a/api/video_codecs/libaom_av1_encoder_factory_test.cc
+++ b/api/video_codecs/libaom_av1_encoder_factory_test.cc
@@ -13,12 +13,12 @@
#include <cstdint>
#include <cstdio>
#include <memory>
+#include <optional>
#include <ostream>
#include <string>
#include <utility>
#include <vector>
-#include "absl/types/optional.h"
#include "absl/types/variant.h"
#include "api/array_view.h"
#include "api/scoped_refptr.h"
@@ -741,7 +741,7 @@
.performance.min_max_effort_level;
// Cbr rc{.duration = TimeDelta::Millis(100),
// .target_bitrate = DataRate::KilobitsPerSec(100)};
- absl::optional<double> psnr_last;
+ std::optional<double> psnr_last;
Av1Decoder dec;
for (int i = effort_range.first; i <= effort_range.second; ++i) {
diff --git a/api/video_codecs/scalability_mode_helper.cc b/api/video_codecs/scalability_mode_helper.cc
index 238bc22..83b7bd4 100644
--- a/api/video_codecs/scalability_mode_helper.cc
+++ b/api/video_codecs/scalability_mode_helper.cc
@@ -10,34 +10,35 @@
#include "api/video_codecs/scalability_mode_helper.h"
+#include <optional>
+
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/video_codecs/scalability_mode.h"
#include "modules/video_coding/svc/scalability_mode_util.h"
namespace webrtc {
-absl::optional<int> ScalabilityModeStringToNumSpatialLayers(
+std::optional<int> ScalabilityModeStringToNumSpatialLayers(
absl::string_view scalability_mode_string) {
- absl::optional<ScalabilityMode> scalability_mode =
+ std::optional<ScalabilityMode> scalability_mode =
ScalabilityModeFromString(scalability_mode_string);
if (!scalability_mode.has_value()) {
- return absl::nullopt;
+ return std::nullopt;
}
return ScalabilityModeToNumSpatialLayers(*scalability_mode);
}
-absl::optional<int> ScalabilityModeStringToNumTemporalLayers(
+std::optional<int> ScalabilityModeStringToNumTemporalLayers(
absl::string_view scalability_mode_string) {
- absl::optional<ScalabilityMode> scalability_mode =
+ std::optional<ScalabilityMode> scalability_mode =
ScalabilityModeFromString(scalability_mode_string);
if (!scalability_mode.has_value()) {
- return absl::nullopt;
+ return std::nullopt;
}
return ScalabilityModeToNumTemporalLayers(*scalability_mode);
}
-absl::optional<ScalabilityMode> ScalabilityModeStringToEnum(
+std::optional<ScalabilityMode> ScalabilityModeStringToEnum(
absl::string_view scalability_mode_string) {
return ScalabilityModeFromString(scalability_mode_string);
}
diff --git a/api/video_codecs/scalability_mode_helper.h b/api/video_codecs/scalability_mode_helper.h
index 21dcfc2..d48d45b 100644
--- a/api/video_codecs/scalability_mode_helper.h
+++ b/api/video_codecs/scalability_mode_helper.h
@@ -11,25 +11,26 @@
#ifndef API_VIDEO_CODECS_SCALABILITY_MODE_HELPER_H_
#define API_VIDEO_CODECS_SCALABILITY_MODE_HELPER_H_
+#include <optional>
+
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/video_codecs/scalability_mode.h"
namespace webrtc {
// Returns the number of spatial layers from the `scalability_mode_string`
// or nullopt if the given mode is unknown.
-absl::optional<int> ScalabilityModeStringToNumSpatialLayers(
+std::optional<int> ScalabilityModeStringToNumSpatialLayers(
absl::string_view scalability_mode_string);
// Returns the number of temporal layers from the `scalability_mode_string`
// or nullopt if the given mode is unknown.
-absl::optional<int> ScalabilityModeStringToNumTemporalLayers(
+std::optional<int> ScalabilityModeStringToNumTemporalLayers(
absl::string_view scalability_mode_string);
// Convert the `scalability_mode_string` to the scalability mode enum value
// or nullopt if the given mode is unknown.
-absl::optional<ScalabilityMode> ScalabilityModeStringToEnum(
+std::optional<ScalabilityMode> ScalabilityModeStringToEnum(
absl::string_view scalability_mode_string);
} // namespace webrtc
diff --git a/api/video_codecs/sdp_video_format.cc b/api/video_codecs/sdp_video_format.cc
index c0838bd..f287d63 100644
--- a/api/video_codecs/sdp_video_format.cc
+++ b/api/video_codecs/sdp_video_format.cc
@@ -10,11 +10,11 @@
#include "api/video_codecs/sdp_video_format.h"
+#include <optional>
#include <string>
#include "absl/container/inlined_vector.h"
#include "absl/strings/match.h"
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/rtp_parameters.h"
#include "api/video/video_codec_type.h"
@@ -259,10 +259,10 @@
{cricket::kAv1FmtpTier, "0"}});
}
-absl::optional<SdpVideoFormat> FuzzyMatchSdpVideoFormat(
+std::optional<SdpVideoFormat> FuzzyMatchSdpVideoFormat(
rtc::ArrayView<const SdpVideoFormat> supported_formats,
const SdpVideoFormat& format) {
- absl::optional<SdpVideoFormat> res;
+ std::optional<SdpVideoFormat> res;
int best_parameter_match = 0;
for (const auto& supported_format : supported_formats) {
if (absl::EqualsIgnoreCase(supported_format.name, format.name)) {
diff --git a/api/video_codecs/sdp_video_format.h b/api/video_codecs/sdp_video_format.h
index e7da486..7c8de66 100644
--- a/api/video_codecs/sdp_video_format.h
+++ b/api/video_codecs/sdp_video_format.h
@@ -12,10 +12,10 @@
#define API_VIDEO_CODECS_SDP_VIDEO_FORMAT_H_
#include <map>
+#include <optional>
#include <string>
#include "absl/container/inlined_vector.h"
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/rtp_parameters.h"
#include "api/video_codecs/scalability_mode.h"
@@ -86,7 +86,7 @@
// SdpVideoFormat, which makes instances that should compare equal to not match
// anymore. Until we stop misusing SdpVideoFormats provide this convenience
// function to perform fuzzy matching.
-absl::optional<SdpVideoFormat> FuzzyMatchSdpVideoFormat(
+std::optional<SdpVideoFormat> FuzzyMatchSdpVideoFormat(
rtc::ArrayView<const SdpVideoFormat> supported_formats,
const SdpVideoFormat& format);
diff --git a/api/video_codecs/simple_encoder_wrapper.cc b/api/video_codecs/simple_encoder_wrapper.cc
index 06ca5af..ae5282d 100644
--- a/api/video_codecs/simple_encoder_wrapper.cc
+++ b/api/video_codecs/simple_encoder_wrapper.cc
@@ -14,13 +14,13 @@
#include <cstddef>
#include <cstdint>
#include <memory>
+#include <optional>
#include <string>
#include <utility>
#include <vector>
#include "absl/algorithm/container.h"
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "absl/types/variant.h"
#include "api/array_view.h"
#include "api/scoped_refptr.h"
@@ -121,7 +121,7 @@
return nullptr;
}
- absl::optional<ScalabilityMode> sm =
+ std::optional<ScalabilityMode> sm =
ScalabilityModeStringToEnum(scalability_mode);
if (!sm) {
return nullptr;
diff --git a/api/video_codecs/simple_encoder_wrapper.h b/api/video_codecs/simple_encoder_wrapper.h
index fd7ca7f..7baefb9 100644
--- a/api/video_codecs/simple_encoder_wrapper.h
+++ b/api/video_codecs/simple_encoder_wrapper.h
@@ -14,11 +14,11 @@
#include <cstdint>
#include <functional>
#include <memory>
+#include <optional>
#include <string>
#include <vector>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/scoped_refptr.h"
#include "api/transport/rtp/dependency_descriptor.h"
#include "api/units/timestamp.h"
@@ -36,7 +36,7 @@
std::vector<uint8_t> bitstream_data;
FrameType frame_type;
GenericFrameInfo generic_frame_info;
- absl::optional<FrameDependencyStructure> dependency_structure;
+ std::optional<FrameDependencyStructure> dependency_structure;
};
using EncodeResultCallback = std::function<void(const EncodeResult& result)>;
diff --git a/api/video_codecs/simple_encoder_wrapper_unittests.cc b/api/video_codecs/simple_encoder_wrapper_unittests.cc
index 14a48f7..c5d9bae 100644
--- a/api/video_codecs/simple_encoder_wrapper_unittests.cc
+++ b/api/video_codecs/simple_encoder_wrapper_unittests.cc
@@ -9,9 +9,9 @@
*/
#include <memory>
+#include <optional>
#include <utility>
-#include "absl/types/optional.h"
#include "api/video/i420_buffer.h" // IWYU pragma: keep
#include "api/video_codecs/libaom_av1_encoder_factory.h"
#include "api/video_codecs/simple_encoder_wrapper.h"
@@ -155,7 +155,7 @@
[&](const SimpleEncoderWrapper::EncodeResult& result) {
++num_callbacks;
ASSERT_THAT(result.oh_no, Eq(false));
- EXPECT_THAT(result.dependency_structure, Ne(absl::nullopt));
+ EXPECT_THAT(result.dependency_structure, Ne(std::nullopt));
EXPECT_THAT(result.bitstream_data, Not(IsEmpty()));
EXPECT_THAT(result.frame_type, Eq(FrameType::kKeyframe));
EXPECT_THAT(result.generic_frame_info.spatial_id, Eq(0));
@@ -167,7 +167,7 @@
[&](const SimpleEncoderWrapper::EncodeResult& result) {
++num_callbacks;
ASSERT_THAT(result.oh_no, Eq(false));
- EXPECT_THAT(result.dependency_structure, Eq(absl::nullopt));
+ EXPECT_THAT(result.dependency_structure, Eq(std::nullopt));
EXPECT_THAT(result.bitstream_data, Not(IsEmpty()));
EXPECT_THAT(result.frame_type, Eq(FrameType::kDeltaFrame));
EXPECT_THAT(result.generic_frame_info.spatial_id, Eq(0));
@@ -200,13 +200,13 @@
ASSERT_THAT(result.oh_no, Eq(false));
if (result.generic_frame_info.spatial_id == 0) {
++num_callbacks;
- EXPECT_THAT(result.dependency_structure, Ne(absl::nullopt));
+ EXPECT_THAT(result.dependency_structure, Ne(std::nullopt));
EXPECT_THAT(result.bitstream_data, Not(IsEmpty()));
EXPECT_THAT(result.frame_type, Eq(FrameType::kKeyframe));
EXPECT_THAT(result.generic_frame_info.temporal_id, Eq(0));
} else if (result.generic_frame_info.spatial_id == 1) {
++num_callbacks;
- EXPECT_THAT(result.dependency_structure, Eq(absl::nullopt));
+ EXPECT_THAT(result.dependency_structure, Eq(std::nullopt));
EXPECT_THAT(result.bitstream_data, Not(IsEmpty()));
EXPECT_THAT(result.frame_type, Eq(FrameType::kDeltaFrame));
EXPECT_THAT(result.generic_frame_info.temporal_id, Eq(0));
@@ -219,13 +219,13 @@
ASSERT_THAT(result.oh_no, Eq(false));
if (result.generic_frame_info.spatial_id == 0) {
++num_callbacks;
- EXPECT_THAT(result.dependency_structure, Eq(absl::nullopt));
+ EXPECT_THAT(result.dependency_structure, Eq(std::nullopt));
EXPECT_THAT(result.bitstream_data, Not(IsEmpty()));
EXPECT_THAT(result.frame_type, Eq(FrameType::kDeltaFrame));
EXPECT_THAT(result.generic_frame_info.temporal_id, Eq(1));
} else if (result.generic_frame_info.spatial_id == 1) {
++num_callbacks;
- EXPECT_THAT(result.dependency_structure, Eq(absl::nullopt));
+ EXPECT_THAT(result.dependency_structure, Eq(std::nullopt));
EXPECT_THAT(result.bitstream_data, Not(IsEmpty()));
EXPECT_THAT(result.frame_type, Eq(FrameType::kDeltaFrame));
EXPECT_THAT(result.generic_frame_info.temporal_id, Eq(1));
diff --git a/api/video_codecs/simulcast_stream.cc b/api/video_codecs/simulcast_stream.cc
index 9ecf746..cda5826 100644
--- a/api/video_codecs/simulcast_stream.cc
+++ b/api/video_codecs/simulcast_stream.cc
@@ -10,7 +10,8 @@
#include "api/video_codecs/simulcast_stream.h"
-#include "absl/types/optional.h"
+#include <optional>
+
#include "api/video_codecs/scalability_mode.h"
#include "rtc_base/checks.h"
@@ -25,14 +26,14 @@
numberOfTemporalLayers = n;
}
-absl::optional<ScalabilityMode> SimulcastStream::GetScalabilityMode() const {
+std::optional<ScalabilityMode> SimulcastStream::GetScalabilityMode() const {
static const ScalabilityMode scalability_modes[3] = {
ScalabilityMode::kL1T1,
ScalabilityMode::kL1T2,
ScalabilityMode::kL1T3,
};
if (numberOfTemporalLayers < 1 || numberOfTemporalLayers > 3) {
- return absl::nullopt;
+ return std::nullopt;
}
return scalability_modes[numberOfTemporalLayers - 1];
}
diff --git a/api/video_codecs/simulcast_stream.h b/api/video_codecs/simulcast_stream.h
index 50ec21e..4dbee5b 100644
--- a/api/video_codecs/simulcast_stream.h
+++ b/api/video_codecs/simulcast_stream.h
@@ -11,7 +11,8 @@
#ifndef API_VIDEO_CODECS_SIMULCAST_STREAM_H_
#define API_VIDEO_CODECS_SIMULCAST_STREAM_H_
-#include "absl/types/optional.h"
+#include <optional>
+
#include "api/video_codecs/scalability_mode.h"
#include "rtc_base/system/rtc_export.h"
@@ -23,7 +24,7 @@
// Temporary utility methods for transition from numberOfTemporalLayers
// setting to ScalabilityMode.
unsigned char GetNumberOfTemporalLayers() const;
- absl::optional<ScalabilityMode> GetScalabilityMode() const;
+ std::optional<ScalabilityMode> GetScalabilityMode() const;
void SetNumberOfTemporalLayers(unsigned char n);
bool operator==(const SimulcastStream& other) const;
diff --git a/api/video_codecs/test/BUILD.gn b/api/video_codecs/test/BUILD.gn
index 95c0975..6b8fe75 100644
--- a/api/video_codecs/test/BUILD.gn
+++ b/api/video_codecs/test/BUILD.gn
@@ -55,7 +55,6 @@
"../../video:video_frame_type",
"../../video:video_rtp_headers",
"//testing/gtest",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -76,7 +75,6 @@
"../../environment",
"../../environment:environment_factory",
"//testing/gtest",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
diff --git a/api/video_codecs/test/h264_profile_level_id_unittest.cc b/api/video_codecs/test/h264_profile_level_id_unittest.cc
index 374c3d5..044765b 100644
--- a/api/video_codecs/test/h264_profile_level_id_unittest.cc
+++ b/api/video_codecs/test/h264_profile_level_id_unittest.cc
@@ -11,8 +11,8 @@
#include "api/video_codecs/h264_profile_level_id.h"
#include <map>
+#include <optional>
-#include "absl/types/optional.h"
#include "api/rtp_parameters.h"
#include "test/gtest.h"
@@ -144,7 +144,7 @@
}
TEST(H264ProfileLevelId, TestParseSdpProfileLevelIdEmpty) {
- const absl::optional<H264ProfileLevelId> profile_level_id =
+ const std::optional<H264ProfileLevelId> profile_level_id =
ParseSdpForH264ProfileLevelId(CodecParameterMap());
EXPECT_TRUE(profile_level_id);
EXPECT_EQ(H264Profile::kProfileConstrainedBaseline,
@@ -155,7 +155,7 @@
TEST(H264ProfileLevelId, TestParseSdpProfileLevelIdConstrainedHigh) {
CodecParameterMap params;
params["profile-level-id"] = "640c2a";
- const absl::optional<H264ProfileLevelId> profile_level_id =
+ const std::optional<H264ProfileLevelId> profile_level_id =
ParseSdpForH264ProfileLevelId(params);
EXPECT_TRUE(profile_level_id);
EXPECT_EQ(H264Profile::kProfileConstrainedHigh, profile_level_id->profile);
diff --git a/api/video_codecs/test/h265_profile_tier_level_unittest.cc b/api/video_codecs/test/h265_profile_tier_level_unittest.cc
index 8fcabe8..e9095e2 100644
--- a/api/video_codecs/test/h265_profile_tier_level_unittest.cc
+++ b/api/video_codecs/test/h265_profile_tier_level_unittest.cc
@@ -10,9 +10,9 @@
#include "api/video_codecs/h265_profile_tier_level.h"
+#include <optional>
#include <string>
-#include "absl/types/optional.h"
#include "api/rtp_parameters.h"
#include "api/video/resolution.h"
#include "test/gtest.h"
@@ -109,7 +109,7 @@
}
TEST(H265ProfileTierLevel, TestParseSdpProfileTierLevelAllEmpty) {
- const absl::optional<H265ProfileTierLevel> profile_tier_level =
+ const std::optional<H265ProfileTierLevel> profile_tier_level =
ParseSdpForH265ProfileTierLevel(CodecParameterMap());
EXPECT_TRUE(profile_tier_level);
EXPECT_EQ(H265Profile::kProfileMain, profile_tier_level->profile);
@@ -121,7 +121,7 @@
CodecParameterMap params;
params["profile-id"] = "1";
params["tier-flag"] = "0";
- absl::optional<H265ProfileTierLevel> profile_tier_level =
+ std::optional<H265ProfileTierLevel> profile_tier_level =
ParseSdpForH265ProfileTierLevel(params);
EXPECT_TRUE(profile_tier_level);
EXPECT_EQ(H265Profile::kProfileMain, profile_tier_level->profile);
@@ -152,7 +152,7 @@
params["profile-id"] = "1";
params["tier-flag"] = "1";
params["level-id"] = "93";
- absl::optional<H265ProfileTierLevel> profile_tier_level =
+ std::optional<H265ProfileTierLevel> profile_tier_level =
ParseSdpForH265ProfileTierLevel(params);
EXPECT_FALSE(profile_tier_level);
params.clear();
@@ -176,7 +176,7 @@
params["profile-id"] = "1";
params["tier-flag"] = "0";
params["level-id"] = "93";
- absl::optional<H265ProfileTierLevel> profile_tier_level =
+ std::optional<H265ProfileTierLevel> profile_tier_level =
ParseSdpForH265ProfileTierLevel(params);
EXPECT_TRUE(profile_tier_level);
EXPECT_EQ("1", H265ProfileToString(profile_tier_level->profile));
@@ -288,12 +288,12 @@
// Test with 64x64 at 30fps
r.width = 64;
r.height = 64;
- EXPECT_EQ(GetSupportedH265Level(r, 30), absl::nullopt);
+ EXPECT_EQ(GetSupportedH265Level(r, 30), std::nullopt);
// Test with extremly large width or height at 15fps
r.width = 16928;
r.height = 64;
- EXPECT_EQ(GetSupportedH265Level(r, 15), absl::nullopt);
+ EXPECT_EQ(GetSupportedH265Level(r, 15), std::nullopt);
}
} // namespace webrtc
diff --git a/api/video_codecs/test/video_decoder_software_fallback_wrapper_unittest.cc b/api/video_codecs/test/video_decoder_software_fallback_wrapper_unittest.cc
index 331f633..8736484 100644
--- a/api/video_codecs/test/video_decoder_software_fallback_wrapper_unittest.cc
+++ b/api/video_codecs/test/video_decoder_software_fallback_wrapper_unittest.cc
@@ -13,9 +13,9 @@
#include <stdint.h>
#include <memory>
+#include <optional>
#include <string>
-#include "absl/types/optional.h"
#include "api/environment/environment.h"
#include "api/environment/environment_factory.h"
#include "api/video/encoded_image.h"
@@ -190,8 +190,8 @@
return -1;
}
void Decoded(webrtc::VideoFrame& decodedImage,
- absl::optional<int32_t> decode_time_ms,
- absl::optional<uint8_t> qp) override {
+ std::optional<int32_t> decode_time_ms,
+ std::optional<uint8_t> qp) override {
RTC_DCHECK_NOTREACHED();
}
} callback;
diff --git a/api/video_codecs/test/video_encoder_factory_template_tests.cc b/api/video_codecs/test/video_encoder_factory_template_tests.cc
index 8b1bf62..cf254ce 100644
--- a/api/video_codecs/test/video_encoder_factory_template_tests.cc
+++ b/api/video_codecs/test/video_encoder_factory_template_tests.cc
@@ -9,9 +9,9 @@
*/
#include <memory>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/environment/environment.h"
#include "api/environment/environment_factory.h"
#include "api/test/mock_video_encoder.h"
@@ -90,13 +90,13 @@
TEST(VideoEncoderFactoryTemplate, OneTemplateAdapterCodecSupport) {
VideoEncoderFactoryTemplate<FooEncoderTemplateAdapter> factory;
- EXPECT_THAT(factory.QueryCodecSupport(kFooSdp, absl::nullopt),
+ EXPECT_THAT(factory.QueryCodecSupport(kFooSdp, std::nullopt),
Field(&CodecSupport::is_supported, true));
EXPECT_THAT(factory.QueryCodecSupport(kFooSdp, "L1T2"),
Field(&CodecSupport::is_supported, true));
EXPECT_THAT(factory.QueryCodecSupport(kFooSdp, "S3T3"),
Field(&CodecSupport::is_supported, false));
- EXPECT_THAT(factory.QueryCodecSupport(SdpVideoFormat("FooX"), absl::nullopt),
+ EXPECT_THAT(factory.QueryCodecSupport(SdpVideoFormat("FooX"), std::nullopt),
Field(&CodecSupport::is_supported, false));
}
@@ -125,15 +125,15 @@
VideoEncoderFactoryTemplate<FooEncoderTemplateAdapter,
BarEncoderTemplateAdapter>
factory;
- EXPECT_THAT(factory.QueryCodecSupport(kFooSdp, absl::nullopt),
+ EXPECT_THAT(factory.QueryCodecSupport(kFooSdp, std::nullopt),
Field(&CodecSupport::is_supported, true));
EXPECT_THAT(factory.QueryCodecSupport(kFooSdp, "L1T2"),
Field(&CodecSupport::is_supported, true));
EXPECT_THAT(factory.QueryCodecSupport(kFooSdp, "S3T3"),
Field(&CodecSupport::is_supported, false));
- EXPECT_THAT(factory.QueryCodecSupport(kBarLowSdp, absl::nullopt),
+ EXPECT_THAT(factory.QueryCodecSupport(kBarLowSdp, std::nullopt),
Field(&CodecSupport::is_supported, true));
- EXPECT_THAT(factory.QueryCodecSupport(kBarHighSdp, absl::nullopt),
+ EXPECT_THAT(factory.QueryCodecSupport(kBarHighSdp, std::nullopt),
Field(&CodecSupport::is_supported, true));
EXPECT_THAT(factory.QueryCodecSupport(kBarLowSdp, "S2T1"),
Field(&CodecSupport::is_supported, true));
diff --git a/api/video_codecs/test/video_encoder_software_fallback_wrapper_unittest.cc b/api/video_codecs/test/video_encoder_software_fallback_wrapper_unittest.cc
index a63e8ba..11f234d 100644
--- a/api/video_codecs/test/video_encoder_software_fallback_wrapper_unittest.cc
+++ b/api/video_codecs/test/video_encoder_software_fallback_wrapper_unittest.cc
@@ -14,11 +14,11 @@
#include <stdint.h>
#include <memory>
+#include <optional>
#include <string>
#include <utility>
#include <vector>
-#include "absl/types/optional.h"
#include "api/environment/environment.h"
#include "api/environment/environment_factory.h"
#include "api/fec_controller_override.h"
@@ -164,7 +164,7 @@
bool supports_native_handle_ = false;
bool is_qp_trusted_ = false;
std::string implementation_name_ = "fake-encoder";
- absl::optional<VideoFrame> last_video_frame_;
+ std::optional<VideoFrame> last_video_frame_;
};
void InitEncode();
diff --git a/api/video_codecs/video_codec.cc b/api/video_codecs/video_codec.cc
index 0cd6b37..8ee5b1f 100644
--- a/api/video_codecs/video_codec.cc
+++ b/api/video_codecs/video_codec.cc
@@ -12,10 +12,10 @@
#include <string.h>
+#include <optional>
#include <string>
#include "absl/strings/match.h"
-#include "absl/types/optional.h"
#include "api/video/video_codec_type.h"
#include "api/video_codecs/scalability_mode.h"
#include "api/video_codecs/simulcast_stream.h"
@@ -86,7 +86,7 @@
<< (mode == VideoCodecMode::kRealtimeVideo ? "RealtimeVideo"
: "Screensharing");
if (IsSinglecast()) {
- absl::optional<ScalabilityMode> scalability_mode = GetScalabilityMode();
+ std::optional<ScalabilityMode> scalability_mode = GetScalabilityMode();
if (scalability_mode.has_value()) {
ss << ", Singlecast: {" << width << "x" << height << " "
<< ScalabilityModeToString(*scalability_mode)
@@ -96,7 +96,7 @@
ss << ", Simulcast: {";
for (size_t i = 0; i < numberOfSimulcastStreams; ++i) {
const SimulcastStream stream = simulcastStream[i];
- absl::optional<ScalabilityMode> scalability_mode =
+ std::optional<ScalabilityMode> scalability_mode =
stream.GetScalabilityMode();
if (scalability_mode.has_value()) {
ss << "[" << stream.width << "x" << stream.height << " "
diff --git a/api/video_codecs/video_codec.h b/api/video_codecs/video_codec.h
index 8872eae..7824d7f 100644
--- a/api/video_codecs/video_codec.h
+++ b/api/video_codecs/video_codec.h
@@ -14,9 +14,9 @@
#include <stddef.h>
#include <stdint.h>
+#include <optional>
#include <string>
-#include "absl/types/optional.h"
#include "api/video/video_codec_constants.h"
#include "api/video/video_codec_type.h"
#include "api/video_codecs/scalability_mode.h"
@@ -127,13 +127,13 @@
// Scalability mode as described in
// https://www.w3.org/TR/webrtc-svc/#scalabilitymodes*
- absl::optional<ScalabilityMode> GetScalabilityMode() const {
+ std::optional<ScalabilityMode> GetScalabilityMode() const {
return scalability_mode_;
}
void SetScalabilityMode(ScalabilityMode scalability_mode) {
scalability_mode_ = scalability_mode;
}
- void UnsetScalabilityMode() { scalability_mode_ = absl::nullopt; }
+ void UnsetScalabilityMode() { scalability_mode_ = std::nullopt; }
VideoCodecComplexity GetVideoEncoderComplexity() const;
void SetVideoEncoderComplexity(VideoCodecComplexity complexity_setting);
@@ -215,7 +215,7 @@
// TODO(hta): Consider replacing the union with a pointer type.
// This will allow removing the VideoCodec* types from this file.
VideoCodecUnion codec_specific_;
- absl::optional<ScalabilityMode> scalability_mode_;
+ std::optional<ScalabilityMode> scalability_mode_;
// 'complexity_' indicates the CPU capability of the client. It's used to
// determine encoder CPU complexity (e.g., cpu_used for VP8, VP9. and AV1).
VideoCodecComplexity complexity_;
diff --git a/api/video_codecs/video_decoder.cc b/api/video_codecs/video_decoder.cc
index b6a7493..d60636c 100644
--- a/api/video_codecs/video_decoder.cc
+++ b/api/video_codecs/video_decoder.cc
@@ -11,9 +11,9 @@
#include "api/video_codecs/video_decoder.h"
#include <cstdint>
+#include <optional>
#include <string>
-#include "absl/types/optional.h"
#include "api/video/video_frame.h"
#include "rtc_base/checks.h"
#include "rtc_base/strings/string_builder.h"
@@ -27,8 +27,8 @@
}
void DecodedImageCallback::Decoded(VideoFrame& decodedImage,
- absl::optional<int32_t> decode_time_ms,
- absl::optional<uint8_t> qp) {
+ std::optional<int32_t> decode_time_ms,
+ std::optional<uint8_t> qp) {
Decoded(decodedImage, decode_time_ms.value_or(-1));
}
diff --git a/api/video_codecs/video_decoder.h b/api/video_codecs/video_decoder.h
index a951f7f..934e224 100644
--- a/api/video_codecs/video_decoder.h
+++ b/api/video_codecs/video_decoder.h
@@ -12,9 +12,9 @@
#define API_VIDEO_CODECS_VIDEO_DECODER_H_
#include <cstdint>
+#include <optional>
#include <string>
-#include "absl/types/optional.h"
#include "api/video/encoded_image.h"
#include "api/video/render_resolution.h"
#include "api/video/video_codec_type.h"
@@ -37,8 +37,8 @@
// TODO(sakal): Remove other implementations when upstream projects have been
// updated.
virtual void Decoded(VideoFrame& decodedImage,
- absl::optional<int32_t> decode_time_ms,
- absl::optional<uint8_t> qp);
+ std::optional<int32_t> decode_time_ms,
+ std::optional<uint8_t> qp);
};
class RTC_EXPORT VideoDecoder {
@@ -66,8 +66,8 @@
// decoder. If value isn't present some codec-default value will be used. If
// value is present and decoder doesn't have buffer pool the value will be
// ignored.
- absl::optional<int> buffer_pool_size() const;
- void set_buffer_pool_size(absl::optional<int> value);
+ std::optional<int> buffer_pool_size() const;
+ void set_buffer_pool_size(std::optional<int> value);
// When valid, user of the VideoDecoder interface shouldn't `Decode`
// encoded images with render resolution larger than width and height
@@ -85,7 +85,7 @@
void set_codec_type(VideoCodecType value) { codec_type_ = value; }
private:
- absl::optional<int> buffer_pool_size_;
+ std::optional<int> buffer_pool_size_;
RenderResolution max_resolution_;
int number_of_cores_ = 1;
VideoCodecType codec_type_ = kVideoCodecGeneric;
@@ -123,12 +123,12 @@
virtual const char* ImplementationName() const;
};
-inline absl::optional<int> VideoDecoder::Settings::buffer_pool_size() const {
+inline std::optional<int> VideoDecoder::Settings::buffer_pool_size() const {
return buffer_pool_size_;
}
inline void VideoDecoder::Settings::set_buffer_pool_size(
- absl::optional<int> value) {
+ std::optional<int> value) {
buffer_pool_size_ = value;
}
diff --git a/api/video_codecs/video_encoder.cc b/api/video_codecs/video_encoder.cc
index 11d6821..c3b5f3a 100644
--- a/api/video_codecs/video_encoder.cc
+++ b/api/video_codecs/video_encoder.cc
@@ -14,12 +14,12 @@
#include <algorithm>
#include <cstdint>
+#include <optional>
#include <string>
#include <tuple>
#include <vector>
#include "absl/container/inlined_vector.h"
-#include "absl/types/optional.h"
#include "api/fec_controller_override.h"
#include "api/units/data_rate.h"
#include "api/video/video_bitrate_allocation.h"
@@ -240,7 +240,7 @@
return true;
}
-absl::optional<VideoEncoder::ResolutionBitrateLimits>
+std::optional<VideoEncoder::ResolutionBitrateLimits>
VideoEncoder::EncoderInfo::GetEncoderBitrateLimitsForResolution(
int frame_size_pixels) const {
std::vector<ResolutionBitrateLimits> bitrate_limits =
@@ -269,11 +269,11 @@
}
if (bitrate_limits[i].frame_size_pixels >= frame_size_pixels) {
- return absl::optional<ResolutionBitrateLimits>(bitrate_limits[i]);
+ return std::optional<ResolutionBitrateLimits>(bitrate_limits[i]);
}
}
- return absl::nullopt;
+ return std::nullopt;
}
VideoEncoder::RateControlParameters::RateControlParameters()
diff --git a/api/video_codecs/video_encoder.h b/api/video_codecs/video_encoder.h
index 77564ea..aadf452 100644
--- a/api/video_codecs/video_encoder.h
+++ b/api/video_codecs/video_encoder.h
@@ -14,11 +14,11 @@
#include <cstddef>
#include <cstdint>
#include <limits>
+#include <optional>
#include <string>
#include <vector>
#include "absl/container/inlined_vector.h"
-#include "absl/types/optional.h"
#include "api/fec_controller_override.h"
#include "api/units/data_rate.h"
#include "api/video/encoded_image.h"
@@ -102,7 +102,7 @@
struct KOff {};
public:
- // TODO(bugs.webrtc.org/9078): Since absl::optional should be trivially copy
+ // TODO(bugs.webrtc.org/9078): Since std::optional should be trivially copy
// constructible, this magic value can likely be replaced by a constexpr
// ScalingSettings value.
static constexpr KOff kOff = {};
@@ -113,7 +113,7 @@
ScalingSettings(KOff); // NOLINT(runtime/explicit)
~ScalingSettings();
- absl::optional<QpThresholds> thresholds;
+ std::optional<QpThresholds> thresholds;
// We will never ask for a resolution lower than this.
// TODO(kthelgason): Lower this limit when better testing
@@ -242,8 +242,8 @@
// Obtains the limits from `resolution_bitrate_limits` that best matches the
// `frame_size_pixels`.
- absl::optional<ResolutionBitrateLimits>
- GetEncoderBitrateLimitsForResolution(int frame_size_pixels) const;
+ std::optional<ResolutionBitrateLimits> GetEncoderBitrateLimitsForResolution(
+ int frame_size_pixels) const;
// If true, this encoder has internal support for generating simulcast
// streams. Otherwise, an adapter class will be needed.
@@ -260,12 +260,12 @@
// Indicates whether or not QP value encoder writes into frame/slice/tile
// header can be interpreted as average frame/slice/tile QP.
- absl::optional<bool> is_qp_trusted;
+ std::optional<bool> is_qp_trusted;
// The minimum QP that the encoder is expected to use with the current
// configuration. This may be used to determine if the encoder has reached
// its target video quality for static screenshare content.
- absl::optional<int> min_qp;
+ std::optional<int> min_qp;
};
struct RTC_EXPORT RateControlParameters {
@@ -307,7 +307,7 @@
// all decodable.
// `false` if some dependencies were undecodable, `true` if all dependencies
// were decodable, and `nullopt` if the dependencies are unknown.
- absl::optional<bool> dependencies_of_last_received_decodable;
+ std::optional<bool> dependencies_of_last_received_decodable;
// Describes whether the received frame was decodable.
// `false` if some dependency was undecodable or if some packet belonging
// to the last received frame was missed.
@@ -315,7 +315,7 @@
// to the last received frame were received.
// `nullopt` if no packet belonging to the last frame was missed, but the
// last packet in the frame was not yet received.
- absl::optional<bool> last_received_decodable;
+ std::optional<bool> last_received_decodable;
};
// Negotiated capabilities which the VideoEncoder may expect the other
@@ -339,7 +339,7 @@
size_t max_payload_size;
// Experimental API - currently only supported by LibvpxVp8Encoder and
// the OpenH264 encoder. If set, limits the number of encoder threads.
- absl::optional<int> encoder_thread_limit;
+ std::optional<int> encoder_thread_limit;
};
static VideoCodecVP8 GetDefaultVp8Settings();
diff --git a/api/video_codecs/video_encoder_factory.h b/api/video_codecs/video_encoder_factory.h
index d6e7ff9..cb00f92 100644
--- a/api/video_codecs/video_encoder_factory.h
+++ b/api/video_codecs/video_encoder_factory.h
@@ -12,10 +12,10 @@
#define API_VIDEO_CODECS_VIDEO_ENCODER_FACTORY_H_
#include <memory>
+#include <optional>
#include <string>
#include <vector>
-#include "absl/types/optional.h"
#include "api/environment/environment.h"
#include "api/units/data_rate.h"
#include "api/video/render_resolution.h"
@@ -48,19 +48,19 @@
// Called every time the available bitrate is updated. Should return a
// non-empty if an encoder switch should be performed.
- virtual absl::optional<SdpVideoFormat> OnAvailableBitrate(
+ virtual std::optional<SdpVideoFormat> OnAvailableBitrate(
const DataRate& rate) = 0;
// Called every time the encoder input resolution change. Should return a
// non-empty if an encoder switch should be performed.
- virtual absl::optional<SdpVideoFormat> OnResolutionChange(
+ virtual std::optional<SdpVideoFormat> OnResolutionChange(
const RenderResolution& resolution) {
- return absl::nullopt;
+ return std::nullopt;
}
// Called if the currently used encoder reports itself as broken. Should
// return a non-empty if an encoder switch should be performed.
- virtual absl::optional<SdpVideoFormat> OnEncoderBroken() = 0;
+ virtual std::optional<SdpVideoFormat> OnEncoderBroken() = 0;
};
// Returns a list of supported video formats in order of preference, to use
@@ -84,7 +84,7 @@
// subject to change without notice.
virtual CodecSupport QueryCodecSupport(
const SdpVideoFormat& format,
- absl::optional<std::string> scalability_mode) const {
+ std::optional<std::string> scalability_mode) const {
// Default implementation, query for supported formats and check if the
// specified format is supported. Returns false if scalability_mode is
// specified.
diff --git a/api/video_codecs/video_encoder_factory_template.h b/api/video_codecs/video_encoder_factory_template.h
index 599c12a..a185d0a 100644
--- a/api/video_codecs/video_encoder_factory_template.h
+++ b/api/video_codecs/video_encoder_factory_template.h
@@ -12,11 +12,11 @@
#define API_VIDEO_CODECS_VIDEO_ENCODER_FACTORY_TEMPLATE_H_
#include <memory>
+#include <optional>
#include <string>
#include <vector>
#include "absl/algorithm/container.h"
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/environment/environment.h"
#include "api/video_codecs/scalability_mode.h"
@@ -63,14 +63,14 @@
// recognize the parameter. The not so valid reason is that we have started
// adding parameters completely unrelated to the SDP to the SdpVideoFormat.
// TODO: bugs.webrtc.org/13868 - Remove FuzzyMatchSdpVideoFormat
- absl::optional<SdpVideoFormat> matched =
+ std::optional<SdpVideoFormat> matched =
FuzzyMatchSdpVideoFormat(GetSupportedFormats(), format);
return CreateInternal<Ts...>(env, matched.value_or(format));
}
CodecSupport QueryCodecSupport(
const SdpVideoFormat& format,
- absl::optional<std::string> scalability_mode) const override {
+ std::optional<std::string> scalability_mode) const override {
return QueryCodecSupportInternal<Ts...>(format, scalability_mode);
}
@@ -87,11 +87,11 @@
template <typename V>
bool IsScalabilityModeSupported(
- const absl::optional<std::string>& scalability_mode_string) const {
+ const std::optional<std::string>& scalability_mode_string) const {
if (!scalability_mode_string.has_value()) {
return true;
}
- absl::optional<ScalabilityMode> scalability_mode =
+ std::optional<ScalabilityMode> scalability_mode =
ScalabilityModeFromString(*scalability_mode_string);
return scalability_mode.has_value() &&
V::IsScalabilityModeSupported(*scalability_mode);
@@ -131,7 +131,7 @@
template <typename V, typename... Vs>
CodecSupport QueryCodecSupportInternal(
const SdpVideoFormat& format,
- const absl::optional<std::string>& scalability_mode) const {
+ const std::optional<std::string>& scalability_mode) const {
if (IsFormatInList(format, V::SupportedFormats())) {
return {.is_supported = IsScalabilityModeSupported<V>(scalability_mode)};
}
diff --git a/api/video_codecs/video_encoder_interface.h b/api/video_codecs/video_encoder_interface.h
index dc64e18..37e44a3 100644
--- a/api/video_codecs/video_encoder_interface.h
+++ b/api/video_codecs/video_encoder_interface.h
@@ -13,9 +13,9 @@
#include <cstdint>
#include <memory>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "absl/types/variant.h"
#include "api/array_view.h"
#include "api/scoped_refptr.h"
@@ -69,7 +69,7 @@
int spatial_id = 0;
Resolution resolution;
std::vector<int> reference_buffers;
- absl::optional<int> update_buffer;
+ std::optional<int> update_buffer;
int effort_level = 0;
std::unique_ptr<FrameOutput> frame_output;
diff --git a/api/video_codecs/video_encoder_software_fallback_wrapper.cc b/api/video_codecs/video_encoder_software_fallback_wrapper.cc
index 2993e16..c3e9ad9 100644
--- a/api/video_codecs/video_encoder_software_fallback_wrapper.cc
+++ b/api/video_codecs/video_encoder_software_fallback_wrapper.cc
@@ -14,12 +14,12 @@
#include <cstdio>
#include <memory>
+#include <optional>
#include <string>
#include <utility>
#include <vector>
#include "absl/strings/match.h"
-#include "absl/types/optional.h"
#include "api/environment/environment.h"
#include "api/fec_controller_override.h"
#include "api/field_trials_view.h"
@@ -83,7 +83,7 @@
const char kVp8ForceFallbackEncoderFieldTrial[] =
"WebRTC-VP8-Forced-Fallback-Encoder-v2";
-absl::optional<ForcedFallbackParams> ParseFallbackParamsFromFieldTrials(
+std::optional<ForcedFallbackParams> ParseFallbackParamsFromFieldTrials(
const FieldTrialsView& field_trials,
const VideoEncoder& main_encoder) {
// Ignore WebRTC-VP8-Forced-Fallback-Encoder-v2 if
@@ -101,7 +101,7 @@
const std::string field_trial =
field_trials.Lookup(kVp8ForceFallbackEncoderFieldTrial);
if (!absl::StartsWith(field_trial, "Enabled")) {
- return absl::nullopt;
+ return std::nullopt;
}
int max_pixels_lower_bound =
@@ -115,23 +115,23 @@
¶ms.max_pixels, &min_bps) != 3) {
RTC_LOG(LS_WARNING)
<< "Invalid number of forced fallback parameters provided.";
- return absl::nullopt;
+ return std::nullopt;
} else if (params.min_pixels <= 0 ||
params.max_pixels < max_pixels_lower_bound ||
params.max_pixels < params.min_pixels || min_bps <= 0) {
RTC_LOG(LS_WARNING) << "Invalid forced fallback parameter value provided.";
- return absl::nullopt;
+ return std::nullopt;
}
params.vp8_specific_resolution_switch = true;
return params;
}
-absl::optional<ForcedFallbackParams> GetForcedFallbackParams(
+std::optional<ForcedFallbackParams> GetForcedFallbackParams(
const FieldTrialsView& field_trials,
bool prefer_temporal_support,
const VideoEncoder& main_encoder) {
- absl::optional<ForcedFallbackParams> params =
+ std::optional<ForcedFallbackParams> params =
ParseFallbackParamsFromFieldTrials(field_trials, main_encoder);
if (prefer_temporal_support) {
if (!params.has_value()) {
@@ -203,15 +203,15 @@
// Settings used in the last InitEncode call and used if a dynamic fallback to
// software is required.
VideoCodec codec_settings_;
- absl::optional<VideoEncoder::Settings> encoder_settings_;
+ std::optional<VideoEncoder::Settings> encoder_settings_;
// The last rate control settings, if set.
- absl::optional<RateControlParameters> rate_control_parameters_;
+ std::optional<RateControlParameters> rate_control_parameters_;
// The last channel parameters set.
- absl::optional<float> packet_loss_;
- absl::optional<int64_t> rtt_;
- absl::optional<LossNotification> loss_notification_;
+ std::optional<float> packet_loss_;
+ std::optional<int64_t> rtt_;
+ std::optional<LossNotification> loss_notification_;
enum class EncoderState {
kUninitialized,
@@ -226,7 +226,7 @@
EncodedImageCallback* callback_;
- const absl::optional<ForcedFallbackParams> fallback_params_;
+ const std::optional<ForcedFallbackParams> fallback_params_;
int32_t EncodeWithMainEncoder(const VideoFrame& frame,
const std::vector<VideoFrameType>* frame_types);
};
@@ -327,7 +327,7 @@
codec_settings_ = *codec_settings;
encoder_settings_ = settings;
// Clear stored rate/channel parameters.
- rate_control_parameters_ = absl::nullopt;
+ rate_control_parameters_ = std::nullopt;
RTC_DCHECK_EQ(encoder_state_, EncoderState::kUninitialized)
<< "InitEncode() should never be called on an active instance!";
diff --git a/api/video_codecs/vp8_frame_buffer_controller.h b/api/video_codecs/vp8_frame_buffer_controller.h
index f9f0e5f..b5dd2c0 100644
--- a/api/video_codecs/vp8_frame_buffer_controller.h
+++ b/api/video_codecs/vp8_frame_buffer_controller.h
@@ -15,9 +15,9 @@
#include <cstddef>
#include <cstdint>
#include <memory>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/fec_controller_override.h"
#include "api/video_codecs/video_codec.h"
#include "api/video_codecs/video_encoder.h"
@@ -82,16 +82,16 @@
std::array<uint32_t, kMaxPeriodicity> ts_layer_id;
};
- absl::optional<TemporalLayerConfig> temporal_layer_config;
+ std::optional<TemporalLayerConfig> temporal_layer_config;
// Target bitrate, in bps.
- absl::optional<uint32_t> rc_target_bitrate;
+ std::optional<uint32_t> rc_target_bitrate;
// Clamp QP to max. Use 0 to disable clamping.
- absl::optional<uint32_t> rc_max_quantizer;
+ std::optional<uint32_t> rc_max_quantizer;
// Error resilience mode.
- absl::optional<uint32_t> g_error_resilient;
+ std::optional<uint32_t> g_error_resilient;
// If set to true, all previous configuration overrides should be reset.
bool reset_previous_configuration_overrides = false;
diff --git a/api/video_codecs/vp9_profile.cc b/api/video_codecs/vp9_profile.cc
index ba694b0..a00849d 100644
--- a/api/video_codecs/vp9_profile.cc
+++ b/api/video_codecs/vp9_profile.cc
@@ -11,10 +11,10 @@
#include "api/video_codecs/vp9_profile.h"
#include <map>
+#include <optional>
#include <string>
#include <utility>
-#include "absl/types/optional.h"
#include "api/rtp_parameters.h"
#include "rtc_base/string_to_number.h"
@@ -37,10 +37,10 @@
return "0";
}
-absl::optional<VP9Profile> StringToVP9Profile(const std::string& str) {
- const absl::optional<int> i = rtc::StringToNumber<int>(str);
+std::optional<VP9Profile> StringToVP9Profile(const std::string& str) {
+ const std::optional<int> i = rtc::StringToNumber<int>(str);
if (!i.has_value())
- return absl::nullopt;
+ return std::nullopt;
switch (i.value()) {
case 0:
@@ -52,11 +52,11 @@
case 3:
return VP9Profile::kProfile3;
default:
- return absl::nullopt;
+ return std::nullopt;
}
}
-absl::optional<VP9Profile> ParseSdpForVP9Profile(
+std::optional<VP9Profile> ParseSdpForVP9Profile(
const CodecParameterMap& params) {
const auto profile_it = params.find(kVP9FmtpProfileId);
if (profile_it == params.end())
@@ -67,8 +67,8 @@
bool VP9IsSameProfile(const CodecParameterMap& params1,
const CodecParameterMap& params2) {
- const absl::optional<VP9Profile> profile = ParseSdpForVP9Profile(params1);
- const absl::optional<VP9Profile> other_profile =
+ const std::optional<VP9Profile> profile = ParseSdpForVP9Profile(params1);
+ const std::optional<VP9Profile> other_profile =
ParseSdpForVP9Profile(params2);
return profile && other_profile && profile == other_profile;
}
diff --git a/api/video_codecs/vp9_profile.h b/api/video_codecs/vp9_profile.h
index 59b665f..94ac01c 100644
--- a/api/video_codecs/vp9_profile.h
+++ b/api/video_codecs/vp9_profile.h
@@ -11,9 +11,9 @@
#ifndef API_VIDEO_CODECS_VP9_PROFILE_H_
#define API_VIDEO_CODECS_VP9_PROFILE_H_
+#include <optional>
#include <string>
-#include "absl/types/optional.h"
#include "api/rtp_parameters.h"
#include "rtc_base/system/rtc_export.h"
@@ -35,13 +35,13 @@
// Helper functions to convert std::string to VP9Profile. Returns null if given
// an invalid profile string.
-absl::optional<VP9Profile> StringToVP9Profile(const std::string& str);
+std::optional<VP9Profile> StringToVP9Profile(const std::string& str);
// Parse profile that is represented as a string of single digit contained in an
// SDP key-value map. A default profile(kProfile0) will be returned if the
// profile key is missing. Nothing will be returned if the key is present but
// the string is invalid.
-RTC_EXPORT absl::optional<VP9Profile> ParseSdpForVP9Profile(
+RTC_EXPORT std::optional<VP9Profile> ParseSdpForVP9Profile(
const CodecParameterMap& params);
// Returns true if the parameters have the same VP9 profile, or neither contains
diff --git a/api/video_track_source_constraints.h b/api/video_track_source_constraints.h
index 55e5396..97b20f0 100644
--- a/api/video_track_source_constraints.h
+++ b/api/video_track_source_constraints.h
@@ -16,15 +16,15 @@
#ifndef API_VIDEO_TRACK_SOURCE_CONSTRAINTS_H_
#define API_VIDEO_TRACK_SOURCE_CONSTRAINTS_H_
-#include "absl/types/optional.h"
+#include <optional>
namespace webrtc {
// This struct definition describes constraints on the video source that may be
// set with VideoTrackSourceInterface::ProcessConstraints.
struct VideoTrackSourceConstraints {
- absl::optional<double> min_fps;
- absl::optional<double> max_fps;
+ std::optional<double> min_fps;
+ std::optional<double> max_fps;
};
} // namespace webrtc
diff --git a/api/voip/BUILD.gn b/api/voip/BUILD.gn
index aa17191..4c0806c 100644
--- a/api/voip/BUILD.gn
+++ b/api/voip/BUILD.gn
@@ -24,7 +24,6 @@
"../audio_codecs:audio_codecs_api",
"../neteq:neteq_api",
"//third_party/abseil-cpp/absl/base:core_headers",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -59,7 +58,6 @@
"..:array_view",
"../../test:test_support",
"../audio_codecs:audio_codecs_api",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
diff --git a/api/voip/test/mock_voip_engine.h b/api/voip/test/mock_voip_engine.h
index 38b857d..9994388 100644
--- a/api/voip/test/mock_voip_engine.h
+++ b/api/voip/test/mock_voip_engine.h
@@ -13,8 +13,8 @@
#include <cstdint>
#include <map>
+#include <optional>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/audio_codecs/audio_format.h"
#include "api/voip/voip_base.h"
@@ -32,7 +32,7 @@
public:
MOCK_METHOD(ChannelId,
CreateChannel,
- (Transport*, absl::optional<uint32_t>),
+ (Transport*, std::optional<uint32_t>),
(override));
MOCK_METHOD(VoipResult, ReleaseChannel, (ChannelId), (override));
MOCK_METHOD(VoipResult, StartSend, (ChannelId), (override));
diff --git a/api/voip/voip_base.h b/api/voip/voip_base.h
index d5ff91b..f7f4432 100644
--- a/api/voip/voip_base.h
+++ b/api/voip/voip_base.h
@@ -12,9 +12,9 @@
#define API_VOIP_VOIP_BASE_H_
#include <cstdint>
+#include <optional>
#include "absl/base/attributes.h"
-#include "absl/types/optional.h"
namespace webrtc {
@@ -67,7 +67,7 @@
// Returns a ChannelId created for caller to handle subsequent Channel
// operations.
virtual ChannelId CreateChannel(Transport* transport,
- absl::optional<uint32_t> local_ssrc) = 0;
+ std::optional<uint32_t> local_ssrc) = 0;
// Releases `channel_id` that no longer has any use.
// Returns following VoipResult;
diff --git a/api/voip/voip_statistics.h b/api/voip/voip_statistics.h
index 430349b..8b994d1 100644
--- a/api/voip/voip_statistics.h
+++ b/api/voip/voip_statistics.h
@@ -12,8 +12,8 @@
#define API_VOIP_VOIP_STATISTICS_H_
#include <cstdint>
+#include <optional>
-#include "absl/types/optional.h"
#include "api/neteq/neteq.h"
#include "api/voip/voip_base.h"
@@ -42,7 +42,7 @@
double fraction_lost = 0.0;
// https://w3c.github.io/webrtc-stats/#dom-rtcremoteinboundrtpstreamstats-roundtriptime
- absl::optional<double> round_trip_time;
+ std::optional<double> round_trip_time;
// Last time (not RTP timestamp) when RTCP report received in milliseconds.
int64_t last_report_received_timestamp_ms;
@@ -69,9 +69,9 @@
// SSRC from remote media endpoint as indicated either by RTP header in RFC
// 3550 [5.1] or RTCP SSRC of sender in RFC 3550 [6.4.1].
- absl::optional<uint32_t> remote_ssrc;
+ std::optional<uint32_t> remote_ssrc;
- absl::optional<RemoteRtcpStatistics> remote_rtcp;
+ std::optional<RemoteRtcpStatistics> remote_rtcp;
};
// VoipStatistics interface provides the interfaces for querying metrics around
diff --git a/audio/BUILD.gn b/audio/BUILD.gn
index e75ca3e..cb83e98 100644
--- a/audio/BUILD.gn
+++ b/audio/BUILD.gn
@@ -116,7 +116,6 @@
"//third_party/abseil-cpp/absl/functional:any_invocable",
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
if (rtc_include_tests) {
@@ -226,7 +225,6 @@
"utility:utility_tests",
"//testing/gtest",
"//third_party/abseil-cpp/absl/memory",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
diff --git a/audio/audio_receive_stream.cc b/audio/audio_receive_stream.cc
index d494b63..44c95f3 100644
--- a/audio/audio_receive_stream.cc
+++ b/audio/audio_receive_stream.cc
@@ -406,7 +406,7 @@
return remote_ssrc();
}
-absl::optional<Syncable::Info> AudioReceiveStreamImpl::GetInfo() const {
+std::optional<Syncable::Info> AudioReceiveStreamImpl::GetInfo() const {
// TODO(bugs.webrtc.org/11993): This is called via RtpStreamsSynchronizer,
// expect to be called on the network thread.
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
diff --git a/audio/audio_receive_stream.h b/audio/audio_receive_stream.h
index 8c094a4..4eac8a7 100644
--- a/audio/audio_receive_stream.h
+++ b/audio/audio_receive_stream.h
@@ -109,7 +109,7 @@
// Syncable
uint32_t id() const override;
- absl::optional<Syncable::Info> GetInfo() const override;
+ std::optional<Syncable::Info> GetInfo() const override;
bool GetPlayoutRtpTimestamp(uint32_t* rtp_timestamp,
int64_t* time_ms) const override;
void SetEstimatedPlayoutNtpTimestampMs(int64_t ntp_timestamp_ms,
diff --git a/audio/audio_send_stream.cc b/audio/audio_send_stream.cc
index fc194c9..6e5bd18 100644
--- a/audio/audio_send_stream.cc
+++ b/audio/audio_send_stream.cc
@@ -49,8 +49,8 @@
const AudioSendStream::Config* old_config) {
using SendCodecSpec = AudioSendStream::Config::SendCodecSpec;
// Only update if any of the things we log have changed.
- auto payload_types_equal = [](const absl::optional<SendCodecSpec>& a,
- const absl::optional<SendCodecSpec>& b) {
+ auto payload_types_equal = [](const std::optional<SendCodecSpec>& a,
+ const std::optional<SendCodecSpec>& b) {
if (a.has_value() && b.has_value()) {
return a->format.name == b->format.name &&
a->payload_type == b->payload_type;
@@ -106,7 +106,7 @@
RtpTransportControllerSendInterface* rtp_transport,
BitrateAllocatorInterface* bitrate_allocator,
RtcpRttStats* rtcp_rtt_stats,
- const absl::optional<RtpState>& suspended_rtp_state)
+ const std::optional<RtpState>& suspended_rtp_state)
: AudioSendStream(env,
config,
audio_state,
@@ -130,7 +130,7 @@
const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
RtpTransportControllerSendInterface* rtp_transport,
BitrateAllocatorInterface* bitrate_allocator,
- const absl::optional<RtpState>& suspended_rtp_state,
+ const std::optional<RtpState>& suspended_rtp_state,
std::unique_ptr<voe::ChannelSendInterface> channel_send)
: env_(env),
allocate_audio_without_feedback_(
@@ -479,7 +479,7 @@
// Pick a target bitrate between the constraints. Overrules the allocator if
// it 1) allocated a bitrate of zero to disable the stream or 2) allocated a
// higher than max to allow for e.g. extra FEC.
- absl::optional<TargetAudioBitrateConstraints> constraints =
+ std::optional<TargetAudioBitrateConstraints> constraints =
GetMinMaxBitrateConstraints();
if (constraints) {
update.target_bitrate.Clamp(constraints->min, constraints->max);
@@ -491,7 +491,7 @@
return 0;
}
-absl::optional<DataRate> AudioSendStream::GetUsedRate() const {
+std::optional<DataRate> AudioSendStream::GetUsedRate() const {
return channel_send_->GetUsedRate();
}
@@ -657,7 +657,7 @@
return SetupSendCodec(new_config);
}
- const absl::optional<int>& new_target_bitrate_bps =
+ const std::optional<int>& new_target_bitrate_bps =
new_config.send_codec_spec->target_bitrate_bps;
// If a bitrate has been specified for the codec, use it over the
// codec's default.
@@ -823,14 +823,14 @@
bitrate_allocator_->RemoveObserver(this);
}
-absl::optional<AudioSendStream::TargetAudioBitrateConstraints>
+std::optional<AudioSendStream::TargetAudioBitrateConstraints>
AudioSendStream::GetMinMaxBitrateConstraints() const {
if (config_.min_bitrate_bps < 0 || config_.max_bitrate_bps < 0) {
RTC_LOG(LS_WARNING) << "Config is invalid: min_bitrate_bps="
<< config_.min_bitrate_bps
<< "; max_bitrate_bps=" << config_.max_bitrate_bps
<< "; both expected greater or equal to 0";
- return absl::nullopt;
+ return std::nullopt;
}
TargetAudioBitrateConstraints constraints{
DataRate::BitsPerSec(config_.min_bitrate_bps),
@@ -853,7 +853,7 @@
if (constraints.max < constraints.min) {
RTC_LOG(LS_WARNING) << "TargetAudioBitrateConstraints::max is less than "
<< "TargetAudioBitrateConstraints::min";
- return absl::nullopt;
+ return std::nullopt;
}
if (use_legacy_overhead_calculation_) {
// OverheadPerPacket = Ipv4(20B) + UDP(8B) + SRTP(10B) + RTP(12)
@@ -866,7 +866,7 @@
} else {
if (!frame_length_range_.has_value()) {
RTC_LOG(LS_WARNING) << "frame_length_range_ is not set";
- return absl::nullopt;
+ return std::nullopt;
}
const DataSize overhead_per_packet = DataSize::Bytes(overhead_per_packet_);
constraints.min += overhead_per_packet / frame_length_range_->second;
diff --git a/audio/audio_send_stream.h b/audio/audio_send_stream.h
index 3845e8d..62a5537 100644
--- a/audio/audio_send_stream.h
+++ b/audio/audio_send_stream.h
@@ -38,13 +38,13 @@
static constexpr char kKey[] = "WebRTC-Audio-Allocation";
// Field Trial configured bitrates to use as overrides over default/user
// configured bitrate range when audio bitrate allocation is enabled.
- absl::optional<DataRate> min_bitrate;
- absl::optional<DataRate> max_bitrate;
+ std::optional<DataRate> min_bitrate;
+ std::optional<DataRate> max_bitrate;
DataRate priority_bitrate = DataRate::Zero();
// By default the priority_bitrate is compensated for packet overhead.
// Use this flag to configure a raw value instead.
- absl::optional<DataRate> priority_bitrate_raw;
- absl::optional<double> bitrate_priority;
+ std::optional<DataRate> priority_bitrate_raw;
+ std::optional<double> bitrate_priority;
std::unique_ptr<StructParametersParser> Parser();
explicit AudioAllocationConfig(const FieldTrialsView& field_trials);
@@ -61,14 +61,14 @@
RtpTransportControllerSendInterface* rtp_transport,
BitrateAllocatorInterface* bitrate_allocator,
RtcpRttStats* rtcp_rtt_stats,
- const absl::optional<RtpState>& suspended_rtp_state);
+ const std::optional<RtpState>& suspended_rtp_state);
// For unit tests, which need to supply a mock ChannelSend.
AudioSendStream(const Environment& env,
const webrtc::AudioSendStream::Config& config,
const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
RtpTransportControllerSendInterface* rtp_transport,
BitrateAllocatorInterface* bitrate_allocator,
- const absl::optional<RtpState>& suspended_rtp_state,
+ const std::optional<RtpState>& suspended_rtp_state,
std::unique_ptr<voe::ChannelSendInterface> channel_send);
AudioSendStream() = delete;
@@ -97,7 +97,7 @@
// Implements BitrateAllocatorObserver.
uint32_t OnBitrateUpdated(BitrateAllocationUpdate update) override;
- absl::optional<DataRate> GetUsedRate() const override;
+ std::optional<DataRate> GetUsedRate() const override;
void SetTransportOverhead(int transport_overhead_per_packet_bytes);
@@ -141,7 +141,7 @@
// Returns bitrate constraints, maybe including overhead when enabled by
// field trial.
- absl::optional<TargetAudioBitrateConstraints> GetMinMaxBitrateConstraints()
+ std::optional<TargetAudioBitrateConstraints> GetMinMaxBitrateConstraints()
const RTC_RUN_ON(worker_thread_checker_);
// Sets per-packet overhead on encoded (for ANA) based on current known values
@@ -181,7 +181,7 @@
RtpTransportControllerSendInterface* const rtp_transport_;
RtpRtcpInterface* const rtp_rtcp_module_;
- absl::optional<RtpState> const suspended_rtp_state_;
+ std::optional<RtpState> const suspended_rtp_state_;
// RFC 5285: Each distinct extension MUST have a unique ID. The value 0 is
// reserved for padding and MUST NOT be used as a local identifier.
@@ -207,9 +207,9 @@
bool registered_with_allocator_ RTC_GUARDED_BY(worker_thread_checker_) =
false;
- absl::optional<std::pair<TimeDelta, TimeDelta>> frame_length_range_
+ std::optional<std::pair<TimeDelta, TimeDelta>> frame_length_range_
RTC_GUARDED_BY(worker_thread_checker_);
- absl::optional<std::pair<DataRate, DataRate>> bitrate_range_
+ std::optional<std::pair<DataRate, DataRate>> bitrate_range_
RTC_GUARDED_BY(worker_thread_checker_);
};
} // namespace internal
diff --git a/audio/audio_send_stream_unittest.cc b/audio/audio_send_stream_unittest.cc
index 6aa95c7..7198a61 100644
--- a/audio/audio_send_stream_unittest.cc
+++ b/audio/audio_send_stream_unittest.cc
@@ -128,13 +128,13 @@
std::begin(kCodecSpecs), std::end(kCodecSpecs))));
ON_CALL(*factory, QueryAudioEncoder)
.WillByDefault(
- [](const SdpAudioFormat& format) -> absl::optional<AudioCodecInfo> {
+ [](const SdpAudioFormat& format) -> std::optional<AudioCodecInfo> {
for (const auto& spec : kCodecSpecs) {
if (format == spec.format) {
return spec.info;
}
}
- return absl::nullopt;
+ return std::nullopt;
});
ON_CALL(*factory, Create).WillByDefault(WithArg<1>(&SetupAudioEncoderMock));
return factory;
@@ -183,7 +183,7 @@
CreateEnvironment(&field_trials, time_controller_.GetClock(),
time_controller_.GetTaskQueueFactory()),
stream_config_, audio_state_, &rtp_transport_, &bitrate_allocator_,
- absl::nullopt,
+ std::nullopt,
std::unique_ptr<voe::ChannelSendInterface>(channel_send_));
}
diff --git a/audio/audio_transport_impl.cc b/audio/audio_transport_impl.cc
index b9d8332..09509cd 100644
--- a/audio/audio_transport_impl.cc
+++ b/audio/audio_transport_impl.cc
@@ -118,7 +118,7 @@
return RecordedDataIsAvailable(
audio_data, number_of_frames, bytes_per_sample, number_of_channels,
sample_rate, audio_delay_milliseconds, clock_drift, volume, key_pressed,
- new_mic_volume, /*estimated_capture_time_ns=*/absl::nullopt);
+ new_mic_volume, /*estimated_capture_time_ns=*/std::nullopt);
}
// Not used in Chromium. Process captured audio and distribute to all sending
@@ -134,7 +134,7 @@
uint32_t /*volume*/,
bool key_pressed,
uint32_t& /*new_mic_volume*/,
- absl::optional<int64_t>
+ std::optional<int64_t>
estimated_capture_time_ns) { // NOLINT: to avoid changing APIs
RTC_DCHECK(audio_data);
RTC_DCHECK_GE(number_of_channels, 1);
diff --git a/audio/audio_transport_impl.h b/audio/audio_transport_impl.h
index 3012cc2..8eac8ef 100644
--- a/audio/audio_transport_impl.h
+++ b/audio/audio_transport_impl.h
@@ -63,7 +63,7 @@
uint32_t currentMicLevel,
bool keyPressed,
uint32_t& newMicLevel,
- absl::optional<int64_t> estimated_capture_time_ns) override;
+ std::optional<int64_t> estimated_capture_time_ns) override;
int32_t NeedMorePlayData(size_t nSamples,
size_t nBytesPerSample,
diff --git a/audio/channel_receive.cc b/audio/channel_receive.cc
index e403cfa..7fcdf66 100644
--- a/audio/channel_receive.cc
+++ b/audio/channel_receive.cc
@@ -70,7 +70,7 @@
acm2::AcmReceiver::Config AcmConfig(
NetEqFactory* neteq_factory,
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory,
- absl::optional<AudioCodecPairId> codec_pair_id,
+ std::optional<AudioCodecPairId> codec_pair_id,
size_t jitter_buffer_max_packets,
bool jitter_buffer_fast_playout,
int jitter_buffer_min_delay_ms) {
@@ -102,7 +102,7 @@
int jitter_buffer_min_delay_ms,
bool enable_non_sender_rtt,
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory,
- absl::optional<AudioCodecPairId> codec_pair_id,
+ std::optional<AudioCodecPairId> codec_pair_id,
rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor,
const webrtc::CryptoOptions& crypto_options,
rtc::scoped_refptr<FrameTransformerInterface> frame_transformer);
@@ -118,7 +118,7 @@
void StopPlayout() override;
// Codecs
- absl::optional<std::pair<int, SdpAudioFormat>> GetReceiveCodec()
+ std::optional<std::pair<int, SdpAudioFormat>> GetReceiveCodec()
const override;
void ReceivedRTCPPacket(const uint8_t* data, size_t length) override;
@@ -146,7 +146,7 @@
int64_t* time_ms) const override;
void SetEstimatedPlayoutNtpTimestampMs(int64_t ntp_timestamp_ms,
int64_t time_ms) override;
- absl::optional<int64_t> GetCurrentEstimatedPlayoutNtpTimestampMs(
+ std::optional<int64_t> GetCurrentEstimatedPlayoutNtpTimestampMs(
int64_t now_ms) const override;
// Audio quality.
@@ -154,7 +154,7 @@
int GetBaseMinimumPlayoutDelayMs() const override;
// Produces the transport-related timestamps; current_delay_ms is left unset.
- absl::optional<Syncable::Info> GetSyncInfo() const override;
+ std::optional<Syncable::Info> GetSyncInfo() const override;
void RegisterReceiverCongestionControlObjects(
PacketRouter* packet_router) override;
@@ -244,9 +244,9 @@
// Info for GetSyncInfo is updated on network or worker thread, and queried on
// the worker thread.
- absl::optional<uint32_t> last_received_rtp_timestamp_
+ std::optional<uint32_t> last_received_rtp_timestamp_
RTC_GUARDED_BY(&worker_thread_checker_);
- absl::optional<int64_t> last_received_rtp_system_time_ms_
+ std::optional<int64_t> last_received_rtp_system_time_ms_
RTC_GUARDED_BY(&worker_thread_checker_);
// The AcmReceiver is thread safe, using its own lock.
@@ -257,15 +257,15 @@
RemoteNtpTimeEstimator ntp_estimator_ RTC_GUARDED_BY(ts_stats_lock_);
// Timestamp of the audio pulled from NetEq.
- absl::optional<uint32_t> jitter_buffer_playout_timestamp_;
+ std::optional<uint32_t> jitter_buffer_playout_timestamp_;
uint32_t playout_timestamp_rtp_ RTC_GUARDED_BY(worker_thread_checker_);
- absl::optional<int64_t> playout_timestamp_rtp_time_ms_
+ std::optional<int64_t> playout_timestamp_rtp_time_ms_
RTC_GUARDED_BY(worker_thread_checker_);
uint32_t playout_delay_ms_ RTC_GUARDED_BY(worker_thread_checker_);
- absl::optional<int64_t> playout_timestamp_ntp_
+ std::optional<int64_t> playout_timestamp_ntp_
RTC_GUARDED_BY(worker_thread_checker_);
- absl::optional<int64_t> playout_timestamp_ntp_time_ms_
+ std::optional<int64_t> playout_timestamp_ntp_time_ms_
RTC_GUARDED_BY(worker_thread_checker_);
mutable Mutex ts_stats_lock_;
@@ -530,7 +530,7 @@
int jitter_buffer_min_delay_ms,
bool enable_non_sender_rtt,
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory,
- absl::optional<AudioCodecPairId> codec_pair_id,
+ std::optional<AudioCodecPairId> codec_pair_id,
rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor,
const webrtc::CryptoOptions& crypto_options,
rtc::scoped_refptr<FrameTransformerInterface> frame_transformer)
@@ -609,7 +609,7 @@
acm_receiver_.FlushBuffers();
}
-absl::optional<std::pair<int, SdpAudioFormat>> ChannelReceive::GetReceiveCodec()
+std::optional<std::pair<int, SdpAudioFormat>> ChannelReceive::GetReceiveCodec()
const {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
return acm_receiver_.LastDecoder();
@@ -736,13 +736,13 @@
// Deliver RTCP packet to RTP/RTCP module for parsing
rtp_rtcp_->IncomingRtcpPacket(rtc::MakeArrayView(data, length));
- absl::optional<TimeDelta> rtt = rtp_rtcp_->LastRtt();
+ std::optional<TimeDelta> rtt = rtp_rtcp_->LastRtt();
if (!rtt.has_value()) {
// Waiting for valid RTT.
return;
}
- absl::optional<RtpRtcpInterface::SenderReportStats> last_sr =
+ std::optional<RtpRtcpInterface::SenderReportStats> last_sr =
rtp_rtcp_->GetSenderReportStats();
if (!last_sr.has_value()) {
// Waiting for RTCP.
@@ -753,7 +753,7 @@
MutexLock lock(&ts_stats_lock_);
ntp_estimator_.UpdateRtcpTimestamp(*rtt, last_sr->last_remote_timestamp,
last_sr->last_remote_rtp_timestamp);
- absl::optional<int64_t> remote_to_local_clock_offset =
+ std::optional<int64_t> remote_to_local_clock_offset =
ntp_estimator_.EstimateRemoteToLocalClockOffset();
if (remote_to_local_clock_offset.has_value()) {
capture_clock_offset_updater_.SetRemoteToLocalClockOffset(
@@ -829,7 +829,7 @@
stats.payload_bytes_received = 0;
stats.header_and_padding_bytes_received = 0;
stats.packetsReceived = 0;
- stats.last_packet_received = absl::nullopt;
+ stats.last_packet_received = std::nullopt;
}
{
@@ -843,7 +843,7 @@
stats.capture_start_ntp_time_ms_ = capture_start_ntp_time_ms_;
}
- absl::optional<RtpRtcpInterface::SenderReportStats> rtcp_sr_stats =
+ std::optional<RtpRtcpInterface::SenderReportStats> rtcp_sr_stats =
rtp_rtcp_->GetSenderReportStats();
if (rtcp_sr_stats.has_value()) {
stats.last_sender_report_timestamp_ms =
@@ -856,7 +856,7 @@
stats.sender_reports_reports_count = rtcp_sr_stats->reports_count;
}
- absl::optional<RtpRtcpInterface::NonSenderRttStats> non_sender_rtt_stats =
+ std::optional<RtpRtcpInterface::NonSenderRttStats> non_sender_rtt_stats =
rtp_rtcp_->GetNonSenderRttStats();
if (non_sender_rtt_stats.has_value()) {
stats.round_trip_time = non_sender_rtt_stats->round_trip_time;
@@ -1007,11 +1007,11 @@
playout_timestamp_ntp_time_ms_ = time_ms;
}
-absl::optional<int64_t>
-ChannelReceive::GetCurrentEstimatedPlayoutNtpTimestampMs(int64_t now_ms) const {
+std::optional<int64_t> ChannelReceive::GetCurrentEstimatedPlayoutNtpTimestampMs(
+ int64_t now_ms) const {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
if (!playout_timestamp_ntp_ || !playout_timestamp_ntp_time_ms_)
- return absl::nullopt;
+ return std::nullopt;
int64_t elapsed_ms = now_ms - *playout_timestamp_ntp_time_ms_;
return *playout_timestamp_ntp_ + elapsed_ms;
@@ -1027,23 +1027,23 @@
return acm_receiver_.GetBaseMinimumDelayMs();
}
-absl::optional<Syncable::Info> ChannelReceive::GetSyncInfo() const {
+std::optional<Syncable::Info> ChannelReceive::GetSyncInfo() const {
// TODO(bugs.webrtc.org/11993): This should run on the network thread.
// We get here via RtpStreamsSynchronizer. Once that's done, many of
// these locks aren't needed.
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
Syncable::Info info;
- absl::optional<RtpRtcpInterface::SenderReportStats> last_sr =
+ std::optional<RtpRtcpInterface::SenderReportStats> last_sr =
rtp_rtcp_->GetSenderReportStats();
if (!last_sr.has_value()) {
- return absl::nullopt;
+ return std::nullopt;
}
info.capture_time_ntp_secs = last_sr->last_remote_timestamp.seconds();
info.capture_time_ntp_frac = last_sr->last_remote_timestamp.fractions();
info.capture_time_source_clock = last_sr->last_remote_rtp_timestamp;
if (!last_received_rtp_timestamp_ || !last_received_rtp_system_time_ms_) {
- return absl::nullopt;
+ return std::nullopt;
}
info.latest_received_capture_timestamp = *last_received_rtp_timestamp_;
info.latest_receive_time_ms = *last_received_rtp_system_time_ms_;
@@ -1116,7 +1116,7 @@
int jitter_buffer_min_delay_ms,
bool enable_non_sender_rtt,
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory,
- absl::optional<AudioCodecPairId> codec_pair_id,
+ std::optional<AudioCodecPairId> codec_pair_id,
rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor,
const webrtc::CryptoOptions& crypto_options,
rtc::scoped_refptr<FrameTransformerInterface> frame_transformer) {
diff --git a/audio/channel_receive.h b/audio/channel_receive.h
index 0e73eac..0a11573 100644
--- a/audio/channel_receive.h
+++ b/audio/channel_receive.h
@@ -13,10 +13,10 @@
#include <map>
#include <memory>
+#include <optional>
#include <utility>
#include <vector>
-#include "absl/types/optional.h"
#include "api/audio/audio_mixer.h"
#include "api/audio_codecs/audio_decoder_factory.h"
#include "api/call/audio_sink.h"
@@ -62,17 +62,17 @@
// The timestamp at which the last packet was received, i.e. the time of the
// local clock when it was received - not the RTP timestamp of that packet.
// https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-lastpacketreceivedtimestamp
- absl::optional<Timestamp> last_packet_received;
+ std::optional<Timestamp> last_packet_received;
// Remote outbound stats derived by the received RTCP sender reports.
// Note that the timestamps below correspond to the time elapsed since the
// Unix epoch.
// https://w3c.github.io/webrtc-stats/#remoteoutboundrtpstats-dict*
- absl::optional<int64_t> last_sender_report_timestamp_ms;
- absl::optional<int64_t> last_sender_report_remote_timestamp_ms;
+ std::optional<int64_t> last_sender_report_timestamp_ms;
+ std::optional<int64_t> last_sender_report_remote_timestamp_ms;
uint64_t sender_reports_packets_sent = 0;
uint64_t sender_reports_bytes_sent = 0;
uint64_t sender_reports_reports_count = 0;
- absl::optional<TimeDelta> round_trip_time;
+ std::optional<TimeDelta> round_trip_time;
TimeDelta total_round_trip_time = TimeDelta::Zero();
int round_trip_time_measurements;
};
@@ -97,7 +97,7 @@
virtual void StopPlayout() = 0;
// Payload type and format of last received RTP packet, if any.
- virtual absl::optional<std::pair<int, SdpAudioFormat>> GetReceiveCodec()
+ virtual std::optional<std::pair<int, SdpAudioFormat>> GetReceiveCodec()
const = 0;
virtual void ReceivedRTCPPacket(const uint8_t* data, size_t length) = 0;
@@ -121,7 +121,7 @@
int64_t* time_ms) const = 0;
virtual void SetEstimatedPlayoutNtpTimestampMs(int64_t ntp_timestamp_ms,
int64_t time_ms) = 0;
- virtual absl::optional<int64_t> GetCurrentEstimatedPlayoutNtpTimestampMs(
+ virtual std::optional<int64_t> GetCurrentEstimatedPlayoutNtpTimestampMs(
int64_t now_ms) const = 0;
// Audio quality.
@@ -131,7 +131,7 @@
virtual int GetBaseMinimumPlayoutDelayMs() const = 0;
// Produces the transport-related timestamps; current_delay_ms is left unset.
- virtual absl::optional<Syncable::Info> GetSyncInfo() const = 0;
+ virtual std::optional<Syncable::Info> GetSyncInfo() const = 0;
virtual void RegisterReceiverCongestionControlObjects(
PacketRouter* packet_router) = 0;
@@ -182,7 +182,7 @@
int jitter_buffer_min_delay_ms,
bool enable_non_sender_rtt,
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory,
- absl::optional<AudioCodecPairId> codec_pair_id,
+ std::optional<AudioCodecPairId> codec_pair_id,
rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor,
const webrtc::CryptoOptions& crypto_options,
rtc::scoped_refptr<FrameTransformerInterface> frame_transformer);
diff --git a/audio/channel_receive_frame_transformer_delegate.cc b/audio/channel_receive_frame_transformer_delegate.cc
index 795f46f..d82925a 100644
--- a/audio/channel_receive_frame_transformer_delegate.cc
+++ b/audio/channel_receive_frame_transformer_delegate.cc
@@ -12,10 +12,10 @@
#include <cstdint>
#include <memory>
+#include <optional>
#include <string>
#include <utility>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/frame_transformer_interface.h"
#include "api/rtp_headers.h"
@@ -62,14 +62,14 @@
Direction GetDirection() const override { return Direction::kReceiver; }
std::string GetMimeType() const override { return codec_mime_type_; }
- const absl::optional<uint16_t> SequenceNumber() const override {
+ const std::optional<uint16_t> SequenceNumber() const override {
return header_.sequenceNumber;
}
- absl::optional<uint64_t> AbsoluteCaptureTimestamp() const override {
+ std::optional<uint64_t> AbsoluteCaptureTimestamp() const override {
// This could be extracted from received header extensions + extrapolation,
// if required in future, eg for being able to re-send received frames.
- return absl::nullopt;
+ return std::nullopt;
}
const RTPHeader& Header() const { return header_; }
@@ -83,17 +83,17 @@
: FrameType::kAudioFrameCN;
}
- absl::optional<uint8_t> AudioLevel() const override {
+ std::optional<uint8_t> AudioLevel() const override {
if (header_.extension.audio_level()) {
return header_.extension.audio_level()->level();
}
- return absl::nullopt;
+ return std::nullopt;
}
- absl::optional<Timestamp> ReceiveTime() const override {
+ std::optional<Timestamp> ReceiveTime() const override {
return receive_time_ == Timestamp::MinusInfinity()
- ? absl::nullopt
- : absl::optional<Timestamp>(receive_time_);
+ ? std::nullopt
+ : std::optional<Timestamp>(receive_time_);
}
private:
diff --git a/audio/channel_receive_unittest.cc b/audio/channel_receive_unittest.cc
index 5bc2222..ef9647d 100644
--- a/audio/channel_receive_unittest.cc
+++ b/audio/channel_receive_unittest.cc
@@ -67,7 +67,7 @@
/* jitter_buffer_fast_playout= */ false,
/* jitter_buffer_min_delay_ms= */ 0,
/* enable_non_sender_rtt= */ false, audio_decoder_factory_,
- /* codec_pair_id= */ absl::nullopt,
+ /* codec_pair_id= */ std::nullopt,
/* frame_decryptor_interface= */ nullptr, crypto_options,
/* frame_transformer= */ nullptr);
channel->SetReceiveCodecs(
diff --git a/audio/channel_send.cc b/audio/channel_send.cc
index 18081e3..d1683b7 100644
--- a/audio/channel_send.cc
+++ b/audio/channel_send.cc
@@ -66,7 +66,7 @@
void Reset() {
rate_last_frame_ = DataRate::BitsPerSec(0);
next_frame_duration_ = TimeDelta::Millis(0);
- report_rate_ = absl::nullopt;
+ report_rate_ = std::nullopt;
}
// A new frame is formed when bytesize is nonzero.
@@ -89,13 +89,13 @@
next_frame_duration_ = TimeDelta::Millis(0);
}
- absl::optional<DataRate> GetUsedRate() const { return report_rate_; }
+ std::optional<DataRate> GetUsedRate() const { return report_rate_; }
private:
TimeDelta next_frame_duration_ = TimeDelta::Millis(0);
DataSize packet_overhead_ = DataSize::Bytes(72);
DataRate rate_last_frame_ = DataRate::BitsPerSec(0);
- absl::optional<DataRate> report_rate_;
+ std::optional<DataRate> report_rate_;
};
class ChannelSend : public ChannelSendInterface,
@@ -194,7 +194,7 @@
void OnReportBlockDataUpdated(ReportBlockData report_block) override;
// Reports actual bitrate used (vs allocated).
- absl::optional<DataRate> GetUsedRate() const override {
+ std::optional<DataRate> GetUsedRate() const override {
MutexLock lock(&bitrate_accountant_mutex_);
return bitrate_accountant_.GetUsedRate();
}
@@ -221,7 +221,7 @@
rtc::ArrayView<const uint8_t> payload,
int64_t absolute_capture_timestamp_ms,
rtc::ArrayView<const uint32_t> csrcs,
- absl::optional<uint8_t> audio_level_dbov)
+ std::optional<uint8_t> audio_level_dbov)
RTC_RUN_ON(encoder_queue_checker_);
void OnReceivedRtt(int64_t rtt_ms);
@@ -254,7 +254,7 @@
// This is just an offset, RTP module will add its own random offset.
uint32_t timestamp_ RTC_GUARDED_BY(audio_thread_race_checker_) = 0;
- absl::optional<int64_t> last_capture_timestamp_ms_
+ std::optional<int64_t> last_capture_timestamp_ms_
RTC_GUARDED_BY(audio_thread_race_checker_);
RmsLevel rms_level_ RTC_GUARDED_BY(encoder_queue_checker_);
@@ -352,7 +352,7 @@
RTC_DCHECK_RUN_ON(&encoder_queue_checker_);
rtc::ArrayView<const uint8_t> payload(payloadData, payloadSize);
- absl::optional<uint8_t> audio_level_dbov;
+ std::optional<uint8_t> audio_level_dbov;
if (include_audio_level_indication_.load()) {
// Take the averaged audio levels from rms_level_ and reset it before
// invoking any async transformer.
@@ -383,7 +383,7 @@
rtc::ArrayView<const uint8_t> payload,
int64_t absolute_capture_timestamp_ms,
rtc::ArrayView<const uint32_t> csrcs,
- absl::optional<uint8_t> audio_level_dbov) {
+ std::optional<uint8_t> audio_level_dbov) {
// E2EE Custom Audio Frame Encryption (This is optional).
// Keep this buffer around for the lifetime of the send call.
rtc::Buffer encrypted_audio_payload;
@@ -942,7 +942,7 @@
rtc::ArrayView<const uint8_t> payload,
int64_t absolute_capture_timestamp_ms,
rtc::ArrayView<const uint32_t> csrcs,
- absl::optional<uint8_t> audio_level_dbov) {
+ std::optional<uint8_t> audio_level_dbov) {
RTC_DCHECK_RUN_ON(&encoder_queue_checker_);
return SendRtpAudio(
frameType, payloadType,
diff --git a/audio/channel_send.h b/audio/channel_send.h
index 6082c0e..02215e4 100644
--- a/audio/channel_send.h
+++ b/audio/channel_send.h
@@ -112,7 +112,7 @@
frame_transformer) = 0;
// Returns payload bitrate actually used.
- virtual absl::optional<DataRate> GetUsedRate() const = 0;
+ virtual std::optional<DataRate> GetUsedRate() const = 0;
// Registers per packet byte overhead.
virtual void RegisterPacketOverhead(int packet_byte_overhead) = 0;
diff --git a/audio/channel_send_frame_transformer_delegate.cc b/audio/channel_send_frame_transformer_delegate.cc
index 64adef9..dacbfcd 100644
--- a/audio/channel_send_frame_transformer_delegate.cc
+++ b/audio/channel_send_frame_transformer_delegate.cc
@@ -56,12 +56,12 @@
uint32_t rtp_timestamp_with_offset,
const uint8_t* payload_data,
size_t payload_size,
- absl::optional<uint64_t> absolute_capture_timestamp_ms,
+ std::optional<uint64_t> absolute_capture_timestamp_ms,
uint32_t ssrc,
std::vector<uint32_t> csrcs,
const std::string& codec_mime_type,
- absl::optional<uint16_t> sequence_number,
- absl::optional<uint8_t> audio_level_dbov)
+ std::optional<uint16_t> sequence_number,
+ std::optional<uint8_t> audio_level_dbov)
: TransformableAudioFrameInterface(Passkey()),
frame_type_(frame_type),
payload_type_(payload_type),
@@ -93,7 +93,7 @@
return csrcs_;
}
- const absl::optional<uint16_t> SequenceNumber() const override {
+ const std::optional<uint16_t> SequenceNumber() const override {
return sequence_number_;
}
@@ -101,29 +101,27 @@
rtp_timestamp_with_offset_ = rtp_timestamp_with_offset;
}
- absl::optional<uint64_t> AbsoluteCaptureTimestamp() const override {
+ std::optional<uint64_t> AbsoluteCaptureTimestamp() const override {
return absolute_capture_timestamp_ms_;
}
- absl::optional<uint8_t> AudioLevel() const override {
+ std::optional<uint8_t> AudioLevel() const override {
return audio_level_dbov_;
}
- absl::optional<Timestamp> ReceiveTime() const override {
- return absl::nullopt;
- }
+ std::optional<Timestamp> ReceiveTime() const override { return std::nullopt; }
private:
AudioFrameType frame_type_;
uint8_t payload_type_;
uint32_t rtp_timestamp_with_offset_;
rtc::Buffer payload_;
- absl::optional<uint64_t> absolute_capture_timestamp_ms_;
+ std::optional<uint64_t> absolute_capture_timestamp_ms_;
uint32_t ssrc_;
std::vector<uint32_t> csrcs_;
std::string codec_mime_type_;
- absl::optional<uint16_t> sequence_number_;
- absl::optional<uint8_t> audio_level_dbov_;
+ std::optional<uint16_t> sequence_number_;
+ std::optional<uint8_t> audio_level_dbov_;
};
ChannelSendFrameTransformerDelegate::ChannelSendFrameTransformerDelegate(
@@ -156,7 +154,7 @@
int64_t absolute_capture_timestamp_ms,
uint32_t ssrc,
const std::string& codec_mimetype,
- absl::optional<uint8_t> audio_level_dbov) {
+ std::optional<uint8_t> audio_level_dbov) {
{
MutexLock lock(&send_lock_);
if (short_circuit_) {
@@ -172,7 +170,7 @@
frame_type, payload_type, rtp_timestamp, payload_data, payload_size,
absolute_capture_timestamp_ms, ssrc,
/*csrcs=*/std::vector<uint32_t>(), codec_mimetype,
- /*sequence_number=*/absl::nullopt, audio_level_dbov));
+ /*sequence_number=*/std::nullopt, audio_level_dbov));
}
void ChannelSendFrameTransformerDelegate::OnTransformedFrame(
diff --git a/audio/channel_send_frame_transformer_delegate.h b/audio/channel_send_frame_transformer_delegate.h
index 5573052..628c403 100644
--- a/audio/channel_send_frame_transformer_delegate.h
+++ b/audio/channel_send_frame_transformer_delegate.h
@@ -37,7 +37,7 @@
rtc::ArrayView<const uint8_t> payload,
int64_t absolute_capture_timestamp_ms,
rtc::ArrayView<const uint32_t> csrcs,
- absl::optional<uint8_t> audio_level_dbov)>;
+ std::optional<uint8_t> audio_level_dbov)>;
ChannelSendFrameTransformerDelegate(
SendFrameCallback send_frame_callback,
rtc::scoped_refptr<FrameTransformerInterface> frame_transformer,
@@ -62,7 +62,7 @@
int64_t absolute_capture_timestamp_ms,
uint32_t ssrc,
const std::string& codec_mime_type,
- absl::optional<uint8_t> audio_level_dbov);
+ std::optional<uint8_t> audio_level_dbov);
// Implements TransformedFrameCallback. Can be called on any thread.
void OnTransformedFrame(
diff --git a/audio/channel_send_frame_transformer_delegate_unittest.cc b/audio/channel_send_frame_transformer_delegate_unittest.cc
index 0ed8677..a4ee9ff 100644
--- a/audio/channel_send_frame_transformer_delegate_unittest.cc
+++ b/audio/channel_send_frame_transformer_delegate_unittest.cc
@@ -12,11 +12,11 @@
#include <cstdint>
#include <memory>
+#include <optional>
#include <utility>
#include <vector>
#include "absl/memory/memory.h"
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/frame_transformer_interface.h"
#include "api/make_ref_counted.h"
@@ -54,14 +54,14 @@
rtc::ArrayView<const uint8_t> payload,
int64_t absolute_capture_timestamp_ms,
rtc::ArrayView<const uint32_t> csrcs,
- absl::optional<uint8_t> audio_level_dbov));
+ std::optional<uint8_t> audio_level_dbov));
ChannelSendFrameTransformerDelegate::SendFrameCallback callback() {
return [this](AudioFrameType frameType, uint8_t payloadType,
uint32_t rtp_timestamp, rtc::ArrayView<const uint8_t> payload,
int64_t absolute_capture_timestamp_ms,
rtc::ArrayView<const uint32_t> csrcs,
- absl::optional<uint8_t> audio_level_dbov) {
+ std::optional<uint8_t> audio_level_dbov) {
return SendFrame(frameType, payloadType, rtp_timestamp, payload,
absolute_capture_timestamp_ms, csrcs, audio_level_dbov);
};
@@ -70,7 +70,7 @@
std::unique_ptr<TransformableAudioFrameInterface> CreateMockReceiverFrame(
const std::vector<uint32_t>& csrcs,
- absl::optional<uint8_t> audio_level_dbov) {
+ std::optional<uint8_t> audio_level_dbov) {
std::unique_ptr<MockTransformableAudioFrame> mock_frame =
std::make_unique<NiceMock<MockTransformableAudioFrame>>();
rtc::ArrayView<const uint8_t> payload(mock_data);
@@ -190,12 +190,12 @@
ON_CALL(*mock_frame_transformer, Transform)
.WillByDefault([&](std::unique_ptr<TransformableFrameInterface> frame) {
callback->OnTransformedFrame(CreateMockReceiverFrame(
- csrcs, absl::optional<uint8_t>(audio_level_dbov)));
+ csrcs, std::optional<uint8_t>(audio_level_dbov)));
});
delegate->Transform(AudioFrameType::kEmptyFrame, 0, 0, mock_data,
sizeof(mock_data), 0,
/*ssrc=*/0, /*mimeType=*/"audio/opus",
- /*audio_level_dbov=*/absl::nullopt);
+ /*audio_level_dbov=*/std::nullopt);
channel_queue.WaitForPreviouslyPostedTasks();
}
@@ -236,7 +236,7 @@
const uint8_t data[] = {1, 2, 3, 4};
delegate->Transform(AudioFrameType::kEmptyFrame, 0, 0, data, sizeof(data), 0,
/*ssrc=*/0, /*mimeType=*/"audio/opus",
- /*audio_level_dbov=*/absl::nullopt);
+ /*audio_level_dbov=*/std::nullopt);
}
TEST(ChannelSendFrameTransformerDelegateTest,
@@ -258,7 +258,7 @@
TEST(ChannelSendFrameTransformerDelegateTest, CloningReceiverFrameWithCsrcs) {
std::unique_ptr<TransformableAudioFrameInterface> frame =
CreateMockReceiverFrame(/*csrcs=*/{123, 234, 345},
- absl::optional<uint8_t>(72));
+ std::optional<uint8_t>(72));
std::unique_ptr<TransformableAudioFrameInterface> cloned_frame =
CloneSenderAudioFrame(frame.get());
diff --git a/audio/channel_send_unittest.cc b/audio/channel_send_unittest.cc
index 3535a99..ac82fbd 100644
--- a/audio/channel_send_unittest.cc
+++ b/audio/channel_send_unittest.cc
@@ -164,7 +164,7 @@
.WillOnce(SaveArg<0>(&callback));
EXPECT_CALL(*mock_frame_transformer, UnregisterTransformedFrameCallback);
- absl::optional<uint32_t> sent_timestamp;
+ std::optional<uint32_t> sent_timestamp;
auto send_rtp = [&](rtc::ArrayView<const uint8_t> data,
const PacketOptions& options) {
RtpPacketReceived packet;
@@ -311,7 +311,7 @@
TEST_F(ChannelSendTest, NoUsedRateInitially) {
channel_->StartSend();
auto used_rate = channel_->GetUsedRate();
- EXPECT_EQ(used_rate, absl::nullopt);
+ EXPECT_EQ(used_rate, std::nullopt);
}
// Ensure that GetUsedRate returns value with one coded frame.
diff --git a/audio/mock_voe_channel_proxy.h b/audio/mock_voe_channel_proxy.h
index 607070a..1a62f02 100644
--- a/audio/mock_voe_channel_proxy.h
+++ b/audio/mock_voe_channel_proxy.h
@@ -75,18 +75,18 @@
SetEstimatedPlayoutNtpTimestampMs,
(int64_t ntp_timestamp_ms, int64_t time_ms),
(override));
- MOCK_METHOD(absl::optional<int64_t>,
+ MOCK_METHOD(std::optional<int64_t>,
GetCurrentEstimatedPlayoutNtpTimestampMs,
(int64_t now_ms),
(const, override));
- MOCK_METHOD(absl::optional<Syncable::Info>,
+ MOCK_METHOD(std::optional<Syncable::Info>,
GetSyncInfo,
(),
(const, override));
MOCK_METHOD(bool, SetMinimumPlayoutDelay, (int delay_ms), (override));
MOCK_METHOD(bool, SetBaseMinimumPlayoutDelayMs, (int delay_ms), (override));
MOCK_METHOD(int, GetBaseMinimumPlayoutDelayMs, (), (const, override));
- MOCK_METHOD((absl::optional<std::pair<int, SdpAudioFormat>>),
+ MOCK_METHOD((std::optional<std::pair<int, SdpAudioFormat>>),
GetReceiveCodec,
(),
(const, override));
@@ -182,7 +182,7 @@
SetEncoderToPacketizerFrameTransformer,
(rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer),
(override));
- MOCK_METHOD(absl::optional<DataRate>, GetUsedRate, (), (const, override));
+ MOCK_METHOD(std::optional<DataRate>, GetUsedRate, (), (const, override));
MOCK_METHOD(void,
RegisterPacketOverhead,
(int packet_byte_overhead),
diff --git a/audio/voip/BUILD.gn b/audio/voip/BUILD.gn
index a49e336..42fc665 100644
--- a/audio/voip/BUILD.gn
+++ b/audio/voip/BUILD.gn
@@ -75,7 +75,6 @@
"../../rtc_base:timeutils",
"../../rtc_base/synchronization:mutex",
"../utility:audio_frame_operations",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
diff --git a/audio/voip/audio_channel.h b/audio/voip/audio_channel.h
index d2452fd..5489109 100644
--- a/audio/voip/audio_channel.h
+++ b/audio/voip/audio_channel.h
@@ -59,7 +59,7 @@
std::unique_ptr<AudioEncoder> encoder) {
egress_->SetEncoder(payload_type, encoder_format, std::move(encoder));
}
- absl::optional<SdpAudioFormat> GetEncoderFormat() const {
+ std::optional<SdpAudioFormat> GetEncoderFormat() const {
return egress_->GetEncoderFormat();
}
void RegisterTelephoneEventType(int rtp_payload_type, int sample_rate_hz) {
diff --git a/audio/voip/audio_egress.h b/audio/voip/audio_egress.h
index aaaea08..7955fed 100644
--- a/audio/voip/audio_egress.h
+++ b/audio/voip/audio_egress.h
@@ -74,7 +74,7 @@
// Retrieve current encoder format info. This returns encoder format set
// by SetEncoder() and if encoder is not set, this will return nullopt.
- absl::optional<SdpAudioFormat> GetEncoderFormat() const {
+ std::optional<SdpAudioFormat> GetEncoderFormat() const {
MutexLock lock(&lock_);
return encoder_format_;
}
@@ -119,7 +119,7 @@
mutable Mutex lock_;
// Current encoder format selected by caller.
- absl::optional<SdpAudioFormat> encoder_format_ RTC_GUARDED_BY(lock_);
+ std::optional<SdpAudioFormat> encoder_format_ RTC_GUARDED_BY(lock_);
// Synchronization is handled internally by RtpRtcp.
RtpRtcpInterface* const rtp_rtcp_;
diff --git a/audio/voip/audio_ingress.cc b/audio/voip/audio_ingress.cc
index 9809c6b..50332c1 100644
--- a/audio/voip/audio_ingress.cc
+++ b/audio/voip/audio_ingress.cc
@@ -103,7 +103,7 @@
}
// For clock rate, default to the playout sampling rate if we haven't
// received any packets yet.
- absl::optional<std::pair<int, SdpAudioFormat>> decoder =
+ std::optional<std::pair<int, SdpAudioFormat>> decoder =
acm_receiver_.LastDecoder();
int clock_rate = decoder ? decoder->second.clockrate_hz
: acm_receiver_.last_output_sample_rate_hz();
@@ -215,13 +215,13 @@
// Deliver RTCP packet to RTP/RTCP module for parsing and processing.
rtp_rtcp_->IncomingRtcpPacket(rtcp_packet);
- absl::optional<TimeDelta> rtt = rtp_rtcp_->LastRtt();
+ std::optional<TimeDelta> rtt = rtp_rtcp_->LastRtt();
if (!rtt.has_value()) {
// Waiting for valid RTT.
return;
}
- absl::optional<RtpRtcpInterface::SenderReportStats> last_sr =
+ std::optional<RtpRtcpInterface::SenderReportStats> last_sr =
rtp_rtcp_->GetSenderReportStats();
if (!last_sr.has_value()) {
// Waiting for RTCP.
@@ -240,7 +240,7 @@
// Get clockrate for current decoder ahead of jitter calculation.
uint32_t clockrate_hz = 0;
- absl::optional<std::pair<int, SdpAudioFormat>> decoder =
+ std::optional<std::pair<int, SdpAudioFormat>> decoder =
acm_receiver_.LastDecoder();
if (decoder) {
clockrate_hz = decoder->second.clockrate_hz;
diff --git a/audio/voip/audio_ingress.h b/audio/voip/audio_ingress.h
index daa21cb..9651871 100644
--- a/audio/voip/audio_ingress.h
+++ b/audio/voip/audio_ingress.h
@@ -15,9 +15,9 @@
#include <atomic>
#include <map>
#include <memory>
+#include <optional>
#include <utility>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/audio/audio_mixer.h"
#include "api/environment/environment.h"
diff --git a/audio/voip/test/audio_channel_unittest.cc b/audio/voip/test/audio_channel_unittest.cc
index 0d4c05f..90407a7 100644
--- a/audio/voip/test/audio_channel_unittest.cc
+++ b/audio/voip/test/audio_channel_unittest.cc
@@ -163,7 +163,7 @@
audio_mixer_->Mix(/*number_of_channels=*/1, &audio_frame);
audio_mixer_->Mix(/*number_of_channels=*/1, &audio_frame);
- absl::optional<IngressStatistics> ingress_stats =
+ std::optional<IngressStatistics> ingress_stats =
audio_channel_->GetIngressStatistics();
EXPECT_TRUE(ingress_stats);
EXPECT_EQ(ingress_stats->neteq_stats.total_samples_received, 160ULL);
@@ -254,7 +254,7 @@
// in loop_rtcp above.
audio_channel_->SendRTCPReportForTesting(kRtcpSr);
- absl::optional<ChannelStatistics> channel_stats =
+ std::optional<ChannelStatistics> channel_stats =
audio_channel_->GetChannelStatistics();
EXPECT_TRUE(channel_stats);
@@ -336,7 +336,7 @@
send_recv_rtcp(audio_channel_, ac_2);
send_recv_rtcp(ac_2, audio_channel_);
- absl::optional<ChannelStatistics> channel_stats =
+ std::optional<ChannelStatistics> channel_stats =
audio_channel_->GetChannelStatistics();
ASSERT_TRUE(channel_stats);
EXPECT_EQ(channel_stats->remote_ssrc, kAc2Ssrc);
diff --git a/audio/voip/test/audio_egress_unittest.cc b/audio/voip/test/audio_egress_unittest.cc
index a5b098e..46b351b 100644
--- a/audio/voip/test/audio_egress_unittest.cc
+++ b/audio/voip/test/audio_egress_unittest.cc
@@ -226,7 +226,7 @@
}
TEST_F(AudioEgressTest, ChangeEncoderFromPcmuToOpus) {
- absl::optional<SdpAudioFormat> pcmu = egress_->GetEncoderFormat();
+ std::optional<SdpAudioFormat> pcmu = egress_->GetEncoderFormat();
EXPECT_TRUE(pcmu);
EXPECT_EQ(pcmu->clockrate_hz, kPcmuFormat.clockrate_hz);
EXPECT_EQ(pcmu->num_channels, kPcmuFormat.num_channels);
@@ -238,7 +238,7 @@
encoder_factory_->Create(env_, kOpusFormat,
{.payload_type = kOpusPayload}));
- absl::optional<SdpAudioFormat> opus = egress_->GetEncoderFormat();
+ std::optional<SdpAudioFormat> opus = egress_->GetEncoderFormat();
EXPECT_TRUE(opus);
EXPECT_EQ(opus->clockrate_hz, kOpusFormat.clockrate_hz);
EXPECT_EQ(opus->num_channels, kOpusFormat.num_channels);
diff --git a/audio/voip/voip_core.cc b/audio/voip/voip_core.cc
index e4aec20..885e9af 100644
--- a/audio/voip/voip_core.cc
+++ b/audio/voip/voip_core.cc
@@ -122,7 +122,7 @@
}
ChannelId VoipCore::CreateChannel(Transport* transport,
- absl::optional<uint32_t> local_ssrc) {
+ std::optional<uint32_t> local_ssrc) {
ChannelId channel_id;
// Set local ssrc to random if not set by caller.
diff --git a/audio/voip/voip_core.h b/audio/voip/voip_core.h
index 3ee1e8c..5d88ba0 100644
--- a/audio/voip/voip_core.h
+++ b/audio/voip/voip_core.h
@@ -69,7 +69,7 @@
// Implements VoipBase interfaces.
ChannelId CreateChannel(Transport* transport,
- absl::optional<uint32_t> local_ssrc) override;
+ std::optional<uint32_t> local_ssrc) override;
VoipResult ReleaseChannel(ChannelId channel_id) override;
VoipResult StartSend(ChannelId channel_id) override;
VoipResult StopSend(ChannelId channel_id) override;
diff --git a/call/BUILD.gn b/call/BUILD.gn
index cc81b38..fd387d9 100644
--- a/call/BUILD.gn
+++ b/call/BUILD.gn
@@ -78,7 +78,6 @@
"../rtc_base/network:sent_packet",
"//third_party/abseil-cpp/absl/functional:any_invocable",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -126,7 +125,6 @@
"../rtc_base:stringutils",
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -220,7 +218,6 @@
"//third_party/abseil-cpp/absl/container:inlined_vector",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
"//third_party/abseil-cpp/absl/types:variant",
]
}
@@ -232,7 +229,6 @@
]
deps = [
":rtp_interfaces",
- "//third_party/abseil-cpp/absl/types:optional",
# For api/bitrate_constraints.h
"../api:libjingle_peerconnection_api",
@@ -334,7 +330,6 @@
"adaptation:resource_adaptation",
"//third_party/abseil-cpp/absl/functional:bind_front",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -381,7 +376,6 @@
"../rtc_base:checks",
"../rtc_base:stringutils",
"../video/config:encoder_config",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -512,7 +506,6 @@
"//third_party/abseil-cpp/absl/container:inlined_vector",
"//third_party/abseil-cpp/absl/functional:any_invocable",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
"//third_party/abseil-cpp/absl/types:variant",
]
}
diff --git a/call/adaptation/BUILD.gn b/call/adaptation/BUILD.gn
index 5dde21d..4743e94 100644
--- a/call/adaptation/BUILD.gn
+++ b/call/adaptation/BUILD.gn
@@ -58,7 +58,6 @@
"../../video/config:encoder_config",
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
"//third_party/abseil-cpp/absl/types:variant",
]
}
@@ -95,7 +94,6 @@
"../../test:scoped_key_value_config",
"../../test:test_support",
"../../video/config:encoder_config",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -125,7 +123,6 @@
"../../video:video_stream_encoder_interface",
"../../video/config:encoder_config",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
}
diff --git a/call/adaptation/encoder_settings.cc b/call/adaptation/encoder_settings.cc
index c894e83..7c00dce 100644
--- a/call/adaptation/encoder_settings.cc
+++ b/call/adaptation/encoder_settings.cc
@@ -46,7 +46,7 @@
}
VideoCodecType GetVideoCodecTypeOrGeneric(
- const absl::optional<EncoderSettings>& settings) {
+ const std::optional<EncoderSettings>& settings) {
return settings.has_value() ? settings->encoder_config().codec_type
: kVideoCodecGeneric;
}
diff --git a/call/adaptation/encoder_settings.h b/call/adaptation/encoder_settings.h
index 30ce0a0..2d5e9bc 100644
--- a/call/adaptation/encoder_settings.h
+++ b/call/adaptation/encoder_settings.h
@@ -11,7 +11,8 @@
#ifndef CALL_ADAPTATION_ENCODER_SETTINGS_H_
#define CALL_ADAPTATION_ENCODER_SETTINGS_H_
-#include "absl/types/optional.h"
+#include <optional>
+
#include "api/video_codecs/video_codec.h"
#include "api/video_codecs/video_encoder.h"
#include "video/config/video_encoder_config.h"
@@ -41,7 +42,7 @@
};
VideoCodecType GetVideoCodecTypeOrGeneric(
- const absl::optional<EncoderSettings>& settings);
+ const std::optional<EncoderSettings>& settings);
} // namespace webrtc
diff --git a/call/adaptation/resource_adaptation_processor.h b/call/adaptation/resource_adaptation_processor.h
index c2d9c3e..a4cc1ca 100644
--- a/call/adaptation/resource_adaptation_processor.h
+++ b/call/adaptation/resource_adaptation_processor.h
@@ -13,12 +13,12 @@
#include <map>
#include <memory>
+#include <optional>
#include <string>
#include <utility>
#include <vector>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/adaptation/resource.h"
#include "api/rtp_parameters.h"
#include "api/scoped_refptr.h"
diff --git a/call/adaptation/resource_adaptation_processor_interface.h b/call/adaptation/resource_adaptation_processor_interface.h
index 4729488..b5c417a 100644
--- a/call/adaptation/resource_adaptation_processor_interface.h
+++ b/call/adaptation/resource_adaptation_processor_interface.h
@@ -12,9 +12,9 @@
#define CALL_ADAPTATION_RESOURCE_ADAPTATION_PROCESSOR_INTERFACE_H_
#include <map>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/adaptation/resource.h"
#include "api/rtp_parameters.h"
#include "api/scoped_refptr.h"
diff --git a/call/adaptation/test/fake_resource.h b/call/adaptation/test/fake_resource.h
index 1119a96..0e02b0c 100644
--- a/call/adaptation/test/fake_resource.h
+++ b/call/adaptation/test/fake_resource.h
@@ -11,11 +11,11 @@
#ifndef CALL_ADAPTATION_TEST_FAKE_RESOURCE_H_
#define CALL_ADAPTATION_TEST_FAKE_RESOURCE_H_
+#include <optional>
#include <string>
#include <vector>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/adaptation/resource.h"
#include "api/scoped_refptr.h"
diff --git a/call/adaptation/video_source_restrictions.cc b/call/adaptation/video_source_restrictions.cc
index 719bc53..8ca40d4 100644
--- a/call/adaptation/video_source_restrictions.cc
+++ b/call/adaptation/video_source_restrictions.cc
@@ -19,14 +19,14 @@
namespace webrtc {
VideoSourceRestrictions::VideoSourceRestrictions()
- : max_pixels_per_frame_(absl::nullopt),
- target_pixels_per_frame_(absl::nullopt),
- max_frame_rate_(absl::nullopt) {}
+ : max_pixels_per_frame_(std::nullopt),
+ target_pixels_per_frame_(std::nullopt),
+ max_frame_rate_(std::nullopt) {}
VideoSourceRestrictions::VideoSourceRestrictions(
- absl::optional<size_t> max_pixels_per_frame,
- absl::optional<size_t> target_pixels_per_frame,
- absl::optional<double> max_frame_rate)
+ std::optional<size_t> max_pixels_per_frame,
+ std::optional<size_t> target_pixels_per_frame,
+ std::optional<double> max_frame_rate)
: max_pixels_per_frame_(std::move(max_pixels_per_frame)),
target_pixels_per_frame_(std::move(target_pixels_per_frame)),
max_frame_rate_(std::move(max_frame_rate)) {
@@ -51,32 +51,32 @@
return ss.Release();
}
-const absl::optional<size_t>& VideoSourceRestrictions::max_pixels_per_frame()
+const std::optional<size_t>& VideoSourceRestrictions::max_pixels_per_frame()
const {
return max_pixels_per_frame_;
}
-const absl::optional<size_t>& VideoSourceRestrictions::target_pixels_per_frame()
+const std::optional<size_t>& VideoSourceRestrictions::target_pixels_per_frame()
const {
return target_pixels_per_frame_;
}
-const absl::optional<double>& VideoSourceRestrictions::max_frame_rate() const {
+const std::optional<double>& VideoSourceRestrictions::max_frame_rate() const {
return max_frame_rate_;
}
void VideoSourceRestrictions::set_max_pixels_per_frame(
- absl::optional<size_t> max_pixels_per_frame) {
+ std::optional<size_t> max_pixels_per_frame) {
max_pixels_per_frame_ = std::move(max_pixels_per_frame);
}
void VideoSourceRestrictions::set_target_pixels_per_frame(
- absl::optional<size_t> target_pixels_per_frame) {
+ std::optional<size_t> target_pixels_per_frame) {
target_pixels_per_frame_ = std::move(target_pixels_per_frame);
}
void VideoSourceRestrictions::set_max_frame_rate(
- absl::optional<double> max_frame_rate) {
+ std::optional<double> max_frame_rate) {
max_frame_rate_ = std::move(max_frame_rate);
}
diff --git a/call/adaptation/video_source_restrictions.h b/call/adaptation/video_source_restrictions.h
index be8520a..6e1b6c6 100644
--- a/call/adaptation/video_source_restrictions.h
+++ b/call/adaptation/video_source_restrictions.h
@@ -11,11 +11,10 @@
#ifndef CALL_ADAPTATION_VIDEO_SOURCE_RESTRICTIONS_H_
#define CALL_ADAPTATION_VIDEO_SOURCE_RESTRICTIONS_H_
+#include <optional>
#include <string>
#include <utility>
-#include "absl/types/optional.h"
-
namespace webrtc {
// Describes optional restrictions to the resolution and frame rate of a video
@@ -26,9 +25,9 @@
VideoSourceRestrictions();
// All values must be positive or nullopt.
// TODO(hbos): Support expressing "disable this stream"?
- VideoSourceRestrictions(absl::optional<size_t> max_pixels_per_frame,
- absl::optional<size_t> target_pixels_per_frame,
- absl::optional<double> max_frame_rate);
+ VideoSourceRestrictions(std::optional<size_t> max_pixels_per_frame,
+ std::optional<size_t> target_pixels_per_frame,
+ std::optional<double> max_frame_rate);
bool operator==(const VideoSourceRestrictions& rhs) const {
return max_pixels_per_frame_ == rhs.max_pixels_per_frame_ &&
@@ -43,7 +42,7 @@
// The source must produce a resolution less than or equal to
// max_pixels_per_frame().
- const absl::optional<size_t>& max_pixels_per_frame() const;
+ const std::optional<size_t>& max_pixels_per_frame() const;
// The source should produce a resolution as close to the
// target_pixels_per_frame() as possible, provided this does not exceed
// max_pixels_per_frame().
@@ -52,13 +51,13 @@
// the camera in the smallest resolution that is greater than or equal to the
// target and scale it down to the target if it is greater. Is this an
// accurate description of what this does today, or do we do something else?
- const absl::optional<size_t>& target_pixels_per_frame() const;
- const absl::optional<double>& max_frame_rate() const;
+ const std::optional<size_t>& target_pixels_per_frame() const;
+ const std::optional<double>& max_frame_rate() const;
- void set_max_pixels_per_frame(absl::optional<size_t> max_pixels_per_frame);
+ void set_max_pixels_per_frame(std::optional<size_t> max_pixels_per_frame);
void set_target_pixels_per_frame(
- absl::optional<size_t> target_pixels_per_frame);
- void set_max_frame_rate(absl::optional<double> max_frame_rate);
+ std::optional<size_t> target_pixels_per_frame);
+ void set_max_frame_rate(std::optional<double> max_frame_rate);
// Update `this` with min(`this`, `other`).
void UpdateMin(const VideoSourceRestrictions& other);
@@ -66,9 +65,9 @@
private:
// These map to rtc::VideoSinkWants's `max_pixel_count` and
// `target_pixel_count`.
- absl::optional<size_t> max_pixels_per_frame_;
- absl::optional<size_t> target_pixels_per_frame_;
- absl::optional<double> max_frame_rate_;
+ std::optional<size_t> max_pixels_per_frame_;
+ std::optional<size_t> target_pixels_per_frame_;
+ std::optional<double> max_frame_rate_;
};
bool DidRestrictionsIncrease(VideoSourceRestrictions before,
diff --git a/call/adaptation/video_source_restrictions_unittest.cc b/call/adaptation/video_source_restrictions_unittest.cc
index 8c1ae4c..1205332 100644
--- a/call/adaptation/video_source_restrictions_unittest.cc
+++ b/call/adaptation/video_source_restrictions_unittest.cc
@@ -19,19 +19,19 @@
const size_t kHdPixels = 1280 * 720;
const VideoSourceRestrictions kUnlimited;
-const VideoSourceRestrictions k15fps(absl::nullopt, absl::nullopt, 15.0);
-const VideoSourceRestrictions kHd(kHdPixels, kHdPixels, absl::nullopt);
+const VideoSourceRestrictions k15fps(std::nullopt, std::nullopt, 15.0);
+const VideoSourceRestrictions kHd(kHdPixels, kHdPixels, std::nullopt);
const VideoSourceRestrictions kHd15fps(kHdPixels, kHdPixels, 15.0);
const VideoSourceRestrictions kVga7fps(kHdPixels / 2, kHdPixels / 2, 7.0);
VideoSourceRestrictions RestrictionsFromMaxPixelsPerFrame(
size_t max_pixels_per_frame) {
- return VideoSourceRestrictions(max_pixels_per_frame, absl::nullopt,
- absl::nullopt);
+ return VideoSourceRestrictions(max_pixels_per_frame, std::nullopt,
+ std::nullopt);
}
VideoSourceRestrictions RestrictionsFromMaxFrameRate(double max_frame_rate) {
- return VideoSourceRestrictions(absl::nullopt, absl::nullopt, max_frame_rate);
+ return VideoSourceRestrictions(std::nullopt, std::nullopt, max_frame_rate);
}
} // namespace
diff --git a/call/adaptation/video_stream_adapter.cc b/call/adaptation/video_stream_adapter.cc
index 5a970fb..eb949c9 100644
--- a/call/adaptation/video_stream_adapter.cc
+++ b/call/adaptation/video_stream_adapter.cc
@@ -12,9 +12,9 @@
#include <algorithm>
#include <limits>
+#include <optional>
#include <utility>
-#include "absl/types/optional.h"
#include "absl/types/variant.h"
#include "api/sequence_checker.h"
#include "api/video/video_adaptation_counters.h"
@@ -38,7 +38,7 @@
RTC_DCHECK(fps != std::numeric_limits<int>::max());
return (fps * 2) / 3;
}
-// TODO(hbos): Use absl::optional<> instead?
+// TODO(hbos): Use std::optional<> instead?
int GetHigherFrameRateThan(int fps) {
return fps != std::numeric_limits<int>::max()
? (fps * 3) / 2
@@ -118,16 +118,16 @@
case DegradationPreference::BALANCED:
break;
case DegradationPreference::MAINTAIN_FRAMERATE:
- source_restrictions.set_max_frame_rate(absl::nullopt);
+ source_restrictions.set_max_frame_rate(std::nullopt);
break;
case DegradationPreference::MAINTAIN_RESOLUTION:
- source_restrictions.set_max_pixels_per_frame(absl::nullopt);
- source_restrictions.set_target_pixels_per_frame(absl::nullopt);
+ source_restrictions.set_max_pixels_per_frame(std::nullopt);
+ source_restrictions.set_target_pixels_per_frame(std::nullopt);
break;
case DegradationPreference::DISABLED:
- source_restrictions.set_max_pixels_per_frame(absl::nullopt);
- source_restrictions.set_target_pixels_per_frame(absl::nullopt);
- source_restrictions.set_max_frame_rate(absl::nullopt);
+ source_restrictions.set_max_pixels_per_frame(std::nullopt);
+ source_restrictions.set_target_pixels_per_frame(std::nullopt);
+ source_restrictions.set_max_frame_rate(std::nullopt);
}
return source_restrictions;
}
@@ -145,7 +145,7 @@
return (pixel_count * 3) / 5;
}
-// TODO(hbos): Use absl::optional<> instead?
+// TODO(hbos): Use std::optional<> instead?
int GetHigherResolutionThan(int pixel_count) {
return pixel_count != std::numeric_limits<int>::max()
? (pixel_count * 5) / 3
@@ -211,7 +211,7 @@
balanced_settings_(field_trials),
adaptation_validation_id_(0),
degradation_preference_(DegradationPreference::DISABLED),
- awaiting_frame_size_change_(absl::nullopt) {
+ awaiting_frame_size_change_(std::nullopt) {
sequence_checker_.Detach();
RTC_DCHECK(input_state_provider_);
RTC_DCHECK(encoder_stats_observer_);
@@ -240,7 +240,7 @@
++adaptation_validation_id_;
current_restrictions_ = {VideoSourceRestrictions(),
VideoAdaptationCounters()};
- awaiting_frame_size_change_ = absl::nullopt;
+ awaiting_frame_size_change_ = std::nullopt;
BroadcastVideoRestrictionsUpdate(input_state_provider_->InputState(),
nullptr);
}
@@ -420,7 +420,7 @@
RTC_DCHECK_EQ(degradation_preference_, DegradationPreference::BALANCED);
int frame_size_pixels = input_state.single_active_stream_pixels().value_or(
input_state.frame_size_pixels().value());
- absl::optional<int> min_fps_diff =
+ std::optional<int> min_fps_diff =
balanced_settings_.MinFpsDiff(frame_size_pixels);
if (current_restrictions_.counters.fps_adaptations <
restrictions.counters.fps_adaptations &&
@@ -492,9 +492,9 @@
RTC_LOG(LS_INFO) << "Scaling down resolution, max pixels: " << target_pixels;
new_restrictions.restrictions.set_max_pixels_per_frame(
target_pixels != std::numeric_limits<int>::max()
- ? absl::optional<size_t>(target_pixels)
- : absl::nullopt);
- new_restrictions.restrictions.set_target_pixels_per_frame(absl::nullopt);
+ ? std::optional<size_t>(target_pixels)
+ : std::nullopt);
+ new_restrictions.restrictions.set_target_pixels_per_frame(std::nullopt);
++new_restrictions.counters.resolution_adaptations;
return new_restrictions;
}
@@ -523,8 +523,8 @@
RTC_LOG(LS_INFO) << "Scaling down framerate: " << max_frame_rate;
new_restrictions.restrictions.set_max_frame_rate(
max_frame_rate != std::numeric_limits<int>::max()
- ? absl::optional<double>(max_frame_rate)
- : absl::nullopt);
+ ? std::optional<double>(max_frame_rate)
+ : std::nullopt);
++new_restrictions.counters.fps_adaptations;
return new_restrictions;
}
@@ -548,12 +548,12 @@
<< max_pixels_wanted;
new_restrictions.restrictions.set_max_pixels_per_frame(
max_pixels_wanted != std::numeric_limits<int>::max()
- ? absl::optional<size_t>(max_pixels_wanted)
- : absl::nullopt);
+ ? std::optional<size_t>(max_pixels_wanted)
+ : std::nullopt);
new_restrictions.restrictions.set_target_pixels_per_frame(
max_pixels_wanted != std::numeric_limits<int>::max()
- ? absl::optional<size_t>(target_pixels)
- : absl::nullopt);
+ ? std::optional<size_t>(target_pixels)
+ : std::nullopt);
--new_restrictions.counters.resolution_adaptations;
RTC_DCHECK_GE(new_restrictions.counters.resolution_adaptations, 0);
return new_restrictions;
@@ -601,8 +601,8 @@
RestrictionsWithCounters new_restrictions = current_restrictions;
new_restrictions.restrictions.set_max_frame_rate(
max_frame_rate != std::numeric_limits<int>::max()
- ? absl::optional<double>(max_frame_rate)
- : absl::nullopt);
+ ? std::optional<double>(max_frame_rate)
+ : std::nullopt);
--new_restrictions.counters.fps_adaptations;
RTC_DCHECK_GE(new_restrictions.counters.fps_adaptations, 0);
return new_restrictions;
@@ -669,7 +669,7 @@
awaiting_frame_size_change_.emplace(
false, adaptation.input_state().frame_size_pixels().value());
} else {
- awaiting_frame_size_change_ = absl::nullopt;
+ awaiting_frame_size_change_ = std::nullopt;
}
current_restrictions_ = {adaptation.restrictions(), adaptation.counters()};
BroadcastVideoRestrictionsUpdate(adaptation.input_state(), resource);
@@ -717,10 +717,10 @@
: pixels_increased(pixels_increased),
frame_size_pixels(frame_size_pixels) {}
-absl::optional<uint32_t> VideoStreamAdapter::GetSingleActiveLayerPixels(
+std::optional<uint32_t> VideoStreamAdapter::GetSingleActiveLayerPixels(
const VideoCodec& codec) {
int num_active = 0;
- absl::optional<uint32_t> pixels;
+ std::optional<uint32_t> pixels;
if (codec.codecType == VideoCodecType::kVideoCodecAV1 &&
codec.GetScalabilityMode().has_value()) {
for (int i = 0;
@@ -747,7 +747,7 @@
}
}
}
- return (num_active > 1) ? absl::nullopt : pixels;
+ return (num_active > 1) ? std::nullopt : pixels;
}
} // namespace webrtc
diff --git a/call/adaptation/video_stream_adapter.h b/call/adaptation/video_stream_adapter.h
index 5c17417..fcf0a6d 100644
--- a/call/adaptation/video_stream_adapter.h
+++ b/call/adaptation/video_stream_adapter.h
@@ -12,10 +12,10 @@
#define CALL_ADAPTATION_VIDEO_STREAM_ADAPTER_H_
#include <memory>
+#include <optional>
#include <utility>
#include <vector>
-#include "absl/types/optional.h"
#include "absl/types/variant.h"
#include "api/adaptation/resource.h"
#include "api/field_trials_view.h"
@@ -165,7 +165,7 @@
VideoAdaptationCounters counters;
};
- static absl::optional<uint32_t> GetSingleActiveLayerPixels(
+ static std::optional<uint32_t> GetSingleActiveLayerPixels(
const VideoCodec& codec);
private:
@@ -249,7 +249,7 @@
const bool pixels_increased;
const int frame_size_pixels;
};
- absl::optional<AwaitingFrameSizeChange> awaiting_frame_size_change_
+ std::optional<AwaitingFrameSizeChange> awaiting_frame_size_change_
RTC_GUARDED_BY(&sequence_checker_);
// The previous restrictions value. Starts as unrestricted.
VideoSourceRestrictions last_video_source_restrictions_
diff --git a/call/adaptation/video_stream_adapter_unittest.cc b/call/adaptation/video_stream_adapter_unittest.cc
index d4bc650..162adf9 100644
--- a/call/adaptation/video_stream_adapter_unittest.cc
+++ b/call/adaptation/video_stream_adapter_unittest.cc
@@ -10,10 +10,10 @@
#include "call/adaptation/video_stream_adapter.h"
+#include <optional>
#include <string>
#include <utility>
-#include "absl/types/optional.h"
#include "api/scoped_refptr.h"
#include "api/video/video_adaptation_reason.h"
#include "api/video_codecs/video_codec.h"
@@ -180,9 +180,9 @@
adapter_.ApplyAdaptation(adaptation, nullptr);
EXPECT_EQ(static_cast<size_t>((kInputPixels * 3) / 5),
adapter_.source_restrictions().max_pixels_per_frame());
- EXPECT_EQ(absl::nullopt,
+ EXPECT_EQ(std::nullopt,
adapter_.source_restrictions().target_pixels_per_frame());
- EXPECT_EQ(absl::nullopt, adapter_.source_restrictions().max_frame_rate());
+ EXPECT_EQ(std::nullopt, adapter_.source_restrictions().max_frame_rate());
EXPECT_EQ(1, adapter_.adaptation_counters().resolution_adaptations);
}
@@ -219,7 +219,7 @@
adapter_.source_restrictions().max_pixels_per_frame());
EXPECT_EQ(static_cast<size_t>(target),
adapter_.source_restrictions().target_pixels_per_frame());
- EXPECT_EQ(absl::nullopt, adapter_.source_restrictions().max_frame_rate());
+ EXPECT_EQ(std::nullopt, adapter_.source_restrictions().max_frame_rate());
EXPECT_EQ(1, adapter_.adaptation_counters().resolution_adaptations);
}
@@ -247,9 +247,9 @@
Adaptation adaptation = adapter_.GetAdaptationDown();
EXPECT_EQ(Adaptation::Status::kValid, adaptation.status());
adapter_.ApplyAdaptation(adaptation, nullptr);
- EXPECT_EQ(absl::nullopt,
+ EXPECT_EQ(std::nullopt,
adapter_.source_restrictions().max_pixels_per_frame());
- EXPECT_EQ(absl::nullopt,
+ EXPECT_EQ(std::nullopt,
adapter_.source_restrictions().target_pixels_per_frame());
EXPECT_EQ(static_cast<double>((kInputFps * 2) / 3),
adapter_.source_restrictions().max_frame_rate());
@@ -286,9 +286,9 @@
Adaptation adaptation = adapter_.GetAdaptationUp();
EXPECT_EQ(Adaptation::Status::kValid, adaptation.status());
fake_stream.ApplyAdaptation(adaptation);
- EXPECT_EQ(absl::nullopt,
+ EXPECT_EQ(std::nullopt,
adapter_.source_restrictions().max_pixels_per_frame());
- EXPECT_EQ(absl::nullopt,
+ EXPECT_EQ(std::nullopt,
adapter_.source_restrictions().target_pixels_per_frame());
EXPECT_EQ(static_cast<double>((input_fps * 3) / 2),
adapter_.source_restrictions().max_frame_rate());
@@ -321,9 +321,9 @@
Adaptation adaptation = adapter_.GetAdaptationDown();
EXPECT_EQ(Adaptation::Status::kValid, adaptation.status());
adapter_.ApplyAdaptation(adaptation, nullptr);
- EXPECT_EQ(absl::nullopt,
+ EXPECT_EQ(std::nullopt,
adapter_.source_restrictions().max_pixels_per_frame());
- EXPECT_EQ(absl::nullopt,
+ EXPECT_EQ(std::nullopt,
adapter_.source_restrictions().target_pixels_per_frame());
EXPECT_EQ(static_cast<double>(kBalancedMediumFrameRateFps),
adapter_.source_restrictions().max_frame_rate());
@@ -349,9 +349,9 @@
EXPECT_EQ(Adaptation::Status::kValid, adaptation.status());
fake_stream.ApplyAdaptation(adaptation);
}
- EXPECT_EQ(absl::nullopt,
+ EXPECT_EQ(std::nullopt,
adapter_.source_restrictions().max_pixels_per_frame());
- EXPECT_EQ(absl::nullopt,
+ EXPECT_EQ(std::nullopt,
adapter_.source_restrictions().target_pixels_per_frame());
EXPECT_EQ(static_cast<double>(kBalancedHighFrameRateFps),
adapter_.source_restrictions().max_frame_rate());
@@ -368,7 +368,7 @@
static_cast<size_t>((kBalancedHighResolutionPixels * 3) / 5);
EXPECT_EQ(kReducedPixelsFirstStep,
adapter_.source_restrictions().max_pixels_per_frame());
- EXPECT_EQ(absl::nullopt,
+ EXPECT_EQ(std::nullopt,
adapter_.source_restrictions().target_pixels_per_frame());
EXPECT_EQ(static_cast<double>(kBalancedHighFrameRateFps),
adapter_.source_restrictions().max_frame_rate());
@@ -387,7 +387,7 @@
}
EXPECT_EQ(kReducedPixelsSecondStep,
adapter_.source_restrictions().max_pixels_per_frame());
- EXPECT_EQ(absl::nullopt,
+ EXPECT_EQ(std::nullopt,
adapter_.source_restrictions().target_pixels_per_frame());
EXPECT_EQ(static_cast<double>(kBalancedHighFrameRateFps),
adapter_.source_restrictions().max_frame_rate());
@@ -482,7 +482,7 @@
Adaptation adaptation = adapter_.GetAdaptationUp();
EXPECT_EQ(Adaptation::Status::kValid, adaptation.status());
fake_stream.ApplyAdaptation(adaptation);
- EXPECT_EQ(absl::nullopt, adapter_.source_restrictions().max_frame_rate());
+ EXPECT_EQ(std::nullopt, adapter_.source_restrictions().max_frame_rate());
EXPECT_EQ(2, adapter_.adaptation_counters().resolution_adaptations);
EXPECT_EQ(0, adapter_.adaptation_counters().fps_adaptations);
}
diff --git a/call/adaptation/video_stream_input_state.cc b/call/adaptation/video_stream_input_state.cc
index 9c0d475..0a12d2e 100644
--- a/call/adaptation/video_stream_input_state.cc
+++ b/call/adaptation/video_stream_input_state.cc
@@ -16,18 +16,18 @@
VideoStreamInputState::VideoStreamInputState()
: has_input_(false),
- frame_size_pixels_(absl::nullopt),
+ frame_size_pixels_(std::nullopt),
frames_per_second_(0),
video_codec_type_(VideoCodecType::kVideoCodecGeneric),
min_pixels_per_frame_(kDefaultMinPixelsPerFrame),
- single_active_stream_pixels_(absl::nullopt) {}
+ single_active_stream_pixels_(std::nullopt) {}
void VideoStreamInputState::set_has_input(bool has_input) {
has_input_ = has_input;
}
void VideoStreamInputState::set_frame_size_pixels(
- absl::optional<int> frame_size_pixels) {
+ std::optional<int> frame_size_pixels) {
frame_size_pixels_ = frame_size_pixels;
}
@@ -45,7 +45,7 @@
}
void VideoStreamInputState::set_single_active_stream_pixels(
- absl::optional<int> single_active_stream_pixels) {
+ std::optional<int> single_active_stream_pixels) {
single_active_stream_pixels_ = single_active_stream_pixels;
}
@@ -53,7 +53,7 @@
return has_input_;
}
-absl::optional<int> VideoStreamInputState::frame_size_pixels() const {
+std::optional<int> VideoStreamInputState::frame_size_pixels() const {
return frame_size_pixels_;
}
@@ -69,7 +69,7 @@
return min_pixels_per_frame_;
}
-absl::optional<int> VideoStreamInputState::single_active_stream_pixels() const {
+std::optional<int> VideoStreamInputState::single_active_stream_pixels() const {
return single_active_stream_pixels_;
}
diff --git a/call/adaptation/video_stream_input_state.h b/call/adaptation/video_stream_input_state.h
index 191e223..744788d 100644
--- a/call/adaptation/video_stream_input_state.h
+++ b/call/adaptation/video_stream_input_state.h
@@ -11,7 +11,8 @@
#ifndef CALL_ADAPTATION_VIDEO_STREAM_INPUT_STATE_H_
#define CALL_ADAPTATION_VIDEO_STREAM_INPUT_STATE_H_
-#include "absl/types/optional.h"
+#include <optional>
+
#include "api/video/video_codec_type.h"
namespace webrtc {
@@ -23,29 +24,29 @@
VideoStreamInputState();
void set_has_input(bool has_input);
- void set_frame_size_pixels(absl::optional<int> frame_size_pixels);
+ void set_frame_size_pixels(std::optional<int> frame_size_pixels);
void set_frames_per_second(int frames_per_second);
void set_video_codec_type(VideoCodecType video_codec_type);
void set_min_pixels_per_frame(int min_pixels_per_frame);
void set_single_active_stream_pixels(
- absl::optional<int> single_active_stream_pixels);
+ std::optional<int> single_active_stream_pixels);
bool has_input() const;
- absl::optional<int> frame_size_pixels() const;
+ std::optional<int> frame_size_pixels() const;
int frames_per_second() const;
VideoCodecType video_codec_type() const;
int min_pixels_per_frame() const;
- absl::optional<int> single_active_stream_pixels() const;
+ std::optional<int> single_active_stream_pixels() const;
bool HasInputFrameSizeAndFramesPerSecond() const;
private:
bool has_input_;
- absl::optional<int> frame_size_pixels_;
+ std::optional<int> frame_size_pixels_;
int frames_per_second_;
VideoCodecType video_codec_type_;
int min_pixels_per_frame_;
- absl::optional<int> single_active_stream_pixels_;
+ std::optional<int> single_active_stream_pixels_;
};
} // namespace webrtc
diff --git a/call/adaptation/video_stream_input_state_provider_unittest.cc b/call/adaptation/video_stream_input_state_provider_unittest.cc
index 5da2ef2..e9cd72b 100644
--- a/call/adaptation/video_stream_input_state_provider_unittest.cc
+++ b/call/adaptation/video_stream_input_state_provider_unittest.cc
@@ -24,11 +24,11 @@
VideoStreamInputStateProvider input_state_provider(&frame_rate_provider);
VideoStreamInputState input_state = input_state_provider.InputState();
EXPECT_EQ(false, input_state.has_input());
- EXPECT_EQ(absl::nullopt, input_state.frame_size_pixels());
+ EXPECT_EQ(std::nullopt, input_state.frame_size_pixels());
EXPECT_EQ(0, input_state.frames_per_second());
EXPECT_EQ(VideoCodecType::kVideoCodecGeneric, input_state.video_codec_type());
EXPECT_EQ(kDefaultMinPixelsPerFrame, input_state.min_pixels_per_frame());
- EXPECT_EQ(absl::nullopt, input_state.single_active_stream_pixels());
+ EXPECT_EQ(std::nullopt, input_state.single_active_stream_pixels());
}
TEST(VideoStreamInputStateProviderTest, ValuesSet) {
diff --git a/call/audio_receive_stream.h b/call/audio_receive_stream.h
index f2118e8..5154360 100644
--- a/call/audio_receive_stream.h
+++ b/call/audio_receive_stream.h
@@ -13,10 +13,10 @@
#include <map>
#include <memory>
+#include <optional>
#include <string>
#include <vector>
-#include "absl/types/optional.h"
#include "api/audio_codecs/audio_decoder_factory.h"
#include "api/call/transport.h"
#include "api/crypto/crypto_options.h"
@@ -42,7 +42,7 @@
uint64_t packets_discarded = 0;
uint32_t nacks_sent = 0;
std::string codec_name;
- absl::optional<int> codec_payload_type;
+ std::optional<int> codec_payload_type;
uint32_t jitter_ms = 0;
uint32_t jitter_buffer_ms = 0;
uint32_t jitter_buffer_preferred_ms = 0;
@@ -84,21 +84,21 @@
// The timestamp at which the last packet was received, i.e. the time of the
// local clock when it was received - not the RTP timestamp of that packet.
// https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-lastpacketreceivedtimestamp
- absl::optional<Timestamp> last_packet_received;
+ std::optional<Timestamp> last_packet_received;
uint64_t jitter_buffer_flushes = 0;
double relative_packet_arrival_delay_seconds = 0.0;
int32_t interruption_count = 0;
int32_t total_interruption_duration_ms = 0;
// https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-estimatedplayouttimestamp
- absl::optional<int64_t> estimated_playout_ntp_timestamp_ms;
+ std::optional<int64_t> estimated_playout_ntp_timestamp_ms;
// Remote outbound stats derived by the received RTCP sender reports.
// https://w3c.github.io/webrtc-stats/#remoteoutboundrtpstats-dict*
- absl::optional<int64_t> last_sender_report_timestamp_ms;
- absl::optional<int64_t> last_sender_report_remote_timestamp_ms;
+ std::optional<int64_t> last_sender_report_timestamp_ms;
+ std::optional<int64_t> last_sender_report_remote_timestamp_ms;
uint64_t sender_reports_packets_sent = 0;
uint64_t sender_reports_bytes_sent = 0;
uint64_t sender_reports_reports_count = 0;
- absl::optional<TimeDelta> round_trip_time;
+ std::optional<TimeDelta> round_trip_time;
TimeDelta total_round_trip_time = TimeDelta::Zero();
int round_trip_time_measurements = 0;
};
@@ -141,7 +141,7 @@
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory;
- absl::optional<AudioCodecPairId> codec_pair_id;
+ std::optional<AudioCodecPairId> codec_pair_id;
// Per PeerConnection crypto options.
webrtc::CryptoOptions crypto_options;
diff --git a/call/audio_send_stream.h b/call/audio_send_stream.h
index 1b7a906..3794dfb 100644
--- a/call/audio_send_stream.h
+++ b/call/audio_send_stream.h
@@ -12,10 +12,10 @@
#define CALL_AUDIO_SEND_STREAM_H_
#include <memory>
+#include <optional>
#include <string>
#include <vector>
-#include "absl/types/optional.h"
#include "api/audio/audio_processing_statistics.h"
#include "api/audio_codecs/audio_codec_pair_id.h"
#include "api/audio_codecs/audio_encoder.h"
@@ -54,7 +54,7 @@
int32_t packets_lost = -1;
float fraction_lost = -1.0f;
std::string codec_name;
- absl::optional<int> codec_payload_type;
+ std::optional<int> codec_payload_type;
int32_t jitter_ms = -1;
int64_t rtt_ms = -1;
int16_t audio_level = 0;
@@ -129,7 +129,7 @@
// Defines whether to turn on audio network adaptor, and defines its config
// string.
- absl::optional<std::string> audio_network_adaptor_config;
+ std::optional<std::string> audio_network_adaptor_config;
struct SendCodecSpec {
SendCodecSpec(int payload_type, const SdpAudioFormat& format);
@@ -146,15 +146,15 @@
bool nack_enabled = false;
bool transport_cc_enabled = false;
bool enable_non_sender_rtt = false;
- absl::optional<int> cng_payload_type;
- absl::optional<int> red_payload_type;
+ std::optional<int> cng_payload_type;
+ std::optional<int> red_payload_type;
// If unset, use the encoder's default target bitrate.
- absl::optional<int> target_bitrate_bps;
+ std::optional<int> target_bitrate_bps;
};
- absl::optional<SendCodecSpec> send_codec_spec;
+ std::optional<SendCodecSpec> send_codec_spec;
rtc::scoped_refptr<AudioEncoderFactory> encoder_factory;
- absl::optional<AudioCodecPairId> codec_pair_id;
+ std::optional<AudioCodecPairId> codec_pair_id;
// Track ID as specified during track creation.
std::string track_id;
diff --git a/call/bitrate_allocator.cc b/call/bitrate_allocator.cc
index e196f56..1b6d63a 100644
--- a/call/bitrate_allocator.cc
+++ b/call/bitrate_allocator.cc
@@ -318,14 +318,14 @@
return allocation;
}
-// Returns new allocation if modified, absl::nullopt otherwise.
-absl::optional<std::map<BitrateAllocatorObserver*, int>> MaybeApplySurplus(
+// Returns new allocation if modified, std::nullopt otherwise.
+std::optional<std::map<BitrateAllocatorObserver*, int>> MaybeApplySurplus(
const std::map<BitrateAllocatorObserver*, int>& allocation,
const std::vector<AllocatableTrack>& allocatable_tracks,
DataRate bitrate,
DataRate upper_elastic_limit) {
if (upper_elastic_limit.IsZero())
- return absl::nullopt;
+ return std::nullopt;
// In this first pass looping over all `allocatable_tracks`, we aggregates
// - `surplus`: sum of unused rates for all kCanContribute* tracks,
@@ -345,12 +345,12 @@
}
const DataRate allocated = DataRate::BitsPerSec(it->second);
sum_allocated += allocated;
- if (const absl::optional<TrackRateElasticity> elasticity =
+ if (const std::optional<TrackRateElasticity> elasticity =
observer_config.config.rate_elasticity) {
bool inactive_can_contribute_and_consume = false;
if (elasticity == TrackRateElasticity::kCanContributeUnusedRate ||
elasticity == TrackRateElasticity::kCanContributeAndConsume) {
- if (const absl::optional<DataRate> used =
+ if (const std::optional<DataRate> used =
observer_config.observer->GetUsedRate()) {
if (*used < allocated) {
surplus += allocated - *used;
@@ -375,7 +375,7 @@
(sum_allocated >= bitrate) ? (sum_allocated - bitrate) : DataRate::Zero();
if (sum_demand < 0.0001 || overshoot > surplus) {
// No demand for extra bitrate or no available surplus.
- return absl::nullopt;
+ return std::nullopt;
}
surplus -= overshoot;
@@ -388,7 +388,7 @@
// No allocation for this track.
continue;
}
- absl::optional<TrackRateElasticity> elasticity =
+ std::optional<TrackRateElasticity> elasticity =
observer_config.config.rate_elasticity;
if (elasticity == TrackRateElasticity::kCanConsumeExtraRate ||
elasticity == TrackRateElasticity::kCanContributeAndConsume) {
diff --git a/call/bitrate_allocator.h b/call/bitrate_allocator.h
index 07d92d1..33dd0e9 100644
--- a/call/bitrate_allocator.h
+++ b/call/bitrate_allocator.h
@@ -40,7 +40,7 @@
// implementation, as bitrate in bps.
virtual uint32_t OnBitrateUpdated(BitrateAllocationUpdate update) = 0;
// Returns the bitrate consumed (vs allocated) by BitrateAllocatorObserver
- virtual absl::optional<DataRate> GetUsedRate() const = 0;
+ virtual std::optional<DataRate> GetUsedRate() const = 0;
protected:
virtual ~BitrateAllocatorObserver() {}
@@ -71,7 +71,7 @@
// observers. If an observer has twice the bitrate_priority of other
// observers, it should be allocated twice the bitrate above its min.
double bitrate_priority;
- absl::optional<TrackRateElasticity> rate_elasticity;
+ std::optional<TrackRateElasticity> rate_elasticity;
};
// Interface used for mocking
@@ -97,7 +97,7 @@
BitrateAllocatorObserver* observer;
MediaStreamAllocationConfig config;
int64_t allocated_bitrate_bps;
- absl::optional<DataRate> last_used_bitrate;
+ std::optional<DataRate> last_used_bitrate;
double media_ratio; // Part of the total bitrate used for media [0.0, 1.0].
uint32_t LastAllocatedBitrate() const;
diff --git a/call/bitrate_allocator_unittest.cc b/call/bitrate_allocator_unittest.cc
index 9c6599a..7b4fff3 100644
--- a/call/bitrate_allocator_unittest.cc
+++ b/call/bitrate_allocator_unittest.cc
@@ -76,9 +76,7 @@
last_probing_interval_ms_ = update.bwe_period.ms();
return update.target_bitrate.bps() * protection_ratio_;
}
- absl::optional<DataRate> GetUsedRate() const override {
- return absl::nullopt;
- }
+ std::optional<DataRate> GetUsedRate() const override { return std::nullopt; }
uint32_t last_bitrate_bps_;
uint8_t last_fraction_loss_;
int64_t last_rtt_ms_;
@@ -89,7 +87,7 @@
class TestContributingBitrateObserver : public TestBitrateObserver {
public:
TestContributingBitrateObserver() : rate_usage_(DataRate::Zero()) {}
- absl::optional<DataRate> GetUsedRate() const override { return rate_usage_; }
+ std::optional<DataRate> GetUsedRate() const override { return rate_usage_; }
DataRate rate_usage_;
};
@@ -129,7 +127,7 @@
uint32_t pad_up_bitrate_bps,
bool enforce_min_bitrate,
double bitrate_priority,
- absl::optional<TrackRateElasticity> rate_elasticity = absl::nullopt) {
+ std::optional<TrackRateElasticity> rate_elasticity = std::nullopt) {
allocator_->AddObserver(
observer, {min_bitrate_bps, max_bitrate_bps, pad_up_bitrate_bps,
/* priority_bitrate */ 0, enforce_min_bitrate,
diff --git a/call/bitrate_estimator_tests.cc b/call/bitrate_estimator_tests.cc
index 2ffba5d..379f947 100644
--- a/call/bitrate_estimator_tests.cc
+++ b/call/bitrate_estimator_tests.cc
@@ -180,8 +180,8 @@
&test->env().clock(),
test::CreateSquareFrameGenerator(
test::VideoTestConstants::kDefaultWidth,
- test::VideoTestConstants::kDefaultHeight, absl::nullopt,
- absl::nullopt),
+ test::VideoTestConstants::kDefaultHeight, std::nullopt,
+ std::nullopt),
test::VideoTestConstants::kDefaultFramerate,
test->env().task_queue_factory());
frame_generator_capturer_->Init();
diff --git a/call/call.cc b/call/call.cc
index 2e4a25e..aa48491 100644
--- a/call/call.cc
+++ b/call/call.cc
@@ -17,13 +17,13 @@
#include <cstdint>
#include <map>
#include <memory>
+#include <optional>
#include <set>
#include <utility>
#include <vector>
#include "absl/functional/bind_front.h"
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/media_types.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/sequence_checker.h"
@@ -288,13 +288,13 @@
RTC_GUARDED_BY(sequence_checker_);
RateCounter received_rtcp_bytes_per_second_counter_
RTC_GUARDED_BY(sequence_checker_);
- absl::optional<Timestamp> first_received_rtp_audio_timestamp_
+ std::optional<Timestamp> first_received_rtp_audio_timestamp_
RTC_GUARDED_BY(sequence_checker_);
- absl::optional<Timestamp> last_received_rtp_audio_timestamp_
+ std::optional<Timestamp> last_received_rtp_audio_timestamp_
RTC_GUARDED_BY(sequence_checker_);
- absl::optional<Timestamp> first_received_rtp_video_timestamp_
+ std::optional<Timestamp> first_received_rtp_video_timestamp_
RTC_GUARDED_BY(sequence_checker_);
- absl::optional<Timestamp> last_received_rtp_video_timestamp_
+ std::optional<Timestamp> last_received_rtp_video_timestamp_
RTC_GUARDED_BY(sequence_checker_);
};
@@ -306,7 +306,7 @@
explicit SendStats(Clock* clock);
~SendStats();
- void SetFirstPacketTime(absl::optional<Timestamp> first_sent_packet_time);
+ void SetFirstPacketTime(std::optional<Timestamp> first_sent_packet_time);
void PauseSendAndPacerBitrateCounters();
void AddTargetBitrateSample(uint32_t target_bitrate_bps);
void SetMinAllocatableRate(BitrateAllocationLimits limits);
@@ -320,7 +320,7 @@
AvgCounter pacer_bitrate_kbps_counter_ RTC_GUARDED_BY(sequence_checker_);
uint32_t min_allocated_send_bitrate_bps_ RTC_GUARDED_BY(sequence_checker_){
0};
- absl::optional<Timestamp> first_sent_packet_time_
+ std::optional<Timestamp> first_sent_packet_time_
RTC_GUARDED_BY(destructor_sequence_checker_);
};
@@ -451,7 +451,7 @@
// Sequence checker for outgoing network traffic. Could be the network thread.
// Could also be a pacer owned thread or TQ such as the TaskQueueSender.
RTC_NO_UNIQUE_ADDRESS SequenceChecker sent_packet_sequence_checker_;
- absl::optional<rtc::SentPacket> last_sent_packet_
+ std::optional<rtc::SentPacket> last_sent_packet_
RTC_GUARDED_BY(sent_packet_sequence_checker_);
};
} // namespace internal
@@ -620,7 +620,7 @@
}
void Call::SendStats::SetFirstPacketTime(
- absl::optional<Timestamp> first_sent_packet_time) {
+ std::optional<Timestamp> first_sent_packet_time) {
RTC_DCHECK_RUN_ON(&destructor_sequence_checker_);
first_sent_packet_time_ = first_sent_packet_time;
}
@@ -771,7 +771,7 @@
// Stream config is logged in AudioSendStream::ConfigureStream, as it may
// change during the stream's lifetime.
- absl::optional<RtpState> suspended_rtp_state;
+ std::optional<RtpState> suspended_rtp_state;
{
const auto& iter = suspended_audio_send_ssrcs_.find(config.rtp.ssrc);
if (iter != suspended_audio_send_ssrcs_.end()) {
diff --git a/call/call_config.h b/call/call_config.h
index 724dad4..74c15509 100644
--- a/call/call_config.h
+++ b/call/call_config.h
@@ -81,7 +81,7 @@
Metronome* encode_metronome = nullptr;
// The burst interval of the pacer, see TaskQueuePacedSender constructor.
- absl::optional<TimeDelta> pacer_burst_interval;
+ std::optional<TimeDelta> pacer_burst_interval;
// Enables send packet batching from the egress RTP sender.
bool enable_send_packet_batching = false;
diff --git a/call/create_call.cc b/call/create_call.cc
index 573033e..f0c4b54 100644
--- a/call/create_call.cc
+++ b/call/create_call.cc
@@ -11,7 +11,10 @@
#include "call/create_call.h"
#include <memory>
+#include <optional>
+#include "api/test/simulated_network.h"
+#include "api/units/time_delta.h"
#include "call/call.h"
namespace webrtc {
diff --git a/call/fake_network_pipe.cc b/call/fake_network_pipe.cc
index 3c7207b..474d2ac 100644
--- a/call/fake_network_pipe.cc
+++ b/call/fake_network_pipe.cc
@@ -33,10 +33,10 @@
NetworkPacket::NetworkPacket(rtc::CopyOnWriteBuffer packet,
int64_t send_time,
int64_t arrival_time,
- absl::optional<PacketOptions> packet_options,
+ std::optional<PacketOptions> packet_options,
bool is_rtcp,
MediaType media_type,
- absl::optional<int64_t> packet_time_us,
+ std::optional<int64_t> packet_time_us,
Transport* transport)
: packet_(std::move(packet)),
send_time_(send_time),
@@ -146,7 +146,7 @@
bool FakeNetworkPipe::SendRtcp(rtc::ArrayView<const uint8_t> packet,
Transport* transport) {
RTC_DCHECK(transport);
- EnqueuePacket(rtc::CopyOnWriteBuffer(packet), absl::nullopt, true, transport);
+ EnqueuePacket(rtc::CopyOnWriteBuffer(packet), std::nullopt, true, transport);
return true;
}
@@ -161,8 +161,8 @@
}
void FakeNetworkPipe::DeliverRtcpPacket(rtc::CopyOnWriteBuffer packet) {
- EnqueuePacket(std::move(packet), absl::nullopt, true, MediaType::ANY,
- absl::nullopt);
+ EnqueuePacket(std::move(packet), std::nullopt, true, MediaType::ANY,
+ std::nullopt);
}
void FakeNetworkPipe::SetClockOffset(int64_t offset_ms) {
@@ -174,10 +174,10 @@
: packet(std::move(packet)) {}
bool FakeNetworkPipe::EnqueuePacket(rtc::CopyOnWriteBuffer packet,
- absl::optional<PacketOptions> options,
+ std::optional<PacketOptions> options,
bool is_rtcp,
MediaType media_type,
- absl::optional<int64_t> packet_time_us) {
+ std::optional<int64_t> packet_time_us) {
MutexLock lock(&process_lock_);
int64_t time_now_us = clock_->TimeInMicroseconds();
return EnqueuePacket(NetworkPacket(std::move(packet), time_now_us,
@@ -186,14 +186,14 @@
}
bool FakeNetworkPipe::EnqueuePacket(rtc::CopyOnWriteBuffer packet,
- absl::optional<PacketOptions> options,
+ std::optional<PacketOptions> options,
bool is_rtcp,
Transport* transport) {
MutexLock lock(&process_lock_);
int64_t time_now_us = clock_->TimeInMicroseconds();
return EnqueuePacket(NetworkPacket(std::move(packet), time_now_us,
time_now_us, options, is_rtcp,
- MediaType::ANY, absl::nullopt, transport));
+ MediaType::ANY, std::nullopt, transport));
}
bool FakeNetworkPipe::EnqueuePacket(NetworkPacket&& net_packet) {
@@ -348,14 +348,14 @@
}
}
-absl::optional<int64_t> FakeNetworkPipe::TimeUntilNextProcess() {
+std::optional<int64_t> FakeNetworkPipe::TimeUntilNextProcess() {
MutexLock lock(&process_lock_);
- absl::optional<int64_t> delivery_us = network_behavior_->NextDeliveryTimeUs();
+ std::optional<int64_t> delivery_us = network_behavior_->NextDeliveryTimeUs();
if (delivery_us) {
int64_t delay_us = *delivery_us - clock_->TimeInMicroseconds();
return std::max<int64_t>((delay_us + 500) / 1000, 0);
}
- return absl::nullopt;
+ return std::nullopt;
}
bool FakeNetworkPipe::HasReceiver() const {
diff --git a/call/fake_network_pipe.h b/call/fake_network_pipe.h
index 7bc7e0f..ea4b1af 100644
--- a/call/fake_network_pipe.h
+++ b/call/fake_network_pipe.h
@@ -37,10 +37,10 @@
NetworkPacket(rtc::CopyOnWriteBuffer packet,
int64_t send_time,
int64_t arrival_time,
- absl::optional<PacketOptions> packet_options,
+ std::optional<PacketOptions> packet_options,
bool is_rtcp,
MediaType media_type,
- absl::optional<int64_t> packet_time_us,
+ std::optional<int64_t> packet_time_us,
Transport* transport);
NetworkPacket(RtpPacketReceived packet,
@@ -69,11 +69,11 @@
}
bool is_rtcp() const { return is_rtcp_; }
MediaType media_type() const { return media_type_; }
- absl::optional<int64_t> packet_time_us() const { return packet_time_us_; }
+ std::optional<int64_t> packet_time_us() const { return packet_time_us_; }
RtpPacketReceived* packet_received() {
return packet_received_ ? &packet_received_.value() : nullptr;
}
- absl::optional<RtpPacketReceived> packet_received() const {
+ std::optional<RtpPacketReceived> packet_received() const {
return packet_received_;
}
Transport* transport() const { return transport_; }
@@ -86,15 +86,15 @@
int64_t arrival_time_;
// If using a Transport for outgoing degradation, populate with
// PacketOptions (transport-wide sequence number) for RTP.
- absl::optional<PacketOptions> packet_options_;
+ std::optional<PacketOptions> packet_options_;
bool is_rtcp_;
// If using a PacketReceiver for incoming degradation, populate with
// appropriate MediaType and packet time. This type/timing will be kept and
// forwarded. The packet time might be altered to reflect time spent in fake
// network pipe.
MediaType media_type_;
- absl::optional<int64_t> packet_time_us_;
- absl::optional<RtpPacketReceived> packet_received_;
+ std::optional<int64_t> packet_time_us_;
+ std::optional<RtpPacketReceived> packet_received_;
Transport* transport_;
};
@@ -150,7 +150,7 @@
// Processes the network queues and trigger PacketReceiver::IncomingPacket for
// packets ready to be delivered.
void Process() override;
- absl::optional<int64_t> TimeUntilNextProcess() override;
+ std::optional<int64_t> TimeUntilNextProcess() override;
// Get statistics.
float PercentageLoss();
@@ -179,15 +179,15 @@
// Returns true if enqueued, or false if packet was dropped. Use this method
// when enqueueing packets that should be received by PacketReceiver instance.
bool EnqueuePacket(rtc::CopyOnWriteBuffer packet,
- absl::optional<PacketOptions> options,
+ std::optional<PacketOptions> options,
bool is_rtcp,
MediaType media_type,
- absl::optional<int64_t> packet_time_us);
+ std::optional<int64_t> packet_time_us);
// Returns true if enqueued, or false if packet was dropped. Use this method
// when enqueueing packets that should be received by Transport instance.
bool EnqueuePacket(rtc::CopyOnWriteBuffer packet,
- absl::optional<PacketOptions> options,
+ std::optional<PacketOptions> options,
bool is_rtcp,
Transport* transport);
diff --git a/call/receive_time_calculator_unittest.cc b/call/receive_time_calculator_unittest.cc
index f2e3d54..312a79d 100644
--- a/call/receive_time_calculator_unittest.cc
+++ b/call/receive_time_calculator_unittest.cc
@@ -15,9 +15,9 @@
#include <algorithm>
#include <cmath>
#include <cstdint>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "rtc_base/random.h"
#include "rtc_base/time_utils.h"
#include "test/gtest.h"
@@ -50,7 +50,7 @@
private:
int64_t clock_us_;
- absl::optional<int64_t> last_query_us_;
+ std::optional<int64_t> last_query_us_;
float drift_;
float accumulated_drift_us_ = 0;
};
diff --git a/call/rtp_bitrate_configurator.cc b/call/rtp_bitrate_configurator.cc
index 264dcdc..f0a36ad 100644
--- a/call/rtp_bitrate_configurator.cc
+++ b/call/rtp_bitrate_configurator.cc
@@ -49,7 +49,7 @@
return bitrate_config_;
}
-absl::optional<BitrateConstraints>
+std::optional<BitrateConstraints>
RtpBitrateConfigurator::UpdateWithSdpParameters(
const BitrateConstraints& bitrate_config) {
RTC_DCHECK_GE(bitrate_config.min_bitrate_bps, 0);
@@ -58,7 +58,7 @@
RTC_DCHECK_GT(bitrate_config.max_bitrate_bps, 0);
}
- absl::optional<int> new_start;
+ std::optional<int> new_start;
// Only update the "start" bitrate if it's set, and different from the old
// value. In practice, this value comes from the x-google-start-bitrate codec
// parameter in SDP, and setting the same remote description twice shouldn't
@@ -72,7 +72,7 @@
return UpdateConstraints(new_start);
}
-absl::optional<BitrateConstraints>
+std::optional<BitrateConstraints>
RtpBitrateConfigurator::UpdateWithClientPreferences(
const BitrateSettings& bitrate_mask) {
bitrate_config_mask_ = bitrate_mask;
@@ -80,17 +80,17 @@
}
// Relay cap can change only max bitrate.
-absl::optional<BitrateConstraints> RtpBitrateConfigurator::UpdateWithRelayCap(
+std::optional<BitrateConstraints> RtpBitrateConfigurator::UpdateWithRelayCap(
DataRate cap) {
if (cap.IsFinite()) {
RTC_DCHECK(!cap.IsZero());
}
max_bitrate_over_relay_ = cap;
- return UpdateConstraints(absl::nullopt);
+ return UpdateConstraints(std::nullopt);
}
-absl::optional<BitrateConstraints> RtpBitrateConfigurator::UpdateConstraints(
- const absl::optional<int>& new_start) {
+std::optional<BitrateConstraints> RtpBitrateConfigurator::UpdateConstraints(
+ const std::optional<int>& new_start) {
BitrateConstraints updated;
updated.min_bitrate_bps =
std::max(bitrate_config_mask_.min_bitrate_bps.value_or(0),
@@ -114,7 +114,7 @@
if (updated.min_bitrate_bps == bitrate_config_.min_bitrate_bps &&
updated.max_bitrate_bps == bitrate_config_.max_bitrate_bps &&
!new_start) {
- return absl::nullopt;
+ return std::nullopt;
}
if (new_start) {
diff --git a/call/rtp_bitrate_configurator.h b/call/rtp_bitrate_configurator.h
index 5cb779a..754629d 100644
--- a/call/rtp_bitrate_configurator.h
+++ b/call/rtp_bitrate_configurator.h
@@ -11,7 +11,8 @@
#ifndef CALL_RTP_BITRATE_CONFIGURATOR_H_
#define CALL_RTP_BITRATE_CONFIGURATOR_H_
-#include "absl/types/optional.h"
+#include <optional>
+
#include "api/transport/bitrate_settings.h"
#include "api/units/data_rate.h"
@@ -36,7 +37,7 @@
// implemented. Passing -1 leaves the start bitrate unchanged. Behavior is not
// guaranteed for other negative values or 0.
// The optional return value is set with new configuration if it was updated.
- absl::optional<BitrateConstraints> UpdateWithSdpParameters(
+ std::optional<BitrateConstraints> UpdateWithSdpParameters(
const BitrateConstraints& bitrate_config_);
// The greater min and smaller max set by this and SetSdpBitrateParameters
@@ -45,17 +46,17 @@
// Assumes 0 <= min <= start <= max holds for set parameters.
// Update the bitrate configuration
// The optional return value is set with new configuration if it was updated.
- absl::optional<BitrateConstraints> UpdateWithClientPreferences(
+ std::optional<BitrateConstraints> UpdateWithClientPreferences(
const BitrateSettings& bitrate_mask);
// Apply a cap for relayed calls.
- absl::optional<BitrateConstraints> UpdateWithRelayCap(DataRate cap);
+ std::optional<BitrateConstraints> UpdateWithRelayCap(DataRate cap);
private:
// Applies update to the BitrateConstraints cached in `config_`, resetting
// with `new_start` if set.
- absl::optional<BitrateConstraints> UpdateConstraints(
- const absl::optional<int>& new_start);
+ std::optional<BitrateConstraints> UpdateConstraints(
+ const std::optional<int>& new_start);
// Bitrate config used until valid bitrate estimates are calculated. Also
// used to cap total bitrate used. This comes from the remote connection.
diff --git a/call/rtp_bitrate_configurator_unittest.cc b/call/rtp_bitrate_configurator_unittest.cc
index 6449a1a..0b019ce 100644
--- a/call/rtp_bitrate_configurator_unittest.cc
+++ b/call/rtp_bitrate_configurator_unittest.cc
@@ -14,7 +14,7 @@
#include "test/gtest.h"
namespace webrtc {
-using absl::nullopt;
+using std::nullopt;
class RtpBitrateConfiguratorTest : public ::testing::Test {
public:
@@ -22,10 +22,10 @@
: configurator_(new RtpBitrateConfigurator(BitrateConstraints())) {}
std::unique_ptr<RtpBitrateConfigurator> configurator_;
void UpdateConfigMatches(BitrateConstraints bitrate_config,
- absl::optional<int> min_bitrate_bps,
- absl::optional<int> start_bitrate_bps,
- absl::optional<int> max_bitrate_bps) {
- absl::optional<BitrateConstraints> result =
+ std::optional<int> min_bitrate_bps,
+ std::optional<int> start_bitrate_bps,
+ std::optional<int> max_bitrate_bps) {
+ std::optional<BitrateConstraints> result =
configurator_->UpdateWithSdpParameters(bitrate_config);
EXPECT_TRUE(result.has_value());
if (start_bitrate_bps.has_value())
@@ -37,10 +37,10 @@
}
void UpdateMaskMatches(BitrateSettings bitrate_mask,
- absl::optional<int> min_bitrate_bps,
- absl::optional<int> start_bitrate_bps,
- absl::optional<int> max_bitrate_bps) {
- absl::optional<BitrateConstraints> result =
+ std::optional<int> min_bitrate_bps,
+ std::optional<int> start_bitrate_bps,
+ std::optional<int> max_bitrate_bps) {
+ std::optional<BitrateConstraints> result =
configurator_->UpdateWithClientPreferences(bitrate_mask);
EXPECT_TRUE(result.has_value());
if (start_bitrate_bps.has_value())
diff --git a/call/rtp_config.cc b/call/rtp_config.cc
index 5457a94..43b615c 100644
--- a/call/rtp_config.cc
+++ b/call/rtp_config.cc
@@ -155,12 +155,12 @@
return flexfec.payload_type != -1 && ssrc == flexfec.ssrc;
}
-absl::optional<uint32_t> RtpConfig::GetRtxSsrcAssociatedWithMediaSsrc(
+std::optional<uint32_t> RtpConfig::GetRtxSsrcAssociatedWithMediaSsrc(
uint32_t media_ssrc) const {
RTC_DCHECK(IsMediaSsrc(media_ssrc));
// If we don't use RTX there is no association.
if (rtx.ssrcs.empty())
- return absl::nullopt;
+ return std::nullopt;
// If we use RTX there MUST be an association ssrcs[i] <-> rtx.ssrcs[i].
RTC_DCHECK_EQ(ssrcs.size(), rtx.ssrcs.size());
return FindAssociatedSsrc(media_ssrc, ssrcs, rtx.ssrcs);
@@ -189,7 +189,7 @@
return media_ssrc;
}
-absl::optional<std::string> RtpConfig::GetRidForSsrc(uint32_t ssrc) const {
+std::optional<std::string> RtpConfig::GetRidForSsrc(uint32_t ssrc) const {
auto it = std::find(ssrcs.begin(), ssrcs.end(), ssrc);
if (it != ssrcs.end()) {
size_t ssrc_index = std::distance(ssrcs.begin(), it);
@@ -197,7 +197,7 @@
return rids[ssrc_index];
}
}
- return absl::nullopt;
+ return std::nullopt;
}
} // namespace webrtc
diff --git a/call/rtp_config.h b/call/rtp_config.h
index ec76d42..0e667b2 100644
--- a/call/rtp_config.h
+++ b/call/rtp_config.h
@@ -14,10 +14,10 @@
#include <stddef.h>
#include <stdint.h>
+#include <optional>
#include <string>
#include <vector>
-#include "absl/types/optional.h"
#include "api/rtp_headers.h"
#include "api/rtp_parameters.h"
@@ -166,11 +166,11 @@
bool IsMediaSsrc(uint32_t ssrc) const;
bool IsRtxSsrc(uint32_t ssrc) const;
bool IsFlexfecSsrc(uint32_t ssrc) const;
- absl::optional<uint32_t> GetRtxSsrcAssociatedWithMediaSsrc(
+ std::optional<uint32_t> GetRtxSsrcAssociatedWithMediaSsrc(
uint32_t media_ssrc) const;
uint32_t GetMediaSsrcAssociatedWithRtxSsrc(uint32_t rtx_ssrc) const;
uint32_t GetMediaSsrcAssociatedWithFlexfecSsrc(uint32_t flexfec_ssrc) const;
- absl::optional<std::string> GetRidForSsrc(uint32_t ssrc) const;
+ std::optional<std::string> GetRidForSsrc(uint32_t ssrc) const;
};
} // namespace webrtc
#endif // CALL_RTP_CONFIG_H_
diff --git a/call/rtp_payload_params.cc b/call/rtp_payload_params.cc
index 33e9c76..00e7509 100644
--- a/call/rtp_payload_params.cc
+++ b/call/rtp_payload_params.cc
@@ -35,7 +35,7 @@
constexpr int kMaxSimulatedSpatialLayers = 3;
void PopulateRtpWithCodecSpecifics(const CodecSpecificInfo& info,
- absl::optional<int> spatial_index,
+ std::optional<int> spatial_index,
RTPVideoHeader* rtp) {
rtp->codec = info.codecType;
rtp->is_last_frame_in_picture = info.end_of_picture;
@@ -198,7 +198,7 @@
RTPVideoHeader RtpPayloadParams::GetRtpVideoHeader(
const EncodedImage& image,
const CodecSpecificInfo* codec_specific_info,
- absl::optional<int64_t> shared_frame_id) {
+ std::optional<int64_t> shared_frame_id) {
int64_t frame_id;
if (shared_frame_id) {
frame_id = *shared_frame_id;
@@ -219,8 +219,8 @@
rtp_video_header.width = image._encodedWidth;
rtp_video_header.height = image._encodedHeight;
rtp_video_header.color_space = image.ColorSpace()
- ? absl::make_optional(*image.ColorSpace())
- : absl::nullopt;
+ ? std::make_optional(*image.ColorSpace())
+ : std::nullopt;
rtp_video_header.video_frame_tracking_id = image.VideoFrameTrackingId();
SetVideoTiming(image, &rtp_video_header.video_timing);
@@ -353,10 +353,10 @@
RTC_DCHECK_NOTREACHED() << "Unsupported codec.";
}
-absl::optional<FrameDependencyStructure> RtpPayloadParams::GenericStructure(
+std::optional<FrameDependencyStructure> RtpPayloadParams::GenericStructure(
const CodecSpecificInfo* codec_specific_info) {
if (codec_specific_info == nullptr) {
- return absl::nullopt;
+ return std::nullopt;
}
// This helper shouldn't be used when template structure is specified
// explicetly.
@@ -367,15 +367,14 @@
return MinimalisticStructure(/*num_spatial_layers=*/1,
/*num_temporal_layer=*/1);
}
- return absl::nullopt;
+ return std::nullopt;
case VideoCodecType::kVideoCodecVP8:
return MinimalisticStructure(/*num_spatial_layers=*/1,
/*num_temporal_layer=*/kMaxTemporalStreams);
case VideoCodecType::kVideoCodecVP9: {
- absl::optional<FrameDependencyStructure> structure =
- MinimalisticStructure(
- /*num_spatial_layers=*/kMaxSimulatedSpatialLayers,
- /*num_temporal_layer=*/kMaxTemporalStreams);
+ std::optional<FrameDependencyStructure> structure = MinimalisticStructure(
+ /*num_spatial_layers=*/kMaxSimulatedSpatialLayers,
+ /*num_temporal_layer=*/kMaxTemporalStreams);
const CodecSpecificInfoVP9& vp9 = codec_specific_info->codecSpecific.VP9;
if (vp9.ss_data_available && vp9.spatial_layer_resolution_present) {
RenderResolution first_valid;
@@ -411,7 +410,7 @@
case VideoCodecType::kVideoCodecAV1:
case VideoCodecType::kVideoCodecH264:
case VideoCodecType::kVideoCodecH265:
- return absl::nullopt;
+ return std::nullopt;
}
RTC_DCHECK_NOTREACHED() << "Unsupported codec.";
}
diff --git a/call/rtp_payload_params.h b/call/rtp_payload_params.h
index ee685d1..870e3c3 100644
--- a/call/rtp_payload_params.h
+++ b/call/rtp_payload_params.h
@@ -12,9 +12,9 @@
#define CALL_RTP_PAYLOAD_PARAMS_H_
#include <array>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/field_trials_view.h"
#include "api/video_codecs/video_encoder.h"
#include "call/rtp_config.h"
@@ -38,11 +38,11 @@
RTPVideoHeader GetRtpVideoHeader(const EncodedImage& image,
const CodecSpecificInfo* codec_specific_info,
- absl::optional<int64_t> shared_frame_id);
+ std::optional<int64_t> shared_frame_id);
// Returns structure that aligns with simulated generic info generated by
// `GetRtpVideoHeader` for the `codec_specific_info`
- absl::optional<FrameDependencyStructure> GenericStructure(
+ std::optional<FrameDependencyStructure> GenericStructure(
const CodecSpecificInfo* codec_specific_info);
uint32_t ssrc() const;
@@ -122,7 +122,7 @@
// that, for a given object, we either always use
// SetDependenciesVp8Deprecated(), or always use SetDependenciesVp8New().
// TODO(bugs.webrtc.org/10242): Remove.
- absl::optional<bool> new_version_used_;
+ std::optional<bool> new_version_used_;
const uint32_t ssrc_;
RtpPayloadState state_;
diff --git a/call/rtp_payload_params_unittest.cc b/call/rtp_payload_params_unittest.cc
index 0099084..15096cf 100644
--- a/call/rtp_payload_params_unittest.cc
+++ b/call/rtp_payload_params_unittest.cc
@@ -13,10 +13,10 @@
#include <string.h>
#include <map>
+#include <optional>
#include <set>
#include "absl/container/inlined_vector.h"
-#include "absl/types/optional.h"
#include "absl/types/variant.h"
#include "api/transport/field_trial_based_config.h"
#include "api/video/video_content_type.h"
@@ -144,7 +144,7 @@
EXPECT_EQ(kVideoRotation_90, header.rotation);
EXPECT_EQ(VideoContentType::SCREENSHARE, header.content_type);
EXPECT_EQ(kVideoCodecVP9, header.codec);
- EXPECT_EQ(absl::make_optional(color_space), header.color_space);
+ EXPECT_EQ(std::make_optional(color_space), header.color_space);
EXPECT_EQ(kPictureId + 1, vp9_header.picture_id);
EXPECT_EQ(kTl0PicIdx, vp9_header.tl0_pic_idx);
EXPECT_EQ(vp9_header.temporal_idx, codec_info.codecSpecific.VP9.temporal_idx);
@@ -371,7 +371,7 @@
RtpPayloadParams params(kSsrc1, &state, FieldTrialBasedConfig());
RTPVideoHeader header =
- params.GetRtpVideoHeader(encoded_image, &codec_info, absl::nullopt);
+ params.GetRtpVideoHeader(encoded_image, &codec_info, std::nullopt);
EXPECT_THAT(header.codec, Eq(kVideoCodecGeneric));
@@ -379,7 +379,7 @@
EXPECT_THAT(header.generic->frame_id, Eq(123));
encoded_image._frameType = VideoFrameType::kVideoFrameDelta;
- header = params.GetRtpVideoHeader(encoded_image, &codec_info, absl::nullopt);
+ header = params.GetRtpVideoHeader(encoded_image, &codec_info, std::nullopt);
ASSERT_TRUE(header.generic);
EXPECT_THAT(header.generic->frame_id, Eq(124));
}
diff --git a/call/rtp_transport_config.h b/call/rtp_transport_config.h
index cce5214..b310710 100644
--- a/call/rtp_transport_config.h
+++ b/call/rtp_transport_config.h
@@ -12,8 +12,8 @@
#define CALL_RTP_TRANSPORT_CONFIG_H_
#include <memory>
+#include <optional>
-#include "absl/types/optional.h"
#include "api/environment/environment.h"
#include "api/network_state_predictor.h"
#include "api/transport/bitrate_settings.h"
@@ -37,7 +37,7 @@
NetworkControllerFactoryInterface* network_controller_factory = nullptr;
// The burst interval of the pacer, see TaskQueuePacedSender constructor.
- absl::optional<TimeDelta> pacer_burst_interval;
+ std::optional<TimeDelta> pacer_burst_interval;
};
} // namespace webrtc
diff --git a/call/rtp_transport_controller_send.cc b/call/rtp_transport_controller_send.cc
index 627adba..34cf06d 100644
--- a/call/rtp_transport_controller_send.cc
+++ b/call/rtp_transport_controller_send.cc
@@ -11,12 +11,12 @@
#include <cstdint>
#include <memory>
+#include <optional>
#include <utility>
#include <vector>
#include "absl/strings/match.h"
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/task_queue/pending_task_safety_flag.h"
#include "api/task_queue/task_queue_base.h"
#include "api/transport/goog_cc_factory.h"
@@ -199,7 +199,7 @@
}
void RtpTransportControllerSend::UpdateControlState() {
- absl::optional<TargetTransferRate> update = control_handler_->GetUpdate();
+ std::optional<TargetTransferRate> update = control_handler_->GetUpdate();
if (!update)
return;
retransmission_rate_limiter_.SetMaxRate(update->target_rate.bps());
@@ -215,13 +215,13 @@
}
}
-absl::optional<bool> RtpTransportControllerSend::GetCongestedStateUpdate()
+std::optional<bool> RtpTransportControllerSend::GetCongestedStateUpdate()
const {
bool congested = transport_feedback_adapter_.GetOutstandingData() >=
congestion_window_size_;
if (congested != is_congested_)
return congested;
- return absl::nullopt;
+ return std::nullopt;
}
PacketRouter* RtpTransportControllerSend::packet_router() {
@@ -326,7 +326,7 @@
return;
}
- absl::optional<BitrateConstraints> relay_constraint_update =
+ std::optional<BitrateConstraints> relay_constraint_update =
ApplyOrLiftRelayCap(IsRelayed(network_route));
// Check whether the network route has changed on each transport.
@@ -413,7 +413,7 @@
int64_t RtpTransportControllerSend::GetPacerQueuingDelayMs() const {
return pacer_.OldestPacketWaitTime().ms();
}
-absl::optional<Timestamp> RtpTransportControllerSend::GetFirstPacketTime()
+std::optional<Timestamp> RtpTransportControllerSend::GetFirstPacketTime()
const {
return pacer_.FirstSentPacketTime();
}
@@ -444,7 +444,7 @@
void RtpTransportControllerSend::ProcessSentPacket(
const rtc::SentPacket& sent_packet) {
RTC_DCHECK_RUN_ON(&sequence_checker_);
- absl::optional<SentPacket> packet_msg =
+ std::optional<SentPacket> packet_msg =
transport_feedback_adapter_.ProcessSentPacket(sent_packet);
if (!packet_msg)
return;
@@ -493,7 +493,7 @@
void RtpTransportControllerSend::SetSdpBitrateParameters(
const BitrateConstraints& constraints) {
RTC_DCHECK_RUN_ON(&sequence_checker_);
- absl::optional<BitrateConstraints> updated =
+ std::optional<BitrateConstraints> updated =
bitrate_configurator_.UpdateWithSdpParameters(constraints);
if (updated.has_value()) {
UpdateBitrateConstraints(*updated);
@@ -507,7 +507,7 @@
void RtpTransportControllerSend::SetClientBitratePreferences(
const BitrateSettings& preferences) {
RTC_DCHECK_RUN_ON(&sequence_checker_);
- absl::optional<BitrateConstraints> updated =
+ std::optional<BitrateConstraints> updated =
bitrate_configurator_.UpdateWithClientPreferences(preferences);
if (updated.has_value()) {
UpdateBitrateConstraints(*updated);
@@ -518,7 +518,7 @@
}
}
-absl::optional<BitrateConstraints>
+std::optional<BitrateConstraints>
RtpTransportControllerSend::ApplyOrLiftRelayCap(bool is_relayed) {
DataRate cap = is_relayed ? relay_bandwidth_cap_ : DataRate::PlusInfinity();
return bitrate_configurator_.UpdateWithRelayCap(cap);
@@ -608,7 +608,7 @@
const rtcp::TransportFeedback& feedback) {
RTC_DCHECK_RUN_ON(&sequence_checker_);
feedback_demuxer_.OnTransportFeedback(feedback);
- absl::optional<TransportPacketsFeedback> feedback_msg =
+ std::optional<TransportPacketsFeedback> feedback_msg =
transport_feedback_adapter_.ProcessTransportFeedback(feedback,
receive_time);
if (feedback_msg) {
diff --git a/call/rtp_transport_controller_send.h b/call/rtp_transport_controller_send.h
index 9b91d9a..602feb3 100644
--- a/call/rtp_transport_controller_send.h
+++ b/call/rtp_transport_controller_send.h
@@ -93,7 +93,7 @@
void OnNetworkAvailability(bool network_available) override;
NetworkLinkRtcpObserver* GetRtcpObserver() override;
int64_t GetPacerQueuingDelayMs() const override;
- absl::optional<Timestamp> GetFirstPacketTime() const override;
+ std::optional<Timestamp> GetFirstPacketTime() const override;
void EnablePeriodicAlrProbing(bool enable) override;
void OnSentPacket(const rtc::SentPacket& sent_packet) override;
void OnReceivedPacket(const ReceivedPacket& packet_msg) override;
@@ -134,7 +134,7 @@
void StartProcessPeriodicTasks() RTC_RUN_ON(sequence_checker_);
void UpdateControllerWithTimeInterval() RTC_RUN_ON(sequence_checker_);
- absl::optional<BitrateConstraints> ApplyOrLiftRelayCap(bool is_relayed);
+ std::optional<BitrateConstraints> ApplyOrLiftRelayCap(bool is_relayed);
bool IsRelevantRouteChange(const rtc::NetworkRoute& old_route,
const rtc::NetworkRoute& new_route) const;
void UpdateBitrateConstraints(const BitrateConstraints& updated);
@@ -142,7 +142,7 @@
void PostUpdates(NetworkControlUpdate update) RTC_RUN_ON(sequence_checker_);
void UpdateControlState() RTC_RUN_ON(sequence_checker_);
void UpdateCongestedState() RTC_RUN_ON(sequence_checker_);
- absl::optional<bool> GetCongestedStateUpdate() const
+ std::optional<bool> GetCongestedStateUpdate() const
RTC_RUN_ON(sequence_checker_);
// Called by packet router just before packet is sent to the RTP modules.
diff --git a/call/rtp_transport_controller_send_interface.h b/call/rtp_transport_controller_send_interface.h
index aa71698..13eede6 100644
--- a/call/rtp_transport_controller_send_interface.h
+++ b/call/rtp_transport_controller_send_interface.h
@@ -15,11 +15,11 @@
#include <map>
#include <memory>
+#include <optional>
#include <string>
#include <vector>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/crypto/crypto_options.h"
#include "api/fec_controller.h"
#include "api/frame_transformer_interface.h"
@@ -138,7 +138,7 @@
virtual void OnNetworkAvailability(bool network_available) = 0;
virtual NetworkLinkRtcpObserver* GetRtcpObserver() = 0;
virtual int64_t GetPacerQueuingDelayMs() const = 0;
- virtual absl::optional<Timestamp> GetFirstPacketTime() const = 0;
+ virtual std::optional<Timestamp> GetFirstPacketTime() const = 0;
virtual void EnablePeriodicAlrProbing(bool enable) = 0;
// Called when a packet has been sent.
diff --git a/call/rtp_video_sender.cc b/call/rtp_video_sender.cc
index f7454ae..431e5c0 100644
--- a/call/rtp_video_sender.cc
+++ b/call/rtp_video_sender.cc
@@ -294,9 +294,9 @@
return rtp_streams;
}
-absl::optional<VideoCodecType> GetVideoCodecType(const RtpConfig& config) {
+std::optional<VideoCodecType> GetVideoCodecType(const RtpConfig& config) {
if (config.raw_payload) {
- return absl::nullopt;
+ return std::nullopt;
}
return PayloadStringToCodecType(config.payload_name);
}
@@ -552,7 +552,7 @@
RTPSenderVideo& sender_video = *rtp_streams_[simulcast_index].sender_video;
if (codec_specific_info && codec_specific_info->template_structure) {
sender_video.SetVideoStructure(&*codec_specific_info->template_structure);
- } else if (absl::optional<FrameDependencyStructure> structure =
+ } else if (std::optional<FrameDependencyStructure> structure =
params_[simulcast_index].GenericStructure(
codec_specific_info)) {
sender_video.SetVideoStructure(&*structure);
@@ -561,7 +561,7 @@
}
}
- absl::optional<int64_t> frame_id;
+ std::optional<int64_t> frame_id;
if (!independent_frame_ids_) {
frame_id = shared_frame_id_;
}
@@ -599,7 +599,7 @@
// If spatial scalability is enabled, it is covered by a single stream.
rtp_streams_[0].rtp_rtcp->SetVideoBitrateAllocation(bitrate);
} else {
- std::vector<absl::optional<VideoBitrateAllocation>> layer_bitrates =
+ std::vector<std::optional<VideoBitrateAllocation>> layer_bitrates =
bitrate.GetSimulcastAllocations();
// Simulcast is in use, split the VideoBitrateAllocation into one struct
// per rtp stream, moving over the temporal layer allocation.
@@ -726,7 +726,7 @@
// Only happens during shutdown, when RTP module is already inactive,
// so OK to call fec generator here.
if (rtp_streams_[i].fec_generator) {
- absl::optional<RtpState> fec_state =
+ std::optional<RtpState> fec_state =
rtp_streams_[i].fec_generator->GetRtpState();
if (fec_state) {
uint32_t ssrc = rtp_config_.flexfec.ssrc;
diff --git a/call/rtp_video_sender.h b/call/rtp_video_sender.h
index c35bc6f..3cb2e9e 100644
--- a/call/rtp_video_sender.h
+++ b/call/rtp_video_sender.h
@@ -13,10 +13,10 @@
#include <map>
#include <memory>
+#include <optional>
#include <unordered_set>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/call/transport.h"
#include "api/environment/environment.h"
@@ -182,7 +182,7 @@
const std::vector<webrtc_internal_rtp_video_sender::RtpStreamSender>
rtp_streams_;
const RtpConfig rtp_config_;
- const absl::optional<VideoCodecType> codec_type_;
+ const std::optional<VideoCodecType> codec_type_;
RtpTransportControllerSendInterface* const transport_;
// When using the generic descriptor we want all simulcast streams to share
diff --git a/call/rtp_video_sender_interface.h b/call/rtp_video_sender_interface.h
index 6bbcb7f..76f7de7 100644
--- a/call/rtp_video_sender_interface.h
+++ b/call/rtp_video_sender_interface.h
@@ -12,9 +12,9 @@
#define CALL_RTP_VIDEO_SENDER_INTERFACE_H_
#include <map>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/call/bitrate_allocation.h"
#include "api/fec_controller_override.h"
diff --git a/call/rtp_video_sender_unittest.cc b/call/rtp_video_sender_unittest.cc
index d3959fb..98cd307 100644
--- a/call/rtp_video_sender_unittest.cc
+++ b/call/rtp_video_sender_unittest.cc
@@ -723,7 +723,7 @@
// Send in delta frame.
encoded_image._frameType = VideoFrameType::kVideoFrameDelta;
- codec_specific.template_structure = absl::nullopt;
+ codec_specific.template_structure = std::nullopt;
codec_specific.generic_frame_info =
GenericFrameInfo::Builder().T(1).Dtis("D").Build();
codec_specific.generic_frame_info->encoder_buffers = {{0, true, false}};
@@ -1000,7 +1000,7 @@
EXPECT_EQ(test.router()->OnEncodedImage(encoded_image, &codec_specific).error,
EncodedImageCallback::Result::OK);
// Send in 2nd spatial layer.
- codec_specific.template_structure = absl::nullopt;
+ codec_specific.template_structure = std::nullopt;
codec_specific.generic_frame_info =
GenericFrameInfo::Builder().S(1).Dtis("-S").Build();
codec_specific.generic_frame_info->encoder_buffers = {{0, true, false},
@@ -1164,7 +1164,7 @@
// Send in a new key frame without the support for the dependency descriptor.
encoded_image._frameType = VideoFrameType::kVideoFrameKey;
- codec_specific.template_structure = absl::nullopt;
+ codec_specific.template_structure = std::nullopt;
EXPECT_EQ(test.router()->OnEncodedImage(encoded_image, &codec_specific).error,
EncodedImageCallback::Result::OK);
test.AdvanceTime(TimeDelta::Millis(33));
diff --git a/call/simulated_packet_receiver.h b/call/simulated_packet_receiver.h
index 2db46e8..e70937d 100644
--- a/call/simulated_packet_receiver.h
+++ b/call/simulated_packet_receiver.h
@@ -34,7 +34,7 @@
// Returns the time until next process or nullopt to indicate that the next
// process time is unknown. If the next process time is unknown, this should
// be checked again any time a packet is enqueued.
- virtual absl::optional<int64_t> TimeUntilNextProcess() = 0;
+ virtual std::optional<int64_t> TimeUntilNextProcess() = 0;
};
} // namespace webrtc
diff --git a/call/syncable.h b/call/syncable.h
index 6817be9..09c0f22 100644
--- a/call/syncable.h
+++ b/call/syncable.h
@@ -16,7 +16,7 @@
#include <stdint.h>
-#include "absl/types/optional.h"
+#include <optional>
namespace webrtc {
@@ -34,7 +34,7 @@
virtual ~Syncable();
virtual uint32_t id() const = 0;
- virtual absl::optional<Info> GetInfo() const = 0;
+ virtual std::optional<Info> GetInfo() const = 0;
virtual bool GetPlayoutRtpTimestamp(uint32_t* rtp_timestamp,
int64_t* time_ms) const = 0;
virtual bool SetMinimumPlayoutDelay(int delay_ms) = 0;
diff --git a/call/test/mock_rtp_transport_controller_send.h b/call/test/mock_rtp_transport_controller_send.h
index d1e1d64..84caccf 100644
--- a/call/test/mock_rtp_transport_controller_send.h
+++ b/call/test/mock_rtp_transport_controller_send.h
@@ -86,7 +86,7 @@
MOCK_METHOD(void, OnNetworkAvailability, (bool), (override));
MOCK_METHOD(NetworkLinkRtcpObserver*, GetRtcpObserver, (), (override));
MOCK_METHOD(int64_t, GetPacerQueuingDelayMs, (), (const, override));
- MOCK_METHOD(absl::optional<Timestamp>,
+ MOCK_METHOD(std::optional<Timestamp>,
GetFirstPacketTime,
(),
(const, override));
diff --git a/call/video_receive_stream.h b/call/video_receive_stream.h
index 4bdf81f..23cf035 100644
--- a/call/video_receive_stream.h
+++ b/call/video_receive_stream.h
@@ -55,7 +55,7 @@
std::function<void(const RecordableEncodedFrame&)> callback;
// Memento of when a keyframe request was last sent. The
// VideoReceiveStreamInterface client should not interpret the attribute.
- absl::optional<int64_t> last_keyframe_request_ms;
+ std::optional<int64_t> last_keyframe_request_ms;
};
// TODO(mflodman) Move all these settings to VideoDecoder and move the
@@ -88,8 +88,8 @@
uint32_t frames_rendered = 0;
// Decoder stats.
- absl::optional<std::string> decoder_implementation_name;
- absl::optional<bool> power_efficient_decoder;
+ std::optional<std::string> decoder_implementation_name;
+ std::optional<bool> power_efficient_decoder;
FrameCounts frame_counts;
int decode_ms = 0;
int max_decode_ms = 0;
@@ -127,7 +127,7 @@
// https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-totalsqauredinterframedelay
double total_squared_inter_frame_delay = 0;
int64_t first_frame_received_to_decoded_ms = -1;
- absl::optional<uint64_t> qp_sum;
+ std::optional<uint64_t> qp_sum;
// TODO(webrtc:357636606): Propagate this score upwards in the chain.
// Corruption score, indicating the probability of corruption. Its value is
@@ -136,8 +136,8 @@
// However, note that the corruption score may not accurately reflect
// corruption. E.g. even if the corruption score is 0, the compressed frame
// may still be corrupted and vice versa.
- absl::optional<double> corruption_score_sum;
- absl::optional<double> corruption_score_squared_sum;
+ std::optional<double> corruption_score_sum;
+ std::optional<double> corruption_score_squared_sum;
// Number of frames the `corruption_score` was calculated on. This is
// usually not the same as `frames_decoded`.
uint32_t corruption_score_count = 0;
@@ -157,23 +157,23 @@
VideoContentType content_type = VideoContentType::UNSPECIFIED;
// https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-estimatedplayouttimestamp
- absl::optional<int64_t> estimated_playout_ntp_timestamp_ms;
+ std::optional<int64_t> estimated_playout_ntp_timestamp_ms;
int sync_offset_ms = std::numeric_limits<int>::max();
uint32_t ssrc = 0;
std::string c_name;
RtpReceiveStats rtp_stats;
RtcpPacketTypeCounter rtcp_packet_type_counts;
- absl::optional<RtpReceiveStats> rtx_rtp_stats;
+ std::optional<RtpReceiveStats> rtx_rtp_stats;
// Timing frame info: all important timestamps for a full lifetime of a
// single 'timing frame'.
- absl::optional<webrtc::TimingFrameInfo> timing_frame_info;
+ std::optional<webrtc::TimingFrameInfo> timing_frame_info;
// Remote outbound stats derived by the received RTCP sender reports.
// https://w3c.github.io/webrtc-stats/#remoteoutboundrtpstats-dict*
- absl::optional<int64_t> last_sender_report_timestamp_ms;
- absl::optional<int64_t> last_sender_report_remote_timestamp_ms;
+ std::optional<int64_t> last_sender_report_timestamp_ms;
+ std::optional<int64_t> last_sender_report_remote_timestamp_ms;
uint32_t sender_reports_packets_sent = 0;
uint64_t sender_reports_bytes_sent = 0;
uint64_t sender_reports_reports_count = 0;
diff --git a/call/video_send_stream.h b/call/video_send_stream.h
index 2c3fe55..bd22096 100644
--- a/call/video_send_stream.h
+++ b/call/video_send_stream.h
@@ -14,10 +14,10 @@
#include <stdint.h>
#include <map>
+#include <optional>
#include <string>
#include <vector>
-#include "absl/types/optional.h"
#include "api/adaptation/resource.h"
#include "api/call/transport.h"
#include "api/crypto/crypto_options.h"
@@ -72,7 +72,7 @@
// If `type` is kRtx or kFlexfec this value is present. The referenced SSRC
// is the kMedia stream that this stream is performing retransmissions or
// FEC for. If `type` is kMedia, this value is null.
- absl::optional<uint32_t> referenced_media_ssrc;
+ std::optional<uint32_t> referenced_media_ssrc;
FrameCounts frame_counts;
int width = 0;
int height = 0;
@@ -87,21 +87,21 @@
RtcpPacketTypeCounter rtcp_packet_type_counts;
// A snapshot of the most recent Report Block with additional data of
// interest to statistics. Used to implement RTCRemoteInboundRtpStreamStats.
- absl::optional<ReportBlockData> report_block_data;
+ std::optional<ReportBlockData> report_block_data;
double encode_frame_rate = 0.0;
int frames_encoded = 0;
- absl::optional<uint64_t> qp_sum;
+ std::optional<uint64_t> qp_sum;
uint64_t total_encode_time_ms = 0;
uint64_t total_encoded_bytes_target = 0;
uint32_t huge_frames_sent = 0;
- absl::optional<ScalabilityMode> scalability_mode;
+ std::optional<ScalabilityMode> scalability_mode;
};
struct Stats {
Stats();
~Stats();
std::string ToString(int64_t time_ms) const;
- absl::optional<std::string> encoder_implementation_name;
+ std::optional<std::string> encoder_implementation_name;
double input_frame_rate = 0;
int encode_frame_rate = 0;
int avg_encode_time_ms = 0;
@@ -145,7 +145,7 @@
webrtc::VideoContentType::UNSPECIFIED;
uint32_t frames_sent = 0;
uint32_t huge_frames_sent = 0;
- absl::optional<bool> power_efficient_encoder;
+ std::optional<bool> power_efficient_encoder;
};
struct Config {
diff --git a/common_audio/BUILD.gn b/common_audio/BUILD.gn
index 3603fdf..3d1688f 100644
--- a/common_audio/BUILD.gn
+++ b/common_audio/BUILD.gn
@@ -58,7 +58,6 @@
"../rtc_base/system:file_wrapper",
"../system_wrappers",
"third_party/ooura:fft_size_256",
- "//third_party/abseil-cpp/absl/types:optional",
]
defines = []
diff --git a/common_audio/mocks/mock_smoothing_filter.h b/common_audio/mocks/mock_smoothing_filter.h
index 9df49dd..0296659 100644
--- a/common_audio/mocks/mock_smoothing_filter.h
+++ b/common_audio/mocks/mock_smoothing_filter.h
@@ -19,7 +19,7 @@
class MockSmoothingFilter : public SmoothingFilter {
public:
MOCK_METHOD(void, AddSample, (float), (override));
- MOCK_METHOD(absl::optional<float>, GetAverage, (), (override));
+ MOCK_METHOD(std::optional<float>, GetAverage, (), (override));
MOCK_METHOD(bool, SetTimeConstantMs, (int), (override));
};
diff --git a/common_audio/smoothing_filter.cc b/common_audio/smoothing_filter.cc
index eaaf3a0..1c603f6 100644
--- a/common_audio/smoothing_filter.cc
+++ b/common_audio/smoothing_filter.cc
@@ -55,10 +55,10 @@
last_sample_ = sample;
}
-absl::optional<float> SmoothingFilterImpl::GetAverage() {
+std::optional<float> SmoothingFilterImpl::GetAverage() {
if (!init_end_time_ms_) {
// `init_end_time_ms_` undefined since we have not received any sample.
- return absl::nullopt;
+ return std::nullopt;
}
ExtrapolateLastSample(rtc::TimeMillis());
return state_;
diff --git a/common_audio/smoothing_filter.h b/common_audio/smoothing_filter.h
index 3419de7..488cc97 100644
--- a/common_audio/smoothing_filter.h
+++ b/common_audio/smoothing_filter.h
@@ -13,7 +13,7 @@
#include <stdint.h>
-#include "absl/types/optional.h"
+#include <optional>
namespace webrtc {
@@ -21,7 +21,7 @@
public:
virtual ~SmoothingFilter() = default;
virtual void AddSample(float sample) = 0;
- virtual absl::optional<float> GetAverage() = 0;
+ virtual std::optional<float> GetAverage() = 0;
virtual bool SetTimeConstantMs(int time_constant_ms) = 0;
};
@@ -49,7 +49,7 @@
~SmoothingFilterImpl() override;
void AddSample(float sample) override;
- absl::optional<float> GetAverage() override;
+ std::optional<float> GetAverage() override;
bool SetTimeConstantMs(int time_constant_ms) override;
// Methods used for unittests.
@@ -63,7 +63,7 @@
const float init_factor_;
const float init_const_;
- absl::optional<int64_t> init_end_time_ms_;
+ std::optional<int64_t> init_end_time_ms_;
float last_sample_;
float alpha_;
float state_;
diff --git a/common_video/BUILD.gn b/common_video/BUILD.gn
index 73a8adb..0270646 100644
--- a/common_video/BUILD.gn
+++ b/common_video/BUILD.gn
@@ -88,7 +88,6 @@
"../system_wrappers:metrics",
"//third_party/abseil-cpp/absl/container:inlined_vector",
"//third_party/abseil-cpp/absl/numeric:bits",
- "//third_party/abseil-cpp/absl/types:optional",
"//third_party/libyuv",
]
if (rtc_use_h265) {
@@ -164,7 +163,6 @@
"../test:test_support",
"../test:video_test_common",
"//testing/gtest",
- "//third_party/abseil-cpp/absl/types:optional",
"//third_party/libyuv",
]
diff --git a/common_video/bitrate_adjuster.cc b/common_video/bitrate_adjuster.cc
index c53c3a0..4053696 100644
--- a/common_video/bitrate_adjuster.cc
+++ b/common_video/bitrate_adjuster.cc
@@ -67,7 +67,7 @@
return adjusted_bitrate_bps_;
}
-absl::optional<uint32_t> BitrateAdjuster::GetEstimatedBitrateBps() {
+std::optional<uint32_t> BitrateAdjuster::GetEstimatedBitrateBps() {
MutexLock lock(&mutex_);
return bitrate_tracker_.Rate(rtc::TimeMillis());
}
diff --git a/common_video/corruption_detection_message.h b/common_video/corruption_detection_message.h
index b6572c9..5fa2aed 100644
--- a/common_video/corruption_detection_message.h
+++ b/common_video/corruption_detection_message.h
@@ -12,9 +12,9 @@
#define COMMON_VIDEO_CORRUPTION_DETECTION_MESSAGE_H_
#include <cstddef>
+#include <optional>
#include "absl/container/inlined_vector.h"
-#include "absl/types/optional.h"
#include "api/array_view.h"
namespace webrtc {
@@ -84,28 +84,28 @@
~Builder() = default;
- absl::optional<CorruptionDetectionMessage> Build() {
+ std::optional<CorruptionDetectionMessage> Build() {
if (message_.sequence_index_ < 0 ||
message_.sequence_index_ > 0b0111'1111) {
- return absl::nullopt;
+ return std::nullopt;
}
if (message_.std_dev_ < 0.0 || message_.std_dev_ > 40.0) {
- return absl::nullopt;
+ return std::nullopt;
}
if (message_.luma_error_threshold_ < 0 ||
message_.luma_error_threshold_ > 15) {
- return absl::nullopt;
+ return std::nullopt;
}
if (message_.chroma_error_threshold_ < 0 ||
message_.chroma_error_threshold_ > 15) {
- return absl::nullopt;
+ return std::nullopt;
}
if (message_.sample_values_.size() > kMaxSampleSize) {
- return absl::nullopt;
+ return std::nullopt;
}
for (double sample_value : message_.sample_values_) {
if (sample_value < 0.0 || sample_value > 255.0) {
- return absl::nullopt;
+ return std::nullopt;
}
}
return message_;
diff --git a/common_video/corruption_detection_message_unittest.cc b/common_video/corruption_detection_message_unittest.cc
index ee11161..4bb47ce 100644
--- a/common_video/corruption_detection_message_unittest.cc
+++ b/common_video/corruption_detection_message_unittest.cc
@@ -10,9 +10,9 @@
#include "common_video/corruption_detection_message.h"
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "test/gtest.h"
namespace webrtc {
@@ -22,36 +22,36 @@
EXPECT_EQ(CorruptionDetectionMessage::Builder()
.WithSequenceIndex(0b1000'0000)
.Build(),
- absl::nullopt);
+ std::nullopt);
}
TEST(CorruptionDetectionMessageTest, FailsToCreateWhenSequenceIndexIsTooSmall) {
EXPECT_EQ(CorruptionDetectionMessage::Builder().WithSequenceIndex(-1).Build(),
- absl::nullopt);
+ std::nullopt);
}
TEST(CorruptionDetectionMessageTest, FailsToCreateWhenStddevIsTooLarge) {
EXPECT_EQ(CorruptionDetectionMessage::Builder().WithStdDev(45.0).Build(),
- absl::nullopt);
+ std::nullopt);
}
TEST(CorruptionDetectionMessageTest, FailsToCreateWhenStddevIsTooSmall) {
EXPECT_EQ(CorruptionDetectionMessage::Builder().WithStdDev(-1.0).Build(),
- absl::nullopt);
+ std::nullopt);
}
TEST(CorruptionDetectionMessageTest,
FailsToCreateWhenLumaErrorThresholdIsTooLarge) {
EXPECT_EQ(
CorruptionDetectionMessage::Builder().WithLumaErrorThreshold(16).Build(),
- absl::nullopt);
+ std::nullopt);
}
TEST(CorruptionDetectionMessageTest,
FailsToCreateWhenLumaErrorThresholdIsTooSmall) {
EXPECT_EQ(
CorruptionDetectionMessage::Builder().WithLumaErrorThreshold(-1).Build(),
- absl::nullopt);
+ std::nullopt);
}
TEST(CorruptionDetectionMessageTest,
@@ -59,7 +59,7 @@
EXPECT_EQ(CorruptionDetectionMessage::Builder()
.WithChromaErrorThreshold(16)
.Build(),
- absl::nullopt);
+ std::nullopt);
}
TEST(CorruptionDetectionMessageTest,
@@ -67,7 +67,7 @@
EXPECT_EQ(CorruptionDetectionMessage::Builder()
.WithChromaErrorThreshold(-1)
.Build(),
- absl::nullopt);
+ std::nullopt);
}
TEST(CorruptionDetectionMessageTest,
@@ -79,7 +79,7 @@
EXPECT_EQ(CorruptionDetectionMessage::Builder()
.WithSampleValues(kSampleValues)
.Build(),
- absl::nullopt);
+ std::nullopt);
}
TEST(CorruptionDetectionMessageTest, FailsToCreateWhenSampleValueIsTooLarge) {
@@ -88,7 +88,7 @@
EXPECT_EQ(CorruptionDetectionMessage::Builder()
.WithSampleValues(kSampleValues)
.Build(),
- absl::nullopt);
+ std::nullopt);
}
TEST(CorruptionDetectionMessageTest, FailsToCreateWhenSampleValueIsTooSmall) {
@@ -97,12 +97,12 @@
EXPECT_EQ(CorruptionDetectionMessage::Builder()
.WithSampleValues(kSampleValues)
.Build(),
- absl::nullopt);
+ std::nullopt);
}
TEST(CorruptionDetectionMessageTest,
CreatesDefaultWhenNoParametersAreSpecified) {
- EXPECT_NE(CorruptionDetectionMessage::Builder().Build(), absl::nullopt);
+ EXPECT_NE(CorruptionDetectionMessage::Builder().Build(), std::nullopt);
}
TEST(CorruptionDetectionMessageTest, CreatesWhenValidParametersAreSpecified) {
@@ -117,7 +117,7 @@
.WithChromaErrorThreshold(15)
.WithSampleValues(kSampleValues)
.Build(),
- absl::nullopt);
+ std::nullopt);
}
} // namespace
diff --git a/common_video/frame_rate_estimator.cc b/common_video/frame_rate_estimator.cc
index 4c5a341..2ac7aee 100644
--- a/common_video/frame_rate_estimator.cc
+++ b/common_video/frame_rate_estimator.cc
@@ -22,13 +22,13 @@
frame_times_.push_back(time);
}
-absl::optional<double> FrameRateEstimator::GetAverageFps() const {
+std::optional<double> FrameRateEstimator::GetAverageFps() const {
if (frame_times_.size() < 2) {
- return absl::nullopt;
+ return std::nullopt;
}
TimeDelta time_span = frame_times_.back() - frame_times_.front();
if (time_span < TimeDelta::Micros(1)) {
- return absl::nullopt;
+ return std::nullopt;
}
TimeDelta avg_frame_interval = time_span / (frame_times_.size() - 1);
@@ -36,7 +36,7 @@
avg_frame_interval.us();
}
-absl::optional<double> FrameRateEstimator::GetAverageFps(Timestamp now) {
+std::optional<double> FrameRateEstimator::GetAverageFps(Timestamp now) {
CullOld(now);
return GetAverageFps();
}
diff --git a/common_video/frame_rate_estimator.h b/common_video/frame_rate_estimator.h
index 95219a5..55e6e20 100644
--- a/common_video/frame_rate_estimator.h
+++ b/common_video/frame_rate_estimator.h
@@ -12,8 +12,8 @@
#define COMMON_VIDEO_FRAME_RATE_ESTIMATOR_H_
#include <deque>
+#include <optional>
-#include "absl/types/optional.h"
#include "api/units/time_delta.h"
#include "api/units/timestamp.h"
@@ -41,10 +41,10 @@
void OnFrame(Timestamp time);
// Get the current average FPS, based on the frames currently in the window.
- absl::optional<double> GetAverageFps() const;
+ std::optional<double> GetAverageFps() const;
// Move the window so it ends at `now`, and return the new fps estimate.
- absl::optional<double> GetAverageFps(Timestamp now);
+ std::optional<double> GetAverageFps(Timestamp now);
// Completely clear the averaging window.
void Reset();
diff --git a/common_video/framerate_controller.cc b/common_video/framerate_controller.cc
index 23e9c70..4e2e0b8 100644
--- a/common_video/framerate_controller.cc
+++ b/common_video/framerate_controller.cc
@@ -71,7 +71,7 @@
void FramerateController::Reset() {
max_framerate_ = std::numeric_limits<double>::max();
- next_frame_timestamp_ns_ = absl::nullopt;
+ next_frame_timestamp_ns_ = std::nullopt;
}
void FramerateController::KeepFrame(int64_t in_timestamp_ns) {
diff --git a/common_video/framerate_controller.h b/common_video/framerate_controller.h
index 371ffd4..44e2e67 100644
--- a/common_video/framerate_controller.h
+++ b/common_video/framerate_controller.h
@@ -13,7 +13,7 @@
#include <stdint.h>
-#include "absl/types/optional.h"
+#include <optional>
namespace webrtc {
@@ -38,7 +38,7 @@
private:
double max_framerate_;
- absl::optional<int64_t> next_frame_timestamp_ns_;
+ std::optional<int64_t> next_frame_timestamp_ns_;
};
} // namespace webrtc
diff --git a/common_video/h264/h264_bitstream_parser.cc b/common_video/h264/h264_bitstream_parser.cc
index 5c0ebb8..96be4c8 100644
--- a/common_video/h264/h264_bitstream_parser.cc
+++ b/common_video/h264/h264_bitstream_parser.cc
@@ -36,7 +36,7 @@
if (!sps_ || !pps_)
return kInvalidStream;
- last_slice_qp_delta_ = absl::nullopt;
+ last_slice_qp_delta_ = std::nullopt;
const std::vector<uint8_t> slice_rbsp = H264::ParseRbsp(source);
if (slice_rbsp.size() < H264::kNaluTypeSize)
return kInvalidStream;
@@ -344,13 +344,13 @@
bitstream.subview(index.payload_start_offset, index.payload_size));
}
-absl::optional<int> H264BitstreamParser::GetLastSliceQp() const {
+std::optional<int> H264BitstreamParser::GetLastSliceQp() const {
if (!last_slice_qp_delta_ || !pps_)
- return absl::nullopt;
+ return std::nullopt;
const int qp = 26 + pps_->pic_init_qp_minus26 + *last_slice_qp_delta_;
if (qp < kMinQpValue || qp > kMaxQpValue) {
RTC_LOG(LS_ERROR) << "Parsed invalid QP from bitstream.";
- return absl::nullopt;
+ return std::nullopt;
}
return qp;
}
diff --git a/common_video/h264/h264_bitstream_parser.h b/common_video/h264/h264_bitstream_parser.h
index 1fad989..5b605b5 100644
--- a/common_video/h264/h264_bitstream_parser.h
+++ b/common_video/h264/h264_bitstream_parser.h
@@ -13,7 +13,8 @@
#include <stddef.h>
#include <stdint.h>
-#include "absl/types/optional.h"
+#include <optional>
+
#include "api/video_codecs/bitstream_parser.h"
#include "common_video/h264/pps_parser.h"
#include "common_video/h264/sps_parser.h"
@@ -32,7 +33,7 @@
~H264BitstreamParser() override;
void ParseBitstream(rtc::ArrayView<const uint8_t> bitstream) override;
- absl::optional<int> GetLastSliceQp() const override;
+ std::optional<int> GetLastSliceQp() const override;
protected:
enum Result {
@@ -45,11 +46,11 @@
uint8_t nalu_type);
// SPS/PPS state, updated when parsing new SPS/PPS, used to parse slices.
- absl::optional<SpsParser::SpsState> sps_;
- absl::optional<PpsParser::PpsState> pps_;
+ std::optional<SpsParser::SpsState> sps_;
+ std::optional<PpsParser::PpsState> pps_;
// Last parsed slice QP.
- absl::optional<int32_t> last_slice_qp_delta_;
+ std::optional<int32_t> last_slice_qp_delta_;
};
} // namespace webrtc
diff --git a/common_video/h264/h264_bitstream_parser_unittest.cc b/common_video/h264/h264_bitstream_parser_unittest.cc
index e03da7a..00186a6 100644
--- a/common_video/h264/h264_bitstream_parser_unittest.cc
+++ b/common_video/h264/h264_bitstream_parser_unittest.cc
@@ -122,7 +122,7 @@
TEST(H264BitstreamParserTest, ReportsLastSliceQpForImageSlices) {
H264BitstreamParser h264_parser;
h264_parser.ParseBitstream(kH264BitstreamChunk);
- absl::optional<int> qp = h264_parser.GetLastSliceQp();
+ std::optional<int> qp = h264_parser.GetLastSliceQp();
ASSERT_TRUE(qp.has_value());
EXPECT_EQ(35, *qp);
@@ -140,7 +140,7 @@
// Parse an additional image slice.
h264_parser.ParseBitstream(kH264BitstreamNextImageSliceChunkCabac);
- absl::optional<int> qp = h264_parser.GetLastSliceQp();
+ std::optional<int> qp = h264_parser.GetLastSliceQp();
ASSERT_TRUE(qp.has_value());
EXPECT_EQ(24, *qp);
}
@@ -149,14 +149,14 @@
H264BitstreamParser h264_parser;
h264_parser.ParseBitstream(kH264BitstreamWeightedPred);
- absl::optional<int> qp = h264_parser.GetLastSliceQp();
+ std::optional<int> qp = h264_parser.GetLastSliceQp();
ASSERT_TRUE(qp.has_value());
EXPECT_EQ(11, *qp);
}
TEST(H264BitstreamParserTest, ReportsLastSliceQpForWeightedPredSlicesL0Active) {
H264BitstreamParser h264_parser;
- absl::optional<int> qp;
+ std::optional<int> qp;
h264_parser.ParseBitstream(H264BitstreamCVWP1SPS);
h264_parser.ParseBitstream(H264BitstreamCVWP1PFrame1);
@@ -182,7 +182,7 @@
TEST(H264BitstreamParserTest, ReportsLastSliceQpForWeightedPredSlicesL1Active) {
H264BitstreamParser h264_parser;
- absl::optional<int> qp;
+ std::optional<int> qp;
h264_parser.ParseBitstream(H264BitstreamCVWP2SPS);
h264_parser.ParseBitstream(H264BitstreamCVWP2BFrame1);
diff --git a/common_video/h264/pps_parser.cc b/common_video/h264/pps_parser.cc
index f12ab9b..2518765 100644
--- a/common_video/h264/pps_parser.cc
+++ b/common_video/h264/pps_parser.cc
@@ -29,7 +29,7 @@
// You can find it on this page:
// http://www.itu.int/rec/T-REC-H.264
-absl::optional<PpsParser::PpsState> PpsParser::ParsePps(
+std::optional<PpsParser::PpsState> PpsParser::ParsePps(
rtc::ArrayView<const uint8_t> data) {
// First, parse out rbsp, which is basically the source buffer minus emulation
// bytes (the last byte of a 0x00 0x00 0x03 sequence). RBSP is defined in
@@ -52,7 +52,7 @@
return reader.Ok();
}
-absl::optional<PpsParser::SliceHeader> PpsParser::ParseSliceHeader(
+std::optional<PpsParser::SliceHeader> PpsParser::ParseSliceHeader(
rtc::ArrayView<const uint8_t> data) {
std::vector<uint8_t> unpacked_buffer = H264::ParseRbsp(data);
BitstreamReader slice_reader(unpacked_buffer);
@@ -68,12 +68,12 @@
// The rest of the slice header requires information from the SPS to parse.
if (!slice_reader.Ok()) {
- return absl::nullopt;
+ return std::nullopt;
}
return slice_header;
}
-absl::optional<PpsParser::PpsState> PpsParser::ParseInternal(
+std::optional<PpsParser::PpsState> PpsParser::ParseInternal(
rtc::ArrayView<const uint8_t> buffer) {
BitstreamReader reader(buffer);
PpsState pps;
@@ -123,7 +123,7 @@
int64_t bits_to_consume =
int64_t{slice_group_id_bits} * pic_size_in_map_units;
if (!reader.Ok() || bits_to_consume > std::numeric_limits<int>::max()) {
- return absl::nullopt;
+ return std::nullopt;
}
reader.ConsumeBits(bits_to_consume);
}
@@ -134,7 +134,7 @@
pps.num_ref_idx_l1_default_active_minus1 = reader.ReadExponentialGolomb();
if (pps.num_ref_idx_l0_default_active_minus1 > H264::kMaxReferenceIndex ||
pps.num_ref_idx_l1_default_active_minus1 > H264::kMaxReferenceIndex) {
- return absl::nullopt;
+ return std::nullopt;
}
// weighted_pred_flag: u(1)
pps.weighted_pred_flag = reader.Read<bool>();
@@ -146,7 +146,7 @@
// Sanity-check parsed value
if (!reader.Ok() || pps.pic_init_qp_minus26 > kMaxPicInitQpDeltaValue ||
pps.pic_init_qp_minus26 < kMinPicInitQpDeltaValue) {
- return absl::nullopt;
+ return std::nullopt;
}
// pic_init_qs_minus26: se(v)
reader.ReadExponentialGolomb();
@@ -158,7 +158,7 @@
// redundant_pic_cnt_present_flag: u(1)
pps.redundant_pic_cnt_present_flag = reader.ReadBit();
if (!reader.Ok()) {
- return absl::nullopt;
+ return std::nullopt;
}
return pps;
diff --git a/common_video/h264/pps_parser.h b/common_video/h264/pps_parser.h
index d2d0609..e4207d7 100644
--- a/common_video/h264/pps_parser.h
+++ b/common_video/h264/pps_parser.h
@@ -14,7 +14,8 @@
#include <stddef.h>
#include <stdint.h>
-#include "absl/types/optional.h"
+#include <optional>
+
#include "api/array_view.h"
namespace webrtc {
@@ -47,10 +48,10 @@
};
// Unpack RBSP and parse PPS state from the supplied buffer.
- static absl::optional<PpsState> ParsePps(rtc::ArrayView<const uint8_t> data);
+ static std::optional<PpsState> ParsePps(rtc::ArrayView<const uint8_t> data);
// TODO: bugs.webrtc.org/42225170 - Deprecate.
- static inline absl::optional<PpsState> ParsePps(const uint8_t* data,
- size_t length) {
+ static inline std::optional<PpsState> ParsePps(const uint8_t* data,
+ size_t length) {
return ParsePps(rtc::MakeArrayView(data, length));
}
@@ -58,13 +59,13 @@
uint32_t* pps_id,
uint32_t* sps_id);
- static absl::optional<SliceHeader> ParseSliceHeader(
+ static std::optional<SliceHeader> ParseSliceHeader(
rtc::ArrayView<const uint8_t> data);
protected:
// Parse the PPS state, for a buffer where RBSP decoding has already been
// performed.
- static absl::optional<PpsState> ParseInternal(
+ static std::optional<PpsState> ParseInternal(
rtc::ArrayView<const uint8_t> buffer);
};
diff --git a/common_video/h264/pps_parser_unittest.cc b/common_video/h264/pps_parser_unittest.cc
index a325e8c..ea8c8ed 100644
--- a/common_video/h264/pps_parser_unittest.cc
+++ b/common_video/h264/pps_parser_unittest.cc
@@ -196,7 +196,7 @@
PpsParser::PpsState generated_pps_;
rtc::Buffer buffer_;
- absl::optional<PpsParser::PpsState> parsed_pps_;
+ std::optional<PpsParser::PpsState> parsed_pps_;
};
TEST_F(PpsParserTest, ZeroPps) {
@@ -227,7 +227,7 @@
H264::ParseNaluType(chunk[index.payload_start_offset]);
if (nalu_type == H264::NaluType::kIdr) {
// Skip NAL type header and parse slice header.
- absl::optional<PpsParser::SliceHeader> slice_header =
+ std::optional<PpsParser::SliceHeader> slice_header =
PpsParser::ParseSliceHeader(chunk.subview(
index.payload_start_offset + 1, index.payload_size - 1));
ASSERT_TRUE(slice_header.has_value());
diff --git a/common_video/h264/sps_parser.cc b/common_video/h264/sps_parser.cc
index 000e028..936080e 100644
--- a/common_video/h264/sps_parser.cc
+++ b/common_video/h264/sps_parser.cc
@@ -32,14 +32,14 @@
// http://www.itu.int/rec/T-REC-H.264
// Unpack RBSP and parse SPS state from the supplied buffer.
-absl::optional<SpsParser::SpsState> SpsParser::ParseSps(
+std::optional<SpsParser::SpsState> SpsParser::ParseSps(
rtc::ArrayView<const uint8_t> data) {
std::vector<uint8_t> unpacked_buffer = H264::ParseRbsp(data);
BitstreamReader reader(unpacked_buffer);
return ParseSpsUpToVui(reader);
}
-absl::optional<SpsParser::SpsState> SpsParser::ParseSpsUpToVui(
+std::optional<SpsParser::SpsState> SpsParser::ParseSpsUpToVui(
BitstreamReader& reader) {
// Now, we need to use a bitstream reader to parse through the actual AVC SPS
// format. See Section 7.3.2.1.1 ("Sequence parameter set data syntax") of the
@@ -102,7 +102,7 @@
int delta_scale = reader.ReadSignedExponentialGolomb();
if (!reader.Ok() || delta_scale < kScalingDeltaMin ||
delta_scale > kScaldingDeltaMax) {
- return absl::nullopt;
+ return std::nullopt;
}
next_scale = (last_scale + delta_scale + 256) % 256;
}
@@ -122,7 +122,7 @@
// log2_max_frame_num_minus4: ue(v)
uint32_t log2_max_frame_num_minus4 = reader.ReadExponentialGolomb();
if (!reader.Ok() || log2_max_frame_num_minus4 > kMaxLog2Minus4) {
- return absl::nullopt;
+ return std::nullopt;
}
sps.log2_max_frame_num = log2_max_frame_num_minus4 + 4;
@@ -132,7 +132,7 @@
// log2_max_pic_order_cnt_lsb_minus4: ue(v)
uint32_t log2_max_pic_order_cnt_lsb_minus4 = reader.ReadExponentialGolomb();
if (!reader.Ok() || log2_max_pic_order_cnt_lsb_minus4 > kMaxLog2Minus4) {
- return absl::nullopt;
+ return std::nullopt;
}
sps.log2_max_pic_order_cnt_lsb = log2_max_pic_order_cnt_lsb_minus4 + 4;
} else if (sps.pic_order_cnt_type == 1) {
@@ -149,7 +149,7 @@
// offset_for_ref_frame[i]: se(v)
reader.ReadExponentialGolomb();
if (!reader.Ok()) {
- return absl::nullopt;
+ return std::nullopt;
}
}
}
@@ -197,7 +197,7 @@
// Far enough! We don't use the rest of the SPS.
if (!reader.Ok()) {
- return absl::nullopt;
+ return std::nullopt;
}
// Figure out the crop units in pixels. That's based on the chroma format's
diff --git a/common_video/h264/sps_parser.h b/common_video/h264/sps_parser.h
index ae8df60..7347e6d 100644
--- a/common_video/h264/sps_parser.h
+++ b/common_video/h264/sps_parser.h
@@ -11,7 +11,8 @@
#ifndef COMMON_VIDEO_H264_SPS_PARSER_H_
#define COMMON_VIDEO_H264_SPS_PARSER_H_
-#include "absl/types/optional.h"
+#include <optional>
+
#include "rtc_base/bitstream_reader.h"
#include "rtc_base/system/rtc_export.h"
@@ -42,17 +43,17 @@
};
// Unpack RBSP and parse SPS state from the supplied buffer.
- static absl::optional<SpsState> ParseSps(rtc::ArrayView<const uint8_t> data);
+ static std::optional<SpsState> ParseSps(rtc::ArrayView<const uint8_t> data);
// TODO: bugs.webrtc.org/42225170 - Deprecate.
- static inline absl::optional<SpsState> ParseSps(const uint8_t* data,
- size_t length) {
+ static inline std::optional<SpsState> ParseSps(const uint8_t* data,
+ size_t length) {
return ParseSps(rtc::MakeArrayView(data, length));
}
protected:
// Parse the SPS state, up till the VUI part, for a buffer where RBSP
// decoding has already been performed.
- static absl::optional<SpsState> ParseSpsUpToVui(BitstreamReader& reader);
+ static std::optional<SpsState> ParseSpsUpToVui(BitstreamReader& reader);
};
} // namespace webrtc
diff --git a/common_video/h264/sps_parser_unittest.cc b/common_video/h264/sps_parser_unittest.cc
index 721aa3f..6904e8e 100644
--- a/common_video/h264/sps_parser_unittest.cc
+++ b/common_video/h264/sps_parser_unittest.cc
@@ -116,7 +116,7 @@
const uint8_t buffer[] = {0x7A, 0x00, 0x1F, 0xBC, 0xD9, 0x40, 0x50, 0x05,
0xBA, 0x10, 0x00, 0x00, 0x03, 0x00, 0xC0, 0x00,
0x00, 0x2A, 0xE0, 0xF1, 0x83, 0x19, 0x60};
- absl::optional<SpsParser::SpsState> sps = SpsParser::ParseSps(buffer);
+ std::optional<SpsParser::SpsState> sps = SpsParser::ParseSps(buffer);
ASSERT_TRUE(sps.has_value());
EXPECT_EQ(1280u, sps->width);
EXPECT_EQ(720u, sps->height);
@@ -128,7 +128,7 @@
const uint8_t buffer[] = {0x7A, 0x00, 0x1E, 0xBC, 0xD9, 0x40, 0xA0, 0x2F,
0xF8, 0x98, 0x40, 0x00, 0x00, 0x03, 0x01, 0x80,
0x00, 0x00, 0x56, 0x83, 0xC5, 0x8B, 0x65, 0x80};
- absl::optional<SpsParser::SpsState> sps = SpsParser::ParseSps(buffer);
+ std::optional<SpsParser::SpsState> sps = SpsParser::ParseSps(buffer);
ASSERT_TRUE(sps.has_value());
EXPECT_EQ(640u, sps->width);
EXPECT_EQ(360u, sps->height);
@@ -140,7 +140,7 @@
const uint8_t buffer[] = {0x7A, 0x00, 0x0D, 0xBC, 0xD9, 0x43, 0x43, 0x3E,
0x5E, 0x10, 0x00, 0x00, 0x03, 0x00, 0x60, 0x00,
0x00, 0x15, 0xA0, 0xF1, 0x42, 0x99, 0x60};
- absl::optional<SpsParser::SpsState> sps = SpsParser::ParseSps(buffer);
+ std::optional<SpsParser::SpsState> sps = SpsParser::ParseSps(buffer);
ASSERT_TRUE(sps.has_value());
EXPECT_EQ(200u, sps->width);
EXPECT_EQ(400u, sps->height);
@@ -149,7 +149,7 @@
TEST(H264SpsParserTest, TestSyntheticSPSQvgaLandscape) {
rtc::Buffer buffer;
GenerateFakeSps(320u, 180u, 1, 0, 0, &buffer);
- absl::optional<SpsParser::SpsState> sps = SpsParser::ParseSps(buffer);
+ std::optional<SpsParser::SpsState> sps = SpsParser::ParseSps(buffer);
ASSERT_TRUE(sps.has_value());
EXPECT_EQ(320u, sps->width);
EXPECT_EQ(180u, sps->height);
@@ -159,7 +159,7 @@
TEST(H264SpsParserTest, TestSyntheticSPSWeirdResolution) {
rtc::Buffer buffer;
GenerateFakeSps(156u, 122u, 2, 0, 0, &buffer);
- absl::optional<SpsParser::SpsState> sps = SpsParser::ParseSps(buffer);
+ std::optional<SpsParser::SpsState> sps = SpsParser::ParseSps(buffer);
ASSERT_TRUE(sps.has_value());
EXPECT_EQ(156u, sps->width);
EXPECT_EQ(122u, sps->height);
@@ -173,7 +173,7 @@
0x10, 0xc2, 0x00, 0x84, 0x3b, 0x50, 0x3c, 0x01,
0x13, 0xf2, 0xcd, 0xc0, 0x40, 0x40, 0x50, 0x00,
0x00, 0x00, 0x10, 0x00, 0x00, 0x01, 0xe8, 0x40};
- absl::optional<SpsParser::SpsState> sps = SpsParser::ParseSps(buffer);
+ std::optional<SpsParser::SpsState> sps = SpsParser::ParseSps(buffer);
ASSERT_TRUE(sps.has_value());
EXPECT_EQ(1920u, sps->width);
EXPECT_EQ(1080u, sps->height);
@@ -182,7 +182,7 @@
TEST(H264SpsParserTest, TestLog2MaxFrameNumMinus4) {
rtc::Buffer buffer;
GenerateFakeSps(320u, 180u, 1, 0, 0, &buffer);
- absl::optional<SpsParser::SpsState> sps = SpsParser::ParseSps(buffer);
+ std::optional<SpsParser::SpsState> sps = SpsParser::ParseSps(buffer);
ASSERT_TRUE(sps.has_value());
EXPECT_EQ(320u, sps->width);
EXPECT_EQ(180u, sps->height);
@@ -204,7 +204,7 @@
TEST(H264SpsParserTest, TestLog2MaxPicOrderCntMinus4) {
rtc::Buffer buffer;
GenerateFakeSps(320u, 180u, 1, 0, 0, &buffer);
- absl::optional<SpsParser::SpsState> sps = SpsParser::ParseSps(buffer);
+ std::optional<SpsParser::SpsState> sps = SpsParser::ParseSps(buffer);
ASSERT_TRUE(sps.has_value());
EXPECT_EQ(320u, sps->width);
EXPECT_EQ(180u, sps->height);
diff --git a/common_video/h264/sps_vui_rewriter.cc b/common_video/h264/sps_vui_rewriter.cc
index fa824a5..236ecb8 100644
--- a/common_video/h264/sps_vui_rewriter.cc
+++ b/common_video/h264/sps_vui_rewriter.cc
@@ -136,14 +136,14 @@
SpsVuiRewriter::ParseResult SpsVuiRewriter::ParseAndRewriteSps(
rtc::ArrayView<const uint8_t> buffer,
- absl::optional<SpsParser::SpsState>* sps,
+ std::optional<SpsParser::SpsState>* sps,
const webrtc::ColorSpace* color_space,
rtc::Buffer* destination) {
// Create temporary RBSP decoded buffer of the payload (exlcuding the
// leading nalu type header byte (the SpsParser uses only the payload).
std::vector<uint8_t> rbsp_buffer = H264::ParseRbsp(buffer);
BitstreamReader source_buffer(rbsp_buffer);
- absl::optional<SpsParser::SpsState> sps_state =
+ std::optional<SpsParser::SpsState> sps_state =
SpsParser::ParseSpsUpToVui(source_buffer);
if (!sps_state)
return ParseResult::kFailure;
@@ -212,7 +212,7 @@
SpsVuiRewriter::ParseResult SpsVuiRewriter::ParseAndRewriteSps(
rtc::ArrayView<const uint8_t> buffer,
- absl::optional<SpsParser::SpsState>* sps,
+ std::optional<SpsParser::SpsState>* sps,
const webrtc::ColorSpace* color_space,
rtc::Buffer* destination,
Direction direction) {
@@ -253,7 +253,7 @@
// protect legacy receive clients) in RtpDepacketizerH264::ParseSingleNalu
// (receive side, in orderer to protect us from unknown or legacy send
// clients).
- absl::optional<SpsParser::SpsState> sps;
+ std::optional<SpsParser::SpsState> sps;
rtc::Buffer output_nalu;
// Add the type header to the output buffer first, so that the rewriter
diff --git a/common_video/h264/sps_vui_rewriter.h b/common_video/h264/sps_vui_rewriter.h
index 66f7e1a..9323219 100644
--- a/common_video/h264/sps_vui_rewriter.h
+++ b/common_video/h264/sps_vui_rewriter.h
@@ -15,7 +15,8 @@
#include <stddef.h>
#include <stdint.h>
-#include "absl/types/optional.h"
+#include <optional>
+
#include "api/video/color_space.h"
#include "common_video/h264/sps_parser.h"
#include "rtc_base/buffer.h"
@@ -42,12 +43,11 @@
// SPS state. This function assumes that any previous headers
// (NALU start, type, Stap-A, etc) have already been parsed and that RBSP
// decoding has been performed.
- static ParseResult ParseAndRewriteSps(
- rtc::ArrayView<const uint8_t> buffer,
- absl::optional<SpsParser::SpsState>* sps,
- const ColorSpace* color_space,
- rtc::Buffer* destination,
- Direction Direction);
+ static ParseResult ParseAndRewriteSps(rtc::ArrayView<const uint8_t> buffer,
+ std::optional<SpsParser::SpsState>* sps,
+ const ColorSpace* color_space,
+ rtc::Buffer* destination,
+ Direction Direction);
// Parses NAL units from `buffer`, strips AUD blocks and rewrites VUI in SPS
// blocks if necessary.
@@ -56,11 +56,10 @@
const ColorSpace* color_space);
private:
- static ParseResult ParseAndRewriteSps(
- rtc::ArrayView<const uint8_t> buffer,
- absl::optional<SpsParser::SpsState>* sps,
- const ColorSpace* color_space,
- rtc::Buffer* destination);
+ static ParseResult ParseAndRewriteSps(rtc::ArrayView<const uint8_t> buffer,
+ std::optional<SpsParser::SpsState>* sps,
+ const ColorSpace* color_space,
+ rtc::Buffer* destination);
static void UpdateStats(ParseResult result, Direction direction);
};
diff --git a/common_video/h264/sps_vui_rewriter_unittest.cc b/common_video/h264/sps_vui_rewriter_unittest.cc
index d440905..4fbfdfe 100644
--- a/common_video/h264/sps_vui_rewriter_unittest.cc
+++ b/common_video/h264/sps_vui_rewriter_unittest.cc
@@ -307,7 +307,7 @@
rtc::Buffer original_sps;
GenerateFakeSps(vui, &original_sps);
- absl::optional<SpsParser::SpsState> sps;
+ std::optional<SpsParser::SpsState> sps;
rtc::Buffer rewritten_sps;
SpsVuiRewriter::ParseResult result = SpsVuiRewriter::ParseAndRewriteSps(
original_sps, &sps, color_space, &rewritten_sps,
diff --git a/common_video/h265/h265_bitstream_parser.cc b/common_video/h265/h265_bitstream_parser.cc
index ee41a79..9ee6ff0 100644
--- a/common_video/h265/h265_bitstream_parser.cc
+++ b/common_video/h265/h265_bitstream_parser.cc
@@ -38,7 +38,7 @@
" to be" \
<< " in range [" << (min) << ":" << (max) << "]" \
<< " found " << (val) << " instead"; \
- return absl::nullopt; \
+ return std::nullopt; \
} \
} while (0)
@@ -82,8 +82,8 @@
H265BitstreamParser::Result H265BitstreamParser::ParseNonParameterSetNalu(
rtc::ArrayView<const uint8_t> source,
uint8_t nalu_type) {
- last_slice_qp_delta_ = absl::nullopt;
- last_slice_pps_id_ = absl::nullopt;
+ last_slice_qp_delta_ = std::nullopt;
+ last_slice_pps_id_ = std::nullopt;
const std::vector<uint8_t> slice_rbsp = H265::ParseRbsp(source);
if (slice_rbsp.size() < H265::kNaluHeaderSize)
return kInvalidStream;
@@ -165,7 +165,7 @@
// short_term_ref_pic_set_sps_flag: u(1)
short_term_ref_pic_set_sps_flag = slice_reader.Read<bool>();
if (!short_term_ref_pic_set_sps_flag) {
- absl::optional<H265SpsParser::ShortTermRefPicSet> ref_pic_set =
+ std::optional<H265SpsParser::ShortTermRefPicSet> ref_pic_set =
H265SpsParser::ParseShortTermRefPicSet(
sps->num_short_term_ref_pic_sets,
sps->num_short_term_ref_pic_sets, sps->short_term_ref_pic_set,
@@ -426,7 +426,7 @@
H265::NaluType nalu_type = H265::ParseNaluType(slice[0]);
switch (nalu_type) {
case H265::NaluType::kVps: {
- absl::optional<H265VpsParser::VpsState> vps_state;
+ std::optional<H265VpsParser::VpsState> vps_state;
if (slice.size() >= H265::kNaluHeaderSize) {
vps_state =
H265VpsParser::ParseVps(slice.subview(H265::kNaluHeaderSize));
@@ -440,7 +440,7 @@
break;
}
case H265::NaluType::kSps: {
- absl::optional<H265SpsParser::SpsState> sps_state;
+ std::optional<H265SpsParser::SpsState> sps_state;
if (slice.size() >= H265::kNaluHeaderSize) {
sps_state =
H265SpsParser::ParseSps(slice.subview(H265::kNaluHeaderSize));
@@ -453,7 +453,7 @@
break;
}
case H265::NaluType::kPps: {
- absl::optional<H265PpsParser::PpsState> pps_state;
+ std::optional<H265PpsParser::PpsState> pps_state;
if (slice.size() >= H265::kNaluHeaderSize) {
std::vector<uint8_t> unpacked_buffer =
H265::ParseRbsp(slice.subview(H265::kNaluHeaderSize));
@@ -490,7 +490,7 @@
}
}
-absl::optional<uint32_t>
+std::optional<uint32_t>
H265BitstreamParser::ParsePpsIdFromSliceSegmentLayerRbsp(
rtc::ArrayView<const uint8_t> data,
uint8_t nalu_type) {
@@ -500,7 +500,7 @@
// first_slice_segment_in_pic_flag: u(1)
slice_reader.ConsumeBits(1);
if (!slice_reader.Ok()) {
- return absl::nullopt;
+ return std::nullopt;
}
if (nalu_type >= H265::NaluType::kBlaWLp &&
@@ -513,7 +513,7 @@
uint32_t slice_pic_parameter_set_id = slice_reader.ReadExponentialGolomb();
IN_RANGE_OR_RETURN_NULL(slice_pic_parameter_set_id, 0, 63);
if (!slice_reader.Ok()) {
- return absl::nullopt;
+ return std::nullopt;
}
return slice_pic_parameter_set_id;
@@ -527,25 +527,25 @@
bitstream.subview(index.payload_start_offset, index.payload_size));
}
-absl::optional<int> H265BitstreamParser::GetLastSliceQp() const {
+std::optional<int> H265BitstreamParser::GetLastSliceQp() const {
if (!last_slice_qp_delta_ || !last_slice_pps_id_) {
- return absl::nullopt;
+ return std::nullopt;
}
const H265PpsParser::PpsState* pps = GetPPS(last_slice_pps_id_.value());
if (!pps)
- return absl::nullopt;
+ return std::nullopt;
const int parsed_qp = 26 + pps->init_qp_minus26 + *last_slice_qp_delta_;
if (parsed_qp < kMinQpValue || parsed_qp > kMaxQpValue) {
RTC_LOG(LS_ERROR) << "Parsed invalid QP from bitstream.";
- return absl::nullopt;
+ return std::nullopt;
}
return parsed_qp;
}
-absl::optional<uint32_t> H265BitstreamParser::GetLastSlicePpsId() const {
+std::optional<uint32_t> H265BitstreamParser::GetLastSlicePpsId() const {
if (!last_slice_pps_id_) {
RTC_LOG(LS_ERROR) << "Failed to parse PPS id from bitstream.";
- return absl::nullopt;
+ return std::nullopt;
}
return last_slice_pps_id_;
diff --git a/common_video/h265/h265_bitstream_parser.h b/common_video/h265/h265_bitstream_parser.h
index cf18ccb..3ac8a5a 100644
--- a/common_video/h265/h265_bitstream_parser.h
+++ b/common_video/h265/h265_bitstream_parser.h
@@ -14,9 +14,9 @@
#include <stddef.h>
#include <stdint.h>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/video_codecs/bitstream_parser.h"
#include "common_video/h265/h265_pps_parser.h"
#include "common_video/h265/h265_sps_parser.h"
@@ -35,11 +35,11 @@
// New interface.
void ParseBitstream(rtc::ArrayView<const uint8_t> bitstream) override;
- absl::optional<int> GetLastSliceQp() const override;
+ std::optional<int> GetLastSliceQp() const override;
- absl::optional<uint32_t> GetLastSlicePpsId() const;
+ std::optional<uint32_t> GetLastSlicePpsId() const;
- static absl::optional<uint32_t> ParsePpsIdFromSliceSegmentLayerRbsp(
+ static std::optional<uint32_t> ParsePpsIdFromSliceSegmentLayerRbsp(
rtc::ArrayView<const uint8_t> data,
uint8_t nalu_type);
@@ -63,8 +63,8 @@
flat_map<uint32_t, H265PpsParser::PpsState> pps_;
// Last parsed slice QP.
- absl::optional<int32_t> last_slice_qp_delta_;
- absl::optional<uint32_t> last_slice_pps_id_;
+ std::optional<int32_t> last_slice_qp_delta_;
+ std::optional<uint32_t> last_slice_pps_id_;
};
} // namespace webrtc
diff --git a/common_video/h265/h265_bitstream_parser_unittest.cc b/common_video/h265/h265_bitstream_parser_unittest.cc
index 5889c70..aed779c 100644
--- a/common_video/h265/h265_bitstream_parser_unittest.cc
+++ b/common_video/h265/h265_bitstream_parser_unittest.cc
@@ -105,7 +105,7 @@
TEST(H265BitstreamParserTest, ReportsLastSliceQpForImageSlices) {
H265BitstreamParser h265_parser;
h265_parser.ParseBitstream(kH265BitstreamChunk);
- absl::optional<int> qp = h265_parser.GetLastSliceQp();
+ std::optional<int> qp = h265_parser.GetLastSliceQp();
ASSERT_TRUE(qp.has_value());
EXPECT_EQ(34, *qp);
@@ -119,14 +119,14 @@
TEST(H265BitstreamParserTest, ReportsLastSliceQpFromShortTermReferenceSlices) {
H265BitstreamParser h265_parser;
h265_parser.ParseBitstream(kH265SliceStrChunk);
- absl::optional<int> qp = h265_parser.GetLastSliceQp();
+ std::optional<int> qp = h265_parser.GetLastSliceQp();
ASSERT_TRUE(qp.has_value());
EXPECT_EQ(33, *qp);
}
TEST(H265BitstreamParserTest, PpsIdFromSlice) {
H265BitstreamParser h265_parser;
- absl::optional<uint32_t> pps_id =
+ std::optional<uint32_t> pps_id =
h265_parser.ParsePpsIdFromSliceSegmentLayerRbsp(kH265SliceChunk,
H265::NaluType::kTrailR);
ASSERT_TRUE(pps_id);
@@ -136,7 +136,7 @@
TEST(H265BitstreamParserTest, ReportsLastSliceQpInvalidQPSlices) {
H265BitstreamParser h265_parser;
h265_parser.ParseBitstream(kH265BitstreamInvalidQPChunk);
- absl::optional<int> qp = h265_parser.GetLastSliceQp();
+ std::optional<int> qp = h265_parser.GetLastSliceQp();
ASSERT_FALSE(qp.has_value());
h265_parser.ParseBitstream(kH265BitstreamInvalidQPChunk52);
diff --git a/common_video/h265/h265_pps_parser.cc b/common_video/h265/h265_pps_parser.cc
index 3339f65..c1c21e4 100644
--- a/common_video/h265/h265_pps_parser.cc
+++ b/common_video/h265/h265_pps_parser.cc
@@ -11,9 +11,9 @@
#include "common_video/h265/h265_pps_parser.h"
#include <memory>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "common_video/h265/h265_common.h"
#include "rtc_base/bit_buffer.h"
#include "rtc_base/bitstream_reader.h"
@@ -26,7 +26,7 @@
" to be" \
<< " in range [" << (min) << ":" << (max) << "]" \
<< " found " << (val) << " instead"; \
- return absl::nullopt; \
+ return std::nullopt; \
} \
} while (0)
@@ -46,7 +46,7 @@
if (!reader.Ok() || !(a)) { \
RTC_LOG(LS_WARNING) << "Error in stream: invalid value, expected " \
<< #a; \
- return absl::nullopt; \
+ return std::nullopt; \
} \
} while (0)
@@ -62,7 +62,7 @@
// You can find it on this page:
// http://www.itu.int/rec/T-REC-H.265
-absl::optional<H265PpsParser::PpsState> H265PpsParser::ParsePps(
+std::optional<H265PpsParser::PpsState> H265PpsParser::ParsePps(
rtc::ArrayView<const uint8_t> data,
const H265SpsParser::SpsState* sps) {
// First, parse out rbsp, which is basically the source buffer minus emulation
@@ -88,18 +88,18 @@
return reader.Ok();
}
-absl::optional<H265PpsParser::PpsState> H265PpsParser::ParseInternal(
+std::optional<H265PpsParser::PpsState> H265PpsParser::ParseInternal(
rtc::ArrayView<const uint8_t> buffer,
const H265SpsParser::SpsState* sps) {
BitstreamReader reader(buffer);
PpsState pps;
if (!sps) {
- return absl::nullopt;
+ return std::nullopt;
}
if (!ParsePpsIdsInternal(reader, pps.pps_id, pps.sps_id)) {
- return absl::nullopt;
+ return std::nullopt;
}
// dependent_slice_segments_enabled_flag: u(1)
@@ -221,14 +221,14 @@
if (pps_scaling_list_data_present_flag) {
// scaling_list_data()
if (!H265SpsParser::ParseScalingListData(reader)) {
- return absl::nullopt;
+ return std::nullopt;
}
}
// lists_modification_present_flag: u(1)
pps.lists_modification_present_flag = reader.Read<bool>();
if (!reader.Ok()) {
- return absl::nullopt;
+ return std::nullopt;
}
return pps;
diff --git a/common_video/h265/h265_pps_parser.h b/common_video/h265/h265_pps_parser.h
index 1bd0ead..11cd286 100644
--- a/common_video/h265/h265_pps_parser.h
+++ b/common_video/h265/h265_pps_parser.h
@@ -11,7 +11,8 @@
#ifndef COMMON_VIDEO_H265_H265_PPS_PARSER_H_
#define COMMON_VIDEO_H265_H265_PPS_PARSER_H_
-#include "absl/types/optional.h"
+#include <optional>
+
#include "api/array_view.h"
#include "common_video/h265/h265_sps_parser.h"
#include "rtc_base/bitstream_reader.h"
@@ -43,10 +44,10 @@
};
// Unpack RBSP and parse PPS state from the supplied buffer.
- static absl::optional<PpsState> ParsePps(rtc::ArrayView<const uint8_t> data,
- const H265SpsParser::SpsState* sps);
+ static std::optional<PpsState> ParsePps(rtc::ArrayView<const uint8_t> data,
+ const H265SpsParser::SpsState* sps);
// TODO: bugs.webrtc.org/42225170 - Deprecate.
- static inline absl::optional<PpsState> ParsePps(
+ static inline std::optional<PpsState> ParsePps(
const uint8_t* data,
size_t length,
const H265SpsParser::SpsState* sps) {
@@ -67,7 +68,7 @@
protected:
// Parse the PPS state, for a bit buffer where RBSP decoding has already been
// performed.
- static absl::optional<PpsState> ParseInternal(
+ static std::optional<PpsState> ParseInternal(
rtc::ArrayView<const uint8_t> buffer,
const H265SpsParser::SpsState* sps);
static bool ParsePpsIdsInternal(BitstreamReader& reader,
diff --git a/common_video/h265/h265_pps_parser_unittest.cc b/common_video/h265/h265_pps_parser_unittest.cc
index c683a43..61cf0a5 100644
--- a/common_video/h265/h265_pps_parser_unittest.cc
+++ b/common_video/h265/h265_pps_parser_unittest.cc
@@ -222,8 +222,8 @@
H265PpsParser::PpsState generated_pps_;
rtc::Buffer buffer_;
- absl::optional<H265PpsParser::PpsState> parsed_pps_;
- absl::optional<H265SpsParser::SpsState> parsed_sps_;
+ std::optional<H265PpsParser::PpsState> parsed_pps_;
+ std::optional<H265SpsParser::SpsState> parsed_sps_;
};
TEST_F(H265PpsParserTest, ZeroPps) {
diff --git a/common_video/h265/h265_sps_parser.cc b/common_video/h265/h265_sps_parser.cc
index cb59d60..fa23f6a 100644
--- a/common_video/h265/h265_sps_parser.cc
+++ b/common_video/h265/h265_sps_parser.cc
@@ -25,7 +25,7 @@
" to be" \
<< " in range [" << (min) << ":" << (max) << "]" \
<< " found " << (val) << " instead"; \
- return absl::nullopt; \
+ return std::nullopt; \
} \
} while (0)
@@ -45,16 +45,16 @@
if (!reader.Ok() || !(a)) { \
RTC_LOG(LS_WARNING) << "Error in stream: invalid value, expected " \
<< #a; \
- return absl::nullopt; \
+ return std::nullopt; \
} \
} while (0)
namespace {
-using OptionalSps = absl::optional<webrtc::H265SpsParser::SpsState>;
+using OptionalSps = std::optional<webrtc::H265SpsParser::SpsState>;
using OptionalShortTermRefPicSet =
- absl::optional<webrtc::H265SpsParser::ShortTermRefPicSet>;
+ std::optional<webrtc::H265SpsParser::ShortTermRefPicSet>;
using OptionalProfileTierLevel =
- absl::optional<webrtc::H265SpsParser::ProfileTierLevel>;
+ std::optional<webrtc::H265SpsParser::ProfileTierLevel>;
constexpr int kMaxNumSizeIds = 4;
constexpr int kMaxNumMatrixIds = 6;
@@ -103,7 +103,7 @@
// http://www.itu.int/rec/T-REC-H.265
// Unpack RBSP and parse SPS state from the supplied buffer.
-absl::optional<H265SpsParser::SpsState> H265SpsParser::ParseSps(
+std::optional<H265SpsParser::SpsState> H265SpsParser::ParseSps(
rtc::ArrayView<const uint8_t> data) {
return ParseSpsInternal(H265::ParseRbsp(data));
}
@@ -146,7 +146,7 @@
return reader.Ok();
}
-absl::optional<H265SpsParser::ShortTermRefPicSet>
+std::optional<H265SpsParser::ShortTermRefPicSet>
H265SpsParser::ParseShortTermRefPicSet(
uint32_t st_rps_idx,
uint32_t num_short_term_ref_pic_sets,
@@ -297,13 +297,13 @@
st_ref_pic_set.num_negative_pics + st_ref_pic_set.num_positive_pics;
if (!reader.Ok()) {
- return absl::nullopt;
+ return std::nullopt;
}
return OptionalShortTermRefPicSet(st_ref_pic_set);
}
-absl::optional<H265SpsParser::ProfileTierLevel>
+std::optional<H265SpsParser::ProfileTierLevel>
H265SpsParser::ParseProfileTierLevel(bool profile_present,
int max_num_sub_layers_minus1,
BitstreamReader& reader) {
@@ -327,7 +327,7 @@
if (!reader.Ok() || (!pf_tier_level.general_progressive_source_flag &&
pf_tier_level.general_interlaced_source_flag)) {
RTC_LOG(LS_WARNING) << "Interlaced streams not supported";
- return absl::nullopt;
+ return std::nullopt;
}
pf_tier_level.general_non_packed_constraint_flag = reader.ReadBits(1);
pf_tier_level.general_frame_only_constraint_flag = reader.ReadBits(1);
@@ -378,13 +378,13 @@
}
if (!reader.Ok()) {
- return absl::nullopt;
+ return std::nullopt;
}
return OptionalProfileTierLevel(pf_tier_level);
}
-absl::optional<H265SpsParser::SpsState> H265SpsParser::ParseSpsInternal(
+std::optional<H265SpsParser::SpsState> H265SpsParser::ParseSpsInternal(
rtc::ArrayView<const uint8_t> buffer) {
BitstreamReader reader(buffer);
@@ -416,7 +416,7 @@
OptionalProfileTierLevel profile_tier_level =
ParseProfileTierLevel(true, sps.sps_max_sub_layers_minus1, reader);
if (!profile_tier_level) {
- return absl::nullopt;
+ return std::nullopt;
}
// sps_seq_parameter_set_id: ue(v)
sps.sps_id = reader.ReadExponentialGolomb();
@@ -572,7 +572,7 @@
if (sps_scaling_list_data_present_flag) {
// scaling_list_data()
if (!ParseScalingListData(reader)) {
- return absl::nullopt;
+ return std::nullopt;
}
}
}
@@ -622,7 +622,7 @@
if (ref_pic_set) {
sps.short_term_ref_pic_set[st_rps_idx] = *ref_pic_set;
} else {
- return absl::nullopt;
+ return std::nullopt;
}
}
@@ -675,7 +675,7 @@
}
if (!reader.Ok()) {
- return absl::nullopt;
+ return std::nullopt;
}
return OptionalSps(sps);
diff --git a/common_video/h265/h265_sps_parser.h b/common_video/h265/h265_sps_parser.h
index 072e2e9..4af1bf3 100644
--- a/common_video/h265/h265_sps_parser.h
+++ b/common_video/h265/h265_sps_parser.h
@@ -11,9 +11,9 @@
#ifndef COMMON_VIDEO_H265_H265_SPS_PARSER_H_
#define COMMON_VIDEO_H265_H265_SPS_PARSER_H_
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "rtc_base/bitstream_reader.h"
#include "rtc_base/system/rtc_export.h"
@@ -104,23 +104,23 @@
};
// Unpack RBSP and parse SPS state from the supplied buffer.
- static absl::optional<SpsState> ParseSps(rtc::ArrayView<const uint8_t> data);
+ static std::optional<SpsState> ParseSps(rtc::ArrayView<const uint8_t> data);
// TODO: bugs.webrtc.org/42225170 - Deprecate.
- static inline absl::optional<SpsState> ParseSps(const uint8_t* data,
- size_t length) {
+ static inline std::optional<SpsState> ParseSps(const uint8_t* data,
+ size_t length) {
return ParseSps(rtc::MakeArrayView(data, length));
}
static bool ParseScalingListData(BitstreamReader& reader);
- static absl::optional<ShortTermRefPicSet> ParseShortTermRefPicSet(
+ static std::optional<ShortTermRefPicSet> ParseShortTermRefPicSet(
uint32_t st_rps_idx,
uint32_t num_short_term_ref_pic_sets,
const std::vector<ShortTermRefPicSet>& ref_pic_sets,
uint32_t sps_max_dec_pic_buffering_minus1,
BitstreamReader& reader);
- static absl::optional<H265SpsParser::ProfileTierLevel> ParseProfileTierLevel(
+ static std::optional<H265SpsParser::ProfileTierLevel> ParseProfileTierLevel(
bool profile_present,
int max_num_sub_layers_minus1,
BitstreamReader& reader);
@@ -128,7 +128,7 @@
protected:
// Parse the SPS state, for a bit buffer where RBSP decoding has already been
// performed.
- static absl::optional<SpsState> ParseSpsInternal(
+ static std::optional<SpsState> ParseSpsInternal(
rtc::ArrayView<const uint8_t> buffer);
// From Table A.8 - General tier and level limits.
diff --git a/common_video/h265/h265_sps_parser_unittest.cc b/common_video/h265/h265_sps_parser_unittest.cc
index 46d8d2d..1c8537b 100644
--- a/common_video/h265/h265_sps_parser_unittest.cc
+++ b/common_video/h265/h265_sps_parser_unittest.cc
@@ -389,7 +389,7 @@
0x02, 0x80, 0x80, 0x2d, 0x16, 0x59, 0x59, 0xa4,
0x93, 0x2b, 0x80, 0x40, 0x00, 0x00, 0x03, 0x00,
0x40, 0x00, 0x00, 0x07, 0x82};
- absl::optional<H265SpsParser::SpsState> sps = H265SpsParser::ParseSps(buffer);
+ std::optional<H265SpsParser::SpsState> sps = H265SpsParser::ParseSps(buffer);
ASSERT_TRUE(sps.has_value());
EXPECT_EQ(1280u, sps->width);
EXPECT_EQ(720u, sps->height);
@@ -417,7 +417,7 @@
0x05, 0x02, 0x01, 0x09, 0xf2, 0xe5, 0x95, 0x9a,
0x49, 0x32, 0xb8, 0x04, 0x00, 0x00, 0x03, 0x00,
0x04, 0x00, 0x00, 0x03, 0x00, 0x78, 0x20};
- absl::optional<H265SpsParser::SpsState> sps = H265SpsParser::ParseSps(buffer);
+ std::optional<H265SpsParser::SpsState> sps = H265SpsParser::ParseSps(buffer);
ASSERT_TRUE(sps.has_value());
EXPECT_EQ(640u, sps->width);
EXPECT_EQ(260u, sps->height);
@@ -444,7 +444,7 @@
0x08, 0x48, 0x04, 0x27, 0x72, 0xe5, 0x95, 0x9a,
0x49, 0x32, 0xb8, 0x04, 0x00, 0x00, 0x03, 0x00,
0x04, 0x00, 0x00, 0x03, 0x00, 0x78, 0x20};
- absl::optional<H265SpsParser::SpsState> sps = H265SpsParser::ParseSps(buffer);
+ std::optional<H265SpsParser::SpsState> sps = H265SpsParser::ParseSps(buffer);
ASSERT_TRUE(sps.has_value());
EXPECT_EQ(260u, sps->width);
EXPECT_EQ(260u, sps->height);
@@ -453,7 +453,7 @@
TEST_F(H265SpsParserTest, TestSyntheticSPSQvgaLandscape) {
rtc::Buffer buffer;
WriteSps(320u, 180u, 1, 0, 1, 0, &buffer);
- absl::optional<H265SpsParser::SpsState> sps = H265SpsParser::ParseSps(buffer);
+ std::optional<H265SpsParser::SpsState> sps = H265SpsParser::ParseSps(buffer);
ASSERT_TRUE(sps.has_value());
EXPECT_EQ(320u, sps->width);
EXPECT_EQ(180u, sps->height);
@@ -463,7 +463,7 @@
TEST_F(H265SpsParserTest, TestSyntheticSPSWeirdResolution) {
rtc::Buffer buffer;
WriteSps(156u, 122u, 2, 0, 1, 0, &buffer);
- absl::optional<H265SpsParser::SpsState> sps = H265SpsParser::ParseSps(buffer);
+ std::optional<H265SpsParser::SpsState> sps = H265SpsParser::ParseSps(buffer);
ASSERT_TRUE(sps.has_value());
EXPECT_EQ(156u, sps->width);
EXPECT_EQ(122u, sps->height);
@@ -473,7 +473,7 @@
TEST_F(H265SpsParserTest, TestLog2MaxSubLayersMinus1) {
rtc::Buffer buffer;
WriteSps(320u, 180u, 1, 0, 1, 0, &buffer);
- absl::optional<H265SpsParser::SpsState> sps = H265SpsParser::ParseSps(buffer);
+ std::optional<H265SpsParser::SpsState> sps = H265SpsParser::ParseSps(buffer);
ASSERT_TRUE(sps.has_value());
EXPECT_EQ(320u, sps->width);
EXPECT_EQ(180u, sps->height);
@@ -481,8 +481,7 @@
EXPECT_EQ(0u, sps->sps_max_sub_layers_minus1);
WriteSps(320u, 180u, 1, 6, 1, 0, &buffer);
- absl::optional<H265SpsParser::SpsState> sps1 =
- H265SpsParser::ParseSps(buffer);
+ std::optional<H265SpsParser::SpsState> sps1 = H265SpsParser::ParseSps(buffer);
ASSERT_TRUE(sps1.has_value());
EXPECT_EQ(320u, sps1->width);
EXPECT_EQ(180u, sps1->height);
@@ -490,7 +489,7 @@
EXPECT_EQ(6u, sps1->sps_max_sub_layers_minus1);
WriteSps(320u, 180u, 1, 7, 1, 0, &buffer);
- absl::optional<H265SpsParser::SpsState> result =
+ std::optional<H265SpsParser::SpsState> result =
H265SpsParser::ParseSps(buffer);
EXPECT_FALSE(result.has_value());
}
@@ -498,7 +497,7 @@
TEST_F(H265SpsParserTest, TestSubLayerOrderingInfoPresentFlag) {
rtc::Buffer buffer;
WriteSps(320u, 180u, 1, 6, 1, 0, &buffer);
- absl::optional<H265SpsParser::SpsState> sps = H265SpsParser::ParseSps(buffer);
+ std::optional<H265SpsParser::SpsState> sps = H265SpsParser::ParseSps(buffer);
ASSERT_TRUE(sps.has_value());
EXPECT_EQ(320u, sps->width);
EXPECT_EQ(180u, sps->height);
@@ -506,8 +505,7 @@
EXPECT_EQ(6u, sps->sps_max_sub_layers_minus1);
WriteSps(320u, 180u, 1, 6, 1, 0, &buffer);
- absl::optional<H265SpsParser::SpsState> sps1 =
- H265SpsParser::ParseSps(buffer);
+ std::optional<H265SpsParser::SpsState> sps1 = H265SpsParser::ParseSps(buffer);
ASSERT_TRUE(sps1.has_value());
EXPECT_EQ(320u, sps1->width);
EXPECT_EQ(180u, sps1->height);
@@ -518,7 +516,7 @@
TEST_F(H265SpsParserTest, TestLongTermRefPicsPresentFlag) {
rtc::Buffer buffer;
WriteSps(320u, 180u, 1, 0, 1, 0, &buffer);
- absl::optional<H265SpsParser::SpsState> sps = H265SpsParser::ParseSps(buffer);
+ std::optional<H265SpsParser::SpsState> sps = H265SpsParser::ParseSps(buffer);
ASSERT_TRUE(sps.has_value());
EXPECT_EQ(320u, sps->width);
EXPECT_EQ(180u, sps->height);
@@ -526,8 +524,7 @@
EXPECT_EQ(0u, sps->long_term_ref_pics_present_flag);
WriteSps(320u, 180u, 1, 6, 1, 1, &buffer);
- absl::optional<H265SpsParser::SpsState> sps1 =
- H265SpsParser::ParseSps(buffer);
+ std::optional<H265SpsParser::SpsState> sps1 = H265SpsParser::ParseSps(buffer);
ASSERT_TRUE(sps1.has_value());
EXPECT_EQ(320u, sps1->width);
EXPECT_EQ(180u, sps1->height);
diff --git a/common_video/h265/h265_vps_parser.cc b/common_video/h265/h265_vps_parser.cc
index 0a00370..fd896ad 100644
--- a/common_video/h265/h265_vps_parser.cc
+++ b/common_video/h265/h265_vps_parser.cc
@@ -24,12 +24,12 @@
// http://www.itu.int/rec/T-REC-H.265
// Unpack RBSP and parse VPS state from the supplied buffer.
-absl::optional<H265VpsParser::VpsState> H265VpsParser::ParseVps(
+std::optional<H265VpsParser::VpsState> H265VpsParser::ParseVps(
rtc::ArrayView<const uint8_t> data) {
return ParseInternal(H265::ParseRbsp(data));
}
-absl::optional<H265VpsParser::VpsState> H265VpsParser::ParseInternal(
+std::optional<H265VpsParser::VpsState> H265VpsParser::ParseInternal(
rtc::ArrayView<const uint8_t> buffer) {
BitstreamReader reader(buffer);
@@ -42,7 +42,7 @@
vps.id = reader.ReadBits(4);
if (!reader.Ok()) {
- return absl::nullopt;
+ return std::nullopt;
}
return vps;
diff --git a/common_video/h265/h265_vps_parser.h b/common_video/h265/h265_vps_parser.h
index e8ca1a0..2cbf5e5 100644
--- a/common_video/h265/h265_vps_parser.h
+++ b/common_video/h265/h265_vps_parser.h
@@ -11,7 +11,8 @@
#ifndef COMMON_VIDEO_H265_H265_VPS_PARSER_H_
#define COMMON_VIDEO_H265_H265_VPS_PARSER_H_
-#include "absl/types/optional.h"
+#include <optional>
+
#include "api/array_view.h"
#include "rtc_base/system/rtc_export.h"
@@ -29,17 +30,17 @@
};
// Unpack RBSP and parse VPS state from the supplied buffer.
- static absl::optional<VpsState> ParseVps(rtc::ArrayView<const uint8_t> data);
+ static std::optional<VpsState> ParseVps(rtc::ArrayView<const uint8_t> data);
// TODO: bugs.webrtc.org/42225170 - Deprecate.
- static inline absl::optional<VpsState> ParseVps(const uint8_t* data,
- size_t length) {
+ static inline std::optional<VpsState> ParseVps(const uint8_t* data,
+ size_t length) {
return ParseVps(rtc::MakeArrayView(data, length));
}
protected:
// Parse the VPS state, for a bit buffer where RBSP decoding has already been
// performed.
- static absl::optional<VpsState> ParseInternal(
+ static std::optional<VpsState> ParseInternal(
rtc::ArrayView<const uint8_t> buffer);
};
diff --git a/common_video/h265/h265_vps_parser_unittest.cc b/common_video/h265/h265_vps_parser_unittest.cc
index 87e1f58..fd2e8a8 100644
--- a/common_video/h265/h265_vps_parser_unittest.cc
+++ b/common_video/h265/h265_vps_parser_unittest.cc
@@ -32,7 +32,7 @@
H265VpsParserTest() {}
~H265VpsParserTest() override {}
- absl::optional<H265VpsParser::VpsState> vps_;
+ std::optional<H265VpsParser::VpsState> vps_;
};
TEST_F(H265VpsParserTest, TestSampleVPSId) {
diff --git a/common_video/include/bitrate_adjuster.h b/common_video/include/bitrate_adjuster.h
index 4b20830..966c35d 100644
--- a/common_video/include/bitrate_adjuster.h
+++ b/common_video/include/bitrate_adjuster.h
@@ -14,7 +14,8 @@
#include <stddef.h>
#include <stdint.h>
-#include "absl/types/optional.h"
+#include <optional>
+
#include "rtc_base/rate_statistics.h"
#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/system/rtc_export.h"
@@ -48,7 +49,7 @@
uint32_t GetAdjustedBitrateBps() const;
// Returns what we think the current bitrate is.
- absl::optional<uint32_t> GetEstimatedBitrateBps();
+ std::optional<uint32_t> GetEstimatedBitrateBps();
// This should be called after each frame is encoded. The timestamp at which
// it is called is used to estimate the output bitrate of the encoder.
diff --git a/examples/BUILD.gn b/examples/BUILD.gn
index ed02eb7..57de7ce 100644
--- a/examples/BUILD.gn
+++ b/examples/BUILD.gn
@@ -720,7 +720,6 @@
"../test:platform_video_capturer",
"../test:rtp_test_utils",
"//third_party/abseil-cpp/absl/memory",
- "//third_party/abseil-cpp/absl/types:optional",
]
if (is_win) {
sources += [
diff --git a/examples/androidvoip/jni/android_voip_client.cc b/examples/androidvoip/jni/android_voip_client.cc
index 183f97a..5004fc0 100644
--- a/examples/androidvoip/jni/android_voip_client.cc
+++ b/examples/androidvoip/jni/android_voip_client.cc
@@ -304,7 +304,7 @@
RUN_ON_VOIP_THREAD(StartSession, env);
// CreateChannel guarantees to return valid channel id.
- channel_ = voip_engine_->Base().CreateChannel(this, absl::nullopt);
+ channel_ = voip_engine_->Base().CreateChannel(this, std::nullopt);
rtp_socket_.reset(rtc::AsyncUDPSocket::Create(voip_thread_->socketserver(),
rtp_local_address_));
@@ -357,7 +357,7 @@
webrtc::VoipResult result = voip_engine_->Base().ReleaseChannel(*channel_);
RTC_CHECK(result == webrtc::VoipResult::kOk);
- channel_ = absl::nullopt;
+ channel_ = std::nullopt;
Java_VoipClient_onStopSessionCompleted(env_, j_voip_client_,
/*isSuccessful=*/true);
}
diff --git a/examples/androidvoip/jni/android_voip_client.h b/examples/androidvoip/jni/android_voip_client.h
index 41b6a69..073613b 100644
--- a/examples/androidvoip/jni/android_voip_client.h
+++ b/examples/androidvoip/jni/android_voip_client.h
@@ -165,7 +165,7 @@
// The entry point to all VoIP APIs.
std::unique_ptr<webrtc::VoipEngine> voip_engine_ RTC_GUARDED_BY(voip_thread_);
// Used by the VoIP API to facilitate a VoIP session.
- absl::optional<webrtc::ChannelId> channel_ RTC_GUARDED_BY(voip_thread_);
+ std::optional<webrtc::ChannelId> channel_ RTC_GUARDED_BY(voip_thread_);
// Members below are used for network related operations.
std::unique_ptr<rtc::AsyncUDPSocket> rtp_socket_ RTC_GUARDED_BY(voip_thread_);
std::unique_ptr<rtc::AsyncUDPSocket> rtcp_socket_
diff --git a/examples/peerconnection/client/conductor.cc b/examples/peerconnection/client/conductor.cc
index 23e5781..cbe11eb 100644
--- a/examples/peerconnection/client/conductor.cc
+++ b/examples/peerconnection/client/conductor.cc
@@ -14,11 +14,11 @@
#include <stdint.h>
#include <memory>
+#include <optional>
#include <utility>
#include <vector>
#include "absl/memory/memory.h"
-#include "absl/types/optional.h"
#include "api/audio/audio_device.h"
#include "api/audio/audio_mixer.h"
#include "api/audio/audio_processing.h"
@@ -358,7 +358,7 @@
}
return;
}
- absl::optional<webrtc::SdpType> type_maybe =
+ std::optional<webrtc::SdpType> type_maybe =
webrtc::SdpTypeFromString(type_str);
if (!type_maybe) {
RTC_LOG(LS_ERROR) << "Unknown SDP type: " << type_str;
diff --git a/logging/BUILD.gn b/logging/BUILD.gn
index 1f70095..39a51719 100644
--- a/logging/BUILD.gn
+++ b/logging/BUILD.gn
@@ -66,7 +66,6 @@
"../rtc_base:logging",
"//third_party/abseil-cpp/absl/base:core_headers",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -176,7 +175,6 @@
"../rtc_base:checks",
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -218,7 +216,6 @@
"../rtc_base:timeutils",
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -318,7 +315,6 @@
"../rtc_base:checks",
"../rtc_base:logging",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -336,7 +332,6 @@
"../rtc_base:safe_conversions",
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -357,7 +352,6 @@
"../rtc_base:checks",
"../rtc_base:logging",
"../rtc_base:safe_conversions",
- "//third_party/abseil-cpp/absl/types:optional",
]
if (rtc_enable_protobuf) {
@@ -418,7 +412,6 @@
"../rtc_base:checks",
"../rtc_base:logging",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
}
@@ -559,7 +552,6 @@
"//third_party/abseil-cpp/absl/base:core_headers",
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -645,7 +637,6 @@
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -670,7 +661,6 @@
"//third_party/abseil-cpp/absl/flags:usage",
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/strings",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
}
diff --git a/logging/rtc_event_log/dependency_descriptor_encoder_decoder.cc b/logging/rtc_event_log/dependency_descriptor_encoder_decoder.cc
index bac57df..ca3c8d1 100644
--- a/logging/rtc_event_log/dependency_descriptor_encoder_decoder.cc
+++ b/logging/rtc_event_log/dependency_descriptor_encoder_decoder.cc
@@ -12,11 +12,11 @@
#include <cstddef>
#include <cstdint>
+#include <optional>
#include <string>
#include <vector>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "logging/rtc_event_log/encoder/delta_encoding.h"
#include "logging/rtc_event_log/encoder/optional_blob_encoding.h"
@@ -28,7 +28,7 @@
namespace webrtc {
// static
-absl::optional<rtclog2::DependencyDescriptorsWireInfo>
+std::optional<rtclog2::DependencyDescriptorsWireInfo>
RtcEventLogDependencyDescriptorEncoderDecoder::Encode(
const std::vector<rtc::ArrayView<const uint8_t>>& raw_dd_data) {
if (raw_dd_data.empty()) {
@@ -49,13 +49,13 @@
// Start and end bit.
{
- absl::optional<uint32_t> start_end_bit;
+ std::optional<uint32_t> start_end_bit;
if (!base_dd.empty()) {
start_end_bit = (base_dd[0] >> 6);
res.set_start_end_bit(*start_end_bit);
}
if (!delta_dds.empty()) {
- std::vector<absl::optional<uint64_t>> values(delta_dds.size());
+ std::vector<std::optional<uint64_t>> values(delta_dds.size());
for (size_t i = 0; i < delta_dds.size(); ++i) {
if (!delta_dds[i].empty()) {
values[i] = delta_dds[i][0] >> 6;
@@ -70,14 +70,14 @@
// Template IDs.
{
- absl::optional<uint32_t> template_id;
+ std::optional<uint32_t> template_id;
if (!base_dd.empty()) {
template_id = (base_dd[0] & 0b0011'1111);
res.set_template_id(*template_id);
}
if (!delta_dds.empty()) {
- std::vector<absl::optional<uint64_t>> values(delta_dds.size());
+ std::vector<std::optional<uint64_t>> values(delta_dds.size());
for (size_t i = 0; i < delta_dds.size(); ++i) {
if (!delta_dds[i].empty()) {
values[i] = delta_dds[i][0] & 0b0011'1111;
@@ -92,14 +92,14 @@
// Frame IDs.
{
- absl::optional<uint32_t> frame_id;
+ std::optional<uint32_t> frame_id;
if (!base_dd.empty()) {
frame_id = (uint16_t{base_dd[1]} << 8) + base_dd[2];
res.set_frame_id(*frame_id);
}
if (!delta_dds.empty()) {
- std::vector<absl::optional<uint64_t>> values(delta_dds.size());
+ std::vector<std::optional<uint64_t>> values(delta_dds.size());
for (size_t i = 0; i < delta_dds.size(); ++i) {
if (!delta_dds[i].empty()) {
values[i] = (uint16_t{delta_dds[i][1]} << 8) + delta_dds[i][2];
@@ -114,7 +114,7 @@
// Extended info
{
- std::vector<absl::optional<std::string>> values(raw_dd_data.size());
+ std::vector<std::optional<std::string>> values(raw_dd_data.size());
for (size_t i = 0; i < raw_dd_data.size(); ++i) {
if (raw_dd_data[i].size() > 3) {
auto extended_info = raw_dd_data[i].subview(3);
@@ -143,49 +143,49 @@
std::vector<std::vector<uint8_t>> res(num_packets);
- absl::optional<uint64_t> start_end_bit_base;
+ std::optional<uint64_t> start_end_bit_base;
if (dd_wire_info.has_start_end_bit()) {
start_end_bit_base = dd_wire_info.start_end_bit();
}
- absl::optional<uint64_t> template_id_base;
+ std::optional<uint64_t> template_id_base;
if (dd_wire_info.has_template_id()) {
template_id_base = dd_wire_info.template_id();
}
- absl::optional<uint64_t> frame_id_base;
+ std::optional<uint64_t> frame_id_base;
if (dd_wire_info.has_frame_id()) {
frame_id_base = dd_wire_info.frame_id();
}
- std::vector<absl::optional<uint64_t>> start_end_bit_deltas;
+ std::vector<std::optional<uint64_t>> start_end_bit_deltas;
if (dd_wire_info.has_start_end_bit_deltas()) {
start_end_bit_deltas = DecodeDeltas(dd_wire_info.start_end_bit_deltas(),
start_end_bit_base, num_packets - 1);
RTC_DCHECK(start_end_bit_deltas.empty() ||
start_end_bit_deltas.size() == (num_packets - 1));
}
- std::vector<absl::optional<uint64_t>> template_id_deltas;
+ std::vector<std::optional<uint64_t>> template_id_deltas;
if (dd_wire_info.has_template_id_deltas()) {
template_id_deltas = DecodeDeltas(dd_wire_info.template_id_deltas(),
template_id_base, num_packets - 1);
RTC_DCHECK(template_id_deltas.empty() ||
template_id_deltas.size() == (num_packets - 1));
}
- std::vector<absl::optional<uint64_t>> frame_id_deltas;
+ std::vector<std::optional<uint64_t>> frame_id_deltas;
if (dd_wire_info.has_frame_id_deltas()) {
frame_id_deltas = DecodeDeltas(dd_wire_info.frame_id_deltas(),
frame_id_base, num_packets - 1);
RTC_DCHECK(frame_id_deltas.empty() ||
frame_id_deltas.size() == (num_packets - 1));
}
- std::vector<absl::optional<std::string>> extended_infos;
+ std::vector<std::optional<std::string>> extended_infos;
if (dd_wire_info.has_extended_infos()) {
extended_infos =
DecodeOptionalBlobs(dd_wire_info.extended_infos(), num_packets);
}
- auto recreate_raw_dd = [&](int i, const absl::optional<uint64_t>& be,
- const absl::optional<uint64_t>& tid,
- const absl::optional<uint64_t>& fid) {
+ auto recreate_raw_dd = [&](int i, const std::optional<uint64_t>& be,
+ const std::optional<uint64_t>& tid,
+ const std::optional<uint64_t>& fid) {
absl::string_view ext;
if (!extended_infos.empty() && extended_infos[i].has_value()) {
ext = *extended_infos[i];
diff --git a/logging/rtc_event_log/dependency_descriptor_encoder_decoder.h b/logging/rtc_event_log/dependency_descriptor_encoder_decoder.h
index 863a3c8..eaa452b 100644
--- a/logging/rtc_event_log/dependency_descriptor_encoder_decoder.h
+++ b/logging/rtc_event_log/dependency_descriptor_encoder_decoder.h
@@ -13,9 +13,9 @@
#include <cstddef>
#include <cstdint>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "logging/rtc_event_log/events/rtc_event_log_parse_status.h"
#include "logging/rtc_event_log/rtc_event_log2_proto_include.h"
@@ -24,7 +24,7 @@
class RtcEventLogDependencyDescriptorEncoderDecoder {
public:
- static absl::optional<rtclog2::DependencyDescriptorsWireInfo> Encode(
+ static std::optional<rtclog2::DependencyDescriptorsWireInfo> Encode(
const std::vector<rtc::ArrayView<const uint8_t>>& raw_dd_data);
static RtcEventLogParseStatusOr<std::vector<std::vector<uint8_t>>> Decode(
const rtclog2::DependencyDescriptorsWireInfo& dd_wire_info,
diff --git a/logging/rtc_event_log/encoder/delta_encoding.cc b/logging/rtc_event_log/encoder/delta_encoding.cc
index 7d62f4a..80fb587 100644
--- a/logging/rtc_event_log/encoder/delta_encoding.cc
+++ b/logging/rtc_event_log/encoder/delta_encoding.cc
@@ -15,12 +15,12 @@
#include <cstdint>
#include <limits>
#include <memory>
+#include <optional>
#include <string>
#include <vector>
#include "absl/memory/memory.h"
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "logging/rtc_event_log/encoder/bit_writer.h"
#include "logging/rtc_event_log/encoder/var_int.h"
#include "rtc_base/bitstream_reader.h"
@@ -186,8 +186,8 @@
// therefore be decoded by FixedLengthDeltaDecoder, or whether it was produced
// by a different encoder.
static std::string EncodeDeltas(
- absl::optional<uint64_t> base,
- const std::vector<absl::optional<uint64_t>>& values);
+ std::optional<uint64_t> base,
+ const std::vector<std::optional<uint64_t>>& values);
FixedLengthDeltaEncoder(const FixedLengthDeltaEncoder&) = delete;
FixedLengthDeltaEncoder& operator=(const FixedLengthDeltaEncoder&) = delete;
@@ -196,8 +196,8 @@
// Calculate min/max values of unsigned/signed deltas, given the bit width
// of all the values in the series.
static void CalculateMinAndMaxDeltas(
- absl::optional<uint64_t> base,
- const std::vector<absl::optional<uint64_t>>& values,
+ std::optional<uint64_t> base,
+ const std::vector<std::optional<uint64_t>>& values,
uint64_t bit_width,
uint64_t* max_unsigned_delta,
uint64_t* max_pos_signed_delta,
@@ -215,8 +215,8 @@
// Therefore, it was deemed acceptable to let them have a reference to
// `values`, whose lifetime must exceed the lifetime of `this`.
FixedLengthDeltaEncoder(const FixedLengthEncodingParameters& params,
- absl::optional<uint64_t> base,
- const std::vector<absl::optional<uint64_t>>& values,
+ std::optional<uint64_t> base,
+ const std::vector<std::optional<uint64_t>>& values,
size_t existent_values_count);
// Perform delta-encoding using the parameters given to the ctor on the
@@ -243,11 +243,11 @@
// The encoding scheme assumes that at least one value is transmitted OOB,
// so that the first value can be encoded as a delta from that OOB value,
// which is `base_`.
- const absl::optional<uint64_t> base_;
+ const std::optional<uint64_t> base_;
// The values to be encoded.
// Note: This is a non-owning reference. See comment above ctor for details.
- const std::vector<absl::optional<uint64_t>>& values_;
+ const std::vector<std::optional<uint64_t>>& values_;
// Buffer into which encoded values will be written.
// This is created dynmically as a way to enforce that the rest of the
@@ -258,8 +258,8 @@
// TODO(eladalon): Reduce the number of passes.
std::string FixedLengthDeltaEncoder::EncodeDeltas(
- absl::optional<uint64_t> base,
- const std::vector<absl::optional<uint64_t>>& values) {
+ std::optional<uint64_t> base,
+ const std::vector<std::optional<uint64_t>>& values) {
RTC_DCHECK(!values.empty());
// As a special case, if all of the elements are identical to the base,
@@ -267,7 +267,7 @@
// the empty string is used to signal that.
if (std::all_of(
values.cbegin(), values.cend(),
- [base](absl::optional<uint64_t> val) { return val == base; })) {
+ [base](std::optional<uint64_t> val) { return val == base; })) {
return std::string();
}
@@ -325,8 +325,8 @@
}
void FixedLengthDeltaEncoder::CalculateMinAndMaxDeltas(
- absl::optional<uint64_t> base,
- const std::vector<absl::optional<uint64_t>>& values,
+ std::optional<uint64_t> base,
+ const std::vector<std::optional<uint64_t>>& values,
uint64_t bit_width,
uint64_t* max_unsigned_delta_out,
uint64_t* max_pos_signed_delta_out,
@@ -342,7 +342,7 @@
uint64_t max_pos_signed_delta = 0;
uint64_t min_neg_signed_delta = 0;
- absl::optional<uint64_t> prev = base;
+ std::optional<uint64_t> prev = base;
for (size_t i = 0; i < values.size(); ++i) {
if (!values[i].has_value()) {
continue;
@@ -394,8 +394,8 @@
FixedLengthDeltaEncoder::FixedLengthDeltaEncoder(
const FixedLengthEncodingParameters& params,
- absl::optional<uint64_t> base,
- const std::vector<absl::optional<uint64_t>>& values,
+ std::optional<uint64_t> base,
+ const std::vector<std::optional<uint64_t>>& values,
size_t existent_values_count)
: params_(params), base_(base), values_(values) {
RTC_DCHECK(!values_.empty());
@@ -408,13 +408,13 @@
if (params_.values_optional()) {
// Encode which values exist and which don't.
- for (absl::optional<uint64_t> value : values_) {
+ for (std::optional<uint64_t> value : values_) {
writer_->WriteBits(value.has_value() ? 1u : 0u, 1);
}
}
- absl::optional<uint64_t> previous = base_;
- for (absl::optional<uint64_t> value : values_) {
+ std::optional<uint64_t> previous = base_;
+ for (std::optional<uint64_t> value : values_) {
if (!value.has_value()) {
RTC_DCHECK(params_.values_optional());
continue;
@@ -459,7 +459,7 @@
return values_.size() * params_.delta_width_bits();
} else {
RTC_DCHECK_EQ(std::count_if(values_.begin(), values_.end(),
- [](absl::optional<uint64_t> val) {
+ [](std::optional<uint64_t> val) {
return val.has_value();
}),
existent_values_count);
@@ -564,9 +564,9 @@
// original values, this will return the sequence of original values.
// If an error occurs (can happen if `input` is corrupt), an empty
// vector will be returned.
- static std::vector<absl::optional<uint64_t>> DecodeDeltas(
+ static std::vector<std::optional<uint64_t>> DecodeDeltas(
absl::string_view input,
- absl::optional<uint64_t> base,
+ std::optional<uint64_t> base,
size_t num_of_deltas);
FixedLengthDeltaDecoder(const FixedLengthDeltaDecoder&) = delete;
@@ -583,7 +583,7 @@
// examined and guaranteed.
static std::unique_ptr<FixedLengthDeltaDecoder> Create(
absl::string_view input,
- absl::optional<uint64_t> base,
+ std::optional<uint64_t> base,
size_t num_of_deltas);
// FixedLengthDeltaDecoder objects are to be created by DecodeDeltas() and
@@ -594,11 +594,11 @@
// of `reader`'s underlying buffer.
FixedLengthDeltaDecoder(BitstreamReader reader,
const FixedLengthEncodingParameters& params,
- absl::optional<uint64_t> base,
+ std::optional<uint64_t> base,
size_t num_of_deltas);
// Perform the decoding using the parameters given to the ctor.
- std::vector<absl::optional<uint64_t>> Decode();
+ std::vector<std::optional<uint64_t>> Decode();
// Add `delta` to `base` to produce the next value in a sequence.
// The delta is applied as signed/unsigned depending on the parameters
@@ -621,7 +621,7 @@
// The encoding scheme assumes that at least one value is transmitted OOB,
// so that the first value can be encoded as a delta from that OOB value,
// which is `base_`.
- const absl::optional<uint64_t> base_;
+ const std::optional<uint64_t> base_;
// The number of values to be known to be decoded.
const size_t num_of_deltas_;
@@ -641,13 +641,13 @@
EncodingType::kFixedSizeSignedDeltasEarlyWrapAndOptSupported;
}
-std::vector<absl::optional<uint64_t>> FixedLengthDeltaDecoder::DecodeDeltas(
+std::vector<std::optional<uint64_t>> FixedLengthDeltaDecoder::DecodeDeltas(
absl::string_view input,
- absl::optional<uint64_t> base,
+ std::optional<uint64_t> base,
size_t num_of_deltas) {
auto decoder = FixedLengthDeltaDecoder::Create(input, base, num_of_deltas);
if (!decoder) {
- return std::vector<absl::optional<uint64_t>>();
+ return std::vector<std::optional<uint64_t>>();
}
return decoder->Decode();
@@ -655,7 +655,7 @@
std::unique_ptr<FixedLengthDeltaDecoder> FixedLengthDeltaDecoder::Create(
absl::string_view input,
- absl::optional<uint64_t> base,
+ std::optional<uint64_t> base,
size_t num_of_deltas) {
BitstreamReader reader(input);
// Encoding type
@@ -715,7 +715,7 @@
FixedLengthDeltaDecoder::FixedLengthDeltaDecoder(
BitstreamReader reader,
const FixedLengthEncodingParameters& params,
- absl::optional<uint64_t> base,
+ std::optional<uint64_t> base,
size_t num_of_deltas)
: reader_(reader),
params_(params),
@@ -724,7 +724,7 @@
RTC_DCHECK(reader_.Ok());
}
-std::vector<absl::optional<uint64_t>> FixedLengthDeltaDecoder::Decode() {
+std::vector<std::optional<uint64_t>> FixedLengthDeltaDecoder::Decode() {
RTC_DCHECK(reader_.Ok());
std::vector<bool> existing_values(num_of_deltas_);
if (params_.values_optional()) {
@@ -735,8 +735,8 @@
std::fill(existing_values.begin(), existing_values.end(), true);
}
- absl::optional<uint64_t> previous = base_;
- std::vector<absl::optional<uint64_t>> values(num_of_deltas_);
+ std::optional<uint64_t> previous = base_;
+ std::vector<std::optional<uint64_t>> values(num_of_deltas_);
for (size_t i = 0; i < num_of_deltas_; ++i) {
if (!existing_values[i]) {
@@ -800,22 +800,21 @@
} // namespace
-std::string EncodeDeltas(absl::optional<uint64_t> base,
- const std::vector<absl::optional<uint64_t>>& values) {
+std::string EncodeDeltas(std::optional<uint64_t> base,
+ const std::vector<std::optional<uint64_t>>& values) {
// TODO(eladalon): Support additional encodings.
return FixedLengthDeltaEncoder::EncodeDeltas(base, values);
}
-std::vector<absl::optional<uint64_t>> DecodeDeltas(
- absl::string_view input,
- absl::optional<uint64_t> base,
- size_t num_of_deltas) {
+std::vector<std::optional<uint64_t>> DecodeDeltas(absl::string_view input,
+ std::optional<uint64_t> base,
+ size_t num_of_deltas) {
RTC_DCHECK_GT(num_of_deltas, 0); // Allows empty vector to indicate error.
// The empty string is a special case indicating that all values were equal
// to the base.
if (input.empty()) {
- std::vector<absl::optional<uint64_t>> result(num_of_deltas);
+ std::vector<std::optional<uint64_t>> result(num_of_deltas);
std::fill(result.begin(), result.end(), base);
return result;
}
@@ -825,7 +824,7 @@
}
RTC_LOG(LS_WARNING) << "Could not decode delta-encoded stream.";
- return std::vector<absl::optional<uint64_t>>();
+ return std::vector<std::optional<uint64_t>>();
}
void SetFixedLengthEncoderDeltaSignednessForTesting(bool signedness) {
diff --git a/logging/rtc_event_log/encoder/delta_encoding.h b/logging/rtc_event_log/encoder/delta_encoding.h
index 779cdc6..d097de7 100644
--- a/logging/rtc_event_log/encoder/delta_encoding.h
+++ b/logging/rtc_event_log/encoder/delta_encoding.h
@@ -14,11 +14,11 @@
#include <stddef.h>
#include <stdint.h>
+#include <optional>
#include <string>
#include <vector>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
namespace webrtc {
@@ -29,8 +29,8 @@
// be provided separately to the decoder.
// This function never fails.
// TODO(eladalon): Split into optional and non-optional variants (efficiency).
-std::string EncodeDeltas(absl::optional<uint64_t> base,
- const std::vector<absl::optional<uint64_t>>& values);
+std::string EncodeDeltas(std::optional<uint64_t> base,
+ const std::vector<std::optional<uint64_t>>& values);
// EncodeDeltas() and DecodeDeltas() are inverse operations;
// invoking DecodeDeltas() over the output of EncodeDeltas(), will return
@@ -39,10 +39,9 @@
// of `num_of_deltas` elements based on `base`, the function returns an empty
// vector, which signals an error.
// TODO(eladalon): Split into optional and non-optional variants (efficiency).
-std::vector<absl::optional<uint64_t>> DecodeDeltas(
- absl::string_view input,
- absl::optional<uint64_t> base,
- size_t num_of_deltas);
+std::vector<std::optional<uint64_t>> DecodeDeltas(absl::string_view input,
+ std::optional<uint64_t> base,
+ size_t num_of_deltas);
} // namespace webrtc
diff --git a/logging/rtc_event_log/encoder/delta_encoding_unittest.cc b/logging/rtc_event_log/encoder/delta_encoding_unittest.cc
index 3a27b24..6f46ff9 100644
--- a/logging/rtc_event_log/encoder/delta_encoding_unittest.cc
+++ b/logging/rtc_event_log/encoder/delta_encoding_unittest.cc
@@ -16,11 +16,11 @@
#include <cstring>
#include <limits>
#include <numeric>
+#include <optional>
#include <string>
#include <tuple>
#include <vector>
-#include "absl/types/optional.h"
#include "rtc_base/arraysize.h"
#include "rtc_base/checks.h"
#include "rtc_base/random.h"
@@ -71,53 +71,52 @@
// that it is equal to the original input.
// If `encoded_string` is non-null, the encoded result will also be written
// into it.
-void TestEncodingAndDecoding(
- absl::optional<uint64_t> base,
- const std::vector<absl::optional<uint64_t>>& values,
- std::string* encoded_string = nullptr) {
+void TestEncodingAndDecoding(std::optional<uint64_t> base,
+ const std::vector<std::optional<uint64_t>>& values,
+ std::string* encoded_string = nullptr) {
const std::string encoded = EncodeDeltas(base, values);
if (encoded_string) {
*encoded_string = encoded;
}
- const std::vector<absl::optional<uint64_t>> decoded =
+ const std::vector<std::optional<uint64_t>> decoded =
DecodeDeltas(encoded, base, values.size());
EXPECT_EQ(decoded, values);
}
-std::vector<absl::optional<uint64_t>> CreateSequenceByFirstValue(
+std::vector<std::optional<uint64_t>> CreateSequenceByFirstValue(
uint64_t first,
size_t sequence_length) {
- std::vector<absl::optional<uint64_t>> sequence(sequence_length);
+ std::vector<std::optional<uint64_t>> sequence(sequence_length);
std::iota(sequence.begin(), sequence.end(), first);
return sequence;
}
-std::vector<absl::optional<uint64_t>> CreateSequenceByLastValue(
+std::vector<std::optional<uint64_t>> CreateSequenceByLastValue(
uint64_t last,
size_t num_values) {
const uint64_t first = last - num_values + 1;
- std::vector<absl::optional<uint64_t>> result(num_values);
+ std::vector<std::optional<uint64_t>> result(num_values);
std::iota(result.begin(), result.end(), first);
return result;
}
// If `sequence_length` is greater than the number of deltas, the sequence of
// deltas will wrap around.
-std::vector<absl::optional<uint64_t>> CreateSequenceByOptionalDeltas(
+std::vector<std::optional<uint64_t>> CreateSequenceByOptionalDeltas(
uint64_t first,
- const std::vector<absl::optional<uint64_t>>& deltas,
+ const std::vector<std::optional<uint64_t>>& deltas,
size_t sequence_length) {
RTC_DCHECK_GE(sequence_length, 1);
- std::vector<absl::optional<uint64_t>> sequence(sequence_length);
+ std::vector<std::optional<uint64_t>> sequence(sequence_length);
uint64_t previous = first;
for (size_t i = 0, next_delta_index = 0; i < sequence.size(); ++i) {
if (deltas[next_delta_index].has_value()) {
sequence[i] =
- absl::optional<uint64_t>(previous + deltas[next_delta_index].value());
+ std::optional<uint64_t>(previous + deltas[next_delta_index].value());
previous = sequence[i].value();
}
next_delta_index = (next_delta_index + 1) % deltas.size();
@@ -129,7 +128,7 @@
size_t EncodingLengthUpperBound(size_t delta_max_bit_width,
size_t num_of_deltas,
DeltaSignedness signedness_override) {
- absl::optional<size_t> smallest_header_size_bytes;
+ std::optional<size_t> smallest_header_size_bytes;
switch (signedness_override) {
case DeltaSignedness::kNoOverride:
case DeltaSignedness::kForceUnsigned:
@@ -146,14 +145,14 @@
// If `sequence_length` is greater than the number of deltas, the sequence of
// deltas will wrap around.
-std::vector<absl::optional<uint64_t>> CreateSequenceByDeltas(
+std::vector<std::optional<uint64_t>> CreateSequenceByDeltas(
uint64_t first,
const std::vector<uint64_t>& deltas,
size_t sequence_length) {
RTC_DCHECK(!deltas.empty());
- std::vector<absl::optional<uint64_t>> optional_deltas(deltas.size());
+ std::vector<std::optional<uint64_t>> optional_deltas(deltas.size());
for (size_t i = 0; i < deltas.size(); ++i) {
- optional_deltas[i] = absl::optional<uint64_t>(deltas[i]);
+ optional_deltas[i] = std::optional<uint64_t>(deltas[i]);
}
return CreateSequenceByOptionalDeltas(first, optional_deltas,
sequence_length);
@@ -190,8 +189,8 @@
};
TEST_P(DeltaEncodingTest, AllValuesEqualToExistentBaseValue) {
- const absl::optional<uint64_t> base(3432);
- std::vector<absl::optional<uint64_t>> values(num_of_values_);
+ const std::optional<uint64_t> base(3432);
+ std::vector<std::optional<uint64_t>> values(num_of_values_);
std::fill(values.begin(), values.end(), base);
std::string encoded;
TestEncodingAndDecoding(base, values, &encoded);
@@ -206,8 +205,8 @@
return; // Test irrelevant for this case.
}
- const absl::optional<uint64_t> base;
- std::vector<absl::optional<uint64_t>> values(num_of_values_);
+ const std::optional<uint64_t> base;
+ std::vector<std::optional<uint64_t>> values(num_of_values_);
std::fill(values.begin(), values.end(), base);
std::string encoded;
TestEncodingAndDecoding(base, values, &encoded);
@@ -222,8 +221,8 @@
return; // Test irrelevant for this case.
}
- const absl::optional<uint64_t> base;
- std::vector<absl::optional<uint64_t>> values(num_of_values_);
+ const std::optional<uint64_t> base;
+ std::vector<std::optional<uint64_t>> values(num_of_values_);
Random prng(Seed());
@@ -248,7 +247,7 @@
}
TEST_P(DeltaEncodingTest, MinDeltaNoWrapAround) {
- const absl::optional<uint64_t> base(3432);
+ const std::optional<uint64_t> base(3432);
auto values = CreateSequenceByFirstValue(base.value() + 1, num_of_values_);
ASSERT_GT(values[values.size() - 1], base) << "Sanity; must not wrap around";
@@ -256,7 +255,7 @@
if (optional_values_) {
// Arbitrarily make one of the values non-existent, to force
// optional-supporting encoding.
- values[0] = absl::optional<uint64_t>();
+ values[0] = std::optional<uint64_t>();
}
TestEncodingAndDecoding(base, values);
@@ -264,7 +263,7 @@
TEST_P(DeltaEncodingTest, BigDeltaNoWrapAround) {
const uint64_t kBigDelta = 132828;
- const absl::optional<uint64_t> base(3432);
+ const std::optional<uint64_t> base(3432);
auto values =
CreateSequenceByFirstValue(base.value() + kBigDelta, num_of_values_);
@@ -273,14 +272,14 @@
if (optional_values_) {
// Arbitrarily make one of the values non-existent, to force
// optional-supporting encoding.
- values[0] = absl::optional<uint64_t>();
+ values[0] = std::optional<uint64_t>();
}
TestEncodingAndDecoding(base, values);
}
TEST_P(DeltaEncodingTest, MaxDeltaNoWrapAround) {
- const absl::optional<uint64_t> base(3432);
+ const std::optional<uint64_t> base(3432);
auto values = CreateSequenceByLastValue(std::numeric_limits<uint64_t>::max(),
num_of_values_);
@@ -289,7 +288,7 @@
if (optional_values_) {
// Arbitrarily make one of the values non-existent, to force
// optional-supporting encoding.
- values[0] = absl::optional<uint64_t>();
+ values[0] = std::optional<uint64_t>();
}
TestEncodingAndDecoding(base, values);
@@ -300,7 +299,7 @@
return; // Inapplicable
}
- const absl::optional<uint64_t> base(std::numeric_limits<uint64_t>::max());
+ const std::optional<uint64_t> base(std::numeric_limits<uint64_t>::max());
auto values = CreateSequenceByDeltas(*base, {1, 10, 3}, num_of_values_);
ASSERT_LT(values[0], base) << "Sanity; must wrap around";
@@ -308,7 +307,7 @@
if (optional_values_) {
// Arbitrarily make one of the values non-existent, to force
// optional-supporting encoding.
- values[1] = absl::optional<uint64_t>();
+ values[1] = std::optional<uint64_t>();
}
TestEncodingAndDecoding(base, values);
@@ -319,7 +318,7 @@
return; // Inapplicable.
}
- const absl::optional<uint64_t> base(std::numeric_limits<uint64_t>::max() - 2);
+ const std::optional<uint64_t> base(std::numeric_limits<uint64_t>::max() - 2);
auto values = CreateSequenceByDeltas(*base, {1, 10, 3}, num_of_values_);
ASSERT_LT(values[values.size() - 1], values[0]) << "Sanity; must wrap around";
@@ -328,7 +327,7 @@
// Arbitrarily make one of the values non-existent, to force
// optional-supporting encoding.
RTC_DCHECK_GT(values.size() - 1, 1u); // Wrap around not cancelled.
- values[1] = absl::optional<uint64_t>();
+ values[1] = std::optional<uint64_t>();
}
TestEncodingAndDecoding(base, values);
@@ -345,8 +344,8 @@
}
const uint64_t kBigDelta = 132828;
- const absl::optional<uint64_t> base(std::numeric_limits<uint64_t>::max() -
- kBigDelta + 3);
+ const std::optional<uint64_t> base(std::numeric_limits<uint64_t>::max() -
+ kBigDelta + 3);
auto values =
CreateSequenceByFirstValue(base.value() + kBigDelta, num_of_values_);
@@ -355,7 +354,7 @@
if (optional_values_) {
// Arbitrarily make one of the values non-existent, to force
// optional-supporting encoding.
- values[1] = absl::optional<uint64_t>();
+ values[1] = std::optional<uint64_t>();
}
TestEncodingAndDecoding(base, values);
@@ -367,8 +366,8 @@
}
const uint64_t kBigDelta = 132828;
- const absl::optional<uint64_t> base(std::numeric_limits<uint64_t>::max() -
- kBigDelta + 3);
+ const std::optional<uint64_t> base(std::numeric_limits<uint64_t>::max() -
+ kBigDelta + 3);
auto values = CreateSequenceByFirstValue(std::numeric_limits<uint64_t>::max(),
num_of_values_);
@@ -378,7 +377,7 @@
// Arbitrarily make one of the values non-existent, to force
// optional-supporting encoding.
RTC_DCHECK_GT(values.size() - 1, 1u); // Wrap around not cancelled.
- values[1] = absl::optional<uint64_t>();
+ values[1] = std::optional<uint64_t>();
}
TestEncodingAndDecoding(base, values);
@@ -392,13 +391,13 @@
return; // Inapplicable
}
- const absl::optional<uint64_t> base(3432);
+ const std::optional<uint64_t> base(3432);
auto values = CreateSequenceByFirstValue(*base - 1, num_of_values_);
if (optional_values_) {
// Arbitrarily make one of the values non-existent, to force
// optional-supporting encoding.
- values[1] = absl::optional<uint64_t>();
+ values[1] = std::optional<uint64_t>();
}
TestEncodingAndDecoding(base, values);
@@ -409,7 +408,7 @@
return; // Inapplicable.
}
- const absl::optional<uint64_t> base(3432);
+ const std::optional<uint64_t> base(3432);
auto values = CreateSequenceByDeltas(
*base, {0, std::numeric_limits<uint64_t>::max(), 3}, num_of_values_);
@@ -419,7 +418,7 @@
// Arbitrarily make one of the values non-existent, to force
// optional-supporting encoding.
RTC_DCHECK_GT(values.size() - 1, 1u); // Wrap around not cancelled.
- values[1] = absl::optional<uint64_t>();
+ values[1] = std::optional<uint64_t>();
}
TestEncodingAndDecoding(base, values);
@@ -429,7 +428,7 @@
// already covered by AllValuesEqualToExistentBaseValue, but it doesn't hurt to
// test again. For all other cases, we have a new test.
TEST_P(DeltaEncodingTest, ZeroDelta) {
- const absl::optional<uint64_t> base(3432);
+ const std::optional<uint64_t> base(3432);
// Arbitrary sequence of deltas with intentional zero deltas, as well as
// consecutive zeros.
@@ -440,7 +439,7 @@
if (optional_values_) {
// Arbitrarily make one of the values non-existent, to force
// optional-supporting encoding.
- values[0] = absl::optional<uint64_t>();
+ values[0] = std::optional<uint64_t>();
}
TestEncodingAndDecoding(base, values);
@@ -604,10 +603,10 @@
};
TEST_P(DeltaEncodingFuzzerLikeTest, Test) {
- const absl::optional<uint64_t> base(3432);
+ const std::optional<uint64_t> base(3432);
Random prng(Seed());
- std::vector<absl::optional<uint64_t>> deltas(num_of_values_);
+ std::vector<std::optional<uint64_t>> deltas(num_of_values_);
for (size_t i = 0; i < deltas.size(); ++i) {
if (!optional_values_ || prng.Rand<bool>()) {
deltas[i] = RandomWithMaxBitWidth(&prng, delta_max_bit_width_);
@@ -646,10 +645,10 @@
TEST_F(DeltaEncodingSpecificEdgeCasesTest, SignedDeltaWithOnlyTopBitOn) {
MaybeSetSignedness(DeltaSignedness::kForceSigned);
- const absl::optional<uint64_t> base(3432);
+ const std::optional<uint64_t> base(3432);
const uint64_t delta = static_cast<uint64_t>(1) << 63;
- const std::vector<absl::optional<uint64_t>> values = {base.value() + delta};
+ const std::vector<std::optional<uint64_t>> values = {base.value() + delta};
TestEncodingAndDecoding(base, values);
}
@@ -657,9 +656,9 @@
TEST_F(DeltaEncodingSpecificEdgeCasesTest, MaximumUnsignedDelta) {
MaybeSetSignedness(DeltaSignedness::kForceUnsigned);
- const absl::optional<uint64_t> base((static_cast<uint64_t>(1) << 63) + 0x123);
+ const std::optional<uint64_t> base((static_cast<uint64_t>(1) << 63) + 0x123);
- const std::vector<absl::optional<uint64_t>> values = {base.value() - 1};
+ const std::vector<std::optional<uint64_t>> values = {base.value() - 1};
TestEncodingAndDecoding(base, values);
}
@@ -675,7 +674,7 @@
: ((static_cast<uint64_t>(1) << width) - 1);
const uint64_t base = wrap_around ? 1u : (0xf82d3 & value_mask);
- const std::vector<absl::optional<uint64_t>> values = {
+ const std::vector<std::optional<uint64_t>> values = {
(base - 1u) & value_mask, (base - 2u) & value_mask,
(base - 3u) & value_mask};
diff --git a/logging/rtc_event_log/encoder/optional_blob_encoding.cc b/logging/rtc_event_log/encoder/optional_blob_encoding.cc
index a0078ee..f93d3ad 100644
--- a/logging/rtc_event_log/encoder/optional_blob_encoding.cc
+++ b/logging/rtc_event_log/encoder/optional_blob_encoding.cc
@@ -12,11 +12,11 @@
#include <cstddef>
#include <cstdint>
+#include <optional>
#include <string>
#include <vector>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "rtc_base/bit_buffer.h"
#include "rtc_base/bitstream_reader.h"
#include "rtc_base/checks.h"
@@ -24,7 +24,7 @@
namespace webrtc {
std::string EncodeOptionalBlobs(
- const std::vector<absl::optional<std::string>>& blobs) {
+ const std::vector<std::optional<std::string>>& blobs) {
if (blobs.empty()) {
return {};
}
@@ -79,10 +79,10 @@
return std::string(buffer.data(), buffer.data() + bytes_written);
}
-std::vector<absl::optional<std::string>> DecodeOptionalBlobs(
+std::vector<std::optional<std::string>> DecodeOptionalBlobs(
absl::string_view encoded_blobs,
size_t num_of_blobs) {
- std::vector<absl::optional<std::string>> res(num_of_blobs);
+ std::vector<std::optional<std::string>> res(num_of_blobs);
if (encoded_blobs.empty() || num_of_blobs == 0) {
return res;
}
diff --git a/logging/rtc_event_log/encoder/optional_blob_encoding.h b/logging/rtc_event_log/encoder/optional_blob_encoding.h
index 32f5278..cd40523 100644
--- a/logging/rtc_event_log/encoder/optional_blob_encoding.h
+++ b/logging/rtc_event_log/encoder/optional_blob_encoding.h
@@ -13,11 +13,11 @@
#include <stddef.h>
+#include <optional>
#include <string>
#include <vector>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
namespace webrtc {
@@ -26,12 +26,12 @@
// in a way that would allow us to separate them again on the decoding side.
// EncodeOptionalBlobs() may not fail but may return an empty string
std::string EncodeOptionalBlobs(
- const std::vector<absl::optional<std::string>>& blobs);
+ const std::vector<std::optional<std::string>>& blobs);
// Calling DecodeOptionalBlobs() on an empty string, or with `num_of_blobs` set
// to 0, is an error. DecodeOptionalBlobs() returns an empty vector if it fails,
// which can happen if `encoded_blobs` is corrupted.
-std::vector<absl::optional<std::string>> DecodeOptionalBlobs(
+std::vector<std::optional<std::string>> DecodeOptionalBlobs(
absl::string_view encoded_blobs,
size_t num_of_blobs);
diff --git a/logging/rtc_event_log/encoder/optional_blob_encoding_unittest.cc b/logging/rtc_event_log/encoder/optional_blob_encoding_unittest.cc
index 8f0e484..7e7bccd 100644
--- a/logging/rtc_event_log/encoder/optional_blob_encoding_unittest.cc
+++ b/logging/rtc_event_log/encoder/optional_blob_encoding_unittest.cc
@@ -11,10 +11,10 @@
#include "logging/rtc_event_log/encoder/optional_blob_encoding.h"
#include <cstdint>
+#include <optional>
#include <string>
#include <vector>
-#include "absl/types/optional.h"
#include "test/gmock.h"
#include "test/gtest.h"
@@ -72,7 +72,7 @@
}
TEST(OptionalBlobEncoding, SomeBlobsPresent) {
- std::string encoded = EncodeOptionalBlobs({"a", absl::nullopt, "c"});
+ std::string encoded = EncodeOptionalBlobs({"a", std::nullopt, "c"});
std::string expected = BitBuilder()
.Bit(0)
.Bit(1)
@@ -87,12 +87,12 @@
TEST(OptionalBlobEncoding, NoBlobsPresent) {
std::string encoded =
- EncodeOptionalBlobs({absl::nullopt, absl::nullopt, absl::nullopt});
+ EncodeOptionalBlobs({std::nullopt, std::nullopt, std::nullopt});
EXPECT_THAT(encoded, IsEmpty());
}
TEST(OptionalBlobEncoding, EmptyBlobsPresent) {
- std::string encoded = EncodeOptionalBlobs({absl::nullopt, "", absl::nullopt});
+ std::string encoded = EncodeOptionalBlobs({std::nullopt, "", std::nullopt});
std::string expected = BitBuilder()
.Bit(0)
.Bit(0)
@@ -148,13 +148,12 @@
.Bytes({0x01, 'c'})
.AsString();
auto decoded = DecodeOptionalBlobs(encoded, 3);
- EXPECT_THAT(decoded, ElementsAre("a", absl::nullopt, "c"));
+ EXPECT_THAT(decoded, ElementsAre("a", std::nullopt, "c"));
}
TEST(OptionalBlobDecoding, NoBlobsPresent) {
auto decoded = DecodeOptionalBlobs("", 3);
- EXPECT_THAT(decoded,
- ElementsAre(absl::nullopt, absl::nullopt, absl::nullopt));
+ EXPECT_THAT(decoded, ElementsAre(std::nullopt, std::nullopt, std::nullopt));
}
TEST(OptionalBlobDecoding, EmptyBlobsPresent) {
@@ -167,7 +166,7 @@
.Bytes({0x0})
.AsString();
auto decoded = DecodeOptionalBlobs(encoded, 3);
- EXPECT_THAT(decoded, ElementsAre(absl::nullopt, "", absl::nullopt));
+ EXPECT_THAT(decoded, ElementsAre(std::nullopt, "", std::nullopt));
}
TEST(OptionalBlobDecoding, ZeroBlobs) {
diff --git a/logging/rtc_event_log/encoder/rtc_event_log_encoder_legacy.cc b/logging/rtc_event_log/encoder/rtc_event_log_encoder_legacy.cc
index 44ef8b3..506abf4 100644
--- a/logging/rtc_event_log/encoder/rtc_event_log_encoder_legacy.cc
+++ b/logging/rtc_event_log/encoder/rtc_event_log_encoder_legacy.cc
@@ -15,10 +15,10 @@
#include <cstdint>
#include <deque>
#include <memory>
+#include <optional>
#include <string>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/candidate.h"
#include "api/rtc_event_log/rtc_event.h"
diff --git a/logging/rtc_event_log/encoder/rtc_event_log_encoder_new_format.cc b/logging/rtc_event_log/encoder/rtc_event_log_encoder_new_format.cc
index 6c57328..ce3b71b 100644
--- a/logging/rtc_event_log/encoder/rtc_event_log_encoder_new_format.cc
+++ b/logging/rtc_event_log/encoder/rtc_event_log_encoder_new_format.cc
@@ -16,11 +16,11 @@
#include <deque>
#include <map>
#include <memory>
+#include <optional>
#include <string>
#include <type_traits>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/candidate.h"
#include "api/dtls_transport_interface.h"
@@ -366,7 +366,7 @@
// Delta encoding
proto_batch->set_number_of_deltas(batch.size() - 1);
- std::vector<absl::optional<uint64_t>> values(batch.size() - 1);
+ std::vector<std::optional<uint64_t>> values(batch.size() - 1);
std::string encoded_deltas;
// timestamp_ms
@@ -417,7 +417,7 @@
proto_batch->set_padding_size(base_event->padding_length());
// Add header extensions (base event).
- absl::optional<uint64_t> base_transport_sequence_number;
+ std::optional<uint64_t> base_transport_sequence_number;
{
uint16_t seqnum;
if (base_event->template GetExtension<TransportSequenceNumber>(&seqnum)) {
@@ -426,7 +426,7 @@
}
}
- absl::optional<uint64_t> unsigned_base_transmission_time_offset;
+ std::optional<uint64_t> unsigned_base_transmission_time_offset;
{
int32_t offset;
if (base_event->template GetExtension<TransmissionOffset>(&offset)) {
@@ -435,7 +435,7 @@
}
}
- absl::optional<uint64_t> base_absolute_send_time;
+ std::optional<uint64_t> base_absolute_send_time;
{
uint32_t sendtime;
if (base_event->template GetExtension<AbsoluteSendTime>(&sendtime)) {
@@ -444,7 +444,7 @@
}
}
- absl::optional<uint64_t> base_video_rotation;
+ std::optional<uint64_t> base_video_rotation;
{
VideoRotation video_rotation;
if (base_event->template GetExtension<VideoOrientation>(&video_rotation)) {
@@ -454,8 +454,8 @@
}
}
- absl::optional<uint64_t> base_audio_level;
- absl::optional<uint64_t> base_voice_activity;
+ std::optional<uint64_t> base_audio_level;
+ std::optional<uint64_t> base_voice_activity;
{
AudioLevel audio_level;
if (base_event->template GetExtension<AudioLevelExtension>(&audio_level)) {
@@ -496,7 +496,7 @@
// Delta encoding
proto_batch->set_number_of_deltas(batch.size() - 1);
- std::vector<absl::optional<uint64_t>> values(batch.size() - 1);
+ std::vector<std::optional<uint64_t>> values(batch.size() - 1);
std::string encoded_deltas;
// timestamp_ms (event)
@@ -998,7 +998,7 @@
proto_batch->set_frame_length_ms(
base_event->config().frame_length_ms.value());
}
- absl::optional<uint64_t> base_uplink_packet_loss_fraction;
+ std::optional<uint64_t> base_uplink_packet_loss_fraction;
if (base_event->config().uplink_packet_loss_fraction.has_value()) {
base_uplink_packet_loss_fraction = ConvertPacketLossFractionToProtoFormat(
base_event->config().uplink_packet_loss_fraction.value());
@@ -1020,7 +1020,7 @@
// Delta encoding
proto_batch->set_number_of_deltas(batch.size() - 1);
- std::vector<absl::optional<uint64_t>> values(batch.size() - 1);
+ std::vector<std::optional<uint64_t>> values(batch.size() - 1);
std::string encoded_deltas;
// timestamp_ms
@@ -1042,10 +1042,10 @@
values[i].reset();
}
}
- const absl::optional<uint64_t> unsigned_base_bitrate_bps =
+ const std::optional<uint64_t> unsigned_base_bitrate_bps =
base_event->config().bitrate_bps.has_value()
? ToUnsigned(base_event->config().bitrate_bps.value())
- : absl::optional<uint64_t>();
+ : std::optional<uint64_t>();
encoded_deltas = EncodeDeltas(unsigned_base_bitrate_bps, values);
if (!encoded_deltas.empty()) {
proto_batch->set_bitrate_bps_deltas(encoded_deltas);
@@ -1060,10 +1060,10 @@
values[i].reset();
}
}
- const absl::optional<uint64_t> unsigned_base_frame_length_ms =
+ const std::optional<uint64_t> unsigned_base_frame_length_ms =
base_event->config().frame_length_ms.has_value()
? ToUnsigned(base_event->config().frame_length_ms.value())
- : absl::optional<uint64_t>();
+ : std::optional<uint64_t>();
encoded_deltas = EncodeDeltas(unsigned_base_frame_length_ms, values);
if (!encoded_deltas.empty()) {
proto_batch->set_frame_length_ms_deltas(encoded_deltas);
@@ -1107,7 +1107,7 @@
// num_channels
for (size_t i = 0; i < values.size(); ++i) {
const RtcEventAudioNetworkAdaptation* event = batch[i + 1];
- const absl::optional<size_t> num_channels = event->config().num_channels;
+ const std::optional<size_t> num_channels = event->config().num_channels;
if (num_channels.has_value()) {
// Since the number of channels is always greater than 0, we can encode
// N channels as N-1, thereby making sure that we get smaller deltas.
@@ -1122,7 +1122,7 @@
}
// In the base event, N channels encoded as N channels, but for delta
// compression purposes, also shifted down by 1.
- absl::optional<size_t> shifted_base_num_channels;
+ std::optional<size_t> shifted_base_num_channels;
if (base_event->config().num_channels.has_value()) {
RTC_DCHECK_GT(base_event->config().num_channels.value(), 0u);
shifted_base_num_channels = base_event->config().num_channels.value() - 1;
@@ -1151,7 +1151,7 @@
// Delta encoding
proto_batch->set_number_of_deltas(batch.size() - 1);
- std::vector<absl::optional<uint64_t>> values(batch.size() - 1);
+ std::vector<std::optional<uint64_t>> values(batch.size() - 1);
std::string encoded_deltas;
// timestamp_ms
@@ -1198,7 +1198,7 @@
// Delta encoding
proto_batch->set_number_of_deltas(batch.size() - 1);
- std::vector<absl::optional<uint64_t>> values(batch.size() - 1);
+ std::vector<std::optional<uint64_t>> values(batch.size() - 1);
std::string encoded_deltas;
// timestamp_ms
@@ -1290,7 +1290,7 @@
// Delta encoding
proto_batch->set_number_of_deltas(batch.size() - 1);
- std::vector<absl::optional<uint64_t>> values(batch.size() - 1);
+ std::vector<std::optional<uint64_t>> values(batch.size() - 1);
std::string encoded_deltas;
// timestamp_ms
@@ -1347,7 +1347,7 @@
// Delta encoding
proto_batch->set_number_of_deltas(batch.size() - 1);
- std::vector<absl::optional<uint64_t>> values(batch.size() - 1);
+ std::vector<std::optional<uint64_t>> values(batch.size() - 1);
std::string encoded_deltas;
// timestamp_ms
@@ -1478,13 +1478,13 @@
proto_batch->set_timestamp_ms(base_event->timestamp_ms());
- absl::optional<uint64_t> base_link_capacity_lower;
+ std::optional<uint64_t> base_link_capacity_lower;
if (base_event->link_capacity_lower_.IsFinite()) {
base_link_capacity_lower =
base_event->link_capacity_lower_.kbps<uint32_t>();
proto_batch->set_link_capacity_lower_kbps(*base_link_capacity_lower);
}
- absl::optional<uint64_t> base_link_capacity_upper;
+ std::optional<uint64_t> base_link_capacity_upper;
if (base_event->link_capacity_upper_.IsFinite()) {
base_link_capacity_upper =
base_event->link_capacity_upper_.kbps<uint32_t>();
@@ -1496,7 +1496,7 @@
// Delta encoding
proto_batch->set_number_of_deltas(batch.size() - 1);
- std::vector<absl::optional<uint64_t>> values(batch.size() - 1);
+ std::vector<std::optional<uint64_t>> values(batch.size() - 1);
std::string encoded_deltas;
// timestamp_ms
@@ -1589,7 +1589,7 @@
// Delta encoding
proto_batch->set_number_of_deltas(batch.size() - 1);
- std::vector<absl::optional<uint64_t>> values(batch.size() - 1);
+ std::vector<std::optional<uint64_t>> values(batch.size() - 1);
std::string encoded_deltas;
// timestamp_ms
@@ -1682,7 +1682,7 @@
// Delta encoding
proto_batch->set_number_of_deltas(batch.size() - 1);
- std::vector<absl::optional<uint64_t>> values(batch.size() - 1);
+ std::vector<std::optional<uint64_t>> values(batch.size() - 1);
std::string encoded_deltas;
if (batch.size() == 1) {
@@ -1756,7 +1756,7 @@
// Delta encoding
proto_batch->set_number_of_deltas(batch.size() - 1);
- std::vector<absl::optional<uint64_t>> values(batch.size() - 1);
+ std::vector<std::optional<uint64_t>> values(batch.size() - 1);
std::string encoded_deltas;
if (batch.size() == 1) {
@@ -1807,7 +1807,7 @@
proto_batch->set_timestamp_ms(base_event->timestamp_ms());
proto_batch->set_packet_number(base_event->packet_number());
proto_batch->set_acked_packet_number(base_event->acked_packet_number());
- absl::optional<uint64_t> base_receive_timestamp;
+ std::optional<uint64_t> base_receive_timestamp;
if (base_event->receive_acked_packet_time_ms()) {
int64_t receive_acked_packet_time_ms =
base_event->receive_acked_packet_time_ms().value();
@@ -1817,7 +1817,7 @@
// Delta encoding
proto_batch->set_number_of_deltas(batch.size() - 1);
- std::vector<absl::optional<uint64_t>> values(batch.size() - 1);
+ std::vector<std::optional<uint64_t>> values(batch.size() - 1);
std::string encoded_deltas;
if (batch.size() == 1) {
@@ -1862,7 +1862,7 @@
if (event->receive_acked_packet_time_ms()) {
values[i] = ToUnsigned(event->receive_acked_packet_time_ms().value());
} else {
- values[i] = absl::nullopt;
+ values[i] = std::nullopt;
}
}
encoded_deltas = EncodeDeltas(base_receive_timestamp, values);
diff --git a/logging/rtc_event_log/events/fixed_length_encoding_parameters_v3.cc b/logging/rtc_event_log/events/fixed_length_encoding_parameters_v3.cc
index 0ad49df..166dedd 100644
--- a/logging/rtc_event_log/events/fixed_length_encoding_parameters_v3.cc
+++ b/logging/rtc_event_log/events/fixed_length_encoding_parameters_v3.cc
@@ -12,8 +12,8 @@
#include <algorithm>
#include <cstdint>
+#include <optional>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "logging/rtc_event_log/events/rtc_event_field_extraction.h"
#include "rtc_base/checks.h"
@@ -110,7 +110,7 @@
return header;
}
-absl::optional<FixedLengthEncodingParametersV3>
+std::optional<FixedLengthEncodingParametersV3>
FixedLengthEncodingParametersV3::ParseDeltaHeader(uint64_t header,
uint64_t value_bit_width) {
uint64_t delta_bit_width = (header & ((1u << 6) - 1)) + 1;
@@ -119,7 +119,7 @@
if (header >= (1u << 8)) {
RTC_LOG(LS_ERROR) << "Failed to parse delta header; unread bits remaining.";
- return absl::nullopt;
+ return std::nullopt;
}
if (!ValidParameters(delta_bit_width, signed_deltas, values_optional,
@@ -129,7 +129,7 @@
<< delta_bit_width << " signed_deltas=" << signed_deltas
<< " values_optional=" << values_optional
<< " value_bit_width=" << value_bit_width;
- return absl::nullopt;
+ return std::nullopt;
}
return FixedLengthEncodingParametersV3(delta_bit_width, signed_deltas,
diff --git a/logging/rtc_event_log/events/fixed_length_encoding_parameters_v3.h b/logging/rtc_event_log/events/fixed_length_encoding_parameters_v3.h
index ce227a0..3640b20 100644
--- a/logging/rtc_event_log/events/fixed_length_encoding_parameters_v3.h
+++ b/logging/rtc_event_log/events/fixed_length_encoding_parameters_v3.h
@@ -12,8 +12,8 @@
#define LOGGING_RTC_EVENT_LOG_EVENTS_FIXED_LENGTH_ENCODING_PARAMETERS_V3_H_
#include <cstdint>
+#include <optional>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "logging/rtc_event_log/events/rtc_event_field_extraction.h"
@@ -38,7 +38,7 @@
rtc::ArrayView<const uint64_t> values,
uint64_t value_bit_width,
bool values_optional);
- static absl::optional<FixedLengthEncodingParametersV3> ParseDeltaHeader(
+ static std::optional<FixedLengthEncodingParametersV3> ParseDeltaHeader(
uint64_t header,
uint64_t value_bit_width);
diff --git a/logging/rtc_event_log/events/rtc_event_field_encoding.cc b/logging/rtc_event_log/events/rtc_event_field_encoding.cc
index 87132c9..08f7e84 100644
--- a/logging/rtc_event_log/events/rtc_event_field_encoding.cc
+++ b/logging/rtc_event_log/events/rtc_event_field_encoding.cc
@@ -13,11 +13,11 @@
#include <cstddef>
#include <cstdint>
#include <limits>
+#include <optional>
#include <string>
#include <vector>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/rtc_event_log/rtc_event.h"
#include "logging/rtc_event_log/encoder/bit_writer.h"
@@ -88,7 +88,7 @@
return std::string();
}
-absl::optional<FieldType> ConvertFieldType(uint64_t value) {
+std::optional<FieldType> ConvertFieldType(uint64_t value) {
switch (value) {
case static_cast<uint64_t>(FieldType::kFixed8):
return FieldType::kFixed8;
@@ -101,7 +101,7 @@
case static_cast<uint64_t>(FieldType::kString):
return FieldType::kString;
default:
- return absl::nullopt;
+ return std::nullopt;
}
}
diff --git a/logging/rtc_event_log/events/rtc_event_field_encoding.h b/logging/rtc_event_log/events/rtc_event_field_encoding.h
index 21a4b41..26c7f15 100644
--- a/logging/rtc_event_log/events/rtc_event_field_encoding.h
+++ b/logging/rtc_event_log/events/rtc_event_field_encoding.h
@@ -13,12 +13,12 @@
#include <cstddef>
#include <cstdint>
+#include <optional>
#include <string>
#include <type_traits>
#include <vector>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/rtc_event_log/rtc_event.h"
#include "logging/rtc_event_log/events/fixed_length_encoding_parameters_v3.h"
@@ -130,13 +130,13 @@
typename E,
std::enable_if_t<std::is_integral<T>::value, bool> = true>
ValuesWithPositions ExtractRtcEventMember(rtc::ArrayView<const RtcEvent*> batch,
- const absl::optional<T> E::*member) {
+ const std::optional<T> E::*member) {
ValuesWithPositions result;
result.position_mask.reserve(batch.size());
result.values.reserve(batch.size());
for (const RtcEvent* event : batch) {
RTC_CHECK_EQ(event->GetType(), E::kType);
- absl::optional<T> field = static_cast<const E*>(event)->*member;
+ std::optional<T> field = static_cast<const E*>(event)->*member;
result.position_mask.push_back(field.has_value());
if (field.has_value()) {
result.values.push_back(EncodeAsUnsigned(field.value()));
diff --git a/logging/rtc_event_log/events/rtc_event_field_encoding_parser.cc b/logging/rtc_event_log/events/rtc_event_field_encoding_parser.cc
index f55b25d..f4e1bb1 100644
--- a/logging/rtc_event_log/events/rtc_event_field_encoding_parser.cc
+++ b/logging/rtc_event_log/events/rtc_event_field_encoding_parser.cc
@@ -14,10 +14,10 @@
#include <algorithm>
#include <cstddef>
#include <cstdint>
+#include <optional>
#include <tuple>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "logging/rtc_event_log/encoder/var_int.h"
#include "logging/rtc_event_log/events/fixed_length_encoding_parameters_v3.h"
@@ -28,7 +28,7 @@
#include "rtc_base/checks.h"
namespace {
-absl::optional<webrtc::FieldType> ConvertFieldType(uint64_t value) {
+std::optional<webrtc::FieldType> ConvertFieldType(uint64_t value) {
switch (value) {
case static_cast<uint64_t>(webrtc::FieldType::kFixed8):
return webrtc::FieldType::kFixed8;
@@ -41,7 +41,7 @@
case static_cast<uint64_t>(webrtc::FieldType::kString):
return webrtc::FieldType::kString;
default:
- return absl::nullopt;
+ return std::nullopt;
}
}
} // namespace
@@ -209,7 +209,7 @@
__FILE__, __LINE__);
// NB: value_bit_width may be incorrect for the field, if this isn't the
// field we are looking for.
- absl::optional<FixedLengthEncodingParametersV3> delta_header =
+ std::optional<FixedLengthEncodingParametersV3> delta_header =
FixedLengthEncodingParametersV3::ParseDeltaHeader(header_value,
value_bit_width);
if (!delta_header.has_value()) {
@@ -318,7 +318,7 @@
__FILE__, __LINE__);
// Split tag into field ID and field type.
field_id = field_tag >> 3;
- absl::optional<FieldType> conversion = ConvertFieldType(field_tag & 7u);
+ std::optional<FieldType> conversion = ConvertFieldType(field_tag & 7u);
if (!conversion.has_value())
return RtcEventLogParseStatus::Error("Failed to parse field type",
__FILE__, __LINE__);
diff --git a/logging/rtc_event_log/events/rtc_event_field_encoding_parser.h b/logging/rtc_event_log/events/rtc_event_field_encoding_parser.h
index a47cc3c..de95160 100644
--- a/logging/rtc_event_log/events/rtc_event_field_encoding_parser.h
+++ b/logging/rtc_event_log/events/rtc_event_field_encoding_parser.h
@@ -13,13 +13,13 @@
#include <cstddef>
#include <cstdint>
+#include <optional>
#include <string>
#include <type_traits>
#include <vector>
#include "absl/base/attributes.h"
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/units/timestamp.h"
#include "logging/rtc_event_log/events/fixed_length_encoding_parameters_v3.h"
@@ -140,7 +140,7 @@
ABSL_MUST_USE_RESULT RtcEventLogParseStatus
PopulateRtcEventMember(const rtc::ArrayView<uint8_t> positions,
const rtc::ArrayView<uint64_t> values,
- absl::optional<T> E::*member,
+ std::optional<T> E::*member,
rtc::ArrayView<E> output) {
size_t batch_size = positions.size();
RTC_CHECK_EQ(output.size(), batch_size);
@@ -152,7 +152,7 @@
output[i].*member = DecodeFromUnsignedToType<T>(value_it);
++value_it;
} else {
- output[i].*member = absl::nullopt;
+ output[i].*member = std::nullopt;
}
}
RTC_CHECK(value_it == values.end());
diff --git a/logging/rtc_event_log/events/rtc_event_field_encoding_unittest.cc b/logging/rtc_event_log/events/rtc_event_field_encoding_unittest.cc
index 62004e8..5fd6928 100644
--- a/logging/rtc_event_log/events/rtc_event_field_encoding_unittest.cc
+++ b/logging/rtc_event_log/events/rtc_event_field_encoding_unittest.cc
@@ -12,13 +12,13 @@
#include <cstdint>
#include <cstdio>
#include <limits>
+#include <optional>
#include <string>
#include <tuple>
#include <type_traits>
#include <vector>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/rtc_event_log/rtc_event.h"
#include "logging/rtc_event_log/encoder/var_int.h"
@@ -93,12 +93,12 @@
template <typename T, std::enable_if_t<std::is_integral<T>::value, bool> = true>
size_t ExpectedEncodingSize(const FieldParameters& params,
- const std::vector<absl::optional<T>>& v,
+ const std::vector<std::optional<T>>& v,
size_t expected_bits_per_delta) {
size_t num_existing_values =
- v.size() - std::count(v.begin(), v.end(), absl::nullopt);
+ v.size() - std::count(v.begin(), v.end(), std::nullopt);
auto first_existing_value = std::find_if(
- v.begin(), v.end(), [](absl::optional<T> x) { return x.has_value(); });
+ v.begin(), v.end(), [](std::optional<T> x) { return x.has_value(); });
if (num_existing_values == 0)
return 0;
@@ -116,7 +116,7 @@
(num_existing_values == v.size() ? 0 : (v.size() + 7) / 8);
// Check if there is an element *not* equal to base.
if (std::all_of(v.begin(), v.end(),
- [base](absl::optional<T> x) { return x == base; })) {
+ [base](std::optional<T> x) { return x == base; })) {
return tag_size + base_size + delta_header_size + positions_size;
}
@@ -164,8 +164,8 @@
uint32_t unsigned32,
int64_t signed64,
uint64_t unsigned64,
- absl::optional<int32_t> optional_signed32,
- absl::optional<int64_t> optional_signed64,
+ std::optional<int32_t> optional_signed32,
+ std::optional<int64_t> optional_signed64,
uint32_t wrapping21,
absl::string_view string)
: b_(b),
@@ -211,8 +211,8 @@
const uint32_t unsigned32_;
const int64_t signed64_;
const uint64_t unsigned64_;
- const absl::optional<int32_t> optional_signed32_ = absl::nullopt;
- const absl::optional<int64_t> optional_signed64_ = absl::nullopt;
+ const std::optional<int32_t> optional_signed32_ = std::nullopt;
+ const std::optional<int64_t> optional_signed64_ = std::nullopt;
const uint32_t wrapping21_ = 0;
const std::string string_;
};
@@ -242,8 +242,8 @@
const std::vector<uint32_t>& unsigned32_values,
const std::vector<int64_t>& signed64_values,
const std::vector<uint64_t>& unsigned64_values,
- const std::vector<absl::optional<int32_t>>& optional32_values,
- const std::vector<absl::optional<int64_t>>& optional64_values,
+ const std::vector<std::optional<int32_t>>& optional32_values,
+ const std::vector<std::optional<int64_t>>& optional64_values,
const std::vector<uint32_t>& wrapping21_values,
const std::vector<std::string>& string_values) {
size_t size = bool_values.size();
@@ -340,7 +340,7 @@
template <typename T>
void ParseAndVerifyOptionalField(
const FieldParameters& params,
- const std::vector<absl::optional<T>>& expected_values,
+ const std::vector<std::optional<T>>& expected_values,
size_t expected_bits_per_delta,
size_t expected_skipped_bytes = 0) {
size_t expected_size =
@@ -361,7 +361,7 @@
expected_values[i].value());
++value_it;
} else {
- EXPECT_EQ(absl::nullopt, expected_values[i]);
+ EXPECT_EQ(std::nullopt, expected_values[i]);
}
}
EXPECT_EQ(value_it, values.end());
@@ -409,8 +409,8 @@
std::vector<uint32_t> unsigned32_values = {123456789};
std::vector<int64_t> signed64_values = {-9876543210};
std::vector<uint64_t> unsigned64_values = {9876543210};
- std::vector<absl::optional<int32_t>> optional32_values = {kInt32Min};
- std::vector<absl::optional<int64_t>> optional64_values = {kInt64Max};
+ std::vector<std::optional<int32_t>> optional32_values = {kInt32Min};
+ std::vector<std::optional<int64_t>> optional64_values = {kInt64Max};
std::vector<uint32_t> wrapping21_values = {(1 << 21) - 1};
std::vector<std::string> string_values = {"foo"};
@@ -478,9 +478,9 @@
-9876543210};
std::vector<uint64_t> unsigned64_values = {9876543210, 9876543210, 9876543210,
9876543210};
- std::vector<absl::optional<int32_t>> optional32_values = {
+ std::vector<std::optional<int32_t>> optional32_values = {
kInt32Min, kInt32Min, kInt32Min, kInt32Min};
- std::vector<absl::optional<int64_t>> optional64_values = {
+ std::vector<std::optional<int64_t>> optional64_values = {
kInt64Max, kInt64Max, kInt64Max, kInt64Max};
std::vector<uint32_t> wrapping21_values = {(1 << 21) - 1, (1 << 21) - 1,
(1 << 21) - 1, (1 << 21) - 1};
@@ -547,9 +547,9 @@
std::vector<int64_t> signed64_values = {kInt64Max - 1, kInt64Max, kInt64Min,
kInt64Min + 1};
std::vector<uint64_t> unsigned64_values = {kUint64Max - 1, kUint64Max, 0, 1};
- std::vector<absl::optional<int32_t>> optional32_values = {
+ std::vector<std::optional<int32_t>> optional32_values = {
kInt32Max - 1, kInt32Max, kInt32Min, kInt32Min + 1};
- std::vector<absl::optional<int64_t>> optional64_values = {
+ std::vector<std::optional<int64_t>> optional64_values = {
kInt64Max - 1, kInt64Max, kInt64Min, kInt64Min + 1};
std::vector<uint32_t> wrapping21_values = {(1 << 21) - 2, (1 << 21) - 1, 0,
1};
@@ -618,9 +618,9 @@
std::vector<int64_t> signed64_values = {kInt64Min + 1, kInt64Min, kInt64Max,
kInt64Max - 1};
std::vector<uint64_t> unsigned64_values = {1, 0, kUint64Max, kUint64Max - 1};
- std::vector<absl::optional<int32_t>> optional32_values = {
+ std::vector<std::optional<int32_t>> optional32_values = {
kInt32Min + 1, kInt32Min, kInt32Max, kInt32Max - 1};
- std::vector<absl::optional<int64_t>> optional64_values = {
+ std::vector<std::optional<int64_t>> optional64_values = {
kInt64Min + 1, kInt64Min, kInt64Max, kInt64Max - 1};
std::vector<uint32_t> wrapping21_values = {1, 0, (1 << 21) - 1,
(1 << 21) - 2};
@@ -690,10 +690,10 @@
std::vector<uint32_t> unsigned32_values = {0, kUint32Max / 2};
std::vector<int64_t> signed64_values = {kInt64Min / 2, kInt64Max / 2};
std::vector<uint64_t> unsigned64_values = {0, kUint64Max / 2};
- std::vector<absl::optional<int32_t>> optional32_values = {kInt32Max / 2,
- kInt32Min / 2};
- std::vector<absl::optional<int64_t>> optional64_values = {kInt64Min / 2,
- kInt64Max / 2};
+ std::vector<std::optional<int32_t>> optional32_values = {kInt32Max / 2,
+ kInt32Min / 2};
+ std::vector<std::optional<int64_t>> optional64_values = {kInt64Min / 2,
+ kInt64Max / 2};
std::vector<uint32_t> wrapping21_values = {0, 1 << 20};
std::vector<std::string> string_values = {"foo", "bar"};
@@ -769,10 +769,10 @@
std::vector<uint32_t> unsigned32_values = {0, kUint32Max / 2};
std::vector<int64_t> signed64_values = {kInt64Min / 2, kInt64Max / 2};
std::vector<uint64_t> unsigned64_values = {0, kUint64Max / 2};
- std::vector<absl::optional<int32_t>> optional32_values = {kInt32Max / 2,
- kInt32Min / 2};
- std::vector<absl::optional<int64_t>> optional64_values = {kInt64Min / 2,
- kInt64Max / 2};
+ std::vector<std::optional<int32_t>> optional32_values = {kInt32Max / 2,
+ kInt32Min / 2};
+ std::vector<std::optional<int64_t>> optional64_values = {kInt64Min / 2,
+ kInt64Max / 2};
std::vector<uint32_t> wrapping21_values = {0, 1 << 20};
std::vector<std::string> string_values = {"foo", "foo"};
@@ -821,10 +821,10 @@
}
TEST_F(RtcEventFieldTest, OptionalFields) {
- std::vector<absl::optional<int32_t>> optional32_values = {
- 2, absl::nullopt, 4, absl::nullopt, 6, absl::nullopt};
- std::vector<absl::optional<int64_t>> optional64_values = {
- absl::nullopt, 1024, absl::nullopt, 1025, absl::nullopt, 1026};
+ std::vector<std::optional<int32_t>> optional32_values = {
+ 2, std::nullopt, 4, std::nullopt, 6, std::nullopt};
+ std::vector<std::optional<int64_t>> optional64_values = {
+ std::nullopt, 1024, std::nullopt, 1025, std::nullopt, 1026};
std::vector<uint32_t> wrapping21_values = {(1 << 21) - 3, 0, 2, 5, 5, 6};
for (size_t i = 0; i < optional32_values.size(); i++) {
@@ -860,11 +860,11 @@
}
TEST_F(RtcEventFieldTest, AllNulloptTreatedAsMissing) {
- std::vector<absl::optional<int32_t>> optional32_values = {
- absl::nullopt, absl::nullopt, absl::nullopt,
- absl::nullopt, absl::nullopt, absl::nullopt};
- std::vector<absl::optional<int64_t>> optional64_values = {
- absl::nullopt, 1024, absl::nullopt, 1025, absl::nullopt, 1026};
+ std::vector<std::optional<int32_t>> optional32_values = {
+ std::nullopt, std::nullopt, std::nullopt,
+ std::nullopt, std::nullopt, std::nullopt};
+ std::vector<std::optional<int64_t>> optional64_values = {
+ std::nullopt, 1024, std::nullopt, 1025, std::nullopt, 1026};
for (size_t i = 0; i < optional32_values.size(); i++) {
batch_.push_back(new RtcTestEvent(0, 0, 0, 0, 0, optional32_values[i],
diff --git a/logging/rtc_event_log/events/rtc_event_generic_ack_received.cc b/logging/rtc_event_log/events/rtc_event_generic_ack_received.cc
index e5aa375..39dda3d 100644
--- a/logging/rtc_event_log/events/rtc_event_generic_ack_received.cc
+++ b/logging/rtc_event_log/events/rtc_event_generic_ack_received.cc
@@ -12,10 +12,10 @@
#include <cstdint>
#include <memory>
+#include <optional>
#include <vector>
#include "absl/memory/memory.h"
-#include "absl/types/optional.h"
#include "api/rtc_event_log/rtc_event.h"
#include "rtc_base/time_utils.h"
@@ -40,7 +40,7 @@
int64_t timestamp_us,
int64_t packet_number,
int64_t acked_packet_number,
- absl::optional<int64_t> receive_acked_packet_time_ms)
+ std::optional<int64_t> receive_acked_packet_time_ms)
: RtcEvent(timestamp_us),
packet_number_(packet_number),
acked_packet_number_(acked_packet_number),
diff --git a/logging/rtc_event_log/events/rtc_event_generic_ack_received.h b/logging/rtc_event_log/events/rtc_event_generic_ack_received.h
index 16b2a9e..62ee5c0 100644
--- a/logging/rtc_event_log/events/rtc_event_generic_ack_received.h
+++ b/logging/rtc_event_log/events/rtc_event_generic_ack_received.h
@@ -13,11 +13,11 @@
#include <cstdint>
#include <memory>
+#include <optional>
#include <string>
#include <vector>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/rtc_event_log/rtc_event.h"
#include "api/units/timestamp.h"
@@ -30,7 +30,7 @@
LoggedGenericAckReceived(Timestamp timestamp,
int64_t packet_number,
int64_t acked_packet_number,
- absl::optional<int64_t> receive_acked_packet_time_ms)
+ std::optional<int64_t> receive_acked_packet_time_ms)
: timestamp(timestamp),
packet_number(packet_number),
acked_packet_number(acked_packet_number),
@@ -43,7 +43,7 @@
Timestamp timestamp = Timestamp::MinusInfinity();
int64_t packet_number;
int64_t acked_packet_number;
- absl::optional<int64_t> receive_acked_packet_time_ms;
+ std::optional<int64_t> receive_acked_packet_time_ms;
};
struct AckedPacket {
@@ -52,7 +52,7 @@
// The time where the packet was received. Not every ACK will
// include the receive timestamp.
- absl::optional<int64_t> receive_acked_packet_time_ms;
+ std::optional<int64_t> receive_acked_packet_time_ms;
};
class RtcEventGenericAckReceived final : public RtcEvent {
@@ -79,7 +79,7 @@
int64_t acked_packet_number() const { return acked_packet_number_; }
// Timestamp when the `acked_packet_number` was received by the remote side.
- absl::optional<int64_t> receive_acked_packet_time_ms() const {
+ std::optional<int64_t> receive_acked_packet_time_ms() const {
return receive_acked_packet_time_ms_;
}
@@ -108,11 +108,11 @@
int64_t timestamp_us,
int64_t packet_number,
int64_t acked_packet_number,
- absl::optional<int64_t> receive_acked_packet_time_ms);
+ std::optional<int64_t> receive_acked_packet_time_ms);
const int64_t packet_number_;
const int64_t acked_packet_number_;
- const absl::optional<int64_t> receive_acked_packet_time_ms_;
+ const std::optional<int64_t> receive_acked_packet_time_ms_;
};
} // namespace webrtc
diff --git a/logging/rtc_event_log/events/rtc_event_remote_estimate.h b/logging/rtc_event_log/events/rtc_event_remote_estimate.h
index 78366aa..d01abe7 100644
--- a/logging/rtc_event_log/events/rtc_event_remote_estimate.h
+++ b/logging/rtc_event_log/events/rtc_event_remote_estimate.h
@@ -11,11 +11,11 @@
#define LOGGING_RTC_EVENT_LOG_EVENTS_RTC_EVENT_REMOTE_ESTIMATE_H_
#include <cstdint>
+#include <optional>
#include <string>
#include <vector>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/rtc_event_log/rtc_event.h"
#include "api/units/data_rate.h"
@@ -32,8 +32,8 @@
Timestamp log_time() const { return timestamp; }
Timestamp timestamp = Timestamp::MinusInfinity();
- absl::optional<DataRate> link_capacity_lower;
- absl::optional<DataRate> link_capacity_upper;
+ std::optional<DataRate> link_capacity_lower;
+ std::optional<DataRate> link_capacity_upper;
};
class RtcEventRemoteEstimate final : public RtcEvent {
diff --git a/logging/rtc_event_log/rtc_event_log2rtp_dump.cc b/logging/rtc_event_log/rtc_event_log2rtp_dump.cc
index df30fe3..aac5fff 100644
--- a/logging/rtc_event_log/rtc_event_log2rtp_dump.cc
+++ b/logging/rtc_event_log/rtc_event_log2rtp_dump.cc
@@ -13,6 +13,7 @@
#include <iostream>
#include <memory>
+#include <optional>
#include <string>
#include <vector>
@@ -20,7 +21,6 @@
#include "absl/flags/parse.h"
#include "absl/flags/usage.h"
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/rtp_headers.h"
#include "logging/rtc_event_log/events/logged_rtp_rtcp.h"
@@ -75,14 +75,14 @@
// The empty string must be validated as true, because it is the default value
// of the command-line flag. In this case, no value is written to the output
// variable.
-absl::optional<uint32_t> ParseSsrc(absl::string_view str) {
+std::optional<uint32_t> ParseSsrc(absl::string_view str) {
// Set `base` to 0 to allow detection of the "0x" prefix in case hex is used.
return rtc::StringToNumber<uint32_t>(str, 0);
}
bool ShouldSkipStream(MediaType media_type,
uint32_t ssrc,
- absl::optional<uint32_t> ssrc_filter) {
+ std::optional<uint32_t> ssrc_filter) {
if (!absl::GetFlag(FLAGS_audio) && media_type == MediaType::AUDIO)
return true;
if (!absl::GetFlag(FLAGS_video) && media_type == MediaType::VIDEO)
@@ -167,7 +167,7 @@
std::string input_file = args[1];
std::string output_file = args[2];
- absl::optional<uint32_t> ssrc_filter;
+ std::optional<uint32_t> ssrc_filter;
if (!absl::GetFlag(FLAGS_ssrc).empty()) {
ssrc_filter = ParseSsrc(absl::GetFlag(FLAGS_ssrc));
RTC_CHECK(ssrc_filter.has_value()) << "Failed to read SSRC filter flag.";
diff --git a/logging/rtc_event_log/rtc_event_log_parser.cc b/logging/rtc_event_log/rtc_event_log_parser.cc
index f1e4878..7197c97 100644
--- a/logging/rtc_event_log/rtc_event_log_parser.cc
+++ b/logging/rtc_event_log/rtc_event_log_parser.cc
@@ -16,6 +16,7 @@
#include <algorithm>
#include <limits>
#include <map>
+#include <optional>
#include <set>
#include <string>
#include <tuple>
@@ -23,7 +24,6 @@
#include <vector>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/candidate.h"
#include "api/dtls_transport_interface.h"
#include "api/rtc_event_log/rtc_event.h"
@@ -431,59 +431,59 @@
}
// timestamp_ms (event)
- std::vector<absl::optional<uint64_t>> timestamp_ms_values =
+ std::vector<std::optional<uint64_t>> timestamp_ms_values =
DecodeDeltas(proto.timestamp_ms_deltas(),
ToUnsigned(proto.timestamp_ms()), number_of_deltas);
RTC_PARSE_CHECK_OR_RETURN_EQ(timestamp_ms_values.size(), number_of_deltas);
// marker (RTP base)
- std::vector<absl::optional<uint64_t>> marker_values =
+ std::vector<std::optional<uint64_t>> marker_values =
DecodeDeltas(proto.marker_deltas(), proto.marker(), number_of_deltas);
RTC_PARSE_CHECK_OR_RETURN_EQ(marker_values.size(), number_of_deltas);
// payload_type (RTP base)
- std::vector<absl::optional<uint64_t>> payload_type_values = DecodeDeltas(
+ std::vector<std::optional<uint64_t>> payload_type_values = DecodeDeltas(
proto.payload_type_deltas(), proto.payload_type(), number_of_deltas);
RTC_PARSE_CHECK_OR_RETURN_EQ(payload_type_values.size(), number_of_deltas);
// sequence_number (RTP base)
- std::vector<absl::optional<uint64_t>> sequence_number_values =
+ std::vector<std::optional<uint64_t>> sequence_number_values =
DecodeDeltas(proto.sequence_number_deltas(), proto.sequence_number(),
number_of_deltas);
RTC_PARSE_CHECK_OR_RETURN_EQ(sequence_number_values.size(), number_of_deltas);
// rtp_timestamp (RTP base)
- std::vector<absl::optional<uint64_t>> rtp_timestamp_values = DecodeDeltas(
+ std::vector<std::optional<uint64_t>> rtp_timestamp_values = DecodeDeltas(
proto.rtp_timestamp_deltas(), proto.rtp_timestamp(), number_of_deltas);
RTC_PARSE_CHECK_OR_RETURN_EQ(rtp_timestamp_values.size(), number_of_deltas);
// ssrc (RTP base)
- std::vector<absl::optional<uint64_t>> ssrc_values =
+ std::vector<std::optional<uint64_t>> ssrc_values =
DecodeDeltas(proto.ssrc_deltas(), proto.ssrc(), number_of_deltas);
RTC_PARSE_CHECK_OR_RETURN_EQ(ssrc_values.size(), number_of_deltas);
// payload_size (RTP base)
- std::vector<absl::optional<uint64_t>> payload_size_values = DecodeDeltas(
+ std::vector<std::optional<uint64_t>> payload_size_values = DecodeDeltas(
proto.payload_size_deltas(), proto.payload_size(), number_of_deltas);
RTC_PARSE_CHECK_OR_RETURN_EQ(payload_size_values.size(), number_of_deltas);
// header_size (RTP base)
- std::vector<absl::optional<uint64_t>> header_size_values = DecodeDeltas(
+ std::vector<std::optional<uint64_t>> header_size_values = DecodeDeltas(
proto.header_size_deltas(), proto.header_size(), number_of_deltas);
RTC_PARSE_CHECK_OR_RETURN_EQ(header_size_values.size(), number_of_deltas);
// padding_size (RTP base)
- std::vector<absl::optional<uint64_t>> padding_size_values = DecodeDeltas(
+ std::vector<std::optional<uint64_t>> padding_size_values = DecodeDeltas(
proto.padding_size_deltas(), proto.padding_size(), number_of_deltas);
RTC_PARSE_CHECK_OR_RETURN_EQ(padding_size_values.size(), number_of_deltas);
// transport_sequence_number (RTP extension)
- std::vector<absl::optional<uint64_t>> transport_sequence_number_values;
+ std::vector<std::optional<uint64_t>> transport_sequence_number_values;
{
- const absl::optional<uint64_t> base_transport_sequence_number =
+ const std::optional<uint64_t> base_transport_sequence_number =
proto.has_transport_sequence_number()
? proto.transport_sequence_number()
- : absl::optional<uint64_t>();
+ : std::optional<uint64_t>();
transport_sequence_number_values =
DecodeDeltas(proto.transport_sequence_number_deltas(),
base_transport_sequence_number, number_of_deltas);
@@ -492,12 +492,12 @@
}
// transmission_time_offset (RTP extension)
- std::vector<absl::optional<uint64_t>> transmission_time_offset_values;
+ std::vector<std::optional<uint64_t>> transmission_time_offset_values;
{
- const absl::optional<uint64_t> unsigned_base_transmission_time_offset =
+ const std::optional<uint64_t> unsigned_base_transmission_time_offset =
proto.has_transmission_time_offset()
? ToUnsigned(proto.transmission_time_offset())
- : absl::optional<uint64_t>();
+ : std::optional<uint64_t>();
transmission_time_offset_values =
DecodeDeltas(proto.transmission_time_offset_deltas(),
unsigned_base_transmission_time_offset, number_of_deltas);
@@ -506,11 +506,11 @@
}
// absolute_send_time (RTP extension)
- std::vector<absl::optional<uint64_t>> absolute_send_time_values;
+ std::vector<std::optional<uint64_t>> absolute_send_time_values;
{
- const absl::optional<uint64_t> base_absolute_send_time =
+ const std::optional<uint64_t> base_absolute_send_time =
proto.has_absolute_send_time() ? proto.absolute_send_time()
- : absl::optional<uint64_t>();
+ : std::optional<uint64_t>();
absolute_send_time_values =
DecodeDeltas(proto.absolute_send_time_deltas(), base_absolute_send_time,
number_of_deltas);
@@ -519,11 +519,11 @@
}
// video_rotation (RTP extension)
- std::vector<absl::optional<uint64_t>> video_rotation_values;
+ std::vector<std::optional<uint64_t>> video_rotation_values;
{
- const absl::optional<uint64_t> base_video_rotation =
+ const std::optional<uint64_t> base_video_rotation =
proto.has_video_rotation() ? proto.video_rotation()
- : absl::optional<uint64_t>();
+ : std::optional<uint64_t>();
video_rotation_values = DecodeDeltas(proto.video_rotation_deltas(),
base_video_rotation, number_of_deltas);
RTC_PARSE_CHECK_OR_RETURN_EQ(video_rotation_values.size(),
@@ -531,22 +531,22 @@
}
// audio_level (RTP extension)
- std::vector<absl::optional<uint64_t>> audio_level_values;
+ std::vector<std::optional<uint64_t>> audio_level_values;
{
- const absl::optional<uint64_t> base_audio_level =
+ const std::optional<uint64_t> base_audio_level =
proto.has_audio_level() ? proto.audio_level()
- : absl::optional<uint64_t>();
+ : std::optional<uint64_t>();
audio_level_values = DecodeDeltas(proto.audio_level_deltas(),
base_audio_level, number_of_deltas);
RTC_PARSE_CHECK_OR_RETURN_EQ(audio_level_values.size(), number_of_deltas);
}
// voice_activity (RTP extension)
- std::vector<absl::optional<uint64_t>> voice_activity_values;
+ std::vector<std::optional<uint64_t>> voice_activity_values;
{
- const absl::optional<uint64_t> base_voice_activity =
+ const std::optional<uint64_t> base_voice_activity =
proto.has_voice_activity() ? proto.voice_activity()
- : absl::optional<uint64_t>();
+ : std::optional<uint64_t>();
voice_activity_values = DecodeDeltas(proto.voice_activity_deltas(),
base_voice_activity, number_of_deltas);
RTC_PARSE_CHECK_OR_RETURN_EQ(voice_activity_values.size(),
@@ -659,7 +659,7 @@
}
// timestamp_ms
- std::vector<absl::optional<uint64_t>> timestamp_ms_values =
+ std::vector<std::optional<uint64_t>> timestamp_ms_values =
DecodeDeltas(proto.timestamp_ms_deltas(),
ToUnsigned(proto.timestamp_ms()), number_of_deltas);
RTC_PARSE_CHECK_OR_RETURN_EQ(timestamp_ms_values.size(), number_of_deltas);
@@ -1178,7 +1178,7 @@
}
// Compute file size.
- absl::optional<size_t> file_size = file.FileSize();
+ std::optional<size_t> file_size = file.FileSize();
RTC_PARSE_CHECK_OR_RETURN(file_size.has_value());
RTC_PARSE_CHECK_OR_RETURN_GE(*file_size, 0u);
RTC_PARSE_CHECK_OR_RETURN_LE(*file_size, kMaxLogSize);
@@ -2685,14 +2685,14 @@
LoggedRemoteEstimateEvent base_event;
base_event.timestamp = Timestamp::Millis(proto.timestamp_ms());
- absl::optional<uint64_t> base_link_capacity_lower_kbps;
+ std::optional<uint64_t> base_link_capacity_lower_kbps;
if (proto.has_link_capacity_lower_kbps()) {
base_link_capacity_lower_kbps = proto.link_capacity_lower_kbps();
base_event.link_capacity_lower =
DataRate::KilobitsPerSec(proto.link_capacity_lower_kbps());
}
- absl::optional<uint64_t> base_link_capacity_upper_kbps;
+ std::optional<uint64_t> base_link_capacity_upper_kbps;
if (proto.has_link_capacity_upper_kbps()) {
base_link_capacity_upper_kbps = proto.link_capacity_upper_kbps();
base_event.link_capacity_upper =
@@ -2759,13 +2759,13 @@
}
// timestamp_ms
- std::vector<absl::optional<uint64_t>> timestamp_ms_values =
+ std::vector<std::optional<uint64_t>> timestamp_ms_values =
DecodeDeltas(proto.timestamp_ms_deltas(),
ToUnsigned(proto.timestamp_ms()), number_of_deltas);
RTC_PARSE_CHECK_OR_RETURN_EQ(timestamp_ms_values.size(), number_of_deltas);
// local_ssrc
- std::vector<absl::optional<uint64_t>> local_ssrc_values = DecodeDeltas(
+ std::vector<std::optional<uint64_t>> local_ssrc_values = DecodeDeltas(
proto.local_ssrc_deltas(), proto.local_ssrc(), number_of_deltas);
RTC_PARSE_CHECK_OR_RETURN_EQ(local_ssrc_values.size(), number_of_deltas);
@@ -2806,18 +2806,18 @@
}
// timestamp_ms
- std::vector<absl::optional<uint64_t>> timestamp_ms_values =
+ std::vector<std::optional<uint64_t>> timestamp_ms_values =
DecodeDeltas(proto.timestamp_ms_deltas(),
ToUnsigned(proto.timestamp_ms()), number_of_deltas);
RTC_PARSE_CHECK_OR_RETURN_EQ(timestamp_ms_values.size(), number_of_deltas);
// remote_ssrc
- std::vector<absl::optional<uint64_t>> remote_ssrc_values = DecodeDeltas(
+ std::vector<std::optional<uint64_t>> remote_ssrc_values = DecodeDeltas(
proto.remote_ssrc_deltas(), proto.remote_ssrc(), number_of_deltas);
RTC_PARSE_CHECK_OR_RETURN_EQ(remote_ssrc_values.size(), number_of_deltas);
// minimum_delay_ms
- std::vector<absl::optional<uint64_t>> minimum_delay_ms_values =
+ std::vector<std::optional<uint64_t>> minimum_delay_ms_values =
DecodeDeltas(proto.minimum_delay_ms_deltas(),
ToUnsigned(proto.minimum_delay_ms()), number_of_deltas);
RTC_PARSE_CHECK_OR_RETURN_EQ(minimum_delay_ms_values.size(),
@@ -2910,23 +2910,23 @@
}
// timestamp_ms
- std::vector<absl::optional<uint64_t>> timestamp_ms_values =
+ std::vector<std::optional<uint64_t>> timestamp_ms_values =
DecodeDeltas(proto.timestamp_ms_deltas(),
ToUnsigned(proto.timestamp_ms()), number_of_deltas);
RTC_PARSE_CHECK_OR_RETURN_EQ(timestamp_ms_values.size(), number_of_deltas);
// bitrate_bps
- std::vector<absl::optional<uint64_t>> bitrate_bps_values = DecodeDeltas(
+ std::vector<std::optional<uint64_t>> bitrate_bps_values = DecodeDeltas(
proto.bitrate_bps_deltas(), proto.bitrate_bps(), number_of_deltas);
RTC_PARSE_CHECK_OR_RETURN_EQ(bitrate_bps_values.size(), number_of_deltas);
// fraction_loss
- std::vector<absl::optional<uint64_t>> fraction_loss_values = DecodeDeltas(
+ std::vector<std::optional<uint64_t>> fraction_loss_values = DecodeDeltas(
proto.fraction_loss_deltas(), proto.fraction_loss(), number_of_deltas);
RTC_PARSE_CHECK_OR_RETURN_EQ(fraction_loss_values.size(), number_of_deltas);
// total_packets
- std::vector<absl::optional<uint64_t>> total_packets_values = DecodeDeltas(
+ std::vector<std::optional<uint64_t>> total_packets_values = DecodeDeltas(
proto.total_packets_deltas(), proto.total_packets(), number_of_deltas);
RTC_PARSE_CHECK_OR_RETURN_EQ(total_packets_values.size(), number_of_deltas);
@@ -2980,18 +2980,18 @@
}
// timestamp_ms
- std::vector<absl::optional<uint64_t>> timestamp_ms_values =
+ std::vector<std::optional<uint64_t>> timestamp_ms_values =
DecodeDeltas(proto.timestamp_ms_deltas(),
ToUnsigned(proto.timestamp_ms()), number_of_deltas);
RTC_PARSE_CHECK_OR_RETURN_EQ(timestamp_ms_values.size(), number_of_deltas);
// bitrate_bps
- std::vector<absl::optional<uint64_t>> bitrate_bps_values = DecodeDeltas(
+ std::vector<std::optional<uint64_t>> bitrate_bps_values = DecodeDeltas(
proto.bitrate_bps_deltas(), proto.bitrate_bps(), number_of_deltas);
RTC_PARSE_CHECK_OR_RETURN_EQ(bitrate_bps_values.size(), number_of_deltas);
// detector_state
- std::vector<absl::optional<uint64_t>> detector_state_values = DecodeDeltas(
+ std::vector<std::optional<uint64_t>> detector_state_values = DecodeDeltas(
proto.detector_state_deltas(),
static_cast<uint64_t>(proto.detector_state()), number_of_deltas);
RTC_PARSE_CHECK_OR_RETURN_EQ(detector_state_values.size(), number_of_deltas);
@@ -3103,40 +3103,40 @@
}
// timestamp_ms
- std::vector<absl::optional<uint64_t>> timestamp_ms_values =
+ std::vector<std::optional<uint64_t>> timestamp_ms_values =
DecodeDeltas(proto.timestamp_ms_deltas(),
ToUnsigned(proto.timestamp_ms()), number_of_deltas);
RTC_PARSE_CHECK_OR_RETURN_EQ(timestamp_ms_values.size(), number_of_deltas);
// SSRC
- std::vector<absl::optional<uint64_t>> ssrc_values =
+ std::vector<std::optional<uint64_t>> ssrc_values =
DecodeDeltas(proto.ssrc_deltas(), proto.ssrc(), number_of_deltas);
RTC_PARSE_CHECK_OR_RETURN_EQ(ssrc_values.size(), number_of_deltas);
// render_time_ms
- std::vector<absl::optional<uint64_t>> render_time_ms_values =
+ std::vector<std::optional<uint64_t>> render_time_ms_values =
DecodeDeltas(proto.render_time_ms_deltas(),
ToUnsigned(proto.render_time_ms()), number_of_deltas);
RTC_PARSE_CHECK_OR_RETURN_EQ(render_time_ms_values.size(), number_of_deltas);
// width
- std::vector<absl::optional<uint64_t>> width_values = DecodeDeltas(
+ std::vector<std::optional<uint64_t>> width_values = DecodeDeltas(
proto.width_deltas(), ToUnsigned(proto.width()), number_of_deltas);
RTC_PARSE_CHECK_OR_RETURN_EQ(width_values.size(), number_of_deltas);
// height
- std::vector<absl::optional<uint64_t>> height_values = DecodeDeltas(
+ std::vector<std::optional<uint64_t>> height_values = DecodeDeltas(
proto.height_deltas(), ToUnsigned(proto.height()), number_of_deltas);
RTC_PARSE_CHECK_OR_RETURN_EQ(height_values.size(), number_of_deltas);
// codec
- std::vector<absl::optional<uint64_t>> codec_values =
+ std::vector<std::optional<uint64_t>> codec_values =
DecodeDeltas(proto.codec_deltas(), static_cast<uint64_t>(proto.codec()),
number_of_deltas);
RTC_PARSE_CHECK_OR_RETURN_EQ(codec_values.size(), number_of_deltas);
// qp
- std::vector<absl::optional<uint64_t>> qp_values =
+ std::vector<std::optional<uint64_t>> qp_values =
DecodeDeltas(proto.qp_deltas(), proto.qp(), number_of_deltas);
RTC_PARSE_CHECK_OR_RETURN_EQ(qp_values.size(), number_of_deltas);
@@ -3187,7 +3187,7 @@
RTC_PARSE_CHECK_OR_RETURN(proto.has_acked_packet_number());
// receive_acked_packet_time_ms is optional.
- absl::optional<int64_t> base_receive_acked_packet_time_ms;
+ std::optional<int64_t> base_receive_acked_packet_time_ms;
if (proto.has_receive_acked_packet_time_ms()) {
base_receive_acked_packet_time_ms = proto.receive_acked_packet_time_ms();
}
@@ -3202,31 +3202,31 @@
}
// timestamp_ms
- std::vector<absl::optional<uint64_t>> timestamp_ms_values =
+ std::vector<std::optional<uint64_t>> timestamp_ms_values =
DecodeDeltas(proto.timestamp_ms_deltas(),
ToUnsigned(proto.timestamp_ms()), number_of_deltas);
RTC_PARSE_CHECK_OR_RETURN_EQ(timestamp_ms_values.size(), number_of_deltas);
// packet_number
- std::vector<absl::optional<uint64_t>> packet_number_values =
+ std::vector<std::optional<uint64_t>> packet_number_values =
DecodeDeltas(proto.packet_number_deltas(),
ToUnsigned(proto.packet_number()), number_of_deltas);
RTC_PARSE_CHECK_OR_RETURN_EQ(packet_number_values.size(), number_of_deltas);
// acked_packet_number
- std::vector<absl::optional<uint64_t>> acked_packet_number_values =
+ std::vector<std::optional<uint64_t>> acked_packet_number_values =
DecodeDeltas(proto.acked_packet_number_deltas(),
ToUnsigned(proto.acked_packet_number()), number_of_deltas);
RTC_PARSE_CHECK_OR_RETURN_EQ(acked_packet_number_values.size(),
number_of_deltas);
// optional receive_acked_packet_time_ms
- const absl::optional<uint64_t> unsigned_receive_acked_packet_time_ms_base =
+ const std::optional<uint64_t> unsigned_receive_acked_packet_time_ms_base =
proto.has_receive_acked_packet_time_ms()
- ? absl::optional<uint64_t>(
+ ? std::optional<uint64_t>(
ToUnsigned(proto.receive_acked_packet_time_ms()))
- : absl::optional<uint64_t>();
- std::vector<absl::optional<uint64_t>> receive_acked_packet_time_ms_values =
+ : std::optional<uint64_t>();
+ std::vector<std::optional<uint64_t>> receive_acked_packet_time_ms_values =
DecodeDeltas(proto.receive_acked_packet_time_ms_deltas(),
unsigned_receive_acked_packet_time_ms_base,
number_of_deltas);
@@ -3243,7 +3243,7 @@
int64_t acked_packet_number;
RTC_PARSE_CHECK_OR_RETURN(
ToSigned(acked_packet_number_values[i].value(), &acked_packet_number));
- absl::optional<int64_t> receive_acked_packet_time_ms;
+ std::optional<int64_t> receive_acked_packet_time_ms;
if (receive_acked_packet_time_ms_values[i].has_value()) {
int64_t value;
@@ -3281,27 +3281,27 @@
}
// timestamp_ms
- std::vector<absl::optional<uint64_t>> timestamp_ms_values =
+ std::vector<std::optional<uint64_t>> timestamp_ms_values =
DecodeDeltas(proto.timestamp_ms_deltas(),
ToUnsigned(proto.timestamp_ms()), number_of_deltas);
RTC_PARSE_CHECK_OR_RETURN_EQ(timestamp_ms_values.size(), number_of_deltas);
// packet_number
- std::vector<absl::optional<uint64_t>> packet_number_values =
+ std::vector<std::optional<uint64_t>> packet_number_values =
DecodeDeltas(proto.packet_number_deltas(),
ToUnsigned(proto.packet_number()), number_of_deltas);
RTC_PARSE_CHECK_OR_RETURN_EQ(packet_number_values.size(), number_of_deltas);
- std::vector<absl::optional<uint64_t>> overhead_length_values =
+ std::vector<std::optional<uint64_t>> overhead_length_values =
DecodeDeltas(proto.overhead_length_deltas(), proto.overhead_length(),
number_of_deltas);
RTC_PARSE_CHECK_OR_RETURN_EQ(overhead_length_values.size(), number_of_deltas);
- std::vector<absl::optional<uint64_t>> payload_length_values = DecodeDeltas(
+ std::vector<std::optional<uint64_t>> payload_length_values = DecodeDeltas(
proto.payload_length_deltas(), proto.payload_length(), number_of_deltas);
RTC_PARSE_CHECK_OR_RETURN_EQ(payload_length_values.size(), number_of_deltas);
- std::vector<absl::optional<uint64_t>> padding_length_values = DecodeDeltas(
+ std::vector<std::optional<uint64_t>> padding_length_values = DecodeDeltas(
proto.padding_length_deltas(), proto.padding_length(), number_of_deltas);
RTC_PARSE_CHECK_OR_RETURN_EQ(padding_length_values.size(), number_of_deltas);
@@ -3344,18 +3344,18 @@
}
// timestamp_ms
- std::vector<absl::optional<uint64_t>> timestamp_ms_values =
+ std::vector<std::optional<uint64_t>> timestamp_ms_values =
DecodeDeltas(proto.timestamp_ms_deltas(),
ToUnsigned(proto.timestamp_ms()), number_of_deltas);
RTC_PARSE_CHECK_OR_RETURN_EQ(timestamp_ms_values.size(), number_of_deltas);
// packet_number
- std::vector<absl::optional<uint64_t>> packet_number_values =
+ std::vector<std::optional<uint64_t>> packet_number_values =
DecodeDeltas(proto.packet_number_deltas(),
ToUnsigned(proto.packet_number()), number_of_deltas);
RTC_PARSE_CHECK_OR_RETURN_EQ(packet_number_values.size(), number_of_deltas);
- std::vector<absl::optional<uint64_t>> packet_length_values = DecodeDeltas(
+ std::vector<std::optional<uint64_t>> packet_length_values = DecodeDeltas(
proto.packet_length_deltas(), proto.packet_length(), number_of_deltas);
RTC_PARSE_CHECK_OR_RETURN_EQ(packet_length_values.size(), number_of_deltas);
@@ -3417,54 +3417,54 @@
}
// timestamp_ms
- std::vector<absl::optional<uint64_t>> timestamp_ms_values =
+ std::vector<std::optional<uint64_t>> timestamp_ms_values =
DecodeDeltas(proto.timestamp_ms_deltas(),
ToUnsigned(proto.timestamp_ms()), number_of_deltas);
RTC_PARSE_CHECK_OR_RETURN_EQ(timestamp_ms_values.size(), number_of_deltas);
// bitrate_bps
- const absl::optional<uint64_t> unsigned_base_bitrate_bps =
+ const std::optional<uint64_t> unsigned_base_bitrate_bps =
proto.has_bitrate_bps()
- ? absl::optional<uint64_t>(ToUnsigned(proto.bitrate_bps()))
- : absl::optional<uint64_t>();
- std::vector<absl::optional<uint64_t>> bitrate_bps_values = DecodeDeltas(
+ ? std::optional<uint64_t>(ToUnsigned(proto.bitrate_bps()))
+ : std::optional<uint64_t>();
+ std::vector<std::optional<uint64_t>> bitrate_bps_values = DecodeDeltas(
proto.bitrate_bps_deltas(), unsigned_base_bitrate_bps, number_of_deltas);
RTC_PARSE_CHECK_OR_RETURN_EQ(bitrate_bps_values.size(), number_of_deltas);
// frame_length_ms
- const absl::optional<uint64_t> unsigned_base_frame_length_ms =
+ const std::optional<uint64_t> unsigned_base_frame_length_ms =
proto.has_frame_length_ms()
- ? absl::optional<uint64_t>(ToUnsigned(proto.frame_length_ms()))
- : absl::optional<uint64_t>();
- std::vector<absl::optional<uint64_t>> frame_length_ms_values =
+ ? std::optional<uint64_t>(ToUnsigned(proto.frame_length_ms()))
+ : std::optional<uint64_t>();
+ std::vector<std::optional<uint64_t>> frame_length_ms_values =
DecodeDeltas(proto.frame_length_ms_deltas(),
unsigned_base_frame_length_ms, number_of_deltas);
RTC_PARSE_CHECK_OR_RETURN_EQ(frame_length_ms_values.size(), number_of_deltas);
// uplink_packet_loss_fraction
- const absl::optional<uint64_t> uplink_packet_loss_fraction =
+ const std::optional<uint64_t> uplink_packet_loss_fraction =
proto.has_uplink_packet_loss_fraction()
- ? absl::optional<uint64_t>(proto.uplink_packet_loss_fraction())
- : absl::optional<uint64_t>();
- std::vector<absl::optional<uint64_t>> uplink_packet_loss_fraction_values =
+ ? std::optional<uint64_t>(proto.uplink_packet_loss_fraction())
+ : std::optional<uint64_t>();
+ std::vector<std::optional<uint64_t>> uplink_packet_loss_fraction_values =
DecodeDeltas(proto.uplink_packet_loss_fraction_deltas(),
uplink_packet_loss_fraction, number_of_deltas);
RTC_PARSE_CHECK_OR_RETURN_EQ(uplink_packet_loss_fraction_values.size(),
number_of_deltas);
// enable_fec
- const absl::optional<uint64_t> enable_fec =
- proto.has_enable_fec() ? absl::optional<uint64_t>(proto.enable_fec())
- : absl::optional<uint64_t>();
- std::vector<absl::optional<uint64_t>> enable_fec_values =
+ const std::optional<uint64_t> enable_fec =
+ proto.has_enable_fec() ? std::optional<uint64_t>(proto.enable_fec())
+ : std::optional<uint64_t>();
+ std::vector<std::optional<uint64_t>> enable_fec_values =
DecodeDeltas(proto.enable_fec_deltas(), enable_fec, number_of_deltas);
RTC_PARSE_CHECK_OR_RETURN_EQ(enable_fec_values.size(), number_of_deltas);
// enable_dtx
- const absl::optional<uint64_t> enable_dtx =
- proto.has_enable_dtx() ? absl::optional<uint64_t>(proto.enable_dtx())
- : absl::optional<uint64_t>();
- std::vector<absl::optional<uint64_t>> enable_dtx_values =
+ const std::optional<uint64_t> enable_dtx =
+ proto.has_enable_dtx() ? std::optional<uint64_t>(proto.enable_dtx())
+ : std::optional<uint64_t>();
+ std::vector<std::optional<uint64_t>> enable_dtx_values =
DecodeDeltas(proto.enable_dtx_deltas(), enable_dtx, number_of_deltas);
RTC_PARSE_CHECK_OR_RETURN_EQ(enable_dtx_values.size(), number_of_deltas);
@@ -3474,12 +3474,12 @@
// We likewise shift the base event down by one, to get the same base as
// encoding had, but then shift all of the values (except the base) back up
// to their original value.
- absl::optional<uint64_t> shifted_base_num_channels;
+ std::optional<uint64_t> shifted_base_num_channels;
if (proto.has_num_channels()) {
shifted_base_num_channels =
- absl::optional<uint64_t>(proto.num_channels() - 1);
+ std::optional<uint64_t>(proto.num_channels() - 1);
}
- std::vector<absl::optional<uint64_t>> num_channels_values = DecodeDeltas(
+ std::vector<std::optional<uint64_t>> num_channels_values = DecodeDeltas(
proto.num_channels_deltas(), shifted_base_num_channels, number_of_deltas);
for (size_t i = 0; i < num_channels_values.size(); ++i) {
if (num_channels_values[i].has_value()) {
diff --git a/logging/rtc_event_log/rtc_event_log_unittest_helper.cc b/logging/rtc_event_log/rtc_event_log_unittest_helper.cc
index 337f7ee..06baf5d 100644
--- a/logging/rtc_event_log/rtc_event_log_unittest_helper.cc
+++ b/logging/rtc_event_log/rtc_event_log_unittest_helper.cc
@@ -17,12 +17,12 @@
#include <limits>
#include <memory>
#include <numeric>
+#include <optional>
#include <utility>
#include <vector>
#include "absl/algorithm/container.h"
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/candidate.h"
#include "api/dtls_transport_interface.h"
@@ -136,13 +136,13 @@
}
}
-absl::optional<int> GetExtensionId(const std::vector<RtpExtension>& extensions,
- absl::string_view uri) {
+std::optional<int> GetExtensionId(const std::vector<RtpExtension>& extensions,
+ absl::string_view uri) {
for (const auto& extension : extensions) {
if (extension.uri == uri)
return extension.id;
}
- return absl::nullopt;
+ return std::nullopt;
}
} // namespace
@@ -629,7 +629,7 @@
}
std::unique_ptr<RtcEventGenericAckReceived>
EventGenerator::NewGenericAckReceived() {
- absl::optional<int64_t> receive_timestamp = absl::nullopt;
+ std::optional<int64_t> receive_timestamp = std::nullopt;
if (prng_.Rand(0, 2) > 0) {
receive_timestamp = prng_.Rand(0, 100000);
}
diff --git a/logging/rtc_event_log/rtc_event_processor.cc b/logging/rtc_event_log/rtc_event_processor.cc
index 433b487..07671cc 100644
--- a/logging/rtc_event_log/rtc_event_processor.cc
+++ b/logging/rtc_event_log/rtc_event_processor.cc
@@ -11,8 +11,8 @@
#include <algorithm>
#include <cstdint>
+#include <optional>
-#include "absl/types/optional.h"
#include "rtc_base/numerics/sequence_number_util.h"
namespace webrtc {
@@ -47,8 +47,8 @@
if (a->GetTypeOrder() != b->GetTypeOrder())
return a->GetTypeOrder() > b->GetTypeOrder();
- absl::optional<uint16_t> wrapped_seq_num_a = a->GetTransportSeqNum();
- absl::optional<uint16_t> wrapped_seq_num_b = b->GetTransportSeqNum();
+ std::optional<uint16_t> wrapped_seq_num_a = a->GetTransportSeqNum();
+ std::optional<uint16_t> wrapped_seq_num_b = b->GetTransportSeqNum();
if (wrapped_seq_num_a && wrapped_seq_num_b) {
return AheadOf<uint16_t>(*wrapped_seq_num_a, *wrapped_seq_num_b);
} else if (wrapped_seq_num_a.has_value() != wrapped_seq_num_b.has_value()) {
diff --git a/logging/rtc_event_log/rtc_event_processor.h b/logging/rtc_event_log/rtc_event_processor.h
index 0dac1c5..846fbb8 100644
--- a/logging/rtc_event_log/rtc_event_processor.h
+++ b/logging/rtc_event_log/rtc_event_processor.h
@@ -15,10 +15,10 @@
#include <functional>
#include <memory>
+#include <optional>
#include <type_traits>
#include <vector>
-#include "absl/types/optional.h"
#include "logging/rtc_event_log/rtc_event_log_parser.h"
#include "logging/rtc_event_log/rtc_event_processor_order.h"
#include "rtc_base/checks.h"
@@ -42,7 +42,7 @@
virtual bool IsEmpty() const = 0;
virtual int64_t GetNextTime() const = 0;
virtual int GetTypeOrder() const = 0;
- virtual absl::optional<uint16_t> GetTransportSeqNum() const = 0;
+ virtual std::optional<uint16_t> GetTransportSeqNum() const = 0;
virtual int GetInsertionOrder() const = 0;
};
@@ -55,7 +55,7 @@
Iterator end,
std::function<void(const T&)> f,
int type_order,
- std::function<absl::optional<uint16_t>(const T&)>
+ std::function<std::optional<uint16_t>(const T&)>
transport_seq_num_accessor,
int insertion_order)
: begin_(begin),
@@ -80,7 +80,7 @@
int GetTypeOrder() const override { return type_order_; }
- absl::optional<uint16_t> GetTransportSeqNum() const override {
+ std::optional<uint16_t> GetTransportSeqNum() const override {
RTC_DCHECK(!IsEmpty());
return transport_seq_num_accessor_(*begin_);
}
@@ -92,7 +92,7 @@
Iterator end_;
std::function<void(const T&)> f_;
int type_order_;
- std::function<absl::optional<uint16_t>(const T&)> transport_seq_num_accessor_;
+ std::function<std::optional<uint16_t>(const T&)> transport_seq_num_accessor_;
int insertion_order_;
};
@@ -152,7 +152,7 @@
const Iterable& iterable,
std::function<void(const typename Iterable::value_type&)> handler,
int type_order,
- std::function<absl::optional<uint16_t>(
+ std::function<std::optional<uint16_t>(
const typename Iterable::value_type&)> transport_seq_num_accessor,
int insertion_order) {
if (iterable.begin() == iterable.end())
diff --git a/logging/rtc_event_log/rtc_event_processor_order.h b/logging/rtc_event_log/rtc_event_processor_order.h
index 5ee1244..262bd10 100644
--- a/logging/rtc_event_log/rtc_event_processor_order.h
+++ b/logging/rtc_event_log/rtc_event_processor_order.h
@@ -13,7 +13,8 @@
#include <stdint.h>
-#include "absl/types/optional.h"
+#include <optional>
+
#include "logging/rtc_event_log/events/logged_rtp_rtcp.h"
#include "logging/rtc_event_log/events/rtc_event_alr_state.h"
#include "logging/rtc_event_log/events/rtc_event_audio_network_adaptation.h"
@@ -101,9 +102,9 @@
class TieBreaker<LoggedStartEvent> {
public:
static constexpr int type_order = static_cast<int>(TypeOrder::Start);
- static absl::optional<uint16_t> transport_seq_num_accessor(
+ static std::optional<uint16_t> transport_seq_num_accessor(
const LoggedStartEvent&) {
- return absl::optional<uint16_t>();
+ return std::optional<uint16_t>();
}
};
@@ -111,9 +112,9 @@
class TieBreaker<LoggedStopEvent> {
public:
static constexpr int type_order = static_cast<int>(TypeOrder::Stop);
- static absl::optional<uint16_t> transport_seq_num_accessor(
+ static std::optional<uint16_t> transport_seq_num_accessor(
const LoggedStopEvent&) {
- return absl::optional<uint16_t>();
+ return std::optional<uint16_t>();
}
};
@@ -121,9 +122,9 @@
class TieBreaker<LoggedAudioRecvConfig> {
public:
static constexpr int type_order = static_cast<int>(TypeOrder::StreamConfig);
- static absl::optional<uint16_t> transport_seq_num_accessor(
+ static std::optional<uint16_t> transport_seq_num_accessor(
const LoggedAudioRecvConfig&) {
- return absl::optional<uint16_t>();
+ return std::optional<uint16_t>();
}
};
@@ -131,9 +132,9 @@
class TieBreaker<LoggedAudioSendConfig> {
public:
static constexpr int type_order = static_cast<int>(TypeOrder::StreamConfig);
- static absl::optional<uint16_t> transport_seq_num_accessor(
+ static std::optional<uint16_t> transport_seq_num_accessor(
const LoggedAudioSendConfig&) {
- return absl::optional<uint16_t>();
+ return std::optional<uint16_t>();
}
};
@@ -141,9 +142,9 @@
class TieBreaker<LoggedVideoRecvConfig> {
public:
static constexpr int type_order = static_cast<int>(TypeOrder::StreamConfig);
- static absl::optional<uint16_t> transport_seq_num_accessor(
+ static std::optional<uint16_t> transport_seq_num_accessor(
const LoggedVideoRecvConfig&) {
- return absl::optional<uint16_t>();
+ return std::optional<uint16_t>();
}
};
@@ -151,9 +152,9 @@
class TieBreaker<LoggedVideoSendConfig> {
public:
static constexpr int type_order = static_cast<int>(TypeOrder::StreamConfig);
- static absl::optional<uint16_t> transport_seq_num_accessor(
+ static std::optional<uint16_t> transport_seq_num_accessor(
const LoggedVideoSendConfig&) {
- return absl::optional<uint16_t>();
+ return std::optional<uint16_t>();
}
};
@@ -162,9 +163,9 @@
public:
static constexpr int type_order =
static_cast<int>(TypeOrder::IceCondidateConfig);
- static absl::optional<uint16_t> transport_seq_num_accessor(
+ static std::optional<uint16_t> transport_seq_num_accessor(
const LoggedIceCandidatePairConfig&) {
- return absl::optional<uint16_t>();
+ return std::optional<uint16_t>();
}
};
@@ -173,9 +174,9 @@
public:
static constexpr int type_order =
static_cast<int>(TypeOrder::IceCandidateEvent);
- static absl::optional<uint16_t> transport_seq_num_accessor(
+ static std::optional<uint16_t> transport_seq_num_accessor(
const LoggedIceCandidatePairEvent&) {
- return absl::optional<uint16_t>();
+ return std::optional<uint16_t>();
}
};
@@ -184,9 +185,9 @@
public:
static constexpr int type_order =
static_cast<int>(TypeOrder::DtlsTransportState);
- static absl::optional<uint16_t> transport_seq_num_accessor(
+ static std::optional<uint16_t> transport_seq_num_accessor(
const LoggedDtlsTransportState&) {
- return absl::optional<uint16_t>();
+ return std::optional<uint16_t>();
}
};
@@ -194,9 +195,9 @@
class TieBreaker<LoggedDtlsWritableState> {
public:
static constexpr int type_order = static_cast<int>(TypeOrder::DtlsWritable);
- static absl::optional<uint16_t> transport_seq_num_accessor(
+ static std::optional<uint16_t> transport_seq_num_accessor(
const LoggedDtlsWritableState&) {
- return absl::optional<uint16_t>();
+ return std::optional<uint16_t>();
}
};
@@ -204,9 +205,9 @@
class TieBreaker<LoggedRouteChangeEvent> {
public:
static constexpr int type_order = static_cast<int>(TypeOrder::RouteChange);
- static absl::optional<uint16_t> transport_seq_num_accessor(
+ static std::optional<uint16_t> transport_seq_num_accessor(
const LoggedRouteChangeEvent&) {
- return absl::optional<uint16_t>();
+ return std::optional<uint16_t>();
}
};
@@ -215,9 +216,9 @@
public:
static constexpr int type_order =
static_cast<int>(TypeOrder::BweRemoteEstimate);
- static absl::optional<uint16_t> transport_seq_num_accessor(
+ static std::optional<uint16_t> transport_seq_num_accessor(
const LoggedRemoteEstimateEvent&) {
- return absl::optional<uint16_t>();
+ return std::optional<uint16_t>();
}
};
@@ -226,9 +227,9 @@
public:
static constexpr int type_order =
static_cast<int>(TypeOrder::BweProbeFailure);
- static absl::optional<uint16_t> transport_seq_num_accessor(
+ static std::optional<uint16_t> transport_seq_num_accessor(
const LoggedBweProbeFailureEvent&) {
- return absl::optional<uint16_t>();
+ return std::optional<uint16_t>();
}
};
@@ -237,9 +238,9 @@
public:
static constexpr int type_order =
static_cast<int>(TypeOrder::BweProbeSuccess);
- static absl::optional<uint16_t> transport_seq_num_accessor(
+ static std::optional<uint16_t> transport_seq_num_accessor(
const LoggedBweProbeSuccessEvent&) {
- return absl::optional<uint16_t>();
+ return std::optional<uint16_t>();
}
};
@@ -247,9 +248,9 @@
class TieBreaker<LoggedBweDelayBasedUpdate> {
public:
static constexpr int type_order = static_cast<int>(TypeOrder::BweDelayBased);
- static absl::optional<uint16_t> transport_seq_num_accessor(
+ static std::optional<uint16_t> transport_seq_num_accessor(
const LoggedBweDelayBasedUpdate&) {
- return absl::optional<uint16_t>();
+ return std::optional<uint16_t>();
}
};
@@ -257,9 +258,9 @@
class TieBreaker<LoggedBweLossBasedUpdate> {
public:
static constexpr int type_order = static_cast<int>(TypeOrder::BweLossBased);
- static absl::optional<uint16_t> transport_seq_num_accessor(
+ static std::optional<uint16_t> transport_seq_num_accessor(
const LoggedBweLossBasedUpdate&) {
- return absl::optional<uint16_t>();
+ return std::optional<uint16_t>();
}
};
@@ -268,9 +269,9 @@
public:
static constexpr int type_order =
static_cast<int>(TypeOrder::BweProbeCreated);
- static absl::optional<uint16_t> transport_seq_num_accessor(
+ static std::optional<uint16_t> transport_seq_num_accessor(
const LoggedBweProbeClusterCreatedEvent&) {
- return absl::optional<uint16_t>();
+ return std::optional<uint16_t>();
}
};
@@ -279,9 +280,9 @@
public:
static constexpr int type_order =
static_cast<int>(TypeOrder::AudioNetworkAdaptation);
- static absl::optional<uint16_t> transport_seq_num_accessor(
+ static std::optional<uint16_t> transport_seq_num_accessor(
const LoggedAudioNetworkAdaptationEvent&) {
- return absl::optional<uint16_t>();
+ return std::optional<uint16_t>();
}
};
@@ -290,9 +291,9 @@
public:
static constexpr int type_order =
static_cast<int>(TypeOrder::NetEqSetMinDelay);
- static absl::optional<uint16_t> transport_seq_num_accessor(
+ static std::optional<uint16_t> transport_seq_num_accessor(
const LoggedNetEqSetMinimumDelayEvent&) {
- return absl::optional<uint16_t>();
+ return std::optional<uint16_t>();
}
};
@@ -300,9 +301,9 @@
class TieBreaker<LoggedAudioPlayoutEvent> {
public:
static constexpr int type_order = static_cast<int>(TypeOrder::AudioPlayout);
- static absl::optional<uint16_t> transport_seq_num_accessor(
+ static std::optional<uint16_t> transport_seq_num_accessor(
const LoggedAudioPlayoutEvent&) {
- return absl::optional<uint16_t>();
+ return std::optional<uint16_t>();
}
};
@@ -310,9 +311,9 @@
class TieBreaker<LoggedFrameDecoded> {
public:
static constexpr int type_order = static_cast<int>(TypeOrder::FrameDecoded);
- static absl::optional<uint16_t> transport_seq_num_accessor(
+ static std::optional<uint16_t> transport_seq_num_accessor(
const LoggedFrameDecoded&) {
- return absl::optional<uint16_t>();
+ return std::optional<uint16_t>();
}
};
@@ -321,9 +322,9 @@
public:
static constexpr int type_order =
static_cast<int>(TypeOrder::GenericPacketIn);
- static absl::optional<uint16_t> transport_seq_num_accessor(
+ static std::optional<uint16_t> transport_seq_num_accessor(
const LoggedGenericPacketReceived&) {
- return absl::optional<uint16_t>();
+ return std::optional<uint16_t>();
}
};
@@ -331,9 +332,9 @@
class TieBreaker<LoggedGenericAckReceived> {
public:
static constexpr int type_order = static_cast<int>(TypeOrder::GenericAckIn);
- static absl::optional<uint16_t> transport_seq_num_accessor(
+ static std::optional<uint16_t> transport_seq_num_accessor(
const LoggedGenericAckReceived&) {
- return absl::optional<uint16_t>();
+ return std::optional<uint16_t>();
}
};
@@ -342,9 +343,9 @@
public:
static constexpr int type_order =
static_cast<int>(TypeOrder::GenericPacketOut);
- static absl::optional<uint16_t> transport_seq_num_accessor(
+ static std::optional<uint16_t> transport_seq_num_accessor(
const LoggedGenericPacketSent&) {
- return absl::optional<uint16_t>();
+ return std::optional<uint16_t>();
}
};
@@ -356,11 +357,11 @@
? TypeOrder::RtpIn
: TypeOrder::RtpOut);
}
- static absl::optional<uint16_t> transport_seq_num_accessor(
+ static std::optional<uint16_t> transport_seq_num_accessor(
const LoggedRtpPacket& p) {
return p.header.extension.hasTransportSequenceNumber
? p.header.extension.transportSequenceNumber
- : absl::optional<uint16_t>();
+ : std::optional<uint16_t>();
}
};
@@ -372,10 +373,10 @@
? TypeOrder::RtpIn
: TypeOrder::RtpOut);
}
- static absl::optional<uint16_t> transport_seq_num_accessor(
+ static std::optional<uint16_t> transport_seq_num_accessor(
const LoggedPacketInfo& p) {
return p.has_transport_seq_no ? p.transport_seq_no
- : absl::optional<uint16_t>();
+ : std::optional<uint16_t>();
}
};
@@ -383,11 +384,11 @@
class TieBreaker<LoggedRtpPacketIncoming> {
public:
static constexpr int type_order = static_cast<int>(TypeOrder::RtpIn);
- static absl::optional<uint16_t> transport_seq_num_accessor(
+ static std::optional<uint16_t> transport_seq_num_accessor(
const LoggedRtpPacketIncoming& p) {
return p.rtp.header.extension.hasTransportSequenceNumber
? p.rtp.header.extension.transportSequenceNumber
- : absl::optional<uint16_t>();
+ : std::optional<uint16_t>();
}
};
@@ -395,11 +396,11 @@
class TieBreaker<LoggedRtpPacketOutgoing> {
public:
static constexpr int type_order = static_cast<int>(TypeOrder::RtpOut);
- static absl::optional<uint16_t> transport_seq_num_accessor(
+ static std::optional<uint16_t> transport_seq_num_accessor(
const LoggedRtpPacketOutgoing& p) {
return p.rtp.header.extension.hasTransportSequenceNumber
? p.rtp.header.extension.transportSequenceNumber
- : absl::optional<uint16_t>();
+ : std::optional<uint16_t>();
}
};
@@ -407,9 +408,9 @@
class TieBreaker<LoggedRtcpPacketIncoming> {
public:
static constexpr int type_order = static_cast<int>(TypeOrder::RtcpIn);
- static absl::optional<uint16_t> transport_seq_num_accessor(
+ static std::optional<uint16_t> transport_seq_num_accessor(
const LoggedRtcpPacketIncoming&) {
- return absl::optional<uint16_t>();
+ return std::optional<uint16_t>();
}
};
@@ -417,9 +418,9 @@
class TieBreaker<LoggedRtcpPacketOutgoing> {
public:
static constexpr int type_order = static_cast<int>(TypeOrder::RtcpOut);
- static absl::optional<uint16_t> transport_seq_num_accessor(
+ static std::optional<uint16_t> transport_seq_num_accessor(
const LoggedRtcpPacketOutgoing&) {
- return absl::optional<uint16_t>();
+ return std::optional<uint16_t>();
}
};
@@ -431,9 +432,9 @@
? TypeOrder::RtcpIn
: TypeOrder::RtcpOut);
}
- static absl::optional<uint16_t> transport_seq_num_accessor(
+ static std::optional<uint16_t> transport_seq_num_accessor(
const LoggedRtcpPacketTransportFeedback&) {
- return absl::optional<uint16_t>();
+ return std::optional<uint16_t>();
}
};
@@ -445,9 +446,9 @@
? TypeOrder::RtcpIn
: TypeOrder::RtcpOut);
}
- static absl::optional<uint16_t> transport_seq_num_accessor(
+ static std::optional<uint16_t> transport_seq_num_accessor(
const LoggedRtcpPacketReceiverReport&) {
- return absl::optional<uint16_t>();
+ return std::optional<uint16_t>();
}
};
@@ -455,9 +456,9 @@
class TieBreaker<LoggedAlrStateEvent> {
public:
static constexpr int type_order = static_cast<int>(TypeOrder::AlrState);
- static absl::optional<uint16_t> transport_seq_num_accessor(
+ static std::optional<uint16_t> transport_seq_num_accessor(
const LoggedAlrStateEvent&) {
- return absl::optional<uint16_t>();
+ return std::optional<uint16_t>();
}
};
diff --git a/logging/rtc_event_log/rtc_event_processor_unittest.cc b/logging/rtc_event_log/rtc_event_processor_unittest.cc
index 53948ad..2427f15 100644
--- a/logging/rtc_event_log/rtc_event_processor_unittest.cc
+++ b/logging/rtc_event_log/rtc_event_processor_unittest.cc
@@ -16,10 +16,10 @@
#include <initializer_list>
#include <limits>
#include <numeric>
+#include <optional>
#include <utility>
#include <vector>
-#include "absl/types/optional.h"
#include "api/rtp_headers.h"
#include "api/units/timestamp.h"
#include "logging/rtc_event_log/events/logged_rtp_rtcp.h"
@@ -69,7 +69,7 @@
LoggedRtpPacket CreateRtpPacket(int64_t time_ms,
uint32_t ssrc,
- absl::optional<uint16_t> transport_seq_num) {
+ std::optional<uint16_t> transport_seq_num) {
RTPHeader header;
header.ssrc = ssrc;
header.timestamp = static_cast<uint32_t>(time_ms);
@@ -237,8 +237,8 @@
TEST(RtcEventProcessor, RtpPacketsInTransportSeqNumOrder) {
std::vector<LoggedRtpPacket> ssrc_1234{
- CreateRtpPacket(1, 1234, absl::nullopt),
- CreateRtpPacket(1, 1234, absl::nullopt)};
+ CreateRtpPacket(1, 1234, std::nullopt),
+ CreateRtpPacket(1, 1234, std::nullopt)};
std::vector<LoggedRtpPacket> ssrc_2345{CreateRtpPacket(1, 2345, 2),
CreateRtpPacket(1, 2345, 3),
CreateRtpPacket(1, 2345, 6)};
@@ -247,9 +247,9 @@
CreateRtpPacket(1, 3456, 5)};
// Store SSRC and transport sequence number for each processed packet.
- std::vector<std::pair<uint32_t, absl::optional<uint16_t>>> results;
+ std::vector<std::pair<uint32_t, std::optional<uint16_t>>> results;
auto get_packet = [&results](const LoggedRtpPacket& packet) {
- absl::optional<uint16_t> transport_seq_num;
+ std::optional<uint16_t> transport_seq_num;
if (packet.header.extension.hasTransportSequenceNumber)
transport_seq_num = packet.header.extension.transportSequenceNumber;
results.emplace_back(packet.header.ssrc, transport_seq_num);
@@ -261,9 +261,9 @@
processor.AddEvents(ssrc_3456, get_packet, PacketDirection::kIncomingPacket);
processor.ProcessEventsInOrder();
- std::vector<std::pair<uint32_t, absl::optional<uint16_t>>> expected{
- {1234, absl::nullopt},
- {1234, absl::nullopt},
+ std::vector<std::pair<uint32_t, std::optional<uint16_t>>> expected{
+ {1234, std::nullopt},
+ {1234, std::nullopt},
{3456, 1},
{2345, 2},
{2345, 3},
@@ -282,9 +282,9 @@
CreateRtpPacket(1, 2345, 0), CreateRtpPacket(1, 2345, 3)};
// Store SSRC and transport sequence number for each processed packet.
- std::vector<std::pair<uint32_t, absl::optional<uint16_t>>> results;
+ std::vector<std::pair<uint32_t, std::optional<uint16_t>>> results;
auto get_packet = [&results](const LoggedRtpPacket& packet) {
- absl::optional<uint16_t> transport_seq_num;
+ std::optional<uint16_t> transport_seq_num;
if (packet.header.extension.hasTransportSequenceNumber)
transport_seq_num = packet.header.extension.transportSequenceNumber;
results.emplace_back(packet.header.ssrc, transport_seq_num);
@@ -295,7 +295,7 @@
processor.AddEvents(ssrc_2345, get_packet, PacketDirection::kOutgoingPacket);
processor.ProcessEventsInOrder();
- std::vector<std::pair<uint32_t, absl::optional<uint16_t>>> expected{
+ std::vector<std::pair<uint32_t, std::optional<uint16_t>>> expected{
{1234, std::numeric_limits<uint16_t>::max() - 1},
{2345, std::numeric_limits<uint16_t>::max()},
{2345, 0},
diff --git a/media/BUILD.gn b/media/BUILD.gn
index 79662b0..ddf33dc 100644
--- a/media/BUILD.gn
+++ b/media/BUILD.gn
@@ -42,7 +42,6 @@
"../rtc_base:checks",
"../rtc_base:stringutils",
"//third_party/abseil-cpp/absl/algorithm:container",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -126,7 +125,6 @@
"../rtc_base/system:rtc_export",
"../rtc_base/third_party/sigslot",
"../video/config:encoder_config",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -147,13 +145,11 @@
"../rtc_base:timeutils",
"../rtc_base/synchronization:mutex",
"../rtc_base/system:rtc_export",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
rtc_source_set("audio_source") {
sources = [ "base/audio_source.h" ]
- deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
}
rtc_library("video_adapter") {
@@ -173,7 +169,6 @@
"../rtc_base/synchronization:mutex",
"../rtc_base/system:rtc_export",
"../system_wrappers:field_trial",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -208,7 +203,6 @@
"../rtc_base:logging",
"../rtc_base:macromagic",
"../rtc_base/synchronization:mutex",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -297,7 +291,6 @@
"../rtc_base/network:sent_packet",
"//third_party/abseil-cpp/absl/functional:any_invocable",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -345,7 +338,6 @@
"../rtc_base:stringutils",
"../rtc_base/network:sent_packet",
"../video/config:encoder_config",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -368,7 +360,6 @@
"//third_party/abseil-cpp/absl/container:inlined_vector",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -475,7 +466,6 @@
"../system_wrappers",
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/base:nullability",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -511,7 +501,6 @@
"../system_wrappers:field_trial",
"../test:fake_video_codecs",
"//third_party/abseil-cpp/absl/strings",
- "//third_party/abseil-cpp/absl/types:optional",
]
if (enable_libaom) {
@@ -652,7 +641,6 @@
"//third_party/abseil-cpp/absl/functional:bind_front",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
sources = [
@@ -733,7 +721,6 @@
"../rtc_base/third_party/sigslot:sigslot",
"../system_wrappers",
"//third_party/abseil-cpp/absl/strings:strings",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
}
@@ -836,7 +823,6 @@
"//third_party/abseil-cpp/absl/functional:any_invocable",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
sources = [
"base/fake_frame_source.cc",
@@ -983,7 +969,6 @@
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
if (enable_libaom) {
diff --git a/media/base/adapted_video_track_source.h b/media/base/adapted_video_track_source.h
index 1c3e0b6..670331d 100644
--- a/media/base/adapted_video_track_source.h
+++ b/media/base/adapted_video_track_source.h
@@ -13,7 +13,8 @@
#include <stdint.h>
-#include "absl/types/optional.h"
+#include <optional>
+
#include "api/media_stream_interface.h"
#include "api/notifier.h"
#include "api/video/video_frame.h"
@@ -94,7 +95,7 @@
cricket::VideoAdapter video_adapter_;
webrtc::Mutex stats_mutex_;
- absl::optional<Stats> stats_ RTC_GUARDED_BY(stats_mutex_);
+ std::optional<Stats> stats_ RTC_GUARDED_BY(stats_mutex_);
VideoBroadcaster broadcaster_;
};
diff --git a/media/base/audio_source.h b/media/base/audio_source.h
index 51fe0e1..3cc0bc4 100644
--- a/media/base/audio_source.h
+++ b/media/base/audio_source.h
@@ -12,8 +12,7 @@
#define MEDIA_BASE_AUDIO_SOURCE_H_
#include <cstddef>
-
-#include "absl/types/optional.h"
+#include <optional>
namespace cricket {
@@ -31,7 +30,7 @@
int sample_rate,
size_t number_of_channels,
size_t number_of_frames,
- absl::optional<int64_t> absolute_capture_timestamp_ms) = 0;
+ std::optional<int64_t> absolute_capture_timestamp_ms) = 0;
// Called when the AudioSource is going away.
virtual void OnClose() = 0;
diff --git a/media/base/codec.cc b/media/base/codec.cc
index 820cb97..e994f0e 100644
--- a/media/base/codec.cc
+++ b/media/base/codec.cc
@@ -13,12 +13,12 @@
#include <algorithm>
#include <cstddef>
#include <iterator>
+#include <optional>
#include <string>
#include <vector>
#include "absl/algorithm/container.h"
#include "absl/strings/match.h"
-#include "absl/types/optional.h"
#include "api/audio_codecs/audio_format.h"
#include "api/media_types.h"
#include "api/rtp_parameters.h"
@@ -498,7 +498,7 @@
for (auto it = supported_formats->cbegin(); it != supported_formats->cend();
++it) {
if (it->name == cricket::kH264CodecName) {
- const absl::optional<webrtc::H264ProfileLevelId> profile_level_id =
+ const std::optional<webrtc::H264ProfileLevelId> profile_level_id =
webrtc::ParseSdpForH264ProfileLevelId(it->parameters);
if (profile_level_id &&
profile_level_id->profile !=
diff --git a/media/base/codec.h b/media/base/codec.h
index 5230779..583a14e 100644
--- a/media/base/codec.h
+++ b/media/base/codec.h
@@ -12,12 +12,12 @@
#define MEDIA_BASE_CODEC_H_
#include <cstddef>
+#include <optional>
#include <string>
#include <vector>
#include "absl/container/inlined_vector.h"
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/audio_codecs/audio_format.h"
#include "api/rtp_parameters.h"
#include "api/video_codecs/scalability_mode.h"
@@ -92,12 +92,12 @@
size_t channels;
// Video only
- absl::optional<std::string> packetization;
+ std::optional<std::string> packetization;
absl::InlinedVector<webrtc::ScalabilityMode, webrtc::kScalabilityModeCount>
scalability_modes;
// H.265 only
- absl::optional<std::string> tx_mode;
+ std::optional<std::string> tx_mode;
// Non key-value parameters such as the telephone-event "0‐15" are
// represented using an empty string as key, i.e. {"": "0-15"}.
diff --git a/media/base/codec_unittest.cc b/media/base/codec_unittest.cc
index 08b37fb..62fbd76 100644
--- a/media/base/codec_unittest.cc
+++ b/media/base/codec_unittest.cc
@@ -555,7 +555,7 @@
EXPECT_EQ(cricket::MEDIA_TYPE_VIDEO, codec_params_1.kind);
EXPECT_EQ("V", codec_params_1.name);
EXPECT_EQ(cricket::kVideoCodecClockrate, codec_params_1.clock_rate);
- EXPECT_EQ(absl::nullopt, codec_params_1.num_channels);
+ EXPECT_EQ(std::nullopt, codec_params_1.num_channels);
ASSERT_EQ(1u, codec_params_1.parameters.size());
EXPECT_EQ("p1", codec_params_1.parameters.begin()->first);
EXPECT_EQ("v1", codec_params_1.parameters.begin()->second);
diff --git a/media/base/fake_media_engine.cc b/media/base/fake_media_engine.cc
index d5c0e02..6cdf64b 100644
--- a/media/base/fake_media_engine.cc
+++ b/media/base/fake_media_engine.cc
@@ -11,11 +11,11 @@
#include "media/base/fake_media_engine.h"
#include <memory>
+#include <optional>
#include <utility>
#include "absl/algorithm/container.h"
#include "absl/strings/match.h"
-#include "absl/types/optional.h"
#include "media/base/media_channel.h"
#include "rtc_base/checks.h"
@@ -43,7 +43,7 @@
int sample_rate,
size_t number_of_channels,
size_t number_of_frames,
- absl::optional<int64_t> absolute_capture_timestamp_ms) {}
+ std::optional<int64_t> absolute_capture_timestamp_ms) {}
void FakeVoiceMediaReceiveChannel::VoiceChannelAudioSink::OnClose() {
source_ = nullptr;
}
@@ -133,13 +133,13 @@
return true;
}
}
-absl::optional<int> FakeVoiceMediaReceiveChannel::GetBaseMinimumPlayoutDelayMs(
+std::optional<int> FakeVoiceMediaReceiveChannel::GetBaseMinimumPlayoutDelayMs(
uint32_t ssrc) const {
const auto it = output_delays_.find(ssrc);
if (it != output_delays_.end()) {
return it->second;
}
- return absl::nullopt;
+ return std::nullopt;
}
bool FakeVoiceMediaReceiveChannel::GetStats(VoiceMediaReceiveInfo* info,
bool get_and_clear_legacy_stats) {
@@ -198,7 +198,7 @@
int sample_rate,
size_t number_of_channels,
size_t number_of_frames,
- absl::optional<int64_t> absolute_capture_timestamp_ms) {}
+ std::optional<int64_t> absolute_capture_timestamp_ms) {}
void FakeVoiceMediaSendChannel::VoiceChannelAudioSink::OnClose() {
source_ = nullptr;
}
@@ -218,11 +218,11 @@
const std::vector<Codec>& FakeVoiceMediaSendChannel::send_codecs() const {
return send_codecs_;
}
-absl::optional<Codec> FakeVoiceMediaSendChannel::GetSendCodec() const {
+std::optional<Codec> FakeVoiceMediaSendChannel::GetSendCodec() const {
if (!send_codecs_.empty()) {
return send_codecs_.front();
}
- return absl::nullopt;
+ return std::nullopt;
}
const std::vector<FakeVoiceMediaSendChannel::DtmfInfo>&
FakeVoiceMediaSendChannel::dtmf_info_queue() const {
@@ -362,9 +362,9 @@
SetSendRtpHeaderExtensions(params.extensions) &&
SetMaxSendBandwidth(params.max_bandwidth_bps));
}
-absl::optional<Codec> FakeVideoMediaSendChannel::GetSendCodec() const {
+std::optional<Codec> FakeVideoMediaSendChannel::GetSendCodec() const {
if (send_codecs_.empty()) {
- return absl::nullopt;
+ return std::nullopt;
}
return send_codecs_[0];
}
@@ -492,13 +492,13 @@
return true;
}
}
-absl::optional<int> FakeVideoMediaReceiveChannel::GetBaseMinimumPlayoutDelayMs(
+std::optional<int> FakeVideoMediaReceiveChannel::GetBaseMinimumPlayoutDelayMs(
uint32_t ssrc) const {
const auto it = output_delays_.find(ssrc);
if (it != output_delays_.end()) {
return it->second;
}
- return absl::nullopt;
+ return std::nullopt;
}
bool FakeVideoMediaReceiveChannel::SetRecvCodecs(
const std::vector<Codec>& codecs) {
@@ -587,9 +587,9 @@
int64_t max_size_bytes) {
return false;
}
-absl::optional<webrtc::AudioDeviceModule::Stats>
+std::optional<webrtc::AudioDeviceModule::Stats>
FakeVoiceEngine::GetAudioDeviceStats() {
- return absl::nullopt;
+ return std::nullopt;
}
void FakeVoiceEngine::StopAecDump() {}
diff --git a/media/base/fake_media_engine.h b/media/base/fake_media_engine.h
index 64af78d..4b07a4a 100644
--- a/media/base/fake_media_engine.h
+++ b/media/base/fake_media_engine.h
@@ -92,8 +92,8 @@
bool CheckNoRtcp() { return rtcp_packets_.empty(); }
void set_fail_set_recv_codecs(bool fail) { fail_set_recv_codecs_ = fail; }
void ResetUnsignaledRecvStream() override {}
- absl::optional<uint32_t> GetUnsignaledSsrc() const override {
- return absl::nullopt;
+ std::optional<uint32_t> GetUnsignaledSsrc() const override {
+ return std::nullopt;
}
void ChooseReceiverReportSsrc(const std::set<uint32_t>& choices) override {}
@@ -481,8 +481,7 @@
bool GetOutputVolume(uint32_t ssrc, double* volume);
bool SetBaseMinimumPlayoutDelayMs(uint32_t ssrc, int delay_ms) override;
- absl::optional<int> GetBaseMinimumPlayoutDelayMs(
- uint32_t ssrc) const override;
+ std::optional<int> GetBaseMinimumPlayoutDelayMs(uint32_t ssrc) const override;
bool GetStats(VoiceMediaReceiveInfo* info,
bool get_and_clear_legacy_stats) override;
@@ -508,7 +507,7 @@
int sample_rate,
size_t number_of_channels,
size_t number_of_frames,
- absl::optional<int64_t> absolute_capture_timestamp_ms) override;
+ std::optional<int64_t> absolute_capture_timestamp_ms) override;
void OnClose() override;
int NumPreferredChannels() const override { return -1; }
AudioSource* source() const;
@@ -577,7 +576,7 @@
bool SendCodecHasNack() const override { return false; }
void SetSendCodecChangedCallback(
absl::AnyInvocable<void()> callback) override {}
- absl::optional<Codec> GetSendCodec() const override;
+ std::optional<Codec> GetSendCodec() const override;
bool GetStats(VoiceMediaSendInfo* stats) override;
@@ -591,7 +590,7 @@
int sample_rate,
size_t number_of_channels,
size_t number_of_frames,
- absl::optional<int64_t> absolute_capture_timestamp_ms) override;
+ std::optional<int64_t> absolute_capture_timestamp_ms) override;
void OnClose() override;
int NumPreferredChannels() const override { return -1; }
AudioSource* source() const;
@@ -662,8 +661,7 @@
std::vector<webrtc::RtpSource> GetSources(uint32_t ssrc) const override;
bool SetBaseMinimumPlayoutDelayMs(uint32_t ssrc, int delay_ms) override;
- absl::optional<int> GetBaseMinimumPlayoutDelayMs(
- uint32_t ssrc) const override;
+ std::optional<int> GetBaseMinimumPlayoutDelayMs(uint32_t ssrc) const override;
void SetRecordableEncodedFrameCallback(
uint32_t ssrc,
@@ -674,7 +672,7 @@
void SetReceiverFeedbackParameters(bool lntf_enabled,
bool nack_enabled,
webrtc::RtcpMode rtcp_mode,
- absl::optional<int> rtx_time) override {}
+ std::optional<int> rtx_time) override {}
bool GetStats(VideoMediaReceiveInfo* info) override;
bool AddDefaultRecvStreamForTesting(const StreamParams& sp) override {
@@ -720,7 +718,7 @@
int max_bps() const;
bool SetSenderParameters(const VideoSenderParameters& params) override;
- absl::optional<Codec> GetSendCodec() const override;
+ std::optional<Codec> GetSendCodec() const override;
bool SetSend(bool send) override;
bool SetVideoSend(
@@ -744,9 +742,7 @@
bool SendCodecHasLntf() const override { return false; }
bool SendCodecHasNack() const override { return false; }
- absl::optional<int> SendCodecRtxTime() const override {
- return absl::nullopt;
- }
+ std::optional<int> SendCodecRtxTime() const override { return std::nullopt; }
bool GetStats(VideoMediaSendInfo* info) override;
private:
@@ -789,7 +785,7 @@
int GetInputLevel();
bool StartAecDump(webrtc::FileWrapper file, int64_t max_size_bytes) override;
void StopAecDump() override;
- absl::optional<webrtc::AudioDeviceModule::Stats> GetAudioDeviceStats()
+ std::optional<webrtc::AudioDeviceModule::Stats> GetAudioDeviceStats()
override;
std::vector<webrtc::RtpHeaderExtensionCapability> GetRtpHeaderExtensions()
const override;
diff --git a/media/base/media_channel.h b/media/base/media_channel.h
index d656304..ed69c6b 100644
--- a/media/base/media_channel.h
+++ b/media/base/media_channel.h
@@ -13,12 +13,12 @@
#include <map>
#include <memory>
+#include <optional>
#include <set>
#include <string>
#include <utility>
#include <vector>
-#include "absl/types/optional.h"
#include "api/audio/audio_processing_statistics.h"
#include "api/audio_codecs/audio_encoder.h"
#include "api/audio_options.h"
@@ -85,8 +85,7 @@
const int kScreencastDefaultFps = 5;
template <class T>
-static std::string ToStringIfSet(const char* key,
- const absl::optional<T>& val) {
+static std::string ToStringIfSet(const char* key, const std::optional<T>& val) {
std::string str;
if (val) {
str = key;
@@ -146,23 +145,23 @@
// Enable denoising? This flag comes from the getUserMedia
// constraint 'googNoiseReduction', and WebRtcVideoEngine passes it
// on to the codec options. Disabled by default.
- absl::optional<bool> video_noise_reduction;
+ std::optional<bool> video_noise_reduction;
// Force screencast to use a minimum bitrate. This flag comes from
// the PeerConnection constraint 'googScreencastMinBitrate'. It is
// copied to the encoder config by WebRtcVideoChannel.
// TODO(https://crbug.com/1315155): Remove the ability to set it in Chromium
// and delete this flag (it should default to 100 kbps).
- absl::optional<int> screencast_min_bitrate_kbps;
+ std::optional<int> screencast_min_bitrate_kbps;
// Set by screencast sources. Implies selection of encoding settings
// suitable for screencast. Most likely not the right way to do
// things, e.g., screencast of a text document and screencast of a
// youtube video have different needs.
- absl::optional<bool> is_screencast;
+ std::optional<bool> is_screencast;
webrtc::VideoTrackInterface::ContentHint content_hint;
private:
template <typename T>
- static void SetFrom(absl::optional<T>* s, const absl::optional<T>& o) {
+ static void SetFrom(std::optional<T>* s, const std::optional<T>& o) {
if (o) {
*s = o;
}
@@ -192,7 +191,7 @@
virtual cricket::MediaType media_type() const = 0;
// Gets the currently set codecs/payload types to be used for outgoing media.
- virtual absl::optional<Codec> GetSendCodec() const = 0;
+ virtual std::optional<Codec> GetSendCodec() const = 0;
// Creates a new outgoing media stream with SSRCs and CNAME as described
// by sp.
@@ -282,7 +281,7 @@
// Called on the network when an RTP packet is received.
virtual void OnPacketReceived(const webrtc::RtpPacketReceived& packet) = 0;
// Gets the current unsignaled receive stream's SSRC, if there is one.
- virtual absl::optional<uint32_t> GetUnsignaledSsrc() const = 0;
+ virtual std::optional<uint32_t> GetUnsignaledSsrc() const = 0;
// Sets the local SSRC for listening to incoming RTCP reports.
virtual void ChooseReceiverReportSsrc(const std::set<uint32_t>& choices) = 0;
// This is currently a workaround because of the demuxer state being managed
@@ -318,7 +317,7 @@
virtual bool SetBaseMinimumPlayoutDelayMs(uint32_t ssrc, int delay_ms) = 0;
// Returns current value of base minimum delay in milliseconds.
- virtual absl::optional<int> GetBaseMinimumPlayoutDelayMs(
+ virtual std::optional<int> GetBaseMinimumPlayoutDelayMs(
uint32_t ssrc) const = 0;
};
@@ -385,12 +384,12 @@
// https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-nackcount
uint32_t nacks_received = 0;
// https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-targetbitrate
- absl::optional<double> target_bitrate;
+ std::optional<double> target_bitrate;
int packets_lost = 0;
float fraction_lost = 0.0f;
int64_t rtt_ms = 0;
std::string codec_name;
- absl::optional<int> codec_payload_type;
+ std::optional<int> codec_payload_type;
std::vector<SsrcSenderInfo> local_stats;
std::vector<SsrcReceiverInfo> remote_stats;
// A snapshot of the most recent Report Block with additional data of interest
@@ -398,7 +397,7 @@
// this list, the `ReportBlockData::source_ssrc()`, which is the SSRC of the
// corresponding outbound RTP stream, is unique.
std::vector<webrtc::ReportBlockData> report_block_datas;
- absl::optional<bool> active;
+ std::optional<bool> active;
// https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-totalpacketsenddelay
webrtc::TimeDelta total_packet_send_delay = webrtc::TimeDelta::Zero();
};
@@ -445,9 +444,9 @@
int packets_received = 0;
int packets_lost = 0;
- absl::optional<uint64_t> retransmitted_bytes_received;
- absl::optional<uint64_t> retransmitted_packets_received;
- absl::optional<uint32_t> nacks_sent;
+ std::optional<uint64_t> retransmitted_bytes_received;
+ std::optional<uint64_t> retransmitted_packets_received;
+ std::optional<uint32_t> nacks_sent;
// Jitter (network-related) latency (cumulative).
// https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-jitterbufferdelay
double jitter_buffer_delay_seconds = 0.0;
@@ -463,32 +462,32 @@
// The timestamp at which the last packet was received, i.e. the time of the
// local clock when it was received - not the RTP timestamp of that packet.
// https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-lastpacketreceivedtimestamp
- absl::optional<webrtc::Timestamp> last_packet_received;
+ std::optional<webrtc::Timestamp> last_packet_received;
// https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-estimatedplayouttimestamp
- absl::optional<int64_t> estimated_playout_ntp_timestamp_ms;
+ std::optional<int64_t> estimated_playout_ntp_timestamp_ms;
std::string codec_name;
- absl::optional<int> codec_payload_type;
+ std::optional<int> codec_payload_type;
std::vector<SsrcReceiverInfo> local_stats;
std::vector<SsrcSenderInfo> remote_stats;
// https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-fecpacketsreceived
- absl::optional<uint64_t> fec_packets_received;
+ std::optional<uint64_t> fec_packets_received;
// https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-fecpacketsdiscarded
- absl::optional<uint64_t> fec_packets_discarded;
+ std::optional<uint64_t> fec_packets_discarded;
// https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-fecbytesreceived
- absl::optional<uint64_t> fec_bytes_received;
+ std::optional<uint64_t> fec_bytes_received;
// https://www.w3.org/TR/webrtc-stats/#dom-rtcinboundrtpstreamstats-totalprocessingdelay
double total_processing_delay_seconds = 0.0;
// Remote outbound stats derived by the received RTCP sender reports.
// https://w3c.github.io/webrtc-stats/#remoteoutboundrtpstats-dict*
- absl::optional<int64_t> last_sender_report_timestamp_ms;
- absl::optional<int64_t> last_sender_report_remote_timestamp_ms;
+ std::optional<int64_t> last_sender_report_timestamp_ms;
+ std::optional<int64_t> last_sender_report_remote_timestamp_ms;
uint64_t sender_reports_packets_sent = 0;
uint64_t sender_reports_bytes_sent = 0;
uint64_t sender_reports_reports_count = 0;
// These require a DLRR block, see
// https://w3c.github.io/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-roundtriptime
- absl::optional<webrtc::TimeDelta> round_trip_time;
+ std::optional<webrtc::TimeDelta> round_trip_time;
webrtc::TimeDelta total_round_trip_time = webrtc::TimeDelta::Zero();
int round_trip_time_measurements = 0;
};
@@ -572,7 +571,7 @@
VideoSenderInfo();
~VideoSenderInfo();
std::vector<SsrcGroup> ssrc_groups;
- absl::optional<std::string> encoder_implementation_name;
+ std::optional<std::string> encoder_implementation_name;
int firs_received = 0;
int plis_received = 0;
int send_frame_width = 0;
@@ -601,23 +600,23 @@
// https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-totalencodedbytestarget
uint64_t total_encoded_bytes_target = 0;
bool has_entered_low_resolution = false;
- absl::optional<uint64_t> qp_sum;
+ std::optional<uint64_t> qp_sum;
webrtc::VideoContentType content_type = webrtc::VideoContentType::UNSPECIFIED;
uint32_t frames_sent = 0;
// https://w3c.github.io/webrtc-stats/#dom-rtcvideosenderstats-hugeframessent
uint32_t huge_frames_sent = 0;
uint32_t aggregated_huge_frames_sent = 0;
- absl::optional<std::string> rid;
- absl::optional<bool> power_efficient_encoder;
- absl::optional<webrtc::ScalabilityMode> scalability_mode;
+ std::optional<std::string> rid;
+ std::optional<bool> power_efficient_encoder;
+ std::optional<webrtc::ScalabilityMode> scalability_mode;
};
struct VideoReceiverInfo : public MediaReceiverInfo {
VideoReceiverInfo();
~VideoReceiverInfo();
std::vector<SsrcGroup> ssrc_groups;
- absl::optional<std::string> decoder_implementation_name;
- absl::optional<bool> power_efficient_decoder;
+ std::optional<std::string> decoder_implementation_name;
+ std::optional<bool> power_efficient_decoder;
int packets_concealed = 0;
int firs_sent = 0;
int plis_sent = 0;
@@ -635,7 +634,7 @@
uint32_t frames_decoded = 0;
uint32_t key_frames_decoded = 0;
uint32_t frames_rendered = 0;
- absl::optional<uint64_t> qp_sum;
+ std::optional<uint64_t> qp_sum;
// https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-totaldecodetime
webrtc::TimeDelta total_decode_time = webrtc::TimeDelta::Zero();
// https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-totalprocessingdelay
@@ -681,7 +680,7 @@
// Timing frame info: all important timestamps for a full lifetime of a
// single 'timing frame'.
- absl::optional<webrtc::TimingFrameInfo> timing_frame_info;
+ std::optional<webrtc::TimingFrameInfo> timing_frame_info;
};
struct BandwidthEstimationInfo {
@@ -971,7 +970,7 @@
// Information queries to support SetReceiverFeedbackParameters
virtual webrtc::RtcpMode SendCodecRtcpMode() const = 0;
virtual bool SendCodecHasLntf() const = 0;
- virtual absl::optional<int> SendCodecRtxTime() const = 0;
+ virtual std::optional<int> SendCodecRtxTime() const = 0;
};
class VideoMediaReceiveChannelInterface : public MediaReceiveChannelInterface {
@@ -1006,7 +1005,7 @@
virtual void SetReceiverFeedbackParameters(bool lntf_enabled,
bool nack_enabled,
webrtc::RtcpMode rtcp_mode,
- absl::optional<int> rtx_time) = 0;
+ std::optional<int> rtx_time) = 0;
virtual bool AddDefaultRecvStreamForTesting(const StreamParams& sp) = 0;
};
diff --git a/media/base/media_channel_impl.h b/media/base/media_channel_impl.h
index eda47af..70bfe28 100644
--- a/media/base/media_channel_impl.h
+++ b/media/base/media_channel_impl.h
@@ -16,6 +16,7 @@
#include <functional>
#include <memory>
+#include <optional>
#include <set>
#include <string>
#include <utility>
@@ -23,7 +24,6 @@
#include "absl/functional/any_invocable.h"
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/audio_options.h"
#include "api/call/audio_sink.h"
#include "api/call/transport.h"
diff --git a/media/base/media_engine.cc b/media/base/media_engine.cc
index c551a58..37ec552 100644
--- a/media/base/media_engine.cc
+++ b/media/base/media_engine.cc
@@ -68,7 +68,7 @@
webrtc::RTCError CheckScalabilityModeValues(
const webrtc::RtpParameters& rtp_parameters,
rtc::ArrayView<cricket::Codec> send_codecs,
- absl::optional<cricket::Codec> send_codec) {
+ std::optional<cricket::Codec> send_codec) {
using webrtc::RTCErrorType;
if (send_codecs.empty()) {
@@ -140,7 +140,7 @@
webrtc::RTCError CheckRtpParametersValues(
const webrtc::RtpParameters& rtp_parameters,
rtc::ArrayView<cricket::Codec> send_codecs,
- absl::optional<cricket::Codec> send_codec) {
+ std::optional<cricket::Codec> send_codec) {
using webrtc::RTCErrorType;
for (size_t i = 0; i < rtp_parameters.encodings.size(); ++i) {
@@ -203,14 +203,14 @@
const webrtc::RtpParameters& old_rtp_parameters,
const webrtc::RtpParameters& rtp_parameters) {
return CheckRtpParametersInvalidModificationAndValues(
- old_rtp_parameters, rtp_parameters, {}, absl::nullopt);
+ old_rtp_parameters, rtp_parameters, {}, std::nullopt);
}
webrtc::RTCError CheckRtpParametersInvalidModificationAndValues(
const webrtc::RtpParameters& old_rtp_parameters,
const webrtc::RtpParameters& rtp_parameters,
rtc::ArrayView<cricket::Codec> send_codecs,
- absl::optional<cricket::Codec> send_codec) {
+ std::optional<cricket::Codec> send_codec) {
using webrtc::RTCErrorType;
if (rtp_parameters.encodings.size() != old_rtp_parameters.encodings.size()) {
LOG_AND_RETURN_ERROR(
diff --git a/media/base/media_engine.h b/media/base/media_engine.h
index 3deda09..f3a7a9c 100644
--- a/media/base/media_engine.h
+++ b/media/base/media_engine.h
@@ -41,14 +41,14 @@
webrtc::RTCError CheckScalabilityModeValues(
const webrtc::RtpParameters& new_parameters,
rtc::ArrayView<cricket::Codec> send_codecs,
- absl::optional<cricket::Codec> send_codec);
+ std::optional<cricket::Codec> send_codec);
// Checks the parameters have valid and supported values, and checks parameters
// with CheckScalabilityModeValues().
webrtc::RTCError CheckRtpParametersValues(
const webrtc::RtpParameters& new_parameters,
rtc::ArrayView<cricket::Codec> send_codecs,
- absl::optional<cricket::Codec> send_codec);
+ std::optional<cricket::Codec> send_codec);
// Checks that the immutable values have not changed in new_parameters and
// checks all parameters with CheckRtpParametersValues().
@@ -56,7 +56,7 @@
const webrtc::RtpParameters& old_parameters,
const webrtc::RtpParameters& new_parameters,
rtc::ArrayView<cricket::Codec> send_codecs,
- absl::optional<cricket::Codec> send_codec);
+ std::optional<cricket::Codec> send_codec);
// Checks that the immutable values have not changed in new_parameters and
// checks parameters (except SVC) with CheckRtpParametersValues(). It should
@@ -130,7 +130,7 @@
// Stops recording AEC dump.
virtual void StopAecDump() = 0;
- virtual absl::optional<webrtc::AudioDeviceModule::Stats>
+ virtual std::optional<webrtc::AudioDeviceModule::Stats>
GetAudioDeviceStats() = 0;
};
diff --git a/media/base/media_engine_unittest.cc b/media/base/media_engine_unittest.cc
index 6f78d61..a223254 100644
--- a/media/base/media_engine_unittest.cc
+++ b/media/base/media_engine_unittest.cc
@@ -77,7 +77,7 @@
(webrtc::FileWrapper file, int64_t max_size_bytes),
(override));
MOCK_METHOD(void, StopAecDump, (), (override));
- MOCK_METHOD(absl::optional<webrtc::AudioDeviceModule::Stats>,
+ MOCK_METHOD(std::optional<webrtc::AudioDeviceModule::Stats>,
GetAudioDeviceStats,
(),
(override));
diff --git a/media/base/sdp_video_format_utils.cc b/media/base/sdp_video_format_utils.cc
index 81ddb7b..c76afad 100644
--- a/media/base/sdp_video_format_utils.cc
+++ b/media/base/sdp_video_format_utils.cc
@@ -57,17 +57,17 @@
return H264LevelIsLess(a, b) ? a : b;
}
-absl::optional<int> ParsePositiveNumberFromParams(
+std::optional<int> ParsePositiveNumberFromParams(
const CodecParameterMap& params,
const char* parameter_name) {
const auto max_frame_rate_it = params.find(parameter_name);
if (max_frame_rate_it == params.end())
- return absl::nullopt;
+ return std::nullopt;
- const absl::optional<int> i =
+ const std::optional<int> i =
rtc::StringToNumber<int>(max_frame_rate_it->second);
if (!i.has_value() || i.value() <= 0)
- return absl::nullopt;
+ return std::nullopt;
return i;
}
@@ -102,9 +102,9 @@
}
// Parse profile-tier-level.
- const absl::optional<H265ProfileTierLevel> local_profile_tier_level =
+ const std::optional<H265ProfileTierLevel> local_profile_tier_level =
ParseSdpForH265ProfileTierLevel(local_supported_params);
- const absl::optional<H265ProfileTierLevel> remote_profile_tier_level =
+ const std::optional<H265ProfileTierLevel> remote_profile_tier_level =
ParseSdpForH265ProfileTierLevel(remote_offered_params);
// Profile and tier for local and remote codec must be valid and equal.
RTC_DCHECK(local_profile_tier_level);
@@ -138,9 +138,9 @@
}
// Parse profile-level-ids.
- const absl::optional<H264ProfileLevelId> local_profile_level_id =
+ const std::optional<H264ProfileLevelId> local_profile_level_id =
ParseSdpForH264ProfileLevelId(local_supported_params);
- const absl::optional<H264ProfileLevelId> remote_profile_level_id =
+ const std::optional<H264ProfileLevelId> remote_profile_level_id =
ParseSdpForH264ProfileLevelId(remote_offered_params);
// The local and remote codec must have valid and equal H264 Profiles.
RTC_DCHECK(local_profile_level_id);
@@ -167,17 +167,15 @@
H264ProfileLevelId(local_profile_level_id->profile, answer_level));
}
-absl::optional<int> ParseSdpForVPxMaxFrameRate(
- const CodecParameterMap& params) {
+std::optional<int> ParseSdpForVPxMaxFrameRate(const CodecParameterMap& params) {
return ParsePositiveNumberFromParams(params, kVPxFmtpMaxFrameRate);
}
-absl::optional<int> ParseSdpForVPxMaxFrameSize(
- const CodecParameterMap& params) {
- const absl::optional<int> i =
+std::optional<int> ParseSdpForVPxMaxFrameSize(const CodecParameterMap& params) {
+ const std::optional<int> i =
ParsePositiveNumberFromParams(params, kVPxFmtpMaxFrameSize);
- return i ? absl::make_optional(i.value() * kVPxFmtpFrameSizeSubBlockPixels)
- : absl::nullopt;
+ return i ? std::make_optional(i.value() * kVPxFmtpFrameSizeSubBlockPixels)
+ : std::nullopt;
}
bool SupportsPerLayerPictureLossIndication(const CodecParameterMap& params) {
diff --git a/media/base/sdp_video_format_utils.h b/media/base/sdp_video_format_utils.h
index e6a4ca7..75b8f71 100644
--- a/media/base/sdp_video_format_utils.h
+++ b/media/base/sdp_video_format_utils.h
@@ -11,7 +11,8 @@
#ifndef MEDIA_BASE_SDP_VIDEO_FORMAT_UTILS_H_
#define MEDIA_BASE_SDP_VIDEO_FORMAT_UTILS_H_
-#include "absl/types/optional.h"
+#include <optional>
+
#include "api/video_codecs/sdp_video_format.h"
namespace webrtc {
@@ -48,14 +49,14 @@
CodecParameterMap* answer_params);
#endif
-// Parse max frame rate from SDP FMTP line. absl::nullopt is returned if the
+// Parse max frame rate from SDP FMTP line. std::nullopt is returned if the
// field is missing or not a number.
-absl::optional<int> ParseSdpForVPxMaxFrameRate(const CodecParameterMap& params);
+std::optional<int> ParseSdpForVPxMaxFrameRate(const CodecParameterMap& params);
-// Parse max frame size from SDP FMTP line. absl::nullopt is returned if the
+// Parse max frame size from SDP FMTP line. std::nullopt is returned if the
// field is missing or not a number. Please note that the value is stored in sub
// blocks but the returned value is in total number of pixels.
-absl::optional<int> ParseSdpForVPxMaxFrameSize(const CodecParameterMap& params);
+std::optional<int> ParseSdpForVPxMaxFrameSize(const CodecParameterMap& params);
// Determines whether the non-standard x-google-per-layer-pli fmtp is present
// in the parameters and has a value of "1".
diff --git a/media/base/sdp_video_format_utils_unittest.cc b/media/base/sdp_video_format_utils_unittest.cc
index 04327d3..d9a0600 100644
--- a/media/base/sdp_video_format_utils_unittest.cc
+++ b/media/base/sdp_video_format_utils_unittest.cc
@@ -106,7 +106,7 @@
TEST(SdpVideoFormatUtilsTest, MaxFrameRateIsMissingOrInvalid) {
CodecParameterMap params;
- absl::optional<int> empty = ParseSdpForVPxMaxFrameRate(params);
+ std::optional<int> empty = ParseSdpForVPxMaxFrameRate(params);
EXPECT_FALSE(empty);
params[kVPxFmtpMaxFrameRate] = "-1";
EXPECT_FALSE(ParseSdpForVPxMaxFrameRate(params));
@@ -126,7 +126,7 @@
TEST(SdpVideoFormatUtilsTest, MaxFrameSizeIsMissingOrInvalid) {
CodecParameterMap params;
- absl::optional<int> empty = ParseSdpForVPxMaxFrameSize(params);
+ std::optional<int> empty = ParseSdpForVPxMaxFrameSize(params);
EXPECT_FALSE(empty);
params[kVPxFmtpMaxFrameSize] = "-1";
EXPECT_FALSE(ParseSdpForVPxMaxFrameSize(params));
diff --git a/media/base/video_adapter.cc b/media/base/video_adapter.cc
index 5eadbf8..1d0beb5 100644
--- a/media/base/video_adapter.cc
+++ b/media/base/video_adapter.cc
@@ -15,9 +15,9 @@
#include <cstdint>
#include <cstdlib>
#include <limits>
+#include <optional>
#include <utility>
-#include "absl/types/optional.h"
#include "media/base/video_common.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
@@ -121,10 +121,10 @@
return best_scale;
}
-absl::optional<std::pair<int, int>> Swap(
- const absl::optional<std::pair<int, int>>& in) {
+std::optional<std::pair<int, int>> Swap(
+ const std::optional<std::pair<int, int>>& in) {
if (!in) {
- return absl::nullopt;
+ return std::nullopt;
}
return std::make_pair(in->second, in->first);
}
@@ -175,7 +175,7 @@
// Select target aspect ratio and max pixel count depending on input frame
// orientation.
- absl::optional<std::pair<int, int>> target_aspect_ratio;
+ std::optional<std::pair<int, int>> target_aspect_ratio;
if (in_width > in_height) {
target_aspect_ratio = output_format_request_.target_landscape_aspect_ratio;
if (output_format_request_.max_landscape_pixel_count)
@@ -268,10 +268,10 @@
}
void VideoAdapter::OnOutputFormatRequest(
- const absl::optional<VideoFormat>& format) {
- absl::optional<std::pair<int, int>> target_aspect_ratio;
- absl::optional<int> max_pixel_count;
- absl::optional<int> max_fps;
+ const std::optional<VideoFormat>& format) {
+ std::optional<std::pair<int, int>> target_aspect_ratio;
+ std::optional<int> max_pixel_count;
+ std::optional<int> max_fps;
if (format) {
target_aspect_ratio = std::make_pair(format->width, format->height);
max_pixel_count = format->width * format->height;
@@ -282,11 +282,11 @@
}
void VideoAdapter::OnOutputFormatRequest(
- const absl::optional<std::pair<int, int>>& target_aspect_ratio,
- const absl::optional<int>& max_pixel_count,
- const absl::optional<int>& max_fps) {
- absl::optional<std::pair<int, int>> target_landscape_aspect_ratio;
- absl::optional<std::pair<int, int>> target_portrait_aspect_ratio;
+ const std::optional<std::pair<int, int>>& target_aspect_ratio,
+ const std::optional<int>& max_pixel_count,
+ const std::optional<int>& max_fps) {
+ std::optional<std::pair<int, int>> target_landscape_aspect_ratio;
+ std::optional<std::pair<int, int>> target_portrait_aspect_ratio;
if (target_aspect_ratio && target_aspect_ratio->first > 0 &&
target_aspect_ratio->second > 0) {
// Maintain input orientation.
@@ -302,11 +302,11 @@
}
void VideoAdapter::OnOutputFormatRequest(
- const absl::optional<std::pair<int, int>>& target_landscape_aspect_ratio,
- const absl::optional<int>& max_landscape_pixel_count,
- const absl::optional<std::pair<int, int>>& target_portrait_aspect_ratio,
- const absl::optional<int>& max_portrait_pixel_count,
- const absl::optional<int>& max_fps) {
+ const std::optional<std::pair<int, int>>& target_landscape_aspect_ratio,
+ const std::optional<int>& max_landscape_pixel_count,
+ const std::optional<std::pair<int, int>>& target_portrait_aspect_ratio,
+ const std::optional<int>& max_portrait_pixel_count,
+ const std::optional<int>& max_fps) {
webrtc::MutexLock lock(&mutex_);
OutputFormatRequest request = {
diff --git a/media/base/video_adapter.h b/media/base/video_adapter.h
index c1d311d..35ed672 100644
--- a/media/base/video_adapter.h
+++ b/media/base/video_adapter.h
@@ -13,10 +13,10 @@
#include <stdint.h>
+#include <optional>
#include <string>
#include <utility>
-#include "absl/types/optional.h"
#include "api/video/video_source_interface.h"
#include "common_video/framerate_controller.h"
#include "media/base/video_common.h"
@@ -62,7 +62,7 @@
// maintain the input orientation, so it doesn't matter if e.g. 1280x720 or
// 720x1280 is requested.
// Note: Should be called from the source only.
- void OnOutputFormatRequest(const absl::optional<VideoFormat>& format)
+ void OnOutputFormatRequest(const std::optional<VideoFormat>& format)
RTC_LOCKS_EXCLUDED(mutex_);
// Requests output frame size and frame interval from `AdaptFrameResolution`.
@@ -74,20 +74,20 @@
// `max_fps`: The maximum output framerate.
// Note: Should be called from the source only.
void OnOutputFormatRequest(
- const absl::optional<std::pair<int, int>>& target_aspect_ratio,
- const absl::optional<int>& max_pixel_count,
- const absl::optional<int>& max_fps) RTC_LOCKS_EXCLUDED(mutex_);
+ const std::optional<std::pair<int, int>>& target_aspect_ratio,
+ const std::optional<int>& max_pixel_count,
+ const std::optional<int>& max_fps) RTC_LOCKS_EXCLUDED(mutex_);
// Same as above, but allows setting two different target aspect ratios
// depending on incoming frame orientation. This gives more fine-grained
// control and can e.g. be used to force landscape video to be cropped to
// portrait video.
void OnOutputFormatRequest(
- const absl::optional<std::pair<int, int>>& target_landscape_aspect_ratio,
- const absl::optional<int>& max_landscape_pixel_count,
- const absl::optional<std::pair<int, int>>& target_portrait_aspect_ratio,
- const absl::optional<int>& max_portrait_pixel_count,
- const absl::optional<int>& max_fps) RTC_LOCKS_EXCLUDED(mutex_);
+ const std::optional<std::pair<int, int>>& target_landscape_aspect_ratio,
+ const std::optional<int>& max_landscape_pixel_count,
+ const std::optional<std::pair<int, int>>& target_portrait_aspect_ratio,
+ const std::optional<int>& max_portrait_pixel_count,
+ const std::optional<int>& max_fps) RTC_LOCKS_EXCLUDED(mutex_);
// Requests the output frame size from `AdaptFrameResolution` to have as close
// as possible to `sink_wants.target_pixel_count` pixels (if set)
@@ -134,11 +134,11 @@
// OnResolutionFramerateRequest respectively.
// The adapted output format is the minimum of these.
struct OutputFormatRequest {
- absl::optional<std::pair<int, int>> target_landscape_aspect_ratio;
- absl::optional<int> max_landscape_pixel_count;
- absl::optional<std::pair<int, int>> target_portrait_aspect_ratio;
- absl::optional<int> max_portrait_pixel_count;
- absl::optional<int> max_fps;
+ std::optional<std::pair<int, int>> target_landscape_aspect_ratio;
+ std::optional<int> max_landscape_pixel_count;
+ std::optional<std::pair<int, int>> target_portrait_aspect_ratio;
+ std::optional<int> max_portrait_pixel_count;
+ std::optional<int> max_fps;
// For logging.
std::string ToString() const;
@@ -157,7 +157,7 @@
// frame). This allows for an application to only use
// RtpEncodingParameters::request_resolution and get the same behavior as if
// it had used VideoAdapter::OnOutputFormatRequest.
- absl::optional<OutputFormatRequest> stashed_output_format_request_
+ std::optional<OutputFormatRequest> stashed_output_format_request_
RTC_GUARDED_BY(mutex_);
webrtc::FramerateController framerate_controller_ RTC_GUARDED_BY(mutex_);
diff --git a/media/base/video_adapter_unittest.cc b/media/base/video_adapter_unittest.cc
index 778e61e..aab96dc 100644
--- a/media/base/video_adapter_unittest.cc
+++ b/media/base/video_adapter_unittest.cc
@@ -36,7 +36,7 @@
using ::testing::Pair;
using webrtc::Resolution;
-rtc::VideoSinkWants BuildSinkWants(absl::optional<int> target_pixel_count,
+rtc::VideoSinkWants BuildSinkWants(std::optional<int> target_pixel_count,
int max_pixel_count,
int max_framerate_fps,
int sink_alignment = 1) {
@@ -52,7 +52,7 @@
}
rtc::VideoSinkWants BuildSinkWants(
- absl::optional<webrtc::Resolution> requested_resolution,
+ std::optional<webrtc::Resolution> requested_resolution,
bool any_active_without_requested_resolution) {
rtc::VideoSinkWants wants;
wants.max_framerate_fps = kDefaultFps;
@@ -153,12 +153,12 @@
void OnOutputFormatRequest(int width,
int height,
- const absl::optional<int>& fps) {
+ const std::optional<int>& fps) {
if (use_new_format_request_) {
- absl::optional<std::pair<int, int>> target_aspect_ratio =
+ std::optional<std::pair<int, int>> target_aspect_ratio =
std::make_pair(width, height);
- absl::optional<int> max_pixel_count = width * height;
- absl::optional<int> max_fps = fps;
+ std::optional<int> max_pixel_count = width * height;
+ std::optional<int> max_fps = fps;
adapter_.OnOutputFormatRequest(target_aspect_ratio, max_pixel_count,
max_fps);
return;
@@ -211,7 +211,7 @@
}
TEST_P(VideoAdapterTest, AdaptZeroInterval) {
- OnOutputFormatRequest(kWidth, kHeight, absl::nullopt);
+ OnOutputFormatRequest(kWidth, kHeight, std::nullopt);
for (int i = 0; i < 40; ++i)
adapter_wrapper_->AdaptFrame(frame_source_->GetFrame());
@@ -265,7 +265,7 @@
// Expect the number of dropped frames to be half of the number the captured
// frames.
TEST_P(VideoAdapterTest, AdaptFramerateToHalfWithNoPixelLimit) {
- adapter_.OnOutputFormatRequest(absl::nullopt, absl::nullopt, kDefaultFps / 2);
+ adapter_.OnOutputFormatRequest(std::nullopt, std::nullopt, kDefaultFps / 2);
// Capture 10 frames and verify that every other frame is dropped. The first
// frame should not be dropped.
@@ -305,7 +305,7 @@
// Do not adapt the frame rate or the resolution. Expect no frame drop, no
// cropping, and no resolution change.
TEST_P(VideoAdapterTest, AdaptFramerateRequestMax) {
- adapter_.OnSinkWants(BuildSinkWants(absl::nullopt,
+ adapter_.OnSinkWants(BuildSinkWants(std::nullopt,
std::numeric_limits<int>::max(),
std::numeric_limits<int>::max()));
@@ -322,7 +322,7 @@
TEST_P(VideoAdapterTest, AdaptFramerateRequestZero) {
adapter_.OnSinkWants(
- BuildSinkWants(absl::nullopt, std::numeric_limits<int>::max(), 0));
+ BuildSinkWants(std::nullopt, std::numeric_limits<int>::max(), 0));
for (int i = 0; i < 10; ++i)
adapter_wrapper_->AdaptFrame(frame_source_->GetFrame());
@@ -336,7 +336,7 @@
// the number of dropped frames to be half of the number the captured frames.
TEST_P(VideoAdapterTest, AdaptFramerateRequestHalf) {
adapter_.OnSinkWants(BuildSinkWants(
- absl::nullopt, std::numeric_limits<int>::max(), kDefaultFps / 2));
+ std::nullopt, std::numeric_limits<int>::max(), kDefaultFps / 2));
for (int i = 0; i < 10; ++i)
adapter_wrapper_->AdaptFrame(frame_source_->GetFrame());
@@ -451,7 +451,7 @@
EXPECT_EQ(400, out_height_);
// Format request 640x400.
- OnOutputFormatRequest(640, 400, absl::nullopt);
+ OnOutputFormatRequest(640, 400, std::nullopt);
EXPECT_TRUE(adapter_.AdaptFrameResolution(640, 400, 0, &cropped_width_,
&cropped_height_, &out_width_,
&out_height_));
@@ -462,7 +462,7 @@
// Request 1280x720, higher than input, but aspect 16:9. Expect cropping but
// no scaling.
- OnOutputFormatRequest(1280, 720, absl::nullopt);
+ OnOutputFormatRequest(1280, 720, std::nullopt);
EXPECT_TRUE(adapter_.AdaptFrameResolution(640, 400, 0, &cropped_width_,
&cropped_height_, &out_width_,
&out_height_));
@@ -472,13 +472,13 @@
EXPECT_EQ(360, out_height_);
// Request 0x0.
- OnOutputFormatRequest(0, 0, absl::nullopt);
+ OnOutputFormatRequest(0, 0, std::nullopt);
EXPECT_FALSE(adapter_.AdaptFrameResolution(640, 400, 0, &cropped_width_,
&cropped_height_, &out_width_,
&out_height_));
// Request 320x200. Expect scaling, but no cropping.
- OnOutputFormatRequest(320, 200, absl::nullopt);
+ OnOutputFormatRequest(320, 200, std::nullopt);
EXPECT_TRUE(adapter_.AdaptFrameResolution(640, 400, 0, &cropped_width_,
&cropped_height_, &out_width_,
&out_height_));
@@ -490,7 +490,7 @@
// Request resolution close to 2/3 scale. Expect adapt down. Scaling to 2/3
// is not optimized and not allowed, therefore 1/2 scaling will be used
// instead.
- OnOutputFormatRequest(424, 265, absl::nullopt);
+ OnOutputFormatRequest(424, 265, std::nullopt);
EXPECT_TRUE(adapter_.AdaptFrameResolution(640, 400, 0, &cropped_width_,
&cropped_height_, &out_width_,
&out_height_));
@@ -500,7 +500,7 @@
EXPECT_EQ(200, out_height_);
// Request resolution of 3 / 8. Expect adapt down.
- OnOutputFormatRequest(640 * 3 / 8, 400 * 3 / 8, absl::nullopt);
+ OnOutputFormatRequest(640 * 3 / 8, 400 * 3 / 8, std::nullopt);
EXPECT_TRUE(adapter_.AdaptFrameResolution(640, 400, 0, &cropped_width_,
&cropped_height_, &out_width_,
&out_height_));
@@ -510,7 +510,7 @@
EXPECT_EQ(400 * 3 / 8, out_height_);
// Switch back up. Expect adapt.
- OnOutputFormatRequest(320, 200, absl::nullopt);
+ OnOutputFormatRequest(320, 200, std::nullopt);
EXPECT_TRUE(adapter_.AdaptFrameResolution(640, 400, 0, &cropped_width_,
&cropped_height_, &out_width_,
&out_height_));
@@ -520,7 +520,7 @@
EXPECT_EQ(200, out_height_);
// Format request 480x300.
- OnOutputFormatRequest(480, 300, absl::nullopt);
+ OnOutputFormatRequest(480, 300, std::nullopt);
EXPECT_TRUE(adapter_.AdaptFrameResolution(640, 400, 0, &cropped_width_,
&cropped_height_, &out_width_,
&out_height_));
@@ -541,7 +541,7 @@
EXPECT_EQ(720, out_height_);
// Format request for VGA.
- OnOutputFormatRequest(640, 360, absl::nullopt);
+ OnOutputFormatRequest(640, 360, std::nullopt);
EXPECT_TRUE(adapter_.AdaptFrameResolution(1280, 720, 0, &cropped_width_,
&cropped_height_, &out_width_,
&out_height_));
@@ -562,7 +562,7 @@
// And another view request comes in for 640x360, which should have no
// real impact.
- OnOutputFormatRequest(640, 360, absl::nullopt);
+ OnOutputFormatRequest(640, 360, std::nullopt);
EXPECT_TRUE(adapter_.AdaptFrameResolution(640, 360, 0, &cropped_width_,
&cropped_height_, &out_width_,
&out_height_));
@@ -574,7 +574,7 @@
TEST_P(VideoAdapterTest, TestVgaWidth) {
// Requested output format is 640x360.
- OnOutputFormatRequest(640, 360, absl::nullopt);
+ OnOutputFormatRequest(640, 360, std::nullopt);
EXPECT_TRUE(adapter_.AdaptFrameResolution(640, 480, 0, &cropped_width_,
&cropped_height_, &out_width_,
@@ -613,7 +613,7 @@
EXPECT_EQ(720, out_height_);
// Adapt down one step.
- adapter_.OnSinkWants(BuildSinkWants(absl::nullopt, 1280 * 720 - 1,
+ adapter_.OnSinkWants(BuildSinkWants(std::nullopt, 1280 * 720 - 1,
std::numeric_limits<int>::max()));
EXPECT_TRUE(adapter_.AdaptFrameResolution(1280, 720, 0, &cropped_width_,
&cropped_height_, &out_width_,
@@ -624,7 +624,7 @@
EXPECT_EQ(540, out_height_);
// Adapt down one step more.
- adapter_.OnSinkWants(BuildSinkWants(absl::nullopt, 960 * 540 - 1,
+ adapter_.OnSinkWants(BuildSinkWants(std::nullopt, 960 * 540 - 1,
std::numeric_limits<int>::max()));
EXPECT_TRUE(adapter_.AdaptFrameResolution(1280, 720, 0, &cropped_width_,
&cropped_height_, &out_width_,
@@ -635,7 +635,7 @@
EXPECT_EQ(360, out_height_);
// Adapt down one step more.
- adapter_.OnSinkWants(BuildSinkWants(absl::nullopt, 640 * 360 - 1,
+ adapter_.OnSinkWants(BuildSinkWants(std::nullopt, 640 * 360 - 1,
std::numeric_limits<int>::max()));
EXPECT_TRUE(adapter_.AdaptFrameResolution(1280, 720, 0, &cropped_width_,
&cropped_height_, &out_width_,
@@ -689,7 +689,7 @@
EXPECT_EQ(720, out_height_);
adapter_.OnSinkWants(
- BuildSinkWants(absl::nullopt, 0, std::numeric_limits<int>::max()));
+ BuildSinkWants(std::nullopt, 0, std::numeric_limits<int>::max()));
EXPECT_FALSE(adapter_.AdaptFrameResolution(1280, 720, 0, &cropped_width_,
&cropped_height_, &out_width_,
&out_height_));
@@ -697,7 +697,7 @@
TEST_P(VideoAdapterTest, TestOnResolutionRequestInLargeSteps) {
// Large step down.
- adapter_.OnSinkWants(BuildSinkWants(absl::nullopt, 640 * 360 - 1,
+ adapter_.OnSinkWants(BuildSinkWants(std::nullopt, 640 * 360 - 1,
std::numeric_limits<int>::max()));
EXPECT_TRUE(adapter_.AdaptFrameResolution(1280, 720, 0, &cropped_width_,
&cropped_height_, &out_width_,
@@ -720,7 +720,7 @@
}
TEST_P(VideoAdapterTest, TestOnOutputFormatRequestCapsMaxResolution) {
- adapter_.OnSinkWants(BuildSinkWants(absl::nullopt, 640 * 360 - 1,
+ adapter_.OnSinkWants(BuildSinkWants(std::nullopt, 640 * 360 - 1,
std::numeric_limits<int>::max()));
EXPECT_TRUE(adapter_.AdaptFrameResolution(1280, 720, 0, &cropped_width_,
&cropped_height_, &out_width_,
@@ -730,7 +730,7 @@
EXPECT_EQ(480, out_width_);
EXPECT_EQ(270, out_height_);
- OnOutputFormatRequest(640, 360, absl::nullopt);
+ OnOutputFormatRequest(640, 360, std::nullopt);
EXPECT_TRUE(adapter_.AdaptFrameResolution(1280, 720, 0, &cropped_width_,
&cropped_height_, &out_width_,
&out_height_));
@@ -739,8 +739,8 @@
EXPECT_EQ(480, out_width_);
EXPECT_EQ(270, out_height_);
- adapter_.OnSinkWants(BuildSinkWants(absl::nullopt, 960 * 720,
- std::numeric_limits<int>::max()));
+ adapter_.OnSinkWants(
+ BuildSinkWants(std::nullopt, 960 * 720, std::numeric_limits<int>::max()));
EXPECT_TRUE(adapter_.AdaptFrameResolution(1280, 720, 0, &cropped_width_,
&cropped_height_, &out_width_,
&out_height_));
@@ -759,7 +759,7 @@
EXPECT_EQ(1280, out_width_);
EXPECT_EQ(720, out_height_);
- adapter_.OnSinkWants(BuildSinkWants(absl::nullopt, 640 * 360 - 1,
+ adapter_.OnSinkWants(BuildSinkWants(std::nullopt, 640 * 360 - 1,
std::numeric_limits<int>::max()));
EXPECT_TRUE(adapter_.AdaptFrameResolution(1280, 720, 0, &cropped_width_,
&cropped_height_, &out_width_,
@@ -769,7 +769,7 @@
EXPECT_EQ(480, out_width_);
EXPECT_EQ(270, out_height_);
- adapter_.OnSinkWants(BuildSinkWants(absl::nullopt,
+ adapter_.OnSinkWants(BuildSinkWants(std::nullopt,
std::numeric_limits<int>::max(),
std::numeric_limits<int>::max()));
EXPECT_TRUE(adapter_.AdaptFrameResolution(1280, 720, 0, &cropped_width_,
@@ -790,7 +790,7 @@
EXPECT_EQ(1280, out_width_);
EXPECT_EQ(720, out_height_);
- adapter_.OnOutputFormatRequest(absl::nullopt, 640 * 360 - 1, absl::nullopt);
+ adapter_.OnOutputFormatRequest(std::nullopt, 640 * 360 - 1, std::nullopt);
EXPECT_TRUE(adapter_.AdaptFrameResolution(1280, 720, 0, &cropped_width_,
&cropped_height_, &out_width_,
&out_height_));
@@ -799,7 +799,7 @@
EXPECT_EQ(480, out_width_);
EXPECT_EQ(270, out_height_);
- adapter_.OnOutputFormatRequest(absl::nullopt, absl::nullopt, absl::nullopt);
+ adapter_.OnOutputFormatRequest(std::nullopt, std::nullopt, std::nullopt);
EXPECT_TRUE(adapter_.AdaptFrameResolution(1280, 720, 0, &cropped_width_,
&cropped_height_, &out_width_,
&out_height_));
@@ -819,7 +819,7 @@
EXPECT_GT(dropped_frames, 0);
// Reset frame rate.
- OnOutputFormatRequest(kWidth, kHeight, absl::nullopt);
+ OnOutputFormatRequest(kWidth, kHeight, std::nullopt);
for (int i = 0; i < 20; ++i)
adapter_wrapper_->AdaptFrame(frame_source_->GetFrame());
@@ -830,8 +830,8 @@
TEST_P(VideoAdapterTest, RequestAspectRatio) {
// Request aspect ratio 320/180 (16:9), smaller than input, but no resolution
// limit. Expect cropping but no scaling.
- adapter_.OnOutputFormatRequest(std::make_pair(320, 180), absl::nullopt,
- absl::nullopt);
+ adapter_.OnOutputFormatRequest(std::make_pair(320, 180), std::nullopt,
+ std::nullopt);
EXPECT_TRUE(adapter_.AdaptFrameResolution(640, 400, 0, &cropped_width_,
&cropped_height_, &out_width_,
&out_height_));
@@ -841,7 +841,7 @@
EXPECT_EQ(360, out_height_);
adapter_.OnOutputFormatRequest(std::make_pair(1280, 720), 1280 * 720 - 1,
- absl::nullopt);
+ std::nullopt);
EXPECT_TRUE(adapter_.AdaptFrameResolution(2592, 1944, 0, &cropped_width_,
&cropped_height_, &out_width_,
&out_height_));
@@ -854,7 +854,7 @@
TEST_P(VideoAdapterTest, RequestAspectRatioWithDifferentOrientation) {
// Request 720x1280, higher than input, but aspect 16:9. Orientation should
// not matter, expect cropping but no scaling.
- OnOutputFormatRequest(720, 1280, absl::nullopt);
+ OnOutputFormatRequest(720, 1280, std::nullopt);
EXPECT_TRUE(adapter_.AdaptFrameResolution(640, 400, 0, &cropped_width_,
&cropped_height_, &out_width_,
&out_height_));
@@ -866,8 +866,8 @@
TEST_P(VideoAdapterTest, InvalidAspectRatioIgnored) {
// Request aspect ratio 320/0. Expect no cropping.
- adapter_.OnOutputFormatRequest(std::make_pair(320, 0), absl::nullopt,
- absl::nullopt);
+ adapter_.OnOutputFormatRequest(std::make_pair(320, 0), std::nullopt,
+ std::nullopt);
EXPECT_TRUE(adapter_.AdaptFrameResolution(640, 400, 0, &cropped_width_,
&cropped_height_, &out_width_,
&out_height_));
@@ -879,7 +879,7 @@
TEST_P(VideoAdapterTest, TestCroppingWithResolutionRequest) {
// Ask for 640x360 (16:9 aspect).
- OnOutputFormatRequest(640, 360, absl::nullopt);
+ OnOutputFormatRequest(640, 360, std::nullopt);
// Send 640x480 (4:3 aspect).
EXPECT_TRUE(adapter_.AdaptFrameResolution(640, 480, 0, &cropped_width_,
&cropped_height_, &out_width_,
@@ -891,7 +891,7 @@
EXPECT_EQ(360, out_height_);
// Adapt down one step.
- adapter_.OnSinkWants(BuildSinkWants(absl::nullopt, 640 * 360 - 1,
+ adapter_.OnSinkWants(BuildSinkWants(std::nullopt, 640 * 360 - 1,
std::numeric_limits<int>::max()));
// Expect cropping to 16:9 format and 3/4 scaling.
EXPECT_TRUE(adapter_.AdaptFrameResolution(640, 480, 0, &cropped_width_,
@@ -903,7 +903,7 @@
EXPECT_EQ(270, out_height_);
// Adapt down one step more.
- adapter_.OnSinkWants(BuildSinkWants(absl::nullopt, 480 * 270 - 1,
+ adapter_.OnSinkWants(BuildSinkWants(std::nullopt, 480 * 270 - 1,
std::numeric_limits<int>::max()));
// Expect cropping to 16:9 format and 1/2 scaling.
EXPECT_TRUE(adapter_.AdaptFrameResolution(640, 480, 0, &cropped_width_,
@@ -953,9 +953,8 @@
TEST_P(VideoAdapterTest, TestCroppingOddResolution) {
// Ask for 640x360 (16:9 aspect), with 3/16 scaling.
- OnOutputFormatRequest(640, 360, absl::nullopt);
- adapter_.OnSinkWants(BuildSinkWants(absl::nullopt,
- 640 * 360 * 3 / 16 * 3 / 16,
+ OnOutputFormatRequest(640, 360, std::nullopt);
+ adapter_.OnSinkWants(BuildSinkWants(std::nullopt, 640 * 360 * 3 / 16 * 3 / 16,
std::numeric_limits<int>::max()));
// Send 640x480 (4:3 aspect).
@@ -975,8 +974,8 @@
// Ask for 1920x1080 (16:9 aspect), with 1/16 scaling.
const int w = 1920;
const int h = 1080;
- OnOutputFormatRequest(w, h, absl::nullopt);
- adapter_.OnSinkWants(BuildSinkWants(absl::nullopt, w * h * 1 / 16 * 1 / 16,
+ OnOutputFormatRequest(w, h, std::nullopt);
+ adapter_.OnSinkWants(BuildSinkWants(std::nullopt, w * h * 1 / 16 * 1 / 16,
std::numeric_limits<int>::max()));
// Send 1920x1080 (16:9 aspect).
@@ -1018,7 +1017,7 @@
&cropped_width_, &cropped_height_,
&out_width_, &out_height_));
- adapter_.OnSinkWants(BuildSinkWants(absl::nullopt, 640 * 480 - 1,
+ adapter_.OnSinkWants(BuildSinkWants(std::nullopt, 640 * 480 - 1,
std::numeric_limits<int>::max()));
// Still expect all frames to be dropped
@@ -1100,7 +1099,7 @@
TEST_P(VideoAdapterTest, AdaptResolutionInStepsFirst3_4) {
const int kWidth = 1280;
const int kHeight = 720;
- OnOutputFormatRequest(kWidth, kHeight, absl::nullopt); // 16:9 aspect.
+ OnOutputFormatRequest(kWidth, kHeight, std::nullopt); // 16:9 aspect.
// Scale factors: 3/4, 2/3, 3/4, 2/3, ...
// Scale : 3/4, 1/2, 3/8, 1/4, 3/16, 1/8.
@@ -1112,7 +1111,7 @@
for (size_t i = 0; i < arraysize(kExpectedWidths); ++i) {
// Adapt down one step.
- adapter_.OnSinkWants(BuildSinkWants(absl::nullopt,
+ adapter_.OnSinkWants(BuildSinkWants(std::nullopt,
request_width * request_height - 1,
std::numeric_limits<int>::max()));
EXPECT_TRUE(adapter_.AdaptFrameResolution(kWidth, kHeight, 0,
@@ -1128,7 +1127,7 @@
TEST_P(VideoAdapterTest, AdaptResolutionInStepsFirst2_3) {
const int kWidth = 1920;
const int kHeight = 1080;
- OnOutputFormatRequest(kWidth, kHeight, absl::nullopt); // 16:9 aspect.
+ OnOutputFormatRequest(kWidth, kHeight, std::nullopt); // 16:9 aspect.
// Scale factors: 2/3, 3/4, 2/3, 3/4, ...
// Scale: 2/3, 1/2, 1/3, 1/4, 1/6, 1/8, 1/12.
@@ -1140,7 +1139,7 @@
for (size_t i = 0; i < arraysize(kExpectedWidths); ++i) {
// Adapt down one step.
- adapter_.OnSinkWants(BuildSinkWants(absl::nullopt,
+ adapter_.OnSinkWants(BuildSinkWants(std::nullopt,
request_width * request_height - 1,
std::numeric_limits<int>::max()));
EXPECT_TRUE(adapter_.AdaptFrameResolution(kWidth, kHeight, 0,
@@ -1156,7 +1155,7 @@
TEST_P(VideoAdapterTest, AdaptResolutionInStepsFirst2x2_3) {
const int kWidth = 1440;
const int kHeight = 1080;
- OnOutputFormatRequest(kWidth, kHeight, absl::nullopt); // 4:3 aspect.
+ OnOutputFormatRequest(kWidth, kHeight, std::nullopt); // 4:3 aspect.
// Scale factors: 2/3, 2/3, 3/4, 2/3, 3/4, ...
// Scale : 2/3, 4/9, 1/3, 2/9, 1/6, 1/9, 1/12, 1/18, 1/24, 1/36.
@@ -1168,7 +1167,7 @@
for (size_t i = 0; i < arraysize(kExpectedWidths); ++i) {
// Adapt down one step.
- adapter_.OnSinkWants(BuildSinkWants(absl::nullopt,
+ adapter_.OnSinkWants(BuildSinkWants(std::nullopt,
request_width * request_height - 1,
std::numeric_limits<int>::max()));
EXPECT_TRUE(adapter_.AdaptFrameResolution(kWidth, kHeight, 0,
@@ -1194,7 +1193,7 @@
int frame_num = 1;
for (const int sink_alignment : {2, 3, 4, 5}) {
adapter_.OnSinkWants(
- BuildSinkWants(absl::nullopt, std::numeric_limits<int>::max(),
+ BuildSinkWants(std::nullopt, std::numeric_limits<int>::max(),
std::numeric_limits<int>::max(), sink_alignment));
EXPECT_TRUE(adapter_.AdaptFrameResolution(
kSourceWidth, kSourceHeight,
@@ -1240,7 +1239,7 @@
// New API inactive, old API inactive, use OnOutputFormatRequest.
OnOutputFormatRequest(640, 360, kDefaultFps);
adapter_.OnSinkWants(
- BuildSinkWants(absl::nullopt,
+ BuildSinkWants(std::nullopt,
/* any_active_without_requested_resolution= */ false));
EXPECT_THAT(
@@ -1272,7 +1271,7 @@
// Disable new API => fallback to last OnOutputFormatRequest.
adapter_.OnSinkWants(
- BuildSinkWants(absl::nullopt,
+ BuildSinkWants(std::nullopt,
/* any_active_without_requested_resolution= */ false));
EXPECT_THAT(
@@ -1318,7 +1317,7 @@
OnOutputFormatRequest(kRequestedWidth, kRequestedHeight, kRequestedFramerate);
adapter_.OnSinkWants(BuildSinkWants(
- absl::nullopt, std::numeric_limits<int>::max(),
+ std::nullopt, std::numeric_limits<int>::max(),
std::numeric_limits<int>::max(), kSinkResolutionAlignment));
EXPECT_TRUE(adapter_.AdaptFrameResolution(
kSourceWidth, kSourceHeight, /*in_timestamp_ns=*/0, &cropped_width_,
diff --git a/media/base/video_broadcaster.cc b/media/base/video_broadcaster.cc
index 43c1773..b34b8db 100644
--- a/media/base/video_broadcaster.cc
+++ b/media/base/video_broadcaster.cc
@@ -11,9 +11,9 @@
#include "media/base/video_broadcaster.h"
#include <algorithm>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/video/i420_buffer.h"
#include "api/video/video_rotation.h"
#include "media/base/video_common.h"
diff --git a/media/base/video_broadcaster.h b/media/base/video_broadcaster.h
index c253d44..6b37cd7 100644
--- a/media/base/video_broadcaster.h
+++ b/media/base/video_broadcaster.h
@@ -73,7 +73,7 @@
rtc::scoped_refptr<webrtc::VideoFrameBuffer> black_frame_buffer_;
bool previous_frame_sent_to_all_sinks_ RTC_GUARDED_BY(sinks_and_wants_lock_) =
true;
- absl::optional<webrtc::VideoTrackSourceConstraints> last_constraints_
+ std::optional<webrtc::VideoTrackSourceConstraints> last_constraints_
RTC_GUARDED_BY(sinks_and_wants_lock_);
};
diff --git a/media/base/video_broadcaster_unittest.cc b/media/base/video_broadcaster_unittest.cc
index bb80c11..7ab122d 100644
--- a/media/base/video_broadcaster_unittest.cc
+++ b/media/base/video_broadcaster_unittest.cc
@@ -11,8 +11,8 @@
#include "media/base/video_broadcaster.h"
#include <limits>
+#include <optional>
-#include "absl/types/optional.h"
#include "api/video/i420_buffer.h"
#include "api/video/video_frame.h"
#include "api/video/video_rotation.h"
@@ -297,21 +297,21 @@
EXPECT_CALL(sink, OnConstraintsChanged(AllOf(
Field(&webrtc::VideoTrackSourceConstraints::min_fps,
- Eq(absl::nullopt)),
+ Eq(std::nullopt)),
Field(&webrtc::VideoTrackSourceConstraints::max_fps,
- Eq(absl::nullopt)))));
+ Eq(std::nullopt)))));
broadcaster.ProcessConstraints(
- webrtc::VideoTrackSourceConstraints{absl::nullopt, absl::nullopt});
+ webrtc::VideoTrackSourceConstraints{std::nullopt, std::nullopt});
Mock::VerifyAndClearExpectations(&sink);
EXPECT_CALL(
sink,
OnConstraintsChanged(AllOf(
Field(&webrtc::VideoTrackSourceConstraints::min_fps,
- Eq(absl::nullopt)),
+ Eq(std::nullopt)),
Field(&webrtc::VideoTrackSourceConstraints::max_fps, Optional(3)))));
broadcaster.ProcessConstraints(
- webrtc::VideoTrackSourceConstraints{absl::nullopt, 3});
+ webrtc::VideoTrackSourceConstraints{std::nullopt, 3});
Mock::VerifyAndClearExpectations(&sink);
EXPECT_CALL(
@@ -319,9 +319,9 @@
OnConstraintsChanged(AllOf(
Field(&webrtc::VideoTrackSourceConstraints::min_fps, Optional(2)),
Field(&webrtc::VideoTrackSourceConstraints::max_fps,
- Eq(absl::nullopt)))));
+ Eq(std::nullopt)))));
broadcaster.ProcessConstraints(
- webrtc::VideoTrackSourceConstraints{2, absl::nullopt});
+ webrtc::VideoTrackSourceConstraints{2, std::nullopt});
Mock::VerifyAndClearExpectations(&sink);
EXPECT_CALL(
diff --git a/media/engine/fake_webrtc_call.h b/media/engine/fake_webrtc_call.h
index 7b76e10..64baab7 100644
--- a/media/engine/fake_webrtc_call.h
+++ b/media/engine/fake_webrtc_call.h
@@ -234,7 +234,7 @@
bool framerate_scaling_enabled_;
rtc::VideoSourceInterface<webrtc::VideoFrame>* source_;
int num_swapped_frames_;
- absl::optional<webrtc::VideoFrame> last_frame_;
+ std::optional<webrtc::VideoFrame> last_frame_;
webrtc::VideoSendStream::Stats stats_;
int num_encoder_reconfigurations_ = 0;
std::vector<std::string> keyframes_requested_by_rid_;
diff --git a/media/engine/fake_webrtc_video_engine.cc b/media/engine/fake_webrtc_video_engine.cc
index cb70268..a701a95 100644
--- a/media/engine/fake_webrtc_video_engine.cc
+++ b/media/engine/fake_webrtc_video_engine.cc
@@ -32,7 +32,7 @@
bool IsScalabilityModeSupported(
const std::vector<webrtc::SdpVideoFormat>& formats,
- absl::optional<std::string> scalability_mode) {
+ std::optional<std::string> scalability_mode) {
if (!scalability_mode.has_value()) {
return true;
}
@@ -222,7 +222,7 @@
webrtc::VideoEncoderFactory::CodecSupport
FakeWebRtcVideoEncoderFactory::QueryCodecSupport(
const webrtc::SdpVideoFormat& format,
- absl::optional<std::string> scalability_mode) const {
+ std::optional<std::string> scalability_mode) const {
std::vector<webrtc::SdpVideoFormat> supported_formats;
for (const auto& f : formats_) {
if (format.IsSameCodec(f))
diff --git a/media/engine/fake_webrtc_video_engine.h b/media/engine/fake_webrtc_video_engine.h
index bf3eed1..237085a 100644
--- a/media/engine/fake_webrtc_video_engine.h
+++ b/media/engine/fake_webrtc_video_engine.h
@@ -118,7 +118,7 @@
std::vector<webrtc::SdpVideoFormat> GetSupportedFormats() const override;
webrtc::VideoEncoderFactory::CodecSupport QueryCodecSupport(
const webrtc::SdpVideoFormat& format,
- absl::optional<std::string> scalability_mode) const override;
+ std::optional<std::string> scalability_mode) const override;
std::unique_ptr<webrtc::VideoEncoder> Create(
const webrtc::Environment& env,
const webrtc::SdpVideoFormat& format) override;
diff --git a/media/engine/internal_encoder_factory.cc b/media/engine/internal_encoder_factory.cc
index 398926a..95ac3d5 100644
--- a/media/engine/internal_encoder_factory.cc
+++ b/media/engine/internal_encoder_factory.cc
@@ -11,10 +11,10 @@
#include "media/engine/internal_encoder_factory.h"
#include <memory>
+#include <optional>
#include <string>
#include <vector>
-#include "absl/types/optional.h"
#include "api/environment/environment.h"
#include "api/video_codecs/sdp_video_format.h"
#include "api/video_codecs/video_encoder_factory.h"
@@ -57,7 +57,7 @@
VideoEncoderFactory::CodecSupport InternalEncoderFactory::QueryCodecSupport(
const SdpVideoFormat& format,
- absl::optional<std::string> scalability_mode) const {
+ std::optional<std::string> scalability_mode) const {
auto original_format =
FuzzyMatchSdpVideoFormat(Factory().GetSupportedFormats(), format);
return original_format
diff --git a/media/engine/internal_encoder_factory.h b/media/engine/internal_encoder_factory.h
index 1d79d3c..ec04f9f 100644
--- a/media/engine/internal_encoder_factory.h
+++ b/media/engine/internal_encoder_factory.h
@@ -12,10 +12,10 @@
#define MEDIA_ENGINE_INTERNAL_ENCODER_FACTORY_H_
#include <memory>
+#include <optional>
#include <string>
#include <vector>
-#include "absl/types/optional.h"
#include "api/environment/environment.h"
#include "api/video_codecs/sdp_video_format.h"
#include "api/video_codecs/video_encoder_factory.h"
@@ -27,7 +27,7 @@
std::vector<SdpVideoFormat> GetSupportedFormats() const override;
CodecSupport QueryCodecSupport(
const SdpVideoFormat& format,
- absl::optional<std::string> scalability_mode) const override;
+ std::optional<std::string> scalability_mode) const override;
std::unique_ptr<VideoEncoder> Create(const Environment& env,
const SdpVideoFormat& format) override;
};
diff --git a/media/engine/internal_encoder_factory_unittest.cc b/media/engine/internal_encoder_factory_unittest.cc
index c13c528..fcd879e 100644
--- a/media/engine/internal_encoder_factory_unittest.cc
+++ b/media/engine/internal_encoder_factory_unittest.cc
@@ -121,17 +121,17 @@
TEST(InternalEncoderFactoryTest, QueryCodecSupportNoScalabilityModeAv1) {
InternalEncoderFactory factory;
EXPECT_THAT(factory.QueryCodecSupport(SdpVideoFormat::AV1Profile0(),
- /*scalability_mode=*/absl::nullopt),
+ /*scalability_mode=*/std::nullopt),
Support(kSupported));
}
TEST(InternalEncoderFactoryTest, QueryCodecSupportNoScalabilityMode) {
InternalEncoderFactory factory;
EXPECT_THAT(factory.QueryCodecSupport(SdpVideoFormat::VP8(),
- /*scalability_mode=*/absl::nullopt),
+ /*scalability_mode=*/std::nullopt),
Support(kSupported));
EXPECT_THAT(factory.QueryCodecSupport(SdpVideoFormat::VP9Profile0(),
- /*scalability_mode=*/absl::nullopt),
+ /*scalability_mode=*/std::nullopt),
Support(kVp9Enabled ? kSupported : kUnsupported));
}
diff --git a/media/engine/payload_type_mapper.cc b/media/engine/payload_type_mapper.cc
index 0f7c00e..9db2281 100644
--- a/media/engine/payload_type_mapper.cc
+++ b/media/engine/payload_type_mapper.cc
@@ -93,7 +93,7 @@
PayloadTypeMapper::~PayloadTypeMapper() = default;
-absl::optional<int> PayloadTypeMapper::GetMappingFor(
+std::optional<int> PayloadTypeMapper::GetMappingFor(
const webrtc::SdpAudioFormat& format) {
auto iter = mappings_.find(format);
if (iter != mappings_.end())
@@ -110,19 +110,19 @@
}
}
- return absl::nullopt;
+ return std::nullopt;
}
-absl::optional<int> PayloadTypeMapper::FindMappingFor(
+std::optional<int> PayloadTypeMapper::FindMappingFor(
const webrtc::SdpAudioFormat& format) const {
auto iter = mappings_.find(format);
if (iter != mappings_.end())
return iter->second;
- return absl::nullopt;
+ return std::nullopt;
}
-absl::optional<Codec> PayloadTypeMapper::ToAudioCodec(
+std::optional<Codec> PayloadTypeMapper::ToAudioCodec(
const webrtc::SdpAudioFormat& format) {
// TODO(ossu): We can safely set bitrate to zero here, since that field is
// not presented in the SDP. It is used to ferry around some target bitrate
@@ -138,7 +138,7 @@
return std::move(codec);
}
- return absl::nullopt;
+ return std::nullopt;
}
bool PayloadTypeMapper::SdpAudioFormatOrdering::operator()(
diff --git a/media/engine/payload_type_mapper.h b/media/engine/payload_type_mapper.h
index 3a79a4d..246b04e 100644
--- a/media/engine/payload_type_mapper.h
+++ b/media/engine/payload_type_mapper.h
@@ -12,9 +12,9 @@
#define MEDIA_ENGINE_PAYLOAD_TYPE_MAPPER_H_
#include <map>
+#include <optional>
#include <set>
-#include "absl/types/optional.h"
#include "api/audio_codecs/audio_format.h"
#include "media/base/codec.h"
@@ -30,16 +30,15 @@
// Finds the current payload type for `format` or assigns a new one, if no
// current mapping exists. Will return an empty value if it was unable to
// create a mapping, i.e. if all dynamic payload type ids have been used up.
- absl::optional<int> GetMappingFor(const webrtc::SdpAudioFormat& format);
+ std::optional<int> GetMappingFor(const webrtc::SdpAudioFormat& format);
// Finds the current payload type for `format`, if any. Returns an empty value
// if no payload type mapping exists for the format.
- absl::optional<int> FindMappingFor(
- const webrtc::SdpAudioFormat& format) const;
+ std::optional<int> FindMappingFor(const webrtc::SdpAudioFormat& format) const;
// Like GetMappingFor, but fills in an AudioCodec structure with the necessary
// information instead.
- absl::optional<Codec> ToAudioCodec(const webrtc::SdpAudioFormat& format);
+ std::optional<Codec> ToAudioCodec(const webrtc::SdpAudioFormat& format);
private:
struct SdpAudioFormatOrdering {
diff --git a/media/engine/simulcast_encoder_adapter.cc b/media/engine/simulcast_encoder_adapter.cc
index 9bdb4d5..db36b73 100644
--- a/media/engine/simulcast_encoder_adapter.cc
+++ b/media/engine/simulcast_encoder_adapter.cc
@@ -17,6 +17,7 @@
#include <cstdint>
#include <iterator>
#include <memory>
+#include <optional>
#include <string>
#include <tuple>
#include <utility>
@@ -24,7 +25,6 @@
#include "absl/algorithm/container.h"
#include "absl/base/nullability.h"
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/environment/environment.h"
#include "api/fec_controller_override.h"
@@ -809,7 +809,7 @@
// By default, `scalability_mode` comes from SimulcastStream when
// SimulcastEncoderAdapter is used. This allows multiple encodings of L1Tx,
// but SimulcastStream currently does not support multiple spatial layers.
- absl::optional<ScalabilityMode> scalability_mode =
+ std::optional<ScalabilityMode> scalability_mode =
stream_params.GetScalabilityMode();
// To support the full set of scalability modes in the event that this is the
// only active encoding, prefer VideoCodec::GetScalabilityMode() if all other
@@ -907,7 +907,7 @@
encoder_info.requested_resolution_alignment = 1;
encoder_info.apply_alignment_to_all_simulcast_layers = false;
encoder_info.supports_native_handle = true;
- encoder_info.scaling_settings.thresholds = absl::nullopt;
+ encoder_info.scaling_settings.thresholds = std::nullopt;
if (stream_contexts_.empty()) {
// GetEncoderInfo queried before InitEncode. Only alignment info is needed
diff --git a/media/engine/simulcast_encoder_adapter.h b/media/engine/simulcast_encoder_adapter.h
index d713043..92d10a4 100644
--- a/media/engine/simulcast_encoder_adapter.h
+++ b/media/engine/simulcast_encoder_adapter.h
@@ -15,13 +15,13 @@
#include <atomic>
#include <list>
#include <memory>
+#include <optional>
#include <stack>
#include <string>
#include <utility>
#include <vector>
#include "absl/base/nullability.h"
-#include "absl/types/optional.h"
#include "api/environment/environment.h"
#include "api/fec_controller_override.h"
#include "api/field_trials_view.h"
@@ -125,10 +125,10 @@
void set_is_keyframe_needed() { is_keyframe_needed_ = true; }
bool is_paused() const { return is_paused_; }
void set_is_paused(bool is_paused) { is_paused_ = is_paused; }
- absl::optional<double> target_fps() const {
+ std::optional<double> target_fps() const {
return framerate_controller_ == nullptr
- ? absl::nullopt
- : absl::optional<double>(
+ ? std::nullopt
+ : std::optional<double>(
framerate_controller_->GetMaxFramerate());
}
diff --git a/media/engine/simulcast_encoder_adapter_unittest.cc b/media/engine/simulcast_encoder_adapter_unittest.cc
index 5d38633..a52ac96 100644
--- a/media/engine/simulcast_encoder_adapter_unittest.cc
+++ b/media/engine/simulcast_encoder_adapter_unittest.cc
@@ -333,7 +333,7 @@
video_format_ = video_format;
}
- void set_is_qp_trusted(absl::optional<bool> is_qp_trusted) {
+ void set_is_qp_trusted(std::optional<bool> is_qp_trusted) {
is_qp_trusted_ = is_qp_trusted;
}
@@ -360,7 +360,7 @@
VideoEncoder::RateControlParameters last_set_rates_;
FramerateFractions fps_allocation_;
bool supports_simulcast_ = false;
- absl::optional<bool> is_qp_trusted_;
+ std::optional<bool> is_qp_trusted_;
SdpVideoFormat video_format_;
std::vector<VideoEncoder::ResolutionBitrateLimits> resolution_bitrate_limits;
@@ -465,9 +465,9 @@
env_, use_fallback_factory_,
SdpVideoFormat("VP8", sdp_video_parameters_));
adapter_ = helper_->CreateMockEncoderAdapter();
- last_encoded_image_width_ = absl::nullopt;
- last_encoded_image_height_ = absl::nullopt;
- last_encoded_image_simulcast_index_ = absl::nullopt;
+ last_encoded_image_width_ = std::nullopt;
+ last_encoded_image_height_ = std::nullopt;
+ last_encoded_image_simulcast_index_ = std::nullopt;
}
void ReSetUp() {
@@ -489,9 +489,9 @@
return Result(Result::OK, encoded_image.RtpTimestamp());
}
- bool GetLastEncodedImageInfo(absl::optional<int>* out_width,
- absl::optional<int>* out_height,
- absl::optional<int>* out_simulcast_index) {
+ bool GetLastEncodedImageInfo(std::optional<int>* out_width,
+ std::optional<int>* out_height,
+ std::optional<int>* out_simulcast_index) {
if (!last_encoded_image_width_.has_value()) {
return false;
}
@@ -621,9 +621,9 @@
std::unique_ptr<TestSimulcastEncoderAdapterFakeHelper> helper_;
std::unique_ptr<VideoEncoder> adapter_;
VideoCodec codec_;
- absl::optional<int> last_encoded_image_width_;
- absl::optional<int> last_encoded_image_height_;
- absl::optional<int> last_encoded_image_simulcast_index_;
+ std::optional<int> last_encoded_image_width_;
+ std::optional<int> last_encoded_image_height_;
+ std::optional<int> last_encoded_image_simulcast_index_;
std::unique_ptr<SimulcastRateAllocator> rate_allocator_;
bool use_fallback_factory_;
CodecParameterMap sdp_video_parameters_;
@@ -672,9 +672,9 @@
std::vector<MockVideoEncoder*> encoders = helper_->factory()->encoders();
ASSERT_EQ(3u, encoders.size());
encoders[0]->SendEncodedImage(1152, 704);
- absl::optional<int> width;
- absl::optional<int> height;
- absl::optional<int> simulcast_index;
+ std::optional<int> width;
+ std::optional<int> height;
+ std::optional<int> simulcast_index;
EXPECT_TRUE(GetLastEncodedImageInfo(&width, &height, &simulcast_index));
ASSERT_TRUE(width.has_value());
EXPECT_EQ(1152, width.value());
@@ -899,9 +899,9 @@
std::vector<MockVideoEncoder*> encoders = helper_->factory()->encoders();
ASSERT_EQ(3u, encoders.size());
encoders[0]->SendEncodedImage(1152, 704);
- absl::optional<int> width;
- absl::optional<int> height;
- absl::optional<int> simulcast_index;
+ std::optional<int> width;
+ std::optional<int> height;
+ std::optional<int> simulcast_index;
EXPECT_TRUE(GetLastEncodedImageInfo(&width, &height, &simulcast_index));
// SEA doesn't intercept frame encode complete callback for the lowest stream.
EXPECT_FALSE(simulcast_index.has_value());
diff --git a/media/engine/webrtc_video_engine.cc b/media/engine/webrtc_video_engine.cc
index fcd7b23..ec51433 100644
--- a/media/engine/webrtc_video_engine.cc
+++ b/media/engine/webrtc_video_engine.cc
@@ -15,6 +15,7 @@
#include <algorithm>
#include <cstdint>
#include <initializer_list>
+#include <optional>
#include <set>
#include <string>
#include <type_traits>
@@ -24,7 +25,6 @@
#include "absl/container/inlined_vector.h"
#include "absl/functional/bind_front.h"
#include "absl/strings/match.h"
-#include "absl/types/optional.h"
#include "api/make_ref_counted.h"
#include "api/media_stream_interface.h"
#include "api/media_types.h"
@@ -422,19 +422,19 @@
return res;
}
-absl::optional<int> NumSpatialLayersFromEncoding(
+std::optional<int> NumSpatialLayersFromEncoding(
const webrtc::RtpParameters& rtp_parameters,
size_t idx) {
if (idx >= rtp_parameters.encodings.size())
- return absl::nullopt;
+ return std::nullopt;
- absl::optional<webrtc::ScalabilityMode> scalability_mode =
+ std::optional<webrtc::ScalabilityMode> scalability_mode =
webrtc::ScalabilityModeFromString(
rtp_parameters.encodings[idx].scalability_mode.value_or(""));
return scalability_mode
- ? absl::optional<int>(
+ ? std::optional<int>(
ScalabilityModeToNumSpatialLayers(*scalability_mode))
- : absl::nullopt;
+ : std::nullopt;
}
std::map<uint32_t, webrtc::VideoSendStream::StreamStats>
@@ -492,7 +492,7 @@
}
bool IsActiveFromEncodings(
- absl::optional<uint32_t> ssrc,
+ std::optional<uint32_t> ssrc,
const std::vector<webrtc::RtpEncodingParameters>& encodings) {
if (ssrc.has_value()) {
// Report the `active` value of a specific ssrc, or false if an encoding
@@ -574,7 +574,7 @@
std::map<int, int> rtx_time_mapping;
webrtc::UlpfecConfig ulpfec_config;
- absl::optional<int> flexfec_payload_type;
+ std::optional<int> flexfec_payload_type;
for (const Codec& in_codec : codecs) {
const int payload_type = in_codec.id;
@@ -965,7 +965,7 @@
vp9_settings.denoisingOn = codec_default_denoising ? true : denoising;
// Disable automatic resize if more than one spatial layer is requested.
bool vp9_automatic_resize = automatic_resize;
- absl::optional<int> num_spatial_layers =
+ std::optional<int> num_spatial_layers =
NumSpatialLayersFromEncoding(rtp_parameters_, /*idx=*/0);
if (num_spatial_layers && *num_spatial_layers > 1) {
vp9_automatic_resize = false;
@@ -1077,7 +1077,7 @@
codec.flexfec_payload_type = -1;
}
- absl::optional<VideoCodecSettings> force_codec;
+ std::optional<VideoCodecSettings> force_codec;
if (!send_streams_.empty()) {
// Since we do not support mixed-codec simulcast yet,
// all send streams must have the same codec value.
@@ -1104,7 +1104,7 @@
if (negotiated_codecs_ != negotiated_codecs) {
if (negotiated_codecs.empty()) {
- changed_params->send_codec = absl::nullopt;
+ changed_params->send_codec = std::nullopt;
} else if (force_codec) {
changed_params->send_codec = force_codec;
} else if (send_codec() != negotiated_codecs.front()) {
@@ -1122,7 +1122,7 @@
call_->trials());
if (send_rtp_extensions_ != filtered_extensions) {
changed_params->rtp_header_extensions =
- absl::optional<std::vector<webrtc::RtpExtension>>(filtered_extensions);
+ std::optional<std::vector<webrtc::RtpExtension>>(filtered_extensions);
}
if (params.mid != send_params_.mid) {
@@ -1404,11 +1404,11 @@
return it->second->SetRtpParameters(parameters, std::move(callback));
}
-absl::optional<Codec> WebRtcVideoSendChannel::GetSendCodec() const {
+std::optional<Codec> WebRtcVideoSendChannel::GetSendCodec() const {
RTC_DCHECK_RUN_ON(&thread_checker_);
if (!send_codec()) {
RTC_LOG(LS_VERBOSE) << "GetSendCodec: No send codec set.";
- return absl::nullopt;
+ return std::nullopt;
}
return send_codec()->codec;
}
@@ -1703,7 +1703,7 @@
webrtc::VideoSendStream::Config config,
const VideoOptions& options,
int max_bitrate_bps,
- const absl::optional<VideoCodecSettings>& codec_settings)
+ const std::optional<VideoCodecSettings>& codec_settings)
: config(std::move(config)),
options(options),
max_bitrate_bps(max_bitrate_bps),
@@ -1717,8 +1717,8 @@
const VideoOptions& options,
bool enable_cpu_overuse_detection,
int max_bitrate_bps,
- const absl::optional<VideoCodecSettings>& codec_settings,
- const absl::optional<std::vector<webrtc::RtpExtension>>& rtp_extensions,
+ const std::optional<VideoCodecSettings>& codec_settings,
+ const std::optional<std::vector<webrtc::RtpExtension>>& rtp_extensions,
// TODO(deadbeef): Don't duplicate information between send_params,
// rtp_extensions, options, etc.
const VideoSenderParameters& send_params)
@@ -2363,7 +2363,7 @@
common_info.add_ssrc(ssrc);
}
common_info.active =
- IsActiveFromEncodings(absl::nullopt, rtp_parameters_.encodings);
+ IsActiveFromEncodings(std::nullopt, rtp_parameters_.encodings);
common_info.framerate_sent = stats.encode_frame_rate;
common_info.frames_encoded = stats.frames_encoded;
common_info.total_encode_time_ms = stats.total_encode_time_ms;
@@ -2390,7 +2390,7 @@
info.add_ssrc(ssrc);
info.rid = parameters_.config.rtp.GetRidForSsrc(ssrc);
info.active = IsActiveFromEncodings(
- !is_svc ? absl::optional<uint32_t>(ssrc) : absl::nullopt,
+ !is_svc ? std::optional<uint32_t>(ssrc) : std::nullopt,
rtp_parameters_.encodings);
auto stream_stats = pair.second;
RTC_DCHECK_EQ(stream_stats.type,
@@ -2624,7 +2624,7 @@
bool lntf_enabled,
bool nack_enabled,
webrtc::RtcpMode rtcp_mode,
- absl::optional<int> rtx_time) {
+ std::optional<int> rtx_time) {
RTC_DCHECK_RUN_ON(&thread_checker_);
// Update receive feedback parameters from new codec or RTCP mode.
@@ -2722,7 +2722,7 @@
if (NonFlexfecReceiveCodecsHaveChanged(recv_codecs_, mapped_codecs)) {
changed_params->codec_settings =
- absl::optional<std::vector<VideoCodecSettings>>(mapped_codecs);
+ std::optional<std::vector<VideoCodecSettings>>(mapped_codecs);
}
// Handle RTP header extensions.
@@ -2731,7 +2731,7 @@
call_->trials());
if (filtered_extensions != recv_rtp_extensions_) {
changed_params->rtp_header_extensions =
- absl::optional<std::vector<webrtc::RtpExtension>>(filtered_extensions);
+ std::optional<std::vector<webrtc::RtpExtension>>(filtered_extensions);
}
int flexfec_payload_type = mapped_codecs.front().flexfec_payload_type;
@@ -2958,7 +2958,7 @@
RTC_DCHECK_RUN_ON(&thread_checker_);
RTC_LOG(LS_INFO) << "ResetUnsignaledRecvStream.";
unsignaled_stream_params_ = StreamParams();
- last_unsignalled_ssrc_creation_time_ms_ = absl::nullopt;
+ last_unsignalled_ssrc_creation_time_ms_ = std::nullopt;
// Delete any created default streams. This is needed to avoid SSRC collisions
// in Call's RtpDemuxer, in the case that `this` has created a default video
@@ -2975,9 +2975,9 @@
}
}
-absl::optional<uint32_t> WebRtcVideoReceiveChannel::GetUnsignaledSsrc() const {
+std::optional<uint32_t> WebRtcVideoReceiveChannel::GetUnsignaledSsrc() const {
RTC_DCHECK_RUN_ON(&thread_checker_);
- absl::optional<uint32_t> ssrc;
+ std::optional<uint32_t> ssrc;
for (auto it = receive_streams_.begin(); it != receive_streams_.end(); ++it) {
if (it->second->IsDefaultStream()) {
ssrc.emplace(it->first);
@@ -3128,7 +3128,7 @@
if (is_rtx_payload) {
// As we don't support receiving simulcast there can only be one RTX
// stream, which will be associated with unsignaled media stream.
- absl::optional<uint32_t> current_default_ssrc = GetUnsignaledSsrc();
+ std::optional<uint32_t> current_default_ssrc = GetUnsignaledSsrc();
if (current_default_ssrc) {
FindReceiveStream(*current_default_ssrc)->UpdateRtxSsrc(packet.Ssrc());
return true;
@@ -3155,17 +3155,17 @@
}
// RTX SSRC not yet known.
- ReCreateDefaultReceiveStream(packet.Ssrc(), absl::nullopt);
+ ReCreateDefaultReceiveStream(packet.Ssrc(), std::nullopt);
last_unsignalled_ssrc_creation_time_ms_ = rtc::TimeMillis();
return true;
}
void WebRtcVideoReceiveChannel::ReCreateDefaultReceiveStream(
uint32_t ssrc,
- absl::optional<uint32_t> rtx_ssrc) {
+ std::optional<uint32_t> rtx_ssrc) {
RTC_DCHECK_RUN_ON(&thread_checker_);
- absl::optional<uint32_t> default_recv_ssrc = GetUnsignaledSsrc();
+ std::optional<uint32_t> default_recv_ssrc = GetUnsignaledSsrc();
if (default_recv_ssrc) {
RTC_LOG(LS_INFO) << "Destroying old default receive stream for SSRC="
<< ssrc << ".";
@@ -3215,7 +3215,7 @@
bool WebRtcVideoReceiveChannel::SetBaseMinimumPlayoutDelayMs(uint32_t ssrc,
int delay_ms) {
RTC_DCHECK_RUN_ON(&thread_checker_);
- absl::optional<uint32_t> default_ssrc = GetUnsignaledSsrc();
+ std::optional<uint32_t> default_ssrc = GetUnsignaledSsrc();
// SSRC of 0 represents the default receive stream.
if (ssrc == 0) {
@@ -3240,7 +3240,7 @@
}
}
-absl::optional<int> WebRtcVideoReceiveChannel::GetBaseMinimumPlayoutDelayMs(
+std::optional<int> WebRtcVideoReceiveChannel::GetBaseMinimumPlayoutDelayMs(
uint32_t ssrc) const {
RTC_DCHECK_RUN_ON(&thread_checker_);
// SSRC of 0 represents the default receive stream.
@@ -3253,7 +3253,7 @@
return stream->second->GetBaseMinimumPlayoutDelayMs();
} else {
RTC_LOG(LS_ERROR) << "No stream found to get base minimum playout delay";
- return absl::nullopt;
+ return std::nullopt;
}
}
@@ -3444,7 +3444,7 @@
bool lntf_enabled,
bool nack_enabled,
webrtc::RtcpMode rtcp_mode,
- absl::optional<int> rtx_time) {
+ std::optional<int> rtx_time) {
RTC_DCHECK(stream_);
if (config_.rtp.rtcp_mode != rtcp_mode) {
@@ -3520,8 +3520,8 @@
RecreateReceiveStream() {
RTC_DCHECK_RUN_ON(&thread_checker_);
RTC_DCHECK(stream_);
- absl::optional<int> base_minimum_playout_delay_ms;
- absl::optional<webrtc::VideoReceiveStreamInterface::RecordingState>
+ std::optional<int> base_minimum_playout_delay_ms;
+ std::optional<webrtc::VideoReceiveStreamInterface::RecordingState>
recording_state;
if (stream_) {
base_minimum_playout_delay_ms = stream_->GetBaseMinimumPlayoutDelayMs();
@@ -3828,7 +3828,7 @@
WebRtcVideoReceiveChannel::WebRtcVideoReceiveStream*
WebRtcVideoReceiveChannel::FindReceiveStream(uint32_t ssrc) {
if (ssrc == 0) {
- absl::optional<uint32_t> default_ssrc = GetUnsignaledSsrc();
+ std::optional<uint32_t> default_ssrc = GetUnsignaledSsrc();
if (!default_ssrc) {
return nullptr;
}
diff --git a/media/engine/webrtc_video_engine.h b/media/engine/webrtc_video_engine.h
index 14df03a..f5b5063 100644
--- a/media/engine/webrtc_video_engine.h
+++ b/media/engine/webrtc_video_engine.h
@@ -17,6 +17,7 @@
#include <functional>
#include <map>
#include <memory>
+#include <optional>
#include <set>
#include <string>
#include <utility>
@@ -24,7 +25,6 @@
#include "absl/functional/any_invocable.h"
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/call/transport.h"
#include "api/crypto/crypto_options.h"
@@ -148,7 +148,7 @@
webrtc::UlpfecConfig ulpfec;
int flexfec_payload_type; // -1 if absent.
int rtx_payload_type; // -1 if absent.
- absl::optional<int> rtx_time;
+ std::optional<int> rtx_time;
};
class WebRtcVideoSendChannel : public MediaChannelUtil,
@@ -192,7 +192,7 @@
const webrtc::RtpParameters& parameters,
webrtc::SetParametersCallback callback) override;
webrtc::RtpParameters GetRtpSendParameters(uint32_t ssrc) const override;
- absl::optional<Codec> GetSendCodec() const override;
+ std::optional<Codec> GetSendCodec() const override;
bool SetSend(bool send) override;
bool SetVideoSend(
uint32_t ssrc,
@@ -279,10 +279,10 @@
}
return HasNack(send_codec()->codec);
}
- absl::optional<int> SendCodecRtxTime() const override {
+ std::optional<int> SendCodecRtxTime() const override {
RTC_DCHECK_RUN_ON(&thread_checker_);
if (!send_codec()) {
- return absl::nullopt;
+ return std::nullopt;
}
return send_codec()->rtx_time;
}
@@ -290,14 +290,14 @@
private:
struct ChangedSenderParameters {
// These optionals are unset if not changed.
- absl::optional<VideoCodecSettings> send_codec;
- absl::optional<std::vector<VideoCodecSettings>> negotiated_codecs;
- absl::optional<std::vector<webrtc::RtpExtension>> rtp_header_extensions;
- absl::optional<std::string> mid;
- absl::optional<bool> extmap_allow_mixed;
- absl::optional<int> max_bandwidth_bps;
- absl::optional<bool> conference_mode;
- absl::optional<webrtc::RtcpMode> rtcp_mode;
+ std::optional<VideoCodecSettings> send_codec;
+ std::optional<std::vector<VideoCodecSettings>> negotiated_codecs;
+ std::optional<std::vector<webrtc::RtpExtension>> rtp_header_extensions;
+ std::optional<std::string> mid;
+ std::optional<bool> extmap_allow_mixed;
+ std::optional<int> max_bandwidth_bps;
+ std::optional<bool> conference_mode;
+ std::optional<webrtc::RtcpMode> rtcp_mode;
};
bool GetChangedSenderParameters(const VideoSenderParameters& params,
@@ -326,8 +326,8 @@
const VideoOptions& options,
bool enable_cpu_overuse_detection,
int max_bitrate_bps,
- const absl::optional<VideoCodecSettings>& codec_settings,
- const absl::optional<std::vector<webrtc::RtpExtension>>& rtp_extensions,
+ const std::optional<VideoCodecSettings>& codec_settings,
+ const std::optional<std::vector<webrtc::RtpExtension>>& rtp_extensions,
const VideoSenderParameters& send_params);
~WebRtcVideoSendStream();
@@ -374,12 +374,12 @@
webrtc::VideoSendStream::Config config,
const VideoOptions& options,
int max_bitrate_bps,
- const absl::optional<VideoCodecSettings>& codec_settings);
+ const std::optional<VideoCodecSettings>& codec_settings);
webrtc::VideoSendStream::Config config;
VideoOptions options;
int max_bitrate_bps;
bool conference_mode;
- absl::optional<VideoCodecSettings> codec_settings;
+ std::optional<VideoCodecSettings> codec_settings;
// Sent resolutions + bitrates etc. by the underlying VideoSendStream,
// typically changes when setting a new resolution or reconfiguring
// bitrates.
@@ -449,8 +449,8 @@
// that a receive channel does not touch the send codec directly.
// Can go away once these are different classes.
// TODO(bugs.webrtc.org/13931): Remove this function
- absl::optional<VideoCodecSettings>& send_codec() { return send_codec_; }
- const absl::optional<VideoCodecSettings>& send_codec() const {
+ std::optional<VideoCodecSettings>& send_codec() { return send_codec_; }
+ const std::optional<VideoCodecSettings>& send_codec() const {
return send_codec_;
}
webrtc::TaskQueueBase* const worker_thread_;
@@ -491,13 +491,12 @@
// is a risk of receiving ssrcs for other, recently added m= sections.
uint32_t demuxer_criteria_id_ RTC_GUARDED_BY(thread_checker_) = 0;
uint32_t demuxer_criteria_completed_id_ RTC_GUARDED_BY(thread_checker_) = 0;
- absl::optional<int64_t> last_unsignalled_ssrc_creation_time_ms_
+ std::optional<int64_t> last_unsignalled_ssrc_creation_time_ms_
RTC_GUARDED_BY(thread_checker_);
std::set<uint32_t> send_ssrcs_ RTC_GUARDED_BY(thread_checker_);
std::set<uint32_t> receive_ssrcs_ RTC_GUARDED_BY(thread_checker_);
- absl::optional<VideoCodecSettings> send_codec_
- RTC_GUARDED_BY(thread_checker_);
+ std::optional<VideoCodecSettings> send_codec_ RTC_GUARDED_BY(thread_checker_);
std::vector<VideoCodecSettings> negotiated_codecs_
RTC_GUARDED_BY(thread_checker_);
@@ -587,7 +586,7 @@
}
bool RemoveRecvStream(uint32_t ssrc) override;
void ResetUnsignaledRecvStream() override;
- absl::optional<uint32_t> GetUnsignaledSsrc() const override;
+ std::optional<uint32_t> GetUnsignaledSsrc() const override;
void OnDemuxerCriteriaUpdatePending() override;
void OnDemuxerCriteriaUpdateComplete() override;
bool SetSink(uint32_t ssrc,
@@ -598,8 +597,7 @@
void OnPacketReceived(const webrtc::RtpPacketReceived& packet) override;
bool SetBaseMinimumPlayoutDelayMs(uint32_t ssrc, int delay_ms) override;
- absl::optional<int> GetBaseMinimumPlayoutDelayMs(
- uint32_t ssrc) const override;
+ std::optional<int> GetBaseMinimumPlayoutDelayMs(uint32_t ssrc) const override;
// Choose one of the available SSRCs (or default if none) as the current
// receiver report SSRC.
@@ -627,18 +625,18 @@
void SetReceiverFeedbackParameters(bool lntf_enabled,
bool nack_enabled,
webrtc::RtcpMode rtcp_mode,
- absl::optional<int> rtx_time) override;
+ std::optional<int> rtx_time) override;
private:
class WebRtcVideoReceiveStream;
struct ChangedReceiverParameters {
// These optionals are unset if not changed.
- absl::optional<std::vector<VideoCodecSettings>> codec_settings;
- absl::optional<std::vector<webrtc::RtpExtension>> rtp_header_extensions;
+ std::optional<std::vector<VideoCodecSettings>> codec_settings;
+ std::optional<std::vector<webrtc::RtpExtension>> rtp_header_extensions;
// Keep track of the FlexFEC payload type separately from `codec_settings`.
// This allows us to recreate the FlexfecReceiveStream separately from the
// VideoReceiveStreamInterface when the FlexFEC payload type is changed.
- absl::optional<int> flexfec_payload_type;
+ std::optional<int> flexfec_payload_type;
};
// Finds VideoReceiveStreamInterface corresponding to ssrc. Aware of
@@ -656,7 +654,7 @@
const webrtc::RtpPacketReceived& parsed_packet)
RTC_EXCLUSIVE_LOCKS_REQUIRED(thread_checker_);
void ReCreateDefaultReceiveStream(uint32_t ssrc,
- absl::optional<uint32_t> rtx_ssrc);
+ std::optional<uint32_t> rtx_ssrc);
// Add a receive stream. Used for testing.
bool AddRecvStream(const StreamParams& sp, bool default_stream);
@@ -704,7 +702,7 @@
void SetFeedbackParameters(bool lntf_enabled,
bool nack_enabled,
webrtc::RtcpMode rtcp_mode,
- absl::optional<int> rtx_time);
+ std::optional<int> rtx_time);
void SetReceiverParameters(const ChangedReceiverParameters& recv_params);
void OnFrame(const webrtc::VideoFrame& frame) override;
@@ -822,13 +820,12 @@
// is a risk of receiving ssrcs for other, recently added m= sections.
uint32_t demuxer_criteria_id_ RTC_GUARDED_BY(thread_checker_) = 0;
uint32_t demuxer_criteria_completed_id_ RTC_GUARDED_BY(thread_checker_) = 0;
- absl::optional<int64_t> last_unsignalled_ssrc_creation_time_ms_
+ std::optional<int64_t> last_unsignalled_ssrc_creation_time_ms_
RTC_GUARDED_BY(thread_checker_);
std::set<uint32_t> send_ssrcs_ RTC_GUARDED_BY(thread_checker_);
std::set<uint32_t> receive_ssrcs_ RTC_GUARDED_BY(thread_checker_);
- absl::optional<VideoCodecSettings> send_codec_
- RTC_GUARDED_BY(thread_checker_);
+ std::optional<VideoCodecSettings> send_codec_ RTC_GUARDED_BY(thread_checker_);
std::vector<VideoCodecSettings> negotiated_codecs_
RTC_GUARDED_BY(thread_checker_);
diff --git a/media/engine/webrtc_video_engine_unittest.cc b/media/engine/webrtc_video_engine_unittest.cc
index 2046f8c..13ffe6f 100644
--- a/media/engine/webrtc_video_engine_unittest.cc
+++ b/media/engine/webrtc_video_engine_unittest.cc
@@ -408,7 +408,7 @@
std::unique_ptr<webrtc::VideoBitrateAllocatorFactory>
video_bitrate_allocator_factory_;
WebRtcVideoEngine engine_;
- absl::optional<Codec> default_codec_;
+ std::optional<Codec> default_codec_;
std::map<int, int> default_apt_rtx_types_;
};
@@ -839,7 +839,7 @@
// The tests only use H264 Constrained Baseline. Make sure we don't return
// an internal H264 codec from the engine with a different H264 profile.
if (absl::EqualsIgnoreCase(name.c_str(), kH264CodecName)) {
- const absl::optional<webrtc::H264ProfileLevelId> profile_level_id =
+ const std::optional<webrtc::H264ProfileLevelId> profile_level_id =
webrtc::ParseSdpForH264ProfileLevelId(engine_codec.params);
if (profile_level_id->profile !=
webrtc::H264Profile::kProfileConstrainedBaseline) {
@@ -2509,7 +2509,7 @@
parameters.codecs.push_back(GetEngineCodec("VP8"));
EXPECT_TRUE(send_channel_->SetSenderParameters(parameters));
- absl::optional<Codec> codec = send_channel_->GetSendCodec();
+ std::optional<Codec> codec = send_channel_->GetSendCodec();
ASSERT_TRUE(codec);
EXPECT_EQ("VP9", codec->name);
@@ -2535,7 +2535,7 @@
parameters.codecs.push_back(GetEngineCodec("VP8"));
EXPECT_TRUE(send_channel_->SetSenderParameters(parameters));
- absl::optional<Codec> codec = send_channel_->GetSendCodec();
+ std::optional<Codec> codec = send_channel_->GetSendCodec();
ASSERT_TRUE(codec);
EXPECT_EQ("VP9", codec->name);
@@ -2561,7 +2561,7 @@
parameters.codecs.push_back(vp9);
EXPECT_TRUE(send_channel_->SetSenderParameters(parameters));
- absl::optional<Codec> codec = send_channel_->GetSendCodec();
+ std::optional<Codec> codec = send_channel_->GetSendCodec();
ASSERT_TRUE(codec);
EXPECT_EQ("VP8", codec->name);
@@ -3662,7 +3662,7 @@
EXPECT_THAT(
rtp_parameters.encodings,
ElementsAre(Field(&webrtc::RtpEncodingParameters::scalability_mode,
- absl::nullopt)));
+ std::nullopt)));
rtp_parameters.encodings[0].scalability_mode = "L2T3";
EXPECT_TRUE(
send_channel_->SetRtpSendParameters(last_ssrc_, rtp_parameters).ok());
@@ -3803,7 +3803,7 @@
EXPECT_THAT(
rtp_parameters.encodings,
ElementsAre(Field(&webrtc::RtpEncodingParameters::scalability_mode,
- absl::nullopt)));
+ std::nullopt)));
rtp_parameters.encodings[0].scalability_mode = "L2T1";
EXPECT_TRUE(
send_channel_->SetRtpSendParameters(last_ssrc_, rtp_parameters).ok());
@@ -4339,7 +4339,7 @@
AssignDefaultAptRtxTypes();
ASSERT_TRUE(send_channel_->SetSenderParameters(send_parameters_));
- absl::optional<Codec> codec = send_channel_->GetSendCodec();
+ std::optional<Codec> codec = send_channel_->GetSendCodec();
ASSERT_TRUE(codec);
EXPECT_TRUE(codec->Matches(engine_.send_codecs()[0]));
@@ -4909,7 +4909,7 @@
// forced codec anymore.
webrtc::RtpParameters new_params =
send_channel_->GetRtpSendParameters(last_ssrc_);
- EXPECT_EQ(new_params.encodings[0].codec, absl::nullopt);
+ EXPECT_EQ(new_params.encodings[0].codec, std::nullopt);
}
// Test that when both the codec-specific bitrate params and max_bandwidth_bps
@@ -5130,7 +5130,7 @@
EXPECT_EQ(atoi(kMaxQuantization),
AddSendStream()->GetVideoStreams().back().max_qp);
- absl::optional<Codec> codec = send_channel_->GetSendCodec();
+ std::optional<Codec> codec = send_channel_->GetSendCodec();
ASSERT_TRUE(codec);
EXPECT_EQ(kMaxQuantization, codec->params[kCodecParamMaxQuantization]);
}
@@ -5954,14 +5954,14 @@
stats.total_encoded_bytes_target);
// Comes from substream only.
EXPECT_EQ(sender.total_packet_send_delay, webrtc::TimeDelta::Zero());
- EXPECT_EQ(sender.qp_sum, absl::nullopt);
+ EXPECT_EQ(sender.qp_sum, std::nullopt);
EXPECT_EQ(sender.has_entered_low_resolution,
stats.has_entered_low_resolution);
EXPECT_EQ(sender.content_type, webrtc::VideoContentType::SCREENSHARE);
EXPECT_EQ(sender.frames_sent, stats.frames_encoded);
EXPECT_EQ(sender.huge_frames_sent, stats.huge_frames_sent);
- EXPECT_EQ(sender.rid, absl::nullopt);
+ EXPECT_EQ(sender.rid, std::nullopt);
}
TEST_F(WebRtcVideoChannelTest, GetAggregatedStatsReportForSubStreams) {
@@ -6085,7 +6085,7 @@
EXPECT_EQ(sender.content_type, webrtc::VideoContentType::SCREENSHARE);
EXPECT_EQ(sender.frames_sent, 2u * substream.frames_encoded);
EXPECT_EQ(sender.huge_frames_sent, stats.huge_frames_sent);
- EXPECT_EQ(sender.rid, absl::nullopt);
+ EXPECT_EQ(sender.rid, std::nullopt);
}
TEST_F(WebRtcVideoChannelTest, GetPerLayerStatsReportForSubStreams) {
@@ -6210,7 +6210,7 @@
EXPECT_EQ(sender.frames_sent,
static_cast<uint32_t>(substream.frames_encoded));
EXPECT_EQ(sender.huge_frames_sent, substream.huge_frames_sent);
- EXPECT_EQ(sender.rid, absl::nullopt);
+ EXPECT_EQ(sender.rid, std::nullopt);
}
TEST_F(WebRtcVideoChannelTest,
@@ -6417,7 +6417,7 @@
substreams[kFirstMediaStreamSsrc].rtp_stats.retransmitted.padding_bytes = 6;
substreams[kFirstMediaStreamSsrc].rtp_stats.retransmitted.payload_bytes = 7;
substreams[kFirstMediaStreamSsrc].rtp_stats.retransmitted.packets = 8;
- substreams[kFirstMediaStreamSsrc].referenced_media_ssrc = absl::nullopt;
+ substreams[kFirstMediaStreamSsrc].referenced_media_ssrc = std::nullopt;
substreams[kFirstMediaStreamSsrc].width = 1280;
substreams[kFirstMediaStreamSsrc].height = 720;
// Second kMedia stream.
@@ -6431,7 +6431,7 @@
substreams[kSecondMediaStreamSsrc].rtp_stats.retransmitted.padding_bytes = 15;
substreams[kSecondMediaStreamSsrc].rtp_stats.retransmitted.payload_bytes = 16;
substreams[kSecondMediaStreamSsrc].rtp_stats.retransmitted.packets = 17;
- substreams[kSecondMediaStreamSsrc].referenced_media_ssrc = absl::nullopt;
+ substreams[kSecondMediaStreamSsrc].referenced_media_ssrc = std::nullopt;
substreams[kSecondMediaStreamSsrc].width = 640;
substreams[kSecondMediaStreamSsrc].height = 480;
// kRtx stream referencing the first kMedia stream.
@@ -6514,7 +6514,7 @@
stats.substreams[101].rtp_stats.retransmitted.padding_bytes = 0;
stats.substreams[101].rtp_stats.retransmitted.payload_bytes = 0;
stats.substreams[101].rtp_stats.retransmitted.packets = 0;
- stats.substreams[101].referenced_media_ssrc = absl::nullopt;
+ stats.substreams[101].referenced_media_ssrc = std::nullopt;
// Simulcast layer 1, RTX stream. header+padding=5, payload=10, packets=1.
stats.substreams[102].type =
webrtc::VideoSendStream::StreamStats::StreamType::kRtx;
@@ -6536,7 +6536,7 @@
stats.substreams[201].rtp_stats.retransmitted.padding_bytes = 0;
stats.substreams[201].rtp_stats.retransmitted.payload_bytes = 0;
stats.substreams[201].rtp_stats.retransmitted.packets = 0;
- stats.substreams[201].referenced_media_ssrc = absl::nullopt;
+ stats.substreams[201].referenced_media_ssrc = std::nullopt;
// Simulcast layer 2, RTX stream. header+padding=10, payload=20, packets=4.
stats.substreams[202].type =
webrtc::VideoSendStream::StreamStats::StreamType::kRtx;
@@ -7373,7 +7373,7 @@
// Test BaseMinimumPlayoutDelayMs on unsignaled receive streams.
TEST_F(WebRtcVideoChannelTest, BaseMinimumPlayoutDelayMsUnsignaledRecvStream) {
- absl::optional<int> delay_ms;
+ std::optional<int> delay_ms;
const FakeVideoReceiveStream* recv_stream;
// Set default stream with SSRC 0
@@ -7931,7 +7931,7 @@
// Check that the vector of VideoStreams also was propagated correctly. Note
// that this is testing the behavior of the FakeVideoSendStream, which mimics
// the calls to CreateEncoderStreams to get the VideoStreams.
- EXPECT_EQ(absl::optional<double>(new_bitrate_priority),
+ EXPECT_EQ(std::optional<double>(new_bitrate_priority),
video_send_stream->GetVideoStreams()[0].bitrate_priority);
}
@@ -7990,13 +7990,13 @@
// FakeVideoSendStream calls CreateEncoderStreams, and we are testing that
// these are created appropriately for the simulcast case.
EXPECT_EQ(kNumSimulcastStreams, video_send_stream->GetVideoStreams().size());
- EXPECT_EQ(absl::optional<double>(new_bitrate_priority),
+ EXPECT_EQ(std::optional<double>(new_bitrate_priority),
video_send_stream->GetVideoStreams()[0].bitrate_priority);
// Since we are only setting bitrate priority per-sender, the other
// VideoStreams should have a bitrate priority of 0.
- EXPECT_EQ(absl::nullopt,
+ EXPECT_EQ(std::nullopt,
video_send_stream->GetVideoStreams()[1].bitrate_priority);
- EXPECT_EQ(absl::nullopt,
+ EXPECT_EQ(std::nullopt,
video_send_stream->GetVideoStreams()[2].bitrate_priority);
EXPECT_TRUE(send_channel_->SetVideoSend(primary_ssrc, nullptr, nullptr));
}
@@ -8332,14 +8332,14 @@
webrtc::RtpParameters parameters =
send_channel_->GetRtpSendParameters(last_ssrc_);
EXPECT_EQ(kNumSimulcastStreams, parameters.encodings.size());
- parameters.encodings[0].scalability_mode = absl::nullopt;
+ parameters.encodings[0].scalability_mode = std::nullopt;
parameters.encodings[1].scalability_mode = "L1T3"; // Supported.
parameters.encodings[2].scalability_mode = "L3T3"; // Unsupported.
EXPECT_TRUE(send_channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
// Verify that the new value is propagated down to the encoder.
// Check that WebRtcVideoSendStream updates VideoEncoderConfig correctly.
- const absl::optional<ScalabilityMode> kDefaultScalabilityMode =
+ const std::optional<ScalabilityMode> kDefaultScalabilityMode =
webrtc::ScalabilityModeFromString(kDefaultScalabilityModeStr);
EXPECT_EQ(2, stream->num_encoder_reconfigurations());
webrtc::VideoEncoderConfig encoder_config = stream->GetEncoderConfig().Copy();
@@ -8413,7 +8413,7 @@
// Verify that the new value is propagated down to the encoder.
// Check that WebRtcVideoSendStream updates VideoEncoderConfig correctly.
- const absl::optional<ScalabilityMode> kDefaultScalabilityMode =
+ const std::optional<ScalabilityMode> kDefaultScalabilityMode =
webrtc::ScalabilityModeFromString(kDefaultScalabilityModeStr);
EXPECT_EQ(2, stream->num_encoder_reconfigurations());
webrtc::VideoEncoderConfig encoder_config = stream->GetEncoderConfig().Copy();
@@ -9589,7 +9589,7 @@
EXPECT_TRUE(send_channel_->SetSenderParameters(parameters));
send_channel_->SetSend(true);
- absl::optional<Codec> codec = send_channel_->GetSendCodec();
+ std::optional<Codec> codec = send_channel_->GetSendCodec();
ASSERT_TRUE(codec);
EXPECT_EQ("VP8", codec->name);
diff --git a/media/engine/webrtc_voice_engine.cc b/media/engine/webrtc_voice_engine.cc
index 95a8f36..542ba57 100644
--- a/media/engine/webrtc_voice_engine.cc
+++ b/media/engine/webrtc_voice_engine.cc
@@ -17,6 +17,7 @@
#include <initializer_list>
#include <iterator>
#include <memory>
+#include <optional>
#include <string>
#include <type_traits>
#include <utility>
@@ -26,7 +27,6 @@
#include "absl/algorithm/container.h"
#include "absl/functional/bind_front.h"
#include "absl/strings/match.h"
-#include "absl/types/optional.h"
#include "api/audio/audio_frame.h"
#include "api/audio/audio_frame_processor.h"
#include "api/audio/audio_processing.h"
@@ -145,14 +145,14 @@
return absl::EqualsIgnoreCase(codec.name, ref_name);
}
-absl::optional<Codec> FindCodec(const std::vector<Codec>& codecs,
- const Codec& codec) {
+std::optional<Codec> FindCodec(const std::vector<Codec>& codecs,
+ const Codec& codec) {
for (const Codec& c : codecs) {
if (c.Matches(codec)) {
return c;
}
}
- return absl::nullopt;
+ return std::nullopt;
}
bool VerifyUniquePayloadTypes(const std::vector<Codec>& codecs) {
@@ -166,7 +166,7 @@
return absl::c_adjacent_find(payload_types) == payload_types.end();
}
-absl::optional<std::string> GetAudioNetworkAdaptorConfig(
+std::optional<std::string> GetAudioNetworkAdaptorConfig(
const AudioOptions& options) {
if (options.audio_network_adaptor && *options.audio_network_adaptor &&
options.audio_network_adaptor_config) {
@@ -174,7 +174,7 @@
// equals true and `options_.audio_network_adaptor_config` has a value.
return options.audio_network_adaptor_config;
}
- return absl::nullopt;
+ return std::nullopt;
}
// Returns its smallest positive argument. If neither argument is positive,
@@ -191,9 +191,9 @@
// `max_send_bitrate_bps` is the bitrate from "b=" in SDP.
// `rtp_max_bitrate_bps` is the bitrate from RtpSender::SetParameters.
-absl::optional<int> ComputeSendBitrate(int max_send_bitrate_bps,
- absl::optional<int> rtp_max_bitrate_bps,
- const webrtc::AudioCodecSpec& spec) {
+std::optional<int> ComputeSendBitrate(int max_send_bitrate_bps,
+ std::optional<int> rtp_max_bitrate_bps,
+ const webrtc::AudioCodecSpec& spec) {
// If application-configured bitrate is set, take minimum of that and SDP
// bitrate.
const int bps = rtp_max_bitrate_bps
@@ -212,7 +212,7 @@
<< " bps"
", requires at least "
<< spec.info.min_bitrate_bps << " bps.";
- return absl::nullopt;
+ return std::nullopt;
}
if (spec.info.HasFixedBitrate()) {
@@ -235,7 +235,7 @@
webrtc::DataRate min_encoder_bitrate = webrtc::DataRate::KilobitsPerSec(16);
bool use_slow_adaptation = true;
- absl::optional<std::string> audio_network_adaptor_config;
+ std::optional<std::string> audio_network_adaptor_config;
std::unique_ptr<webrtc::StructParametersParser> Parser() {
return webrtc::StructParametersParser::Create( //
@@ -272,7 +272,7 @@
webrtc::Transport* rtcp_send_transport,
const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory,
const std::map<int, webrtc::SdpAudioFormat>& decoder_map,
- absl::optional<webrtc::AudioCodecPairId> codec_pair_id,
+ std::optional<webrtc::AudioCodecPairId> codec_pair_id,
size_t jitter_buffer_max_packets,
bool jitter_buffer_fast_accelerate,
int jitter_buffer_min_delay_ms,
@@ -686,7 +686,7 @@
}
}
-absl::optional<webrtc::AudioDeviceModule::Stats>
+std::optional<webrtc::AudioDeviceModule::Stats>
WebRtcVoiceEngine::GetAudioDeviceStats() {
return adm()->GetStats();
}
@@ -722,7 +722,7 @@
auto map_format = [&mapper](const webrtc::SdpAudioFormat& format,
std::vector<Codec>* out) {
- absl::optional<Codec> opt_codec = mapper.ToAudioCodec(format);
+ std::optional<Codec> opt_codec = mapper.ToAudioCodec(format);
if (opt_codec) {
if (out) {
out->push_back(*opt_codec);
@@ -737,7 +737,7 @@
for (const auto& spec : specs) {
// We need to do some extra stuff before adding the main codecs to out.
- absl::optional<Codec> opt_codec = map_format(spec.format, nullptr);
+ std::optional<Codec> opt_codec = map_format(spec.format, nullptr);
if (opt_codec) {
Codec& codec = *opt_codec;
if (spec.info.supports_network_adaption) {
@@ -796,17 +796,17 @@
const std::string& mid,
const std::string& c_name,
const std::string track_id,
- const absl::optional<webrtc::AudioSendStream::Config::SendCodecSpec>&
+ const std::optional<webrtc::AudioSendStream::Config::SendCodecSpec>&
send_codec_spec,
bool extmap_allow_mixed,
const std::vector<webrtc::RtpExtension>& extensions,
int max_send_bitrate_bps,
int rtcp_report_interval_ms,
- const absl::optional<std::string>& audio_network_adaptor_config,
+ const std::optional<std::string>& audio_network_adaptor_config,
webrtc::Call* call,
webrtc::Transport* send_transport,
const rtc::scoped_refptr<webrtc::AudioEncoderFactory>& encoder_factory,
- const absl::optional<webrtc::AudioCodecPairId> codec_pair_id,
+ const std::optional<webrtc::AudioCodecPairId> codec_pair_id,
rtc::scoped_refptr<webrtc::FrameEncryptorInterface> frame_encryptor,
const webrtc::CryptoOptions& crypto_options)
: adaptive_ptime_config_(call->trials()),
@@ -899,7 +899,7 @@
}
void SetAudioNetworkAdaptorConfig(
- const absl::optional<std::string>& audio_network_adaptor_config) {
+ const std::optional<std::string>& audio_network_adaptor_config) {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
if (audio_network_adaptor_config_from_options_ ==
audio_network_adaptor_config) {
@@ -1000,7 +1000,7 @@
int sample_rate,
size_t number_of_channels,
size_t number_of_frames,
- absl::optional<int64_t> absolute_capture_timestamp_ms) override {
+ std::optional<int64_t> absolute_capture_timestamp_ms) override {
TRACE_EVENT_BEGIN2("webrtc", "WebRtcAudioSendStream::OnData", "sample_rate",
sample_rate, "number_of_frames", number_of_frames);
RTC_DCHECK_EQ(16, bits_per_sample);
@@ -1044,7 +1044,7 @@
return webrtc::InvokeSetParametersCallback(callback, error);
}
- absl::optional<int> send_rate;
+ std::optional<int> send_rate;
if (audio_codec_spec_) {
send_rate = ComputeSendBitrate(max_send_bitrate_bps_,
parameters.encodings[0].max_bitrate_bps,
@@ -1055,7 +1055,7 @@
}
}
- const absl::optional<int> old_rtp_max_bitrate =
+ const std::optional<int> old_rtp_max_bitrate =
rtp_parameters_.encodings[0].max_bitrate_bps;
double old_priority = rtp_parameters_.encodings[0].bitrate_priority;
webrtc::Priority old_dscp = rtp_parameters_.encodings[0].network_priority;
@@ -1207,10 +1207,10 @@
bool muted_ = false;
int max_send_bitrate_bps_;
webrtc::RtpParameters rtp_parameters_;
- absl::optional<webrtc::AudioCodecSpec> audio_codec_spec_;
+ std::optional<webrtc::AudioCodecSpec> audio_codec_spec_;
// TODO(webrtc:11717): Remove this once audio_network_adaptor in AudioOptions
// has been removed.
- absl::optional<std::string> audio_network_adaptor_config_from_options_;
+ std::optional<std::string> audio_network_adaptor_config_from_options_;
std::atomic<int> num_encoded_channels_{-1};
};
@@ -1254,7 +1254,7 @@
options_.SetAll(options);
engine()->ApplyOptions(options_);
- absl::optional<std::string> audio_network_adaptor_config =
+ std::optional<std::string> audio_network_adaptor_config =
GetAudioNetworkAdaptorConfig(options_);
for (auto& it : send_streams_) {
it.second->SetAudioNetworkAdaptorConfig(audio_network_adaptor_config);
@@ -1275,7 +1275,7 @@
// all the information at once.
// Finding if the RtpParameters force a specific codec
- absl::optional<Codec> force_codec;
+ std::optional<Codec> force_codec;
if (send_streams_.size() == 1) {
// Since audio simulcast is not supported, currently, only PlanB
// has multiple tracks and we don't care about getting the
@@ -1343,11 +1343,11 @@
return SetOptions(params.options);
}
-absl::optional<Codec> WebRtcVoiceSendChannel::GetSendCodec() const {
+std::optional<Codec> WebRtcVoiceSendChannel::GetSendCodec() const {
if (send_codec_spec_) {
return CreateAudioCodec(send_codec_spec_->format);
}
- return absl::nullopt;
+ return std::nullopt;
}
// Utility function called from SetSenderParameters() to extract current send
@@ -1355,9 +1355,9 @@
// and receive streams may be reconfigured based on the new settings.
bool WebRtcVoiceSendChannel::SetSendCodecs(
const std::vector<Codec>& codecs,
- absl::optional<Codec> preferred_codec) {
+ std::optional<Codec> preferred_codec) {
RTC_DCHECK_RUN_ON(worker_thread_);
- dtmf_payload_type_ = absl::nullopt;
+ dtmf_payload_type_ = std::nullopt;
dtmf_payload_freq_ = -1;
// Validate supplied codecs list.
@@ -1387,10 +1387,9 @@
}
// Scan through the list to figure out the codec to use for sending.
- absl::optional<webrtc::AudioSendStream::Config::SendCodecSpec>
- send_codec_spec;
+ std::optional<webrtc::AudioSendStream::Config::SendCodecSpec> send_codec_spec;
webrtc::BitrateConstraints bitrate_config;
- absl::optional<webrtc::AudioCodecInfo> voice_codec_info;
+ std::optional<webrtc::AudioCodecInfo> voice_codec_info;
size_t send_codec_position = 0;
for (const Codec& voice_codec : codecs) {
if (!(IsCodec(voice_codec, kCnCodecName) ||
@@ -1554,7 +1553,7 @@
return false;
}
- absl::optional<std::string> audio_network_adaptor_config =
+ std::optional<std::string> audio_network_adaptor_config =
GetAudioNetworkAdaptorConfig(options_);
WebRtcAudioSendStream* stream = new WebRtcAudioSendStream(
ssrc, mid_, sp.cname, sp.id, send_codec_spec_, ExtmapAllowMixed(),
@@ -1878,7 +1877,7 @@
}
SetPreferredDscp(new_dscp);
- absl::optional<cricket::Codec> send_codec = GetSendCodec();
+ std::optional<cricket::Codec> send_codec = GetSendCodec();
// Since we validate that all layers have the same value, we can just check
// the first layer.
// TODO(orphis): Support mixed-codec simulcast
@@ -2168,7 +2167,7 @@
for (const Codec& codec : codecs) {
// Log a warning if a codec's payload type is changing. This used to be
// treated as an error. It's abnormal, but not really illegal.
- absl::optional<Codec> old_codec = FindCodec(recv_codecs_, codec);
+ std::optional<Codec> old_codec = FindCodec(recv_codecs_, codec);
if (old_codec && old_codec->id != codec.id) {
RTC_LOG(LS_WARNING) << codec.name << " mapped to a second payload type ("
<< codec.id << ", was already mapped to "
@@ -2357,9 +2356,9 @@
}
}
-absl::optional<uint32_t> WebRtcVoiceReceiveChannel::GetUnsignaledSsrc() const {
+std::optional<uint32_t> WebRtcVoiceReceiveChannel::GetUnsignaledSsrc() const {
if (unsignaled_recv_ssrcs_.empty()) {
- return absl::nullopt;
+ return std::nullopt;
}
// In the event of multiple unsignaled ssrcs, the last in the vector will be
// the most recent one (the one forwarded to the MediaStreamTrack).
@@ -2447,7 +2446,7 @@
return true;
}
-absl::optional<int> WebRtcVoiceReceiveChannel::GetBaseMinimumPlayoutDelayMs(
+std::optional<int> WebRtcVoiceReceiveChannel::GetBaseMinimumPlayoutDelayMs(
uint32_t ssrc) const {
// SSRC of 0 represents the default receive stream.
if (ssrc == 0) {
@@ -2459,7 +2458,7 @@
if (it != recv_streams_.end()) {
return it->second->GetBaseMinimumPlayoutDelayMs();
}
- return absl::nullopt;
+ return std::nullopt;
}
void WebRtcVoiceReceiveChannel::SetFrameDecryptor(
diff --git a/media/engine/webrtc_voice_engine.h b/media/engine/webrtc_voice_engine.h
index 13ec681..942c569 100644
--- a/media/engine/webrtc_voice_engine.h
+++ b/media/engine/webrtc_voice_engine.h
@@ -16,6 +16,7 @@
#include <map>
#include <memory>
+#include <optional>
#include <set>
#include <string>
#include <utility>
@@ -23,7 +24,6 @@
#include "absl/functional/any_invocable.h"
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/audio/audio_device.h"
#include "api/audio/audio_frame_processor.h"
#include "api/audio/audio_mixer.h"
@@ -130,7 +130,7 @@
// Stops AEC dump.
void StopAecDump() override;
- absl::optional<webrtc::AudioDeviceModule::Stats> GetAudioDeviceStats()
+ std::optional<webrtc::AudioDeviceModule::Stats> GetAudioDeviceStats()
override;
private:
@@ -202,7 +202,7 @@
}
VoiceMediaSendChannelInterface* AsVoiceSendChannel() override { return this; }
- absl::optional<Codec> GetSendCodec() const override;
+ std::optional<Codec> GetSendCodec() const override;
// Functions imported from MediaChannelUtil
void SetInterface(MediaChannelNetworkInterface* iface) override {
@@ -285,7 +285,7 @@
private:
bool SetOptions(const AudioOptions& options);
bool SetSendCodecs(const std::vector<Codec>& codecs,
- absl::optional<Codec> preferred_codec);
+ std::optional<Codec> preferred_codec);
bool SetLocalSource(uint32_t ssrc, AudioSource* source);
bool MuteStream(uint32_t ssrc, bool mute);
@@ -303,7 +303,7 @@
int max_send_bitrate_bps_ = 0;
AudioOptions options_;
- absl::optional<int> dtmf_payload_type_;
+ std::optional<int> dtmf_payload_type_;
int dtmf_payload_freq_ = -1;
bool enable_non_sender_rtt_ = false;
bool send_ = false;
@@ -318,7 +318,7 @@
std::string mid_;
webrtc::RtcpMode rtcp_mode_;
- absl::optional<webrtc::AudioSendStream::Config::SendCodecSpec>
+ std::optional<webrtc::AudioSendStream::Config::SendCodecSpec>
send_codec_spec_;
// TODO(kwiberg): Per-SSRC codec pair IDs?
@@ -381,7 +381,7 @@
bool AddRecvStream(const StreamParams& sp) override;
bool RemoveRecvStream(uint32_t ssrc) override;
void ResetUnsignaledRecvStream() override;
- absl::optional<uint32_t> GetUnsignaledSsrc() const override;
+ std::optional<uint32_t> GetUnsignaledSsrc() const override;
void ChooseReceiverReportSsrc(const std::set<uint32_t>& choices) override;
@@ -401,8 +401,7 @@
bool SetDefaultOutputVolume(double volume) override;
bool SetBaseMinimumPlayoutDelayMs(uint32_t ssrc, int delay_ms) override;
- absl::optional<int> GetBaseMinimumPlayoutDelayMs(
- uint32_t ssrc) const override;
+ std::optional<int> GetBaseMinimumPlayoutDelayMs(uint32_t ssrc) const override;
void OnPacketReceived(const webrtc::RtpPacketReceived& packet) override;
bool GetStats(VoiceMediaReceiveInfo* info,
@@ -496,7 +495,7 @@
std::vector<webrtc::RtpExtension> recv_rtp_extensions_;
webrtc::RtpHeaderExtensionMap recv_rtp_extension_map_;
- absl::optional<webrtc::AudioSendStream::Config::SendCodecSpec>
+ std::optional<webrtc::AudioSendStream::Config::SendCodecSpec>
send_codec_spec_;
// TODO(kwiberg): Per-SSRC codec pair IDs?
diff --git a/media/engine/webrtc_voice_engine_unittest.cc b/media/engine/webrtc_voice_engine_unittest.cc
index 2189a67..e15690d 100644
--- a/media/engine/webrtc_voice_engine_unittest.cc
+++ b/media/engine/webrtc_voice_engine_unittest.cc
@@ -11,11 +11,11 @@
#include "media/engine/webrtc_voice_engine.h"
#include <memory>
+#include <optional>
#include <utility>
#include "absl/memory/memory.h"
#include "absl/strings/match.h"
-#include "absl/types/optional.h"
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
#include "api/environment/environment.h"
@@ -468,7 +468,7 @@
EXPECT_EQ(expected_bitrate, spec->target_bitrate_bps);
}
- absl::optional<int> GetCodecBitrate(int32_t ssrc) {
+ std::optional<int> GetCodecBitrate(int32_t ssrc) {
return GetSendStreamConfig(ssrc).send_codec_spec->target_bitrate_bps;
}
@@ -476,8 +476,7 @@
return GetSendStreamConfig(ssrc).max_bitrate_bps;
}
- const absl::optional<std::string>& GetAudioNetworkAdaptorConfig(
- int32_t ssrc) {
+ const std::optional<std::string>& GetAudioNetworkAdaptorConfig(int32_t ssrc) {
return GetSendStreamConfig(ssrc).audio_network_adaptor_config;
}
@@ -1230,7 +1229,7 @@
// Now change it back to active and verify we resume sending.
// This should occur even when other parameters are updated.
parameters.encodings[0].active = true;
- parameters.encodings[0].max_bitrate_bps = absl::optional<int>(6000);
+ parameters.encodings[0].max_bitrate_bps = std::optional<int>(6000);
EXPECT_TRUE(send_channel_->SetRtpSendParameters(kSsrcX, parameters).ok());
EXPECT_TRUE(GetSendStream(kSsrcX).IsSending());
}
@@ -1408,7 +1407,7 @@
// forced codec anymore.
webrtc::RtpParameters new_params =
send_channel_->GetRtpSendParameters(kSsrcX);
- EXPECT_EQ(new_params.encodings[0].codec, absl::nullopt);
+ EXPECT_EQ(new_params.encodings[0].codec, std::nullopt);
}
// Test that max_bitrate_bps in send stream config gets updated correctly when
@@ -1590,7 +1589,7 @@
EXPECT_EQ(22000, send_codec_spec.target_bitrate_bps);
EXPECT_STRCASEEQ("OPUS", send_codec_spec.format.name.c_str());
EXPECT_NE(send_codec_spec.format.clockrate_hz, 8000);
- EXPECT_EQ(absl::nullopt, send_codec_spec.cng_payload_type);
+ EXPECT_EQ(std::nullopt, send_codec_spec.cng_payload_type);
EXPECT_FALSE(send_channel_->CanInsertDtmf());
}
@@ -1620,7 +1619,7 @@
const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
EXPECT_EQ(111, send_codec_spec.payload_type);
EXPECT_STRCASEEQ("opus", send_codec_spec.format.name.c_str());
- EXPECT_EQ(absl::nullopt, send_codec_spec.red_payload_type);
+ EXPECT_EQ(std::nullopt, send_codec_spec.red_payload_type);
}
// Test that we do not use Opus/Red by default.
@@ -1634,7 +1633,7 @@
const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
EXPECT_EQ(111, send_codec_spec.payload_type);
EXPECT_STRCASEEQ("opus", send_codec_spec.format.name.c_str());
- EXPECT_EQ(absl::nullopt, send_codec_spec.red_payload_type);
+ EXPECT_EQ(std::nullopt, send_codec_spec.red_payload_type);
}
// Test that the RED fmtp line must match the payload type.
@@ -1648,7 +1647,7 @@
const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
EXPECT_EQ(111, send_codec_spec.payload_type);
EXPECT_STRCASEEQ("opus", send_codec_spec.format.name.c_str());
- EXPECT_EQ(absl::nullopt, send_codec_spec.red_payload_type);
+ EXPECT_EQ(std::nullopt, send_codec_spec.red_payload_type);
}
// Test that the RED fmtp line must show 2..32 payloads.
@@ -1662,7 +1661,7 @@
const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
EXPECT_EQ(111, send_codec_spec.payload_type);
EXPECT_STRCASEEQ("opus", send_codec_spec.format.name.c_str());
- EXPECT_EQ(absl::nullopt, send_codec_spec.red_payload_type);
+ EXPECT_EQ(std::nullopt, send_codec_spec.red_payload_type);
for (int i = 1; i < 32; i++) {
parameters.codecs[0].params[""] += "/111";
SetSenderParameters(parameters);
@@ -1676,7 +1675,7 @@
const auto& send_codec_spec3 = *GetSendStreamConfig(kSsrcX).send_codec_spec;
EXPECT_EQ(111, send_codec_spec3.payload_type);
EXPECT_STRCASEEQ("opus", send_codec_spec3.format.name.c_str());
- EXPECT_EQ(absl::nullopt, send_codec_spec3.red_payload_type);
+ EXPECT_EQ(std::nullopt, send_codec_spec3.red_payload_type);
}
// Test that WebRtcVoiceEngine reconfigures, rather than recreates its
@@ -1709,7 +1708,7 @@
parameters.codecs[0].bitrate = 0;
parameters.codecs[0].clockrate = 50000;
EXPECT_TRUE(send_channel_->SetSenderParameters(parameters));
- EXPECT_EQ(send_channel_->GetSendCodec(), absl::nullopt);
+ EXPECT_EQ(send_channel_->GetSendCodec(), std::nullopt);
}
// Test that if channels=0 for opus, we do not have a send codec.
@@ -1720,7 +1719,7 @@
parameters.codecs[0].bitrate = 0;
parameters.codecs[0].channels = 0;
EXPECT_TRUE(send_channel_->SetSenderParameters(parameters));
- EXPECT_EQ(send_channel_->GetSendCodec(), absl::nullopt);
+ EXPECT_EQ(send_channel_->GetSendCodec(), std::nullopt);
}
// Test that if channels=0 for opus, we do not have a send codec.
@@ -1732,7 +1731,7 @@
parameters.codecs[0].channels = 0;
parameters.codecs[0].params["stereo"] = "1";
EXPECT_TRUE(send_channel_->SetSenderParameters(parameters));
- EXPECT_EQ(send_channel_->GetSendCodec(), absl::nullopt);
+ EXPECT_EQ(send_channel_->GetSendCodec(), std::nullopt);
}
// Test that if channel is 1 for opus and there's no stereo, we do not have a
@@ -1744,7 +1743,7 @@
parameters.codecs[0].bitrate = 0;
parameters.codecs[0].channels = 1;
EXPECT_TRUE(send_channel_->SetSenderParameters(parameters));
- EXPECT_EQ(send_channel_->GetSendCodec(), absl::nullopt);
+ EXPECT_EQ(send_channel_->GetSendCodec(), std::nullopt);
}
// Test that if channel is 1 for opus and stereo=0, we do not have a send codec.
@@ -1756,7 +1755,7 @@
parameters.codecs[0].channels = 1;
parameters.codecs[0].params["stereo"] = "0";
EXPECT_TRUE(send_channel_->SetSenderParameters(parameters));
- EXPECT_EQ(send_channel_->GetSendCodec(), absl::nullopt);
+ EXPECT_EQ(send_channel_->GetSendCodec(), std::nullopt);
}
// Test that if channel is 1 for opus and stereo=1, we do not have a send codec.
@@ -1768,7 +1767,7 @@
parameters.codecs[0].channels = 1;
parameters.codecs[0].params["stereo"] = "1";
EXPECT_TRUE(send_channel_->SetSenderParameters(parameters));
- EXPECT_EQ(send_channel_->GetSendCodec(), absl::nullopt);
+ EXPECT_EQ(send_channel_->GetSendCodec(), std::nullopt);
}
// Test that with bitrate=0 and no stereo, bitrate is 32000.
@@ -2046,7 +2045,7 @@
EXPECT_TRUE(SetupSendStream());
cricket::AudioSenderParameter parameters;
EXPECT_TRUE(send_channel_->SetSenderParameters(parameters));
- EXPECT_EQ(send_channel_->GetSendCodec(), absl::nullopt);
+ EXPECT_EQ(send_channel_->GetSendCodec(), std::nullopt);
}
// Test that we can set send codecs even with telephone-event codec as the first
@@ -2178,7 +2177,7 @@
{
const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
EXPECT_STRCASEEQ("PCMU", send_codec_spec.format.name.c_str());
- EXPECT_EQ(absl::nullopt, send_codec_spec.cng_payload_type);
+ EXPECT_EQ(std::nullopt, send_codec_spec.cng_payload_type);
}
// Set PCMU(8K) and CN(8K). VAD should be activated.
parameters.codecs[1] = kCn8000Codec;
@@ -2195,7 +2194,7 @@
{
const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
EXPECT_STRCASEEQ("OPUS", send_codec_spec.format.name.c_str());
- EXPECT_EQ(absl::nullopt, send_codec_spec.cng_payload_type);
+ EXPECT_EQ(std::nullopt, send_codec_spec.cng_payload_type);
}
}
@@ -2366,7 +2365,7 @@
const auto& send_codec_spec =
*call_.GetAudioSendStream(ssrc)->GetConfig().send_codec_spec;
EXPECT_STRCASEEQ("PCMU", send_codec_spec.format.name.c_str());
- EXPECT_EQ(absl::nullopt, send_codec_spec.cng_payload_type);
+ EXPECT_EQ(std::nullopt, send_codec_spec.cng_payload_type);
}
}
@@ -2532,7 +2531,7 @@
cricket::AudioOptions options;
options.audio_network_adaptor = false;
SetAudioSend(kSsrcX, true, nullptr, &options);
- EXPECT_EQ(absl::nullopt, GetAudioNetworkAdaptorConfig(kSsrcX));
+ EXPECT_EQ(std::nullopt, GetAudioNetworkAdaptorConfig(kSsrcX));
}
TEST_P(WebRtcVoiceEngineTestFake, AudioNetworkAdaptorNotGetOverridden) {
@@ -2544,7 +2543,7 @@
GetAudioNetworkAdaptorConfig(kSsrcX));
const int initial_num = call_.GetNumCreatedSendStreams();
cricket::AudioOptions options;
- options.audio_network_adaptor = absl::nullopt;
+ options.audio_network_adaptor = std::nullopt;
// Unvalued `options.audio_network_adaptor` should not reset audio network
// adaptor.
SetAudioSend(kSsrcX, true, nullptr, &options);
diff --git a/media/sctp/dcsctp_transport.cc b/media/sctp/dcsctp_transport.cc
index 4623435..ae8d778 100644
--- a/media/sctp/dcsctp_transport.cc
+++ b/media/sctp/dcsctp_transport.cc
@@ -13,11 +13,11 @@
#include <atomic>
#include <cstdint>
#include <limits>
+#include <optional>
#include <utility>
#include <vector>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/data_channel_interface.h"
#include "api/environment/environment.h"
@@ -72,7 +72,7 @@
}
}
-absl::optional<DataMessageType> ToDataMessageType(dcsctp::PPID ppid) {
+std::optional<DataMessageType> ToDataMessageType(dcsctp::PPID ppid) {
switch (static_cast<WebrtcPPID>(ppid.value())) {
case WebrtcPPID::kDCEP:
return DataMessageType::kControl;
@@ -85,10 +85,10 @@
case WebrtcPPID::kBinaryEmpty:
return DataMessageType::kBinary;
}
- return absl::nullopt;
+ return std::nullopt;
}
-absl::optional<cricket::SctpErrorCauseCode> ToErrorCauseCode(
+std::optional<cricket::SctpErrorCauseCode> ToErrorCauseCode(
dcsctp::ErrorKind error) {
switch (error) {
case dcsctp::ErrorKind::kParseFailed:
@@ -107,7 +107,7 @@
// No SCTP error cause code matches those
break;
}
- return absl::nullopt;
+ return std::nullopt;
}
bool IsEmptyPPID(dcsctp::PPID ppid) {
@@ -192,8 +192,8 @@
options.max_message_size = max_message_size;
options.max_timer_backoff_duration = kMaxTimerBackoffDuration;
// Don't close the connection automatically on too many retransmissions.
- options.max_retransmissions = absl::nullopt;
- options.max_init_retransmits = absl::nullopt;
+ options.max_retransmissions = std::nullopt;
+ options.max_init_retransmits = std::nullopt;
options.per_stream_send_queue_limit =
DataChannelInterface::MaxSendQueueSize();
// This is just set to avoid denial-of-service. Practically unlimited.
@@ -382,15 +382,15 @@
return socket_->options().max_message_size;
}
-absl::optional<int> DcSctpTransport::max_outbound_streams() const {
+std::optional<int> DcSctpTransport::max_outbound_streams() const {
if (!socket_)
- return absl::nullopt;
+ return std::nullopt;
return socket_->options().announced_maximum_outgoing_streams;
}
-absl::optional<int> DcSctpTransport::max_inbound_streams() const {
+std::optional<int> DcSctpTransport::max_inbound_streams() const {
if (!socket_)
- return absl::nullopt;
+ return std::nullopt;
return socket_->options().announced_maximum_incoming_streams;
}
diff --git a/media/sctp/dcsctp_transport.h b/media/sctp/dcsctp_transport.h
index 7f3e7df..030babe 100644
--- a/media/sctp/dcsctp_transport.h
+++ b/media/sctp/dcsctp_transport.h
@@ -12,10 +12,10 @@
#define MEDIA_SCTP_DCSCTP_TRANSPORT_H_
#include <memory>
+#include <optional>
#include <string>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/environment/environment.h"
#include "api/priority.h"
@@ -65,8 +65,8 @@
const rtc::CopyOnWriteBuffer& payload) override;
bool ReadyToSendData() override;
int max_message_size() const override;
- absl::optional<int> max_outbound_streams() const override;
- absl::optional<int> max_inbound_streams() const override;
+ std::optional<int> max_outbound_streams() const override;
+ std::optional<int> max_inbound_streams() const override;
size_t buffered_amount(int sid) const override;
size_t buffered_amount_low_threshold(int sid) const override;
void SetBufferedAmountLowThreshold(int sid, size_t bytes) override;
diff --git a/media/sctp/sctp_transport_internal.h b/media/sctp/sctp_transport_internal.h
index 7596d67..2dad0cd 100644
--- a/media/sctp/sctp_transport_internal.h
+++ b/media/sctp/sctp_transport_internal.h
@@ -137,10 +137,10 @@
// Returns the current max message size, set with Start().
virtual int max_message_size() const = 0;
// Returns the current negotiated max # of outbound streams.
- // Will return absl::nullopt if negotiation is incomplete.
- virtual absl::optional<int> max_outbound_streams() const = 0;
+ // Will return std::nullopt if negotiation is incomplete.
+ virtual std::optional<int> max_outbound_streams() const = 0;
// Returns the current negotiated max # of inbound streams.
- virtual absl::optional<int> max_inbound_streams() const = 0;
+ virtual std::optional<int> max_inbound_streams() const = 0;
// Returns the amount of buffered data in the send queue for a stream.
virtual size_t buffered_amount(int sid) const = 0;
virtual size_t buffered_amount_low_threshold(int sid) const = 0;
diff --git a/modules/BUILD.gn b/modules/BUILD.gn
index 540e856..21f866d 100644
--- a/modules/BUILD.gn
+++ b/modules/BUILD.gn
@@ -30,7 +30,6 @@
rtc_source_set("module_api_public") {
sources = [ "include/module_common_types_public.h" ]
- deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
}
rtc_source_set("module_api") {
diff --git a/modules/audio_coding/BUILD.gn b/modules/audio_coding/BUILD.gn
index a28152b..087043b 100644
--- a/modules/audio_coding/BUILD.gn
+++ b/modules/audio_coding/BUILD.gn
@@ -62,7 +62,6 @@
"../../system_wrappers:metrics",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -76,7 +75,6 @@
"../../api/audio_codecs:audio_codecs_api",
"../../rtc_base:buffer",
"../../rtc_base:checks",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -109,7 +107,6 @@
"../../api/units:time_delta",
"../../common_audio",
"../../rtc_base:checks",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -131,7 +128,6 @@
"../../rtc_base:checks",
"../../rtc_base:logging",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -152,7 +148,6 @@
"../../api/units:time_delta",
"../../rtc_base:buffer",
"../../rtc_base:checks",
- "//third_party/abseil-cpp/absl/types:optional",
]
public_deps += [ ":g711_c" ] # no-presubmit-check TODO(webrtc:8603)
}
@@ -185,7 +180,6 @@
"../../rtc_base:buffer",
"../../rtc_base:checks",
"../../rtc_base:safe_conversions",
- "//third_party/abseil-cpp/absl/types:optional",
]
public_deps += [ ":g722_c" ] # no-presubmit-check TODO(webrtc:8603)
}
@@ -220,7 +214,6 @@
"../../rtc_base:checks",
"../../rtc_base:logging",
"../../rtc_base:safe_conversions",
- "//third_party/abseil-cpp/absl/types:optional",
]
public_deps += [ ":ilbc_c" ] # no-presubmit-check TODO(webrtc:8603)
}
@@ -452,7 +445,6 @@
"../../rtc_base:checks",
"../../rtc_base:stringutils",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -488,7 +480,6 @@
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
public_deps += # no-presubmit-check TODO(webrtc:8603)
[ ":webrtc_opus_wrapper" ]
@@ -522,7 +513,6 @@
"../../rtc_base:stringutils",
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/strings",
- "//third_party/abseil-cpp/absl/types:optional",
]
public_deps += # no-presubmit-check TODO(webrtc:8603)
[ ":webrtc_opus_wrapper" ]
@@ -574,7 +564,6 @@
"audio_network_adaptor/audio_network_adaptor_config.cc",
"audio_network_adaptor/include/audio_network_adaptor_config.h",
]
- deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
}
rtc_library("audio_network_adaptor") {
@@ -624,7 +613,6 @@
"../../system_wrappers:field_trial",
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
if (rtc_enable_protobuf) {
@@ -739,7 +727,6 @@
"../../system_wrappers:metrics",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -802,7 +789,6 @@
"../../rtc_base:stringutils",
"../../system_wrappers",
"../rtp_rtcp:rtp_rtcp_format",
- "//third_party/abseil-cpp/absl/types:optional",
]
defines = audio_codec_defines
}
@@ -843,7 +829,6 @@
"../../test:rtp_test_utils",
"../rtp_rtcp:rtp_rtcp_format",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -876,7 +861,6 @@
"../rtp_rtcp",
"../rtp_rtcp:rtp_rtcp_format",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
if (rtc_enable_protobuf) {
@@ -961,7 +945,6 @@
"//testing/gtest",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
defines = audio_coding_defines
@@ -1244,7 +1227,6 @@
"../../rtc_base:refcount",
"../../test:fileutils",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
sources = [
"neteq/tools/neteq_test_factory.cc",
@@ -1278,7 +1260,6 @@
"//third_party/abseil-cpp/absl/flags:flag",
"//third_party/abseil-cpp/absl/flags:parse",
"//third_party/abseil-cpp/absl/strings",
- "//third_party/abseil-cpp/absl/types:optional",
]
sources = [ "neteq/tools/neteq_rtpplay.cc" ]
}
@@ -1751,7 +1732,6 @@
"//third_party/abseil-cpp/absl/flags:flag",
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
defines = audio_coding_defines
diff --git a/modules/audio_coding/acm2/acm_receiver.cc b/modules/audio_coding/acm2/acm_receiver.cc
index e203a32..0946d79 100644
--- a/modules/audio_coding/acm2/acm_receiver.cc
+++ b/modules/audio_coding/acm2/acm_receiver.cc
@@ -90,11 +90,11 @@
return neteq_->GetBaseMinimumDelayMs();
}
-absl::optional<int> AcmReceiver::last_packet_sample_rate_hz() const {
- absl::optional<NetEq::DecoderFormat> decoder =
+std::optional<int> AcmReceiver::last_packet_sample_rate_hz() const {
+ std::optional<NetEq::DecoderFormat> decoder =
neteq_->GetCurrentDecoderFormat();
if (!decoder) {
- return absl::nullopt;
+ return std::nullopt;
}
return decoder->sample_rate_hz;
}
@@ -194,7 +194,7 @@
neteq_->FlushBuffers();
}
-absl::optional<uint32_t> AcmReceiver::GetPlayoutTimestamp() {
+std::optional<uint32_t> AcmReceiver::GetPlayoutTimestamp() {
return neteq_->GetPlayoutTimestamp();
}
@@ -206,12 +206,11 @@
return neteq_->TargetDelayMs();
}
-absl::optional<std::pair<int, SdpAudioFormat>> AcmReceiver::LastDecoder()
- const {
- absl::optional<NetEq::DecoderFormat> decoder =
+std::optional<std::pair<int, SdpAudioFormat>> AcmReceiver::LastDecoder() const {
+ std::optional<NetEq::DecoderFormat> decoder =
neteq_->GetCurrentDecoderFormat();
if (!decoder) {
- return absl::nullopt;
+ return std::nullopt;
}
return std::make_pair(decoder->payload_type, decoder->sdp_format);
}
diff --git a/modules/audio_coding/acm2/acm_receiver.h b/modules/audio_coding/acm2/acm_receiver.h
index 1e14248..2d0dffb 100644
--- a/modules/audio_coding/acm2/acm_receiver.h
+++ b/modules/audio_coding/acm2/acm_receiver.h
@@ -16,11 +16,11 @@
#include <array>
#include <map>
#include <memory>
+#include <optional>
#include <string>
#include <utility>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/audio/audio_frame.h"
#include "api/audio_codecs/audio_decoder.h"
@@ -147,7 +147,7 @@
// packet. If no packet of a registered non-CNG codec has been received, the
// return value is empty. Also, if the decoder was unregistered since the last
// packet was inserted, the return value is empty.
- absl::optional<int> last_packet_sample_rate_hz() const;
+ std::optional<int> last_packet_sample_rate_hz() const;
// Returns last_output_sample_rate_hz from the NetEq instance.
int last_output_sample_rate_hz() const;
@@ -168,7 +168,7 @@
// Returns the RTP timestamp for the last sample delivered by GetAudio().
// The return value will be empty if no valid timestamp is available.
- absl::optional<uint32_t> GetPlayoutTimestamp();
+ std::optional<uint32_t> GetPlayoutTimestamp();
// Returns the current total delay from NetEq (packet buffer and sync buffer)
// in ms, with smoothing applied to even out short-time fluctuations due to
@@ -183,9 +183,9 @@
//
// Get payload type and format of the last non-CNG/non-DTMF received payload.
- // If no non-CNG/non-DTMF packet is received absl::nullopt is returned.
+ // If no non-CNG/non-DTMF packet is received std::nullopt is returned.
//
- absl::optional<std::pair<int, SdpAudioFormat>> LastDecoder() const;
+ std::optional<std::pair<int, SdpAudioFormat>> LastDecoder() const;
//
// Enable NACK and set the maximum size of the NACK list. If NACK is already
diff --git a/modules/audio_coding/acm2/acm_receiver_unittest.cc b/modules/audio_coding/acm2/acm_receiver_unittest.cc
index 07fcb05..55b5c49 100644
--- a/modules/audio_coding/acm2/acm_receiver_unittest.cc
+++ b/modules/audio_coding/acm2/acm_receiver_unittest.cc
@@ -12,8 +12,8 @@
#include <algorithm> // std::min
#include <memory>
+#include <optional>
-#include "absl/types/optional.h"
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
#include "api/environment/environment.h"
@@ -67,7 +67,7 @@
const SdpAudioFormat& format,
const std::map<int, int> cng_payload_types = {}) {
// Create the speech encoder.
- absl::optional<AudioCodecInfo> info =
+ std::optional<AudioCodecInfo> info =
encoder_factory_->QueryAudioEncoder(format);
RTC_CHECK(info.has_value());
std::unique_ptr<AudioEncoder> enc =
@@ -264,7 +264,7 @@
}
// No audio payload is received.
- EXPECT_EQ(absl::nullopt, receiver_->LastDecoder());
+ EXPECT_EQ(std::nullopt, receiver_->LastDecoder());
// Start with sending DTX.
packet_sent_ = false;
@@ -275,8 +275,8 @@
EXPECT_EQ(AudioFrameType::kAudioFrameCN, last_frame_type_);
// Has received, only, DTX. Last Audio codec is undefined.
- EXPECT_EQ(absl::nullopt, receiver_->LastDecoder());
- EXPECT_EQ(absl::nullopt, receiver_->last_packet_sample_rate_hz());
+ EXPECT_EQ(std::nullopt, receiver_->LastDecoder());
+ EXPECT_EQ(std::nullopt, receiver_->last_packet_sample_rate_hz());
for (size_t i = 0; i < codecs.size(); ++i) {
// Set DTX off to send audio payload.
diff --git a/modules/audio_coding/acm2/audio_coding_module.cc b/modules/audio_coding/acm2/audio_coding_module.cc
index 3635701..e1a775d 100644
--- a/modules/audio_coding/acm2/audio_coding_module.cc
+++ b/modules/audio_coding/acm2/audio_coding_module.cc
@@ -108,7 +108,7 @@
// TODO(bugs.webrtc.org/10739): change `absolute_capture_timestamp_ms` to
// int64_t when it always receives a valid value.
int Encode(const InputData& input_data,
- absl::optional<int64_t> absolute_capture_timestamp_ms)
+ std::optional<int64_t> absolute_capture_timestamp_ms)
RTC_EXCLUSIVE_LOCKS_REQUIRED(acm_mutex_);
bool HaveValidEncoder(absl::string_view caller_name) const
@@ -152,7 +152,7 @@
bool first_frame_ RTC_GUARDED_BY(acm_mutex_);
uint32_t last_timestamp_ RTC_GUARDED_BY(acm_mutex_);
uint32_t last_rtp_timestamp_ RTC_GUARDED_BY(acm_mutex_);
- absl::optional<int64_t> absolute_capture_timestamp_ms_
+ std::optional<int64_t> absolute_capture_timestamp_ms_
RTC_GUARDED_BY(acm_mutex_);
Mutex callback_mutex_;
@@ -198,7 +198,7 @@
int32_t AudioCodingModuleImpl::Encode(
const InputData& input_data,
- absl::optional<int64_t> absolute_capture_timestamp_ms) {
+ std::optional<int64_t> absolute_capture_timestamp_ms) {
// TODO(bugs.webrtc.org/10739): add dcheck that
// `audio_frame.absolute_capture_timestamp_ms()` always has a value.
AudioEncoder::EncodedInfo encoded_info;
@@ -280,7 +280,7 @@
absolute_capture_timestamp_ms_.value_or(-1));
}
}
- absolute_capture_timestamp_ms_ = absl::nullopt;
+ absolute_capture_timestamp_ms_ = std::nullopt;
previous_pltype_ = encoded_info.payload_type;
return static_cast<int32_t>(encode_buffer_.size());
}
diff --git a/modules/audio_coding/acm2/audio_coding_module_unittest.cc b/modules/audio_coding/acm2/audio_coding_module_unittest.cc
index b30788c..c8ef9df 100644
--- a/modules/audio_coding/acm2/audio_coding_module_unittest.cc
+++ b/modules/audio_coding/acm2/audio_coding_module_unittest.cc
@@ -249,7 +249,7 @@
RTPHeader rtp_header_;
AudioFrame input_frame_;
- absl::optional<SdpAudioFormat> audio_format_;
+ std::optional<SdpAudioFormat> audio_format_;
int pac_size_ = -1;
};
diff --git a/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.cc b/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.cc
index 64163f9..3cc80da 100644
--- a/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.cc
+++ b/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.cc
@@ -107,7 +107,7 @@
controller->MakeDecision(&config);
// Update ANA stats.
- auto increment_opt = [](absl::optional<uint32_t>& a) {
+ auto increment_opt = [](std::optional<uint32_t>& a) {
a = a.value_or(0) + 1;
};
if (prev_config_) {
diff --git a/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h b/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h
index 664e76b..b4ec355 100644
--- a/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h
+++ b/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h
@@ -14,8 +14,8 @@
#include <stdio.h>
#include <memory>
+#include <optional>
-#include "absl/types/optional.h"
#include "api/audio_codecs/audio_encoder.h"
#include "modules/audio_coding/audio_network_adaptor/controller.h"
#include "modules/audio_coding/audio_network_adaptor/debug_dump_writer.h"
@@ -79,7 +79,7 @@
Controller::NetworkMetrics last_metrics_;
- absl::optional<AudioEncoderRuntimeConfig> prev_config_;
+ std::optional<AudioEncoderRuntimeConfig> prev_config_;
ANAStats stats_;
};
diff --git a/modules/audio_coding/audio_network_adaptor/bitrate_controller.h b/modules/audio_coding/audio_network_adaptor/bitrate_controller.h
index c103214..6e91ec2 100644
--- a/modules/audio_coding/audio_network_adaptor/bitrate_controller.h
+++ b/modules/audio_coding/audio_network_adaptor/bitrate_controller.h
@@ -13,7 +13,8 @@
#include <stddef.h>
-#include "absl/types/optional.h"
+#include <optional>
+
#include "modules/audio_coding/audio_network_adaptor/controller.h"
#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h"
@@ -49,8 +50,8 @@
const Config config_;
int bitrate_bps_;
int frame_length_ms_;
- absl::optional<int> target_audio_bitrate_bps_;
- absl::optional<size_t> overhead_bytes_per_packet_;
+ std::optional<int> target_audio_bitrate_bps_;
+ std::optional<size_t> overhead_bytes_per_packet_;
};
} // namespace audio_network_adaptor
diff --git a/modules/audio_coding/audio_network_adaptor/bitrate_controller_unittest.cc b/modules/audio_coding/audio_network_adaptor/bitrate_controller_unittest.cc
index 9c593b8..e847f87 100644
--- a/modules/audio_coding/audio_network_adaptor/bitrate_controller_unittest.cc
+++ b/modules/audio_coding/audio_network_adaptor/bitrate_controller_unittest.cc
@@ -21,8 +21,8 @@
void UpdateNetworkMetrics(
BitrateController* controller,
- const absl::optional<int>& target_audio_bitrate_bps,
- const absl::optional<size_t>& overhead_bytes_per_packet) {
+ const std::optional<int>& target_audio_bitrate_bps,
+ const std::optional<size_t>& overhead_bytes_per_packet) {
// UpdateNetworkMetrics can accept multiple network metric updates at once.
// However, currently, the most used case is to update one metric at a time.
// To reflect this fact, we separate the calls.
@@ -39,7 +39,7 @@
}
void CheckDecision(BitrateController* controller,
- const absl::optional<int>& frame_length_ms,
+ const std::optional<int>& frame_length_ms,
int expected_bitrate_bps) {
AudioEncoderRuntimeConfig config;
config.frame_length_ms = frame_length_ms;
@@ -59,7 +59,7 @@
constexpr size_t kOverheadBytesPerPacket = 64;
BitrateController controller(BitrateController::Config(
kInitialBitrateBps, kInitialFrameLengthMs, 0, 0));
- UpdateNetworkMetrics(&controller, absl::nullopt, kOverheadBytesPerPacket);
+ UpdateNetworkMetrics(&controller, std::nullopt, kOverheadBytesPerPacket);
CheckDecision(&controller, kInitialFrameLengthMs * 2, kInitialBitrateBps);
}
@@ -69,7 +69,7 @@
constexpr int kTargetBitrateBps = 48000;
BitrateController controller(BitrateController::Config(
kInitialBitrateBps, kInitialFrameLengthMs, 0, 0));
- UpdateNetworkMetrics(&controller, kTargetBitrateBps, absl::nullopt);
+ UpdateNetworkMetrics(&controller, kTargetBitrateBps, std::nullopt);
CheckDecision(&controller, kInitialFrameLengthMs * 2, kInitialBitrateBps);
}
@@ -121,7 +121,7 @@
1000 /
kInitialFrameLengthMs;
UpdateNetworkMetrics(&controller, kTargetBitrateBps, kOverheadBytesPerPacket);
- CheckDecision(&controller, absl::nullopt, kBitrateBps);
+ CheckDecision(&controller, std::nullopt, kBitrateBps);
}
TEST(AnaBitrateControllerTest, IncreaseBitrateOnFrameLengthIncreased) {
@@ -135,7 +135,7 @@
1000 /
kInitialFrameLengthMs;
UpdateNetworkMetrics(&controller, kTargetBitrateBps, kOverheadBytesPerPacket);
- CheckDecision(&controller, absl::nullopt, kBitrateBps);
+ CheckDecision(&controller, std::nullopt, kBitrateBps);
constexpr int kFrameLengthMs = 60;
constexpr size_t kPacketOverheadRateDiff =
@@ -157,7 +157,7 @@
1000 /
kInitialFrameLengthMs;
UpdateNetworkMetrics(&controller, kTargetBitrateBps, kOverheadBytesPerPacket);
- CheckDecision(&controller, absl::nullopt, kBitrateBps);
+ CheckDecision(&controller, std::nullopt, kBitrateBps);
constexpr int kFrameLengthMs = 20;
constexpr size_t kPacketOverheadRateDiff =
diff --git a/modules/audio_coding/audio_network_adaptor/channel_controller.h b/modules/audio_coding/audio_network_adaptor/channel_controller.h
index 3cd4bb7..a877d77 100644
--- a/modules/audio_coding/audio_network_adaptor/channel_controller.h
+++ b/modules/audio_coding/audio_network_adaptor/channel_controller.h
@@ -13,7 +13,8 @@
#include <stddef.h>
-#include "absl/types/optional.h"
+#include <optional>
+
#include "modules/audio_coding/audio_network_adaptor/controller.h"
#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h"
@@ -50,7 +51,7 @@
private:
const Config config_;
size_t channels_to_encode_;
- absl::optional<int> uplink_bandwidth_bps_;
+ std::optional<int> uplink_bandwidth_bps_;
};
} // namespace webrtc
diff --git a/modules/audio_coding/audio_network_adaptor/channel_controller_unittest.cc b/modules/audio_coding/audio_network_adaptor/channel_controller_unittest.cc
index 21504bc..1803d44 100644
--- a/modules/audio_coding/audio_network_adaptor/channel_controller_unittest.cc
+++ b/modules/audio_coding/audio_network_adaptor/channel_controller_unittest.cc
@@ -33,7 +33,7 @@
}
void CheckDecision(ChannelController* controller,
- const absl::optional<int>& uplink_bandwidth_bps,
+ const std::optional<int>& uplink_bandwidth_bps,
size_t expected_num_channels) {
if (uplink_bandwidth_bps) {
Controller::NetworkMetrics network_metrics;
@@ -50,7 +50,7 @@
TEST(ChannelControllerTest, OutputInitValueWhenUplinkBandwidthUnknown) {
constexpr int kInitChannels = 2;
auto controller = CreateChannelController(kInitChannels);
- CheckDecision(controller.get(), absl::nullopt, kInitChannels);
+ CheckDecision(controller.get(), std::nullopt, kInitChannels);
}
TEST(ChannelControllerTest, SwitchTo2ChannelsOnHighUplinkBandwidth) {
diff --git a/modules/audio_coding/audio_network_adaptor/controller.h b/modules/audio_coding/audio_network_adaptor/controller.h
index b70ada0..fc2cba7 100644
--- a/modules/audio_coding/audio_network_adaptor/controller.h
+++ b/modules/audio_coding/audio_network_adaptor/controller.h
@@ -11,7 +11,8 @@
#ifndef MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_CONTROLLER_H_
#define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_CONTROLLER_H_
-#include "absl/types/optional.h"
+#include <optional>
+
#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
namespace webrtc {
@@ -21,11 +22,11 @@
struct NetworkMetrics {
NetworkMetrics();
~NetworkMetrics();
- absl::optional<int> uplink_bandwidth_bps;
- absl::optional<float> uplink_packet_loss_fraction;
- absl::optional<int> target_audio_bitrate_bps;
- absl::optional<int> rtt_ms;
- absl::optional<size_t> overhead_bytes_per_packet;
+ std::optional<int> uplink_bandwidth_bps;
+ std::optional<float> uplink_packet_loss_fraction;
+ std::optional<int> target_audio_bitrate_bps;
+ std::optional<int> rtt_ms;
+ std::optional<size_t> overhead_bytes_per_packet;
};
virtual ~Controller() = default;
diff --git a/modules/audio_coding/audio_network_adaptor/controller_manager.cc b/modules/audio_coding/audio_network_adaptor/controller_manager.cc
index 793c73a..40a48a0 100644
--- a/modules/audio_coding/audio_network_adaptor/controller_manager.cc
+++ b/modules/audio_coding/audio_network_adaptor/controller_manager.cc
@@ -338,7 +338,7 @@
const std::map<const Controller*, std::pair<int, float>>& scoring_points)
: config_(config),
controllers_(std::move(controllers)),
- last_reordering_time_ms_(absl::nullopt),
+ last_reordering_time_ms_(std::nullopt),
last_scoring_point_(0, 0.0) {
for (auto& controller : controllers_)
default_sorted_controllers_.push_back(controller.get());
diff --git a/modules/audio_coding/audio_network_adaptor/controller_manager.h b/modules/audio_coding/audio_network_adaptor/controller_manager.h
index 47e8e0f..e03bd05 100644
--- a/modules/audio_coding/audio_network_adaptor/controller_manager.h
+++ b/modules/audio_coding/audio_network_adaptor/controller_manager.h
@@ -107,7 +107,7 @@
std::vector<std::unique_ptr<Controller>> controllers_;
- absl::optional<int64_t> last_reordering_time_ms_;
+ std::optional<int64_t> last_reordering_time_ms_;
ScoringPoint last_scoring_point_;
std::vector<Controller*> default_sorted_controllers_;
diff --git a/modules/audio_coding/audio_network_adaptor/controller_manager_unittest.cc b/modules/audio_coding/audio_network_adaptor/controller_manager_unittest.cc
index f399511..dfc3d6f 100644
--- a/modules/audio_coding/audio_network_adaptor/controller_manager_unittest.cc
+++ b/modules/audio_coding/audio_network_adaptor/controller_manager_unittest.cc
@@ -86,8 +86,8 @@
// exists in the vector.
void CheckControllersOrder(
ControllerManagerStates* states,
- const absl::optional<int>& uplink_bandwidth_bps,
- const absl::optional<float>& uplink_packet_loss_fraction,
+ const std::optional<int>& uplink_bandwidth_bps,
+ const std::optional<float>& uplink_packet_loss_fraction,
const std::vector<int>& expected_order) {
RTC_DCHECK_EQ(kNumControllers, expected_order.size());
Controller::NetworkMetrics metrics;
@@ -123,7 +123,7 @@
auto states = CreateControllerManager();
// `network_metrics` are empty, and the controllers are supposed to follow the
// default order.
- CheckControllersOrder(&states, absl::nullopt, absl::nullopt, {0, 1, 2, 3});
+ CheckControllersOrder(&states, std::nullopt, std::nullopt, {0, 1, 2, 3});
}
TEST(ControllerManagerTest, ControllersWithoutCharPointAtEndAndInDefaultOrder) {
diff --git a/modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc b/modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc
index 5ffbee2..f5fdca7 100644
--- a/modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc
+++ b/modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc
@@ -10,9 +10,9 @@
#include "modules/audio_coding/audio_network_adaptor/debug_dump_writer.h"
+#include <optional>
#include <string>
-#include "absl/types/optional.h"
#include "rtc_base/checks.h"
#include "rtc_base/numerics/safe_conversions.h"
#include "rtc_base/system/file_wrapper.h"
diff --git a/modules/audio_coding/audio_network_adaptor/dtx_controller.h b/modules/audio_coding/audio_network_adaptor/dtx_controller.h
index b8a8e47..f43fd02 100644
--- a/modules/audio_coding/audio_network_adaptor/dtx_controller.h
+++ b/modules/audio_coding/audio_network_adaptor/dtx_controller.h
@@ -11,7 +11,8 @@
#ifndef MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_DTX_CONTROLLER_H_
#define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_DTX_CONTROLLER_H_
-#include "absl/types/optional.h"
+#include <optional>
+
#include "modules/audio_coding/audio_network_adaptor/controller.h"
#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h"
@@ -44,7 +45,7 @@
private:
const Config config_;
bool dtx_enabled_;
- absl::optional<int> uplink_bandwidth_bps_;
+ std::optional<int> uplink_bandwidth_bps_;
};
} // namespace webrtc
diff --git a/modules/audio_coding/audio_network_adaptor/dtx_controller_unittest.cc b/modules/audio_coding/audio_network_adaptor/dtx_controller_unittest.cc
index 567df6f..46c8f8b 100644
--- a/modules/audio_coding/audio_network_adaptor/dtx_controller_unittest.cc
+++ b/modules/audio_coding/audio_network_adaptor/dtx_controller_unittest.cc
@@ -31,7 +31,7 @@
}
void CheckDecision(DtxController* controller,
- const absl::optional<int>& uplink_bandwidth_bps,
+ const std::optional<int>& uplink_bandwidth_bps,
bool expected_dtx_enabled) {
if (uplink_bandwidth_bps) {
Controller::NetworkMetrics network_metrics;
@@ -48,7 +48,7 @@
TEST(DtxControllerTest, OutputInitValueWhenUplinkBandwidthUnknown) {
constexpr bool kInitialDtxEnabled = true;
auto controller = CreateController(kInitialDtxEnabled);
- CheckDecision(controller.get(), absl::nullopt, kInitialDtxEnabled);
+ CheckDecision(controller.get(), std::nullopt, kInitialDtxEnabled);
}
TEST(DtxControllerTest, TurnOnDtxForLowUplinkBandwidth) {
diff --git a/modules/audio_coding/audio_network_adaptor/event_log_writer.cc b/modules/audio_coding/audio_network_adaptor/event_log_writer.cc
index 0a79484..1bef64e 100644
--- a/modules/audio_coding/audio_network_adaptor/event_log_writer.cc
+++ b/modules/audio_coding/audio_network_adaptor/event_log_writer.cc
@@ -15,9 +15,9 @@
#include <algorithm>
#include <cstdlib>
#include <memory>
+#include <optional>
#include <utility>
-#include "absl/types/optional.h"
#include "api/rtc_event_log/rtc_event.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "logging/rtc_event_log/events/rtc_event_audio_network_adaptation.h"
diff --git a/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.cc b/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.cc
index 2134dad..bcfbd75 100644
--- a/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.cc
+++ b/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.cc
@@ -68,7 +68,7 @@
}
bool FecControllerPlrBased::FecEnablingDecision(
- const absl::optional<float>& packet_loss) const {
+ const std::optional<float>& packet_loss) const {
if (!uplink_bandwidth_bps_ || !packet_loss) {
return false;
} else {
@@ -79,7 +79,7 @@
}
bool FecControllerPlrBased::FecDisablingDecision(
- const absl::optional<float>& packet_loss) const {
+ const std::optional<float>& packet_loss) const {
if (!uplink_bandwidth_bps_ || !packet_loss) {
return false;
} else {
diff --git a/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.h b/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.h
index 0c57ad1..71b2d69 100644
--- a/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.h
+++ b/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.h
@@ -12,8 +12,8 @@
#define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_FEC_CONTROLLER_PLR_BASED_H_
#include <memory>
+#include <optional>
-#include "absl/types/optional.h"
#include "common_audio/smoothing_filter.h"
#include "modules/audio_coding/audio_network_adaptor/controller.h"
#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h"
@@ -60,12 +60,12 @@
void MakeDecision(AudioEncoderRuntimeConfig* config) override;
private:
- bool FecEnablingDecision(const absl::optional<float>& packet_loss) const;
- bool FecDisablingDecision(const absl::optional<float>& packet_loss) const;
+ bool FecEnablingDecision(const std::optional<float>& packet_loss) const;
+ bool FecDisablingDecision(const std::optional<float>& packet_loss) const;
const Config config_;
bool fec_enabled_;
- absl::optional<int> uplink_bandwidth_bps_;
+ std::optional<int> uplink_bandwidth_bps_;
const std::unique_ptr<SmoothingFilter> packet_loss_smoother_;
};
diff --git a/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based_unittest.cc b/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based_unittest.cc
index 743b087..f110477 100644
--- a/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based_unittest.cc
+++ b/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based_unittest.cc
@@ -79,8 +79,8 @@
}
void UpdateNetworkMetrics(FecControllerPlrBasedTestStates* states,
- const absl::optional<int>& uplink_bandwidth_bps,
- const absl::optional<float>& uplink_packet_loss) {
+ const std::optional<int>& uplink_bandwidth_bps,
+ const std::optional<float>& uplink_packet_loss) {
// UpdateNetworkMetrics can accept multiple network metric updates at once.
// However, currently, the most used case is to update one metric at a time.
// To reflect this fact, we separate the calls.
@@ -132,7 +132,7 @@
kEnablingPacketLossAtLowBw - kEpsilon, kEnablingPacketLossAtLowBw,
kEnablingPacketLossAtLowBw + kEpsilon}) {
auto states = CreateFecControllerPlrBased(initial_fec_enabled);
- UpdateNetworkMetrics(&states, absl::nullopt, packet_loss);
+ UpdateNetworkMetrics(&states, std::nullopt, packet_loss);
CheckDecision(&states, initial_fec_enabled, packet_loss);
}
}
@@ -147,7 +147,7 @@
kDisablingBandwidthLow + 1, kEnablingBandwidthLow - 1,
kEnablingBandwidthLow, kEnablingBandwidthLow + 1}) {
auto states = CreateFecControllerPlrBased(initial_fec_enabled);
- UpdateNetworkMetrics(&states, bandwidth, absl::nullopt);
+ UpdateNetworkMetrics(&states, bandwidth, std::nullopt);
CheckDecision(&states, initial_fec_enabled, 0.0);
}
}
diff --git a/modules/audio_coding/audio_network_adaptor/frame_length_controller.h b/modules/audio_coding/audio_network_adaptor/frame_length_controller.h
index 04693f8..5abfb83 100644
--- a/modules/audio_coding/audio_network_adaptor/frame_length_controller.h
+++ b/modules/audio_coding/audio_network_adaptor/frame_length_controller.h
@@ -14,9 +14,9 @@
#include <stddef.h>
#include <map>
+#include <optional>
#include <set>
-#include "absl/types/optional.h"
#include "modules/audio_coding/audio_network_adaptor/controller.h"
#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h"
@@ -77,11 +77,11 @@
std::set<int>::const_iterator frame_length_ms_;
- absl::optional<int> uplink_bandwidth_bps_;
+ std::optional<int> uplink_bandwidth_bps_;
- absl::optional<float> uplink_packet_loss_fraction_;
+ std::optional<float> uplink_packet_loss_fraction_;
- absl::optional<size_t> overhead_bytes_per_packet_;
+ std::optional<size_t> overhead_bytes_per_packet_;
// True if the previous frame length decision was an increase, otherwise
// false.
diff --git a/modules/audio_coding/audio_network_adaptor/frame_length_controller_unittest.cc b/modules/audio_coding/audio_network_adaptor/frame_length_controller_unittest.cc
index 2312393..2bf52b2 100644
--- a/modules/audio_coding/audio_network_adaptor/frame_length_controller_unittest.cc
+++ b/modules/audio_coding/audio_network_adaptor/frame_length_controller_unittest.cc
@@ -124,9 +124,9 @@
void UpdateNetworkMetrics(
FrameLengthController* controller,
- const absl::optional<int>& uplink_bandwidth_bps,
- const absl::optional<float>& uplink_packet_loss_fraction,
- const absl::optional<size_t>& overhead_bytes_per_packet) {
+ const std::optional<int>& uplink_bandwidth_bps,
+ const std::optional<float>& uplink_packet_loss_fraction,
+ const std::optional<size_t>& overhead_bytes_per_packet) {
// UpdateNetworkMetrics can accept multiple network metric updates at once.
// However, currently, the most used case is to update one metric at a time.
// To reflect this fact, we separate the calls.
@@ -160,14 +160,14 @@
auto controller = CreateController(CreateChangeCriteriaFor20msAnd60ms(),
kDefaultEncoderFrameLengthsMs, 60);
UpdateNetworkMetrics(controller.get(), kFl60msTo20msBandwidthBps,
- absl::nullopt, kOverheadBytesPerPacket);
+ std::nullopt, kOverheadBytesPerPacket);
CheckDecision(controller.get(), 20);
}
TEST(FrameLengthControllerTest, DecreaseTo20MsOnHighUplinkPacketLossFraction) {
auto controller = CreateController(CreateChangeCriteriaFor20msAnd60ms(),
kDefaultEncoderFrameLengthsMs, 60);
- UpdateNetworkMetrics(controller.get(), absl::nullopt,
+ UpdateNetworkMetrics(controller.get(), std::nullopt,
kFlDecreasingPacketLossFraction,
kOverheadBytesPerPacket);
CheckDecision(controller.get(), 20);
@@ -200,7 +200,7 @@
auto controller = CreateController(CreateChangeCriteriaFor40msAnd60ms(),
kDefaultEncoderFrameLengthsMs, 40);
UpdateNetworkMetrics(controller.get(), kFl60msTo40msBandwidthBps,
- absl::nullopt, kOverheadBytesPerPacket);
+ std::nullopt, kOverheadBytesPerPacket);
CheckDecision(controller.get(), 40);
}
@@ -316,11 +316,11 @@
kDefaultEncoderFrameLengthsMs, 120);
// It takes two steps for frame length to go from 120ms to 20ms.
UpdateNetworkMetrics(controller.get(), kFl60msTo20msBandwidthBps,
- absl::nullopt, kOverheadBytesPerPacket);
+ std::nullopt, kOverheadBytesPerPacket);
CheckDecision(controller.get(), 60);
UpdateNetworkMetrics(controller.get(), kFl60msTo20msBandwidthBps,
- absl::nullopt, kOverheadBytesPerPacket);
+ std::nullopt, kOverheadBytesPerPacket);
CheckDecision(controller.get(), 20);
}
@@ -328,12 +328,12 @@
auto controller = CreateController(CreateChangeCriteriaFor20ms60msAnd120ms(),
kDefaultEncoderFrameLengthsMs, 120);
// It takes two steps for frame length to go from 120ms to 20ms.
- UpdateNetworkMetrics(controller.get(), absl::nullopt,
+ UpdateNetworkMetrics(controller.get(), std::nullopt,
kFlDecreasingPacketLossFraction,
kOverheadBytesPerPacket);
CheckDecision(controller.get(), 60);
- UpdateNetworkMetrics(controller.get(), absl::nullopt,
+ UpdateNetworkMetrics(controller.get(), std::nullopt,
kFlDecreasingPacketLossFraction,
kOverheadBytesPerPacket);
CheckDecision(controller.get(), 20);
diff --git a/modules/audio_coding/audio_network_adaptor/frame_length_controller_v2.h b/modules/audio_coding/audio_network_adaptor/frame_length_controller_v2.h
index d7102b0..dc936c9 100644
--- a/modules/audio_coding/audio_network_adaptor/frame_length_controller_v2.h
+++ b/modules/audio_coding/audio_network_adaptor/frame_length_controller_v2.h
@@ -11,9 +11,9 @@
#ifndef MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_FRAME_LENGTH_CONTROLLER_V2_H_
#define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_FRAME_LENGTH_CONTROLLER_V2_H_
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "modules/audio_coding/audio_network_adaptor/controller.h"
#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
@@ -34,9 +34,9 @@
const int min_payload_bitrate_bps_;
const bool use_slow_adaptation_;
- absl::optional<int> uplink_bandwidth_bps_;
- absl::optional<int> target_bitrate_bps_;
- absl::optional<int> overhead_bytes_per_packet_;
+ std::optional<int> uplink_bandwidth_bps_;
+ std::optional<int> target_bitrate_bps_;
+ std::optional<int> overhead_bytes_per_packet_;
};
} // namespace webrtc
diff --git a/modules/audio_coding/audio_network_adaptor/frame_length_controller_v2_unittest.cc b/modules/audio_coding/audio_network_adaptor/frame_length_controller_v2_unittest.cc
index 1c88f47..1bec1ac 100644
--- a/modules/audio_coding/audio_network_adaptor/frame_length_controller_v2_unittest.cc
+++ b/modules/audio_coding/audio_network_adaptor/frame_length_controller_v2_unittest.cc
@@ -62,8 +62,8 @@
std::make_unique<FrameLengthControllerV2>(kANASupportedFrameLengths,
kMinPayloadBitrateBps,
/*use_slow_adaptation=*/false);
- absl::optional<int> target_audio_bitrate_bps_;
- absl::optional<int> overhead_bytes_per_packet_;
+ std::optional<int> target_audio_bitrate_bps_;
+ std::optional<int> overhead_bytes_per_packet_;
};
// Don't return any decision if we haven't received all required network
diff --git a/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h b/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h
index 346ed5d..ed07298 100644
--- a/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h
+++ b/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h
@@ -11,7 +11,8 @@
#ifndef MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_INCLUDE_AUDIO_NETWORK_ADAPTOR_H_
#define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_INCLUDE_AUDIO_NETWORK_ADAPTOR_H_
-#include "absl/types/optional.h"
+#include <optional>
+
#include "api/audio_codecs/audio_encoder.h"
#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h"
diff --git a/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h b/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h
index bd16292..5934f6e 100644
--- a/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h
+++ b/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h
@@ -13,7 +13,7 @@
#include <stddef.h>
-#include "absl/types/optional.h"
+#include <optional>
namespace webrtc {
@@ -23,18 +23,18 @@
~AudioEncoderRuntimeConfig();
AudioEncoderRuntimeConfig& operator=(const AudioEncoderRuntimeConfig& other);
bool operator==(const AudioEncoderRuntimeConfig& other) const;
- absl::optional<int> bitrate_bps;
- absl::optional<int> frame_length_ms;
+ std::optional<int> bitrate_bps;
+ std::optional<int> frame_length_ms;
// Note: This is what we tell the encoder. It doesn't have to reflect
// the actual NetworkMetrics; it's subject to our decision.
- absl::optional<float> uplink_packet_loss_fraction;
- absl::optional<bool> enable_fec;
- absl::optional<bool> enable_dtx;
+ std::optional<float> uplink_packet_loss_fraction;
+ std::optional<bool> enable_fec;
+ std::optional<bool> enable_dtx;
// Some encoders can encode fewer channels than the actual input to make
// better use of the bandwidth. `num_channels` sets the number of channels
// to encode.
- absl::optional<size_t> num_channels;
+ std::optional<size_t> num_channels;
// This is true if the last frame length change was an increase, and otherwise
// false.
diff --git a/modules/audio_coding/codecs/builtin_audio_decoder_factory_unittest.cc b/modules/audio_coding/codecs/builtin_audio_decoder_factory_unittest.cc
index d61c471..3383ffb 100644
--- a/modules/audio_coding/codecs/builtin_audio_decoder_factory_unittest.cc
+++ b/modules/audio_coding/codecs/builtin_audio_decoder_factory_unittest.cc
@@ -23,7 +23,7 @@
rtc::scoped_refptr<AudioDecoderFactory> adf =
CreateBuiltinAudioDecoderFactory();
ASSERT_TRUE(adf);
- EXPECT_FALSE(adf->Create(env, SdpAudioFormat("rey", 8000, 1), absl::nullopt));
+ EXPECT_FALSE(adf->Create(env, SdpAudioFormat("rey", 8000, 1), std::nullopt));
}
TEST(AudioDecoderFactoryTest, CreatePcmu) {
@@ -32,13 +32,12 @@
CreateBuiltinAudioDecoderFactory();
ASSERT_TRUE(adf);
// PCMu supports 8 kHz, and any number of channels.
+ EXPECT_FALSE(adf->Create(env, SdpAudioFormat("pcmu", 8000, 0), std::nullopt));
+ EXPECT_TRUE(adf->Create(env, SdpAudioFormat("pcmu", 8000, 1), std::nullopt));
+ EXPECT_TRUE(adf->Create(env, SdpAudioFormat("pcmu", 8000, 2), std::nullopt));
+ EXPECT_TRUE(adf->Create(env, SdpAudioFormat("pcmu", 8000, 3), std::nullopt));
EXPECT_FALSE(
- adf->Create(env, SdpAudioFormat("pcmu", 8000, 0), absl::nullopt));
- EXPECT_TRUE(adf->Create(env, SdpAudioFormat("pcmu", 8000, 1), absl::nullopt));
- EXPECT_TRUE(adf->Create(env, SdpAudioFormat("pcmu", 8000, 2), absl::nullopt));
- EXPECT_TRUE(adf->Create(env, SdpAudioFormat("pcmu", 8000, 3), absl::nullopt));
- EXPECT_FALSE(
- adf->Create(env, SdpAudioFormat("pcmu", 16000, 1), absl::nullopt));
+ adf->Create(env, SdpAudioFormat("pcmu", 16000, 1), std::nullopt));
}
TEST(AudioDecoderFactoryTest, CreatePcma) {
@@ -47,13 +46,12 @@
CreateBuiltinAudioDecoderFactory();
ASSERT_TRUE(adf);
// PCMa supports 8 kHz, and any number of channels.
+ EXPECT_FALSE(adf->Create(env, SdpAudioFormat("pcma", 8000, 0), std::nullopt));
+ EXPECT_TRUE(adf->Create(env, SdpAudioFormat("pcma", 8000, 1), std::nullopt));
+ EXPECT_TRUE(adf->Create(env, SdpAudioFormat("pcma", 8000, 2), std::nullopt));
+ EXPECT_TRUE(adf->Create(env, SdpAudioFormat("pcma", 8000, 3), std::nullopt));
EXPECT_FALSE(
- adf->Create(env, SdpAudioFormat("pcma", 8000, 0), absl::nullopt));
- EXPECT_TRUE(adf->Create(env, SdpAudioFormat("pcma", 8000, 1), absl::nullopt));
- EXPECT_TRUE(adf->Create(env, SdpAudioFormat("pcma", 8000, 2), absl::nullopt));
- EXPECT_TRUE(adf->Create(env, SdpAudioFormat("pcma", 8000, 3), absl::nullopt));
- EXPECT_FALSE(
- adf->Create(env, SdpAudioFormat("pcma", 16000, 1), absl::nullopt));
+ adf->Create(env, SdpAudioFormat("pcma", 16000, 1), std::nullopt));
}
TEST(AudioDecoderFactoryTest, CreateIlbc) {
@@ -62,15 +60,13 @@
CreateBuiltinAudioDecoderFactory();
ASSERT_TRUE(adf);
// iLBC supports 8 kHz, 1 channel.
- EXPECT_FALSE(
- adf->Create(env, SdpAudioFormat("ilbc", 8000, 0), absl::nullopt));
+ EXPECT_FALSE(adf->Create(env, SdpAudioFormat("ilbc", 8000, 0), std::nullopt));
#ifdef WEBRTC_CODEC_ILBC
- EXPECT_TRUE(adf->Create(env, SdpAudioFormat("ilbc", 8000, 1), absl::nullopt));
+ EXPECT_TRUE(adf->Create(env, SdpAudioFormat("ilbc", 8000, 1), std::nullopt));
#endif
+ EXPECT_FALSE(adf->Create(env, SdpAudioFormat("ilbc", 8000, 2), std::nullopt));
EXPECT_FALSE(
- adf->Create(env, SdpAudioFormat("ilbc", 8000, 2), absl::nullopt));
- EXPECT_FALSE(
- adf->Create(env, SdpAudioFormat("ilbc", 16000, 1), absl::nullopt));
+ adf->Create(env, SdpAudioFormat("ilbc", 16000, 1), std::nullopt));
}
TEST(AudioDecoderFactoryTest, CreateL16) {
@@ -83,10 +79,10 @@
const int num_channels[] = {1, 2, 3, 24};
for (int clockrate : clockrates) {
EXPECT_FALSE(
- adf->Create(env, SdpAudioFormat("l16", clockrate, 0), absl::nullopt));
+ adf->Create(env, SdpAudioFormat("l16", clockrate, 0), std::nullopt));
for (int channels : num_channels) {
EXPECT_TRUE(adf->Create(env, SdpAudioFormat("l16", clockrate, channels),
- absl::nullopt));
+ std::nullopt));
}
}
}
@@ -110,7 +106,7 @@
EXPECT_FALSE(adf->Create(
env,
SdpAudioFormat(codec, 32000, AudioDecoder::kMaxNumberOfChannels + 1),
- absl::nullopt));
+ std::nullopt));
}
}
@@ -120,20 +116,18 @@
CreateBuiltinAudioDecoderFactory();
ASSERT_TRUE(adf);
// g722 supports 8 kHz, 1-2 channels.
+ EXPECT_FALSE(adf->Create(env, SdpAudioFormat("g722", 8000, 0), std::nullopt));
+ EXPECT_TRUE(adf->Create(env, SdpAudioFormat("g722", 8000, 1), std::nullopt));
+ EXPECT_TRUE(adf->Create(env, SdpAudioFormat("g722", 8000, 2), std::nullopt));
+ EXPECT_FALSE(adf->Create(env, SdpAudioFormat("g722", 8000, 3), std::nullopt));
EXPECT_FALSE(
- adf->Create(env, SdpAudioFormat("g722", 8000, 0), absl::nullopt));
- EXPECT_TRUE(adf->Create(env, SdpAudioFormat("g722", 8000, 1), absl::nullopt));
- EXPECT_TRUE(adf->Create(env, SdpAudioFormat("g722", 8000, 2), absl::nullopt));
+ adf->Create(env, SdpAudioFormat("g722", 16000, 1), std::nullopt));
EXPECT_FALSE(
- adf->Create(env, SdpAudioFormat("g722", 8000, 3), absl::nullopt));
- EXPECT_FALSE(
- adf->Create(env, SdpAudioFormat("g722", 16000, 1), absl::nullopt));
- EXPECT_FALSE(
- adf->Create(env, SdpAudioFormat("g722", 32000, 1), absl::nullopt));
+ adf->Create(env, SdpAudioFormat("g722", 32000, 1), std::nullopt));
// g722 actually uses a 16 kHz sample rate instead of the nominal 8 kHz.
std::unique_ptr<AudioDecoder> dec =
- adf->Create(env, SdpAudioFormat("g722", 8000, 1), absl::nullopt);
+ adf->Create(env, SdpAudioFormat("g722", 8000, 1), std::nullopt);
EXPECT_EQ(16000, dec->SampleRateHz());
}
@@ -157,7 +151,7 @@
good,
static_cast<bool>(adf->Create(
env, SdpAudioFormat("opus", hz, channels, std::move(params)),
- absl::nullopt)));
+ std::nullopt)));
}
}
}
diff --git a/modules/audio_coding/codecs/cng/audio_encoder_cng.cc b/modules/audio_coding/codecs/cng/audio_encoder_cng.cc
index de82cd9..cbb0ba9 100644
--- a/modules/audio_coding/codecs/cng/audio_encoder_cng.cc
+++ b/modules/audio_coding/codecs/cng/audio_encoder_cng.cc
@@ -12,9 +12,9 @@
#include <cstdint>
#include <memory>
+#include <optional>
#include <utility>
-#include "absl/types/optional.h"
#include "api/units/time_delta.h"
#include "modules/audio_coding/codecs/cng/webrtc_cng.h"
#include "rtc_base/checks.h"
@@ -54,10 +54,9 @@
override;
void OnReceivedUplinkPacketLossFraction(
float uplink_packet_loss_fraction) override;
- void OnReceivedUplinkBandwidth(
- int target_audio_bitrate_bps,
- absl::optional<int64_t> bwe_period_ms) override;
- absl::optional<std::pair<TimeDelta, TimeDelta>> GetFrameLengthRange()
+ void OnReceivedUplinkBandwidth(int target_audio_bitrate_bps,
+ std::optional<int64_t> bwe_period_ms) override;
+ std::optional<std::pair<TimeDelta, TimeDelta>> GetFrameLengthRange()
const override;
private:
@@ -225,12 +224,12 @@
void AudioEncoderCng::OnReceivedUplinkBandwidth(
int target_audio_bitrate_bps,
- absl::optional<int64_t> bwe_period_ms) {
+ std::optional<int64_t> bwe_period_ms) {
speech_encoder_->OnReceivedUplinkBandwidth(target_audio_bitrate_bps,
bwe_period_ms);
}
-absl::optional<std::pair<TimeDelta, TimeDelta>>
+std::optional<std::pair<TimeDelta, TimeDelta>>
AudioEncoderCng::GetFrameLengthRange() const {
return speech_encoder_->GetFrameLengthRange();
}
diff --git a/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc b/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc
index c688004..4895c24 100644
--- a/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc
+++ b/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc
@@ -226,8 +226,8 @@
TEST_F(AudioEncoderCngTest, CheckTargetAudioBitratePropagation) {
CreateCng(MakeCngConfig());
EXPECT_CALL(*mock_encoder_,
- OnReceivedUplinkBandwidth(4711, absl::optional<int64_t>()));
- cng_->OnReceivedUplinkBandwidth(4711, absl::nullopt);
+ OnReceivedUplinkBandwidth(4711, std::optional<int64_t>()));
+ cng_->OnReceivedUplinkBandwidth(4711, std::nullopt);
}
TEST_F(AudioEncoderCngTest, CheckPacketLossFractionPropagation) {
@@ -241,7 +241,7 @@
auto expected_range =
std::make_pair(TimeDelta::Millis(20), TimeDelta::Millis(20));
EXPECT_CALL(*mock_encoder_, GetFrameLengthRange())
- .WillRepeatedly(Return(absl::make_optional(expected_range)));
+ .WillRepeatedly(Return(std::make_optional(expected_range)));
EXPECT_THAT(cng_->GetFrameLengthRange(), Optional(Eq(expected_range)));
}
diff --git a/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc b/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
index 65e2da4..42aa25c 100644
--- a/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
+++ b/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
@@ -89,7 +89,7 @@
speech_buffer_.clear();
}
-absl::optional<std::pair<TimeDelta, TimeDelta>>
+std::optional<std::pair<TimeDelta, TimeDelta>>
AudioEncoderPcm::GetFrameLengthRange() const {
return {{TimeDelta::Millis(num_10ms_frames_per_packet_ * 10),
TimeDelta::Millis(num_10ms_frames_per_packet_ * 10)}};
diff --git a/modules/audio_coding/codecs/g711/audio_encoder_pcm.h b/modules/audio_coding/codecs/g711/audio_encoder_pcm.h
index d50be4b..ad9a2ae 100644
--- a/modules/audio_coding/codecs/g711/audio_encoder_pcm.h
+++ b/modules/audio_coding/codecs/g711/audio_encoder_pcm.h
@@ -11,10 +11,10 @@
#ifndef MODULES_AUDIO_CODING_CODECS_G711_AUDIO_ENCODER_PCM_H_
#define MODULES_AUDIO_CODING_CODECS_G711_AUDIO_ENCODER_PCM_H_
+#include <optional>
#include <utility>
#include <vector>
-#include "absl/types/optional.h"
#include "api/audio_codecs/audio_encoder.h"
#include "api/units/time_delta.h"
@@ -43,7 +43,7 @@
size_t Max10MsFramesInAPacket() const override;
int GetTargetBitrate() const override;
void Reset() override;
- absl::optional<std::pair<TimeDelta, TimeDelta>> GetFrameLengthRange()
+ std::optional<std::pair<TimeDelta, TimeDelta>> GetFrameLengthRange()
const override;
protected:
diff --git a/modules/audio_coding/codecs/g722/audio_encoder_g722.cc b/modules/audio_coding/codecs/g722/audio_encoder_g722.cc
index b7d34ba..61d275e 100644
--- a/modules/audio_coding/codecs/g722/audio_encoder_g722.cc
+++ b/modules/audio_coding/codecs/g722/audio_encoder_g722.cc
@@ -79,7 +79,7 @@
RTC_CHECK_EQ(0, WebRtcG722_EncoderInit(encoders_[i].encoder));
}
-absl::optional<std::pair<TimeDelta, TimeDelta>>
+std::optional<std::pair<TimeDelta, TimeDelta>>
AudioEncoderG722Impl::GetFrameLengthRange() const {
return {{TimeDelta::Millis(num_10ms_frames_per_packet_ * 10),
TimeDelta::Millis(num_10ms_frames_per_packet_ * 10)}};
diff --git a/modules/audio_coding/codecs/g722/audio_encoder_g722.h b/modules/audio_coding/codecs/g722/audio_encoder_g722.h
index a932aa8..4db7a7d 100644
--- a/modules/audio_coding/codecs/g722/audio_encoder_g722.h
+++ b/modules/audio_coding/codecs/g722/audio_encoder_g722.h
@@ -12,9 +12,9 @@
#define MODULES_AUDIO_CODING_CODECS_G722_AUDIO_ENCODER_G722_H_
#include <memory>
+#include <optional>
#include <utility>
-#include "absl/types/optional.h"
#include "api/audio_codecs/audio_encoder.h"
#include "api/audio_codecs/g722/audio_encoder_g722_config.h"
#include "api/units/time_delta.h"
@@ -38,7 +38,7 @@
size_t Max10MsFramesInAPacket() const override;
int GetTargetBitrate() const override;
void Reset() override;
- absl::optional<std::pair<TimeDelta, TimeDelta>> GetFrameLengthRange()
+ std::optional<std::pair<TimeDelta, TimeDelta>> GetFrameLengthRange()
const override;
protected:
diff --git a/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc b/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc
index 9fbf42c..afe3aff 100644
--- a/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc
+++ b/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc
@@ -127,7 +127,7 @@
num_10ms_frames_buffered_ = 0;
}
-absl::optional<std::pair<TimeDelta, TimeDelta>>
+std::optional<std::pair<TimeDelta, TimeDelta>>
AudioEncoderIlbcImpl::GetFrameLengthRange() const {
return {{TimeDelta::Millis(num_10ms_frames_per_packet_ * 10),
TimeDelta::Millis(num_10ms_frames_per_packet_ * 10)}};
diff --git a/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h b/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h
index c8dfa2c..31c2047 100644
--- a/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h
+++ b/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h
@@ -14,9 +14,9 @@
#include <stddef.h>
#include <stdint.h>
+#include <optional>
#include <utility>
-#include "absl/types/optional.h"
#include "api/audio_codecs/audio_encoder.h"
#include "api/audio_codecs/ilbc/audio_encoder_ilbc_config.h"
#include "api/units/time_delta.h"
@@ -41,7 +41,7 @@
rtc::ArrayView<const int16_t> audio,
rtc::Buffer* encoded) override;
void Reset() override;
- absl::optional<std::pair<TimeDelta, TimeDelta>> GetFrameLengthRange()
+ std::optional<std::pair<TimeDelta, TimeDelta>> GetFrameLengthRange()
const override;
private:
diff --git a/modules/audio_coding/codecs/legacy_encoded_audio_frame.cc b/modules/audio_coding/codecs/legacy_encoded_audio_frame.cc
index dacf325..82417f8 100644
--- a/modules/audio_coding/codecs/legacy_encoded_audio_frame.cc
+++ b/modules/audio_coding/codecs/legacy_encoded_audio_frame.cc
@@ -29,7 +29,7 @@
return (ret < 0) ? 0 : static_cast<size_t>(ret);
}
-absl::optional<AudioDecoder::EncodedAudioFrame::DecodeResult>
+std::optional<AudioDecoder::EncodedAudioFrame::DecodeResult>
LegacyEncodedAudioFrame::Decode(rtc::ArrayView<int16_t> decoded) const {
AudioDecoder::SpeechType speech_type = AudioDecoder::kSpeech;
const int ret = decoder_->Decode(
@@ -37,7 +37,7 @@
decoded.size() * sizeof(int16_t), decoded.data(), &speech_type);
if (ret < 0)
- return absl::nullopt;
+ return std::nullopt;
return DecodeResult{static_cast<size_t>(ret), speech_type};
}
diff --git a/modules/audio_coding/codecs/legacy_encoded_audio_frame.h b/modules/audio_coding/codecs/legacy_encoded_audio_frame.h
index 21da136..1063637 100644
--- a/modules/audio_coding/codecs/legacy_encoded_audio_frame.h
+++ b/modules/audio_coding/codecs/legacy_encoded_audio_frame.h
@@ -14,9 +14,9 @@
#include <stddef.h>
#include <stdint.h>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/audio_codecs/audio_decoder.h"
#include "rtc_base/buffer.h"
@@ -37,7 +37,7 @@
size_t Duration() const override;
- absl::optional<DecodeResult> Decode(
+ std::optional<DecodeResult> Decode(
rtc::ArrayView<int16_t> decoded) const override;
// For testing:
diff --git a/modules/audio_coding/codecs/opus/audio_coder_opus_common.cc b/modules/audio_coding/codecs/opus/audio_coder_opus_common.cc
index 03c0218..7e127cc 100644
--- a/modules/audio_coding/codecs/opus/audio_coder_opus_common.cc
+++ b/modules/audio_coding/codecs/opus/audio_coder_opus_common.cc
@@ -14,18 +14,18 @@
namespace webrtc {
-absl::optional<std::string> GetFormatParameter(const SdpAudioFormat& format,
- absl::string_view param) {
+std::optional<std::string> GetFormatParameter(const SdpAudioFormat& format,
+ absl::string_view param) {
auto it = format.parameters.find(std::string(param));
if (it == format.parameters.end())
- return absl::nullopt;
+ return std::nullopt;
return it->second;
}
// Parses a comma-separated string "1,2,0,6" into a std::vector<unsigned char>.
template <>
-absl::optional<std::vector<unsigned char>> GetFormatParameter(
+std::optional<std::vector<unsigned char>> GetFormatParameter(
const SdpAudioFormat& format,
absl::string_view param) {
std::vector<unsigned char> result;
@@ -41,7 +41,7 @@
comma_separated_list.substr(pos, distance_to_next_comma);
auto conv = rtc::StringToNumber<int>(substring_with_number);
if (!conv.has_value()) {
- return absl::nullopt;
+ return std::nullopt;
}
result.push_back(*conv);
pos += substring_with_number.size() + 1;
diff --git a/modules/audio_coding/codecs/opus/audio_coder_opus_common.h b/modules/audio_coding/codecs/opus/audio_coder_opus_common.h
index 5ebb51b..708ae4a 100644
--- a/modules/audio_coding/codecs/opus/audio_coder_opus_common.h
+++ b/modules/audio_coding/codecs/opus/audio_coder_opus_common.h
@@ -11,29 +11,29 @@
#ifndef MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_CODER_OPUS_COMMON_H_
#define MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_CODER_OPUS_COMMON_H_
+#include <optional>
#include <string>
#include <utility>
#include <vector>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/audio_codecs/audio_decoder.h"
#include "api/audio_codecs/audio_format.h"
#include "rtc_base/string_to_number.h"
namespace webrtc {
-absl::optional<std::string> GetFormatParameter(const SdpAudioFormat& format,
- absl::string_view param);
+std::optional<std::string> GetFormatParameter(const SdpAudioFormat& format,
+ absl::string_view param);
template <typename T>
-absl::optional<T> GetFormatParameter(const SdpAudioFormat& format,
- absl::string_view param) {
+std::optional<T> GetFormatParameter(const SdpAudioFormat& format,
+ absl::string_view param) {
return rtc::StringToNumber<T>(GetFormatParameter(format, param).value_or(""));
}
template <>
-absl::optional<std::vector<unsigned char>> GetFormatParameter(
+std::optional<std::vector<unsigned char>> GetFormatParameter(
const SdpAudioFormat& format,
absl::string_view param);
@@ -58,7 +58,7 @@
bool IsDtxPacket() const override { return payload_.size() <= 2; }
- absl::optional<DecodeResult> Decode(
+ std::optional<DecodeResult> Decode(
rtc::ArrayView<int16_t> decoded) const override {
AudioDecoder::SpeechType speech_type = AudioDecoder::kSpeech;
int ret;
@@ -73,7 +73,7 @@
}
if (ret < 0)
- return absl::nullopt;
+ return std::nullopt;
return DecodeResult{static_cast<size_t>(ret), speech_type};
}
diff --git a/modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_impl.cc b/modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_impl.cc
index 285ea89..7123b92 100644
--- a/modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_impl.cc
+++ b/modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_impl.cc
@@ -57,30 +57,30 @@
WebRtcOpus_DecoderFree(dec_state_);
}
-absl::optional<AudioDecoderMultiChannelOpusConfig>
+std::optional<AudioDecoderMultiChannelOpusConfig>
AudioDecoderMultiChannelOpusImpl::SdpToConfig(const SdpAudioFormat& format) {
AudioDecoderMultiChannelOpusConfig config;
config.num_channels = format.num_channels;
auto num_streams = GetFormatParameter<int>(format, "num_streams");
if (!num_streams.has_value()) {
- return absl::nullopt;
+ return std::nullopt;
}
config.num_streams = *num_streams;
auto coupled_streams = GetFormatParameter<int>(format, "coupled_streams");
if (!coupled_streams.has_value()) {
- return absl::nullopt;
+ return std::nullopt;
}
config.coupled_streams = *coupled_streams;
auto channel_mapping =
GetFormatParameter<std::vector<unsigned char>>(format, "channel_mapping");
if (!channel_mapping.has_value()) {
- return absl::nullopt;
+ return std::nullopt;
}
config.channel_mapping = *channel_mapping;
if (!config.IsOk()) {
- return absl::nullopt;
+ return std::nullopt;
}
return config;
}
diff --git a/modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_impl.h b/modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_impl.h
index 2ff47a8..85dc5e1 100644
--- a/modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_impl.h
+++ b/modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_impl.h
@@ -46,7 +46,7 @@
int SampleRateHz() const override;
size_t Channels() const override;
- static absl::optional<AudioDecoderMultiChannelOpusConfig> SdpToConfig(
+ static std::optional<AudioDecoderMultiChannelOpusConfig> SdpToConfig(
const SdpAudioFormat& format);
protected:
diff --git a/modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_unittest.cc b/modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_unittest.cc
index 57e2107..7b30a39 100644
--- a/modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_unittest.cc
+++ b/modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_unittest.cc
@@ -25,15 +25,15 @@
{"num_streams", "2"}});
EXPECT_EQ(GetFormatParameter(sdp_format, "channel_mapping"),
- absl::optional<std::string>("0,1,2,3"));
+ std::optional<std::string>("0,1,2,3"));
EXPECT_EQ(GetFormatParameter<int>(sdp_format, "coupled_streams"),
- absl::optional<int>(2));
+ std::optional<int>(2));
- EXPECT_EQ(GetFormatParameter(sdp_format, "missing"), absl::nullopt);
+ EXPECT_EQ(GetFormatParameter(sdp_format, "missing"), std::nullopt);
EXPECT_EQ(GetFormatParameter<int>(sdp_format, "channel_mapping"),
- absl::nullopt);
+ std::nullopt);
}
TEST(AudioDecoderMultiOpusTest, InvalidChannelMappings) {
@@ -43,7 +43,7 @@
{{"channel_mapping", "3,0"},
{"coupled_streams", "1"},
{"num_streams", "2"}});
- const absl::optional<AudioDecoderMultiChannelOpus::Config> decoder_config =
+ const std::optional<AudioDecoderMultiChannelOpus::Config> decoder_config =
AudioDecoderMultiChannelOpus::SdpToConfig(sdp_format);
EXPECT_FALSE(decoder_config.has_value());
}
@@ -54,7 +54,7 @@
{{"channel_mapping", "0,1,2,3,4,5"},
{"coupled_streams", "0"},
{"num_streams", "2"}});
- const absl::optional<AudioDecoderMultiChannelOpus::Config> decoder_config =
+ const std::optional<AudioDecoderMultiChannelOpus::Config> decoder_config =
AudioDecoderMultiChannelOpus::SdpToConfig(sdp_format);
EXPECT_FALSE(decoder_config.has_value());
}
@@ -63,7 +63,7 @@
const SdpAudioFormat sdp_format(
"multiopus", 48000, 5,
{{"channel_mapping", "0,1,two,3,4"}, {"coupled_streams", "0"}});
- const absl::optional<AudioDecoderMultiChannelOpus::Config> decoder_config =
+ const std::optional<AudioDecoderMultiChannelOpus::Config> decoder_config =
AudioDecoderMultiChannelOpus::SdpToConfig(sdp_format);
EXPECT_FALSE(decoder_config.has_value());
}
@@ -75,7 +75,7 @@
{"coupled_streams", "2"},
{"num_streams", "2"}});
- const absl::optional<AudioDecoderMultiChannelOpus::Config> decoder_config =
+ const std::optional<AudioDecoderMultiChannelOpus::Config> decoder_config =
AudioDecoderMultiChannelOpus::SdpToConfig(sdp_format);
ASSERT_TRUE(decoder_config.has_value());
@@ -92,7 +92,7 @@
{"coupled_stream", "2"},
{"num_streams", "2"}});
- const absl::optional<AudioDecoderMultiChannelOpus::Config> decoder_config =
+ const std::optional<AudioDecoderMultiChannelOpus::Config> decoder_config =
AudioDecoderMultiChannelOpus::SdpToConfig(sdp_format);
EXPECT_FALSE(decoder_config.has_value());
@@ -104,7 +104,7 @@
{"coupled_streams", "2"},
{"num_streams", "2"}});
- const absl::optional<AudioDecoderMultiChannelOpus::Config> decoder_config =
+ const std::optional<AudioDecoderMultiChannelOpus::Config> decoder_config =
AudioDecoderMultiChannelOpus::SdpToConfig(sdp_format);
EXPECT_FALSE(decoder_config.has_value());
@@ -117,7 +117,7 @@
{"coupled_streams", "2"},
{"num_streams", "2"}});
- const absl::optional<AudioDecoderMultiChannelOpus::Config> decoder_config =
+ const std::optional<AudioDecoderMultiChannelOpus::Config> decoder_config =
AudioDecoderMultiChannelOpus::SdpToConfig(sdp_format);
ASSERT_TRUE(decoder_config.has_value());
@@ -135,7 +135,7 @@
EXPECT_FALSE(specs.empty());
for (const AudioCodecSpec& spec : specs) {
- const absl::optional<AudioDecoderMultiChannelOpus::Config> decoder_config =
+ const std::optional<AudioDecoderMultiChannelOpus::Config> decoder_config =
AudioDecoderMultiChannelOpus::SdpToConfig(spec.format);
ASSERT_TRUE(decoder_config.has_value());
diff --git a/modules/audio_coding/codecs/opus/audio_decoder_opus.cc b/modules/audio_coding/codecs/opus/audio_decoder_opus.cc
index 7d5e3cf..9eed1c6 100644
--- a/modules/audio_coding/codecs/opus/audio_decoder_opus.cc
+++ b/modules/audio_coding/codecs/opus/audio_decoder_opus.cc
@@ -11,9 +11,9 @@
#include "modules/audio_coding/codecs/opus/audio_decoder_opus.h"
#include <memory>
+#include <optional>
#include <utility>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/field_trials_view.h"
#include "modules/audio_coding/codecs/opus/audio_coder_opus_common.h"
diff --git a/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.cc b/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.cc
index 9d140f7..45f3be1 100644
--- a/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.cc
+++ b/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.cc
@@ -103,7 +103,7 @@
// out how invalid it is and accurately log invalid values.
int CalculateBitrate(int max_playback_rate_hz,
size_t num_channels,
- absl::optional<std::string> bitrate_param) {
+ std::optional<std::string> bitrate_param) {
const int default_bitrate =
CalculateDefaultBitrate(max_playback_rate_hz, num_channels);
@@ -167,7 +167,7 @@
RTC_CHECK(RecreateEncoderInstance(config_));
}
-absl::optional<std::pair<TimeDelta, TimeDelta>>
+std::optional<std::pair<TimeDelta, TimeDelta>>
AudioEncoderMultiChannelOpusImpl::GetFrameLengthRange() const {
return {{TimeDelta::Millis(config_.frame_size_ms),
TimeDelta::Millis(config_.frame_size_ms)}};
@@ -237,11 +237,11 @@
return true;
}
-absl::optional<AudioEncoderMultiChannelOpusConfig>
+std::optional<AudioEncoderMultiChannelOpusConfig>
AudioEncoderMultiChannelOpusImpl::SdpToConfig(const SdpAudioFormat& format) {
if (!absl::EqualsIgnoreCase(format.name, "multiopus") ||
format.clockrate_hz != 48000) {
- return absl::nullopt;
+ return std::nullopt;
}
AudioEncoderMultiChannelOpusConfig config;
@@ -266,25 +266,25 @@
auto num_streams = GetFormatParameter<int>(format, "num_streams");
if (!num_streams.has_value()) {
- return absl::nullopt;
+ return std::nullopt;
}
config.num_streams = *num_streams;
auto coupled_streams = GetFormatParameter<int>(format, "coupled_streams");
if (!coupled_streams.has_value()) {
- return absl::nullopt;
+ return std::nullopt;
}
config.coupled_streams = *coupled_streams;
auto channel_mapping =
GetFormatParameter<std::vector<unsigned char>>(format, "channel_mapping");
if (!channel_mapping.has_value()) {
- return absl::nullopt;
+ return std::nullopt;
}
config.channel_mapping = *channel_mapping;
if (!config.IsOk()) {
- return absl::nullopt;
+ return std::nullopt;
}
return config;
}
diff --git a/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.h b/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.h
index 6d6e63b..9def211 100644
--- a/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.h
+++ b/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.h
@@ -15,10 +15,10 @@
#include <stdint.h>
#include <memory>
+#include <optional>
#include <utility>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/audio_codecs/audio_encoder.h"
#include "api/audio_codecs/audio_format.h"
@@ -45,7 +45,7 @@
// Static interface for use by BuiltinAudioEncoderFactory.
static constexpr const char* GetPayloadName() { return "multiopus"; }
- static absl::optional<AudioCodecInfo> QueryAudioEncoder(
+ static std::optional<AudioCodecInfo> QueryAudioEncoder(
const SdpAudioFormat& format);
int SampleRateHz() const override;
@@ -55,7 +55,7 @@
int GetTargetBitrate() const override;
void Reset() override;
- absl::optional<std::pair<TimeDelta, TimeDelta>> GetFrameLengthRange()
+ std::optional<std::pair<TimeDelta, TimeDelta>> GetFrameLengthRange()
const override;
protected:
@@ -64,7 +64,7 @@
rtc::Buffer* encoded) override;
private:
- static absl::optional<AudioEncoderMultiChannelOpusConfig> SdpToConfig(
+ static std::optional<AudioEncoderMultiChannelOpusConfig> SdpToConfig(
const SdpAudioFormat& format);
static AudioCodecInfo QueryAudioEncoder(
const AudioEncoderMultiChannelOpusConfig& config);
diff --git a/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_unittest.cc b/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_unittest.cc
index 92f6f2c..7601589 100644
--- a/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_unittest.cc
+++ b/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_unittest.cc
@@ -26,7 +26,7 @@
{{"channel_mapping", "3,0"},
{"coupled_streams", "1"},
{"num_streams", "2"}});
- const absl::optional<AudioEncoderMultiChannelOpus::Config> encoder_config =
+ const std::optional<AudioEncoderMultiChannelOpus::Config> encoder_config =
AudioEncoderMultiChannelOpus::SdpToConfig(sdp_format);
// Maps input channel 0 to coded channel 3, which doesn't exist.
@@ -38,7 +38,7 @@
{{"channel_mapping", "0"},
{"coupled_streams", "1"},
{"num_streams", "2"}});
- const absl::optional<AudioEncoderMultiChannelOpus::Config> encoder_config =
+ const std::optional<AudioEncoderMultiChannelOpus::Config> encoder_config =
AudioEncoderMultiChannelOpus::SdpToConfig(sdp_format);
// The mapping is too short.
@@ -49,7 +49,7 @@
{{"channel_mapping", "0,0,0"},
{"coupled_streams", "0"},
{"num_streams", "1"}});
- const absl::optional<AudioEncoderMultiChannelOpus::Config> encoder_config =
+ const std::optional<AudioEncoderMultiChannelOpus::Config> encoder_config =
AudioEncoderMultiChannelOpus::SdpToConfig(sdp_format);
// Coded channel 0 comes from both input channels 0, 1 and 2.
@@ -60,7 +60,7 @@
{{"channel_mapping", "0,255,255"},
{"coupled_streams", "0"},
{"num_streams", "1"}});
- const absl::optional<AudioEncoderMultiChannelOpus::Config> encoder_config =
+ const std::optional<AudioEncoderMultiChannelOpus::Config> encoder_config =
AudioEncoderMultiChannelOpus::SdpToConfig(sdp_format);
ASSERT_TRUE(encoder_config.has_value());
@@ -72,7 +72,7 @@
{{"channel_mapping", "0,255,255"},
{"coupled_streams", "0"},
{"num_streams", "2"}});
- const absl::optional<AudioEncoderMultiChannelOpus::Config> encoder_config =
+ const std::optional<AudioEncoderMultiChannelOpus::Config> encoder_config =
AudioEncoderMultiChannelOpus::SdpToConfig(sdp_format);
// This is NOT fine, because channels nothing says how coded channel 1
@@ -90,7 +90,7 @@
{"channel_mapping", "0,4,1,2,3,5"},
{"num_streams", "4"},
{"coupled_streams", "2"}}});
- const absl::optional<AudioEncoderMultiChannelOpus::Config> encoder_config =
+ const std::optional<AudioEncoderMultiChannelOpus::Config> encoder_config =
AudioEncoderMultiChannelOpus::SdpToConfig(sdp_format);
ASSERT_TRUE(encoder_config.has_value());
@@ -107,7 +107,7 @@
{{"channel_mapping", "0,255,255"},
{"coupled_streams", "0"},
{"num_streams", "2"}});
- const absl::optional<AudioEncoderMultiChannelOpus::Config> encoder_config =
+ const std::optional<AudioEncoderMultiChannelOpus::Config> encoder_config =
AudioEncoderMultiChannelOpus::SdpToConfig(sdp_format);
ASSERT_FALSE(encoder_config.has_value());
}
@@ -116,7 +116,7 @@
{{"channel_mapping", "1,255,0"},
{"coupled_streams", "1"},
{"num_streams", "1"}});
- const absl::optional<AudioEncoderMultiChannelOpus::Config> encoder_config =
+ const std::optional<AudioEncoderMultiChannelOpus::Config> encoder_config =
AudioEncoderMultiChannelOpus::SdpToConfig(sdp_format);
ASSERT_TRUE(encoder_config.has_value());
@@ -141,7 +141,7 @@
EXPECT_FALSE(specs.empty());
for (const AudioCodecSpec& spec : specs) {
- const absl::optional<AudioEncoderMultiChannelOpus::Config> encoder_config =
+ const std::optional<AudioEncoderMultiChannelOpus::Config> encoder_config =
AudioEncoderMultiChannelOpus::SdpToConfig(spec.format);
ASSERT_TRUE(encoder_config.has_value());
diff --git a/modules/audio_coding/codecs/opus/audio_encoder_opus.cc b/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
index 7eb8bbd..b2e44c7 100644
--- a/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
+++ b/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
@@ -89,7 +89,7 @@
// out how invalid it is and accurately log invalid values.
int CalculateBitrate(int max_playback_rate_hz,
size_t num_channels,
- absl::optional<std::string> bitrate_param) {
+ std::optional<std::string> bitrate_param) {
const int default_bitrate =
CalculateDefaultBitrate(max_playback_rate_hz, num_channels);
@@ -228,11 +228,11 @@
return info;
}
-absl::optional<AudioEncoderOpusConfig> AudioEncoderOpusImpl::SdpToConfig(
+std::optional<AudioEncoderOpusConfig> AudioEncoderOpusImpl::SdpToConfig(
const SdpAudioFormat& format) {
if (!absl::EqualsIgnoreCase(format.name, "opus") ||
format.clockrate_hz != kRtpTimestampRateHz || format.num_channels != 2) {
- return absl::nullopt;
+ return std::nullopt;
}
AudioEncoderOpusConfig config;
@@ -265,12 +265,12 @@
&config.supported_frame_lengths_ms);
if (!config.IsOk()) {
RTC_DCHECK_NOTREACHED();
- return absl::nullopt;
+ return std::nullopt;
}
return config;
}
-absl::optional<int> AudioEncoderOpusImpl::GetNewComplexity(
+std::optional<int> AudioEncoderOpusImpl::GetNewComplexity(
const AudioEncoderOpusConfig& config) {
RTC_DCHECK(config.IsOk());
const int bitrate_bps = GetBitrateBps(config);
@@ -279,7 +279,7 @@
bitrate_bps <= config.complexity_threshold_bps +
config.complexity_threshold_window_bps) {
// Within the hysteresis window; make no change.
- return absl::nullopt;
+ return std::nullopt;
} else {
return bitrate_bps <= config.complexity_threshold_bps
? config.low_rate_complexity
@@ -287,7 +287,7 @@
}
}
-absl::optional<int> AudioEncoderOpusImpl::GetNewBandwidth(
+std::optional<int> AudioEncoderOpusImpl::GetNewBandwidth(
const AudioEncoderOpusConfig& config,
OpusEncInst* inst) {
constexpr int kMinWidebandBitrate = 8000;
@@ -296,17 +296,17 @@
RTC_DCHECK(config.IsOk());
const int bitrate = GetBitrateBps(config);
if (bitrate > kAutomaticThreshold) {
- return absl::optional<int>(OPUS_AUTO);
+ return std::optional<int>(OPUS_AUTO);
}
const int bandwidth = WebRtcOpus_GetBandwidth(inst);
RTC_DCHECK_GE(bandwidth, 0);
if (bitrate > kMaxNarrowbandBitrate && bandwidth < OPUS_BANDWIDTH_WIDEBAND) {
- return absl::optional<int>(OPUS_BANDWIDTH_WIDEBAND);
+ return std::optional<int>(OPUS_BANDWIDTH_WIDEBAND);
} else if (bitrate < kMinWidebandBitrate &&
bandwidth > OPUS_BANDWIDTH_NARROWBAND) {
- return absl::optional<int>(OPUS_BANDWIDTH_NARROWBAND);
+ return std::optional<int>(OPUS_BANDWIDTH_NARROWBAND);
}
- return absl::optional<int>();
+ return std::optional<int>();
}
class AudioEncoderOpusImpl::PacketLossFractionSmoother {
@@ -496,8 +496,8 @@
void AudioEncoderOpusImpl::OnReceivedUplinkBandwidth(
int target_audio_bitrate_bps,
- absl::optional<int64_t> bwe_period_ms,
- absl::optional<int64_t> stable_target_bitrate_bps) {
+ std::optional<int64_t> bwe_period_ms,
+ std::optional<int64_t> stable_target_bitrate_bps) {
if (audio_network_adaptor_) {
audio_network_adaptor_->SetTargetAudioBitrate(target_audio_bitrate_bps);
if (use_stable_target_for_adaptation_) {
@@ -538,9 +538,9 @@
}
void AudioEncoderOpusImpl::OnReceivedUplinkBandwidth(
int target_audio_bitrate_bps,
- absl::optional<int64_t> bwe_period_ms) {
+ std::optional<int64_t> bwe_period_ms) {
OnReceivedUplinkBandwidth(target_audio_bitrate_bps, bwe_period_ms,
- absl::nullopt);
+ std::nullopt);
}
void AudioEncoderOpusImpl::OnReceivedUplinkAllocation(
@@ -794,7 +794,7 @@
if (!bitrate_smoother_last_update_time_ ||
now_ms - *bitrate_smoother_last_update_time_ >=
config_.uplink_bandwidth_update_interval_ms) {
- absl::optional<float> smoothed_bitrate = bitrate_smoother_->GetAverage();
+ std::optional<float> smoothed_bitrate = bitrate_smoother_->GetAverage();
if (smoothed_bitrate)
audio_network_adaptor_->SetUplinkBandwidth(*smoothed_bitrate);
bitrate_smoother_last_update_time_ = now_ms;
@@ -809,11 +809,11 @@
return ANAStats();
}
-absl::optional<std::pair<TimeDelta, TimeDelta> >
+std::optional<std::pair<TimeDelta, TimeDelta> >
AudioEncoderOpusImpl::GetFrameLengthRange() const {
if (audio_network_adaptor_) {
if (config_.supported_frame_lengths_ms.empty()) {
- return absl::nullopt;
+ return std::nullopt;
}
return {{TimeDelta::Millis(config_.supported_frame_lengths_ms.front()),
TimeDelta::Millis(config_.supported_frame_lengths_ms.back())}};
diff --git a/modules/audio_coding/codecs/opus/audio_encoder_opus.h b/modules/audio_coding/codecs/opus/audio_encoder_opus.h
index b09fbf1..f1c9c65 100644
--- a/modules/audio_coding/codecs/opus/audio_encoder_opus.h
+++ b/modules/audio_coding/codecs/opus/audio_encoder_opus.h
@@ -13,11 +13,11 @@
#include <functional>
#include <memory>
+#include <optional>
#include <string>
#include <vector>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/audio_codecs/audio_encoder.h"
#include "api/audio_codecs/audio_format.h"
#include "api/audio_codecs/opus/audio_encoder_opus_config.h"
@@ -36,13 +36,13 @@
// defined by complexity_threshold_bps +/- complexity_threshold_window_bps.
// Otherwise, returns the current complexity depending on whether the
// current bitrate is above or below complexity_threshold_bps.
- static absl::optional<int> GetNewComplexity(
+ static std::optional<int> GetNewComplexity(
const AudioEncoderOpusConfig& config);
// Returns OPUS_AUTO if the the current bitrate is above wideband threshold.
// Returns empty if it is below, but bandwidth coincides with the desired one.
// Otherwise returns the desired bandwidth.
- static absl::optional<int> GetNewBandwidth(
+ static std::optional<int> GetNewBandwidth(
const AudioEncoderOpusConfig& config,
OpusEncInst* inst);
@@ -90,16 +90,15 @@
void OnReceivedUplinkPacketLossFraction(
float uplink_packet_loss_fraction) override;
void OnReceivedTargetAudioBitrate(int target_audio_bitrate_bps) override;
- void OnReceivedUplinkBandwidth(
- int target_audio_bitrate_bps,
- absl::optional<int64_t> bwe_period_ms) override;
+ void OnReceivedUplinkBandwidth(int target_audio_bitrate_bps,
+ std::optional<int64_t> bwe_period_ms) override;
void OnReceivedUplinkAllocation(BitrateAllocationUpdate update) override;
void OnReceivedRtt(int rtt_ms) override;
void OnReceivedOverhead(size_t overhead_bytes_per_packet) override;
void SetReceiverFrameLengthRange(int min_frame_length_ms,
int max_frame_length_ms) override;
ANAStats GetANAStats() const override;
- absl::optional<std::pair<TimeDelta, TimeDelta> > GetFrameLengthRange()
+ std::optional<std::pair<TimeDelta, TimeDelta> > GetFrameLengthRange()
const override;
rtc::ArrayView<const int> supported_frame_lengths_ms() const {
return config_.supported_frame_lengths_ms;
@@ -129,7 +128,7 @@
const AudioNetworkAdaptorCreator& audio_network_adaptor_creator,
std::unique_ptr<SmoothingFilter> bitrate_smoother);
- static absl::optional<AudioEncoderOpusConfig> SdpToConfig(
+ static std::optional<AudioEncoderOpusConfig> SdpToConfig(
const SdpAudioFormat& format);
static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs);
static AudioCodecInfo QueryAudioEncoder(const AudioEncoderOpusConfig& config);
@@ -144,8 +143,8 @@
void OnReceivedUplinkBandwidth(
int target_audio_bitrate_bps,
- absl::optional<int64_t> bwe_period_ms,
- absl::optional<int64_t> link_capacity_allocation);
+ std::optional<int64_t> bwe_period_ms,
+ std::optional<int64_t> link_capacity_allocation);
// TODO(minyue): remove "override" when we can deprecate
// `AudioEncoder::SetTargetBitrate`.
@@ -177,9 +176,9 @@
std::unique_ptr<PacketLossFractionSmoother> packet_loss_fraction_smoother_;
const AudioNetworkAdaptorCreator audio_network_adaptor_creator_;
std::unique_ptr<AudioNetworkAdaptor> audio_network_adaptor_;
- absl::optional<size_t> overhead_bytes_per_packet_;
+ std::optional<size_t> overhead_bytes_per_packet_;
const std::unique_ptr<SmoothingFilter> bitrate_smoother_;
- absl::optional<int64_t> bitrate_smoother_last_update_time_;
+ std::optional<int64_t> bitrate_smoother_last_update_time_;
int consecutive_dtx_frames_;
friend struct AudioEncoderOpus;
diff --git a/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc b/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
index c12f70f..fc37a21 100644
--- a/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
+++ b/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
@@ -208,24 +208,24 @@
rtc::CheckedDivExact(48000, kDefaultOpusPacSize);
// Set a too low bitrate.
states->encoder->OnReceivedUplinkBandwidth(kMinBitrateBps + kOverheadBps - 1,
- absl::nullopt);
+ std::nullopt);
EXPECT_EQ(kMinBitrateBps, states->encoder->GetTargetBitrate());
// Set a too high bitrate.
states->encoder->OnReceivedUplinkBandwidth(kMaxBitrateBps + kOverheadBps + 1,
- absl::nullopt);
+ std::nullopt);
EXPECT_EQ(kMaxBitrateBps, states->encoder->GetTargetBitrate());
// Set the minimum rate.
states->encoder->OnReceivedUplinkBandwidth(kMinBitrateBps + kOverheadBps,
- absl::nullopt);
+ std::nullopt);
EXPECT_EQ(kMinBitrateBps, states->encoder->GetTargetBitrate());
// Set the maximum rate.
states->encoder->OnReceivedUplinkBandwidth(kMaxBitrateBps + kOverheadBps,
- absl::nullopt);
+ std::nullopt);
EXPECT_EQ(kMaxBitrateBps, states->encoder->GetTargetBitrate());
// Set rates from kMaxBitrateBps up to 32000 bps.
for (int rate = kMinBitrateBps + kOverheadBps; rate <= 32000 + kOverheadBps;
rate += 1000) {
- states->encoder->OnReceivedUplinkBandwidth(rate, absl::nullopt);
+ states->encoder->OnReceivedUplinkBandwidth(rate, std::nullopt);
EXPECT_EQ(rate - kOverheadBps, states->encoder->GetTargetBitrate());
}
}
@@ -379,7 +379,7 @@
auto states = CreateCodec(sample_rate_hz_, 2);
states->encoder->OnReceivedUplinkBandwidth(kDefaultOpusRate * 2,
- absl::nullopt);
+ std::nullopt);
// Since `OnReceivedOverhead` has not been called, the codec bitrate should
// not change.
@@ -394,7 +394,7 @@
// Bitrate within hysteresis window. Expect empty output.
config.bitrate_bps = 12500;
- EXPECT_EQ(absl::nullopt, AudioEncoderOpusImpl::GetNewComplexity(config));
+ EXPECT_EQ(std::nullopt, AudioEncoderOpusImpl::GetNewComplexity(config));
// Bitrate below hysteresis window. Expect higher complexity.
config.bitrate_bps = 10999;
@@ -402,7 +402,7 @@
// Bitrate within hysteresis window. Expect empty output.
config.bitrate_bps = 12500;
- EXPECT_EQ(absl::nullopt, AudioEncoderOpusImpl::GetNewComplexity(config));
+ EXPECT_EQ(std::nullopt, AudioEncoderOpusImpl::GetNewComplexity(config));
// Bitrate above hysteresis window. Expect lower complexity.
config.bitrate_bps = 14001;
@@ -428,9 +428,9 @@
sample_rate_hz_));
// Bitrate below minmum wideband. Expect narrowband.
- config.bitrate_bps = absl::optional<int>(7999);
+ config.bitrate_bps = std::optional<int>(7999);
auto bandwidth = AudioEncoderOpusImpl::GetNewBandwidth(config, inst);
- EXPECT_EQ(absl::optional<int>(OPUS_BANDWIDTH_NARROWBAND), bandwidth);
+ EXPECT_EQ(std::optional<int>(OPUS_BANDWIDTH_NARROWBAND), bandwidth);
WebRtcOpus_SetBandwidth(inst, *bandwidth);
// It is necessary to encode here because Opus has some logic in the encoder
// that goes from the user-set bandwidth to the used and returned one.
@@ -439,14 +439,14 @@
kMaxBytes, bitstream);
// Bitrate not yet above maximum narrowband. Expect empty.
- config.bitrate_bps = absl::optional<int>(9000);
+ config.bitrate_bps = std::optional<int>(9000);
bandwidth = AudioEncoderOpusImpl::GetNewBandwidth(config, inst);
- EXPECT_EQ(absl::optional<int>(), bandwidth);
+ EXPECT_EQ(std::optional<int>(), bandwidth);
// Bitrate above maximum narrowband. Expect wideband.
- config.bitrate_bps = absl::optional<int>(9001);
+ config.bitrate_bps = std::optional<int>(9001);
bandwidth = AudioEncoderOpusImpl::GetNewBandwidth(config, inst);
- EXPECT_EQ(absl::optional<int>(OPUS_BANDWIDTH_WIDEBAND), bandwidth);
+ EXPECT_EQ(std::optional<int>(OPUS_BANDWIDTH_WIDEBAND), bandwidth);
WebRtcOpus_SetBandwidth(inst, *bandwidth);
// It is necessary to encode here because Opus has some logic in the encoder
// that goes from the user-set bandwidth to the used and returned one.
@@ -455,14 +455,14 @@
kMaxBytes, bitstream);
// Bitrate not yet below minimum wideband. Expect empty.
- config.bitrate_bps = absl::optional<int>(8000);
+ config.bitrate_bps = std::optional<int>(8000);
bandwidth = AudioEncoderOpusImpl::GetNewBandwidth(config, inst);
- EXPECT_EQ(absl::optional<int>(), bandwidth);
+ EXPECT_EQ(std::optional<int>(), bandwidth);
// Bitrate above automatic threshold. Expect automatic.
- config.bitrate_bps = absl::optional<int>(12001);
+ config.bitrate_bps = std::optional<int>(12001);
bandwidth = AudioEncoderOpusImpl::GetNewBandwidth(config, inst);
- EXPECT_EQ(absl::optional<int>(OPUS_AUTO), bandwidth);
+ EXPECT_EQ(std::optional<int>(OPUS_AUTO), bandwidth);
EXPECT_EQ(0, WebRtcOpus_EncoderFree(inst));
}
@@ -528,7 +528,7 @@
rtc::Buffer encoded;
uint32_t rtp_timestamp = 12345; // Just a number not important to this test.
- states->encoder->OnReceivedUplinkBandwidth(0, absl::nullopt);
+ states->encoder->OnReceivedUplinkBandwidth(0, std::nullopt);
for (int packet_index = 0; packet_index < kNumPacketsToEncode;
packet_index++) {
// Make sure we are not encoding before we have enough data for
@@ -676,7 +676,7 @@
std::unique_ptr<AudioEncoder> encoder = AudioEncoderOpus::MakeAudioEncoder(
CreateEnvironment(), config, {.payload_type = kDefaultOpusPayloadType});
auto ptime = webrtc::TimeDelta::Millis(10);
- absl::optional<std::pair<webrtc::TimeDelta, webrtc::TimeDelta>> range = {
+ std::optional<std::pair<webrtc::TimeDelta, webrtc::TimeDelta>> range = {
{ptime, ptime}};
EXPECT_EQ(encoder->GetFrameLengthRange(), range);
}
diff --git a/modules/audio_coding/codecs/opus/opus_bandwidth_unittest.cc b/modules/audio_coding/codecs/opus/opus_bandwidth_unittest.cc
index 85b9293..2e05447 100644
--- a/modules/audio_coding/codecs/opus/opus_bandwidth_unittest.cc
+++ b/modules/audio_coding/codecs/opus/opus_bandwidth_unittest.cc
@@ -114,7 +114,7 @@
// Create encoder.
AudioEncoderOpusConfig enc_config;
- enc_config.bitrate_bps = absl::optional<int>(7999);
+ enc_config.bitrate_bps = std::optional<int>(7999);
enc_config.num_channels = kNumChannels;
auto encoder =
AudioEncoderOpus::MakeAudioEncoder(env, enc_config, {.payload_type = 17});
diff --git a/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc b/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc
index 634f14d..a622de7 100644
--- a/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc
+++ b/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc
@@ -237,7 +237,7 @@
void AudioEncoderCopyRed::OnReceivedUplinkBandwidth(
int target_audio_bitrate_bps,
- absl::optional<int64_t> bwe_period_ms) {
+ std::optional<int64_t> bwe_period_ms) {
speech_encoder_->OnReceivedUplinkBandwidth(target_audio_bitrate_bps,
bwe_period_ms);
}
@@ -247,7 +247,7 @@
speech_encoder_->OnReceivedUplinkAllocation(update);
}
-absl::optional<std::pair<TimeDelta, TimeDelta>>
+std::optional<std::pair<TimeDelta, TimeDelta>>
AudioEncoderCopyRed::GetFrameLengthRange() const {
return speech_encoder_->GetFrameLengthRange();
}
diff --git a/modules/audio_coding/codecs/red/audio_encoder_copy_red.h b/modules/audio_coding/codecs/red/audio_encoder_copy_red.h
index 359b5ea..dee6a42 100644
--- a/modules/audio_coding/codecs/red/audio_encoder_copy_red.h
+++ b/modules/audio_coding/codecs/red/audio_encoder_copy_red.h
@@ -16,9 +16,9 @@
#include <list>
#include <memory>
+#include <optional>
#include <utility>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/audio_codecs/audio_encoder.h"
#include "api/field_trials_view.h"
@@ -70,16 +70,15 @@
void DisableAudioNetworkAdaptor() override;
void OnReceivedUplinkPacketLossFraction(
float uplink_packet_loss_fraction) override;
- void OnReceivedUplinkBandwidth(
- int target_audio_bitrate_bps,
- absl::optional<int64_t> bwe_period_ms) override;
+ void OnReceivedUplinkBandwidth(int target_audio_bitrate_bps,
+ std::optional<int64_t> bwe_period_ms) override;
void OnReceivedUplinkAllocation(BitrateAllocationUpdate update) override;
void OnReceivedRtt(int rtt_ms) override;
void OnReceivedOverhead(size_t overhead_bytes_per_packet) override;
void SetReceiverFrameLengthRange(int min_frame_length_ms,
int max_frame_length_ms) override;
ANAStats GetANAStats() const override;
- absl::optional<std::pair<TimeDelta, TimeDelta>> GetFrameLengthRange()
+ std::optional<std::pair<TimeDelta, TimeDelta>> GetFrameLengthRange()
const override;
rtc::ArrayView<std::unique_ptr<AudioEncoder>> ReclaimContainedEncoders()
override;
diff --git a/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc b/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc
index e9b1b07..8f2703e 100644
--- a/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc
+++ b/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc
@@ -106,8 +106,8 @@
TEST_F(AudioEncoderCopyRedTest, CheckTargetAudioBitratePropagation) {
EXPECT_CALL(*mock_encoder_,
- OnReceivedUplinkBandwidth(4711, absl::optional<int64_t>()));
- red_->OnReceivedUplinkBandwidth(4711, absl::nullopt);
+ OnReceivedUplinkBandwidth(4711, std::optional<int64_t>()));
+ red_->OnReceivedUplinkBandwidth(4711, std::nullopt);
}
TEST_F(AudioEncoderCopyRedTest, CheckPacketLossFractionPropagation) {
@@ -119,7 +119,7 @@
auto expected_range =
std::make_pair(TimeDelta::Millis(20), TimeDelta::Millis(20));
EXPECT_CALL(*mock_encoder_, GetFrameLengthRange())
- .WillRepeatedly(Return(absl::make_optional(expected_range)));
+ .WillRepeatedly(Return(std::make_optional(expected_range)));
EXPECT_THAT(red_->GetFrameLengthRange(), Optional(Eq(expected_range)));
}
diff --git a/modules/audio_coding/include/audio_coding_module.h b/modules/audio_coding/include/audio_coding_module.h
index 5c5bd16..ec8255e 100644
--- a/modules/audio_coding/include/audio_coding_module.h
+++ b/modules/audio_coding/include/audio_coding_module.h
@@ -12,11 +12,11 @@
#define MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_H_
#include <memory>
+#include <optional>
#include <string>
#include <utility>
#include <vector>
-#include "absl/types/optional.h"
#include "api/audio_codecs/audio_encoder.h"
#include "api/function_view.h"
#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
diff --git a/modules/audio_coding/neteq/audio_decoder_unittest.cc b/modules/audio_coding/neteq/audio_decoder_unittest.cc
index a39b46b..ddcf020 100644
--- a/modules/audio_coding/neteq/audio_decoder_unittest.cc
+++ b/modules/audio_coding/neteq/audio_decoder_unittest.cc
@@ -419,7 +419,7 @@
namespace {
int SetAndGetTargetBitrate(AudioEncoder* audio_encoder, int rate) {
- audio_encoder->OnReceivedUplinkBandwidth(rate, absl::nullopt);
+ audio_encoder->OnReceivedUplinkBandwidth(rate, std::nullopt);
return audio_encoder->GetTargetBitrate();
}
void TestSetAndGetTargetBitratesWithFixedCodec(AudioEncoder* audio_encoder,
diff --git a/modules/audio_coding/neteq/decision_logic.cc b/modules/audio_coding/neteq/decision_logic.cc
index c05bea5..7d9f3a9 100644
--- a/modules/audio_coding/neteq/decision_logic.cc
+++ b/modules/audio_coding/neteq/decision_logic.cc
@@ -14,8 +14,8 @@
#include <cstdint>
#include <memory>
+#include <optional>
-#include "absl/types/optional.h"
#include "api/neteq/neteq.h"
#include "api/neteq/neteq_controller.h"
#include "modules/audio_coding/neteq/packet_arrival_history.h"
@@ -171,13 +171,12 @@
return buffer_level_filter_->filtered_current_level();
}
-absl::optional<int> DecisionLogic::PacketArrived(
- int fs_hz,
- bool should_update_stats,
- const PacketArrivedInfo& info) {
+std::optional<int> DecisionLogic::PacketArrived(int fs_hz,
+ bool should_update_stats,
+ const PacketArrivedInfo& info) {
buffer_flush_ = buffer_flush_ || info.buffer_flush;
if (!should_update_stats || info.is_cng_or_dtmf) {
- return absl::nullopt;
+ return std::nullopt;
}
if (info.packet_length_samples > 0 && fs_hz > 0 &&
info.packet_length_samples != packet_length_samples_) {
@@ -188,7 +187,7 @@
info.packet_length_samples);
if (!inserted || packet_arrival_history_->size() < 2) {
// No meaningful delay estimate unless at least 2 packets have arrived.
- return absl::nullopt;
+ return std::nullopt;
}
int arrival_delay_ms =
packet_arrival_history_->GetDelayMs(info.main_timestamp);
diff --git a/modules/audio_coding/neteq/decision_logic.h b/modules/audio_coding/neteq/decision_logic.h
index c94aa49..21cbc56 100644
--- a/modules/audio_coding/neteq/decision_logic.h
+++ b/modules/audio_coding/neteq/decision_logic.h
@@ -70,9 +70,9 @@
int UnlimitedTargetLevelMs() const override;
- absl::optional<int> PacketArrived(int fs_hz,
- bool should_update_stats,
- const PacketArrivedInfo& info) override;
+ std::optional<int> PacketArrived(int fs_hz,
+ bool should_update_stats,
+ const PacketArrivedInfo& info) override;
void RegisterEmptyPacket() override {}
diff --git a/modules/audio_coding/neteq/decision_logic_unittest.cc b/modules/audio_coding/neteq/decision_logic_unittest.cc
index d7c06ed..ecf353c 100644
--- a/modules/audio_coding/neteq/decision_logic_unittest.cc
+++ b/modules/audio_coding/neteq/decision_logic_unittest.cc
@@ -195,7 +195,7 @@
TEST_F(DecisionLogicTest, CngTimeout) {
auto status = CreateNetEqStatus(NetEq::Mode::kCodecInternalCng, 0);
- status.next_packet = absl::nullopt;
+ status.next_packet = std::nullopt;
status.generated_noise_samples = kSamplesPerMs * 500;
bool reset_decoder = false;
EXPECT_EQ(decision_logic_->GetDecision(status, &reset_decoder),
diff --git a/modules/audio_coding/neteq/decoder_database.cc b/modules/audio_coding/neteq/decoder_database.cc
index 28a985d..e97d59f 100644
--- a/modules/audio_coding/neteq/decoder_database.cc
+++ b/modules/audio_coding/neteq/decoder_database.cc
@@ -30,7 +30,7 @@
DecoderDatabase::DecoderDatabase(
const Environment& env,
scoped_refptr<AudioDecoderFactory> decoder_factory,
- absl::optional<AudioCodecPairId> codec_pair_id)
+ std::optional<AudioCodecPairId> codec_pair_id)
: env_(env),
active_decoder_type_(-1),
active_cng_decoder_type_(-1),
@@ -42,7 +42,7 @@
DecoderDatabase::DecoderInfo::DecoderInfo(
const Environment& env,
const SdpAudioFormat& audio_format,
- absl::optional<AudioCodecPairId> codec_pair_id,
+ std::optional<AudioCodecPairId> codec_pair_id,
AudioDecoderFactory* factory)
: env_(env),
audio_format_(audio_format),
@@ -73,7 +73,7 @@
return absl::EqualsIgnoreCase(audio_format_.name, name);
}
-absl::optional<DecoderDatabase::DecoderInfo::CngDecoder>
+std::optional<DecoderDatabase::DecoderInfo::CngDecoder>
DecoderDatabase::DecoderInfo::CngDecoder::Create(const SdpAudioFormat& format) {
if (absl::EqualsIgnoreCase(format.name, "CN")) {
// CN has a 1:1 RTP clock rate to sample rate ratio.
@@ -82,7 +82,7 @@
sample_rate_hz == 32000 || sample_rate_hz == 48000);
return DecoderDatabase::DecoderInfo::CngDecoder{sample_rate_hz};
} else {
- return absl::nullopt;
+ return std::nullopt;
}
}
diff --git a/modules/audio_coding/neteq/decoder_database.h b/modules/audio_coding/neteq/decoder_database.h
index f70d255..9c1a1ec 100644
--- a/modules/audio_coding/neteq/decoder_database.h
+++ b/modules/audio_coding/neteq/decoder_database.h
@@ -42,7 +42,7 @@
public:
DecoderInfo(const Environment& env,
const SdpAudioFormat& audio_format,
- absl::optional<AudioCodecPairId> codec_pair_id,
+ std::optional<AudioCodecPairId> codec_pair_id,
AudioDecoderFactory* factory);
DecoderInfo(DecoderInfo&&);
~DecoderInfo();
@@ -86,16 +86,16 @@
private:
const Environment env_;
const SdpAudioFormat audio_format_;
- const absl::optional<AudioCodecPairId> codec_pair_id_;
+ const std::optional<AudioCodecPairId> codec_pair_id_;
AudioDecoderFactory* const factory_;
mutable std::unique_ptr<AudioDecoder> decoder_;
// Set iff this is a comfort noise decoder.
struct CngDecoder {
- static absl::optional<CngDecoder> Create(const SdpAudioFormat& format);
+ static std::optional<CngDecoder> Create(const SdpAudioFormat& format);
int sample_rate_hz;
};
- const absl::optional<CngDecoder> cng_decoder_;
+ const std::optional<CngDecoder> cng_decoder_;
enum class Subtype : int8_t { kNormal, kComfortNoise, kDtmf, kRed };
@@ -110,7 +110,7 @@
DecoderDatabase(const Environment& env,
scoped_refptr<AudioDecoderFactory> decoder_factory,
- absl::optional<AudioCodecPairId> codec_pair_id);
+ std::optional<AudioCodecPairId> codec_pair_id);
virtual ~DecoderDatabase();
@@ -192,7 +192,7 @@
int active_cng_decoder_type_;
mutable std::unique_ptr<ComfortNoiseDecoder> active_cng_decoder_;
scoped_refptr<AudioDecoderFactory> decoder_factory_;
- const absl::optional<AudioCodecPairId> codec_pair_id_;
+ const std::optional<AudioCodecPairId> codec_pair_id_;
};
} // namespace webrtc
diff --git a/modules/audio_coding/neteq/decoder_database_unittest.cc b/modules/audio_coding/neteq/decoder_database_unittest.cc
index 1dd84cf..fe551b6 100644
--- a/modules/audio_coding/neteq/decoder_database_unittest.cc
+++ b/modules/audio_coding/neteq/decoder_database_unittest.cc
@@ -28,16 +28,14 @@
TEST(DecoderDatabase, CreateAndDestroy) {
DecoderDatabase db(CreateEnvironment(),
- make_ref_counted<MockAudioDecoderFactory>(),
- absl::nullopt);
+ make_ref_counted<MockAudioDecoderFactory>(), std::nullopt);
EXPECT_EQ(0, db.Size());
EXPECT_TRUE(db.Empty());
}
TEST(DecoderDatabase, InsertAndRemove) {
DecoderDatabase db(CreateEnvironment(),
- make_ref_counted<MockAudioDecoderFactory>(),
- absl::nullopt);
+ make_ref_counted<MockAudioDecoderFactory>(), std::nullopt);
const uint8_t kPayloadType = 0;
const std::string kCodecName = "Robert\'); DROP TABLE Students;";
EXPECT_EQ(
@@ -52,8 +50,7 @@
TEST(DecoderDatabase, InsertAndRemoveAll) {
DecoderDatabase db(CreateEnvironment(),
- make_ref_counted<MockAudioDecoderFactory>(),
- absl::nullopt);
+ make_ref_counted<MockAudioDecoderFactory>(), std::nullopt);
const std::string kCodecName1 = "Robert\'); DROP TABLE Students;";
const std::string kCodecName2 = "https://xkcd.com/327/";
EXPECT_EQ(DecoderDatabase::kOK,
@@ -75,7 +72,7 @@
EXPECT_EQ("pcmu", format.name);
return absl::WrapUnique(decoder);
}));
- DecoderDatabase db(CreateEnvironment(), std::move(factory), absl::nullopt);
+ DecoderDatabase db(CreateEnvironment(), std::move(factory), std::nullopt);
const uint8_t kPayloadType = 0;
const std::string kCodecName = "pcmu";
EXPECT_EQ(
@@ -93,7 +90,7 @@
TEST(DecoderDatabase, GetDecoder) {
DecoderDatabase db(CreateEnvironment(), CreateBuiltinAudioDecoderFactory(),
- absl::nullopt);
+ std::nullopt);
const uint8_t kPayloadType = 0;
EXPECT_EQ(DecoderDatabase::kOK,
db.RegisterPayload(kPayloadType, SdpAudioFormat("l16", 8000, 1)));
@@ -103,8 +100,7 @@
TEST(DecoderDatabase, TypeTests) {
DecoderDatabase db(CreateEnvironment(),
- make_ref_counted<MockAudioDecoderFactory>(),
- absl::nullopt);
+ make_ref_counted<MockAudioDecoderFactory>(), std::nullopt);
const uint8_t kPayloadTypePcmU = 0;
const uint8_t kPayloadTypeCng = 13;
const uint8_t kPayloadTypeDtmf = 100;
@@ -138,8 +134,7 @@
TEST(DecoderDatabase, CheckPayloadTypes) {
constexpr int kNumPayloads = 10;
DecoderDatabase db(CreateEnvironment(),
- make_ref_counted<MockAudioDecoderFactory>(),
- absl::nullopt);
+ make_ref_counted<MockAudioDecoderFactory>(), std::nullopt);
// Load a number of payloads into the database. Payload types are 0, 1, ...,
// while the decoder type is the same for all payload types (this does not
// matter for the test).
@@ -181,7 +176,7 @@
// Test the methods for setting and getting active speech and CNG decoders.
TEST(DecoderDatabase, IF_ISAC(ActiveDecoders)) {
DecoderDatabase db(CreateEnvironment(), CreateBuiltinAudioDecoderFactory(),
- absl::nullopt);
+ std::nullopt);
// Load payload types.
ASSERT_EQ(DecoderDatabase::kOK,
db.RegisterPayload(0, SdpAudioFormat("pcmu", 8000, 1)));
diff --git a/modules/audio_coding/neteq/delay_manager.h b/modules/audio_coding/neteq/delay_manager.h
index d678dcf..8cee36a 100644
--- a/modules/audio_coding/neteq/delay_manager.h
+++ b/modules/audio_coding/neteq/delay_manager.h
@@ -15,8 +15,8 @@
#include <deque>
#include <memory>
+#include <optional>
-#include "absl/types/optional.h"
#include "api/field_trials_view.h"
#include "api/neteq/tick_timer.h"
#include "modules/audio_coding/neteq/histogram.h"
@@ -34,8 +34,8 @@
// Options that can be configured via field trial.
double quantile = 0.95;
double forget_factor = 0.983;
- absl::optional<double> start_forget_weight = 2;
- absl::optional<int> resample_interval_ms = 500;
+ std::optional<double> start_forget_weight = 2;
+ std::optional<int> resample_interval_ms = 500;
bool use_reorder_optimizer = true;
double reorder_forget_factor = 0.9993;
diff --git a/modules/audio_coding/neteq/delay_manager_unittest.cc b/modules/audio_coding/neteq/delay_manager_unittest.cc
index 35ae22c..d59a038 100644
--- a/modules/audio_coding/neteq/delay_manager_unittest.cc
+++ b/modules/audio_coding/neteq/delay_manager_unittest.cc
@@ -15,8 +15,8 @@
#include <math.h>
#include <memory>
+#include <optional>
-#include "absl/types/optional.h"
#include "modules/audio_coding/neteq/histogram.h"
#include "modules/audio_coding/neteq/mock/mock_histogram.h"
#include "modules/audio_coding/neteq/mock/mock_statistics_calculator.h"
diff --git a/modules/audio_coding/neteq/expand_uma_logger.cc b/modules/audio_coding/neteq/expand_uma_logger.cc
index a91358b..e735a45 100644
--- a/modules/audio_coding/neteq/expand_uma_logger.cc
+++ b/modules/audio_coding/neteq/expand_uma_logger.cc
@@ -47,7 +47,7 @@
last_value_ = samples;
sample_rate_hz_ = sample_rate_hz;
if (!last_logged_value_) {
- last_logged_value_ = absl::optional<uint64_t>(samples);
+ last_logged_value_ = std::optional<uint64_t>(samples);
}
if (!timer_->Finished()) {
@@ -58,7 +58,7 @@
RTC_DCHECK(last_logged_value_);
RTC_DCHECK_GE(last_value_, *last_logged_value_);
const uint64_t diff = last_value_ - *last_logged_value_;
- last_logged_value_ = absl::optional<uint64_t>(last_value_);
+ last_logged_value_ = std::optional<uint64_t>(last_value_);
// Calculate rate in percent.
RTC_DCHECK_GT(sample_rate_hz, 0);
const int rate = (100 * diff) / (sample_rate_hz * logging_period_s_);
diff --git a/modules/audio_coding/neteq/expand_uma_logger.h b/modules/audio_coding/neteq/expand_uma_logger.h
index cc5c20a..d6f8371 100644
--- a/modules/audio_coding/neteq/expand_uma_logger.h
+++ b/modules/audio_coding/neteq/expand_uma_logger.h
@@ -13,10 +13,10 @@
#include <stdint.h>
#include <memory>
+#include <optional>
#include <string>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/neteq/tick_timer.h"
namespace webrtc {
@@ -48,7 +48,7 @@
const int logging_period_s_;
const TickTimer& tick_timer_;
std::unique_ptr<TickTimer::Countdown> timer_;
- absl::optional<uint64_t> last_logged_value_;
+ std::optional<uint64_t> last_logged_value_;
uint64_t last_value_ = 0;
int sample_rate_hz_ = 0;
};
diff --git a/modules/audio_coding/neteq/histogram.cc b/modules/audio_coding/neteq/histogram.cc
index 4360d1a..825c8ac 100644
--- a/modules/audio_coding/neteq/histogram.cc
+++ b/modules/audio_coding/neteq/histogram.cc
@@ -13,8 +13,8 @@
#include <algorithm>
#include <cstdlib>
#include <numeric>
+#include <optional>
-#include "absl/types/optional.h"
#include "rtc_base/checks.h"
#include "rtc_base/numerics/safe_conversions.h"
@@ -22,7 +22,7 @@
Histogram::Histogram(size_t num_buckets,
int forget_factor,
- absl::optional<double> start_forget_weight)
+ std::optional<double> start_forget_weight)
: buckets_(num_buckets, 0),
forget_factor_(0),
base_forget_factor_(forget_factor),
diff --git a/modules/audio_coding/neteq/histogram.h b/modules/audio_coding/neteq/histogram.h
index 265a10e..169eb33 100644
--- a/modules/audio_coding/neteq/histogram.h
+++ b/modules/audio_coding/neteq/histogram.h
@@ -13,10 +13,9 @@
#include <string.h> // Provide access to size_t.
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
-
namespace webrtc {
class Histogram {
@@ -24,7 +23,7 @@
// Creates histogram with capacity `num_buckets` and `forget_factor` in Q15.
Histogram(size_t num_buckets,
int forget_factor,
- absl::optional<double> start_forget_weight = absl::nullopt);
+ std::optional<double> start_forget_weight = std::nullopt);
virtual ~Histogram();
@@ -47,7 +46,7 @@
// Accessors only intended for testing purposes.
int base_forget_factor_for_testing() const { return base_forget_factor_; }
int forget_factor_for_testing() const { return forget_factor_; }
- absl::optional<double> start_forget_weight_for_testing() const {
+ std::optional<double> start_forget_weight_for_testing() const {
return start_forget_weight_;
}
@@ -56,7 +55,7 @@
int forget_factor_; // Q15
const int base_forget_factor_;
int add_count_;
- const absl::optional<double> start_forget_weight_;
+ const std::optional<double> start_forget_weight_;
};
} // namespace webrtc
diff --git a/modules/audio_coding/neteq/mock/mock_decoder_database.h b/modules/audio_coding/neteq/mock/mock_decoder_database.h
index 3adae87..08e1215 100644
--- a/modules/audio_coding/neteq/mock/mock_decoder_database.h
+++ b/modules/audio_coding/neteq/mock/mock_decoder_database.h
@@ -24,7 +24,7 @@
MockDecoderDatabase()
: DecoderDatabase(CreateEnvironment(),
/*decoder_factory=*/nullptr,
- /*codec_pair_id=*/absl::nullopt) {}
+ /*codec_pair_id=*/std::nullopt) {}
~MockDecoderDatabase() override { Die(); }
MOCK_METHOD(void, Die, ());
MOCK_METHOD(bool, Empty, (), (const, override));
diff --git a/modules/audio_coding/neteq/mock/mock_neteq_controller.h b/modules/audio_coding/neteq/mock/mock_neteq_controller.h
index dc5cab4..ea7809a 100644
--- a/modules/audio_coding/neteq/mock/mock_neteq_controller.h
+++ b/modules/audio_coding/neteq/mock/mock_neteq_controller.h
@@ -39,7 +39,7 @@
MOCK_METHOD(void, ExpandDecision, (NetEq::Operation operation), (override));
MOCK_METHOD(void, AddSampleMemory, (int32_t value), (override));
MOCK_METHOD(int, TargetLevelMs, (), (const, override));
- MOCK_METHOD(absl::optional<int>,
+ MOCK_METHOD(std::optional<int>,
PacketArrived,
(int fs_hz,
bool should_update_stats,
diff --git a/modules/audio_coding/neteq/mock/mock_packet_buffer.h b/modules/audio_coding/neteq/mock/mock_packet_buffer.h
index fa44f60..85bc82b 100644
--- a/modules/audio_coding/neteq/mock/mock_packet_buffer.h
+++ b/modules/audio_coding/neteq/mock/mock_packet_buffer.h
@@ -36,7 +36,7 @@
(uint32_t timestamp, uint32_t* next_timestamp),
(const, override));
MOCK_METHOD(const Packet*, PeekNextPacket, (), (const, override));
- MOCK_METHOD(absl::optional<Packet>, GetNextPacket, (), (override));
+ MOCK_METHOD(std::optional<Packet>, GetNextPacket, (), (override));
MOCK_METHOD(int, DiscardNextPacket, (), (override));
MOCK_METHOD(void,
DiscardOldPackets,
diff --git a/modules/audio_coding/neteq/nack_tracker.cc b/modules/audio_coding/neteq/nack_tracker.cc
index a1e973b..f6f6b80 100644
--- a/modules/audio_coding/neteq/nack_tracker.cc
+++ b/modules/audio_coding/neteq/nack_tracker.cc
@@ -98,7 +98,7 @@
LimitNackListSize();
}
-absl::optional<int> NackTracker::GetSamplesPerPacket(
+std::optional<int> NackTracker::GetSamplesPerPacket(
uint16_t sequence_number_current_received_rtp,
uint32_t timestamp_current_received_rtp) const {
uint32_t timestamp_increase =
@@ -110,7 +110,7 @@
if (samples_per_packet == 0 ||
samples_per_packet > kMaxPacketSizeMs * sample_rate_khz_) {
// Not a valid samples per packet.
- return absl::nullopt;
+ return std::nullopt;
}
return samples_per_packet;
}
@@ -125,7 +125,7 @@
IsNewerSequenceNumber(sequence_number_current_received_rtp,
sequence_num_last_decoded_rtp_));
- absl::optional<int> samples_per_packet = GetSamplesPerPacket(
+ std::optional<int> samples_per_packet = GetSamplesPerPacket(
sequence_number_current_received_rtp, timestamp_current_received_rtp);
if (!samples_per_packet) {
return;
diff --git a/modules/audio_coding/neteq/nack_tracker.h b/modules/audio_coding/neteq/nack_tracker.h
index c95496d..a11f483 100644
--- a/modules/audio_coding/neteq/nack_tracker.h
+++ b/modules/audio_coding/neteq/nack_tracker.h
@@ -15,9 +15,9 @@
#include <stdint.h>
#include <map>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/field_trials_view.h"
#include "modules/include/module_common_types_public.h"
#include "rtc_base/gtest_prod_util.h"
@@ -151,7 +151,7 @@
// Returns a valid number of samples per packet given the current received
// sequence number and timestamp or nullopt of none could be computed.
- absl::optional<int> GetSamplesPerPacket(
+ std::optional<int> GetSamplesPerPacket(
uint16_t sequence_number_current_received_rtp,
uint32_t timestamp_current_received_rtp) const;
diff --git a/modules/audio_coding/neteq/neteq_decoder_plc_unittest.cc b/modules/audio_coding/neteq/neteq_decoder_plc_unittest.cc
index ec6ade9..223dfe4 100644
--- a/modules/audio_coding/neteq/neteq_decoder_plc_unittest.cc
+++ b/modules/audio_coding/neteq/neteq_decoder_plc_unittest.cc
@@ -11,10 +11,10 @@
// Test to verify correct operation when using the decoder-internal PLC.
#include <memory>
+#include <optional>
#include <utility>
#include <vector>
-#include "absl/types/optional.h"
#include "modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h"
#include "modules/audio_coding/neteq/tools/audio_checksum.h"
#include "modules/audio_coding/neteq/tools/audio_sink.h"
@@ -119,15 +119,15 @@
burst_length_(burst_length),
input_(std::move(input)) {}
- absl::optional<int64_t> NextPacketTime() const override {
+ std::optional<int64_t> NextPacketTime() const override {
return input_->NextPacketTime();
}
- absl::optional<int64_t> NextOutputEventTime() const override {
+ std::optional<int64_t> NextOutputEventTime() const override {
return input_->NextOutputEventTime();
}
- absl::optional<SetMinimumDelayInfo> NextSetMinimumDelayInfo() const override {
+ std::optional<SetMinimumDelayInfo> NextSetMinimumDelayInfo() const override {
return input_->NextSetMinimumDelayInfo();
}
@@ -151,7 +151,7 @@
bool ended() const override { return input_->ended(); }
- absl::optional<RTPHeader> NextHeader() const override {
+ std::optional<RTPHeader> NextHeader() const override {
return input_->NextHeader();
}
diff --git a/modules/audio_coding/neteq/neteq_impl.cc b/modules/audio_coding/neteq/neteq_impl.cc
index a93954f..3507510 100644
--- a/modules/audio_coding/neteq/neteq_impl.cc
+++ b/modules/audio_coding/neteq/neteq_impl.cc
@@ -201,7 +201,7 @@
int NetEqImpl::GetAudio(AudioFrame* audio_frame,
bool* muted,
int* current_sample_rate_hz,
- absl::optional<Operation> action_override) {
+ std::optional<Operation> action_override) {
TRACE_EVENT0("webrtc", "NetEqImpl::GetAudio");
MutexLock lock(&mutex_);
if (GetAudioInternal(audio_frame, action_override) != 0) {
@@ -357,14 +357,14 @@
return result;
}
-absl::optional<uint32_t> NetEqImpl::GetPlayoutTimestamp() const {
+std::optional<uint32_t> NetEqImpl::GetPlayoutTimestamp() const {
MutexLock lock(&mutex_);
if (first_packet_ || last_mode_ == Mode::kRfc3389Cng ||
last_mode_ == Mode::kCodecInternalCng) {
// We don't have a valid RTP timestamp until we have decoded our first
// RTP packet. Also, the RTP timestamp is not accurate while playing CNG,
// which is indicated by returning an empty value.
- return absl::nullopt;
+ return std::nullopt;
}
return timestamp_scaler_->ToExternal(playout_timestamp_);
}
@@ -374,16 +374,15 @@
return last_output_sample_rate_hz_;
}
-absl::optional<NetEq::DecoderFormat> NetEqImpl::GetCurrentDecoderFormat()
- const {
+std::optional<NetEq::DecoderFormat> NetEqImpl::GetCurrentDecoderFormat() const {
MutexLock lock(&mutex_);
if (!current_rtp_payload_type_.has_value()) {
- return absl::nullopt;
+ return std::nullopt;
}
const DecoderDatabase::DecoderInfo* di =
decoder_database_->GetDecoderInfo(*current_rtp_payload_type_);
if (di == nullptr) {
- return absl::nullopt;
+ return std::nullopt;
}
return DecoderFormat{
/*payload_type=*/*current_rtp_payload_type_,
@@ -713,7 +712,7 @@
if (current_cng_rtp_payload_type_ &&
*current_cng_rtp_payload_type_ != payload_type) {
// New CNG payload type implies new codec type.
- current_rtp_payload_type_ = absl::nullopt;
+ current_rtp_payload_type_ = std::nullopt;
changed = true;
}
current_cng_rtp_payload_type_ = payload_type;
@@ -724,7 +723,7 @@
(current_cng_rtp_payload_type_ &&
!EqualSampleRates(payload_type, *current_cng_rtp_payload_type_,
*decoder_database_))) {
- current_cng_rtp_payload_type_ = absl::nullopt;
+ current_cng_rtp_payload_type_ = std::nullopt;
changed = true;
}
current_rtp_payload_type_ = payload_type;
@@ -733,7 +732,7 @@
}
int NetEqImpl::GetAudioInternal(AudioFrame* audio_frame,
- absl::optional<Operation> action_override) {
+ std::optional<Operation> action_override) {
PacketList packet_list;
DtmfEvent dtmf_event;
Operation operation;
@@ -987,7 +986,7 @@
PacketList* packet_list,
DtmfEvent* dtmf_event,
bool* play_dtmf,
- absl::optional<Operation> action_override) {
+ std::optional<Operation> action_override) {
// Initialize output variables.
*play_dtmf = false;
*operation = Operation::kUndefined;
@@ -1912,7 +1911,7 @@
// Packet extraction loop.
do {
timestamp_ = next_packet->timestamp;
- absl::optional<Packet> packet = packet_buffer_->GetNextPacket();
+ std::optional<Packet> packet = packet_buffer_->GetNextPacket();
// `next_packet` may be invalid after the `packet_buffer_` operation.
next_packet = nullptr;
if (!packet) {
@@ -1980,7 +1979,7 @@
!has_cng_packet;
packet_list->push_back(std::move(*packet)); // Store packet in list.
- packet = absl::nullopt; // Ensure it's never used after the move.
+ packet = std::nullopt; // Ensure it's never used after the move.
} while (extracted_samples < required_samples && next_packet_available);
if (extracted_samples > 0) {
diff --git a/modules/audio_coding/neteq/neteq_impl.h b/modules/audio_coding/neteq/neteq_impl.h
index 0ce80b7..2571927 100644
--- a/modules/audio_coding/neteq/neteq_impl.h
+++ b/modules/audio_coding/neteq/neteq_impl.h
@@ -18,7 +18,6 @@
#include <utility>
#include <vector>
-#include "absl/types/optional.h"
#include "api/audio/audio_frame.h"
#include "api/environment/environment.h"
#include "api/neteq/neteq.h"
@@ -138,7 +137,7 @@
AudioFrame* audio_frame,
bool* muted = nullptr,
int* current_sample_rate_hz = nullptr,
- absl::optional<Operation> action_override = absl::nullopt) override;
+ std::optional<Operation> action_override = std::nullopt) override;
void SetCodecs(const std::map<int, SdpAudioFormat>& codecs) override;
@@ -173,11 +172,11 @@
NetEqOperationsAndState GetOperationsAndState() const override;
- absl::optional<uint32_t> GetPlayoutTimestamp() const override;
+ std::optional<uint32_t> GetPlayoutTimestamp() const override;
int last_output_sample_rate_hz() const override;
- absl::optional<DecoderFormat> GetCurrentDecoderFormat() const override;
+ std::optional<DecoderFormat> GetCurrentDecoderFormat() const override;
// Flushes both the packet buffer and the sync buffer.
void FlushBuffers() override;
@@ -219,7 +218,7 @@
// Delivers 10 ms of audio data. The data is written to `audio_frame`.
// Returns 0 on success, otherwise an error code.
int GetAudioInternal(AudioFrame* audio_frame,
- absl::optional<Operation> action_override)
+ std::optional<Operation> action_override)
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
// Provides a decision to the GetAudioInternal method. The decision what to
@@ -231,7 +230,7 @@
PacketList* packet_list,
DtmfEvent* dtmf_event,
bool* play_dtmf,
- absl::optional<Operation> action_override)
+ std::optional<Operation> action_override)
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
// Decodes the speech packets in `packet_list`, and writes the results to
@@ -378,7 +377,7 @@
size_t decoder_frame_length_ RTC_GUARDED_BY(mutex_);
Mode last_mode_ RTC_GUARDED_BY(mutex_);
Operation last_operation_ RTC_GUARDED_BY(mutex_);
- absl::optional<AudioDecoder::SpeechType> last_decoded_type_
+ std::optional<AudioDecoder::SpeechType> last_decoded_type_
RTC_GUARDED_BY(mutex_);
size_t decoded_buffer_length_ RTC_GUARDED_BY(mutex_);
std::unique_ptr<int16_t[]> decoded_buffer_ RTC_GUARDED_BY(mutex_);
@@ -386,8 +385,8 @@
bool new_codec_ RTC_GUARDED_BY(mutex_);
uint32_t timestamp_ RTC_GUARDED_BY(mutex_);
bool reset_decoder_ RTC_GUARDED_BY(mutex_);
- absl::optional<uint8_t> current_rtp_payload_type_ RTC_GUARDED_BY(mutex_);
- absl::optional<uint8_t> current_cng_rtp_payload_type_ RTC_GUARDED_BY(mutex_);
+ std::optional<uint8_t> current_rtp_payload_type_ RTC_GUARDED_BY(mutex_);
+ std::optional<uint8_t> current_cng_rtp_payload_type_ RTC_GUARDED_BY(mutex_);
bool first_packet_ RTC_GUARDED_BY(mutex_);
bool enable_fast_accelerate_ RTC_GUARDED_BY(mutex_);
std::unique_ptr<NackTracker> nack_ RTC_GUARDED_BY(mutex_);
diff --git a/modules/audio_coding/neteq/neteq_impl_unittest.cc b/modules/audio_coding/neteq/neteq_impl_unittest.cc
index 561a10e..740d969 100644
--- a/modules/audio_coding/neteq/neteq_impl_unittest.cc
+++ b/modules/audio_coding/neteq/neteq_impl_unittest.cc
@@ -320,7 +320,7 @@
return mock_decoder;
}));
DecoderDatabase::DecoderInfo info(env, SdpAudioFormat("pcmu", 8000, 1),
- absl::nullopt, mock_decoder_factory.get());
+ std::nullopt, mock_decoder_factory.get());
// Expectations for decoder database.
EXPECT_CALL(*mock_decoder_database_, GetDecoderInfo(kPayloadType))
@@ -532,10 +532,10 @@
// The value of the last of the output samples is the same as the number of
// samples played from the decoded packet. Thus, this number + the RTP
// timestamp should match the playout timestamp.
- // Wrap the expected value in an absl::optional to compare them as such.
+ // Wrap the expected value in an std::optional to compare them as such.
EXPECT_EQ(
- absl::optional<uint32_t>(rtp_header.timestamp +
- output.data()[output.samples_per_channel_ - 1]),
+ std::optional<uint32_t>(rtp_header.timestamp +
+ output.data()[output.samples_per_channel_ - 1]),
neteq_->GetPlayoutTimestamp());
// Check the timestamp for the last value in the sync buffer. This should
@@ -1652,7 +1652,7 @@
return dec;
}));
DecoderDatabase::DecoderInfo info(env, SdpAudioFormat("pcmu", 8000, 1),
- absl::nullopt, mock_decoder_factory.get());
+ std::nullopt, mock_decoder_factory.get());
// Expectations for decoder database.
EXPECT_CALL(*mock_decoder_database_, GetDecoderInfo(kPayloadType))
.WillRepeatedly(Return(&info));
@@ -1734,7 +1734,7 @@
header.timestamp = 1234;
uint8_t payload[160] = {0};
- absl::optional<NetEq::DecoderFormat> decoder =
+ std::optional<NetEq::DecoderFormat> decoder =
neteq_->GetCurrentDecoderFormat();
EXPECT_FALSE(decoder.has_value());
diff --git a/modules/audio_coding/neteq/neteq_network_stats_unittest.cc b/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
index 171bcae..a145bbb 100644
--- a/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
+++ b/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
@@ -58,7 +58,7 @@
size_t Duration() const override { return kPacketDuration; }
- absl::optional<DecodeResult> Decode(
+ std::optional<DecodeResult> Decode(
rtc::ArrayView<int16_t> decoded) const override {
const size_t output_size =
sizeof(int16_t) * kPacketDuration * num_channels_;
@@ -69,7 +69,7 @@
} else {
ADD_FAILURE() << "Expected decoded.size() to be >= output_size ("
<< decoded.size() << " vs. " << output_size << ")";
- return absl::nullopt;
+ return std::nullopt;
}
}
diff --git a/modules/audio_coding/neteq/neteq_unittest.cc b/modules/audio_coding/neteq/neteq_unittest.cc
index 2ddd9e3..1fc9710 100644
--- a/modules/audio_coding/neteq/neteq_unittest.cc
+++ b/modules/audio_coding/neteq/neteq_unittest.cc
@@ -513,7 +513,7 @@
ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
EXPECT_EQ(AudioFrame::kNormalSpeech, out_frame_.speech_type_);
- absl::optional<uint32_t> playout_timestamp = neteq_->GetPlayoutTimestamp();
+ std::optional<uint32_t> playout_timestamp = neteq_->GetPlayoutTimestamp();
ASSERT_TRUE(playout_timestamp);
EXPECT_EQ(first_speech_timestamp + kSamples - algorithmic_delay_samples,
*playout_timestamp);
@@ -1005,7 +1005,7 @@
{8, kRtpExtensionVideoTiming}};
std::unique_ptr<NetEqInput> input = CreateNetEqRtpDumpInput(
webrtc::test::ResourcePath("audio_coding/neteq_universal_new", "rtp"),
- rtp_ext_map, absl::nullopt /*No SSRC filter*/);
+ rtp_ext_map, std::nullopt /*No SSRC filter*/);
std::unique_ptr<TimeLimitedNetEqInput> input_time_limit(
new TimeLimitedNetEqInput(std::move(input), 20000));
std::unique_ptr<AudioSink> output(new VoidAudioSink);
diff --git a/modules/audio_coding/neteq/packet.h b/modules/audio_coding/neteq/packet.h
index 795c36d..9a1f919 100644
--- a/modules/audio_coding/neteq/packet.h
+++ b/modules/audio_coding/neteq/packet.h
@@ -74,7 +74,7 @@
// Datagram excluding RTP header and header extension.
rtc::Buffer payload;
Priority priority;
- absl::optional<RtpPacketInfo> packet_info;
+ std::optional<RtpPacketInfo> packet_info;
std::unique_ptr<TickTimer::Stopwatch> waiting_time;
std::unique_ptr<AudioDecoder::EncodedAudioFrame> frame;
diff --git a/modules/audio_coding/neteq/packet_buffer.cc b/modules/audio_coding/neteq/packet_buffer.cc
index 2abbbd9d..bcc6a85 100644
--- a/modules/audio_coding/neteq/packet_buffer.cc
+++ b/modules/audio_coding/neteq/packet_buffer.cc
@@ -151,13 +151,13 @@
return buffer_.empty() ? nullptr : &buffer_.front();
}
-absl::optional<Packet> PacketBuffer::GetNextPacket() {
+std::optional<Packet> PacketBuffer::GetNextPacket() {
if (Empty()) {
// Buffer is empty.
- return absl::nullopt;
+ return std::nullopt;
}
- absl::optional<Packet> packet(std::move(buffer_.front()));
+ std::optional<Packet> packet(std::move(buffer_.front()));
// Assert that the packet sanity checks in InsertPacket method works.
RTC_DCHECK(!packet->empty());
buffer_.pop_front();
diff --git a/modules/audio_coding/neteq/packet_buffer.h b/modules/audio_coding/neteq/packet_buffer.h
index 795dd4e..547e993 100644
--- a/modules/audio_coding/neteq/packet_buffer.h
+++ b/modules/audio_coding/neteq/packet_buffer.h
@@ -11,7 +11,8 @@
#ifndef MODULES_AUDIO_CODING_NETEQ_PACKET_BUFFER_H_
#define MODULES_AUDIO_CODING_NETEQ_PACKET_BUFFER_H_
-#include "absl/types/optional.h"
+#include <optional>
+
#include "modules/audio_coding/neteq/decoder_database.h"
#include "modules/audio_coding/neteq/packet.h"
#include "modules/include/module_common_types_public.h" // IsNewerTimestamp
@@ -78,7 +79,7 @@
// Extracts the first packet in the buffer and returns it.
// Returns an empty optional if the buffer is empty.
- virtual absl::optional<Packet> GetNextPacket();
+ virtual std::optional<Packet> GetNextPacket();
// Discards the first packet in the buffer. The packet is deleted.
// Returns PacketBuffer::kBufferEmpty if the buffer is empty,
diff --git a/modules/audio_coding/neteq/packet_buffer_unittest.cc b/modules/audio_coding/neteq/packet_buffer_unittest.cc
index 8f307a9..5bdb47d 100644
--- a/modules/audio_coding/neteq/packet_buffer_unittest.cc
+++ b/modules/audio_coding/neteq/packet_buffer_unittest.cc
@@ -36,7 +36,7 @@
MOCK_METHOD(bool, IsDtxPacket, (), (const, override));
- MOCK_METHOD(absl::optional<DecodeResult>,
+ MOCK_METHOD(std::optional<DecodeResult>,
Decode,
(rtc::ArrayView<int16_t> decoded),
(const, override));
@@ -253,7 +253,7 @@
EXPECT_EQ(kExpectPacketsInBuffer, buffer.NumPacketsInBuffer());
for (size_t i = 0; i < kExpectPacketsInBuffer; ++i) {
- const absl::optional<Packet> packet = buffer.GetNextPacket();
+ const std::optional<Packet> packet = buffer.GetNextPacket();
EXPECT_EQ(packet, expect_order[i]); // Compare contents.
}
EXPECT_TRUE(buffer.Empty());
@@ -356,7 +356,7 @@
// Extract them and make sure that come out in the right order.
uint32_t current_ts = start_ts;
for (int i = 0; i < 10; ++i) {
- const absl::optional<Packet> packet = buffer.GetNextPacket();
+ const std::optional<Packet> packet = buffer.GetNextPacket();
ASSERT_TRUE(packet);
EXPECT_EQ(current_ts, packet->timestamp);
current_ts += ts_increment;
diff --git a/modules/audio_coding/neteq/red_payload_splitter_unittest.cc b/modules/audio_coding/neteq/red_payload_splitter_unittest.cc
index 5e84c75..8a17d85 100644
--- a/modules/audio_coding/neteq/red_payload_splitter_unittest.cc
+++ b/modules/audio_coding/neteq/red_payload_splitter_unittest.cc
@@ -299,7 +299,7 @@
// easier to just register the payload types and let the actual implementation
// do its job.
DecoderDatabase decoder_database(
- env, make_ref_counted<MockAudioDecoderFactory>(), absl::nullopt);
+ env, make_ref_counted<MockAudioDecoderFactory>(), std::nullopt);
decoder_database.RegisterPayload(0, SdpAudioFormat("cn", 8000, 1));
decoder_database.RegisterPayload(1, SdpAudioFormat("pcmu", 8000, 1));
decoder_database.RegisterPayload(2,
@@ -335,7 +335,7 @@
// easier to just register the payload types and let the actual implementation
// do its job.
DecoderDatabase decoder_database(
- env, make_ref_counted<MockAudioDecoderFactory>(), absl::nullopt);
+ env, make_ref_counted<MockAudioDecoderFactory>(), std::nullopt);
decoder_database.RegisterPayload(kRedPayloadType,
SdpAudioFormat("red", 8000, 1));
diff --git a/modules/audio_coding/neteq/reorder_optimizer.cc b/modules/audio_coding/neteq/reorder_optimizer.cc
index f6e073f..be96de1 100644
--- a/modules/audio_coding/neteq/reorder_optimizer.cc
+++ b/modules/audio_coding/neteq/reorder_optimizer.cc
@@ -25,7 +25,7 @@
ReorderOptimizer::ReorderOptimizer(int forget_factor,
int ms_per_loss_percent,
- absl::optional<int> start_forget_weight)
+ std::optional<int> start_forget_weight)
: histogram_(kDelayBuckets, forget_factor, start_forget_weight),
ms_per_loss_percent_(ms_per_loss_percent) {}
diff --git a/modules/audio_coding/neteq/reorder_optimizer.h b/modules/audio_coding/neteq/reorder_optimizer.h
index 06f6bc7..3c5e8fa 100644
--- a/modules/audio_coding/neteq/reorder_optimizer.h
+++ b/modules/audio_coding/neteq/reorder_optimizer.h
@@ -11,7 +11,8 @@
#ifndef MODULES_AUDIO_CODING_NETEQ_REORDER_OPTIMIZER_H_
#define MODULES_AUDIO_CODING_NETEQ_REORDER_OPTIMIZER_H_
-#include "absl/types/optional.h"
+#include <optional>
+
#include "modules/audio_coding/neteq/histogram.h"
namespace webrtc {
@@ -23,11 +24,11 @@
public:
ReorderOptimizer(int forget_factor,
int ms_per_loss_percent,
- absl::optional<int> start_forget_weight);
+ std::optional<int> start_forget_weight);
void Update(int relative_delay_ms, bool reordered, int base_delay_ms);
- absl::optional<int> GetOptimalDelayMs() const { return optimal_delay_ms_; }
+ std::optional<int> GetOptimalDelayMs() const { return optimal_delay_ms_; }
void Reset();
@@ -36,7 +37,7 @@
Histogram histogram_;
const int ms_per_loss_percent_;
- absl::optional<int> optimal_delay_ms_;
+ std::optional<int> optimal_delay_ms_;
};
} // namespace webrtc
diff --git a/modules/audio_coding/neteq/test/neteq_decoding_test.cc b/modules/audio_coding/neteq/test/neteq_decoding_test.cc
index b5f7919..a817276 100644
--- a/modules/audio_coding/neteq/test/neteq_decoding_test.cc
+++ b/modules/audio_coding/neteq/test/neteq_decoding_test.cc
@@ -269,7 +269,7 @@
ASSERT_EQ(1u, output.num_channels_);
// Expect delay (in samples) to be less than 2 packets.
- absl::optional<uint32_t> playout_timestamp = neteq_->GetPlayoutTimestamp();
+ std::optional<uint32_t> playout_timestamp = neteq_->GetPlayoutTimestamp();
ASSERT_TRUE(playout_timestamp);
EXPECT_LE(timestamp - *playout_timestamp,
static_cast<uint32_t>(kSamples * 2));
@@ -314,7 +314,7 @@
}
EXPECT_EQ(AudioFrame::kNormalSpeech, out_frame_.speech_type_);
- absl::optional<uint32_t> playout_timestamp = neteq_->GetPlayoutTimestamp();
+ std::optional<uint32_t> playout_timestamp = neteq_->GetPlayoutTimestamp();
ASSERT_TRUE(playout_timestamp);
int32_t delay_before = timestamp - *playout_timestamp;
diff --git a/modules/audio_coding/neteq/timestamp_scaler_unittest.cc b/modules/audio_coding/neteq/timestamp_scaler_unittest.cc
index 804e4d8..e880004 100644
--- a/modules/audio_coding/neteq/timestamp_scaler_unittest.cc
+++ b/modules/audio_coding/neteq/timestamp_scaler_unittest.cc
@@ -30,7 +30,7 @@
auto factory = CreateBuiltinAudioDecoderFactory();
// Use PCMu, because it doesn't use scaled timestamps.
const DecoderDatabase::DecoderInfo info(env, SdpAudioFormat("pcmu", 8000, 1),
- absl::nullopt, factory.get());
+ std::nullopt, factory.get());
static const uint8_t kRtpPayloadType = 0;
EXPECT_CALL(db, GetDecoderInfo(kRtpPayloadType))
.WillRepeatedly(Return(&info));
@@ -53,7 +53,7 @@
auto factory = CreateBuiltinAudioDecoderFactory();
// Use PCMu, because it doesn't use scaled timestamps.
const DecoderDatabase::DecoderInfo info(env, SdpAudioFormat("pcmu", 8000, 1),
- absl::nullopt, factory.get());
+ std::nullopt, factory.get());
static const uint8_t kRtpPayloadType = 0;
EXPECT_CALL(db, GetDecoderInfo(kRtpPayloadType))
.WillRepeatedly(Return(&info));
@@ -81,7 +81,7 @@
auto factory = CreateBuiltinAudioDecoderFactory();
// Use G722, which has a factor 2 scaling.
const DecoderDatabase::DecoderInfo info(env, SdpAudioFormat("g722", 8000, 1),
- absl::nullopt, factory.get());
+ std::nullopt, factory.get());
static const uint8_t kRtpPayloadType = 17;
EXPECT_CALL(db, GetDecoderInfo(kRtpPayloadType))
.WillRepeatedly(Return(&info));
@@ -108,7 +108,7 @@
auto factory = CreateBuiltinAudioDecoderFactory();
// Use G722, which has a factor 2 scaling.
const DecoderDatabase::DecoderInfo info(env, SdpAudioFormat("g722", 8000, 1),
- absl::nullopt, factory.get());
+ std::nullopt, factory.get());
static const uint8_t kRtpPayloadType = 17;
EXPECT_CALL(db, GetDecoderInfo(kRtpPayloadType))
.WillRepeatedly(Return(&info));
@@ -139,9 +139,9 @@
auto factory = CreateBuiltinAudioDecoderFactory();
// Use G722, which has a factor 2 scaling.
const DecoderDatabase::DecoderInfo info_g722(
- env, SdpAudioFormat("g722", 8000, 1), absl::nullopt, factory.get());
+ env, SdpAudioFormat("g722", 8000, 1), std::nullopt, factory.get());
const DecoderDatabase::DecoderInfo info_cng(
- env, SdpAudioFormat("cn", 16000, 1), absl::nullopt, factory.get());
+ env, SdpAudioFormat("cn", 16000, 1), std::nullopt, factory.get());
static const uint8_t kRtpPayloadTypeG722 = 17;
static const uint8_t kRtpPayloadTypeCng = 13;
EXPECT_CALL(db, GetDecoderInfo(kRtpPayloadTypeG722))
@@ -184,7 +184,7 @@
auto factory = CreateBuiltinAudioDecoderFactory();
// Use G722, which has a factor 2 scaling.
const DecoderDatabase::DecoderInfo info(env, SdpAudioFormat("g722", 8000, 1),
- absl::nullopt, factory.get());
+ std::nullopt, factory.get());
static const uint8_t kRtpPayloadType = 17;
EXPECT_CALL(db, GetDecoderInfo(kRtpPayloadType))
.WillRepeatedly(Return(&info));
@@ -215,7 +215,7 @@
auto factory = CreateBuiltinAudioDecoderFactory();
// Use G722, which has a factor 2 scaling.
const DecoderDatabase::DecoderInfo info(env, SdpAudioFormat("g722", 8000, 1),
- absl::nullopt, factory.get());
+ std::nullopt, factory.get());
static const uint8_t kRtpPayloadType = 17;
EXPECT_CALL(db, GetDecoderInfo(kRtpPayloadType))
.WillRepeatedly(Return(&info));
@@ -250,7 +250,7 @@
auto factory = CreateBuiltinAudioDecoderFactory();
// Use G722, which has a factor 2 scaling.
const DecoderDatabase::DecoderInfo info(env, SdpAudioFormat("g722", 8000, 1),
- absl::nullopt, factory.get());
+ std::nullopt, factory.get());
static const uint8_t kRtpPayloadType = 17;
EXPECT_CALL(db, GetDecoderInfo(kRtpPayloadType))
.WillRepeatedly(Return(&info));
@@ -292,7 +292,7 @@
MockDecoderDatabase db;
auto factory = CreateBuiltinAudioDecoderFactory();
const DecoderDatabase::DecoderInfo info(env, SdpAudioFormat("opus", 48000, 2),
- absl::nullopt, factory.get());
+ std::nullopt, factory.get());
static const uint8_t kRtpPayloadType = 17;
EXPECT_CALL(db, GetDecoderInfo(kRtpPayloadType))
.WillRepeatedly(Return(&info));
diff --git a/modules/audio_coding/neteq/tools/encode_neteq_input.cc b/modules/audio_coding/neteq/tools/encode_neteq_input.cc
index 87b987d..f0c9102 100644
--- a/modules/audio_coding/neteq/tools/encode_neteq_input.cc
+++ b/modules/audio_coding/neteq/tools/encode_neteq_input.cc
@@ -29,12 +29,12 @@
EncodeNetEqInput::~EncodeNetEqInput() = default;
-absl::optional<int64_t> EncodeNetEqInput::NextPacketTime() const {
+std::optional<int64_t> EncodeNetEqInput::NextPacketTime() const {
RTC_DCHECK(packet_data_);
return static_cast<int64_t>(packet_data_->time_ms);
}
-absl::optional<int64_t> EncodeNetEqInput::NextOutputEventTime() const {
+std::optional<int64_t> EncodeNetEqInput::NextOutputEventTime() const {
return next_output_event_ms_;
}
@@ -56,7 +56,7 @@
return next_output_event_ms_ > input_duration_ms_;
}
-absl::optional<RTPHeader> EncodeNetEqInput::NextHeader() const {
+std::optional<RTPHeader> EncodeNetEqInput::NextHeader() const {
RTC_DCHECK(packet_data_);
return packet_data_->header;
}
diff --git a/modules/audio_coding/neteq/tools/encode_neteq_input.h b/modules/audio_coding/neteq/tools/encode_neteq_input.h
index f2ed4b1..a92d1b1 100644
--- a/modules/audio_coding/neteq/tools/encode_neteq_input.h
+++ b/modules/audio_coding/neteq/tools/encode_neteq_input.h
@@ -37,12 +37,12 @@
int64_t input_duration_ms);
~EncodeNetEqInput() override;
- absl::optional<int64_t> NextPacketTime() const override;
+ std::optional<int64_t> NextPacketTime() const override;
- absl::optional<int64_t> NextOutputEventTime() const override;
+ std::optional<int64_t> NextOutputEventTime() const override;
- absl::optional<SetMinimumDelayInfo> NextSetMinimumDelayInfo() const override {
- return absl::nullopt;
+ std::optional<SetMinimumDelayInfo> NextSetMinimumDelayInfo() const override {
+ return std::nullopt;
}
std::unique_ptr<PacketData> PopPacket() override;
@@ -53,7 +53,7 @@
bool ended() const override;
- absl::optional<RTPHeader> NextHeader() const override;
+ std::optional<RTPHeader> NextHeader() const override;
private:
static constexpr int64_t kOutputPeriodMs = 10;
diff --git a/modules/audio_coding/neteq/tools/fake_decode_from_file.cc b/modules/audio_coding/neteq/tools/fake_decode_from_file.cc
index ad52239..bd6c204 100644
--- a/modules/audio_coding/neteq/tools/fake_decode_from_file.cc
+++ b/modules/audio_coding/neteq/tools/fake_decode_from_file.cc
@@ -32,7 +32,7 @@
size_t Duration() const override { return duration_; }
- absl::optional<DecodeResult> Decode(
+ std::optional<DecodeResult> Decode(
rtc::ArrayView<int16_t> decoded) const override {
if (is_dtx_) {
std::fill_n(decoded.data(), duration_, 0);
diff --git a/modules/audio_coding/neteq/tools/fake_decode_from_file.h b/modules/audio_coding/neteq/tools/fake_decode_from_file.h
index 050a29d..e62950a 100644
--- a/modules/audio_coding/neteq/tools/fake_decode_from_file.h
+++ b/modules/audio_coding/neteq/tools/fake_decode_from_file.h
@@ -12,8 +12,8 @@
#define MODULES_AUDIO_CODING_NETEQ_TOOLS_FAKE_DECODE_FROM_FILE_H_
#include <memory>
+#include <optional>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/audio_codecs/audio_decoder.h"
#include "modules/audio_coding/neteq/tools/input_audio_file.h"
@@ -67,7 +67,7 @@
private:
std::unique_ptr<InputAudioFile> input_;
- absl::optional<uint32_t> next_timestamp_from_input_;
+ std::optional<uint32_t> next_timestamp_from_input_;
const int sample_rate_hz_;
const bool stereo_;
};
diff --git a/modules/audio_coding/neteq/tools/initial_packet_inserter_neteq_input.cc b/modules/audio_coding/neteq/tools/initial_packet_inserter_neteq_input.cc
index 763078e..b41d773 100644
--- a/modules/audio_coding/neteq/tools/initial_packet_inserter_neteq_input.cc
+++ b/modules/audio_coding/neteq/tools/initial_packet_inserter_neteq_input.cc
@@ -27,17 +27,16 @@
packets_to_insert_(number_of_initial_packets),
sample_rate_hz_(sample_rate_hz) {}
-absl::optional<int64_t> InitialPacketInserterNetEqInput::NextPacketTime()
- const {
+std::optional<int64_t> InitialPacketInserterNetEqInput::NextPacketTime() const {
return source_->NextPacketTime();
}
-absl::optional<int64_t> InitialPacketInserterNetEqInput::NextOutputEventTime()
+std::optional<int64_t> InitialPacketInserterNetEqInput::NextOutputEventTime()
const {
return source_->NextOutputEventTime();
}
-absl::optional<NetEqInput::SetMinimumDelayInfo>
+std::optional<NetEqInput::SetMinimumDelayInfo>
InitialPacketInserterNetEqInput::NextSetMinimumDelayInfo() const {
return source_->NextSetMinimumDelayInfo();
}
@@ -80,7 +79,7 @@
return source_->ended();
}
-absl::optional<RTPHeader> InitialPacketInserterNetEqInput::NextHeader() const {
+std::optional<RTPHeader> InitialPacketInserterNetEqInput::NextHeader() const {
return source_->NextHeader();
}
diff --git a/modules/audio_coding/neteq/tools/initial_packet_inserter_neteq_input.h b/modules/audio_coding/neteq/tools/initial_packet_inserter_neteq_input.h
index 7246949..f72c21c 100644
--- a/modules/audio_coding/neteq/tools/initial_packet_inserter_neteq_input.h
+++ b/modules/audio_coding/neteq/tools/initial_packet_inserter_neteq_input.h
@@ -27,14 +27,14 @@
InitialPacketInserterNetEqInput(std::unique_ptr<NetEqInput> source,
int number_of_initial_packets,
int sample_rate_hz);
- absl::optional<int64_t> NextPacketTime() const override;
- absl::optional<int64_t> NextOutputEventTime() const override;
- absl::optional<SetMinimumDelayInfo> NextSetMinimumDelayInfo() const override;
+ std::optional<int64_t> NextPacketTime() const override;
+ std::optional<int64_t> NextOutputEventTime() const override;
+ std::optional<SetMinimumDelayInfo> NextSetMinimumDelayInfo() const override;
std::unique_ptr<PacketData> PopPacket() override;
void AdvanceOutputEvent() override;
void AdvanceSetMinimumDelay() override;
bool ended() const override;
- absl::optional<RTPHeader> NextHeader() const override;
+ std::optional<RTPHeader> NextHeader() const override;
private:
const std::unique_ptr<NetEqInput> source_;
diff --git a/modules/audio_coding/neteq/tools/neteq_delay_analyzer.h b/modules/audio_coding/neteq/tools/neteq_delay_analyzer.h
index ffcba58..cdf178b 100644
--- a/modules/audio_coding/neteq/tools/neteq_delay_analyzer.h
+++ b/modules/audio_coding/neteq/tools/neteq_delay_analyzer.h
@@ -12,12 +12,12 @@
#define MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_DELAY_ANALYZER_H_
#include <map>
+#include <optional>
#include <set>
#include <string>
#include <vector>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "modules/audio_coding/neteq/tools/neteq_input.h"
#include "modules/audio_coding/neteq/tools/neteq_test.h"
@@ -57,10 +57,10 @@
struct TimingData {
explicit TimingData(int64_t at) : arrival_time_ms(at) {}
int64_t arrival_time_ms;
- absl::optional<int64_t> decode_get_audio_count;
- absl::optional<int64_t> sync_delay_ms;
- absl::optional<int> target_delay_ms;
- absl::optional<int> current_delay_ms;
+ std::optional<int64_t> decode_get_audio_count;
+ std::optional<int64_t> sync_delay_ms;
+ std::optional<int> target_delay_ms;
+ std::optional<int> current_delay_ms;
};
std::map<uint32_t, TimingData> data_;
std::vector<int64_t> get_audio_time_ms_;
diff --git a/modules/audio_coding/neteq/tools/neteq_event_log_input.cc b/modules/audio_coding/neteq/tools/neteq_event_log_input.cc
index 0ca855b..d42a31b 100644
--- a/modules/audio_coding/neteq/tools/neteq_event_log_input.cc
+++ b/modules/audio_coding/neteq/tools/neteq_event_log_input.cc
@@ -26,7 +26,7 @@
const std::vector<LoggedAudioPlayoutEvent>& output_events,
const std::vector<LoggedNetEqSetMinimumDelayEvent>&
neteq_set_minimum_delay_events,
- absl::optional<int64_t> end_time_ms)
+ std::optional<int64_t> end_time_ms)
: packet_stream_(packet_stream),
packet_stream_it_(packet_stream_.begin()),
output_events_(output_events),
@@ -43,34 +43,34 @@
}
}
- absl::optional<int64_t> NextPacketTime() const override {
+ std::optional<int64_t> NextPacketTime() const override {
if (packet_stream_it_ == packet_stream_.end()) {
- return absl::nullopt;
+ return std::nullopt;
}
if (end_time_ms_ && packet_stream_it_->rtp.log_time_ms() > *end_time_ms_) {
- return absl::nullopt;
+ return std::nullopt;
}
return packet_stream_it_->rtp.log_time_ms();
}
- absl::optional<int64_t> NextOutputEventTime() const override {
+ std::optional<int64_t> NextOutputEventTime() const override {
if (output_events_it_ == output_events_.end()) {
- return absl::nullopt;
+ return std::nullopt;
}
if (end_time_ms_ && output_events_it_->log_time_ms() > *end_time_ms_) {
- return absl::nullopt;
+ return std::nullopt;
}
return output_events_it_->log_time_ms();
}
- absl::optional<SetMinimumDelayInfo> NextSetMinimumDelayInfo() const override {
+ std::optional<SetMinimumDelayInfo> NextSetMinimumDelayInfo() const override {
if (neteq_set_minimum_delay_events_it_ ==
neteq_set_minimum_delay_events_.end()) {
- return absl::nullopt;
+ return std::nullopt;
}
if (end_time_ms_ &&
neteq_set_minimum_delay_events_it_->log_time_ms() > *end_time_ms_) {
- return absl::nullopt;
+ return std::nullopt;
}
return SetMinimumDelayInfo(
neteq_set_minimum_delay_events_it_->log_time_ms(),
@@ -110,9 +110,9 @@
bool ended() const override { return !NextEventTime(); }
- absl::optional<RTPHeader> NextHeader() const override {
+ std::optional<RTPHeader> NextHeader() const override {
if (packet_stream_it_ == packet_stream_.end()) {
- return absl::nullopt;
+ return std::nullopt;
}
return packet_stream_it_->rtp.header;
}
@@ -126,14 +126,14 @@
neteq_set_minimum_delay_events_;
std::vector<LoggedNetEqSetMinimumDelayEvent>::const_iterator
neteq_set_minimum_delay_events_it_;
- const absl::optional<int64_t> end_time_ms_;
+ const std::optional<int64_t> end_time_ms_;
};
} // namespace
std::unique_ptr<NetEqInput> CreateNetEqEventLogInput(
const ParsedRtcEventLog& parsed_log,
- absl::optional<uint32_t> ssrc) {
+ std::optional<uint32_t> ssrc) {
if (parsed_log.incoming_audio_ssrcs().empty()) {
return nullptr;
}
diff --git a/modules/audio_coding/neteq/tools/neteq_event_log_input.h b/modules/audio_coding/neteq/tools/neteq_event_log_input.h
index a84de84..cdfd62d 100644
--- a/modules/audio_coding/neteq/tools/neteq_event_log_input.h
+++ b/modules/audio_coding/neteq/tools/neteq_event_log_input.h
@@ -11,8 +11,9 @@
#ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_EVENT_LOG_INPUT_H_
#define MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_EVENT_LOG_INPUT_H_
+#include <optional>
+
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "logging/rtc_event_log/rtc_event_log_parser.h"
#include "modules/audio_coding/neteq/tools/neteq_input.h"
@@ -21,7 +22,7 @@
std::unique_ptr<NetEqInput> CreateNetEqEventLogInput(
const ParsedRtcEventLog& parsed_log,
- absl::optional<uint32_t> ssrc);
+ std::optional<uint32_t> ssrc);
} // namespace test
} // namespace webrtc
diff --git a/modules/audio_coding/neteq/tools/neteq_input.cc b/modules/audio_coding/neteq/tools/neteq_input.cc
index 57160b9..2bcc59b 100644
--- a/modules/audio_coding/neteq/tools/neteq_input.cc
+++ b/modules/audio_coding/neteq/tools/neteq_input.cc
@@ -50,17 +50,17 @@
TimeLimitedNetEqInput::~TimeLimitedNetEqInput() = default;
-absl::optional<int64_t> TimeLimitedNetEqInput::NextPacketTime() const {
- return ended_ ? absl::nullopt : input_->NextPacketTime();
+std::optional<int64_t> TimeLimitedNetEqInput::NextPacketTime() const {
+ return ended_ ? std::nullopt : input_->NextPacketTime();
}
-absl::optional<int64_t> TimeLimitedNetEqInput::NextOutputEventTime() const {
- return ended_ ? absl::nullopt : input_->NextOutputEventTime();
+std::optional<int64_t> TimeLimitedNetEqInput::NextOutputEventTime() const {
+ return ended_ ? std::nullopt : input_->NextOutputEventTime();
}
-absl::optional<NetEqInput::SetMinimumDelayInfo>
+std::optional<NetEqInput::SetMinimumDelayInfo>
TimeLimitedNetEqInput::NextSetMinimumDelayInfo() const {
- return ended_ ? absl::nullopt : input_->NextSetMinimumDelayInfo();
+ return ended_ ? std::nullopt : input_->NextSetMinimumDelayInfo();
}
std::unique_ptr<NetEqInput::PacketData> TimeLimitedNetEqInput::PopPacket() {
@@ -90,8 +90,8 @@
return ended_ || input_->ended();
}
-absl::optional<RTPHeader> TimeLimitedNetEqInput::NextHeader() const {
- return ended_ ? absl::nullopt : input_->NextHeader();
+std::optional<RTPHeader> TimeLimitedNetEqInput::NextHeader() const {
+ return ended_ ? std::nullopt : input_->NextHeader();
}
void TimeLimitedNetEqInput::MaybeSetEnded() {
diff --git a/modules/audio_coding/neteq/tools/neteq_input.h b/modules/audio_coding/neteq/tools/neteq_input.h
index 56b0212..127cf19 100644
--- a/modules/audio_coding/neteq/tools/neteq_input.h
+++ b/modules/audio_coding/neteq/tools/neteq_input.h
@@ -13,9 +13,9 @@
#include <algorithm>
#include <memory>
+#include <optional>
#include <string>
-#include "absl/types/optional.h"
#include "modules/audio_coding/neteq/tools/packet.h"
#include "modules/audio_coding/neteq/tools/packet_source.h"
#include "rtc_base/buffer.h"
@@ -47,21 +47,21 @@
// Returns at what time (in ms) NetEq::InsertPacket should be called next, or
// empty if the source is out of packets.
- virtual absl::optional<int64_t> NextPacketTime() const = 0;
+ virtual std::optional<int64_t> NextPacketTime() const = 0;
// Returns at what time (in ms) NetEq::GetAudio should be called next, or
// empty if no more output events are available.
- virtual absl::optional<int64_t> NextOutputEventTime() const = 0;
+ virtual std::optional<int64_t> NextOutputEventTime() const = 0;
// Returns the information related to the next NetEq set minimum delay event
// if available.
- virtual absl::optional<SetMinimumDelayInfo> NextSetMinimumDelayInfo()
+ virtual std::optional<SetMinimumDelayInfo> NextSetMinimumDelayInfo()
const = 0;
// Returns the time (in ms) for the next event (packet, output or set minimum
// delay event) or empty if there are no more events.
- absl::optional<int64_t> NextEventTime() const {
- absl::optional<int64_t> next_event_time = NextPacketTime();
+ std::optional<int64_t> NextEventTime() const {
+ std::optional<int64_t> next_event_time = NextPacketTime();
const auto next_output_time = NextOutputEventTime();
// Return the minimum of non-empty `a` and `b`, or empty if both are empty.
if (next_output_time) {
@@ -102,7 +102,7 @@
// Returns the RTP header for the next packet, i.e., the packet that will be
// delivered next by PopPacket().
- virtual absl::optional<RTPHeader> NextHeader() const = 0;
+ virtual std::optional<RTPHeader> NextHeader() const = 0;
};
// Wrapper class to impose a time limit on a NetEqInput object, typically
@@ -112,20 +112,20 @@
public:
TimeLimitedNetEqInput(std::unique_ptr<NetEqInput> input, int64_t duration_ms);
~TimeLimitedNetEqInput() override;
- absl::optional<int64_t> NextPacketTime() const override;
- absl::optional<int64_t> NextOutputEventTime() const override;
- absl::optional<SetMinimumDelayInfo> NextSetMinimumDelayInfo() const override;
+ std::optional<int64_t> NextPacketTime() const override;
+ std::optional<int64_t> NextOutputEventTime() const override;
+ std::optional<SetMinimumDelayInfo> NextSetMinimumDelayInfo() const override;
std::unique_ptr<PacketData> PopPacket() override;
void AdvanceOutputEvent() override;
void AdvanceSetMinimumDelay() override;
bool ended() const override;
- absl::optional<RTPHeader> NextHeader() const override;
+ std::optional<RTPHeader> NextHeader() const override;
private:
void MaybeSetEnded();
std::unique_ptr<NetEqInput> input_;
- const absl::optional<int64_t> start_time_ms_;
+ const std::optional<int64_t> start_time_ms_;
const int64_t duration_ms_;
bool ended_ = false;
};
diff --git a/modules/audio_coding/neteq/tools/neteq_replacement_input.cc b/modules/audio_coding/neteq/tools/neteq_replacement_input.cc
index f2648d7..2fb53d8 100644
--- a/modules/audio_coding/neteq/tools/neteq_replacement_input.cc
+++ b/modules/audio_coding/neteq/tools/neteq_replacement_input.cc
@@ -30,17 +30,17 @@
ReplacePacket();
}
-absl::optional<int64_t> NetEqReplacementInput::NextPacketTime() const {
+std::optional<int64_t> NetEqReplacementInput::NextPacketTime() const {
return packet_
- ? absl::optional<int64_t>(static_cast<int64_t>(packet_->time_ms))
- : absl::nullopt;
+ ? std::optional<int64_t>(static_cast<int64_t>(packet_->time_ms))
+ : std::nullopt;
}
-absl::optional<int64_t> NetEqReplacementInput::NextOutputEventTime() const {
+std::optional<int64_t> NetEqReplacementInput::NextOutputEventTime() const {
return source_->NextOutputEventTime();
}
-absl::optional<NetEqInput::SetMinimumDelayInfo>
+std::optional<NetEqInput::SetMinimumDelayInfo>
NetEqReplacementInput::NextSetMinimumDelayInfo() const {
return source_->NextSetMinimumDelayInfo();
}
@@ -72,7 +72,7 @@
return source_->ended();
}
-absl::optional<RTPHeader> NetEqReplacementInput::NextHeader() const {
+std::optional<RTPHeader> NetEqReplacementInput::NextHeader() const {
return source_->NextHeader();
}
@@ -98,7 +98,7 @@
return;
}
- absl::optional<RTPHeader> next_hdr = source_->NextHeader();
+ std::optional<RTPHeader> next_hdr = source_->NextHeader();
RTC_DCHECK(next_hdr);
uint8_t payload[12];
constexpr uint32_t kMaxFrameSize = 120 * 48;
diff --git a/modules/audio_coding/neteq/tools/neteq_replacement_input.h b/modules/audio_coding/neteq/tools/neteq_replacement_input.h
index 23e4bea..33d8138 100644
--- a/modules/audio_coding/neteq/tools/neteq_replacement_input.h
+++ b/modules/audio_coding/neteq/tools/neteq_replacement_input.h
@@ -28,14 +28,14 @@
const std::set<uint8_t>& comfort_noise_types,
const std::set<uint8_t>& forbidden_types);
- absl::optional<int64_t> NextPacketTime() const override;
- absl::optional<int64_t> NextOutputEventTime() const override;
- absl::optional<SetMinimumDelayInfo> NextSetMinimumDelayInfo() const override;
+ std::optional<int64_t> NextPacketTime() const override;
+ std::optional<int64_t> NextOutputEventTime() const override;
+ std::optional<SetMinimumDelayInfo> NextSetMinimumDelayInfo() const override;
std::unique_ptr<PacketData> PopPacket() override;
void AdvanceOutputEvent() override;
void AdvanceSetMinimumDelay() override;
bool ended() const override;
- absl::optional<RTPHeader> NextHeader() const override;
+ std::optional<RTPHeader> NextHeader() const override;
private:
void ReplacePacket();
diff --git a/modules/audio_coding/neteq/tools/neteq_rtp_dump_input.cc b/modules/audio_coding/neteq/tools/neteq_rtp_dump_input.cc
index 20e092b..b3c29c1 100644
--- a/modules/audio_coding/neteq/tools/neteq_rtp_dump_input.cc
+++ b/modules/audio_coding/neteq/tools/neteq_rtp_dump_input.cc
@@ -22,7 +22,7 @@
public:
NetEqRtpDumpInput(absl::string_view file_name,
const std::map<int, RTPExtensionType>& hdr_ext_map,
- absl::optional<uint32_t> ssrc_filter)
+ std::optional<uint32_t> ssrc_filter)
: source_(RtpFileSource::Create(file_name, ssrc_filter)) {
for (const auto& ext_pair : hdr_ext_map) {
source_->RegisterRtpHeaderExtension(ext_pair.second, ext_pair.first);
@@ -30,12 +30,12 @@
LoadNextPacket();
}
- absl::optional<int64_t> NextOutputEventTime() const override {
+ std::optional<int64_t> NextOutputEventTime() const override {
return next_output_event_ms_;
}
- absl::optional<SetMinimumDelayInfo> NextSetMinimumDelayInfo() const override {
- return absl::nullopt;
+ std::optional<SetMinimumDelayInfo> NextSetMinimumDelayInfo() const override {
+ return std::nullopt;
}
void AdvanceOutputEvent() override {
@@ -43,16 +43,16 @@
*next_output_event_ms_ += kOutputPeriodMs;
}
if (!NextPacketTime()) {
- next_output_event_ms_ = absl::nullopt;
+ next_output_event_ms_ = std::nullopt;
}
}
void AdvanceSetMinimumDelay() override {}
- absl::optional<int64_t> NextPacketTime() const override {
- return packet_ ? absl::optional<int64_t>(
+ std::optional<int64_t> NextPacketTime() const override {
+ return packet_ ? std::optional<int64_t>(
static_cast<int64_t>(packet_->time_ms()))
- : absl::nullopt;
+ : std::nullopt;
}
std::unique_ptr<PacketData> PopPacket() override {
@@ -78,9 +78,8 @@
return packet_data;
}
- absl::optional<RTPHeader> NextHeader() const override {
- return packet_ ? absl::optional<RTPHeader>(packet_->header())
- : absl::nullopt;
+ std::optional<RTPHeader> NextHeader() const override {
+ return packet_ ? std::optional<RTPHeader>(packet_->header()) : std::nullopt;
}
bool ended() const override { return !next_output_event_ms_; }
@@ -88,7 +87,7 @@
private:
void LoadNextPacket() { packet_ = source_->NextPacket(); }
- absl::optional<int64_t> next_output_event_ms_ = 0;
+ std::optional<int64_t> next_output_event_ms_ = 0;
static constexpr int64_t kOutputPeriodMs = 10;
std::unique_ptr<RtpFileSource> source_;
@@ -100,7 +99,7 @@
std::unique_ptr<NetEqInput> CreateNetEqRtpDumpInput(
absl::string_view file_name,
const std::map<int, RTPExtensionType>& hdr_ext_map,
- absl::optional<uint32_t> ssrc_filter) {
+ std::optional<uint32_t> ssrc_filter) {
return std::make_unique<NetEqRtpDumpInput>(file_name, hdr_ext_map,
ssrc_filter);
}
diff --git a/modules/audio_coding/neteq/tools/neteq_rtp_dump_input.h b/modules/audio_coding/neteq/tools/neteq_rtp_dump_input.h
index e68ebb2..da03732 100644
--- a/modules/audio_coding/neteq/tools/neteq_rtp_dump_input.h
+++ b/modules/audio_coding/neteq/tools/neteq_rtp_dump_input.h
@@ -13,9 +13,9 @@
#include <map>
#include <memory>
+#include <optional>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "modules/audio_coding/neteq/tools/neteq_input.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
@@ -25,7 +25,7 @@
std::unique_ptr<NetEqInput> CreateNetEqRtpDumpInput(
absl::string_view file_name,
const std::map<int, RTPExtensionType>& hdr_ext_map,
- absl::optional<uint32_t> ssrc_filter);
+ std::optional<uint32_t> ssrc_filter);
} // namespace test
} // namespace webrtc
diff --git a/modules/audio_coding/neteq/tools/neteq_rtpplay.cc b/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
index b274069..a7db337 100644
--- a/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
+++ b/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
@@ -9,12 +9,12 @@
*/
#include <iostream>
+#include <optional>
#include <string>
#include "absl/flags/flag.h"
#include "absl/flags/parse.h"
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "modules/audio_coding/neteq/tools/neteq_test.h"
#include "modules/audio_coding/neteq/tools/neteq_test_factory.h"
#include "rtc_base/strings/string_builder.h"
@@ -262,13 +262,13 @@
return true;
}
-absl::optional<std::string> CreateOptionalOutputFileName(
+std::optional<std::string> CreateOptionalOutputFileName(
bool output_requested,
absl::string_view basename,
absl::string_view output_audio_filename,
absl::string_view suffix) {
if (!output_requested) {
- return absl::nullopt;
+ return std::nullopt;
}
if (!basename.empty()) {
// Override the automatic assignment.
@@ -283,7 +283,7 @@
return sb.str();
}
std::cout << "Error: invalid text log file parameters.";
- return absl::nullopt;
+ return std::nullopt;
}
} // namespace
@@ -394,7 +394,7 @@
uint32_t ssrc;
RTC_CHECK(ParseSsrc(absl::GetFlag(FLAGS_ssrc), &ssrc))
<< "Flag verification has failed.";
- config.ssrc_filter = absl::make_optional(ssrc);
+ config.ssrc_filter = std::make_optional(ssrc);
}
std::unique_ptr<webrtc::test::NetEqTest> test =
diff --git a/modules/audio_coding/neteq/tools/neteq_test.cc b/modules/audio_coding/neteq/tools/neteq_test.cc
index 4487ac5..76ab544 100644
--- a/modules/audio_coding/neteq/tools/neteq_test.cc
+++ b/modules/audio_coding/neteq/tools/neteq_test.cc
@@ -25,20 +25,20 @@
namespace test {
namespace {
-absl::optional<NetEq::Operation> ActionToOperations(
- absl::optional<NetEqSimulator::Action> a) {
+std::optional<NetEq::Operation> ActionToOperations(
+ std::optional<NetEqSimulator::Action> a) {
if (!a) {
- return absl::nullopt;
+ return std::nullopt;
}
switch (*a) {
case NetEqSimulator::Action::kAccelerate:
- return absl::make_optional(NetEq::Operation::kAccelerate);
+ return std::make_optional(NetEq::Operation::kAccelerate);
case NetEqSimulator::Action::kExpand:
- return absl::make_optional(NetEq::Operation::kExpand);
+ return std::make_optional(NetEq::Operation::kExpand);
case NetEqSimulator::Action::kNormal:
- return absl::make_optional(NetEq::Operation::kNormal);
+ return std::make_optional(NetEq::Operation::kNormal);
case NetEqSimulator::Action::kPreemptiveExpand:
- return absl::make_optional(NetEq::Operation::kPreemptiveExpand);
+ return std::make_optional(NetEq::Operation::kPreemptiveExpand);
}
}
@@ -166,9 +166,9 @@
<< ", buffer size: " << std::setw(4)
<< ops_state.current_buffer_size_ms << std::endl;
}
- last_packet_time_ms_ = absl::make_optional<int>(time_now_ms);
+ last_packet_time_ms_ = std::make_optional<int>(time_now_ms);
last_packet_timestamp_ =
- absl::make_optional<uint32_t>(packet_data->header.timestamp);
+ std::make_optional<uint32_t>(packet_data->header.timestamp);
}
if (input_->NextSetMinimumDelayInfo().has_value() &&
@@ -187,7 +187,7 @@
AudioFrame out_frame;
int error = neteq_->GetAudio(&out_frame, nullptr, nullptr,
ActionToOperations(next_action_));
- next_action_ = absl::nullopt;
+ next_action_ = std::nullopt;
if (error != NetEq::kOK) {
if (callbacks_.error_callback) {
callbacks_.error_callback->OnGetAudioError();
@@ -292,7 +292,7 @@
}
void NetEqTest::SetNextAction(NetEqTest::Action next_operation) {
- next_action_ = absl::optional<Action>(next_operation);
+ next_action_ = std::optional<Action>(next_operation);
}
NetEqTest::NetEqState NetEqTest::GetNetEqState() {
diff --git a/modules/audio_coding/neteq/tools/neteq_test.h b/modules/audio_coding/neteq/tools/neteq_test.h
index 89506a8..5f826e2 100644
--- a/modules/audio_coding/neteq/tools/neteq_test.h
+++ b/modules/audio_coding/neteq/tools/neteq_test.h
@@ -14,10 +14,10 @@
#include <fstream>
#include <map>
#include <memory>
+#include <optional>
#include <string>
#include <utility>
-#include "absl/types/optional.h"
#include "api/audio_codecs/audio_decoder_factory.h"
#include "api/environment/environment.h"
#include "api/neteq/neteq.h"
@@ -115,8 +115,8 @@
std::unique_ptr<NetEqInput> input_;
SimulatedClock clock_;
const Environment env_;
- absl::optional<Action> next_action_;
- absl::optional<int> last_packet_time_ms_;
+ std::optional<Action> next_action_;
+ std::optional<int> last_packet_time_ms_;
std::unique_ptr<NetEq> neteq_;
std::unique_ptr<AudioSink> output_;
Callbacks callbacks_;
@@ -124,7 +124,7 @@
NetEqState current_state_;
NetEqOperationsAndState prev_ops_state_;
NetEqLifetimeStatistics prev_lifetime_stats_;
- absl::optional<uint32_t> last_packet_timestamp_;
+ std::optional<uint32_t> last_packet_timestamp_;
std::unique_ptr<std::ofstream> text_log_;
};
diff --git a/modules/audio_coding/neteq/tools/neteq_test_factory.cc b/modules/audio_coding/neteq/tools/neteq_test_factory.cc
index 766a092..1f79c44 100644
--- a/modules/audio_coding/neteq/tools/neteq_test_factory.cc
+++ b/modules/audio_coding/neteq/tools/neteq_test_factory.cc
@@ -47,7 +47,7 @@
namespace test {
namespace {
-absl::optional<int> CodecSampleRate(
+std::optional<int> CodecSampleRate(
uint8_t payload_type,
webrtc::test::NetEqTestFactory::Config config) {
if (payload_type == config.pcmu || payload_type == config.pcma ||
@@ -66,7 +66,7 @@
return 48000;
if (payload_type == config.red)
return 0;
- return absl::nullopt;
+ return std::nullopt;
}
} // namespace
@@ -98,7 +98,7 @@
private:
NetEqPostInsertPacket* other_callback_;
- absl::optional<uint32_t> last_ssrc_;
+ std::optional<uint32_t> last_ssrc_;
};
NetEqTestFactory::NetEqTestFactory() = default;
@@ -197,9 +197,9 @@
}
// Check the sample rate.
- absl::optional<int> sample_rate_hz;
+ std::optional<int> sample_rate_hz;
std::set<std::pair<int, uint32_t>> discarded_pt_and_ssrc;
- while (absl::optional<RTPHeader> first_rtp_header = input->NextHeader()) {
+ while (std::optional<RTPHeader> first_rtp_header = input->NextHeader()) {
RTC_DCHECK(first_rtp_header);
sample_rate_hz = CodecSampleRate(first_rtp_header->payloadType, config);
if (sample_rate_hz) {
@@ -295,7 +295,7 @@
decoder_factory = rtc::make_ref_counted<FunctionAudioDecoderFactory>(
[decoder_factory, config](
const Environment& env, const SdpAudioFormat& format,
- absl::optional<AudioCodecPairId> codec_pair_id) {
+ std::optional<AudioCodecPairId> codec_pair_id) {
std::unique_ptr<AudioDecoder> decoder =
decoder_factory->Create(env, format, codec_pair_id);
if (!decoder && format.name == "replacement") {
diff --git a/modules/audio_coding/neteq/tools/neteq_test_factory.h b/modules/audio_coding/neteq/tools/neteq_test_factory.h
index 6091749..3a34171 100644
--- a/modules/audio_coding/neteq/tools/neteq_test_factory.h
+++ b/modules/audio_coding/neteq/tools/neteq_test_factory.h
@@ -12,10 +12,10 @@
#define MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_TEST_FACTORY_H_
#include <memory>
+#include <optional>
#include <string>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "modules/audio_coding/neteq/tools/neteq_test.h"
namespace webrtc {
@@ -98,7 +98,7 @@
// A PCM file that will be used to populate dummy RTP packets.
std::string replacement_audio_file;
// Only use packets with this SSRC.
- absl::optional<uint32_t> ssrc_filter;
+ std::optional<uint32_t> ssrc_filter;
// Extension ID for audio level (RFC 6464).
static constexpr int default_audio_level() { return 1; }
int audio_level = default_audio_level();
@@ -136,11 +136,11 @@
bool textlog = false;
// If specified and `textlog` is true, the output of `textlog` is written to
// the specified file name.
- absl::optional<std::string> textlog_filename;
+ std::optional<std::string> textlog_filename;
// Base name for the output script files for plotting the delay profile.
- absl::optional<std::string> plot_scripts_basename;
+ std::optional<std::string> plot_scripts_basename;
// Path to the output audio file.
- absl::optional<std::string> output_audio_filename;
+ std::optional<std::string> output_audio_filename;
// Field trials to use during the simulation.
std::string field_trial_string;
};
diff --git a/modules/audio_coding/neteq/tools/rtp_file_source.cc b/modules/audio_coding/neteq/tools/rtp_file_source.cc
index 7a8daef..056b30d 100644
--- a/modules/audio_coding/neteq/tools/rtp_file_source.cc
+++ b/modules/audio_coding/neteq/tools/rtp_file_source.cc
@@ -27,7 +27,7 @@
namespace test {
RtpFileSource* RtpFileSource::Create(absl::string_view file_name,
- absl::optional<uint32_t> ssrc_filter) {
+ std::optional<uint32_t> ssrc_filter) {
RtpFileSource* source = new RtpFileSource(ssrc_filter);
RTC_CHECK(source->OpenFile(file_name));
return source;
@@ -79,7 +79,7 @@
}
}
-RtpFileSource::RtpFileSource(absl::optional<uint32_t> ssrc_filter)
+RtpFileSource::RtpFileSource(std::optional<uint32_t> ssrc_filter)
: PacketSource(), ssrc_filter_(ssrc_filter) {}
bool RtpFileSource::OpenFile(absl::string_view file_name) {
diff --git a/modules/audio_coding/neteq/tools/rtp_file_source.h b/modules/audio_coding/neteq/tools/rtp_file_source.h
index 55505be..300981a 100644
--- a/modules/audio_coding/neteq/tools/rtp_file_source.h
+++ b/modules/audio_coding/neteq/tools/rtp_file_source.h
@@ -14,10 +14,10 @@
#include <stdio.h>
#include <memory>
+#include <optional>
#include <string>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "modules/audio_coding/neteq/tools/packet_source.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
@@ -33,7 +33,7 @@
// opened, or has the wrong format, NULL will be returned.
static RtpFileSource* Create(
absl::string_view file_name,
- absl::optional<uint32_t> ssrc_filter = absl::nullopt);
+ std::optional<uint32_t> ssrc_filter = std::nullopt);
// Checks whether a files is a valid RTP dump or PCAP (Wireshark) file.
static bool ValidRtpDump(absl::string_view file_name);
@@ -54,12 +54,12 @@
static const int kRtpFileHeaderSize = 4 + 4 + 4 + 2 + 2;
static const size_t kPacketHeaderSize = 8;
- explicit RtpFileSource(absl::optional<uint32_t> ssrc_filter);
+ explicit RtpFileSource(std::optional<uint32_t> ssrc_filter);
bool OpenFile(absl::string_view file_name);
std::unique_ptr<RtpFileReader> rtp_reader_;
- const absl::optional<uint32_t> ssrc_filter_;
+ const std::optional<uint32_t> ssrc_filter_;
RtpHeaderExtensionMap rtp_header_extension_map_;
};
diff --git a/modules/audio_coding/neteq/underrun_optimizer.cc b/modules/audio_coding/neteq/underrun_optimizer.cc
index baed812..d1f12e9 100644
--- a/modules/audio_coding/neteq/underrun_optimizer.cc
+++ b/modules/audio_coding/neteq/underrun_optimizer.cc
@@ -24,15 +24,15 @@
UnderrunOptimizer::UnderrunOptimizer(const TickTimer* tick_timer,
int histogram_quantile,
int forget_factor,
- absl::optional<int> start_forget_weight,
- absl::optional<int> resample_interval_ms)
+ std::optional<int> start_forget_weight,
+ std::optional<int> resample_interval_ms)
: tick_timer_(tick_timer),
histogram_(kDelayBuckets, forget_factor, start_forget_weight),
histogram_quantile_(histogram_quantile),
resample_interval_ms_(resample_interval_ms) {}
void UnderrunOptimizer::Update(int relative_delay_ms) {
- absl::optional<int> histogram_update;
+ std::optional<int> histogram_update;
if (resample_interval_ms_) {
if (!resample_stopwatch_) {
resample_stopwatch_ = tick_timer_->GetNewStopwatch();
diff --git a/modules/audio_coding/neteq/underrun_optimizer.h b/modules/audio_coding/neteq/underrun_optimizer.h
index b37ce18..20a46ff 100644
--- a/modules/audio_coding/neteq/underrun_optimizer.h
+++ b/modules/audio_coding/neteq/underrun_optimizer.h
@@ -12,8 +12,8 @@
#define MODULES_AUDIO_CODING_NETEQ_UNDERRUN_OPTIMIZER_H_
#include <memory>
+#include <optional>
-#include "absl/types/optional.h"
#include "api/neteq/tick_timer.h"
#include "modules/audio_coding/neteq/histogram.h"
@@ -27,12 +27,12 @@
UnderrunOptimizer(const TickTimer* tick_timer,
int histogram_quantile,
int forget_factor,
- absl::optional<int> start_forget_weight,
- absl::optional<int> resample_interval_ms);
+ std::optional<int> start_forget_weight,
+ std::optional<int> resample_interval_ms);
void Update(int relative_delay_ms);
- absl::optional<int> GetOptimalDelayMs() const { return optimal_delay_ms_; }
+ std::optional<int> GetOptimalDelayMs() const { return optimal_delay_ms_; }
void Reset();
@@ -40,10 +40,10 @@
const TickTimer* tick_timer_;
Histogram histogram_;
const int histogram_quantile_; // In Q30.
- const absl::optional<int> resample_interval_ms_;
+ const std::optional<int> resample_interval_ms_;
std::unique_ptr<TickTimer::Stopwatch> resample_stopwatch_;
int max_delay_in_interval_ms_ = 0;
- absl::optional<int> optimal_delay_ms_;
+ std::optional<int> optimal_delay_ms_;
};
} // namespace webrtc
diff --git a/modules/audio_coding/neteq/underrun_optimizer_unittest.cc b/modules/audio_coding/neteq/underrun_optimizer_unittest.cc
index a86e9cf..0988e03 100644
--- a/modules/audio_coding/neteq/underrun_optimizer_unittest.cc
+++ b/modules/audio_coding/neteq/underrun_optimizer_unittest.cc
@@ -25,7 +25,7 @@
TickTimer tick_timer;
constexpr int kResampleIntervalMs = 500;
UnderrunOptimizer underrun_optimizer(&tick_timer, kDefaultHistogramQuantile,
- kForgetFactor, absl::nullopt,
+ kForgetFactor, std::nullopt,
kResampleIntervalMs);
// The histogram should be updated once with the maximum delay observed for
diff --git a/modules/audio_coding/test/PCMFile.h b/modules/audio_coding/test/PCMFile.h
index 5320aa6..64af5df 100644
--- a/modules/audio_coding/test/PCMFile.h
+++ b/modules/audio_coding/test/PCMFile.h
@@ -14,10 +14,10 @@
#include <stdio.h>
#include <stdlib.h>
+#include <optional>
#include <string>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/audio/audio_frame.h"
namespace webrtc {
@@ -68,7 +68,7 @@
uint32_t timestamp_;
bool read_stereo_;
bool save_stereo_;
- absl::optional<int> num_10ms_blocks_to_read_;
+ std::optional<int> num_10ms_blocks_to_read_;
int blocks_read_ = 0;
};
diff --git a/modules/audio_coding/test/TestRedFec.cc b/modules/audio_coding/test/TestRedFec.cc
index c2b42cc..b4d138b 100644
--- a/modules/audio_coding/test/TestRedFec.cc
+++ b/modules/audio_coding/test/TestRedFec.cc
@@ -103,7 +103,7 @@
_outFileB.Close();
#endif
- RegisterSendCodec(_acmA, {"opus", 48000, 2}, absl::nullopt, false);
+ RegisterSendCodec(_acmA, {"opus", 48000, 2}, std::nullopt, false);
// _channelA2B imposes 25% packet loss rate.
EXPECT_EQ(0, _acmA->SetPacketLossRate(25));
@@ -116,11 +116,11 @@
Run();
// Switch to L16 with RED.
- RegisterSendCodec(_acmA, {"L16", 8000, 1}, absl::nullopt, true);
+ RegisterSendCodec(_acmA, {"L16", 8000, 1}, std::nullopt, true);
Run();
// Switch to Opus again.
- RegisterSendCodec(_acmA, {"opus", 48000, 2}, absl::nullopt, false);
+ RegisterSendCodec(_acmA, {"opus", 48000, 2}, std::nullopt, false);
_acmA->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* enc) {
EXPECT_EQ(true, (*enc)->SetFec(false));
});
@@ -135,7 +135,7 @@
void TestRedFec::RegisterSendCodec(
const std::unique_ptr<AudioCodingModule>& acm,
const SdpAudioFormat& codec_format,
- absl::optional<Vad::Aggressiveness> vad_mode,
+ std::optional<Vad::Aggressiveness> vad_mode,
bool use_red) {
constexpr int payload_type = 17, cn_payload_type = 27, red_payload_type = 37;
diff --git a/modules/audio_coding/test/TestRedFec.h b/modules/audio_coding/test/TestRedFec.h
index 8c43c1d..56a6495 100644
--- a/modules/audio_coding/test/TestRedFec.h
+++ b/modules/audio_coding/test/TestRedFec.h
@@ -35,7 +35,7 @@
private:
void RegisterSendCodec(const std::unique_ptr<AudioCodingModule>& acm,
const SdpAudioFormat& codec_format,
- absl::optional<Vad::Aggressiveness> vad_mode,
+ std::optional<Vad::Aggressiveness> vad_mode,
bool use_red);
void Run();
void OpenOutFile(int16_t testNumber);
diff --git a/modules/audio_coding/test/TestVADDTX.cc b/modules/audio_coding/test/TestVADDTX.cc
index 50781a7..ac87ae1 100644
--- a/modules/audio_coding/test/TestVADDTX.cc
+++ b/modules/audio_coding/test/TestVADDTX.cc
@@ -86,7 +86,7 @@
}
bool TestVadDtx::RegisterCodec(const SdpAudioFormat& codec_format,
- absl::optional<Vad::Aggressiveness> vad_mode) {
+ std::optional<Vad::Aggressiveness> vad_mode) {
constexpr int payload_type = 17, cn_payload_type = 117;
bool added_comfort_noise = false;
@@ -190,7 +190,7 @@
// Test various configurations on VAD/DTX.
void TestWebRtcVadDtx::RunTestCases(const SdpAudioFormat& codec_format) {
Test(/*new_outfile=*/true,
- /*expect_dtx_enabled=*/RegisterCodec(codec_format, absl::nullopt));
+ /*expect_dtx_enabled=*/RegisterCodec(codec_format, std::nullopt));
Test(/*new_outfile=*/false,
/*expect_dtx_enabled=*/RegisterCodec(codec_format, Vad::kVadAggressive));
@@ -225,7 +225,7 @@
// Register Opus as send codec
std::string out_filename =
webrtc::test::OutputPath() + "testOpusDtx_outFile_mono.pcm";
- RegisterCodec({"opus", 48000, 2}, absl::nullopt);
+ RegisterCodec({"opus", 48000, 2}, std::nullopt);
acm_send_->ModifyEncoder([](std::unique_ptr<AudioEncoder>* encoder_ptr) {
(*encoder_ptr)->SetDtx(false);
});
diff --git a/modules/audio_coding/test/TestVADDTX.h b/modules/audio_coding/test/TestVADDTX.h
index 427c94c..b7f90a3 100644
--- a/modules/audio_coding/test/TestVADDTX.h
+++ b/modules/audio_coding/test/TestVADDTX.h
@@ -64,7 +64,7 @@
protected:
// Returns true iff CN was added.
bool RegisterCodec(const SdpAudioFormat& codec_format,
- absl::optional<Vad::Aggressiveness> vad_mode);
+ std::optional<Vad::Aggressiveness> vad_mode);
// Encoding a file and see if the numbers that various packets occur follow
// the expectation. Saves result to a file.
diff --git a/modules/audio_device/BUILD.gn b/modules/audio_device/BUILD.gn
index d4eee0b..b69ada7 100644
--- a/modules/audio_device/BUILD.gn
+++ b/modules/audio_device/BUILD.gn
@@ -78,7 +78,6 @@
"../../rtc_base/synchronization:mutex",
"../../system_wrappers",
"../../system_wrappers:metrics",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -166,7 +165,6 @@
"../../rtc_base/win:scoped_com_initializer",
"../../rtc_base/win:windows_version",
"//third_party/abseil-cpp/absl/strings:strings",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
}
@@ -204,7 +202,6 @@
"../../rtc_base/synchronization:mutex",
"../../rtc_base/task_utils:repeating_task",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
}
@@ -437,7 +434,6 @@
"../../api:make_ref_counted",
"../../api/audio:audio_device",
"../../test:test_support",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -479,7 +475,6 @@
"../../test:fileutils",
"../../test:test_support",
"../../test/time_controller",
- "//third_party/abseil-cpp/absl/types:optional",
]
if (is_linux || is_chromeos || is_mac || is_win) {
sources += [ "audio_device_unittest.cc" ]
diff --git a/modules/audio_device/audio_device_buffer.cc b/modules/audio_device/audio_device_buffer.cc
index 38ed633..95096da 100644
--- a/modules/audio_device/audio_device_buffer.cc
+++ b/modules/audio_device/audio_device_buffer.cc
@@ -245,13 +245,13 @@
int32_t AudioDeviceBuffer::SetRecordedBuffer(const void* audio_buffer,
size_t samples_per_channel) {
- return SetRecordedBuffer(audio_buffer, samples_per_channel, absl::nullopt);
+ return SetRecordedBuffer(audio_buffer, samples_per_channel, std::nullopt);
}
int32_t AudioDeviceBuffer::SetRecordedBuffer(
const void* audio_buffer,
size_t samples_per_channel,
- absl::optional<int64_t> capture_timestamp_ns) {
+ std::optional<int64_t> capture_timestamp_ns) {
// Copy the complete input buffer to the local buffer.
const size_t old_size = rec_buffer_.size();
rec_buffer_.SetData(static_cast<const int16_t*>(audio_buffer),
diff --git a/modules/audio_device/audio_device_buffer.h b/modules/audio_device/audio_device_buffer.h
index e9a90ab..99a4cb7 100644
--- a/modules/audio_device/audio_device_buffer.h
+++ b/modules/audio_device/audio_device_buffer.h
@@ -110,7 +110,7 @@
virtual int32_t SetRecordedBuffer(
const void* audio_buffer,
size_t samples_per_channel,
- absl::optional<int64_t> capture_timestamp_ns);
+ std::optional<int64_t> capture_timestamp_ns);
virtual void SetVQEData(int play_delay_ms, int rec_delay_ms);
virtual int32_t DeliverRecordedData();
uint32_t NewMicLevel() const;
@@ -200,11 +200,11 @@
int rec_delay_ms_;
// Capture timestamp.
- absl::optional<int64_t> capture_timestamp_ns_;
+ std::optional<int64_t> capture_timestamp_ns_;
// The last time the Timestamp Aligner was used to estimate clock offset
// between system clock and capture time from audio.
// This is used to prevent estimating the clock offset too often.
- absl::optional<int64_t> align_offsync_estimation_time_;
+ std::optional<int64_t> align_offsync_estimation_time_;
// Counts number of times LogStats() has been called.
size_t num_stat_reports_ RTC_GUARDED_BY(task_queue_);
diff --git a/modules/audio_device/audio_device_data_observer.cc b/modules/audio_device/audio_device_data_observer.cc
index 9643f75..6f4a177 100644
--- a/modules/audio_device/audio_device_data_observer.cc
+++ b/modules/audio_device/audio_device_data_observer.cc
@@ -58,7 +58,7 @@
return RecordedDataIsAvailable(
audioSamples, nSamples, nBytesPerSample, nChannels, samples_per_sec,
total_delay_ms, clockDrift, currentMicLevel, keyPressed, newMicLevel,
- /*capture_timestamp_ns=*/absl::nullopt);
+ /*capture_timestamp_ns=*/std::nullopt);
}
// AudioTransport methods overrides.
@@ -73,7 +73,7 @@
uint32_t currentMicLevel,
bool keyPressed,
uint32_t& newMicLevel,
- absl::optional<int64_t> capture_timestamp_ns) override {
+ std::optional<int64_t> capture_timestamp_ns) override {
int32_t res = 0;
// Capture PCM data of locally captured audio.
if (observer_) {
diff --git a/modules/audio_device/audio_device_unittest.cc b/modules/audio_device/audio_device_unittest.cc
index 8275f21..8090cc5 100644
--- a/modules/audio_device/audio_device_unittest.cc
+++ b/modules/audio_device/audio_device_unittest.cc
@@ -15,8 +15,8 @@
#include <list>
#include <memory>
#include <numeric>
+#include <optional>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/scoped_refptr.h"
#include "api/sequence_checker.h"
@@ -320,7 +320,7 @@
SequenceChecker read_thread_checker_;
SequenceChecker write_thread_checker_;
- absl::optional<int64_t> pulse_time_ RTC_GUARDED_BY(lock_);
+ std::optional<int64_t> pulse_time_ RTC_GUARDED_BY(lock_);
std::vector<int> latencies_ RTC_GUARDED_BY(race_checker_);
size_t read_count_ RTC_GUARDED_BY(read_thread_checker_) = 0;
size_t write_count_ RTC_GUARDED_BY(write_thread_checker_) = 0;
diff --git a/modules/audio_device/fine_audio_buffer.cc b/modules/audio_device/fine_audio_buffer.cc
index f483b8d..c872988 100644
--- a/modules/audio_device/fine_audio_buffer.cc
+++ b/modules/audio_device/fine_audio_buffer.cc
@@ -109,7 +109,7 @@
void FineAudioBuffer::DeliverRecordedData(
rtc::ArrayView<const int16_t> audio_buffer,
int record_delay_ms,
- absl::optional<int64_t> capture_time_ns) {
+ std::optional<int64_t> capture_time_ns) {
RTC_DCHECK(IsReadyForRecord());
// Always append new data and grow the buffer when needed.
record_buffer_.AppendData(audio_buffer.data(), audio_buffer.size());
diff --git a/modules/audio_device/fine_audio_buffer.h b/modules/audio_device/fine_audio_buffer.h
index 7af41d3..cd5dedd 100644
--- a/modules/audio_device/fine_audio_buffer.h
+++ b/modules/audio_device/fine_audio_buffer.h
@@ -13,8 +13,8 @@
#include <cstddef>
#include <cstdint>
+#include <optional>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "rtc_base/buffer.h"
@@ -66,11 +66,11 @@
// cache. Call #3 restarts the scheme above.
void DeliverRecordedData(rtc::ArrayView<const int16_t> audio_buffer,
int record_delay_ms) {
- DeliverRecordedData(audio_buffer, record_delay_ms, absl::nullopt);
+ DeliverRecordedData(audio_buffer, record_delay_ms, std::nullopt);
}
void DeliverRecordedData(rtc::ArrayView<const int16_t> audio_buffer,
int record_delay_ms,
- absl::optional<int64_t> capture_time_ns);
+ std::optional<int64_t> capture_time_ns);
private:
// Device buffer that works with 10ms chunks of data both for playout and
diff --git a/modules/audio_device/include/mock_audio_transport.h b/modules/audio_device/include/mock_audio_transport.h
index cde0e53..56f6905 100644
--- a/modules/audio_device/include/mock_audio_transport.h
+++ b/modules/audio_device/include/mock_audio_transport.h
@@ -48,7 +48,7 @@
uint32_t currentMicLevel,
bool keyPressed,
uint32_t& newMicLevel,
- absl::optional<int64_t> estimated_capture_time_ns),
+ std::optional<int64_t> estimated_capture_time_ns),
(override));
MOCK_METHOD(int32_t,
diff --git a/modules/audio_device/include/test_audio_device_unittest.cc b/modules/audio_device/include/test_audio_device_unittest.cc
index 165a4fb..a0eb240 100644
--- a/modules/audio_device/include/test_audio_device_unittest.cc
+++ b/modules/audio_device/include/test_audio_device_unittest.cc
@@ -13,9 +13,9 @@
#include <algorithm>
#include <array>
#include <memory>
+#include <optional>
#include <utility>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/audio/audio_device_defines.h"
#include "api/task_queue/task_queue_factory.h"
@@ -366,7 +366,7 @@
uint32_t current_mic_level,
bool key_pressed,
uint32_t& new_mic_level,
- absl::optional<int64_t> estimated_capture_time_ns) override {
+ std::optional<int64_t> estimated_capture_time_ns) override {
new_mic_level = 1;
if (mode_ != Mode::kRecording) {
diff --git a/modules/audio_device/mock_audio_device_buffer.h b/modules/audio_device/mock_audio_device_buffer.h
index 0b27618..d9c9a48 100644
--- a/modules/audio_device/mock_audio_device_buffer.h
+++ b/modules/audio_device/mock_audio_device_buffer.h
@@ -11,7 +11,8 @@
#ifndef MODULES_AUDIO_DEVICE_MOCK_AUDIO_DEVICE_BUFFER_H_
#define MODULES_AUDIO_DEVICE_MOCK_AUDIO_DEVICE_BUFFER_H_
-#include "absl/types/optional.h"
+#include <optional>
+
#include "modules/audio_device/audio_device_buffer.h"
#include "test/gmock.h"
@@ -27,7 +28,7 @@
SetRecordedBuffer,
(const void* audioBuffer,
size_t nSamples,
- absl::optional<int64_t> capture_time_ns),
+ std::optional<int64_t> capture_time_ns),
(override));
MOCK_METHOD(void, SetVQEData, (int playDelayMS, int recDelayMS), (override));
MOCK_METHOD(int32_t, DeliverRecordedData, (), (override));
diff --git a/modules/audio_device/test_audio_device_impl.cc b/modules/audio_device/test_audio_device_impl.cc
index a3742ea..914a04d 100644
--- a/modules/audio_device/test_audio_device_impl.cc
+++ b/modules/audio_device/test_audio_device_impl.cc
@@ -10,9 +10,9 @@
#include "modules/audio_device/test_audio_device_impl.h"
#include <memory>
+#include <optional>
#include <utility>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/task_queue/task_queue_factory.h"
#include "api/units/time_delta.h"
@@ -169,7 +169,7 @@
audio_buffer_->SetRecordedBuffer(
recording_buffer_.data(),
recording_buffer_.size() / capturer_->NumChannels(),
- absl::make_optional(rtc::TimeNanos()));
+ std::make_optional(rtc::TimeNanos()));
audio_buffer_->DeliverRecordedData();
}
if (!keep_capturing) {
diff --git a/modules/audio_device/test_audio_device_impl_test.cc b/modules/audio_device/test_audio_device_impl_test.cc
index ad15ad5..6b6b0a6 100644
--- a/modules/audio_device/test_audio_device_impl_test.cc
+++ b/modules/audio_device/test_audio_device_impl_test.cc
@@ -10,9 +10,9 @@
#include "modules/audio_device/test_audio_device_impl.h"
#include <memory>
+#include <optional>
#include <utility>
-#include "absl/types/optional.h"
#include "api/audio/audio_device.h"
#include "api/audio/audio_device_defines.h"
#include "api/task_queue/task_queue_factory.h"
@@ -52,7 +52,7 @@
uint32_t current_mic_level,
bool key_pressed,
uint32_t& new_mic_level,
- absl::optional<int64_t> estimated_capture_time_ns) override {
+ std::optional<int64_t> estimated_capture_time_ns) override {
new_mic_level = 1;
if (mode_ != Mode::kRecording) {
diff --git a/modules/audio_device/win/core_audio_base_win.h b/modules/audio_device/win/core_audio_base_win.h
index 6c1357e..2d42dfc 100644
--- a/modules/audio_device/win/core_audio_base_win.h
+++ b/modules/audio_device/win/core_audio_base_win.h
@@ -14,10 +14,10 @@
#include <atomic>
#include <functional>
#include <memory>
+#include <optional>
#include <string>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/sequence_checker.h"
#include "modules/audio_device/win/core_audio_utility_win.h"
#include "rtc_base/platform_thread.h"
@@ -140,7 +140,7 @@
bool is_active_ = false;
int64_t num_data_callbacks_ = 0;
int latency_ms_ = 0;
- absl::optional<uint32_t> sample_rate_;
+ std::optional<uint32_t> sample_rate_;
private:
const Direction direction_;
diff --git a/modules/audio_device/win/core_audio_input_win.cc b/modules/audio_device/win/core_audio_input_win.cc
index 17790da..f70330e 100644
--- a/modules/audio_device/win/core_audio_input_win.cc
+++ b/modules/audio_device/win/core_audio_input_win.cc
@@ -317,7 +317,7 @@
// Update input delay estimate but only about once per second to save
// resources. The estimate is usually stable.
if (num_data_callbacks_ % 100 == 0) {
- absl::optional<int> opt_record_delay_ms;
+ std::optional<int> opt_record_delay_ms;
// TODO(henrika): note that FineAudioBuffer adds latency as well.
opt_record_delay_ms = EstimateLatencyMillis(capture_time_100ns);
if (opt_record_delay_ms) {
@@ -392,10 +392,10 @@
return true;
}
-absl::optional<int> CoreAudioInput::EstimateLatencyMillis(
+std::optional<int> CoreAudioInput::EstimateLatencyMillis(
uint64_t capture_time_100ns) {
if (!qpc_to_100ns_) {
- return absl::nullopt;
+ return std::nullopt;
}
// Input parameter `capture_time_100ns` contains the performance counter at
// the time that the audio endpoint device recorded the device position of
@@ -406,7 +406,7 @@
// - subtracting `capture_time_100ns` from now_time_100ns.
LARGE_INTEGER perf_counter_now = {};
if (!::QueryPerformanceCounter(&perf_counter_now)) {
- return absl::nullopt;
+ return std::nullopt;
}
uint64_t qpc_now_raw = perf_counter_now.QuadPart;
uint64_t now_time_100ns = qpc_now_raw * (*qpc_to_100ns_);
diff --git a/modules/audio_device/win/core_audio_input_win.h b/modules/audio_device/win/core_audio_input_win.h
index be290f9..ecf536a 100644
--- a/modules/audio_device/win/core_audio_input_win.h
+++ b/modules/audio_device/win/core_audio_input_win.h
@@ -12,9 +12,9 @@
#define MODULES_AUDIO_DEVICE_WIN_CORE_AUDIO_INPUT_WIN_H_
#include <memory>
+#include <optional>
#include <string>
-#include "absl/types/optional.h"
#include "modules/audio_device/win/audio_device_module_win.h"
#include "modules/audio_device/win/core_audio_base_win.h"
@@ -58,12 +58,12 @@
void ReleaseCOMObjects();
bool OnDataCallback(uint64_t device_frequency);
bool OnErrorCallback(ErrorType error);
- absl::optional<int> EstimateLatencyMillis(uint64_t capture_time_100ns);
+ std::optional<int> EstimateLatencyMillis(uint64_t capture_time_100ns);
bool HandleStreamDisconnected();
std::unique_ptr<FineAudioBuffer> fine_audio_buffer_;
Microsoft::WRL::ComPtr<IAudioCaptureClient> audio_capture_client_;
- absl::optional<double> qpc_to_100ns_;
+ std::optional<double> qpc_to_100ns_;
};
} // namespace webrtc_win
diff --git a/modules/audio_mixer/BUILD.gn b/modules/audio_mixer/BUILD.gn
index bb6d080..38c4e95 100644
--- a/modules/audio_mixer/BUILD.gn
+++ b/modules/audio_mixer/BUILD.gn
@@ -124,7 +124,6 @@
"../../rtc_base:task_queue_for_test",
"../../system_wrappers:metrics",
"../../test:test_support",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
diff --git a/modules/audio_mixer/audio_mixer_impl_unittest.cc b/modules/audio_mixer/audio_mixer_impl_unittest.cc
index b04b706..00e2f7a 100644
--- a/modules/audio_mixer/audio_mixer_impl_unittest.cc
+++ b/modules/audio_mixer/audio_mixer_impl_unittest.cc
@@ -15,11 +15,11 @@
#include <cstdint>
#include <limits>
#include <memory>
+#include <optional>
#include <string>
#include <utility>
#include <vector>
-#include "absl/types/optional.h"
#include "api/audio/audio_mixer.h"
#include "api/rtp_packet_info.h"
#include "api/rtp_packet_infos.h"
@@ -448,7 +448,7 @@
const uint32_t kCsrc3 = 23;
const int kAudioLevel0 = 10;
const int kAudioLevel1 = 40;
- const absl::optional<uint32_t> kAudioLevel2 = absl::nullopt;
+ const std::optional<uint32_t> kAudioLevel2 = std::nullopt;
const uint32_t kRtpTimestamp0 = 300;
const uint32_t kRtpTimestamp1 = 400;
const Timestamp kReceiveTime0 = Timestamp::Millis(10);
diff --git a/modules/audio_mixer/frame_combiner_unittest.cc b/modules/audio_mixer/frame_combiner_unittest.cc
index 80c2b99..4218f4e 100644
--- a/modules/audio_mixer/frame_combiner_unittest.cc
+++ b/modules/audio_mixer/frame_combiner_unittest.cc
@@ -13,11 +13,11 @@
#include <cstdint>
#include <initializer_list>
#include <numeric>
+#include <optional>
#include <string>
#include <type_traits>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/rtp_packet_info.h"
#include "api/rtp_packet_infos.h"
diff --git a/modules/audio_processing/BUILD.gn b/modules/audio_processing/BUILD.gn
index c403a95..1f4bc7e 100644
--- a/modules/audio_processing/BUILD.gn
+++ b/modules/audio_processing/BUILD.gn
@@ -93,7 +93,6 @@
":audio_frame_view",
"../../api/audio:audio_processing",
"//third_party/abseil-cpp/absl/base:core_headers",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -194,7 +193,6 @@
"//third_party/abseil-cpp/absl/base:nullability",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
deps += [
@@ -233,7 +231,6 @@
"../../rtc_base:checks",
"../../rtc_base:logging",
"../../system_wrappers:metrics",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -246,7 +243,6 @@
deps = [
"../../api:array_view",
"../../rtc_base:checks",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -281,7 +277,6 @@
"../../rtc_base:checks",
"../../rtc_base:stringutils",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
defines = []
}
@@ -404,7 +399,6 @@
"vad:vad_unittests",
"//testing/gtest",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
defines = []
@@ -495,7 +489,6 @@
"../../rtc_base:safe_conversions",
"../../rtc_base:safe_minmax",
"agc2:gain_map",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -547,7 +540,6 @@
"//third_party/abseil-cpp/absl/flags:flag",
"//third_party/abseil-cpp/absl/flags:parse",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
} # audioproc_f_impl
}
@@ -625,7 +617,6 @@
"../audio_coding:neteq_input_audio_tools",
"//testing/gtest",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
diff --git a/modules/audio_processing/aec3/BUILD.gn b/modules/audio_processing/aec3/BUILD.gn
index 0f84e24..12a32b8 100644
--- a/modules/audio_processing/aec3/BUILD.gn
+++ b/modules/audio_processing/aec3/BUILD.gn
@@ -157,7 +157,6 @@
"../../../system_wrappers:metrics",
"../utility:cascaded_biquad_filter",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
if (current_cpu == "x86" || current_cpu == "x64") {
@@ -228,7 +227,6 @@
"../../../api:array_view",
"../../../rtc_base:gtest_prod",
"../../../rtc_base/system:arch",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -326,7 +324,6 @@
"../../../test:field_trial",
"../../../test:test_support",
"../utility:cascaded_biquad_filter",
- "//third_party/abseil-cpp/absl/types:optional",
]
defines = []
diff --git a/modules/audio_processing/aec3/adaptive_fir_filter_unittest.cc b/modules/audio_processing/aec3/adaptive_fir_filter_unittest.cc
index a13764c..162cf54 100644
--- a/modules/audio_processing/aec3/adaptive_fir_filter_unittest.cc
+++ b/modules/audio_processing/aec3/adaptive_fir_filter_unittest.cc
@@ -484,7 +484,7 @@
std::vector<float> y(kBlockSize, 0.f);
AecState aec_state(EchoCanceller3Config{}, num_capture_channels);
RenderSignalAnalyzer render_signal_analyzer(config);
- absl::optional<DelayEstimate> delay_estimate;
+ std::optional<DelayEstimate> delay_estimate;
std::vector<float> e(kBlockSize, 0.f);
std::array<float, kFftLength> s_scratch;
std::vector<SubtractorOutput> output(num_capture_channels);
diff --git a/modules/audio_processing/aec3/aec_state.cc b/modules/audio_processing/aec3/aec_state.cc
index 81fd91f..e3e9795 100644
--- a/modules/audio_processing/aec3/aec_state.cc
+++ b/modules/audio_processing/aec3/aec_state.cc
@@ -14,9 +14,9 @@
#include <algorithm>
#include <numeric>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "modules/audio_processing/aec3/aec3_common.h"
#include "modules/audio_processing/logging/apm_data_dumper.h"
@@ -170,7 +170,7 @@
}
void AecState::Update(
- const absl::optional<DelayEstimate>& external_delay,
+ const std::optional<DelayEstimate>& external_delay,
rtc::ArrayView<const std::vector<std::array<float, kFftLengthBy2Plus1>>>
adaptive_filter_frequency_responses,
rtc::ArrayView<const std::vector<float>> adaptive_filter_impulse_responses,
@@ -359,7 +359,7 @@
void AecState::FilterDelay::Update(
rtc::ArrayView<const int> analyzer_filter_delay_estimates_blocks,
- const absl::optional<DelayEstimate>& external_delay,
+ const std::optional<DelayEstimate>& external_delay,
size_t blocks_with_proper_filter_adaptation) {
// Update the delay based on the external delay.
if (external_delay &&
@@ -405,7 +405,7 @@
bool active_render,
bool transparent_mode,
bool saturated_capture,
- const absl::optional<DelayEstimate>& external_delay,
+ const std::optional<DelayEstimate>& external_delay,
bool any_filter_converged) {
// Update blocks counter.
const bool filter_update = active_render && !saturated_capture;
diff --git a/modules/audio_processing/aec3/aec_state.h b/modules/audio_processing/aec3/aec_state.h
index a39325c..e80b377 100644
--- a/modules/audio_processing/aec3/aec_state.h
+++ b/modules/audio_processing/aec3/aec_state.h
@@ -16,9 +16,9 @@
#include <array>
#include <atomic>
#include <memory>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/audio/echo_canceller3_config.h"
#include "modules/audio_processing/aec3/aec3_common.h"
@@ -138,7 +138,7 @@
// Updates the aec state.
// TODO(bugs.webrtc.org/10913): Compute multi-channel ERL.
void Update(
- const absl::optional<DelayEstimate>& external_delay,
+ const std::optional<DelayEstimate>& external_delay,
rtc::ArrayView<const std::vector<std::array<float, kFftLengthBy2Plus1>>>
adaptive_filter_frequency_responses,
rtc::ArrayView<const std::vector<float>>
@@ -213,7 +213,7 @@
// Updates the delay estimates based on new data.
void Update(
rtc::ArrayView<const int> analyzer_filter_delay_estimates_blocks,
- const absl::optional<DelayEstimate>& external_delay,
+ const std::optional<DelayEstimate>& external_delay,
size_t blocks_with_proper_filter_adaptation);
private:
@@ -221,7 +221,7 @@
bool external_delay_reported_ = false;
std::vector<int> filter_delays_blocks_;
int min_filter_delay_;
- absl::optional<DelayEstimate> external_delay_;
+ std::optional<DelayEstimate> external_delay_;
} delay_state_;
// Classifier for toggling transparent mode when there is no echo.
@@ -253,7 +253,7 @@
void Update(bool active_render,
bool transparent_mode,
bool saturated_capture,
- const absl::optional<DelayEstimate>& external_delay,
+ const std::optional<DelayEstimate>& external_delay,
bool any_filter_converged);
private:
diff --git a/modules/audio_processing/aec3/aec_state_unittest.cc b/modules/audio_processing/aec3/aec_state_unittest.cc
index 6662c8f..2caf677 100644
--- a/modules/audio_processing/aec3/aec_state_unittest.cc
+++ b/modules/audio_processing/aec3/aec_state_unittest.cc
@@ -28,7 +28,7 @@
ApmDataDumper data_dumper(42);
EchoCanceller3Config config;
AecState state(config, num_capture_channels);
- absl::optional<DelayEstimate> delay_estimate =
+ std::optional<DelayEstimate> delay_estimate =
DelayEstimate(DelayEstimate::Quality::kRefined, 10);
std::unique_ptr<RenderDelayBuffer> render_delay_buffer(
RenderDelayBuffer::Create(config, kSampleRateHz, num_render_channels));
@@ -247,7 +247,7 @@
AecState state(config, kNumCaptureChannels);
std::unique_ptr<RenderDelayBuffer> render_delay_buffer(
RenderDelayBuffer::Create(config, 48000, 1));
- absl::optional<DelayEstimate> delay_estimate;
+ std::optional<DelayEstimate> delay_estimate;
std::vector<std::array<float, kFftLengthBy2Plus1>> E2_refined(
kNumCaptureChannels);
std::vector<std::array<float, kFftLengthBy2Plus1>> Y2(kNumCaptureChannels);
diff --git a/modules/audio_processing/aec3/block_processor.cc b/modules/audio_processing/aec3/block_processor.cc
index 63e3d9c..ad3702e 100644
--- a/modules/audio_processing/aec3/block_processor.cc
+++ b/modules/audio_processing/aec3/block_processor.cc
@@ -13,10 +13,10 @@
#include <atomic>
#include <memory>
+#include <optional>
#include <utility>
#include <vector>
-#include "absl/types/optional.h"
#include "api/audio/echo_canceller3_config.h"
#include "api/audio/echo_control.h"
#include "modules/audio_processing/aec3/aec3_common.h"
@@ -76,7 +76,7 @@
BlockProcessorMetrics metrics_;
RenderDelayBuffer::BufferingEvent render_event_;
size_t capture_call_counter_ = 0;
- absl::optional<DelayEstimate> estimated_delay_;
+ std::optional<DelayEstimate> estimated_delay_;
};
std::atomic<int> BlockProcessorImpl::instance_count_(0);
@@ -223,7 +223,7 @@
void BlockProcessorImpl::GetMetrics(EchoControl::Metrics* metrics) const {
echo_remover_->GetMetrics(metrics);
constexpr int block_size_ms = 4;
- absl::optional<size_t> delay = render_buffer_->Delay();
+ std::optional<size_t> delay = render_buffer_->Delay();
metrics->delay_ms = delay ? static_cast<int>(*delay) * block_size_ms : 0;
}
diff --git a/modules/audio_processing/aec3/block_processor_unittest.cc b/modules/audio_processing/aec3/block_processor_unittest.cc
index aba5c41..123dffd 100644
--- a/modules/audio_processing/aec3/block_processor_unittest.cc
+++ b/modules/audio_processing/aec3/block_processor_unittest.cc
@@ -315,19 +315,18 @@
}
EXPECT_CALL(*echo_remover_mock_pointer, ProcessCapture)
- .WillRepeatedly(
- [](EchoPathVariability /*echo_path_variability*/,
- bool /*capture_signal_saturation*/,
- const absl::optional<DelayEstimate>& /*external_delay*/,
- RenderBuffer* render_buffer, Block* /*linear_output*/,
- Block* capture) {
- const auto& render = render_buffer->GetBlock(0);
- const auto render_view = render.View(/*band=*/0, /*channel=*/0);
- const auto capture_view = capture->View(/*band=*/0, /*channel=*/0);
- for (size_t i = 0; i < kBlockSize; ++i) {
- EXPECT_FLOAT_EQ(render_view[i], capture_view[i]);
- }
- });
+ .WillRepeatedly([](EchoPathVariability /*echo_path_variability*/,
+ bool /*capture_signal_saturation*/,
+ const std::optional<DelayEstimate>& /*external_delay*/,
+ RenderBuffer* render_buffer, Block* /*linear_output*/,
+ Block* capture) {
+ const auto& render = render_buffer->GetBlock(0);
+ const auto render_view = render.View(/*band=*/0, /*channel=*/0);
+ const auto capture_view = capture->View(/*band=*/0, /*channel=*/0);
+ for (size_t i = 0; i < kBlockSize; ++i) {
+ EXPECT_FLOAT_EQ(render_view[i], capture_view[i]);
+ }
+ });
FillSampleVector(++capture_call_counter, kDelayInBlocks,
capture_block.View(/*band=*/0, /*capture=*/0));
diff --git a/modules/audio_processing/aec3/config_selector.cc b/modules/audio_processing/aec3/config_selector.cc
index c55344d..e6bd829 100644
--- a/modules/audio_processing/aec3/config_selector.cc
+++ b/modules/audio_processing/aec3/config_selector.cc
@@ -47,7 +47,7 @@
ConfigSelector::ConfigSelector(
const EchoCanceller3Config& config,
- const absl::optional<EchoCanceller3Config>& multichannel_config,
+ const std::optional<EchoCanceller3Config>& multichannel_config,
int num_render_input_channels)
: config_(config), multichannel_config_(multichannel_config) {
if (multichannel_config_.has_value()) {
diff --git a/modules/audio_processing/aec3/config_selector.h b/modules/audio_processing/aec3/config_selector.h
index 3b3f94e..f56f7ba 100644
--- a/modules/audio_processing/aec3/config_selector.h
+++ b/modules/audio_processing/aec3/config_selector.h
@@ -11,7 +11,8 @@
#ifndef MODULES_AUDIO_PROCESSING_AEC3_CONFIG_SELECTOR_H_
#define MODULES_AUDIO_PROCESSING_AEC3_CONFIG_SELECTOR_H_
-#include "absl/types/optional.h"
+#include <optional>
+
#include "api/audio/echo_canceller3_config.h"
namespace webrtc {
@@ -19,10 +20,9 @@
// Selects the config to use.
class ConfigSelector {
public:
- ConfigSelector(
- const EchoCanceller3Config& config,
- const absl::optional<EchoCanceller3Config>& multichannel_config,
- int num_render_input_channels);
+ ConfigSelector(const EchoCanceller3Config& config,
+ const std::optional<EchoCanceller3Config>& multichannel_config,
+ int num_render_input_channels);
// Updates the config selection based on the detection of multichannel
// content.
@@ -32,7 +32,7 @@
private:
const EchoCanceller3Config config_;
- const absl::optional<EchoCanceller3Config> multichannel_config_;
+ const std::optional<EchoCanceller3Config> multichannel_config_;
const EchoCanceller3Config* active_config_ = nullptr;
};
diff --git a/modules/audio_processing/aec3/config_selector_unittest.cc b/modules/audio_processing/aec3/config_selector_unittest.cc
index 1826bfc..91a60d8 100644
--- a/modules/audio_processing/aec3/config_selector_unittest.cc
+++ b/modules/audio_processing/aec3/config_selector_unittest.cc
@@ -10,9 +10,9 @@
#include "modules/audio_processing/aec3/config_selector.h"
+#include <optional>
#include <tuple>
-#include "absl/types/optional.h"
#include "api/audio/echo_canceller3_config.h"
#include "test/gtest.h"
@@ -39,7 +39,7 @@
const auto [num_channels, detect_stereo_content] = GetParam();
EchoCanceller3Config config;
config.multi_channel.detect_stereo_content = detect_stereo_content;
- absl::optional<EchoCanceller3Config> multichannel_config;
+ std::optional<EchoCanceller3Config> multichannel_config;
config.delay.default_delay = config.delay.default_delay + 1;
const size_t custom_delay_value_in_config = config.delay.default_delay;
@@ -63,7 +63,7 @@
const auto [num_channels, detect_stereo_content] = GetParam();
EchoCanceller3Config config;
config.multi_channel.detect_stereo_content = detect_stereo_content;
- absl::optional<EchoCanceller3Config> multichannel_config = config;
+ std::optional<EchoCanceller3Config> multichannel_config = config;
config.delay.default_delay += 1;
const size_t custom_delay_value_in_config = config.delay.default_delay;
@@ -87,7 +87,7 @@
const int num_channels = GetParam();
EchoCanceller3Config config;
config.multi_channel.detect_stereo_content = true;
- absl::optional<EchoCanceller3Config> multichannel_config = config;
+ std::optional<EchoCanceller3Config> multichannel_config = config;
config.delay.default_delay += 1;
const size_t custom_delay_value_in_config = config.delay.default_delay;
diff --git a/modules/audio_processing/aec3/echo_audibility.cc b/modules/audio_processing/aec3/echo_audibility.cc
index 142a33d..d52b5ff 100644
--- a/modules/audio_processing/aec3/echo_audibility.cc
+++ b/modules/audio_processing/aec3/echo_audibility.cc
@@ -45,7 +45,7 @@
void EchoAudibility::Reset() {
render_stationarity_.Reset();
non_zero_render_seen_ = false;
- render_spectrum_write_prev_ = absl::nullopt;
+ render_spectrum_write_prev_ = std::nullopt;
}
void EchoAudibility::UpdateRenderStationarityFlags(
diff --git a/modules/audio_processing/aec3/echo_audibility.h b/modules/audio_processing/aec3/echo_audibility.h
index b9d6f87..0bf20cd 100644
--- a/modules/audio_processing/aec3/echo_audibility.h
+++ b/modules/audio_processing/aec3/echo_audibility.h
@@ -13,7 +13,8 @@
#include <stddef.h>
-#include "absl/types/optional.h"
+#include <optional>
+
#include "api/array_view.h"
#include "modules/audio_processing/aec3/block_buffer.h"
#include "modules/audio_processing/aec3/render_buffer.h"
@@ -73,7 +74,7 @@
// values.
bool IsRenderTooLow(const BlockBuffer& block_buffer);
- absl::optional<int> render_spectrum_write_prev_;
+ std::optional<int> render_spectrum_write_prev_;
int render_block_write_prev_;
bool non_zero_render_seen_;
const bool use_render_stationarity_at_init_;
diff --git a/modules/audio_processing/aec3/echo_canceller3.cc b/modules/audio_processing/aec3/echo_canceller3.cc
index 4c500f4..00010ce 100644
--- a/modules/audio_processing/aec3/echo_canceller3.cc
+++ b/modules/audio_processing/aec3/echo_canceller3.cc
@@ -718,7 +718,7 @@
EchoCanceller3::EchoCanceller3(
const EchoCanceller3Config& config,
- const absl::optional<EchoCanceller3Config>& multichannel_config,
+ const std::optional<EchoCanceller3Config>& multichannel_config,
int sample_rate_hz,
size_t num_render_channels,
size_t num_capture_channels)
diff --git a/modules/audio_processing/aec3/echo_canceller3.h b/modules/audio_processing/aec3/echo_canceller3.h
index 7bf8e51..b05b3d9 100644
--- a/modules/audio_processing/aec3/echo_canceller3.h
+++ b/modules/audio_processing/aec3/echo_canceller3.h
@@ -15,9 +15,9 @@
#include <atomic>
#include <memory>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/audio/echo_canceller3_config.h"
#include "api/audio/echo_control.h"
@@ -88,12 +88,11 @@
// AnalyzeRender call which can be called concurrently with the other methods.
class EchoCanceller3 : public EchoControl {
public:
- EchoCanceller3(
- const EchoCanceller3Config& config,
- const absl::optional<EchoCanceller3Config>& multichannel_config,
- int sample_rate_hz,
- size_t num_render_channels,
- size_t num_capture_channels);
+ EchoCanceller3(const EchoCanceller3Config& config,
+ const std::optional<EchoCanceller3Config>& multichannel_config,
+ int sample_rate_hz,
+ size_t num_render_channels,
+ size_t num_capture_channels);
~EchoCanceller3() override;
diff --git a/modules/audio_processing/aec3/echo_canceller3_unittest.cc b/modules/audio_processing/aec3/echo_canceller3_unittest.cc
index 10957fd..9cc11df 100644
--- a/modules/audio_processing/aec3/echo_canceller3_unittest.cc
+++ b/modules/audio_processing/aec3/echo_canceller3_unittest.cc
@@ -241,7 +241,7 @@
// output.
void RunCaptureTransportVerificationTest() {
EchoCanceller3 aec3(EchoCanceller3Config(),
- /*multichannel_config=*/absl::nullopt, sample_rate_hz_,
+ /*multichannel_config=*/std::nullopt, sample_rate_hz_,
1, 1);
aec3.SetBlockProcessorForTesting(
std::make_unique<CaptureTransportVerificationProcessor>(num_bands_));
@@ -267,7 +267,7 @@
// block processor.
void RunRenderTransportVerificationTest() {
EchoCanceller3 aec3(EchoCanceller3Config(),
- /*multichannel_config=*/absl::nullopt, sample_rate_hz_,
+ /*multichannel_config=*/std::nullopt, sample_rate_hz_,
1, 1);
aec3.SetBlockProcessorForTesting(
std::make_unique<RenderTransportVerificationProcessor>(num_bands_));
@@ -338,7 +338,7 @@
}
EchoCanceller3 aec3(EchoCanceller3Config(),
- /*multichannel_config=*/absl::nullopt, sample_rate_hz_,
+ /*multichannel_config=*/std::nullopt, sample_rate_hz_,
1, 1);
aec3.SetBlockProcessorForTesting(std::move(block_processor_mock));
@@ -420,7 +420,7 @@
}
EchoCanceller3 aec3(EchoCanceller3Config(),
- /*multichannel_config=*/absl::nullopt, sample_rate_hz_,
+ /*multichannel_config=*/std::nullopt, sample_rate_hz_,
1, 1);
aec3.SetBlockProcessorForTesting(std::move(block_processor_mock));
@@ -508,7 +508,7 @@
}
EchoCanceller3 aec3(EchoCanceller3Config(),
- /*multichannel_config=*/absl::nullopt, sample_rate_hz_,
+ /*multichannel_config=*/std::nullopt, sample_rate_hz_,
1, 1);
aec3.SetBlockProcessorForTesting(std::move(block_processor_mock));
for (size_t frame_index = 0; frame_index < kNumFramesToProcess;
@@ -548,7 +548,7 @@
// capture and render API calls.
void RunRenderSwapQueueVerificationTest() {
const EchoCanceller3Config config;
- EchoCanceller3 aec3(config, /*multichannel_config=*/absl::nullopt,
+ EchoCanceller3 aec3(config, /*multichannel_config=*/std::nullopt,
sample_rate_hz_, 1, 1);
aec3.SetBlockProcessorForTesting(
std::make_unique<RenderTransportVerificationProcessor>(num_bands_));
@@ -598,7 +598,7 @@
// properly reported.
void RunRenderPipelineSwapQueueOverrunReturnValueTest() {
EchoCanceller3 aec3(EchoCanceller3Config(),
- /*multichannel_config=*/absl::nullopt, sample_rate_hz_,
+ /*multichannel_config=*/std::nullopt, sample_rate_hz_,
1, 1);
constexpr size_t kRenderTransferQueueSize = 30;
@@ -625,7 +625,7 @@
// way that the number of bands for the rates are different.
const int aec3_sample_rate_hz = sample_rate_hz_ == 48000 ? 32000 : 48000;
EchoCanceller3 aec3(EchoCanceller3Config(),
- /*multichannel_config=*/absl::nullopt,
+ /*multichannel_config=*/std::nullopt,
aec3_sample_rate_hz, 1, 1);
PopulateInputFrame(frame_length_, 0, &render_buffer_.channels_f()[0][0], 0);
@@ -640,7 +640,7 @@
// way that the number of bands for the rates are different.
const int aec3_sample_rate_hz = sample_rate_hz_ == 48000 ? 32000 : 48000;
EchoCanceller3 aec3(EchoCanceller3Config(),
- /*multichannel_config=*/absl::nullopt,
+ /*multichannel_config=*/std::nullopt,
aec3_sample_rate_hz, 1, 1);
PopulateInputFrame(frame_length_, num_bands_, 0,
&capture_buffer_.split_bands_f(0)[0], 100);
@@ -947,7 +947,7 @@
constexpr size_t kNumBlocksForMonoConfig = 1;
constexpr size_t kNumBlocksForSurroundConfig = 2;
EchoCanceller3Config mono_config;
- absl::optional<EchoCanceller3Config> multichannel_config;
+ std::optional<EchoCanceller3Config> multichannel_config;
mono_config.multi_channel.detect_stereo_content = true;
mono_config.multi_channel.stereo_detection_threshold = 0.0f;
@@ -993,7 +993,7 @@
constexpr size_t kNumBlocksForMonoConfig = 1;
constexpr size_t kNumBlocksForSurroundConfig = 2;
EchoCanceller3Config mono_config;
- absl::optional<EchoCanceller3Config> multichannel_config;
+ std::optional<EchoCanceller3Config> multichannel_config;
constexpr float kStereoDetectionThreshold = 2.0f;
mono_config.multi_channel.detect_stereo_content = true;
@@ -1043,7 +1043,7 @@
constexpr size_t kNumBlocksForMonoConfig = 1;
constexpr size_t kNumBlocksForSurroundConfig = 2;
EchoCanceller3Config mono_config;
- absl::optional<EchoCanceller3Config> surround_config;
+ std::optional<EchoCanceller3Config> surround_config;
mono_config.multi_channel.detect_stereo_content = true;
mono_config.multi_channel.stereo_detection_hysteresis_seconds = 0.5f;
@@ -1101,7 +1101,7 @@
constexpr size_t kNumBlocksForMonoConfig = 1;
constexpr size_t kNumBlocksForSurroundConfig = 2;
EchoCanceller3Config mono_config;
- absl::optional<EchoCanceller3Config> multichannel_config;
+ std::optional<EchoCanceller3Config> multichannel_config;
for (bool detect_stereo_content : {false, true}) {
mono_config.multi_channel.detect_stereo_content = detect_stereo_content;
@@ -1149,7 +1149,7 @@
TEST(EchoCanceller3InputCheckDeathTest, NullCaptureProcessingParameter) {
EXPECT_DEATH(
EchoCanceller3(EchoCanceller3Config(),
- /*multichannel_config_=*/absl::nullopt, 16000, 1, 1)
+ /*multichannel_config_=*/std::nullopt, 16000, 1, 1)
.ProcessCapture(nullptr, false),
"");
}
@@ -1161,7 +1161,7 @@
ApmDataDumper data_dumper(0);
EXPECT_DEATH(
EchoCanceller3(EchoCanceller3Config(),
- /*multichannel_config_=*/absl::nullopt, 8001, 1, 1),
+ /*multichannel_config_=*/std::nullopt, 8001, 1, 1),
"");
}
diff --git a/modules/audio_processing/aec3/echo_path_delay_estimator.cc b/modules/audio_processing/aec3/echo_path_delay_estimator.cc
index 510e4b8..1326779 100644
--- a/modules/audio_processing/aec3/echo_path_delay_estimator.cc
+++ b/modules/audio_processing/aec3/echo_path_delay_estimator.cc
@@ -58,7 +58,7 @@
Reset(true, reset_delay_confidence);
}
-absl::optional<DelayEstimate> EchoPathDelayEstimator::EstimateDelay(
+std::optional<DelayEstimate> EchoPathDelayEstimator::EstimateDelay(
const DownsampledRenderBuffer& render_buffer,
const Block& capture) {
std::array<float, kBlockSize> downsampled_capture_data;
@@ -74,7 +74,7 @@
matched_filter_.Update(render_buffer, downsampled_capture,
matched_filter_lag_aggregator_.ReliableDelayFound());
- absl::optional<DelayEstimate> aggregated_matched_filter_lag =
+ std::optional<DelayEstimate> aggregated_matched_filter_lag =
matched_filter_lag_aggregator_.Aggregate(
matched_filter_.GetBestLagEstimate());
@@ -121,7 +121,7 @@
matched_filter_lag_aggregator_.Reset(reset_delay_confidence);
}
matched_filter_.Reset(/*full_reset=*/reset_lag_aggregator);
- old_aggregated_lag_ = absl::nullopt;
+ old_aggregated_lag_ = std::nullopt;
consistent_estimate_counter_ = 0;
}
} // namespace webrtc
diff --git a/modules/audio_processing/aec3/echo_path_delay_estimator.h b/modules/audio_processing/aec3/echo_path_delay_estimator.h
index b24d0a2..bd5c3b0 100644
--- a/modules/audio_processing/aec3/echo_path_delay_estimator.h
+++ b/modules/audio_processing/aec3/echo_path_delay_estimator.h
@@ -13,7 +13,8 @@
#include <stddef.h>
-#include "absl/types/optional.h"
+#include <optional>
+
#include "api/array_view.h"
#include "modules/audio_processing/aec3/alignment_mixer.h"
#include "modules/audio_processing/aec3/block.h"
@@ -45,7 +46,7 @@
void Reset(bool reset_delay_confidence);
// Produce a delay estimate if such is avaliable.
- absl::optional<DelayEstimate> EstimateDelay(
+ std::optional<DelayEstimate> EstimateDelay(
const DownsampledRenderBuffer& render_buffer,
const Block& capture);
@@ -68,7 +69,7 @@
Decimator capture_decimator_;
MatchedFilter matched_filter_;
MatchedFilterLagAggregator matched_filter_lag_aggregator_;
- absl::optional<DelayEstimate> old_aggregated_lag_;
+ std::optional<DelayEstimate> old_aggregated_lag_;
size_t consistent_estimate_counter_ = 0;
ClockdriftDetector clockdrift_detector_;
diff --git a/modules/audio_processing/aec3/echo_path_delay_estimator_unittest.cc b/modules/audio_processing/aec3/echo_path_delay_estimator_unittest.cc
index e2c101f..4b8fdc9 100644
--- a/modules/audio_processing/aec3/echo_path_delay_estimator_unittest.cc
+++ b/modules/audio_processing/aec3/echo_path_delay_estimator_unittest.cc
@@ -88,7 +88,7 @@
EchoPathDelayEstimator estimator(&data_dumper, config,
kNumCaptureChannels);
- absl::optional<DelayEstimate> estimated_delay_samples;
+ std::optional<DelayEstimate> estimated_delay_samples;
for (size_t k = 0; k < (500 + (delay_samples) / kBlockSize); ++k) {
RandomizeSampleVector(&random_generator,
render.View(/*band=*/0, /*channel=*/0));
diff --git a/modules/audio_processing/aec3/echo_remover.cc b/modules/audio_processing/aec3/echo_remover.cc
index 673d88a..e5c1b4c 100644
--- a/modules/audio_processing/aec3/echo_remover.cc
+++ b/modules/audio_processing/aec3/echo_remover.cc
@@ -120,7 +120,7 @@
// signal.
void ProcessCapture(EchoPathVariability echo_path_variability,
bool capture_signal_saturation,
- const absl::optional<DelayEstimate>& external_delay,
+ const std::optional<DelayEstimate>& external_delay,
RenderBuffer* render_buffer,
Block* linear_output,
Block* capture) override;
@@ -239,7 +239,7 @@
void EchoRemoverImpl::ProcessCapture(
EchoPathVariability echo_path_variability,
bool capture_signal_saturation,
- const absl::optional<DelayEstimate>& external_delay,
+ const std::optional<DelayEstimate>& external_delay,
RenderBuffer* render_buffer,
Block* linear_output,
Block* capture) {
diff --git a/modules/audio_processing/aec3/echo_remover.h b/modules/audio_processing/aec3/echo_remover.h
index f2f4f5e..db3992c 100644
--- a/modules/audio_processing/aec3/echo_remover.h
+++ b/modules/audio_processing/aec3/echo_remover.h
@@ -11,9 +11,9 @@
#ifndef MODULES_AUDIO_PROCESSING_AEC3_ECHO_REMOVER_H_
#define MODULES_AUDIO_PROCESSING_AEC3_ECHO_REMOVER_H_
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/audio/echo_canceller3_config.h"
#include "api/audio/echo_control.h"
#include "modules/audio_processing/aec3/block.h"
@@ -41,7 +41,7 @@
virtual void ProcessCapture(
EchoPathVariability echo_path_variability,
bool capture_signal_saturation,
- const absl::optional<DelayEstimate>& external_delay,
+ const std::optional<DelayEstimate>& external_delay,
RenderBuffer* render_buffer,
Block* linear_output,
Block* capture) = 0;
diff --git a/modules/audio_processing/aec3/echo_remover_unittest.cc b/modules/audio_processing/aec3/echo_remover_unittest.cc
index 66168ab..9934a11 100644
--- a/modules/audio_processing/aec3/echo_remover_unittest.cc
+++ b/modules/audio_processing/aec3/echo_remover_unittest.cc
@@ -53,7 +53,7 @@
TEST_P(EchoRemoverMultiChannel, BasicApiCalls) {
const size_t num_render_channels = std::get<0>(GetParam());
const size_t num_capture_channels = std::get<1>(GetParam());
- absl::optional<DelayEstimate> delay_estimate;
+ std::optional<DelayEstimate> delay_estimate;
for (auto rate : {16000, 32000, 48000}) {
SCOPED_TRACE(ProduceDebugText(rate));
std::unique_ptr<EchoRemover> remover(
@@ -95,7 +95,7 @@
// TODO(peah): Re-enable the test once the issue with memory leaks during DEATH
// tests on test bots has been fixed.c
TEST(EchoRemoverDeathTest, DISABLED_WrongCaptureNumBands) {
- absl::optional<DelayEstimate> delay_estimate;
+ std::optional<DelayEstimate> delay_estimate;
for (auto rate : {16000, 32000, 48000}) {
SCOPED_TRACE(ProduceDebugText(rate));
std::unique_ptr<EchoRemover> remover(
@@ -114,7 +114,7 @@
// Verifies the check for non-null capture block.
TEST(EchoRemoverDeathTest, NullCapture) {
- absl::optional<DelayEstimate> delay_estimate;
+ std::optional<DelayEstimate> delay_estimate;
std::unique_ptr<EchoRemover> remover(
EchoRemover::Create(EchoCanceller3Config(), 16000, 1, 1));
std::unique_ptr<RenderDelayBuffer> render_buffer(
@@ -134,7 +134,7 @@
TEST(EchoRemover, BasicEchoRemoval) {
constexpr int kNumBlocksToProcess = 500;
Random random_generator(42U);
- absl::optional<DelayEstimate> delay_estimate;
+ std::optional<DelayEstimate> delay_estimate;
for (size_t num_channels : {1, 2, 4}) {
for (auto rate : {16000, 32000, 48000}) {
Block x(NumBandsForRate(rate), num_channels);
diff --git a/modules/audio_processing/aec3/erle_estimator.h b/modules/audio_processing/aec3/erle_estimator.h
index 5579759..ac9c738 100644
--- a/modules/audio_processing/aec3/erle_estimator.h
+++ b/modules/audio_processing/aec3/erle_estimator.h
@@ -15,9 +15,9 @@
#include <array>
#include <memory>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/audio/echo_canceller3_config.h"
#include "modules/audio_processing/aec3/aec3_common.h"
@@ -91,7 +91,7 @@
// vector with content between 0 and 1 where 1 indicates that, at this current
// time instant, the linear filter is reaching its maximum subtraction
// performance.
- rtc::ArrayView<const absl::optional<float>> GetInstLinearQualityEstimates()
+ rtc::ArrayView<const std::optional<float>> GetInstLinearQualityEstimates()
const {
return fullband_erle_estimator_.GetInstLinearQualityEstimates();
}
diff --git a/modules/audio_processing/aec3/fullband_erle_estimator.cc b/modules/audio_processing/aec3/fullband_erle_estimator.cc
index e56674e..5545b93 100644
--- a/modules/audio_processing/aec3/fullband_erle_estimator.cc
+++ b/modules/audio_processing/aec3/fullband_erle_estimator.cc
@@ -13,8 +13,8 @@
#include <algorithm>
#include <memory>
#include <numeric>
+#include <optional>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "modules/audio_processing/aec3/aec3_common.h"
#include "modules/audio_processing/logging/apm_data_dumper.h"
@@ -142,7 +142,7 @@
}
void FullBandErleEstimator::ErleInstantaneous::ResetAccumulators() {
- erle_log2_ = absl::nullopt;
+ erle_log2_ = std::nullopt;
inst_quality_estimate_ = 0.f;
num_points_ = 0;
E2_acum_ = 0.f;
diff --git a/modules/audio_processing/aec3/fullband_erle_estimator.h b/modules/audio_processing/aec3/fullband_erle_estimator.h
index 7a08217..cc828b2 100644
--- a/modules/audio_processing/aec3/fullband_erle_estimator.h
+++ b/modules/audio_processing/aec3/fullband_erle_estimator.h
@@ -12,9 +12,9 @@
#define MODULES_AUDIO_PROCESSING_AEC3_FULLBAND_ERLE_ESTIMATOR_H_
#include <memory>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/audio/echo_canceller3_config.h"
#include "modules/audio_processing/aec3/aec3_common.h"
@@ -49,7 +49,7 @@
// Returns an estimation of the current linear filter quality. It returns a
// float number between 0 and 1 mapping 1 to the highest possible quality.
- rtc::ArrayView<const absl::optional<float>> GetInstLinearQualityEstimates()
+ rtc::ArrayView<const std::optional<float>> GetInstLinearQualityEstimates()
const {
return linear_filters_qualities_;
}
@@ -73,10 +73,10 @@
// Resets the members related with an instantaneous estimate.
void ResetAccumulators();
// Returns the instantaneous ERLE in log2 units.
- absl::optional<float> GetInstErleLog2() const { return erle_log2_; }
+ std::optional<float> GetInstErleLog2() const { return erle_log2_; }
// Gets an indication between 0 and 1 of the performance of the linear
// filter for the current time instant.
- absl::optional<float> GetQualityEstimate() const {
+ std::optional<float> GetQualityEstimate() const {
if (erle_log2_) {
float value = inst_quality_estimate_;
if (clamp_inst_quality_to_zero_) {
@@ -85,9 +85,9 @@
if (clamp_inst_quality_to_one_) {
value = std::min(1.f, value);
}
- return absl::optional<float>(value);
+ return std::optional<float>(value);
}
- return absl::nullopt;
+ return std::nullopt;
}
void Dump(const std::unique_ptr<ApmDataDumper>& data_dumper) const;
@@ -96,7 +96,7 @@
void UpdateQualityEstimate();
const bool clamp_inst_quality_to_zero_;
const bool clamp_inst_quality_to_one_;
- absl::optional<float> erle_log2_;
+ std::optional<float> erle_log2_;
float inst_quality_estimate_;
float max_erle_log2_;
float min_erle_log2_;
@@ -110,7 +110,7 @@
std::vector<int> hold_counters_instantaneous_erle_;
std::vector<float> erle_time_domain_log2_;
std::vector<ErleInstantaneous> instantaneous_erle_;
- std::vector<absl::optional<float>> linear_filters_qualities_;
+ std::vector<std::optional<float>> linear_filters_qualities_;
};
} // namespace webrtc
diff --git a/modules/audio_processing/aec3/matched_filter.cc b/modules/audio_processing/aec3/matched_filter.cc
index a22c952..59a3b46 100644
--- a/modules/audio_processing/aec3/matched_filter.cc
+++ b/modules/audio_processing/aec3/matched_filter.cc
@@ -23,8 +23,8 @@
#include <initializer_list>
#include <iterator>
#include <numeric>
+#include <optional>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "modules/audio_processing/aec3/downsampled_render_buffer.h"
#include "modules/audio_processing/logging/apm_data_dumper.h"
@@ -647,8 +647,8 @@
std::fill(f.begin(), f.end(), 0.f);
}
- winner_lag_ = absl::nullopt;
- reported_lag_estimate_ = absl::nullopt;
+ winner_lag_ = std::nullopt;
+ reported_lag_estimate_ = std::nullopt;
if (full_reset) {
for (auto& e : accumulated_error_) {
std::fill(e.begin(), e.end(), 1.0f);
@@ -677,10 +677,10 @@
// Apply all matched filters.
float winner_error_sum = error_sum_anchor;
- winner_lag_ = absl::nullopt;
- reported_lag_estimate_ = absl::nullopt;
+ winner_lag_ = std::nullopt;
+ reported_lag_estimate_ = std::nullopt;
size_t alignment_shift = 0;
- absl::optional<size_t> previous_lag_estimate;
+ std::optional<size_t> previous_lag_estimate;
const int num_filters = static_cast<int>(filters_.size());
int winner_index = -1;
for (int n = 0; n < num_filters; ++n) {
diff --git a/modules/audio_processing/aec3/matched_filter.h b/modules/audio_processing/aec3/matched_filter.h
index fcdecbd..bb111c5 100644
--- a/modules/audio_processing/aec3/matched_filter.h
+++ b/modules/audio_processing/aec3/matched_filter.h
@@ -13,9 +13,9 @@
#include <stddef.h>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "modules/audio_processing/aec3/aec3_common.h"
#include "rtc_base/gtest_prod_util.h"
@@ -133,7 +133,7 @@
void Reset(bool full_reset);
// Returns the current lag estimates.
- absl::optional<const MatchedFilter::LagEstimate> GetBestLagEstimate() const {
+ std::optional<const MatchedFilter::LagEstimate> GetBestLagEstimate() const {
return reported_lag_estimate_;
}
@@ -158,8 +158,8 @@
std::vector<std::vector<float>> accumulated_error_;
std::vector<float> instantaneous_accumulated_error_;
std::vector<float> scratch_memory_;
- absl::optional<MatchedFilter::LagEstimate> reported_lag_estimate_;
- absl::optional<size_t> winner_lag_;
+ std::optional<MatchedFilter::LagEstimate> reported_lag_estimate_;
+ std::optional<size_t> winner_lag_;
int last_detected_best_lag_filter_ = -1;
std::vector<size_t> filters_offsets_;
int number_pre_echo_updates_ = 0;
diff --git a/modules/audio_processing/aec3/matched_filter_lag_aggregator.cc b/modules/audio_processing/aec3/matched_filter_lag_aggregator.cc
index 0ece2a6..a7c886a 100644
--- a/modules/audio_processing/aec3/matched_filter_lag_aggregator.cc
+++ b/modules/audio_processing/aec3/matched_filter_lag_aggregator.cc
@@ -62,8 +62,8 @@
}
}
-absl::optional<DelayEstimate> MatchedFilterLagAggregator::Aggregate(
- const absl::optional<const MatchedFilter::LagEstimate>& lag_estimate) {
+std::optional<DelayEstimate> MatchedFilterLagAggregator::Aggregate(
+ const std::optional<const MatchedFilter::LagEstimate>& lag_estimate) {
if (lag_estimate && pre_echo_lag_aggregator_) {
pre_echo_lag_aggregator_->Dump(data_dumper_);
pre_echo_lag_aggregator_->Aggregate(
@@ -90,7 +90,7 @@
}
}
- return absl::nullopt;
+ return std::nullopt;
}
MatchedFilterLagAggregator::HighestPeakAggregator::HighestPeakAggregator(
diff --git a/modules/audio_processing/aec3/matched_filter_lag_aggregator.h b/modules/audio_processing/aec3/matched_filter_lag_aggregator.h
index 1e80a21..9022424 100644
--- a/modules/audio_processing/aec3/matched_filter_lag_aggregator.h
+++ b/modules/audio_processing/aec3/matched_filter_lag_aggregator.h
@@ -11,9 +11,9 @@
#ifndef MODULES_AUDIO_PROCESSING_AEC3_MATCHED_FILTER_LAG_AGGREGATOR_H_
#define MODULES_AUDIO_PROCESSING_AEC3_MATCHED_FILTER_LAG_AGGREGATOR_H_
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/audio/echo_canceller3_config.h"
#include "modules/audio_processing/aec3/delay_estimate.h"
#include "modules/audio_processing/aec3/matched_filter.h"
@@ -41,8 +41,8 @@
void Reset(bool hard_reset);
// Aggregates the provided lag estimates.
- absl::optional<DelayEstimate> Aggregate(
- const absl::optional<const MatchedFilter::LagEstimate>& lag_estimate);
+ std::optional<DelayEstimate> Aggregate(
+ const std::optional<const MatchedFilter::LagEstimate>& lag_estimate);
// Returns whether a reliable delay estimate has been found.
bool ReliableDelayFound() const { return significant_candidate_found_; }
diff --git a/modules/audio_processing/aec3/matched_filter_lag_aggregator_unittest.cc b/modules/audio_processing/aec3/matched_filter_lag_aggregator_unittest.cc
index 6804102..25c62fe 100644
--- a/modules/audio_processing/aec3/matched_filter_lag_aggregator_unittest.cc
+++ b/modules/audio_processing/aec3/matched_filter_lag_aggregator_unittest.cc
@@ -36,7 +36,7 @@
MatchedFilterLagAggregator aggregator(&data_dumper, /*max_filter_lag=*/100,
config.delay);
- absl::optional<DelayEstimate> aggregated_lag;
+ std::optional<DelayEstimate> aggregated_lag;
for (size_t k = 0; k < kNumLagsBeforeDetection; ++k) {
aggregated_lag = aggregator.Aggregate(
MatchedFilter::LagEstimate(/*lag=*/10, /*pre_echo_lag=*/10));
@@ -66,7 +66,7 @@
MatchedFilterLagAggregator aggregator(&data_dumper, /*max_filter_lag=*/kLag,
config.delay);
for (size_t k = 0; k < kNumLagsBeforeDetection * 10; ++k) {
- absl::optional<DelayEstimate> aggregated_lag = aggregator.Aggregate(
+ std::optional<DelayEstimate> aggregated_lag = aggregator.Aggregate(
MatchedFilter::LagEstimate(/*lag=*/kLag, /*pre_echo_lag=*/kLag));
EXPECT_FALSE(aggregated_lag);
EXPECT_EQ(kLag, aggregated_lag->delay);
@@ -84,7 +84,7 @@
std::vector<MatchedFilter::LagEstimate> lag_estimates(1);
MatchedFilterLagAggregator aggregator(&data_dumper, std::max(kLag1, kLag2),
config.delay);
- absl::optional<DelayEstimate> aggregated_lag;
+ std::optional<DelayEstimate> aggregated_lag;
for (size_t k = 0; k < kNumLagsBeforeDetection; ++k) {
aggregated_lag = aggregator.Aggregate(
MatchedFilter::LagEstimate(/*lag=*/kLag1, /*pre_echo_lag=*/kLag1));
diff --git a/modules/audio_processing/aec3/mock/mock_echo_remover.h b/modules/audio_processing/aec3/mock/mock_echo_remover.h
index 31f075e..0be4a65 100644
--- a/modules/audio_processing/aec3/mock/mock_echo_remover.h
+++ b/modules/audio_processing/aec3/mock/mock_echo_remover.h
@@ -11,9 +11,9 @@
#ifndef MODULES_AUDIO_PROCESSING_AEC3_MOCK_MOCK_ECHO_REMOVER_H_
#define MODULES_AUDIO_PROCESSING_AEC3_MOCK_MOCK_ECHO_REMOVER_H_
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "modules/audio_processing/aec3/echo_path_variability.h"
#include "modules/audio_processing/aec3/echo_remover.h"
#include "modules/audio_processing/aec3/render_buffer.h"
@@ -31,7 +31,7 @@
ProcessCapture,
(EchoPathVariability echo_path_variability,
bool capture_signal_saturation,
- const absl::optional<DelayEstimate>& delay_estimate,
+ const std::optional<DelayEstimate>& delay_estimate,
RenderBuffer* render_buffer,
Block* linear_output,
Block* capture),
diff --git a/modules/audio_processing/aec3/mock/mock_render_delay_controller.h b/modules/audio_processing/aec3/mock/mock_render_delay_controller.h
index 14d499d..3395044 100644
--- a/modules/audio_processing/aec3/mock/mock_render_delay_controller.h
+++ b/modules/audio_processing/aec3/mock/mock_render_delay_controller.h
@@ -11,7 +11,8 @@
#ifndef MODULES_AUDIO_PROCESSING_AEC3_MOCK_MOCK_RENDER_DELAY_CONTROLLER_H_
#define MODULES_AUDIO_PROCESSING_AEC3_MOCK_MOCK_RENDER_DELAY_CONTROLLER_H_
-#include "absl/types/optional.h"
+#include <optional>
+
#include "api/array_view.h"
#include "modules/audio_processing/aec3/downsampled_render_buffer.h"
#include "modules/audio_processing/aec3/render_delay_controller.h"
@@ -27,7 +28,7 @@
MOCK_METHOD(void, Reset, (bool reset_delay_statistics), (override));
MOCK_METHOD(void, LogRenderCall, (), (override));
- MOCK_METHOD(absl::optional<DelayEstimate>,
+ MOCK_METHOD(std::optional<DelayEstimate>,
GetDelay,
(const DownsampledRenderBuffer& render_buffer,
size_t render_delay_buffer_delay,
diff --git a/modules/audio_processing/aec3/multi_channel_content_detector.cc b/modules/audio_processing/aec3/multi_channel_content_detector.cc
index 9806896..2b07585 100644
--- a/modules/audio_processing/aec3/multi_channel_content_detector.cc
+++ b/modules/audio_processing/aec3/multi_channel_content_detector.cc
@@ -94,9 +94,9 @@
detection_threshold_(detection_threshold),
detection_timeout_threshold_frames_(
stereo_detection_timeout_threshold_seconds > 0
- ? absl::make_optional(stereo_detection_timeout_threshold_seconds *
- kNumFramesPerSecond)
- : absl::nullopt),
+ ? std::make_optional(stereo_detection_timeout_threshold_seconds *
+ kNumFramesPerSecond)
+ : std::nullopt),
stereo_detection_hysteresis_frames_(static_cast<int>(
stereo_detection_hysteresis_seconds * kNumFramesPerSecond)),
metrics_logger_((detect_stereo_content && num_render_input_channels > 1)
diff --git a/modules/audio_processing/aec3/multi_channel_content_detector.h b/modules/audio_processing/aec3/multi_channel_content_detector.h
index be8717f..2b2f3b8 100644
--- a/modules/audio_processing/aec3/multi_channel_content_detector.h
+++ b/modules/audio_processing/aec3/multi_channel_content_detector.h
@@ -14,10 +14,9 @@
#include <stddef.h>
#include <memory>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
-
namespace webrtc {
// Analyzes audio content to determine whether the contained audio is proper
@@ -76,7 +75,7 @@
const bool detect_stereo_content_;
const float detection_threshold_;
- const absl::optional<int> detection_timeout_threshold_frames_;
+ const std::optional<int> detection_timeout_threshold_frames_;
const int stereo_detection_hysteresis_frames_;
// Collects and reports metrics on the amount of multichannel content
diff --git a/modules/audio_processing/aec3/refined_filter_update_gain_unittest.cc b/modules/audio_processing/aec3/refined_filter_update_gain_unittest.cc
index c77c5b5..2a05f18 100644
--- a/modules/audio_processing/aec3/refined_filter_update_gain_unittest.cc
+++ b/modules/audio_processing/aec3/refined_filter_update_gain_unittest.cc
@@ -91,7 +91,7 @@
RenderDelayBuffer::Create(config, kSampleRateHz, kNumRenderChannels));
AecState aec_state(config, kNumCaptureChannels);
RenderSignalAnalyzer render_signal_analyzer(config);
- absl::optional<DelayEstimate> delay_estimate;
+ std::optional<DelayEstimate> delay_estimate;
std::array<float, kFftLength> s_scratch;
std::array<float, kBlockSize> s;
FftData S;
diff --git a/modules/audio_processing/aec3/render_delay_buffer.cc b/modules/audio_processing/aec3/render_delay_buffer.cc
index ca77a58..f4721d4 100644
--- a/modules/audio_processing/aec3/render_delay_buffer.cc
+++ b/modules/audio_processing/aec3/render_delay_buffer.cc
@@ -17,9 +17,9 @@
#include <cmath>
#include <memory>
#include <numeric>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/audio/echo_canceller3_config.h"
#include "modules/audio_processing/aec3/aec3_common.h"
@@ -80,7 +80,7 @@
BlockBuffer blocks_;
SpectrumBuffer spectra_;
FftBuffer ffts_;
- absl::optional<size_t> delay_;
+ std::optional<size_t> delay_;
RenderBuffer echo_remover_buffer_;
DownsampledRenderBuffer low_rate_;
AlignmentMixer render_mixer_;
@@ -95,7 +95,7 @@
int64_t render_call_counter_ = 0;
bool render_activity_ = false;
size_t render_activity_counter_ = 0;
- absl::optional<int> external_audio_buffer_delay_;
+ std::optional<int> external_audio_buffer_delay_;
bool external_audio_buffer_delay_verified_after_reset_ = false;
size_t min_latency_blocks_ = 0;
size_t excess_render_detection_counter_ = 0;
@@ -193,7 +193,7 @@
ApplyTotalDelay(config_.delay.default_delay);
// Unset the delays which are set by AlignFromDelay.
- delay_ = absl::nullopt;
+ delay_ = std::nullopt;
}
}
diff --git a/modules/audio_processing/aec3/render_delay_controller.cc b/modules/audio_processing/aec3/render_delay_controller.cc
index 465e77f..df594ea 100644
--- a/modules/audio_processing/aec3/render_delay_controller.cc
+++ b/modules/audio_processing/aec3/render_delay_controller.cc
@@ -14,8 +14,8 @@
#include <algorithm>
#include <atomic>
#include <memory>
+#include <optional>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/audio/echo_canceller3_config.h"
#include "modules/audio_processing/aec3/aec3_common.h"
@@ -44,7 +44,7 @@
~RenderDelayControllerImpl() override;
void Reset(bool reset_delay_confidence) override;
void LogRenderCall() override;
- absl::optional<DelayEstimate> GetDelay(
+ std::optional<DelayEstimate> GetDelay(
const DownsampledRenderBuffer& render_buffer,
size_t render_delay_buffer_delay,
const Block& capture) override;
@@ -54,17 +54,17 @@
static std::atomic<int> instance_count_;
std::unique_ptr<ApmDataDumper> data_dumper_;
const int hysteresis_limit_blocks_;
- absl::optional<DelayEstimate> delay_;
+ std::optional<DelayEstimate> delay_;
EchoPathDelayEstimator delay_estimator_;
RenderDelayControllerMetrics metrics_;
- absl::optional<DelayEstimate> delay_samples_;
+ std::optional<DelayEstimate> delay_samples_;
size_t capture_call_counter_ = 0;
int delay_change_counter_ = 0;
DelayEstimate::Quality last_delay_estimate_quality_;
};
DelayEstimate ComputeBufferDelay(
- const absl::optional<DelayEstimate>& current_delay,
+ const std::optional<DelayEstimate>& current_delay,
int hysteresis_limit_blocks,
DelayEstimate estimated_delay) {
// Compute the buffer delay increase required to achieve the desired latency.
@@ -100,8 +100,8 @@
RenderDelayControllerImpl::~RenderDelayControllerImpl() = default;
void RenderDelayControllerImpl::Reset(bool reset_delay_confidence) {
- delay_ = absl::nullopt;
- delay_samples_ = absl::nullopt;
+ delay_ = std::nullopt;
+ delay_samples_ = std::nullopt;
delay_estimator_.Reset(reset_delay_confidence);
delay_change_counter_ = 0;
if (reset_delay_confidence) {
@@ -111,7 +111,7 @@
void RenderDelayControllerImpl::LogRenderCall() {}
-absl::optional<DelayEstimate> RenderDelayControllerImpl::GetDelay(
+std::optional<DelayEstimate> RenderDelayControllerImpl::GetDelay(
const DownsampledRenderBuffer& render_buffer,
size_t render_delay_buffer_delay,
const Block& capture) {
@@ -155,11 +155,10 @@
last_delay_estimate_quality_ = delay_samples_->quality;
}
- metrics_.Update(
- delay_samples_ ? absl::optional<size_t>(delay_samples_->delay)
- : absl::nullopt,
- delay_ ? absl::optional<size_t>(delay_->delay) : absl::nullopt,
- delay_estimator_.Clockdrift());
+ metrics_.Update(delay_samples_ ? std::optional<size_t>(delay_samples_->delay)
+ : std::nullopt,
+ delay_ ? std::optional<size_t>(delay_->delay) : std::nullopt,
+ delay_estimator_.Clockdrift());
data_dumper_->DumpRaw("aec3_render_delay_controller_delay",
delay_samples ? delay_samples->delay : 0);
diff --git a/modules/audio_processing/aec3/render_delay_controller.h b/modules/audio_processing/aec3/render_delay_controller.h
index 4a18a11..b74c161 100644
--- a/modules/audio_processing/aec3/render_delay_controller.h
+++ b/modules/audio_processing/aec3/render_delay_controller.h
@@ -11,7 +11,8 @@
#ifndef MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_CONTROLLER_H_
#define MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_CONTROLLER_H_
-#include "absl/types/optional.h"
+#include <optional>
+
#include "api/array_view.h"
#include "api/audio/echo_canceller3_config.h"
#include "modules/audio_processing/aec3/block.h"
@@ -38,7 +39,7 @@
virtual void LogRenderCall() = 0;
// Aligns the render buffer content with the capture signal.
- virtual absl::optional<DelayEstimate> GetDelay(
+ virtual std::optional<DelayEstimate> GetDelay(
const DownsampledRenderBuffer& render_buffer,
size_t render_delay_buffer_delay,
const Block& capture) = 0;
diff --git a/modules/audio_processing/aec3/render_delay_controller_metrics.cc b/modules/audio_processing/aec3/render_delay_controller_metrics.cc
index 1e0a0f4..8d3f415 100644
--- a/modules/audio_processing/aec3/render_delay_controller_metrics.cc
+++ b/modules/audio_processing/aec3/render_delay_controller_metrics.cc
@@ -42,8 +42,8 @@
RenderDelayControllerMetrics::RenderDelayControllerMetrics() = default;
void RenderDelayControllerMetrics::Update(
- absl::optional<size_t> delay_samples,
- absl::optional<size_t> buffer_delay_blocks,
+ std::optional<size_t> delay_samples,
+ std::optional<size_t> buffer_delay_blocks,
ClockdriftDetector::Level clockdrift) {
++call_counter_;
diff --git a/modules/audio_processing/aec3/render_delay_controller_metrics.h b/modules/audio_processing/aec3/render_delay_controller_metrics.h
index b81833b..0a6f30e 100644
--- a/modules/audio_processing/aec3/render_delay_controller_metrics.h
+++ b/modules/audio_processing/aec3/render_delay_controller_metrics.h
@@ -13,7 +13,8 @@
#include <stddef.h>
-#include "absl/types/optional.h"
+#include <optional>
+
#include "modules/audio_processing/aec3/clockdrift_detector.h"
namespace webrtc {
@@ -28,8 +29,8 @@
delete;
// Updates the metric with new data.
- void Update(absl::optional<size_t> delay_samples,
- absl::optional<size_t> buffer_delay_blocks,
+ void Update(std::optional<size_t> delay_samples,
+ std::optional<size_t> buffer_delay_blocks,
ClockdriftDetector::Level clockdrift);
private:
diff --git a/modules/audio_processing/aec3/render_delay_controller_metrics_unittest.cc b/modules/audio_processing/aec3/render_delay_controller_metrics_unittest.cc
index cf9df6b..22739e7 100644
--- a/modules/audio_processing/aec3/render_delay_controller_metrics_unittest.cc
+++ b/modules/audio_processing/aec3/render_delay_controller_metrics_unittest.cc
@@ -10,7 +10,8 @@
#include "modules/audio_processing/aec3/render_delay_controller_metrics.h"
-#include "absl/types/optional.h"
+#include <optional>
+
#include "modules/audio_processing/aec3/aec3_common.h"
#include "system_wrappers/include/metrics.h"
#include "test/gtest.h"
@@ -27,7 +28,7 @@
for (int j = 0; j < 3; ++j) {
for (int k = 0; k < kMetricsReportingIntervalBlocks - 1; ++k) {
- metrics.Update(absl::nullopt, absl::nullopt,
+ metrics.Update(std::nullopt, std::nullopt,
ClockdriftDetector::Level::kNone);
}
EXPECT_METRIC_EQ(
@@ -49,7 +50,7 @@
// We expect metric reports every kMetricsReportingIntervalBlocks blocks.
++expected_num_metric_reports;
- metrics.Update(absl::nullopt, absl::nullopt,
+ metrics.Update(std::nullopt, std::nullopt,
ClockdriftDetector::Level::kNone);
EXPECT_METRIC_EQ(
metrics::NumSamples("WebRTC.Audio.EchoCanceller.EchoPathDelay"),
diff --git a/modules/audio_processing/aec3/render_delay_controller_unittest.cc b/modules/audio_processing/aec3/render_delay_controller_unittest.cc
index e1a54fc..adb5344 100644
--- a/modules/audio_processing/aec3/render_delay_controller_unittest.cc
+++ b/modules/audio_processing/aec3/render_delay_controller_unittest.cc
@@ -85,7 +85,7 @@
for (size_t num_capture_channels : {1, 2, 4}) {
for (size_t num_render_channels : {1, 2, 8}) {
Block capture_block(/*num_bands=*/1, num_capture_channels);
- absl::optional<DelayEstimate> delay_blocks;
+ std::optional<DelayEstimate> delay_blocks;
for (size_t num_matched_filters = 4; num_matched_filters <= 10;
num_matched_filters++) {
for (auto down_sampling_factor : kDownSamplingFactors) {
@@ -140,7 +140,7 @@
Block render_block(NumBandsForRate(rate), num_render_channels);
for (size_t delay_samples : {15, 50, 150, 200, 800, 4000}) {
- absl::optional<DelayEstimate> delay_blocks;
+ std::optional<DelayEstimate> delay_blocks;
SCOPED_TRACE(ProduceDebugText(rate, delay_samples,
num_render_channels,
num_capture_channels));
@@ -203,7 +203,7 @@
Block capture_block(NumBandsForRate(rate), num_capture_channels);
for (int delay_samples : {-15, -50, -150, -200}) {
- absl::optional<DelayEstimate> delay_blocks;
+ std::optional<DelayEstimate> delay_blocks;
SCOPED_TRACE(ProduceDebugText(rate, -delay_samples,
num_render_channels,
num_capture_channels));
@@ -259,7 +259,7 @@
for (auto rate : {16000, 32000, 48000}) {
Block render_block(NumBandsForRate(rate), num_render_channels);
for (size_t delay_samples : {15, 50, 300, 800}) {
- absl::optional<DelayEstimate> delay_blocks;
+ std::optional<DelayEstimate> delay_blocks;
SCOPED_TRACE(ProduceDebugText(rate, delay_samples,
num_render_channels,
num_capture_channels));
diff --git a/modules/audio_processing/aec3/render_signal_analyzer.cc b/modules/audio_processing/aec3/render_signal_analyzer.cc
index bfbeb0e..a3e78e4 100644
--- a/modules/audio_processing/aec3/render_signal_analyzer.cc
+++ b/modules/audio_processing/aec3/render_signal_analyzer.cc
@@ -27,7 +27,7 @@
// Identifies local bands with narrow characteristics.
void IdentifySmallNarrowBandRegions(
const RenderBuffer& render_buffer,
- const absl::optional<size_t>& delay_partitions,
+ const std::optional<size_t>& delay_partitions,
std::array<size_t, kFftLengthBy2 - 1>* narrow_band_counters) {
RTC_DCHECK(narrow_band_counters);
@@ -56,14 +56,14 @@
// Identifies whether the signal has a single strong narrow-band component.
void IdentifyStrongNarrowBandComponent(const RenderBuffer& render_buffer,
int strong_peak_freeze_duration,
- absl::optional<int>* narrow_peak_band,
+ std::optional<int>* narrow_peak_band,
size_t* narrow_peak_counter) {
RTC_DCHECK(narrow_peak_band);
RTC_DCHECK(narrow_peak_counter);
if (*narrow_peak_band &&
++(*narrow_peak_counter) >
static_cast<size_t>(strong_peak_freeze_duration)) {
- *narrow_peak_band = absl::nullopt;
+ *narrow_peak_band = std::nullopt;
}
const Block& x_latest = render_buffer.GetBlock(0);
@@ -125,7 +125,7 @@
void RenderSignalAnalyzer::Update(
const RenderBuffer& render_buffer,
- const absl::optional<size_t>& delay_partitions) {
+ const std::optional<size_t>& delay_partitions) {
// Identify bands of narrow nature.
IdentifySmallNarrowBandRegions(render_buffer, delay_partitions,
&narrow_band_counters_);
diff --git a/modules/audio_processing/aec3/render_signal_analyzer.h b/modules/audio_processing/aec3/render_signal_analyzer.h
index 2e4aaa4..058b292 100644
--- a/modules/audio_processing/aec3/render_signal_analyzer.h
+++ b/modules/audio_processing/aec3/render_signal_analyzer.h
@@ -14,8 +14,8 @@
#include <algorithm>
#include <array>
#include <cstddef>
+#include <optional>
-#include "absl/types/optional.h"
#include "api/audio/echo_canceller3_config.h"
#include "modules/audio_processing/aec3/aec3_common.h"
#include "modules/audio_processing/aec3/render_buffer.h"
@@ -34,7 +34,7 @@
// Updates the render signal analysis with the most recent render signal.
void Update(const RenderBuffer& render_buffer,
- const absl::optional<size_t>& delay_partitions);
+ const std::optional<size_t>& delay_partitions);
// Returns true if the render signal is poorly exciting.
bool PoorSignalExcitation() const {
@@ -48,12 +48,12 @@
void MaskRegionsAroundNarrowBands(
std::array<float, kFftLengthBy2Plus1>* v) const;
- absl::optional<int> NarrowPeakBand() const { return narrow_peak_band_; }
+ std::optional<int> NarrowPeakBand() const { return narrow_peak_band_; }
private:
const int strong_peak_freeze_duration_;
std::array<size_t, kFftLengthBy2 - 1> narrow_band_counters_;
- absl::optional<int> narrow_peak_band_;
+ std::optional<int> narrow_peak_band_;
size_t narrow_peak_counter_;
};
diff --git a/modules/audio_processing/aec3/render_signal_analyzer_unittest.cc b/modules/audio_processing/aec3/render_signal_analyzer_unittest.cc
index 16f6280..ca428ba 100644
--- a/modules/audio_processing/aec3/render_signal_analyzer_unittest.cc
+++ b/modules/audio_processing/aec3/render_signal_analyzer_unittest.cc
@@ -84,7 +84,7 @@
render_delay_buffer->PrepareCaptureProcessing();
analyzer.Update(*render_delay_buffer->GetRenderBuffer(),
- known_delay ? absl::optional<size_t>(0) : absl::nullopt);
+ known_delay ? std::optional<size_t>(0) : std::nullopt);
}
};
@@ -149,7 +149,7 @@
render_delay_buffer->PrepareCaptureProcessing();
analyzer.Update(*render_delay_buffer->GetRenderBuffer(),
- absl::optional<size_t>(0));
+ std::optional<size_t>(0));
}
mask.fill(1.f);
diff --git a/modules/audio_processing/aec3/residual_echo_estimator.h b/modules/audio_processing/aec3/residual_echo_estimator.h
index c468764..3bed698 100644
--- a/modules/audio_processing/aec3/residual_echo_estimator.h
+++ b/modules/audio_processing/aec3/residual_echo_estimator.h
@@ -13,8 +13,8 @@
#include <array>
#include <memory>
+#include <optional>
-#include "absl/types/optional.h"
#include "api/audio/echo_canceller3_config.h"
#include "modules/audio_processing/aec3/aec3_common.h"
#include "modules/audio_processing/aec3/aec_state.h"
diff --git a/modules/audio_processing/aec3/residual_echo_estimator_unittest.cc b/modules/audio_processing/aec3/residual_echo_estimator_unittest.cc
index 9a7bf0a..6ca5bd4 100644
--- a/modules/audio_processing/aec3/residual_echo_estimator_unittest.cc
+++ b/modules/audio_processing/aec3/residual_echo_estimator_unittest.cc
@@ -120,7 +120,7 @@
Random random_generator_;
std::vector<SubtractorOutput> output_;
std::array<float, kBlockSize> y_;
- absl::optional<DelayEstimate> delay_estimate_;
+ std::optional<DelayEstimate> delay_estimate_;
bool first_frame_ = true;
};
diff --git a/modules/audio_processing/aec3/reverb_decay_estimator.cc b/modules/audio_processing/aec3/reverb_decay_estimator.cc
index 2daf376..ddff1c9 100644
--- a/modules/audio_processing/aec3/reverb_decay_estimator.cc
+++ b/modules/audio_processing/aec3/reverb_decay_estimator.cc
@@ -102,7 +102,7 @@
ReverbDecayEstimator::~ReverbDecayEstimator() = default;
void ReverbDecayEstimator::Update(rtc::ArrayView<const float> filter,
- const absl::optional<float>& filter_quality,
+ const std::optional<float>& filter_quality,
int filter_delay_blocks,
bool usable_linear_filter,
bool stationary_signal) {
diff --git a/modules/audio_processing/aec3/reverb_decay_estimator.h b/modules/audio_processing/aec3/reverb_decay_estimator.h
index fee5421..6d9d849 100644
--- a/modules/audio_processing/aec3/reverb_decay_estimator.h
+++ b/modules/audio_processing/aec3/reverb_decay_estimator.h
@@ -12,9 +12,9 @@
#define MODULES_AUDIO_PROCESSING_AEC3_REVERB_DECAY_ESTIMATOR_H_
#include <array>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "modules/audio_processing/aec3/aec3_common.h" // kMaxAdaptiveFilter...
@@ -30,7 +30,7 @@
~ReverbDecayEstimator();
// Updates the decay estimate.
void Update(rtc::ArrayView<const float> filter,
- const absl::optional<float>& filter_quality,
+ const std::optional<float>& filter_quality,
int filter_delay_blocks,
bool usable_linear_filter,
bool stationary_signal);
diff --git a/modules/audio_processing/aec3/reverb_frequency_response.cc b/modules/audio_processing/aec3/reverb_frequency_response.cc
index 6e7282a..e96930a 100644
--- a/modules/audio_processing/aec3/reverb_frequency_response.cc
+++ b/modules/audio_processing/aec3/reverb_frequency_response.cc
@@ -62,7 +62,7 @@
const std::vector<std::array<float, kFftLengthBy2Plus1>>&
frequency_response,
int filter_delay_blocks,
- const absl::optional<float>& linear_filter_quality,
+ const std::optional<float>& linear_filter_quality,
bool stationary_block) {
if (stationary_block || !linear_filter_quality) {
return;
diff --git a/modules/audio_processing/aec3/reverb_frequency_response.h b/modules/audio_processing/aec3/reverb_frequency_response.h
index 69b16b5..8bb9313 100644
--- a/modules/audio_processing/aec3/reverb_frequency_response.h
+++ b/modules/audio_processing/aec3/reverb_frequency_response.h
@@ -12,9 +12,9 @@
#define MODULES_AUDIO_PROCESSING_AEC3_REVERB_FREQUENCY_RESPONSE_H_
#include <array>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "modules/audio_processing/aec3/aec3_common.h"
@@ -31,7 +31,7 @@
void Update(const std::vector<std::array<float, kFftLengthBy2Plus1>>&
frequency_response,
int filter_delay_blocks,
- const absl::optional<float>& linear_filter_quality,
+ const std::optional<float>& linear_filter_quality,
bool stationary_block);
// Returns the estimated frequency response for the reverb.
diff --git a/modules/audio_processing/aec3/reverb_model_estimator.cc b/modules/audio_processing/aec3/reverb_model_estimator.cc
index 5cd7a78..8a815e6 100644
--- a/modules/audio_processing/aec3/reverb_model_estimator.cc
+++ b/modules/audio_processing/aec3/reverb_model_estimator.cc
@@ -31,7 +31,7 @@
rtc::ArrayView<const std::vector<float>> impulse_responses,
rtc::ArrayView<const std::vector<std::array<float, kFftLengthBy2Plus1>>>
frequency_responses,
- rtc::ArrayView<const absl::optional<float>> linear_filter_qualities,
+ rtc::ArrayView<const std::optional<float>> linear_filter_qualities,
rtc::ArrayView<const int> filter_delays_blocks,
const std::vector<bool>& usable_linear_estimates,
bool stationary_block) {
diff --git a/modules/audio_processing/aec3/reverb_model_estimator.h b/modules/audio_processing/aec3/reverb_model_estimator.h
index 63bade9..550239c 100644
--- a/modules/audio_processing/aec3/reverb_model_estimator.h
+++ b/modules/audio_processing/aec3/reverb_model_estimator.h
@@ -13,9 +13,9 @@
#include <array>
#include <memory>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/audio/echo_canceller3_config.h"
#include "modules/audio_processing/aec3/aec3_common.h" // kFftLengthBy2Plus1
@@ -38,7 +38,7 @@
rtc::ArrayView<const std::vector<float>> impulse_responses,
rtc::ArrayView<const std::vector<std::array<float, kFftLengthBy2Plus1>>>
frequency_responses,
- rtc::ArrayView<const absl::optional<float>> linear_filter_qualities,
+ rtc::ArrayView<const std::optional<float>> linear_filter_qualities,
rtc::ArrayView<const int> filter_delays_blocks,
const std::vector<bool>& usable_linear_estimates,
bool stationary_block);
diff --git a/modules/audio_processing/aec3/reverb_model_estimator_unittest.cc b/modules/audio_processing/aec3/reverb_model_estimator_unittest.cc
index fb7dcef..f1c962a 100644
--- a/modules/audio_processing/aec3/reverb_model_estimator_unittest.cc
+++ b/modules/audio_processing/aec3/reverb_model_estimator_unittest.cc
@@ -14,9 +14,9 @@
#include <array>
#include <cmath>
#include <numeric>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/audio/echo_canceller3_config.h"
#include "modules/audio_processing/aec3/aec3_common.h"
@@ -74,7 +74,7 @@
float true_power_tail_ = 0.f;
std::vector<std::vector<float>> h_;
std::vector<std::vector<std::array<float, kFftLengthBy2Plus1>>> H2_;
- std::vector<absl::optional<float>> quality_linear_;
+ std::vector<std::optional<float>> quality_linear_;
};
void ReverbModelEstimatorTest::CreateImpulseResponseWithDecay() {
diff --git a/modules/audio_processing/aec3/subtractor_unittest.cc b/modules/audio_processing/aec3/subtractor_unittest.cc
index 56b9cec..718bee0 100644
--- a/modules/audio_processing/aec3/subtractor_unittest.cc
+++ b/modules/audio_processing/aec3/subtractor_unittest.cc
@@ -44,7 +44,7 @@
Subtractor subtractor(config, num_render_channels, num_capture_channels,
&data_dumper, DetectOptimization());
- absl::optional<DelayEstimate> delay_estimate;
+ std::optional<DelayEstimate> delay_estimate;
Block x(kNumBands, num_render_channels);
Block y(/*num_bands=*/1, num_capture_channels);
std::array<float, kBlockSize> x_old;
diff --git a/modules/audio_processing/aec3/suppression_gain.cc b/modules/audio_processing/aec3/suppression_gain.cc
index 037daba..26797fc 100644
--- a/modules/audio_processing/aec3/suppression_gain.cc
+++ b/modules/audio_processing/aec3/suppression_gain.cc
@@ -107,7 +107,7 @@
rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>> echo_spectrum,
rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>>
comfort_noise_spectrum,
- const absl::optional<int>& narrow_peak_band,
+ const std::optional<int>& narrow_peak_band,
bool saturated_echo,
const Block& render,
const std::array<float, kFftLengthBy2Plus1>& low_band_gain) const {
@@ -394,7 +394,7 @@
low_band_gain);
// Compute the gain for the upper bands.
- const absl::optional<int> narrow_peak_band =
+ const std::optional<int> narrow_peak_band =
render_signal_analyzer.NarrowPeakBand();
*high_bands_gain =
diff --git a/modules/audio_processing/aec3/suppression_gain.h b/modules/audio_processing/aec3/suppression_gain.h
index c19ddd7..806334c 100644
--- a/modules/audio_processing/aec3/suppression_gain.h
+++ b/modules/audio_processing/aec3/suppression_gain.h
@@ -14,9 +14,9 @@
#include <array>
#include <atomic>
#include <memory>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/audio/echo_canceller3_config.h"
#include "modules/audio_processing/aec3/aec3_common.h"
@@ -70,7 +70,7 @@
rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>> echo_spectrum,
rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>>
comfort_noise_spectrum,
- const absl::optional<int>& narrow_peak_band,
+ const std::optional<int>& narrow_peak_band,
bool saturated_echo,
const Block& render,
const std::array<float, kFftLengthBy2Plus1>& low_band_gain) const;
diff --git a/modules/audio_processing/aec3/suppression_gain_unittest.cc b/modules/audio_processing/aec3/suppression_gain_unittest.cc
index 02de706..bb50200 100644
--- a/modules/audio_processing/aec3/suppression_gain_unittest.cc
+++ b/modules/audio_processing/aec3/suppression_gain_unittest.cc
@@ -81,7 +81,7 @@
&data_dumper, DetectOptimization());
std::unique_ptr<RenderDelayBuffer> render_delay_buffer(
RenderDelayBuffer::Create(config, kSampleRateHz, kNumRenderChannels));
- absl::optional<DelayEstimate> delay_estimate;
+ std::optional<DelayEstimate> delay_estimate;
// Ensure that a strong noise is detected to mask any echoes.
for (size_t ch = 0; ch < kNumCaptureChannels; ++ch) {
diff --git a/modules/audio_processing/agc/BUILD.gn b/modules/audio_processing/agc/BUILD.gn
index 1616f54..f5bb272 100644
--- a/modules/audio_processing/agc/BUILD.gn
+++ b/modules/audio_processing/agc/BUILD.gn
@@ -38,7 +38,6 @@
"../agc2:gain_map",
"../agc2:input_volume_stats_reporter",
"../vad",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
diff --git a/modules/audio_processing/agc/agc_manager_direct.cc b/modules/audio_processing/agc/agc_manager_direct.cc
index b8ad4a8..18d84d8 100644
--- a/modules/audio_processing/agc/agc_manager_direct.cc
+++ b/modules/audio_processing/agc/agc_manager_direct.cc
@@ -69,11 +69,11 @@
// string. Returns an unspecified value if the field trial is not specified, if
// disabled or if it cannot be parsed. Example:
// 'WebRTC-Audio-2ndAgcMinMicLevelExperiment/Enabled-80' => returns 80.
-absl::optional<int> GetMinMicLevelOverride() {
+std::optional<int> GetMinMicLevelOverride() {
constexpr char kMinMicLevelFieldTrial[] =
"WebRTC-Audio-2ndAgcMinMicLevelExperiment";
if (!webrtc::field_trial::IsEnabled(kMinMicLevelFieldTrial)) {
- return absl::nullopt;
+ return std::nullopt;
}
const auto field_trial_string =
webrtc::field_trial::FindFullName(kMinMicLevelFieldTrial);
@@ -84,7 +84,7 @@
} else {
RTC_LOG(LS_WARNING) << "[agc] Invalid parameter for "
<< kMinMicLevelFieldTrial << ", ignored.";
- return absl::nullopt;
+ return std::nullopt;
}
}
@@ -189,8 +189,8 @@
}
void MonoAgc::Process(rtc::ArrayView<const int16_t> audio,
- absl::optional<int> rms_error_override) {
- new_compression_to_set_ = absl::nullopt;
+ std::optional<int> rms_error_override) {
+ new_compression_to_set_ = std::nullopt;
if (check_volume_on_next_process_) {
check_volume_on_next_process_ = false;
@@ -617,13 +617,13 @@
}
void AgcManagerDirect::Process(const AudioBuffer& audio_buffer) {
- Process(audio_buffer, /*speech_probability=*/absl::nullopt,
- /*speech_level_dbfs=*/absl::nullopt);
+ Process(audio_buffer, /*speech_probability=*/std::nullopt,
+ /*speech_level_dbfs=*/std::nullopt);
}
void AgcManagerDirect::Process(const AudioBuffer& audio_buffer,
- absl::optional<float> speech_probability,
- absl::optional<float> speech_level_dbfs) {
+ std::optional<float> speech_probability,
+ std::optional<float> speech_level_dbfs) {
AggregateChannelLevels();
const int volume_after_clipping_handling = recommended_input_volume_;
@@ -632,7 +632,7 @@
}
const size_t num_frames_per_band = audio_buffer.num_frames_per_band();
- absl::optional<int> rms_error_override = absl::nullopt;
+ std::optional<int> rms_error_override = std::nullopt;
if (speech_probability.has_value() && speech_level_dbfs.has_value()) {
rms_error_override =
GetSpeechLevelErrorDb(*speech_level_dbfs, *speech_probability);
@@ -656,7 +656,7 @@
}
}
-absl::optional<int> AgcManagerDirect::GetDigitalComressionGain() {
+std::optional<int> AgcManagerDirect::GetDigitalComressionGain() {
return new_compressions_to_set_[channel_controlling_gain_];
}
diff --git a/modules/audio_processing/agc/agc_manager_direct.h b/modules/audio_processing/agc/agc_manager_direct.h
index 62b0e0e..db2d27b 100644
--- a/modules/audio_processing/agc/agc_manager_direct.h
+++ b/modules/audio_processing/agc/agc_manager_direct.h
@@ -13,8 +13,8 @@
#include <atomic>
#include <memory>
+#include <optional>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/audio/audio_processing.h"
#include "modules/audio_processing/agc/agc.h"
@@ -75,8 +75,8 @@
// TODO(webrtc:7494): This signature is needed for testing purposes, unify
// the signatures when the clean-up is done.
void Process(const AudioBuffer& audio_buffer,
- absl::optional<float> speech_probability,
- absl::optional<float> speech_level_dbfs);
+ std::optional<float> speech_probability,
+ std::optional<float> speech_level_dbfs);
// Processes `audio_buffer`. Chooses a digital compression gain and the new
// input volume to recommend. Must be called after `AnalyzePreProcess()`.
@@ -100,7 +100,7 @@
// If available, returns the latest digital compression gain that has been
// chosen.
- absl::optional<int> GetDigitalComressionGain();
+ std::optional<int> GetDigitalComressionGain();
// Returns true if clipping prediction is enabled.
bool clipping_predictor_enabled() const { return !!clipping_predictor_; }
@@ -150,7 +150,7 @@
const bool analog_controller_enabled_;
- const absl::optional<int> min_mic_level_override_;
+ const std::optional<int> min_mic_level_override_;
std::unique_ptr<ApmDataDumper> data_dumper_;
static std::atomic<int> instance_counter_;
const int num_capture_channels_;
@@ -176,7 +176,7 @@
const int clipped_wait_frames_;
std::vector<std::unique_ptr<MonoAgc>> channel_agcs_;
- std::vector<absl::optional<int>> new_compressions_to_set_;
+ std::vector<std::optional<int>> new_compressions_to_set_;
const std::unique_ptr<ClippingPredictor> clipping_predictor_;
const bool use_clipping_predictor_step_;
@@ -213,16 +213,14 @@
// after `HandleClipping()`. If `rms_error_override` has a value, RMS error
// from AGC is overridden by it.
void Process(rtc::ArrayView<const int16_t> audio,
- absl::optional<int> rms_error_override);
+ std::optional<int> rms_error_override);
// Returns the recommended input volume. Must be called after `Process()`.
int recommended_analog_level() const { return recommended_input_volume_; }
float voice_probability() const { return agc_->voice_probability(); }
void ActivateLogging() { log_to_histograms_ = true; }
- absl::optional<int> new_compression() const {
- return new_compression_to_set_;
- }
+ std::optional<int> new_compression() const { return new_compression_to_set_; }
// Only used for testing.
void set_agc(Agc* agc) { agc_.reset(agc); }
@@ -263,7 +261,7 @@
// recommended input volume.
int recommended_input_volume_ = 0;
- absl::optional<int> new_compression_to_set_;
+ std::optional<int> new_compression_to_set_;
bool log_to_histograms_ = false;
const int clipped_level_min_;
diff --git a/modules/audio_processing/agc/agc_manager_direct_unittest.cc b/modules/audio_processing/agc/agc_manager_direct_unittest.cc
index 70ac0b5..3b6ae72 100644
--- a/modules/audio_processing/agc/agc_manager_direct_unittest.cc
+++ b/modules/audio_processing/agc/agc_manager_direct_unittest.cc
@@ -160,7 +160,7 @@
}
std::string GetAgcMinMicLevelExperimentFieldTrial(
- absl::optional<int> min_mic_level) {
+ std::optional<int> min_mic_level) {
if (min_mic_level.has_value()) {
return GetAgcMinMicLevelExperimentFieldTrialEnabled(*min_mic_level);
}
@@ -196,8 +196,8 @@
// `AgcManagerDirectTestHelper::CallAgcSequence()` instead.
void CallPreProcessAndProcess(int num_calls,
const AudioBuffer& audio_buffer,
- absl::optional<float> speech_probability_override,
- absl::optional<float> speech_level_override,
+ std::optional<float> speech_probability_override,
+ std::optional<float> speech_level_override,
AgcManagerDirect& manager) {
for (int n = 0; n < num_calls; ++n) {
manager.AnalyzePreProcess(audio_buffer);
@@ -266,8 +266,8 @@
// have a value.
void Feed(int num_frames,
int gain_db,
- absl::optional<float> speech_probability_override,
- absl::optional<float> speech_level_override,
+ std::optional<float> speech_probability_override,
+ std::optional<float> speech_level_override,
AgcManagerDirect& agc) {
float gain = std::pow(10.0f, gain_db / 20.0f); // From dB to linear gain.
is_.seekg(0, is_.beg); // Start from the beginning of the PCM file.
@@ -353,13 +353,13 @@
// AGC is replaced by an override value if `speech_probability_override`
// and `speech_level_override` have a value.
int CallAgcSequence(int applied_input_volume,
- absl::optional<float> speech_probability_override,
- absl::optional<float> speech_level_override) {
+ std::optional<float> speech_probability_override,
+ std::optional<float> speech_level_override) {
manager.set_stream_analog_level(applied_input_volume);
manager.AnalyzePreProcess(audio_buffer);
manager.Process(audio_buffer, speech_probability_override,
speech_level_override);
- absl::optional<int> digital_gain = manager.GetDigitalComressionGain();
+ std::optional<int> digital_gain = manager.GetDigitalComressionGain();
if (digital_gain) {
mock_gain_control.set_compression_gain_db(*digital_gain);
}
@@ -372,13 +372,13 @@
// value if `speech_probability_override` and `speech_level_override` have
// a value.
void CallProcess(int num_calls,
- absl::optional<float> speech_probability_override,
- absl::optional<float> speech_level_override) {
+ std::optional<float> speech_probability_override,
+ std::optional<float> speech_level_override) {
for (int i = 0; i < num_calls; ++i) {
EXPECT_CALL(*mock_agc, Process(_)).WillOnce(Return());
manager.Process(audio_buffer, speech_probability_override,
speech_level_override);
- absl::optional<int> new_digital_gain = manager.GetDigitalComressionGain();
+ std::optional<int> new_digital_gain = manager.GetDigitalComressionGain();
if (new_digital_gain) {
mock_gain_control.set_compression_gain_db(*new_digital_gain);
}
@@ -435,7 +435,7 @@
};
class AgcManagerDirectParametrizedTest
- : public ::testing::TestWithParam<std::tuple<absl::optional<int>, bool>> {
+ : public ::testing::TestWithParam<std::tuple<std::optional<int>, bool>> {
protected:
AgcManagerDirectParametrizedTest()
: field_trials_(
@@ -449,9 +449,8 @@
}
bool IsRmsErrorOverridden() const { return std::get<1>(GetParam()); }
- absl::optional<float> GetOverrideOrEmpty(float value) const {
- return IsRmsErrorOverridden() ? absl::optional<float>(value)
- : absl::nullopt;
+ std::optional<float> GetOverrideOrEmpty(float value) const {
+ return IsRmsErrorOverridden() ? std::optional<float>(value) : std::nullopt;
}
private:
@@ -461,8 +460,7 @@
INSTANTIATE_TEST_SUITE_P(
,
AgcManagerDirectParametrizedTest,
- ::testing::Combine(testing::Values(absl::nullopt, 12, 20),
- testing::Bool()));
+ ::testing::Combine(testing::Values(std::nullopt, 12, 20), testing::Bool()));
// Checks that when the analog controller is disabled, no downward adaptation
// takes place.
@@ -706,7 +704,7 @@
}
TEST_P(AgcManagerDirectParametrizedTest, CompressorStepsTowardsTarget) {
- constexpr absl::optional<float> kNoOverride = absl::nullopt;
+ constexpr std::optional<float> kNoOverride = std::nullopt;
const auto speech_probability_override =
GetOverrideOrEmpty(kHighSpeechProbability);
@@ -723,7 +721,7 @@
GetOverrideOrEmpty(-23.0f));
EXPECT_CALL(helper.mock_gain_control, set_compression_gain_db(_)).Times(0);
// The mock `GetRmsErrorDb()` returns false; mimic this by passing
- // absl::nullopt as an override.
+ // std::nullopt as an override.
helper.CallProcess(/*num_calls=*/19, kNoOverride, kNoOverride);
// Moves slowly upwards.
@@ -778,7 +776,7 @@
}
TEST_P(AgcManagerDirectParametrizedTest, CompressorErrorIsDeemphasized) {
- constexpr absl::optional<float> kNoOverride = absl::nullopt;
+ constexpr std::optional<float> kNoOverride = std::nullopt;
const auto speech_probability_override =
GetOverrideOrEmpty(kHighSpeechProbability);
@@ -792,7 +790,7 @@
helper.CallProcess(/*num_calls=*/1, speech_probability_override,
GetOverrideOrEmpty(-28.0f));
// The mock `GetRmsErrorDb()` returns false; mimic this by passing
- // absl::nullopt as an override.
+ // std::nullopt as an override.
helper.CallProcess(/*num_calls=*/18, kNoOverride, kNoOverride);
EXPECT_CALL(helper.mock_gain_control, set_compression_gain_db(8))
.WillOnce(Return(0));
@@ -823,7 +821,7 @@
}
TEST_P(AgcManagerDirectParametrizedTest, CompressorReachesMaximum) {
- constexpr absl::optional<float> kNoOverride = absl::nullopt;
+ constexpr std::optional<float> kNoOverride = std::nullopt;
const auto speech_probability_override =
GetOverrideOrEmpty(kHighSpeechProbability);
@@ -840,7 +838,7 @@
helper.CallProcess(/*num_calls=*/4, speech_probability_override,
GetOverrideOrEmpty(-28.0f));
// The mock `GetRmsErrorDb()` returns false; mimic this by passing
- // absl::nullopt as an override.
+ // std::nullopt as an override.
helper.CallProcess(/*num_calls=*/15, kNoOverride, kNoOverride);
EXPECT_CALL(helper.mock_gain_control, set_compression_gain_db(8))
.WillOnce(Return(0));
@@ -860,7 +858,7 @@
}
TEST_P(AgcManagerDirectParametrizedTest, CompressorReachesMinimum) {
- constexpr absl::optional<float> kNoOverride = absl::nullopt;
+ constexpr std::optional<float> kNoOverride = std::nullopt;
const auto speech_probability_override =
GetOverrideOrEmpty(kHighSpeechProbability);
@@ -877,7 +875,7 @@
helper.CallProcess(/*num_calls=*/4, speech_probability_override,
GetOverrideOrEmpty(-18.0f));
// The mock `GetRmsErrorDb()` returns false; mimic this by passing
- // absl::nullopt as an override.
+ // std::nullopt as an override.
helper.CallProcess(/*num_calls=*/15, kNoOverride, kNoOverride);
EXPECT_CALL(helper.mock_gain_control, set_compression_gain_db(6))
.WillOnce(Return(0));
@@ -907,7 +905,7 @@
GetOverrideOrEmpty(kHighSpeechProbability),
GetOverrideOrEmpty(kSpeechLevelDbfs));
- absl::optional<int> new_digital_gain =
+ std::optional<int> new_digital_gain =
helper.manager.GetDigitalComressionGain();
if (new_digital_gain) {
helper.mock_gain_control.set_compression_gain_db(*new_digital_gain);
@@ -1225,7 +1223,7 @@
TEST_P(AgcManagerDirectParametrizedTest,
MaxCompressionIsIncreasedAfterClipping) {
- constexpr absl::optional<float> kNoOverride = absl::nullopt;
+ constexpr std::optional<float> kNoOverride = std::nullopt;
const auto speech_probability_override =
GetOverrideOrEmpty(kHighSpeechProbability);
@@ -1248,7 +1246,7 @@
helper.CallProcess(/*num_calls=*/5, speech_probability_override,
GetOverrideOrEmpty(-29.0f));
// The mock `GetRmsErrorDb()` returns false; mimic this by passing
- // absl::nullopt as an override.
+ // std::nullopt as an override.
helper.CallProcess(/*num_calls=*/14, kNoOverride, kNoOverride);
EXPECT_CALL(helper.mock_gain_control, set_compression_gain_db(8))
.WillOnce(Return(0));
@@ -1529,11 +1527,11 @@
// Simulate 4 seconds of clipping; it is expected to trigger a downward
// adjustment of the analog gain.
CallPreProcessAndProcess(/*num_calls=*/400, audio_buffer,
- /*speech_probability_override=*/absl::nullopt,
- /*speech_level_override=*/absl::nullopt, *manager);
+ /*speech_probability_override=*/std::nullopt,
+ /*speech_level_override=*/std::nullopt, *manager);
CallPreProcessAndProcess(/*num_calls=*/400, audio_buffer,
- /*speech_probability_override=*/absl::nullopt,
- /*speech_level_override=*/absl::nullopt,
+ /*speech_probability_override=*/std::nullopt,
+ /*speech_level_override=*/std::nullopt,
*manager_with_override);
// Make sure that an adaptation occurred.
@@ -1591,8 +1589,8 @@
/*speech_probability_level=*/-18.0f, *manager);
CallPreProcessAndProcess(
/*num_calls=*/400, audio_buffer,
- /*speech_probability_override=*/absl::optional<float>(0.7f),
- /*speech_probability_level=*/absl::optional<float>(-18.0f),
+ /*speech_probability_override=*/std::optional<float>(0.7f),
+ /*speech_probability_level=*/std::optional<float>(-18.0f),
*manager_with_override);
// Make sure that an adaptation occurred.
@@ -1652,11 +1650,11 @@
// Simulate 4 seconds of clipping; it is expected to trigger a downward
// adjustment of the analog gain.
CallPreProcessAndProcess(/*num_calls=*/400, audio_buffer,
- /*speech_probability_override=*/absl::nullopt,
- /*speech_level_override=*/absl::nullopt, *manager);
+ /*speech_probability_override=*/std::nullopt,
+ /*speech_level_override=*/std::nullopt, *manager);
CallPreProcessAndProcess(/*num_calls=*/400, audio_buffer,
- /*speech_probability_override=*/absl::nullopt,
- /*speech_level_override=*/absl::nullopt,
+ /*speech_probability_override=*/std::nullopt,
+ /*speech_level_override=*/std::nullopt,
*manager_with_override);
// Make sure that an adaptation occurred.
@@ -1718,12 +1716,12 @@
CallPreProcessAndProcess(
/*num_calls=*/400, audio_buffer,
- /*speech_probability_override=*/absl::optional<float>(0.7f),
- /*speech_level_override=*/absl::optional<float>(-18.0f), *manager);
+ /*speech_probability_override=*/std::optional<float>(0.7f),
+ /*speech_level_override=*/std::optional<float>(-18.0f), *manager);
CallPreProcessAndProcess(
/*num_calls=*/400, audio_buffer,
- /*speech_probability_override=*/absl::optional<float>(0.7f),
- /*speech_level_override=*/absl::optional<float>(-18.0f),
+ /*speech_probability_override=*/std::optional<float>(0.7f),
+ /*speech_level_override=*/std::optional<float>(-18.0f),
*manager_with_override);
// Make sure that an adaptation occurred.
@@ -2059,7 +2057,7 @@
ASSERT_EQ(manager_1.recommended_analog_level(), kAnalogLevel);
ASSERT_EQ(manager_2.recommended_analog_level(), kAnalogLevel);
- reader.Feed(kNumFrames, kGainDb, absl::nullopt, absl::nullopt, manager_1);
+ reader.Feed(kNumFrames, kGainDb, std::nullopt, std::nullopt, manager_1);
reader.Feed(kNumFrames, kGainDb, manager_2);
// Check that the states are the same and adaptation occurs.
diff --git a/modules/audio_processing/agc2/BUILD.gn b/modules/audio_processing/agc2/BUILD.gn
index 73b2beb..5da168a 100644
--- a/modules/audio_processing/agc2/BUILD.gn
+++ b/modules/audio_processing/agc2/BUILD.gn
@@ -80,7 +80,6 @@
"../../../rtc_base:checks",
"../../../rtc_base:safe_compare",
"../../../rtc_base:safe_minmax",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -117,7 +116,6 @@
"../../../rtc_base:checks",
"../../../rtc_base:logging",
"../../../rtc_base:safe_minmax",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -220,7 +218,6 @@
"../../../rtc_base:safe_minmax",
"../../../system_wrappers:field_trial",
"../../../system_wrappers:metrics",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -486,7 +483,6 @@
"../../../rtc_base:stringutils",
"../../../system_wrappers:metrics",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
diff --git a/modules/audio_processing/agc2/clipping_predictor.cc b/modules/audio_processing/agc2/clipping_predictor.cc
index fd759c6..06115a30 100644
--- a/modules/audio_processing/agc2/clipping_predictor.cc
+++ b/modules/audio_processing/agc2/clipping_predictor.cc
@@ -131,11 +131,11 @@
// if at least `GetMinFramesProcessed()` frames have been processed since the
// last reset and a clipping event is predicted. `level`, `min_mic_level`, and
// `max_mic_level` are limited to [0, 255] and `default_step` to [1, 255].
- absl::optional<int> EstimateClippedLevelStep(int channel,
- int level,
- int default_step,
- int min_mic_level,
- int max_mic_level) const {
+ std::optional<int> EstimateClippedLevelStep(int channel,
+ int level,
+ int default_step,
+ int min_mic_level,
+ int max_mic_level) const {
RTC_CHECK_GE(channel, 0);
RTC_CHECK_LT(channel, ch_buffers_.size());
RTC_DCHECK_GE(level, 0);
@@ -147,7 +147,7 @@
RTC_DCHECK_GE(max_mic_level, 0);
RTC_DCHECK_LE(max_mic_level, 255);
if (level <= min_mic_level) {
- return absl::nullopt;
+ return std::nullopt;
}
if (PredictClippingEvent(channel)) {
const int new_level =
@@ -157,7 +157,7 @@
return step;
}
}
- return absl::nullopt;
+ return std::nullopt;
}
private:
@@ -271,11 +271,11 @@
// least `GetMinFramesProcessed()` frames have been processed since the last
// reset and a clipping event is predicted. `level`, `min_mic_level`, and
// `max_mic_level` are limited to [0, 255] and `default_step` to [1, 255].
- absl::optional<int> EstimateClippedLevelStep(int channel,
- int level,
- int default_step,
- int min_mic_level,
- int max_mic_level) const {
+ std::optional<int> EstimateClippedLevelStep(int channel,
+ int level,
+ int default_step,
+ int min_mic_level,
+ int max_mic_level) const {
RTC_DCHECK_GE(channel, 0);
RTC_DCHECK_LT(channel, ch_buffers_.size());
RTC_DCHECK_GE(level, 0);
@@ -287,9 +287,9 @@
RTC_DCHECK_GE(max_mic_level, 0);
RTC_DCHECK_LE(max_mic_level, 255);
if (level <= min_mic_level) {
- return absl::nullopt;
+ return std::nullopt;
}
- absl::optional<float> estimate_db = EstimatePeakValue(channel);
+ std::optional<float> estimate_db = EstimatePeakValue(channel);
if (estimate_db.has_value() && estimate_db.value() > clipping_threshold_) {
int step = 0;
if (!adaptive_step_estimation_) {
@@ -309,7 +309,7 @@
return level - new_level;
}
}
- return absl::nullopt;
+ return std::nullopt;
}
private:
@@ -319,18 +319,18 @@
// Predicts clipping sample peaks based on the processed audio frames.
// Returns the estimated peak value if clipping is predicted. Otherwise
- // returns absl::nullopt.
- absl::optional<float> EstimatePeakValue(int channel) const {
+ // returns std::nullopt.
+ std::optional<float> EstimatePeakValue(int channel) const {
const auto reference_metrics = ch_buffers_[channel]->ComputePartialMetrics(
reference_window_delay_, reference_window_length_);
if (!reference_metrics.has_value()) {
- return absl::nullopt;
+ return std::nullopt;
}
const auto metrics =
ch_buffers_[channel]->ComputePartialMetrics(0, window_length_);
if (!metrics.has_value() ||
!(FloatS16ToDbfs(metrics.value().max) > clipping_threshold_)) {
- return absl::nullopt;
+ return std::nullopt;
}
const float reference_crest_factor =
ComputeCrestFactor(reference_metrics.value());
diff --git a/modules/audio_processing/agc2/clipping_predictor.h b/modules/audio_processing/agc2/clipping_predictor.h
index 5332719..3fd1086 100644
--- a/modules/audio_processing/agc2/clipping_predictor.h
+++ b/modules/audio_processing/agc2/clipping_predictor.h
@@ -12,9 +12,9 @@
#define MODULES_AUDIO_PROCESSING_AGC2_CLIPPING_PREDICTOR_H_
#include <memory>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/audio/audio_processing.h"
#include "modules/audio_processing/include/audio_frame_view.h"
@@ -35,12 +35,12 @@
// Predicts if clipping is going to occur for the specified `channel` in the
// near-future and, if so, it returns a recommended analog mic level decrease
- // step. Returns absl::nullopt if clipping is not predicted.
+ // step. Returns std::nullopt if clipping is not predicted.
// `level` is the current analog mic level, `default_step` is the amount the
// mic level is lowered by the analog controller with every clipping event and
// `min_mic_level` and `max_mic_level` is the range of allowed analog mic
// levels.
- virtual absl::optional<int> EstimateClippedLevelStep(
+ virtual std::optional<int> EstimateClippedLevelStep(
int channel,
int level,
int default_step,
diff --git a/modules/audio_processing/agc2/clipping_predictor_level_buffer.cc b/modules/audio_processing/agc2/clipping_predictor_level_buffer.cc
index fe4cf2a..acc114c 100644
--- a/modules/audio_processing/agc2/clipping_predictor_level_buffer.cc
+++ b/modules/audio_processing/agc2/clipping_predictor_level_buffer.cc
@@ -50,7 +50,7 @@
}
// TODO(bugs.webrtc.org/12774): Optimize partial computation for long buffers.
-absl::optional<ClippingPredictorLevelBuffer::Level>
+std::optional<ClippingPredictorLevelBuffer::Level>
ClippingPredictorLevelBuffer::ComputePartialMetrics(int delay,
int num_items) const {
RTC_DCHECK_GE(delay, 0);
@@ -59,7 +59,7 @@
RTC_DCHECK_LE(num_items, Capacity());
RTC_DCHECK_LE(delay + num_items, Capacity());
if (delay + num_items > Size()) {
- return absl::nullopt;
+ return std::nullopt;
}
float sum = 0.0f;
float max = 0.0f;
@@ -71,7 +71,7 @@
sum += data_[idx].average;
max = std::fmax(data_[idx].max, max);
}
- return absl::optional<Level>({sum / static_cast<float>(num_items), max});
+ return std::optional<Level>({sum / static_cast<float>(num_items), max});
}
} // namespace webrtc
diff --git a/modules/audio_processing/agc2/clipping_predictor_level_buffer.h b/modules/audio_processing/agc2/clipping_predictor_level_buffer.h
index c903277..21e9b46 100644
--- a/modules/audio_processing/agc2/clipping_predictor_level_buffer.h
+++ b/modules/audio_processing/agc2/clipping_predictor_level_buffer.h
@@ -12,10 +12,9 @@
#define MODULES_AUDIO_PROCESSING_AGC2_CLIPPING_PREDICTOR_LEVEL_BUFFER_H_
#include <memory>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
-
namespace webrtc {
// A circular buffer to store frame-wise `Level` items for clipping prediction.
@@ -58,7 +57,7 @@
// from `delay` to `delay` - `num_items` (a delay equal to zero corresponds
// to the most recently pushed item). The value of `delay` is limited to
// [0, N] and `num_items` to [1, M] where N + M is the capacity of the buffer.
- absl::optional<Level> ComputePartialMetrics(int delay, int num_items) const;
+ std::optional<Level> ComputePartialMetrics(int delay, int num_items) const;
private:
int tail_;
diff --git a/modules/audio_processing/agc2/clipping_predictor_level_buffer_unittest.cc b/modules/audio_processing/agc2/clipping_predictor_level_buffer_unittest.cc
index 7af9a43..4f20ddb 100644
--- a/modules/audio_processing/agc2/clipping_predictor_level_buffer_unittest.cc
+++ b/modules/audio_processing/agc2/clipping_predictor_level_buffer_unittest.cc
@@ -108,9 +108,9 @@
buffer.Push({1, 2});
buffer.Push({3, 6});
EXPECT_EQ(buffer.ComputePartialMetrics(/*delay=*/0, /*num_items=*/3),
- absl::nullopt);
+ std::nullopt);
EXPECT_EQ(buffer.ComputePartialMetrics(/*delay=*/2, /*num_items=*/1),
- absl::nullopt);
+ std::nullopt);
}
TEST(ClippingPredictorLevelBufferTest, CheckMetricsAfterReset) {
diff --git a/modules/audio_processing/agc2/clipping_predictor_unittest.cc b/modules/audio_processing/agc2/clipping_predictor_unittest.cc
index af73107..dbb6c23 100644
--- a/modules/audio_processing/agc2/clipping_predictor_unittest.cc
+++ b/modules/audio_processing/agc2/clipping_predictor_unittest.cc
@@ -105,7 +105,7 @@
SCOPED_TRACE(i);
EXPECT_EQ(predictor.EstimateClippedLevelStep(i, level, default_step,
min_mic_level, max_mic_level),
- absl::nullopt);
+ std::nullopt);
}
}
diff --git a/modules/audio_processing/agc2/input_volume_controller.cc b/modules/audio_processing/agc2/input_volume_controller.cc
index bcc650f..fcb8f1d 100644
--- a/modules/audio_processing/agc2/input_volume_controller.cc
+++ b/modules/audio_processing/agc2/input_volume_controller.cc
@@ -173,7 +173,7 @@
// previous update and the ratio of non-silence frames (i.e., frames with a
// `speech_probability` higher than `speech_probability_threshold_`) is at least
// `speech_ratio_threshold_`.
-void MonoInputVolumeController::Process(absl::optional<int> rms_error_db,
+void MonoInputVolumeController::Process(std::optional<int> rms_error_db,
float speech_probability) {
if (check_volume_on_next_process_) {
check_volume_on_next_process_ = false;
@@ -404,7 +404,7 @@
clipping_rate_log_ = 0.0f;
clipping_rate_log_counter_ = 0;
- applied_input_volume_ = absl::nullopt;
+ applied_input_volume_ = std::nullopt;
}
void InputVolumeController::AnalyzeInputAudio(int applied_input_volume,
@@ -498,13 +498,13 @@
AggregateChannelLevels();
}
-absl::optional<int> InputVolumeController::RecommendInputVolume(
+std::optional<int> InputVolumeController::RecommendInputVolume(
float speech_probability,
- absl::optional<float> speech_level_dbfs) {
+ std::optional<float> speech_level_dbfs) {
// Only process if applied input volume is set.
if (!applied_input_volume_.has_value()) {
RTC_LOG(LS_ERROR) << "[AGC2] Applied input volume not set.";
- return absl::nullopt;
+ return std::nullopt;
}
AggregateChannelLevels();
@@ -514,7 +514,7 @@
return applied_input_volume_;
}
- absl::optional<int> rms_error_db;
+ std::optional<int> rms_error_db;
if (speech_level_dbfs.has_value()) {
// Compute the error for all frames (both speech and non-speech frames).
rms_error_db = GetSpeechLevelRmsErrorDb(
@@ -533,7 +533,7 @@
recommended_input_volume_);
}
- applied_input_volume_ = absl::nullopt;
+ applied_input_volume_ = std::nullopt;
return recommended_input_volume();
}
diff --git a/modules/audio_processing/agc2/input_volume_controller.h b/modules/audio_processing/agc2/input_volume_controller.h
index b61f68f..60e76d8 100644
--- a/modules/audio_processing/agc2/input_volume_controller.h
+++ b/modules/audio_processing/agc2/input_volume_controller.h
@@ -12,9 +12,9 @@
#define MODULES_AUDIO_PROCESSING_AGC2_INPUT_VOLUME_CONTROLLER_H_
#include <memory>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/audio/audio_processing.h"
#include "modules/audio_processing/agc2/clipping_predictor.h"
@@ -95,9 +95,9 @@
// suppression are applied. Returns a non-empty input volume recommendation if
// available. If `capture_output_used_` is true, returns the applied input
// volume.
- absl::optional<int> RecommendInputVolume(
+ std::optional<int> RecommendInputVolume(
float speech_probability,
- absl::optional<float> speech_level_dbfs);
+ std::optional<float> speech_level_dbfs);
// Stores whether the capture output will be used or not. Call when the
// capture stream output has been flagged to be used/not-used. If unused, the
@@ -155,7 +155,7 @@
int recommended_input_volume_ = 0;
// Applied input volume. After `SetAppliedInputVolume()` is called it holds
// the current applied volume.
- absl::optional<int> applied_input_volume_;
+ std::optional<int> applied_input_volume_;
bool capture_output_used_;
@@ -213,7 +213,7 @@
// result of `HandleClipping()` and on `rms_error_dbfs`. Updates are only
// allowed for active speech segments and when `rms_error_dbfs` is not empty.
// Must be called after `HandleClipping()`.
- void Process(absl::optional<int> rms_error_dbfs, float speech_probability);
+ void Process(std::optional<int> rms_error_dbfs, float speech_probability);
// Returns the recommended input volume. Must be called after `Process()`.
int recommended_analog_level() const { return recommended_input_volume_; }
diff --git a/modules/audio_processing/agc2/input_volume_controller_unittest.cc b/modules/audio_processing/agc2/input_volume_controller_unittest.cc
index d1bdcf2..eaa8471 100644
--- a/modules/audio_processing/agc2/input_volume_controller_unittest.cc
+++ b/modules/audio_processing/agc2/input_volume_controller_unittest.cc
@@ -156,7 +156,7 @@
int applied_input_volume,
int gain_db,
float speech_probability,
- absl::optional<float> speech_level_dbfs,
+ std::optional<float> speech_level_dbfs,
InputVolumeController& controller) {
RTC_DCHECK(controller.capture_output_used());
@@ -201,7 +201,7 @@
float UpdateRecommendedInputVolume(MonoInputVolumeController& mono_controller,
int applied_input_volume,
float speech_probability,
- absl::optional<float> rms_error_dbfs) {
+ std::optional<float> rms_error_dbfs) {
mono_controller.set_stream_analog_level(applied_input_volume);
EXPECT_EQ(mono_controller.recommended_analog_level(), applied_input_volume);
mono_controller.Process(rms_error_dbfs, speech_probability);
@@ -256,12 +256,12 @@
// - Uses `audio_buffer` to call `AnalyzeInputAudio()` and
// `RecommendInputVolume()`;
// Returns the recommended input volume.
- absl::optional<int> CallAgcSequence(int applied_input_volume,
- float speech_probability,
- absl::optional<float> speech_level_dbfs,
- int num_calls = 1) {
+ std::optional<int> CallAgcSequence(int applied_input_volume,
+ float speech_probability,
+ std::optional<float> speech_level_dbfs,
+ int num_calls = 1) {
RTC_DCHECK_GE(num_calls, 1);
- absl::optional<int> volume = applied_input_volume;
+ std::optional<int> volume = applied_input_volume;
for (int i = 0; i < num_calls; ++i) {
// Repeat the initial volume if `RecommendInputVolume()` doesn't return a
// value.
@@ -285,7 +285,7 @@
int CallRecommendInputVolume(int num_calls,
int initial_volume,
float speech_probability,
- absl::optional<float> speech_level_dbfs) {
+ std::optional<float> speech_level_dbfs) {
RTC_DCHECK(controller.capture_output_used());
// Create non-clipping audio for `AnalyzeInputAudio()`.
@@ -1189,7 +1189,7 @@
constexpr int kGainDb = -20;
SpeechSamplesReader reader;
int volume = reader.Feed(kNumFrames, kInitialInputVolume, kGainDb,
- kLowSpeechProbability, absl::nullopt, controller);
+ kLowSpeechProbability, std::nullopt, controller);
// Check that no adaptation occurs.
ASSERT_EQ(volume, kInitialInputVolume);
@@ -1845,8 +1845,8 @@
EXPECT_EQ(volume_1, kInitialInputVolume);
EXPECT_EQ(volume_2, kInitialInputVolume);
- volume_1 = UpdateRecommendedInputVolume(
- mono_controller_1, volume_1, kHighSpeechProbability, absl::nullopt);
+ volume_1 = UpdateRecommendedInputVolume(mono_controller_1, volume_1,
+ kHighSpeechProbability, std::nullopt);
volume_2 = UpdateRecommendedInputVolume(mono_controller_2, volume_2,
kHighSpeechProbability, -10.0f);
diff --git a/modules/audio_processing/agc2/input_volume_stats_reporter.h b/modules/audio_processing/agc2/input_volume_stats_reporter.h
index 31b1100..2f31aa0 100644
--- a/modules/audio_processing/agc2/input_volume_stats_reporter.h
+++ b/modules/audio_processing/agc2/input_volume_stats_reporter.h
@@ -11,7 +11,8 @@
#ifndef MODULES_AUDIO_PROCESSING_AGC2_INPUT_VOLUME_STATS_REPORTER_H_
#define MODULES_AUDIO_PROCESSING_AGC2_INPUT_VOLUME_STATS_REPORTER_H_
-#include "absl/types/optional.h"
+#include <optional>
+
#include "rtc_base/gtest_prod_util.h"
#include "system_wrappers/include/metrics.h"
@@ -83,7 +84,7 @@
const bool cannot_log_stats_;
int log_volume_update_stats_counter_ = 0;
- absl::optional<int> previous_input_volume_ = absl::nullopt;
+ std::optional<int> previous_input_volume_ = std::nullopt;
};
// Updates the histogram that keeps track of recommended input volume changes
diff --git a/modules/audio_processing/agc2/saturation_protector_buffer.cc b/modules/audio_processing/agc2/saturation_protector_buffer.cc
index 41efdad..1f9083f 100644
--- a/modules/audio_processing/agc2/saturation_protector_buffer.cc
+++ b/modules/audio_processing/agc2/saturation_protector_buffer.cc
@@ -62,9 +62,9 @@
}
}
-absl::optional<float> SaturationProtectorBuffer::Front() const {
+std::optional<float> SaturationProtectorBuffer::Front() const {
if (size_ == 0) {
- return absl::nullopt;
+ return std::nullopt;
}
RTC_DCHECK_LT(FrontIndex(), buffer_.size());
return buffer_[FrontIndex()];
diff --git a/modules/audio_processing/agc2/saturation_protector_buffer.h b/modules/audio_processing/agc2/saturation_protector_buffer.h
index e17d099..3965e93 100644
--- a/modules/audio_processing/agc2/saturation_protector_buffer.h
+++ b/modules/audio_processing/agc2/saturation_protector_buffer.h
@@ -12,8 +12,8 @@
#define MODULES_AUDIO_PROCESSING_AGC2_SATURATION_PROTECTOR_BUFFER_H_
#include <array>
+#include <optional>
-#include "absl/types/optional.h"
#include "modules/audio_processing/agc2/agc2_common.h"
namespace webrtc {
@@ -43,7 +43,7 @@
// Returns the oldest item in the buffer. Returns an empty value if the
// buffer is empty.
- absl::optional<float> Front() const;
+ std::optional<float> Front() const;
private:
int FrontIndex() const;
diff --git a/modules/audio_processing/audio_processing_impl.cc b/modules/audio_processing/audio_processing_impl.cc
index 9a831d9..a1cba51 100644
--- a/modules/audio_processing/audio_processing_impl.cc
+++ b/modules/audio_processing/audio_processing_impl.cc
@@ -14,6 +14,7 @@
#include <cstdint>
#include <cstring>
#include <memory>
+#include <optional>
#include <string>
#include <type_traits>
#include <utility>
@@ -21,7 +22,6 @@
#include "absl/base/nullability.h"
#include "absl/strings/match.h"
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/audio/audio_frame.h"
#include "api/task_queue/task_queue_base.h"
@@ -1400,7 +1400,7 @@
if (submodules_.agc_manager) {
submodules_.agc_manager->Process(*capture_buffer);
- absl::optional<int> new_digital_gain =
+ std::optional<int> new_digital_gain =
submodules_.agc_manager->GetDigitalComressionGain();
if (new_digital_gain && submodules_.gain_control) {
submodules_.gain_control->set_compression_gain_db(*new_digital_gain);
@@ -1725,7 +1725,7 @@
// Invalidate any previously recommended input volume which will be updated by
// `ProcessStream()`.
- capture_.recommended_input_volume = absl::nullopt;
+ capture_.recommended_input_volume = std::nullopt;
if (submodules_.agc_manager) {
submodules_.agc_manager->set_stream_analog_level(level);
@@ -1758,7 +1758,7 @@
if (!capture_.applied_input_volume.has_value()) {
// When `set_stream_analog_level()` is not called, no input level can be
// recommended.
- capture_.recommended_input_volume = absl::nullopt;
+ capture_.recommended_input_volume = std::nullopt;
return;
}
@@ -1890,7 +1890,7 @@
RTC_DCHECK(submodules_.echo_controller);
} else {
EchoCanceller3Config config;
- absl::optional<EchoCanceller3Config> multichannel_config;
+ std::optional<EchoCanceller3Config> multichannel_config;
if (use_setup_specific_default_aec3_config_) {
multichannel_config = EchoCanceller3::CreateDefaultMultichannelConfig();
}
diff --git a/modules/audio_processing/audio_processing_impl.h b/modules/audio_processing/audio_processing_impl.h
index 5388c84..ecdc055 100644
--- a/modules/audio_processing/audio_processing_impl.h
+++ b/modules/audio_processing/audio_processing_impl.h
@@ -16,12 +16,12 @@
#include <atomic>
#include <list>
#include <memory>
+#include <optional>
#include <string>
#include <vector>
#include "absl/base/nullability.h"
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/audio/audio_processing.h"
#include "api/audio/audio_processing_statistics.h"
@@ -440,12 +440,12 @@
AudioProcessingStats stats;
// Input volume applied on the audio input device when the audio is
// acquired. Unspecified when unknown.
- absl::optional<int> applied_input_volume;
+ std::optional<int> applied_input_volume;
bool applied_input_volume_changed;
// Recommended input volume to apply on the audio input device the next time
// that audio is acquired. Unspecified when no input volume can be
// recommended.
- absl::optional<int> recommended_input_volume;
+ std::optional<int> recommended_input_volume;
} capture_ RTC_GUARDED_BY(mutex_capture_);
struct ApmCaptureNonLockedState {
diff --git a/modules/audio_processing/audio_processing_impl_unittest.cc b/modules/audio_processing/audio_processing_impl_unittest.cc
index 82e6550..5b00410 100644
--- a/modules/audio_processing/audio_processing_impl_unittest.cc
+++ b/modules/audio_processing/audio_processing_impl_unittest.cc
@@ -13,9 +13,9 @@
#include <algorithm>
#include <array>
#include <memory>
+#include <optional>
#include <tuple>
-#include "absl/types/optional.h"
#include "api/audio/audio_processing.h"
#include "api/make_ref_counted.h"
#include "api/scoped_refptr.h"
diff --git a/modules/audio_processing/echo_detector/circular_buffer.cc b/modules/audio_processing/echo_detector/circular_buffer.cc
index a6d10ed..c24f920 100644
--- a/modules/audio_processing/echo_detector/circular_buffer.cc
+++ b/modules/audio_processing/echo_detector/circular_buffer.cc
@@ -28,9 +28,9 @@
RTC_DCHECK_LE(nr_elements_in_buffer_, buffer_.size());
}
-absl::optional<float> CircularBuffer::Pop() {
+std::optional<float> CircularBuffer::Pop() {
if (nr_elements_in_buffer_ == 0) {
- return absl::nullopt;
+ return std::nullopt;
}
const size_t index =
(buffer_.size() + next_insertion_index_ - nr_elements_in_buffer_) %
diff --git a/modules/audio_processing/echo_detector/circular_buffer.h b/modules/audio_processing/echo_detector/circular_buffer.h
index db1aeae..5cce0d7 100644
--- a/modules/audio_processing/echo_detector/circular_buffer.h
+++ b/modules/audio_processing/echo_detector/circular_buffer.h
@@ -13,10 +13,9 @@
#include <stddef.h>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
-
namespace webrtc {
// Ring buffer containing floating point values.
@@ -26,7 +25,7 @@
~CircularBuffer();
void Push(float value);
- absl::optional<float> Pop();
+ std::optional<float> Pop();
size_t Size() const { return nr_elements_in_buffer_; }
// This function fills the buffer with zeros, but does not change its size.
void Clear();
diff --git a/modules/audio_processing/echo_detector/circular_buffer_unittest.cc b/modules/audio_processing/echo_detector/circular_buffer_unittest.cc
index 7a234d4..95bea5e 100644
--- a/modules/audio_processing/echo_detector/circular_buffer_unittest.cc
+++ b/modules/audio_processing/echo_detector/circular_buffer_unittest.cc
@@ -47,7 +47,7 @@
TEST(CircularBufferTests, ReadFromEmpty) {
CircularBuffer test_buffer(3);
- EXPECT_EQ(absl::nullopt, test_buffer.Pop());
+ EXPECT_EQ(std::nullopt, test_buffer.Pop());
}
} // namespace webrtc
diff --git a/modules/audio_processing/gain_control_impl.cc b/modules/audio_processing/gain_control_impl.cc
index 2a7f527..56f56b5 100644
--- a/modules/audio_processing/gain_control_impl.cc
+++ b/modules/audio_processing/gain_control_impl.cc
@@ -11,8 +11,8 @@
#include "modules/audio_processing/gain_control_impl.h"
#include <cstdint>
+#include <optional>
-#include "absl/types/optional.h"
#include "api/audio/audio_processing.h"
#include "modules/audio_processing/agc/legacy/gain_control.h"
#include "modules/audio_processing/audio_buffer.h"
diff --git a/modules/audio_processing/gain_control_impl.h b/modules/audio_processing/gain_control_impl.h
index 8aea8f2..1a2cafe 100644
--- a/modules/audio_processing/gain_control_impl.h
+++ b/modules/audio_processing/gain_control_impl.h
@@ -15,9 +15,9 @@
#include <stdint.h>
#include <memory>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "modules/audio_processing/agc/gain_control.h"
@@ -81,8 +81,8 @@
std::vector<std::unique_ptr<MonoAgcState>> mono_agcs_;
std::vector<int> capture_levels_;
- absl::optional<size_t> num_proc_channels_;
- absl::optional<int> sample_rate_hz_;
+ std::optional<size_t> num_proc_channels_;
+ std::optional<int> sample_rate_hz_;
static int instance_counter_;
};
diff --git a/modules/audio_processing/gain_controller2.cc b/modules/audio_processing/gain_controller2.cc
index 9bfae52..b8f72ce 100644
--- a/modules/audio_processing/gain_controller2.cc
+++ b/modules/audio_processing/gain_controller2.cc
@@ -156,7 +156,7 @@
void GainController2::Analyze(int applied_input_volume,
const AudioBuffer& audio_buffer) {
- recommended_input_volume_ = absl::nullopt;
+ recommended_input_volume_ = std::nullopt;
RTC_DCHECK_GE(applied_input_volume, 0);
RTC_DCHECK_LE(applied_input_volume, 255);
@@ -167,10 +167,10 @@
}
}
-void GainController2::Process(absl::optional<float> speech_probability,
+void GainController2::Process(std::optional<float> speech_probability,
bool input_volume_changed,
AudioBuffer* audio) {
- recommended_input_volume_ = absl::nullopt;
+ recommended_input_volume_ = std::nullopt;
data_dumper_.DumpRaw("agc2_applied_input_volume_changed",
input_volume_changed);
@@ -203,13 +203,13 @@
// Compute audio, noise and speech levels.
AudioLevels audio_levels = ComputeAudioLevels(float_frame, data_dumper_);
- absl::optional<float> noise_rms_dbfs;
+ std::optional<float> noise_rms_dbfs;
if (noise_level_estimator_) {
// TODO(bugs.webrtc.org/7494): Pass `audio_levels` to remove duplicated
// computation in `noise_level_estimator_`.
noise_rms_dbfs = noise_level_estimator_->Analyze(float_frame);
}
- absl::optional<SpeechLevel> speech_level;
+ std::optional<SpeechLevel> speech_level;
if (speech_level_estimator_) {
RTC_DCHECK(speech_probability.has_value());
speech_level_estimator_->Update(
@@ -228,8 +228,8 @@
input_volume_controller_->RecommendInputVolume(
*speech_probability,
speech_level->is_confident
- ? absl::optional<float>(speech_level->rms_dbfs)
- : absl::nullopt);
+ ? std::optional<float>(speech_level->rms_dbfs)
+ : std::nullopt);
}
}
diff --git a/modules/audio_processing/gain_controller2.h b/modules/audio_processing/gain_controller2.h
index dbf192b..d192a20 100644
--- a/modules/audio_processing/gain_controller2.h
+++ b/modules/audio_processing/gain_controller2.h
@@ -69,7 +69,7 @@
// Handles input volume changes; if the caller cannot determine whether an
// input volume change occurred, set `input_volume_changed` to false.
// TODO(bugs.webrtc.org/7494): Remove `speech_probability`.
- void Process(absl::optional<float> speech_probability,
+ void Process(std::optional<float> speech_probability,
bool input_volume_changed,
AudioBuffer* audio);
@@ -77,7 +77,7 @@
AvailableCpuFeatures GetCpuFeatures() const { return cpu_features_; }
- absl::optional<int> recommended_input_volume() const {
+ std::optional<int> recommended_input_volume() const {
return recommended_input_volume_;
}
@@ -103,7 +103,7 @@
// Recommended input volume from `InputVolumecontroller`. Non-empty after
// `Process()` if input volume controller is enabled and
// `InputVolumeController::Process()` has returned a non-empty value.
- absl::optional<int> recommended_input_volume_;
+ std::optional<int> recommended_input_volume_;
};
} // namespace webrtc
diff --git a/modules/audio_processing/gain_controller2_unittest.cc b/modules/audio_processing/gain_controller2_unittest.cc
index bccab8d..0af8365 100644
--- a/modules/audio_processing/gain_controller2_unittest.cc
+++ b/modules/audio_processing/gain_controller2_unittest.cc
@@ -57,7 +57,7 @@
const auto applied_volume = agc2.recommended_input_volume();
agc2.Analyze(applied_volume.value_or(applied_initial_volume), ab);
- agc2.Process(/*speech_probability=*/absl::nullopt,
+ agc2.Process(/*speech_probability=*/std::nullopt,
/*input_volume_changed=*/false, &ab);
}
@@ -438,13 +438,13 @@
x *= gain;
}
test::CopyVectorToAudioBuffer(stream_config, frame, &audio_buffer);
- agc2.Process(/*speech_probability=*/absl::nullopt,
+ agc2.Process(/*speech_probability=*/std::nullopt,
/*input_volume_changed=*/false, &audio_buffer);
}
// Estimate the applied gain by processing a probing frame.
SetAudioBufferSamples(/*value=*/1.0f, audio_buffer);
- agc2.Process(/*speech_probability=*/absl::nullopt,
+ agc2.Process(/*speech_probability=*/std::nullopt,
/*input_volume_changed=*/false, &audio_buffer);
const float applied_gain_db =
20.0f * std::log10(audio_buffer.channels_const()[0][0]);
@@ -527,7 +527,7 @@
&audio_buffer);
test::CopyVectorToAudioBuffer(stream_config, frame,
&audio_buffer_reference);
- agc2_reference.Process(/*speech_probability=*/absl::nullopt,
+ agc2_reference.Process(/*speech_probability=*/std::nullopt,
/*input_volume_changed=*/false,
&audio_buffer_reference);
// Check the output buffers.
@@ -592,7 +592,7 @@
}
test::CopyVectorToAudioBuffer(stream_config, frame,
&audio_buffer_reference);
- agc2_reference.Process(absl::nullopt, /*input_volume_changed=*/false,
+ agc2_reference.Process(std::nullopt, /*input_volume_changed=*/false,
&audio_buffer_reference);
test::CopyVectorToAudioBuffer(stream_config, frame, &audio_buffer);
float speech_probability = vad.Analyze(audio_buffer.view());
diff --git a/modules/audio_processing/include/aec_dump.h b/modules/audio_processing/include/aec_dump.h
index f07b911..532fa21 100644
--- a/modules/audio_processing/include/aec_dump.h
+++ b/modules/audio_processing/include/aec_dump.h
@@ -13,10 +13,10 @@
#include <stdint.h>
+#include <optional>
#include <string>
#include "absl/base/attributes.h"
-#include "absl/types/optional.h"
#include "api/audio/audio_processing.h"
#include "modules/audio_processing/include/audio_frame_view.h"
@@ -68,7 +68,7 @@
struct AudioProcessingState {
int delay;
int drift;
- absl::optional<int> applied_input_volume;
+ std::optional<int> applied_input_volume;
bool keypress;
};
diff --git a/modules/audio_processing/logging/apm_data_dumper.cc b/modules/audio_processing/logging/apm_data_dumper.cc
index 65d2167..283f739 100644
--- a/modules/audio_processing/logging/apm_data_dumper.cc
+++ b/modules/audio_processing/logging/apm_data_dumper.cc
@@ -61,7 +61,7 @@
#if WEBRTC_APM_DEBUG_DUMP == 1
bool ApmDataDumper::recording_activated_ = false;
-absl::optional<int> ApmDataDumper::dump_set_to_use_;
+std::optional<int> ApmDataDumper::dump_set_to_use_;
char ApmDataDumper::output_dir_[] = "";
FILE* ApmDataDumper::GetRawFile(absl::string_view name) {
diff --git a/modules/audio_processing/logging/apm_data_dumper.h b/modules/audio_processing/logging/apm_data_dumper.h
index 76f8b34..5186854 100644
--- a/modules/audio_processing/logging/apm_data_dumper.h
+++ b/modules/audio_processing/logging/apm_data_dumper.h
@@ -20,8 +20,9 @@
#include <unordered_map>
#endif
+#include <optional>
+
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/array_view.h"
#if WEBRTC_APM_DEBUG_DUMP == 1
#include "common_audio/wav_file.h"
@@ -391,7 +392,7 @@
private:
#if WEBRTC_APM_DEBUG_DUMP == 1
static bool recording_activated_;
- static absl::optional<int> dump_set_to_use_;
+ static std::optional<int> dump_set_to_use_;
static constexpr size_t kOutputDirMaxLength = 1024;
static char output_dir_[kOutputDirMaxLength];
const int instance_index_;
diff --git a/modules/audio_processing/residual_echo_detector.cc b/modules/audio_processing/residual_echo_detector.cc
index 2a564fc..06ac149 100644
--- a/modules/audio_processing/residual_echo_detector.cc
+++ b/modules/audio_processing/residual_echo_detector.cc
@@ -12,8 +12,8 @@
#include <algorithm>
#include <numeric>
+#include <optional>
-#include "absl/types/optional.h"
#include "modules/audio_processing/logging/apm_data_dumper.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
@@ -89,7 +89,7 @@
}
// Get the next render value.
- const absl::optional<float> buffered_render_power = render_buffer_.Pop();
+ const std::optional<float> buffered_render_power = render_buffer_.Pop();
if (!buffered_render_power) {
// This can happen in a few cases: at the start of a call, due to a glitch
// or due to clock drift. The excess capture value will be ignored.
diff --git a/modules/audio_processing/rms_level.cc b/modules/audio_processing/rms_level.cc
index b0a45cb..1c00e83 100644
--- a/modules/audio_processing/rms_level.cc
+++ b/modules/audio_processing/rms_level.cc
@@ -54,7 +54,7 @@
sum_square_ = 0.f;
sample_count_ = 0;
max_sum_square_ = 0.f;
- block_size_ = absl::nullopt;
+ block_size_ = std::nullopt;
}
void RmsLevel::Analyze(rtc::ArrayView<const int16_t> data) {
@@ -119,7 +119,7 @@
RmsLevel::Levels RmsLevel::AverageAndPeak() {
// Note that block_size_ should by design always be non-empty when
- // sample_count_ != 0. Also, the * operator of absl::optional enforces this
+ // sample_count_ != 0. Also, the * operator of std::optional enforces this
// with a DCHECK.
Levels levels = (sample_count_ == 0)
? Levels{RmsLevel::kMinLevelDb, RmsLevel::kMinLevelDb}
diff --git a/modules/audio_processing/rms_level.h b/modules/audio_processing/rms_level.h
index fbece19..b2a5ba0 100644
--- a/modules/audio_processing/rms_level.h
+++ b/modules/audio_processing/rms_level.h
@@ -14,7 +14,8 @@
#include <stddef.h>
#include <stdint.h>
-#include "absl/types/optional.h"
+#include <optional>
+
#include "api/array_view.h"
namespace webrtc {
@@ -69,7 +70,7 @@
float sum_square_;
size_t sample_count_;
float max_sum_square_;
- absl::optional<size_t> block_size_;
+ std::optional<size_t> block_size_;
};
} // namespace webrtc
diff --git a/modules/audio_processing/test/aec_dump_based_simulator.cc b/modules/audio_processing/test/aec_dump_based_simulator.cc
index 416e287..84ad7de 100644
--- a/modules/audio_processing/test/aec_dump_based_simulator.cc
+++ b/modules/audio_processing/test/aec_dump_based_simulator.cc
@@ -177,8 +177,8 @@
// Set the applied input level if available.
aec_dump_applied_input_level_ =
msg.has_applied_input_volume()
- ? absl::optional<int>(msg.applied_input_volume())
- : absl::nullopt;
+ ? std::optional<int>(msg.applied_input_volume())
+ : std::nullopt;
}
void AecDumpBasedSimulator::VerifyProcessStreamBitExactness(
diff --git a/modules/audio_processing/test/audio_processing_simulator.h b/modules/audio_processing/test/audio_processing_simulator.h
index 082ccb1..314a70d 100644
--- a/modules/audio_processing/test/audio_processing_simulator.h
+++ b/modules/audio_processing/test/audio_processing_simulator.h
@@ -15,9 +15,9 @@
#include <fstream>
#include <limits>
#include <memory>
+#include <optional>
#include <string>
-#include "absl/types/optional.h"
#include "api/audio/audio_processing.h"
#include "common_audio/channel_buffer.h"
#include "common_audio/include/audio_util.h"
@@ -80,79 +80,79 @@
SimulationSettings();
SimulationSettings(const SimulationSettings&);
~SimulationSettings();
- absl::optional<int> stream_delay;
- absl::optional<bool> use_stream_delay;
- absl::optional<int> output_sample_rate_hz;
- absl::optional<int> output_num_channels;
- absl::optional<int> reverse_output_sample_rate_hz;
- absl::optional<int> reverse_output_num_channels;
- absl::optional<std::string> output_filename;
- absl::optional<std::string> reverse_output_filename;
- absl::optional<std::string> input_filename;
- absl::optional<std::string> reverse_input_filename;
- absl::optional<std::string> artificial_nearend_filename;
- absl::optional<std::string> linear_aec_output_filename;
- absl::optional<bool> use_aec;
- absl::optional<bool> use_aecm;
- absl::optional<bool> use_ed; // Residual Echo Detector.
- absl::optional<std::string> ed_graph_output_filename;
- absl::optional<bool> use_agc;
- absl::optional<bool> use_agc2;
- absl::optional<bool> use_pre_amplifier;
- absl::optional<bool> use_capture_level_adjustment;
- absl::optional<bool> use_analog_mic_gain_emulation;
- absl::optional<bool> use_hpf;
- absl::optional<bool> use_ns;
- absl::optional<int> use_ts;
- absl::optional<bool> use_analog_agc;
- absl::optional<bool> use_all;
- absl::optional<bool> analog_agc_use_digital_adaptive_controller;
- absl::optional<int> agc_mode;
- absl::optional<int> agc_target_level;
- absl::optional<bool> use_agc_limiter;
- absl::optional<int> agc_compression_gain;
- absl::optional<bool> agc2_use_adaptive_gain;
- absl::optional<float> agc2_fixed_gain_db;
- absl::optional<bool> agc2_use_input_volume_controller;
- absl::optional<float> pre_amplifier_gain_factor;
- absl::optional<float> pre_gain_factor;
- absl::optional<float> post_gain_factor;
- absl::optional<float> analog_mic_gain_emulation_initial_level;
- absl::optional<int> ns_level;
- absl::optional<bool> ns_analysis_on_linear_aec_output;
- absl::optional<bool> override_key_pressed;
- absl::optional<int> maximum_internal_processing_rate;
+ std::optional<int> stream_delay;
+ std::optional<bool> use_stream_delay;
+ std::optional<int> output_sample_rate_hz;
+ std::optional<int> output_num_channels;
+ std::optional<int> reverse_output_sample_rate_hz;
+ std::optional<int> reverse_output_num_channels;
+ std::optional<std::string> output_filename;
+ std::optional<std::string> reverse_output_filename;
+ std::optional<std::string> input_filename;
+ std::optional<std::string> reverse_input_filename;
+ std::optional<std::string> artificial_nearend_filename;
+ std::optional<std::string> linear_aec_output_filename;
+ std::optional<bool> use_aec;
+ std::optional<bool> use_aecm;
+ std::optional<bool> use_ed; // Residual Echo Detector.
+ std::optional<std::string> ed_graph_output_filename;
+ std::optional<bool> use_agc;
+ std::optional<bool> use_agc2;
+ std::optional<bool> use_pre_amplifier;
+ std::optional<bool> use_capture_level_adjustment;
+ std::optional<bool> use_analog_mic_gain_emulation;
+ std::optional<bool> use_hpf;
+ std::optional<bool> use_ns;
+ std::optional<int> use_ts;
+ std::optional<bool> use_analog_agc;
+ std::optional<bool> use_all;
+ std::optional<bool> analog_agc_use_digital_adaptive_controller;
+ std::optional<int> agc_mode;
+ std::optional<int> agc_target_level;
+ std::optional<bool> use_agc_limiter;
+ std::optional<int> agc_compression_gain;
+ std::optional<bool> agc2_use_adaptive_gain;
+ std::optional<float> agc2_fixed_gain_db;
+ std::optional<bool> agc2_use_input_volume_controller;
+ std::optional<float> pre_amplifier_gain_factor;
+ std::optional<float> pre_gain_factor;
+ std::optional<float> post_gain_factor;
+ std::optional<float> analog_mic_gain_emulation_initial_level;
+ std::optional<int> ns_level;
+ std::optional<bool> ns_analysis_on_linear_aec_output;
+ std::optional<bool> override_key_pressed;
+ std::optional<int> maximum_internal_processing_rate;
int initial_mic_level;
bool simulate_mic_gain = false;
- absl::optional<bool> multi_channel_render;
- absl::optional<bool> multi_channel_capture;
- absl::optional<int> simulated_mic_kind;
- absl::optional<int> frame_for_sending_capture_output_used_false;
- absl::optional<int> frame_for_sending_capture_output_used_true;
+ std::optional<bool> multi_channel_render;
+ std::optional<bool> multi_channel_capture;
+ std::optional<int> simulated_mic_kind;
+ std::optional<int> frame_for_sending_capture_output_used_false;
+ std::optional<int> frame_for_sending_capture_output_used_true;
bool report_performance = false;
- absl::optional<std::string> performance_report_output_filename;
+ std::optional<std::string> performance_report_output_filename;
bool report_bitexactness = false;
bool use_verbose_logging = false;
bool use_quiet_output = false;
bool discard_all_settings_in_aecdump = true;
- absl::optional<std::string> aec_dump_input_filename;
- absl::optional<std::string> aec_dump_output_filename;
+ std::optional<std::string> aec_dump_input_filename;
+ std::optional<std::string> aec_dump_output_filename;
bool fixed_interface = false;
bool store_intermediate_output = false;
bool print_aec_parameter_values = false;
bool dump_internal_data = false;
WavFile::SampleFormat wav_output_format = WavFile::SampleFormat::kInt16;
- absl::optional<std::string> dump_internal_data_output_dir;
- absl::optional<int> dump_set_to_use;
- absl::optional<std::string> call_order_input_filename;
- absl::optional<std::string> call_order_output_filename;
- absl::optional<std::string> aec_settings_filename;
- absl::optional<absl::string_view> aec_dump_input_string;
+ std::optional<std::string> dump_internal_data_output_dir;
+ std::optional<int> dump_set_to_use;
+ std::optional<std::string> call_order_input_filename;
+ std::optional<std::string> call_order_output_filename;
+ std::optional<std::string> aec_settings_filename;
+ std::optional<absl::string_view> aec_dump_input_string;
std::vector<float>* processed_capture_samples = nullptr;
bool analysis_only = false;
- absl::optional<int> dump_start_frame;
- absl::optional<int> dump_end_frame;
- absl::optional<int> init_to_process;
+ std::optional<int> dump_start_frame;
+ std::optional<int> dump_end_frame;
+ std::optional<int> init_to_process;
};
// Provides common functionality for performing audioprocessing simulations.
@@ -220,7 +220,7 @@
Int16Frame rev_frame_;
Int16Frame fwd_frame_;
bool bitexact_output_ = true;
- absl::optional<int> aec_dump_applied_input_level_ = 0;
+ std::optional<int> aec_dump_applied_input_level_ = 0;
protected:
size_t output_reset_counter_ = 0;
diff --git a/modules/audio_processing/test/audioproc_float_impl.cc b/modules/audio_processing/test/audioproc_float_impl.cc
index 5d3c75a..f9b0f4b 100644
--- a/modules/audio_processing/test/audioproc_float_impl.cc
+++ b/modules/audio_processing/test/audioproc_float_impl.cc
@@ -339,19 +339,19 @@
"protobuf debug dump recordings.\n";
void SetSettingIfSpecified(absl::string_view value,
- absl::optional<std::string>* parameter) {
+ std::optional<std::string>* parameter) {
if (value.compare("") != 0) {
*parameter = std::string(value);
}
}
-void SetSettingIfSpecified(int value, absl::optional<int>* parameter) {
+void SetSettingIfSpecified(int value, std::optional<int>* parameter) {
if (value != kParameterNotSpecifiedValue) {
*parameter = value;
}
}
-void SetSettingIfSpecified(float value, absl::optional<float>* parameter) {
+void SetSettingIfSpecified(float value, std::optional<float>* parameter) {
constexpr float kFloatParameterNotSpecifiedValue =
kParameterNotSpecifiedValue;
if (value != kFloatParameterNotSpecifiedValue) {
@@ -359,7 +359,7 @@
}
}
-void SetSettingIfFlagSet(int32_t flag, absl::optional<bool>* parameter) {
+void SetSettingIfFlagSet(int32_t flag, std::optional<bool>* parameter) {
if (flag == 0) {
*parameter = false;
} else if (flag == 1) {
@@ -507,14 +507,14 @@
&settings.dump_end_frame);
constexpr int kFramesPerSecond = 100;
- absl::optional<float> start_seconds;
+ std::optional<float> start_seconds;
SetSettingIfSpecified(absl::GetFlag(FLAGS_dump_start_seconds),
&start_seconds);
if (start_seconds) {
settings.dump_start_frame = *start_seconds * kFramesPerSecond;
}
- absl::optional<float> end_seconds;
+ std::optional<float> end_seconds;
SetSettingIfSpecified(absl::GetFlag(FLAGS_dump_end_seconds), &end_seconds);
if (end_seconds) {
settings.dump_end_frame = *end_seconds * kFramesPerSecond;
diff --git a/modules/audio_processing/test/conversational_speech/BUILD.gn b/modules/audio_processing/test/conversational_speech/BUILD.gn
index 9d387d2..543306e 100644
--- a/modules/audio_processing/test/conversational_speech/BUILD.gn
+++ b/modules/audio_processing/test/conversational_speech/BUILD.gn
@@ -74,6 +74,5 @@
"../../../../test:test_support",
"//testing/gtest",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
diff --git a/modules/audio_processing/test/conversational_speech/generator_unittest.cc b/modules/audio_processing/test/conversational_speech/generator_unittest.cc
index 1771444..bc7fc26 100644
--- a/modules/audio_processing/test/conversational_speech/generator_unittest.cc
+++ b/modules/audio_processing/test/conversational_speech/generator_unittest.cc
@@ -41,10 +41,10 @@
#include <cmath>
#include <map>
#include <memory>
+#include <optional>
#include <vector>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "common_audio/wav_file.h"
#include "modules/audio_processing/test/conversational_speech/config.h"
#include "modules/audio_processing/test/conversational_speech/mock_wavreader_factory.h"
@@ -152,7 +152,7 @@
if (!DirExists(dir)) {
return;
}
- absl::optional<std::vector<std::string>> dir_content = ReadDirectory(dir);
+ std::optional<std::vector<std::string>> dir_content = ReadDirectory(dir);
EXPECT_TRUE(dir_content);
for (const auto& path : *dir_content) {
if (DirExists(path)) {
diff --git a/modules/audio_processing/test/debug_dump_replayer.cc b/modules/audio_processing/test/debug_dump_replayer.cc
index 2f483f5..fb45463 100644
--- a/modules/audio_processing/test/debug_dump_replayer.cc
+++ b/modules/audio_processing/test/debug_dump_replayer.cc
@@ -54,9 +54,9 @@
}
// Get next event that has not run.
-absl::optional<audioproc::Event> DebugDumpReplayer::GetNextEvent() const {
+std::optional<audioproc::Event> DebugDumpReplayer::GetNextEvent() const {
if (!has_next_event_)
- return absl::nullopt;
+ return std::nullopt;
else
return next_event_;
}
diff --git a/modules/audio_processing/test/debug_dump_replayer.h b/modules/audio_processing/test/debug_dump_replayer.h
index 2768f18..efde413 100644
--- a/modules/audio_processing/test/debug_dump_replayer.h
+++ b/modules/audio_processing/test/debug_dump_replayer.h
@@ -32,7 +32,7 @@
bool SetDumpFile(absl::string_view filename);
// Return next event.
- absl::optional<audioproc::Event> GetNextEvent() const;
+ std::optional<audioproc::Event> GetNextEvent() const;
// Run the next event. Returns true if succeeded.
bool RunNextEvent();
diff --git a/modules/audio_processing/test/debug_dump_test.cc b/modules/audio_processing/test/debug_dump_test.cc
index 93f892d..1e6ba11 100644
--- a/modules/audio_processing/test/debug_dump_test.cc
+++ b/modules/audio_processing/test/debug_dump_test.cc
@@ -264,7 +264,7 @@
void DebugDumpTest::VerifyDebugDump(absl::string_view in_filename) {
ASSERT_TRUE(debug_dump_replayer_.SetDumpFile(in_filename));
- while (const absl::optional<audioproc::Event> event =
+ while (const std::optional<audioproc::Event> event =
debug_dump_replayer_.GetNextEvent()) {
debug_dump_replayer_.RunNextEvent();
if (event->type() == audioproc::Event::STREAM) {
@@ -371,7 +371,7 @@
ASSERT_TRUE(debug_dump_replayer_.SetDumpFile(generator.dump_file_name()));
- while (const absl::optional<audioproc::Event> event =
+ while (const std::optional<audioproc::Event> event =
debug_dump_replayer_.GetNextEvent()) {
debug_dump_replayer_.RunNextEvent();
if (event->type() == audioproc::Event::CONFIG) {
@@ -395,7 +395,7 @@
ASSERT_TRUE(debug_dump_replayer_.SetDumpFile(generator.dump_file_name()));
- while (const absl::optional<audioproc::Event> event =
+ while (const std::optional<audioproc::Event> event =
debug_dump_replayer_.GetNextEvent()) {
debug_dump_replayer_.RunNextEvent();
if (event->type() == audioproc::Event::CONFIG) {
@@ -420,7 +420,7 @@
ASSERT_TRUE(debug_dump_replayer_.SetDumpFile(generator.dump_file_name()));
- while (const absl::optional<audioproc::Event> event =
+ while (const std::optional<audioproc::Event> event =
debug_dump_replayer_.GetNextEvent()) {
debug_dump_replayer_.RunNextEvent();
if (event->type() == audioproc::Event::CONFIG) {
@@ -442,7 +442,7 @@
ASSERT_TRUE(debug_dump_replayer_.SetDumpFile(generator.dump_file_name()));
- while (const absl::optional<audioproc::Event> event =
+ while (const std::optional<audioproc::Event> event =
debug_dump_replayer_.GetNextEvent()) {
debug_dump_replayer_.RunNextEvent();
if (event->type() == audioproc::Event::CONFIG) {
diff --git a/modules/audio_processing/test/fake_recording_device.cc b/modules/audio_processing/test/fake_recording_device.cc
index 3fd80b2..476ab8a 100644
--- a/modules/audio_processing/test/fake_recording_device.cc
+++ b/modules/audio_processing/test/fake_recording_device.cc
@@ -12,8 +12,8 @@
#include <algorithm>
#include <memory>
+#include <optional>
-#include "absl/types/optional.h"
#include "modules/audio_processing/agc2/gain_map_internal.h"
#include "rtc_base/logging.h"
#include "rtc_base/numerics/safe_conversions.h"
@@ -45,7 +45,7 @@
// Mic level to simulate.
int mic_level_;
// Optional mic level to undo.
- absl::optional<int> undo_mic_level_;
+ std::optional<int> undo_mic_level_;
};
namespace {
@@ -94,7 +94,7 @@
}
};
-float ComputeAgcLinearFactor(const absl::optional<int>& undo_mic_level,
+float ComputeAgcLinearFactor(const std::optional<int>& undo_mic_level,
int mic_level) {
// If an undo level is specified, virtually restore the unmodified
// microphone level; otherwise simulate the mic gain only.
diff --git a/modules/audio_processing/test/performance_timer.h b/modules/audio_processing/test/performance_timer.h
index 5375ba7..b97a67d 100644
--- a/modules/audio_processing/test/performance_timer.h
+++ b/modules/audio_processing/test/performance_timer.h
@@ -11,9 +11,9 @@
#ifndef MODULES_AUDIO_PROCESSING_TEST_PERFORMANCE_TIMER_H_
#define MODULES_AUDIO_PROCESSING_TEST_PERFORMANCE_TIMER_H_
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "system_wrappers/include/clock.h"
namespace webrtc {
@@ -37,7 +37,7 @@
private:
webrtc::Clock* clock_;
- absl::optional<int64_t> start_timestamp_us_;
+ std::optional<int64_t> start_timestamp_us_;
std::vector<int64_t> timestamps_us_;
};
diff --git a/modules/audio_processing/transient/BUILD.gn b/modules/audio_processing/transient/BUILD.gn
index ac1b13d..75e2761 100644
--- a/modules/audio_processing/transient/BUILD.gn
+++ b/modules/audio_processing/transient/BUILD.gn
@@ -96,7 +96,6 @@
"//testing/gtest",
"//third_party/abseil-cpp/absl/flags:flag",
"//third_party/abseil-cpp/absl/flags:parse",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
}
@@ -125,7 +124,6 @@
"../../../test:test_support",
"//testing/gtest",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
}
diff --git a/modules/audio_processing/transient/transient_suppressor_unittest.cc b/modules/audio_processing/transient/transient_suppressor_unittest.cc
index ab48504..1a368d9 100644
--- a/modules/audio_processing/transient/transient_suppressor_unittest.cc
+++ b/modules/audio_processing/transient/transient_suppressor_unittest.cc
@@ -10,9 +10,9 @@
#include "modules/audio_processing/transient/transient_suppressor.h"
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "modules/audio_processing/transient/common.h"
#include "modules/audio_processing/transient/transient_suppressor_impl.h"
#include "test/gtest.h"
@@ -23,13 +23,13 @@
// Returns the index of the first non-zero sample in `samples` or an unspecified
// value if no value is zero.
-absl::optional<int> FindFirstNonZeroSample(const std::vector<float>& samples) {
+std::optional<int> FindFirstNonZeroSample(const std::vector<float>& samples) {
for (size_t i = 0; i < samples.size(); ++i) {
if (samples[i] != 0.0f) {
return i;
}
}
- return absl::nullopt;
+ return std::nullopt;
}
} // namespace
@@ -138,7 +138,7 @@
/*key_pressed=*/false);
// Detect the algorithmic delay of `TransientSuppressorImpl`.
- absl::optional<int> frame_delay = FindFirstNonZeroSample(frame);
+ std::optional<int> frame_delay = FindFirstNonZeroSample(frame);
// Check that the delayed voice probability is delayed according to the
// measured delay.
diff --git a/modules/congestion_controller/goog_cc/BUILD.gn b/modules/congestion_controller/goog_cc/BUILD.gn
index a152729..f82a54a 100644
--- a/modules/congestion_controller/goog_cc/BUILD.gn
+++ b/modules/congestion_controller/goog_cc/BUILD.gn
@@ -43,7 +43,6 @@
"../../remote_bitrate_estimator",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -55,7 +54,6 @@
deps = [
"../../../api/units:data_rate",
"../../../rtc_base:safe_minmax",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -70,7 +68,6 @@
"../../../api/units:data_size",
"../../../rtc_base:checks",
"../../../rtc_base/experiments:rate_control_settings",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -90,7 +87,6 @@
"../../../rtc_base/experiments:alr_experiment",
"../../../rtc_base/experiments:field_trial_parser",
"../../pacing:interval_budget",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
rtc_library("estimators") {
@@ -130,7 +126,6 @@
"../../../rtc_base/experiments:field_trial_parser",
"../../remote_bitrate_estimator",
"//third_party/abseil-cpp/absl/strings",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -151,7 +146,6 @@
"../../../rtc_base/experiments:field_trial_parser",
"../../remote_bitrate_estimator",
"//third_party/abseil-cpp/absl/algorithm:container",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -195,7 +189,6 @@
"../../../system_wrappers:metrics",
"../../remote_bitrate_estimator",
"//third_party/abseil-cpp/absl/strings",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -227,7 +220,6 @@
"../../../system_wrappers:metrics",
"../../pacing",
"../../remote_bitrate_estimator",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -255,7 +247,6 @@
"../../../system_wrappers:metrics",
"//third_party/abseil-cpp/absl/base:core_headers",
"//third_party/abseil-cpp/absl/strings",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -283,7 +274,6 @@
"../../../test/logging:log_writer",
"../../remote_bitrate_estimator",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
if (!build_with_chromium) {
@@ -347,7 +337,6 @@
"../../pacing",
"//testing/gmock",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
}
diff --git a/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.cc b/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.cc
index 11bae88..4310970 100644
--- a/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.cc
+++ b/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.cc
@@ -12,10 +12,10 @@
#include <algorithm>
#include <memory>
+#include <optional>
#include <utility>
#include <vector>
-#include "absl/types/optional.h"
#include "api/field_trials_view.h"
#include "api/transport/network_types.h"
#include "api/units/data_rate.h"
@@ -56,11 +56,11 @@
}
}
-absl::optional<DataRate> AcknowledgedBitrateEstimator::bitrate() const {
+std::optional<DataRate> AcknowledgedBitrateEstimator::bitrate() const {
return bitrate_estimator_->bitrate();
}
-absl::optional<DataRate> AcknowledgedBitrateEstimator::PeekRate() const {
+std::optional<DataRate> AcknowledgedBitrateEstimator::PeekRate() const {
return bitrate_estimator_->PeekRate();
}
diff --git a/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.h b/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.h
index a31c4df..68766f3 100644
--- a/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.h
+++ b/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.h
@@ -12,9 +12,9 @@
#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_ACKNOWLEDGED_BITRATE_ESTIMATOR_H_
#include <memory>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/field_trials_view.h"
#include "api/transport/network_types.h"
#include "api/units/data_rate.h"
@@ -37,13 +37,13 @@
void IncomingPacketFeedbackVector(
const std::vector<PacketResult>& packet_feedback_vector) override;
- absl::optional<DataRate> bitrate() const override;
- absl::optional<DataRate> PeekRate() const override;
+ std::optional<DataRate> bitrate() const override;
+ std::optional<DataRate> PeekRate() const override;
void SetAlr(bool in_alr) override;
void SetAlrEndedTime(Timestamp alr_ended_time) override;
private:
- absl::optional<Timestamp> alr_ended_time_;
+ std::optional<Timestamp> alr_ended_time_;
bool in_alr_;
std::unique_ptr<BitrateEstimator> bitrate_estimator_;
};
diff --git a/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator_interface.h b/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator_interface.h
index 44e455d..4ff8dc1 100644
--- a/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator_interface.h
+++ b/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator_interface.h
@@ -11,11 +11,10 @@
#ifndef MODULES_CONGESTION_CONTROLLER_GOOG_CC_ACKNOWLEDGED_BITRATE_ESTIMATOR_INTERFACE_H_
#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_ACKNOWLEDGED_BITRATE_ESTIMATOR_INTERFACE_H_
-
#include <memory>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/field_trials_view.h"
#include "api/transport/network_types.h"
#include "api/units/data_rate.h"
@@ -75,8 +74,8 @@
virtual void IncomingPacketFeedbackVector(
const std::vector<PacketResult>& packet_feedback_vector) = 0;
- virtual absl::optional<DataRate> bitrate() const = 0;
- virtual absl::optional<DataRate> PeekRate() const = 0;
+ virtual std::optional<DataRate> bitrate() const = 0;
+ virtual std::optional<DataRate> PeekRate() const = 0;
virtual void SetAlr(bool in_alr) = 0;
virtual void SetAlrEndedTime(Timestamp alr_ended_time) = 0;
};
diff --git a/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator_unittest.cc b/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator_unittest.cc
index 6e28f8e..81fb4fb 100644
--- a/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator_unittest.cc
+++ b/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator_unittest.cc
@@ -13,10 +13,10 @@
#include <cstddef>
#include <cstdint>
#include <memory>
+#include <optional>
#include <utility>
#include <vector>
-#include "absl/types/optional.h"
#include "api/transport/field_trial_based_config.h"
#include "api/transport/network_types.h"
#include "api/units/data_rate.h"
@@ -47,7 +47,7 @@
Update,
(Timestamp at_time, DataSize data_size, bool in_alr),
(override));
- MOCK_METHOD(absl::optional<DataRate>, bitrate, (), (const, override));
+ MOCK_METHOD(std::optional<DataRate>, bitrate, (), (const, override));
MOCK_METHOD(void, ExpectFastRateChange, (), (override));
};
@@ -134,7 +134,7 @@
TEST(TestAcknowledgedBitrateEstimator, ReturnBitrate) {
auto states = CreateTestStates();
- absl::optional<DataRate> return_value = DataRate::KilobitsPerSec(42);
+ std::optional<DataRate> return_value = DataRate::KilobitsPerSec(42);
EXPECT_CALL(*states.mock_bitrate_estimator, bitrate())
.Times(1)
.WillOnce(Return(return_value));
diff --git a/modules/congestion_controller/goog_cc/alr_detector.cc b/modules/congestion_controller/goog_cc/alr_detector.cc
index 33dbdef..6cc7429 100644
--- a/modules/congestion_controller/goog_cc/alr_detector.cc
+++ b/modules/congestion_controller/goog_cc/alr_detector.cc
@@ -13,8 +13,8 @@
#include <cstdint>
#include <cstdio>
#include <memory>
+#include <optional>
-#include "absl/types/optional.h"
#include "api/field_trials_view.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "logging/rtc_event_log/events/rtc_event_alr_state.h"
@@ -28,7 +28,7 @@
namespace {
AlrDetectorConfig GetConfigFromTrials(const FieldTrialsView* key_value_config) {
RTC_CHECK(AlrExperimentSettings::MaxOneFieldTrialEnabled(*key_value_config));
- absl::optional<AlrExperimentSettings> experiment_settings =
+ std::optional<AlrExperimentSettings> experiment_settings =
AlrExperimentSettings::CreateFromFieldTrial(
*key_value_config,
AlrExperimentSettings::kScreenshareProbingBweExperimentName);
@@ -105,7 +105,7 @@
alr_budget_.set_target_rate_kbps(target_rate_kbps);
}
-absl::optional<int64_t> AlrDetector::GetApplicationLimitedRegionStartTime()
+std::optional<int64_t> AlrDetector::GetApplicationLimitedRegionStartTime()
const {
return alr_started_time_ms_;
}
diff --git a/modules/congestion_controller/goog_cc/alr_detector.h b/modules/congestion_controller/goog_cc/alr_detector.h
index bddcab3..e758b26 100644
--- a/modules/congestion_controller/goog_cc/alr_detector.h
+++ b/modules/congestion_controller/goog_cc/alr_detector.h
@@ -15,8 +15,8 @@
#include <stdint.h>
#include <memory>
+#include <optional>
-#include "absl/types/optional.h"
#include "api/field_trials_view.h"
#include "modules/pacing/interval_budget.h"
#include "rtc_base/experiments/struct_parameters_parser.h"
@@ -57,16 +57,16 @@
// Returns time in milliseconds when the current application-limited region
// started or empty result if the sender is currently not application-limited.
- absl::optional<int64_t> GetApplicationLimitedRegionStartTime() const;
+ std::optional<int64_t> GetApplicationLimitedRegionStartTime() const;
private:
friend class GoogCcStatePrinter;
const AlrDetectorConfig conf_;
- absl::optional<int64_t> last_send_time_ms_;
+ std::optional<int64_t> last_send_time_ms_;
IntervalBudget alr_budget_;
- absl::optional<int64_t> alr_started_time_ms_;
+ std::optional<int64_t> alr_started_time_ms_;
RtcEventLog* event_log_;
};
diff --git a/modules/congestion_controller/goog_cc/alr_detector_unittest.cc b/modules/congestion_controller/goog_cc/alr_detector_unittest.cc
index da5852c..57c583d 100644
--- a/modules/congestion_controller/goog_cc/alr_detector_unittest.cc
+++ b/modules/congestion_controller/goog_cc/alr_detector_unittest.cc
@@ -11,8 +11,8 @@
#include "modules/congestion_controller/goog_cc/alr_detector.h"
#include <cstdint>
+#include <optional>
-#include "absl/types/optional.h"
#include "api/transport/field_trial_based_config.h"
#include "rtc_base/checks.h"
#include "rtc_base/experiments/alr_experiment.h"
@@ -68,8 +68,8 @@
}
AlrDetector* const alr_detector_;
int64_t* timestamp_ms_;
- absl::optional<int> interval_ms_;
- absl::optional<int> usage_percentage_;
+ std::optional<int> interval_ms_;
+ std::optional<int> usage_percentage_;
};
} // namespace
@@ -158,7 +158,7 @@
TEST(AlrDetectorTest, ParseControlFieldTrial) {
webrtc::test::ScopedFieldTrials scoped_field_trial(
"WebRTC-ProbingScreenshareBwe/Control/");
- absl::optional<AlrExperimentSettings> parsed_params =
+ std::optional<AlrExperimentSettings> parsed_params =
AlrExperimentSettings::CreateFromFieldTrial(
FieldTrialBasedConfig(), "WebRTC-ProbingScreenshareBwe");
EXPECT_FALSE(static_cast<bool>(parsed_params));
@@ -167,7 +167,7 @@
TEST(AlrDetectorTest, ParseActiveFieldTrial) {
webrtc::test::ScopedFieldTrials scoped_field_trial(
"WebRTC-ProbingScreenshareBwe/1.1,2875,85,20,-20,1/");
- absl::optional<AlrExperimentSettings> parsed_params =
+ std::optional<AlrExperimentSettings> parsed_params =
AlrExperimentSettings::CreateFromFieldTrial(
FieldTrialBasedConfig(), "WebRTC-ProbingScreenshareBwe");
ASSERT_TRUE(static_cast<bool>(parsed_params));
diff --git a/modules/congestion_controller/goog_cc/bitrate_estimator.cc b/modules/congestion_controller/goog_cc/bitrate_estimator.cc
index 3ec9780..9140b8e 100644
--- a/modules/congestion_controller/goog_cc/bitrate_estimator.cc
+++ b/modules/congestion_controller/goog_cc/bitrate_estimator.cc
@@ -13,8 +13,8 @@
#include <algorithm>
#include <cmath>
#include <cstdint>
+#include <optional>
-#include "absl/types/optional.h"
#include "api/field_trials_view.h"
#include "api/units/data_rate.h"
#include "api/units/data_size.h"
@@ -146,16 +146,16 @@
return bitrate_sample;
}
-absl::optional<DataRate> BitrateEstimator::bitrate() const {
+std::optional<DataRate> BitrateEstimator::bitrate() const {
if (bitrate_estimate_kbps_ < 0.f)
- return absl::nullopt;
+ return std::nullopt;
return DataRate::KilobitsPerSec(bitrate_estimate_kbps_);
}
-absl::optional<DataRate> BitrateEstimator::PeekRate() const {
+std::optional<DataRate> BitrateEstimator::PeekRate() const {
if (current_window_ms_ > 0)
return DataSize::Bytes(sum_) / TimeDelta::Millis(current_window_ms_);
- return absl::nullopt;
+ return std::nullopt;
}
void BitrateEstimator::ExpectFastRateChange() {
diff --git a/modules/congestion_controller/goog_cc/bitrate_estimator.h b/modules/congestion_controller/goog_cc/bitrate_estimator.h
index 5ca8234..a808a82 100644
--- a/modules/congestion_controller/goog_cc/bitrate_estimator.h
+++ b/modules/congestion_controller/goog_cc/bitrate_estimator.h
@@ -13,7 +13,8 @@
#include <stdint.h>
-#include "absl/types/optional.h"
+#include <optional>
+
#include "api/field_trials_view.h"
#include "api/units/data_rate.h"
#include "api/units/data_size.h"
@@ -33,8 +34,8 @@
virtual ~BitrateEstimator();
virtual void Update(Timestamp at_time, DataSize amount, bool in_alr);
- virtual absl::optional<DataRate> bitrate() const;
- absl::optional<DataRate> PeekRate() const;
+ virtual std::optional<DataRate> bitrate() const;
+ std::optional<DataRate> PeekRate() const;
virtual void ExpectFastRateChange();
diff --git a/modules/congestion_controller/goog_cc/congestion_window_pushback_controller.h b/modules/congestion_controller/goog_cc/congestion_window_pushback_controller.h
index a344fa1..208331f 100644
--- a/modules/congestion_controller/goog_cc/congestion_window_pushback_controller.h
+++ b/modules/congestion_controller/goog_cc/congestion_window_pushback_controller.h
@@ -13,7 +13,8 @@
#include <stdint.h>
-#include "absl/types/optional.h"
+#include <optional>
+
#include "api/field_trials_view.h"
#include "api/units/data_size.h"
@@ -36,7 +37,7 @@
private:
const bool add_pacing_;
const uint32_t min_pushback_target_bitrate_bps_;
- absl::optional<DataSize> current_data_window_;
+ std::optional<DataSize> current_data_window_;
int64_t outstanding_bytes_ = 0;
int64_t pacing_bytes_ = 0;
double encoding_rate_ratio_ = 1.0;
diff --git a/modules/congestion_controller/goog_cc/delay_based_bwe.cc b/modules/congestion_controller/goog_cc/delay_based_bwe.cc
index 187ef69..e2ab104 100644
--- a/modules/congestion_controller/goog_cc/delay_based_bwe.cc
+++ b/modules/congestion_controller/goog_cc/delay_based_bwe.cc
@@ -13,10 +13,10 @@
#include <algorithm>
#include <cstdint>
#include <memory>
+#include <optional>
#include <utility>
#include <vector>
-#include "absl/types/optional.h"
#include "api/field_trials_view.h"
#include "api/network_state_predictor.h"
#include "api/rtc_event_log/rtc_event_log.h"
@@ -98,9 +98,9 @@
DelayBasedBwe::Result DelayBasedBwe::IncomingPacketFeedbackVector(
const TransportPacketsFeedback& msg,
- absl::optional<DataRate> acked_bitrate,
- absl::optional<DataRate> probe_bitrate,
- absl::optional<NetworkStateEstimate> network_estimate,
+ std::optional<DataRate> acked_bitrate,
+ std::optional<DataRate> probe_bitrate,
+ std::optional<NetworkStateEstimate> network_estimate,
bool in_alr) {
RTC_DCHECK_RUNS_SERIALIZED(&network_race_);
@@ -204,15 +204,15 @@
}
DataRate DelayBasedBwe::TriggerOveruse(Timestamp at_time,
- absl::optional<DataRate> link_capacity) {
+ std::optional<DataRate> link_capacity) {
RateControlInput input(BandwidthUsage::kBwOverusing, link_capacity);
return rate_control_.Update(input, at_time);
}
DelayBasedBwe::Result DelayBasedBwe::MaybeUpdateEstimate(
- absl::optional<DataRate> acked_bitrate,
- absl::optional<DataRate> probe_bitrate,
- absl::optional<NetworkStateEstimate> state_estimate,
+ std::optional<DataRate> acked_bitrate,
+ std::optional<DataRate> probe_bitrate,
+ std::optional<NetworkStateEstimate> state_estimate,
bool recovered_from_overuse,
bool in_alr,
Timestamp at_time) {
@@ -266,7 +266,7 @@
}
bool DelayBasedBwe::UpdateEstimate(Timestamp at_time,
- absl::optional<DataRate> acked_bitrate,
+ std::optional<DataRate> acked_bitrate,
DataRate* target_rate) {
const RateControlInput input(active_delay_detector_->State(), acked_bitrate);
*target_rate = rate_control_.Update(input, at_time);
diff --git a/modules/congestion_controller/goog_cc/delay_based_bwe.h b/modules/congestion_controller/goog_cc/delay_based_bwe.h
index 3bdab4a..50dcae9 100644
--- a/modules/congestion_controller/goog_cc/delay_based_bwe.h
+++ b/modules/congestion_controller/goog_cc/delay_based_bwe.h
@@ -14,9 +14,9 @@
#include <stdint.h>
#include <memory>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/field_trials_view.h"
#include "api/network_state_predictor.h"
#include "api/transport/bandwidth_usage.h"
@@ -74,9 +74,9 @@
Result IncomingPacketFeedbackVector(
const TransportPacketsFeedback& msg,
- absl::optional<DataRate> acked_bitrate,
- absl::optional<DataRate> probe_bitrate,
- absl::optional<NetworkStateEstimate> network_estimate,
+ std::optional<DataRate> acked_bitrate,
+ std::optional<DataRate> probe_bitrate,
+ std::optional<NetworkStateEstimate> network_estimate,
bool in_alr);
void OnRttUpdate(TimeDelta avg_rtt);
bool LatestEstimate(std::vector<uint32_t>* ssrcs, DataRate* bitrate) const;
@@ -84,7 +84,7 @@
void SetMinBitrate(DataRate min_bitrate);
TimeDelta GetExpectedBwePeriod() const;
DataRate TriggerOveruse(Timestamp at_time,
- absl::optional<DataRate> link_capacity);
+ std::optional<DataRate> link_capacity);
DataRate last_estimate() const { return prev_bitrate_; }
BandwidthUsage last_state() const { return prev_state_; }
@@ -92,17 +92,16 @@
friend class GoogCcStatePrinter;
void IncomingPacketFeedback(const PacketResult& packet_feedback,
Timestamp at_time);
- Result MaybeUpdateEstimate(
- absl::optional<DataRate> acked_bitrate,
- absl::optional<DataRate> probe_bitrate,
- absl::optional<NetworkStateEstimate> state_estimate,
- bool recovered_from_overuse,
- bool in_alr,
- Timestamp at_time);
+ Result MaybeUpdateEstimate(std::optional<DataRate> acked_bitrate,
+ std::optional<DataRate> probe_bitrate,
+ std::optional<NetworkStateEstimate> state_estimate,
+ bool recovered_from_overuse,
+ bool in_alr,
+ Timestamp at_time);
// Updates the current remote rate estimate and returns true if a valid
// estimate exists.
bool UpdateEstimate(Timestamp at_time,
- absl::optional<DataRate> acked_bitrate,
+ std::optional<DataRate> acked_bitrate,
DataRate* target_rate);
rtc::RaceChecker network_race_;
diff --git a/modules/congestion_controller/goog_cc/delay_based_bwe_unittest_helper.cc b/modules/congestion_controller/goog_cc/delay_based_bwe_unittest_helper.cc
index e56aa4a..433d3cb 100644
--- a/modules/congestion_controller/goog_cc/delay_based_bwe_unittest_helper.cc
+++ b/modules/congestion_controller/goog_cc/delay_based_bwe_unittest_helper.cc
@@ -13,9 +13,9 @@
#include <cstddef>
#include <cstdint>
#include <memory>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/transport/network_types.h"
#include "api/units/data_rate.h"
#include "api/units/data_size.h"
@@ -220,7 +220,7 @@
bitrate_estimator_->IncomingPacketFeedbackVector(
msg, acknowledged_bitrate_estimator_->bitrate(),
probe_bitrate_estimator_->FetchAndResetLastEstimatedBitrate(),
- /*network_estimate*/ absl::nullopt, /*in_alr*/ false);
+ /*network_estimate*/ std::nullopt, /*in_alr*/ false);
if (result.updated) {
bitrate_observer_.OnReceiveBitrateChanged(result.target_bitrate.bps());
}
@@ -264,7 +264,7 @@
bitrate_estimator_->IncomingPacketFeedbackVector(
msg, acknowledged_bitrate_estimator_->bitrate(),
probe_bitrate_estimator_->FetchAndResetLastEstimatedBitrate(),
- /*network_estimate*/ absl::nullopt, /*in_alr*/ false);
+ /*network_estimate*/ std::nullopt, /*in_alr*/ false);
if (result.updated) {
bitrate_observer_.OnReceiveBitrateChanged(result.target_bitrate.bps());
if (!first_update_ && result.target_bitrate.bps() < bitrate_bps)
diff --git a/modules/congestion_controller/goog_cc/goog_cc_network_control.cc b/modules/congestion_controller/goog_cc/goog_cc_network_control.cc
index ddc4aa2..9b00511 100644
--- a/modules/congestion_controller/goog_cc/goog_cc_network_control.cc
+++ b/modules/congestion_controller/goog_cc/goog_cc_network_control.cc
@@ -16,11 +16,11 @@
#include <cstdint>
#include <memory>
#include <numeric>
+#include <optional>
#include <utility>
#include <vector>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/environment/environment.h"
#include "api/field_trials_view.h"
#include "api/transport/bandwidth_usage.h"
@@ -163,7 +163,7 @@
NetworkControlUpdate GoogCcNetworkController::OnNetworkRouteChange(
NetworkRouteChange msg) {
if (safe_reset_on_route_change_) {
- absl::optional<DataRate> estimated_bitrate;
+ std::optional<DataRate> estimated_bitrate;
if (safe_reset_acknowledged_rate_) {
estimated_bitrate = acknowledged_bitrate_estimator_->bitrate();
if (!estimated_bitrate)
@@ -213,7 +213,7 @@
*initial_config_->stream_based_config
.enable_repeated_initial_probing);
}
- absl::optional<DataRate> total_bitrate =
+ std::optional<DataRate> total_bitrate =
initial_config_->stream_based_config.max_total_allocated_bitrate;
if (total_bitrate) {
auto probes = probe_controller_->OnMaxTotalAllocatedBitrate(
@@ -228,7 +228,7 @@
msg.pacer_queue->bytes());
}
bandwidth_estimation_->UpdateEstimate(msg.at_time);
- absl::optional<int64_t> start_time_ms =
+ std::optional<int64_t> start_time_ms =
alr_detector_->GetApplicationLimitedRegionStartTime();
probe_controller_->SetAlrStartTimeMs(start_time_ms);
@@ -494,7 +494,7 @@
lost_packets_since_last_loss_update_ = 0;
}
}
- absl::optional<int64_t> alr_start_time =
+ std::optional<int64_t> alr_start_time =
alr_detector_->GetApplicationLimitedRegionStartTime();
if (previously_in_alr_ && !alr_start_time.has_value()) {
@@ -519,7 +519,7 @@
network_estimator_->OnTransportPacketsFeedback(report);
SetNetworkStateEstimate(network_estimator_->GetCurrentEstimate());
}
- absl::optional<DataRate> probe_bitrate =
+ std::optional<DataRate> probe_bitrate =
probe_bitrate_estimator_->FetchAndResetLastEstimatedBitrate();
if (ignore_probes_lower_than_network_estimate_ && probe_bitrate &&
estimate_ && *probe_bitrate < delay_based_bwe_->last_estimate() &&
@@ -601,7 +601,7 @@
}
void GoogCcNetworkController::SetNetworkStateEstimate(
- absl::optional<NetworkStateEstimate> estimate) {
+ std::optional<NetworkStateEstimate> estimate) {
auto prev_estimate = estimate_;
estimate_ = estimate;
if (estimate_ && (!prev_estimate ||
diff --git a/modules/congestion_controller/goog_cc/goog_cc_network_control.h b/modules/congestion_controller/goog_cc/goog_cc_network_control.h
index 46ef23f..7e7fe07 100644
--- a/modules/congestion_controller/goog_cc/goog_cc_network_control.h
+++ b/modules/congestion_controller/goog_cc/goog_cc_network_control.h
@@ -15,9 +15,9 @@
#include <deque>
#include <memory>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/environment/environment.h"
#include "api/network_state_predictor.h"
#include "api/transport/network_control.h"
@@ -83,7 +83,7 @@
Timestamp at_time);
void UpdateCongestionWindowSize();
PacerConfig GetPacingRates(Timestamp at_time) const;
- void SetNetworkStateEstimate(absl::optional<NetworkStateEstimate> estimate);
+ void SetNetworkStateEstimate(std::optional<NetworkStateEstimate> estimate);
const Environment env_;
const bool packet_feedback_only_;
@@ -109,16 +109,16 @@
std::unique_ptr<AcknowledgedBitrateEstimatorInterface>
acknowledged_bitrate_estimator_;
- absl::optional<NetworkControllerConfig> initial_config_;
+ std::optional<NetworkControllerConfig> initial_config_;
DataRate min_target_rate_ = DataRate::Zero();
DataRate min_data_rate_ = DataRate::Zero();
DataRate max_data_rate_ = DataRate::PlusInfinity();
- absl::optional<DataRate> starting_rate_;
+ std::optional<DataRate> starting_rate_;
bool first_packet_sent_ = false;
- absl::optional<NetworkStateEstimate> estimate_;
+ std::optional<NetworkStateEstimate> estimate_;
Timestamp next_loss_update_ = Timestamp::MinusInfinity();
int lost_packets_since_last_loss_update_ = 0;
@@ -131,7 +131,7 @@
DataRate last_stable_target_rate_;
LossBasedState last_loss_base_state_;
- absl::optional<uint8_t> last_estimated_fraction_loss_ = 0;
+ std::optional<uint8_t> last_estimated_fraction_loss_ = 0;
TimeDelta last_estimated_round_trip_time_ = TimeDelta::PlusInfinity();
double pacing_factor_;
@@ -140,7 +140,7 @@
bool previously_in_alr_ = false;
- absl::optional<DataSize> current_data_window_;
+ std::optional<DataSize> current_data_window_;
};
} // namespace webrtc
diff --git a/modules/congestion_controller/goog_cc/goog_cc_network_control_unittest.cc b/modules/congestion_controller/goog_cc/goog_cc_network_control_unittest.cc
index aee5bd1..e34dfee 100644
--- a/modules/congestion_controller/goog_cc/goog_cc_network_control_unittest.cc
+++ b/modules/congestion_controller/goog_cc/goog_cc_network_control_unittest.cc
@@ -12,13 +12,13 @@
#include <cstddef>
#include <cstdint>
#include <memory>
+#include <optional>
#include <queue>
#include <string>
#include <utility>
#include <vector>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/environment/environment.h"
#include "api/environment/environment_factory.h"
#include "api/test/network_emulation/create_cross_traffic.h"
@@ -94,9 +94,9 @@
NetworkRouteChange CreateRouteChange(
Timestamp time,
- absl::optional<DataRate> start_rate = absl::nullopt,
- absl::optional<DataRate> min_rate = absl::nullopt,
- absl::optional<DataRate> max_rate = absl::nullopt) {
+ std::optional<DataRate> start_rate = std::nullopt,
+ std::optional<DataRate> min_rate = std::nullopt,
+ std::optional<DataRate> max_rate = std::nullopt) {
NetworkRouteChange route_change;
route_change.at_time = time;
route_change.constraints.at_time = time;
@@ -121,13 +121,13 @@
// Simulate sending packets and receiving transport feedback during
// `runtime_ms`, then return the final target birate.
-absl::optional<DataRate> PacketTransmissionAndFeedbackBlock(
+std::optional<DataRate> PacketTransmissionAndFeedbackBlock(
NetworkControllerInterface* controller,
int64_t runtime_ms,
int64_t delay,
Timestamp& current_time) {
NetworkControlUpdate update;
- absl::optional<DataRate> target_bitrate;
+ std::optional<DataRate> target_bitrate;
int64_t delay_buildup = 0;
int64_t start_time_ms = current_time.ms();
while (current_time.ms() - start_time_ms < runtime_ms) {
@@ -423,14 +423,14 @@
// The test must run and insert packets/feedback long enough that the
// BWE computes a valid estimate. This is first done in an environment which
// simulates no bandwidth limitation, and therefore not built-up delay.
- absl::optional<DataRate> target_bitrate_before_delay =
+ std::optional<DataRate> target_bitrate_before_delay =
PacketTransmissionAndFeedbackBlock(controller.get(), kRunTimeMs, 0,
current_time);
ASSERT_TRUE(target_bitrate_before_delay.has_value());
// Repeat, but this time with a building delay, and make sure that the
// estimation is adjusted downwards.
- absl::optional<DataRate> target_bitrate_after_delay =
+ std::optional<DataRate> target_bitrate_after_delay =
PacketTransmissionAndFeedbackBlock(controller.get(), kRunTimeMs, 50,
current_time);
EXPECT_LT(*target_bitrate_after_delay, *target_bitrate_before_delay);
@@ -1098,7 +1098,7 @@
fixture.CreateController();
Timestamp current_time = Timestamp::Millis(123);
TimeDelta one_way_delay = TimeDelta::Millis(10);
- absl::optional<TimeDelta> rtt = absl::nullopt;
+ std::optional<TimeDelta> rtt = std::nullopt;
TransportPacketsFeedback feedback = CreateTransportPacketsFeedback(
/*per_packet_network_delay=*/TimeDelta::Millis(50), one_way_delay,
diff --git a/modules/congestion_controller/goog_cc/link_capacity_estimator.h b/modules/congestion_controller/goog_cc/link_capacity_estimator.h
index aa23491..5c4cbd4 100644
--- a/modules/congestion_controller/goog_cc/link_capacity_estimator.h
+++ b/modules/congestion_controller/goog_cc/link_capacity_estimator.h
@@ -10,7 +10,8 @@
#ifndef MODULES_CONGESTION_CONTROLLER_GOOG_CC_LINK_CAPACITY_ESTIMATOR_H_
#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_LINK_CAPACITY_ESTIMATOR_H_
-#include "absl/types/optional.h"
+#include <optional>
+
#include "api/units/data_rate.h"
namespace webrtc {
@@ -30,7 +31,7 @@
void Update(DataRate capacity_sample, double alpha);
double deviation_estimate_kbps() const;
- absl::optional<double> estimate_kbps_;
+ std::optional<double> estimate_kbps_;
double deviation_kbps_ = 0.4;
};
} // namespace webrtc
diff --git a/modules/congestion_controller/goog_cc/loss_based_bwe_v2.cc b/modules/congestion_controller/goog_cc/loss_based_bwe_v2.cc
index 80a1888..559f23a 100644
--- a/modules/congestion_controller/goog_cc/loss_based_bwe_v2.cc
+++ b/modules/congestion_controller/goog_cc/loss_based_bwe_v2.cc
@@ -15,10 +15,10 @@
#include <cstddef>
#include <cstdlib>
#include <limits>
+#include <optional>
#include <vector>
#include "absl/algorithm/container.h"
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/field_trials_view.h"
#include "api/transport/network_types.h"
@@ -42,7 +42,7 @@
return datarate.IsFinite();
}
-bool IsValid(absl::optional<DataRate> datarate) {
+bool IsValid(std::optional<DataRate> datarate) {
return datarate.has_value() && IsValid(datarate.value());
}
@@ -396,7 +396,7 @@
// Returns a `LossBasedBweV2::Config` iff the `key_value_config` specifies a
// configuration for the `LossBasedBweV2` which is explicitly enabled.
-absl::optional<LossBasedBweV2::Config> LossBasedBweV2::CreateConfig(
+std::optional<LossBasedBweV2::Config> LossBasedBweV2::CreateConfig(
const FieldTrialsView* key_value_config) {
FieldTrialParameter<bool> enabled("Enabled", true);
FieldTrialParameter<double> bandwidth_rampup_upper_bound_factor(
@@ -514,7 +514,7 @@
}
if (!enabled.Get()) {
- return absl::nullopt;
+ return std::nullopt;
}
Config config;
config.bandwidth_rampup_upper_bound_factor =
diff --git a/modules/congestion_controller/goog_cc/loss_based_bwe_v2.h b/modules/congestion_controller/goog_cc/loss_based_bwe_v2.h
index c855f58..14e9d3d 100644
--- a/modules/congestion_controller/goog_cc/loss_based_bwe_v2.h
+++ b/modules/congestion_controller/goog_cc/loss_based_bwe_v2.h
@@ -12,10 +12,10 @@
#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_LOSS_BASED_BWE_V2_H_
#include <cstdint>
+#include <optional>
#include <unordered_map>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/field_trials_view.h"
#include "api/transport/network_types.h"
@@ -164,7 +164,7 @@
DataRate rate = DataRate::PlusInfinity();
};
- static absl::optional<Config> CreateConfig(
+ static std::optional<Config> CreateConfig(
const FieldTrialsView* key_value_config);
bool IsConfigValid() const;
@@ -200,16 +200,16 @@
bool IsInLossLimitedState() const;
bool CanKeepIncreasingState(DataRate estimate) const;
- absl::optional<DataRate> acknowledged_bitrate_;
- absl::optional<Config> config_;
+ std::optional<DataRate> acknowledged_bitrate_;
+ std::optional<Config> config_;
ChannelParameters current_best_estimate_;
int num_observations_ = 0;
std::vector<Observation> observations_;
PartialObservation partial_observation_;
Timestamp last_send_time_most_recent_observation_ = Timestamp::PlusInfinity();
Timestamp last_time_estimate_reduced_ = Timestamp::MinusInfinity();
- absl::optional<DataRate> cached_instant_upper_bound_;
- absl::optional<DataRate> cached_instant_lower_bound_;
+ std::optional<DataRate> cached_instant_upper_bound_;
+ std::optional<DataRate> cached_instant_lower_bound_;
std::vector<double> instant_upper_bound_temporal_weights_;
std::vector<double> temporal_weights_;
Timestamp recovering_after_loss_timestamp_ = Timestamp::MinusInfinity();
diff --git a/modules/congestion_controller/goog_cc/probe_bitrate_estimator.cc b/modules/congestion_controller/goog_cc/probe_bitrate_estimator.cc
index 99d7d4c..62b6a6b 100644
--- a/modules/congestion_controller/goog_cc/probe_bitrate_estimator.cc
+++ b/modules/congestion_controller/goog_cc/probe_bitrate_estimator.cc
@@ -12,8 +12,8 @@
#include <algorithm>
#include <memory>
+#include <optional>
-#include "absl/types/optional.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/transport/network_types.h"
#include "api/units/data_rate.h"
@@ -64,7 +64,7 @@
ProbeBitrateEstimator::~ProbeBitrateEstimator() = default;
-absl::optional<DataRate> ProbeBitrateEstimator::HandleProbeAndEstimateBitrate(
+std::optional<DataRate> ProbeBitrateEstimator::HandleProbeAndEstimateBitrate(
const PacketResult& packet_feedback) {
int cluster_id = packet_feedback.sent_packet.pacing_info.probe_cluster_id;
RTC_DCHECK_NE(cluster_id, PacedPacketInfo::kNotAProbe);
@@ -103,7 +103,7 @@
packet_feedback.sent_packet.pacing_info.probe_cluster_min_bytes) *
kMinReceivedBytesRatio;
if (cluster->num_probes < min_probes || cluster->size_total < min_size)
- return absl::nullopt;
+ return std::nullopt;
TimeDelta send_interval = cluster->last_send - cluster->first_send;
TimeDelta receive_interval = cluster->last_receive - cluster->first_receive;
@@ -122,7 +122,7 @@
event_log_->Log(std::make_unique<RtcEventProbeResultFailure>(
cluster_id, ProbeFailureReason::kInvalidSendReceiveInterval));
}
- return absl::nullopt;
+ return std::nullopt;
}
// Since the `send_interval` does not include the time it takes to actually
// send the last packet the size of the last sent packet should not be
@@ -159,7 +159,7 @@
event_log_->Log(std::make_unique<RtcEventProbeResultFailure>(
cluster_id, ProbeFailureReason::kInvalidSendReceiveRatio));
}
- return absl::nullopt;
+ return std::nullopt;
}
RTC_LOG(LS_INFO) << "Probing successful"
" [cluster id: "
@@ -187,9 +187,9 @@
return estimated_data_rate_;
}
-absl::optional<DataRate>
+std::optional<DataRate>
ProbeBitrateEstimator::FetchAndResetLastEstimatedBitrate() {
- absl::optional<DataRate> estimated_data_rate = estimated_data_rate_;
+ std::optional<DataRate> estimated_data_rate = estimated_data_rate_;
estimated_data_rate_.reset();
return estimated_data_rate;
}
diff --git a/modules/congestion_controller/goog_cc/probe_bitrate_estimator.h b/modules/congestion_controller/goog_cc/probe_bitrate_estimator.h
index f11aece..fdb5540 100644
--- a/modules/congestion_controller/goog_cc/probe_bitrate_estimator.h
+++ b/modules/congestion_controller/goog_cc/probe_bitrate_estimator.h
@@ -12,8 +12,8 @@
#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_PROBE_BITRATE_ESTIMATOR_H_
#include <map>
+#include <optional>
-#include "absl/types/optional.h"
#include "api/transport/network_types.h"
#include "api/units/data_rate.h"
#include "api/units/data_size.h"
@@ -29,10 +29,10 @@
// Should be called for every probe packet we receive feedback about.
// Returns the estimated bitrate if the probe completes a valid cluster.
- absl::optional<DataRate> HandleProbeAndEstimateBitrate(
+ std::optional<DataRate> HandleProbeAndEstimateBitrate(
const PacketResult& packet_feedback);
- absl::optional<DataRate> FetchAndResetLastEstimatedBitrate();
+ std::optional<DataRate> FetchAndResetLastEstimatedBitrate();
private:
struct AggregatedCluster {
@@ -51,7 +51,7 @@
std::map<int, AggregatedCluster> clusters_;
RtcEventLog* const event_log_;
- absl::optional<DataRate> estimated_data_rate_;
+ std::optional<DataRate> estimated_data_rate_;
};
} // namespace webrtc
diff --git a/modules/congestion_controller/goog_cc/probe_bitrate_estimator_unittest.cc b/modules/congestion_controller/goog_cc/probe_bitrate_estimator_unittest.cc
index efb9725..a8113c9 100644
--- a/modules/congestion_controller/goog_cc/probe_bitrate_estimator_unittest.cc
+++ b/modules/congestion_controller/goog_cc/probe_bitrate_estimator_unittest.cc
@@ -13,8 +13,8 @@
#include <stddef.h>
#include <cstdint>
+#include <optional>
-#include "absl/types/optional.h"
#include "api/transport/network_types.h"
#include "api/units/data_rate.h"
#include "api/units/data_size.h"
@@ -55,7 +55,7 @@
}
protected:
- absl::optional<DataRate> measured_data_rate_;
+ std::optional<DataRate> measured_data_rate_;
ProbeBitrateEstimator probe_bitrate_estimator_;
};
diff --git a/modules/congestion_controller/goog_cc/probe_controller.cc b/modules/congestion_controller/goog_cc/probe_controller.cc
index 44cb0a7..bc56ce0 100644
--- a/modules/congestion_controller/goog_cc/probe_controller.cc
+++ b/modules/congestion_controller/goog_cc/probe_controller.cc
@@ -14,10 +14,10 @@
#include <cstdint>
#include <initializer_list>
#include <memory>
+#include <optional>
#include <vector>
#include "absl/strings/match.h"
-#include "absl/types/optional.h"
#include "api/field_trials_view.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/transport/network_types.h"
@@ -375,11 +375,11 @@
}
void ProbeController::SetAlrStartTimeMs(
- absl::optional<int64_t> alr_start_time_ms) {
+ std::optional<int64_t> alr_start_time_ms) {
if (alr_start_time_ms) {
alr_start_time_ = Timestamp::Millis(*alr_start_time_ms);
} else {
- alr_start_time_ = absl::nullopt;
+ alr_start_time_ = std::nullopt;
}
}
void ProbeController::SetAlrEndedTimeMs(int64_t alr_end_time_ms) {
@@ -433,7 +433,7 @@
min_bitrate_to_probe_further_ = DataRate::PlusInfinity();
time_last_probing_initiated_ = Timestamp::Zero();
estimated_bitrate_ = DataRate::Zero();
- network_estimate_ = absl::nullopt;
+ network_estimate_ = std::nullopt;
start_bitrate_ = DataRate::Zero();
max_bitrate_ = kDefaultMaxProbingBitrate;
Timestamp now = at_time;
diff --git a/modules/congestion_controller/goog_cc/probe_controller.h b/modules/congestion_controller/goog_cc/probe_controller.h
index 96c6091..7d769da 100644
--- a/modules/congestion_controller/goog_cc/probe_controller.h
+++ b/modules/congestion_controller/goog_cc/probe_controller.h
@@ -13,10 +13,10 @@
#include <stdint.h>
+#include <optional>
#include <vector>
#include "absl/base/attributes.h"
-#include "absl/types/optional.h"
#include "api/field_trials_view.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/transport/network_types.h"
@@ -138,7 +138,7 @@
// SetBitrates.
void EnableRepeatedInitialProbing(bool enable);
- void SetAlrStartTimeMs(absl::optional<int64_t> alr_start_time);
+ void SetAlrStartTimeMs(std::optional<int64_t> alr_start_time);
void SetAlrEndedTimeMs(int64_t alr_end_time);
ABSL_MUST_USE_RESULT std::vector<ProbeClusterConfig> RequestProbe(
@@ -187,12 +187,12 @@
DataRate min_bitrate_to_probe_further_ = DataRate::PlusInfinity();
Timestamp time_last_probing_initiated_ = Timestamp::MinusInfinity();
DataRate estimated_bitrate_ = DataRate::Zero();
- absl::optional<webrtc::NetworkStateEstimate> network_estimate_;
+ std::optional<webrtc::NetworkStateEstimate> network_estimate_;
DataRate start_bitrate_ = DataRate::Zero();
DataRate max_bitrate_ = DataRate::PlusInfinity();
Timestamp last_bwe_drop_probing_time_ = Timestamp::Zero();
- absl::optional<Timestamp> alr_start_time_;
- absl::optional<Timestamp> alr_end_time_;
+ std::optional<Timestamp> alr_start_time_;
+ std::optional<Timestamp> alr_end_time_;
bool enable_periodic_alr_probing_;
Timestamp time_of_last_large_drop_ = Timestamp::MinusInfinity();
DataRate bitrate_before_last_large_drop_ = DataRate::Zero();
diff --git a/modules/congestion_controller/goog_cc/probe_controller_unittest.cc b/modules/congestion_controller/goog_cc/probe_controller_unittest.cc
index 5ee7cc7..01d4a6d 100644
--- a/modules/congestion_controller/goog_cc/probe_controller_unittest.cc
+++ b/modules/congestion_controller/goog_cc/probe_controller_unittest.cc
@@ -10,10 +10,10 @@
#include "modules/congestion_controller/goog_cc/probe_controller.h"
#include <memory>
+#include <optional>
#include <vector>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/transport/network_types.h"
#include "api/units/data_rate.h"
#include "api/units/time_delta.h"
@@ -210,7 +210,7 @@
EXPECT_EQ(probes.at(0).target_data_rate, kMaxBitrate);
// Do not probe when not in alr.
- probe_controller->SetAlrStartTimeMs(absl::nullopt);
+ probe_controller->SetAlrStartTimeMs(std::nullopt);
probes = probe_controller->OnMaxTotalAllocatedBitrate(
kMaxBitrate + DataRate::BitsPerSec(2), fixture.CurrentTime());
EXPECT_TRUE(probes.empty());
@@ -520,7 +520,7 @@
DataRate::BitsPerSec(500), BandwidthLimitedCause::kDelayBasedLimited,
fixture.CurrentTime());
- probe_controller->SetAlrStartTimeMs(absl::nullopt);
+ probe_controller->SetAlrStartTimeMs(std::nullopt);
fixture.AdvanceTime(kAlrProbeInterval + TimeDelta::Millis(1));
probes = probe_controller->Process(fixture.CurrentTime());
probes = probe_controller->SetEstimatedBitrate(
@@ -548,7 +548,7 @@
DataRate::BitsPerSec(500), BandwidthLimitedCause::kDelayBasedLimited,
fixture.CurrentTime());
- probe_controller->SetAlrStartTimeMs(absl::nullopt);
+ probe_controller->SetAlrStartTimeMs(std::nullopt);
fixture.AdvanceTime(kAlrProbeInterval + TimeDelta::Millis(1));
probes = probe_controller->Process(fixture.CurrentTime());
probes = probe_controller->SetEstimatedBitrate(
@@ -1066,7 +1066,7 @@
probes = probe_controller->Process(fixture.CurrentTime());
ASSERT_TRUE(probes.empty());
- probe_controller->SetAlrStartTimeMs(absl::nullopt);
+ probe_controller->SetAlrStartTimeMs(std::nullopt);
fixture.AdvanceTime(kAlrProbeInterval + TimeDelta::Millis(1));
probes = probe_controller->Process(fixture.CurrentTime());
EXPECT_TRUE(probes.empty());
diff --git a/modules/congestion_controller/goog_cc/robust_throughput_estimator.cc b/modules/congestion_controller/goog_cc/robust_throughput_estimator.cc
index 5a22910..3aad0c9 100644
--- a/modules/congestion_controller/goog_cc/robust_throughput_estimator.cc
+++ b/modules/congestion_controller/goog_cc/robust_throughput_estimator.cc
@@ -13,10 +13,10 @@
#include <stddef.h>
#include <algorithm>
+#include <optional>
#include <utility>
#include <vector>
-#include "absl/types/optional.h"
#include "api/transport/network_types.h"
#include "api/units/data_rate.h"
#include "api/units/data_size.h"
@@ -101,9 +101,9 @@
}
}
-absl::optional<DataRate> RobustThroughputEstimator::bitrate() const {
+std::optional<DataRate> RobustThroughputEstimator::bitrate() const {
if (window_.empty() || window_.size() < settings_.required_packets)
- return absl::nullopt;
+ return std::nullopt;
TimeDelta largest_recv_gap(TimeDelta::Zero());
TimeDelta second_largest_recv_gap(TimeDelta::Zero());
diff --git a/modules/congestion_controller/goog_cc/robust_throughput_estimator.h b/modules/congestion_controller/goog_cc/robust_throughput_estimator.h
index 9d89856..a1dc139 100644
--- a/modules/congestion_controller/goog_cc/robust_throughput_estimator.h
+++ b/modules/congestion_controller/goog_cc/robust_throughput_estimator.h
@@ -12,9 +12,9 @@
#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_ROBUST_THROUGHPUT_ESTIMATOR_H_
#include <deque>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/transport/network_types.h"
#include "api/units/data_rate.h"
#include "api/units/timestamp.h"
@@ -31,9 +31,9 @@
void IncomingPacketFeedbackVector(
const std::vector<PacketResult>& packet_feedback_vector) override;
- absl::optional<DataRate> bitrate() const override;
+ std::optional<DataRate> bitrate() const override;
- absl::optional<DataRate> PeekRate() const override { return bitrate(); }
+ std::optional<DataRate> PeekRate() const override { return bitrate(); }
void SetAlr(bool /*in_alr*/) override {}
void SetAlrEndedTime(Timestamp /*alr_ended_time*/) override {}
diff --git a/modules/congestion_controller/goog_cc/send_side_bandwidth_estimation.cc b/modules/congestion_controller/goog_cc/send_side_bandwidth_estimation.cc
index c5e1117..4a32609 100644
--- a/modules/congestion_controller/goog_cc/send_side_bandwidth_estimation.cc
+++ b/modules/congestion_controller/goog_cc/send_side_bandwidth_estimation.cc
@@ -16,10 +16,10 @@
#include <cstdio>
#include <limits>
#include <memory>
+#include <optional>
#include <string>
#include <utility>
-#include "absl/types/optional.h"
#include "api/field_trials_view.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/transport/bandwidth_usage.h"
@@ -125,7 +125,7 @@
capacity_estimate_bps_ = start_rate.bps<double>();
}
-void LinkCapacityTracker::OnRateUpdate(absl::optional<DataRate> acknowledged,
+void LinkCapacityTracker::OnRateUpdate(std::optional<DataRate> acknowledged,
DataRate target,
Timestamp at_time) {
if (!acknowledged)
@@ -278,7 +278,7 @@
}
void SendSideBandwidthEstimation::SetBitrates(
- absl::optional<DataRate> send_bitrate,
+ std::optional<DataRate> send_bitrate,
DataRate min_bitrate,
DataRate max_bitrate,
Timestamp at_time) {
@@ -354,7 +354,7 @@
}
void SendSideBandwidthEstimation::SetAcknowledgedRate(
- absl::optional<DataRate> acknowledged_rate,
+ std::optional<DataRate> acknowledged_rate,
Timestamp at_time) {
acknowledged_rate_ = acknowledged_rate;
if (!acknowledged_rate.has_value()) {
@@ -373,7 +373,7 @@
void SendSideBandwidthEstimation::UpdateLossBasedEstimator(
const TransportPacketsFeedback& report,
BandwidthUsage delay_detector_state,
- absl::optional<DataRate> probe_bitrate,
+ std::optional<DataRate> probe_bitrate,
bool in_alr) {
if (LossBasedBandwidthEstimatorV1Enabled()) {
loss_based_bandwidth_estimator_v1_.UpdateLossStatistics(
diff --git a/modules/congestion_controller/goog_cc/send_side_bandwidth_estimation.h b/modules/congestion_controller/goog_cc/send_side_bandwidth_estimation.h
index 3f26fdf..9a56a28 100644
--- a/modules/congestion_controller/goog_cc/send_side_bandwidth_estimation.h
+++ b/modules/congestion_controller/goog_cc/send_side_bandwidth_estimation.h
@@ -17,10 +17,10 @@
#include <deque>
#include <memory>
+#include <optional>
#include <utility>
#include <vector>
-#include "absl/types/optional.h"
#include "api/field_trials_view.h"
#include "api/transport/bandwidth_usage.h"
#include "api/transport/network_types.h"
@@ -43,7 +43,7 @@
void UpdateDelayBasedEstimate(Timestamp at_time,
DataRate delay_based_bitrate);
void OnStartingRate(DataRate start_rate);
- void OnRateUpdate(absl::optional<DataRate> acknowledged,
+ void OnRateUpdate(std::optional<DataRate> acknowledged,
DataRate target,
Timestamp at_time);
void OnRttBackoff(DataRate backoff_rate, Timestamp at_time);
@@ -115,18 +115,18 @@
// Call when we receive a RTCP message with a ReceiveBlock.
void UpdateRtt(TimeDelta rtt, Timestamp at_time);
- void SetBitrates(absl::optional<DataRate> send_bitrate,
+ void SetBitrates(std::optional<DataRate> send_bitrate,
DataRate min_bitrate,
DataRate max_bitrate,
Timestamp at_time);
void SetSendBitrate(DataRate bitrate, Timestamp at_time);
void SetMinMaxBitrate(DataRate min_bitrate, DataRate max_bitrate);
int GetMinBitrate() const;
- void SetAcknowledgedRate(absl::optional<DataRate> acknowledged_rate,
+ void SetAcknowledgedRate(std::optional<DataRate> acknowledged_rate,
Timestamp at_time);
void UpdateLossBasedEstimator(const TransportPacketsFeedback& report,
BandwidthUsage delay_detector_state,
- absl::optional<DataRate> probe_bitrate,
+ std::optional<DataRate> probe_bitrate,
bool in_alr);
bool PaceAtLossBasedEstimate() const;
@@ -178,7 +178,7 @@
int lost_packets_since_last_loss_update_;
int expected_packets_since_last_loss_update_;
- absl::optional<DataRate> acknowledged_rate_;
+ std::optional<DataRate> acknowledged_rate_;
DataRate current_target_;
DataRate last_logged_target_;
DataRate min_bitrate_configured_;
diff --git a/modules/congestion_controller/goog_cc/test/goog_cc_printer.cc b/modules/congestion_controller/goog_cc/test/goog_cc_printer.cc
index 51a67de..669fa4d 100644
--- a/modules/congestion_controller/goog_cc/test/goog_cc_printer.cc
+++ b/modules/congestion_controller/goog_cc/test/goog_cc_printer.cc
@@ -13,11 +13,11 @@
#include <deque>
#include <memory>
+#include <optional>
#include <string>
#include <utility>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/rtc_event_log_output.h"
#include "api/transport/goog_cc_factory.h"
#include "api/transport/network_control.h"
@@ -42,16 +42,16 @@
void WriteTypedValue(RtcEventLogOutput* out, double value) {
LogWriteFormat(out, "%.6f", value);
}
-void WriteTypedValue(RtcEventLogOutput* out, absl::optional<DataRate> value) {
+void WriteTypedValue(RtcEventLogOutput* out, std::optional<DataRate> value) {
LogWriteFormat(out, "%.0f", value ? value->bytes_per_sec<double>() : NAN);
}
-void WriteTypedValue(RtcEventLogOutput* out, absl::optional<DataSize> value) {
+void WriteTypedValue(RtcEventLogOutput* out, std::optional<DataSize> value) {
LogWriteFormat(out, "%.0f", value ? value->bytes<double>() : NAN);
}
-void WriteTypedValue(RtcEventLogOutput* out, absl::optional<TimeDelta> value) {
+void WriteTypedValue(RtcEventLogOutput* out, std::optional<TimeDelta> value) {
LogWriteFormat(out, "%.3f", value ? value->seconds<double>() : NAN);
}
-void WriteTypedValue(RtcEventLogOutput* out, absl::optional<Timestamp> value) {
+void WriteTypedValue(RtcEventLogOutput* out, std::optional<Timestamp> value) {
LogWriteFormat(out, "%.3f", value ? value->seconds<double>() : NAN);
}
diff --git a/modules/congestion_controller/goog_cc/trendline_estimator.cc b/modules/congestion_controller/goog_cc/trendline_estimator.cc
index 309afaa..4fb11b3 100644
--- a/modules/congestion_controller/goog_cc/trendline_estimator.cc
+++ b/modules/congestion_controller/goog_cc/trendline_estimator.cc
@@ -18,11 +18,11 @@
#include <cstdio>
#include <deque>
#include <memory>
+#include <optional>
#include <string>
#include <utility>
#include "absl/strings/match.h"
-#include "absl/types/optional.h"
#include "api/field_trials_view.h"
#include "api/network_state_predictor.h"
#include "api/transport/bandwidth_usage.h"
@@ -57,7 +57,7 @@
return TrendlineEstimatorSettings::kDefaultTrendlineWindowSize;
}
-absl::optional<double> LinearFitSlope(
+std::optional<double> LinearFitSlope(
const std::deque<TrendlineEstimator::PacketTiming>& packets) {
RTC_DCHECK(packets.size() >= 2);
// Compute the "center of mass".
@@ -79,11 +79,11 @@
denominator += (x - x_avg) * (x - x_avg);
}
if (denominator == 0)
- return absl::nullopt;
+ return std::nullopt;
return numerator / denominator;
}
-absl::optional<double> ComputeSlopeCap(
+std::optional<double> ComputeSlopeCap(
const std::deque<TrendlineEstimator::PacketTiming>& packets,
const TrendlineEstimatorSettings& settings) {
RTC_DCHECK(1 <= settings.beginning_packets &&
@@ -104,7 +104,7 @@
late = packets[i];
}
if (late.arrival_time_ms - early.arrival_time_ms < 1) {
- return absl::nullopt;
+ return std::nullopt;
}
return (late.raw_delay_ms - early.raw_delay_ms) /
(late.arrival_time_ms - early.arrival_time_ms) +
@@ -238,7 +238,7 @@
// trend < 0 -> the delay decreases, queues are being emptied
trend = LinearFitSlope(delay_hist_).value_or(trend);
if (settings_.enable_cap) {
- absl::optional<double> cap = ComputeSlopeCap(delay_hist_, settings_);
+ std::optional<double> cap = ComputeSlopeCap(delay_hist_, settings_);
// We only use the cap to filter out overuse detections, not
// to detect additional underuses.
if (trend >= 0 && cap.has_value() && trend > cap.value()) {
diff --git a/modules/congestion_controller/pcc/BUILD.gn b/modules/congestion_controller/pcc/BUILD.gn
index 1ae7523..0af1baf 100644
--- a/modules/congestion_controller/pcc/BUILD.gn
+++ b/modules/congestion_controller/pcc/BUILD.gn
@@ -36,7 +36,6 @@
"../../../api/units:timestamp",
"../../../rtc_base:checks",
"../../../rtc_base:random",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -90,7 +89,6 @@
":utility_function",
"../../../api/transport:network_control",
"../../../api/units:data_rate",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
diff --git a/modules/congestion_controller/pcc/bitrate_controller.cc b/modules/congestion_controller/pcc/bitrate_controller.cc
index 1a9cddb..3c907f7 100644
--- a/modules/congestion_controller/pcc/bitrate_controller.cc
+++ b/modules/congestion_controller/pcc/bitrate_controller.cc
@@ -107,12 +107,11 @@
return rate_change;
}
-absl::optional<DataRate>
-PccBitrateController::ComputeRateUpdateForSlowStartMode(
+std::optional<DataRate> PccBitrateController::ComputeRateUpdateForSlowStartMode(
const PccMonitorInterval& monitor_interval) {
double utility_value = utility_function_->Compute(monitor_interval);
if (previous_utility_.has_value() && utility_value <= previous_utility_) {
- return absl::nullopt;
+ return std::nullopt;
}
previous_utility_ = utility_value;
return monitor_interval.GetTargetSendingRate();
diff --git a/modules/congestion_controller/pcc/bitrate_controller.h b/modules/congestion_controller/pcc/bitrate_controller.h
index fadeea1..a34d429 100644
--- a/modules/congestion_controller/pcc/bitrate_controller.h
+++ b/modules/congestion_controller/pcc/bitrate_controller.h
@@ -14,9 +14,9 @@
#include <stdint.h>
#include <memory>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/units/data_rate.h"
#include "modules/congestion_controller/pcc/monitor_interval.h"
#include "modules/congestion_controller/pcc/utility_function.h"
@@ -42,7 +42,7 @@
double dynamic_boundary_increment,
std::unique_ptr<PccUtilityFunctionInterface> utility_function);
- absl::optional<DataRate> ComputeRateUpdateForSlowStartMode(
+ std::optional<DataRate> ComputeRateUpdateForSlowStartMode(
const PccMonitorInterval& monitor_interval);
DataRate ComputeRateUpdateForOnlineLearningMode(
@@ -65,7 +65,7 @@
int64_t step_size_adjustments_number_;
const double initial_conversion_factor_;
- absl::optional<double> previous_utility_;
+ std::optional<double> previous_utility_;
};
} // namespace pcc
diff --git a/modules/congestion_controller/pcc/bitrate_controller_unittest.cc b/modules/congestion_controller/pcc/bitrate_controller_unittest.cc
index 957d99b..b6e1ee5 100644
--- a/modules/congestion_controller/pcc/bitrate_controller_unittest.cc
+++ b/modules/congestion_controller/pcc/bitrate_controller_unittest.cc
@@ -253,7 +253,7 @@
kTargetSendingRate * 2);
EXPECT_EQ(
bitrate_controller.ComputeRateUpdateForSlowStartMode(monitor_block[0]),
- absl::nullopt);
+ std::nullopt);
}
TEST(PccBitrateControllerTest, StepSizeIncrease) {
diff --git a/modules/congestion_controller/pcc/pcc_network_controller.cc b/modules/congestion_controller/pcc/pcc_network_controller.cc
index 8653470..1705afb 100644
--- a/modules/congestion_controller/pcc/pcc_network_controller.cc
+++ b/modules/congestion_controller/pcc/pcc_network_controller.cc
@@ -11,8 +11,8 @@
#include "modules/congestion_controller/pcc/pcc_network_controller.h"
#include <algorithm>
+#include <optional>
-#include "absl/types/optional.h"
#include "api/units/data_size.h"
#include "rtc_base/checks.h"
diff --git a/modules/congestion_controller/rtp/BUILD.gn b/modules/congestion_controller/rtp/BUILD.gn
index 8435ddb..531a026 100644
--- a/modules/congestion_controller/rtp/BUILD.gn
+++ b/modules/congestion_controller/rtp/BUILD.gn
@@ -27,7 +27,6 @@
"../../../rtc_base:safe_minmax",
"../../../rtc_base/system:no_unique_address",
"../../pacing",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
rtc_library("transport_feedback") {
diff --git a/modules/congestion_controller/rtp/control_handler.cc b/modules/congestion_controller/rtp/control_handler.cc
index 357a0f0..93e18a8 100644
--- a/modules/congestion_controller/rtp/control_handler.cc
+++ b/modules/congestion_controller/rtp/control_handler.cc
@@ -38,10 +38,10 @@
pacer_expected_queue_ms_ = expected_queue_time.ms();
}
-absl::optional<TargetTransferRate> CongestionControlHandler::GetUpdate() {
+std::optional<TargetTransferRate> CongestionControlHandler::GetUpdate() {
RTC_DCHECK_RUN_ON(&sequenced_checker_);
if (!last_incoming_.has_value())
- return absl::nullopt;
+ return std::nullopt;
TargetTransferRate new_outgoing = *last_incoming_;
DataRate log_target_rate = new_outgoing.target_rate;
bool pause_encoding = false;
@@ -67,7 +67,7 @@
last_reported_ = new_outgoing;
return new_outgoing;
}
- return absl::nullopt;
+ return std::nullopt;
}
} // namespace webrtc
diff --git a/modules/congestion_controller/rtp/control_handler.h b/modules/congestion_controller/rtp/control_handler.h
index 649d2f9..f492199 100644
--- a/modules/congestion_controller/rtp/control_handler.h
+++ b/modules/congestion_controller/rtp/control_handler.h
@@ -13,7 +13,8 @@
#include <stdint.h>
-#include "absl/types/optional.h"
+#include <optional>
+
#include "api/sequence_checker.h"
#include "api/transport/network_types.h"
#include "api/units/data_size.h"
@@ -38,11 +39,11 @@
void SetTargetRate(TargetTransferRate new_target_rate);
void SetNetworkAvailability(bool network_available);
void SetPacerQueue(TimeDelta expected_queue_time);
- absl::optional<TargetTransferRate> GetUpdate();
+ std::optional<TargetTransferRate> GetUpdate();
private:
- absl::optional<TargetTransferRate> last_incoming_;
- absl::optional<TargetTransferRate> last_reported_;
+ std::optional<TargetTransferRate> last_incoming_;
+ std::optional<TargetTransferRate> last_reported_;
bool network_available_ = true;
bool encoder_paused_in_last_report_ = false;
diff --git a/modules/congestion_controller/rtp/transport_feedback_adapter.cc b/modules/congestion_controller/rtp/transport_feedback_adapter.cc
index 081e016..2f5e6a7 100644
--- a/modules/congestion_controller/rtp/transport_feedback_adapter.cc
+++ b/modules/congestion_controller/rtp/transport_feedback_adapter.cc
@@ -108,7 +108,7 @@
history_.insert(std::make_pair(packet.sent.sequence_number, packet));
}
-absl::optional<SentPacket> TransportFeedbackAdapter::ProcessSentPacket(
+std::optional<SentPacket> TransportFeedbackAdapter::ProcessSentPacket(
const rtc::SentPacket& sent_packet) {
auto send_time = Timestamp::Millis(sent_packet.send_time_ms);
// TODO(srte): Only use one way to indicate that packet feedback is used.
@@ -144,16 +144,16 @@
DataSize::Bytes(sent_packet.info.packet_size_bytes);
last_untracked_send_time_ = std::max(last_untracked_send_time_, send_time);
}
- return absl::nullopt;
+ return std::nullopt;
}
-absl::optional<TransportPacketsFeedback>
+std::optional<TransportPacketsFeedback>
TransportFeedbackAdapter::ProcessTransportFeedback(
const rtcp::TransportFeedback& feedback,
Timestamp feedback_receive_time) {
if (feedback.GetPacketStatusCount() == 0) {
RTC_LOG(LS_INFO) << "Empty transport feedback packet received.";
- return absl::nullopt;
+ return std::nullopt;
}
TransportPacketsFeedback msg;
@@ -161,7 +161,7 @@
msg.packet_feedbacks =
ProcessTransportFeedbackInner(feedback, feedback_receive_time);
if (msg.packet_feedbacks.empty())
- return absl::nullopt;
+ return std::nullopt;
msg.data_in_flight = in_flight_.GetOutstandingData(network_route_);
return msg;
diff --git a/modules/congestion_controller/rtp/transport_feedback_adapter.h b/modules/congestion_controller/rtp/transport_feedback_adapter.h
index 7b1243b..fb48b17 100644
--- a/modules/congestion_controller/rtp/transport_feedback_adapter.h
+++ b/modules/congestion_controller/rtp/transport_feedback_adapter.h
@@ -62,10 +62,10 @@
void AddPacket(const RtpPacketSendInfo& packet_info,
size_t overhead_bytes,
Timestamp creation_time);
- absl::optional<SentPacket> ProcessSentPacket(
+ std::optional<SentPacket> ProcessSentPacket(
const rtc::SentPacket& sent_packet);
- absl::optional<TransportPacketsFeedback> ProcessTransportFeedback(
+ std::optional<TransportPacketsFeedback> ProcessTransportFeedback(
const rtcp::TransportFeedback& feedback,
Timestamp feedback_receive_time);
diff --git a/modules/congestion_controller/rtp/transport_feedback_adapter_unittest.cc b/modules/congestion_controller/rtp/transport_feedback_adapter_unittest.cc
index 5aad74c..c6a6b6d 100644
--- a/modules/congestion_controller/rtp/transport_feedback_adapter_unittest.cc
+++ b/modules/congestion_controller/rtp/transport_feedback_adapter_unittest.cc
@@ -385,14 +385,14 @@
packet_info.pacing_info = packet.sent_packet.pacing_info;
packet_info.packet_type = RtpPacketMediaType::kVideo;
adapter_->AddPacket(packet_info, 0u, clock_.CurrentTime());
- absl::optional<SentPacket> sent_packet = adapter_->ProcessSentPacket(
+ std::optional<SentPacket> sent_packet = adapter_->ProcessSentPacket(
rtc::SentPacket(packet.sent_packet.sequence_number,
packet.sent_packet.send_time.ms(), rtc::PacketInfo()));
EXPECT_TRUE(sent_packet.has_value());
// Call ProcessSentPacket() again with the same sequence number. This packet
// has already been marked as sent and the call should be ignored.
- absl::optional<SentPacket> duplicate_packet = adapter_->ProcessSentPacket(
+ std::optional<SentPacket> duplicate_packet = adapter_->ProcessSentPacket(
rtc::SentPacket(packet.sent_packet.sequence_number,
packet.sent_packet.send_time.ms(), rtc::PacketInfo()));
EXPECT_FALSE(duplicate_packet.has_value());
diff --git a/modules/desktop_capture/BUILD.gn b/modules/desktop_capture/BUILD.gn
index ddbf3c3..4ea31b9 100644
--- a/modules/desktop_capture/BUILD.gn
+++ b/modules/desktop_capture/BUILD.gn
@@ -280,7 +280,6 @@
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
public_configs = [ ":x11_config" ]
sources = [
diff --git a/modules/desktop_capture/linux/wayland/egl_dmabuf.cc b/modules/desktop_capture/linux/wayland/egl_dmabuf.cc
index b529077..fff9764 100644
--- a/modules/desktop_capture/linux/wayland/egl_dmabuf.cc
+++ b/modules/desktop_capture/linux/wayland/egl_dmabuf.cc
@@ -19,8 +19,9 @@
#include <unistd.h>
#include <xf86drm.h>
+#include <optional>
+
#include "absl/memory/memory.h"
-#include "absl/types/optional.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/sanitizer.h"
@@ -369,7 +370,7 @@
RTC_LOG(LS_ERROR) << "Failed to obtain default EGL display: "
<< FormatEGLError(EglGetError()) << "\n"
<< "Defaulting to using first available render node";
- absl::optional<std::string> render_node = GetRenderNode();
+ std::optional<std::string> render_node = GetRenderNode();
if (!render_node) {
return;
}
@@ -724,19 +725,19 @@
return modifiers;
}
-absl::optional<std::string> EglDmaBuf::GetRenderNode() {
+std::optional<std::string> EglDmaBuf::GetRenderNode() {
int max_devices = drmGetDevices2(0, nullptr, 0);
if (max_devices <= 0) {
RTC_LOG(LS_ERROR) << "drmGetDevices2() has not found any devices (errno="
<< -max_devices << ")";
- return absl::nullopt;
+ return std::nullopt;
}
std::vector<drmDevicePtr> devices(max_devices);
int ret = drmGetDevices2(0, devices.data(), max_devices);
if (ret < 0) {
RTC_LOG(LS_ERROR) << "drmGetDevices2() returned an error " << ret;
- return absl::nullopt;
+ return std::nullopt;
}
std::string render_node;
diff --git a/modules/desktop_capture/linux/wayland/egl_dmabuf.h b/modules/desktop_capture/linux/wayland/egl_dmabuf.h
index 22a8f5a..61dd2b1 100644
--- a/modules/desktop_capture/linux/wayland/egl_dmabuf.h
+++ b/modules/desktop_capture/linux/wayland/egl_dmabuf.h
@@ -16,10 +16,10 @@
#include <gbm.h>
#include <memory>
+#include <optional>
#include <string>
#include <vector>
-#include "absl/types/optional.h"
#include "modules/desktop_capture/desktop_geometry.h"
namespace webrtc {
@@ -66,7 +66,7 @@
GLuint texture_ = 0;
EGLStruct egl_;
- absl::optional<std::string> GetRenderNode();
+ std::optional<std::string> GetRenderNode();
};
} // namespace webrtc
diff --git a/modules/desktop_capture/linux/wayland/mouse_cursor_monitor_pipewire.cc b/modules/desktop_capture/linux/wayland/mouse_cursor_monitor_pipewire.cc
index 00b07f3..21f550b 100644
--- a/modules/desktop_capture/linux/wayland/mouse_cursor_monitor_pipewire.cc
+++ b/modules/desktop_capture/linux/wayland/mouse_cursor_monitor_pipewire.cc
@@ -40,7 +40,7 @@
RTC_DCHECK_RUN_ON(&sequence_checker_);
RTC_DCHECK(callback_);
- absl::optional<DesktopVector> mouse_cursor_position =
+ std::optional<DesktopVector> mouse_cursor_position =
options_.screencast_stream()->CaptureCursorPosition();
// Invalid cursor or position
if (!mouse_cursor_position) {
diff --git a/modules/desktop_capture/linux/wayland/screencast_stream_utils.cc b/modules/desktop_capture/linux/wayland/screencast_stream_utils.cc
index 8177cfd..b0bac69 100644
--- a/modules/desktop_capture/linux/wayland/screencast_stream_utils.cc
+++ b/modules/desktop_capture/linux/wayland/screencast_stream_utils.cc
@@ -34,9 +34,9 @@
return {};
}
- absl::optional<int> major = rtc::StringToNumber<int>(parsed_version.at(0));
- absl::optional<int> minor = rtc::StringToNumber<int>(parsed_version.at(1));
- absl::optional<int> micro = rtc::StringToNumber<int>(parsed_version.at(2));
+ std::optional<int> major = rtc::StringToNumber<int>(parsed_version.at(0));
+ std::optional<int> minor = rtc::StringToNumber<int>(parsed_version.at(1));
+ std::optional<int> micro = rtc::StringToNumber<int>(parsed_version.at(2));
// Return invalid version if we failed to parse it
if (!major || !minor || !micro) {
diff --git a/modules/desktop_capture/linux/wayland/shared_screencast_stream.cc b/modules/desktop_capture/linux/wayland/shared_screencast_stream.cc
index 047ad5a..9aa50e1 100644
--- a/modules/desktop_capture/linux/wayland/shared_screencast_stream.cc
+++ b/modules/desktop_capture/linux/wayland/shared_screencast_stream.cc
@@ -1034,12 +1034,12 @@
return private_->CaptureCursor();
}
-absl::optional<DesktopVector> SharedScreenCastStream::CaptureCursorPosition() {
+std::optional<DesktopVector> SharedScreenCastStream::CaptureCursorPosition() {
DesktopVector position = private_->CaptureCursorPosition();
// Consider only (x >= 0 and y >= 0) a valid position
if (position.x() < 0 || position.y() < 0) {
- return absl::nullopt;
+ return std::nullopt;
}
return position;
diff --git a/modules/desktop_capture/linux/wayland/shared_screencast_stream.h b/modules/desktop_capture/linux/wayland/shared_screencast_stream.h
index 1b00e27..1784ad3 100644
--- a/modules/desktop_capture/linux/wayland/shared_screencast_stream.h
+++ b/modules/desktop_capture/linux/wayland/shared_screencast_stream.h
@@ -12,8 +12,8 @@
#define MODULES_DESKTOP_CAPTURE_LINUX_WAYLAND_SHARED_SCREENCAST_STREAM_H_
#include <memory>
+#include <optional>
-#include "absl/types/optional.h"
#include "api/ref_counted_base.h"
#include "api/scoped_refptr.h"
#include "modules/desktop_capture/desktop_capturer.h"
@@ -81,7 +81,7 @@
// Returns the most recent mouse cursor position. Will not return a value in
// case we didn't manage to get it from PipeWire buffer.
- absl::optional<DesktopVector> CaptureCursorPosition();
+ std::optional<DesktopVector> CaptureCursorPosition();
~SharedScreenCastStream();
diff --git a/modules/desktop_capture/win/wgc_capture_source.h b/modules/desktop_capture/win/wgc_capture_source.h
index 57b105c..f155cdd 100644
--- a/modules/desktop_capture/win/wgc_capture_source.h
+++ b/modules/desktop_capture/win/wgc_capture_source.h
@@ -16,8 +16,8 @@
#include <wrl/client.h>
#include <memory>
+#include <optional>
-#include "absl/types/optional.h"
#include "modules/desktop_capture/desktop_capturer.h"
#include "modules/desktop_capture/desktop_geometry.h"
@@ -138,7 +138,7 @@
// device index as it's SourceId. However, WGC requires we use an HMONITOR to
// describe which screen to capture. So, we internally convert the supplied
// device index into an HMONITOR when `IsCapturable()` is called.
- absl::optional<HMONITOR> hmonitor_;
+ std::optional<HMONITOR> hmonitor_;
};
} // namespace webrtc
diff --git a/modules/include/module_common_types_public.h b/modules/include/module_common_types_public.h
index 5a6f634..289b5d0 100644
--- a/modules/include/module_common_types_public.h
+++ b/modules/include/module_common_types_public.h
@@ -12,8 +12,7 @@
#define MODULES_INCLUDE_MODULE_COMMON_TYPES_PUBLIC_H_
#include <limits>
-
-#include "absl/types/optional.h"
+#include <optional>
namespace webrtc {
diff --git a/modules/pacing/BUILD.gn b/modules/pacing/BUILD.gn
index b560557..7878676 100644
--- a/modules/pacing/BUILD.gn
+++ b/modules/pacing/BUILD.gn
@@ -64,7 +64,6 @@
"//third_party/abseil-cpp/absl/container:inlined_vector",
"//third_party/abseil-cpp/absl/functional:any_invocable",
"//third_party/abseil-cpp/absl/strings",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
diff --git a/modules/pacing/bitrate_prober.cc b/modules/pacing/bitrate_prober.cc
index 29b4f4c..5b6142a 100644
--- a/modules/pacing/bitrate_prober.cc
+++ b/modules/pacing/bitrate_prober.cc
@@ -139,9 +139,9 @@
return next_probe_time_;
}
-absl::optional<PacedPacketInfo> BitrateProber::CurrentCluster(Timestamp now) {
+std::optional<PacedPacketInfo> BitrateProber::CurrentCluster(Timestamp now) {
if (clusters_.empty() || probing_state_ != ProbingState::kActive) {
- return absl::nullopt;
+ return std::nullopt;
}
if (next_probe_time_.IsFinite() &&
@@ -153,7 +153,7 @@
clusters_.pop();
if (clusters_.empty()) {
probing_state_ = ProbingState::kInactive;
- return absl::nullopt;
+ return std::nullopt;
}
}
diff --git a/modules/pacing/bitrate_prober.h b/modules/pacing/bitrate_prober.h
index 149d24f..7ceb129 100644
--- a/modules/pacing/bitrate_prober.h
+++ b/modules/pacing/bitrate_prober.h
@@ -72,7 +72,7 @@
Timestamp NextProbeTime(Timestamp now) const;
// Information about the current probing cluster.
- absl::optional<PacedPacketInfo> CurrentCluster(Timestamp now);
+ std::optional<PacedPacketInfo> CurrentCluster(Timestamp now);
// Returns the minimum number of bytes that the prober recommends for
// the next probe, or zero if not probing. A probe can consist of multiple
diff --git a/modules/pacing/pacing_controller.cc b/modules/pacing/pacing_controller.cc
index a45c5d8..5999fa5 100644
--- a/modules/pacing/pacing_controller.cc
+++ b/modules/pacing/pacing_controller.cc
@@ -204,7 +204,7 @@
// queue). Flush any pending packets currently in the queue for that stream
// in order to get the new keyframe out as quickly as possible.
packet_queue_.RemovePacketsForSsrc(packet->Ssrc());
- absl::optional<uint32_t> rtx_ssrc =
+ std::optional<uint32_t> rtx_ssrc =
packet_sender_->GetRtxSsrcForMedia(packet->Ssrc());
if (rtx_ssrc) {
packet_queue_.RemovePacketsForSsrc(*rtx_ssrc);
@@ -283,7 +283,7 @@
return std::max(media_debt_, padding_debt_);
}
-absl::optional<Timestamp> PacingController::FirstSentPacketTime() const {
+std::optional<Timestamp> PacingController::FirstSentPacketTime() const {
return first_sent_packet_time_;
}
diff --git a/modules/pacing/pacing_controller.h b/modules/pacing/pacing_controller.h
index 9d78912..db70ec9 100644
--- a/modules/pacing/pacing_controller.h
+++ b/modules/pacing/pacing_controller.h
@@ -17,9 +17,9 @@
#include <array>
#include <atomic>
#include <memory>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/field_trials_view.h"
#include "api/function_view.h"
#include "api/rtp_packet_sender.h"
@@ -62,8 +62,8 @@
virtual void OnAbortedRetransmissions(
uint32_t ssrc,
rtc::ArrayView<const uint16_t> sequence_numbers) {}
- virtual absl::optional<uint32_t> GetRtxSsrcForMedia(uint32_t ssrc) const {
- return absl::nullopt;
+ virtual std::optional<uint32_t> GetRtxSsrcForMedia(uint32_t ssrc) const {
+ return std::nullopt;
}
};
@@ -179,7 +179,7 @@
DataSize CurrentBufferLevel() const;
// Returns the time when the first packet was sent.
- absl::optional<Timestamp> FirstSentPacketTime() const;
+ std::optional<Timestamp> FirstSentPacketTime() const;
// Returns the number of milliseconds it will take to send the current
// packets in the queue, given the current size and bitrate, ignoring prio.
@@ -275,7 +275,7 @@
Timestamp last_process_time_;
Timestamp last_send_time_;
- absl::optional<Timestamp> first_sent_packet_time_;
+ std::optional<Timestamp> first_sent_packet_time_;
bool seen_first_packet_;
PrioritizedPacketQueue packet_queue_;
diff --git a/modules/pacing/pacing_controller_unittest.cc b/modules/pacing/pacing_controller_unittest.cc
index 8a37292..01430ff 100644
--- a/modules/pacing/pacing_controller_unittest.cc
+++ b/modules/pacing/pacing_controller_unittest.cc
@@ -134,7 +134,7 @@
OnAbortedRetransmissions,
(uint32_t, rtc::ArrayView<const uint16_t>),
(override));
- MOCK_METHOD(absl::optional<uint32_t>,
+ MOCK_METHOD(std::optional<uint32_t>,
GetRtxSsrcForMedia,
(uint32_t),
(const, override));
@@ -162,7 +162,7 @@
OnAbortedRetransmissions,
(uint32_t, rtc::ArrayView<const uint16_t>),
(override));
- MOCK_METHOD(absl::optional<uint32_t>,
+ MOCK_METHOD(std::optional<uint32_t>,
GetRtxSsrcForMedia,
(uint32_t),
(const, override));
@@ -200,8 +200,8 @@
void OnAbortedRetransmissions(uint32_t,
rtc::ArrayView<const uint16_t>) override {}
- absl::optional<uint32_t> GetRtxSsrcForMedia(uint32_t) const override {
- return absl::nullopt;
+ std::optional<uint32_t> GetRtxSsrcForMedia(uint32_t) const override {
+ return std::nullopt;
}
void OnBatchComplete() override {}
@@ -261,8 +261,8 @@
void OnAbortedRetransmissions(uint32_t,
rtc::ArrayView<const uint16_t>) override {}
- absl::optional<uint32_t> GetRtxSsrcForMedia(uint32_t) const override {
- return absl::nullopt;
+ std::optional<uint32_t> GetRtxSsrcForMedia(uint32_t) const override {
+ return std::nullopt;
}
void OnBatchComplete() override {}
diff --git a/modules/pacing/packet_router.cc b/modules/pacing/packet_router.cc
index 5124e94..8f4d492 100644
--- a/modules/pacing/packet_router.cc
+++ b/modules/pacing/packet_router.cc
@@ -14,10 +14,10 @@
#include <cstdint>
#include <limits>
#include <memory>
+#include <optional>
#include <utility>
#include "absl/functional/any_invocable.h"
-#include "absl/types/optional.h"
#include "api/transport/network_types.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/rtcp_packet.h"
@@ -49,10 +49,10 @@
RTC_DCHECK_RUN_ON(&thread_checker_);
AddSendRtpModuleToMap(rtp_module, rtp_module->SSRC());
- if (absl::optional<uint32_t> rtx_ssrc = rtp_module->RtxSsrc()) {
+ if (std::optional<uint32_t> rtx_ssrc = rtp_module->RtxSsrc()) {
AddSendRtpModuleToMap(rtp_module, *rtx_ssrc);
}
- if (absl::optional<uint32_t> flexfec_ssrc = rtp_module->FlexfecSsrc()) {
+ if (std::optional<uint32_t> flexfec_ssrc = rtp_module->FlexfecSsrc()) {
AddSendRtpModuleToMap(rtp_module, *flexfec_ssrc);
}
@@ -119,10 +119,10 @@
MaybeRemoveRembModuleCandidate(rtp_module, /* media_sender = */ true);
RemoveSendRtpModuleFromMap(rtp_module->SSRC());
- if (absl::optional<uint32_t> rtx_ssrc = rtp_module->RtxSsrc()) {
+ if (std::optional<uint32_t> rtx_ssrc = rtp_module->RtxSsrc()) {
RemoveSendRtpModuleFromMap(*rtx_ssrc);
}
- if (absl::optional<uint32_t> flexfec_ssrc = rtp_module->FlexfecSsrc()) {
+ if (std::optional<uint32_t> flexfec_ssrc = rtp_module->FlexfecSsrc()) {
RemoveSendRtpModuleFromMap(*flexfec_ssrc);
}
@@ -275,7 +275,7 @@
}
}
-absl::optional<uint32_t> PacketRouter::GetRtxSsrcForMedia(uint32_t ssrc) const {
+std::optional<uint32_t> PacketRouter::GetRtxSsrcForMedia(uint32_t ssrc) const {
RTC_DCHECK_RUN_ON(&thread_checker_);
auto it = send_modules_map_.find(ssrc);
if (it != send_modules_map_.end() && it->second->SSRC() == ssrc) {
@@ -283,7 +283,7 @@
// media SSRC for that RTP module.
return it->second->RtxSsrc();
}
- return absl::nullopt;
+ return std::nullopt;
}
void PacketRouter::SendRemb(int64_t bitrate_bps, std::vector<uint32_t> ssrcs) {
diff --git a/modules/pacing/packet_router.h b/modules/pacing/packet_router.h
index ae10637..1422645 100644
--- a/modules/pacing/packet_router.h
+++ b/modules/pacing/packet_router.h
@@ -71,7 +71,7 @@
void OnAbortedRetransmissions(
uint32_t ssrc,
rtc::ArrayView<const uint16_t> sequence_numbers) override;
- absl::optional<uint32_t> GetRtxSsrcForMedia(uint32_t ssrc) const override;
+ std::optional<uint32_t> GetRtxSsrcForMedia(uint32_t ssrc) const override;
void OnBatchComplete() override;
// Send REMB feedback.
diff --git a/modules/pacing/packet_router_unittest.cc b/modules/pacing/packet_router_unittest.cc
index fdd45cc..20e9622 100644
--- a/modules/pacing/packet_router_unittest.cc
+++ b/modules/pacing/packet_router_unittest.cc
@@ -420,7 +420,7 @@
&RtpPacketToSend::HasExtension<TransportSequenceNumber>,
false)),
Pointee(Property(&RtpPacketToSend::transport_sequence_number,
- absl::nullopt))),
+ std::nullopt))),
_));
packet_router_.SendPacket(std::move(packet), PacedPacketInfo());
packet_router_.OnBatchComplete();
@@ -517,9 +517,9 @@
packet_router_.AddSendRtpModule(&rtp_2, false);
EXPECT_EQ(packet_router_.GetRtxSsrcForMedia(kSsrc1), kRtxSsrc1);
- EXPECT_EQ(packet_router_.GetRtxSsrcForMedia(kRtxSsrc1), absl::nullopt);
- EXPECT_EQ(packet_router_.GetRtxSsrcForMedia(kSsrc2), absl::nullopt);
- EXPECT_EQ(packet_router_.GetRtxSsrcForMedia(kInvalidSsrc), absl::nullopt);
+ EXPECT_EQ(packet_router_.GetRtxSsrcForMedia(kRtxSsrc1), std::nullopt);
+ EXPECT_EQ(packet_router_.GetRtxSsrcForMedia(kSsrc2), std::nullopt);
+ EXPECT_EQ(packet_router_.GetRtxSsrcForMedia(kInvalidSsrc), std::nullopt);
packet_router_.RemoveSendRtpModule(&rtp_1);
packet_router_.RemoveSendRtpModule(&rtp_2);
diff --git a/modules/pacing/prioritized_packet_queue.cc b/modules/pacing/prioritized_packet_queue.cc
index 2d0d829..fa2ad7d 100644
--- a/modules/pacing/prioritized_packet_queue.cc
+++ b/modules/pacing/prioritized_packet_queue.cc
@@ -12,10 +12,10 @@
#include <algorithm>
#include <array>
+#include <optional>
#include <utility>
#include "absl/container/inlined_vector.h"
-#include "absl/types/optional.h"
#include "api/units/time_delta.h"
#include "api/units/timestamp.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
@@ -30,7 +30,7 @@
int GetPriorityForType(
RtpPacketMediaType type,
- absl::optional<RtpPacketToSend::OriginalType> original_type) {
+ std::optional<RtpPacketToSend::OriginalType> original_type) {
// Lower number takes priority over higher.
switch (type) {
case RtpPacketMediaType::kAudio:
@@ -69,7 +69,7 @@
ttl_per_prio[GetPriorityForType(RtpPacketMediaType::kRetransmission,
RtpPacketToSend::OriginalType::kVideo)] =
packet_queue_ttl.video_retransmission;
- ttl_per_prio[GetPriorityForType(RtpPacketMediaType::kVideo, absl::nullopt)] =
+ ttl_per_prio[GetPriorityForType(RtpPacketMediaType::kVideo, std::nullopt)] =
packet_queue_ttl.video;
return ttl_per_prio;
}
@@ -170,7 +170,7 @@
int prio_level =
GetPriorityForType(packet_type, prioritize_audio_retransmission_
? packet->original_packet_type()
- : absl::nullopt);
+ : std::nullopt);
PurgeOldPacketsAtPriorityLevel(prio_level, enqueue_time);
RTC_DCHECK_GE(prio_level, 0);
RTC_DCHECK_LT(prio_level, kNumPriorityLevels);
@@ -253,7 +253,7 @@
Timestamp PrioritizedPacketQueue::LeadingPacketEnqueueTime(
RtpPacketMediaType type) const {
RTC_DCHECK(type != RtpPacketMediaType::kRetransmission);
- const int priority_level = GetPriorityForType(type, absl::nullopt);
+ const int priority_level = GetPriorityForType(type, std::nullopt);
if (streams_by_prio_[priority_level].empty()) {
return Timestamp::MinusInfinity();
}
@@ -265,7 +265,7 @@
const {
if (!prioritize_audio_retransmission_) {
const int priority_level =
- GetPriorityForType(RtpPacketMediaType::kRetransmission, absl::nullopt);
+ GetPriorityForType(RtpPacketMediaType::kRetransmission, std::nullopt);
if (streams_by_prio_[priority_level].empty()) {
return Timestamp::PlusInfinity();
}
diff --git a/modules/pacing/rtp_packet_pacer.h b/modules/pacing/rtp_packet_pacer.h
index 2caa665..f388c07 100644
--- a/modules/pacing/rtp_packet_pacer.h
+++ b/modules/pacing/rtp_packet_pacer.h
@@ -13,9 +13,9 @@
#include <stdint.h>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/transport/network_types.h"
#include "api/units/data_rate.h"
#include "api/units/data_size.h"
@@ -49,7 +49,7 @@
virtual DataSize QueueSizeData() const = 0;
// Returns the time when the first packet was sent.
- virtual absl::optional<Timestamp> FirstSentPacketTime() const = 0;
+ virtual std::optional<Timestamp> FirstSentPacketTime() const = 0;
// Returns the expected number of milliseconds it will take to send the
// current packets in the queue, given the current size and bitrate, ignoring
diff --git a/modules/pacing/task_queue_paced_sender.cc b/modules/pacing/task_queue_paced_sender.cc
index 5559153..ca89770 100644
--- a/modules/pacing/task_queue_paced_sender.cc
+++ b/modules/pacing/task_queue_paced_sender.cc
@@ -160,7 +160,7 @@
return GetStats().queue_size;
}
-absl::optional<Timestamp> TaskQueuePacedSender::FirstSentPacketTime() const {
+std::optional<Timestamp> TaskQueuePacedSender::FirstSentPacketTime() const {
return GetStats().first_sent_packet_time;
}
diff --git a/modules/pacing/task_queue_paced_sender.h b/modules/pacing/task_queue_paced_sender.h
index a1d2474..be2bc84 100644
--- a/modules/pacing/task_queue_paced_sender.h
+++ b/modules/pacing/task_queue_paced_sender.h
@@ -15,9 +15,9 @@
#include <stdint.h>
#include <memory>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/field_trials_view.h"
#include "api/sequence_checker.h"
#include "api/task_queue/pending_task_safety_flag.h"
@@ -107,7 +107,7 @@
DataSize QueueSizeData() const override;
// Returns the time when the first packet was sent;
- absl::optional<Timestamp> FirstSentPacketTime() const override;
+ std::optional<Timestamp> FirstSentPacketTime() const override;
// Returns the number of milliseconds it will take to send the current
// packets in the queue, given the current size and bitrate, ignoring prio.
@@ -127,7 +127,7 @@
Timestamp oldest_packet_enqueue_time;
DataSize queue_size;
TimeDelta expected_queue_time;
- absl::optional<Timestamp> first_sent_packet_time;
+ std::optional<Timestamp> first_sent_packet_time;
};
void OnStatsUpdated(const Stats& stats);
diff --git a/modules/remote_bitrate_estimator/BUILD.gn b/modules/remote_bitrate_estimator/BUILD.gn
index 2cb8700..453ae66 100644
--- a/modules/remote_bitrate_estimator/BUILD.gn
+++ b/modules/remote_bitrate_estimator/BUILD.gn
@@ -51,7 +51,6 @@
"../rtp_rtcp:rtp_rtcp_format",
"//third_party/abseil-cpp/absl/base:nullability",
"//third_party/abseil-cpp/absl/strings",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -89,7 +88,6 @@
"../../rtc_base/synchronization:mutex",
"../../system_wrappers",
"../rtp_rtcp:rtp_rtcp_format",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -182,7 +180,6 @@
"../../test:test_support",
"../pacing",
"../rtp_rtcp:rtp_rtcp_format",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
}
diff --git a/modules/remote_bitrate_estimator/aimd_rate_control.cc b/modules/remote_bitrate_estimator/aimd_rate_control.cc
index 576d307..26ff73c 100644
--- a/modules/remote_bitrate_estimator/aimd_rate_control.cc
+++ b/modules/remote_bitrate_estimator/aimd_rate_control.cc
@@ -183,7 +183,7 @@
}
void AimdRateControl::SetNetworkStateEstimate(
- const absl::optional<NetworkStateEstimate>& estimate) {
+ const std::optional<NetworkStateEstimate>& estimate) {
network_estimate_ = estimate;
}
@@ -221,7 +221,7 @@
void AimdRateControl::ChangeBitrate(const RateControlInput& input,
Timestamp at_time) {
- absl::optional<DataRate> new_bitrate;
+ std::optional<DataRate> new_bitrate;
DataRate estimated_throughput =
input.estimated_throughput.value_or(latest_estimated_throughput_);
if (input.estimated_throughput)
diff --git a/modules/remote_bitrate_estimator/aimd_rate_control.h b/modules/remote_bitrate_estimator/aimd_rate_control.h
index b7a8ef6..55f1e37 100644
--- a/modules/remote_bitrate_estimator/aimd_rate_control.h
+++ b/modules/remote_bitrate_estimator/aimd_rate_control.h
@@ -13,7 +13,8 @@
#include <stdint.h>
-#include "absl/types/optional.h"
+#include <optional>
+
#include "api/field_trials_view.h"
#include "api/transport/network_types.h"
#include "api/units/data_rate.h"
@@ -57,7 +58,7 @@
void SetInApplicationLimitedRegion(bool in_alr);
void SetEstimate(DataRate bitrate, Timestamp at_time);
void SetNetworkStateEstimate(
- const absl::optional<NetworkStateEstimate>& estimate);
+ const std::optional<NetworkStateEstimate>& estimate);
// Returns the increase rate when used bandwidth is near the link capacity.
double GetNearMaxIncreaseRateBpsPerSecond() const;
@@ -90,7 +91,7 @@
DataRate current_bitrate_;
DataRate latest_estimated_throughput_;
LinkCapacityEstimator link_capacity_;
- absl::optional<NetworkStateEstimate> network_estimate_;
+ std::optional<NetworkStateEstimate> network_estimate_;
RateControlState rate_control_state_;
Timestamp time_last_bitrate_change_;
Timestamp time_last_bitrate_decrease_;
@@ -107,7 +108,7 @@
FieldTrialFlag disable_estimate_bounded_increase_{"Disabled"};
FieldTrialParameter<bool> use_current_estimate_as_min_upper_bound_{"c_upper",
true};
- absl::optional<DataRate> last_decrease_;
+ std::optional<DataRate> last_decrease_;
};
} // namespace webrtc
diff --git a/modules/remote_bitrate_estimator/aimd_rate_control_unittest.cc b/modules/remote_bitrate_estimator/aimd_rate_control_unittest.cc
index 401e87e..6c3c985 100644
--- a/modules/remote_bitrate_estimator/aimd_rate_control_unittest.cc
+++ b/modules/remote_bitrate_estimator/aimd_rate_control_unittest.cc
@@ -165,7 +165,7 @@
now += (kInitializationTime + TimeDelta::Millis(1));
aimd_rate_control.Update({BandwidthUsage::kBwNormal, kInitialBitrate}, now);
for (int i = 0; i < 100; ++i) {
- aimd_rate_control.Update({BandwidthUsage::kBwNormal, absl::nullopt}, now);
+ aimd_rate_control.Update({BandwidthUsage::kBwNormal, std::nullopt}, now);
now += TimeDelta::Millis(100);
}
EXPECT_LE(aimd_rate_control.LatestEstimate(),
@@ -187,7 +187,7 @@
ASSERT_EQ(aimd_rate_control.LatestEstimate(), kInitialBitrate);
for (int i = 0; i < 100; ++i) {
- aimd_rate_control.Update({BandwidthUsage::kBwNormal, absl::nullopt}, now);
+ aimd_rate_control.Update({BandwidthUsage::kBwNormal, std::nullopt}, now);
now += TimeDelta::Millis(100);
}
EXPECT_EQ(aimd_rate_control.LatestEstimate(), kInitialBitrate);
@@ -287,7 +287,7 @@
aimd_rate_control.SetInApplicationLimitedRegion(false);
aimd_rate_control.Update({BandwidthUsage::kBwNormal, kInitialBitrate}, now);
for (int i = 0; i < 100; ++i) {
- aimd_rate_control.Update({BandwidthUsage::kBwNormal, absl::nullopt}, now);
+ aimd_rate_control.Update({BandwidthUsage::kBwNormal, std::nullopt}, now);
now += TimeDelta::Millis(100);
}
EXPECT_GT(aimd_rate_control.LatestEstimate(), kInitialBitrate);
@@ -306,7 +306,7 @@
aimd_rate_control.SetNetworkStateEstimate(network_estimate);
for (int i = 0; i < 100; ++i) {
- aimd_rate_control.Update({BandwidthUsage::kBwNormal, absl::nullopt}, now);
+ aimd_rate_control.Update({BandwidthUsage::kBwNormal, std::nullopt}, now);
now += TimeDelta::Millis(100);
}
EXPECT_GT(aimd_rate_control.LatestEstimate(),
diff --git a/modules/remote_bitrate_estimator/bwe_defines.cc b/modules/remote_bitrate_estimator/bwe_defines.cc
index db92f46..ca41732 100644
--- a/modules/remote_bitrate_estimator/bwe_defines.cc
+++ b/modules/remote_bitrate_estimator/bwe_defines.cc
@@ -16,7 +16,7 @@
RateControlInput::RateControlInput(
BandwidthUsage bw_state,
- const absl::optional<DataRate>& estimated_throughput)
+ const std::optional<DataRate>& estimated_throughput)
: bw_state(bw_state), estimated_throughput(estimated_throughput) {}
RateControlInput::~RateControlInput() = default;
diff --git a/modules/remote_bitrate_estimator/congestion_control_feedback_generator.cc b/modules/remote_bitrate_estimator/congestion_control_feedback_generator.cc
index aba3586..7c5c648 100644
--- a/modules/remote_bitrate_estimator/congestion_control_feedback_generator.cc
+++ b/modules/remote_bitrate_estimator/congestion_control_feedback_generator.cc
@@ -103,8 +103,8 @@
std::vector<rtcp::CongestionControlFeedback::PacketInfo> rtcp_packet_info;
rtcp_packet_info.reserve(packets_.size());
- absl::optional<uint32_t> previous_ssrc;
- absl::optional<int64_t> previous_seq_no;
+ std::optional<uint32_t> previous_ssrc;
+ std::optional<int64_t> previous_seq_no;
for (const PacketInfo packet : packets_) {
if (previous_ssrc == packet.ssrc &&
previous_seq_no == packet.unwrapped_sequence_number) {
diff --git a/modules/remote_bitrate_estimator/include/bwe_defines.h b/modules/remote_bitrate_estimator/include/bwe_defines.h
index 02646ef..e3f4e8c 100644
--- a/modules/remote_bitrate_estimator/include/bwe_defines.h
+++ b/modules/remote_bitrate_estimator/include/bwe_defines.h
@@ -13,7 +13,8 @@
#include <stdint.h>
-#include "absl/types/optional.h"
+#include <optional>
+
#include "api/transport/bandwidth_usage.h"
#include "api/units/data_rate.h"
#include "api/units/time_delta.h"
@@ -36,11 +37,11 @@
struct RateControlInput {
RateControlInput(BandwidthUsage bw_state,
- const absl::optional<DataRate>& estimated_throughput);
+ const std::optional<DataRate>& estimated_throughput);
~RateControlInput();
BandwidthUsage bw_state;
- absl::optional<DataRate> estimated_throughput;
+ std::optional<DataRate> estimated_throughput;
};
} // namespace webrtc
diff --git a/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.cc b/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.cc
index 278ad9d..af2cd5b 100644
--- a/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.cc
+++ b/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.cc
@@ -229,7 +229,7 @@
// here.
// Check if incoming bitrate estimate is valid, and if it needs to be reset.
- absl::optional<DataRate> incoming_bitrate =
+ std::optional<DataRate> incoming_bitrate =
incoming_bitrate_.Rate(arrival_time);
if (incoming_bitrate) {
incoming_bitrate_initialized_ = true;
@@ -302,7 +302,7 @@
remote_rate_.GetFeedbackInterval().ms()) {
update_estimate = true;
} else if (detector_.State() == BandwidthUsage::kBwOverusing) {
- absl::optional<DataRate> incoming_rate =
+ std::optional<DataRate> incoming_rate =
incoming_bitrate_.Rate(arrival_time);
if (incoming_rate.has_value() &&
remote_rate_.TimeToReduceFurther(now, *incoming_rate)) {
diff --git a/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.cc b/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.cc
index 9db92ac..60c0852 100644
--- a/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.cc
+++ b/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.cc
@@ -11,10 +11,10 @@
#include "modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.h"
#include <cstdint>
+#include <optional>
#include <utility>
#include "absl/base/nullability.h"
-#include "absl/types/optional.h"
#include "api/environment/environment.h"
#include "modules/remote_bitrate_estimator/aimd_rate_control.h"
#include "modules/remote_bitrate_estimator/include/bwe_defines.h"
@@ -58,7 +58,7 @@
void RemoteBitrateEstimatorSingleStream::IncomingPacket(
const RtpPacketReceived& rtp_packet) {
- absl::optional<int32_t> transmission_time_offset =
+ std::optional<int32_t> transmission_time_offset =
rtp_packet.GetExtension<TransmissionOffset>();
if (!uma_recorded_) {
BweNames type = transmission_time_offset.has_value()
@@ -75,7 +75,7 @@
estimator.last_packet_time = now;
// Check if incoming bitrate estimate is valid, and if it needs to be reset.
- absl::optional<DataRate> incoming_bitrate = incoming_bitrate_.Rate(now);
+ std::optional<DataRate> incoming_bitrate = incoming_bitrate_.Rate(now);
if (incoming_bitrate) {
last_valid_incoming_bitrate_ = *incoming_bitrate;
} else if (last_valid_incoming_bitrate_ > DataRate::Zero()) {
@@ -103,7 +103,7 @@
estimator.estimator.num_of_deltas(), now_ms);
}
if (estimator.detector.State() == BandwidthUsage::kBwOverusing) {
- absl::optional<DataRate> incoming_bitrate = incoming_bitrate_.Rate(now);
+ std::optional<DataRate> incoming_bitrate = incoming_bitrate_.Rate(now);
if (incoming_bitrate.has_value() &&
(prior_state != BandwidthUsage::kBwOverusing ||
remote_rate_.TimeToReduceFurther(now, *incoming_bitrate))) {
diff --git a/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.h b/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.h
index 16a0097..1efd807 100644
--- a/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.h
+++ b/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.h
@@ -15,10 +15,10 @@
#include <stdint.h>
#include <map>
+#include <optional>
#include <vector>
#include "absl/base/nullability.h"
-#include "absl/types/optional.h"
#include "api/environment/environment.h"
#include "api/units/data_rate.h"
#include "api/units/time_delta.h"
@@ -73,7 +73,7 @@
BitrateTracker incoming_bitrate_;
DataRate last_valid_incoming_bitrate_;
AimdRateControl remote_rate_;
- absl::optional<Timestamp> last_process_time_;
+ std::optional<Timestamp> last_process_time_;
TimeDelta process_interval_;
bool uma_recorded_;
};
diff --git a/modules/remote_bitrate_estimator/transport_sequence_number_feedback_generator.cc b/modules/remote_bitrate_estimator/transport_sequence_number_feedback_generator.cc
index b764013..4433b35 100644
--- a/modules/remote_bitrate_estimator/transport_sequence_number_feedback_generator.cc
+++ b/modules/remote_bitrate_estimator/transport_sequence_number_feedback_generator.cc
@@ -14,9 +14,9 @@
#include <cstdint>
#include <limits>
#include <memory>
+#include <optional>
#include <utility>
-#include "absl/types/optional.h"
#include "api/units/data_size.h"
#include "api/units/time_delta.h"
#include "modules/rtp_rtcp/source/rtcp_packet/remote_estimate.h"
@@ -94,7 +94,7 @@
}
uint16_t seqnum = 0;
- absl::optional<FeedbackRequest> feedback_request;
+ std::optional<FeedbackRequest> feedback_request;
if (!packet.GetExtension<TransportSequenceNumber>(&seqnum) &&
!packet.GetExtension<TransportSequenceNumberV2>(&seqnum,
&feedback_request)) {
@@ -139,7 +139,7 @@
SendFeedbackOnRequest(seq, *feedback_request);
}
- absl::optional<uint32_t> absolute_send_time_24bits =
+ std::optional<uint32_t> absolute_send_time_24bits =
packet.GetExtension<AbsoluteSendTime>();
if (network_state_estimator_ && absolute_send_time_24bits.has_value()) {
PacketResult packet_result;
@@ -217,7 +217,7 @@
std::unique_ptr<rtcp::RemoteEstimate> remote_estimate;
if (network_state_estimator_) {
- absl::optional<NetworkStateEstimate> state_estimate =
+ std::optional<NetworkStateEstimate> state_estimate =
network_state_estimator_->GetCurrentEstimate();
if (state_estimate) {
remote_estimate = std::make_unique<rtcp::RemoteEstimate>();
diff --git a/modules/remote_bitrate_estimator/transport_sequence_number_feedback_generator.h b/modules/remote_bitrate_estimator/transport_sequence_number_feedback_generator.h
index 4c2fb31..be9412d 100644
--- a/modules/remote_bitrate_estimator/transport_sequence_number_feedback_generator.h
+++ b/modules/remote_bitrate_estimator/transport_sequence_number_feedback_generator.h
@@ -14,9 +14,9 @@
#include <deque>
#include <functional>
#include <memory>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/field_trials_view.h"
#include "api/rtp_headers.h"
#include "api/transport/network_control.h"
@@ -95,8 +95,8 @@
DataSize packet_overhead_ RTC_GUARDED_BY(&lock_);
// The next sequence number that should be the start sequence number during
- // periodic reporting. Will be absl::nullopt before the first seen packet.
- absl::optional<int64_t> periodic_window_start_seq_ RTC_GUARDED_BY(&lock_);
+ // periodic reporting. Will be std::nullopt before the first seen packet.
+ std::optional<int64_t> periodic_window_start_seq_ RTC_GUARDED_BY(&lock_);
// Packet arrival times, by sequence number.
PacketArrivalTimeMap packet_arrival_times_ RTC_GUARDED_BY(&lock_);
diff --git a/modules/remote_bitrate_estimator/transport_sequence_number_feedback_generator_unittest.cc b/modules/remote_bitrate_estimator/transport_sequence_number_feedback_generator_unittest.cc
index bc1ce81..5d628d1 100644
--- a/modules/remote_bitrate_estimator/transport_sequence_number_feedback_generator_unittest.cc
+++ b/modules/remote_bitrate_estimator/transport_sequence_number_feedback_generator_unittest.cc
@@ -12,9 +12,9 @@
#include <cstdint>
#include <memory>
+#include <optional>
#include <utility>
-#include "absl/types/optional.h"
#include "api/transport/test/mock_network_control.h"
#include "api/units/data_rate.h"
#include "api/units/data_size.h"
@@ -86,7 +86,7 @@
protected:
void IncomingPacket(uint16_t seq,
Timestamp arrival_time,
- absl::optional<uint32_t> abs_send_time = absl::nullopt) {
+ std::optional<uint32_t> abs_send_time = std::nullopt) {
RtpHeaderExtensionMap map;
map.Register<TransportSequenceNumber>(1);
map.Register<AbsoluteSendTime>(2);
@@ -102,7 +102,7 @@
void IncomingPacketV2(
uint16_t seq,
Timestamp arrival_time,
- absl::optional<FeedbackRequest> feedback_request = absl::nullopt) {
+ std::optional<FeedbackRequest> feedback_request = std::nullopt) {
RtpHeaderExtensionMap map;
map.Register<TransportSequenceNumberV2>(1);
RtpPacketReceived packet(&map, arrival_time);
diff --git a/modules/rtp_rtcp/BUILD.gn b/modules/rtp_rtcp/BUILD.gn
index ec05361..d0e6567 100644
--- a/modules/rtp_rtcp/BUILD.gn
+++ b/modules/rtp_rtcp/BUILD.gn
@@ -146,7 +146,6 @@
"//third_party/abseil-cpp/absl/base:core_headers",
"//third_party/abseil-cpp/absl/container:inlined_vector",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
"//third_party/abseil-cpp/absl/types:variant",
]
}
@@ -357,7 +356,6 @@
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
"//third_party/abseil-cpp/absl/types:variant",
]
}
@@ -393,7 +391,6 @@
"//third_party/abseil-cpp/absl/base:core_headers",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -449,7 +446,6 @@
"//third_party/abseil-cpp/absl/cleanup",
"//third_party/abseil-cpp/absl/functional:any_invocable",
"//third_party/abseil-cpp/absl/memory",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -468,7 +464,6 @@
"../../api/video:video_rtp_headers",
"../../modules/video_coding:codec_globals_headers",
"//third_party/abseil-cpp/absl/container:inlined_vector",
- "//third_party/abseil-cpp/absl/types:optional",
"//third_party/abseil-cpp/absl/types:variant",
]
}
@@ -517,7 +512,6 @@
"../../rtc_base:checks",
"../../test:test_support",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -738,7 +732,6 @@
"//third_party/abseil-cpp/absl/base:core_headers",
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -767,7 +760,6 @@
":rtp_rtcp_format",
"../../common_video:common_video",
"../../test:test_support",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
}
diff --git a/modules/rtp_rtcp/include/flexfec_sender.h b/modules/rtp_rtcp/include/flexfec_sender.h
index 65ac0e9..83034cc 100644
--- a/modules/rtp_rtcp/include/flexfec_sender.h
+++ b/modules/rtp_rtcp/include/flexfec_sender.h
@@ -51,7 +51,7 @@
FecType GetFecType() const override {
return VideoFecGenerator::FecType::kFlexFec;
}
- absl::optional<uint32_t> FecSsrc() override { return ssrc_; }
+ std::optional<uint32_t> FecSsrc() override { return ssrc_; }
// Sets the FEC rate, max frames sent before FEC packets are sent,
// and what type of generator matrices are used.
@@ -72,7 +72,7 @@
DataRate CurrentFecRate() const override;
// Only called on the VideoSendStream queue, after operation has shut down.
- absl::optional<RtpState> GetRtpState() override;
+ std::optional<RtpState> GetRtpState() override;
private:
// Utility.
diff --git a/modules/rtp_rtcp/include/receive_statistics.h b/modules/rtp_rtcp/include/receive_statistics.h
index 827fd3a..5faea15 100644
--- a/modules/rtp_rtcp/include/receive_statistics.h
+++ b/modules/rtp_rtcp/include/receive_statistics.h
@@ -13,9 +13,9 @@
#include <map>
#include <memory>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "call/rtp_packet_sink_interface.h"
#include "modules/rtp_rtcp/include/rtcp_statistics.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
@@ -41,7 +41,7 @@
virtual RtpReceiveStats GetStats() const = 0;
// Returns average over the stream life time.
- virtual absl::optional<int> GetFractionLostInPercent() const = 0;
+ virtual std::optional<int> GetFractionLostInPercent() const = 0;
// TODO(bugs.webrtc.org/10679): Delete, migrate users to the above GetStats
// method (and extend RtpReceiveStats if needed).
diff --git a/modules/rtp_rtcp/include/remote_ntp_time_estimator.h b/modules/rtp_rtcp/include/remote_ntp_time_estimator.h
index 01d0c85..f60763a 100644
--- a/modules/rtp_rtcp/include/remote_ntp_time_estimator.h
+++ b/modules/rtp_rtcp/include/remote_ntp_time_estimator.h
@@ -13,7 +13,8 @@
#include <stdint.h>
-#include "absl/types/optional.h"
+#include <optional>
+
#include "api/units/time_delta.h"
#include "api/units/timestamp.h"
#include "rtc_base/numerics/moving_percentile_filter.h"
@@ -58,7 +59,7 @@
// local one. This is equal to local NTP clock - remote NTP clock.
// The offset is returned in ntp time resolution, i.e. 1/2^32 sec ~= 0.2 ns.
// Returns nullopt on failure.
- absl::optional<int64_t> EstimateRemoteToLocalClockOffset();
+ std::optional<int64_t> EstimateRemoteToLocalClockOffset();
private:
Clock* clock_;
diff --git a/modules/rtp_rtcp/include/rtp_rtcp_defines.h b/modules/rtp_rtcp/include/rtp_rtcp_defines.h
index 911a4eb..eb4d1d0 100644
--- a/modules/rtp_rtcp/include/rtp_rtcp_defines.h
+++ b/modules/rtp_rtcp/include/rtp_rtcp_defines.h
@@ -17,12 +17,12 @@
#include <cstddef>
#include <cstdint>
#include <memory>
+#include <optional>
#include <utility>
#include <vector>
#include "absl/algorithm/container.h"
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "absl/types/variant.h"
#include "api/array_view.h"
#include "api/audio_codecs/audio_format.h"
@@ -200,11 +200,11 @@
const PacedPacketInfo& pacing_info);
uint16_t transport_sequence_number = 0;
- absl::optional<uint32_t> media_ssrc;
+ std::optional<uint32_t> media_ssrc;
uint16_t rtp_sequence_number = 0; // Only valid if `media_ssrc` is set.
uint32_t rtp_timestamp = 0;
size_t length = 0;
- absl::optional<RtpPacketMediaType> packet_type;
+ std::optional<RtpPacketMediaType> packet_type;
PacedPacketInfo pacing_info;
};
@@ -241,7 +241,7 @@
// `rtp_sequence_number` and `is_retransmission` are only valid if `ssrc`
// is populated.
- absl::optional<uint32_t> ssrc;
+ std::optional<uint32_t> ssrc;
uint16_t rtp_sequence_number;
bool is_retransmission;
};
@@ -403,7 +403,7 @@
// Time of the last packet received in unix epoch,
// i.e. Timestamp::Zero() represents 1st Jan 1970 00:00
- absl::optional<Timestamp> last_packet_received;
+ std::optional<Timestamp> last_packet_received;
// Counters exposed in RTCInboundRtpStreamStats, see
// https://w3c.github.io/webrtc-stats/#inboundrtpstats-dict*
@@ -425,7 +425,7 @@
class SendPacketObserver {
public:
virtual ~SendPacketObserver() = default;
- virtual void OnSendPacket(absl::optional<uint16_t> packet_id,
+ virtual void OnSendPacket(std::optional<uint16_t> packet_id,
Timestamp capture_time,
uint32_t ssrc) = 0;
};
diff --git a/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h b/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
index 6aec723..4406ad6 100644
--- a/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
+++ b/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
@@ -12,13 +12,13 @@
#define MODULES_RTP_RTCP_MOCKS_MOCK_RTP_RTCP_H_
#include <memory>
+#include <optional>
#include <set>
#include <string>
#include <utility>
#include <vector>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/video/video_bitrate_allocation.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
@@ -69,9 +69,9 @@
MOCK_METHOD(void, SetMid, (absl::string_view mid), (override));
MOCK_METHOD(void, SetRtxSendStatus, (int modes), (override));
MOCK_METHOD(int, RtxSendStatus, (), (const, override));
- MOCK_METHOD(absl::optional<uint32_t>, RtxSsrc, (), (const, override));
+ MOCK_METHOD(std::optional<uint32_t>, RtxSsrc, (), (const, override));
MOCK_METHOD(void, SetRtxSendPayloadType, (int, int), (override));
- MOCK_METHOD(absl::optional<uint32_t>, FlexfecSsrc, (), (const, override));
+ MOCK_METHOD(std::optional<uint32_t>, FlexfecSsrc, (), (const, override));
MOCK_METHOD(int32_t, SetSendingStatus, (bool sending), (override));
MOCK_METHOD(bool, Sending, (), (const, override));
MOCK_METHOD(void, SetSendingMediaStatus, (bool sending), (override));
@@ -132,7 +132,7 @@
MOCK_METHOD(RtcpMode, RTCP, (), (const, override));
MOCK_METHOD(void, SetRTCPStatus, (RtcpMode method), (override));
MOCK_METHOD(int32_t, SetCNAME, (absl::string_view cname), (override));
- MOCK_METHOD(absl::optional<TimeDelta>, LastRtt, (), (const, override));
+ MOCK_METHOD(std::optional<TimeDelta>, LastRtt, (), (const, override));
MOCK_METHOD(TimeDelta, ExpectedRetransmissionTime, (), (const, override));
MOCK_METHOD(int32_t, SendRTCP, (RTCPPacketType packet_type), (override));
MOCK_METHOD(void,
@@ -143,11 +143,11 @@
GetLatestReportBlockData,
(),
(const, override));
- MOCK_METHOD(absl::optional<SenderReportStats>,
+ MOCK_METHOD(std::optional<SenderReportStats>,
GetSenderReportStats,
(),
(const, override));
- MOCK_METHOD(absl::optional<NonSenderRttStats>,
+ MOCK_METHOD(std::optional<NonSenderRttStats>,
GetNonSenderRttStats,
(),
(const, override));
diff --git a/modules/rtp_rtcp/source/absolute_capture_time_interpolator.cc b/modules/rtp_rtcp/source/absolute_capture_time_interpolator.cc
index f151084..5da0ce4 100644
--- a/modules/rtp_rtcp/source/absolute_capture_time_interpolator.cc
+++ b/modules/rtp_rtcp/source/absolute_capture_time_interpolator.cc
@@ -29,21 +29,21 @@
return csrcs[0];
}
-absl::optional<AbsoluteCaptureTime>
+std::optional<AbsoluteCaptureTime>
AbsoluteCaptureTimeInterpolator::OnReceivePacket(
uint32_t source,
uint32_t rtp_timestamp,
int rtp_clock_frequency_hz,
- const absl::optional<AbsoluteCaptureTime>& received_extension) {
+ const std::optional<AbsoluteCaptureTime>& received_extension) {
const Timestamp receive_time = clock_->CurrentTime();
MutexLock lock(&mutex_);
- if (received_extension == absl::nullopt) {
+ if (received_extension == std::nullopt) {
if (!ShouldInterpolateExtension(receive_time, source, rtp_timestamp,
rtp_clock_frequency_hz)) {
last_receive_time_ = Timestamp::MinusInfinity();
- return absl::nullopt;
+ return std::nullopt;
}
return AbsoluteCaptureTime{
diff --git a/modules/rtp_rtcp/source/absolute_capture_time_interpolator.h b/modules/rtp_rtcp/source/absolute_capture_time_interpolator.h
index c830686..59ddead 100644
--- a/modules/rtp_rtcp/source/absolute_capture_time_interpolator.h
+++ b/modules/rtp_rtcp/source/absolute_capture_time_interpolator.h
@@ -44,12 +44,12 @@
rtc::ArrayView<const uint32_t> csrcs);
// Returns a received header extension, an interpolated header extension, or
- // `absl::nullopt` if it's not possible to interpolate a header extension.
- absl::optional<AbsoluteCaptureTime> OnReceivePacket(
+ // `std::nullopt` if it's not possible to interpolate a header extension.
+ std::optional<AbsoluteCaptureTime> OnReceivePacket(
uint32_t source,
uint32_t rtp_timestamp,
int rtp_clock_frequency_hz,
- const absl::optional<AbsoluteCaptureTime>& received_extension);
+ const std::optional<AbsoluteCaptureTime>& received_extension);
private:
friend class AbsoluteCaptureTimeSender;
diff --git a/modules/rtp_rtcp/source/absolute_capture_time_interpolator_unittest.cc b/modules/rtp_rtcp/source/absolute_capture_time_interpolator_unittest.cc
index 4a48054..7471496 100644
--- a/modules/rtp_rtcp/source/absolute_capture_time_interpolator_unittest.cc
+++ b/modules/rtp_rtcp/source/absolute_capture_time_interpolator_unittest.cc
@@ -38,7 +38,7 @@
const AbsoluteCaptureTime kExtension0 = {Int64MsToUQ32x32(9000),
Int64MsToQ32x32(-350)};
const AbsoluteCaptureTime kExtension1 = {Int64MsToUQ32x32(9020),
- absl::nullopt};
+ std::nullopt};
SimulatedClock clock(0);
AbsoluteCaptureTimeInterpolator interpolator(&clock);
@@ -64,13 +64,13 @@
EXPECT_EQ(
interpolator.OnReceivePacket(kSource, kRtpTimestamp0, kRtpClockFrequency,
- /*received_extension=*/absl::nullopt),
- absl::nullopt);
+ /*received_extension=*/std::nullopt),
+ std::nullopt);
EXPECT_EQ(
interpolator.OnReceivePacket(kSource, kRtpTimestamp1, kRtpClockFrequency,
- /*received_extension=*/absl::nullopt),
- absl::nullopt);
+ /*received_extension=*/std::nullopt),
+ std::nullopt);
}
TEST(AbsoluteCaptureTimeInterpolatorTest, InterpolateLaterPacketArrivingLater) {
@@ -89,9 +89,9 @@
kRtpClockFrequency, kExtension),
kExtension);
- absl::optional<AbsoluteCaptureTime> extension =
+ std::optional<AbsoluteCaptureTime> extension =
interpolator.OnReceivePacket(kSource, kRtpTimestamp1, kRtpClockFrequency,
- /*received_extension=*/absl::nullopt);
+ /*received_extension=*/std::nullopt);
ASSERT_TRUE(extension.has_value());
EXPECT_EQ(UQ32x32ToInt64Ms(extension->absolute_capture_timestamp),
UQ32x32ToInt64Ms(kExtension.absolute_capture_timestamp) + 20);
@@ -100,7 +100,7 @@
extension =
interpolator.OnReceivePacket(kSource, kRtpTimestamp2, kRtpClockFrequency,
- /*received_extension=*/absl::nullopt);
+ /*received_extension=*/std::nullopt);
ASSERT_TRUE(extension.has_value());
EXPECT_EQ(UQ32x32ToInt64Ms(extension->absolute_capture_timestamp),
UQ32x32ToInt64Ms(kExtension.absolute_capture_timestamp) + 40);
@@ -125,9 +125,9 @@
kRtpClockFrequency, kExtension),
kExtension);
- absl::optional<AbsoluteCaptureTime> extension =
+ std::optional<AbsoluteCaptureTime> extension =
interpolator.OnReceivePacket(kSource, kRtpTimestamp1, kRtpClockFrequency,
- /*received_extension=*/absl::nullopt);
+ /*received_extension=*/std::nullopt);
ASSERT_TRUE(extension.has_value());
EXPECT_EQ(UQ32x32ToInt64Ms(extension->absolute_capture_timestamp),
UQ32x32ToInt64Ms(kExtension.absolute_capture_timestamp) - 20);
@@ -136,7 +136,7 @@
extension =
interpolator.OnReceivePacket(kSource, kRtpTimestamp2, kRtpClockFrequency,
- /*received_extension=*/absl::nullopt);
+ /*received_extension=*/std::nullopt);
ASSERT_TRUE(extension.has_value());
EXPECT_EQ(UQ32x32ToInt64Ms(extension->absolute_capture_timestamp),
UQ32x32ToInt64Ms(kExtension.absolute_capture_timestamp) - 40);
@@ -161,9 +161,9 @@
kRtpClockFrequency, kExtension),
kExtension);
- absl::optional<AbsoluteCaptureTime> extension =
+ std::optional<AbsoluteCaptureTime> extension =
interpolator.OnReceivePacket(kSource, kRtpTimestamp1, kRtpClockFrequency,
- /*received_extension=*/absl::nullopt);
+ /*received_extension=*/std::nullopt);
ASSERT_TRUE(extension.has_value());
EXPECT_EQ(UQ32x32ToInt64Ms(extension->absolute_capture_timestamp),
UQ32x32ToInt64Ms(kExtension.absolute_capture_timestamp) + 20);
@@ -172,7 +172,7 @@
extension =
interpolator.OnReceivePacket(kSource, kRtpTimestamp2, kRtpClockFrequency,
- /*received_extension=*/absl::nullopt);
+ /*received_extension=*/std::nullopt);
ASSERT_TRUE(extension.has_value());
EXPECT_EQ(UQ32x32ToInt64Ms(extension->absolute_capture_timestamp),
UQ32x32ToInt64Ms(kExtension.absolute_capture_timestamp) + 40);
@@ -197,9 +197,9 @@
kRtpClockFrequency, kExtension),
kExtension);
- absl::optional<AbsoluteCaptureTime> extension =
+ std::optional<AbsoluteCaptureTime> extension =
interpolator.OnReceivePacket(kSource, kRtpTimestamp1, kRtpClockFrequency,
- /*received_extension=*/absl::nullopt);
+ /*received_extension=*/std::nullopt);
ASSERT_TRUE(extension.has_value());
EXPECT_EQ(UQ32x32ToInt64Ms(extension->absolute_capture_timestamp),
UQ32x32ToInt64Ms(kExtension.absolute_capture_timestamp) - 20);
@@ -208,7 +208,7 @@
extension =
interpolator.OnReceivePacket(kSource, kRtpTimestamp2, kRtpClockFrequency,
- /*received_extension=*/absl::nullopt);
+ /*received_extension=*/std::nullopt);
ASSERT_TRUE(extension.has_value());
EXPECT_EQ(UQ32x32ToInt64Ms(extension->absolute_capture_timestamp),
UQ32x32ToInt64Ms(kExtension.absolute_capture_timestamp) - 40);
@@ -236,15 +236,15 @@
EXPECT_NE(
interpolator.OnReceivePacket(kSource, kRtpTimestamp1, kRtpClockFrequency,
- /*received_extension=*/absl::nullopt),
- absl::nullopt);
+ /*received_extension=*/std::nullopt),
+ std::nullopt);
clock.AdvanceTime(TimeDelta::Millis(1));
EXPECT_EQ(
interpolator.OnReceivePacket(kSource, kRtpTimestamp2, kRtpClockFrequency,
- /*received_extension=*/absl::nullopt),
- absl::nullopt);
+ /*received_extension=*/std::nullopt),
+ std::nullopt);
}
TEST(AbsoluteCaptureTimeInterpolatorTest, SkipInterpolateIfSourceChanged) {
@@ -265,8 +265,8 @@
EXPECT_EQ(
interpolator.OnReceivePacket(kSource1, kRtpTimestamp1, kRtpClockFrequency,
- /*received_extension=*/absl::nullopt),
- absl::nullopt);
+ /*received_extension=*/std::nullopt),
+ std::nullopt);
}
TEST(AbsoluteCaptureTimeInterpolatorTest,
@@ -288,8 +288,8 @@
EXPECT_EQ(
interpolator.OnReceivePacket(kSource, kRtpTimestamp1, kRtpClockFrequency1,
- /*received_extension=*/absl::nullopt),
- absl::nullopt);
+ /*received_extension=*/std::nullopt),
+ std::nullopt);
}
TEST(AbsoluteCaptureTimeInterpolatorTest,
@@ -310,8 +310,8 @@
EXPECT_EQ(interpolator.OnReceivePacket(kSource, kRtpTimestamp1,
/*rtp_clock_frequency_hz=*/0,
- /*received_extension=*/absl::nullopt),
- absl::nullopt);
+ /*received_extension=*/std::nullopt),
+ std::nullopt);
}
TEST(AbsoluteCaptureTimeInterpolatorTest, SkipInterpolateIsSticky) {
@@ -333,13 +333,13 @@
EXPECT_EQ(
interpolator.OnReceivePacket(kSource1, kRtpTimestamp1, kRtpClockFrequency,
- /*received_extension=*/absl::nullopt),
- absl::nullopt);
+ /*received_extension=*/std::nullopt),
+ std::nullopt);
EXPECT_EQ(
interpolator.OnReceivePacket(kSource0, kRtpTimestamp2, kRtpClockFrequency,
- /*received_extension=*/absl::nullopt),
- absl::nullopt);
+ /*received_extension=*/std::nullopt),
+ std::nullopt);
}
} // namespace webrtc
diff --git a/modules/rtp_rtcp/source/absolute_capture_time_sender.cc b/modules/rtp_rtcp/source/absolute_capture_time_sender.cc
index 09f61a0..ae7fe6e 100644
--- a/modules/rtp_rtcp/source/absolute_capture_time_sender.cc
+++ b/modules/rtp_rtcp/source/absolute_capture_time_sender.cc
@@ -31,29 +31,29 @@
return AbsoluteCaptureTimeInterpolator::GetSource(ssrc, csrcs);
}
-absl::optional<AbsoluteCaptureTime> AbsoluteCaptureTimeSender::OnSendPacket(
+std::optional<AbsoluteCaptureTime> AbsoluteCaptureTimeSender::OnSendPacket(
uint32_t source,
uint32_t rtp_timestamp,
uint32_t rtp_clock_frequency,
uint64_t absolute_capture_timestamp,
- absl::optional<int64_t> estimated_capture_clock_offset) {
+ std::optional<int64_t> estimated_capture_clock_offset) {
return OnSendPacket(source, rtp_timestamp, rtp_clock_frequency,
NtpTime(absolute_capture_timestamp),
estimated_capture_clock_offset, /*force=*/false);
}
-absl::optional<AbsoluteCaptureTime> AbsoluteCaptureTimeSender::OnSendPacket(
+std::optional<AbsoluteCaptureTime> AbsoluteCaptureTimeSender::OnSendPacket(
uint32_t source,
uint32_t rtp_timestamp,
int rtp_clock_frequency_hz,
NtpTime absolute_capture_time,
- absl::optional<int64_t> estimated_capture_clock_offset,
+ std::optional<int64_t> estimated_capture_clock_offset,
bool force) {
Timestamp send_time = clock_->CurrentTime();
if (!(force || ShouldSendExtension(
send_time, source, rtp_timestamp, rtp_clock_frequency_hz,
absolute_capture_time, estimated_capture_clock_offset))) {
- return absl::nullopt;
+ return std::nullopt;
}
last_source_ = source;
@@ -75,7 +75,7 @@
uint32_t rtp_timestamp,
int rtp_clock_frequency_hz,
NtpTime absolute_capture_time,
- absl::optional<int64_t> estimated_capture_clock_offset) const {
+ std::optional<int64_t> estimated_capture_clock_offset) const {
// Should if the last sent extension is too old, in particular if we've never
// sent anything before.
if (send_time - last_send_time_ > kInterpolationMaxInterval) {
diff --git a/modules/rtp_rtcp/source/absolute_capture_time_sender.h b/modules/rtp_rtcp/source/absolute_capture_time_sender.h
index a961c3b..595920f 100644
--- a/modules/rtp_rtcp/source/absolute_capture_time_sender.h
+++ b/modules/rtp_rtcp/source/absolute_capture_time_sender.h
@@ -49,7 +49,7 @@
rtc::ArrayView<const uint32_t> csrcs);
// Returns value to write into AbsoluteCaptureTime RTP header extension to be
- // sent, or `absl::nullopt` if the header extension shouldn't be attached to
+ // sent, or `std::nullopt` if the header extension shouldn't be attached to
// the outgoing packet.
//
// - `source` - id of the capture system.
@@ -65,22 +65,22 @@
// i.e. delta of 2^32 represents 1 second. See AbsoluteCaptureTime type
// comments for more details.
// - `force` - when set to true, OnSendPacket is forced to return non-nullopt.
- absl::optional<AbsoluteCaptureTime> OnSendPacket(
+ std::optional<AbsoluteCaptureTime> OnSendPacket(
uint32_t source,
uint32_t rtp_timestamp,
int rtp_clock_frequency_hz,
NtpTime absolute_capture_time,
- absl::optional<int64_t> estimated_capture_clock_offset,
+ std::optional<int64_t> estimated_capture_clock_offset,
bool force = false);
- // Returns a header extension to be sent, or `absl::nullopt` if the header
+ // Returns a header extension to be sent, or `std::nullopt` if the header
// extension shouldn't be sent.
- [[deprecated]] absl::optional<AbsoluteCaptureTime> OnSendPacket(
+ [[deprecated]] std::optional<AbsoluteCaptureTime> OnSendPacket(
uint32_t source,
uint32_t rtp_timestamp,
uint32_t rtp_clock_frequency,
uint64_t absolute_capture_timestamp,
- absl::optional<int64_t> estimated_capture_clock_offset);
+ std::optional<int64_t> estimated_capture_clock_offset);
private:
bool ShouldSendExtension(
@@ -89,7 +89,7 @@
uint32_t rtp_timestamp,
int rtp_clock_frequency_hz,
NtpTime absolute_capture_time,
- absl::optional<int64_t> estimated_capture_clock_offset) const;
+ std::optional<int64_t> estimated_capture_clock_offset) const;
Clock* const clock_;
@@ -99,7 +99,7 @@
uint32_t last_rtp_timestamp_;
int last_rtp_clock_frequency_hz_;
NtpTime last_absolute_capture_time_;
- absl::optional<int64_t> last_estimated_capture_clock_offset_;
+ std::optional<int64_t> last_estimated_capture_clock_offset_;
};
} // namespace webrtc
diff --git a/modules/rtp_rtcp/source/absolute_capture_time_sender_unittest.cc b/modules/rtp_rtcp/source/absolute_capture_time_sender_unittest.cc
index d9146e4..dcb951e 100644
--- a/modules/rtp_rtcp/source/absolute_capture_time_sender_unittest.cc
+++ b/modules/rtp_rtcp/source/absolute_capture_time_sender_unittest.cc
@@ -52,12 +52,12 @@
EXPECT_EQ(sender.OnSendPacket(kSource, kRtpTimestamp1, kRtpClockFrequency,
NtpTime(kExtension1.absolute_capture_timestamp),
kExtension1.estimated_capture_clock_offset),
- absl::nullopt);
+ std::nullopt);
EXPECT_EQ(sender.OnSendPacket(kSource, kRtpTimestamp2, kRtpClockFrequency,
NtpTime(kExtension2.absolute_capture_timestamp),
kExtension2.estimated_capture_clock_offset),
- absl::nullopt);
+ std::nullopt);
}
TEST(AbsoluteCaptureTimeSenderTest, InterpolateEarlierPacketSentLater) {
@@ -84,12 +84,12 @@
EXPECT_EQ(sender.OnSendPacket(kSource, kRtpTimestamp1, kRtpClockFrequency,
NtpTime(kExtension1.absolute_capture_timestamp),
kExtension1.estimated_capture_clock_offset),
- absl::nullopt);
+ std::nullopt);
EXPECT_EQ(sender.OnSendPacket(kSource, kRtpTimestamp2, kRtpClockFrequency,
NtpTime(kExtension2.absolute_capture_timestamp),
kExtension2.estimated_capture_clock_offset),
- absl::nullopt);
+ std::nullopt);
}
TEST(AbsoluteCaptureTimeSenderTest,
@@ -117,12 +117,12 @@
EXPECT_EQ(sender.OnSendPacket(kSource, kRtpTimestamp1, kRtpClockFrequency,
NtpTime(kExtension1.absolute_capture_timestamp),
kExtension1.estimated_capture_clock_offset),
- absl::nullopt);
+ std::nullopt);
EXPECT_EQ(sender.OnSendPacket(kSource, kRtpTimestamp2, kRtpClockFrequency,
NtpTime(kExtension2.absolute_capture_timestamp),
kExtension2.estimated_capture_clock_offset),
- absl::nullopt);
+ std::nullopt);
}
TEST(AbsoluteCaptureTimeSenderTest,
@@ -150,12 +150,12 @@
EXPECT_EQ(sender.OnSendPacket(kSource, kRtpTimestamp1, kRtpClockFrequency,
NtpTime(kExtension1.absolute_capture_timestamp),
kExtension1.estimated_capture_clock_offset),
- absl::nullopt);
+ std::nullopt);
EXPECT_EQ(sender.OnSendPacket(kSource, kRtpTimestamp2, kRtpClockFrequency,
NtpTime(kExtension2.absolute_capture_timestamp),
kExtension2.estimated_capture_clock_offset),
- absl::nullopt);
+ std::nullopt);
}
TEST(AbsoluteCaptureTimeSenderTest, SkipInterpolateIfTooLate) {
@@ -184,7 +184,7 @@
EXPECT_EQ(sender.OnSendPacket(kSource, kRtpTimestamp1, kRtpClockFrequency,
NtpTime(kExtension1.absolute_capture_timestamp),
kExtension1.estimated_capture_clock_offset),
- absl::nullopt);
+ std::nullopt);
clock.AdvanceTime(TimeDelta::Millis(1));
@@ -224,7 +224,7 @@
EXPECT_EQ(sender.OnSendPacket(kSource1, kRtpTimestamp2, kRtpClockFrequency,
NtpTime(kExtension2.absolute_capture_timestamp),
kExtension2.estimated_capture_clock_offset),
- absl::nullopt);
+ std::nullopt);
}
TEST(AbsoluteCaptureTimeSenderTest, SkipInterpolateWhenForced) {
@@ -258,7 +258,7 @@
NtpTime(kExtension2.absolute_capture_timestamp),
kExtension2.estimated_capture_clock_offset,
/*force=*/false),
- absl::nullopt);
+ std::nullopt);
}
TEST(AbsoluteCaptureTimeSenderTest, SkipInterpolateIfRtpClockFrequencyChanged) {
@@ -291,7 +291,7 @@
EXPECT_EQ(sender.OnSendPacket(kSource, kRtpTimestamp2, kRtpClockFrequency1,
NtpTime(kExtension2.absolute_capture_timestamp),
kExtension2.estimated_capture_clock_offset),
- absl::nullopt);
+ std::nullopt);
}
TEST(AbsoluteCaptureTimeSenderTest,
@@ -337,7 +337,7 @@
const AbsoluteCaptureTime kExtension1 = {Int64MsToUQ32x32(9000 + 20),
Int64MsToQ32x32(370)};
const AbsoluteCaptureTime kExtension2 = {Int64MsToUQ32x32(9000 + 40),
- absl::nullopt};
+ std::nullopt};
SimulatedClock clock(0);
AbsoluteCaptureTimeSender sender(&clock);
@@ -388,7 +388,7 @@
EXPECT_EQ(sender.OnSendPacket(kSource, kRtpTimestamp1, kRtpClockFrequency,
NtpTime(kExtension1.absolute_capture_timestamp),
kExtension1.estimated_capture_clock_offset),
- absl::nullopt);
+ std::nullopt);
EXPECT_EQ(sender.OnSendPacket(kSource, kRtpTimestamp2, kRtpClockFrequency,
NtpTime(kExtension2.absolute_capture_timestamp),
diff --git a/modules/rtp_rtcp/source/active_decode_targets_helper.h b/modules/rtp_rtcp/source/active_decode_targets_helper.h
index 13755e8..319fcd6 100644
--- a/modules/rtp_rtcp/source/active_decode_targets_helper.h
+++ b/modules/rtp_rtcp/source/active_decode_targets_helper.h
@@ -14,8 +14,8 @@
#include <stdint.h>
#include <bitset>
+#include <optional>
-#include "absl/types/optional.h"
#include "api/array_view.h"
namespace webrtc {
@@ -41,9 +41,9 @@
rtc::ArrayView<const int> chain_diffs);
// Returns active decode target to attach to the dependency descriptor.
- absl::optional<uint32_t> ActiveDecodeTargetsBitmask() const {
+ std::optional<uint32_t> ActiveDecodeTargetsBitmask() const {
if (unsent_on_chain_.none())
- return absl::nullopt;
+ return std::nullopt;
return last_active_decode_targets_.to_ulong();
}
diff --git a/modules/rtp_rtcp/source/active_decode_targets_helper_unittest.cc b/modules/rtp_rtcp/source/active_decode_targets_helper_unittest.cc
index 6f64fd1..2e7f687 100644
--- a/modules/rtp_rtcp/source/active_decode_targets_helper_unittest.cc
+++ b/modules/rtp_rtcp/source/active_decode_targets_helper_unittest.cc
@@ -10,9 +10,9 @@
#include "modules/rtp_rtcp/source/active_decode_targets_helper.h"
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "test/gtest.h"
namespace webrtc {
@@ -29,7 +29,7 @@
/*active_decode_targets=*/0b11,
/*is_keyframe=*/true, /*frame_id=*/1, chain_diffs);
- EXPECT_EQ(helper.ActiveDecodeTargetsBitmask(), absl::nullopt);
+ EXPECT_EQ(helper.ActiveDecodeTargetsBitmask(), std::nullopt);
}
TEST(ActiveDecodeTargetsHelperTest,
@@ -50,7 +50,7 @@
/*active_decode_targets=*/0b11,
/*is_keyframe=*/true, /*frame_id=*/3, chain_diffs_key);
- EXPECT_EQ(helper.ActiveDecodeTargetsBitmask(), absl::nullopt);
+ EXPECT_EQ(helper.ActiveDecodeTargetsBitmask(), std::nullopt);
}
TEST(ActiveDecodeTargetsHelperTest,
@@ -78,7 +78,7 @@
/*active_decode_targets=*/0b01,
/*is_keyframe=*/false, /*frame_id=*/2, chain_diffs_delta);
- ASSERT_EQ(helper.ActiveDecodeTargetsBitmask(), absl::nullopt);
+ ASSERT_EQ(helper.ActiveDecodeTargetsBitmask(), std::nullopt);
helper.OnFrame(kDecodeTargetProtectedByChain,
/*active_decode_targets=*/0b01,
/*is_keyframe=*/true, /*frame_id=*/3, chain_diffs_key);
@@ -94,12 +94,12 @@
helper.OnFrame(kDecodeTargetProtectedByChain,
/*active_decode_targets=*/kAll,
/*is_keyframe=*/true, /*frame_id=*/1, chain_diffs);
- EXPECT_EQ(helper.ActiveDecodeTargetsBitmask(), absl::nullopt);
+ EXPECT_EQ(helper.ActiveDecodeTargetsBitmask(), std::nullopt);
helper.OnFrame(kDecodeTargetProtectedByChain,
/*active_decode_targets=*/kAll,
/*is_keyframe=*/false, /*frame_id=*/2, chain_diffs);
- EXPECT_EQ(helper.ActiveDecodeTargetsBitmask(), absl::nullopt);
+ EXPECT_EQ(helper.ActiveDecodeTargetsBitmask(), std::nullopt);
}
TEST(ActiveDecodeTargetsHelperTest,
@@ -115,7 +115,7 @@
/*active_decode_targets=*/0b01,
/*is_keyframe=*/false, /*frame_id=*/2, chain_diffs_delta);
- EXPECT_EQ(helper.ActiveDecodeTargetsBitmask(), absl::nullopt);
+ EXPECT_EQ(helper.ActiveDecodeTargetsBitmask(), std::nullopt);
}
TEST(ActiveDecodeTargetsHelperTest, ReturnsNewBitmaskOnDeltaFrame) {
@@ -125,7 +125,7 @@
helper.OnFrame(kDecodeTargetProtectedByChain,
/*active_decode_targets=*/0b11,
/*is_keyframe=*/true, /*frame_id=*/1, chain_diffs_key);
- ASSERT_EQ(helper.ActiveDecodeTargetsBitmask(), absl::nullopt);
+ ASSERT_EQ(helper.ActiveDecodeTargetsBitmask(), std::nullopt);
int chain_diffs_delta[] = {1};
helper.OnFrame(kDecodeTargetProtectedByChain,
/*active_decode_targets=*/0b01,
@@ -142,12 +142,12 @@
helper.OnFrame(kDecodeTargetProtectedByChain,
/*active_decode_targets=*/0b01,
/*is_keyframe=*/true, /*frame_id=*/1, chain_diffs_key);
- ASSERT_NE(helper.ActiveDecodeTargetsBitmask(), absl::nullopt);
+ ASSERT_NE(helper.ActiveDecodeTargetsBitmask(), std::nullopt);
int chain_diffs_delta[] = {1};
helper.OnFrame(kDecodeTargetProtectedByChain,
/*active_decode_targets=*/0b01,
/*is_keyframe=*/false, /*frame_id=*/2, chain_diffs_delta);
- ASSERT_EQ(helper.ActiveDecodeTargetsBitmask(), absl::nullopt);
+ ASSERT_EQ(helper.ActiveDecodeTargetsBitmask(), std::nullopt);
// Reactive all the decode targets
helper.OnFrame(kDecodeTargetProtectedByChain,
@@ -167,7 +167,7 @@
/*active_decode_targets=*/0b111,
/*is_keyframe=*/true,
/*frame_id=*/0, chain_diffs_key);
- ASSERT_EQ(helper.ActiveDecodeTargetsBitmask(), absl::nullopt);
+ ASSERT_EQ(helper.ActiveDecodeTargetsBitmask(), std::nullopt);
int chain_diffs_delta1[] = {1, 1, 1};
helper.OnFrame(kDecodeTargetProtectedByChain,
@@ -190,7 +190,7 @@
/*active_decode_targets=*/kSome,
/*is_keyframe=*/false,
/*frame_id=*/3, chain_diffs_delta3);
- EXPECT_EQ(helper.ActiveDecodeTargetsBitmask(), absl::nullopt);
+ EXPECT_EQ(helper.ActiveDecodeTargetsBitmask(), std::nullopt);
}
TEST(ActiveDecodeTargetsHelperTest, ReturnsBitmaskWhenChanged) {
@@ -232,13 +232,13 @@
helper.OnFrame(kDecodeTargetProtectedByChain, /*active_decode_targets=*/kAll,
/*is_keyframe=*/true,
/*frame_id=*/0, kNoChainDiffs);
- EXPECT_EQ(helper.ActiveDecodeTargetsBitmask(), absl::nullopt);
+ EXPECT_EQ(helper.ActiveDecodeTargetsBitmask(), std::nullopt);
helper.OnFrame(kDecodeTargetProtectedByChain,
/*active_decode_targets=*/0b101,
/*is_keyframe=*/false,
/*frame_id=*/1, kNoChainDiffs);
- EXPECT_EQ(helper.ActiveDecodeTargetsBitmask(), absl::nullopt);
+ EXPECT_EQ(helper.ActiveDecodeTargetsBitmask(), std::nullopt);
}
TEST(ActiveDecodeTargetsHelperTest, Supports32DecodeTargets) {
@@ -261,7 +261,7 @@
/*active_decode_targets=*/some,
/*is_keyframe=*/false,
/*frame_id=*/2, chain_diffs_delta);
- EXPECT_EQ(helper.ActiveDecodeTargetsBitmask(), absl::nullopt);
+ EXPECT_EQ(helper.ActiveDecodeTargetsBitmask(), std::nullopt);
helper.OnFrame(decode_target_protected_by_chain,
/*active_decode_targets=*/kAll,
/*is_keyframe=*/false,
diff --git a/modules/rtp_rtcp/source/capture_clock_offset_updater.cc b/modules/rtp_rtcp/source/capture_clock_offset_updater.cc
index ad935a9..9378520 100644
--- a/modules/rtp_rtcp/source/capture_clock_offset_updater.cc
+++ b/modules/rtp_rtcp/source/capture_clock_offset_updater.cc
@@ -14,12 +14,12 @@
namespace webrtc {
-absl::optional<int64_t>
+std::optional<int64_t>
CaptureClockOffsetUpdater::AdjustEstimatedCaptureClockOffset(
- absl::optional<int64_t> remote_capture_clock_offset) const {
- if (remote_capture_clock_offset == absl::nullopt ||
- remote_to_local_clock_offset_ == absl::nullopt) {
- return absl::nullopt;
+ std::optional<int64_t> remote_capture_clock_offset) const {
+ if (remote_capture_clock_offset == std::nullopt ||
+ remote_to_local_clock_offset_ == std::nullopt) {
+ return std::nullopt;
}
// Do calculations as "unsigned" to make overflows deterministic.
@@ -27,16 +27,16 @@
static_cast<uint64_t>(*remote_to_local_clock_offset_);
}
-absl::optional<TimeDelta> CaptureClockOffsetUpdater::ConvertsToTimeDela(
- absl::optional<int64_t> q32x32) {
- if (q32x32 == absl::nullopt) {
- return absl::nullopt;
+std::optional<TimeDelta> CaptureClockOffsetUpdater::ConvertsToTimeDela(
+ std::optional<int64_t> q32x32) {
+ if (q32x32 == std::nullopt) {
+ return std::nullopt;
}
return TimeDelta::Millis(Q32x32ToInt64Ms(*q32x32));
}
void CaptureClockOffsetUpdater::SetRemoteToLocalClockOffset(
- absl::optional<int64_t> offset_q32x32) {
+ std::optional<int64_t> offset_q32x32) {
remote_to_local_clock_offset_ = offset_q32x32;
}
diff --git a/modules/rtp_rtcp/source/capture_clock_offset_updater.h b/modules/rtp_rtcp/source/capture_clock_offset_updater.h
index 9b28848..5bae8b8 100644
--- a/modules/rtp_rtcp/source/capture_clock_offset_updater.h
+++ b/modules/rtp_rtcp/source/capture_clock_offset_updater.h
@@ -13,7 +13,8 @@
#include <stdint.h>
-#include "absl/types/optional.h"
+#include <optional>
+
#include "api/units/time_delta.h"
namespace webrtc {
@@ -32,8 +33,8 @@
// Adjusts remote_capture_clock_offset, which originates from Absolute Capture
// Time RTP header extension, to get the local clock offset against the
// capturer's clock.
- absl::optional<int64_t> AdjustEstimatedCaptureClockOffset(
- absl::optional<int64_t> remote_capture_clock_offset) const;
+ std::optional<int64_t> AdjustEstimatedCaptureClockOffset(
+ std::optional<int64_t> remote_capture_clock_offset) const;
// Sets the NTP clock offset between the sender system (which may be different
// from the capture system) and the local system. This information is normally
@@ -41,14 +42,14 @@
// by RTCP sender reports (see DLSR/DLRR).
//
// Note that the value must be in Q32.32-formatted fixed-point seconds.
- void SetRemoteToLocalClockOffset(absl::optional<int64_t> offset_q32x32);
+ void SetRemoteToLocalClockOffset(std::optional<int64_t> offset_q32x32);
// Converts a signed Q32.32-formatted fixed-point to a TimeDelta.
- static absl::optional<TimeDelta> ConvertsToTimeDela(
- absl::optional<int64_t> q32x32);
+ static std::optional<TimeDelta> ConvertsToTimeDela(
+ std::optional<int64_t> q32x32);
private:
- absl::optional<int64_t> remote_to_local_clock_offset_;
+ std::optional<int64_t> remote_to_local_clock_offset_;
};
} // namespace webrtc
diff --git a/modules/rtp_rtcp/source/capture_clock_offset_updater_unittest.cc b/modules/rtp_rtcp/source/capture_clock_offset_updater_unittest.cc
index f6bea4b..f05dae2 100644
--- a/modules/rtp_rtcp/source/capture_clock_offset_updater_unittest.cc
+++ b/modules/rtp_rtcp/source/capture_clock_offset_updater_unittest.cc
@@ -18,24 +18,24 @@
TEST(AbsoluteCaptureTimeReceiverTest,
SkipEstimatedCaptureClockOffsetIfRemoteToLocalClockOffsetIsUnknown) {
- static const absl::optional<int64_t> kRemoteCaptureClockOffset =
+ static const std::optional<int64_t> kRemoteCaptureClockOffset =
Int64MsToQ32x32(-350);
CaptureClockOffsetUpdater updater;
- updater.SetRemoteToLocalClockOffset(absl::nullopt);
+ updater.SetRemoteToLocalClockOffset(std::nullopt);
EXPECT_EQ(
updater.AdjustEstimatedCaptureClockOffset(kRemoteCaptureClockOffset),
- absl::nullopt);
+ std::nullopt);
}
TEST(AbsoluteCaptureTimeReceiverTest,
SkipEstimatedCaptureClockOffsetIfRemoteCaptureClockOffsetIsUnknown) {
- static const absl::optional<int64_t> kCaptureClockOffsetNull = absl::nullopt;
+ static const std::optional<int64_t> kCaptureClockOffsetNull = std::nullopt;
CaptureClockOffsetUpdater updater;
updater.SetRemoteToLocalClockOffset(0);
EXPECT_EQ(updater.AdjustEstimatedCaptureClockOffset(kCaptureClockOffsetNull),
kCaptureClockOffsetNull);
- static const absl::optional<int64_t> kRemoteCaptureClockOffset =
+ static const std::optional<int64_t> kRemoteCaptureClockOffset =
Int64MsToQ32x32(-350);
EXPECT_EQ(
updater.AdjustEstimatedCaptureClockOffset(kRemoteCaptureClockOffset),
@@ -43,9 +43,9 @@
}
TEST(AbsoluteCaptureTimeReceiverTest, EstimatedCaptureClockOffsetArithmetic) {
- static const absl::optional<int64_t> kRemoteCaptureClockOffset =
+ static const std::optional<int64_t> kRemoteCaptureClockOffset =
Int64MsToQ32x32(-350);
- static const absl::optional<int64_t> kRemoteToLocalClockOffset =
+ static const std::optional<int64_t> kRemoteToLocalClockOffset =
Int64MsToQ32x32(-7000007);
CaptureClockOffsetUpdater updater;
updater.SetRemoteToLocalClockOffset(kRemoteToLocalClockOffset);
@@ -63,7 +63,7 @@
constexpr int64_t kPositiveQ32x32 =
kPositive.ms() * (NtpTime::kFractionsPerSecond / 1000);
constexpr TimeDelta kEpsilon = TimeDelta::Millis(1);
- absl::optional<TimeDelta> converted =
+ std::optional<TimeDelta> converted =
CaptureClockOffsetUpdater::ConvertsToTimeDela(kNegativeQ32x32);
EXPECT_GT(converted, kNegative - kEpsilon);
EXPECT_LT(converted, kNegative + kEpsilon);
@@ -73,7 +73,7 @@
EXPECT_LT(converted, kPositive + kEpsilon);
EXPECT_FALSE(
- CaptureClockOffsetUpdater::ConvertsToTimeDela(absl::nullopt).has_value());
+ CaptureClockOffsetUpdater::ConvertsToTimeDela(std::nullopt).has_value());
}
} // namespace webrtc
diff --git a/modules/rtp_rtcp/source/corruption_detection_extension_unittest.cc b/modules/rtp_rtcp/source/corruption_detection_extension_unittest.cc
index be47e84..26d7006 100644
--- a/modules/rtp_rtcp/source/corruption_detection_extension_unittest.cc
+++ b/modules/rtp_rtcp/source/corruption_detection_extension_unittest.cc
@@ -12,8 +12,8 @@
#include <cstddef>
#include <cstdint>
+#include <optional>
-#include "absl/types/optional.h"
#include "common_video/corruption_detection_message.h"
#include "test/gmock.h"
#include "test/gtest.h"
@@ -25,7 +25,7 @@
using ::testing::ElementsAre;
TEST(CorruptionDetectionExtensionTest, ValueSizeIs1UnlessSamplesAreSpecified) {
- const absl::optional<CorruptionDetectionMessage> kMessage =
+ const std::optional<CorruptionDetectionMessage> kMessage =
CorruptionDetectionMessage::Builder()
.WithSequenceIndex(0b0110'1111)
.WithInterpretSequenceIndexAsMostSignificantBits(true)
@@ -33,53 +33,53 @@
.WithSampleValues({})
.Build();
- ASSERT_NE(kMessage, absl::nullopt);
+ ASSERT_NE(kMessage, std::nullopt);
EXPECT_EQ(CorruptionDetectionExtension::ValueSize(*kMessage), size_t{1});
}
TEST(CorruptionDetectionExtensionTest,
GivenSamplesTheValueSizeIsTheSumOfTheNumberOfSamplesPlus3) {
const double kSampleValues[] = {1.0, 2.0, 3.0, 4.0};
- const absl::optional<CorruptionDetectionMessage> kMessage =
+ const std::optional<CorruptionDetectionMessage> kMessage =
CorruptionDetectionMessage::Builder()
.WithSampleValues(kSampleValues)
.Build();
- ASSERT_NE(kMessage, absl::nullopt);
+ ASSERT_NE(kMessage, std::nullopt);
EXPECT_EQ(CorruptionDetectionExtension::ValueSize(*kMessage), size_t{7});
}
TEST(CorruptionDetectionExtensionTest,
WritesMandatoryWhenEnoughMemoryIsAllocatedWithoutSamples) {
- const absl::optional<CorruptionDetectionMessage> kMessage =
+ const std::optional<CorruptionDetectionMessage> kMessage =
CorruptionDetectionMessage::Builder()
.WithSequenceIndex(0b0110'1111)
.WithInterpretSequenceIndexAsMostSignificantBits(true)
.Build();
uint8_t data[] = {0};
- ASSERT_NE(kMessage, absl::nullopt);
+ ASSERT_NE(kMessage, std::nullopt);
EXPECT_TRUE(CorruptionDetectionExtension::Write(data, *kMessage));
EXPECT_THAT(data, ElementsAre(0b1110'1111));
}
TEST(CorruptionDetectionExtensionTest,
FailsToWriteWhenTooMuchMemoryIsAllocatedWithoutSamples) {
- const absl::optional<CorruptionDetectionMessage> kMessage =
+ const std::optional<CorruptionDetectionMessage> kMessage =
CorruptionDetectionMessage::Builder()
.WithSequenceIndex(0b0110'1111)
.WithInterpretSequenceIndexAsMostSignificantBits(true)
.Build();
uint8_t data[] = {0, 0, 0};
- ASSERT_NE(kMessage, absl::nullopt);
+ ASSERT_NE(kMessage, std::nullopt);
EXPECT_FALSE(CorruptionDetectionExtension::Write(data, *kMessage));
}
TEST(CorruptionDetectionExtensionTest,
FailsToWriteWhenTooMuchMemoryIsAllocatedWithSamples) {
const double kSampleValues[] = {1.0};
- const absl::optional<CorruptionDetectionMessage> kMessage =
+ const std::optional<CorruptionDetectionMessage> kMessage =
CorruptionDetectionMessage::Builder()
.WithSequenceIndex(0b0110'1111)
.WithInterpretSequenceIndexAsMostSignificantBits(true)
@@ -88,14 +88,14 @@
.Build();
uint8_t data[] = {0, 0, 0, 0, 0};
- ASSERT_NE(kMessage, absl::nullopt);
+ ASSERT_NE(kMessage, std::nullopt);
EXPECT_FALSE(CorruptionDetectionExtension::Write(data, *kMessage));
}
TEST(CorruptionDetectionExtensionTest,
WritesEverythingWhenEnoughMemoryIsAllocatedWithSamples) {
const double kSampleValues[] = {1.0};
- const absl::optional<CorruptionDetectionMessage> kMessage =
+ const std::optional<CorruptionDetectionMessage> kMessage =
CorruptionDetectionMessage::Builder()
.WithSequenceIndex(0b0110'1111)
.WithInterpretSequenceIndexAsMostSignificantBits(true)
@@ -104,7 +104,7 @@
.Build();
uint8_t data[] = {0, 0, 0, 0};
- ASSERT_NE(kMessage, absl::nullopt);
+ ASSERT_NE(kMessage, std::nullopt);
EXPECT_TRUE(CorruptionDetectionExtension::Write(data, *kMessage));
EXPECT_THAT(data, ElementsAre(0b1110'1111, 51, 0, 1));
}
@@ -113,7 +113,7 @@
WritesEverythingToExtensionWhenUpperBitsAreUsedForSequenceIndex) {
const double kSampleValues[] = {1.0, 2.0, 3.0, 4.0, 5.0, 6.0, 7.0,
8.0, 9.0, 10.0, 11.0, 12.0, 13.0};
- const absl::optional<CorruptionDetectionMessage> kMessage =
+ const std::optional<CorruptionDetectionMessage> kMessage =
CorruptionDetectionMessage::Builder()
.WithSequenceIndex(0b0110'1111)
.WithInterpretSequenceIndexAsMostSignificantBits(true)
@@ -124,7 +124,7 @@
.Build();
uint8_t data[16];
- ASSERT_NE(kMessage, absl::nullopt);
+ ASSERT_NE(kMessage, std::nullopt);
EXPECT_TRUE(CorruptionDetectionExtension::Write(data, *kMessage));
EXPECT_THAT(data, ElementsAre(0b1110'1111, 220, 0b1110'1111, 1, 2, 3, 4, 5, 6,
7, 8, 9, 10, 11, 12, 13));
@@ -134,7 +134,7 @@
WritesEverythingToExtensionWhenLowerBitsAreUsedForSequenceIndex) {
const double kSampleValues[] = {1.0, 2.0, 3.0, 4.0, 5.0, 6.0, 7.0,
8.0, 9.0, 10.0, 11.0, 12.0, 13.0};
- const absl::optional<CorruptionDetectionMessage> kMessage =
+ const std::optional<CorruptionDetectionMessage> kMessage =
CorruptionDetectionMessage::Builder()
.WithSequenceIndex(0b0110'1111)
.WithInterpretSequenceIndexAsMostSignificantBits(false)
@@ -145,7 +145,7 @@
.Build();
uint8_t data[16];
- ASSERT_NE(kMessage, absl::nullopt);
+ ASSERT_NE(kMessage, std::nullopt);
EXPECT_TRUE(CorruptionDetectionExtension::Write(data, *kMessage));
EXPECT_THAT(data, ElementsAre(0b0110'1111, 220, 0b1110'1111, 1, 2, 3, 4, 5, 6,
7, 8, 9, 10, 11, 12, 13));
@@ -153,13 +153,13 @@
TEST(CorruptionDetectionExtensionTest, TruncatesSampleValuesWhenWriting) {
const double kSampleValues[] = {1.4, 2.5, 3.6};
- const absl::optional<CorruptionDetectionMessage> kMessage =
+ const std::optional<CorruptionDetectionMessage> kMessage =
CorruptionDetectionMessage::Builder()
.WithSampleValues(kSampleValues)
.Build();
uint8_t data[6];
- ASSERT_NE(kMessage, absl::nullopt);
+ ASSERT_NE(kMessage, std::nullopt);
EXPECT_TRUE(CorruptionDetectionExtension::Write(data, *kMessage));
EXPECT_THAT(data, ElementsAre(0, 0, 0, 1, 2, 3));
}
diff --git a/modules/rtp_rtcp/source/deprecated/deprecated_rtp_sender_egress.cc b/modules/rtp_rtcp/source/deprecated/deprecated_rtp_sender_egress.cc
index 445b141..5566aee 100644
--- a/modules/rtp_rtcp/source/deprecated/deprecated_rtp_sender_egress.cc
+++ b/modules/rtp_rtcp/source/deprecated/deprecated_rtp_sender_egress.cc
@@ -62,7 +62,7 @@
: ssrc_(config.local_media_ssrc),
rtx_ssrc_(config.rtx_send_ssrc),
flexfec_ssrc_(config.fec_generator ? config.fec_generator->FecSsrc()
- : absl::nullopt),
+ : std::nullopt),
populate_network2_timestamp_(config.populate_network2_timestamp),
clock_(config.clock),
packet_history_(packet_history),
diff --git a/modules/rtp_rtcp/source/deprecated/deprecated_rtp_sender_egress.h b/modules/rtp_rtcp/source/deprecated/deprecated_rtp_sender_egress.h
index d29a434..7ed0d6c 100644
--- a/modules/rtp_rtcp/source/deprecated/deprecated_rtp_sender_egress.h
+++ b/modules/rtp_rtcp/source/deprecated/deprecated_rtp_sender_egress.h
@@ -13,9 +13,9 @@
#include <map>
#include <memory>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/call/transport.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/units/data_rate.h"
@@ -58,8 +58,8 @@
void SendPacket(RtpPacketToSend* packet, const PacedPacketInfo& pacing_info)
RTC_LOCKS_EXCLUDED(lock_);
uint32_t Ssrc() const { return ssrc_; }
- absl::optional<uint32_t> RtxSsrc() const { return rtx_ssrc_; }
- absl::optional<uint32_t> FlexFecSsrc() const { return flexfec_ssrc_; }
+ std::optional<uint32_t> RtxSsrc() const { return rtx_ssrc_; }
+ std::optional<uint32_t> FlexFecSsrc() const { return flexfec_ssrc_; }
void ProcessBitrateAndNotifyObservers() RTC_LOCKS_EXCLUDED(lock_);
RtpSendRates GetSendRates() const RTC_LOCKS_EXCLUDED(lock_);
@@ -99,8 +99,8 @@
RTC_EXCLUSIVE_LOCKS_REQUIRED(lock_);
const uint32_t ssrc_;
- const absl::optional<uint32_t> rtx_ssrc_;
- const absl::optional<uint32_t> flexfec_ssrc_;
+ const std::optional<uint32_t> rtx_ssrc_;
+ const std::optional<uint32_t> flexfec_ssrc_;
const bool populate_network2_timestamp_;
Clock* const clock_;
RtpPacketHistory* const packet_history_;
diff --git a/modules/rtp_rtcp/source/flexfec_sender.cc b/modules/rtp_rtcp/source/flexfec_sender.cc
index ab99d28..0f3c661 100644
--- a/modules/rtp_rtcp/source/flexfec_sender.cc
+++ b/modules/rtp_rtcp/source/flexfec_sender.cc
@@ -196,7 +196,7 @@
.value_or(DataRate::Zero());
}
-absl::optional<RtpState> FlexfecSender::GetRtpState() {
+std::optional<RtpState> FlexfecSender::GetRtpState() {
RtpState rtp_state;
rtp_state.sequence_number = seq_num_;
rtp_state.start_timestamp = timestamp_offset_;
diff --git a/modules/rtp_rtcp/source/frame_object.cc b/modules/rtp_rtcp/source/frame_object.cc
index 23abe3a..5fa8225 100644
--- a/modules/rtp_rtcp/source/frame_object.cc
+++ b/modules/rtp_rtcp/source/frame_object.cc
@@ -34,7 +34,7 @@
VideoRotation rotation,
VideoContentType content_type,
const RTPVideoHeader& video_header,
- const absl::optional<webrtc::ColorSpace>& color_space,
+ const std::optional<webrtc::ColorSpace>& color_space,
RtpPacketInfos packet_infos,
rtc::scoped_refptr<EncodedImageBuffer> image_buffer)
: image_buffer_(image_buffer),
diff --git a/modules/rtp_rtcp/source/frame_object.h b/modules/rtp_rtcp/source/frame_object.h
index 481c561..4c57b69 100644
--- a/modules/rtp_rtcp/source/frame_object.h
+++ b/modules/rtp_rtcp/source/frame_object.h
@@ -11,9 +11,9 @@
#ifndef MODULES_RTP_RTCP_SOURCE_FRAME_OBJECT_H_
#define MODULES_RTP_RTCP_SOURCE_FRAME_OBJECT_H_
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/video/encoded_frame.h"
#include "api/video/video_frame_metadata.h"
@@ -35,7 +35,7 @@
VideoRotation rotation,
VideoContentType content_type,
const RTPVideoHeader& video_header,
- const absl::optional<webrtc::ColorSpace>& color_space,
+ const std::optional<webrtc::ColorSpace>& color_space,
RtpPacketInfos packet_infos,
rtc::scoped_refptr<EncodedImageBuffer> image_buffer);
diff --git a/modules/rtp_rtcp/source/packet_sequencer.cc b/modules/rtp_rtcp/source/packet_sequencer.cc
index f640351..4d1ac6c 100644
--- a/modules/rtp_rtcp/source/packet_sequencer.cc
+++ b/modules/rtp_rtcp/source/packet_sequencer.cc
@@ -24,7 +24,7 @@
} // namespace
PacketSequencer::PacketSequencer(uint32_t media_ssrc,
- absl::optional<uint32_t> rtx_ssrc,
+ std::optional<uint32_t> rtx_ssrc,
bool require_marker_before_media_padding,
Clock* clock)
: media_ssrc_(media_ssrc),
diff --git a/modules/rtp_rtcp/source/packet_sequencer.h b/modules/rtp_rtcp/source/packet_sequencer.h
index 0ae069d..a6e1ba7 100644
--- a/modules/rtp_rtcp/source/packet_sequencer.h
+++ b/modules/rtp_rtcp/source/packet_sequencer.h
@@ -28,7 +28,7 @@
// last packets of a video frame).
// Packets with unknown SSRCs will be ignored.
PacketSequencer(uint32_t media_ssrc,
- absl::optional<uint32_t> rtx_ssrc,
+ std::optional<uint32_t> rtx_ssrc,
bool require_marker_before_media_padding,
Clock* clock);
@@ -58,7 +58,7 @@
void PopulatePaddingFields(RtpPacketToSend& packet);
const uint32_t media_ssrc_;
- const absl::optional<uint32_t> rtx_ssrc_;
+ const std::optional<uint32_t> rtx_ssrc_;
const bool require_marker_before_media_padding_;
Clock* const clock_;
diff --git a/modules/rtp_rtcp/source/receive_statistics_impl.cc b/modules/rtp_rtcp/source/receive_statistics_impl.cc
index 1ca7795..abdd76c 100644
--- a/modules/rtp_rtcp/source/receive_statistics_impl.cc
+++ b/modules/rtp_rtcp/source/receive_statistics_impl.cc
@@ -72,7 +72,7 @@
--cumulative_loss_;
uint16_t expected_sequence_number = *received_seq_out_of_order_ + 1;
- received_seq_out_of_order_ = absl::nullopt;
+ received_seq_out_of_order_ = std::nullopt;
if (packet.SequenceNumber() == expected_sequence_number) {
// Ignore sequence number gap caused by stream restart for packet loss
// calculation, by setting received_seq_max_ to the sequence number just
@@ -283,13 +283,13 @@
last_report_seq_max_ = received_seq_max_;
}
-absl::optional<int> StreamStatisticianImpl::GetFractionLostInPercent() const {
+std::optional<int> StreamStatisticianImpl::GetFractionLostInPercent() const {
if (!ReceivedRtpPacket()) {
- return absl::nullopt;
+ return std::nullopt;
}
int64_t expected_packets = 1 + received_seq_max_ - received_seq_first_;
if (expected_packets <= 0) {
- return absl::nullopt;
+ return std::nullopt;
}
if (cumulative_loss_ <= 0) {
return 0;
diff --git a/modules/rtp_rtcp/source/receive_statistics_impl.h b/modules/rtp_rtcp/source/receive_statistics_impl.h
index ccac2d5..a6e2d8b 100644
--- a/modules/rtp_rtcp/source/receive_statistics_impl.h
+++ b/modules/rtp_rtcp/source/receive_statistics_impl.h
@@ -14,10 +14,10 @@
#include <algorithm>
#include <functional>
#include <memory>
+#include <optional>
#include <utility>
#include <vector>
-#include "absl/types/optional.h"
#include "api/units/time_delta.h"
#include "api/units/timestamp.h"
#include "modules/rtp_rtcp/include/receive_statistics.h"
@@ -51,7 +51,7 @@
// Implements StreamStatistician
RtpReceiveStats GetStats() const override;
- absl::optional<int> GetFractionLostInPercent() const override;
+ std::optional<int> GetFractionLostInPercent() const override;
StreamDataCounters GetReceiveStreamDataCounters() const override;
uint32_t BitrateReceived() const override;
@@ -96,14 +96,14 @@
// senders, in particular, our own loss-based bandwidth estimator.
int32_t cumulative_loss_rtcp_offset_;
- absl::optional<Timestamp> last_receive_time_;
+ std::optional<Timestamp> last_receive_time_;
uint32_t last_received_timestamp_;
RtpSequenceNumberUnwrapper seq_unwrapper_;
int64_t received_seq_first_;
int64_t received_seq_max_;
// Assume that the other side restarted when there are two sequential packets
// with large jump from received_seq_max_.
- absl::optional<uint16_t> received_seq_out_of_order_;
+ std::optional<uint16_t> received_seq_out_of_order_;
// Current counter values.
StreamDataCounters receive_counters_;
@@ -129,7 +129,7 @@
MutexLock lock(&stream_lock_);
return impl_.GetStats();
}
- absl::optional<int> GetFractionLostInPercent() const override {
+ std::optional<int> GetFractionLostInPercent() const override {
MutexLock lock(&stream_lock_);
return impl_.GetFractionLostInPercent();
}
diff --git a/modules/rtp_rtcp/source/remote_ntp_time_estimator.cc b/modules/rtp_rtcp/source/remote_ntp_time_estimator.cc
index d70882c..21ef24a 100644
--- a/modules/rtp_rtcp/source/remote_ntp_time_estimator.cc
+++ b/modules/rtp_rtcp/source/remote_ntp_time_estimator.cc
@@ -97,11 +97,11 @@
return receiver_capture;
}
-absl::optional<int64_t>
+std::optional<int64_t>
RemoteNtpTimeEstimator::EstimateRemoteToLocalClockOffset() {
if (ntp_clocks_offset_estimator_.GetNumberOfSamplesStored() <
kMinimumNumberOfSamples) {
- return absl::nullopt;
+ return std::nullopt;
}
return ntp_clocks_offset_estimator_.GetFilteredValue();
}
diff --git a/modules/rtp_rtcp/source/remote_ntp_time_estimator_unittest.cc b/modules/rtp_rtcp/source/remote_ntp_time_estimator_unittest.cc
index b356da7..e9e8ce9 100644
--- a/modules/rtp_rtcp/source/remote_ntp_time_estimator_unittest.cc
+++ b/modules/rtp_rtcp/source/remote_ntp_time_estimator_unittest.cc
@@ -10,7 +10,8 @@
#include "modules/rtp_rtcp/include/remote_ntp_time_estimator.h"
-#include "absl/types/optional.h"
+#include <optional>
+
#include "modules/rtp_rtcp/source/ntp_time_util.h"
#include "system_wrappers/include/clock.h"
#include "system_wrappers/include/ntp_time.h"
@@ -78,7 +79,7 @@
// Local peer needs at least 2 RTCP SR to calculate the capture time.
const int64_t kNotEnoughRtcpSr = -1;
EXPECT_EQ(kNotEnoughRtcpSr, estimator_.Estimate(rtp_timestamp));
- EXPECT_EQ(estimator_.EstimateRemoteToLocalClockOffset(), absl::nullopt);
+ EXPECT_EQ(estimator_.EstimateRemoteToLocalClockOffset(), std::nullopt);
AdvanceTime(TimeDelta::Millis(800));
// Remote sends second RTCP SR.
diff --git a/modules/rtp_rtcp/source/rtcp_packet/extended_reports.cc b/modules/rtp_rtcp/source/rtcp_packet/extended_reports.cc
index ce57bd5..84a3a53 100644
--- a/modules/rtp_rtcp/source/rtcp_packet/extended_reports.cc
+++ b/modules/rtp_rtcp/source/rtcp_packet/extended_reports.cc
@@ -59,7 +59,7 @@
SetSenderSsrc(ByteReader<uint32_t>::ReadBigEndian(packet.payload()));
rrtr_block_.reset();
dlrr_block_.ClearItems();
- target_bitrate_ = absl::nullopt;
+ target_bitrate_ = std::nullopt;
const uint8_t* current_block = packet.payload() + kXrBaseLength;
const uint8_t* const packet_end =
diff --git a/modules/rtp_rtcp/source/rtcp_packet/extended_reports.h b/modules/rtp_rtcp/source/rtcp_packet/extended_reports.h
index 6c804bb..d71e759 100644
--- a/modules/rtp_rtcp/source/rtcp_packet/extended_reports.h
+++ b/modules/rtp_rtcp/source/rtcp_packet/extended_reports.h
@@ -11,9 +11,9 @@
#ifndef MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_EXTENDED_REPORTS_H_
#define MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_EXTENDED_REPORTS_H_
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "modules/rtp_rtcp/source/rtcp_packet.h"
#include "modules/rtp_rtcp/source/rtcp_packet/dlrr.h"
#include "modules/rtp_rtcp/source/rtcp_packet/rrtr.h"
@@ -40,9 +40,9 @@
bool AddDlrrItem(const ReceiveTimeInfo& time_info);
void SetTargetBitrate(const TargetBitrate& target_bitrate);
- const absl::optional<Rrtr>& rrtr() const { return rrtr_block_; }
+ const std::optional<Rrtr>& rrtr() const { return rrtr_block_; }
const Dlrr& dlrr() const { return dlrr_block_; }
- const absl::optional<TargetBitrate>& target_bitrate() const {
+ const std::optional<TargetBitrate>& target_bitrate() const {
return target_bitrate_;
}
@@ -64,9 +64,9 @@
void ParseDlrrBlock(const uint8_t* block, uint16_t block_length);
void ParseTargetBitrateBlock(const uint8_t* block, uint16_t block_length);
- absl::optional<Rrtr> rrtr_block_;
+ std::optional<Rrtr> rrtr_block_;
Dlrr dlrr_block_; // Dlrr without items treated same as no dlrr block.
- absl::optional<TargetBitrate> target_bitrate_;
+ std::optional<TargetBitrate> target_bitrate_;
};
} // namespace rtcp
} // namespace webrtc
diff --git a/modules/rtp_rtcp/source/rtcp_packet/target_bitrate_unittest.cc b/modules/rtp_rtcp/source/rtcp_packet/target_bitrate_unittest.cc
index b16bb5b..0073e0e 100644
--- a/modules/rtp_rtcp/source/rtcp_packet/target_bitrate_unittest.cc
+++ b/modules/rtp_rtcp/source/rtcp_packet/target_bitrate_unittest.cc
@@ -67,7 +67,7 @@
rtcp::ExtendedReports xr;
EXPECT_TRUE(ParseSinglePacket(kRtcpPacket, &xr));
EXPECT_EQ(kSsrc, xr.sender_ssrc());
- const absl::optional<TargetBitrate>& target_bitrate = xr.target_bitrate();
+ const std::optional<TargetBitrate>& target_bitrate = xr.target_bitrate();
ASSERT_TRUE(static_cast<bool>(target_bitrate));
CheckBitrateItems(target_bitrate->GetTargetBitrates());
}
diff --git a/modules/rtp_rtcp/source/rtcp_packet/transport_feedback_unittest.cc b/modules/rtp_rtcp/source/rtcp_packet/transport_feedback_unittest.cc
index 92ca464..264d613 100644
--- a/modules/rtp_rtcp/source/rtcp_packet/transport_feedback_unittest.cc
+++ b/modules/rtp_rtcp/source/rtcp_packet/transport_feedback_unittest.cc
@@ -12,9 +12,9 @@
#include <limits>
#include <memory>
+#include <optional>
#include <utility>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/units/time_delta.h"
#include "api/units/timestamp.h"
@@ -170,7 +170,7 @@
std::vector<TimeDelta> expected_deltas_;
size_t expected_size_;
TimeDelta default_delta_;
- absl::optional<TransportFeedback> feedback_;
+ std::optional<TransportFeedback> feedback_;
rtc::Buffer serialized_;
bool include_timestamps_;
};
diff --git a/modules/rtp_rtcp/source/rtcp_receiver.cc b/modules/rtp_rtcp/source/rtcp_receiver.cc
index 983545a..14653cb 100644
--- a/modules/rtp_rtcp/source/rtcp_receiver.cc
+++ b/modules/rtp_rtcp/source/rtcp_receiver.cc
@@ -101,7 +101,7 @@
ssrcs_.push_back(*config.rtx_send_ssrc);
}
if (config.fec_generator) {
- absl::optional<uint32_t> flexfec_ssrc = config.fec_generator->FecSsrc();
+ std::optional<uint32_t> flexfec_ssrc = config.fec_generator->FecSsrc();
if (flexfec_ssrc) {
ssrcs_.push_back(*flexfec_ssrc);
}
@@ -131,12 +131,12 @@
uint32_t remote_ssrc = 0;
std::vector<uint16_t> nack_sequence_numbers;
std::vector<ReportBlockData> report_block_datas;
- absl::optional<TimeDelta> rtt;
+ std::optional<TimeDelta> rtt;
uint32_t receiver_estimated_max_bitrate_bps = 0;
std::unique_ptr<rtcp::TransportFeedback> transport_feedback;
- absl::optional<rtcp::CongestionControlFeedback> congestion_control_feedback;
- absl::optional<VideoBitrateAllocation> target_bitrate_allocation;
- absl::optional<NetworkStateEstimate> network_state_estimate;
+ std::optional<rtcp::CongestionControlFeedback> congestion_control_feedback;
+ std::optional<VideoBitrateAllocation> target_bitrate_allocation;
+ std::optional<NetworkStateEstimate> network_state_estimate;
std::unique_ptr<rtcp::LossNotification> loss_notification;
};
@@ -258,20 +258,20 @@
++num_rtts_;
}
-absl::optional<TimeDelta> RTCPReceiver::AverageRtt() const {
+std::optional<TimeDelta> RTCPReceiver::AverageRtt() const {
MutexLock lock(&rtcp_receiver_lock_);
auto it = rtts_.find(remote_ssrc_);
if (it == rtts_.end()) {
- return absl::nullopt;
+ return std::nullopt;
}
return it->second.average_rtt();
}
-absl::optional<TimeDelta> RTCPReceiver::LastRtt() const {
+std::optional<TimeDelta> RTCPReceiver::LastRtt() const {
MutexLock lock(&rtcp_receiver_lock_);
auto it = rtts_.find(remote_ssrc_);
if (it == rtts_.end()) {
- return absl::nullopt;
+ return std::nullopt;
}
return it->second.last_rtt();
}
@@ -290,19 +290,18 @@
xr_rrtr_status_ = enabled;
}
-absl::optional<TimeDelta> RTCPReceiver::GetAndResetXrRrRtt() {
+std::optional<TimeDelta> RTCPReceiver::GetAndResetXrRrRtt() {
MutexLock lock(&rtcp_receiver_lock_);
- absl::optional<TimeDelta> rtt = xr_rr_rtt_;
- xr_rr_rtt_ = absl::nullopt;
+ std::optional<TimeDelta> rtt = xr_rr_rtt_;
+ xr_rr_rtt_ = std::nullopt;
return rtt;
}
// Called regularly (1/sec) on the worker thread to do rtt calculations.
-absl::optional<TimeDelta> RTCPReceiver::OnPeriodicRttUpdate(
- Timestamp newer_than,
- bool sending) {
+std::optional<TimeDelta> RTCPReceiver::OnPeriodicRttUpdate(Timestamp newer_than,
+ bool sending) {
// Running on the worker thread (same as construction thread).
- absl::optional<TimeDelta> rtt;
+ std::optional<TimeDelta> rtt;
if (sending) {
// Check if we've received a report block within the last kRttUpdateInterval
@@ -336,11 +335,11 @@
return rtt;
}
-absl::optional<RtpRtcpInterface::SenderReportStats>
+std::optional<RtpRtcpInterface::SenderReportStats>
RTCPReceiver::GetSenderReportStats() const {
MutexLock lock(&rtcp_receiver_lock_);
if (!remote_sender_.last_arrival_timestamp.Valid()) {
- return absl::nullopt;
+ return std::nullopt;
}
return remote_sender_;
@@ -799,7 +798,7 @@
received_rrtrs_.erase(it->second);
received_rrtrs_ssrc_it_.erase(it);
}
- xr_rr_rtt_ = absl::nullopt;
+ xr_rr_rtt_ = std::nullopt;
return true;
}
diff --git a/modules/rtp_rtcp/source/rtcp_receiver.h b/modules/rtp_rtcp/source/rtcp_receiver.h
index 6bfd1e1..361822a 100644
--- a/modules/rtp_rtcp/source/rtcp_receiver.h
+++ b/modules/rtp_rtcp/source/rtcp_receiver.h
@@ -13,10 +13,10 @@
#include <list>
#include <map>
+#include <optional>
#include <string>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/sequence_checker.h"
#include "api/units/time_delta.h"
@@ -75,7 +75,7 @@
}
void Invalidate() { round_trip_time_.reset(); }
// https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-roundtriptime
- absl::optional<TimeDelta> round_trip_time() const {
+ std::optional<TimeDelta> round_trip_time() const {
return round_trip_time_;
}
// https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-totalroundtriptime
@@ -86,7 +86,7 @@
}
private:
- absl::optional<TimeDelta> round_trip_time_;
+ std::optional<TimeDelta> round_trip_time_;
TimeDelta total_round_trip_time_ = TimeDelta::Zero();
int round_trip_time_measurements_ = 0;
};
@@ -112,24 +112,24 @@
bool receiver_only() const { return receiver_only_; }
// Returns stats based on the received RTCP Sender Reports.
- absl::optional<RtpRtcpInterface::SenderReportStats> GetSenderReportStats()
+ std::optional<RtpRtcpInterface::SenderReportStats> GetSenderReportStats()
const;
std::vector<rtcp::ReceiveTimeInfo> ConsumeReceivedXrReferenceTimeInfo();
- absl::optional<TimeDelta> AverageRtt() const;
- absl::optional<TimeDelta> LastRtt() const;
+ std::optional<TimeDelta> AverageRtt() const;
+ std::optional<TimeDelta> LastRtt() const;
// Returns non-sender RTT metrics for the remote SSRC.
NonSenderRttStats GetNonSenderRTT() const;
void SetNonSenderRttMeasurement(bool enabled);
- absl::optional<TimeDelta> GetAndResetXrRrRtt();
+ std::optional<TimeDelta> GetAndResetXrRrRtt();
// Called once per second on the worker thread to do rtt calculations.
// Returns an optional rtt value if one is available.
- absl::optional<TimeDelta> OnPeriodicRttUpdate(Timestamp newer_than,
- bool sending);
+ std::optional<TimeDelta> OnPeriodicRttUpdate(Timestamp newer_than,
+ bool sending);
// A snapshot of Report Blocks with additional data of interest to statistics.
// Within this list, the source SSRC is unique and ReportBlockData represents
@@ -382,7 +382,7 @@
// Estimated rtt, nullopt when there is no valid estimate.
bool xr_rrtr_status_ RTC_GUARDED_BY(rtcp_receiver_lock_);
- absl::optional<TimeDelta> xr_rr_rtt_;
+ std::optional<TimeDelta> xr_rr_rtt_;
Timestamp oldest_tmmbr_info_ RTC_GUARDED_BY(rtcp_receiver_lock_);
// Mapped by remote ssrc.
diff --git a/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc b/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc
index 2194ec2..7b8a15f 100644
--- a/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc
+++ b/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc
@@ -237,7 +237,7 @@
const uint32_t kDelayNtp = 0x4321;
const TimeDelta kDelay = CompactNtpRttToTimeDelta(kDelayNtp);
- EXPECT_EQ(receiver.LastRtt(), absl::nullopt);
+ EXPECT_EQ(receiver.LastRtt(), std::nullopt);
uint32_t sent_ntp = CompactNtp(mocks.clock.CurrentNtpTime());
mocks.clock.AdvanceTime(kRtt + kDelay);
@@ -266,7 +266,7 @@
const uint32_t kDelayNtp = 0x4321;
const TimeDelta kDelay = CompactNtpRttToTimeDelta(kDelayNtp);
- EXPECT_EQ(receiver.LastRtt(), absl::nullopt);
+ EXPECT_EQ(receiver.LastRtt(), std::nullopt);
uint32_t sent_ntp = CompactNtp(mocks.clock.CurrentNtpTime());
mocks.clock.AdvanceTime(kRtt + kDelay);
@@ -565,8 +565,8 @@
receiver.SetRemoteSSRC(kSenderSsrc);
// No report block received.
- EXPECT_EQ(receiver.LastRtt(), absl::nullopt);
- EXPECT_EQ(receiver.AverageRtt(), absl::nullopt);
+ EXPECT_EQ(receiver.LastRtt(), std::nullopt);
+ EXPECT_EQ(receiver.AverageRtt(), std::nullopt);
rtcp::ReportBlock rb;
rb.SetMediaSsrc(kReceiverMainSsrc);
@@ -584,8 +584,8 @@
receiver.IncomingPacket(rr.Build());
EXPECT_EQ(receiver.LastReceivedReportBlockMs(), now.ms());
- EXPECT_NE(receiver.LastRtt(), absl::nullopt);
- EXPECT_NE(receiver.AverageRtt(), absl::nullopt);
+ EXPECT_NE(receiver.LastRtt(), std::nullopt);
+ EXPECT_NE(receiver.AverageRtt(), std::nullopt);
}
// App packets are ignored.
diff --git a/modules/rtp_rtcp/source/rtcp_sender.cc b/modules/rtp_rtcp/source/rtcp_sender.cc
index 7ec3b17..cea49e4 100644
--- a/modules/rtp_rtcp/source/rtcp_sender.cc
+++ b/modules/rtp_rtcp/source/rtcp_sender.cc
@@ -14,10 +14,10 @@
#include <algorithm> // std::min
#include <memory>
+#include <optional>
#include <utility>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/rtp_headers.h"
#include "api/units/data_rate.h"
@@ -197,7 +197,7 @@
MutexLock lock(&mutex_rtcp_sender_);
if (new_method == RtcpMode::kOff) {
- next_time_to_send_rtcp_ = absl::nullopt;
+ next_time_to_send_rtcp_ = std::nullopt;
} else if (method_ == RtcpMode::kOff) {
// When switching on, reschedule the next packet
SetNextRtcpSendEvaluationDuration(report_interval_ / 2);
@@ -234,7 +234,7 @@
event_log_->Log(std::make_unique<RtcEventRtcpPacketOutgoing>(packet));
}
};
- absl::optional<PacketSender> sender;
+ std::optional<PacketSender> sender;
{
MutexLock lock(&mutex_rtcp_sender_);
@@ -302,8 +302,8 @@
}
void RTCPSender::SetLastRtpTime(uint32_t rtp_timestamp,
- absl::optional<Timestamp> capture_time,
- absl::optional<int8_t> payload_type) {
+ std::optional<Timestamp> capture_time,
+ std::optional<int8_t> payload_type) {
MutexLock lock(&mutex_rtcp_sender_);
// For compatibility with clients who don't set payload type correctly on all
// calls.
@@ -594,7 +594,7 @@
}
}
};
- absl::optional<PacketSender> sender;
+ std::optional<PacketSender> sender;
{
MutexLock lock(&mutex_rtcp_sender_);
sender.emplace(callback, max_packet_size_);
@@ -609,7 +609,7 @@
return error_code;
}
-absl::optional<int32_t> RTCPSender::ComputeCompoundRTCPPacket(
+std::optional<int32_t> RTCPSender::ComputeCompoundRTCPPacket(
const FeedbackState& feedback_state,
RTCPPacketType packet_type,
int32_t nack_size,
@@ -684,7 +684,7 @@
}
RTC_DCHECK(AllVolatileFlagsConsumed());
- return absl::nullopt;
+ return std::nullopt;
}
TimeDelta RTCPSender::ComputeTimeUntilNextReport(DataRate send_bitrate) {
@@ -831,7 +831,7 @@
// Check if this allocation is first ever, or has a different set of
// spatial/temporal layers signaled and enabled, if so trigger an rtcp report
// as soon as possible.
- absl::optional<VideoBitrateAllocation> new_bitrate =
+ std::optional<VideoBitrateAllocation> new_bitrate =
CheckAndUpdateLayerStructure(bitrate);
if (new_bitrate) {
video_bitrate_allocation_ = *new_bitrate;
@@ -847,9 +847,9 @@
SetFlag(kRtcpAnyExtendedReports, true);
}
-absl::optional<VideoBitrateAllocation> RTCPSender::CheckAndUpdateLayerStructure(
+std::optional<VideoBitrateAllocation> RTCPSender::CheckAndUpdateLayerStructure(
const VideoBitrateAllocation& bitrate) const {
- absl::optional<VideoBitrateAllocation> updated_bitrate;
+ std::optional<VideoBitrateAllocation> updated_bitrate;
for (size_t si = 0; si < kMaxSpatialLayers; ++si) {
for (size_t ti = 0; ti < kMaxTemporalStreams; ++ti) {
if (!updated_bitrate &&
diff --git a/modules/rtp_rtcp/source/rtcp_sender.h b/modules/rtp_rtcp/source/rtcp_sender.h
index fe2ec9b..1274aea 100644
--- a/modules/rtp_rtcp/source/rtcp_sender.h
+++ b/modules/rtp_rtcp/source/rtcp_sender.h
@@ -13,12 +13,12 @@
#include <map>
#include <memory>
+#include <optional>
#include <set>
#include <string>
#include <vector>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/call/transport.h"
#include "api/units/data_rate.h"
#include "api/units/time_delta.h"
@@ -79,7 +79,7 @@
std::function<void(TimeDelta)> schedule_next_rtcp_send_evaluation_function;
RtcEventLog* event_log = nullptr;
- absl::optional<TimeDelta> rtcp_report_interval;
+ std::optional<TimeDelta> rtcp_report_interval;
ReceiveStatisticsProvider* receive_statistics = nullptr;
RtcpPacketTypeCounterObserver* rtcp_packet_type_counter_observer = nullptr;
};
@@ -126,8 +126,8 @@
RTC_LOCKS_EXCLUDED(mutex_rtcp_sender_);
void SetLastRtpTime(uint32_t rtp_timestamp,
- absl::optional<Timestamp> capture_time,
- absl::optional<int8_t> payload_type)
+ std::optional<Timestamp> capture_time,
+ std::optional<int8_t> payload_type)
RTC_LOCKS_EXCLUDED(mutex_rtcp_sender_);
void SetRtpClockRate(int8_t payload_type, int rtp_clock_rate_hz)
@@ -185,7 +185,7 @@
class RtcpContext;
class PacketSender;
- absl::optional<int32_t> ComputeCompoundRTCPPacket(
+ std::optional<int32_t> ComputeCompoundRTCPPacket(
const FeedbackState& feedback_state,
RTCPPacketType packet_type,
int32_t nack_size,
@@ -256,12 +256,12 @@
mutable Mutex mutex_rtcp_sender_;
bool sending_ RTC_GUARDED_BY(mutex_rtcp_sender_);
- absl::optional<Timestamp> next_time_to_send_rtcp_
+ std::optional<Timestamp> next_time_to_send_rtcp_
RTC_GUARDED_BY(mutex_rtcp_sender_);
uint32_t timestamp_offset_ RTC_GUARDED_BY(mutex_rtcp_sender_);
uint32_t last_rtp_timestamp_ RTC_GUARDED_BY(mutex_rtcp_sender_);
- absl::optional<Timestamp> last_frame_capture_time_
+ std::optional<Timestamp> last_frame_capture_time_
RTC_GUARDED_BY(mutex_rtcp_sender_);
// SSRC that we receive on our RTP channel
uint32_t remote_ssrc_ RTC_GUARDED_BY(mutex_rtcp_sender_);
@@ -303,7 +303,7 @@
std::map<int8_t, int> rtp_clock_rates_khz_ RTC_GUARDED_BY(mutex_rtcp_sender_);
int8_t last_payload_type_ RTC_GUARDED_BY(mutex_rtcp_sender_);
- absl::optional<VideoBitrateAllocation> CheckAndUpdateLayerStructure(
+ std::optional<VideoBitrateAllocation> CheckAndUpdateLayerStructure(
const VideoBitrateAllocation& bitrate) const
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_rtcp_sender_);
diff --git a/modules/rtp_rtcp/source/rtcp_sender_unittest.cc b/modules/rtp_rtcp/source/rtcp_sender_unittest.cc
index f58098b..023f923 100644
--- a/modules/rtp_rtcp/source/rtcp_sender_unittest.cc
+++ b/modules/rtp_rtcp/source/rtcp_sender_unittest.cc
@@ -720,7 +720,7 @@
EXPECT_EQ(0, rtcp_sender->SendRTCP(feedback_state(), kRtcpReport));
EXPECT_EQ(1, parser()->xr()->num_packets());
EXPECT_EQ(kSenderSsrc, parser()->xr()->sender_ssrc());
- const absl::optional<rtcp::TargetBitrate>& target_bitrate =
+ const std::optional<rtcp::TargetBitrate>& target_bitrate =
parser()->xr()->target_bitrate();
ASSERT_TRUE(target_bitrate);
const std::vector<rtcp::TargetBitrate::BitrateItem>& bitrates =
@@ -786,7 +786,7 @@
allocation.SetBitrate(1, 0, 200000);
rtcp_sender->SetVideoBitrateAllocation(allocation);
EXPECT_EQ(0, rtcp_sender->SendRTCP(feedback_state(), kRtcpReport));
- absl::optional<rtcp::TargetBitrate> target_bitrate =
+ std::optional<rtcp::TargetBitrate> target_bitrate =
parser()->xr()->target_bitrate();
ASSERT_TRUE(target_bitrate);
std::vector<rtcp::TargetBitrate::BitrateItem> bitrates =
diff --git a/modules/rtp_rtcp/source/rtcp_transceiver_impl.cc b/modules/rtp_rtcp/source/rtcp_transceiver_impl.cc
index 9ada3cdc..f07d765 100644
--- a/modules/rtp_rtcp/source/rtcp_transceiver_impl.cc
+++ b/modules/rtp_rtcp/source/rtcp_transceiver_impl.cc
@@ -11,11 +11,11 @@
#include "modules/rtp_rtcp/source/rtcp_transceiver_impl.h"
#include <algorithm>
+#include <optional>
#include <utility>
#include "absl/algorithm/container.h"
#include "absl/memory/memory.h"
-#include "absl/types/optional.h"
#include "api/video/video_bitrate_allocation.h"
#include "modules/rtp_rtcp/include/receive_statistics.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
@@ -68,7 +68,7 @@
struct RtcpTransceiverImpl::RemoteSenderState {
uint8_t fir_sequence_number = 0;
- absl::optional<SenderReportTimes> last_received_sender_report;
+ std::optional<SenderReportTimes> last_received_sender_report;
std::vector<MediaReceiverRtcpObserver*> observers;
};
@@ -234,7 +234,7 @@
// immideately on large bitrate change when there is one RtcpTransceiver per
// rtp transport.
if (send_now) {
- absl::optional<rtcp::Remb> remb;
+ std::optional<rtcp::Remb> remb;
remb.swap(remb_);
SendImmediateFeedback(*remb);
remb.swap(remb_);
@@ -366,7 +366,7 @@
Timestamp::Millis(now_ntp.ToMs() - rtc::kNtpJan1970Millisecs);
for (const rtcp::ReportBlock& block : rtcp_report_blocks) {
- absl::optional<TimeDelta> rtt;
+ std::optional<TimeDelta> rtt;
if (block.last_sr() != 0) {
rtt = CompactNtpRttToTimeDelta(
receive_time_ntp - block.delay_since_last_sr() - block.last_sr());
@@ -736,7 +736,7 @@
PacketSender& sender) {
RTC_DCHECK(sender.IsEmpty());
ReservedBytes reserved = {.per_packet = reserved_bytes};
- absl::optional<rtcp::Sdes> sdes;
+ std::optional<rtcp::Sdes> sdes;
if (!config_.cname.empty()) {
sdes.emplace();
bool added = sdes->AddCName(config_.feedback_ssrc, config_.cname);
@@ -747,7 +747,7 @@
if (remb_.has_value()) {
reserved.per_packet += remb_->BlockLength();
}
- absl::optional<rtcp::ExtendedReports> xr_with_dlrr;
+ std::optional<rtcp::ExtendedReports> xr_with_dlrr;
if (!received_rrtrs_.empty()) {
RTC_DCHECK(config_.reply_to_non_sender_rtt_measurement);
xr_with_dlrr.emplace();
diff --git a/modules/rtp_rtcp/source/rtcp_transceiver_impl.h b/modules/rtp_rtcp/source/rtcp_transceiver_impl.h
index c73b292..d4f12dc 100644
--- a/modules/rtp_rtcp/source/rtcp_transceiver_impl.h
+++ b/modules/rtp_rtcp/source/rtcp_transceiver_impl.h
@@ -13,10 +13,10 @@
#include <list>
#include <memory>
+#include <optional>
#include <string>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/units/timestamp.h"
#include "modules/rtp_rtcp/source/rtcp_packet/common_header.h"
@@ -154,7 +154,7 @@
std::function<void(rtc::ArrayView<const uint8_t>)> rtcp_transport_;
bool ready_to_send_;
- absl::optional<rtcp::Remb> remb_;
+ std::optional<rtcp::Remb> remb_;
// TODO(bugs.webrtc.org/8239): Remove entries from remote_senders_ that are no
// longer needed.
flat_map<uint32_t, RemoteSenderState> remote_senders_;
diff --git a/modules/rtp_rtcp/source/rtcp_transceiver_impl_unittest.cc b/modules/rtp_rtcp/source/rtcp_transceiver_impl_unittest.cc
index 8bd3652..988bccc 100644
--- a/modules/rtp_rtcp/source/rtcp_transceiver_impl_unittest.cc
+++ b/modules/rtp_rtcp/source/rtcp_transceiver_impl_unittest.cc
@@ -221,7 +221,7 @@
config.rtcp_transport = transport.AsStdFunction();
config.initial_report_delay = TimeDelta::Millis(10);
config.task_queue = queue.get();
- absl::optional<RtcpTransceiverImpl> rtcp_transceiver;
+ std::optional<RtcpTransceiverImpl> rtcp_transceiver;
Timestamp started = CurrentTime();
queue->PostTask([&] { rtcp_transceiver.emplace(config); });
@@ -248,7 +248,7 @@
config.initial_report_delay = TimeDelta::Zero();
config.report_period = kReportPeriod;
config.task_queue = queue.get();
- absl::optional<RtcpTransceiverImpl> rtcp_transceiver;
+ std::optional<RtcpTransceiverImpl> rtcp_transceiver;
Timestamp time_just_before_1st_packet = Timestamp::MinusInfinity();
queue->PostTask([&] {
// Because initial_report_delay_ms is set to 0, time_just_before_the_packet
@@ -283,7 +283,7 @@
config.initial_report_delay = TimeDelta::Zero();
config.report_period = kReportPeriod;
config.task_queue = queue.get();
- absl::optional<RtcpTransceiverImpl> rtcp_transceiver;
+ std::optional<RtcpTransceiverImpl> rtcp_transceiver;
queue->PostTask([&] { rtcp_transceiver.emplace(config); });
// Wait for the first packet.
@@ -365,7 +365,7 @@
config.initial_ready_to_send = false;
config.rtcp_transport = transport.AsStdFunction();
config.task_queue = queue.get();
- absl::optional<RtcpTransceiverImpl> rtcp_transceiver;
+ std::optional<RtcpTransceiverImpl> rtcp_transceiver;
rtcp_transceiver.emplace(config);
queue->PostTask([&] { rtcp_transceiver->SetReadyToSend(true); });
diff --git a/modules/rtp_rtcp/source/rtp_dependency_descriptor_reader.cc b/modules/rtp_rtcp/source/rtp_dependency_descriptor_reader.cc
index 1a56efd..e1b05b0 100644
--- a/modules/rtp_rtcp/source/rtp_dependency_descriptor_reader.cc
+++ b/modules/rtp_rtcp/source/rtp_dependency_descriptor_reader.cc
@@ -199,7 +199,7 @@
ReadFrameChains();
if (structure_->resolutions.empty()) {
- descriptor_->resolution = absl::nullopt;
+ descriptor_->resolution = std::nullopt;
} else {
// Format guarantees that if there were resolutions in the last structure,
// then each spatial layer got one.
diff --git a/modules/rtp_rtcp/source/rtp_format.cc b/modules/rtp_rtcp/source/rtp_format.cc
index f4e0514..dffa50e 100644
--- a/modules/rtp_rtcp/source/rtp_format.cc
+++ b/modules/rtp_rtcp/source/rtp_format.cc
@@ -29,7 +29,7 @@
namespace webrtc {
std::unique_ptr<RtpPacketizer> RtpPacketizer::Create(
- absl::optional<VideoCodecType> type,
+ std::optional<VideoCodecType> type,
rtc::ArrayView<const uint8_t> payload,
PayloadSizeLimits limits,
// Codec-specific details.
diff --git a/modules/rtp_rtcp/source/rtp_format.h b/modules/rtp_rtcp/source/rtp_format.h
index 91b0367..097481d 100644
--- a/modules/rtp_rtcp/source/rtp_format.h
+++ b/modules/rtp_rtcp/source/rtp_format.h
@@ -14,9 +14,9 @@
#include <stdint.h>
#include <memory>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "modules/rtp_rtcp/source/rtp_video_header.h"
@@ -36,7 +36,7 @@
// If type is not set, returns a raw packetizer.
static std::unique_ptr<RtpPacketizer> Create(
- absl::optional<VideoCodecType> type,
+ std::optional<VideoCodecType> type,
rtc::ArrayView<const uint8_t> payload,
PayloadSizeLimits limits,
// Codec-specific details.
diff --git a/modules/rtp_rtcp/source/rtp_format_h264.cc b/modules/rtp_rtcp/source/rtp_format_h264.cc
index 7e11add..917cb1b 100644
--- a/modules/rtp_rtcp/source/rtp_format_h264.cc
+++ b/modules/rtp_rtcp/source/rtp_format_h264.cc
@@ -16,11 +16,11 @@
#include <cstdint>
#include <iterator>
#include <memory>
+#include <optional>
#include <utility>
#include <vector>
#include "absl/algorithm/container.h"
-#include "absl/types/optional.h"
#include "absl/types/variant.h"
#include "common_video/h264/h264_common.h"
#include "common_video/h264/pps_parser.h"
diff --git a/modules/rtp_rtcp/source/rtp_format_video_generic.cc b/modules/rtp_rtcp/source/rtp_format_video_generic.cc
index f5c7f2e..d91e6b4 100644
--- a/modules/rtp_rtcp/source/rtp_format_video_generic.cc
+++ b/modules/rtp_rtcp/source/rtp_format_video_generic.cc
@@ -12,7 +12,8 @@
#include <string.h>
-#include "absl/types/optional.h"
+#include <optional>
+
#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
diff --git a/modules/rtp_rtcp/source/rtp_generic_frame_descriptor.h b/modules/rtp_rtcp/source/rtp_generic_frame_descriptor.h
index 8760acc..b48f1302 100644
--- a/modules/rtp_rtcp/source/rtp_generic_frame_descriptor.h
+++ b/modules/rtp_rtcp/source/rtp_generic_frame_descriptor.h
@@ -13,9 +13,9 @@
#include <stddef.h>
#include <stdint.h>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
namespace webrtc {
diff --git a/modules/rtp_rtcp/source/rtp_header_extensions.cc b/modules/rtp_rtcp/source/rtp_header_extensions.cc
index 43770a7..46c8218 100644
--- a/modules/rtp_rtcp/source/rtp_header_extensions.cc
+++ b/modules/rtp_rtcp/source/rtp_header_extensions.cc
@@ -16,11 +16,11 @@
#include <cstddef>
#include <cstdint>
#include <limits>
+#include <optional>
#include <string>
#include <vector>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/rtp_headers.h"
#include "api/video/color_space.h"
@@ -119,7 +119,7 @@
size_t AbsoluteCaptureTimeExtension::ValueSize(
const AbsoluteCaptureTime& extension) {
- if (extension.estimated_capture_clock_offset != absl::nullopt) {
+ if (extension.estimated_capture_clock_offset != std::nullopt) {
return kValueSizeBytes;
} else {
return kValueSizeBytesWithoutEstimatedCaptureClockOffset;
@@ -304,14 +304,14 @@
bool TransportSequenceNumberV2::Parse(
rtc::ArrayView<const uint8_t> data,
uint16_t* transport_sequence_number,
- absl::optional<FeedbackRequest>* feedback_request) {
+ std::optional<FeedbackRequest>* feedback_request) {
if (data.size() != kValueSizeBytes &&
data.size() != kValueSizeBytesWithoutFeedbackRequest)
return false;
*transport_sequence_number = ByteReader<uint16_t>::ReadBigEndian(data.data());
- *feedback_request = absl::nullopt;
+ *feedback_request = std::nullopt;
if (data.size() == kValueSizeBytes) {
uint16_t feedback_request_raw =
ByteReader<uint16_t>::ReadBigEndian(data.data() + 2);
@@ -330,7 +330,7 @@
bool TransportSequenceNumberV2::Write(
rtc::ArrayView<uint8_t> data,
uint16_t transport_sequence_number,
- const absl::optional<FeedbackRequest>& feedback_request) {
+ const std::optional<FeedbackRequest>& feedback_request) {
RTC_DCHECK_EQ(data.size(),
ValueSize(transport_sequence_number, feedback_request));
@@ -792,17 +792,17 @@
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
// Sample Audio Level Encoding Using the Two-Byte Header Format
bool InbandComfortNoiseExtension::Parse(rtc::ArrayView<const uint8_t> data,
- absl::optional<uint8_t>* level) {
+ std::optional<uint8_t>* level) {
if (data.size() != kValueSizeBytes)
return false;
*level = (data[0] & 0b1000'0000) != 0
- ? absl::nullopt
- : absl::make_optional(data[0] & 0b0111'1111);
+ ? std::nullopt
+ : std::make_optional(data[0] & 0b0111'1111);
return true;
}
bool InbandComfortNoiseExtension::Write(rtc::ArrayView<uint8_t> data,
- absl::optional<uint8_t> level) {
+ std::optional<uint8_t> level) {
RTC_DCHECK_EQ(data.size(), kValueSizeBytes);
data[0] = 0b0000'0000;
if (level) {
diff --git a/modules/rtp_rtcp/source/rtp_header_extensions.h b/modules/rtp_rtcp/source/rtp_header_extensions.h
index 02a1543..b8495f4 100644
--- a/modules/rtp_rtcp/source/rtp_header_extensions.h
+++ b/modules/rtp_rtcp/source/rtp_header_extensions.h
@@ -163,16 +163,16 @@
static bool Parse(rtc::ArrayView<const uint8_t> data,
uint16_t* transport_sequence_number,
- absl::optional<FeedbackRequest>* feedback_request);
+ std::optional<FeedbackRequest>* feedback_request);
static size_t ValueSize(
uint16_t /*transport_sequence_number*/,
- const absl::optional<FeedbackRequest>& feedback_request) {
+ const std::optional<FeedbackRequest>& feedback_request) {
return feedback_request ? kValueSizeBytes
: kValueSizeBytesWithoutFeedbackRequest;
}
static bool Write(rtc::ArrayView<uint8_t> data,
uint16_t transport_sequence_number,
- const absl::optional<FeedbackRequest>& feedback_request);
+ const std::optional<FeedbackRequest>& feedback_request);
private:
static constexpr uint16_t kIncludeTimestampsBit = 1 << 15;
@@ -344,7 +344,7 @@
class InbandComfortNoiseExtension {
public:
- using value_type = absl::optional<uint8_t>;
+ using value_type = std::optional<uint8_t>;
static constexpr RTPExtensionType kId = kRtpExtensionInbandComfortNoise;
static constexpr uint8_t kValueSizeBytes = 1;
@@ -353,12 +353,11 @@
static constexpr absl::string_view Uri() { return kUri; }
static bool Parse(rtc::ArrayView<const uint8_t> data,
- absl::optional<uint8_t>* level);
- static size_t ValueSize(absl::optional<uint8_t> level) {
+ std::optional<uint8_t>* level);
+ static size_t ValueSize(std::optional<uint8_t> level) {
return kValueSizeBytes;
}
- static bool Write(rtc::ArrayView<uint8_t> data,
- absl::optional<uint8_t> level);
+ static bool Write(rtc::ArrayView<uint8_t> data, std::optional<uint8_t> level);
};
class VideoFrameTrackingIdExtension {
diff --git a/modules/rtp_rtcp/source/rtp_packet.h b/modules/rtp_rtcp/source/rtp_packet.h
index c002e51..872d2da 100644
--- a/modules/rtp_rtcp/source/rtp_packet.h
+++ b/modules/rtp_rtcp/source/rtp_packet.h
@@ -10,11 +10,11 @@
#ifndef MODULES_RTP_RTCP_SOURCE_RTP_PACKET_H_
#define MODULES_RTP_RTCP_SOURCE_RTP_PACKET_H_
+#include <optional>
#include <string>
#include <utility>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
@@ -131,7 +131,7 @@
bool GetExtension(FirstValue&&, Values&&...) const;
template <typename Extension>
- absl::optional<typename Extension::value_type> GetExtension() const;
+ std::optional<typename Extension::value_type> GetExtension() const;
// Returns view of the raw extension or empty view on failure.
template <typename Extension>
@@ -241,11 +241,11 @@
}
template <typename Extension>
-absl::optional<typename Extension::value_type> RtpPacket::GetExtension() const {
- absl::optional<typename Extension::value_type> result;
+std::optional<typename Extension::value_type> RtpPacket::GetExtension() const {
+ std::optional<typename Extension::value_type> result;
auto raw = FindExtension(Extension::kId);
if (raw.empty() || !Extension::Parse(raw, &result.emplace()))
- result = absl::nullopt;
+ result = std::nullopt;
return result;
}
diff --git a/modules/rtp_rtcp/source/rtp_packet_history.cc b/modules/rtp_rtcp/source/rtp_packet_history.cc
index 93bd4ab..936f6f2 100644
--- a/modules/rtp_rtcp/source/rtp_packet_history.cc
+++ b/modules/rtp_rtcp/source/rtp_packet_history.cc
@@ -309,7 +309,7 @@
void RtpPacketHistory::Reset() {
packet_history_.clear();
- large_payload_packet_ = absl::nullopt;
+ large_payload_packet_ = std::nullopt;
}
void RtpPacketHistory::CullOldPackets() {
diff --git a/modules/rtp_rtcp/source/rtp_packet_history.h b/modules/rtp_rtcp/source/rtp_packet_history.h
index 46db9d7..512ffda 100644
--- a/modules/rtp_rtcp/source/rtp_packet_history.h
+++ b/modules/rtp_rtcp/source/rtp_packet_history.h
@@ -14,11 +14,11 @@
#include <deque>
#include <map>
#include <memory>
+#include <optional>
#include <set>
#include <utility>
#include <vector>
-#include "absl/types/optional.h"
#include "api/function_view.h"
#include "api/units/time_delta.h"
#include "api/units/timestamp.h"
@@ -190,7 +190,7 @@
// Total number of packets with inserted.
uint64_t packets_inserted_ RTC_GUARDED_BY(lock_);
- absl::optional<RtpPacketToSend> large_payload_packet_ RTC_GUARDED_BY(lock_);
+ std::optional<RtpPacketToSend> large_payload_packet_ RTC_GUARDED_BY(lock_);
};
} // namespace webrtc
#endif // MODULES_RTP_RTCP_SOURCE_RTP_PACKET_HISTORY_H_
diff --git a/modules/rtp_rtcp/source/rtp_packet_send_info.cc b/modules/rtp_rtcp/source/rtp_packet_send_info.cc
index 6ec7671..2fc5ce5 100644
--- a/modules/rtp_rtcp/source/rtp_packet_send_info.cc
+++ b/modules/rtp_rtcp/source/rtp_packet_send_info.cc
@@ -9,8 +9,8 @@
*/
#include <cstdint>
+#include <optional>
-#include "absl/types/optional.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
@@ -24,7 +24,7 @@
packet_info.transport_sequence_number =
*packet.transport_sequence_number() & 0xFFFF;
} else {
- absl::optional<uint16_t> packet_id =
+ std::optional<uint16_t> packet_id =
packet.GetExtension<TransportSequenceNumber>();
if (packet_id) {
packet_info.transport_sequence_number = *packet_id;
diff --git a/modules/rtp_rtcp/source/rtp_packet_send_info_unittest.cc b/modules/rtp_rtcp/source/rtp_packet_send_info_unittest.cc
index 223ce26..4dcbe17 100644
--- a/modules/rtp_rtcp/source/rtp_packet_send_info_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_packet_send_info_unittest.cc
@@ -9,8 +9,8 @@
*/
#include <cstdint>
+#include <optional>
-#include "absl/types/optional.h"
#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
@@ -70,7 +70,7 @@
paced_info.probe_cluster_id = 8;
RtpPacketSendInfo send_info = RtpPacketSendInfo::From(packet, paced_info);
- EXPECT_EQ(send_info.media_ssrc, absl::nullopt);
+ EXPECT_EQ(send_info.media_ssrc, std::nullopt);
VerifyDefaultProperties(send_info, packet, paced_info);
}
@@ -81,7 +81,7 @@
paced_info.probe_cluster_id = 8;
RtpPacketSendInfo send_info = RtpPacketSendInfo::From(packet, paced_info);
- EXPECT_EQ(send_info.media_ssrc, absl::nullopt);
+ EXPECT_EQ(send_info.media_ssrc, std::nullopt);
VerifyDefaultProperties(send_info, packet, paced_info);
}
diff --git a/modules/rtp_rtcp/source/rtp_packet_to_send.h b/modules/rtp_rtcp/source/rtp_packet_to_send.h
index 47747a0..7eaf1cd 100644
--- a/modules/rtp_rtcp/source/rtp_packet_to_send.h
+++ b/modules/rtp_rtcp/source/rtp_packet_to_send.h
@@ -13,9 +13,9 @@
#include <stddef.h>
#include <stdint.h>
+#include <optional>
#include <utility>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/ref_counted_base.h"
#include "api/scoped_refptr.h"
@@ -51,13 +51,11 @@
void set_packet_type(RtpPacketMediaType type);
- absl::optional<RtpPacketMediaType> packet_type() const {
- return packet_type_;
- }
+ std::optional<RtpPacketMediaType> packet_type() const { return packet_type_; }
enum class OriginalType { kAudio, kVideo };
// Original type does not change if packet type is changed to kRetransmission.
- absl::optional<OriginalType> original_packet_type() const {
+ std::optional<OriginalType> original_packet_type() const {
return original_packet_type_;
}
@@ -67,14 +65,14 @@
void set_retransmitted_sequence_number(uint16_t sequence_number) {
retransmitted_sequence_number_ = sequence_number;
}
- absl::optional<uint16_t> retransmitted_sequence_number() const {
+ std::optional<uint16_t> retransmitted_sequence_number() const {
return retransmitted_sequence_number_;
}
// If this is a retransmission, indicates the SSRC of the original
// media packet that this packet represents.
void set_original_ssrc(uint32_t ssrc) { original_ssrc_ = ssrc; }
- absl::optional<uint32_t> original_ssrc() const { return original_ssrc_; }
+ std::optional<uint32_t> original_ssrc() const { return original_ssrc_; }
void set_allow_retransmission(bool allow_retransmission) {
allow_retransmission_ = allow_retransmission;
@@ -138,12 +136,12 @@
void set_time_in_send_queue(TimeDelta time_in_send_queue) {
time_in_send_queue_ = time_in_send_queue;
}
- absl::optional<TimeDelta> time_in_send_queue() const {
+ std::optional<TimeDelta> time_in_send_queue() const {
return time_in_send_queue_;
}
// A sequence number guaranteed to be monotically increasing by one for all
// packets where transport feedback is expected.
- absl::optional<int64_t> transport_sequence_number() const {
+ std::optional<int64_t> transport_sequence_number() const {
return transport_sequence_number_;
}
void set_transport_sequence_number(int64_t transport_sequence_number) {
@@ -152,18 +150,18 @@
private:
webrtc::Timestamp capture_time_ = webrtc::Timestamp::Zero();
- absl::optional<RtpPacketMediaType> packet_type_;
- absl::optional<OriginalType> original_packet_type_;
- absl::optional<uint32_t> original_ssrc_;
- absl::optional<int64_t> transport_sequence_number_;
+ std::optional<RtpPacketMediaType> packet_type_;
+ std::optional<OriginalType> original_packet_type_;
+ std::optional<uint32_t> original_ssrc_;
+ std::optional<int64_t> transport_sequence_number_;
bool allow_retransmission_ = false;
- absl::optional<uint16_t> retransmitted_sequence_number_;
+ std::optional<uint16_t> retransmitted_sequence_number_;
rtc::scoped_refptr<rtc::RefCountedBase> additional_data_;
bool is_first_packet_of_frame_ = false;
bool is_key_frame_ = false;
bool fec_protect_packet_ = false;
bool is_red_ = false;
- absl::optional<TimeDelta> time_in_send_queue_;
+ std::optional<TimeDelta> time_in_send_queue_;
};
} // namespace webrtc
diff --git a/modules/rtp_rtcp/source/rtp_packet_unittest.cc b/modules/rtp_rtcp/source/rtp_packet_unittest.cc
index ccdb5f8..5f9544d 100644
--- a/modules/rtp_rtcp/source/rtp_packet_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_packet_unittest.cc
@@ -399,7 +399,7 @@
EXPECT_FALSE(packet.SetExtension<TransportSequenceNumber>(42));
EXPECT_FALSE(packet.HasExtension<TransportSequenceNumber>());
- EXPECT_EQ(packet.GetExtension<TransportSequenceNumber>(), absl::nullopt);
+ EXPECT_EQ(packet.GetExtension<TransportSequenceNumber>(), std::nullopt);
}
TEST(RtpPacketTest, SetReservedExtensionsAfterPayload) {
@@ -605,7 +605,7 @@
auto time_offset = packet.GetExtension<TransmissionOffset>();
static_assert(
std::is_same<decltype(time_offset),
- absl::optional<TransmissionOffset::value_type>>::value,
+ std::optional<TransmissionOffset::value_type>>::value,
"");
EXPECT_EQ(time_offset, kTimeOffset);
EXPECT_FALSE(packet.GetExtension<RtpStreamId>().has_value());
@@ -1043,7 +1043,7 @@
constexpr AbsoluteCaptureTime kAbsoluteCaptureTime{
/*absolute_capture_timestamp=*/9876543210123456789ULL,
- /*estimated_capture_clock_offset=*/absl::nullopt};
+ /*estimated_capture_clock_offset=*/std::nullopt};
ASSERT_TRUE(send_packet.SetExtension<AbsoluteCaptureTimeExtension>(
kAbsoluteCaptureTime));
@@ -1099,7 +1099,7 @@
constexpr int kTransportSequenceNumber = 12345;
send_packet.SetExtension<TransportSequenceNumberV2>(kTransportSequenceNumber,
- absl::nullopt);
+ std::nullopt);
EXPECT_EQ(send_packet.GetRawExtension<TransportSequenceNumberV2>().size(),
2u);
@@ -1108,7 +1108,7 @@
EXPECT_TRUE(receive_packet.Parse(send_packet.Buffer()));
uint16_t received_transport_sequeunce_number;
- absl::optional<FeedbackRequest> received_feedback_request;
+ std::optional<FeedbackRequest> received_feedback_request;
EXPECT_TRUE(receive_packet.GetExtension<TransportSequenceNumberV2>(
&received_transport_sequeunce_number, &received_feedback_request));
EXPECT_EQ(received_transport_sequeunce_number, kTransportSequenceNumber);
@@ -1130,7 +1130,7 @@
send_packet.SetSsrc(kSsrc);
constexpr int kTransportSequenceNumber = 12345;
- constexpr absl::optional<FeedbackRequest> kNoFeedbackRequest =
+ constexpr std::optional<FeedbackRequest> kNoFeedbackRequest =
FeedbackRequest{/*include_timestamps=*/false, /*sequence_count=*/0};
send_packet.ReserveExtension<TransportSequenceNumberV2>();
send_packet.SetExtension<TransportSequenceNumberV2>(kTransportSequenceNumber,
@@ -1143,7 +1143,7 @@
EXPECT_TRUE(receive_packet.Parse(send_packet.Buffer()));
uint16_t received_transport_sequeunce_number;
- absl::optional<FeedbackRequest> received_feedback_request;
+ std::optional<FeedbackRequest> received_feedback_request;
EXPECT_TRUE(receive_packet.GetExtension<TransportSequenceNumberV2>(
&received_transport_sequeunce_number, &received_feedback_request));
EXPECT_EQ(received_transport_sequeunce_number, kTransportSequenceNumber);
@@ -1163,7 +1163,7 @@
send_packet.SetSsrc(kSsrc);
constexpr int kTransportSequenceNumber = 12345;
- constexpr absl::optional<FeedbackRequest> kFeedbackRequest =
+ constexpr std::optional<FeedbackRequest> kFeedbackRequest =
FeedbackRequest{/*include_timestamps=*/true, /*sequence_count=*/3};
send_packet.SetExtension<TransportSequenceNumberV2>(kTransportSequenceNumber,
kFeedbackRequest);
@@ -1174,7 +1174,7 @@
// Parse transport sequence number and feedback request.
uint16_t received_transport_sequeunce_number;
- absl::optional<FeedbackRequest> received_feedback_request;
+ std::optional<FeedbackRequest> received_feedback_request;
EXPECT_TRUE(receive_packet.GetExtension<TransportSequenceNumberV2>(
&received_transport_sequeunce_number, &received_feedback_request));
EXPECT_EQ(received_transport_sequeunce_number, kTransportSequenceNumber);
diff --git a/modules/rtp_rtcp/source/rtp_packetizer_h265.cc b/modules/rtp_rtcp/source/rtp_packetizer_h265.cc
index 901b68b..efc0d8a 100644
--- a/modules/rtp_rtcp/source/rtp_packetizer_h265.cc
+++ b/modules/rtp_rtcp/source/rtp_packetizer_h265.cc
@@ -10,9 +10,9 @@
#include "modules/rtp_rtcp/source/rtp_packetizer_h265.h"
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "common_video/h264/h264_common.h"
#include "common_video/h265/h265_common.h"
#include "modules/rtp_rtcp/source/byte_io.h"
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
index e13b043..a2c309e 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
@@ -15,12 +15,12 @@
#include <algorithm>
#include <cstdint>
#include <memory>
+#include <optional>
#include <set>
#include <string>
#include <utility>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "modules/rtp_rtcp/source/ntp_time_util.h"
#include "modules/rtp_rtcp/source/rtcp_packet/dlrr.h"
#include "modules/rtp_rtcp/source/rtcp_sender.h"
@@ -155,7 +155,7 @@
} else {
// Report rtt from receiver.
if (process_rtt && rtt_stats_ != nullptr) {
- absl::optional<TimeDelta> rtt = rtcp_receiver_.GetAndResetXrRrRtt();
+ std::optional<TimeDelta> rtt = rtcp_receiver_.GetAndResetXrRrRtt();
if (rtt.has_value()) {
rtt_stats_->OnRttUpdate(rtt->ms());
}
@@ -195,15 +195,15 @@
associated_payload_type);
}
-absl::optional<uint32_t> ModuleRtpRtcpImpl::RtxSsrc() const {
- return rtp_sender_ ? rtp_sender_->packet_generator.RtxSsrc() : absl::nullopt;
+std::optional<uint32_t> ModuleRtpRtcpImpl::RtxSsrc() const {
+ return rtp_sender_ ? rtp_sender_->packet_generator.RtxSsrc() : std::nullopt;
}
-absl::optional<uint32_t> ModuleRtpRtcpImpl::FlexfecSsrc() const {
+std::optional<uint32_t> ModuleRtpRtcpImpl::FlexfecSsrc() const {
if (rtp_sender_) {
return rtp_sender_->packet_generator.FlexfecSsrc();
}
- return absl::nullopt;
+ return std::nullopt;
}
void ModuleRtpRtcpImpl::IncomingRtcpPacket(
@@ -295,7 +295,7 @@
}
state.receiver = &rtcp_receiver_;
- if (absl::optional<RtpRtcpInterface::SenderReportStats> last_sr =
+ if (std::optional<RtpRtcpInterface::SenderReportStats> last_sr =
rtcp_receiver_.GetSenderReportStats();
last_sr.has_value()) {
state.remote_sr = CompactNtp(last_sr->last_remote_timestamp);
@@ -346,11 +346,11 @@
// TODO(bugs.webrtc.org/12873): Migrate this method and it's users to use
// optional Timestamps.
- absl::optional<Timestamp> capture_time;
+ std::optional<Timestamp> capture_time;
if (capture_time_ms > 0) {
capture_time = Timestamp::Millis(capture_time_ms);
}
- absl::optional<int> payload_type_optional;
+ std::optional<int> payload_type_optional;
if (payload_type >= 0)
payload_type_optional = payload_type;
rtcp_sender_.SetLastRtpTime(timestamp, capture_time, payload_type_optional);
@@ -475,8 +475,8 @@
return rtcp_sender_.SetCNAME(c_name);
}
-absl::optional<TimeDelta> ModuleRtpRtcpImpl::LastRtt() const {
- absl::optional<TimeDelta> rtt = rtcp_receiver_.LastRtt();
+std::optional<TimeDelta> ModuleRtpRtcpImpl::LastRtt() const {
+ std::optional<TimeDelta> rtt = rtcp_receiver_.LastRtt();
if (!rtt.has_value()) {
MutexLock lock(&mutex_rtt_);
if (rtt_ms_ > 0) {
@@ -493,7 +493,7 @@
}
// No rtt available (`kRtpRtcpRttProcessTimeMs` not yet passed?), so try to
// poll avg_rtt_ms directly from rtcp receiver.
- if (absl::optional<TimeDelta> rtt = rtcp_receiver_.AverageRtt()) {
+ if (std::optional<TimeDelta> rtt = rtcp_receiver_.AverageRtt()) {
return *rtt;
}
return kDefaultExpectedRetransmissionTime;
@@ -517,15 +517,15 @@
return rtcp_receiver_.GetLatestReportBlockData();
}
-absl::optional<RtpRtcpInterface::SenderReportStats>
+std::optional<RtpRtcpInterface::SenderReportStats>
ModuleRtpRtcpImpl::GetSenderReportStats() const {
return rtcp_receiver_.GetSenderReportStats();
}
-absl::optional<RtpRtcpInterface::NonSenderRttStats>
+std::optional<RtpRtcpInterface::NonSenderRttStats>
ModuleRtpRtcpImpl::GetNonSenderRttStats() const {
// This is not implemented for this legacy class.
- return absl::nullopt;
+ return std::nullopt;
}
// (REMB) Receiver Estimated Max Bitrate.
@@ -603,7 +603,7 @@
// Use RTT from RtcpRttStats class if provided.
int64_t rtt = rtt_ms();
if (rtt == 0) {
- if (absl::optional<TimeDelta> average_rtt = rtcp_receiver_.AverageRtt()) {
+ if (std::optional<TimeDelta> average_rtt = rtcp_receiver_.AverageRtt()) {
rtt = average_rtt->ms();
}
}
@@ -676,7 +676,7 @@
// Use RTT from RtcpRttStats class if provided.
int64_t rtt = rtt_ms();
if (rtt == 0) {
- if (absl::optional<TimeDelta> average_rtt = rtcp_receiver_.AverageRtt()) {
+ if (std::optional<TimeDelta> average_rtt = rtcp_receiver_.AverageRtt()) {
rtt = average_rtt->ms();
}
}
@@ -687,7 +687,7 @@
rtc::ArrayView<const ReportBlockData> report_blocks) {
if (rtp_sender_) {
uint32_t ssrc = SSRC();
- absl::optional<uint32_t> rtx_ssrc;
+ std::optional<uint32_t> rtx_ssrc;
if (rtp_sender_->packet_generator.RtxStatus() != kRtxOff) {
rtx_ssrc = rtp_sender_->packet_generator.RtxSsrc();
}
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl.h b/modules/rtp_rtcp/source/rtp_rtcp_impl.h
index 143b5f1..ba703bb 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl.h
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl.h
@@ -15,12 +15,12 @@
#include <stdint.h>
#include <memory>
+#include <optional>
#include <set>
#include <string>
#include <vector>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/rtp_headers.h"
#include "api/video/video_bitrate_allocation.h"
#include "modules/include/module_fec_types.h"
@@ -107,12 +107,12 @@
void SetRtxSendStatus(int mode) override;
int RtxSendStatus() const override;
- absl::optional<uint32_t> RtxSsrc() const override;
+ std::optional<uint32_t> RtxSsrc() const override;
void SetRtxSendPayloadType(int payload_type,
int associated_payload_type) override;
- absl::optional<uint32_t> FlexfecSsrc() const override;
+ std::optional<uint32_t> FlexfecSsrc() const override;
// Sends kRtcpByeCode when going from true to false.
int32_t SetSendingStatus(bool sending) override;
@@ -185,7 +185,7 @@
int32_t SetCNAME(absl::string_view c_name) override;
// Get RoundTripTime.
- absl::optional<TimeDelta> LastRtt() const override;
+ std::optional<TimeDelta> LastRtt() const override;
TimeDelta ExpectedRetransmissionTime() const override;
@@ -202,10 +202,10 @@
// Within this list, the `ReportBlockData::source_ssrc()`, which is the SSRC
// of the corresponding outbound RTP stream, is unique.
std::vector<ReportBlockData> GetLatestReportBlockData() const override;
- absl::optional<SenderReportStats> GetSenderReportStats() const override;
+ std::optional<SenderReportStats> GetSenderReportStats() const override;
// Round trip time statistics computed from the XR block contained in the last
// report.
- absl::optional<NonSenderRttStats> GetNonSenderRttStats() const override;
+ std::optional<NonSenderRttStats> GetNonSenderRttStats() const override;
// (REMB) Receiver Estimated Max Bitrate.
void SetRemb(int64_t bitrate_bps, std::vector<uint32_t> ssrcs) override;
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc b/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc
index a4aae47..b8cf585 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc
@@ -15,13 +15,13 @@
#include <algorithm>
#include <cstdint>
#include <memory>
+#include <optional>
#include <set>
#include <string>
#include <utility>
#include "absl/memory/memory.h"
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/sequence_checker.h"
#include "api/units/time_delta.h"
#include "api/units/timestamp.h"
@@ -156,15 +156,15 @@
associated_payload_type);
}
-absl::optional<uint32_t> ModuleRtpRtcpImpl2::RtxSsrc() const {
- return rtp_sender_ ? rtp_sender_->packet_generator.RtxSsrc() : absl::nullopt;
+std::optional<uint32_t> ModuleRtpRtcpImpl2::RtxSsrc() const {
+ return rtp_sender_ ? rtp_sender_->packet_generator.RtxSsrc() : std::nullopt;
}
-absl::optional<uint32_t> ModuleRtpRtcpImpl2::FlexfecSsrc() const {
+std::optional<uint32_t> ModuleRtpRtcpImpl2::FlexfecSsrc() const {
if (rtp_sender_) {
return rtp_sender_->packet_generator.FlexfecSsrc();
}
- return absl::nullopt;
+ return std::nullopt;
}
void ModuleRtpRtcpImpl2::IncomingRtcpPacket(
@@ -277,7 +277,7 @@
}
state.receiver = &rtcp_receiver_;
- if (absl::optional<RtpRtcpInterface::SenderReportStats> last_sr =
+ if (std::optional<RtpRtcpInterface::SenderReportStats> last_sr =
rtcp_receiver_.GetSenderReportStats();
last_sr.has_value()) {
state.remote_sr = CompactNtp(last_sr->last_remote_timestamp);
@@ -329,11 +329,11 @@
}
// TODO(bugs.webrtc.org/12873): Migrate this method and it's users to use
// optional Timestamps.
- absl::optional<Timestamp> capture_time;
+ std::optional<Timestamp> capture_time;
if (capture_time_ms > 0) {
capture_time = Timestamp::Millis(capture_time_ms);
}
- absl::optional<int> payload_type_optional;
+ std::optional<int> payload_type_optional;
if (payload_type >= 0)
payload_type_optional = payload_type;
@@ -497,8 +497,8 @@
return rtcp_sender_.SetCNAME(c_name);
}
-absl::optional<TimeDelta> ModuleRtpRtcpImpl2::LastRtt() const {
- absl::optional<TimeDelta> rtt = rtcp_receiver_.LastRtt();
+std::optional<TimeDelta> ModuleRtpRtcpImpl2::LastRtt() const {
+ std::optional<TimeDelta> rtt = rtcp_receiver_.LastRtt();
if (!rtt.has_value()) {
MutexLock lock(&mutex_rtt_);
if (rtt_ms_ > 0) {
@@ -515,7 +515,7 @@
}
// No rtt available (`kRttUpdateInterval` not yet passed?), so try to
// poll avg_rtt_ms directly from rtcp receiver.
- if (absl::optional<TimeDelta> rtt = rtcp_receiver_.AverageRtt()) {
+ if (std::optional<TimeDelta> rtt = rtcp_receiver_.AverageRtt()) {
return *rtt;
}
return kDefaultExpectedRetransmissionTime;
@@ -539,12 +539,12 @@
return rtcp_receiver_.GetLatestReportBlockData();
}
-absl::optional<RtpRtcpInterface::SenderReportStats>
+std::optional<RtpRtcpInterface::SenderReportStats>
ModuleRtpRtcpImpl2::GetSenderReportStats() const {
return rtcp_receiver_.GetSenderReportStats();
}
-absl::optional<RtpRtcpInterface::NonSenderRttStats>
+std::optional<RtpRtcpInterface::NonSenderRttStats>
ModuleRtpRtcpImpl2::GetNonSenderRttStats() const {
RTCPReceiver::NonSenderRttStats non_sender_rtt_stats =
rtcp_receiver_.GetNonSenderRTT();
@@ -630,7 +630,7 @@
// Use RTT from RtcpRttStats class if provided.
int64_t rtt = rtt_ms();
if (rtt == 0) {
- if (absl::optional<TimeDelta> average_rtt = rtcp_receiver_.AverageRtt()) {
+ if (std::optional<TimeDelta> average_rtt = rtcp_receiver_.AverageRtt()) {
rtt = average_rtt->ms();
}
}
@@ -705,7 +705,7 @@
// Use RTT from RtcpRttStats class if provided.
int64_t rtt = rtt_ms();
if (rtt == 0) {
- if (absl::optional<TimeDelta> average_rtt = rtcp_receiver_.AverageRtt()) {
+ if (std::optional<TimeDelta> average_rtt = rtcp_receiver_.AverageRtt()) {
rtt = average_rtt->ms();
}
}
@@ -716,7 +716,7 @@
rtc::ArrayView<const ReportBlockData> report_blocks) {
if (rtp_sender_) {
uint32_t ssrc = SSRC();
- absl::optional<uint32_t> rtx_ssrc;
+ std::optional<uint32_t> rtx_ssrc;
if (rtp_sender_->packet_generator.RtxStatus() != kRtxOff) {
rtx_ssrc = rtp_sender_->packet_generator.RtxSsrc();
}
@@ -766,7 +766,7 @@
RTC_DCHECK_RUN_ON(worker_queue_);
Timestamp check_since = clock_->CurrentTime() - kRttUpdateInterval;
- absl::optional<TimeDelta> rtt =
+ std::optional<TimeDelta> rtt =
rtcp_receiver_.OnPeriodicRttUpdate(check_since, rtcp_sender_.Sending());
if (rtt) {
if (rtt_stats_) {
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl2.h b/modules/rtp_rtcp/source/rtp_rtcp_impl2.h
index cd365b2..b2c0537 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl2.h
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl2.h
@@ -15,12 +15,12 @@
#include <stdint.h>
#include <memory>
+#include <optional>
#include <set>
#include <string>
#include <vector>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/environment/environment.h"
#include "api/rtp_headers.h"
#include "api/sequence_checker.h"
@@ -119,12 +119,12 @@
void SetRtxSendStatus(int mode) override;
int RtxSendStatus() const override;
- absl::optional<uint32_t> RtxSsrc() const override;
+ std::optional<uint32_t> RtxSsrc() const override;
void SetRtxSendPayloadType(int payload_type,
int associated_payload_type) override;
- absl::optional<uint32_t> FlexfecSsrc() const override;
+ std::optional<uint32_t> FlexfecSsrc() const override;
// Sends kRtcpByeCode when going from true to false.
int32_t SetSendingStatus(bool sending) override;
@@ -189,7 +189,7 @@
int32_t SetCNAME(absl::string_view c_name) override;
// Get RoundTripTime.
- absl::optional<TimeDelta> LastRtt() const override;
+ std::optional<TimeDelta> LastRtt() const override;
TimeDelta ExpectedRetransmissionTime() const override;
@@ -207,8 +207,8 @@
// Within this list, the `ReportBlockData::source_ssrc()`, which is the SSRC
// of the corresponding outbound RTP stream, is unique.
std::vector<ReportBlockData> GetLatestReportBlockData() const override;
- absl::optional<SenderReportStats> GetSenderReportStats() const override;
- absl::optional<NonSenderRttStats> GetNonSenderRttStats() const override;
+ std::optional<SenderReportStats> GetSenderReportStats() const override;
+ std::optional<NonSenderRttStats> GetNonSenderRttStats() const override;
// (REMB) Receiver Estimated Max Bitrate.
void SetRemb(int64_t bitrate_bps, std::vector<uint32_t> ssrcs) override;
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl2_unittest.cc b/modules/rtp_rtcp/source/rtp_rtcp_impl2_unittest.cc
index 9332813..63224e7 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl2_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl2_unittest.cc
@@ -13,10 +13,10 @@
#include <deque>
#include <map>
#include <memory>
+#include <optional>
#include <set>
#include <utility>
-#include "absl/types/optional.h"
#include "api/environment/environment.h"
#include "api/environment/environment_factory.h"
#include "api/field_trials_registry.h"
@@ -194,7 +194,7 @@
counter_map_[ssrc] = packet_counter;
}
- void OnSendPacket(absl::optional<uint16_t> packet_id,
+ void OnSendPacket(std::optional<uint16_t> packet_id,
Timestamp capture_time,
uint32_t ssrc) override {
if (packet_id.has_value()) {
@@ -202,7 +202,7 @@
}
}
- absl::optional<SentPacket> last_sent_packet() const {
+ std::optional<SentPacket> last_sent_packet() const {
return last_sent_packet_;
}
@@ -245,7 +245,7 @@
config.rtcp_report_interval_ms = rtcp_report_interval_.ms();
config.local_media_ssrc = is_sender_ ? kSenderSsrc : kReceiverSsrc;
config.rtx_send_ssrc =
- is_sender_ ? absl::make_optional(kRtxSenderSsrc) : absl::nullopt;
+ is_sender_ ? std::make_optional(kRtxSenderSsrc) : std::nullopt;
config.need_rtp_packet_infos = true;
config.non_sender_rtt_measurement = true;
config.send_packet_observer = this;
@@ -257,7 +257,7 @@
private:
std::map<uint32_t, RtcpPacketTypeCounter> counter_map_;
- absl::optional<SentPacket> last_sent_packet_;
+ std::optional<SentPacket> last_sent_packet_;
VideoFecGenerator* fec_generator_ = nullptr;
TimeDelta rtcp_report_interval_ = kDefaultReportInterval;
};
@@ -302,7 +302,7 @@
}
void ReinitWithFec(VideoFecGenerator* fec_generator,
- absl::optional<int> red_payload_type) {
+ std::optional<int> red_payload_type) {
sender_.ReinintWithFec(fec_generator);
EXPECT_EQ(0, sender_.impl_->SetSendingStatus(true));
sender_.impl_->SetSendingMediaStatus(true);
@@ -765,7 +765,7 @@
// Checks that the sender report stats are not available if no RTCP SR was sent.
TEST_F(RtpRtcpImpl2Test, SenderReportStatsNotAvailable) {
- EXPECT_THAT(receiver_.impl_->GetSenderReportStats(), Eq(absl::nullopt));
+ EXPECT_THAT(receiver_.impl_->GetSenderReportStats(), Eq(std::nullopt));
}
// Checks that the sender report stats are available if an RTCP SR was sent.
@@ -775,7 +775,7 @@
// Send an SR.
ASSERT_THAT(sender_.impl_->SendRTCP(kRtcpReport), Eq(0));
AdvanceTime(kOneWayNetworkDelay);
- EXPECT_THAT(receiver_.impl_->GetSenderReportStats(), Not(Eq(absl::nullopt)));
+ EXPECT_THAT(receiver_.impl_->GetSenderReportStats(), Not(Eq(std::nullopt)));
}
// Checks that the sender report stats are not available if an RTCP SR with an
@@ -792,7 +792,7 @@
sr.SetOctetCount(456u);
auto raw_packet = sr.Build();
receiver_.impl_->IncomingRtcpPacket(raw_packet);
- EXPECT_THAT(receiver_.impl_->GetSenderReportStats(), Eq(absl::nullopt));
+ EXPECT_THAT(receiver_.impl_->GetSenderReportStats(), Eq(std::nullopt));
}
// Checks the stats derived from the last received RTCP SR are set correctly.
@@ -844,7 +844,7 @@
ASSERT_THAT(sender_.impl_->SendRTCP(kRtcpReport), Eq(0));
AdvanceTime(kOneWayNetworkDelay);
auto stats = receiver_.impl_->GetSenderReportStats();
- ASSERT_THAT(stats, Not(Eq(absl::nullopt)));
+ ASSERT_THAT(stats, Not(Eq(std::nullopt)));
EXPECT_TRUE(stats->last_arrival_timestamp.Valid());
}
@@ -1006,7 +1006,7 @@
FlexfecSender flexfec_sender(env_, kFlexfecPayloadType, kFlexfecSsrc,
kSenderSsrc, kNoMid, kNoRtpExtensions,
kNoRtpExtensionSizes, &start_state);
- ReinitWithFec(&flexfec_sender, /*red_payload_type=*/absl::nullopt);
+ ReinitWithFec(&flexfec_sender, /*red_payload_type=*/std::nullopt);
// Parameters selected to generate a single FEC packet per media packet.
FecProtectionParams params;
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc b/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
index abf3b63..1ed37e0 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
@@ -599,7 +599,7 @@
// Checks that the remote sender stats are not available if no RTCP SR was sent.
TEST_F(RtpRtcpImplTest, SenderReportStatsNotAvailable) {
- EXPECT_THAT(receiver_.impl_->GetSenderReportStats(), Eq(absl::nullopt));
+ EXPECT_THAT(receiver_.impl_->GetSenderReportStats(), Eq(std::nullopt));
}
// Checks that the remote sender stats are available if an RTCP SR was sent.
@@ -608,7 +608,7 @@
SendFrame(&sender_, sender_video_.get(), kBaseLayerTid);
// Send an SR.
ASSERT_THAT(sender_.impl_->SendRTCP(kRtcpReport), Eq(0));
- EXPECT_THAT(receiver_.impl_->GetSenderReportStats(), Not(Eq(absl::nullopt)));
+ EXPECT_THAT(receiver_.impl_->GetSenderReportStats(), Not(Eq(std::nullopt)));
}
// Checks that the remote sender stats are not available if an RTCP SR with an
@@ -624,7 +624,7 @@
sr.SetPacketCount(123u);
sr.SetOctetCount(456u);
receiver_.impl_->IncomingRtcpPacket(sr.Build());
- EXPECT_THAT(receiver_.impl_->GetSenderReportStats(), Eq(absl::nullopt));
+ EXPECT_THAT(receiver_.impl_->GetSenderReportStats(), Eq(std::nullopt));
}
// Checks the stats derived from the last received RTCP SR are set correctly.
@@ -672,7 +672,7 @@
// Send an SR.
ASSERT_THAT(sender_.impl_->SendRTCP(kRtcpReport), Eq(0));
auto stats = receiver_.impl_->GetSenderReportStats();
- ASSERT_THAT(stats, Not(Eq(absl::nullopt)));
+ ASSERT_THAT(stats, Not(Eq(std::nullopt)));
EXPECT_TRUE(stats->last_arrival_timestamp.Valid());
}
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_interface.h b/modules/rtp_rtcp/source/rtp_rtcp_interface.h
index dfdc13d..d400b76 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_interface.h
+++ b/modules/rtp_rtcp/source/rtp_rtcp_interface.h
@@ -12,11 +12,11 @@
#define MODULES_RTP_RTCP_SOURCE_RTP_RTCP_INTERFACE_H_
#include <memory>
+#include <optional>
#include <string>
#include <vector>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/field_trials_view.h"
#include "api/frame_transformer_interface.h"
#include "api/rtp_packet_sender.h"
@@ -127,7 +127,7 @@
// SSRCs for media and retransmission, respectively.
// FlexFec SSRC is fetched from `flexfec_sender`.
uint32_t local_media_ssrc = 0;
- absl::optional<uint32_t> rtx_send_ssrc;
+ std::optional<uint32_t> rtx_send_ssrc;
bool need_rtp_packet_infos = false;
@@ -171,7 +171,7 @@
// Refer to https://datatracker.ietf.org/doc/html/rfc3611#section-2.
struct NonSenderRttStats {
// https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-roundtriptime
- absl::optional<TimeDelta> round_trip_time;
+ std::optional<TimeDelta> round_trip_time;
// https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-totalroundtriptime
TimeDelta total_round_trip_time = TimeDelta::Zero();
// https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-roundtriptimemeasurements
@@ -263,7 +263,7 @@
virtual int RtxSendStatus() const = 0;
// Returns the SSRC used for RTX if set, otherwise a nullopt.
- virtual absl::optional<uint32_t> RtxSsrc() const = 0;
+ virtual std::optional<uint32_t> RtxSsrc() const = 0;
// Sets the payload type to use when sending RTX packets. Note that this
// doesn't enable RTX, only the payload type is set.
@@ -271,7 +271,7 @@
int associated_payload_type) = 0;
// Returns the FlexFEC SSRC, if there is one.
- virtual absl::optional<uint32_t> FlexfecSsrc() const = 0;
+ virtual std::optional<uint32_t> FlexfecSsrc() const = 0;
// Sets sending status.
// Returns -1 on failure else 0.
@@ -386,7 +386,7 @@
virtual int32_t SetCNAME(absl::string_view cname) = 0;
// Returns current RTT (round-trip time) estimate.
- virtual absl::optional<TimeDelta> LastRtt() const = 0;
+ virtual std::optional<TimeDelta> LastRtt() const = 0;
// Returns the estimated RTT, with fallback to a default value.
virtual TimeDelta ExpectedRetransmissionTime() const = 0;
@@ -407,9 +407,9 @@
// that pair.
virtual std::vector<ReportBlockData> GetLatestReportBlockData() const = 0;
// Returns stats based on the received RTCP SRs.
- virtual absl::optional<SenderReportStats> GetSenderReportStats() const = 0;
+ virtual std::optional<SenderReportStats> GetSenderReportStats() const = 0;
// Returns non-sender RTT stats, based on DLRR.
- virtual absl::optional<NonSenderRttStats> GetNonSenderRttStats() const = 0;
+ virtual std::optional<NonSenderRttStats> GetNonSenderRttStats() const = 0;
// (REMB) Receiver Estimated Max Bitrate.
// Schedules sending REMB on next and following sender/receiver reports.
diff --git a/modules/rtp_rtcp/source/rtp_sender.cc b/modules/rtp_rtcp/source/rtp_sender.cc
index 2abbfe5..bfdaca0 100644
--- a/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/modules/rtp_rtcp/source/rtp_sender.cc
@@ -15,12 +15,12 @@
#include <cstdint>
#include <cstring>
#include <memory>
+#include <optional>
#include <string>
#include <utility>
#include <vector>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/rtp_headers.h"
#include "api/rtp_packet_sender.h"
@@ -161,7 +161,7 @@
ssrc_(config.local_media_ssrc),
rtx_ssrc_(config.rtx_send_ssrc),
flexfec_ssrc_(config.fec_generator ? config.fec_generator->FecSsrc()
- : absl::nullopt),
+ : std::nullopt),
packet_history_(packet_history),
paced_sender_(packet_sender),
sending_media_(true), // Default to sending media.
diff --git a/modules/rtp_rtcp/source/rtp_sender.h b/modules/rtp_rtcp/source/rtp_sender.h
index 61a1a5f..0877857 100644
--- a/modules/rtp_rtcp/source/rtp_sender.h
+++ b/modules/rtp_rtcp/source/rtp_sender.h
@@ -13,12 +13,12 @@
#include <map>
#include <memory>
+#include <optional>
#include <string>
#include <utility>
#include <vector>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/call/transport.h"
#include "api/field_trials_view.h"
@@ -102,7 +102,7 @@
// RTX.
void SetRtxStatus(int mode) RTC_LOCKS_EXCLUDED(send_mutex_);
int RtxStatus() const RTC_LOCKS_EXCLUDED(send_mutex_);
- absl::optional<uint32_t> RtxSsrc() const RTC_LOCKS_EXCLUDED(send_mutex_) {
+ std::optional<uint32_t> RtxSsrc() const RTC_LOCKS_EXCLUDED(send_mutex_) {
return rtx_ssrc_;
}
// Returns expected size difference between an RTX packet and media packet
@@ -140,7 +140,7 @@
uint32_t SSRC() const RTC_LOCKS_EXCLUDED(send_mutex_) { return ssrc_; }
- absl::optional<uint32_t> FlexfecSsrc() const RTC_LOCKS_EXCLUDED(send_mutex_) {
+ std::optional<uint32_t> FlexfecSsrc() const RTC_LOCKS_EXCLUDED(send_mutex_) {
return flexfec_ssrc_;
}
@@ -172,8 +172,8 @@
const bool audio_configured_;
const uint32_t ssrc_;
- const absl::optional<uint32_t> rtx_ssrc_;
- const absl::optional<uint32_t> flexfec_ssrc_;
+ const std::optional<uint32_t> rtx_ssrc_;
+ const std::optional<uint32_t> flexfec_ssrc_;
RtpPacketHistory* const packet_history_;
RtpPacketSender* const paced_sender_;
diff --git a/modules/rtp_rtcp/source/rtp_sender_audio.cc b/modules/rtp_rtcp/source/rtp_sender_audio.cc
index aec54d3..3330f39 100644
--- a/modules/rtp_rtcp/source/rtp_sender_audio.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_audio.cc
@@ -13,11 +13,11 @@
#include <string.h>
#include <memory>
+#include <optional>
#include <utility>
#include <vector>
#include "absl/strings/match.h"
-#include "absl/types/optional.h"
#include "api/audio_codecs/audio_format.h"
#include "api/rtp_headers.h"
#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
@@ -137,7 +137,7 @@
// updates, with a value of 50 ms RECOMMENDED.
constexpr int kDtmfIntervalTimeMs = 50;
uint32_t dtmf_payload_freq = 0;
- absl::optional<AbsoluteCaptureTime> absolute_capture_time;
+ std::optional<AbsoluteCaptureTime> absolute_capture_time;
{
MutexLock lock(&send_audio_mutex_);
dtmf_payload_freq = dtmf_payload_freq_;
diff --git a/modules/rtp_rtcp/source/rtp_sender_audio.h b/modules/rtp_rtcp/source/rtp_sender_audio.h
index 83a2cb2..d17c96c 100644
--- a/modules/rtp_rtcp/source/rtp_sender_audio.h
+++ b/modules/rtp_rtcp/source/rtp_sender_audio.h
@@ -55,12 +55,12 @@
uint32_t rtp_timestamp = 0;
// capture time of the audio frame in the same epoch as `clock->CurrentTime`
- absl::optional<Timestamp> capture_time;
+ std::optional<Timestamp> capture_time;
// Audio level in dBov for
// header-extension-for-audio-level-indication.
// Valid range is [0,127]. Actual value is negative.
- absl::optional<int> audio_level_dbov;
+ std::optional<int> audio_level_dbov;
// Contributing sources list.
rtc::ArrayView<const uint32_t> csrcs;
@@ -107,7 +107,7 @@
OneTimeEvent first_packet_sent_;
- absl::optional<int> encoder_rtp_timestamp_frequency_
+ std::optional<int> encoder_rtp_timestamp_frequency_
RTC_GUARDED_BY(send_audio_mutex_);
AbsoluteCaptureTimeSender absolute_capture_time_sender_
diff --git a/modules/rtp_rtcp/source/rtp_sender_egress.cc b/modules/rtp_rtcp/source/rtp_sender_egress.cc
index 5c97ef8..0753ec3 100644
--- a/modules/rtp_rtcp/source/rtp_sender_egress.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_egress.cc
@@ -94,7 +94,7 @@
ssrc_(config.local_media_ssrc),
rtx_ssrc_(config.rtx_send_ssrc),
flexfec_ssrc_(config.fec_generator ? config.fec_generator->FecSsrc()
- : absl::nullopt),
+ : std::nullopt),
populate_network2_timestamp_(config.populate_network2_timestamp),
clock_(config.clock),
packet_history_(packet_history),
@@ -175,7 +175,7 @@
// This packet should be protected by FEC, add it to packet generator.
RTC_DCHECK(fec_generator_);
RTC_DCHECK(packet->packet_type() == RtpPacketMediaType::kVideo);
- absl::optional<std::pair<FecProtectionParams, FecProtectionParams>>
+ std::optional<std::pair<FecProtectionParams, FecProtectionParams>>
new_fec_params;
new_fec_params.swap(pending_fec_params_);
if (new_fec_params) {
@@ -275,7 +275,7 @@
// bit and will wrap. We should be able to use the 64bit value as id, but in
// order to not change behaviour we use the 16bit extension value if it is
// used.
- absl::optional<uint16_t> packet_id =
+ std::optional<uint16_t> packet_id =
packet->GetExtension<TransportSequenceNumber>();
if (packet_id.has_value()) {
options.packet_id = *packet_id;
diff --git a/modules/rtp_rtcp/source/rtp_sender_egress.h b/modules/rtp_rtcp/source/rtp_sender_egress.h
index 49ee8c1..3e4fa66 100644
--- a/modules/rtp_rtcp/source/rtp_sender_egress.h
+++ b/modules/rtp_rtcp/source/rtp_sender_egress.h
@@ -13,10 +13,10 @@
#include <map>
#include <memory>
+#include <optional>
#include <utility>
#include <vector>
-#include "absl/types/optional.h"
#include "api/call/transport.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/sequence_checker.h"
@@ -72,8 +72,8 @@
const PacedPacketInfo& pacing_info);
void OnBatchComplete();
uint32_t Ssrc() const { return ssrc_; }
- absl::optional<uint32_t> RtxSsrc() const { return rtx_ssrc_; }
- absl::optional<uint32_t> FlexFecSsrc() const { return flexfec_ssrc_; }
+ std::optional<uint32_t> RtxSsrc() const { return rtx_ssrc_; }
+ std::optional<uint32_t> FlexFecSsrc() const { return flexfec_ssrc_; }
RtpSendRates GetSendRates(Timestamp now) const;
void GetDataCounters(StreamDataCounters* rtp_stats,
@@ -126,8 +126,8 @@
const bool enable_send_packet_batching_;
TaskQueueBase* const worker_queue_;
const uint32_t ssrc_;
- const absl::optional<uint32_t> rtx_ssrc_;
- const absl::optional<uint32_t> flexfec_ssrc_;
+ const std::optional<uint32_t> rtx_ssrc_;
+ const std::optional<uint32_t> flexfec_ssrc_;
const bool populate_network2_timestamp_;
Clock* const clock_;
RtpPacketHistory* const packet_history_ RTC_GUARDED_BY(worker_queue_);
@@ -136,8 +136,8 @@
const bool is_audio_;
const bool need_rtp_packet_infos_;
VideoFecGenerator* const fec_generator_ RTC_GUARDED_BY(worker_queue_);
- absl::optional<uint16_t> last_sent_seq_ RTC_GUARDED_BY(worker_queue_);
- absl::optional<uint16_t> last_sent_rtx_seq_ RTC_GUARDED_BY(worker_queue_);
+ std::optional<uint16_t> last_sent_seq_ RTC_GUARDED_BY(worker_queue_);
+ std::optional<uint16_t> last_sent_rtx_seq_ RTC_GUARDED_BY(worker_queue_);
SendPacketObserver* const send_packet_observer_;
StreamDataCountersCallback* const rtp_stats_callback_;
@@ -151,7 +151,7 @@
StreamDataCounters rtx_rtp_stats_ RTC_GUARDED_BY(worker_queue_);
// One element per value in RtpPacketMediaType, with index matching value.
std::vector<BitrateTracker> send_rates_ RTC_GUARDED_BY(worker_queue_);
- absl::optional<std::pair<FecProtectionParams, FecProtectionParams>>
+ std::optional<std::pair<FecProtectionParams, FecProtectionParams>>
pending_fec_params_ RTC_GUARDED_BY(worker_queue_);
// Maps sent packets' sequence numbers to a tuple consisting of:
diff --git a/modules/rtp_rtcp/source/rtp_sender_egress_unittest.cc b/modules/rtp_rtcp/source/rtp_sender_egress_unittest.cc
index 1835627..af88f0b 100644
--- a/modules/rtp_rtcp/source/rtp_sender_egress_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_egress_unittest.cc
@@ -11,9 +11,9 @@
#include "modules/rtp_rtcp/source/rtp_sender_egress.h"
#include <cstdint>
+#include <optional>
#include <string>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/call/transport.h"
#include "api/environment/environment.h"
@@ -62,7 +62,7 @@
public:
MOCK_METHOD(void,
OnSendPacket,
- (absl::optional<uint16_t>, Timestamp, uint32_t),
+ (std::optional<uint16_t>, Timestamp, uint32_t),
(override));
};
@@ -102,11 +102,11 @@
RTC_CHECK_NOTREACHED();
}
- absl::optional<TransmittedPacket> last_packet() { return last_packet_; }
+ std::optional<TransmittedPacket> last_packet() { return last_packet_; }
private:
DataSize total_data_sent_;
- absl::optional<TransmittedPacket> last_packet_;
+ std::optional<TransmittedPacket> last_packet_;
RtpHeaderExtensionMap* const extensions_;
};
@@ -289,7 +289,7 @@
std::unique_ptr<RtpSenderEgress> sender = CreateRtpSenderEgress();
sender->SendPacket(std::move(padding), PacedPacketInfo());
- absl::optional<int32_t> offset =
+ std::optional<int32_t> offset =
transport_.last_packet()->packet.GetExtension<TransmissionOffset>();
EXPECT_EQ(offset, 0);
}
@@ -375,7 +375,7 @@
std::unique_ptr<RtpSenderEgress> sender = CreateRtpSenderEgress();
EXPECT_CALL(send_packet_observer_,
- OnSendPacket(Eq(absl::nullopt), clock_->CurrentTime(), kSsrc));
+ OnSendPacket(Eq(std::nullopt), clock_->CurrentTime(), kSsrc));
sender->SendPacket(BuildRtpPacket(), PacedPacketInfo());
}
diff --git a/modules/rtp_rtcp/source/rtp_sender_unittest.cc b/modules/rtp_rtcp/source/rtp_sender_unittest.cc
index bd434dd..bfe71d4 100644
--- a/modules/rtp_rtcp/source/rtp_sender_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_unittest.cc
@@ -174,7 +174,7 @@
RateLimiter retransmission_rate_limiter_;
FlexfecSender flexfec_sender_;
- absl::optional<PacketSequencer> sequencer_;
+ std::optional<PacketSequencer> sequencer_;
std::unique_ptr<RtpPacketHistory> packet_history_;
std::unique_ptr<RTPSender> rtp_sender_;
@@ -434,7 +434,7 @@
TEST_F(RtpSenderTest, RequiresRtxSsrcToEnableRtx) {
RtpRtcpInterface::Configuration config = GetDefaultConfig();
- config.rtx_send_ssrc = absl::nullopt;
+ config.rtx_send_ssrc = std::nullopt;
RTPSender rtp_sender(config, packet_history_.get(), config.paced_sender);
rtp_sender.SetRtxPayloadType(kRtxPayload, kPayload);
@@ -1369,7 +1369,7 @@
RTPSenderVideo rtp_sender_video(video_config);
const uint8_t kPayloadType = 127;
- const absl::optional<VideoCodecType> kCodecType =
+ const std::optional<VideoCodecType> kCodecType =
VideoCodecType::kVideoCodecGeneric;
const uint32_t kCaptureTimeMsToRtpTimestamp = 90; // 90 kHz clock
diff --git a/modules/rtp_rtcp/source/rtp_sender_video.cc b/modules/rtp_rtcp/source/rtp_sender_video.cc
index fc39225..e505e28 100644
--- a/modules/rtp_rtcp/source/rtp_sender_video.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_video.cc
@@ -100,18 +100,18 @@
return true;
}
-absl::optional<VideoPlayoutDelay> LoadVideoPlayoutDelayOverride(
+std::optional<VideoPlayoutDelay> LoadVideoPlayoutDelayOverride(
const FieldTrialsView* key_value_config) {
RTC_DCHECK(key_value_config);
- FieldTrialOptional<int> playout_delay_min_ms("min_ms", absl::nullopt);
- FieldTrialOptional<int> playout_delay_max_ms("max_ms", absl::nullopt);
+ FieldTrialOptional<int> playout_delay_min_ms("min_ms", std::nullopt);
+ FieldTrialOptional<int> playout_delay_max_ms("max_ms", std::nullopt);
ParseFieldTrial({&playout_delay_max_ms, &playout_delay_min_ms},
key_value_config->Lookup("WebRTC-ForceSendPlayoutDelay"));
return playout_delay_max_ms && playout_delay_min_ms
- ? absl::make_optional<VideoPlayoutDelay>(
+ ? std::make_optional<VideoPlayoutDelay>(
TimeDelta::Millis(*playout_delay_min_ms),
TimeDelta::Millis(*playout_delay_max_ms))
- : absl::nullopt;
+ : std::nullopt;
}
// Some packets can be skipped and the stream can still be decoded. Those
@@ -456,7 +456,7 @@
}
bool RTPSenderVideo::SendVideo(int payload_type,
- absl::optional<VideoCodecType> codec_type,
+ std::optional<VideoCodecType> codec_type,
uint32_t rtp_timestamp,
Timestamp capture_time,
rtc::ArrayView<const uint8_t> payload,
@@ -746,7 +746,7 @@
}
bool RTPSenderVideo::SendEncodedImage(int payload_type,
- absl::optional<VideoCodecType> codec_type,
+ std::optional<VideoCodecType> codec_type,
uint32_t rtp_timestamp,
const EncodedImage& encoded_image,
RTPVideoHeader video_header,
@@ -832,7 +832,7 @@
Timestamp expected_next_frame_time = Timestamp::PlusInfinity();
for (int i = temporal_id - 1; i >= 0; --i) {
TemporalLayerStats* stats = &frame_stats_by_temporal_layer_[i];
- absl::optional<Frequency> rate = stats->frame_rate.Rate(now);
+ std::optional<Frequency> rate = stats->frame_rate.Rate(now);
if (rate > Frequency::Zero()) {
Timestamp tl_next = stats->last_frame_time + 1 / *rate;
if (tl_next - now > -expected_retransmission_time &&
@@ -856,7 +856,7 @@
void RTPSenderVideo::MaybeUpdateCurrentPlayoutDelay(
const RTPVideoHeader& header) {
- absl::optional<VideoPlayoutDelay> requested_delay =
+ std::optional<VideoPlayoutDelay> requested_delay =
forced_playout_delay_.has_value() ? forced_playout_delay_
: header.playout_delay;
diff --git a/modules/rtp_rtcp/source/rtp_sender_video.h b/modules/rtp_rtcp/source/rtp_sender_video.h
index fa5f3b6..4e8abf8 100644
--- a/modules/rtp_rtcp/source/rtp_sender_video.h
+++ b/modules/rtp_rtcp/source/rtp_sender_video.h
@@ -13,10 +13,10 @@
#include <map>
#include <memory>
+#include <optional>
#include <vector>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/frame_transformer_interface.h"
#include "api/scoped_refptr.h"
@@ -77,12 +77,12 @@
RTPSender* rtp_sender = nullptr;
// Some FEC data is duplicated here in preparation of moving FEC to
// the egress stage.
- absl::optional<VideoFecGenerator::FecType> fec_type;
+ std::optional<VideoFecGenerator::FecType> fec_type;
size_t fec_overhead_bytes = 0; // Per packet max FEC overhead.
FrameEncryptorInterface* frame_encryptor = nullptr;
bool require_frame_encryption = false;
bool enable_retransmit_all_layers = false;
- absl::optional<int> red_payload_type;
+ std::optional<int> red_payload_type;
const FieldTrialsView* field_trials = nullptr;
rtc::scoped_refptr<FrameTransformerInterface> frame_transformer;
TaskQueueFactory* task_queue_factory = nullptr;
@@ -98,7 +98,7 @@
// video encoder, excluding any additional overhead.
// Calls to this method are assumed to be externally serialized.
bool SendVideo(int payload_type,
- absl::optional<VideoCodecType> codec_type,
+ std::optional<VideoCodecType> codec_type,
uint32_t rtp_timestamp,
Timestamp capture_time,
rtc::ArrayView<const uint8_t> payload,
@@ -108,7 +108,7 @@
std::vector<uint32_t> csrcs) override;
bool SendEncodedImage(int payload_type,
- absl::optional<VideoCodecType> codec_type,
+ std::optional<VideoCodecType> codec_type,
uint32_t rtp_timestamp,
const EncodedImage& encoded_image,
RTPVideoHeader video_header,
@@ -199,32 +199,32 @@
rtc::RaceChecker send_checker_;
int32_t retransmission_settings_ RTC_GUARDED_BY(send_checker_);
VideoRotation last_rotation_ RTC_GUARDED_BY(send_checker_);
- absl::optional<ColorSpace> last_color_space_ RTC_GUARDED_BY(send_checker_);
+ std::optional<ColorSpace> last_color_space_ RTC_GUARDED_BY(send_checker_);
bool transmit_color_space_next_frame_ RTC_GUARDED_BY(send_checker_);
std::unique_ptr<FrameDependencyStructure> video_structure_
RTC_GUARDED_BY(send_checker_);
- absl::optional<VideoLayersAllocation> allocation_
+ std::optional<VideoLayersAllocation> allocation_
RTC_GUARDED_BY(send_checker_);
// Flag indicating if we should send `allocation_`.
SendVideoLayersAllocation send_allocation_ RTC_GUARDED_BY(send_checker_);
- absl::optional<VideoLayersAllocation> last_full_sent_allocation_
+ std::optional<VideoLayersAllocation> last_full_sent_allocation_
RTC_GUARDED_BY(send_checker_);
// Current target playout delay.
- absl::optional<VideoPlayoutDelay> current_playout_delay_
+ std::optional<VideoPlayoutDelay> current_playout_delay_
RTC_GUARDED_BY(send_checker_);
// Flag indicating if we need to send `current_playout_delay_` in order
// to guarantee it gets delivered.
bool playout_delay_pending_;
// Set by the field trial WebRTC-ForceSendPlayoutDelay to override the playout
// delay of outgoing video frames.
- const absl::optional<VideoPlayoutDelay> forced_playout_delay_;
+ const std::optional<VideoPlayoutDelay> forced_playout_delay_;
// Should never be held when calling out of this class.
Mutex mutex_;
- const absl::optional<int> red_payload_type_;
- absl::optional<VideoFecGenerator::FecType> fec_type_;
+ const std::optional<int> red_payload_type_;
+ std::optional<VideoFecGenerator::FecType> fec_type_;
const size_t fec_overhead_bytes_; // Per packet max FEC overhead.
mutable Mutex stats_mutex_;
diff --git a/modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate.cc b/modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate.cc
index 8c24160..b49468b 100644
--- a/modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate.cc
@@ -13,11 +13,11 @@
#include <cstddef>
#include <cstdint>
#include <memory>
+#include <optional>
#include <string>
#include <utility>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/frame_transformer_interface.h"
#include "api/scoped_refptr.h"
@@ -51,7 +51,7 @@
TransformableVideoSenderFrame(const EncodedImage& encoded_image,
const RTPVideoHeader& video_header,
int payload_type,
- absl::optional<VideoCodecType> codec_type,
+ std::optional<VideoCodecType> codec_type,
uint32_t rtp_timestamp,
TimeDelta expected_retransmission_time,
uint32_t ssrc,
@@ -112,9 +112,9 @@
const RTPVideoHeader& GetHeader() const { return header_; }
uint8_t GetPayloadType() const override { return payload_type_; }
- absl::optional<VideoCodecType> GetCodecType() const { return codec_type_; }
+ std::optional<VideoCodecType> GetCodecType() const { return codec_type_; }
Timestamp GetCaptureTime() const { return capture_time_; }
- absl::optional<Timestamp> GetCaptureTimeIdentifier() const override {
+ std::optional<Timestamp> GetCaptureTimeIdentifier() const override {
return capture_time_identifier_;
}
@@ -137,10 +137,10 @@
RTPVideoHeader header_;
const VideoFrameType frame_type_;
const uint8_t payload_type_;
- const absl::optional<VideoCodecType> codec_type_ = absl::nullopt;
+ const std::optional<VideoCodecType> codec_type_ = std::nullopt;
uint32_t timestamp_;
const Timestamp capture_time_;
- const absl::optional<Timestamp> capture_time_identifier_;
+ const std::optional<Timestamp> capture_time_identifier_;
const TimeDelta expected_retransmission_time_;
uint32_t ssrc_;
@@ -166,7 +166,7 @@
bool RTPSenderVideoFrameTransformerDelegate::TransformFrame(
int payload_type,
- absl::optional<VideoCodecType> codec_type,
+ std::optional<VideoCodecType> codec_type,
uint32_t rtp_timestamp,
const EncodedImage& encoded_image,
RTPVideoHeader video_header,
diff --git a/modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate.h b/modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate.h
index a32c708..3580d84 100644
--- a/modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate.h
+++ b/modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate.h
@@ -33,7 +33,7 @@
class RTPVideoFrameSenderInterface {
public:
virtual bool SendVideo(int payload_type,
- absl::optional<VideoCodecType> codec_type,
+ std::optional<VideoCodecType> codec_type,
uint32_t rtp_timestamp,
Timestamp capture_time,
rtc::ArrayView<const uint8_t> payload,
@@ -66,7 +66,7 @@
// Delegates the call to FrameTransformerInterface::TransformFrame.
bool TransformFrame(int payload_type,
- absl::optional<VideoCodecType> codec_type,
+ std::optional<VideoCodecType> codec_type,
uint32_t rtp_timestamp,
const EncodedImage& encoded_image,
RTPVideoHeader video_header,
diff --git a/modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate_unittest.cc b/modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate_unittest.cc
index 586836a..4224b8c 100644
--- a/modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate_unittest.cc
@@ -33,7 +33,7 @@
MOCK_METHOD(bool,
SendVideo,
(int payload_type,
- absl::optional<VideoCodecType> codec_type,
+ std::optional<VideoCodecType> codec_type,
uint32_t rtp_timestamp,
Timestamp capture_time,
rtc::ArrayView<const uint8_t> payload,
@@ -258,7 +258,7 @@
rtc::Event event;
EXPECT_CALL(
test_sender_,
- SendVideo(payload_type, absl::make_optional(kVideoCodecVP8), timestamp,
+ SendVideo(payload_type, std::make_optional(kVideoCodecVP8), timestamp,
/*capture_time=*/Timestamp::MinusInfinity(), buffer, _, _,
/*expected_retransmission_time=*/TimeDelta::Millis(10),
frame_csrcs))
diff --git a/modules/rtp_rtcp/source/rtp_sender_video_unittest.cc b/modules/rtp_rtcp/source/rtp_sender_video_unittest.cc
index ee7d18b..8c8091f 100644
--- a/modules/rtp_rtcp/source/rtp_sender_video_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_video_unittest.cc
@@ -1268,7 +1268,7 @@
kPayload, kType, kTimestamp, kAbsoluteCaptureTimestamp, kFrame,
sizeof(kFrame), hdr, kDefaultExpectedRetransmissionTime, {});
- absl::optional<AbsoluteCaptureTime> absolute_capture_time;
+ std::optional<AbsoluteCaptureTime> absolute_capture_time;
// It is expected that one and only one of the packets sent on this video
// frame has absolute capture time header extension.
@@ -1293,7 +1293,7 @@
TEST_F(RtpSenderVideoTest, AbsoluteCaptureTimeWithExtensionProvided) {
constexpr AbsoluteCaptureTime kAbsoluteCaptureTime = {
123,
- absl::optional<int64_t>(456),
+ std::optional<int64_t>(456),
};
uint8_t kFrame[kMaxPacketLength];
rtp_module_.RegisterRtpHeaderExtension(AbsoluteCaptureTimeExtension::Uri(),
@@ -1307,7 +1307,7 @@
sizeof(kFrame), hdr,
kDefaultExpectedRetransmissionTime, {});
- absl::optional<AbsoluteCaptureTime> absolute_capture_time;
+ std::optional<AbsoluteCaptureTime> absolute_capture_time;
// It is expected that one and only one of the packets sent on this video
// frame has absolute capture time header extension.
@@ -1360,7 +1360,7 @@
// Set playout delay on a non-discardable frame, the extension should still
// be populated since dilvery wasn't guaranteed on the last one.
- hdr.playout_delay = absl::nullopt; // Indicates "no change".
+ hdr.playout_delay = std::nullopt; // Indicates "no change".
vp8_header.temporalIdx = 0;
rtp_sender_video_->SendVideo(
kPayload, kType, kTimestamp, fake_clock_.CurrentTime(), kFrame,
@@ -1428,7 +1428,7 @@
RTPVideoHeader video_header;
video_header.frame_type = VideoFrameType::kVideoFrameKey;
ASSERT_TRUE(rtp_sender_video_->SendVideo(
- kPayloadType, absl::nullopt, 1234, fake_clock_.CurrentTime(), kPayload,
+ kPayloadType, std::nullopt, 1234, fake_clock_.CurrentTime(), kPayload,
sizeof(kPayload), video_header, TimeDelta::PlusInfinity(), {}));
rtc::ArrayView<const uint8_t> sent_payload =
diff --git a/modules/rtp_rtcp/source/rtp_sequence_number_map.cc b/modules/rtp_rtcp/source/rtp_sequence_number_map.cc
index 441429d..99824da 100644
--- a/modules/rtp_rtcp/source/rtp_sequence_number_map.cc
+++ b/modules/rtp_rtcp/source/rtp_sequence_number_map.cc
@@ -94,7 +94,7 @@
}
}
-absl::optional<RtpSequenceNumberMap::Info> RtpSequenceNumberMap::Get(
+std::optional<RtpSequenceNumberMap::Info> RtpSequenceNumberMap::Get(
uint16_t sequence_number) const {
// To make the binary search easier to understand, we use the fact that
// adding a constant offset to all elements, as well as to the searched
@@ -105,7 +105,7 @@
// element to 0 would serve this purpose.
if (associations_.empty()) {
- return absl::nullopt;
+ return std::nullopt;
}
const uint16_t offset =
@@ -118,8 +118,8 @@
const auto elem = absl::c_lower_bound(associations_, sequence_number, cmp);
return elem != associations_.end() && elem->sequence_number == sequence_number
- ? absl::optional<Info>(elem->info)
- : absl::nullopt;
+ ? std::optional<Info>(elem->info)
+ : std::nullopt;
}
size_t RtpSequenceNumberMap::AssociationCountForTesting() const {
diff --git a/modules/rtp_rtcp/source/rtp_sequence_number_map.h b/modules/rtp_rtcp/source/rtp_sequence_number_map.h
index 8a036c2..d2bbd18 100644
--- a/modules/rtp_rtcp/source/rtp_sequence_number_map.h
+++ b/modules/rtp_rtcp/source/rtp_sequence_number_map.h
@@ -14,8 +14,7 @@
#include <cstddef>
#include <cstdint>
#include <deque>
-
-#include "absl/types/optional.h"
+#include <optional>
namespace webrtc {
@@ -57,7 +56,7 @@
size_t packet_count,
uint32_t timestamp);
- absl::optional<Info> Get(uint16_t sequence_number) const;
+ std::optional<Info> Get(uint16_t sequence_number) const;
size_t AssociationCountForTesting() const;
diff --git a/modules/rtp_rtcp/source/rtp_sequence_number_map_unittest.cc b/modules/rtp_rtcp/source/rtp_sequence_number_map_unittest.cc
index 78c9e4a..e90f45b 100644
--- a/modules/rtp_rtcp/source/rtp_sequence_number_map_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_sequence_number_map_unittest.cc
@@ -14,11 +14,11 @@
#include <iterator>
#include <limits>
#include <memory>
+#include <optional>
#include <tuple>
#include <vector>
#include "absl/memory/memory.h"
-#include "absl/types/optional.h"
#include "rtc_base/checks.h"
#include "rtc_base/numerics/sequence_number_util.h"
#include "rtc_base/random.h"
diff --git a/modules/rtp_rtcp/source/rtp_video_header.cc b/modules/rtp_rtcp/source/rtp_video_header.cc
index c4ab609..c37cc65 100644
--- a/modules/rtp_rtcp/source/rtp_video_header.cc
+++ b/modules/rtp_rtcp/source/rtp_video_header.cc
@@ -76,7 +76,7 @@
rotation = metadata.GetRotation();
content_type = metadata.GetContentType();
if (!metadata.GetFrameId().has_value()) {
- generic = absl::nullopt;
+ generic = std::nullopt;
} else {
generic.emplace();
generic->frame_id = metadata.GetFrameId().value();
diff --git a/modules/rtp_rtcp/source/rtp_video_header.h b/modules/rtp_rtcp/source/rtp_video_header.h
index 3100d4d..20ad4b2 100644
--- a/modules/rtp_rtcp/source/rtp_video_header.h
+++ b/modules/rtp_rtcp/source/rtp_video_header.h
@@ -12,9 +12,9 @@
#include <bitset>
#include <cstdint>
+#include <optional>
#include "absl/container/inlined_vector.h"
-#include "absl/types/optional.h"
#include "absl/types/variant.h"
#include "api/rtp_headers.h"
#include "api/transport/rtp/dependency_descriptor.h"
@@ -69,7 +69,7 @@
VideoFrameMetadata GetAsMetadata() const;
void SetFromMetadata(const VideoFrameMetadata& metadata);
- absl::optional<GenericDescriptorInfo> generic;
+ std::optional<GenericDescriptorInfo> generic;
VideoFrameType frame_type = VideoFrameType::kEmptyFrame;
uint16_t width = 0;
@@ -82,18 +82,18 @@
uint8_t simulcastIdx = 0;
VideoCodecType codec = VideoCodecType::kVideoCodecGeneric;
- absl::optional<VideoPlayoutDelay> playout_delay;
+ std::optional<VideoPlayoutDelay> playout_delay;
VideoSendTiming video_timing;
- absl::optional<ColorSpace> color_space;
+ std::optional<ColorSpace> color_space;
// This field is meant for media quality testing purpose only. When enabled it
// carries the webrtc::VideoFrame id field from the sender to the receiver.
- absl::optional<uint16_t> video_frame_tracking_id;
+ std::optional<uint16_t> video_frame_tracking_id;
RTPVideoTypeHeader video_type_header;
// When provided, is sent as is as an RTP header extension according to
// http://www.webrtc.org/experiments/rtp-hdrext/abs-capture-time.
// Otherwise, it is derived from other relevant information.
- absl::optional<AbsoluteCaptureTime> absolute_capture_time;
+ std::optional<AbsoluteCaptureTime> absolute_capture_time;
};
} // namespace webrtc
diff --git a/modules/rtp_rtcp/source/rtp_video_header_unittest.cc b/modules/rtp_rtcp/source/rtp_video_header_unittest.cc
index 335fa1a..d87a2b5 100644
--- a/modules/rtp_rtcp/source/rtp_video_header_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_video_header_unittest.cc
@@ -117,7 +117,7 @@
TEST(RTPVideoHeaderTest, FrameId_FromMetadataWhenFrameIdIsMissing) {
VideoFrameMetadata metadata;
- metadata.SetFrameId(absl::nullopt);
+ metadata.SetFrameId(std::nullopt);
RTPVideoHeader video_header = RTPVideoHeader::FromMetadata(metadata);
EXPECT_FALSE(video_header.generic.has_value());
}
diff --git a/modules/rtp_rtcp/source/rtp_video_stream_receiver_frame_transformer_delegate_unittest.cc b/modules/rtp_rtcp/source/rtp_video_stream_receiver_frame_transformer_delegate_unittest.cc
index 192e239..6fd467e 100644
--- a/modules/rtp_rtcp/source/rtp_video_stream_receiver_frame_transformer_delegate_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_video_stream_receiver_frame_transformer_delegate_unittest.cc
@@ -50,7 +50,7 @@
/*last_packet_received_time=*/5, /*rtp_timestamp=*/6, /*ntp_time_ms=*/7,
VideoSendTiming(), /*payload_type=*/8, video_header.codec,
kVideoRotation_0, VideoContentType::UNSPECIFIED, video_header,
- absl::nullopt, RtpPacketInfos({packet_info}),
+ std::nullopt, RtpPacketInfos({packet_info}),
EncodedImageBuffer::Create(0));
}
@@ -222,8 +222,8 @@
// Checks that the recieved RTPFrameObject has the new metadata.
EXPECT_CALL(receiver, ManageFrame)
.WillOnce([&](std::unique_ptr<RtpFrameObject> frame) {
- const absl::optional<RTPVideoHeader::GenericDescriptorInfo>&
- descriptor = frame->GetRtpVideoHeader().generic;
+ const std::optional<RTPVideoHeader::GenericDescriptorInfo>& descriptor =
+ frame->GetRtpVideoHeader().generic;
if (!descriptor.has_value()) {
ADD_FAILURE() << "GenericDescriptorInfo in RTPVideoHeader doesn't "
"have a value.";
diff --git a/modules/rtp_rtcp/source/source_tracker.h b/modules/rtp_rtcp/source/source_tracker.h
index 30a5b8a..9d39599 100644
--- a/modules/rtp_rtcp/source/source_tracker.h
+++ b/modules/rtp_rtcp/source/source_tracker.h
@@ -13,11 +13,11 @@
#include <cstdint>
#include <list>
+#include <optional>
#include <unordered_map>
#include <utility>
#include <vector>
-#include "absl/types/optional.h"
#include "api/rtp_packet_infos.h"
#include "api/task_queue/pending_task_safety_flag.h"
#include "api/task_queue/task_queue_base.h"
@@ -91,19 +91,19 @@
// the most recent packet used to assemble the frame associated with
// `timestamp`. May be absent. Only relevant for audio receivers. See the
// specs for `RTCRtpContributingSource` for more info.
- absl::optional<uint8_t> audio_level;
+ std::optional<uint8_t> audio_level;
// Absolute capture time header extension received or interpolated from the
// most recent packet used to assemble the frame. For more info see
// https://webrtc.org/experiments/rtp-hdrext/abs-capture-time/
- absl::optional<AbsoluteCaptureTime> absolute_capture_time;
+ std::optional<AbsoluteCaptureTime> absolute_capture_time;
// Clock offset between the local clock and the capturer's clock.
// Do not confuse with `AbsoluteCaptureTime::estimated_capture_clock_offset`
// which instead represents the clock offset between a remote sender and the
// capturer. The following holds:
// Capture's NTP Clock = Local NTP Clock + Local-Capture Clock Offset
- absl::optional<TimeDelta> local_capture_clock_offset;
+ std::optional<TimeDelta> local_capture_clock_offset;
// RTP timestamp of the most recent packet used to assemble the frame
// associated with `timestamp`.
diff --git a/modules/rtp_rtcp/source/source_tracker_unittest.cc b/modules/rtp_rtcp/source/source_tracker_unittest.cc
index e14a389..32ac711 100644
--- a/modules/rtp_rtcp/source/source_tracker_unittest.cc
+++ b/modules/rtp_rtcp/source/source_tracker_unittest.cc
@@ -12,13 +12,13 @@
#include <algorithm>
#include <list>
+#include <optional>
#include <random>
#include <set>
#include <tuple>
#include <utility>
#include <vector>
-#include "absl/types/optional.h"
#include "api/rtp_headers.h"
#include "api/rtp_packet_info.h"
#include "api/rtp_packet_infos.h"
@@ -167,9 +167,9 @@
return std::uniform_int_distribution<uint32_t>()(generator_);
}
- absl::optional<uint8_t> GenerateAudioLevel() {
+ std::optional<uint8_t> GenerateAudioLevel() {
if (std::bernoulli_distribution(0.25)(generator_)) {
- return absl::nullopt;
+ return std::nullopt;
}
// Workaround for std::uniform_int_distribution<uint8_t> not being allowed.
@@ -177,9 +177,9 @@
std::uniform_int_distribution<uint16_t>()(generator_));
}
- absl::optional<AbsoluteCaptureTime> GenerateAbsoluteCaptureTime() {
+ std::optional<AbsoluteCaptureTime> GenerateAbsoluteCaptureTime() {
if (std::bernoulli_distribution(0.25)(generator_)) {
- return absl::nullopt;
+ return std::nullopt;
}
AbsoluteCaptureTime value;
@@ -188,7 +188,7 @@
std::uniform_int_distribution<uint64_t>()(generator_);
if (std::bernoulli_distribution(0.5)(generator_)) {
- value.estimated_capture_clock_offset = absl::nullopt;
+ value.estimated_capture_clock_offset = std::nullopt;
} else {
value.estimated_capture_clock_offset =
std::uniform_int_distribution<int64_t>()(generator_);
@@ -197,9 +197,9 @@
return value;
}
- absl::optional<TimeDelta> GenerateLocalCaptureClockOffset() {
+ std::optional<TimeDelta> GenerateLocalCaptureClockOffset() {
if (std::bernoulli_distribution(0.5)(generator_)) {
- return absl::nullopt;
+ return std::nullopt;
}
return TimeDelta::Millis(
UQ32x32ToInt64Ms(std::uniform_int_distribution<int64_t>()(generator_)));
@@ -263,12 +263,12 @@
constexpr uint32_t kCsrcs2 = 22;
constexpr uint32_t kRtpTimestamp0 = 40;
constexpr uint32_t kRtpTimestamp1 = 50;
- constexpr absl::optional<uint8_t> kAudioLevel0 = 50;
- constexpr absl::optional<uint8_t> kAudioLevel1 = 20;
- constexpr absl::optional<AbsoluteCaptureTime> kAbsoluteCaptureTime =
+ constexpr std::optional<uint8_t> kAudioLevel0 = 50;
+ constexpr std::optional<uint8_t> kAudioLevel1 = 20;
+ constexpr std::optional<AbsoluteCaptureTime> kAbsoluteCaptureTime =
AbsoluteCaptureTime{/*absolute_capture_timestamp=*/12,
- /*estimated_capture_clock_offset=*/absl::nullopt};
- constexpr absl::optional<TimeDelta> kLocalCaptureClockOffset = absl::nullopt;
+ /*estimated_capture_clock_offset=*/std::nullopt};
+ constexpr std::optional<TimeDelta> kLocalCaptureClockOffset = std::nullopt;
constexpr Timestamp kReceiveTime0 = Timestamp::Millis(60);
constexpr Timestamp kReceiveTime1 = Timestamp::Millis(70);
@@ -318,13 +318,13 @@
constexpr uint32_t kRtpTimestamp0 = 40;
constexpr uint32_t kRtpTimestamp1 = 45;
constexpr uint32_t kRtpTimestamp2 = 50;
- constexpr absl::optional<uint8_t> kAudioLevel0 = 50;
- constexpr absl::optional<uint8_t> kAudioLevel1 = 20;
- constexpr absl::optional<uint8_t> kAudioLevel2 = 10;
- constexpr absl::optional<AbsoluteCaptureTime> kAbsoluteCaptureTime =
+ constexpr std::optional<uint8_t> kAudioLevel0 = 50;
+ constexpr std::optional<uint8_t> kAudioLevel1 = 20;
+ constexpr std::optional<uint8_t> kAudioLevel2 = 10;
+ constexpr std::optional<AbsoluteCaptureTime> kAbsoluteCaptureTime =
AbsoluteCaptureTime{/*absolute_capture_timestamp=*/12,
- /*estimated_capture_clock_offset=*/absl::nullopt};
- constexpr absl::optional<TimeDelta> kLocalCaptureClockOffset = absl::nullopt;
+ /*estimated_capture_clock_offset=*/std::nullopt};
+ constexpr std::optional<TimeDelta> kLocalCaptureClockOffset = std::nullopt;
constexpr Timestamp kReceiveTime0 = Timestamp::Millis(60);
constexpr Timestamp kReceiveTime1 = Timestamp::Millis(70);
constexpr Timestamp kReceiveTime2 = Timestamp::Millis(80);
@@ -382,20 +382,20 @@
constexpr uint32_t kRtpTimestamp0 = 40;
constexpr uint32_t kRtpTimestamp1 = 41;
constexpr uint32_t kRtpTimestamp2 = 42;
- constexpr absl::optional<uint8_t> kAudioLevel0 = 50;
- constexpr absl::optional<uint8_t> kAudioLevel1 = absl::nullopt;
- constexpr absl::optional<uint8_t> kAudioLevel2 = 10;
- constexpr absl::optional<AbsoluteCaptureTime> kAbsoluteCaptureTime0 =
+ constexpr std::optional<uint8_t> kAudioLevel0 = 50;
+ constexpr std::optional<uint8_t> kAudioLevel1 = std::nullopt;
+ constexpr std::optional<uint8_t> kAudioLevel2 = 10;
+ constexpr std::optional<AbsoluteCaptureTime> kAbsoluteCaptureTime0 =
AbsoluteCaptureTime{12, 34};
- constexpr absl::optional<AbsoluteCaptureTime> kAbsoluteCaptureTime1 =
+ constexpr std::optional<AbsoluteCaptureTime> kAbsoluteCaptureTime1 =
AbsoluteCaptureTime{56, 78};
- constexpr absl::optional<AbsoluteCaptureTime> kAbsoluteCaptureTime2 =
+ constexpr std::optional<AbsoluteCaptureTime> kAbsoluteCaptureTime2 =
AbsoluteCaptureTime{89, 90};
- constexpr absl::optional<TimeDelta> kLocalCaptureClockOffset0 =
+ constexpr std::optional<TimeDelta> kLocalCaptureClockOffset0 =
TimeDelta::Millis(123);
- constexpr absl::optional<TimeDelta> kLocalCaptureClockOffset1 =
+ constexpr std::optional<TimeDelta> kLocalCaptureClockOffset1 =
TimeDelta::Millis(456);
- constexpr absl::optional<TimeDelta> kLocalCaptureClockOffset2 =
+ constexpr std::optional<TimeDelta> kLocalCaptureClockOffset2 =
TimeDelta::Millis(789);
constexpr Timestamp kReceiveTime0 = Timestamp::Millis(60);
constexpr Timestamp kReceiveTime1 = Timestamp::Millis(61);
@@ -487,15 +487,15 @@
constexpr uint32_t kCsrcs2 = 22;
constexpr uint32_t kRtpTimestamp0 = 40;
constexpr uint32_t kRtpTimestamp1 = 41;
- constexpr absl::optional<uint8_t> kAudioLevel0 = 50;
- constexpr absl::optional<uint8_t> kAudioLevel1 = absl::nullopt;
- constexpr absl::optional<AbsoluteCaptureTime> kAbsoluteCaptureTime0 =
+ constexpr std::optional<uint8_t> kAudioLevel0 = 50;
+ constexpr std::optional<uint8_t> kAudioLevel1 = std::nullopt;
+ constexpr std::optional<AbsoluteCaptureTime> kAbsoluteCaptureTime0 =
AbsoluteCaptureTime{12, 34};
- constexpr absl::optional<AbsoluteCaptureTime> kAbsoluteCaptureTime1 =
+ constexpr std::optional<AbsoluteCaptureTime> kAbsoluteCaptureTime1 =
AbsoluteCaptureTime{56, 78};
- constexpr absl::optional<TimeDelta> kLocalCaptureClockOffset0 =
+ constexpr std::optional<TimeDelta> kLocalCaptureClockOffset0 =
TimeDelta::Millis(123);
- constexpr absl::optional<TimeDelta> kLocalCaptureClockOffset1 =
+ constexpr std::optional<TimeDelta> kLocalCaptureClockOffset1 =
TimeDelta::Millis(456);
constexpr Timestamp kReceiveTime0 = Timestamp::Millis(60);
constexpr Timestamp kReceiveTime1 = Timestamp::Millis(61);
diff --git a/modules/rtp_rtcp/source/ulpfec_generator.h b/modules/rtp_rtcp/source/ulpfec_generator.h
index 64d5034..bf82878 100644
--- a/modules/rtp_rtcp/source/ulpfec_generator.h
+++ b/modules/rtp_rtcp/source/ulpfec_generator.h
@@ -42,7 +42,7 @@
FecType GetFecType() const override {
return VideoFecGenerator::FecType::kUlpFec;
}
- absl::optional<uint32_t> FecSsrc() override { return absl::nullopt; }
+ std::optional<uint32_t> FecSsrc() override { return std::nullopt; }
void SetProtectionParameters(const FecProtectionParams& delta_params,
const FecProtectionParams& key_params) override;
@@ -60,7 +60,7 @@
// Current rate of FEC packets generated, including all RTP-level headers.
DataRate CurrentFecRate() const override;
- absl::optional<RtpState> GetRtpState() override { return absl::nullopt; }
+ std::optional<RtpState> GetRtpState() override { return std::nullopt; }
// Currently used protection params.
const FecProtectionParams& CurrentParams() const;
@@ -108,7 +108,7 @@
RTC_GUARDED_BY(race_checker_);
ForwardErrorCorrection::PacketList media_packets_
RTC_GUARDED_BY(race_checker_);
- absl::optional<RtpPacketToSend> last_media_packet_
+ std::optional<RtpPacketToSend> last_media_packet_
RTC_GUARDED_BY(race_checker_);
std::list<ForwardErrorCorrection::Packet*> generated_fec_packets_
RTC_GUARDED_BY(race_checker_);
@@ -118,7 +118,7 @@
bool media_contains_keyframe_ RTC_GUARDED_BY(race_checker_);
mutable Mutex mutex_;
- absl::optional<Params> pending_params_ RTC_GUARDED_BY(mutex_);
+ std::optional<Params> pending_params_ RTC_GUARDED_BY(mutex_);
BitrateTracker fec_bitrate_ RTC_GUARDED_BY(mutex_);
};
diff --git a/modules/rtp_rtcp/source/video_fec_generator.h b/modules/rtp_rtcp/source/video_fec_generator.h
index 38e4103..aff0ae9 100644
--- a/modules/rtp_rtcp/source/video_fec_generator.h
+++ b/modules/rtp_rtcp/source/video_fec_generator.h
@@ -28,7 +28,7 @@
enum class FecType { kFlexFec, kUlpFec };
virtual FecType GetFecType() const = 0;
// Returns the SSRC used for FEC packets (i.e. FlexFec SSRC).
- virtual absl::optional<uint32_t> FecSsrc() = 0;
+ virtual std::optional<uint32_t> FecSsrc() = 0;
// Returns the overhead, in bytes per packet, for FEC (and possibly RED).
virtual size_t MaxPacketOverhead() const = 0;
// Current rate of FEC packets generated, including all RTP-level headers.
@@ -47,7 +47,7 @@
virtual std::vector<std::unique_ptr<RtpPacketToSend>> GetFecPackets() = 0;
// Only called on the VideoSendStream queue, after operation has shut down,
// and only populated if there is an RtpState (e.g. FlexFec).
- virtual absl::optional<RtpState> GetRtpState() = 0;
+ virtual std::optional<RtpState> GetRtpState() = 0;
};
} // namespace webrtc
diff --git a/modules/rtp_rtcp/source/video_rtp_depacketizer.h b/modules/rtp_rtcp/source/video_rtp_depacketizer.h
index 2266120..adf6dc6 100644
--- a/modules/rtp_rtcp/source/video_rtp_depacketizer.h
+++ b/modules/rtp_rtcp/source/video_rtp_depacketizer.h
@@ -13,7 +13,8 @@
#include <stdint.h>
-#include "absl/types/optional.h"
+#include <optional>
+
#include "api/array_view.h"
#include "api/scoped_refptr.h"
#include "api/video/encoded_image.h"
@@ -30,7 +31,7 @@
};
virtual ~VideoRtpDepacketizer() = default;
- virtual absl::optional<ParsedRtpPayload> Parse(
+ virtual std::optional<ParsedRtpPayload> Parse(
rtc::CopyOnWriteBuffer rtp_payload) = 0;
virtual rtc::scoped_refptr<EncodedImageBuffer> AssembleFrame(
rtc::ArrayView<const rtc::ArrayView<const uint8_t>> rtp_payloads);
diff --git a/modules/rtp_rtcp/source/video_rtp_depacketizer_av1.cc b/modules/rtp_rtcp/source/video_rtp_depacketizer_av1.cc
index 30bbbc5..6f70297 100644
--- a/modules/rtp_rtcp/source/video_rtp_depacketizer_av1.cc
+++ b/modules/rtp_rtcp/source/video_rtp_depacketizer_av1.cc
@@ -355,19 +355,19 @@
return bitstream;
}
-absl::optional<VideoRtpDepacketizer::ParsedRtpPayload>
+std::optional<VideoRtpDepacketizer::ParsedRtpPayload>
VideoRtpDepacketizerAv1::Parse(rtc::CopyOnWriteBuffer rtp_payload) {
if (rtp_payload.size() == 0) {
RTC_DLOG(LS_ERROR) << "Empty rtp payload.";
- return absl::nullopt;
+ return std::nullopt;
}
uint8_t aggregation_header = rtp_payload.cdata()[0];
if (RtpStartsNewCodedVideoSequence(aggregation_header) &&
RtpStartsWithFragment(aggregation_header)) {
// new coded video sequence can't start from an OBU fragment.
- return absl::nullopt;
+ return std::nullopt;
}
- absl::optional<ParsedRtpPayload> parsed(absl::in_place);
+ std::optional<ParsedRtpPayload> parsed(absl::in_place);
// To assemble frame, all of the rtp payload is required, including
// aggregation header.
diff --git a/modules/rtp_rtcp/source/video_rtp_depacketizer_av1.h b/modules/rtp_rtcp/source/video_rtp_depacketizer_av1.h
index ac8c7e6..ed998d6 100644
--- a/modules/rtp_rtcp/source/video_rtp_depacketizer_av1.h
+++ b/modules/rtp_rtcp/source/video_rtp_depacketizer_av1.h
@@ -14,7 +14,8 @@
#include <stddef.h>
#include <stdint.h>
-#include "absl/types/optional.h"
+#include <optional>
+
#include "api/array_view.h"
#include "api/scoped_refptr.h"
#include "api/video/encoded_image.h"
@@ -34,7 +35,7 @@
rtc::ArrayView<const rtc::ArrayView<const uint8_t>> rtp_payloads)
override;
- absl::optional<ParsedRtpPayload> Parse(
+ std::optional<ParsedRtpPayload> Parse(
rtc::CopyOnWriteBuffer rtp_payload) override;
};
diff --git a/modules/rtp_rtcp/source/video_rtp_depacketizer_av1_unittest.cc b/modules/rtp_rtcp/source/video_rtp_depacketizer_av1_unittest.cc
index e9ad1a1..94cbbe4 100644
--- a/modules/rtp_rtcp/source/video_rtp_depacketizer_av1_unittest.cc
+++ b/modules/rtp_rtcp/source/video_rtp_depacketizer_av1_unittest.cc
@@ -30,7 +30,7 @@
const uint8_t packet[] = {(uint8_t{1} << 7) | kObuCountOne, 1, 2, 3, 4};
rtc::CopyOnWriteBuffer rtp_payload(packet);
VideoRtpDepacketizerAv1 depacketizer;
- absl::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed =
+ std::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed =
depacketizer.Parse(rtp_payload);
ASSERT_TRUE(parsed);
EXPECT_EQ(parsed->video_payload.size(), sizeof(packet));
@@ -44,7 +44,7 @@
kObuHeaderFrame}; // Value doesn't matter since it is a
// continuation of the OBU from previous packet.
VideoRtpDepacketizerAv1 depacketizer;
- absl::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed =
+ std::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed =
depacketizer.Parse(rtc::CopyOnWriteBuffer(packet));
ASSERT_TRUE(parsed);
EXPECT_FALSE(parsed->video_header.is_first_packet_in_frame);
@@ -54,7 +54,7 @@
ParseTreatsNoContinuationFlagAsBeginningOfFrame) {
const uint8_t packet[] = {(uint8_t{0} << 7) | kObuCountOne, kObuHeaderFrame};
VideoRtpDepacketizerAv1 depacketizer;
- absl::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed =
+ std::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed =
depacketizer.Parse(rtc::CopyOnWriteBuffer(packet));
ASSERT_TRUE(parsed);
EXPECT_TRUE(parsed->video_header.is_first_packet_in_frame);
@@ -64,7 +64,7 @@
const uint8_t packet[] = {(uint8_t{1} << 6) | kObuCountOne, kObuHeaderFrame};
rtc::CopyOnWriteBuffer rtp_payload(packet);
VideoRtpDepacketizerAv1 depacketizer;
- absl::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed =
+ std::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed =
depacketizer.Parse(rtp_payload);
ASSERT_TRUE(parsed);
EXPECT_FALSE(parsed->video_header.is_last_packet_in_frame);
@@ -73,7 +73,7 @@
TEST(VideoRtpDepacketizerAv1Test, ParseTreatsNoWillContinueFlagAsEndOfFrame) {
const uint8_t packet[] = {(uint8_t{0} << 6) | kObuCountOne, kObuHeaderFrame};
VideoRtpDepacketizerAv1 depacketizer;
- absl::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed =
+ std::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed =
depacketizer.Parse(rtc::CopyOnWriteBuffer(packet));
ASSERT_TRUE(parsed);
EXPECT_TRUE(parsed->video_header.is_last_packet_in_frame);
@@ -84,7 +84,7 @@
const uint8_t packet[] = {(uint8_t{1} << 3) | kObuCountOne,
kObuHeaderSequenceHeader};
VideoRtpDepacketizerAv1 depacketizer;
- absl::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed =
+ std::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed =
depacketizer.Parse(rtc::CopyOnWriteBuffer(packet));
ASSERT_TRUE(parsed);
EXPECT_TRUE(parsed->video_header.is_first_packet_in_frame);
@@ -97,7 +97,7 @@
const uint8_t packet[] = {(uint8_t{0} << 3) | kObuCountOne,
kObuHeaderSequenceHeader};
VideoRtpDepacketizerAv1 depacketizer;
- absl::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed =
+ std::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed =
depacketizer.Parse(rtc::CopyOnWriteBuffer(packet));
ASSERT_TRUE(parsed);
EXPECT_TRUE(parsed->video_header.is_first_packet_in_frame);
diff --git a/modules/rtp_rtcp/source/video_rtp_depacketizer_generic.cc b/modules/rtp_rtcp/source/video_rtp_depacketizer_generic.cc
index 6010771..ec8993b 100644
--- a/modules/rtp_rtcp/source/video_rtp_depacketizer_generic.cc
+++ b/modules/rtp_rtcp/source/video_rtp_depacketizer_generic.cc
@@ -13,9 +13,9 @@
#include <stddef.h>
#include <stdint.h>
+#include <optional>
#include <utility>
-#include "absl/types/optional.h"
#include "modules/rtp_rtcp/source/rtp_video_header.h"
#include "modules/rtp_rtcp/source/video_rtp_depacketizer.h"
#include "rtc_base/copy_on_write_buffer.h"
@@ -33,13 +33,13 @@
constexpr size_t kExtendedHeaderLength = 2;
} // namespace
-absl::optional<VideoRtpDepacketizer::ParsedRtpPayload>
+std::optional<VideoRtpDepacketizer::ParsedRtpPayload>
VideoRtpDepacketizerGeneric::Parse(rtc::CopyOnWriteBuffer rtp_payload) {
if (rtp_payload.size() == 0) {
RTC_LOG(LS_WARNING) << "Empty payload.";
- return absl::nullopt;
+ return std::nullopt;
}
- absl::optional<ParsedRtpPayload> parsed(absl::in_place);
+ std::optional<ParsedRtpPayload> parsed(absl::in_place);
const uint8_t* payload_data = rtp_payload.cdata();
uint8_t generic_header = payload_data[0];
@@ -57,7 +57,7 @@
if (generic_header & kExtendedHeaderBit) {
if (rtp_payload.size() < offset + kExtendedHeaderLength) {
RTC_LOG(LS_WARNING) << "Too short payload for generic header.";
- return absl::nullopt;
+ return std::nullopt;
}
parsed->video_header.video_type_header
.emplace<RTPVideoHeaderLegacyGeneric>()
diff --git a/modules/rtp_rtcp/source/video_rtp_depacketizer_generic.h b/modules/rtp_rtcp/source/video_rtp_depacketizer_generic.h
index 27056da..a85bd43 100644
--- a/modules/rtp_rtcp/source/video_rtp_depacketizer_generic.h
+++ b/modules/rtp_rtcp/source/video_rtp_depacketizer_generic.h
@@ -11,7 +11,8 @@
#ifndef MODULES_RTP_RTCP_SOURCE_VIDEO_RTP_DEPACKETIZER_GENERIC_H_
#define MODULES_RTP_RTCP_SOURCE_VIDEO_RTP_DEPACKETIZER_GENERIC_H_
-#include "absl/types/optional.h"
+#include <optional>
+
#include "modules/rtp_rtcp/source/video_rtp_depacketizer.h"
#include "rtc_base/copy_on_write_buffer.h"
@@ -21,7 +22,7 @@
public:
~VideoRtpDepacketizerGeneric() override = default;
- absl::optional<ParsedRtpPayload> Parse(
+ std::optional<ParsedRtpPayload> Parse(
rtc::CopyOnWriteBuffer rtp_payload) override;
};
diff --git a/modules/rtp_rtcp/source/video_rtp_depacketizer_generic_unittest.cc b/modules/rtp_rtcp/source/video_rtp_depacketizer_generic_unittest.cc
index 860ddab..7f4b83a 100644
--- a/modules/rtp_rtcp/source/video_rtp_depacketizer_generic_unittest.cc
+++ b/modules/rtp_rtcp/source/video_rtp_depacketizer_generic_unittest.cc
@@ -12,7 +12,8 @@
#include <stdint.h>
-#include "absl/types/optional.h"
+#include <optional>
+
#include "rtc_base/copy_on_write_buffer.h"
#include "test/gmock.h"
#include "test/gtest.h"
@@ -28,11 +29,11 @@
rtc::CopyOnWriteBuffer rtp_payload(kPayload);
VideoRtpDepacketizerGeneric depacketizer;
- absl::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed =
+ std::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed =
depacketizer.Parse(rtp_payload);
ASSERT_TRUE(parsed);
- EXPECT_EQ(parsed->video_header.generic, absl::nullopt);
+ EXPECT_EQ(parsed->video_header.generic, std::nullopt);
EXPECT_THAT(parsed->video_payload, SizeIs(kRtpPayloadSize - 1));
}
@@ -42,7 +43,7 @@
rtc::CopyOnWriteBuffer rtp_payload(kPayload);
VideoRtpDepacketizerGeneric depacketizer;
- absl::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed =
+ std::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed =
depacketizer.Parse(rtp_payload);
ASSERT_TRUE(parsed);
@@ -58,7 +59,7 @@
rtc::CopyOnWriteBuffer rtp_payload(kPayload);
VideoRtpDepacketizerGeneric depacketizer;
- absl::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed =
+ std::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed =
depacketizer.Parse(rtp_payload);
ASSERT_TRUE(parsed);
diff --git a/modules/rtp_rtcp/source/video_rtp_depacketizer_h264.cc b/modules/rtp_rtcp/source/video_rtp_depacketizer_h264.cc
index 084b6ac..e8fd911 100644
--- a/modules/rtp_rtcp/source/video_rtp_depacketizer_h264.cc
+++ b/modules/rtp_rtcp/source/video_rtp_depacketizer_h264.cc
@@ -12,10 +12,10 @@
#include <cstddef>
#include <cstdint>
+#include <optional>
#include <utility>
#include <vector>
-#include "absl/types/optional.h"
#include "absl/types/variant.h"
#include "common_video/h264/h264_common.h"
#include "common_video/h264/pps_parser.h"
@@ -59,10 +59,10 @@
return true;
}
-absl::optional<VideoRtpDepacketizer::ParsedRtpPayload> ProcessStapAOrSingleNalu(
+std::optional<VideoRtpDepacketizer::ParsedRtpPayload> ProcessStapAOrSingleNalu(
rtc::CopyOnWriteBuffer rtp_payload) {
const uint8_t* const payload_data = rtp_payload.cdata();
- absl::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed_payload(
+ std::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed_payload(
absl::in_place);
bool modified_buffer = false;
parsed_payload->video_payload = rtp_payload;
@@ -82,12 +82,12 @@
// Skip the StapA header (StapA NAL type + length).
if (rtp_payload.size() <= kStapAHeaderSize) {
RTC_LOG(LS_ERROR) << "StapA header truncated.";
- return absl::nullopt;
+ return std::nullopt;
}
if (!ParseStapAStartOffsets(nalu_start, nalu_length, &nalu_start_offsets)) {
RTC_LOG(LS_ERROR) << "StapA packet with incorrect NALU packet lengths.";
- return absl::nullopt;
+ return std::nullopt;
}
h264_header.packetization_type = kH264StapA;
@@ -108,7 +108,7 @@
size_t end_offset = nalu_start_offsets[i + 1] - kLengthFieldSize;
if (end_offset - start_offset < H264::kNaluTypeSize) {
RTC_LOG(LS_ERROR) << "STAP-A packet too short";
- return absl::nullopt;
+ return std::nullopt;
}
NaluInfo nalu;
@@ -128,7 +128,7 @@
if (start_offset)
output_buffer.AppendData(payload_data, start_offset);
- absl::optional<SpsParser::SpsState> sps;
+ std::optional<SpsParser::SpsState> sps;
SpsVuiRewriter::ParseResult result = SpsVuiRewriter::ParseAndRewriteSps(
nalu_data, &sps, nullptr, &output_buffer,
@@ -136,7 +136,7 @@
switch (result) {
case SpsVuiRewriter::ParseResult::kFailure:
RTC_LOG(LS_WARNING) << "Failed to parse SPS NAL unit.";
- return absl::nullopt;
+ return std::nullopt;
case SpsVuiRewriter::ParseResult::kVuiRewritten:
if (modified_buffer) {
RTC_LOG(LS_WARNING)
@@ -186,7 +186,7 @@
} else {
RTC_LOG(LS_WARNING)
<< "Failed to parse PPS id and SPS id from PPS slice.";
- return absl::nullopt;
+ return std::nullopt;
}
break;
}
@@ -195,7 +195,7 @@
VideoFrameType::kVideoFrameKey;
[[fallthrough]];
case H264::NaluType::kSlice: {
- absl::optional<PpsParser::SliceHeader> slice_header =
+ std::optional<PpsParser::SliceHeader> slice_header =
PpsParser::ParseSliceHeader(nalu_data);
if (slice_header) {
nalu.pps_id = slice_header->pic_parameter_set_id;
@@ -205,7 +205,7 @@
} else {
RTC_LOG(LS_WARNING) << "Failed to parse PPS id from slice of type: "
<< static_cast<int>(nalu.type);
- return absl::nullopt;
+ return std::nullopt;
}
break;
}
@@ -219,7 +219,7 @@
case H264::NaluType::kStapA:
case H264::NaluType::kFuA:
RTC_LOG(LS_WARNING) << "Unexpected STAP-A or FU-A received.";
- return absl::nullopt;
+ return std::nullopt;
}
h264_header.nalus.push_back(nalu);
@@ -228,13 +228,13 @@
return parsed_payload;
}
-absl::optional<VideoRtpDepacketizer::ParsedRtpPayload> ParseFuaNalu(
+std::optional<VideoRtpDepacketizer::ParsedRtpPayload> ParseFuaNalu(
rtc::CopyOnWriteBuffer rtp_payload) {
if (rtp_payload.size() < kFuAHeaderSize) {
RTC_LOG(LS_ERROR) << "FU-A NAL units truncated.";
- return absl::nullopt;
+ return std::nullopt;
}
- absl::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed_payload(
+ std::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed_payload(
absl::in_place);
uint8_t fnri = rtp_payload.cdata()[0] & (kH264FBit | kH264NriMask);
uint8_t original_nal_type = rtp_payload.cdata()[1] & kH264TypeMask;
@@ -247,7 +247,7 @@
if (first_fragment) {
if (original_nal_type == H264::NaluType::kIdr ||
original_nal_type == H264::NaluType::kSlice) {
- absl::optional<PpsParser::SliceHeader> slice_header =
+ std::optional<PpsParser::SliceHeader> slice_header =
PpsParser::ParseSliceHeader(rtc::ArrayView<const uint8_t>(rtp_payload)
.subview(2 * kNalHeaderSize));
if (slice_header) {
@@ -293,11 +293,11 @@
} // namespace
-absl::optional<VideoRtpDepacketizer::ParsedRtpPayload>
+std::optional<VideoRtpDepacketizer::ParsedRtpPayload>
VideoRtpDepacketizerH264::Parse(rtc::CopyOnWriteBuffer rtp_payload) {
if (rtp_payload.size() == 0) {
RTC_LOG(LS_ERROR) << "Empty payload.";
- return absl::nullopt;
+ return std::nullopt;
}
uint8_t nal_type = rtp_payload.cdata()[0] & kH264TypeMask;
diff --git a/modules/rtp_rtcp/source/video_rtp_depacketizer_h264.h b/modules/rtp_rtcp/source/video_rtp_depacketizer_h264.h
index cbea860..0953a12 100644
--- a/modules/rtp_rtcp/source/video_rtp_depacketizer_h264.h
+++ b/modules/rtp_rtcp/source/video_rtp_depacketizer_h264.h
@@ -11,7 +11,8 @@
#ifndef MODULES_RTP_RTCP_SOURCE_VIDEO_RTP_DEPACKETIZER_H264_H_
#define MODULES_RTP_RTCP_SOURCE_VIDEO_RTP_DEPACKETIZER_H264_H_
-#include "absl/types/optional.h"
+#include <optional>
+
#include "modules/rtp_rtcp/source/video_rtp_depacketizer.h"
#include "rtc_base/copy_on_write_buffer.h"
@@ -20,7 +21,7 @@
public:
~VideoRtpDepacketizerH264() override = default;
- absl::optional<ParsedRtpPayload> Parse(
+ std::optional<ParsedRtpPayload> Parse(
rtc::CopyOnWriteBuffer rtp_payload) override;
};
} // namespace webrtc
diff --git a/modules/rtp_rtcp/source/video_rtp_depacketizer_h264_unittest.cc b/modules/rtp_rtcp/source/video_rtp_depacketizer_h264_unittest.cc
index d279d24..1e10ebc 100644
--- a/modules/rtp_rtcp/source/video_rtp_depacketizer_h264_unittest.cc
+++ b/modules/rtp_rtcp/source/video_rtp_depacketizer_h264_unittest.cc
@@ -11,9 +11,9 @@
#include "modules/rtp_rtcp/source/video_rtp_depacketizer_h264.h"
#include <cstdint>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "common_video/h264/h264_common.h"
#include "modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
@@ -56,7 +56,7 @@
rtc::CopyOnWriteBuffer rtp_payload(packet);
VideoRtpDepacketizerH264 depacketizer;
- absl::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed =
+ std::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed =
depacketizer.Parse(rtp_payload);
ASSERT_TRUE(parsed);
@@ -77,7 +77,7 @@
rtc::CopyOnWriteBuffer rtp_payload(packet);
VideoRtpDepacketizerH264 depacketizer;
- absl::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed =
+ std::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed =
depacketizer.Parse(rtp_payload);
ASSERT_TRUE(parsed);
@@ -112,7 +112,7 @@
rtc::CopyOnWriteBuffer rtp_payload(packet);
VideoRtpDepacketizerH264 depacketizer;
- absl::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed =
+ std::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed =
depacketizer.Parse(rtp_payload);
ASSERT_TRUE(parsed);
@@ -139,7 +139,7 @@
rtc::CopyOnWriteBuffer rtp_payload(packet);
VideoRtpDepacketizerH264 depacketizer;
- absl::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed =
+ std::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed =
depacketizer.Parse(rtp_payload);
ASSERT_TRUE(parsed);
@@ -260,7 +260,7 @@
rtc::CopyOnWriteBuffer rtp_payload(packet);
VideoRtpDepacketizerH264 depacketizer;
- absl::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed =
+ std::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed =
depacketizer.Parse(rtp_payload);
ASSERT_TRUE(parsed);
@@ -303,7 +303,7 @@
const uint8_t kExpected3[] = {0x03};
VideoRtpDepacketizerH264 depacketizer;
- absl::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed1 =
+ std::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed1 =
depacketizer.Parse(rtc::CopyOnWriteBuffer(packet1));
ASSERT_TRUE(parsed1);
// We expect that the first packet is one byte shorter since the FU-A header
@@ -476,7 +476,7 @@
};
VideoRtpDepacketizerH264 depacketizer;
- absl::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed =
+ std::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed =
depacketizer.Parse(rtc::CopyOnWriteBuffer(kPayload));
ASSERT_TRUE(parsed);
EXPECT_TRUE(parsed->video_header.is_first_packet_in_frame);
@@ -497,7 +497,7 @@
};
VideoRtpDepacketizerH264 depacketizer;
- absl::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed =
+ std::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed =
depacketizer.Parse(rtc::CopyOnWriteBuffer(kPayload));
ASSERT_TRUE(parsed);
EXPECT_FALSE(parsed->video_header.is_first_packet_in_frame);
diff --git a/modules/rtp_rtcp/source/video_rtp_depacketizer_h265.cc b/modules/rtp_rtcp/source/video_rtp_depacketizer_h265.cc
index b415ae7..d16e638 100644
--- a/modules/rtp_rtcp/source/video_rtp_depacketizer_h265.cc
+++ b/modules/rtp_rtcp/source/video_rtp_depacketizer_h265.cc
@@ -13,11 +13,11 @@
#include <cstddef>
#include <cstdint>
#include <memory>
+#include <optional>
#include <utility>
#include <vector>
#include "absl/base/macros.h"
-#include "absl/types/optional.h"
#include "absl/types/variant.h"
#include "api/video/video_codec_type.h"
#include "common_video/h264/h264_common.h"
@@ -61,16 +61,16 @@
// https://datatracker.ietf.org/doc/html/rfc7798#section-4.4.1
// Aggregation Packet (AP) strcture
// https://datatracker.ietf.org/doc/html/rfc7798#section-4.4.2
-absl::optional<VideoRtpDepacketizer::ParsedRtpPayload> ProcessApOrSingleNalu(
+std::optional<VideoRtpDepacketizer::ParsedRtpPayload> ProcessApOrSingleNalu(
rtc::CopyOnWriteBuffer rtp_payload) {
// Skip the single NALU header (payload header), aggregated packet case will
// be checked later.
if (rtp_payload.size() <= kH265PayloadHeaderSizeBytes) {
RTC_LOG(LS_ERROR) << "Single NALU header truncated.";
- return absl::nullopt;
+ return std::nullopt;
}
const uint8_t* const payload_data = rtp_payload.cdata();
- absl::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed_payload(
+ std::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed_payload(
absl::in_place);
parsed_payload->video_header.width = 0;
parsed_payload->video_header.height = 0;
@@ -86,13 +86,13 @@
// Skip the aggregated packet header (Aggregated packet NAL type + length).
if (rtp_payload.size() <= kH265ApHeaderSizeBytes) {
RTC_LOG(LS_ERROR) << "Aggregated packet header truncated.";
- return absl::nullopt;
+ return std::nullopt;
}
if (!ParseApStartOffsets(nalu_start, nalu_length, &nalu_start_offsets)) {
RTC_LOG(LS_ERROR)
<< "Aggregated packet with incorrect NALU packet lengths.";
- return absl::nullopt;
+ return std::nullopt;
}
nal_type = (payload_data[kH265ApHeaderSizeBytes] & kH265TypeMask) >> 1;
@@ -110,7 +110,7 @@
size_t end_offset = nalu_start_offsets[i + 1] - kH265LengthFieldSizeBytes;
if (end_offset - start_offset < kH265NalHeaderSizeBytes) {
RTC_LOG(LS_ERROR) << "Aggregated packet too short";
- return absl::nullopt;
+ return std::nullopt;
}
// Insert start code before each NALU in aggregated packet.
@@ -143,7 +143,7 @@
if (start_offset)
output_buffer->AppendData(payload_data, start_offset);
- absl::optional<H265SpsParser::SpsState> sps =
+ std::optional<H265SpsParser::SpsState> sps =
H265SpsParser::ParseSps(nalu_data);
if (sps) {
@@ -175,7 +175,7 @@
case H265::NaluType::kFu:
case H265::NaluType::kPaci:
RTC_LOG(LS_WARNING) << "Unexpected AP, FU or PACI received.";
- return absl::nullopt;
+ return std::nullopt;
}
}
parsed_payload->video_payload = video_payload;
@@ -184,13 +184,13 @@
// Fragmentation Unit (FU) structure:
// https://datatracker.ietf.org/doc/html/rfc7798#section-4.4.3
-absl::optional<VideoRtpDepacketizer::ParsedRtpPayload> ParseFuNalu(
+std::optional<VideoRtpDepacketizer::ParsedRtpPayload> ParseFuNalu(
rtc::CopyOnWriteBuffer rtp_payload) {
if (rtp_payload.size() < kH265FuHeaderSizeBytes + kH265NalHeaderSizeBytes) {
RTC_LOG(LS_ERROR) << "FU NAL units truncated.";
- return absl::nullopt;
+ return std::nullopt;
}
- absl::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed_payload(
+ std::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed_payload(
absl::in_place);
uint8_t f = rtp_payload.cdata()[0] & kH265FBit;
@@ -234,11 +234,11 @@
} // namespace
-absl::optional<VideoRtpDepacketizer::ParsedRtpPayload>
+std::optional<VideoRtpDepacketizer::ParsedRtpPayload>
VideoRtpDepacketizerH265::Parse(rtc::CopyOnWriteBuffer rtp_payload) {
if (rtp_payload.empty()) {
RTC_LOG(LS_ERROR) << "Empty payload.";
- return absl::nullopt;
+ return std::nullopt;
}
uint8_t nal_type = (rtp_payload.cdata()[0] & kH265TypeMask) >> 1;
@@ -249,7 +249,7 @@
} else if (nal_type == H265::NaluType::kPaci) {
// TODO(bugs.webrtc.org/13485): Implement PACI parse for H265
RTC_LOG(LS_ERROR) << "Not support type:" << nal_type;
- return absl::nullopt;
+ return std::nullopt;
} else {
// Single NAL unit packet or Aggregated packets (AP).
return ProcessApOrSingleNalu(std::move(rtp_payload));
diff --git a/modules/rtp_rtcp/source/video_rtp_depacketizer_h265.h b/modules/rtp_rtcp/source/video_rtp_depacketizer_h265.h
index ed5290d..27f3e53 100644
--- a/modules/rtp_rtcp/source/video_rtp_depacketizer_h265.h
+++ b/modules/rtp_rtcp/source/video_rtp_depacketizer_h265.h
@@ -11,7 +11,8 @@
#ifndef MODULES_RTP_RTCP_SOURCE_VIDEO_RTP_DEPACKETIZER_H265_H_
#define MODULES_RTP_RTCP_SOURCE_VIDEO_RTP_DEPACKETIZER_H265_H_
-#include "absl/types/optional.h"
+#include <optional>
+
#include "modules/rtp_rtcp/source/video_rtp_depacketizer.h"
#include "rtc_base/copy_on_write_buffer.h"
@@ -20,7 +21,7 @@
public:
~VideoRtpDepacketizerH265() override = default;
- absl::optional<ParsedRtpPayload> Parse(
+ std::optional<ParsedRtpPayload> Parse(
rtc::CopyOnWriteBuffer rtp_payload) override;
};
} // namespace webrtc
diff --git a/modules/rtp_rtcp/source/video_rtp_depacketizer_h265_unittest.cc b/modules/rtp_rtcp/source/video_rtp_depacketizer_h265_unittest.cc
index c94e0d5..8b7d080 100644
--- a/modules/rtp_rtcp/source/video_rtp_depacketizer_h265_unittest.cc
+++ b/modules/rtp_rtcp/source/video_rtp_depacketizer_h265_unittest.cc
@@ -11,9 +11,9 @@
#include "modules/rtp_rtcp/source/video_rtp_depacketizer_h265.h"
#include <cstdint>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "common_video/h265/h265_common.h"
#include "modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
@@ -40,7 +40,7 @@
rtc::CopyOnWriteBuffer rtp_payload(packet);
VideoRtpDepacketizerH265 depacketizer;
- absl::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed =
+ std::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed =
depacketizer.Parse(rtp_payload);
ASSERT_TRUE(parsed);
@@ -74,7 +74,7 @@
rtc::CopyOnWriteBuffer rtp_payload(packet);
VideoRtpDepacketizerH265 depacketizer;
- absl::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed =
+ std::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed =
depacketizer.Parse(rtp_payload);
ASSERT_TRUE(parsed);
@@ -92,7 +92,7 @@
rtc::CopyOnWriteBuffer rtp_payload(packet);
VideoRtpDepacketizerH265 depacketizer;
- absl::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed =
+ std::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed =
depacketizer.Parse(rtp_payload);
ASSERT_FALSE(parsed);
}
@@ -160,7 +160,7 @@
rtc::CopyOnWriteBuffer rtp_payload(packet);
VideoRtpDepacketizerH265 depacketizer;
- absl::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed =
+ std::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed =
depacketizer.Parse(rtp_payload);
ASSERT_TRUE(parsed);
@@ -219,7 +219,7 @@
rtc::CopyOnWriteBuffer rtp_payload(packet);
VideoRtpDepacketizerH265 depacketizer;
- absl::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed =
+ std::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed =
depacketizer.Parse(rtp_payload);
ASSERT_TRUE(parsed);
@@ -270,7 +270,7 @@
rtc::CopyOnWriteBuffer rtp_payload(packet);
VideoRtpDepacketizerH265 depacketizer;
- absl::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed =
+ std::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed =
depacketizer.Parse(rtp_payload);
ASSERT_TRUE(parsed);
@@ -311,7 +311,7 @@
const uint8_t kExpected3[] = {0x03};
VideoRtpDepacketizerH265 depacketizer;
- absl::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed1 =
+ std::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed1 =
depacketizer.Parse(rtc::CopyOnWriteBuffer(packet1));
ASSERT_TRUE(parsed1);
// We expect that the first packet is one byte shorter since the FU header
diff --git a/modules/rtp_rtcp/source/video_rtp_depacketizer_raw.cc b/modules/rtp_rtcp/source/video_rtp_depacketizer_raw.cc
index 81b4e4a..f4f4a79 100644
--- a/modules/rtp_rtcp/source/video_rtp_depacketizer_raw.cc
+++ b/modules/rtp_rtcp/source/video_rtp_depacketizer_raw.cc
@@ -10,17 +10,17 @@
#include "modules/rtp_rtcp/source/video_rtp_depacketizer_raw.h"
+#include <optional>
#include <utility>
-#include "absl/types/optional.h"
#include "modules/rtp_rtcp/source/video_rtp_depacketizer.h"
#include "rtc_base/copy_on_write_buffer.h"
namespace webrtc {
-absl::optional<VideoRtpDepacketizer::ParsedRtpPayload>
+std::optional<VideoRtpDepacketizer::ParsedRtpPayload>
VideoRtpDepacketizerRaw::Parse(rtc::CopyOnWriteBuffer rtp_payload) {
- absl::optional<ParsedRtpPayload> parsed(absl::in_place);
+ std::optional<ParsedRtpPayload> parsed(absl::in_place);
parsed->video_payload = std::move(rtp_payload);
return parsed;
}
diff --git a/modules/rtp_rtcp/source/video_rtp_depacketizer_raw.h b/modules/rtp_rtcp/source/video_rtp_depacketizer_raw.h
index 59c8695..abd2d77 100644
--- a/modules/rtp_rtcp/source/video_rtp_depacketizer_raw.h
+++ b/modules/rtp_rtcp/source/video_rtp_depacketizer_raw.h
@@ -11,7 +11,8 @@
#ifndef MODULES_RTP_RTCP_SOURCE_VIDEO_RTP_DEPACKETIZER_RAW_H_
#define MODULES_RTP_RTCP_SOURCE_VIDEO_RTP_DEPACKETIZER_RAW_H_
-#include "absl/types/optional.h"
+#include <optional>
+
#include "modules/rtp_rtcp/source/video_rtp_depacketizer.h"
#include "rtc_base/copy_on_write_buffer.h"
@@ -21,7 +22,7 @@
public:
~VideoRtpDepacketizerRaw() override = default;
- absl::optional<ParsedRtpPayload> Parse(
+ std::optional<ParsedRtpPayload> Parse(
rtc::CopyOnWriteBuffer rtp_payload) override;
};
diff --git a/modules/rtp_rtcp/source/video_rtp_depacketizer_raw_unittest.cc b/modules/rtp_rtcp/source/video_rtp_depacketizer_raw_unittest.cc
index 36c826a..218ee09 100644
--- a/modules/rtp_rtcp/source/video_rtp_depacketizer_raw_unittest.cc
+++ b/modules/rtp_rtcp/source/video_rtp_depacketizer_raw_unittest.cc
@@ -11,8 +11,8 @@
#include "modules/rtp_rtcp/source/video_rtp_depacketizer_raw.h"
#include <cstdint>
+#include <optional>
-#include "absl/types/optional.h"
#include "rtc_base/copy_on_write_buffer.h"
#include "test/gtest.h"
@@ -24,7 +24,7 @@
rtc::CopyOnWriteBuffer rtp_payload(kPayload);
VideoRtpDepacketizerRaw depacketizer;
- absl::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed =
+ std::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed =
depacketizer.Parse(rtp_payload);
ASSERT_TRUE(parsed);
@@ -39,7 +39,7 @@
rtc::CopyOnWriteBuffer rtp_payload(kPayload);
VideoRtpDepacketizerRaw depacketizer;
- absl::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed =
+ std::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed =
depacketizer.Parse(rtp_payload);
ASSERT_TRUE(parsed);
diff --git a/modules/rtp_rtcp/source/video_rtp_depacketizer_vp8.cc b/modules/rtp_rtcp/source/video_rtp_depacketizer_vp8.cc
index d6bd33c..157ad37 100644
--- a/modules/rtp_rtcp/source/video_rtp_depacketizer_vp8.cc
+++ b/modules/rtp_rtcp/source/video_rtp_depacketizer_vp8.cc
@@ -13,7 +13,8 @@
#include <stddef.h>
#include <stdint.h>
-#include "absl/types/optional.h"
+#include <optional>
+
#include "api/array_view.h"
#include "modules/rtp_rtcp/source/rtp_video_header.h"
#include "rtc_base/checks.h"
@@ -131,14 +132,14 @@
} // namespace
-absl::optional<VideoRtpDepacketizer::ParsedRtpPayload>
+std::optional<VideoRtpDepacketizer::ParsedRtpPayload>
VideoRtpDepacketizerVp8::Parse(rtc::CopyOnWriteBuffer rtp_payload) {
rtc::ArrayView<const uint8_t> payload(rtp_payload.cdata(),
rtp_payload.size());
- absl::optional<ParsedRtpPayload> result(absl::in_place);
+ std::optional<ParsedRtpPayload> result(absl::in_place);
int offset = ParseRtpPayload(payload, &result->video_header);
if (offset == kFailedToParse)
- return absl::nullopt;
+ return std::nullopt;
RTC_DCHECK_LT(offset, rtp_payload.size());
result->video_payload =
rtp_payload.Slice(offset, rtp_payload.size() - offset);
diff --git a/modules/rtp_rtcp/source/video_rtp_depacketizer_vp8.h b/modules/rtp_rtcp/source/video_rtp_depacketizer_vp8.h
index 3d7cb32..d303020 100644
--- a/modules/rtp_rtcp/source/video_rtp_depacketizer_vp8.h
+++ b/modules/rtp_rtcp/source/video_rtp_depacketizer_vp8.h
@@ -12,8 +12,8 @@
#define MODULES_RTP_RTCP_SOURCE_VIDEO_RTP_DEPACKETIZER_VP8_H_
#include <cstdint>
+#include <optional>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "modules/rtp_rtcp/source/rtp_video_header.h"
#include "modules/rtp_rtcp/source/video_rtp_depacketizer.h"
@@ -33,7 +33,7 @@
static int ParseRtpPayload(rtc::ArrayView<const uint8_t> rtp_payload,
RTPVideoHeader* video_header);
- absl::optional<ParsedRtpPayload> Parse(
+ std::optional<ParsedRtpPayload> Parse(
rtc::CopyOnWriteBuffer rtp_payload) override;
};
diff --git a/modules/rtp_rtcp/source/video_rtp_depacketizer_vp8_unittest.cc b/modules/rtp_rtcp/source/video_rtp_depacketizer_vp8_unittest.cc
index 77469cf..7bc9e97 100644
--- a/modules/rtp_rtcp/source/video_rtp_depacketizer_vp8_unittest.cc
+++ b/modules/rtp_rtcp/source/video_rtp_depacketizer_vp8_unittest.cc
@@ -201,7 +201,7 @@
ASSERT_TRUE(packetizer.NextPacket(&packet));
VideoRtpDepacketizerVp8 depacketizer;
- absl::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed =
+ std::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed =
depacketizer.Parse(packet.PayloadBuffer());
ASSERT_TRUE(parsed);
@@ -225,7 +225,7 @@
rtc::CopyOnWriteBuffer rtp_payload(packet);
VideoRtpDepacketizerVp8 depacketizer;
- absl::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed =
+ std::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed =
depacketizer.Parse(rtp_payload);
ASSERT_TRUE(parsed);
diff --git a/modules/rtp_rtcp/source/video_rtp_depacketizer_vp9.cc b/modules/rtp_rtcp/source/video_rtp_depacketizer_vp9.cc
index a8b04db..ebd27ff 100644
--- a/modules/rtp_rtcp/source/video_rtp_depacketizer_vp9.cc
+++ b/modules/rtp_rtcp/source/video_rtp_depacketizer_vp9.cc
@@ -146,12 +146,12 @@
}
} // namespace
-absl::optional<VideoRtpDepacketizer::ParsedRtpPayload>
+std::optional<VideoRtpDepacketizer::ParsedRtpPayload>
VideoRtpDepacketizerVp9::Parse(rtc::CopyOnWriteBuffer rtp_payload) {
- absl::optional<ParsedRtpPayload> result(absl::in_place);
+ std::optional<ParsedRtpPayload> result(absl::in_place);
int offset = ParseRtpPayload(rtp_payload, &result->video_header);
if (offset == 0)
- return absl::nullopt;
+ return std::nullopt;
RTC_DCHECK_LT(offset, rtp_payload.size());
result->video_payload =
rtp_payload.Slice(offset, rtp_payload.size() - offset);
diff --git a/modules/rtp_rtcp/source/video_rtp_depacketizer_vp9.h b/modules/rtp_rtcp/source/video_rtp_depacketizer_vp9.h
index 4bb358a..fb0dd14 100644
--- a/modules/rtp_rtcp/source/video_rtp_depacketizer_vp9.h
+++ b/modules/rtp_rtcp/source/video_rtp_depacketizer_vp9.h
@@ -12,8 +12,8 @@
#define MODULES_RTP_RTCP_SOURCE_VIDEO_RTP_DEPACKETIZER_VP9_H_
#include <cstdint>
+#include <optional>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "modules/rtp_rtcp/source/rtp_video_header.h"
#include "modules/rtp_rtcp/source/video_rtp_depacketizer.h"
@@ -33,7 +33,7 @@
static int ParseRtpPayload(rtc::ArrayView<const uint8_t> rtp_payload,
RTPVideoHeader* video_header);
- absl::optional<ParsedRtpPayload> Parse(
+ std::optional<ParsedRtpPayload> Parse(
rtc::CopyOnWriteBuffer rtp_payload) override;
};
diff --git a/modules/rtp_rtcp/source/video_rtp_depacketizer_vp9_unittest.cc b/modules/rtp_rtcp/source/video_rtp_depacketizer_vp9_unittest.cc
index 7f89812..fb5fd27 100644
--- a/modules/rtp_rtcp/source/video_rtp_depacketizer_vp9_unittest.cc
+++ b/modules/rtp_rtcp/source/video_rtp_depacketizer_vp9_unittest.cc
@@ -361,7 +361,7 @@
rtc::CopyOnWriteBuffer rtp_payload(packet);
VideoRtpDepacketizerVp9 depacketizer;
- absl::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed =
+ std::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed =
depacketizer.Parse(rtp_payload);
ASSERT_TRUE(parsed);
diff --git a/modules/video_capture/linux/pipewire_session.cc b/modules/video_capture/linux/pipewire_session.cc
index fd93b31..49421de 100644
--- a/modules/video_capture/linux/pipewire_session.cc
+++ b/modules/video_capture/linux/pipewire_session.cc
@@ -96,13 +96,13 @@
if (info->change_mask & PW_NODE_CHANGE_MASK_PROPS) {
const char* vid_str;
const char* pid_str;
- absl::optional<int> vid;
- absl::optional<int> pid;
+ std::optional<int> vid;
+ std::optional<int> pid;
vid_str = spa_dict_lookup(info->props, SPA_KEY_DEVICE_VENDOR_ID);
pid_str = spa_dict_lookup(info->props, SPA_KEY_DEVICE_PRODUCT_ID);
- vid = vid_str ? rtc::StringToNumber<int>(vid_str) : absl::nullopt;
- pid = pid_str ? rtc::StringToNumber<int>(pid_str) : absl::nullopt;
+ vid = vid_str ? rtc::StringToNumber<int>(vid_str) : std::nullopt;
+ pid = pid_str ? rtc::StringToNumber<int>(pid_str) : std::nullopt;
if (vid && pid) {
char model_str[10];
diff --git a/modules/video_capture/linux/video_capture_pipewire.cc b/modules/video_capture/linux/video_capture_pipewire.cc
index 940db43..b21b55f 100644
--- a/modules/video_capture/linux/video_capture_pipewire.cc
+++ b/modules/video_capture/linux/video_capture_pipewire.cc
@@ -73,9 +73,9 @@
RTC_CHECK_RUNS_SERIALIZED(&capture_checker_);
RTC_DCHECK_RUN_ON(&api_checker_);
- absl::optional<int> id;
+ std::optional<int> id;
id = rtc::StringToNumber<int>(deviceUniqueId);
- if (id == absl::nullopt)
+ if (id == std::nullopt)
return -1;
node_id_ = id.value();
diff --git a/modules/video_coding/BUILD.gn b/modules/video_coding/BUILD.gn
index 2e6b1c3..344f811 100644
--- a/modules/video_coding/BUILD.gn
+++ b/modules/video_coding/BUILD.gn
@@ -42,7 +42,6 @@
"../../rtc_base:checks",
"../../rtc_base:logging",
"//third_party/abseil-cpp/absl/container:inlined_vector",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -59,7 +58,6 @@
"../../rtc_base:logging",
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/container:inlined_vector",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -137,7 +135,6 @@
"../rtp_rtcp:rtp_rtcp_format",
"../rtp_rtcp:rtp_video_header",
"//third_party/abseil-cpp/absl/base:core_headers",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -272,7 +269,6 @@
"timing:timing_module",
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/container:inlined_vector",
- "//third_party/abseil-cpp/absl/types:optional",
"//third_party/abseil-cpp/absl/types:variant",
]
}
@@ -297,7 +293,6 @@
"../../common_video",
"../../common_video/generic_frame_descriptor",
"../../rtc_base/system:rtc_export",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -351,7 +346,6 @@
"timing:inter_frame_delay_variation_calculator",
"timing:jitter_estimator",
"timing:timing_module",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -447,7 +441,6 @@
"svc:scalability_mode_util",
"//third_party/abseil-cpp/absl/numeric:bits",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -491,7 +484,6 @@
"//third_party/abseil-cpp/absl/base:nullability",
"//third_party/abseil-cpp/absl/container:inlined_vector",
"//third_party/abseil-cpp/absl/strings",
- "//third_party/abseil-cpp/absl/types:optional",
"//third_party/libyuv",
]
@@ -576,7 +568,6 @@
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/base:nullability",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
"//third_party/libyuv",
]
if (rtc_build_libvpx) {
@@ -617,7 +608,6 @@
"../../rtc_base:rtc_numerics",
"../../rtc_base:timeutils",
"../../system_wrappers:metrics",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -697,7 +687,6 @@
"//third_party/abseil-cpp/absl/container:inlined_vector",
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/strings",
- "//third_party/abseil-cpp/absl/types:optional",
"//third_party/libyuv",
]
if (rtc_build_libvpx) {
@@ -843,7 +832,6 @@
"../../test:video_test_common",
"../../test:video_test_support",
"../rtp_rtcp:rtp_rtcp_format",
- "//third_party/abseil-cpp/absl/types:optional",
"//third_party/libyuv",
]
}
@@ -934,7 +922,6 @@
"../../video/config:streams_config",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -954,7 +941,6 @@
"../../rtc_base:stringutils",
"../../test:test_common",
"../rtp_rtcp:rtp_rtcp_format",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -1070,7 +1056,6 @@
"codecs/av1:dav1d_decoder",
"svc:scalability_mode_util",
"//third_party/abseil-cpp/absl/memory",
- "//third_party/abseil-cpp/absl/types:optional",
"//third_party/libyuv",
]
@@ -1252,7 +1237,6 @@
"svc:svc_rate_allocator_tests",
"timing:jitter_estimator",
"timing:timing_module",
- "//third_party/abseil-cpp/absl/types:optional",
"//third_party/abseil-cpp/absl/types:variant",
]
if (rtc_build_libvpx) {
diff --git a/modules/video_coding/chain_diff_calculator.cc b/modules/video_coding/chain_diff_calculator.cc
index 5f85271..d8ce44a 100644
--- a/modules/video_coding/chain_diff_calculator.cc
+++ b/modules/video_coding/chain_diff_calculator.cc
@@ -13,10 +13,10 @@
#include <stdint.h>
#include <algorithm>
+#include <optional>
#include <vector>
#include "absl/container/inlined_vector.h"
-#include "absl/types/optional.h"
#include "rtc_base/logging.h"
namespace webrtc {
@@ -25,7 +25,7 @@
last_frame_in_chain_.resize(chains.size());
for (size_t i = 0; i < chains.size(); ++i) {
if (chains[i]) {
- last_frame_in_chain_[i] = absl::nullopt;
+ last_frame_in_chain_[i] = std::nullopt;
}
}
}
diff --git a/modules/video_coding/chain_diff_calculator.h b/modules/video_coding/chain_diff_calculator.h
index bca7340..b757544 100644
--- a/modules/video_coding/chain_diff_calculator.h
+++ b/modules/video_coding/chain_diff_calculator.h
@@ -13,10 +13,10 @@
#include <stdint.h>
+#include <optional>
#include <vector>
#include "absl/container/inlined_vector.h"
-#include "absl/types/optional.h"
namespace webrtc {
@@ -38,7 +38,7 @@
private:
absl::InlinedVector<int, 4> ChainDiffs(int64_t frame_id) const;
- absl::InlinedVector<absl::optional<int64_t>, 4> last_frame_in_chain_;
+ absl::InlinedVector<std::optional<int64_t>, 4> last_frame_in_chain_;
};
} // namespace webrtc
diff --git a/modules/video_coding/codecs/av1/BUILD.gn b/modules/video_coding/codecs/av1/BUILD.gn
index 197e1f3..8a2d490 100644
--- a/modules/video_coding/codecs/av1/BUILD.gn
+++ b/modules/video_coding/codecs/av1/BUILD.gn
@@ -71,7 +71,6 @@
"//third_party/abseil-cpp/absl/base:core_headers",
"//third_party/abseil-cpp/absl/base:nullability",
"//third_party/abseil-cpp/absl/strings",
- "//third_party/abseil-cpp/absl/types:optional",
"//third_party/libaom",
]
}
@@ -110,7 +109,6 @@
"../../svc:scalability_mode_util",
"../../svc:scalability_structures",
"../../svc:scalable_video_controller",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
}
diff --git a/modules/video_coding/codecs/av1/av1_svc_config.cc b/modules/video_coding/codecs/av1/av1_svc_config.cc
index 09b840c..59b79dc 100644
--- a/modules/video_coding/codecs/av1/av1_svc_config.cc
+++ b/modules/video_coding/codecs/av1/av1_svc_config.cc
@@ -39,8 +39,8 @@
return std::min(num_layers_fit_horz, num_layers_fit_vert);
}
-absl::optional<ScalabilityMode> BuildScalabilityMode(int num_temporal_layers,
- int num_spatial_layers) {
+std::optional<ScalabilityMode> BuildScalabilityMode(int num_temporal_layers,
+ int num_spatial_layers) {
char name[20];
rtc::SimpleStringBuilder ss(name);
ss << "L" << num_spatial_layers << "T" << num_temporal_layers;
@@ -56,7 +56,7 @@
LibaomAv1EncoderSupportedScalabilityModes() {
absl::InlinedVector<ScalabilityMode, kScalabilityModeCount> scalability_modes;
for (ScalabilityMode scalability_mode : kAllScalabilityModes) {
- if (ScalabilityStructureConfig(scalability_mode) != absl::nullopt) {
+ if (ScalabilityStructureConfig(scalability_mode) != std::nullopt) {
scalability_modes.push_back(scalability_mode);
}
}
@@ -66,7 +66,7 @@
bool LibaomAv1EncoderSupportsScalabilityMode(ScalabilityMode scalability_mode) {
// For libaom AV1, the scalability mode is supported if we can create the
// scalability structure.
- return ScalabilityStructureConfig(scalability_mode) != absl::nullopt;
+ return ScalabilityStructureConfig(scalability_mode) != std::nullopt;
}
bool SetAv1SvcConfig(VideoCodec& video_codec,
@@ -74,7 +74,7 @@
int num_spatial_layers) {
RTC_DCHECK_EQ(video_codec.codecType, kVideoCodecAV1);
- absl::optional<ScalabilityMode> scalability_mode =
+ std::optional<ScalabilityMode> scalability_mode =
video_codec.GetScalabilityMode();
if (!scalability_mode.has_value()) {
scalability_mode =
diff --git a/modules/video_coding/codecs/av1/dav1d_decoder.cc b/modules/video_coding/codecs/av1/dav1d_decoder.cc
index 5a60c7f..1e93925 100644
--- a/modules/video_coding/codecs/av1/dav1d_decoder.cc
+++ b/modules/video_coding/codecs/av1/dav1d_decoder.cc
@@ -197,9 +197,9 @@
// doi: 10.1109/JPROC.2021.3058584. keywords:
// {Encoding;Codecs;Decoding;Streaming media;Video compression;Media;Alliance
// of Open Media;AV1;video compression},
- absl::optional<uint8_t> qp = dav1d_picture.frame_hdr->quant.yac;
+ std::optional<uint8_t> qp = dav1d_picture.frame_hdr->quant.yac;
decode_complete_callback_->Decoded(decoded_frame,
- /*decode_time_ms=*/absl::nullopt, qp);
+ /*decode_time_ms=*/std::nullopt, qp);
return WEBRTC_VIDEO_CODEC_OK;
}
diff --git a/modules/video_coding/codecs/av1/libaom_av1_encoder.cc b/modules/video_coding/codecs/av1/libaom_av1_encoder.cc
index c067922..6b71874 100644
--- a/modules/video_coding/codecs/av1/libaom_av1_encoder.cc
+++ b/modules/video_coding/codecs/av1/libaom_av1_encoder.cc
@@ -13,6 +13,7 @@
#include <stdint.h>
#include <memory>
+#include <optional>
#include <utility>
#include <vector>
@@ -20,7 +21,6 @@
#include "absl/base/macros.h"
#include "absl/base/nullability.h"
#include "absl/strings/match.h"
-#include "absl/types/optional.h"
#include "api/environment/environment.h"
#include "api/field_trials_view.h"
#include "api/scoped_refptr.h"
@@ -120,10 +120,10 @@
void MaybeRewrapImgWithFormat(const aom_img_fmt_t fmt);
std::unique_ptr<ScalableVideoController> svc_controller_;
- absl::optional<ScalabilityMode> scalability_mode_;
+ std::optional<ScalabilityMode> scalability_mode_;
bool inited_;
bool rates_configured_;
- absl::optional<aom_svc_params_t> svc_params_;
+ std::optional<aom_svc_params_t> svc_params_;
VideoCodec encoder_settings_;
LibaomAv1EncoderSettings settings_;
aom_image_t* frame_for_encode_;
@@ -434,7 +434,7 @@
bool svc_enabled =
svc_config.num_spatial_layers > 1 || svc_config.num_temporal_layers > 1;
if (!svc_enabled) {
- svc_params_ = absl::nullopt;
+ svc_params_ = std::nullopt;
return true;
}
if (svc_config.num_spatial_layers < 1 || svc_config.num_spatial_layers > 4) {
@@ -652,7 +652,7 @@
// The libaom AV1 encoder requires that `aom_codec_encode` is called for
// every spatial layer, even if the configured bitrate for that layer is
// zero. For zero bitrate spatial layers no frames will be produced.
- absl::optional<ScalableVideoController::LayerFrameConfig>
+ std::optional<ScalableVideoController::LayerFrameConfig>
non_encoded_layer_frame;
ScalableVideoController::LayerFrameConfig* layer_frame;
if (next_layer_frame != layer_frames.end() &&
diff --git a/modules/video_coding/codecs/av1/libaom_av1_encoder_unittest.cc b/modules/video_coding/codecs/av1/libaom_av1_encoder_unittest.cc
index a00b03a..37477be 100644
--- a/modules/video_coding/codecs/av1/libaom_av1_encoder_unittest.cc
+++ b/modules/video_coding/codecs/av1/libaom_av1_encoder_unittest.cc
@@ -13,10 +13,10 @@
#include <algorithm>
#include <limits>
#include <memory>
+#include <optional>
#include <utility>
#include <vector>
-#include "absl/types/optional.h"
#include "api/environment/environment.h"
#include "api/environment/environment_factory.h"
#include "api/test/create_frame_generator.h"
@@ -93,7 +93,7 @@
EncodedVideoFrameProducer(*encoder).SetNumInputFrames(1).Encode();
ASSERT_THAT(encoded_frames, SizeIs(1));
ASSERT_NE(encoded_frames[0].codec_specific_info.generic_frame_info,
- absl::nullopt);
+ std::nullopt);
// Assuming L1T2 structure uses 1st decode target for T0 and 2nd decode target
// for T0+T1 frames, expect only 1st decode target is active.
EXPECT_EQ(encoded_frames[0]
@@ -477,7 +477,7 @@
EXPECT_EQ(encoder->InitEncode(&codec_settings, DefaultEncoderSettings()),
WEBRTC_VIDEO_CODEC_OK);
EXPECT_EQ(encoder->GetEncoderInfo().scaling_settings.thresholds,
- absl::nullopt);
+ std::nullopt);
}
} // namespace
diff --git a/modules/video_coding/codecs/av1/libaom_av1_unittest.cc b/modules/video_coding/codecs/av1/libaom_av1_unittest.cc
index 763e399..6f71bd1 100644
--- a/modules/video_coding/codecs/av1/libaom_av1_unittest.cc
+++ b/modules/video_coding/codecs/av1/libaom_av1_unittest.cc
@@ -13,11 +13,11 @@
#include <map>
#include <memory>
+#include <optional>
#include <ostream>
#include <tuple>
#include <vector>
-#include "absl/types/optional.h"
#include "api/environment/environment.h"
#include "api/environment/environment_factory.h"
#include "api/units/data_size.h"
@@ -121,8 +121,8 @@
return 0;
}
void Decoded(VideoFrame& /*decoded_image*/,
- absl::optional<int32_t> /*decode_time_ms*/,
- absl::optional<uint8_t> /*qp*/) override {
+ std::optional<int32_t> /*decode_time_ms*/,
+ std::optional<uint8_t> /*qp*/) override {
++num_called_;
}
@@ -182,7 +182,7 @@
struct SvcTestParam {
ScalabilityMode GetScalabilityMode() const {
- absl::optional<ScalabilityMode> scalability_mode =
+ std::optional<ScalabilityMode> scalability_mode =
ScalabilityModeFromString(name);
RTC_CHECK(scalability_mode.has_value());
return *scalability_mode;
diff --git a/modules/video_coding/codecs/h264/h264.cc b/modules/video_coding/codecs/h264/h264.cc
index 60142c8..d46503b 100644
--- a/modules/video_coding/codecs/h264/h264.cc
+++ b/modules/video_coding/codecs/h264/h264.cc
@@ -12,10 +12,10 @@
#include "modules/video_coding/codecs/h264/include/h264.h"
#include <memory>
+#include <optional>
#include <string>
#include "absl/container/inlined_vector.h"
-#include "absl/types/optional.h"
#include "api/video_codecs/sdp_video_format.h"
#include "media/base/media_constants.h"
#include "rtc_base/trace_event.h"
@@ -54,7 +54,7 @@
H264Level level,
const std::string& packetization_mode,
bool add_scalability_modes) {
- const absl::optional<std::string> profile_string =
+ const std::optional<std::string> profile_string =
H264ProfileLevelIdToString(H264ProfileLevelId(profile, level));
RTC_CHECK(profile_string);
absl::InlinedVector<ScalabilityMode, kScalabilityModeCount> scalability_modes;
diff --git a/modules/video_coding/codecs/h264/h264_decoder_impl.cc b/modules/video_coding/codecs/h264/h264_decoder_impl.cc
index 7e01d6a..2ebb4cb 100644
--- a/modules/video_coding/codecs/h264/h264_decoder_impl.cc
+++ b/modules/video_coding/codecs/h264/h264_decoder_impl.cc
@@ -328,7 +328,7 @@
av_frame_.reset(av_frame_alloc());
- if (absl::optional<int> buffer_pool_size = settings.buffer_pool_size()) {
+ if (std::optional<int> buffer_pool_size = settings.buffer_pool_size()) {
if (!ffmpeg_buffer_pool_.Resize(*buffer_pool_size)) {
return false;
}
@@ -399,7 +399,7 @@
// TODO(sakal): Maybe it is possible to get QP directly from FFmpeg.
h264_bitstream_parser_.ParseBitstream(input_image);
- absl::optional<int> qp = h264_bitstream_parser_.GetLastSliceQp();
+ std::optional<int> qp = h264_bitstream_parser_.GetLastSliceQp();
// Obtain the `video_frame` containing the decoded image.
VideoFrame* input_frame =
@@ -617,7 +617,7 @@
// Return decoded frame.
// TODO(nisse): Timestamp and rotation are all zero here. Change decoder
// interface to pass a VideoFrameBuffer instead of a VideoFrame?
- decoded_image_callback_->Decoded(decoded_frame, absl::nullopt, qp);
+ decoded_image_callback_->Decoded(decoded_frame, std::nullopt, qp);
// Stop referencing it, possibly freeing `input_frame`.
av_frame_unref(av_frame_.get());
diff --git a/modules/video_coding/codecs/h264/h264_encoder_impl.cc b/modules/video_coding/codecs/h264/h264_encoder_impl.cc
index cfd89d1..bc81005 100644
--- a/modules/video_coding/codecs/h264/h264_encoder_impl.cc
+++ b/modules/video_coding/codecs/h264/h264_encoder_impl.cc
@@ -18,10 +18,10 @@
#include <algorithm>
#include <limits>
+#include <optional>
#include <string>
#include "absl/strings/match.h"
-#include "absl/types/optional.h"
#include "api/video/video_codec_constants.h"
#include "api/video_codecs/scalability_mode.h"
#include "common_video/libyuv/include/webrtc_libyuv.h"
@@ -58,7 +58,7 @@
kH264EncoderEventMax = 16,
};
-int NumberOfThreads(absl::optional<int> encoder_thread_limit,
+int NumberOfThreads(std::optional<int> encoder_thread_limit,
int width,
int height,
int number_of_cores) {
@@ -102,7 +102,7 @@
return VideoFrameType::kEmptyFrame;
}
-absl::optional<ScalabilityMode> ScalabilityModeFromTemporalLayers(
+std::optional<ScalabilityMode> ScalabilityModeFromTemporalLayers(
int num_temporal_layers) {
switch (num_temporal_layers) {
case 0:
@@ -116,7 +116,7 @@
default:
RTC_DCHECK_NOTREACHED();
}
- return absl::nullopt;
+ return std::nullopt;
}
} // namespace
diff --git a/modules/video_coding/codecs/h264/h264_encoder_impl.h b/modules/video_coding/codecs/h264/h264_encoder_impl.h
index 2818711..f04b649 100644
--- a/modules/video_coding/codecs/h264/h264_encoder_impl.h
+++ b/modules/video_coding/codecs/h264/h264_encoder_impl.h
@@ -103,7 +103,7 @@
std::vector<LayerConfig> configurations_;
std::vector<EncodedImage> encoded_images_;
std::vector<std::unique_ptr<ScalableVideoController>> svc_controllers_;
- absl::InlinedVector<absl::optional<ScalabilityMode>, kMaxSimulcastStreams>
+ absl::InlinedVector<std::optional<ScalabilityMode>, kMaxSimulcastStreams>
scalability_modes_;
const Environment env_;
@@ -111,7 +111,7 @@
H264PacketizationMode packetization_mode_;
size_t max_payload_size_;
int32_t number_of_cores_;
- absl::optional<int> encoder_thread_limit_;
+ std::optional<int> encoder_thread_limit_;
EncodedImageCallback* encoded_image_callback_;
bool has_reported_init_;
diff --git a/modules/video_coding/codecs/h264/test/h264_impl_unittest.cc b/modules/video_coding/codecs/h264/test/h264_impl_unittest.cc
index 9bc72b2..985224d 100644
--- a/modules/video_coding/codecs/h264/test/h264_impl_unittest.cc
+++ b/modules/video_coding/codecs/h264/test/h264_impl_unittest.cc
@@ -11,8 +11,8 @@
#include <stdint.h>
#include <memory>
+#include <optional>
-#include "absl/types/optional.h"
#include "api/video/color_space.h"
#include "api/video/encoded_image.h"
#include "api/video/video_frame.h"
@@ -64,7 +64,7 @@
encoded_frame._frameType = VideoFrameType::kVideoFrameKey;
EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, decoder_->Decode(encoded_frame, 0));
std::unique_ptr<VideoFrame> decoded_frame;
- absl::optional<uint8_t> decoded_qp;
+ std::optional<uint8_t> decoded_qp;
ASSERT_TRUE(WaitForDecodedFrame(&decoded_frame, &decoded_qp));
ASSERT_TRUE(decoded_frame);
EXPECT_GT(I420PSNR(&input_frame, decoded_frame.get()), 36);
@@ -89,7 +89,7 @@
encoded_frame._frameType = VideoFrameType::kVideoFrameKey;
EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, decoder_->Decode(encoded_frame, 0));
std::unique_ptr<VideoFrame> decoded_frame;
- absl::optional<uint8_t> decoded_qp;
+ std::optional<uint8_t> decoded_qp;
ASSERT_TRUE(WaitForDecodedFrame(&decoded_frame, &decoded_qp));
ASSERT_TRUE(decoded_frame);
ASSERT_TRUE(decoded_qp);
diff --git a/modules/video_coding/codecs/test/encoded_video_frame_producer.cc b/modules/video_coding/codecs/test/encoded_video_frame_producer.cc
index 1087963..cd77052 100644
--- a/modules/video_coding/codecs/test/encoded_video_frame_producer.cc
+++ b/modules/video_coding/codecs/test/encoded_video_frame_producer.cc
@@ -49,7 +49,7 @@
std::unique_ptr<test::FrameGeneratorInterface> frame_buffer_generator =
test::CreateSquareFrameGenerator(
resolution_.Width(), resolution_.Height(),
- test::FrameGeneratorInterface::OutputType::kI420, absl::nullopt);
+ test::FrameGeneratorInterface::OutputType::kI420, std::nullopt);
std::vector<EncodedFrame> encoded_frames;
EncoderCallback encoder_callback(encoded_frames);
diff --git a/modules/video_coding/codecs/test/video_codec_test.cc b/modules/video_coding/codecs/test/video_codec_test.cc
index 2b144ac..48d6bec 100644
--- a/modules/video_coding/codecs/test/video_codec_test.cc
+++ b/modules/video_coding/codecs/test/video_codec_test.cc
@@ -53,16 +53,16 @@
"Decoder: dav1d, libvpx-vp9, libvpx-vp8, ffmpeg-h264, hw-vp8, "
"hw-vp9, hw-av1, hw-h264, hw-h265");
ABSL_FLAG(std::string, scalability_mode, "L1T1", "Scalability mode.");
-ABSL_FLAG(absl::optional<int>, width, absl::nullopt, "Encode width.");
-ABSL_FLAG(absl::optional<int>, height, absl::nullopt, "Encode height.");
+ABSL_FLAG(std::optional<int>, width, std::nullopt, "Encode width.");
+ABSL_FLAG(std::optional<int>, height, std::nullopt, "Encode height.");
ABSL_FLAG(std::vector<std::string>,
bitrate_kbps,
{"1024"},
"Encode target bitrate per layer (l0t0,l0t1,...l1t0,l1t1 and so on) "
"in kbps.");
-ABSL_FLAG(absl::optional<double>,
+ABSL_FLAG(std::optional<double>,
framerate_fps,
- absl::nullopt,
+ std::nullopt,
"Encode target frame rate of the top temporal layer in fps.");
ABSL_FLAG(bool, screencast, false, "Enable screen encoding mode.");
ABSL_FLAG(bool, frame_drop, true, "Enable frame dropping.");
@@ -193,7 +193,7 @@
CreateEncoderFactory(encoder_impl);
if (!encoder_factory
->QueryCodecSupport(sdp_video_format,
- /*scalability_mode=*/absl::nullopt)
+ /*scalability_mode=*/std::nullopt)
.is_supported) {
RTC_LOG(LS_WARNING) << "No " << encoder_impl << " encoder for video format "
<< sdp_video_format.ToString();
@@ -262,7 +262,7 @@
CreateEncoderFactory(encoder_impl);
if (!encoder_factory
->QueryCodecSupport(sdp_video_format,
- /*scalability_mode=*/absl::nullopt)
+ /*scalability_mode=*/std::nullopt)
.is_supported) {
RTC_LOG(LS_WARNING) << "No encoder for video format "
<< sdp_video_format.ToString();
diff --git a/modules/video_coding/codecs/test/video_codec_unittest.cc b/modules/video_coding/codecs/test/video_codec_unittest.cc
index 556a167..f367469 100644
--- a/modules/video_coding/codecs/test/video_codec_unittest.cc
+++ b/modules/video_coding/codecs/test/video_codec_unittest.cc
@@ -57,8 +57,8 @@
void VideoCodecUnitTest::FakeDecodeCompleteCallback::Decoded(
VideoFrame& frame,
- absl::optional<int32_t> decode_time_ms,
- absl::optional<uint8_t> qp) {
+ std::optional<int32_t> decode_time_ms,
+ std::optional<uint8_t> qp) {
MutexLock lock(&test_->decoded_frame_section_);
test_->decoded_frame_.emplace(frame);
test_->decoded_qp_ = qp;
@@ -77,7 +77,7 @@
input_frame_generator_ = test::CreateSquareFrameGenerator(
codec_settings_.width, codec_settings_.height,
- test::FrameGeneratorInterface::OutputType::kI420, absl::optional<int>());
+ test::FrameGeneratorInterface::OutputType::kI420, std::optional<int>());
encoder_ = CreateEncoder();
decoder_ = CreateDecoder();
@@ -159,7 +159,7 @@
}
bool VideoCodecUnitTest::WaitForDecodedFrame(std::unique_ptr<VideoFrame>* frame,
- absl::optional<uint8_t>* qp) {
+ std::optional<uint8_t>* qp) {
bool ret = decoded_frame_event_.Wait(kDecodeTimeout);
EXPECT_TRUE(ret) << "Timed out while waiting for a decoded frame.";
// This becomes unsafe if there are multiple threads waiting for frames.
diff --git a/modules/video_coding/codecs/test/video_codec_unittest.h b/modules/video_coding/codecs/test/video_codec_unittest.h
index 6ad46ba..b154a4b 100644
--- a/modules/video_coding/codecs/test/video_codec_unittest.h
+++ b/modules/video_coding/codecs/test/video_codec_unittest.h
@@ -65,8 +65,8 @@
return -1;
}
void Decoded(VideoFrame& frame,
- absl::optional<int32_t> decode_time_ms,
- absl::optional<uint8_t> qp) override;
+ std::optional<int32_t> decode_time_ms,
+ std::optional<uint8_t> qp) override;
private:
VideoCodecUnitTest* const test_;
@@ -95,7 +95,7 @@
// Helper method for waiting a single decoded frame.
bool WaitForDecodedFrame(std::unique_ptr<VideoFrame>* frame,
- absl::optional<uint8_t>* qp);
+ std::optional<uint8_t>* qp);
size_t GetNumEncodedFrames();
@@ -120,9 +120,9 @@
rtc::Event decoded_frame_event_;
Mutex decoded_frame_section_;
- absl::optional<VideoFrame> decoded_frame_
+ std::optional<VideoFrame> decoded_frame_
RTC_GUARDED_BY(decoded_frame_section_);
- absl::optional<uint8_t> decoded_qp_ RTC_GUARDED_BY(decoded_frame_section_);
+ std::optional<uint8_t> decoded_qp_ RTC_GUARDED_BY(decoded_frame_section_);
uint32_t last_input_frame_timestamp_;
};
diff --git a/modules/video_coding/codecs/test/videocodec_test_fixture_impl.cc b/modules/video_coding/codecs/test/videocodec_test_fixture_impl.cc
index 4056959..414f22d 100644
--- a/modules/video_coding/codecs/test/videocodec_test_fixture_impl.cc
+++ b/modules/video_coding/codecs/test/videocodec_test_fixture_impl.cc
@@ -16,6 +16,7 @@
#include <algorithm>
#include <cmath>
#include <memory>
+#include <optional>
#include <string>
#include <utility>
#include <vector>
@@ -23,7 +24,6 @@
#include "absl/strings/match.h"
#include "absl/strings/str_replace.h"
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/environment/environment.h"
#include "api/environment/environment_factory.h"
diff --git a/modules/video_coding/codecs/test/videocodec_test_stats_impl.cc b/modules/video_coding/codecs/test/videocodec_test_stats_impl.cc
index 390348b..9e1a61f 100644
--- a/modules/video_coding/codecs/test/videocodec_test_stats_impl.cc
+++ b/modules/video_coding/codecs/test/videocodec_test_stats_impl.cc
@@ -106,7 +106,7 @@
++temporal_idx) {
VideoStatistics layer_stat = SliceAndCalcVideoStatistic(
first_frame_num, last_frame_num, spatial_idx, temporal_idx, false,
- /*target_bitrate=*/absl::nullopt, /*target_framerate=*/absl::nullopt);
+ /*target_bitrate=*/std::nullopt, /*target_framerate=*/std::nullopt);
layer_stats.push_back(layer_stat);
}
}
@@ -126,8 +126,8 @@
return SliceAndCalcVideoStatistic(
first_frame_num, last_frame_num, num_spatial_layers - 1,
- num_temporal_layers - 1, true, /*target_bitrate=*/absl::nullopt,
- /*target_framerate=*/absl::nullopt);
+ num_temporal_layers - 1, true, /*target_bitrate=*/std::nullopt,
+ /*target_framerate=*/std::nullopt);
}
VideoStatistics VideoCodecTestStatsImpl::CalcVideoStatistic(
@@ -205,8 +205,8 @@
size_t spatial_idx,
size_t temporal_idx,
bool aggregate_independent_layers,
- absl::optional<DataRate> target_bitrate,
- absl::optional<Frequency> target_framerate) {
+ std::optional<DataRate> target_bitrate,
+ std::optional<Frequency> target_framerate) {
VideoStatistics video_stat;
float buffer_level_bits = 0.0f;
diff --git a/modules/video_coding/codecs/test/videocodec_test_stats_impl.h b/modules/video_coding/codecs/test/videocodec_test_stats_impl.h
index 4212cfa..7040efb 100644
--- a/modules/video_coding/codecs/test/videocodec_test_stats_impl.h
+++ b/modules/video_coding/codecs/test/videocodec_test_stats_impl.h
@@ -14,10 +14,10 @@
#include <stddef.h>
#include <map>
+#include <optional>
#include <string>
#include <vector>
-#include "absl/types/optional.h"
#include "api/test/videocodec_test_stats.h" // NOLINT(build/include)
namespace webrtc {
@@ -76,8 +76,8 @@
size_t spatial_idx,
size_t temporal_idx,
bool aggregate_independent_layers,
- absl::optional<DataRate> target_bitrate,
- absl::optional<Frequency> target_framerate);
+ std::optional<DataRate> target_bitrate,
+ std::optional<Frequency> target_framerate);
void GetNumberOfEncodedLayers(size_t first_frame_num,
size_t last_frame_num,
diff --git a/modules/video_coding/codecs/test/videoprocessor.h b/modules/video_coding/codecs/test/videoprocessor.h
index e6ecdff..35a925b 100644
--- a/modules/video_coding/codecs/test/videoprocessor.h
+++ b/modules/video_coding/codecs/test/videoprocessor.h
@@ -16,10 +16,10 @@
#include <map>
#include <memory>
+#include <optional>
#include <utility>
#include <vector>
-#include "absl/types/optional.h"
#include "api/environment/environment.h"
#include "api/sequence_checker.h"
#include "api/task_queue/task_queue_base.h"
@@ -146,8 +146,8 @@
}
void Decoded(webrtc::VideoFrame& image,
- absl::optional<int32_t> decode_time_ms,
- absl::optional<uint8_t> qp) override {
+ std::optional<int32_t> decode_time_ms,
+ std::optional<uint8_t> qp) override {
Decoded(image);
}
diff --git a/modules/video_coding/codecs/vp8/default_temporal_layers.h b/modules/video_coding/codecs/vp8/default_temporal_layers.h
index bc6574c..d6b2491 100644
--- a/modules/video_coding/codecs/vp8/default_temporal_layers.h
+++ b/modules/video_coding/codecs/vp8/default_temporal_layers.h
@@ -19,11 +19,11 @@
#include <deque>
#include <limits>
#include <memory>
+#include <optional>
#include <set>
#include <utility>
#include <vector>
-#include "absl/types/optional.h"
#include "api/video_codecs/vp8_frame_config.h"
#include "api/video_codecs/vp8_temporal_layers.h"
#include "modules/video_coding/codecs/vp8/include/temporal_layers_checker.h"
@@ -124,7 +124,7 @@
uint8_t pattern_idx_;
// Updated cumulative bitrates, per temporal layer.
- absl::optional<std::vector<uint32_t>> new_bitrates_bps_;
+ std::optional<std::vector<uint32_t>> new_bitrates_bps_;
// Status for each pending frame, in
std::deque<PendingFrame> pending_frames_;
diff --git a/modules/video_coding/codecs/vp8/libvpx_vp8_decoder.cc b/modules/video_coding/codecs/vp8/libvpx_vp8_decoder.cc
index 9a7a6f1..fc6cad6 100644
--- a/modules/video_coding/codecs/vp8/libvpx_vp8_decoder.cc
+++ b/modules/video_coding/codecs/vp8/libvpx_vp8_decoder.cc
@@ -15,9 +15,9 @@
#include <algorithm>
#include <memory>
+#include <optional>
#include <string>
-#include "absl/types/optional.h"
#include "api/environment/environment.h"
#include "api/field_trials_view.h"
#include "api/scoped_refptr.h"
@@ -53,13 +53,13 @@
constexpr bool kIsArm = false;
#endif
-absl::optional<LibvpxVp8Decoder::DeblockParams> DefaultDeblockParams() {
+std::optional<LibvpxVp8Decoder::DeblockParams> DefaultDeblockParams() {
return LibvpxVp8Decoder::DeblockParams(/*max_level=*/8,
/*degrade_qp=*/60,
/*min_qp=*/30);
}
-absl::optional<LibvpxVp8Decoder::DeblockParams>
+std::optional<LibvpxVp8Decoder::DeblockParams>
GetPostProcParamsFromFieldTrialGroup(const FieldTrialsView& field_trials) {
std::string group = field_trials.Lookup(kIsArm ? kVp8PostProcArmFieldTrial
: kVp8PostProcFieldTrial);
@@ -127,7 +127,7 @@
key_frame_required_(true),
deblock_params_(use_postproc_ ? GetPostProcParamsFromFieldTrialGroup(
env.field_trials())
- : absl::nullopt),
+ : std::nullopt),
qp_smoother_(use_postproc_ ? new QpSmoother() : nullptr) {}
LibvpxVp8Decoder::~LibvpxVp8Decoder() {
@@ -160,7 +160,7 @@
// Always start with a complete key frame.
key_frame_required_ = true;
- if (absl::optional<int> buffer_pool_size = settings.buffer_pool_size()) {
+ if (std::optional<int> buffer_pool_size = settings.buffer_pool_size()) {
if (!buffer_pool_.Resize(*buffer_pool_size)) {
return false;
}
@@ -305,7 +305,7 @@
.set_rtp_timestamp(timestamp)
.set_color_space(explicit_color_space)
.build();
- decode_complete_callback_->Decoded(decoded_image, absl::nullopt, qp);
+ decode_complete_callback_->Decoded(decoded_image, std::nullopt, qp);
return WEBRTC_VIDEO_CODEC_OK;
}
diff --git a/modules/video_coding/codecs/vp8/libvpx_vp8_decoder.h b/modules/video_coding/codecs/vp8/libvpx_vp8_decoder.h
index c14a2b7..725ad71 100644
--- a/modules/video_coding/codecs/vp8/libvpx_vp8_decoder.h
+++ b/modules/video_coding/codecs/vp8/libvpx_vp8_decoder.h
@@ -12,8 +12,8 @@
#define MODULES_VIDEO_CODING_CODECS_VP8_LIBVPX_VP8_DECODER_H_
#include <memory>
+#include <optional>
-#include "absl/types/optional.h"
#include "api/environment/environment.h"
#include "api/field_trials_view.h"
#include "api/video/encoded_image.h"
@@ -72,7 +72,7 @@
int last_frame_width_;
int last_frame_height_;
bool key_frame_required_;
- const absl::optional<DeblockParams> deblock_params_;
+ const std::optional<DeblockParams> deblock_params_;
const std::unique_ptr<QpSmoother> qp_smoother_;
};
diff --git a/modules/video_coding/codecs/vp8/libvpx_vp8_encoder.cc b/modules/video_coding/codecs/vp8/libvpx_vp8_encoder.cc
index 7b83e93..7edd3fe 100644
--- a/modules/video_coding/codecs/vp8/libvpx_vp8_encoder.cc
+++ b/modules/video_coding/codecs/vp8/libvpx_vp8_encoder.cc
@@ -117,8 +117,8 @@
// Allow a newer value to override a current value only if the new value
// is set.
template <typename T>
-bool MaybeSetNewValue(const absl::optional<T>& new_value,
- absl::optional<T>* base_value) {
+bool MaybeSetNewValue(const std::optional<T>& new_value,
+ std::optional<T>* base_value) {
if (new_value.has_value() && new_value != *base_value) {
*base_value = new_value;
return true;
@@ -254,7 +254,7 @@
const uint32_t original_frame_drop_threshold_;
};
-absl::optional<TimeDelta> ParseFrameDropInterval(
+std::optional<TimeDelta> ParseFrameDropInterval(
const FieldTrialsView& field_trials) {
FieldTrialFlag disabled = FieldTrialFlag("Disabled");
FieldTrialParameter<TimeDelta> interval("interval",
@@ -263,7 +263,7 @@
field_trials.Lookup("WebRTC-VP8-MaxFrameInterval"));
if (disabled.Get()) {
// Kill switch set, don't use any max frame interval.
- return absl::nullopt;
+ return std::nullopt;
}
return interval.Get();
}
@@ -494,7 +494,7 @@
return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
}
- if (absl::optional<ScalabilityMode> scalability_mode =
+ if (std::optional<ScalabilityMode> scalability_mode =
inst->GetScalabilityMode();
scalability_mode.has_value() &&
!VP8SupportsScalabilityMode(*scalability_mode)) {
@@ -1155,7 +1155,7 @@
frame_buffer_controller_->OnEncodeDone(stream_idx, timestamp,
encoded_images_[encoder_idx].size(),
is_keyframe, qp, codec_specific);
- if (is_keyframe && codec_specific->template_structure != absl::nullopt) {
+ if (is_keyframe && codec_specific->template_structure != std::nullopt) {
// Number of resolutions must match number of spatial layers, VP8 structures
// expected to use single spatial layer. Templates must be ordered by
// spatial_id, so assumption there is exactly one spatial layer is same as
diff --git a/modules/video_coding/codecs/vp8/libvpx_vp8_encoder.h b/modules/video_coding/codecs/vp8/libvpx_vp8_encoder.h
index 1800741..3f26efc 100644
--- a/modules/video_coding/codecs/vp8/libvpx_vp8_encoder.h
+++ b/modules/video_coding/codecs/vp8/libvpx_vp8_encoder.h
@@ -145,7 +145,7 @@
const LibvpxVp8EncoderInfoSettings encoder_info_override_;
- absl::optional<TimeDelta> max_frame_drop_interval_;
+ std::optional<TimeDelta> max_frame_drop_interval_;
bool android_specific_threading_settings_;
};
diff --git a/modules/video_coding/codecs/vp8/screenshare_layers.cc b/modules/video_coding/codecs/vp8/screenshare_layers.cc
index 71db0b2..ccec4b4 100644
--- a/modules/video_coding/codecs/vp8/screenshare_layers.cc
+++ b/modules/video_coding/codecs/vp8/screenshare_layers.cc
@@ -302,7 +302,7 @@
return;
}
- absl::optional<DependencyInfo> dependency_info;
+ std::optional<DependencyInfo> dependency_info;
auto it = pending_frame_configs_.find(rtp_timestamp);
if (it != pending_frame_configs_.end()) {
dependency_info = it->second;
@@ -510,9 +510,8 @@
static_cast<float>(*capture_framerate_) / *target_framerate_;
}
- if (bitrate_updated_ ||
- encoder_config_.rc_target_bitrate !=
- absl::make_optional(encoder_config_bitrate_kbps)) {
+ if (bitrate_updated_ || encoder_config_.rc_target_bitrate !=
+ std::make_optional(encoder_config_bitrate_kbps)) {
encoder_config_.rc_target_bitrate = encoder_config_bitrate_kbps;
// Don't reconfigure qp limits during quality boost frames.
diff --git a/modules/video_coding/codecs/vp8/screenshare_layers.h b/modules/video_coding/codecs/vp8/screenshare_layers.h
index 47d6b40..7f721ab 100644
--- a/modules/video_coding/codecs/vp8/screenshare_layers.h
+++ b/modules/video_coding/codecs/vp8/screenshare_layers.h
@@ -91,8 +91,8 @@
const int number_of_temporal_layers_;
// TODO(eladalon/sprang): These should be made into const-int set in the ctor.
- absl::optional<int> min_qp_;
- absl::optional<int> max_qp_;
+ std::optional<int> min_qp_;
+ std::optional<int> max_qp_;
int active_layer_;
int64_t last_timestamp_;
@@ -105,9 +105,9 @@
std::map<uint32_t, DependencyInfo> pending_frame_configs_;
// Configured max framerate.
- absl::optional<uint32_t> target_framerate_;
+ std::optional<uint32_t> target_framerate_;
// Incoming framerate from capturer.
- absl::optional<uint32_t> capture_framerate_;
+ std::optional<uint32_t> capture_framerate_;
// Tracks what framerate we actually encode, and drops frames on overshoot.
RateStatistics encode_framerate_;
diff --git a/modules/video_coding/codecs/vp8/screenshare_layers_unittest.cc b/modules/video_coding/codecs/vp8/screenshare_layers_unittest.cc
index e5b3bd4..4fb599f 100644
--- a/modules/video_coding/codecs/vp8/screenshare_layers_unittest.cc
+++ b/modules/video_coding/codecs/vp8/screenshare_layers_unittest.cc
@@ -161,12 +161,10 @@
// Adds frames until we get one in the specified temporal layer. The last
// FrameEncoded() call will be omitted and needs to be done by the caller.
// Returns the flags for the last frame.
- int SkipUntilTl(int layer) {
- return SkipUntilTlAndSync(layer, absl::nullopt);
- }
+ int SkipUntilTl(int layer) { return SkipUntilTlAndSync(layer, std::nullopt); }
// Same as SkipUntilTl, but also waits until the sync bit condition is met.
- int SkipUntilTlAndSync(int layer, absl::optional<bool> sync) {
+ int SkipUntilTlAndSync(int layer, std::optional<bool> sync) {
int flags = 0;
const int kMaxFramesToSkip =
1 + (sync.value_or(false) ? kMaxSyncPeriodSeconds : 1) * kFrameRate;
diff --git a/modules/video_coding/codecs/vp8/test/vp8_impl_unittest.cc b/modules/video_coding/codecs/vp8/test/vp8_impl_unittest.cc
index 8c03148..cb1b918 100644
--- a/modules/video_coding/codecs/vp8/test/vp8_impl_unittest.cc
+++ b/modules/video_coding/codecs/vp8/test/vp8_impl_unittest.cc
@@ -214,7 +214,7 @@
CodecSpecificInfo codec_specific_info;
input_frame_generator_ = test::CreateSquareFrameGenerator(
kWidth, kHeight, test::FrameGeneratorInterface::OutputType::kNV12,
- absl::nullopt);
+ std::nullopt);
EncodeAndWaitForFrame(NextInputFrame(), &encoded_frame, &codec_specific_info);
EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, encoder_->Release());
@@ -231,11 +231,11 @@
CodecSpecificInfo codec_specific_info;
input_frame_generator_ = test::CreateSquareFrameGenerator(
kWidth, kHeight, test::FrameGeneratorInterface::OutputType::kNV12,
- absl::nullopt);
+ std::nullopt);
EncodeAndWaitForFrame(NextInputFrame(), &encoded_frame, &codec_specific_info);
input_frame_generator_ = test::CreateSquareFrameGenerator(
kWidth, kHeight, test::FrameGeneratorInterface::OutputType::kI420,
- absl::nullopt);
+ std::nullopt);
EncodeAndWaitForFrame(NextInputFrame(), &encoded_frame, &codec_specific_info);
EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, encoder_->Release());
@@ -288,7 +288,7 @@
encoded_frame._frameType = VideoFrameType::kVideoFrameKey;
EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, decoder_->Decode(encoded_frame, -1));
std::unique_ptr<VideoFrame> decoded_frame;
- absl::optional<uint8_t> decoded_qp;
+ std::optional<uint8_t> decoded_qp;
ASSERT_TRUE(WaitForDecodedFrame(&decoded_frame, &decoded_qp));
ASSERT_TRUE(decoded_frame);
ASSERT_TRUE(decoded_qp);
@@ -504,7 +504,7 @@
EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, decoder_->Decode(encoded_frame, -1));
std::unique_ptr<VideoFrame> decoded_frame;
- absl::optional<uint8_t> decoded_qp;
+ std::optional<uint8_t> decoded_qp;
ASSERT_TRUE(WaitForDecodedFrame(&decoded_frame, &decoded_qp));
ASSERT_TRUE(decoded_frame);
// Compute PSNR on all planes (faster than SSIM).
@@ -574,7 +574,7 @@
input_frame_generator_ = test::CreateSquareFrameGenerator(
codec_settings_.width, codec_settings_.height,
test::FrameGeneratorInterface::OutputType::kI420,
- /* num_squares = */ absl::optional<int>(300));
+ /* num_squares = */ std::optional<int>(300));
EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
encoder_->InitEncode(&codec_settings_, kSettings));
diff --git a/modules/video_coding/codecs/vp9/libvpx_vp9_decoder.cc b/modules/video_coding/codecs/vp9/libvpx_vp9_decoder.cc
index c83f929..8be1757 100644
--- a/modules/video_coding/codecs/vp9/libvpx_vp9_decoder.cc
+++ b/modules/video_coding/codecs/vp9/libvpx_vp9_decoder.cc
@@ -169,7 +169,7 @@
inited_ = true;
// Always start with a complete key frame.
key_frame_required_ = true;
- if (absl::optional<int> buffer_pool_size = settings.buffer_pool_size()) {
+ if (std::optional<int> buffer_pool_size = settings.buffer_pool_size()) {
if (!libvpx_buffer_pool_.Resize(*buffer_pool_size)) {
return false;
}
@@ -196,7 +196,7 @@
}
if (input_image._frameType == VideoFrameType::kVideoFrameKey) {
- absl::optional<Vp9UncompressedHeader> frame_info =
+ std::optional<Vp9UncompressedHeader> frame_info =
ParseUncompressedVp9Header(
rtc::MakeArrayView(input_image.data(), input_image.size()));
if (frame_info) {
@@ -353,7 +353,7 @@
}
VideoFrame decoded_image = builder.build();
- decode_complete_callback_->Decoded(decoded_image, absl::nullopt, qp);
+ decode_complete_callback_->Decoded(decoded_image, std::nullopt, qp);
return WEBRTC_VIDEO_CODEC_OK;
}
diff --git a/modules/video_coding/codecs/vp9/libvpx_vp9_encoder.cc b/modules/video_coding/codecs/vp9/libvpx_vp9_encoder.cc
index 7a2b494..c9129ca 100644
--- a/modules/video_coding/codecs/vp9/libvpx_vp9_encoder.cc
+++ b/modules/video_coding/codecs/vp9/libvpx_vp9_encoder.cc
@@ -15,13 +15,13 @@
#include <algorithm>
#include <limits>
+#include <optional>
#include <utility>
#include <vector>
#include "absl/algorithm/container.h"
#include "absl/memory/memory.h"
#include "absl/strings/match.h"
-#include "absl/types/optional.h"
#include "api/video/color_space.h"
#include "api/video/i010_buffer.h"
#include "api/video_codecs/scalability_mode.h"
@@ -158,7 +158,7 @@
}
}
- absl::optional<ScalabilityMode> scalability_mode =
+ std::optional<ScalabilityMode> scalability_mode =
ScalabilityModeFromString(name);
if (!scalability_mode.has_value()) {
RTC_LOG(LS_WARNING) << "Invalid scalability mode " << name;
@@ -504,8 +504,8 @@
return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
}
- absl::optional<vpx_img_fmt_t> previous_img_fmt =
- raw_ ? absl::make_optional<vpx_img_fmt_t>(raw_->fmt) : absl::nullopt;
+ std::optional<vpx_img_fmt_t> previous_img_fmt =
+ raw_ ? std::make_optional<vpx_img_fmt_t>(raw_->fmt) : std::nullopt;
int ret_val = Release();
if (ret_val < 0) {
@@ -1214,8 +1214,8 @@
}
bool LibvpxVp9Encoder::PopulateCodecSpecific(CodecSpecificInfo* codec_specific,
- absl::optional<int>* spatial_idx,
- absl::optional<int>* temporal_idx,
+ std::optional<int>* spatial_idx,
+ std::optional<int>* temporal_idx,
const vpx_codec_cx_pkt& pkt) {
RTC_CHECK(codec_specific != nullptr);
codec_specific->codecType = kVideoCodecVP9;
@@ -1241,14 +1241,14 @@
if (num_temporal_layers_ == 1) {
RTC_CHECK_EQ(layer_id.temporal_layer_id, 0);
vp9_info->temporal_idx = kNoTemporalIdx;
- *temporal_idx = absl::nullopt;
+ *temporal_idx = std::nullopt;
} else {
vp9_info->temporal_idx = layer_id.temporal_layer_id;
*temporal_idx = layer_id.temporal_layer_id;
}
if (num_active_spatial_layers_ == 1) {
RTC_CHECK_EQ(layer_id.spatial_layer_id, 0);
- *spatial_idx = absl::nullopt;
+ *spatial_idx = std::nullopt;
} else {
*spatial_idx = layer_id.spatial_layer_id;
}
@@ -1395,8 +1395,8 @@
codec_specific_.scalability_mode = MakeScalabilityMode(
num_active_spatial_layers_, num_temporal_layers_, inter_layer_pred_,
num_active_spatial_layers_ > 1
- ? absl::make_optional(ScalabilityModeResolutionRatio::kTwoToOne)
- : absl::nullopt,
+ ? std::make_optional(ScalabilityModeResolutionRatio::kTwoToOne)
+ : std::nullopt,
/*shift=*/false);
}
@@ -1564,7 +1564,7 @@
const bool is_inter_layer_pred_allowed =
inter_layer_pred_ == InterLayerPredMode::kOn ||
(inter_layer_pred_ == InterLayerPredMode::kOnKeyPic && is_key_pic);
- absl::optional<int> last_updated_buf_idx;
+ std::optional<int> last_updated_buf_idx;
// Put temporal reference to LAST and spatial reference to GOLDEN. Update
// frame buffer (i.e. store encoded frame) if current frame is a temporal
@@ -1660,8 +1660,8 @@
static_cast<const uint8_t*>(pkt->data.frame.buf), pkt->data.frame.sz));
codec_specific_ = {};
- absl::optional<int> spatial_index;
- absl::optional<int> temporal_index;
+ std::optional<int> spatial_index;
+ std::optional<int> temporal_index;
if (!PopulateCodecSpecific(&codec_specific_, &spatial_index, &temporal_index,
*pkt)) {
// Drop the frame.
diff --git a/modules/video_coding/codecs/vp9/libvpx_vp9_encoder.h b/modules/video_coding/codecs/vp9/libvpx_vp9_encoder.h
index f83fe2d..6454c98 100644
--- a/modules/video_coding/codecs/vp9/libvpx_vp9_encoder.h
+++ b/modules/video_coding/codecs/vp9/libvpx_vp9_encoder.h
@@ -69,8 +69,8 @@
int InitAndSetControlSettings(const VideoCodec* inst);
bool PopulateCodecSpecific(CodecSpecificInfo* codec_specific,
- absl::optional<int>* spatial_idx,
- absl::optional<int>* temporal_idx,
+ std::optional<int>* spatial_idx,
+ std::optional<int>* temporal_idx,
const vpx_codec_cx_pkt& pkt);
void FillReferenceIndices(const vpx_codec_cx_pkt& pkt,
size_t pic_num,
@@ -152,7 +152,7 @@
bool force_all_active_layers_;
std::unique_ptr<ScalableVideoController> svc_controller_;
- absl::optional<ScalabilityMode> scalability_mode_;
+ std::optional<ScalabilityMode> scalability_mode_;
std::vector<FramerateControllerDeprecated> framerate_controller_;
// Used for flexible mode.
diff --git a/modules/video_coding/codecs/vp9/svc_config.cc b/modules/video_coding/codecs/vp9/svc_config.cc
index 555af83..cd3990d 100644
--- a/modules/video_coding/codecs/vp9/svc_config.cc
+++ b/modules/video_coding/codecs/vp9/svc_config.cc
@@ -83,7 +83,7 @@
size_t first_active_layer,
size_t num_spatial_layers,
size_t num_temporal_layers,
- absl::optional<ScalableVideoController::StreamLayersConfig> config) {
+ std::optional<ScalableVideoController::StreamLayersConfig> config) {
RTC_DCHECK_LT(first_active_layer, num_spatial_layers);
// Limit number of layers for given resolution.
@@ -169,7 +169,7 @@
std::vector<SpatialLayer> GetVp9SvcConfig(VideoCodec& codec) {
RTC_DCHECK_EQ(codec.codecType, kVideoCodecVP9);
- absl::optional<ScalabilityMode> scalability_mode = codec.GetScalabilityMode();
+ std::optional<ScalabilityMode> scalability_mode = codec.GetScalabilityMode();
RTC_DCHECK(scalability_mode.has_value());
bool requested_single_spatial_layer =
@@ -193,7 +193,7 @@
codec.VP9()->interLayerPred =
ScalabilityModeToInterLayerPredMode(*scalability_mode);
- absl::optional<ScalableVideoController::StreamLayersConfig> info =
+ std::optional<ScalableVideoController::StreamLayersConfig> info =
ScalabilityStructureConfig(*scalability_mode);
if (!info.has_value()) {
RTC_LOG(LS_WARNING) << "Failed to create structure "
@@ -206,7 +206,7 @@
GetSvcConfig(codec.width, codec.height, codec.maxFramerate,
/*first_active_layer=*/0, info->num_spatial_layers,
info->num_temporal_layers, /*is_screen_sharing=*/false,
- codec.GetScalabilityMode() ? info : absl::nullopt);
+ codec.GetScalabilityMode() ? info : std::nullopt);
RTC_DCHECK(!spatial_layers.empty());
spatial_layers[0].minBitrate = kMinVp9SvcBitrateKbps;
@@ -230,7 +230,7 @@
size_t num_spatial_layers,
size_t num_temporal_layers,
bool is_screen_sharing,
- absl::optional<ScalableVideoController::StreamLayersConfig> config) {
+ std::optional<ScalableVideoController::StreamLayersConfig> config) {
RTC_DCHECK_GT(input_width, 0);
RTC_DCHECK_GT(input_height, 0);
RTC_DCHECK_GT(num_spatial_layers, 0);
diff --git a/modules/video_coding/codecs/vp9/svc_config.h b/modules/video_coding/codecs/vp9/svc_config.h
index adeaf0f..eaf316b 100644
--- a/modules/video_coding/codecs/vp9/svc_config.h
+++ b/modules/video_coding/codecs/vp9/svc_config.h
@@ -31,8 +31,8 @@
size_t num_spatial_layers,
size_t num_temporal_layers,
bool is_screen_sharing,
- absl::optional<ScalableVideoController::StreamLayersConfig> config =
- absl::nullopt);
+ std::optional<ScalableVideoController::StreamLayersConfig> config =
+ std::nullopt);
} // namespace webrtc
diff --git a/modules/video_coding/codecs/vp9/test/vp9_impl_unittest.cc b/modules/video_coding/codecs/vp9/test/vp9_impl_unittest.cc
index 2632610..3a1f351 100644
--- a/modules/video_coding/codecs/vp9/test/vp9_impl_unittest.cc
+++ b/modules/video_coding/codecs/vp9/test/vp9_impl_unittest.cc
@@ -125,7 +125,7 @@
protected:
void SetUp() override {
input_frame_generator_ = test::CreateSquareFrameGenerator(
- kWidth, kHeight, GetParam(), absl::optional<int>());
+ kWidth, kHeight, GetParam(), std::optional<int>());
TestVp9Impl::SetUp();
}
};
@@ -145,7 +145,7 @@
encoded_frame._frameType = VideoFrameType::kVideoFrameKey;
EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, decoder_->Decode(encoded_frame, 0));
std::unique_ptr<VideoFrame> decoded_frame;
- absl::optional<uint8_t> decoded_qp;
+ std::optional<uint8_t> decoded_qp;
ASSERT_TRUE(WaitForDecodedFrame(&decoded_frame, &decoded_qp));
ASSERT_TRUE(decoded_frame);
EXPECT_GT(I420PSNR(&input_frame, decoded_frame.get()), 36);
@@ -195,7 +195,7 @@
// Encoded frame without explicit color space information.
EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, decoder_->Decode(encoded_frame, 0));
std::unique_ptr<VideoFrame> decoded_frame;
- absl::optional<uint8_t> decoded_qp;
+ std::optional<uint8_t> decoded_qp;
ASSERT_TRUE(WaitForDecodedFrame(&decoded_frame, &decoded_qp));
ASSERT_TRUE(decoded_frame);
// Color space present from encoded bitstream.
@@ -213,7 +213,7 @@
encoded_frame._frameType = VideoFrameType::kVideoFrameKey;
EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, decoder_->Decode(encoded_frame, 0));
std::unique_ptr<VideoFrame> decoded_frame;
- absl::optional<uint8_t> decoded_qp;
+ std::optional<uint8_t> decoded_qp;
ASSERT_TRUE(WaitForDecodedFrame(&decoded_frame, &decoded_qp));
ASSERT_TRUE(decoded_frame);
ASSERT_TRUE(decoded_qp);
@@ -242,14 +242,14 @@
// Change the input frame type from I420 to NV12, encoding should still work.
input_frame_generator_ = test::CreateSquareFrameGenerator(
kWidth, kHeight, test::FrameGeneratorInterface::OutputType::kNV12,
- absl::optional<int>());
+ std::optional<int>());
EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, encoder_->Encode(NextInputFrame(), nullptr));
ASSERT_TRUE(WaitForEncodedFrame(&encoded_frame, &codec_specific_info));
// Flipping back to I420, encoding should still work.
input_frame_generator_ = test::CreateSquareFrameGenerator(
kWidth, kHeight, test::FrameGeneratorInterface::OutputType::kI420,
- absl::optional<int>());
+ std::optional<int>());
EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, encoder_->Encode(NextInputFrame(), nullptr));
ASSERT_TRUE(WaitForEncodedFrame(&encoded_frame, &codec_specific_info));
}
@@ -531,7 +531,7 @@
}
MATCHER_P2(GenericLayerIs, spatial_id, temporal_id, "") {
- if (arg.codec_specific_info.generic_frame_info == absl::nullopt) {
+ if (arg.codec_specific_info.generic_frame_info == std::nullopt) {
*result_listener << " miss generic_frame_info";
return false;
}
@@ -1861,8 +1861,8 @@
frame.codec_specific_info.codecSpecific.VP9;
EXPECT_EQ(frame.encoded_image.SpatialIndex(),
num_spatial_layers_ == 1
- ? absl::nullopt
- : absl::optional<int>(i % num_spatial_layers_))
+ ? std::nullopt
+ : std::optional<int>(i % num_spatial_layers_))
<< "Frame " << i;
EXPECT_EQ(vp9.temporal_idx, num_temporal_layers_ == 1
? kNoTemporalIdx
@@ -2009,8 +2009,7 @@
TestVp9Impl::SetUp();
input_frame_generator_ = test::CreateSquareFrameGenerator(
codec_settings_.width, codec_settings_.height,
- test::FrameGeneratorInterface::OutputType::kI010,
- absl::optional<int>());
+ test::FrameGeneratorInterface::OutputType::kI010, std::optional<int>());
}
std::unique_ptr<VideoEncoder> CreateEncoder() override {
@@ -2035,7 +2034,7 @@
encoded_frame._frameType = VideoFrameType::kVideoFrameKey;
EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, decoder_->Decode(encoded_frame, 0));
std::unique_ptr<VideoFrame> decoded_frame;
- absl::optional<uint8_t> decoded_qp;
+ std::optional<uint8_t> decoded_qp;
ASSERT_TRUE(WaitForDecodedFrame(&decoded_frame, &decoded_qp));
ASSERT_TRUE(decoded_frame);
@@ -2431,7 +2430,7 @@
struct SvcFrameDropConfigTestParameters {
bool flexible_mode;
- absl::optional<ScalabilityMode> scalability_mode;
+ std::optional<ScalabilityMode> scalability_mode;
std::string field_trial;
int expected_framedrop_mode;
int expected_max_consec_drop;
@@ -2508,7 +2507,7 @@
// controller is not enabled). Layer drop is not allowed.
SvcFrameDropConfigTestParameters{
.flexible_mode = true,
- .scalability_mode = absl::nullopt,
+ .scalability_mode = std::nullopt,
.expected_framedrop_mode = FULL_SUPERFRAME_DROP,
.expected_max_consec_drop = 2}));
diff --git a/modules/video_coding/decoder_database.cc b/modules/video_coding/decoder_database.cc
index dabef41..4dba5c3 100644
--- a/modules/video_coding/decoder_database.cc
+++ b/modules/video_coding/decoder_database.cc
@@ -34,7 +34,7 @@
// frame after RegisterReceiveCodec).
if (current_decoder_ && current_decoder_->IsSameDecoder(it->second.get())) {
// Release it if it was registered and in use.
- current_decoder_ = absl::nullopt;
+ current_decoder_ = std::nullopt;
}
decoders_.erase(it);
}
@@ -64,7 +64,7 @@
const VideoDecoder::Settings& settings) {
// If payload value already exists, erase old and insert new.
if (payload_type == current_payload_type_) {
- current_payload_type_ = absl::nullopt;
+ current_payload_type_ = std::nullopt;
}
decoder_settings_[payload_type] = settings;
}
@@ -75,13 +75,13 @@
}
if (payload_type == current_payload_type_) {
// This codec is currently in use.
- current_payload_type_ = absl::nullopt;
+ current_payload_type_ = std::nullopt;
}
return true;
}
void VCMDecoderDatabase::DeregisterReceiveCodecs() {
- current_payload_type_ = absl::nullopt;
+ current_payload_type_ = std::nullopt;
decoder_settings_.clear();
}
@@ -96,12 +96,12 @@
}
// If decoder exists - delete.
if (current_decoder_.has_value()) {
- current_decoder_ = absl::nullopt;
- current_payload_type_ = absl::nullopt;
+ current_decoder_ = std::nullopt;
+ current_payload_type_ = std::nullopt;
}
CreateAndInitDecoder(frame);
- if (current_decoder_ == absl::nullopt) {
+ if (current_decoder_ == std::nullopt) {
return nullptr;
}
@@ -109,7 +109,7 @@
callback->OnIncomingPayloadType(payload_type);
if (current_decoder_->RegisterDecodeCompleteCallback(decoded_frame_callback) <
0) {
- current_decoder_ = absl::nullopt;
+ current_decoder_ = std::nullopt;
return nullptr;
}
@@ -144,7 +144,7 @@
decoder_item->second.set_max_render_resolution(frame_resolution);
}
if (!current_decoder_->Configure(decoder_item->second)) {
- current_decoder_ = absl::nullopt;
+ current_decoder_ = std::nullopt;
RTC_LOG(LS_ERROR) << "Failed to initialize decoder.";
}
}
diff --git a/modules/video_coding/decoder_database.h b/modules/video_coding/decoder_database.h
index 87edcd0..951be7d 100644
--- a/modules/video_coding/decoder_database.h
+++ b/modules/video_coding/decoder_database.h
@@ -15,8 +15,8 @@
#include <map>
#include <memory>
+#include <optional>
-#include "absl/types/optional.h"
#include "api/sequence_checker.h"
#include "api/video/encoded_frame.h"
#include "api/video_codecs/video_decoder.h"
@@ -58,8 +58,8 @@
SequenceChecker decoder_sequence_checker_;
- absl::optional<uint8_t> current_payload_type_;
- absl::optional<VCMGenericDecoder> current_decoder_
+ std::optional<uint8_t> current_payload_type_;
+ std::optional<VCMGenericDecoder> current_decoder_
RTC_GUARDED_BY(decoder_sequence_checker_);
// Initialization paramaters for decoders keyed by payload type.
std::map<uint8_t, VideoDecoder::Settings> decoder_settings_;
diff --git a/modules/video_coding/deprecated/BUILD.gn b/modules/video_coding/deprecated/BUILD.gn
index dacde52..43c05fa 100644
--- a/modules/video_coding/deprecated/BUILD.gn
+++ b/modules/video_coding/deprecated/BUILD.gn
@@ -98,7 +98,6 @@
"../../../api/video:video_frame_type",
"../../../modules/rtp_rtcp:rtp_rtcp_format",
"../../../modules/rtp_rtcp:rtp_video_header",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
diff --git a/modules/video_coding/deprecated/jitter_buffer.cc b/modules/video_coding/deprecated/jitter_buffer.cc
index 66f4cea..eff9edd 100644
--- a/modules/video_coding/deprecated/jitter_buffer.cc
+++ b/modules/video_coding/deprecated/jitter_buffer.cc
@@ -575,7 +575,7 @@
uint32_t VCMJitterBuffer::EstimatedJitterMs() {
MutexLock lock(&mutex_);
const double rtt_mult = 1.0f;
- return jitter_estimate_.GetJitterEstimate(rtt_mult, absl::nullopt).ms();
+ return jitter_estimate_.GetJitterEstimate(rtt_mult, std::nullopt).ms();
}
void VCMJitterBuffer::SetNackSettings(size_t max_nack_list_size,
diff --git a/modules/video_coding/deprecated/packet.cc b/modules/video_coding/deprecated/packet.cc
index 110f38e..d89933e 100644
--- a/modules/video_coding/deprecated/packet.cc
+++ b/modules/video_coding/deprecated/packet.cc
@@ -59,7 +59,7 @@
// Playout decisions are made entirely based on first packet in a frame.
if (!is_first_packet_in_frame()) {
- video_header.playout_delay = absl::nullopt;
+ video_header.playout_delay = std::nullopt;
}
}
diff --git a/modules/video_coding/deprecated/packet.h b/modules/video_coding/deprecated/packet.h
index de69ff4..9a0e5ca 100644
--- a/modules/video_coding/deprecated/packet.h
+++ b/modules/video_coding/deprecated/packet.h
@@ -14,7 +14,8 @@
#include <stddef.h>
#include <stdint.h>
-#include "absl/types/optional.h"
+#include <optional>
+
#include "api/rtp_headers.h"
#include "api/rtp_packet_info.h"
#include "api/units/timestamp.h"
@@ -71,7 +72,7 @@
bool insertStartCode; // True if a start code should be inserted before this
// packet.
RTPVideoHeader video_header;
- absl::optional<RtpGenericFrameDescriptor> generic_descriptor;
+ std::optional<RtpGenericFrameDescriptor> generic_descriptor;
RtpPacketInfo packet_info;
};
diff --git a/modules/video_coding/deprecated/receiver.cc b/modules/video_coding/deprecated/receiver.cc
index a048f84..524b1cb 100644
--- a/modules/video_coding/deprecated/receiver.cc
+++ b/modules/video_coding/deprecated/receiver.cc
@@ -90,7 +90,7 @@
}
uint32_t frame_timestamp = found_frame->RtpTimestamp();
- if (absl::optional<VideoPlayoutDelay> playout_delay =
+ if (std::optional<VideoPlayoutDelay> playout_delay =
found_frame->EncodedImage().PlayoutDelay()) {
timing_->set_min_playout_delay(playout_delay->min());
timing_->set_max_playout_delay(playout_delay->max());
diff --git a/modules/video_coding/encoded_frame.cc b/modules/video_coding/encoded_frame.cc
index dbac39c..6bbd218 100644
--- a/modules/video_coding/encoded_frame.cc
+++ b/modules/video_coding/encoded_frame.cc
@@ -37,7 +37,7 @@
void VCMEncodedFrame::Reset() {
SetRtpTimestamp(0);
- SetSpatialIndex(absl::nullopt);
+ SetSpatialIndex(std::nullopt);
_renderTimeMs = -1;
_payloadType = 0;
_frameType = VideoFrameType::kVideoFrameDelta;
diff --git a/modules/video_coding/frame_dependencies_calculator.cc b/modules/video_coding/frame_dependencies_calculator.cc
index 7ca59f7..39793db 100644
--- a/modules/video_coding/frame_dependencies_calculator.cc
+++ b/modules/video_coding/frame_dependencies_calculator.cc
@@ -41,7 +41,7 @@
continue;
}
const BufferUsage& buffer = buffers_[buffer_usage.id];
- if (buffer.frame_id == absl::nullopt) {
+ if (buffer.frame_id == std::nullopt) {
RTC_LOG(LS_ERROR) << "Odd configuration: frame " << frame_id
<< " references buffer #" << buffer_usage.id
<< " that was never updated.";
diff --git a/modules/video_coding/frame_dependencies_calculator.h b/modules/video_coding/frame_dependencies_calculator.h
index 2c4a850..a296f21 100644
--- a/modules/video_coding/frame_dependencies_calculator.h
+++ b/modules/video_coding/frame_dependencies_calculator.h
@@ -13,10 +13,10 @@
#include <stdint.h>
+#include <optional>
#include <vector>
#include "absl/container/inlined_vector.h"
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "common_video/generic_frame_descriptor/generic_frame_info.h"
@@ -37,7 +37,7 @@
private:
struct BufferUsage {
- absl::optional<int64_t> frame_id;
+ std::optional<int64_t> frame_id;
absl::InlinedVector<int64_t, 4> dependencies;
};
diff --git a/modules/video_coding/generic_decoder.cc b/modules/video_coding/generic_decoder.cc
index 0bfc628..1f2c600 100644
--- a/modules/video_coding/generic_decoder.cc
+++ b/modules/video_coding/generic_decoder.cc
@@ -15,10 +15,10 @@
#include <algorithm>
#include <cmath>
#include <iterator>
+#include <optional>
#include <utility>
#include "absl/algorithm/container.h"
-#include "absl/types/optional.h"
#include "api/video/video_timing.h"
#include "api/video_codecs/video_decoder.h"
#include "modules/include/module_common_types_public.h"
@@ -73,15 +73,15 @@
int32_t VCMDecodedFrameCallback::Decoded(VideoFrame& decodedImage,
int64_t decode_time_ms) {
Decoded(decodedImage,
- decode_time_ms >= 0 ? absl::optional<int32_t>(decode_time_ms)
- : absl::nullopt,
- absl::nullopt);
+ decode_time_ms >= 0 ? std::optional<int32_t>(decode_time_ms)
+ : std::nullopt,
+ std::nullopt);
return WEBRTC_VIDEO_CODEC_OK;
}
-std::pair<absl::optional<FrameInfo>, size_t>
+std::pair<std::optional<FrameInfo>, size_t>
VCMDecodedFrameCallback::FindFrameInfo(uint32_t rtp_timestamp) {
- absl::optional<FrameInfo> frame_info;
+ std::optional<FrameInfo> frame_info;
auto it = absl::c_find_if(frame_infos_, [rtp_timestamp](const auto& entry) {
return entry.rtp_timestamp == rtp_timestamp ||
@@ -100,15 +100,15 @@
}
void VCMDecodedFrameCallback::Decoded(VideoFrame& decodedImage,
- absl::optional<int32_t> decode_time_ms,
- absl::optional<uint8_t> qp) {
+ std::optional<int32_t> decode_time_ms,
+ std::optional<uint8_t> qp) {
RTC_DCHECK(_receiveCallback) << "Callback must not be null at this point";
TRACE_EVENT(
"webrtc", "VCMDecodedFrameCallback::Decoded",
perfetto::TerminatingFlow::ProcessScoped(decodedImage.rtp_timestamp()));
// TODO(holmer): We should improve this so that we can handle multiple
// callbacks from one call to Decode().
- absl::optional<FrameInfo> frame_info;
+ std::optional<FrameInfo> frame_info;
int timestamp_map_size = 0;
int dropped_frames = 0;
{
@@ -300,8 +300,8 @@
frame_info.decode_start = now;
frame_info.render_time =
render_time_ms >= 0
- ? absl::make_optional(Timestamp::Millis(render_time_ms))
- : absl::nullopt;
+ ? std::make_optional(Timestamp::Millis(render_time_ms))
+ : std::nullopt;
frame_info.rotation = frame.rotation();
frame_info.timing = frame.video_timing();
frame_info.ntp_time_ms = frame.ntp_time_ms_;
diff --git a/modules/video_coding/generic_decoder.h b/modules/video_coding/generic_decoder.h
index b1fb1f3..4859031 100644
--- a/modules/video_coding/generic_decoder.h
+++ b/modules/video_coding/generic_decoder.h
@@ -39,8 +39,8 @@
// This is likely not optional, but some inputs seem to sometimes be negative.
// TODO(bugs.webrtc.org/13756): See if this can be replaced with Timestamp
// once all inputs to this field use Timestamp instead of an integer.
- absl::optional<Timestamp> render_time;
- absl::optional<Timestamp> decode_start;
+ std::optional<Timestamp> render_time;
+ std::optional<Timestamp> decode_start;
VideoRotation rotation;
VideoContentType content_type;
EncodedImage::Timing timing;
@@ -62,8 +62,8 @@
int32_t Decoded(VideoFrame& decodedImage) override;
int32_t Decoded(VideoFrame& decodedImage, int64_t decode_time_ms) override;
void Decoded(VideoFrame& decodedImage,
- absl::optional<int32_t> decode_time_ms,
- absl::optional<uint8_t> qp) override;
+ std::optional<int32_t> decode_time_ms,
+ std::optional<uint8_t> qp) override;
void OnDecoderInfoChanged(const VideoDecoder::DecoderInfo& decoder_info);
@@ -71,7 +71,7 @@
void ClearTimestampMap();
private:
- std::pair<absl::optional<FrameInfo>, size_t> FindFrameInfo(
+ std::pair<std::optional<FrameInfo>, size_t> FindFrameInfo(
uint32_t rtp_timestamp) RTC_EXCLUSIVE_LOCKS_REQUIRED(lock_);
SequenceChecker construction_thread_;
diff --git a/modules/video_coding/generic_decoder_unittest.cc b/modules/video_coding/generic_decoder_unittest.cc
index 1142304..7edc961 100644
--- a/modules/video_coding/generic_decoder_unittest.cc
+++ b/modules/video_coding/generic_decoder_unittest.cc
@@ -12,9 +12,9 @@
#include <cstdint>
#include <memory>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/rtp_packet_infos.h"
#include "api/video_codecs/video_decoder.h"
@@ -33,7 +33,7 @@
class ReceiveCallback : public VCMReceiveCallback {
public:
int32_t FrameToRender(VideoFrame& frame,
- absl::optional<uint8_t> qp,
+ std::optional<uint8_t> qp,
TimeDelta decode_time,
VideoContentType content_type,
VideoFrameType frame_type) override {
@@ -41,9 +41,9 @@
return 0;
}
- absl::optional<VideoFrame> PopLastFrame() {
+ std::optional<VideoFrame> PopLastFrame() {
if (frames_.empty())
- return absl::nullopt;
+ return std::nullopt;
auto ret = frames_.front();
frames_.pop_back();
return ret;
@@ -98,7 +98,7 @@
encoded_frame.SetPacketInfos(packet_infos);
generic_decoder_.Decode(encoded_frame, clock_->CurrentTime());
time_controller_.AdvanceTime(TimeDelta::Millis(10));
- absl::optional<VideoFrame> decoded_frame = user_callback_.PopLastFrame();
+ std::optional<VideoFrame> decoded_frame = user_callback_.PopLastFrame();
ASSERT_TRUE(decoded_frame.has_value());
EXPECT_EQ(decoded_frame->packet_infos().size(), 3U);
}
@@ -134,7 +134,7 @@
}
time_controller_.AdvanceTime(TimeDelta::Millis(200));
- absl::optional<VideoFrame> decoded_frame = user_callback_.PopLastFrame();
+ std::optional<VideoFrame> decoded_frame = user_callback_.PopLastFrame();
ASSERT_TRUE(decoded_frame.has_value());
EXPECT_EQ(decoded_frame->packet_infos().size(), 3U);
}
@@ -143,11 +143,11 @@
EncodedFrame encoded_frame;
generic_decoder_.Decode(encoded_frame, clock_->CurrentTime());
time_controller_.AdvanceTime(TimeDelta::Millis(10));
- absl::optional<VideoFrame> decoded_frame = user_callback_.PopLastFrame();
+ std::optional<VideoFrame> decoded_frame = user_callback_.PopLastFrame();
ASSERT_TRUE(decoded_frame.has_value());
EXPECT_THAT(
decoded_frame->render_parameters().max_composition_delay_in_frames,
- testing::Eq(absl::nullopt));
+ testing::Eq(std::nullopt));
}
TEST_F(GenericDecoderTest, MaxCompositionDelayActivatedByPlayoutDelay) {
@@ -156,10 +156,10 @@
// is specified as X,Y, where X=0, Y>0.
constexpr int kMaxCompositionDelayInFrames = 3; // ~50 ms at 60 fps.
timing_.SetMaxCompositionDelayInFrames(
- absl::make_optional(kMaxCompositionDelayInFrames));
+ std::make_optional(kMaxCompositionDelayInFrames));
generic_decoder_.Decode(encoded_frame, clock_->CurrentTime());
time_controller_.AdvanceTime(TimeDelta::Millis(10));
- absl::optional<VideoFrame> decoded_frame = user_callback_.PopLastFrame();
+ std::optional<VideoFrame> decoded_frame = user_callback_.PopLastFrame();
ASSERT_TRUE(decoded_frame.has_value());
EXPECT_THAT(
decoded_frame->render_parameters().max_composition_delay_in_frames,
@@ -170,7 +170,7 @@
EncodedFrame encoded_frame;
generic_decoder_.Decode(encoded_frame, clock_->CurrentTime());
time_controller_.AdvanceTime(TimeDelta::Millis(10));
- absl::optional<VideoFrame> decoded_frame = user_callback_.PopLastFrame();
+ std::optional<VideoFrame> decoded_frame = user_callback_.PopLastFrame();
ASSERT_TRUE(decoded_frame.has_value());
EXPECT_FALSE(decoded_frame->render_parameters().use_low_latency_rendering);
}
@@ -183,7 +183,7 @@
timing_.set_max_playout_delay(kPlayoutDelay.max());
generic_decoder_.Decode(encoded_frame, clock_->CurrentTime());
time_controller_.AdvanceTime(TimeDelta::Millis(10));
- absl::optional<VideoFrame> decoded_frame = user_callback_.PopLastFrame();
+ std::optional<VideoFrame> decoded_frame = user_callback_.PopLastFrame();
ASSERT_TRUE(decoded_frame.has_value());
EXPECT_TRUE(decoded_frame->render_parameters().use_low_latency_rendering);
}
diff --git a/modules/video_coding/h264_sps_pps_tracker.cc b/modules/video_coding/h264_sps_pps_tracker.cc
index 834a36d..0d7931c 100644
--- a/modules/video_coding/h264_sps_pps_tracker.cc
+++ b/modules/video_coding/h264_sps_pps_tracker.cc
@@ -216,9 +216,9 @@
RTC_LOG(LS_WARNING) << "SPS Nalu header missing";
return;
}
- absl::optional<SpsParser::SpsState> parsed_sps = SpsParser::ParseSps(
+ std::optional<SpsParser::SpsState> parsed_sps = SpsParser::ParseSps(
rtc::ArrayView<const uint8_t>(sps).subview(kNaluHeaderOffset));
- absl::optional<PpsParser::PpsState> parsed_pps = PpsParser::ParsePps(
+ std::optional<PpsParser::PpsState> parsed_pps = PpsParser::ParsePps(
rtc::ArrayView<const uint8_t>(pps).subview(kNaluHeaderOffset));
if (!parsed_sps) {
diff --git a/modules/video_coding/h26x_packet_buffer.cc b/modules/video_coding/h26x_packet_buffer.cc
index e9f6353..9020bb7 100644
--- a/modules/video_coding/h26x_packet_buffer.cc
+++ b/modules/video_coding/h26x_packet_buffer.cc
@@ -338,9 +338,9 @@
RTC_LOG(LS_WARNING) << "SPS Nalu header missing";
return;
}
- absl::optional<SpsParser::SpsState> parsed_sps = SpsParser::ParseSps(
+ std::optional<SpsParser::SpsState> parsed_sps = SpsParser::ParseSps(
rtc::ArrayView<const uint8_t>(sps).subview(kNaluHeaderOffset));
- absl::optional<PpsParser::PpsState> parsed_pps = PpsParser::ParsePps(
+ std::optional<PpsParser::PpsState> parsed_pps = PpsParser::ParsePps(
rtc::ArrayView<const uint8_t>(pps).subview(kNaluHeaderOffset));
if (!parsed_sps) {
diff --git a/modules/video_coding/h26x_packet_buffer.h b/modules/video_coding/h26x_packet_buffer.h
index f199341..62b97d3 100644
--- a/modules/video_coding/h26x_packet_buffer.h
+++ b/modules/video_coding/h26x_packet_buffer.h
@@ -14,11 +14,11 @@
#include <array>
#include <map>
#include <memory>
+#include <optional>
#include <string>
#include <vector>
#include "absl/base/attributes.h"
-#include "absl/types/optional.h"
#include "modules/video_coding/packet_buffer.h"
#include "rtc_base/numerics/sequence_number_unwrapper.h"
diff --git a/modules/video_coding/include/video_codec_interface.h b/modules/video_coding/include/video_codec_interface.h
index 987e1b6..243ac07 100644
--- a/modules/video_coding/include/video_codec_interface.h
+++ b/modules/video_coding/include/video_codec_interface.h
@@ -11,9 +11,9 @@
#ifndef MODULES_VIDEO_CODING_INCLUDE_VIDEO_CODEC_INTERFACE_H_
#define MODULES_VIDEO_CODING_INCLUDE_VIDEO_CODEC_INTERFACE_H_
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/video/video_frame.h"
#include "api/video_codecs/scalability_mode.h"
#include "api/video_codecs/video_decoder.h"
@@ -116,9 +116,9 @@
VideoCodecType codecType;
CodecSpecificInfoUnion codecSpecific;
bool end_of_picture = true;
- absl::optional<GenericFrameInfo> generic_frame_info;
- absl::optional<FrameDependencyStructure> template_structure;
- absl::optional<ScalabilityMode> scalability_mode;
+ std::optional<GenericFrameInfo> generic_frame_info;
+ std::optional<FrameDependencyStructure> template_structure;
+ std::optional<ScalabilityMode> scalability_mode;
};
} // namespace webrtc
diff --git a/modules/video_coding/include/video_coding_defines.h b/modules/video_coding/include/video_coding_defines.h
index cbd4207..4821d5e 100644
--- a/modules/video_coding/include/video_coding_defines.h
+++ b/modules/video_coding/include/video_coding_defines.h
@@ -14,7 +14,8 @@
#include <stddef.h>
#include <stdint.h>
-#include "absl/types/optional.h"
+#include <optional>
+
#include "api/video/video_content_type.h"
#include "api/video/video_frame.h"
#include "api/video/video_timing.h"
@@ -51,7 +52,7 @@
class VCMReceiveCallback {
public:
virtual int32_t FrameToRender(VideoFrame& videoFrame, // NOLINT
- absl::optional<uint8_t> qp,
+ std::optional<uint8_t> qp,
TimeDelta decode_time,
VideoContentType content_type,
VideoFrameType frame_type) = 0;
diff --git a/modules/video_coding/loss_notification_controller.h b/modules/video_coding/loss_notification_controller.h
index ecba412..d777884 100644
--- a/modules/video_coding/loss_notification_controller.h
+++ b/modules/video_coding/loss_notification_controller.h
@@ -13,9 +13,9 @@
#include <stdint.h>
+#include <optional>
#include <set>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/sequence_checker.h"
#include "modules/include/module_common_types.h"
@@ -75,11 +75,11 @@
RTC_GUARDED_BY(sequence_checker_);
// Tracked to avoid processing repeated frames (buggy/malicious remote).
- absl::optional<int64_t> last_received_frame_id_
+ std::optional<int64_t> last_received_frame_id_
RTC_GUARDED_BY(sequence_checker_);
// Tracked to avoid processing repeated packets.
- absl::optional<uint16_t> last_received_seq_num_
+ std::optional<uint16_t> last_received_seq_num_
RTC_GUARDED_BY(sequence_checker_);
// Tracked in order to correctly report the potential-decodability of
@@ -95,7 +95,7 @@
explicit FrameInfo(uint16_t first_seq_num) : first_seq_num(first_seq_num) {}
uint16_t first_seq_num;
};
- absl::optional<FrameInfo> last_decodable_non_discardable_
+ std::optional<FrameInfo> last_decodable_non_discardable_
RTC_GUARDED_BY(sequence_checker_);
// Track which frames are decodable. Later frames are also decodable if
diff --git a/modules/video_coding/loss_notification_controller_unittest.cc b/modules/video_coding/loss_notification_controller_unittest.cc
index 9c4e715..83beb63 100644
--- a/modules/video_coding/loss_notification_controller_unittest.cc
+++ b/modules/video_coding/loss_notification_controller_unittest.cc
@@ -13,12 +13,12 @@
#include <stdint.h>
#include <limits>
+#include <optional>
#include <string>
#include <tuple>
#include <utility>
#include <vector>
-#include "absl/types/optional.h"
#include "test/gtest.h"
namespace webrtc {
@@ -138,7 +138,7 @@
}
void ExpectKeyFrameRequest() {
- EXPECT_EQ(LastLossNotification(), absl::nullopt);
+ EXPECT_EQ(LastLossNotification(), std::nullopt);
EXPECT_TRUE(LastKeyFrameRequest());
}
@@ -188,9 +188,9 @@
return result;
}
- absl::optional<LossNotification> LastLossNotification() {
- const absl::optional<LossNotification> result = last_loss_notification_;
- last_loss_notification_ = absl::nullopt;
+ std::optional<LossNotification> LastLossNotification() {
+ const std::optional<LossNotification> result = last_loss_notification_;
+ last_loss_notification_ = std::nullopt;
return result;
}
@@ -198,13 +198,13 @@
bool key_frame_requested_;
- absl::optional<LossNotification> last_loss_notification_;
+ std::optional<LossNotification> last_loss_notification_;
// First packet of last frame. (Note that if a test skips the first packet
// of a subsequent frame, OnAssembledFrame is not called, and so this is
// note read. Therefore, it's not a problem if it is not cleared when
// the frame changes.)
- absl::optional<Packet> previous_first_packet_in_frame_;
+ std::optional<Packet> previous_first_packet_in_frame_;
};
class LossNotificationControllerTest
diff --git a/modules/video_coding/packet_buffer.h b/modules/video_coding/packet_buffer.h
index 9e143b8..495f64b 100644
--- a/modules/video_coding/packet_buffer.h
+++ b/modules/video_coding/packet_buffer.h
@@ -117,7 +117,7 @@
// determine continuity between them.
std::vector<std::unique_ptr<Packet>> buffer_;
- absl::optional<uint16_t> newest_inserted_seq_num_;
+ std::optional<uint16_t> newest_inserted_seq_num_;
std::set<uint16_t, DescendingSeqNumComp<uint16_t>> missing_packets_;
std::set<uint16_t, DescendingSeqNumComp<uint16_t>> received_padding_;
diff --git a/modules/video_coding/rtp_frame_reference_finder_unittest.cc b/modules/video_coding/rtp_frame_reference_finder_unittest.cc
index 0ca2cf0..8ade214 100644
--- a/modules/video_coding/rtp_frame_reference_finder_unittest.cc
+++ b/modules/video_coding/rtp_frame_reference_finder_unittest.cc
@@ -54,7 +54,7 @@
kVideoRotation_0,
VideoContentType::UNSPECIFIED,
video_header,
- /*color_space=*/absl::nullopt,
+ /*color_space=*/std::nullopt,
RtpPacketInfos(),
EncodedImageBuffer::Create(/*size=*/0));
// clang-format on
diff --git a/modules/video_coding/rtp_vp8_ref_finder_unittest.cc b/modules/video_coding/rtp_vp8_ref_finder_unittest.cc
index a27085e..01ff233 100644
--- a/modules/video_coding/rtp_vp8_ref_finder_unittest.cc
+++ b/modules/video_coding/rtp_vp8_ref_finder_unittest.cc
@@ -92,7 +92,7 @@
kVideoRotation_0,
VideoContentType::UNSPECIFIED,
video_header,
- /*color_space=*/absl::nullopt,
+ /*color_space=*/std::nullopt,
RtpPacketInfos(),
EncodedImageBuffer::Create(/*size=*/0));
// clang-format on
@@ -100,9 +100,9 @@
private:
bool is_keyframe_ = false;
- absl::optional<int> picture_id_;
- absl::optional<int> temporal_id_;
- absl::optional<int> tl0_idx_;
+ std::optional<int> picture_id_;
+ std::optional<int> temporal_id_;
+ std::optional<int> tl0_idx_;
bool sync = false;
};
diff --git a/modules/video_coding/rtp_vp9_ref_finder_unittest.cc b/modules/video_coding/rtp_vp9_ref_finder_unittest.cc
index 23b6cfc..35d30ad 100644
--- a/modules/video_coding/rtp_vp9_ref_finder_unittest.cc
+++ b/modules/video_coding/rtp_vp9_ref_finder_unittest.cc
@@ -128,7 +128,7 @@
kVideoRotation_0,
VideoContentType::UNSPECIFIED,
video_header,
- /*color_space=*/absl::nullopt,
+ /*color_space=*/std::nullopt,
RtpPacketInfos(),
EncodedImageBuffer::Create(/*size=*/0));
// clang-format on
@@ -138,10 +138,10 @@
uint16_t seq_num_start = 0;
uint16_t seq_num_end = 0;
bool keyframe = false;
- absl::optional<int> picture_id;
- absl::optional<int> spatial_id;
- absl::optional<int> temporal_id;
- absl::optional<int> tl0_idx;
+ std::optional<int> picture_id;
+ std::optional<int> spatial_id;
+ std::optional<int> temporal_id;
+ std::optional<int> tl0_idx;
bool up_switch = false;
bool inter_layer = false;
bool inter_pic = true;
diff --git a/modules/video_coding/svc/BUILD.gn b/modules/video_coding/svc/BUILD.gn
index b80a5e4..79ebec7 100644
--- a/modules/video_coding/svc/BUILD.gn
+++ b/modules/video_coding/svc/BUILD.gn
@@ -19,7 +19,6 @@
"../../../rtc_base:checks",
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -61,7 +60,6 @@
"../../../rtc_base:logging",
"//third_party/abseil-cpp/absl/base:core_headers",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -109,7 +107,6 @@
"../../../rtc_base:stringutils",
"../../../test:test_support",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
diff --git a/modules/video_coding/svc/create_scalability_structure.cc b/modules/video_coding/svc/create_scalability_structure.cc
index fbcd27b..33370c7 100644
--- a/modules/video_coding/svc/create_scalability_structure.cc
+++ b/modules/video_coding/svc/create_scalability_structure.cc
@@ -282,14 +282,14 @@
return nullptr;
}
-absl::optional<ScalableVideoController::StreamLayersConfig>
+std::optional<ScalableVideoController::StreamLayersConfig>
ScalabilityStructureConfig(ScalabilityMode name) {
for (const auto& entry : kFactories) {
if (entry.name == name) {
return entry.config;
}
}
- return absl::nullopt;
+ return std::nullopt;
}
} // namespace webrtc
diff --git a/modules/video_coding/svc/create_scalability_structure.h b/modules/video_coding/svc/create_scalability_structure.h
index 3b67443..2544b66 100644
--- a/modules/video_coding/svc/create_scalability_structure.h
+++ b/modules/video_coding/svc/create_scalability_structure.h
@@ -11,9 +11,9 @@
#define MODULES_VIDEO_CODING_SVC_CREATE_SCALABILITY_STRUCTURE_H_
#include <memory>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/video_codecs/scalability_mode.h"
#include "modules/video_coding/svc/scalable_video_controller.h"
@@ -27,7 +27,7 @@
// Returns description of the scalability structure identified by 'name',
// Return nullopt for unknown name.
-absl::optional<ScalableVideoController::StreamLayersConfig>
+std::optional<ScalableVideoController::StreamLayersConfig>
ScalabilityStructureConfig(ScalabilityMode name);
} // namespace webrtc
diff --git a/modules/video_coding/svc/scalability_mode_util.cc b/modules/video_coding/svc/scalability_mode_util.cc
index 580500f..9ee30f4 100644
--- a/modules/video_coding/svc/scalability_mode_util.cc
+++ b/modules/video_coding/svc/scalability_mode_util.cc
@@ -12,11 +12,11 @@
#include <array>
#include <cstddef>
+#include <optional>
#include <type_traits>
#include "absl/algorithm/container.h"
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/video_codecs/scalability_mode.h"
#include "api/video_codecs/video_codec.h"
#include "rtc_base/checks.h"
@@ -31,7 +31,7 @@
const int num_spatial_layers;
const int num_temporal_layers;
const InterLayerPredMode inter_layer_pred;
- const absl::optional<ScalabilityModeResolutionRatio> ratio =
+ const std::optional<ScalabilityModeResolutionRatio> ratio =
ScalabilityModeResolutionRatio::kTwoToOne;
const bool shift = false;
};
@@ -45,19 +45,19 @@
.num_spatial_layers = 1,
.num_temporal_layers = 1,
.inter_layer_pred = InterLayerPredMode::kOff,
- .ratio = absl::nullopt},
+ .ratio = std::nullopt},
ScalabilityModeParameters{.scalability_mode = ScalabilityMode::kL1T2,
.name = "L1T2",
.num_spatial_layers = 1,
.num_temporal_layers = 2,
.inter_layer_pred = InterLayerPredMode::kOff,
- .ratio = absl::nullopt},
+ .ratio = std::nullopt},
ScalabilityModeParameters{.scalability_mode = ScalabilityMode::kL1T3,
.name = "L1T3",
.num_spatial_layers = 1,
.num_temporal_layers = 3,
.inter_layer_pred = InterLayerPredMode::kOff,
- .ratio = absl::nullopt},
+ .ratio = std::nullopt},
ScalabilityModeParameters{
.scalability_mode = ScalabilityMode::kL2T1,
.name = "L2T1",
@@ -312,11 +312,11 @@
} // namespace
-absl::optional<ScalabilityMode> MakeScalabilityMode(
+std::optional<ScalabilityMode> MakeScalabilityMode(
int num_spatial_layers,
int num_temporal_layers,
InterLayerPredMode inter_layer_pred,
- absl::optional<ScalabilityModeResolutionRatio> ratio,
+ std::optional<ScalabilityModeResolutionRatio> ratio,
bool shift) {
for (const auto& candidate_mode : kScalabilityModeParams) {
if (candidate_mode.num_spatial_layers == num_spatial_layers &&
@@ -328,10 +328,10 @@
}
}
}
- return absl::nullopt;
+ return std::nullopt;
}
-absl::optional<ScalabilityMode> ScalabilityModeFromString(
+std::optional<ScalabilityMode> ScalabilityModeFromString(
absl::string_view mode_string) {
const auto it =
absl::c_find_if(kScalabilityModeParams,
@@ -341,7 +341,7 @@
if (it != std::end(kScalabilityModeParams)) {
return it->scalability_mode;
}
- return absl::nullopt;
+ return std::nullopt;
}
InterLayerPredMode ScalabilityModeToInterLayerPredMode(
@@ -357,7 +357,7 @@
return kScalabilityModeParams[Idx(scalability_mode)].num_temporal_layers;
}
-absl::optional<ScalabilityModeResolutionRatio> ScalabilityModeToResolutionRatio(
+std::optional<ScalabilityModeResolutionRatio> ScalabilityModeToResolutionRatio(
ScalabilityMode scalability_mode) {
return kScalabilityModeParams[Idx(scalability_mode)].ratio;
}
diff --git a/modules/video_coding/svc/scalability_mode_util.h b/modules/video_coding/svc/scalability_mode_util.h
index cfc247b..4e80796 100644
--- a/modules/video_coding/svc/scalability_mode_util.h
+++ b/modules/video_coding/svc/scalability_mode_util.h
@@ -11,8 +11,9 @@
#ifndef MODULES_VIDEO_CODING_SVC_SCALABILITY_MODE_UTIL_H_
#define MODULES_VIDEO_CODING_SVC_SCALABILITY_MODE_UTIL_H_
+#include <optional>
+
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/video_codecs/scalability_mode.h"
#include "api/video_codecs/video_codec.h"
@@ -25,14 +26,14 @@
static constexpr char kDefaultScalabilityModeStr[] = "L1T2";
-RTC_EXPORT absl::optional<ScalabilityMode> MakeScalabilityMode(
+RTC_EXPORT std::optional<ScalabilityMode> MakeScalabilityMode(
int num_spatial_layers,
int num_temporal_layers,
InterLayerPredMode inter_layer_pred,
- absl::optional<ScalabilityModeResolutionRatio> ratio,
+ std::optional<ScalabilityModeResolutionRatio> ratio,
bool shift);
-absl::optional<ScalabilityMode> ScalabilityModeFromString(
+std::optional<ScalabilityMode> ScalabilityModeFromString(
absl::string_view scalability_mode_string);
InterLayerPredMode ScalabilityModeToInterLayerPredMode(
@@ -42,7 +43,7 @@
int ScalabilityModeToNumTemporalLayers(ScalabilityMode scalability_mode);
-absl::optional<ScalabilityModeResolutionRatio> ScalabilityModeToResolutionRatio(
+std::optional<ScalabilityModeResolutionRatio> ScalabilityModeToResolutionRatio(
ScalabilityMode scalability_mode);
bool ScalabilityModeIsShiftMode(ScalabilityMode scalability_mode);
diff --git a/modules/video_coding/svc/scalability_mode_util_unittest.cc b/modules/video_coding/svc/scalability_mode_util_unittest.cc
index b023a12..346944c 100644
--- a/modules/video_coding/svc/scalability_mode_util_unittest.cc
+++ b/modules/video_coding/svc/scalability_mode_util_unittest.cc
@@ -10,12 +10,12 @@
#include "modules/video_coding/svc/scalability_mode_util.h"
+#include <optional>
#include <string>
#include <tuple>
#include <vector>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/video_codecs/scalability_mode.h"
#include "test/gmock.h"
#include "test/gtest.h"
@@ -30,8 +30,8 @@
}
TEST(ScalabilityModeUtil, RejectsUnknownString) {
- EXPECT_EQ(ScalabilityModeFromString(""), absl::nullopt);
- EXPECT_EQ(ScalabilityModeFromString("not-a-mode"), absl::nullopt);
+ EXPECT_EQ(ScalabilityModeFromString(""), std::nullopt);
+ EXPECT_EQ(ScalabilityModeFromString("not-a-mode"), std::nullopt);
}
TEST(ScalabilityModeUtil, MakeScalabilityModeRoundTrip) {
@@ -40,7 +40,7 @@
numerical_enum++) {
ScalabilityMode scalability_mode =
static_cast<ScalabilityMode>(numerical_enum);
- absl::optional<ScalabilityMode> created_mode = MakeScalabilityMode(
+ std::optional<ScalabilityMode> created_mode = MakeScalabilityMode(
ScalabilityModeToNumSpatialLayers(scalability_mode),
ScalabilityModeToNumTemporalLayers(scalability_mode),
ScalabilityModeToInterLayerPredMode(scalability_mode),
diff --git a/modules/video_coding/svc/scalability_structure_full_svc.cc b/modules/video_coding/svc/scalability_structure_full_svc.cc
index a262317..d440b23 100644
--- a/modules/video_coding/svc/scalability_structure_full_svc.cc
+++ b/modules/video_coding/svc/scalability_structure_full_svc.cc
@@ -9,11 +9,11 @@
*/
#include "modules/video_coding/svc/scalability_structure_full_svc.h"
+#include <optional>
#include <utility>
#include <vector>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/transport/rtp/dependency_descriptor.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
@@ -142,7 +142,7 @@
}
FramePattern current_pattern = NextPattern();
- absl::optional<int> spatial_dependency_buffer_id;
+ std::optional<int> spatial_dependency_buffer_id;
switch (current_pattern) {
case kDeltaT0:
case kKey:
diff --git a/modules/video_coding/svc/scalability_structure_key_svc.cc b/modules/video_coding/svc/scalability_structure_key_svc.cc
index 0e6fecf..4e771bb 100644
--- a/modules/video_coding/svc/scalability_structure_key_svc.cc
+++ b/modules/video_coding/svc/scalability_structure_key_svc.cc
@@ -10,10 +10,10 @@
#include "modules/video_coding/svc/scalability_structure_key_svc.h"
#include <bitset>
+#include <optional>
#include <utility>
#include <vector>
-#include "absl/types/optional.h"
#include "api/transport/rtp/dependency_descriptor.h"
#include "api/video/video_bitrate_allocation.h"
#include "common_video/generic_frame_descriptor/generic_frame_info.h"
@@ -90,7 +90,7 @@
ScalabilityStructureKeySvc::KeyframeConfig() {
std::vector<LayerFrameConfig> configs;
configs.reserve(num_spatial_layers_);
- absl::optional<int> spatial_dependency_buffer_id;
+ std::optional<int> spatial_dependency_buffer_id;
spatial_id_is_enabled_.reset();
// Disallow temporal references cross T0 on higher temporal layers.
can_reference_t1_frame_for_spatial_id_.reset();
diff --git a/modules/video_coding/svc/scalability_structure_unittest.cc b/modules/video_coding/svc/scalability_structure_unittest.cc
index 2d517c5..9c350ed 100644
--- a/modules/video_coding/svc/scalability_structure_unittest.cc
+++ b/modules/video_coding/svc/scalability_structure_unittest.cc
@@ -12,10 +12,10 @@
#include <stdint.h>
#include <memory>
+#include <optional>
#include <ostream>
#include <string>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/transport/rtp/dependency_descriptor.h"
#include "modules/video_coding/svc/create_scalability_structure.h"
@@ -85,7 +85,7 @@
}
ScalabilityMode GetScalabilityMode() const {
- absl::optional<ScalabilityMode> scalability_mode =
+ std::optional<ScalabilityMode> scalability_mode =
ScalabilityModeFromString(name);
RTC_CHECK(scalability_mode.has_value());
return *scalability_mode;
@@ -101,10 +101,10 @@
StaticConfigMatchesConfigReturnedByController) {
std::unique_ptr<ScalableVideoController> controller =
CreateScalabilityStructure(GetParam().GetScalabilityMode());
- absl::optional<ScalableVideoController::StreamLayersConfig> static_config =
+ std::optional<ScalableVideoController::StreamLayersConfig> static_config =
ScalabilityStructureConfig(GetParam().GetScalabilityMode());
ASSERT_THAT(controller, NotNull());
- ASSERT_NE(static_config, absl::nullopt);
+ ASSERT_NE(static_config, std::nullopt);
ScalableVideoController::StreamLayersConfig config =
controller->StreamConfig();
EXPECT_EQ(config.num_spatial_layers, static_config->num_spatial_layers);
diff --git a/modules/video_coding/svc/svc_rate_allocator.cc b/modules/video_coding/svc/svc_rate_allocator.cc
index f3514a1..0f29fca 100644
--- a/modules/video_coding/svc/svc_rate_allocator.cc
+++ b/modules/video_coding/svc/svc_rate_allocator.cc
@@ -174,7 +174,7 @@
SvcRateAllocator::NumLayers SvcRateAllocator::GetNumLayers(
const VideoCodec& codec) {
NumLayers layers;
- if (absl::optional<ScalabilityMode> scalability_mode =
+ if (std::optional<ScalabilityMode> scalability_mode =
codec.GetScalabilityMode();
scalability_mode.has_value()) {
if (auto structure = CreateScalabilityStructure(*scalability_mode)) {
diff --git a/modules/video_coding/timing/BUILD.gn b/modules/video_coding/timing/BUILD.gn
index 2cace29..f6fdc54 100644
--- a/modules/video_coding/timing/BUILD.gn
+++ b/modules/video_coding/timing/BUILD.gn
@@ -27,7 +27,6 @@
"../../../api/units:time_delta",
"../../../api/units:timestamp",
"../../../rtc_base:rtc_numerics",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -67,7 +66,6 @@
"../../../rtc_base/experiments:field_trial_parser",
"../../../system_wrappers",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -92,7 +90,6 @@
"../../../api/units:timestamp",
"../../../modules:module_api_public",
"../../../rtc_base:rtc_numerics",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -114,7 +111,6 @@
"../../../rtc_base/experiments:field_trial_parser",
"../../../rtc_base/synchronization:mutex",
"../../../system_wrappers",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -146,6 +142,5 @@
"../../../system_wrappers:system_wrappers",
"../../../test:scoped_key_value_config",
"../../../test:test_support",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
diff --git a/modules/video_coding/timing/inter_frame_delay_variation_calculator.cc b/modules/video_coding/timing/inter_frame_delay_variation_calculator.cc
index 69bc916..3b464ee 100644
--- a/modules/video_coding/timing/inter_frame_delay_variation_calculator.cc
+++ b/modules/video_coding/timing/inter_frame_delay_variation_calculator.cc
@@ -10,7 +10,8 @@
#include "modules/video_coding/timing/inter_frame_delay_variation_calculator.h"
-#include "absl/types/optional.h"
+#include <optional>
+
#include "api/units/frequency.h"
#include "api/units/time_delta.h"
#include "modules/include/module_common_types_public.h"
@@ -26,11 +27,11 @@
}
void InterFrameDelayVariationCalculator::Reset() {
- prev_wall_clock_ = absl::nullopt;
+ prev_wall_clock_ = std::nullopt;
prev_rtp_timestamp_unwrapped_ = 0;
}
-absl::optional<TimeDelta> InterFrameDelayVariationCalculator::Calculate(
+std::optional<TimeDelta> InterFrameDelayVariationCalculator::Calculate(
uint32_t rtp_timestamp,
Timestamp now) {
int64_t rtp_timestamp_unwrapped = unwrapper_.Unwrap(rtp_timestamp);
@@ -39,7 +40,7 @@
prev_wall_clock_ = now;
prev_rtp_timestamp_unwrapped_ = rtp_timestamp_unwrapped;
// Inter-frame delay variation is undefined for a single frame.
- // TODO(brandtr): Should this return absl::nullopt instead?
+ // TODO(brandtr): Should this return std::nullopt instead?
return TimeDelta::Zero();
}
@@ -49,7 +50,7 @@
uint32_t cropped_prev = static_cast<uint32_t>(prev_rtp_timestamp_unwrapped_);
if (rtp_timestamp_unwrapped < prev_rtp_timestamp_unwrapped_ ||
!IsNewerTimestamp(rtp_timestamp, cropped_prev)) {
- return absl::nullopt;
+ return std::nullopt;
}
// Compute the compensated timestamp difference.
diff --git a/modules/video_coding/timing/inter_frame_delay_variation_calculator.h b/modules/video_coding/timing/inter_frame_delay_variation_calculator.h
index c1d3d20..d8a8c83 100644
--- a/modules/video_coding/timing/inter_frame_delay_variation_calculator.h
+++ b/modules/video_coding/timing/inter_frame_delay_variation_calculator.h
@@ -13,7 +13,8 @@
#include <stdint.h>
-#include "absl/types/optional.h"
+#include <optional>
+
#include "api/units/time_delta.h"
#include "api/units/timestamp.h"
#include "rtc_base/numerics/sequence_number_unwrapper.h"
@@ -32,11 +33,11 @@
// Calculates the inter-frame delay variation of a frame with the given
// RTP timestamp. This method is called when the frame is complete.
- absl::optional<TimeDelta> Calculate(uint32_t rtp_timestamp, Timestamp now);
+ std::optional<TimeDelta> Calculate(uint32_t rtp_timestamp, Timestamp now);
private:
// The previous wall clock timestamp used in the calculation.
- absl::optional<Timestamp> prev_wall_clock_;
+ std::optional<Timestamp> prev_wall_clock_;
// The previous RTP timestamp used in the calculation.
int64_t prev_rtp_timestamp_unwrapped_;
diff --git a/modules/video_coding/timing/inter_frame_delay_variation_calculator_unittest.cc b/modules/video_coding/timing/inter_frame_delay_variation_calculator_unittest.cc
index ea719b6..d193318 100644
--- a/modules/video_coding/timing/inter_frame_delay_variation_calculator_unittest.cc
+++ b/modules/video_coding/timing/inter_frame_delay_variation_calculator_unittest.cc
@@ -11,8 +11,8 @@
#include "modules/video_coding/timing/inter_frame_delay_variation_calculator.h"
#include <limits>
+#include <optional>
-#include "absl/types/optional.h"
#include "api/units/frequency.h"
#include "api/units/time_delta.h"
#include "api/units/timestamp.h"
@@ -40,7 +40,7 @@
InterFrameDelayVariationCalculator ifdv_calculator;
EXPECT_THAT(ifdv_calculator.Calculate(180000, kStartTime),
Optional(TimeDelta::Zero()));
- EXPECT_THAT(ifdv_calculator.Calculate(90000, kStartTime), Eq(absl::nullopt));
+ EXPECT_THAT(ifdv_calculator.Calculate(90000, kStartTime), Eq(std::nullopt));
}
TEST(InterFrameDelayVariationCalculatorTest,
@@ -51,7 +51,7 @@
Optional(TimeDelta::Zero()));
// RTP has wrapped around backwards.
rtp -= 3000;
- EXPECT_THAT(ifdv_calculator.Calculate(rtp, kStartTime), Eq(absl::nullopt));
+ EXPECT_THAT(ifdv_calculator.Calculate(rtp, kStartTime), Eq(std::nullopt));
}
TEST(InterFrameDelayVariationCalculatorTest, CorrectDelayForFrames) {
@@ -184,7 +184,7 @@
// Frame delay should be as normal, in this case simulated as 1ms late.
clock.AdvanceTime(kFrameDelay);
EXPECT_THAT(ifdv_calculator.Calculate(rtp, clock.CurrentTime()),
- Eq(absl::nullopt));
+ Eq(std::nullopt));
}
} // namespace webrtc
diff --git a/modules/video_coding/timing/jitter_estimator.cc b/modules/video_coding/timing/jitter_estimator.cc
index 6275778..b10048d 100644
--- a/modules/video_coding/timing/jitter_estimator.cc
+++ b/modules/video_coding/timing/jitter_estimator.cc
@@ -15,8 +15,8 @@
#include <algorithm>
#include <cstdint>
+#include <optional>
-#include "absl/types/optional.h"
#include "api/field_trials_view.h"
#include "api/units/data_size.h"
#include "api/units/frequency.h"
@@ -155,9 +155,9 @@
var_frame_size_bytes2_ = 100;
avg_frame_size_median_bytes_.Reset();
max_frame_size_bytes_percentile_.Reset();
- last_update_time_ = absl::nullopt;
- prev_estimate_ = absl::nullopt;
- prev_frame_size_ = absl::nullopt;
+ last_update_time_ = std::nullopt;
+ prev_estimate_ = std::nullopt;
+ prev_frame_size_ = std::nullopt;
avg_noise_ms_ = 0.0;
var_noise_ms2_ = 4.0;
alpha_count_ = 1;
@@ -424,7 +424,7 @@
// otherwise tries to calculate an estimate.
TimeDelta JitterEstimator::GetJitterEstimate(
double rtt_multiplier,
- absl::optional<TimeDelta> rtt_mult_add_cap) {
+ std::optional<TimeDelta> rtt_mult_add_cap) {
TimeDelta jitter = CalculateEstimate() + OPERATING_SYSTEM_JITTER;
Timestamp now = clock_->CurrentTime();
diff --git a/modules/video_coding/timing/jitter_estimator.h b/modules/video_coding/timing/jitter_estimator.h
index 89dc649..44a95be 100644
--- a/modules/video_coding/timing/jitter_estimator.h
+++ b/modules/video_coding/timing/jitter_estimator.h
@@ -13,10 +13,10 @@
#include <algorithm>
#include <memory>
+#include <optional>
#include <queue>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/field_trials_view.h"
#include "api/units/data_size.h"
#include "api/units/frequency.h"
@@ -66,37 +66,37 @@
// If set, the "max" frame size is calculated as this percentile over a
// window of recent frame sizes.
- absl::optional<double> max_frame_size_percentile = absl::nullopt;
+ std::optional<double> max_frame_size_percentile = std::nullopt;
// The length of the percentile filters' window, in number of frames.
- absl::optional<int> frame_size_window = absl::nullopt;
+ std::optional<int> frame_size_window = std::nullopt;
// The incoming frame delay variation samples are clamped to be at most
// this number of standard deviations away from zero.
//
// Increasing this value clamps fewer samples.
- absl::optional<double> num_stddev_delay_clamp = absl::nullopt;
+ std::optional<double> num_stddev_delay_clamp = std::nullopt;
// A (relative) frame delay variation sample is an outlier if its absolute
// deviation from the Kalman filter model falls outside this number of
// sample standard deviations.
//
// Increasing this value rejects fewer samples.
- absl::optional<double> num_stddev_delay_outlier = absl::nullopt;
+ std::optional<double> num_stddev_delay_outlier = std::nullopt;
// An (absolute) frame size sample is an outlier if its positive deviation
// from the estimated average frame size falls outside this number of sample
// standard deviations.
//
// Increasing this value rejects fewer samples.
- absl::optional<double> num_stddev_size_outlier = absl::nullopt;
+ std::optional<double> num_stddev_size_outlier = std::nullopt;
// A (relative) frame size variation sample is deemed "congested", and is
// thus rejected, if its value is less than this factor times the estimated
// max frame size.
//
// Decreasing this value rejects fewer samples.
- absl::optional<double> congestion_rejection_factor = absl::nullopt;
+ std::optional<double> congestion_rejection_factor = std::nullopt;
// If true, the noise estimate will be updated for congestion rejected
// frames. This is currently enabled by default, but that may not be optimal
@@ -128,7 +128,7 @@
//
// Return value : Jitter estimate.
TimeDelta GetJitterEstimate(double rtt_multiplier,
- absl::optional<TimeDelta> rtt_mult_add_cap);
+ std::optional<TimeDelta> rtt_mult_add_cap);
// Updates the nack counter.
void FrameNacked();
@@ -187,11 +187,11 @@
double startup_frame_size_sum_bytes_;
size_t startup_frame_size_count_;
- absl::optional<Timestamp> last_update_time_;
+ std::optional<Timestamp> last_update_time_;
// The previously returned jitter estimate
- absl::optional<TimeDelta> prev_estimate_;
+ std::optional<TimeDelta> prev_estimate_;
// Frame size of the previous frame
- absl::optional<DataSize> prev_frame_size_;
+ std::optional<DataSize> prev_frame_size_;
// Average of the random jitter. Unit is milliseconds.
double avg_noise_ms_;
// Variance of the time-deviation from the line. Unit is milliseconds^2.
diff --git a/modules/video_coding/timing/jitter_estimator_unittest.cc b/modules/video_coding/timing/jitter_estimator_unittest.cc
index 8e0c015..d8f89d2 100644
--- a/modules/video_coding/timing/jitter_estimator_unittest.cc
+++ b/modules/video_coding/timing/jitter_estimator_unittest.cc
@@ -12,11 +12,11 @@
#include <stdint.h>
#include <memory>
+#include <optional>
#include <string>
#include <utility>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/field_trials.h"
#include "api/units/data_size.h"
@@ -79,7 +79,7 @@
TEST_F(JitterEstimatorTest, SteadyStateConvergence) {
ValueGenerator gen(10);
Run(/*duration_s=*/60, /*framerate_fps=*/30, gen);
- EXPECT_EQ(estimator_.GetJitterEstimate(0, absl::nullopt).ms(), 54);
+ EXPECT_EQ(estimator_.GetJitterEstimate(0, std::nullopt).ms(), 54);
}
TEST_F(JitterEstimatorTest,
@@ -88,12 +88,11 @@
// Steady state.
Run(/*duration_s=*/60, /*framerate_fps=*/30, gen);
- TimeDelta steady_state_jitter =
- estimator_.GetJitterEstimate(0, absl::nullopt);
+ TimeDelta steady_state_jitter = estimator_.GetJitterEstimate(0, std::nullopt);
// A single outlier frame size...
estimator_.UpdateEstimate(gen.Delay(), 10 * gen.FrameSize());
- TimeDelta outlier_jitter = estimator_.GetJitterEstimate(0, absl::nullopt);
+ TimeDelta outlier_jitter = estimator_.GetJitterEstimate(0, std::nullopt);
// ...changes the estimate.
EXPECT_GT(outlier_jitter.ms(), 1.25 * steady_state_jitter.ms());
@@ -107,7 +106,7 @@
estimator_.UpdateEstimate(gen.Delay(), gen.FrameSize());
fake_clock_.AdvanceTime(time_delta);
if (i > 2)
- EXPECT_EQ(estimator_.GetJitterEstimate(0, absl::nullopt),
+ EXPECT_EQ(estimator_.GetJitterEstimate(0, std::nullopt),
TimeDelta::Zero());
gen.Advance();
}
@@ -150,12 +149,11 @@
// Steady state.
Run(/*duration_s=*/60, /*framerate_fps=*/30, gen);
- TimeDelta steady_state_jitter =
- estimator_.GetJitterEstimate(0, absl::nullopt);
+ TimeDelta steady_state_jitter = estimator_.GetJitterEstimate(0, std::nullopt);
// A single outlier frame size...
estimator_.UpdateEstimate(gen.Delay(), 2 * gen.FrameSize());
- TimeDelta outlier_jitter = estimator_.GetJitterEstimate(0, absl::nullopt);
+ TimeDelta outlier_jitter = estimator_.GetJitterEstimate(0, std::nullopt);
// ...impacts the estimate.
EXPECT_GT(outlier_jitter.ms(), steady_state_jitter.ms());
@@ -168,12 +166,11 @@
// Steady state.
Run(/*duration_s=*/10, /*framerate_fps=*/30, gen);
- TimeDelta steady_state_jitter =
- estimator_.GetJitterEstimate(0, absl::nullopt);
+ TimeDelta steady_state_jitter = estimator_.GetJitterEstimate(0, std::nullopt);
// Congested frame...
estimator_.UpdateEstimate(-10 * gen.Delay(), 0.1 * gen.FrameSize());
- TimeDelta outlier_jitter = estimator_.GetJitterEstimate(0, absl::nullopt);
+ TimeDelta outlier_jitter = estimator_.GetJitterEstimate(0, std::nullopt);
// ...impacts the estimate.
EXPECT_GT(outlier_jitter.ms(), steady_state_jitter.ms());
@@ -225,12 +222,11 @@
// Steady state.
Run(/*duration_s=*/60, /*framerate_fps=*/30, gen);
- TimeDelta steady_state_jitter =
- estimator_.GetJitterEstimate(0, absl::nullopt);
+ TimeDelta steady_state_jitter = estimator_.GetJitterEstimate(0, std::nullopt);
// A single outlier frame size...
estimator_.UpdateEstimate(10 * gen.Delay(), gen.FrameSize());
- TimeDelta outlier_jitter = estimator_.GetJitterEstimate(0, absl::nullopt);
+ TimeDelta outlier_jitter = estimator_.GetJitterEstimate(0, std::nullopt);
// ...does not change the estimate.
EXPECT_EQ(outlier_jitter.ms(), steady_state_jitter.ms());
@@ -244,19 +240,18 @@
// Steady state.
Run(/*duration_s=*/60, /*framerate_fps=*/30, gen);
- TimeDelta steady_state_jitter =
- estimator_.GetJitterEstimate(0, absl::nullopt);
+ TimeDelta steady_state_jitter = estimator_.GetJitterEstimate(0, std::nullopt);
// Three outlier frames do not impact the jitter estimate.
for (int i = 0; i < 3; ++i) {
estimator_.UpdateEstimate(gen.Delay(), 2 * gen.FrameSize());
}
- TimeDelta outlier_jitter_3x = estimator_.GetJitterEstimate(0, absl::nullopt);
+ TimeDelta outlier_jitter_3x = estimator_.GetJitterEstimate(0, std::nullopt);
EXPECT_EQ(outlier_jitter_3x.ms(), steady_state_jitter.ms());
// Four outlier frames do impact the jitter estimate.
estimator_.UpdateEstimate(gen.Delay(), 2 * gen.FrameSize());
- TimeDelta outlier_jitter_4x = estimator_.GetJitterEstimate(0, absl::nullopt);
+ TimeDelta outlier_jitter_4x = estimator_.GetJitterEstimate(0, std::nullopt);
EXPECT_GT(outlier_jitter_4x.ms(), outlier_jitter_3x.ms());
}
@@ -268,12 +263,11 @@
// Steady state.
Run(/*duration_s=*/10, /*framerate_fps=*/30, gen);
- TimeDelta steady_state_jitter =
- estimator_.GetJitterEstimate(0, absl::nullopt);
+ TimeDelta steady_state_jitter = estimator_.GetJitterEstimate(0, std::nullopt);
// Congested frame...
estimator_.UpdateEstimate(-10 * gen.Delay(), 0.1 * gen.FrameSize());
- TimeDelta outlier_jitter = estimator_.GetJitterEstimate(0, absl::nullopt);
+ TimeDelta outlier_jitter = estimator_.GetJitterEstimate(0, std::nullopt);
// ...does not impact the estimate.
EXPECT_EQ(outlier_jitter.ms(), steady_state_jitter.ms());
diff --git a/modules/video_coding/timing/timestamp_extrapolator.cc b/modules/video_coding/timing/timestamp_extrapolator.cc
index cc7eff9..e87a46a 100644
--- a/modules/video_coding/timing/timestamp_extrapolator.cc
+++ b/modules/video_coding/timing/timestamp_extrapolator.cc
@@ -11,8 +11,8 @@
#include "modules/video_coding/timing/timestamp_extrapolator.h"
#include <algorithm>
+#include <optional>
-#include "absl/types/optional.h"
#include "rtc_base/numerics/sequence_number_unwrapper.h"
namespace webrtc {
@@ -41,8 +41,8 @@
void TimestampExtrapolator::Reset(Timestamp start) {
start_ = start;
prev_ = start_;
- first_unwrapped_timestamp_ = absl::nullopt;
- prev_unwrapped_timestamp_ = absl::nullopt;
+ first_unwrapped_timestamp_ = std::nullopt;
+ prev_unwrapped_timestamp_ = std::nullopt;
w_[0] = 90.0;
w_[1] = 0;
p_[0][0] = 1;
@@ -123,12 +123,12 @@
}
}
-absl::optional<Timestamp> TimestampExtrapolator::ExtrapolateLocalTime(
+std::optional<Timestamp> TimestampExtrapolator::ExtrapolateLocalTime(
uint32_t timestamp90khz) const {
int64_t unwrapped_ts90khz = unwrapper_.PeekUnwrap(timestamp90khz);
if (!first_unwrapped_timestamp_) {
- return absl::nullopt;
+ return std::nullopt;
} else if (packet_count_ < kStartUpFilterDelayInPackets) {
constexpr double kRtpTicksPerMs = 90;
TimeDelta diff = TimeDelta::Millis(
@@ -136,7 +136,7 @@
if (prev_.us() + diff.us() < 0) {
// Prevent the construction of a negative Timestamp.
// This scenario can occur when the RTP timestamp wraps around.
- return absl::nullopt;
+ return std::nullopt;
}
return prev_ + diff;
} else if (w_[0] < 1e-3) {
@@ -148,7 +148,7 @@
if (start_.us() + diff.us() < 0) {
// Prevent the construction of a negative Timestamp.
// This scenario can occur when the RTP timestamp wraps around.
- return absl::nullopt;
+ return std::nullopt;
}
return start_ + diff;
}
diff --git a/modules/video_coding/timing/timestamp_extrapolator.h b/modules/video_coding/timing/timestamp_extrapolator.h
index 6a97639..f3072fa 100644
--- a/modules/video_coding/timing/timestamp_extrapolator.h
+++ b/modules/video_coding/timing/timestamp_extrapolator.h
@@ -13,7 +13,8 @@
#include <stdint.h>
-#include "absl/types/optional.h"
+#include <optional>
+
#include "api/units/timestamp.h"
#include "rtc_base/numerics/sequence_number_unwrapper.h"
@@ -24,7 +25,7 @@
public:
explicit TimestampExtrapolator(Timestamp start);
void Update(Timestamp now, uint32_t ts90khz);
- absl::optional<Timestamp> ExtrapolateLocalTime(uint32_t timestamp90khz) const;
+ std::optional<Timestamp> ExtrapolateLocalTime(uint32_t timestamp90khz) const;
void Reset(Timestamp start);
private:
@@ -35,9 +36,9 @@
double p_[2][2];
Timestamp start_;
Timestamp prev_;
- absl::optional<int64_t> first_unwrapped_timestamp_;
+ std::optional<int64_t> first_unwrapped_timestamp_;
RtpTimestampUnwrapper unwrapper_;
- absl::optional<int64_t> prev_unwrapped_timestamp_;
+ std::optional<int64_t> prev_unwrapped_timestamp_;
uint32_t packet_count_;
double detector_accumulator_pos_;
double detector_accumulator_neg_;
diff --git a/modules/video_coding/timing/timestamp_extrapolator_unittest.cc b/modules/video_coding/timing/timestamp_extrapolator_unittest.cc
index fa1e23a..bf0c78a 100644
--- a/modules/video_coding/timing/timestamp_extrapolator_unittest.cc
+++ b/modules/video_coding/timing/timestamp_extrapolator_unittest.cc
@@ -13,8 +13,8 @@
#include <stdint.h>
#include <limits>
+#include <optional>
-#include "absl/types/optional.h"
#include "api/units/frequency.h"
#include "api/units/time_delta.h"
#include "api/units/timestamp.h"
@@ -40,7 +40,7 @@
TimestampExtrapolator ts_extrapolator(clock.CurrentTime());
// No packets so no timestamp.
- EXPECT_THAT(ts_extrapolator.ExtrapolateLocalTime(90000), Eq(absl::nullopt));
+ EXPECT_THAT(ts_extrapolator.ExtrapolateLocalTime(90000), Eq(std::nullopt));
uint32_t rtp = 90000;
clock.AdvanceTime(k25FpsDelay);
@@ -143,7 +143,7 @@
// Go backwards! Static cast to avoid undefined behaviour with -=.
rtp -= static_cast<uint32_t>(kRtpHz * TimeDelta::Seconds(10));
ts_extrapolator.Update(clock.CurrentTime(), rtp);
- EXPECT_THAT(ts_extrapolator.ExtrapolateLocalTime(rtp), absl::nullopt);
+ EXPECT_THAT(ts_extrapolator.ExtrapolateLocalTime(rtp), std::nullopt);
}
TEST(TimestampExtrapolatorTest, Slow90KHzClock) {
diff --git a/modules/video_coding/timing/timing.cc b/modules/video_coding/timing/timing.cc
index 735f632..e791612 100644
--- a/modules/video_coding/timing/timing.cc
+++ b/modules/video_coding/timing/timing.cc
@@ -292,13 +292,13 @@
timing_frame_info_.emplace(info);
}
-absl::optional<TimingFrameInfo> VCMTiming::GetTimingFrameInfo() {
+std::optional<TimingFrameInfo> VCMTiming::GetTimingFrameInfo() {
MutexLock lock(&mutex_);
return timing_frame_info_;
}
void VCMTiming::SetMaxCompositionDelayInFrames(
- absl::optional<int> max_composition_delay_in_frames) {
+ std::optional<int> max_composition_delay_in_frames) {
MutexLock lock(&mutex_);
max_composition_delay_in_frames_ = max_composition_delay_in_frames;
}
diff --git a/modules/video_coding/timing/timing.h b/modules/video_coding/timing/timing.h
index 9e7fb87..2b2a05e 100644
--- a/modules/video_coding/timing/timing.h
+++ b/modules/video_coding/timing/timing.h
@@ -12,8 +12,8 @@
#define MODULES_VIDEO_CODING_TIMING_TIMING_H_
#include <memory>
+#include <optional>
-#include "absl/types/optional.h"
#include "api/field_trials_view.h"
#include "api/units/time_delta.h"
#include "api/video/video_frame.h"
@@ -120,10 +120,10 @@
VideoDelayTimings GetTimings() const;
void SetTimingFrameInfo(const TimingFrameInfo& info);
- absl::optional<TimingFrameInfo> GetTimingFrameInfo();
+ std::optional<TimingFrameInfo> GetTimingFrameInfo();
void SetMaxCompositionDelayInFrames(
- absl::optional<int> max_composition_delay_in_frames);
+ std::optional<int> max_composition_delay_in_frames);
VideoFrame::RenderParameters RenderParameters() const;
@@ -156,9 +156,9 @@
TimeDelta jitter_delay_ RTC_GUARDED_BY(mutex_);
TimeDelta current_delay_ RTC_GUARDED_BY(mutex_);
uint32_t prev_frame_timestamp_ RTC_GUARDED_BY(mutex_);
- absl::optional<TimingFrameInfo> timing_frame_info_ RTC_GUARDED_BY(mutex_);
+ std::optional<TimingFrameInfo> timing_frame_info_ RTC_GUARDED_BY(mutex_);
size_t num_decoded_frames_ RTC_GUARDED_BY(mutex_);
- absl::optional<int> max_composition_delay_in_frames_ RTC_GUARDED_BY(mutex_);
+ std::optional<int> max_composition_delay_in_frames_ RTC_GUARDED_BY(mutex_);
// Set by the field trial WebRTC-ZeroPlayoutDelay. The parameter min_pacing
// determines the minimum delay between frames scheduled for decoding that is
// used when min playout delay=0 and max playout delay>=0.
diff --git a/modules/video_coding/utility/bandwidth_quality_scaler.cc b/modules/video_coding/utility/bandwidth_quality_scaler.cc
index f81ae3b..054a157 100644
--- a/modules/video_coding/utility/bandwidth_quality_scaler.cc
+++ b/modules/video_coding/utility/bandwidth_quality_scaler.cc
@@ -109,13 +109,13 @@
return BandwidthQualityScaler::CheckBitrateResult::kInsufficientSamples;
}
- absl::optional<int64_t> current_bitrate_bps =
+ std::optional<int64_t> current_bitrate_bps =
encoded_bitrate_.Rate(last_time_sent_in_ms_.value());
if (!current_bitrate_bps.has_value()) {
// We can't get a valid bitrate due to not enough data points.
return BandwidthQualityScaler::CheckBitrateResult::kInsufficientSamples;
}
- absl::optional<VideoEncoder::ResolutionBitrateLimits> suitable_bitrate_limit =
+ std::optional<VideoEncoder::ResolutionBitrateLimits> suitable_bitrate_limit =
EncoderInfoSettings::
GetSinglecastBitrateLimitForResolutionWhenQpIsUntrusted(
last_frame_size_pixels_, resolution_bitrate_limits_);
diff --git a/modules/video_coding/utility/bandwidth_quality_scaler.h b/modules/video_coding/utility/bandwidth_quality_scaler.h
index 89e171c..17eafb0 100644
--- a/modules/video_coding/utility/bandwidth_quality_scaler.h
+++ b/modules/video_coding/utility/bandwidth_quality_scaler.h
@@ -15,9 +15,9 @@
#include <stdint.h>
#include <memory>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/scoped_refptr.h"
#include "api/sequence_checker.h"
#include "api/video_codecs/video_encoder.h"
@@ -82,9 +82,9 @@
BandwidthQualityScalerUsageHandlerInterface* const handler_
RTC_GUARDED_BY(&task_checker_);
- absl::optional<int64_t> last_time_sent_in_ms_ RTC_GUARDED_BY(&task_checker_);
+ std::optional<int64_t> last_time_sent_in_ms_ RTC_GUARDED_BY(&task_checker_);
RateStatistics encoded_bitrate_ RTC_GUARDED_BY(&task_checker_);
- absl::optional<int> last_frame_size_pixels_ RTC_GUARDED_BY(&task_checker_);
+ std::optional<int> last_frame_size_pixels_ RTC_GUARDED_BY(&task_checker_);
rtc::WeakPtrFactory<BandwidthQualityScaler> weak_ptr_factory_;
std::vector<VideoEncoder::ResolutionBitrateLimits> resolution_bitrate_limits_;
diff --git a/modules/video_coding/utility/bandwidth_quality_scaler_unittest.cc b/modules/video_coding/utility/bandwidth_quality_scaler_unittest.cc
index 67620e4..9c5fe31 100644
--- a/modules/video_coding/utility/bandwidth_quality_scaler_unittest.cc
+++ b/modules/video_coding/utility/bandwidth_quality_scaler_unittest.cc
@@ -125,7 +125,7 @@
return -1;
}
- absl::optional<VideoEncoder::ResolutionBitrateLimits>
+ std::optional<VideoEncoder::ResolutionBitrateLimits>
GetDefaultSuitableBitrateLimit(int frame_size_pixels) {
return EncoderInfoSettings::
GetSinglecastBitrateLimitForResolutionWhenQpIsUntrusted(
@@ -150,7 +150,7 @@
TimeDelta delay = TimeDelta::Zero();
int num_delayed_tasks = 0;
for (const FrameConfig& config : frame_configs) {
- absl::optional<VideoEncoder::ResolutionBitrateLimits> suitable_bitrate =
+ std::optional<VideoEncoder::ResolutionBitrateLimits> suitable_bitrate =
GetDefaultSuitableBitrateLimit(config.actual_width *
config.actual_height);
EXPECT_TRUE(suitable_bitrate);
diff --git a/modules/video_coding/utility/decoded_frames_history.cc b/modules/video_coding/utility/decoded_frames_history.cc
index 1138aa8..6b48d89 100644
--- a/modules/video_coding/utility/decoded_frames_history.cc
+++ b/modules/video_coding/utility/decoded_frames_history.cc
@@ -74,11 +74,11 @@
last_frame_id_.reset();
}
-absl::optional<int64_t> DecodedFramesHistory::GetLastDecodedFrameId() const {
+std::optional<int64_t> DecodedFramesHistory::GetLastDecodedFrameId() const {
return last_decoded_frame_;
}
-absl::optional<uint32_t> DecodedFramesHistory::GetLastDecodedFrameTimestamp()
+std::optional<uint32_t> DecodedFramesHistory::GetLastDecodedFrameTimestamp()
const {
return last_decoded_frame_timestamp_;
}
diff --git a/modules/video_coding/utility/decoded_frames_history.h b/modules/video_coding/utility/decoded_frames_history.h
index 9b8bf65..39337c0 100644
--- a/modules/video_coding/utility/decoded_frames_history.h
+++ b/modules/video_coding/utility/decoded_frames_history.h
@@ -14,9 +14,9 @@
#include <stdint.h>
#include <bitset>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/video/encoded_frame.h"
namespace webrtc {
@@ -35,16 +35,16 @@
void Clear();
- absl::optional<int64_t> GetLastDecodedFrameId() const;
- absl::optional<uint32_t> GetLastDecodedFrameTimestamp() const;
+ std::optional<int64_t> GetLastDecodedFrameId() const;
+ std::optional<uint32_t> GetLastDecodedFrameTimestamp() const;
private:
int FrameIdToIndex(int64_t frame_id) const;
std::vector<bool> buffer_;
- absl::optional<int64_t> last_frame_id_;
- absl::optional<int64_t> last_decoded_frame_;
- absl::optional<uint32_t> last_decoded_frame_timestamp_;
+ std::optional<int64_t> last_frame_id_;
+ std::optional<int64_t> last_decoded_frame_;
+ std::optional<uint32_t> last_decoded_frame_timestamp_;
};
} // namespace video_coding
diff --git a/modules/video_coding/utility/decoded_frames_history_unittest.cc b/modules/video_coding/utility/decoded_frames_history_unittest.cc
index ac09a42..bfe5467 100644
--- a/modules/video_coding/utility/decoded_frames_history_unittest.cc
+++ b/modules/video_coding/utility/decoded_frames_history_unittest.cc
@@ -48,8 +48,8 @@
history.InsertDecoded(1234, 0);
history.Clear();
EXPECT_EQ(history.WasDecoded(1234), false);
- EXPECT_EQ(history.GetLastDecodedFrameId(), absl::nullopt);
- EXPECT_EQ(history.GetLastDecodedFrameTimestamp(), absl::nullopt);
+ EXPECT_EQ(history.GetLastDecodedFrameId(), std::nullopt);
+ EXPECT_EQ(history.GetLastDecodedFrameTimestamp(), std::nullopt);
}
TEST(DecodedFramesHistory, HandlesBigJumpInPictureId) {
@@ -74,7 +74,7 @@
TEST(DecodedFramesHistory, ReturnsLastDecodedFrameId) {
DecodedFramesHistory history(kHistorySize);
- EXPECT_EQ(history.GetLastDecodedFrameId(), absl::nullopt);
+ EXPECT_EQ(history.GetLastDecodedFrameId(), std::nullopt);
history.InsertDecoded(1234, 0);
EXPECT_EQ(history.GetLastDecodedFrameId(), 1234);
history.InsertDecoded(1235, 0);
@@ -83,7 +83,7 @@
TEST(DecodedFramesHistory, ReturnsLastDecodedFrameTimestamp) {
DecodedFramesHistory history(kHistorySize);
- EXPECT_EQ(history.GetLastDecodedFrameTimestamp(), absl::nullopt);
+ EXPECT_EQ(history.GetLastDecodedFrameTimestamp(), std::nullopt);
history.InsertDecoded(1234, 12345);
EXPECT_EQ(history.GetLastDecodedFrameTimestamp(), 12345u);
history.InsertDecoded(1235, 12366);
diff --git a/modules/video_coding/utility/framerate_controller_deprecated.cc b/modules/video_coding/utility/framerate_controller_deprecated.cc
index 5978adc..0f31030 100644
--- a/modules/video_coding/utility/framerate_controller_deprecated.cc
+++ b/modules/video_coding/utility/framerate_controller_deprecated.cc
@@ -77,7 +77,7 @@
last_timestamp_ms_ = timestamp_ms;
}
-absl::optional<float> FramerateControllerDeprecated::Rate(
+std::optional<float> FramerateControllerDeprecated::Rate(
uint32_t timestamp_ms) const {
return framerate_estimator_.Rate(timestamp_ms);
}
diff --git a/modules/video_coding/utility/framerate_controller_deprecated.h b/modules/video_coding/utility/framerate_controller_deprecated.h
index ca0cbea..6fcda3d 100644
--- a/modules/video_coding/utility/framerate_controller_deprecated.h
+++ b/modules/video_coding/utility/framerate_controller_deprecated.h
@@ -13,7 +13,8 @@
#include <stdint.h>
-#include "absl/types/optional.h"
+#include <optional>
+
#include "rtc_base/rate_statistics.h"
namespace webrtc {
@@ -34,10 +35,10 @@
void Reset();
private:
- absl::optional<float> Rate(uint32_t timestamp_ms) const;
+ std::optional<float> Rate(uint32_t timestamp_ms) const;
- absl::optional<float> target_framerate_fps_;
- absl::optional<uint32_t> last_timestamp_ms_;
+ std::optional<float> target_framerate_fps_;
+ std::optional<uint32_t> last_timestamp_ms_;
uint32_t min_frame_interval_ms_;
RateStatistics framerate_estimator_;
};
diff --git a/modules/video_coding/utility/ivf_file_reader.cc b/modules/video_coding/utility/ivf_file_reader.cc
index 4c08ca6..958cc84 100644
--- a/modules/video_coding/utility/ivf_file_reader.cc
+++ b/modules/video_coding/utility/ivf_file_reader.cc
@@ -68,7 +68,7 @@
return false;
}
- absl::optional<VideoCodecType> codec_type = ParseCodecType(ivf_header, 8);
+ std::optional<VideoCodecType> codec_type = ParseCodecType(ivf_header, 8);
if (!codec_type) {
return false;
}
@@ -111,9 +111,9 @@
return true;
}
-absl::optional<EncodedImage> IvfFileReader::NextFrame() {
+std::optional<EncodedImage> IvfFileReader::NextFrame() {
if (has_error_ || !HasMoreFrames()) {
- return absl::nullopt;
+ return std::nullopt;
}
rtc::scoped_refptr<EncodedImageBuffer> payload = EncodedImageBuffer::Create();
@@ -139,7 +139,7 @@
RTC_LOG(LS_ERROR) << "Frame #" << num_read_frames_
<< ": failed to read frame payload";
has_error_ = true;
- return absl::nullopt;
+ return std::nullopt;
}
num_read_frames_++;
@@ -151,7 +151,7 @@
if (!has_error_ && num_read_frames_ != num_frames_) {
RTC_LOG(LS_ERROR) << "Unexpected EOF";
has_error_ = true;
- return absl::nullopt;
+ return std::nullopt;
}
}
@@ -179,8 +179,8 @@
return true;
}
-absl::optional<VideoCodecType> IvfFileReader::ParseCodecType(uint8_t* buffer,
- size_t start_pos) {
+std::optional<VideoCodecType> IvfFileReader::ParseCodecType(uint8_t* buffer,
+ size_t start_pos) {
if (memcmp(&buffer[start_pos], kVp8Header, kCodecTypeBytesCount) == 0) {
return VideoCodecType::kVideoCodecVP8;
}
@@ -201,11 +201,10 @@
<< std::string(
reinterpret_cast<char const*>(&buffer[start_pos]),
kCodecTypeBytesCount);
- return absl::nullopt;
+ return std::nullopt;
}
-absl::optional<IvfFileReader::FrameHeader>
-IvfFileReader::ReadNextFrameHeader() {
+std::optional<IvfFileReader::FrameHeader> IvfFileReader::ReadNextFrameHeader() {
uint8_t ivf_frame_header[kIvfFrameHeaderSize] = {0};
size_t read = file_.Read(&ivf_frame_header, kIvfFrameHeaderSize);
if (read != kIvfFrameHeaderSize) {
@@ -214,7 +213,7 @@
RTC_LOG(LS_ERROR) << "Frame #" << num_read_frames_
<< ": failed to read IVF frame header";
}
- return absl::nullopt;
+ return std::nullopt;
}
FrameHeader header;
header.frame_size = static_cast<size_t>(
@@ -226,14 +225,14 @@
has_error_ = true;
RTC_LOG(LS_ERROR) << "Frame #" << num_read_frames_
<< ": invalid frame size";
- return absl::nullopt;
+ return std::nullopt;
}
if (header.timestamp < 0) {
has_error_ = true;
RTC_LOG(LS_ERROR) << "Frame #" << num_read_frames_
<< ": negative timestamp";
- return absl::nullopt;
+ return std::nullopt;
}
return header;
diff --git a/modules/video_coding/utility/ivf_file_reader.h b/modules/video_coding/utility/ivf_file_reader.h
index db4fc25..901751a 100644
--- a/modules/video_coding/utility/ivf_file_reader.h
+++ b/modules/video_coding/utility/ivf_file_reader.h
@@ -12,9 +12,9 @@
#define MODULES_VIDEO_CODING_UTILITY_IVF_FILE_READER_H_
#include <memory>
+#include <optional>
#include <utility>
-#include "absl/types/optional.h"
#include "api/video/encoded_image.h"
#include "api/video_codecs/video_codec.h"
#include "rtc_base/system/file_wrapper.h"
@@ -39,9 +39,9 @@
// Returns count of frames in this file.
size_t GetFramesCount() const { return num_frames_; }
- // Returns next frame or absl::nullopt if any error acquired. Always returns
- // absl::nullopt after first error was spotted.
- absl::optional<EncodedImage> NextFrame();
+ // Returns next frame or std::nullopt if any error acquired. Always returns
+ // std::nullopt after first error was spotted.
+ std::optional<EncodedImage> NextFrame();
bool HasMoreFrames() const { return num_read_frames_ < num_frames_; }
bool HasError() const { return has_error_; }
@@ -61,9 +61,9 @@
// Parses codec type from specified position of the buffer. Codec type
// contains kCodecTypeBytesCount bytes and caller has to ensure that buffer
// won't overflow.
- absl::optional<VideoCodecType> ParseCodecType(uint8_t* buffer,
- size_t start_pos);
- absl::optional<FrameHeader> ReadNextFrameHeader();
+ std::optional<VideoCodecType> ParseCodecType(uint8_t* buffer,
+ size_t start_pos);
+ std::optional<FrameHeader> ReadNextFrameHeader();
VideoCodecType codec_type_;
size_t num_frames_;
@@ -73,7 +73,7 @@
uint32_t time_scale_;
FileWrapper file_;
- absl::optional<FrameHeader> next_frame_header_;
+ std::optional<FrameHeader> next_frame_header_;
bool has_error_;
};
diff --git a/modules/video_coding/utility/ivf_file_reader_unittest.cc b/modules/video_coding/utility/ivf_file_reader_unittest.cc
index 14bfdca..cf615c5 100644
--- a/modules/video_coding/utility/ivf_file_reader_unittest.cc
+++ b/modules/video_coding/utility/ivf_file_reader_unittest.cc
@@ -78,7 +78,7 @@
ASSERT_TRUE(file_writer->Close());
}
- void ValidateFrame(absl::optional<EncodedImage> frame,
+ void ValidateFrame(std::optional<EncodedImage> frame,
int frame_index,
bool use_capture_tims_ms,
int spatial_layers_count) {
diff --git a/modules/video_coding/utility/qp_parser.cc b/modules/video_coding/utility/qp_parser.cc
index a531b54..fe9193d 100644
--- a/modules/video_coding/utility/qp_parser.cc
+++ b/modules/video_coding/utility/qp_parser.cc
@@ -15,13 +15,13 @@
namespace webrtc {
-absl::optional<uint32_t> QpParser::Parse(VideoCodecType codec_type,
- size_t spatial_idx,
- const uint8_t* frame_data,
- size_t frame_size) {
+std::optional<uint32_t> QpParser::Parse(VideoCodecType codec_type,
+ size_t spatial_idx,
+ const uint8_t* frame_data,
+ size_t frame_size) {
if (frame_data == nullptr || frame_size == 0 ||
spatial_idx >= kMaxSimulcastStreams) {
- return absl::nullopt;
+ return std::nullopt;
}
if (codec_type == kVideoCodecVP8) {
@@ -43,12 +43,11 @@
#endif
}
- return absl::nullopt;
+ return std::nullopt;
}
-absl::optional<uint32_t> QpParser::H264QpParser::Parse(
- const uint8_t* frame_data,
- size_t frame_size) {
+std::optional<uint32_t> QpParser::H264QpParser::Parse(const uint8_t* frame_data,
+ size_t frame_size) {
MutexLock lock(&mutex_);
bitstream_parser_.ParseBitstream(
rtc::ArrayView<const uint8_t>(frame_data, frame_size));
@@ -56,9 +55,8 @@
}
#ifdef RTC_ENABLE_H265
-absl::optional<uint32_t> QpParser::H265QpParser::Parse(
- const uint8_t* frame_data,
- size_t frame_size) {
+std::optional<uint32_t> QpParser::H265QpParser::Parse(const uint8_t* frame_data,
+ size_t frame_size) {
MutexLock lock(&mutex_);
bitstream_parser_.ParseBitstream(
rtc::ArrayView<const uint8_t>(frame_data, frame_size));
diff --git a/modules/video_coding/utility/qp_parser.h b/modules/video_coding/utility/qp_parser.h
index 210fe02..c397e0a 100644
--- a/modules/video_coding/utility/qp_parser.h
+++ b/modules/video_coding/utility/qp_parser.h
@@ -11,7 +11,8 @@
#ifndef MODULES_VIDEO_CODING_UTILITY_QP_PARSER_H_
#define MODULES_VIDEO_CODING_UTILITY_QP_PARSER_H_
-#include "absl/types/optional.h"
+#include <optional>
+
#include "api/video/video_codec_constants.h"
#include "api/video/video_codec_type.h"
#include "common_video/h264/h264_bitstream_parser.h"
@@ -23,17 +24,16 @@
namespace webrtc {
class QpParser {
public:
- absl::optional<uint32_t> Parse(VideoCodecType codec_type,
- size_t spatial_idx,
- const uint8_t* frame_data,
- size_t frame_size);
+ std::optional<uint32_t> Parse(VideoCodecType codec_type,
+ size_t spatial_idx,
+ const uint8_t* frame_data,
+ size_t frame_size);
private:
// A thread safe wrapper for H264 bitstream parser.
class H264QpParser {
public:
- absl::optional<uint32_t> Parse(const uint8_t* frame_data,
- size_t frame_size);
+ std::optional<uint32_t> Parse(const uint8_t* frame_data, size_t frame_size);
private:
Mutex mutex_;
@@ -46,8 +46,7 @@
// A thread safe wrapper for H.265 bitstream parser.
class H265QpParser {
public:
- absl::optional<uint32_t> Parse(const uint8_t* frame_data,
- size_t frame_size);
+ std::optional<uint32_t> Parse(const uint8_t* frame_data, size_t frame_size);
private:
Mutex mutex_;
diff --git a/modules/video_coding/utility/qp_parser_unittest.cc b/modules/video_coding/utility/qp_parser_unittest.cc
index 1131288..c9ae7e7 100644
--- a/modules/video_coding/utility/qp_parser_unittest.cc
+++ b/modules/video_coding/utility/qp_parser_unittest.cc
@@ -55,21 +55,21 @@
TEST(QpParserTest, ParseQpVp8) {
QpParser parser;
- absl::optional<uint32_t> qp = parser.Parse(
+ std::optional<uint32_t> qp = parser.Parse(
kVideoCodecVP8, 0, kCodedFrameVp8Qp25, sizeof(kCodedFrameVp8Qp25));
EXPECT_EQ(qp, 25u);
}
TEST(QpParserTest, ParseQpVp9) {
QpParser parser;
- absl::optional<uint32_t> qp = parser.Parse(
+ std::optional<uint32_t> qp = parser.Parse(
kVideoCodecVP9, 0, kCodedFrameVp9Qp96, sizeof(kCodedFrameVp9Qp96));
EXPECT_EQ(qp, 96u);
}
TEST(QpParserTest, ParseQpH264) {
QpParser parser;
- absl::optional<uint32_t> qp = parser.Parse(
+ std::optional<uint32_t> qp = parser.Parse(
VideoCodecType::kVideoCodecH264, 0, kCodedFrameH264SpsPpsIdrQp38,
sizeof(kCodedFrameH264SpsPpsIdrQp38));
EXPECT_EQ(qp, 38u);
@@ -89,27 +89,27 @@
TEST(QpParserTest, ParseQpUnsupportedCodecType) {
QpParser parser;
- absl::optional<uint32_t> qp = parser.Parse(
+ std::optional<uint32_t> qp = parser.Parse(
kVideoCodecGeneric, 0, kCodedFrameVp8Qp25, sizeof(kCodedFrameVp8Qp25));
EXPECT_FALSE(qp.has_value());
}
TEST(QpParserTest, ParseQpNullData) {
QpParser parser;
- absl::optional<uint32_t> qp = parser.Parse(kVideoCodecVP8, 0, nullptr, 100);
+ std::optional<uint32_t> qp = parser.Parse(kVideoCodecVP8, 0, nullptr, 100);
EXPECT_FALSE(qp.has_value());
}
TEST(QpParserTest, ParseQpEmptyData) {
QpParser parser;
- absl::optional<uint32_t> qp =
+ std::optional<uint32_t> qp =
parser.Parse(kVideoCodecVP8, 0, kCodedFrameVp8Qp25, 0);
EXPECT_FALSE(qp.has_value());
}
TEST(QpParserTest, ParseQpSpatialIdxExceedsMax) {
QpParser parser;
- absl::optional<uint32_t> qp =
+ std::optional<uint32_t> qp =
parser.Parse(kVideoCodecVP8, kMaxSimulcastStreams, kCodedFrameVp8Qp25,
sizeof(kCodedFrameVp8Qp25));
EXPECT_FALSE(qp.has_value());
diff --git a/modules/video_coding/utility/quality_scaler.cc b/modules/video_coding/utility/quality_scaler.cc
index 4b71f90..6cb4dc9 100644
--- a/modules/video_coding/utility/quality_scaler.cc
+++ b/modules/video_coding/utility/quality_scaler.cc
@@ -42,10 +42,10 @@
last_sample_ms_(0),
smoother_(alpha) {}
- absl::optional<int> GetAvg() const {
+ std::optional<int> GetAvg() const {
float value = smoother_.filtered();
if (value == rtc::ExpFilter::kValueUndefined) {
- return absl::nullopt;
+ return std::nullopt;
}
return static_cast<int>(value);
}
@@ -275,7 +275,7 @@
}
// Check if we should scale down due to high frame drop.
- const absl::optional<int> drop_rate =
+ const std::optional<int> drop_rate =
config_.use_all_drop_reasons
? framedrop_percent_all_.GetAverageRoundedDown()
: framedrop_percent_media_opt_.GetAverageRoundedDown();
@@ -285,10 +285,10 @@
}
// Check if we should scale up or down based on QP.
- const absl::optional<int> avg_qp_high =
+ const std::optional<int> avg_qp_high =
qp_smoother_high_ ? qp_smoother_high_->GetAvg()
: average_qp_.GetAverageRoundedDown();
- const absl::optional<int> avg_qp_low =
+ const std::optional<int> avg_qp_low =
qp_smoother_low_ ? qp_smoother_low_->GetAvg()
: average_qp_.GetAverageRoundedDown();
if (avg_qp_high && avg_qp_low) {
diff --git a/modules/video_coding/utility/quality_scaler.h b/modules/video_coding/utility/quality_scaler.h
index 7939c9e..eeb878a 100644
--- a/modules/video_coding/utility/quality_scaler.h
+++ b/modules/video_coding/utility/quality_scaler.h
@@ -15,8 +15,8 @@
#include <stdint.h>
#include <memory>
+#include <optional>
-#include "absl/types/optional.h"
#include "api/field_trials_view.h"
#include "api/scoped_refptr.h"
#include "api/sequence_checker.h"
@@ -101,7 +101,7 @@
const size_t min_frames_needed_;
const double initial_scale_factor_;
- const absl::optional<double> scale_factor_;
+ const std::optional<double> scale_factor_;
};
// Reacts to QP being too high or too low. For best quality, when QP is high it
diff --git a/modules/video_coding/utility/simulcast_test_fixture_impl.cc b/modules/video_coding/utility/simulcast_test_fixture_impl.cc
index 7180993..7cba7e6 100644
--- a/modules/video_coding/utility/simulcast_test_fixture_impl.cc
+++ b/modules/video_coding/utility/simulcast_test_fixture_impl.cc
@@ -150,8 +150,8 @@
return -1;
}
void Decoded(VideoFrame& decoded_image,
- absl::optional<int32_t> decode_time_ms,
- absl::optional<uint8_t> qp) override {
+ std::optional<int32_t> decode_time_ms,
+ std::optional<uint8_t> qp) override {
Decoded(decoded_image);
}
int DecodedFrames() { return decoded_frames_; }
@@ -930,8 +930,8 @@
EXPECT_CALL(decoder_callback, Decoded(_, _, _))
.WillOnce(::testing::Invoke([](VideoFrame& decodedImage,
- absl::optional<int32_t> decode_time_ms,
- absl::optional<uint8_t> qp) {
+ std::optional<int32_t> decode_time_ms,
+ std::optional<uint8_t> qp) {
EXPECT_EQ(decodedImage.width(), kDefaultWidth / 4);
EXPECT_EQ(decodedImage.height(), kDefaultHeight / 4);
}));
@@ -939,8 +939,8 @@
EXPECT_CALL(decoder_callback, Decoded(_, _, _))
.WillOnce(::testing::Invoke([](VideoFrame& decodedImage,
- absl::optional<int32_t> decode_time_ms,
- absl::optional<uint8_t> qp) {
+ std::optional<int32_t> decode_time_ms,
+ std::optional<uint8_t> qp) {
EXPECT_EQ(decodedImage.width(), kDefaultWidth / 2);
EXPECT_EQ(decodedImage.height(), kDefaultHeight / 2);
}));
@@ -948,8 +948,8 @@
EXPECT_CALL(decoder_callback, Decoded(_, _, _))
.WillOnce(::testing::Invoke([](VideoFrame& decodedImage,
- absl::optional<int32_t> decode_time_ms,
- absl::optional<uint8_t> qp) {
+ std::optional<int32_t> decode_time_ms,
+ std::optional<uint8_t> qp) {
EXPECT_EQ(decodedImage.width(), kDefaultWidth);
EXPECT_EQ(decodedImage.height(), kDefaultHeight);
}));
diff --git a/modules/video_coding/utility/vp9_uncompressed_header_parser.cc b/modules/video_coding/utility/vp9_uncompressed_header_parser.cc
index 9e8526a..b1e4b08 100644
--- a/modules/video_coding/utility/vp9_uncompressed_header_parser.cc
+++ b/modules/video_coding/utility/vp9_uncompressed_header_parser.cc
@@ -507,7 +507,7 @@
(total_buffer_size_bits / 8) - (br.RemainingBitCount() / 8);
}
-absl::optional<Vp9UncompressedHeader> ParseUncompressedVp9Header(
+std::optional<Vp9UncompressedHeader> ParseUncompressedVp9Header(
rtc::ArrayView<const uint8_t> buf) {
BitstreamReader reader(buf);
Vp9UncompressedHeader frame_info;
@@ -515,7 +515,7 @@
if (reader.Ok() && frame_info.frame_width > 0) {
return frame_info;
}
- return absl::nullopt;
+ return std::nullopt;
}
namespace vp9 {
diff --git a/modules/video_coding/utility/vp9_uncompressed_header_parser.h b/modules/video_coding/utility/vp9_uncompressed_header_parser.h
index 14b55c4..524d6cf 100644
--- a/modules/video_coding/utility/vp9_uncompressed_header_parser.h
+++ b/modules/video_coding/utility/vp9_uncompressed_header_parser.h
@@ -16,9 +16,9 @@
#include <array>
#include <bitset>
+#include <optional>
#include <string>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "modules/video_coding/utility/vp9_constants.h"
@@ -90,14 +90,14 @@
struct Vp9UncompressedHeader {
int profile = 0; // Profiles 0-3 are valid.
- absl::optional<uint8_t> show_existing_frame;
+ std::optional<uint8_t> show_existing_frame;
bool is_keyframe = false;
bool show_frame = false;
bool error_resilient = false;
Vp9BitDept bit_detph = Vp9BitDept::k8Bit;
- absl::optional<Vp9ColorSpace> color_space;
- absl::optional<Vp9ColorRange> color_range;
- absl::optional<Vp9YuvSubsampling> sub_sampling;
+ std::optional<Vp9ColorSpace> color_space;
+ std::optional<Vp9ColorRange> color_range;
+ std::optional<Vp9YuvSubsampling> sub_sampling;
int frame_width = 0;
int frame_height = 0;
int render_width = 0;
@@ -105,10 +105,10 @@
// Width/height of the tiles used (in units of 8x8 blocks).
size_t tile_cols_log2 = 0; // tile_cols = 1 << tile_cols_log2
size_t tile_rows_log2 = 0; // tile_rows = 1 << tile_rows_log2
- absl::optional<size_t> render_size_offset_bits;
+ std::optional<size_t> render_size_offset_bits;
// Number of bits from the start of the frame header to where the loop filter
// parameters are located.
- absl::optional<size_t> loop_filter_params_offset_bits;
+ std::optional<size_t> loop_filter_params_offset_bits;
Vp9InterpolationFilter interpolation_filter =
Vp9InterpolationFilter::kEightTap;
bool allow_high_precision_mv = false;
@@ -117,10 +117,10 @@
uint8_t frame_context_idx = 0;
bool segmentation_enabled = false;
- absl::optional<std::array<uint8_t, 7>> segmentation_tree_probs;
- absl::optional<std::array<uint8_t, 3>> segmentation_pred_prob;
+ std::optional<std::array<uint8_t, 7>> segmentation_tree_probs;
+ std::optional<std::array<uint8_t, 3>> segmentation_pred_prob;
bool segmentation_is_delta = false;
- std::array<std::array<absl::optional<int>, kVp9SegLvlMax>, kVp9MaxSegments>
+ std::array<std::array<std::optional<int>, kVp9SegLvlMax>, kVp9MaxSegments>
segmentation_features;
// Which of the 8 reference buffers may be used as references for this frame.
@@ -132,7 +132,7 @@
std::bitset<kVp9MaxRefFrames> reference_buffers_sign_bias = 0;
// Indicates which reference buffer [0,7] to infer the frame size from.
- absl::optional<int> infer_size_from_reference;
+ std::optional<int> infer_size_from_reference;
// Which of the 8 reference buffers are updated by this frame.
std::bitset<kVp9NumRefFrames> updated_buffers = 0;
@@ -150,7 +150,7 @@
// Parses the uncompressed header and populates (most) values in a
// UncompressedHeader struct. Returns nullopt on failure.
-absl::optional<Vp9UncompressedHeader> ParseUncompressedVp9Header(
+std::optional<Vp9UncompressedHeader> ParseUncompressedVp9Header(
rtc::ArrayView<const uint8_t> buf);
} // namespace webrtc
diff --git a/modules/video_coding/utility/vp9_uncompressed_header_parser_unittest.cc b/modules/video_coding/utility/vp9_uncompressed_header_parser_unittest.cc
index 25ab656..685a7dd 100644
--- a/modules/video_coding/utility/vp9_uncompressed_header_parser_unittest.cc
+++ b/modules/video_coding/utility/vp9_uncompressed_header_parser_unittest.cc
@@ -31,7 +31,7 @@
0x2e, 0x73, 0xb7, 0xee, 0x22, 0x06, 0x81, 0x82, 0xd4, 0xef, 0xc3, 0x58,
0x1f, 0x12, 0xd2, 0x7b, 0x28, 0x1f, 0x80, 0xfc, 0x07, 0xe0, 0x00, 0x00};
- absl::optional<Vp9UncompressedHeader> frame_info =
+ std::optional<Vp9UncompressedHeader> frame_info =
ParseUncompressedVp9Header(kHeader);
ASSERT_TRUE(frame_info.has_value());
@@ -74,7 +74,7 @@
const uint8_t kHeader[] = {0x90, 0x49, 0x83, 0x42, 0x80, 0x2e,
0x30, 0x0, 0xb0, 0x0, 0x37, 0xff,
0x06, 0x80, 0x0, 0x0, 0x0, 0x0};
- absl::optional<Vp9UncompressedHeader> frame_info =
+ std::optional<Vp9UncompressedHeader> frame_info =
ParseUncompressedVp9Header(kHeader);
ASSERT_TRUE(frame_info.has_value());
EXPECT_THAT(frame_info->segmentation_pred_prob,
@@ -85,7 +85,7 @@
const uint8_t kHeader[] = {0x90, 0x49, 0x83, 0x42, 0x80, 0x2e, 0x30, 0x00,
0xb0, 0x00, 0x37, 0xff, 0x06, 0x80, 0x01, 0x08,
0x00, 0x00, 0x00, 0x00, 0x00, 0x00};
- absl::optional<Vp9UncompressedHeader> frame_info =
+ std::optional<Vp9UncompressedHeader> frame_info =
ParseUncompressedVp9Header(kHeader);
ASSERT_TRUE(frame_info.has_value());
EXPECT_THAT(frame_info->segmentation_features[0][kVp9SegLvlSkip], Eq(1));
diff --git a/modules/video_coding/video_codec_initializer.cc b/modules/video_coding/video_codec_initializer.cc
index 2c6a255..fda0005 100644
--- a/modules/video_coding/video_codec_initializer.cc
+++ b/modules/video_coding/video_codec_initializer.cc
@@ -14,8 +14,8 @@
#include <string.h>
#include <algorithm>
+#include <optional>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/field_trials_view.h"
#include "api/scoped_refptr.h"
@@ -81,8 +81,7 @@
int max_framerate = 0;
- absl::optional<ScalabilityMode> scalability_mode =
- streams[0].scalability_mode;
+ std::optional<ScalabilityMode> scalability_mode = streams[0].scalability_mode;
for (size_t i = 0; i < streams.size(); ++i) {
SimulcastStream* sim_stream = &video_codec.simulcastStream[i];
RTC_DCHECK_GT(streams[i].width, 0);
@@ -338,7 +337,7 @@
break;
}
- const absl::optional<DataRate> experimental_min_bitrate =
+ const std::optional<DataRate> experimental_min_bitrate =
GetExperimentalMinVideoBitrate(field_trials, video_codec.codecType);
if (experimental_min_bitrate) {
const int experimental_min_bitrate_kbps =
diff --git a/modules/video_coding/video_codec_initializer_unittest.cc b/modules/video_coding/video_codec_initializer_unittest.cc
index de88a64..d7c9171 100644
--- a/modules/video_coding/video_codec_initializer_unittest.cc
+++ b/modules/video_coding/video_codec_initializer_unittest.cc
@@ -14,8 +14,8 @@
#include <stdint.h>
#include <memory>
+#include <optional>
-#include "absl/types/optional.h"
#include "api/environment/environment.h"
#include "api/environment/environment_factory.h"
#include "api/scoped_refptr.h"
@@ -61,8 +61,8 @@
protected:
void SetUpFor(VideoCodecType type,
- absl::optional<int> num_simulcast_streams,
- absl::optional<int> num_spatial_streams,
+ std::optional<int> num_simulcast_streams,
+ std::optional<int> num_spatial_streams,
int num_temporal_streams,
bool screenshare) {
config_ = VideoEncoderConfig();
@@ -113,7 +113,7 @@
VideoStream DefaultStream(
int width = kDefaultWidth,
int height = kDefaultHeight,
- absl::optional<ScalabilityMode> scalability_mode = absl::nullopt) {
+ std::optional<ScalabilityMode> scalability_mode = std::nullopt) {
VideoStream stream;
stream.width = width;
stream.height = height;
@@ -153,7 +153,7 @@
};
TEST_F(VideoCodecInitializerTest, SingleStreamVp8Screenshare) {
- SetUpFor(VideoCodecType::kVideoCodecVP8, 1, absl::nullopt, 1, true);
+ SetUpFor(VideoCodecType::kVideoCodecVP8, 1, std::nullopt, 1, true);
streams_.push_back(DefaultStream());
InitializeCodec();
@@ -166,7 +166,7 @@
}
TEST_F(VideoCodecInitializerTest, SingleStreamVp8ScreenshareInactive) {
- SetUpFor(VideoCodecType::kVideoCodecVP8, 1, absl::nullopt, 1, true);
+ SetUpFor(VideoCodecType::kVideoCodecVP8, 1, std::nullopt, 1, true);
VideoStream inactive_stream = DefaultStream();
inactive_stream.active = false;
streams_.push_back(inactive_stream);
@@ -181,7 +181,7 @@
}
TEST_F(VideoCodecInitializerTest, TemporalLayeredVp8ScreenshareConference) {
- SetUpFor(VideoCodecType::kVideoCodecVP8, 1, absl::nullopt, 2, true);
+ SetUpFor(VideoCodecType::kVideoCodecVP8, 1, std::nullopt, 2, true);
streams_.push_back(DefaultScreenshareStream());
InitializeCodec();
bitrate_allocator_->SetLegacyConferenceMode(true);
@@ -198,7 +198,7 @@
}
TEST_F(VideoCodecInitializerTest, TemporalLayeredVp8Screenshare) {
- SetUpFor(VideoCodecType::kVideoCodecVP8, 1, absl::nullopt, 2, true);
+ SetUpFor(VideoCodecType::kVideoCodecVP8, 1, std::nullopt, 2, true);
streams_.push_back(DefaultScreenshareStream());
InitializeCodec();
@@ -213,7 +213,7 @@
}
TEST_F(VideoCodecInitializerTest, SimulcastVp8Screenshare) {
- SetUpFor(VideoCodecType::kVideoCodecVP8, 2, absl::nullopt, 1, true);
+ SetUpFor(VideoCodecType::kVideoCodecVP8, 2, std::nullopt, 1, true);
streams_.push_back(DefaultScreenshareStream());
VideoStream video_stream = DefaultStream();
video_stream.max_framerate = kScreenshareDefaultFramerate;
@@ -237,7 +237,7 @@
// Tests that when a video stream is inactive, then the bitrate allocation will
// be 0 for that stream.
TEST_F(VideoCodecInitializerTest, SimulcastVp8ScreenshareInactive) {
- SetUpFor(VideoCodecType::kVideoCodecVP8, 2, absl::nullopt, 1, true);
+ SetUpFor(VideoCodecType::kVideoCodecVP8, 2, std::nullopt, 1, true);
streams_.push_back(DefaultScreenshareStream());
VideoStream inactive_video_stream = DefaultStream();
inactive_video_stream.active = false;
@@ -262,7 +262,7 @@
TEST_F(VideoCodecInitializerTest, HighFpsSimulcastVp8Screenshare) {
// Two simulcast streams, the lower one using legacy settings (two temporal
// streams, 5fps), the higher one using 3 temporal streams and 30fps.
- SetUpFor(VideoCodecType::kVideoCodecVP8, 2, absl::nullopt, 3, true);
+ SetUpFor(VideoCodecType::kVideoCodecVP8, 2, std::nullopt, 3, true);
streams_.push_back(DefaultScreenshareStream());
VideoStream video_stream = DefaultStream();
video_stream.num_temporal_layers = 3;
@@ -286,7 +286,7 @@
}
TEST_F(VideoCodecInitializerTest, Vp9SvcDefaultLayering) {
- SetUpFor(VideoCodecType::kVideoCodecVP9, absl::nullopt, 3, 3, false);
+ SetUpFor(VideoCodecType::kVideoCodecVP9, std::nullopt, 3, 3, false);
VideoStream stream = DefaultStream();
stream.num_temporal_layers = 3;
streams_.push_back(stream);
@@ -297,7 +297,7 @@
}
TEST_F(VideoCodecInitializerTest, Vp9SvcAdjustedLayering) {
- SetUpFor(VideoCodecType::kVideoCodecVP9, absl::nullopt, 3, 3, false);
+ SetUpFor(VideoCodecType::kVideoCodecVP9, std::nullopt, 3, 3, false);
VideoStream stream = DefaultStream();
stream.num_temporal_layers = 3;
// Set resolution which is only enough to produce 2 spatial layers.
@@ -312,7 +312,7 @@
TEST_F(VideoCodecInitializerTest,
Vp9SingleSpatialLayerMaxBitrateIsEqualToCodecMaxBitrate) {
- SetUpFor(VideoCodecType::kVideoCodecVP9, absl::nullopt, 1, 3, false);
+ SetUpFor(VideoCodecType::kVideoCodecVP9, std::nullopt, 1, 3, false);
VideoStream stream = DefaultStream();
stream.num_temporal_layers = 3;
streams_.push_back(stream);
@@ -340,7 +340,7 @@
TEST_F(VideoCodecInitializerTest,
Vp9SingleSpatialLayerTargetBitrateIsEqualToCodecMaxBitrate) {
- SetUpFor(VideoCodecType::kVideoCodecVP9, absl::nullopt, 1, 1, true);
+ SetUpFor(VideoCodecType::kVideoCodecVP9, std::nullopt, 1, 1, true);
VideoStream stream = DefaultStream();
stream.num_temporal_layers = 1;
streams_.push_back(stream);
@@ -355,7 +355,7 @@
// Request 3 spatial layers for 320x180 input. Actual number of layers will be
// reduced to 1 due to low input resolution but SVC bitrate limits should be
// applied.
- SetUpFor(VideoCodecType::kVideoCodecVP9, absl::nullopt, 3, 3, false);
+ SetUpFor(VideoCodecType::kVideoCodecVP9, std::nullopt, 3, 3, false);
VideoStream stream = DefaultStream();
stream.width = 320;
stream.height = 180;
@@ -389,7 +389,7 @@
}
TEST_F(VideoCodecInitializerTest, Vp9DeactivateLayers) {
- SetUpFor(VideoCodecType::kVideoCodecVP9, absl::nullopt, 3, 1, false);
+ SetUpFor(VideoCodecType::kVideoCodecVP9, std::nullopt, 3, 1, false);
VideoStream stream = DefaultStream();
streams_.push_back(stream);
@@ -463,7 +463,7 @@
}
TEST_F(VideoCodecInitializerTest, Vp9SvcResolutionAlignment) {
- SetUpFor(VideoCodecType::kVideoCodecVP9, absl::nullopt, 3, 3, false);
+ SetUpFor(VideoCodecType::kVideoCodecVP9, std::nullopt, 3, 3, false);
VideoStream stream = DefaultStream();
stream.width = 1281;
stream.height = 721;
diff --git a/modules/video_coding/video_coding_impl.cc b/modules/video_coding/video_coding_impl.cc
index 8799d9b..5c62c49 100644
--- a/modules/video_coding/video_coding_impl.cc
+++ b/modules/video_coding/video_coding_impl.cc
@@ -55,7 +55,7 @@
// frame after RegisterReceiveCodec).
if (current_decoder_ && current_decoder_->IsSameDecoder(it->second)) {
// Release it if it was registered and in use.
- current_decoder_ = absl::nullopt;
+ current_decoder_ = std::nullopt;
}
VideoDecoder* ret = it->second;
decoders_.erase(it);
@@ -85,7 +85,7 @@
const VideoDecoder::Settings& settings) {
// If payload value already exists, erase old and insert new.
if (payload_type == current_payload_type_) {
- current_payload_type_ = absl::nullopt;
+ current_payload_type_ = std::nullopt;
}
decoder_settings_[payload_type] = settings;
}
@@ -97,7 +97,7 @@
}
if (payload_type == current_payload_type_) {
// This codec is currently in use.
- current_payload_type_ = absl::nullopt;
+ current_payload_type_ = std::nullopt;
}
return true;
}
@@ -113,12 +113,12 @@
}
// If decoder exists - delete.
if (current_decoder_.has_value()) {
- current_decoder_ = absl::nullopt;
- current_payload_type_ = absl::nullopt;
+ current_decoder_ = std::nullopt;
+ current_payload_type_ = std::nullopt;
}
CreateAndInitDecoder(frame);
- if (current_decoder_ == absl::nullopt) {
+ if (current_decoder_ == std::nullopt) {
return nullptr;
}
@@ -126,7 +126,7 @@
callback->OnIncomingPayloadType(payload_type);
if (current_decoder_->RegisterDecodeCompleteCallback(decoded_frame_callback) <
0) {
- current_decoder_ = absl::nullopt;
+ current_decoder_ = std::nullopt;
return nullptr;
}
@@ -162,7 +162,7 @@
decoder_item->second.set_max_render_resolution(frame_resolution);
}
if (!current_decoder_->Configure(decoder_item->second)) {
- current_decoder_ = absl::nullopt;
+ current_decoder_ = std::nullopt;
RTC_LOG(LS_ERROR) << "Failed to initialize decoder.";
}
}
diff --git a/modules/video_coding/video_coding_impl.h b/modules/video_coding/video_coding_impl.h
index b715d58..7f6bb10 100644
--- a/modules/video_coding/video_coding_impl.h
+++ b/modules/video_coding/video_coding_impl.h
@@ -13,10 +13,10 @@
#include <map>
#include <memory>
+#include <optional>
#include <string>
#include <vector>
-#include "absl/types/optional.h"
#include "api/field_trials_view.h"
#include "api/sequence_checker.h"
#include "modules/video_coding/deprecated/frame_buffer.h"
@@ -89,8 +89,8 @@
SequenceChecker decoder_sequence_checker_;
- absl::optional<uint8_t> current_payload_type_;
- absl::optional<VCMGenericDecoder> current_decoder_
+ std::optional<uint8_t> current_payload_type_;
+ std::optional<VCMGenericDecoder> current_decoder_
RTC_GUARDED_BY(decoder_sequence_checker_);
// Initialization paramaters for decoders keyed by payload type.
std::map<uint8_t, VideoDecoder::Settings> decoder_settings_;
diff --git a/modules/video_coding/video_receiver2_unittest.cc b/modules/video_coding/video_receiver2_unittest.cc
index 88a19df..48ae8af 100644
--- a/modules/video_coding/video_receiver2_unittest.cc
+++ b/modules/video_coding/video_receiver2_unittest.cc
@@ -38,7 +38,7 @@
MOCK_METHOD(int32_t,
FrameToRender,
(VideoFrame&,
- absl::optional<uint8_t>,
+ std::optional<uint8_t>,
TimeDelta,
VideoContentType,
VideoFrameType),
diff --git a/modules/video_coding/video_receiver_unittest.cc b/modules/video_coding/video_receiver_unittest.cc
index 2063653..7a04826 100644
--- a/modules/video_coding/video_receiver_unittest.cc
+++ b/modules/video_coding/video_receiver_unittest.cc
@@ -41,7 +41,7 @@
MOCK_METHOD(int32_t,
FrameToRender,
(VideoFrame&,
- absl::optional<uint8_t>,
+ std::optional<uint8_t>,
TimeDelta,
VideoContentType,
VideoFrameType),
diff --git a/net/dcsctp/fuzzers/dcsctp_fuzzers.cc b/net/dcsctp/fuzzers/dcsctp_fuzzers.cc
index e8fcacf..b70c6c3 100644
--- a/net/dcsctp/fuzzers/dcsctp_fuzzers.cc
+++ b/net/dcsctp/fuzzers/dcsctp_fuzzers.cc
@@ -434,7 +434,7 @@
SendOptions options;
options.unordered = IsUnordered(flags & 0x01);
options.max_retransmissions =
- (flags & 0x02) != 0 ? absl::make_optional(0) : absl::nullopt;
+ (flags & 0x02) != 0 ? std::make_optional(0) : std::nullopt;
options.lifecycle_id = LifecycleId(42);
size_t payload_exponent = (flags >> 2) % 16;
size_t payload_size = static_cast<size_t>(1) << payload_exponent;
@@ -446,7 +446,7 @@
case 7: {
// Expire an active timeout/timer.
uint8_t timeout_idx = state.GetByte();
- absl::optional<TimeoutID> timeout_id = cb.ExpireTimeout(timeout_idx);
+ std::optional<TimeoutID> timeout_id = cb.ExpireTimeout(timeout_idx);
if (timeout_id.has_value()) {
socket.HandleTimeout(*timeout_id);
}
diff --git a/net/dcsctp/fuzzers/dcsctp_fuzzers.h b/net/dcsctp/fuzzers/dcsctp_fuzzers.h
index 49aa7f0..d76d0c0 100644
--- a/net/dcsctp/fuzzers/dcsctp_fuzzers.h
+++ b/net/dcsctp/fuzzers/dcsctp_fuzzers.h
@@ -44,13 +44,13 @@
// expired.
RTC_DCHECK(timeout_id_.has_value());
RTC_DCHECK(active_timeouts_.erase(*timeout_id_) == 1);
- timeout_id_ = absl::nullopt;
+ timeout_id_ = std::nullopt;
}
// A set of all active timeouts, managed by `FuzzerCallbacks`.
std::set<TimeoutID>& active_timeouts_;
// If present, the timout is active and will expire reported as `timeout_id`.
- absl::optional<TimeoutID> timeout_id_;
+ std::optional<TimeoutID> timeout_id_;
};
class FuzzerCallbacks : public DcSctpSocketCallbacks {
@@ -91,7 +91,7 @@
}
// Given an index among the active timeouts, will expire that one.
- absl::optional<TimeoutID> ExpireTimeout(size_t index) {
+ std::optional<TimeoutID> ExpireTimeout(size_t index) {
if (index < active_timeouts_.size()) {
auto it = active_timeouts_.begin();
std::advance(it, index);
@@ -99,7 +99,7 @@
active_timeouts_.erase(it);
return timeout_id;
}
- return absl::nullopt;
+ return std::nullopt;
}
private:
diff --git a/net/dcsctp/packet/BUILD.gn b/net/dcsctp/packet/BUILD.gn
index 942f039..dcdff08 100644
--- a/net/dcsctp/packet/BUILD.gn
+++ b/net/dcsctp/packet/BUILD.gn
@@ -29,7 +29,6 @@
"../../../api:array_view",
"../../../rtc_base:checks",
"../../../rtc_base:logging",
- "//third_party/abseil-cpp/absl/types:optional",
]
sources = [
"tlv_trait.cc",
@@ -73,7 +72,6 @@
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
sources = [
"parameter/add_incoming_streams_request_parameter.cc",
@@ -118,7 +116,6 @@
"../public:types",
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
sources = [
"error_cause/cookie_received_while_shutting_down_cause.cc",
@@ -166,7 +163,6 @@
"../packet:bounded_io",
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
sources = [
"chunk/abort_chunk.cc",
@@ -235,7 +231,6 @@
"../common:math",
"../public:types",
"//third_party/abseil-cpp/absl/memory",
- "//third_party/abseil-cpp/absl/types:optional",
]
sources = [
"sctp_packet.cc",
@@ -265,7 +260,6 @@
"../common:math",
"../public:types",
"../testing:testing_macros",
- "//third_party/abseil-cpp/absl/types:optional",
]
sources = [
"bounded_byte_reader_test.cc",
diff --git a/net/dcsctp/packet/chunk/abort_chunk.cc b/net/dcsctp/packet/chunk/abort_chunk.cc
index 8348eb9..3d0e469 100644
--- a/net/dcsctp/packet/chunk/abort_chunk.cc
+++ b/net/dcsctp/packet/chunk/abort_chunk.cc
@@ -11,10 +11,10 @@
#include <stdint.h>
+#include <optional>
#include <utility>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "net/dcsctp/packet/bounded_byte_reader.h"
#include "net/dcsctp/packet/bounded_byte_writer.h"
@@ -36,16 +36,16 @@
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
constexpr int AbortChunk::kType;
-absl::optional<AbortChunk> AbortChunk::Parse(
+std::optional<AbortChunk> AbortChunk::Parse(
rtc::ArrayView<const uint8_t> data) {
- absl::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
+ std::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
if (!reader.has_value()) {
- return absl::nullopt;
+ return std::nullopt;
}
- absl::optional<Parameters> error_causes =
+ std::optional<Parameters> error_causes =
Parameters::Parse(reader->variable_data());
if (!error_causes.has_value()) {
- return absl::nullopt;
+ return std::nullopt;
}
uint8_t flags = reader->Load8<1>();
bool filled_in_verification_tag = (flags & (1 << kFlagsBitT)) == 0;
diff --git a/net/dcsctp/packet/chunk/abort_chunk.h b/net/dcsctp/packet/chunk/abort_chunk.h
index 1408a75..2072a9b 100644
--- a/net/dcsctp/packet/chunk/abort_chunk.h
+++ b/net/dcsctp/packet/chunk/abort_chunk.h
@@ -42,7 +42,7 @@
AbortChunk(AbortChunk&& other) = default;
AbortChunk& operator=(AbortChunk&& other) = default;
- static absl::optional<AbortChunk> Parse(rtc::ArrayView<const uint8_t> data);
+ static std::optional<AbortChunk> Parse(rtc::ArrayView<const uint8_t> data);
void SerializeTo(std::vector<uint8_t>& out) const override;
std::string ToString() const override;
diff --git a/net/dcsctp/packet/chunk/chunk.cc b/net/dcsctp/packet/chunk/chunk.cc
index 832ab82..b2059b4 100644
--- a/net/dcsctp/packet/chunk/chunk.cc
+++ b/net/dcsctp/packet/chunk/chunk.cc
@@ -11,9 +11,9 @@
#include <cstdint>
#include <memory>
+#include <optional>
#include <utility>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "net/dcsctp/common/math.h"
#include "net/dcsctp/packet/chunk/abort_chunk.h"
@@ -42,7 +42,7 @@
rtc::ArrayView<const uint8_t> data,
rtc::StringBuilder& sb) {
if (chunk_type == Chunk::kType) {
- absl::optional<Chunk> c = Chunk::Parse(data);
+ std::optional<Chunk> c = Chunk::Parse(data);
if (c.has_value()) {
sb << c->ToString();
} else {
diff --git a/net/dcsctp/packet/chunk/chunk.h b/net/dcsctp/packet/chunk/chunk.h
index 687aa1d..6fea10a 100644
--- a/net/dcsctp/packet/chunk/chunk.h
+++ b/net/dcsctp/packet/chunk/chunk.h
@@ -16,13 +16,13 @@
#include <cstdint>
#include <iterator>
#include <memory>
+#include <optional>
#include <string>
#include <utility>
#include <vector>
#include "absl/algorithm/container.h"
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "net/dcsctp/packet/data.h"
#include "net/dcsctp/packet/error_cause/error_cause.h"
diff --git a/net/dcsctp/packet/chunk/cookie_ack_chunk.cc b/net/dcsctp/packet/chunk/cookie_ack_chunk.cc
index 4839969..a6d28df 100644
--- a/net/dcsctp/packet/chunk/cookie_ack_chunk.cc
+++ b/net/dcsctp/packet/chunk/cookie_ack_chunk.cc
@@ -11,9 +11,9 @@
#include <stdint.h>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
namespace dcsctp {
@@ -27,10 +27,10 @@
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
constexpr int CookieAckChunk::kType;
-absl::optional<CookieAckChunk> CookieAckChunk::Parse(
+std::optional<CookieAckChunk> CookieAckChunk::Parse(
rtc::ArrayView<const uint8_t> data) {
if (!ParseTLV(data).has_value()) {
- return absl::nullopt;
+ return std::nullopt;
}
return CookieAckChunk();
}
diff --git a/net/dcsctp/packet/chunk/cookie_ack_chunk.h b/net/dcsctp/packet/chunk/cookie_ack_chunk.h
index f7d4a33..631e7a2 100644
--- a/net/dcsctp/packet/chunk/cookie_ack_chunk.h
+++ b/net/dcsctp/packet/chunk/cookie_ack_chunk.h
@@ -35,7 +35,7 @@
CookieAckChunk() {}
- static absl::optional<CookieAckChunk> Parse(
+ static std::optional<CookieAckChunk> Parse(
rtc::ArrayView<const uint8_t> data);
void SerializeTo(std::vector<uint8_t>& out) const override;
diff --git a/net/dcsctp/packet/chunk/cookie_echo_chunk.cc b/net/dcsctp/packet/chunk/cookie_echo_chunk.cc
index a01d0b1..30e2b89 100644
--- a/net/dcsctp/packet/chunk/cookie_echo_chunk.cc
+++ b/net/dcsctp/packet/chunk/cookie_echo_chunk.cc
@@ -11,10 +11,10 @@
#include <stdint.h>
+#include <optional>
#include <type_traits>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "net/dcsctp/packet/bounded_byte_reader.h"
#include "net/dcsctp/packet/bounded_byte_writer.h"
@@ -34,11 +34,11 @@
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
constexpr int CookieEchoChunk::kType;
-absl::optional<CookieEchoChunk> CookieEchoChunk::Parse(
+std::optional<CookieEchoChunk> CookieEchoChunk::Parse(
rtc::ArrayView<const uint8_t> data) {
- absl::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
+ std::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
if (!reader.has_value()) {
- return absl::nullopt;
+ return std::nullopt;
}
return CookieEchoChunk(reader->variable_data());
}
diff --git a/net/dcsctp/packet/chunk/cookie_echo_chunk.h b/net/dcsctp/packet/chunk/cookie_echo_chunk.h
index 8cb8052..12f819b 100644
--- a/net/dcsctp/packet/chunk/cookie_echo_chunk.h
+++ b/net/dcsctp/packet/chunk/cookie_echo_chunk.h
@@ -36,7 +36,7 @@
explicit CookieEchoChunk(rtc::ArrayView<const uint8_t> cookie)
: cookie_(cookie.begin(), cookie.end()) {}
- static absl::optional<CookieEchoChunk> Parse(
+ static std::optional<CookieEchoChunk> Parse(
rtc::ArrayView<const uint8_t> data);
void SerializeTo(std::vector<uint8_t>& out) const override;
diff --git a/net/dcsctp/packet/chunk/data_chunk.cc b/net/dcsctp/packet/chunk/data_chunk.cc
index cf866b7..92a988d 100644
--- a/net/dcsctp/packet/chunk/data_chunk.cc
+++ b/net/dcsctp/packet/chunk/data_chunk.cc
@@ -11,11 +11,11 @@
#include <stdint.h>
+#include <optional>
#include <string>
#include <type_traits>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "net/dcsctp/packet/bounded_byte_reader.h"
#include "net/dcsctp/packet/bounded_byte_writer.h"
@@ -43,10 +43,10 @@
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
constexpr int DataChunk::kType;
-absl::optional<DataChunk> DataChunk::Parse(rtc::ArrayView<const uint8_t> data) {
- absl::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
+std::optional<DataChunk> DataChunk::Parse(rtc::ArrayView<const uint8_t> data) {
+ std::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
if (!reader.has_value()) {
- return absl::nullopt;
+ return std::nullopt;
}
uint8_t flags = reader->Load8<1>();
diff --git a/net/dcsctp/packet/chunk/data_chunk.h b/net/dcsctp/packet/chunk/data_chunk.h
index 12bb05f..515fd9e 100644
--- a/net/dcsctp/packet/chunk/data_chunk.h
+++ b/net/dcsctp/packet/chunk/data_chunk.h
@@ -59,7 +59,7 @@
DataChunk(TSN tsn, Data&& data, bool immediate_ack)
: AnyDataChunk(tsn, std::move(data), immediate_ack) {}
- static absl::optional<DataChunk> Parse(rtc::ArrayView<const uint8_t> data);
+ static std::optional<DataChunk> Parse(rtc::ArrayView<const uint8_t> data);
void SerializeTo(std::vector<uint8_t>& out) const override;
std::string ToString() const override;
diff --git a/net/dcsctp/packet/chunk/error_chunk.cc b/net/dcsctp/packet/chunk/error_chunk.cc
index baac0c5..8311ad2 100644
--- a/net/dcsctp/packet/chunk/error_chunk.cc
+++ b/net/dcsctp/packet/chunk/error_chunk.cc
@@ -11,10 +11,10 @@
#include <stdint.h>
+#include <optional>
#include <utility>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "net/dcsctp/packet/bounded_byte_reader.h"
#include "net/dcsctp/packet/bounded_byte_writer.h"
@@ -36,16 +36,16 @@
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
constexpr int ErrorChunk::kType;
-absl::optional<ErrorChunk> ErrorChunk::Parse(
+std::optional<ErrorChunk> ErrorChunk::Parse(
rtc::ArrayView<const uint8_t> data) {
- absl::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
+ std::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
if (!reader.has_value()) {
- return absl::nullopt;
+ return std::nullopt;
}
- absl::optional<Parameters> error_causes =
+ std::optional<Parameters> error_causes =
Parameters::Parse(reader->variable_data());
if (!error_causes.has_value()) {
- return absl::nullopt;
+ return std::nullopt;
}
return ErrorChunk(*std::move(error_causes));
}
diff --git a/net/dcsctp/packet/chunk/error_chunk.h b/net/dcsctp/packet/chunk/error_chunk.h
index 96122cf..f982547 100644
--- a/net/dcsctp/packet/chunk/error_chunk.h
+++ b/net/dcsctp/packet/chunk/error_chunk.h
@@ -41,7 +41,7 @@
ErrorChunk(ErrorChunk&& other) = default;
ErrorChunk& operator=(ErrorChunk&& other) = default;
- static absl::optional<ErrorChunk> Parse(rtc::ArrayView<const uint8_t> data);
+ static std::optional<ErrorChunk> Parse(rtc::ArrayView<const uint8_t> data);
void SerializeTo(std::vector<uint8_t>& out) const override;
std::string ToString() const override;
diff --git a/net/dcsctp/packet/chunk/forward_tsn_chunk.cc b/net/dcsctp/packet/chunk/forward_tsn_chunk.cc
index e432114..1f22e35 100644
--- a/net/dcsctp/packet/chunk/forward_tsn_chunk.cc
+++ b/net/dcsctp/packet/chunk/forward_tsn_chunk.cc
@@ -12,11 +12,11 @@
#include <stddef.h>
#include <stdint.h>
+#include <optional>
#include <string>
#include <utility>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "net/dcsctp/packet/bounded_byte_reader.h"
#include "net/dcsctp/packet/bounded_byte_writer.h"
@@ -44,11 +44,11 @@
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
constexpr int ForwardTsnChunk::kType;
-absl::optional<ForwardTsnChunk> ForwardTsnChunk::Parse(
+std::optional<ForwardTsnChunk> ForwardTsnChunk::Parse(
rtc::ArrayView<const uint8_t> data) {
- absl::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
+ std::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
if (!reader.has_value()) {
- return absl::nullopt;
+ return std::nullopt;
}
TSN new_cumulative_tsn(reader->Load32<4>());
diff --git a/net/dcsctp/packet/chunk/forward_tsn_chunk.h b/net/dcsctp/packet/chunk/forward_tsn_chunk.h
index b9ef666..caa3379 100644
--- a/net/dcsctp/packet/chunk/forward_tsn_chunk.h
+++ b/net/dcsctp/packet/chunk/forward_tsn_chunk.h
@@ -40,7 +40,7 @@
std::vector<SkippedStream> skipped_streams)
: AnyForwardTsnChunk(new_cumulative_tsn, std::move(skipped_streams)) {}
- static absl::optional<ForwardTsnChunk> Parse(
+ static std::optional<ForwardTsnChunk> Parse(
rtc::ArrayView<const uint8_t> data);
void SerializeTo(std::vector<uint8_t>& out) const override;
diff --git a/net/dcsctp/packet/chunk/heartbeat_ack_chunk.cc b/net/dcsctp/packet/chunk/heartbeat_ack_chunk.cc
index 3cbcd09..ba98436 100644
--- a/net/dcsctp/packet/chunk/heartbeat_ack_chunk.cc
+++ b/net/dcsctp/packet/chunk/heartbeat_ack_chunk.cc
@@ -11,10 +11,10 @@
#include <stdint.h>
+#include <optional>
#include <utility>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "net/dcsctp/packet/bounded_byte_reader.h"
#include "net/dcsctp/packet/bounded_byte_writer.h"
@@ -36,17 +36,17 @@
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
constexpr int HeartbeatAckChunk::kType;
-absl::optional<HeartbeatAckChunk> HeartbeatAckChunk::Parse(
+std::optional<HeartbeatAckChunk> HeartbeatAckChunk::Parse(
rtc::ArrayView<const uint8_t> data) {
- absl::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
+ std::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
if (!reader.has_value()) {
- return absl::nullopt;
+ return std::nullopt;
}
- absl::optional<Parameters> parameters =
+ std::optional<Parameters> parameters =
Parameters::Parse(reader->variable_data());
if (!parameters.has_value()) {
- return absl::nullopt;
+ return std::nullopt;
}
return HeartbeatAckChunk(*std::move(parameters));
}
diff --git a/net/dcsctp/packet/chunk/heartbeat_ack_chunk.h b/net/dcsctp/packet/chunk/heartbeat_ack_chunk.h
index a6479f7..8a63205 100644
--- a/net/dcsctp/packet/chunk/heartbeat_ack_chunk.h
+++ b/net/dcsctp/packet/chunk/heartbeat_ack_chunk.h
@@ -43,7 +43,7 @@
HeartbeatAckChunk(HeartbeatAckChunk&& other) = default;
HeartbeatAckChunk& operator=(HeartbeatAckChunk&& other) = default;
- static absl::optional<HeartbeatAckChunk> Parse(
+ static std::optional<HeartbeatAckChunk> Parse(
rtc::ArrayView<const uint8_t> data);
void SerializeTo(std::vector<uint8_t>& out) const override;
@@ -51,7 +51,7 @@
const Parameters& parameters() const { return parameters_; }
- absl::optional<HeartbeatInfoParameter> info() const {
+ std::optional<HeartbeatInfoParameter> info() const {
return parameters_.get<HeartbeatInfoParameter>();
}
diff --git a/net/dcsctp/packet/chunk/heartbeat_request_chunk.cc b/net/dcsctp/packet/chunk/heartbeat_request_chunk.cc
index d759d6b..a94a7e3 100644
--- a/net/dcsctp/packet/chunk/heartbeat_request_chunk.cc
+++ b/net/dcsctp/packet/chunk/heartbeat_request_chunk.cc
@@ -11,10 +11,10 @@
#include <stdint.h>
+#include <optional>
#include <utility>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "net/dcsctp/packet/bounded_byte_reader.h"
#include "net/dcsctp/packet/bounded_byte_writer.h"
@@ -36,17 +36,17 @@
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
constexpr int HeartbeatRequestChunk::kType;
-absl::optional<HeartbeatRequestChunk> HeartbeatRequestChunk::Parse(
+std::optional<HeartbeatRequestChunk> HeartbeatRequestChunk::Parse(
rtc::ArrayView<const uint8_t> data) {
- absl::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
+ std::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
if (!reader.has_value()) {
- return absl::nullopt;
+ return std::nullopt;
}
- absl::optional<Parameters> parameters =
+ std::optional<Parameters> parameters =
Parameters::Parse(reader->variable_data());
if (!parameters.has_value()) {
- return absl::nullopt;
+ return std::nullopt;
}
return HeartbeatRequestChunk(*std::move(parameters));
}
diff --git a/net/dcsctp/packet/chunk/heartbeat_request_chunk.h b/net/dcsctp/packet/chunk/heartbeat_request_chunk.h
index fe2ce19..218bd6a 100644
--- a/net/dcsctp/packet/chunk/heartbeat_request_chunk.h
+++ b/net/dcsctp/packet/chunk/heartbeat_request_chunk.h
@@ -42,7 +42,7 @@
HeartbeatRequestChunk(HeartbeatRequestChunk&& other) = default;
HeartbeatRequestChunk& operator=(HeartbeatRequestChunk&& other) = default;
- static absl::optional<HeartbeatRequestChunk> Parse(
+ static std::optional<HeartbeatRequestChunk> Parse(
rtc::ArrayView<const uint8_t> data);
void SerializeTo(std::vector<uint8_t>& out) const override;
@@ -50,7 +50,7 @@
const Parameters& parameters() const { return parameters_; }
Parameters extract_parameters() && { return std::move(parameters_); }
- absl::optional<HeartbeatInfoParameter> info() const {
+ std::optional<HeartbeatInfoParameter> info() const {
return parameters_.get<HeartbeatInfoParameter>();
}
diff --git a/net/dcsctp/packet/chunk/idata_chunk.cc b/net/dcsctp/packet/chunk/idata_chunk.cc
index e9a777d..b6d3b3e 100644
--- a/net/dcsctp/packet/chunk/idata_chunk.cc
+++ b/net/dcsctp/packet/chunk/idata_chunk.cc
@@ -11,11 +11,11 @@
#include <stdint.h>
+#include <optional>
#include <string>
#include <type_traits>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "net/dcsctp/packet/bounded_byte_reader.h"
#include "net/dcsctp/packet/bounded_byte_writer.h"
@@ -45,11 +45,11 @@
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
constexpr int IDataChunk::kType;
-absl::optional<IDataChunk> IDataChunk::Parse(
+std::optional<IDataChunk> IDataChunk::Parse(
rtc::ArrayView<const uint8_t> data) {
- absl::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
+ std::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
if (!reader.has_value()) {
- return absl::nullopt;
+ return std::nullopt;
}
uint8_t flags = reader->Load8<1>();
TSN tsn(reader->Load32<4>());
diff --git a/net/dcsctp/packet/chunk/idata_chunk.h b/net/dcsctp/packet/chunk/idata_chunk.h
index 36b7e7e..c8b6709 100644
--- a/net/dcsctp/packet/chunk/idata_chunk.h
+++ b/net/dcsctp/packet/chunk/idata_chunk.h
@@ -59,7 +59,7 @@
explicit IDataChunk(TSN tsn, Data&& data, bool immediate_ack)
: AnyDataChunk(tsn, std::move(data), immediate_ack) {}
- static absl::optional<IDataChunk> Parse(rtc::ArrayView<const uint8_t> data);
+ static std::optional<IDataChunk> Parse(rtc::ArrayView<const uint8_t> data);
void SerializeTo(std::vector<uint8_t>& out) const override;
std::string ToString() const override;
diff --git a/net/dcsctp/packet/chunk/iforward_tsn_chunk.cc b/net/dcsctp/packet/chunk/iforward_tsn_chunk.cc
index 2b8e9c9..cc83b3b 100644
--- a/net/dcsctp/packet/chunk/iforward_tsn_chunk.cc
+++ b/net/dcsctp/packet/chunk/iforward_tsn_chunk.cc
@@ -12,11 +12,11 @@
#include <stddef.h>
#include <stdint.h>
+#include <optional>
#include <string>
#include <utility>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "net/dcsctp/packet/bounded_byte_reader.h"
#include "net/dcsctp/packet/bounded_byte_writer.h"
@@ -48,11 +48,11 @@
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
constexpr int IForwardTsnChunk::kType;
-absl::optional<IForwardTsnChunk> IForwardTsnChunk::Parse(
+std::optional<IForwardTsnChunk> IForwardTsnChunk::Parse(
rtc::ArrayView<const uint8_t> data) {
- absl::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
+ std::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
if (!reader.has_value()) {
- return absl::nullopt;
+ return std::nullopt;
}
TSN new_cumulative_tsn(reader->Load32<4>());
diff --git a/net/dcsctp/packet/chunk/iforward_tsn_chunk.h b/net/dcsctp/packet/chunk/iforward_tsn_chunk.h
index 54d23f7..fe05221 100644
--- a/net/dcsctp/packet/chunk/iforward_tsn_chunk.h
+++ b/net/dcsctp/packet/chunk/iforward_tsn_chunk.h
@@ -40,7 +40,7 @@
std::vector<SkippedStream> skipped_streams)
: AnyForwardTsnChunk(new_cumulative_tsn, std::move(skipped_streams)) {}
- static absl::optional<IForwardTsnChunk> Parse(
+ static std::optional<IForwardTsnChunk> Parse(
rtc::ArrayView<const uint8_t> data);
void SerializeTo(std::vector<uint8_t>& out) const override;
diff --git a/net/dcsctp/packet/chunk/init_ack_chunk.cc b/net/dcsctp/packet/chunk/init_ack_chunk.cc
index c7ef9da..7977faa 100644
--- a/net/dcsctp/packet/chunk/init_ack_chunk.cc
+++ b/net/dcsctp/packet/chunk/init_ack_chunk.cc
@@ -11,11 +11,11 @@
#include <stdint.h>
+#include <optional>
#include <string>
#include <utility>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "net/dcsctp/packet/bounded_byte_reader.h"
#include "net/dcsctp/packet/bounded_byte_writer.h"
@@ -46,11 +46,11 @@
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
constexpr int InitAckChunk::kType;
-absl::optional<InitAckChunk> InitAckChunk::Parse(
+std::optional<InitAckChunk> InitAckChunk::Parse(
rtc::ArrayView<const uint8_t> data) {
- absl::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
+ std::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
if (!reader.has_value()) {
- return absl::nullopt;
+ return std::nullopt;
}
VerificationTag initiate_tag(reader->Load32<4>());
@@ -58,10 +58,10 @@
uint16_t nbr_outbound_streams = reader->Load16<12>();
uint16_t nbr_inbound_streams = reader->Load16<14>();
TSN initial_tsn(reader->Load32<16>());
- absl::optional<Parameters> parameters =
+ std::optional<Parameters> parameters =
Parameters::Parse(reader->variable_data());
if (!parameters.has_value()) {
- return absl::nullopt;
+ return std::nullopt;
}
return InitAckChunk(initiate_tag, a_rwnd, nbr_outbound_streams,
nbr_inbound_streams, initial_tsn, *std::move(parameters));
diff --git a/net/dcsctp/packet/chunk/init_ack_chunk.h b/net/dcsctp/packet/chunk/init_ack_chunk.h
index 6fcf64b..5058cf1 100644
--- a/net/dcsctp/packet/chunk/init_ack_chunk.h
+++ b/net/dcsctp/packet/chunk/init_ack_chunk.h
@@ -51,7 +51,7 @@
InitAckChunk(InitAckChunk&& other) = default;
InitAckChunk& operator=(InitAckChunk&& other) = default;
- static absl::optional<InitAckChunk> Parse(rtc::ArrayView<const uint8_t> data);
+ static std::optional<InitAckChunk> Parse(rtc::ArrayView<const uint8_t> data);
void SerializeTo(std::vector<uint8_t>& out) const override;
std::string ToString() const override;
diff --git a/net/dcsctp/packet/chunk/init_chunk.cc b/net/dcsctp/packet/chunk/init_chunk.cc
index 8030107..dbb3382 100644
--- a/net/dcsctp/packet/chunk/init_chunk.cc
+++ b/net/dcsctp/packet/chunk/init_chunk.cc
@@ -11,11 +11,11 @@
#include <stdint.h>
+#include <optional>
#include <string>
#include <utility>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "net/dcsctp/packet/bounded_byte_reader.h"
#include "net/dcsctp/packet/bounded_byte_writer.h"
@@ -46,10 +46,10 @@
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
constexpr int InitChunk::kType;
-absl::optional<InitChunk> InitChunk::Parse(rtc::ArrayView<const uint8_t> data) {
- absl::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
+std::optional<InitChunk> InitChunk::Parse(rtc::ArrayView<const uint8_t> data) {
+ std::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
if (!reader.has_value()) {
- return absl::nullopt;
+ return std::nullopt;
}
VerificationTag initiate_tag(reader->Load32<4>());
@@ -58,10 +58,10 @@
uint16_t nbr_inbound_streams = reader->Load16<14>();
TSN initial_tsn(reader->Load32<16>());
- absl::optional<Parameters> parameters =
+ std::optional<Parameters> parameters =
Parameters::Parse(reader->variable_data());
if (!parameters.has_value()) {
- return absl::nullopt;
+ return std::nullopt;
}
return InitChunk(initiate_tag, a_rwnd, nbr_outbound_streams,
nbr_inbound_streams, initial_tsn, *std::move(parameters));
diff --git a/net/dcsctp/packet/chunk/init_chunk.h b/net/dcsctp/packet/chunk/init_chunk.h
index 38f9994..4f3a9b9 100644
--- a/net/dcsctp/packet/chunk/init_chunk.h
+++ b/net/dcsctp/packet/chunk/init_chunk.h
@@ -51,7 +51,7 @@
InitChunk(InitChunk&& other) = default;
InitChunk& operator=(InitChunk&& other) = default;
- static absl::optional<InitChunk> Parse(rtc::ArrayView<const uint8_t> data);
+ static std::optional<InitChunk> Parse(rtc::ArrayView<const uint8_t> data);
void SerializeTo(std::vector<uint8_t>& out) const override;
std::string ToString() const override;
diff --git a/net/dcsctp/packet/chunk/reconfig_chunk.cc b/net/dcsctp/packet/chunk/reconfig_chunk.cc
index f39f3b6..91fa7d7 100644
--- a/net/dcsctp/packet/chunk/reconfig_chunk.cc
+++ b/net/dcsctp/packet/chunk/reconfig_chunk.cc
@@ -11,10 +11,10 @@
#include <stdint.h>
+#include <optional>
#include <utility>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "net/dcsctp/packet/bounded_byte_reader.h"
#include "net/dcsctp/packet/bounded_byte_writer.h"
@@ -40,17 +40,17 @@
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
constexpr int ReConfigChunk::kType;
-absl::optional<ReConfigChunk> ReConfigChunk::Parse(
+std::optional<ReConfigChunk> ReConfigChunk::Parse(
rtc::ArrayView<const uint8_t> data) {
- absl::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
+ std::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
if (!reader.has_value()) {
- return absl::nullopt;
+ return std::nullopt;
}
- absl::optional<Parameters> parameters =
+ std::optional<Parameters> parameters =
Parameters::Parse(reader->variable_data());
if (!parameters.has_value()) {
- return absl::nullopt;
+ return std::nullopt;
}
return ReConfigChunk(*std::move(parameters));
diff --git a/net/dcsctp/packet/chunk/reconfig_chunk.h b/net/dcsctp/packet/chunk/reconfig_chunk.h
index 9d2539a..90df22e 100644
--- a/net/dcsctp/packet/chunk/reconfig_chunk.h
+++ b/net/dcsctp/packet/chunk/reconfig_chunk.h
@@ -38,8 +38,7 @@
explicit ReConfigChunk(Parameters parameters)
: parameters_(std::move(parameters)) {}
- static absl::optional<ReConfigChunk> Parse(
- rtc::ArrayView<const uint8_t> data);
+ static std::optional<ReConfigChunk> Parse(rtc::ArrayView<const uint8_t> data);
void SerializeTo(std::vector<uint8_t>& out) const override;
std::string ToString() const override;
diff --git a/net/dcsctp/packet/chunk/sack_chunk.cc b/net/dcsctp/packet/chunk/sack_chunk.cc
index 179f7ea..75fd9b5 100644
--- a/net/dcsctp/packet/chunk/sack_chunk.cc
+++ b/net/dcsctp/packet/chunk/sack_chunk.cc
@@ -12,11 +12,11 @@
#include <stddef.h>
#include <cstdint>
+#include <optional>
#include <string>
#include <type_traits>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "net/dcsctp/packet/bounded_byte_reader.h"
#include "net/dcsctp/packet/bounded_byte_writer.h"
@@ -58,10 +58,10 @@
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
constexpr int SackChunk::kType;
-absl::optional<SackChunk> SackChunk::Parse(rtc::ArrayView<const uint8_t> data) {
- absl::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
+std::optional<SackChunk> SackChunk::Parse(rtc::ArrayView<const uint8_t> data) {
+ std::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
if (!reader.has_value()) {
- return absl::nullopt;
+ return std::nullopt;
}
TSN tsn_ack(reader->Load32<4>());
@@ -72,7 +72,7 @@
if (reader->variable_data_size() != nbr_of_gap_blocks * kGapAckBlockSize +
nbr_of_dup_tsns * kDupTsnBlockSize) {
RTC_DLOG(LS_WARNING) << "Invalid number of gap blocks or duplicate TSNs";
- return absl::nullopt;
+ return std::nullopt;
}
std::vector<GapAckBlock> gap_ack_blocks;
diff --git a/net/dcsctp/packet/chunk/sack_chunk.h b/net/dcsctp/packet/chunk/sack_chunk.h
index e6758fa..ae4807d 100644
--- a/net/dcsctp/packet/chunk/sack_chunk.h
+++ b/net/dcsctp/packet/chunk/sack_chunk.h
@@ -54,7 +54,7 @@
a_rwnd_(a_rwnd),
gap_ack_blocks_(std::move(gap_ack_blocks)),
duplicate_tsns_(std::move(duplicate_tsns)) {}
- static absl::optional<SackChunk> Parse(rtc::ArrayView<const uint8_t> data);
+ static std::optional<SackChunk> Parse(rtc::ArrayView<const uint8_t> data);
void SerializeTo(std::vector<uint8_t>& out) const override;
std::string ToString() const override;
diff --git a/net/dcsctp/packet/chunk/shutdown_ack_chunk.cc b/net/dcsctp/packet/chunk/shutdown_ack_chunk.cc
index d42acee..3df4e5a 100644
--- a/net/dcsctp/packet/chunk/shutdown_ack_chunk.cc
+++ b/net/dcsctp/packet/chunk/shutdown_ack_chunk.cc
@@ -11,9 +11,9 @@
#include <stdint.h>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
namespace dcsctp {
@@ -27,10 +27,10 @@
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
constexpr int ShutdownAckChunk::kType;
-absl::optional<ShutdownAckChunk> ShutdownAckChunk::Parse(
+std::optional<ShutdownAckChunk> ShutdownAckChunk::Parse(
rtc::ArrayView<const uint8_t> data) {
if (!ParseTLV(data).has_value()) {
- return absl::nullopt;
+ return std::nullopt;
}
return ShutdownAckChunk();
}
diff --git a/net/dcsctp/packet/chunk/shutdown_ack_chunk.h b/net/dcsctp/packet/chunk/shutdown_ack_chunk.h
index 29c1a98..4ed9755 100644
--- a/net/dcsctp/packet/chunk/shutdown_ack_chunk.h
+++ b/net/dcsctp/packet/chunk/shutdown_ack_chunk.h
@@ -35,7 +35,7 @@
ShutdownAckChunk() {}
- static absl::optional<ShutdownAckChunk> Parse(
+ static std::optional<ShutdownAckChunk> Parse(
rtc::ArrayView<const uint8_t> data);
void SerializeTo(std::vector<uint8_t>& out) const override;
diff --git a/net/dcsctp/packet/chunk/shutdown_chunk.cc b/net/dcsctp/packet/chunk/shutdown_chunk.cc
index 59f806f..c7f64dd 100644
--- a/net/dcsctp/packet/chunk/shutdown_chunk.cc
+++ b/net/dcsctp/packet/chunk/shutdown_chunk.cc
@@ -11,10 +11,10 @@
#include <stdint.h>
+#include <optional>
#include <type_traits>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "net/dcsctp/packet/bounded_byte_reader.h"
#include "net/dcsctp/packet/bounded_byte_writer.h"
@@ -33,11 +33,11 @@
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
constexpr int ShutdownChunk::kType;
-absl::optional<ShutdownChunk> ShutdownChunk::Parse(
+std::optional<ShutdownChunk> ShutdownChunk::Parse(
rtc::ArrayView<const uint8_t> data) {
- absl::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
+ std::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
if (!reader.has_value()) {
- return absl::nullopt;
+ return std::nullopt;
}
TSN cumulative_tsn_ack(reader->Load32<4>());
diff --git a/net/dcsctp/packet/chunk/shutdown_chunk.h b/net/dcsctp/packet/chunk/shutdown_chunk.h
index 8148cca..f1090ae 100644
--- a/net/dcsctp/packet/chunk/shutdown_chunk.h
+++ b/net/dcsctp/packet/chunk/shutdown_chunk.h
@@ -36,8 +36,7 @@
explicit ShutdownChunk(TSN cumulative_tsn_ack)
: cumulative_tsn_ack_(cumulative_tsn_ack) {}
- static absl::optional<ShutdownChunk> Parse(
- rtc::ArrayView<const uint8_t> data);
+ static std::optional<ShutdownChunk> Parse(rtc::ArrayView<const uint8_t> data);
void SerializeTo(std::vector<uint8_t>& out) const override;
std::string ToString() const override;
diff --git a/net/dcsctp/packet/chunk/shutdown_complete_chunk.cc b/net/dcsctp/packet/chunk/shutdown_complete_chunk.cc
index 3f54857..abb04b6 100644
--- a/net/dcsctp/packet/chunk/shutdown_complete_chunk.cc
+++ b/net/dcsctp/packet/chunk/shutdown_complete_chunk.cc
@@ -11,10 +11,10 @@
#include <stdint.h>
+#include <optional>
#include <type_traits>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "net/dcsctp/packet/bounded_byte_reader.h"
#include "net/dcsctp/packet/bounded_byte_writer.h"
@@ -31,11 +31,11 @@
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
constexpr int ShutdownCompleteChunk::kType;
-absl::optional<ShutdownCompleteChunk> ShutdownCompleteChunk::Parse(
+std::optional<ShutdownCompleteChunk> ShutdownCompleteChunk::Parse(
rtc::ArrayView<const uint8_t> data) {
- absl::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
+ std::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
if (!reader.has_value()) {
- return absl::nullopt;
+ return std::nullopt;
}
uint8_t flags = reader->Load8<1>();
bool tag_reflected = (flags & (1 << kFlagsBitT)) != 0;
diff --git a/net/dcsctp/packet/chunk/shutdown_complete_chunk.h b/net/dcsctp/packet/chunk/shutdown_complete_chunk.h
index 46d28e8..6ab7eb9 100644
--- a/net/dcsctp/packet/chunk/shutdown_complete_chunk.h
+++ b/net/dcsctp/packet/chunk/shutdown_complete_chunk.h
@@ -37,7 +37,7 @@
explicit ShutdownCompleteChunk(bool tag_reflected)
: tag_reflected_(tag_reflected) {}
- static absl::optional<ShutdownCompleteChunk> Parse(
+ static std::optional<ShutdownCompleteChunk> Parse(
rtc::ArrayView<const uint8_t> data);
void SerializeTo(std::vector<uint8_t>& out) const override;
diff --git a/net/dcsctp/packet/error_cause/cookie_received_while_shutting_down_cause.cc b/net/dcsctp/packet/error_cause/cookie_received_while_shutting_down_cause.cc
index ef67c2a..f675be2 100644
--- a/net/dcsctp/packet/error_cause/cookie_received_while_shutting_down_cause.cc
+++ b/net/dcsctp/packet/error_cause/cookie_received_while_shutting_down_cause.cc
@@ -11,9 +11,9 @@
#include <stdint.h>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
namespace dcsctp {
@@ -25,11 +25,11 @@
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
constexpr int CookieReceivedWhileShuttingDownCause::kType;
-absl::optional<CookieReceivedWhileShuttingDownCause>
+std::optional<CookieReceivedWhileShuttingDownCause>
CookieReceivedWhileShuttingDownCause::Parse(
rtc::ArrayView<const uint8_t> data) {
if (!ParseTLV(data).has_value()) {
- return absl::nullopt;
+ return std::nullopt;
}
return CookieReceivedWhileShuttingDownCause();
}
diff --git a/net/dcsctp/packet/error_cause/cookie_received_while_shutting_down_cause.h b/net/dcsctp/packet/error_cause/cookie_received_while_shutting_down_cause.h
index 362f181..c2a44c9 100644
--- a/net/dcsctp/packet/error_cause/cookie_received_while_shutting_down_cause.h
+++ b/net/dcsctp/packet/error_cause/cookie_received_while_shutting_down_cause.h
@@ -38,7 +38,7 @@
CookieReceivedWhileShuttingDownCause() {}
- static absl::optional<CookieReceivedWhileShuttingDownCause> Parse(
+ static std::optional<CookieReceivedWhileShuttingDownCause> Parse(
rtc::ArrayView<const uint8_t> data);
void SerializeTo(std::vector<uint8_t>& out) const override;
diff --git a/net/dcsctp/packet/error_cause/error_cause.cc b/net/dcsctp/packet/error_cause/error_cause.cc
index dcd0747..c28675a 100644
--- a/net/dcsctp/packet/error_cause/error_cause.cc
+++ b/net/dcsctp/packet/error_cause/error_cause.cc
@@ -13,11 +13,11 @@
#include <cstdint>
#include <memory>
+#include <optional>
#include <string>
#include <utility>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "net/dcsctp/common/math.h"
#include "net/dcsctp/packet/error_cause/cookie_received_while_shutting_down_cause.h"
@@ -40,7 +40,7 @@
template <class ErrorCause>
bool ParseAndPrint(ParameterDescriptor descriptor, rtc::StringBuilder& sb) {
if (descriptor.type == ErrorCause::kType) {
- absl::optional<ErrorCause> p = ErrorCause::Parse(descriptor.data);
+ std::optional<ErrorCause> p = ErrorCause::Parse(descriptor.data);
if (p.has_value()) {
sb << p->ToString();
} else {
diff --git a/net/dcsctp/packet/error_cause/error_cause.h b/net/dcsctp/packet/error_cause/error_cause.h
index fa2bf81..a4247cb 100644
--- a/net/dcsctp/packet/error_cause/error_cause.h
+++ b/net/dcsctp/packet/error_cause/error_cause.h
@@ -15,6 +15,7 @@
#include <cstdint>
#include <iosfwd>
#include <memory>
+#include <optional>
#include <string>
#include <type_traits>
#include <utility>
@@ -22,7 +23,6 @@
#include "absl/algorithm/container.h"
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "net/dcsctp/packet/parameter/parameter.h"
#include "net/dcsctp/packet/tlv_trait.h"
diff --git a/net/dcsctp/packet/error_cause/invalid_mandatory_parameter_cause.cc b/net/dcsctp/packet/error_cause/invalid_mandatory_parameter_cause.cc
index 0187544..9bcba4f 100644
--- a/net/dcsctp/packet/error_cause/invalid_mandatory_parameter_cause.cc
+++ b/net/dcsctp/packet/error_cause/invalid_mandatory_parameter_cause.cc
@@ -11,9 +11,9 @@
#include <stdint.h>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
namespace dcsctp {
@@ -25,10 +25,10 @@
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
constexpr int InvalidMandatoryParameterCause::kType;
-absl::optional<InvalidMandatoryParameterCause>
+std::optional<InvalidMandatoryParameterCause>
InvalidMandatoryParameterCause::Parse(rtc::ArrayView<const uint8_t> data) {
if (!ParseTLV(data).has_value()) {
- return absl::nullopt;
+ return std::nullopt;
}
return InvalidMandatoryParameterCause();
}
diff --git a/net/dcsctp/packet/error_cause/invalid_mandatory_parameter_cause.h b/net/dcsctp/packet/error_cause/invalid_mandatory_parameter_cause.h
index e192b5a..9f4f0c9 100644
--- a/net/dcsctp/packet/error_cause/invalid_mandatory_parameter_cause.h
+++ b/net/dcsctp/packet/error_cause/invalid_mandatory_parameter_cause.h
@@ -37,7 +37,7 @@
InvalidMandatoryParameterCause() {}
- static absl::optional<InvalidMandatoryParameterCause> Parse(
+ static std::optional<InvalidMandatoryParameterCause> Parse(
rtc::ArrayView<const uint8_t> data);
void SerializeTo(std::vector<uint8_t>& out) const override;
diff --git a/net/dcsctp/packet/error_cause/invalid_stream_identifier_cause.cc b/net/dcsctp/packet/error_cause/invalid_stream_identifier_cause.cc
index b2ddd6f..9979b0c 100644
--- a/net/dcsctp/packet/error_cause/invalid_stream_identifier_cause.cc
+++ b/net/dcsctp/packet/error_cause/invalid_stream_identifier_cause.cc
@@ -11,11 +11,11 @@
#include <stdint.h>
+#include <optional>
#include <string>
#include <type_traits>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "net/dcsctp/packet/bounded_byte_reader.h"
#include "net/dcsctp/packet/bounded_byte_writer.h"
@@ -33,11 +33,11 @@
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
constexpr int InvalidStreamIdentifierCause::kType;
-absl::optional<InvalidStreamIdentifierCause>
-InvalidStreamIdentifierCause::Parse(rtc::ArrayView<const uint8_t> data) {
- absl::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
+std::optional<InvalidStreamIdentifierCause> InvalidStreamIdentifierCause::Parse(
+ rtc::ArrayView<const uint8_t> data) {
+ std::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
if (!reader.has_value()) {
- return absl::nullopt;
+ return std::nullopt;
}
StreamID stream_id(reader->Load16<4>());
diff --git a/net/dcsctp/packet/error_cause/invalid_stream_identifier_cause.h b/net/dcsctp/packet/error_cause/invalid_stream_identifier_cause.h
index b7dfe17..2f4b33e 100644
--- a/net/dcsctp/packet/error_cause/invalid_stream_identifier_cause.h
+++ b/net/dcsctp/packet/error_cause/invalid_stream_identifier_cause.h
@@ -39,7 +39,7 @@
explicit InvalidStreamIdentifierCause(StreamID stream_id)
: stream_id_(stream_id) {}
- static absl::optional<InvalidStreamIdentifierCause> Parse(
+ static std::optional<InvalidStreamIdentifierCause> Parse(
rtc::ArrayView<const uint8_t> data);
void SerializeTo(std::vector<uint8_t>& out) const override;
diff --git a/net/dcsctp/packet/error_cause/missing_mandatory_parameter_cause.cc b/net/dcsctp/packet/error_cause/missing_mandatory_parameter_cause.cc
index 679439d..e4a8b2d 100644
--- a/net/dcsctp/packet/error_cause/missing_mandatory_parameter_cause.cc
+++ b/net/dcsctp/packet/error_cause/missing_mandatory_parameter_cause.cc
@@ -12,11 +12,11 @@
#include <stddef.h>
#include <cstdint>
+#include <optional>
#include <string>
#include <type_traits>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "net/dcsctp/packet/bounded_byte_reader.h"
#include "net/dcsctp/packet/bounded_byte_writer.h"
@@ -40,17 +40,17 @@
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
constexpr int MissingMandatoryParameterCause::kType;
-absl::optional<MissingMandatoryParameterCause>
+std::optional<MissingMandatoryParameterCause>
MissingMandatoryParameterCause::Parse(rtc::ArrayView<const uint8_t> data) {
- absl::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
+ std::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
if (!reader.has_value()) {
- return absl::nullopt;
+ return std::nullopt;
}
uint32_t count = reader->Load32<4>();
if (reader->variable_data_size() / kMissingParameterSize != count) {
RTC_DLOG(LS_WARNING) << "Invalid number of missing parameters";
- return absl::nullopt;
+ return std::nullopt;
}
std::vector<uint16_t> missing_parameter_types;
diff --git a/net/dcsctp/packet/error_cause/missing_mandatory_parameter_cause.h b/net/dcsctp/packet/error_cause/missing_mandatory_parameter_cause.h
index 4435424..03903a9 100644
--- a/net/dcsctp/packet/error_cause/missing_mandatory_parameter_cause.h
+++ b/net/dcsctp/packet/error_cause/missing_mandatory_parameter_cause.h
@@ -40,7 +40,7 @@
: missing_parameter_types_(missing_parameter_types.begin(),
missing_parameter_types.end()) {}
- static absl::optional<MissingMandatoryParameterCause> Parse(
+ static std::optional<MissingMandatoryParameterCause> Parse(
rtc::ArrayView<const uint8_t> data);
void SerializeTo(std::vector<uint8_t>& out) const override;
diff --git a/net/dcsctp/packet/error_cause/no_user_data_cause.cc b/net/dcsctp/packet/error_cause/no_user_data_cause.cc
index 2853915..3555e30 100644
--- a/net/dcsctp/packet/error_cause/no_user_data_cause.cc
+++ b/net/dcsctp/packet/error_cause/no_user_data_cause.cc
@@ -11,11 +11,11 @@
#include <stdint.h>
+#include <optional>
#include <string>
#include <type_traits>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "net/dcsctp/packet/bounded_byte_reader.h"
#include "net/dcsctp/packet/bounded_byte_writer.h"
@@ -34,11 +34,11 @@
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
constexpr int NoUserDataCause::kType;
-absl::optional<NoUserDataCause> NoUserDataCause::Parse(
+std::optional<NoUserDataCause> NoUserDataCause::Parse(
rtc::ArrayView<const uint8_t> data) {
- absl::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
+ std::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
if (!reader.has_value()) {
- return absl::nullopt;
+ return std::nullopt;
}
TSN tsn(reader->Load32<4>());
return NoUserDataCause(tsn);
diff --git a/net/dcsctp/packet/error_cause/no_user_data_cause.h b/net/dcsctp/packet/error_cause/no_user_data_cause.h
index 1087dcc..e4092b0 100644
--- a/net/dcsctp/packet/error_cause/no_user_data_cause.h
+++ b/net/dcsctp/packet/error_cause/no_user_data_cause.h
@@ -36,7 +36,7 @@
explicit NoUserDataCause(TSN tsn) : tsn_(tsn) {}
- static absl::optional<NoUserDataCause> Parse(
+ static std::optional<NoUserDataCause> Parse(
rtc::ArrayView<const uint8_t> data);
void SerializeTo(std::vector<uint8_t>& out) const override;
diff --git a/net/dcsctp/packet/error_cause/out_of_resource_error_cause.cc b/net/dcsctp/packet/error_cause/out_of_resource_error_cause.cc
index e5c7c0e..552f48e 100644
--- a/net/dcsctp/packet/error_cause/out_of_resource_error_cause.cc
+++ b/net/dcsctp/packet/error_cause/out_of_resource_error_cause.cc
@@ -11,9 +11,9 @@
#include <stdint.h>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
namespace dcsctp {
@@ -25,10 +25,10 @@
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
constexpr int OutOfResourceErrorCause::kType;
-absl::optional<OutOfResourceErrorCause> OutOfResourceErrorCause::Parse(
+std::optional<OutOfResourceErrorCause> OutOfResourceErrorCause::Parse(
rtc::ArrayView<const uint8_t> data) {
if (!ParseTLV(data).has_value()) {
- return absl::nullopt;
+ return std::nullopt;
}
return OutOfResourceErrorCause();
}
diff --git a/net/dcsctp/packet/error_cause/out_of_resource_error_cause.h b/net/dcsctp/packet/error_cause/out_of_resource_error_cause.h
index fc798ca..4234d73 100644
--- a/net/dcsctp/packet/error_cause/out_of_resource_error_cause.h
+++ b/net/dcsctp/packet/error_cause/out_of_resource_error_cause.h
@@ -36,7 +36,7 @@
OutOfResourceErrorCause() {}
- static absl::optional<OutOfResourceErrorCause> Parse(
+ static std::optional<OutOfResourceErrorCause> Parse(
rtc::ArrayView<const uint8_t> data);
void SerializeTo(std::vector<uint8_t>& out) const override;
diff --git a/net/dcsctp/packet/error_cause/protocol_violation_cause.cc b/net/dcsctp/packet/error_cause/protocol_violation_cause.cc
index 1b8d423..00360cc 100644
--- a/net/dcsctp/packet/error_cause/protocol_violation_cause.cc
+++ b/net/dcsctp/packet/error_cause/protocol_violation_cause.cc
@@ -11,11 +11,11 @@
#include <stdint.h>
+#include <optional>
#include <string>
#include <type_traits>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "net/dcsctp/packet/bounded_byte_reader.h"
#include "net/dcsctp/packet/bounded_byte_writer.h"
@@ -36,11 +36,11 @@
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
constexpr int ProtocolViolationCause::kType;
-absl::optional<ProtocolViolationCause> ProtocolViolationCause::Parse(
+std::optional<ProtocolViolationCause> ProtocolViolationCause::Parse(
rtc::ArrayView<const uint8_t> data) {
- absl::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
+ std::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
if (!reader.has_value()) {
- return absl::nullopt;
+ return std::nullopt;
}
return ProtocolViolationCause(
std::string(reinterpret_cast<const char*>(reader->variable_data().data()),
diff --git a/net/dcsctp/packet/error_cause/protocol_violation_cause.h b/net/dcsctp/packet/error_cause/protocol_violation_cause.h
index 3081e1f..e4e3db8 100644
--- a/net/dcsctp/packet/error_cause/protocol_violation_cause.h
+++ b/net/dcsctp/packet/error_cause/protocol_violation_cause.h
@@ -37,7 +37,7 @@
explicit ProtocolViolationCause(absl::string_view additional_information)
: additional_information_(additional_information) {}
- static absl::optional<ProtocolViolationCause> Parse(
+ static std::optional<ProtocolViolationCause> Parse(
rtc::ArrayView<const uint8_t> data);
void SerializeTo(std::vector<uint8_t>& out) const override;
diff --git a/net/dcsctp/packet/error_cause/restart_of_an_association_with_new_address_cause.cc b/net/dcsctp/packet/error_cause/restart_of_an_association_with_new_address_cause.cc
index abe5de6..5cb2d64 100644
--- a/net/dcsctp/packet/error_cause/restart_of_an_association_with_new_address_cause.cc
+++ b/net/dcsctp/packet/error_cause/restart_of_an_association_with_new_address_cause.cc
@@ -11,10 +11,10 @@
#include <stdint.h>
+#include <optional>
#include <type_traits>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "net/dcsctp/packet/bounded_byte_reader.h"
#include "net/dcsctp/packet/bounded_byte_writer.h"
@@ -34,12 +34,12 @@
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
constexpr int RestartOfAnAssociationWithNewAddressesCause::kType;
-absl::optional<RestartOfAnAssociationWithNewAddressesCause>
+std::optional<RestartOfAnAssociationWithNewAddressesCause>
RestartOfAnAssociationWithNewAddressesCause::Parse(
rtc::ArrayView<const uint8_t> data) {
- absl::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
+ std::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
if (!reader.has_value()) {
- return absl::nullopt;
+ return std::nullopt;
}
return RestartOfAnAssociationWithNewAddressesCause(reader->variable_data());
}
diff --git a/net/dcsctp/packet/error_cause/restart_of_an_association_with_new_address_cause.h b/net/dcsctp/packet/error_cause/restart_of_an_association_with_new_address_cause.h
index a1cccdc..20d7163 100644
--- a/net/dcsctp/packet/error_cause/restart_of_an_association_with_new_address_cause.h
+++ b/net/dcsctp/packet/error_cause/restart_of_an_association_with_new_address_cause.h
@@ -41,7 +41,7 @@
rtc::ArrayView<const uint8_t> new_address_tlvs)
: new_address_tlvs_(new_address_tlvs.begin(), new_address_tlvs.end()) {}
- static absl::optional<RestartOfAnAssociationWithNewAddressesCause> Parse(
+ static std::optional<RestartOfAnAssociationWithNewAddressesCause> Parse(
rtc::ArrayView<const uint8_t> data);
void SerializeTo(std::vector<uint8_t>& out) const override;
diff --git a/net/dcsctp/packet/error_cause/stale_cookie_error_cause.cc b/net/dcsctp/packet/error_cause/stale_cookie_error_cause.cc
index d77d848..3bd6f2c 100644
--- a/net/dcsctp/packet/error_cause/stale_cookie_error_cause.cc
+++ b/net/dcsctp/packet/error_cause/stale_cookie_error_cause.cc
@@ -11,11 +11,11 @@
#include <stdint.h>
+#include <optional>
#include <string>
#include <type_traits>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "net/dcsctp/packet/bounded_byte_reader.h"
#include "net/dcsctp/packet/bounded_byte_writer.h"
@@ -33,11 +33,11 @@
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
constexpr int StaleCookieErrorCause::kType;
-absl::optional<StaleCookieErrorCause> StaleCookieErrorCause::Parse(
+std::optional<StaleCookieErrorCause> StaleCookieErrorCause::Parse(
rtc::ArrayView<const uint8_t> data) {
- absl::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
+ std::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
if (!reader.has_value()) {
- return absl::nullopt;
+ return std::nullopt;
}
uint32_t staleness_us = reader->Load32<4>();
return StaleCookieErrorCause(staleness_us);
diff --git a/net/dcsctp/packet/error_cause/stale_cookie_error_cause.h b/net/dcsctp/packet/error_cause/stale_cookie_error_cause.h
index d8b7b5b..c40fcbc 100644
--- a/net/dcsctp/packet/error_cause/stale_cookie_error_cause.h
+++ b/net/dcsctp/packet/error_cause/stale_cookie_error_cause.h
@@ -37,7 +37,7 @@
explicit StaleCookieErrorCause(uint32_t staleness_us)
: staleness_us_(staleness_us) {}
- static absl::optional<StaleCookieErrorCause> Parse(
+ static std::optional<StaleCookieErrorCause> Parse(
rtc::ArrayView<const uint8_t> data);
void SerializeTo(std::vector<uint8_t>& out) const override;
diff --git a/net/dcsctp/packet/error_cause/unrecognized_chunk_type_cause.cc b/net/dcsctp/packet/error_cause/unrecognized_chunk_type_cause.cc
index 04b960d..2b64dfd 100644
--- a/net/dcsctp/packet/error_cause/unrecognized_chunk_type_cause.cc
+++ b/net/dcsctp/packet/error_cause/unrecognized_chunk_type_cause.cc
@@ -10,11 +10,11 @@
#include "net/dcsctp/packet/error_cause/unrecognized_chunk_type_cause.h"
#include <cstdint>
+#include <optional>
#include <string>
#include <utility>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "net/dcsctp/packet/bounded_byte_reader.h"
#include "net/dcsctp/packet/bounded_byte_writer.h"
@@ -33,11 +33,11 @@
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
constexpr int UnrecognizedChunkTypeCause::kType;
-absl::optional<UnrecognizedChunkTypeCause> UnrecognizedChunkTypeCause::Parse(
+std::optional<UnrecognizedChunkTypeCause> UnrecognizedChunkTypeCause::Parse(
rtc::ArrayView<const uint8_t> data) {
- absl::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
+ std::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
if (!reader.has_value()) {
- return absl::nullopt;
+ return std::nullopt;
}
std::vector<uint8_t> unrecognized_chunk(reader->variable_data().begin(),
reader->variable_data().end());
diff --git a/net/dcsctp/packet/error_cause/unrecognized_chunk_type_cause.h b/net/dcsctp/packet/error_cause/unrecognized_chunk_type_cause.h
index 26d3d3b..d16e21b 100644
--- a/net/dcsctp/packet/error_cause/unrecognized_chunk_type_cause.h
+++ b/net/dcsctp/packet/error_cause/unrecognized_chunk_type_cause.h
@@ -40,7 +40,7 @@
explicit UnrecognizedChunkTypeCause(std::vector<uint8_t> unrecognized_chunk)
: unrecognized_chunk_(std::move(unrecognized_chunk)) {}
- static absl::optional<UnrecognizedChunkTypeCause> Parse(
+ static std::optional<UnrecognizedChunkTypeCause> Parse(
rtc::ArrayView<const uint8_t> data);
void SerializeTo(std::vector<uint8_t>& out) const override;
diff --git a/net/dcsctp/packet/error_cause/unrecognized_parameter_cause.cc b/net/dcsctp/packet/error_cause/unrecognized_parameter_cause.cc
index 80001a9..448dcb7 100644
--- a/net/dcsctp/packet/error_cause/unrecognized_parameter_cause.cc
+++ b/net/dcsctp/packet/error_cause/unrecognized_parameter_cause.cc
@@ -11,10 +11,10 @@
#include <stdint.h>
+#include <optional>
#include <type_traits>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "net/dcsctp/packet/bounded_byte_reader.h"
#include "net/dcsctp/packet/bounded_byte_writer.h"
@@ -32,11 +32,11 @@
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
constexpr int UnrecognizedParametersCause::kType;
-absl::optional<UnrecognizedParametersCause> UnrecognizedParametersCause::Parse(
+std::optional<UnrecognizedParametersCause> UnrecognizedParametersCause::Parse(
rtc::ArrayView<const uint8_t> data) {
- absl::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
+ std::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
if (!reader.has_value()) {
- return absl::nullopt;
+ return std::nullopt;
}
return UnrecognizedParametersCause(reader->variable_data());
}
diff --git a/net/dcsctp/packet/error_cause/unrecognized_parameter_cause.h b/net/dcsctp/packet/error_cause/unrecognized_parameter_cause.h
index ebec5ed..166cc97 100644
--- a/net/dcsctp/packet/error_cause/unrecognized_parameter_cause.h
+++ b/net/dcsctp/packet/error_cause/unrecognized_parameter_cause.h
@@ -41,7 +41,7 @@
: unrecognized_parameters_(unrecognized_parameters.begin(),
unrecognized_parameters.end()) {}
- static absl::optional<UnrecognizedParametersCause> Parse(
+ static std::optional<UnrecognizedParametersCause> Parse(
rtc::ArrayView<const uint8_t> data);
void SerializeTo(std::vector<uint8_t>& out) const override;
diff --git a/net/dcsctp/packet/error_cause/unresolvable_address_cause.cc b/net/dcsctp/packet/error_cause/unresolvable_address_cause.cc
index 8108d31..5e0f273 100644
--- a/net/dcsctp/packet/error_cause/unresolvable_address_cause.cc
+++ b/net/dcsctp/packet/error_cause/unresolvable_address_cause.cc
@@ -11,10 +11,10 @@
#include <stdint.h>
+#include <optional>
#include <type_traits>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "net/dcsctp/packet/bounded_byte_reader.h"
#include "net/dcsctp/packet/bounded_byte_writer.h"
@@ -32,11 +32,11 @@
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
constexpr int UnresolvableAddressCause::kType;
-absl::optional<UnresolvableAddressCause> UnresolvableAddressCause::Parse(
+std::optional<UnresolvableAddressCause> UnresolvableAddressCause::Parse(
rtc::ArrayView<const uint8_t> data) {
- absl::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
+ std::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
if (!reader.has_value()) {
- return absl::nullopt;
+ return std::nullopt;
}
return UnresolvableAddressCause(reader->variable_data());
}
diff --git a/net/dcsctp/packet/error_cause/unresolvable_address_cause.h b/net/dcsctp/packet/error_cause/unresolvable_address_cause.h
index c63b377..cea735c 100644
--- a/net/dcsctp/packet/error_cause/unresolvable_address_cause.h
+++ b/net/dcsctp/packet/error_cause/unresolvable_address_cause.h
@@ -41,7 +41,7 @@
: unresolvable_address_(unresolvable_address.begin(),
unresolvable_address.end()) {}
- static absl::optional<UnresolvableAddressCause> Parse(
+ static std::optional<UnresolvableAddressCause> Parse(
rtc::ArrayView<const uint8_t> data);
void SerializeTo(std::vector<uint8_t>& out) const override;
diff --git a/net/dcsctp/packet/error_cause/user_initiated_abort_cause.cc b/net/dcsctp/packet/error_cause/user_initiated_abort_cause.cc
index da99aac..e4cd267 100644
--- a/net/dcsctp/packet/error_cause/user_initiated_abort_cause.cc
+++ b/net/dcsctp/packet/error_cause/user_initiated_abort_cause.cc
@@ -11,11 +11,11 @@
#include <stdint.h>
+#include <optional>
#include <string>
#include <type_traits>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "net/dcsctp/packet/bounded_byte_reader.h"
#include "net/dcsctp/packet/bounded_byte_writer.h"
@@ -36,11 +36,11 @@
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
constexpr int UserInitiatedAbortCause::kType;
-absl::optional<UserInitiatedAbortCause> UserInitiatedAbortCause::Parse(
+std::optional<UserInitiatedAbortCause> UserInitiatedAbortCause::Parse(
rtc::ArrayView<const uint8_t> data) {
- absl::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
+ std::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
if (!reader.has_value()) {
- return absl::nullopt;
+ return std::nullopt;
}
if (reader->variable_data().empty()) {
return UserInitiatedAbortCause("");
diff --git a/net/dcsctp/packet/error_cause/user_initiated_abort_cause.h b/net/dcsctp/packet/error_cause/user_initiated_abort_cause.h
index 9eb1665..e0a6109 100644
--- a/net/dcsctp/packet/error_cause/user_initiated_abort_cause.h
+++ b/net/dcsctp/packet/error_cause/user_initiated_abort_cause.h
@@ -37,7 +37,7 @@
explicit UserInitiatedAbortCause(absl::string_view upper_layer_abort_reason)
: upper_layer_abort_reason_(upper_layer_abort_reason) {}
- static absl::optional<UserInitiatedAbortCause> Parse(
+ static std::optional<UserInitiatedAbortCause> Parse(
rtc::ArrayView<const uint8_t> data);
void SerializeTo(std::vector<uint8_t>& out) const override;
diff --git a/net/dcsctp/packet/parameter/add_incoming_streams_request_parameter.cc b/net/dcsctp/packet/parameter/add_incoming_streams_request_parameter.cc
index c33e3e1..e8d82b54 100644
--- a/net/dcsctp/packet/parameter/add_incoming_streams_request_parameter.cc
+++ b/net/dcsctp/packet/parameter/add_incoming_streams_request_parameter.cc
@@ -11,11 +11,11 @@
#include <stdint.h>
+#include <optional>
#include <string>
#include <type_traits>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "net/dcsctp/common/internal_types.h"
#include "net/dcsctp/packet/bounded_byte_reader.h"
@@ -38,11 +38,11 @@
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
constexpr int AddIncomingStreamsRequestParameter::kType;
-absl::optional<AddIncomingStreamsRequestParameter>
+std::optional<AddIncomingStreamsRequestParameter>
AddIncomingStreamsRequestParameter::Parse(rtc::ArrayView<const uint8_t> data) {
- absl::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
+ std::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
if (!reader.has_value()) {
- return absl::nullopt;
+ return std::nullopt;
}
ReconfigRequestSN request_sequence_number(reader->Load32<4>());
uint16_t nbr_of_new_streams = reader->Load16<8>();
diff --git a/net/dcsctp/packet/parameter/add_incoming_streams_request_parameter.h b/net/dcsctp/packet/parameter/add_incoming_streams_request_parameter.h
index 3859eb3..8997586 100644
--- a/net/dcsctp/packet/parameter/add_incoming_streams_request_parameter.h
+++ b/net/dcsctp/packet/parameter/add_incoming_streams_request_parameter.h
@@ -42,7 +42,7 @@
: request_sequence_number_(request_sequence_number),
nbr_of_new_streams_(nbr_of_new_streams) {}
- static absl::optional<AddIncomingStreamsRequestParameter> Parse(
+ static std::optional<AddIncomingStreamsRequestParameter> Parse(
rtc::ArrayView<const uint8_t> data);
void SerializeTo(std::vector<uint8_t>& out) const override;
diff --git a/net/dcsctp/packet/parameter/add_outgoing_streams_request_parameter.cc b/net/dcsctp/packet/parameter/add_outgoing_streams_request_parameter.cc
index 4787ee9..1a8f4f9 100644
--- a/net/dcsctp/packet/parameter/add_outgoing_streams_request_parameter.cc
+++ b/net/dcsctp/packet/parameter/add_outgoing_streams_request_parameter.cc
@@ -11,11 +11,11 @@
#include <stdint.h>
+#include <optional>
#include <string>
#include <type_traits>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "net/dcsctp/packet/bounded_byte_reader.h"
#include "net/dcsctp/packet/bounded_byte_writer.h"
@@ -37,11 +37,11 @@
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
constexpr int AddOutgoingStreamsRequestParameter::kType;
-absl::optional<AddOutgoingStreamsRequestParameter>
+std::optional<AddOutgoingStreamsRequestParameter>
AddOutgoingStreamsRequestParameter::Parse(rtc::ArrayView<const uint8_t> data) {
- absl::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
+ std::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
if (!reader.has_value()) {
- return absl::nullopt;
+ return std::nullopt;
}
ReconfigRequestSN request_sequence_number(reader->Load32<4>());
uint16_t nbr_of_new_streams = reader->Load16<8>();
diff --git a/net/dcsctp/packet/parameter/add_outgoing_streams_request_parameter.h b/net/dcsctp/packet/parameter/add_outgoing_streams_request_parameter.h
index 01e8f91..adfcb48 100644
--- a/net/dcsctp/packet/parameter/add_outgoing_streams_request_parameter.h
+++ b/net/dcsctp/packet/parameter/add_outgoing_streams_request_parameter.h
@@ -42,7 +42,7 @@
: request_sequence_number_(request_sequence_number),
nbr_of_new_streams_(nbr_of_new_streams) {}
- static absl::optional<AddOutgoingStreamsRequestParameter> Parse(
+ static std::optional<AddOutgoingStreamsRequestParameter> Parse(
rtc::ArrayView<const uint8_t> data);
void SerializeTo(std::vector<uint8_t>& out) const override;
diff --git a/net/dcsctp/packet/parameter/forward_tsn_supported_parameter.cc b/net/dcsctp/packet/parameter/forward_tsn_supported_parameter.cc
index 7dd8e19..96c1629 100644
--- a/net/dcsctp/packet/parameter/forward_tsn_supported_parameter.cc
+++ b/net/dcsctp/packet/parameter/forward_tsn_supported_parameter.cc
@@ -11,9 +11,9 @@
#include <stdint.h>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
namespace dcsctp {
@@ -26,10 +26,10 @@
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
constexpr int ForwardTsnSupportedParameter::kType;
-absl::optional<ForwardTsnSupportedParameter>
-ForwardTsnSupportedParameter::Parse(rtc::ArrayView<const uint8_t> data) {
+std::optional<ForwardTsnSupportedParameter> ForwardTsnSupportedParameter::Parse(
+ rtc::ArrayView<const uint8_t> data) {
if (!ParseTLV(data).has_value()) {
- return absl::nullopt;
+ return std::nullopt;
}
return ForwardTsnSupportedParameter();
}
diff --git a/net/dcsctp/packet/parameter/forward_tsn_supported_parameter.h b/net/dcsctp/packet/parameter/forward_tsn_supported_parameter.h
index d4cff4a..45b9dcf 100644
--- a/net/dcsctp/packet/parameter/forward_tsn_supported_parameter.h
+++ b/net/dcsctp/packet/parameter/forward_tsn_supported_parameter.h
@@ -37,7 +37,7 @@
ForwardTsnSupportedParameter() {}
- static absl::optional<ForwardTsnSupportedParameter> Parse(
+ static std::optional<ForwardTsnSupportedParameter> Parse(
rtc::ArrayView<const uint8_t> data);
void SerializeTo(std::vector<uint8_t>& out) const override;
diff --git a/net/dcsctp/packet/parameter/heartbeat_info_parameter.cc b/net/dcsctp/packet/parameter/heartbeat_info_parameter.cc
index 918976d..5435c07 100644
--- a/net/dcsctp/packet/parameter/heartbeat_info_parameter.cc
+++ b/net/dcsctp/packet/parameter/heartbeat_info_parameter.cc
@@ -11,11 +11,11 @@
#include <stdint.h>
+#include <optional>
#include <string>
#include <type_traits>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "net/dcsctp/packet/bounded_byte_reader.h"
#include "net/dcsctp/packet/bounded_byte_writer.h"
@@ -37,11 +37,11 @@
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
constexpr int HeartbeatInfoParameter::kType;
-absl::optional<HeartbeatInfoParameter> HeartbeatInfoParameter::Parse(
+std::optional<HeartbeatInfoParameter> HeartbeatInfoParameter::Parse(
rtc::ArrayView<const uint8_t> data) {
- absl::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
+ std::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
if (!reader.has_value()) {
- return absl::nullopt;
+ return std::nullopt;
}
return HeartbeatInfoParameter(reader->variable_data());
}
diff --git a/net/dcsctp/packet/parameter/heartbeat_info_parameter.h b/net/dcsctp/packet/parameter/heartbeat_info_parameter.h
index ec503a94..8622d9a 100644
--- a/net/dcsctp/packet/parameter/heartbeat_info_parameter.h
+++ b/net/dcsctp/packet/parameter/heartbeat_info_parameter.h
@@ -37,7 +37,7 @@
explicit HeartbeatInfoParameter(rtc::ArrayView<const uint8_t> info)
: info_(info.begin(), info.end()) {}
- static absl::optional<HeartbeatInfoParameter> Parse(
+ static std::optional<HeartbeatInfoParameter> Parse(
rtc::ArrayView<const uint8_t> data);
void SerializeTo(std::vector<uint8_t>& out) const override;
diff --git a/net/dcsctp/packet/parameter/incoming_ssn_reset_request_parameter.cc b/net/dcsctp/packet/parameter/incoming_ssn_reset_request_parameter.cc
index 6191adf..3935f8e 100644
--- a/net/dcsctp/packet/parameter/incoming_ssn_reset_request_parameter.cc
+++ b/net/dcsctp/packet/parameter/incoming_ssn_reset_request_parameter.cc
@@ -12,11 +12,11 @@
#include <stddef.h>
#include <cstdint>
+#include <optional>
#include <string>
#include <utility>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "net/dcsctp/packet/bounded_byte_reader.h"
#include "net/dcsctp/packet/bounded_byte_writer.h"
@@ -42,11 +42,11 @@
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
constexpr int IncomingSSNResetRequestParameter::kType;
-absl::optional<IncomingSSNResetRequestParameter>
+std::optional<IncomingSSNResetRequestParameter>
IncomingSSNResetRequestParameter::Parse(rtc::ArrayView<const uint8_t> data) {
- absl::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
+ std::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
if (!reader.has_value()) {
- return absl::nullopt;
+ return std::nullopt;
}
ReconfigRequestSN request_sequence_number(reader->Load32<4>());
diff --git a/net/dcsctp/packet/parameter/incoming_ssn_reset_request_parameter.h b/net/dcsctp/packet/parameter/incoming_ssn_reset_request_parameter.h
index 18963ef..eb14eb6 100644
--- a/net/dcsctp/packet/parameter/incoming_ssn_reset_request_parameter.h
+++ b/net/dcsctp/packet/parameter/incoming_ssn_reset_request_parameter.h
@@ -43,7 +43,7 @@
: request_sequence_number_(request_sequence_number),
stream_ids_(std::move(stream_ids)) {}
- static absl::optional<IncomingSSNResetRequestParameter> Parse(
+ static std::optional<IncomingSSNResetRequestParameter> Parse(
rtc::ArrayView<const uint8_t> data);
void SerializeTo(std::vector<uint8_t>& out) const override;
diff --git a/net/dcsctp/packet/parameter/outgoing_ssn_reset_request_parameter.cc b/net/dcsctp/packet/parameter/outgoing_ssn_reset_request_parameter.cc
index c25a242..a1391bd 100644
--- a/net/dcsctp/packet/parameter/outgoing_ssn_reset_request_parameter.cc
+++ b/net/dcsctp/packet/parameter/outgoing_ssn_reset_request_parameter.cc
@@ -12,11 +12,11 @@
#include <stddef.h>
#include <cstdint>
+#include <optional>
#include <string>
#include <utility>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "net/dcsctp/common/internal_types.h"
#include "net/dcsctp/packet/bounded_byte_reader.h"
@@ -48,11 +48,11 @@
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
constexpr int OutgoingSSNResetRequestParameter::kType;
-absl::optional<OutgoingSSNResetRequestParameter>
+std::optional<OutgoingSSNResetRequestParameter>
OutgoingSSNResetRequestParameter::Parse(rtc::ArrayView<const uint8_t> data) {
- absl::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
+ std::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
if (!reader.has_value()) {
- return absl::nullopt;
+ return std::nullopt;
}
ReconfigRequestSN request_sequence_number(reader->Load32<4>());
diff --git a/net/dcsctp/packet/parameter/outgoing_ssn_reset_request_parameter.h b/net/dcsctp/packet/parameter/outgoing_ssn_reset_request_parameter.h
index 6eb44e0..3a9855c 100644
--- a/net/dcsctp/packet/parameter/outgoing_ssn_reset_request_parameter.h
+++ b/net/dcsctp/packet/parameter/outgoing_ssn_reset_request_parameter.h
@@ -49,7 +49,7 @@
sender_last_assigned_tsn_(sender_last_assigned_tsn),
stream_ids_(std::move(stream_ids)) {}
- static absl::optional<OutgoingSSNResetRequestParameter> Parse(
+ static std::optional<OutgoingSSNResetRequestParameter> Parse(
rtc::ArrayView<const uint8_t> data);
void SerializeTo(std::vector<uint8_t>& out) const override;
diff --git a/net/dcsctp/packet/parameter/parameter.cc b/net/dcsctp/packet/parameter/parameter.cc
index b3b2bff..e300f0a 100644
--- a/net/dcsctp/packet/parameter/parameter.cc
+++ b/net/dcsctp/packet/parameter/parameter.cc
@@ -13,12 +13,12 @@
#include <cstdint>
#include <memory>
+#include <optional>
#include <string>
#include <utility>
#include <vector>
#include "absl/memory/memory.h"
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "net/dcsctp/common/math.h"
#include "net/dcsctp/packet/bounded_byte_reader.h"
@@ -70,20 +70,20 @@
return result;
}
-absl::optional<Parameters> Parameters::Parse(
+std::optional<Parameters> Parameters::Parse(
rtc::ArrayView<const uint8_t> data) {
// Validate the parameter descriptors
rtc::ArrayView<const uint8_t> span(data);
while (!span.empty()) {
if (span.size() < kParameterHeaderSize) {
RTC_DLOG(LS_WARNING) << "Insufficient parameter length";
- return absl::nullopt;
+ return std::nullopt;
}
BoundedByteReader<kParameterHeaderSize> header(span);
uint16_t length = header.Load16<2>();
if (length < kParameterHeaderSize || length > span.size()) {
RTC_DLOG(LS_WARNING) << "Invalid parameter length field";
- return absl::nullopt;
+ return std::nullopt;
}
size_t length_with_padding = RoundUpTo4(length);
if (length_with_padding > span.size()) {
diff --git a/net/dcsctp/packet/parameter/parameter.h b/net/dcsctp/packet/parameter/parameter.h
index e8fa67c..d32157d 100644
--- a/net/dcsctp/packet/parameter/parameter.h
+++ b/net/dcsctp/packet/parameter/parameter.h
@@ -16,6 +16,7 @@
#include <cstdint>
#include <iterator>
#include <memory>
+#include <optional>
#include <string>
#include <type_traits>
#include <utility>
@@ -23,7 +24,6 @@
#include "absl/algorithm/container.h"
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "net/dcsctp/packet/tlv_trait.h"
#include "rtc_base/strings/string_builder.h"
@@ -61,7 +61,7 @@
std::vector<uint8_t> data_;
};
- static absl::optional<Parameters> Parse(rtc::ArrayView<const uint8_t> data);
+ static std::optional<Parameters> Parse(rtc::ArrayView<const uint8_t> data);
Parameters() {}
Parameters(Parameters&& other) = default;
@@ -71,7 +71,7 @@
std::vector<ParameterDescriptor> descriptors() const;
template <typename P>
- absl::optional<P> get() const {
+ std::optional<P> get() const {
static_assert(std::is_base_of<Parameter, P>::value,
"Template parameter not derived from Parameter");
for (const auto& p : descriptors()) {
@@ -79,7 +79,7 @@
return P::Parse(p.data);
}
}
- return absl::nullopt;
+ return std::nullopt;
}
private:
diff --git a/net/dcsctp/packet/parameter/reconfiguration_response_parameter.cc b/net/dcsctp/packet/parameter/reconfiguration_response_parameter.cc
index fafb204..eed1e02 100644
--- a/net/dcsctp/packet/parameter/reconfiguration_response_parameter.cc
+++ b/net/dcsctp/packet/parameter/reconfiguration_response_parameter.cc
@@ -12,12 +12,12 @@
#include <stddef.h>
#include <stdint.h>
+#include <optional>
#include <string>
#include <type_traits>
#include <vector>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "net/dcsctp/packet/bounded_byte_reader.h"
#include "net/dcsctp/packet/bounded_byte_writer.h"
@@ -64,11 +64,11 @@
}
}
-absl::optional<ReconfigurationResponseParameter>
+std::optional<ReconfigurationResponseParameter>
ReconfigurationResponseParameter::Parse(rtc::ArrayView<const uint8_t> data) {
- absl::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
+ std::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
if (!reader.has_value()) {
- return absl::nullopt;
+ return std::nullopt;
}
ReconfigRequestSN response_sequence_number(reader->Load32<4>());
@@ -101,14 +101,14 @@
default:
RTC_DLOG(LS_WARNING) << "Invalid reconfig response result: "
<< result_nbr;
- return absl::nullopt;
+ return std::nullopt;
}
if (reader->variable_data().empty()) {
return ReconfigurationResponseParameter(response_sequence_number, result);
} else if (reader->variable_data_size() != kNextTsnHeaderSize) {
RTC_DLOG(LS_WARNING) << "Invalid parameter size";
- return absl::nullopt;
+ return std::nullopt;
}
BoundedByteReader<kNextTsnHeaderSize> sub_reader =
diff --git a/net/dcsctp/packet/parameter/reconfiguration_response_parameter.h b/net/dcsctp/packet/parameter/reconfiguration_response_parameter.h
index c5a68ac..a8cf4dd 100644
--- a/net/dcsctp/packet/parameter/reconfiguration_response_parameter.h
+++ b/net/dcsctp/packet/parameter/reconfiguration_response_parameter.h
@@ -12,11 +12,11 @@
#include <stddef.h>
#include <cstdint>
+#include <optional>
#include <string>
#include <vector>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "net/dcsctp/common/internal_types.h"
#include "net/dcsctp/packet/parameter/parameter.h"
@@ -51,8 +51,8 @@
Result result)
: response_sequence_number_(response_sequence_number),
result_(result),
- sender_next_tsn_(absl::nullopt),
- receiver_next_tsn_(absl::nullopt) {}
+ sender_next_tsn_(std::nullopt),
+ receiver_next_tsn_(std::nullopt) {}
explicit ReconfigurationResponseParameter(
ReconfigRequestSN response_sequence_number,
@@ -64,7 +64,7 @@
sender_next_tsn_(sender_next_tsn),
receiver_next_tsn_(receiver_next_tsn) {}
- static absl::optional<ReconfigurationResponseParameter> Parse(
+ static std::optional<ReconfigurationResponseParameter> Parse(
rtc::ArrayView<const uint8_t> data);
void SerializeTo(std::vector<uint8_t>& out) const override;
@@ -74,15 +74,15 @@
return response_sequence_number_;
}
Result result() const { return result_; }
- absl::optional<TSN> sender_next_tsn() const { return sender_next_tsn_; }
- absl::optional<TSN> receiver_next_tsn() const { return receiver_next_tsn_; }
+ std::optional<TSN> sender_next_tsn() const { return sender_next_tsn_; }
+ std::optional<TSN> receiver_next_tsn() const { return receiver_next_tsn_; }
private:
static constexpr size_t kNextTsnHeaderSize = 8;
ReconfigRequestSN response_sequence_number_;
Result result_;
- absl::optional<TSN> sender_next_tsn_;
- absl::optional<TSN> receiver_next_tsn_;
+ std::optional<TSN> sender_next_tsn_;
+ std::optional<TSN> receiver_next_tsn_;
};
absl::string_view ToString(ReconfigurationResponseParameter::Result result);
diff --git a/net/dcsctp/packet/parameter/reconfiguration_response_parameter_test.cc b/net/dcsctp/packet/parameter/reconfiguration_response_parameter_test.cc
index 8125d93..b784545 100644
--- a/net/dcsctp/packet/parameter/reconfiguration_response_parameter_test.cc
+++ b/net/dcsctp/packet/parameter/reconfiguration_response_parameter_test.cc
@@ -11,10 +11,10 @@
#include <stdint.h>
+#include <optional>
#include <type_traits>
#include <vector>
-#include "absl/types/optional.h"
#include "net/dcsctp/testing/testing_macros.h"
#include "rtc_base/gunit.h"
#include "test/gmock.h"
@@ -37,8 +37,8 @@
EXPECT_EQ(*deserialized.response_sequence_number(), 1u);
EXPECT_EQ(deserialized.result(),
ReconfigurationResponseParameter::Result::kSuccessPerformed);
- EXPECT_EQ(deserialized.sender_next_tsn(), absl::nullopt);
- EXPECT_EQ(deserialized.receiver_next_tsn(), absl::nullopt);
+ EXPECT_EQ(deserialized.sender_next_tsn(), std::nullopt);
+ EXPECT_EQ(deserialized.receiver_next_tsn(), std::nullopt);
}
TEST(ReconfigurationResponseParameterTest,
diff --git a/net/dcsctp/packet/parameter/ssn_tsn_reset_request_parameter.cc b/net/dcsctp/packet/parameter/ssn_tsn_reset_request_parameter.cc
index d656e0d..1e37ea7 100644
--- a/net/dcsctp/packet/parameter/ssn_tsn_reset_request_parameter.cc
+++ b/net/dcsctp/packet/parameter/ssn_tsn_reset_request_parameter.cc
@@ -11,11 +11,11 @@
#include <stdint.h>
+#include <optional>
#include <string>
#include <type_traits>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "net/dcsctp/packet/bounded_byte_reader.h"
#include "net/dcsctp/packet/bounded_byte_writer.h"
@@ -35,11 +35,11 @@
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
constexpr int SSNTSNResetRequestParameter::kType;
-absl::optional<SSNTSNResetRequestParameter> SSNTSNResetRequestParameter::Parse(
+std::optional<SSNTSNResetRequestParameter> SSNTSNResetRequestParameter::Parse(
rtc::ArrayView<const uint8_t> data) {
- absl::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
+ std::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
if (!reader.has_value()) {
- return absl::nullopt;
+ return std::nullopt;
}
ReconfigRequestSN request_sequence_number(reader->Load32<4>());
diff --git a/net/dcsctp/packet/parameter/ssn_tsn_reset_request_parameter.h b/net/dcsctp/packet/parameter/ssn_tsn_reset_request_parameter.h
index e31d7eb..eb156a7 100644
--- a/net/dcsctp/packet/parameter/ssn_tsn_reset_request_parameter.h
+++ b/net/dcsctp/packet/parameter/ssn_tsn_reset_request_parameter.h
@@ -40,7 +40,7 @@
ReconfigRequestSN request_sequence_number)
: request_sequence_number_(request_sequence_number) {}
- static absl::optional<SSNTSNResetRequestParameter> Parse(
+ static std::optional<SSNTSNResetRequestParameter> Parse(
rtc::ArrayView<const uint8_t> data);
void SerializeTo(std::vector<uint8_t>& out) const override;
diff --git a/net/dcsctp/packet/parameter/state_cookie_parameter.cc b/net/dcsctp/packet/parameter/state_cookie_parameter.cc
index 9777aa6..be4b6a5 100644
--- a/net/dcsctp/packet/parameter/state_cookie_parameter.cc
+++ b/net/dcsctp/packet/parameter/state_cookie_parameter.cc
@@ -11,11 +11,11 @@
#include <stdint.h>
+#include <optional>
#include <string>
#include <type_traits>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "net/dcsctp/packet/bounded_byte_reader.h"
#include "net/dcsctp/packet/bounded_byte_writer.h"
@@ -28,11 +28,11 @@
constexpr int StateCookieParameter::kType;
-absl::optional<StateCookieParameter> StateCookieParameter::Parse(
+std::optional<StateCookieParameter> StateCookieParameter::Parse(
rtc::ArrayView<const uint8_t> data) {
- absl::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
+ std::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
if (!reader.has_value()) {
- return absl::nullopt;
+ return std::nullopt;
}
return StateCookieParameter(reader->variable_data());
}
diff --git a/net/dcsctp/packet/parameter/state_cookie_parameter.h b/net/dcsctp/packet/parameter/state_cookie_parameter.h
index f435549..473ee4f 100644
--- a/net/dcsctp/packet/parameter/state_cookie_parameter.h
+++ b/net/dcsctp/packet/parameter/state_cookie_parameter.h
@@ -38,7 +38,7 @@
explicit StateCookieParameter(rtc::ArrayView<const uint8_t> data)
: data_(data.begin(), data.end()) {}
- static absl::optional<StateCookieParameter> Parse(
+ static std::optional<StateCookieParameter> Parse(
rtc::ArrayView<const uint8_t> data);
void SerializeTo(std::vector<uint8_t>& out) const override;
diff --git a/net/dcsctp/packet/parameter/supported_extensions_parameter.cc b/net/dcsctp/packet/parameter/supported_extensions_parameter.cc
index 87a5bd9b..a21819f 100644
--- a/net/dcsctp/packet/parameter/supported_extensions_parameter.cc
+++ b/net/dcsctp/packet/parameter/supported_extensions_parameter.cc
@@ -10,11 +10,11 @@
#include "net/dcsctp/packet/parameter/supported_extensions_parameter.h"
#include <cstdint>
+#include <optional>
#include <string>
#include <utility>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "net/dcsctp/packet/bounded_byte_reader.h"
#include "net/dcsctp/packet/bounded_byte_writer.h"
@@ -39,11 +39,11 @@
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
constexpr int SupportedExtensionsParameter::kType;
-absl::optional<SupportedExtensionsParameter>
-SupportedExtensionsParameter::Parse(rtc::ArrayView<const uint8_t> data) {
- absl::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
+std::optional<SupportedExtensionsParameter> SupportedExtensionsParameter::Parse(
+ rtc::ArrayView<const uint8_t> data) {
+ std::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
if (!reader.has_value()) {
- return absl::nullopt;
+ return std::nullopt;
}
std::vector<uint8_t> chunk_types(reader->variable_data().begin(),
diff --git a/net/dcsctp/packet/parameter/supported_extensions_parameter.h b/net/dcsctp/packet/parameter/supported_extensions_parameter.h
index 5689fd8..6a024ad 100644
--- a/net/dcsctp/packet/parameter/supported_extensions_parameter.h
+++ b/net/dcsctp/packet/parameter/supported_extensions_parameter.h
@@ -41,7 +41,7 @@
explicit SupportedExtensionsParameter(std::vector<uint8_t> chunk_types)
: chunk_types_(std::move(chunk_types)) {}
- static absl::optional<SupportedExtensionsParameter> Parse(
+ static std::optional<SupportedExtensionsParameter> Parse(
rtc::ArrayView<const uint8_t> data);
void SerializeTo(std::vector<uint8_t>& out) const override;
diff --git a/net/dcsctp/packet/parameter/zero_checksum_acceptable_chunk_parameter.cc b/net/dcsctp/packet/parameter/zero_checksum_acceptable_chunk_parameter.cc
index a846d6d..04b7412 100644
--- a/net/dcsctp/packet/parameter/zero_checksum_acceptable_chunk_parameter.cc
+++ b/net/dcsctp/packet/parameter/zero_checksum_acceptable_chunk_parameter.cc
@@ -11,9 +11,9 @@
#include <stdint.h>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "rtc_base/strings/string_builder.h"
@@ -29,17 +29,17 @@
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
constexpr int ZeroChecksumAcceptableChunkParameter::kType;
-absl::optional<ZeroChecksumAcceptableChunkParameter>
+std::optional<ZeroChecksumAcceptableChunkParameter>
ZeroChecksumAcceptableChunkParameter::Parse(
rtc::ArrayView<const uint8_t> data) {
- absl::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
+ std::optional<BoundedByteReader<kHeaderSize>> reader = ParseTLV(data);
if (!reader.has_value()) {
- return absl::nullopt;
+ return std::nullopt;
}
ZeroChecksumAlternateErrorDetectionMethod method(reader->Load32<4>());
if (method == ZeroChecksumAlternateErrorDetectionMethod::None()) {
- return absl::nullopt;
+ return std::nullopt;
}
return ZeroChecksumAcceptableChunkParameter(method);
}
diff --git a/net/dcsctp/packet/parameter/zero_checksum_acceptable_chunk_parameter.h b/net/dcsctp/packet/parameter/zero_checksum_acceptable_chunk_parameter.h
index 18c98c9..4b24692 100644
--- a/net/dcsctp/packet/parameter/zero_checksum_acceptable_chunk_parameter.h
+++ b/net/dcsctp/packet/parameter/zero_checksum_acceptable_chunk_parameter.h
@@ -41,7 +41,7 @@
ZeroChecksumAlternateErrorDetectionMethod error_detection_method)
: error_detection_method_(error_detection_method) {}
- static absl::optional<ZeroChecksumAcceptableChunkParameter> Parse(
+ static std::optional<ZeroChecksumAcceptableChunkParameter> Parse(
rtc::ArrayView<const uint8_t> data);
void SerializeTo(std::vector<uint8_t>& out) const override;
diff --git a/net/dcsctp/packet/sctp_packet.cc b/net/dcsctp/packet/sctp_packet.cc
index 978181f..f9c8812 100644
--- a/net/dcsctp/packet/sctp_packet.cc
+++ b/net/dcsctp/packet/sctp_packet.cc
@@ -12,12 +12,12 @@
#include <stddef.h>
#include <cstdint>
+#include <optional>
#include <string>
#include <utility>
#include <vector>
#include "absl/memory/memory.h"
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "net/dcsctp/common/math.h"
#include "net/dcsctp/packet/bounded_byte_reader.h"
@@ -105,12 +105,12 @@
return out;
}
-absl::optional<SctpPacket> SctpPacket::Parse(rtc::ArrayView<const uint8_t> data,
- const DcSctpOptions& options) {
+std::optional<SctpPacket> SctpPacket::Parse(rtc::ArrayView<const uint8_t> data,
+ const DcSctpOptions& options) {
if (data.size() < kHeaderSize + kChunkTlvHeaderSize ||
data.size() > kMaxUdpPacketSize) {
RTC_DLOG(LS_WARNING) << "Invalid packet size";
- return absl::nullopt;
+ return std::nullopt;
}
BoundedByteReader<kHeaderSize> reader(data);
@@ -143,7 +143,7 @@
"Invalid packet checksum, packet_checksum=0x%08x, "
"calculated_checksum=0x%08x",
common_header.checksum, calculated_checksum);
- return absl::nullopt;
+ return std::nullopt;
}
// Restore the checksum in the header.
BoundedByteWriter<kHeaderSize>(data_copy).Store32<8>(
@@ -166,7 +166,7 @@
while (!descriptor_data.empty()) {
if (descriptor_data.size() < kChunkTlvHeaderSize) {
RTC_DLOG(LS_WARNING) << "Too small chunk";
- return absl::nullopt;
+ return std::nullopt;
}
BoundedByteReader<kChunkTlvHeaderSize> chunk_header(descriptor_data);
uint8_t type = chunk_header.Load8<0>();
@@ -176,10 +176,10 @@
if (padded_length > descriptor_data.size()) {
RTC_DLOG(LS_WARNING) << "Too large chunk. length=" << length
<< ", remaining=" << descriptor_data.size();
- return absl::nullopt;
+ return std::nullopt;
} else if (padded_length < kChunkTlvHeaderSize) {
RTC_DLOG(LS_WARNING) << "Too small chunk. length=" << length;
- return absl::nullopt;
+ return std::nullopt;
}
descriptors.emplace_back(type, flags,
descriptor_data.subview(0, padded_length));
diff --git a/net/dcsctp/packet/sctp_packet.h b/net/dcsctp/packet/sctp_packet.h
index 0d348b4..9c06106 100644
--- a/net/dcsctp/packet/sctp_packet.h
+++ b/net/dcsctp/packet/sctp_packet.h
@@ -89,8 +89,8 @@
};
// Parses `data` as an SCTP packet and returns it if it validates.
- static absl::optional<SctpPacket> Parse(rtc::ArrayView<const uint8_t> data,
- const DcSctpOptions& options);
+ static std::optional<SctpPacket> Parse(rtc::ArrayView<const uint8_t> data,
+ const DcSctpOptions& options);
// Returns the SCTP common header.
const CommonHeader& common_header() const { return common_header_; }
diff --git a/net/dcsctp/packet/tlv_trait.h b/net/dcsctp/packet/tlv_trait.h
index a3c728e..37b3628 100644
--- a/net/dcsctp/packet/tlv_trait.h
+++ b/net/dcsctp/packet/tlv_trait.h
@@ -16,10 +16,10 @@
#include <algorithm>
#include <cstddef>
#include <cstdint>
+#include <optional>
#include <string>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "net/dcsctp/packet/bounded_byte_reader.h"
#include "net/dcsctp/packet/bounded_byte_writer.h"
@@ -80,11 +80,11 @@
// Validates the data with regards to size, alignment and type.
// If valid, returns a bounded buffer.
- static absl::optional<BoundedByteReader<Config::kHeaderSize>> ParseTLV(
+ static std::optional<BoundedByteReader<Config::kHeaderSize>> ParseTLV(
rtc::ArrayView<const uint8_t> data) {
if (data.size() < Config::kHeaderSize) {
tlv_trait_impl::ReportInvalidSize(data.size(), Config::kHeaderSize);
- return absl::nullopt;
+ return std::nullopt;
}
BoundedByteReader<kTlvHeaderSize> tlv_header(data);
@@ -94,7 +94,7 @@
if (type != Config::kType) {
tlv_trait_impl::ReportInvalidType(type, Config::kType);
- return absl::nullopt;
+ return std::nullopt;
}
const uint16_t length = tlv_header.template Load16<2>();
if (Config::kVariableLengthAlignment == 0) {
@@ -102,25 +102,25 @@
if (length != Config::kHeaderSize || data.size() != Config::kHeaderSize) {
tlv_trait_impl::ReportInvalidFixedLengthField(length,
Config::kHeaderSize);
- return absl::nullopt;
+ return std::nullopt;
}
} else {
// Expect variable length data - verify its size alignment.
if (length > data.size() || length < Config::kHeaderSize) {
tlv_trait_impl::ReportInvalidVariableLengthField(length, data.size());
- return absl::nullopt;
+ return std::nullopt;
}
const size_t padding = data.size() - length;
if (padding > 3) {
// https://tools.ietf.org/html/rfc4960#section-3.2
// "This padding MUST NOT be more than 3 bytes in total"
tlv_trait_impl::ReportInvalidPadding(padding);
- return absl::nullopt;
+ return std::nullopt;
}
if (!ValidateLengthAlignment(length, Config::kVariableLengthAlignment)) {
tlv_trait_impl::ReportInvalidLengthMultiple(
length, Config::kVariableLengthAlignment);
- return absl::nullopt;
+ return std::nullopt;
}
}
return BoundedByteReader<Config::kHeaderSize>(data.subview(0, length));
diff --git a/net/dcsctp/packet/tlv_trait_test.cc b/net/dcsctp/packet/tlv_trait_test.cc
index a0dd1a1..e0fabd9 100644
--- a/net/dcsctp/packet/tlv_trait_test.cc
+++ b/net/dcsctp/packet/tlv_trait_test.cc
@@ -44,8 +44,8 @@
writer.CopyToVariableData(rtc::ArrayView<const uint8_t>(variable_data));
}
- static absl::optional<BoundedByteReader<OneByteTypeConfig::kHeaderSize>>
- Parse(rtc::ArrayView<const uint8_t> data) {
+ static std::optional<BoundedByteReader<OneByteTypeConfig::kHeaderSize>> Parse(
+ rtc::ArrayView<const uint8_t> data) {
return ParseTLV(data);
}
};
@@ -64,7 +64,7 @@
uint8_t data[] = {0x49, 0x00, 0x00, 0x10, 0x01, 0x02, 0x03, 0x04,
0x05, 0x06, 0x07, 0x08, 0xDE, 0xAD, 0xBE, 0xEF};
- absl::optional<BoundedByteReader<OneByteTypeConfig::kHeaderSize>> reader =
+ std::optional<BoundedByteReader<OneByteTypeConfig::kHeaderSize>> reader =
OneByteChunk::Parse(data);
ASSERT_TRUE(reader.has_value());
EXPECT_EQ(reader->Load32<4>(), 0x01020304U);
@@ -93,8 +93,8 @@
writer.CopyToVariableData(rtc::ArrayView<const uint8_t>(variable_data));
}
- static absl::optional<BoundedByteReader<TwoByteTypeConfig::kHeaderSize>>
- Parse(rtc::ArrayView<const uint8_t> data) {
+ static std::optional<BoundedByteReader<TwoByteTypeConfig::kHeaderSize>> Parse(
+ rtc::ArrayView<const uint8_t> data) {
return ParseTLV(data);
}
};
@@ -114,7 +114,7 @@
uint8_t data[] = {0x7A, 0x69, 0x00, 0x10, 0x01, 0x02, 0x03, 0x04,
0x05, 0x06, 0x07, 0x08, 0xDE, 0xAD, 0xBE, 0xEF};
- absl::optional<BoundedByteReader<TwoByteTypeConfig::kHeaderSize>> reader =
+ std::optional<BoundedByteReader<TwoByteTypeConfig::kHeaderSize>> reader =
TwoByteChunk::Parse(data);
EXPECT_TRUE(reader.has_value());
EXPECT_EQ(reader->Load32<4>(), 0x01020304U);
diff --git a/net/dcsctp/public/BUILD.gn b/net/dcsctp/public/BUILD.gn
index 316af7b..97f7651 100644
--- a/net/dcsctp/public/BUILD.gn
+++ b/net/dcsctp/public/BUILD.gn
@@ -13,7 +13,6 @@
"../../../api:array_view",
"../../../api/units:time_delta",
"../../../rtc_base:strong_alias",
- "//third_party/abseil-cpp/absl/types:optional",
]
sources = [
"dcsctp_message.h",
@@ -31,7 +30,6 @@
"../../../rtc_base:checks",
"../../../rtc_base:strong_alias",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
sources = [
"dcsctp_handover_state.cc",
diff --git a/net/dcsctp/public/dcsctp_options.h b/net/dcsctp/public/dcsctp_options.h
index 8b4ed3c..115cf5e 100644
--- a/net/dcsctp/public/dcsctp_options.h
+++ b/net/dcsctp/public/dcsctp_options.h
@@ -13,7 +13,8 @@
#include <stddef.h>
#include <stdint.h>
-#include "absl/types/optional.h"
+#include <optional>
+
#include "net/dcsctp/public/types.h"
namespace dcsctp {
@@ -128,7 +129,7 @@
// transient network issues. Setting this value may require changing
// `max_retransmissions` and `max_init_retransmits` to ensure that the
// connection is not closed too quickly.
- absl::optional<DurationMs> max_timer_backoff_duration = absl::nullopt;
+ std::optional<DurationMs> max_timer_backoff_duration = std::nullopt;
// Hearbeat interval (on idle connections only). Set to zero to disable.
DurationMs heartbeat_interval = DurationMs(30000);
@@ -183,13 +184,13 @@
// retransmission scenarios.
int max_burst = 4;
- // Maximum Data Retransmit Attempts (per DATA chunk). Set to absl::nullopt for
+ // Maximum Data Retransmit Attempts (per DATA chunk). Set to std::nullopt for
// no limit.
- absl::optional<int> max_retransmissions = 10;
+ std::optional<int> max_retransmissions = 10;
// Max.Init.Retransmits (https://tools.ietf.org/html/rfc4960#section-15). Set
- // to absl::nullopt for no limit.
- absl::optional<int> max_init_retransmits = 8;
+ // to std::nullopt for no limit.
+ std::optional<int> max_init_retransmits = 8;
// RFC3758 Partial Reliability Extension
bool enable_partial_reliability = true;
diff --git a/net/dcsctp/public/dcsctp_socket.h b/net/dcsctp/public/dcsctp_socket.h
index 9989ae8..bf53ebd 100644
--- a/net/dcsctp/public/dcsctp_socket.h
+++ b/net/dcsctp/public/dcsctp_socket.h
@@ -12,11 +12,11 @@
#include <cstdint>
#include <memory>
+#include <optional>
#include <utility>
#include <vector>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/task_queue/task_queue_base.h"
#include "api/units/timestamp.h"
@@ -51,11 +51,11 @@
// If set, will discard messages that haven't been correctly sent and
// received before the lifetime has expired. This is only available if the
// peer supports Partial Reliability Extension (RFC3758).
- absl::optional<DurationMs> lifetime = absl::nullopt;
+ std::optional<DurationMs> lifetime = std::nullopt;
// If set, limits the number of retransmissions. This is only available
// if the peer supports Partial Reliability Extension (RFC3758).
- absl::optional<size_t> max_retransmissions = absl::nullopt;
+ std::optional<size_t> max_retransmissions = std::nullopt;
// If set, will generate lifecycle events for this message. See e.g.
// `DcSctpSocketCallbacks::OnLifecycleMessageFullySent`. This value is decided
@@ -621,10 +621,10 @@
size_t bytes) = 0;
// Retrieves the latest metrics. If the socket is not fully connected,
- // `absl::nullopt` will be returned. Note that metrics are not guaranteed to
+ // `std::nullopt` will be returned. Note that metrics are not guaranteed to
// be carried over if this socket is handed over by calling
// `GetHandoverStateAndClose`.
- virtual absl::optional<Metrics> GetMetrics() const = 0;
+ virtual std::optional<Metrics> GetMetrics() const = 0;
// Returns empty bitmask if the socket is in the state in which a snapshot of
// the state can be made by `GetHandoverStateAndClose()`. Return value is
@@ -637,7 +637,7 @@
// The method fails if the socket is not in a state ready for handover.
// nullopt indicates the failure. `DcSctpSocketCallbacks::OnClosed` will be
// called on success.
- virtual absl::optional<DcSctpSocketHandoverState>
+ virtual std::optional<DcSctpSocketHandoverState>
GetHandoverStateAndClose() = 0;
// Returns the detected SCTP implementation of the peer. As this is not
diff --git a/net/dcsctp/public/mock_dcsctp_socket.h b/net/dcsctp/public/mock_dcsctp_socket.h
index c71c3ae..627b20d 100644
--- a/net/dcsctp/public/mock_dcsctp_socket.h
+++ b/net/dcsctp/public/mock_dcsctp_socket.h
@@ -81,13 +81,13 @@
(StreamID stream_id, size_t bytes),
(override));
- MOCK_METHOD(absl::optional<Metrics>, GetMetrics, (), (const, override));
+ MOCK_METHOD(std::optional<Metrics>, GetMetrics, (), (const, override));
MOCK_METHOD(HandoverReadinessStatus,
GetHandoverReadiness,
(),
(const, override));
- MOCK_METHOD(absl::optional<DcSctpSocketHandoverState>,
+ MOCK_METHOD(std::optional<DcSctpSocketHandoverState>,
GetHandoverStateAndClose,
(),
(override));
diff --git a/net/dcsctp/rx/BUILD.gn b/net/dcsctp/rx/BUILD.gn
index a39441f..2046a27 100644
--- a/net/dcsctp/rx/BUILD.gn
+++ b/net/dcsctp/rx/BUILD.gn
@@ -21,7 +21,6 @@
"../timer",
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
sources = [
"data_tracker.cc",
@@ -72,7 +71,6 @@
"../public:types",
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
sources = [
"traditional_reassembly_streams.cc",
@@ -99,7 +97,6 @@
"../public:types",
"//third_party/abseil-cpp/absl/functional:any_invocable",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
sources = [
"reassembly_queue.cc",
@@ -129,7 +126,6 @@
"../public:types",
"../testing:data_generator",
"../timer",
- "//third_party/abseil-cpp/absl/types:optional",
]
sources = [
"data_tracker_test.cc",
diff --git a/net/dcsctp/rx/data_tracker.cc b/net/dcsctp/rx/data_tracker.cc
index 70b7587..627478a 100644
--- a/net/dcsctp/rx/data_tracker.cc
+++ b/net/dcsctp/rx/data_tracker.cc
@@ -12,6 +12,7 @@
#include <algorithm>
#include <cstdint>
#include <iterator>
+#include <optional>
#include <set>
#include <string>
#include <utility>
@@ -19,7 +20,6 @@
#include "absl/algorithm/container.h"
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "net/dcsctp/common/sequence_numbers.h"
#include "net/dcsctp/packet/chunk/sack_chunk.h"
#include "net/dcsctp/timer/timer.h"
diff --git a/net/dcsctp/rx/data_tracker_test.cc b/net/dcsctp/rx/data_tracker_test.cc
index 0e9e4fc..e0bf4cf 100644
--- a/net/dcsctp/rx/data_tracker_test.cc
+++ b/net/dcsctp/rx/data_tracker_test.cc
@@ -12,8 +12,8 @@
#include <cstdint>
#include <initializer_list>
#include <memory>
+#include <optional>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/task_queue/task_queue_base.h"
#include "net/dcsctp/common/handover_testing.h"
diff --git a/net/dcsctp/rx/reassembly_queue.cc b/net/dcsctp/rx/reassembly_queue.cc
index 6f22806..4947d74 100644
--- a/net/dcsctp/rx/reassembly_queue.cc
+++ b/net/dcsctp/rx/reassembly_queue.cc
@@ -14,13 +14,13 @@
#include <algorithm>
#include <cstdint>
#include <memory>
+#include <optional>
#include <set>
#include <string>
#include <utility>
#include <vector>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "net/dcsctp/common/sequence_numbers.h"
#include "net/dcsctp/packet/chunk/forward_tsn_common.h"
@@ -135,7 +135,7 @@
// normally."
auto deferred_actions =
std::move(deferred_reset_streams_->deferred_actions);
- deferred_reset_streams_ = absl::nullopt;
+ deferred_reset_streams_ = std::nullopt;
for (auto& action : deferred_actions) {
action();
@@ -152,7 +152,7 @@
RTC_DLOG(LS_VERBOSE) << log_prefix_
<< "Entering deferred reset; sender_last_assigned_tsn="
<< *sender_last_assigned_tsn;
- deferred_reset_streams_ = absl::make_optional<DeferredResetStreams>(
+ deferred_reset_streams_ = std::make_optional<DeferredResetStreams>(
tsn_unwrapper_.Unwrap(sender_last_assigned_tsn),
webrtc::flat_set<StreamID>(streams.begin(), streams.end()));
}
diff --git a/net/dcsctp/rx/reassembly_queue.h b/net/dcsctp/rx/reassembly_queue.h
index 9e0b6e2..ef6f0fe 100644
--- a/net/dcsctp/rx/reassembly_queue.h
+++ b/net/dcsctp/rx/reassembly_queue.h
@@ -151,7 +151,7 @@
std::vector<DcSctpMessage> reassembled_messages_;
// If present, "deferred reset processing" mode is active.
- absl::optional<DeferredResetStreams> deferred_reset_streams_;
+ std::optional<DeferredResetStreams> deferred_reset_streams_;
// The number of "payload bytes" that are in this queue, in total.
size_t queued_bytes_ = 0;
diff --git a/net/dcsctp/rx/traditional_reassembly_streams.cc b/net/dcsctp/rx/traditional_reassembly_streams.cc
index c94691f..296258c 100644
--- a/net/dcsctp/rx/traditional_reassembly_streams.cc
+++ b/net/dcsctp/rx/traditional_reassembly_streams.cc
@@ -16,11 +16,11 @@
#include <iterator>
#include <map>
#include <numeric>
+#include <optional>
#include <utility>
#include <vector>
#include "absl/algorithm/container.h"
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "net/dcsctp/common/sequence_numbers.h"
#include "net/dcsctp/packet/chunk/forward_tsn_common.h"
@@ -34,8 +34,8 @@
// Given a map (`chunks`) and an iterator to within that map (`iter`), this
// function will return an iterator to the first chunk in that message, which
// has the `is_beginning` flag set. If there are any gaps, or if the beginning
-// can't be found, `absl::nullopt` is returned.
-absl::optional<std::map<UnwrappedTSN, Data>::iterator> FindBeginning(
+// can't be found, `std::nullopt` is returned.
+std::optional<std::map<UnwrappedTSN, Data>::iterator> FindBeginning(
const std::map<UnwrappedTSN, Data>& chunks,
std::map<UnwrappedTSN, Data>::iterator iter) {
UnwrappedTSN prev_tsn = iter->first;
@@ -44,11 +44,11 @@
return iter;
}
if (iter == chunks.begin()) {
- return absl::nullopt;
+ return std::nullopt;
}
--iter;
if (iter->first.next_value() != prev_tsn) {
- return absl::nullopt;
+ return std::nullopt;
}
prev_tsn = iter->first;
}
@@ -57,8 +57,8 @@
// Given a map (`chunks`) and an iterator to within that map (`iter`), this
// function will return an iterator to the chunk after the last chunk in that
// message, which has the `is_end` flag set. If there are any gaps, or if the
-// end can't be found, `absl::nullopt` is returned.
-absl::optional<std::map<UnwrappedTSN, Data>::iterator> FindEnd(
+// end can't be found, `std::nullopt` is returned.
+std::optional<std::map<UnwrappedTSN, Data>::iterator> FindEnd(
std::map<UnwrappedTSN, Data>& chunks,
std::map<UnwrappedTSN, Data>::iterator iter) {
UnwrappedTSN prev_tsn = iter->first;
@@ -68,10 +68,10 @@
}
++iter;
if (iter == chunks.end()) {
- return absl::nullopt;
+ return std::nullopt;
}
if (iter->first != prev_tsn.next_value()) {
- return absl::nullopt;
+ return std::nullopt;
}
prev_tsn = iter->first;
}
@@ -108,11 +108,11 @@
// message, which can be inefficient for very large values of N. This could be
// optimized by e.g. only trying to assemble a message once _any_ beginning
// and _any_ end has been found.
- absl::optional<ChunkMap::iterator> start = FindBeginning(chunks_, iter);
+ std::optional<ChunkMap::iterator> start = FindBeginning(chunks_, iter);
if (!start.has_value()) {
return 0;
}
- absl::optional<ChunkMap::iterator> end = FindEnd(chunks_, iter);
+ std::optional<ChunkMap::iterator> end = FindEnd(chunks_, iter);
if (!end.has_value()) {
return 0;
}
diff --git a/net/dcsctp/socket/BUILD.gn b/net/dcsctp/socket/BUILD.gn
index 56c7907..1f05c2a 100644
--- a/net/dcsctp/socket/BUILD.gn
+++ b/net/dcsctp/socket/BUILD.gn
@@ -36,7 +36,6 @@
"../timer",
"//third_party/abseil-cpp/absl/functional:bind_front",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
sources = [
"heartbeat_handler.cc",
@@ -66,7 +65,6 @@
"../tx:retransmission_queue",
"//third_party/abseil-cpp/absl/functional:bind_front",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
sources = [
"stream_reset_handler.cc",
@@ -113,7 +111,6 @@
"../tx:send_queue",
"//third_party/abseil-cpp/absl/functional:bind_front",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
sources = [
"capabilities.h",
@@ -159,7 +156,6 @@
"//third_party/abseil-cpp/absl/functional:bind_front",
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
"//third_party/abseil-cpp/absl/types:variant",
]
sources = [
@@ -186,7 +182,6 @@
"../public:types",
"../timer",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -202,7 +197,6 @@
"../public:socket",
"../public:types",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -256,7 +250,6 @@
"//third_party/abseil-cpp/absl/flags:flag",
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
sources = [
"dcsctp_socket_network_test.cc",
diff --git a/net/dcsctp/socket/dcsctp_socket.cc b/net/dcsctp/socket/dcsctp_socket.cc
index 48b6edb..0db5ce1 100644
--- a/net/dcsctp/socket/dcsctp_socket.cc
+++ b/net/dcsctp/socket/dcsctp_socket.cc
@@ -13,6 +13,7 @@
#include <cstdint>
#include <limits>
#include <memory>
+#include <optional>
#include <string>
#include <utility>
#include <vector>
@@ -20,7 +21,6 @@
#include "absl/functional/bind_front.h"
#include "absl/memory/memory.h"
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "net/dcsctp/packet/chunk/abort_chunk.h"
#include "net/dcsctp/packet/chunk/chunk.h"
@@ -97,7 +97,7 @@
uint16_t peer_nbr_inbound_streams,
const Parameters& parameters) {
Capabilities capabilities;
- absl::optional<SupportedExtensionsParameter> supported_extensions =
+ std::optional<SupportedExtensionsParameter> supported_extensions =
parameters.get<SupportedExtensionsParameter>();
if (options.enable_partial_reliability) {
@@ -612,9 +612,9 @@
send_queue_.SetBufferedAmountLowThreshold(stream_id, bytes);
}
-absl::optional<Metrics> DcSctpSocket::GetMetrics() const {
+std::optional<Metrics> DcSctpSocket::GetMetrics() const {
if (tcb_ == nullptr) {
- return absl::nullopt;
+ return std::nullopt;
}
Metrics metrics = metrics_;
@@ -657,7 +657,7 @@
}
void DcSctpSocket::MaybeSendResetStreamsRequest() {
- absl::optional<ReConfigChunk> reconfig =
+ std::optional<ReConfigChunk> reconfig =
tcb_->stream_reset_handler().MakeStreamResetRequest();
if (reconfig.has_value()) {
SctpPacket::Builder builder = tcb_->PacketBuilder();
@@ -788,7 +788,7 @@
packet_observer_->OnReceivedPacket(TimeMs(callbacks_.Now().ms()), data);
}
- absl::optional<SctpPacket> packet = SctpPacket::Parse(data, options_);
+ std::optional<SctpPacket> packet = SctpPacket::Parse(data, options_);
if (!packet.has_value()) {
// https://tools.ietf.org/html/rfc4960#section-6.8
// "The default procedure for handling invalid SCTP packets is to
@@ -1044,7 +1044,7 @@
void DcSctpSocket::HandleData(const CommonHeader& header,
const SctpPacket::ChunkDescriptor& descriptor) {
- absl::optional<DataChunk> chunk = DataChunk::Parse(descriptor.data);
+ std::optional<DataChunk> chunk = DataChunk::Parse(descriptor.data);
if (ValidateParseSuccess(chunk) && ValidateHasTCB()) {
HandleDataCommon(*chunk);
}
@@ -1052,7 +1052,7 @@
void DcSctpSocket::HandleIData(const CommonHeader& header,
const SctpPacket::ChunkDescriptor& descriptor) {
- absl::optional<IDataChunk> chunk = IDataChunk::Parse(descriptor.data);
+ std::optional<IDataChunk> chunk = IDataChunk::Parse(descriptor.data);
if (ValidateParseSuccess(chunk) && ValidateHasTCB()) {
HandleDataCommon(*chunk);
}
@@ -1119,7 +1119,7 @@
void DcSctpSocket::HandleInit(const CommonHeader& header,
const SctpPacket::ChunkDescriptor& descriptor) {
- absl::optional<InitChunk> chunk = InitChunk::Parse(descriptor.data);
+ std::optional<InitChunk> chunk = InitChunk::Parse(descriptor.data);
if (!ValidateParseSuccess(chunk)) {
return;
}
@@ -1249,7 +1249,7 @@
void DcSctpSocket::HandleInitAck(
const CommonHeader& header,
const SctpPacket::ChunkDescriptor& descriptor) {
- absl::optional<InitAckChunk> chunk = InitAckChunk::Parse(descriptor.data);
+ std::optional<InitAckChunk> chunk = InitAckChunk::Parse(descriptor.data);
if (!ValidateParseSuccess(chunk)) {
return;
}
@@ -1307,14 +1307,13 @@
void DcSctpSocket::HandleCookieEcho(
const CommonHeader& header,
const SctpPacket::ChunkDescriptor& descriptor) {
- absl::optional<CookieEchoChunk> chunk =
+ std::optional<CookieEchoChunk> chunk =
CookieEchoChunk::Parse(descriptor.data);
if (!ValidateParseSuccess(chunk)) {
return;
}
- absl::optional<StateCookie> cookie =
- StateCookie::Deserialize(chunk->cookie());
+ std::optional<StateCookie> cookie = StateCookie::Deserialize(chunk->cookie());
if (!cookie.has_value()) {
callbacks_.OnError(ErrorKind::kParseFailed, "Failed to parse state cookie");
return;
@@ -1449,7 +1448,7 @@
void DcSctpSocket::HandleCookieAck(
const CommonHeader& header,
const SctpPacket::ChunkDescriptor& descriptor) {
- absl::optional<CookieAckChunk> chunk = CookieAckChunk::Parse(descriptor.data);
+ std::optional<CookieAckChunk> chunk = CookieAckChunk::Parse(descriptor.data);
if (!ValidateParseSuccess(chunk)) {
return;
}
@@ -1480,7 +1479,7 @@
void DcSctpSocket::HandleSack(const CommonHeader& header,
const SctpPacket::ChunkDescriptor& descriptor) {
- absl::optional<SackChunk> chunk = SackChunk::Parse(descriptor.data);
+ std::optional<SackChunk> chunk = SackChunk::Parse(descriptor.data);
if (ValidateParseSuccess(chunk) && ValidateHasTCB()) {
Timestamp now = callbacks_.Now();
@@ -1513,7 +1512,7 @@
void DcSctpSocket::HandleHeartbeatRequest(
const CommonHeader& header,
const SctpPacket::ChunkDescriptor& descriptor) {
- absl::optional<HeartbeatRequestChunk> chunk =
+ std::optional<HeartbeatRequestChunk> chunk =
HeartbeatRequestChunk::Parse(descriptor.data);
if (ValidateParseSuccess(chunk) && ValidateHasTCB()) {
@@ -1524,7 +1523,7 @@
void DcSctpSocket::HandleHeartbeatAck(
const CommonHeader& header,
const SctpPacket::ChunkDescriptor& descriptor) {
- absl::optional<HeartbeatAckChunk> chunk =
+ std::optional<HeartbeatAckChunk> chunk =
HeartbeatAckChunk::Parse(descriptor.data);
if (ValidateParseSuccess(chunk) && ValidateHasTCB()) {
@@ -1534,7 +1533,7 @@
void DcSctpSocket::HandleAbort(const CommonHeader& header,
const SctpPacket::ChunkDescriptor& descriptor) {
- absl::optional<AbortChunk> chunk = AbortChunk::Parse(descriptor.data);
+ std::optional<AbortChunk> chunk = AbortChunk::Parse(descriptor.data);
if (ValidateParseSuccess(chunk)) {
std::string error_string = ErrorCausesToString(chunk->error_causes());
if (tcb_ == nullptr) {
@@ -1554,7 +1553,7 @@
void DcSctpSocket::HandleError(const CommonHeader& header,
const SctpPacket::ChunkDescriptor& descriptor) {
- absl::optional<ErrorChunk> chunk = ErrorChunk::Parse(descriptor.data);
+ std::optional<ErrorChunk> chunk = ErrorChunk::Parse(descriptor.data);
if (ValidateParseSuccess(chunk)) {
std::string error_string = ErrorCausesToString(chunk->error_causes());
if (tcb_ == nullptr) {
@@ -1573,7 +1572,7 @@
const CommonHeader& header,
const SctpPacket::ChunkDescriptor& descriptor) {
Timestamp now = callbacks_.Now();
- absl::optional<ReConfigChunk> chunk = ReConfigChunk::Parse(descriptor.data);
+ std::optional<ReConfigChunk> chunk = ReConfigChunk::Parse(descriptor.data);
if (ValidateParseSuccess(chunk) && ValidateHasTCB()) {
tcb_->stream_reset_handler().HandleReConfig(*std::move(chunk));
// Handling this response may result in outgoing stream resets finishing
@@ -1694,7 +1693,7 @@
void DcSctpSocket::HandleForwardTsn(
const CommonHeader& header,
const SctpPacket::ChunkDescriptor& descriptor) {
- absl::optional<ForwardTsnChunk> chunk =
+ std::optional<ForwardTsnChunk> chunk =
ForwardTsnChunk::Parse(descriptor.data);
if (ValidateParseSuccess(chunk) && ValidateHasTCB()) {
HandleForwardTsnCommon(*chunk);
@@ -1704,7 +1703,7 @@
void DcSctpSocket::HandleIForwardTsn(
const CommonHeader& header,
const SctpPacket::ChunkDescriptor& descriptor) {
- absl::optional<IForwardTsnChunk> chunk =
+ std::optional<IForwardTsnChunk> chunk =
IForwardTsnChunk::Parse(descriptor.data);
if (ValidateParseSuccess(chunk) && ValidateHasTCB()) {
HandleForwardTsnCommon(*chunk);
@@ -1788,12 +1787,12 @@
return status;
}
-absl::optional<DcSctpSocketHandoverState>
+std::optional<DcSctpSocketHandoverState>
DcSctpSocket::GetHandoverStateAndClose() {
CallbackDeferrer::ScopedDeferrer deferrer(callbacks_);
if (!GetHandoverReadiness().IsReady()) {
- return absl::nullopt;
+ return std::nullopt;
}
DcSctpSocketHandoverState state;
diff --git a/net/dcsctp/socket/dcsctp_socket.h b/net/dcsctp/socket/dcsctp_socket.h
index c65571a..f694307 100644
--- a/net/dcsctp/socket/dcsctp_socket.h
+++ b/net/dcsctp/socket/dcsctp_socket.h
@@ -103,9 +103,9 @@
size_t buffered_amount(StreamID stream_id) const override;
size_t buffered_amount_low_threshold(StreamID stream_id) const override;
void SetBufferedAmountLowThreshold(StreamID stream_id, size_t bytes) override;
- absl::optional<Metrics> GetMetrics() const override;
+ std::optional<Metrics> GetMetrics() const override;
HandoverReadinessStatus GetHandoverReadiness() const override;
- absl::optional<DcSctpSocketHandoverState> GetHandoverStateAndClose() override;
+ std::optional<DcSctpSocketHandoverState> GetHandoverStateAndClose() override;
SctpImplementation peer_implementation() const override {
return metrics_.peer_implementation;
}
@@ -191,7 +191,7 @@
// Returns true if the parsing of a chunk of type `T` succeeded. If it didn't,
// it reports an error and returns false.
template <class T>
- bool ValidateParseSuccess(const absl::optional<T>& c) {
+ bool ValidateParseSuccess(const std::optional<T>& c) {
if (c.has_value()) {
return true;
}
diff --git a/net/dcsctp/socket/dcsctp_socket_network_test.cc b/net/dcsctp/socket/dcsctp_socket_network_test.cc
index f497cfa..679ed17 100644
--- a/net/dcsctp/socket/dcsctp_socket_network_test.cc
+++ b/net/dcsctp/socket/dcsctp_socket_network_test.cc
@@ -10,13 +10,13 @@
#include <cstdint>
#include <deque>
#include <memory>
+#include <optional>
#include <string>
#include <utility>
#include <vector>
#include "absl/memory/memory.h"
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/task_queue/pending_task_safety_flag.h"
#include "api/task_queue/task_queue_base.h"
@@ -243,7 +243,7 @@
std::vector<uint8_t>(kLargePayloadSize)),
send_options);
- send_options.max_retransmissions = absl::nullopt;
+ send_options.max_retransmissions = std::nullopt;
sctp_socket_.Send(
DcSctpMessage(kStreamId, kPpid,
std::vector<uint8_t>(kSmallPayloadSize)),
@@ -252,12 +252,12 @@
}
}
- absl::optional<DcSctpMessage> ConsumeReceivedMessage() {
+ std::optional<DcSctpMessage> ConsumeReceivedMessage() {
if (!last_received_message_.has_value()) {
- return absl::nullopt;
+ return std::nullopt;
}
DcSctpMessage ret = *std::move(last_received_message_);
- last_received_message_ = absl::nullopt;
+ last_received_message_ = std::nullopt;
return ret;
}
@@ -317,7 +317,7 @@
webrtc::Random random_;
DcSctpSocket sctp_socket_;
size_t received_bytes_ = 0;
- absl::optional<DcSctpMessage> last_received_message_;
+ std::optional<DcSctpMessage> last_received_message_;
Timestamp last_bandwidth_printout_;
// Per-second received bitrates, in Mbps
std::vector<double> received_bitrate_mbps_;
diff --git a/net/dcsctp/socket/dcsctp_socket_test.cc b/net/dcsctp/socket/dcsctp_socket_test.cc
index 7266dc8..cea30a9 100644
--- a/net/dcsctp/socket/dcsctp_socket_test.cc
+++ b/net/dcsctp/socket/dcsctp_socket_test.cc
@@ -13,6 +13,7 @@
#include <cstdint>
#include <deque>
#include <memory>
+#include <optional>
#include <string>
#include <utility>
#include <vector>
@@ -20,7 +21,6 @@
#include "absl/flags/flag.h"
#include "absl/memory/memory.h"
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "net/dcsctp/common/handover_testing.h"
#include "net/dcsctp/common/math.h"
@@ -84,7 +84,7 @@
constexpr DcSctpOptions kDefaultOptions;
MATCHER_P(HasChunks, chunks, "") {
- absl::optional<SctpPacket> packet = SctpPacket::Parse(arg, kDefaultOptions);
+ std::optional<SctpPacket> packet = SctpPacket::Parse(arg, kDefaultOptions);
if (!packet.has_value()) {
*result_listener << "data didn't parse as an SctpPacket";
return false;
@@ -103,7 +103,7 @@
return false;
}
- absl::optional<DataChunk> chunk = DataChunk::Parse(arg.data);
+ std::optional<DataChunk> chunk = DataChunk::Parse(arg.data);
if (!chunk.has_value()) {
*result_listener << "The chunk didn't parse as a data chunk";
return false;
@@ -118,7 +118,7 @@
return false;
}
- absl::optional<SackChunk> chunk = SackChunk::Parse(arg.data);
+ std::optional<SackChunk> chunk = SackChunk::Parse(arg.data);
if (!chunk.has_value()) {
*result_listener << "The chunk didn't parse as a sack chunk";
return false;
@@ -133,7 +133,7 @@
return false;
}
- absl::optional<ReConfigChunk> chunk = ReConfigChunk::Parse(arg.data);
+ std::optional<ReConfigChunk> chunk = ReConfigChunk::Parse(arg.data);
if (!chunk.has_value()) {
*result_listener << "The chunk didn't parse as a re-config chunk";
return false;
@@ -148,7 +148,7 @@
return false;
}
- absl::optional<HeartbeatAckChunk> chunk = HeartbeatAckChunk::Parse(arg.data);
+ std::optional<HeartbeatAckChunk> chunk = HeartbeatAckChunk::Parse(arg.data);
if (!chunk.has_value()) {
*result_listener << "The chunk didn't parse as a HeartbeatAckChunk";
return false;
@@ -163,7 +163,7 @@
return false;
}
- absl::optional<HeartbeatRequestChunk> chunk =
+ std::optional<HeartbeatRequestChunk> chunk =
HeartbeatRequestChunk::Parse(arg.data);
if (!chunk.has_value()) {
*result_listener << "The chunk didn't parse as a HeartbeatRequestChunk";
@@ -185,7 +185,7 @@
return false;
}
- absl::optional<OutgoingSSNResetRequestParameter> parameter =
+ std::optional<OutgoingSSNResetRequestParameter> parameter =
OutgoingSSNResetRequestParameter::Parse(arg.data);
if (!parameter.has_value()) {
*result_listener
@@ -203,7 +203,7 @@
return false;
}
- absl::optional<ReconfigurationResponseParameter> parameter =
+ std::optional<ReconfigurationResponseParameter> parameter =
ReconfigurationResponseParameter::Parse(arg.data);
if (!parameter.has_value()) {
*result_listener
@@ -264,7 +264,7 @@
void RunTimers(SocketUnderTest& s) {
for (;;) {
- absl::optional<TimeoutID> timeout_id = s.cb.GetNextExpiredTimeout();
+ std::optional<TimeoutID> timeout_id = s.cb.GetNextExpiredTimeout();
if (!timeout_id.has_value()) {
break;
}
@@ -324,7 +324,7 @@
if (!is_closed) {
EXPECT_CALL(sut->cb, OnClosed).Times(1);
}
- absl::optional<DcSctpSocketHandoverState> handover_state =
+ std::optional<DcSctpSocketHandoverState> handover_state =
sut->socket.GetHandoverStateAndClose();
EXPECT_TRUE(handover_state.has_value());
g_handover_state_transformer_for_test(&*handover_state);
@@ -340,7 +340,7 @@
std::vector<uint32_t> GetReceivedMessagePpids(SocketUnderTest& z) {
std::vector<uint32_t> ppids;
for (;;) {
- absl::optional<DcSctpMessage> msg = z.cb.ConsumeReceivedMessage();
+ std::optional<DcSctpMessage> msg = z.cb.ConsumeReceivedMessage();
if (!msg.has_value()) {
break;
}
@@ -382,7 +382,7 @@
a.socket.Send(DcSctpMessage(StreamID(1), PPID(53), {1, 2}), kSendOptions);
ExchangeMessages(a, *z);
- absl::optional<DcSctpMessage> msg = z->cb.ConsumeReceivedMessage();
+ std::optional<DcSctpMessage> msg = z->cb.ConsumeReceivedMessage();
ASSERT_TRUE(msg.has_value());
EXPECT_EQ(msg->stream_id(), StreamID(1));
}
@@ -792,7 +792,7 @@
EXPECT_EQ(a.socket.state(), SocketState::kConnected);
EXPECT_EQ(z.socket.state(), SocketState::kConnected);
- absl::optional<DcSctpMessage> msg = z.cb.ConsumeReceivedMessage();
+ std::optional<DcSctpMessage> msg = z.cb.ConsumeReceivedMessage();
ASSERT_TRUE(msg.has_value());
EXPECT_EQ(msg->stream_id(), StreamID(1));
}
@@ -806,7 +806,7 @@
a.socket.Send(DcSctpMessage(StreamID(1), PPID(53), {1, 2}), kSendOptions);
z.socket.ReceivePacket(a.cb.ConsumeSentPacket());
- absl::optional<DcSctpMessage> msg = z.cb.ConsumeReceivedMessage();
+ std::optional<DcSctpMessage> msg = z.cb.ConsumeReceivedMessage();
ASSERT_TRUE(msg.has_value());
EXPECT_EQ(msg->stream_id(), StreamID(1));
}
@@ -826,7 +826,7 @@
z->socket.ReceivePacket(a.cb.ConsumeSentPacket());
- absl::optional<DcSctpMessage> msg = z->cb.ConsumeReceivedMessage();
+ std::optional<DcSctpMessage> msg = z->cb.ConsumeReceivedMessage();
ASSERT_TRUE(msg.has_value());
EXPECT_EQ(msg->stream_id(), StreamID(1));
@@ -851,7 +851,7 @@
// Retransmit and handle the rest
ExchangeMessages(a, *z);
- absl::optional<DcSctpMessage> msg = z->cb.ConsumeReceivedMessage();
+ std::optional<DcSctpMessage> msg = z->cb.ConsumeReceivedMessage();
ASSERT_TRUE(msg.has_value());
EXPECT_EQ(msg->stream_id(), StreamID(1));
EXPECT_THAT(msg->payload(), testing::ElementsAreArray(payload));
@@ -1051,7 +1051,7 @@
a.socket.Send(DcSctpMessage(StreamID(1), PPID(53), {1, 2}), {});
z->socket.ReceivePacket(a.cb.ConsumeSentPacket());
- absl::optional<DcSctpMessage> msg = z->cb.ConsumeReceivedMessage();
+ std::optional<DcSctpMessage> msg = z->cb.ConsumeReceivedMessage();
ASSERT_TRUE(msg.has_value());
EXPECT_EQ(msg->stream_id(), StreamID(1));
@@ -1100,11 +1100,11 @@
// Handle SACK
a.socket.ReceivePacket(z->cb.ConsumeSentPacket());
- absl::optional<DcSctpMessage> msg1 = z->cb.ConsumeReceivedMessage();
+ std::optional<DcSctpMessage> msg1 = z->cb.ConsumeReceivedMessage();
ASSERT_TRUE(msg1.has_value());
EXPECT_EQ(msg1->stream_id(), StreamID(1));
- absl::optional<DcSctpMessage> msg2 = z->cb.ConsumeReceivedMessage();
+ std::optional<DcSctpMessage> msg2 = z->cb.ConsumeReceivedMessage();
ASSERT_TRUE(msg2.has_value());
EXPECT_EQ(msg2->stream_id(), StreamID(1));
@@ -1182,19 +1182,19 @@
a.socket.ReceivePacket(z->cb.ConsumeSentPacket());
// Receive all messages.
- absl::optional<DcSctpMessage> msg1 = z->cb.ConsumeReceivedMessage();
+ std::optional<DcSctpMessage> msg1 = z->cb.ConsumeReceivedMessage();
ASSERT_TRUE(msg1.has_value());
EXPECT_EQ(msg1->stream_id(), StreamID(1));
- absl::optional<DcSctpMessage> msg2 = z->cb.ConsumeReceivedMessage();
+ std::optional<DcSctpMessage> msg2 = z->cb.ConsumeReceivedMessage();
ASSERT_TRUE(msg2.has_value());
EXPECT_EQ(msg2->stream_id(), StreamID(1));
- absl::optional<DcSctpMessage> msg3 = z->cb.ConsumeReceivedMessage();
+ std::optional<DcSctpMessage> msg3 = z->cb.ConsumeReceivedMessage();
ASSERT_TRUE(msg3.has_value());
EXPECT_EQ(msg3->stream_id(), StreamID(3));
- absl::optional<DcSctpMessage> msg4 = z->cb.ConsumeReceivedMessage();
+ std::optional<DcSctpMessage> msg4 = z->cb.ConsumeReceivedMessage();
ASSERT_TRUE(msg4.has_value());
EXPECT_EQ(msg4->stream_id(), StreamID(3));
@@ -1253,7 +1253,7 @@
// have the wrong verification tag, those will yield errors.
ExchangeMessages(a, z2);
- absl::optional<DcSctpMessage> msg = z2.cb.ConsumeReceivedMessage();
+ std::optional<DcSctpMessage> msg = z2.cb.ConsumeReceivedMessage();
ASSERT_TRUE(msg.has_value());
EXPECT_EQ(msg->stream_id(), StreamID(1));
EXPECT_THAT(msg->payload(), testing::ElementsAreArray(payload));
@@ -1304,15 +1304,15 @@
// Which will trigger a SACK
a.socket.ReceivePacket(z->cb.ConsumeSentPacket());
- absl::optional<DcSctpMessage> msg1 = z->cb.ConsumeReceivedMessage();
+ std::optional<DcSctpMessage> msg1 = z->cb.ConsumeReceivedMessage();
ASSERT_TRUE(msg1.has_value());
EXPECT_EQ(msg1->ppid(), PPID(51));
- absl::optional<DcSctpMessage> msg2 = z->cb.ConsumeReceivedMessage();
+ std::optional<DcSctpMessage> msg2 = z->cb.ConsumeReceivedMessage();
ASSERT_TRUE(msg2.has_value());
EXPECT_EQ(msg2->ppid(), PPID(53));
- absl::optional<DcSctpMessage> msg3 = z->cb.ConsumeReceivedMessage();
+ std::optional<DcSctpMessage> msg3 = z->cb.ConsumeReceivedMessage();
EXPECT_FALSE(msg3.has_value());
MaybeHandoverSocketAndSendMessage(a, std::move(z));
@@ -1395,7 +1395,7 @@
ExchangeMessages(a, *z);
- absl::optional<DcSctpMessage> msg1 = z->cb.ConsumeReceivedMessage();
+ std::optional<DcSctpMessage> msg1 = z->cb.ConsumeReceivedMessage();
ASSERT_TRUE(msg1.has_value());
EXPECT_EQ(msg1->ppid(), PPID(54));
@@ -2154,7 +2154,7 @@
z = HandoverSocket(std::move(z));
- absl::optional<DcSctpMessage> msg;
+ std::optional<DcSctpMessage> msg;
RTC_LOG(LS_INFO) << "Sending A #1";
@@ -2241,7 +2241,7 @@
ExchangeMessages(a, *z);
// The Z socket should receive the second message, but not the first.
- absl::optional<DcSctpMessage> msg = z->cb.ConsumeReceivedMessage();
+ std::optional<DcSctpMessage> msg = z->cb.ConsumeReceivedMessage();
ASSERT_TRUE(msg.has_value());
EXPECT_EQ(msg->ppid(), PPID(52));
@@ -2380,13 +2380,13 @@
ExchangeMessages(a, z);
// Receive these messages
- absl::optional<DcSctpMessage> msg1 = z.cb.ConsumeReceivedMessage();
+ std::optional<DcSctpMessage> msg1 = z.cb.ConsumeReceivedMessage();
ASSERT_TRUE(msg1.has_value());
EXPECT_EQ(msg1->stream_id(), StreamID(1));
- absl::optional<DcSctpMessage> msg2 = z.cb.ConsumeReceivedMessage();
+ std::optional<DcSctpMessage> msg2 = z.cb.ConsumeReceivedMessage();
ASSERT_TRUE(msg2.has_value());
EXPECT_EQ(msg2->stream_id(), StreamID(2));
- absl::optional<DcSctpMessage> msg3 = z.cb.ConsumeReceivedMessage();
+ std::optional<DcSctpMessage> msg3 = z.cb.ConsumeReceivedMessage();
ASSERT_TRUE(msg3.has_value());
EXPECT_EQ(msg3->stream_id(), StreamID(3));
@@ -2415,13 +2415,13 @@
ExchangeMessages(a, z);
// Receive these messages
- absl::optional<DcSctpMessage> msg4 = z.cb.ConsumeReceivedMessage();
+ std::optional<DcSctpMessage> msg4 = z.cb.ConsumeReceivedMessage();
ASSERT_TRUE(msg4.has_value());
EXPECT_EQ(msg4->stream_id(), StreamID(1));
- absl::optional<DcSctpMessage> msg5 = z.cb.ConsumeReceivedMessage();
+ std::optional<DcSctpMessage> msg5 = z.cb.ConsumeReceivedMessage();
ASSERT_TRUE(msg5.has_value());
EXPECT_EQ(msg5->stream_id(), StreamID(2));
- absl::optional<DcSctpMessage> msg6 = z.cb.ConsumeReceivedMessage();
+ std::optional<DcSctpMessage> msg6 = z.cb.ConsumeReceivedMessage();
ASSERT_TRUE(msg6.has_value());
EXPECT_EQ(msg6->stream_id(), StreamID(3));
}
@@ -2527,7 +2527,7 @@
std::vector<uint32_t> received_ppids;
for (;;) {
- absl::optional<DcSctpMessage> msg = z.cb.ConsumeReceivedMessage();
+ std::optional<DcSctpMessage> msg = z.cb.ConsumeReceivedMessage();
if (!msg.has_value()) {
break;
}
@@ -2800,13 +2800,13 @@
z.socket.ReceivePacket(data2);
z.socket.ReceivePacket(data3);
- absl::optional<DcSctpMessage> msg1 = z.cb.ConsumeReceivedMessage();
+ std::optional<DcSctpMessage> msg1 = z.cb.ConsumeReceivedMessage();
ASSERT_TRUE(msg1.has_value());
EXPECT_EQ(msg1->stream_id(), StreamID(1));
EXPECT_EQ(msg1->ppid(), PPID(53));
EXPECT_EQ(msg1->payload().size(), kTwoFragmentsSize);
- absl::optional<DcSctpMessage> msg2 = z.cb.ConsumeReceivedMessage();
+ std::optional<DcSctpMessage> msg2 = z.cb.ConsumeReceivedMessage();
ASSERT_TRUE(msg2.has_value());
EXPECT_EQ(msg2->stream_id(), StreamID(1));
EXPECT_EQ(msg2->ppid(), PPID(54));
@@ -2855,7 +2855,7 @@
{});
ExchangeMessages(a, z);
- absl::optional<DcSctpMessage> msg3 = z.cb.ConsumeReceivedMessage();
+ std::optional<DcSctpMessage> msg3 = z.cb.ConsumeReceivedMessage();
ASSERT_TRUE(msg3.has_value());
EXPECT_EQ(msg3->stream_id(), StreamID(1));
EXPECT_EQ(msg3->ppid(), PPID(55));
@@ -2883,13 +2883,13 @@
EXPECT_CALL(z.cb, OnIncomingStreamsReset(ElementsAre(StreamID(1))));
ExchangeMessages(a, z);
- absl::optional<DcSctpMessage> msg1 = z.cb.ConsumeReceivedMessage();
+ std::optional<DcSctpMessage> msg1 = z.cb.ConsumeReceivedMessage();
ASSERT_TRUE(msg1.has_value());
EXPECT_EQ(msg1->stream_id(), StreamID(1));
EXPECT_EQ(msg1->ppid(), PPID(51));
EXPECT_EQ(msg1->payload().size(), kSmallMessageSize);
- absl::optional<DcSctpMessage> msg2 = z.cb.ConsumeReceivedMessage();
+ std::optional<DcSctpMessage> msg2 = z.cb.ConsumeReceivedMessage();
ASSERT_TRUE(msg2.has_value());
EXPECT_EQ(msg2->stream_id(), StreamID(1));
EXPECT_EQ(msg2->ppid(), PPID(52));
@@ -3057,11 +3057,11 @@
}
}
- absl::optional<DcSctpMessage> msg1 = z->cb.ConsumeReceivedMessage();
+ std::optional<DcSctpMessage> msg1 = z->cb.ConsumeReceivedMessage();
ASSERT_TRUE(msg1.has_value());
EXPECT_THAT(msg1->payload(), SizeIs(kLargeMessageSize));
- absl::optional<DcSctpMessage> msg2 = a.cb.ConsumeReceivedMessage();
+ std::optional<DcSctpMessage> msg2 = a.cb.ConsumeReceivedMessage();
ASSERT_TRUE(msg2.has_value());
EXPECT_THAT(msg2->payload(), SizeIs(kLargeMessageSize));
@@ -3223,7 +3223,7 @@
// Then let the rest continue.
ExchangeMessages(a, z);
- absl::optional<DcSctpMessage> msg = z.cb.ConsumeReceivedMessage();
+ std::optional<DcSctpMessage> msg = z.cb.ConsumeReceivedMessage();
ASSERT_TRUE(msg.has_value());
EXPECT_EQ(msg->stream_id(), StreamID(1));
EXPECT_THAT(msg->payload(), SizeIs(kLargeMessageSize));
@@ -3260,7 +3260,7 @@
// Then let the rest continue.
ExchangeMessages(a, z);
- absl::optional<DcSctpMessage> msg = z.cb.ConsumeReceivedMessage();
+ std::optional<DcSctpMessage> msg = z.cb.ConsumeReceivedMessage();
ASSERT_TRUE(msg.has_value());
EXPECT_EQ(msg->stream_id(), StreamID(1));
EXPECT_THAT(msg->payload(), SizeIs(kLargeMessageSize));
diff --git a/net/dcsctp/socket/heartbeat_handler.cc b/net/dcsctp/socket/heartbeat_handler.cc
index 31211e0..d5ff6fc 100644
--- a/net/dcsctp/socket/heartbeat_handler.cc
+++ b/net/dcsctp/socket/heartbeat_handler.cc
@@ -13,13 +13,13 @@
#include <cstdint>
#include <memory>
+#include <optional>
#include <string>
#include <utility>
#include <vector>
#include "absl/functional/bind_front.h"
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/units/time_delta.h"
#include "net/dcsctp/packet/bounded_byte_reader.h"
@@ -65,12 +65,12 @@
return data;
}
- static absl::optional<HeartbeatInfo> Deserialize(
+ static std::optional<HeartbeatInfo> Deserialize(
rtc::ArrayView<const uint8_t> data) {
if (data.size() != kBufferSize) {
RTC_LOG(LS_WARNING) << "Invalid heartbeat info: " << data.size()
<< " bytes";
- return absl::nullopt;
+ return std::nullopt;
}
BoundedByteReader<kBufferSize> reader(data);
@@ -142,14 +142,14 @@
void HeartbeatHandler::HandleHeartbeatAck(HeartbeatAckChunk chunk) {
timeout_timer_->Stop();
- absl::optional<HeartbeatInfoParameter> info_param = chunk.info();
+ std::optional<HeartbeatInfoParameter> info_param = chunk.info();
if (!info_param.has_value()) {
ctx_->callbacks().OnError(
ErrorKind::kParseFailed,
"Failed to parse HEARTBEAT-ACK; No Heartbeat Info parameter");
return;
}
- absl::optional<HeartbeatInfo> info =
+ std::optional<HeartbeatInfo> info =
HeartbeatInfo::Deserialize(info_param->info());
if (!info.has_value()) {
ctx_->callbacks().OnError(ErrorKind::kParseFailed,
diff --git a/net/dcsctp/socket/heartbeat_handler_test.cc b/net/dcsctp/socket/heartbeat_handler_test.cc
index b7f3a60..8f6af0c 100644
--- a/net/dcsctp/socket/heartbeat_handler_test.cc
+++ b/net/dcsctp/socket/heartbeat_handler_test.cc
@@ -56,7 +56,7 @@
void AdvanceTime(webrtc::TimeDelta duration) {
callbacks_.AdvanceTime(duration);
for (;;) {
- absl::optional<TimeoutID> timeout_id = callbacks_.GetNextExpiredTimeout();
+ std::optional<TimeoutID> timeout_id = callbacks_.GetNextExpiredTimeout();
if (!timeout_id.has_value()) {
break;
}
diff --git a/net/dcsctp/socket/mock_context.h b/net/dcsctp/socket/mock_context.h
index bbd9cd1..5980169 100644
--- a/net/dcsctp/socket/mock_context.h
+++ b/net/dcsctp/socket/mock_context.h
@@ -11,9 +11,9 @@
#define NET_DCSCTP_SOCKET_MOCK_CONTEXT_H_
#include <cstdint>
+#include <optional>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "net/dcsctp/packet/sctp_packet.h"
#include "net/dcsctp/public/dcsctp_options.h"
#include "net/dcsctp/public/dcsctp_socket.h"
diff --git a/net/dcsctp/socket/mock_dcsctp_socket_callbacks.h b/net/dcsctp/socket/mock_dcsctp_socket_callbacks.h
index 972e547..827609f 100644
--- a/net/dcsctp/socket/mock_dcsctp_socket_callbacks.h
+++ b/net/dcsctp/socket/mock_dcsctp_socket_callbacks.h
@@ -13,12 +13,12 @@
#include <cstdint>
#include <deque>
#include <memory>
+#include <optional>
#include <string>
#include <utility>
#include <vector>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/task_queue/task_queue_base.h"
#include "net/dcsctp/public/dcsctp_message.h"
@@ -150,9 +150,9 @@
sent_packets_.pop_front();
return ret;
}
- absl::optional<DcSctpMessage> ConsumeReceivedMessage() {
+ std::optional<DcSctpMessage> ConsumeReceivedMessage() {
if (received_messages_.empty()) {
- return absl::nullopt;
+ return std::nullopt;
}
DcSctpMessage ret = std::move(received_messages_.front());
received_messages_.pop_front();
@@ -162,7 +162,7 @@
void AdvanceTime(webrtc::TimeDelta duration) { now_ = now_ + duration; }
void SetTime(webrtc::Timestamp now) { now_ = now; }
- absl::optional<TimeoutID> GetNextExpiredTimeout() {
+ std::optional<TimeoutID> GetNextExpiredTimeout() {
return timeout_manager_.GetNextExpiredTimeout();
}
diff --git a/net/dcsctp/socket/state_cookie.cc b/net/dcsctp/socket/state_cookie.cc
index 1e7c43f..31488c3 100644
--- a/net/dcsctp/socket/state_cookie.cc
+++ b/net/dcsctp/socket/state_cookie.cc
@@ -10,9 +10,9 @@
#include "net/dcsctp/socket/state_cookie.h"
#include <cstdint>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "net/dcsctp/packet/bounded_byte_reader.h"
#include "net/dcsctp/packet/bounded_byte_writer.h"
@@ -48,12 +48,12 @@
return cookie;
}
-absl::optional<StateCookie> StateCookie::Deserialize(
+std::optional<StateCookie> StateCookie::Deserialize(
rtc::ArrayView<const uint8_t> cookie) {
if (cookie.size() != kCookieSize) {
RTC_DLOG(LS_WARNING) << "Invalid state cookie: " << cookie.size()
<< " bytes";
- return absl::nullopt;
+ return std::nullopt;
}
BoundedByteReader<kCookieSize> buffer(cookie);
@@ -61,7 +61,7 @@
uint32_t magic2 = buffer.Load32<4>();
if (magic1 != kMagic1 || magic2 != kMagic2) {
RTC_DLOG(LS_WARNING) << "Invalid state cookie; wrong magic";
- return absl::nullopt;
+ return std::nullopt;
}
VerificationTag peer_tag(buffer.Load32<8>());
diff --git a/net/dcsctp/socket/state_cookie.h b/net/dcsctp/socket/state_cookie.h
index 734d64d..731a68e 100644
--- a/net/dcsctp/socket/state_cookie.h
+++ b/net/dcsctp/socket/state_cookie.h
@@ -11,9 +11,9 @@
#define NET_DCSCTP_SOCKET_STATE_COOKIE_H_
#include <cstdint>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "net/dcsctp/common/internal_types.h"
#include "net/dcsctp/socket/capabilities.h"
@@ -47,8 +47,8 @@
// Returns a serialized version of this cookie.
std::vector<uint8_t> Serialize();
- // Deserializes the cookie, and returns absl::nullopt if that failed.
- static absl::optional<StateCookie> Deserialize(
+ // Deserializes the cookie, and returns std::nullopt if that failed.
+ static std::optional<StateCookie> Deserialize(
rtc::ArrayView<const uint8_t> cookie);
VerificationTag peer_tag() const { return peer_tag_; }
diff --git a/net/dcsctp/socket/stream_reset_handler.cc b/net/dcsctp/socket/stream_reset_handler.cc
index fafb993..322ecd5 100644
--- a/net/dcsctp/socket/stream_reset_handler.cc
+++ b/net/dcsctp/socket/stream_reset_handler.cc
@@ -11,10 +11,10 @@
#include <cstdint>
#include <memory>
+#include <optional>
#include <utility>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/units/time_delta.h"
#include "net/dcsctp/common/internal_types.h"
@@ -84,10 +84,10 @@
return false;
}
-absl::optional<std::vector<ReconfigurationResponseParameter>>
+std::optional<std::vector<ReconfigurationResponseParameter>>
StreamResetHandler::Process(const ReConfigChunk& chunk) {
if (!Validate(chunk)) {
- return absl::nullopt;
+ return std::nullopt;
}
std::vector<ReconfigurationResponseParameter> responses;
@@ -112,7 +112,7 @@
}
void StreamResetHandler::HandleReConfig(ReConfigChunk chunk) {
- absl::optional<std::vector<ReconfigurationResponseParameter>> responses =
+ std::optional<std::vector<ReconfigurationResponseParameter>> responses =
Process(chunk);
if (!responses.has_value()) {
@@ -168,7 +168,7 @@
void StreamResetHandler::HandleResetOutgoing(
const ParameterDescriptor& descriptor,
std::vector<ReconfigurationResponseParameter>& responses) {
- absl::optional<OutgoingSSNResetRequestParameter> req =
+ std::optional<OutgoingSSNResetRequestParameter> req =
OutgoingSSNResetRequestParameter::Parse(descriptor.data);
if (!req.has_value()) {
ctx_->callbacks().OnError(ErrorKind::kParseFailed,
@@ -222,7 +222,7 @@
void StreamResetHandler::HandleResetIncoming(
const ParameterDescriptor& descriptor,
std::vector<ReconfigurationResponseParameter>& responses) {
- absl::optional<IncomingSSNResetRequestParameter> req =
+ std::optional<IncomingSSNResetRequestParameter> req =
IncomingSSNResetRequestParameter::Parse(descriptor.data);
if (!req.has_value()) {
ctx_->callbacks().OnError(ErrorKind::kParseFailed,
@@ -242,7 +242,7 @@
}
void StreamResetHandler::HandleResponse(const ParameterDescriptor& descriptor) {
- absl::optional<ReconfigurationResponseParameter> resp =
+ std::optional<ReconfigurationResponseParameter> resp =
ReconfigurationResponseParameter::Parse(descriptor.data);
if (!resp.has_value()) {
ctx_->callbacks().OnError(
@@ -266,7 +266,7 @@
sb << *stream_id;
});
ctx_->callbacks().OnStreamsResetPerformed(current_request_->streams());
- current_request_ = absl::nullopt;
+ current_request_ = std::nullopt;
retransmission_queue_->CommitResetStreams();
break;
case ResponseResult::kInProgress:
@@ -296,20 +296,20 @@
});
ctx_->callbacks().OnStreamsResetFailed(current_request_->streams(),
ToString(resp->result()));
- current_request_ = absl::nullopt;
+ current_request_ = std::nullopt;
retransmission_queue_->RollbackResetStreams();
break;
}
}
}
-absl::optional<ReConfigChunk> StreamResetHandler::MakeStreamResetRequest() {
+std::optional<ReConfigChunk> StreamResetHandler::MakeStreamResetRequest() {
// Only send stream resets if there are streams to reset, and no current
// ongoing request (there can only be one at a time), and if the stream
// can be reset.
if (current_request_.has_value() ||
!retransmission_queue_->HasStreamsReadyToBeReset()) {
- return absl::nullopt;
+ return std::nullopt;
}
current_request_.emplace(retransmission_queue_->last_assigned_tsn(),
diff --git a/net/dcsctp/socket/stream_reset_handler.h b/net/dcsctp/socket/stream_reset_handler.h
index 77e8f3b..ac0deb7 100644
--- a/net/dcsctp/socket/stream_reset_handler.h
+++ b/net/dcsctp/socket/stream_reset_handler.h
@@ -12,13 +12,13 @@
#include <cstdint>
#include <memory>
+#include <optional>
#include <string>
#include <utility>
#include <vector>
#include "absl/functional/bind_front.h"
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/units/time_delta.h"
#include "net/dcsctp/common/internal_types.h"
@@ -103,10 +103,10 @@
void ResetStreams(rtc::ArrayView<const StreamID> outgoing_streams);
// Creates a Reset Streams request that must be sent if returned. Will start
- // the reconfig timer. Will return absl::nullopt if there is no need to
+ // the reconfig timer. Will return std::nullopt if there is no need to
// create a request (no streams to reset) or if there already is an ongoing
// stream reset request that hasn't completed yet.
- absl::optional<ReConfigChunk> MakeStreamResetRequest();
+ std::optional<ReConfigChunk> MakeStreamResetRequest();
// Called when handling and incoming RE-CONFIG chunk.
void HandleReConfig(ReConfigChunk chunk);
@@ -124,7 +124,7 @@
class CurrentRequest {
public:
CurrentRequest(TSN sender_last_assigned_tsn, std::vector<StreamID> streams)
- : req_seq_nbr_(absl::nullopt),
+ : req_seq_nbr_(std::nullopt),
sender_last_assigned_tsn_(sender_last_assigned_tsn),
streams_(std::move(streams)) {}
@@ -152,7 +152,7 @@
// If the receiver can't apply the request yet (and answered "In Progress"),
// this will be called to prepare the request to be retransmitted at a later
// time.
- void PrepareRetransmission() { req_seq_nbr_ = absl::nullopt; }
+ void PrepareRetransmission() { req_seq_nbr_ = std::nullopt; }
// If the request hasn't been sent yet, this assigns it a request number.
void PrepareToSend(ReconfigRequestSN new_req_seq_nbr) {
@@ -164,7 +164,7 @@
// has been prepared, but has not yet been sent. This is typically used when
// the peer responded "in progress" and the same request (but a different
// request number) must be sent again.
- absl::optional<ReconfigRequestSN> req_seq_nbr_;
+ std::optional<ReconfigRequestSN> req_seq_nbr_;
// The sender's (that's us) last assigned TSN, from the retransmission
// queue.
TSN sender_last_assigned_tsn_;
@@ -176,9 +176,9 @@
bool Validate(const ReConfigChunk& chunk);
// Processes a stream stream reconfiguration chunk and may either return
- // absl::nullopt (on protocol errors), or a list of responses - either 0, 1
+ // std::nullopt (on protocol errors), or a list of responses - either 0, 1
// or 2.
- absl::optional<std::vector<ReconfigurationResponseParameter>> Process(
+ std::optional<std::vector<ReconfigurationResponseParameter>> Process(
const ReConfigChunk& chunk);
// Creates the actual RE-CONFIG chunk. A request (which set `current_request`)
@@ -226,7 +226,7 @@
ReconfigRequestSN next_outgoing_req_seq_nbr_;
// The current stream request operation.
- absl::optional<CurrentRequest> current_request_;
+ std::optional<CurrentRequest> current_request_;
// For incoming requests - last processed request sequence number.
UnwrappedReconfigRequestSn last_processed_req_seq_nbr_;
diff --git a/net/dcsctp/socket/stream_reset_handler_test.cc b/net/dcsctp/socket/stream_reset_handler_test.cc
index a5a326d..480acbe 100644
--- a/net/dcsctp/socket/stream_reset_handler_test.cc
+++ b/net/dcsctp/socket/stream_reset_handler_test.cc
@@ -12,10 +12,10 @@
#include <array>
#include <cstdint>
#include <memory>
+#include <optional>
#include <type_traits>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/task_queue/task_queue_base.h"
#include "net/dcsctp/common/handover_testing.h"
@@ -131,7 +131,7 @@
void AdvanceTime(TimeDelta duration) {
callbacks_.AdvanceTime(duration);
for (;;) {
- absl::optional<TimeoutID> timeout_id = callbacks_.GetNextExpiredTimeout();
+ std::optional<TimeoutID> timeout_id = callbacks_.GetNextExpiredTimeout();
if (!timeout_id.has_value()) {
break;
}
@@ -152,7 +152,7 @@
}
std::vector<ReconfigurationResponseParameter> responses;
- absl::optional<SctpPacket> p = SctpPacket::Parse(payload, DcSctpOptions());
+ std::optional<SctpPacket> p = SctpPacket::Parse(payload, DcSctpOptions());
if (!p.has_value()) {
EXPECT_TRUE(false);
return {};
@@ -161,7 +161,7 @@
EXPECT_TRUE(false);
return {};
}
- absl::optional<ReConfigChunk> response_chunk =
+ std::optional<ReConfigChunk> response_chunk =
ReConfigChunk::Parse(p->descriptors()[0].data);
if (!response_chunk.has_value()) {
EXPECT_TRUE(false);
@@ -169,7 +169,7 @@
}
for (const auto& desc : response_chunk->parameters().descriptors()) {
if (desc.type == ReconfigurationResponseParameter::kType) {
- absl::optional<ReconfigurationResponseParameter> response =
+ std::optional<ReconfigurationResponseParameter> response =
ReconfigurationResponseParameter::Parse(desc.data);
if (!response.has_value()) {
EXPECT_TRUE(false);
@@ -482,7 +482,7 @@
EXPECT_CALL(producer_, GetStreamsReadyToBeReset())
.WillOnce(Return(std::vector<StreamID>({StreamID(42)})));
- absl::optional<ReConfigChunk> reconfig = handler_->MakeStreamResetRequest();
+ std::optional<ReConfigChunk> reconfig = handler_->MakeStreamResetRequest();
ASSERT_TRUE(reconfig.has_value());
ASSERT_HAS_VALUE_AND_ASSIGN(
OutgoingSSNResetRequestParameter req,
@@ -510,7 +510,7 @@
.WillOnce(Return(
std::vector<StreamID>({StreamID(40), StreamID(41), StreamID(42),
StreamID(43), StreamID(44)})));
- absl::optional<ReConfigChunk> reconfig = handler_->MakeStreamResetRequest();
+ std::optional<ReConfigChunk> reconfig = handler_->MakeStreamResetRequest();
ASSERT_TRUE(reconfig.has_value());
ASSERT_HAS_VALUE_AND_ASSIGN(
OutgoingSSNResetRequestParameter req,
@@ -546,7 +546,7 @@
EXPECT_CALL(producer_, GetStreamsReadyToBeReset())
.WillOnce(Return(std::vector<StreamID>({StreamID(42)})));
- absl::optional<ReConfigChunk> reconfig = handler_->MakeStreamResetRequest();
+ std::optional<ReConfigChunk> reconfig = handler_->MakeStreamResetRequest();
ASSERT_TRUE(reconfig.has_value());
ASSERT_HAS_VALUE_AND_ASSIGN(
OutgoingSSNResetRequestParameter req,
@@ -574,7 +574,7 @@
EXPECT_CALL(producer_, GetStreamsReadyToBeReset())
.WillOnce(Return(std::vector<StreamID>({StreamID(42)})));
- absl::optional<ReConfigChunk> reconfig = handler_->MakeStreamResetRequest();
+ std::optional<ReConfigChunk> reconfig = handler_->MakeStreamResetRequest();
ASSERT_TRUE(reconfig.has_value());
ASSERT_HAS_VALUE_AND_ASSIGN(
OutgoingSSNResetRequestParameter req,
@@ -604,7 +604,7 @@
EXPECT_CALL(producer_, GetStreamsReadyToBeReset())
.WillOnce(Return(std::vector<StreamID>({kStreamToReset})));
- absl::optional<ReConfigChunk> reconfig1 = handler_->MakeStreamResetRequest();
+ std::optional<ReConfigChunk> reconfig1 = handler_->MakeStreamResetRequest();
ASSERT_TRUE(reconfig1.has_value());
ASSERT_HAS_VALUE_AND_ASSIGN(
OutgoingSSNResetRequestParameter req1,
@@ -656,7 +656,7 @@
EXPECT_CALL(producer_, GetStreamsReadyToBeReset())
.WillOnce(Return(std::vector<StreamID>({StreamID(42)})));
- absl::optional<ReConfigChunk> reconfig1 = handler_->MakeStreamResetRequest();
+ std::optional<ReConfigChunk> reconfig1 = handler_->MakeStreamResetRequest();
ASSERT_TRUE(reconfig1.has_value());
ASSERT_HAS_VALUE_AND_ASSIGN(
OutgoingSSNResetRequestParameter req1,
@@ -671,7 +671,7 @@
EXPECT_CALL(producer_, PrepareResetStream(StreamID(43)));
StreamID stream_ids[] = {StreamID(41), StreamID(43)};
handler_->ResetStreams(stream_ids);
- EXPECT_EQ(handler_->MakeStreamResetRequest(), absl::nullopt);
+ EXPECT_EQ(handler_->MakeStreamResetRequest(), std::nullopt);
Parameters::Builder builder;
builder.Add(ReconfigurationResponseParameter(
@@ -691,7 +691,7 @@
EXPECT_CALL(producer_, GetStreamsReadyToBeReset())
.WillOnce(Return(std::vector<StreamID>({StreamID(41), StreamID(43)})));
- absl::optional<ReConfigChunk> reconfig2 = handler_->MakeStreamResetRequest();
+ std::optional<ReConfigChunk> reconfig2 = handler_->MakeStreamResetRequest();
ASSERT_TRUE(reconfig2.has_value());
ASSERT_HAS_VALUE_AND_ASSIGN(
OutgoingSSNResetRequestParameter req2,
@@ -811,7 +811,7 @@
EXPECT_CALL(producer_, GetStreamsReadyToBeReset())
.WillOnce(Return(std::vector<StreamID>({StreamID(42)})));
- absl::optional<ReConfigChunk> reconfig = handler_->MakeStreamResetRequest();
+ std::optional<ReConfigChunk> reconfig = handler_->MakeStreamResetRequest();
ASSERT_TRUE(reconfig.has_value());
ASSERT_HAS_VALUE_AND_ASSIGN(
OutgoingSSNResetRequestParameter req,
diff --git a/net/dcsctp/socket/transmission_control_block.cc b/net/dcsctp/socket/transmission_control_block.cc
index cf62c15..26ca6f6 100644
--- a/net/dcsctp/socket/transmission_control_block.cc
+++ b/net/dcsctp/socket/transmission_control_block.cc
@@ -12,11 +12,11 @@
#include <algorithm>
#include <cstdint>
#include <memory>
+#include <optional>
#include <string>
#include <utility>
#include <vector>
-#include "absl/types/optional.h"
#include "api/units/time_delta.h"
#include "net/dcsctp/packet/chunk/data_chunk.h"
#include "net/dcsctp/packet/chunk/forward_tsn_chunk.h"
@@ -66,7 +66,7 @@
absl::bind_front(&TransmissionControlBlock::OnRtxTimerExpiry, this),
TimerOptions(options.rto_initial.ToTimeDelta(),
TimerBackoffAlgorithm::kExponential,
- /*max_restarts=*/absl::nullopt,
+ /*max_restarts=*/std::nullopt,
options.max_timer_backoff_duration.has_value()
? options.max_timer_backoff_duration->ToTimeDelta()
: TimeDelta::PlusInfinity()))),
@@ -233,7 +233,7 @@
reassembly_queue_.remaining_bytes()));
}
MaybeSendForwardTsn(builder, now);
- absl::optional<ReConfigChunk> reconfig =
+ std::optional<ReConfigChunk> reconfig =
stream_reset_handler_.MakeStreamResetRequest();
if (reconfig.has_value()) {
builder.Add(*reconfig);
diff --git a/net/dcsctp/socket/transmission_control_block.h b/net/dcsctp/socket/transmission_control_block.h
index f8b2445..e4c5821 100644
--- a/net/dcsctp/socket/transmission_control_block.h
+++ b/net/dcsctp/socket/transmission_control_block.h
@@ -120,7 +120,7 @@
// Called when the COOKIE ACK chunk has been received, to allow further
// packets to be sent.
- void ClearCookieEchoChunk() { cookie_echo_chunk_ = absl::nullopt; }
+ void ClearCookieEchoChunk() { cookie_echo_chunk_ = std::nullopt; }
bool has_cookie_echo_chunk() const { return cookie_echo_chunk_.has_value(); }
@@ -187,7 +187,7 @@
// including a COOKIE ECHO). So if `cookie_echo_chunk_` is present, the
// SendBufferedChunks will always only just send one packet, with this chunk
// as the first chunk in the packet.
- absl::optional<CookieEchoChunk> cookie_echo_chunk_ = absl::nullopt;
+ std::optional<CookieEchoChunk> cookie_echo_chunk_ = std::nullopt;
};
} // namespace dcsctp
diff --git a/net/dcsctp/socket/transmission_control_block_test.cc b/net/dcsctp/socket/transmission_control_block_test.cc
index 6106fbb..472d572 100644
--- a/net/dcsctp/socket/transmission_control_block_test.cc
+++ b/net/dcsctp/socket/transmission_control_block_test.cc
@@ -12,10 +12,10 @@
#include <array>
#include <cstdint>
#include <memory>
+#include <optional>
#include <type_traits>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/task_queue/task_queue_base.h"
#include "net/dcsctp/common/handover_testing.h"
diff --git a/net/dcsctp/testing/BUILD.gn b/net/dcsctp/testing/BUILD.gn
index 573f32a..069d486 100644
--- a/net/dcsctp/testing/BUILD.gn
+++ b/net/dcsctp/testing/BUILD.gn
@@ -22,7 +22,6 @@
"../packet:data",
"../public:types",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
sources = [
"data_generator.cc",
diff --git a/net/dcsctp/testing/data_generator.h b/net/dcsctp/testing/data_generator.h
index 52f98dd..f885329 100644
--- a/net/dcsctp/testing/data_generator.h
+++ b/net/dcsctp/testing/data_generator.h
@@ -11,10 +11,10 @@
#define NET_DCSCTP_TESTING_DATA_GENERATOR_H_
#include <cstdint>
+#include <optional>
#include <vector>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "net/dcsctp/common/internal_types.h"
#include "net/dcsctp/packet/data.h"
@@ -23,7 +23,7 @@
struct DataGeneratorOptions {
StreamID stream_id = StreamID(1);
- absl::optional<MID> mid = absl::nullopt;
+ std::optional<MID> mid = std::nullopt;
PPID ppid = PPID(53);
};
diff --git a/net/dcsctp/testing/testing_macros.h b/net/dcsctp/testing/testing_macros.h
index 5cbdfff..dafa76f 100644
--- a/net/dcsctp/testing/testing_macros.h
+++ b/net/dcsctp/testing/testing_macros.h
@@ -17,7 +17,7 @@
#define DCSCTP_CONCAT_INNER_(x, y) x##y
#define DCSCTP_CONCAT_(x, y) DCSCTP_CONCAT_INNER_(x, y)
-// Similar to ASSERT_OK_AND_ASSIGN, this works with an absl::optional<> instead
+// Similar to ASSERT_OK_AND_ASSIGN, this works with an std::optional<> instead
// of an absl::StatusOr<>.
#define ASSERT_HAS_VALUE_AND_ASSIGN(lhs, rexpr) \
auto DCSCTP_CONCAT_(tmp_opt_val__, __LINE__) = rexpr; \
diff --git a/net/dcsctp/timer/BUILD.gn b/net/dcsctp/timer/BUILD.gn
index d123f64..5b2eb1e 100644
--- a/net/dcsctp/timer/BUILD.gn
+++ b/net/dcsctp/timer/BUILD.gn
@@ -22,7 +22,6 @@
"../public:types",
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
sources = [
"fake_timeout.h",
@@ -66,7 +65,6 @@
"../../../test:test_support",
"../../../test/time_controller:time_controller",
"../public:socket",
- "//third_party/abseil-cpp/absl/types:optional",
]
sources = [
"task_queue_timeout_test.cc",
diff --git a/net/dcsctp/timer/fake_timeout.h b/net/dcsctp/timer/fake_timeout.h
index cac4928..b94129b 100644
--- a/net/dcsctp/timer/fake_timeout.h
+++ b/net/dcsctp/timer/fake_timeout.h
@@ -14,10 +14,10 @@
#include <functional>
#include <limits>
#include <memory>
+#include <optional>
#include <utility>
#include <vector>
-#include "absl/types/optional.h"
#include "api/task_queue/task_queue_base.h"
#include "api/units/timestamp.h"
#include "net/dcsctp/public/timeout.h"
@@ -89,7 +89,7 @@
// still believes it's running, and it needs to be updated to set
// Timer::is_running_ to false before you operate on the Timer or Timeout
// again.
- absl::optional<TimeoutID> GetNextExpiredTimeout() {
+ std::optional<TimeoutID> GetNextExpiredTimeout() {
webrtc::Timestamp now = get_time_();
std::vector<TimeoutID> expired_timers;
for (auto& timer : timers_) {
@@ -97,7 +97,7 @@
return timer->timeout_id();
}
}
- return absl::nullopt;
+ return std::nullopt;
}
webrtc::TimeDelta GetTimeToNextTimeout() const {
diff --git a/net/dcsctp/timer/timer.h b/net/dcsctp/timer/timer.h
index 30b07f9..cfacdda 100644
--- a/net/dcsctp/timer/timer.h
+++ b/net/dcsctp/timer/timer.h
@@ -16,11 +16,11 @@
#include <functional>
#include <map>
#include <memory>
+#include <optional>
#include <string>
#include <utility>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/task_queue/task_queue_base.h"
#include "api/units/time_delta.h"
#include "net/dcsctp/public/timeout.h"
@@ -45,17 +45,17 @@
: TimerOptions(duration, TimerBackoffAlgorithm::kExponential) {}
TimerOptions(webrtc::TimeDelta duration,
TimerBackoffAlgorithm backoff_algorithm)
- : TimerOptions(duration, backoff_algorithm, absl::nullopt) {}
+ : TimerOptions(duration, backoff_algorithm, std::nullopt) {}
TimerOptions(webrtc::TimeDelta duration,
TimerBackoffAlgorithm backoff_algorithm,
- absl::optional<int> max_restarts)
+ std::optional<int> max_restarts)
: TimerOptions(duration,
backoff_algorithm,
max_restarts,
webrtc::TimeDelta::PlusInfinity()) {}
TimerOptions(webrtc::TimeDelta duration,
TimerBackoffAlgorithm backoff_algorithm,
- absl::optional<int> max_restarts,
+ std::optional<int> max_restarts,
webrtc::TimeDelta max_backoff_duration)
: TimerOptions(duration,
backoff_algorithm,
@@ -64,7 +64,7 @@
webrtc::TaskQueueBase::DelayPrecision::kLow) {}
TimerOptions(webrtc::TimeDelta duration,
TimerBackoffAlgorithm backoff_algorithm,
- absl::optional<int> max_restarts,
+ std::optional<int> max_restarts,
webrtc::TimeDelta max_backoff_duration,
webrtc::TaskQueueBase::DelayPrecision precision)
: duration(duration),
@@ -79,8 +79,8 @@
// restarted. If not set, the same duration will be used.
const TimerBackoffAlgorithm backoff_algorithm;
// The maximum number of times that the timer will be automatically restarted,
- // or absl::nullopt if there is no limit.
- const absl::optional<int> max_restarts;
+ // or std::nullopt if there is no limit.
+ const std::optional<int> max_restarts;
// The maximum timeout value for exponential backoff.
const webrtc::TimeDelta max_backoff_duration;
// The precision of the webrtc::TaskQueueBase used for scheduling.
diff --git a/net/dcsctp/timer/timer_test.cc b/net/dcsctp/timer/timer_test.cc
index 9a7c029..5d5091e 100644
--- a/net/dcsctp/timer/timer_test.cc
+++ b/net/dcsctp/timer/timer_test.cc
@@ -10,8 +10,8 @@
#include "net/dcsctp/timer/timer.h"
#include <memory>
+#include <optional>
-#include "absl/types/optional.h"
#include "api/task_queue/task_queue_base.h"
#include "api/units/time_delta.h"
#include "net/dcsctp/public/timeout.h"
@@ -39,7 +39,7 @@
now_ = now_ + duration;
for (;;) {
- absl::optional<TimeoutID> timeout_id =
+ std::optional<TimeoutID> timeout_id =
timeout_manager_.GetNextExpiredTimeout();
if (!timeout_id.has_value()) {
break;
@@ -396,7 +396,7 @@
std::unique_ptr<Timer> t1 = manager_.CreateTimer(
"t1", on_expired_.AsStdFunction(),
TimerOptions(TimeDelta::Seconds(1), TimerBackoffAlgorithm::kExponential,
- /*max_restarts=*/absl::nullopt, TimeDelta::Seconds(5)));
+ /*max_restarts=*/std::nullopt, TimeDelta::Seconds(5)));
t1->Start();
@@ -431,7 +431,7 @@
TEST(TimerManagerTest, TimerManagerPassesPrecisionToCreateTimeoutMethod) {
FakeTimeoutManager timeout_manager([&]() { return Timestamp::Zero(); });
- absl::optional<webrtc::TaskQueueBase::DelayPrecision> create_timer_precison;
+ std::optional<webrtc::TaskQueueBase::DelayPrecision> create_timer_precison;
TimerManager manager([&](webrtc::TaskQueueBase::DelayPrecision precision) {
create_timer_precison = precision;
return timeout_manager.CreateTimeout(precision);
@@ -445,7 +445,7 @@
manager.CreateTimer(
"test_timer", []() { return TimeDelta::Zero(); },
TimerOptions(TimeDelta::Millis(123), TimerBackoffAlgorithm::kExponential,
- absl::nullopt, TimeDelta::PlusInfinity(),
+ std::nullopt, TimeDelta::PlusInfinity(),
webrtc::TaskQueueBase::DelayPrecision::kHigh));
EXPECT_EQ(create_timer_precison,
webrtc::TaskQueueBase::DelayPrecision::kHigh);
@@ -453,7 +453,7 @@
manager.CreateTimer(
"test_timer", []() { return TimeDelta::Zero(); },
TimerOptions(TimeDelta::Millis(123), TimerBackoffAlgorithm::kExponential,
- absl::nullopt, TimeDelta::PlusInfinity(),
+ std::nullopt, TimeDelta::PlusInfinity(),
webrtc::TaskQueueBase::DelayPrecision::kLow));
EXPECT_EQ(create_timer_precison, webrtc::TaskQueueBase::DelayPrecision::kLow);
}
diff --git a/net/dcsctp/tx/BUILD.gn b/net/dcsctp/tx/BUILD.gn
index b28aa0e..0e21d91 100644
--- a/net/dcsctp/tx/BUILD.gn
+++ b/net/dcsctp/tx/BUILD.gn
@@ -17,7 +17,6 @@
"../packet:data",
"../public:socket",
"../public:types",
- "//third_party/abseil-cpp/absl/types:optional",
]
sources = [ "send_queue.h" ]
}
@@ -37,7 +36,6 @@
"../public:types",
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
sources = [
"rr_send_queue.cc",
@@ -62,7 +60,6 @@
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
sources = [
"stream_scheduler.cc",
@@ -114,7 +111,6 @@
"../public:socket",
"../public:types",
"../timer",
- "//third_party/abseil-cpp/absl/types:optional",
]
sources = [
"outstanding_data.cc",
@@ -140,7 +136,6 @@
"../timer",
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
sources = [
"retransmission_queue.cc",
@@ -156,7 +151,6 @@
"../../../api:array_view",
"../../../api/units:timestamp",
"../../../test:test_support",
- "//third_party/abseil-cpp/absl/types:optional",
]
sources = [ "mock_send_queue.h" ]
}
@@ -192,7 +186,6 @@
"../testing:data_generator",
"../testing:testing_macros",
"../timer",
- "//third_party/abseil-cpp/absl/types:optional",
]
sources = [
"outstanding_data_test.cc",
diff --git a/net/dcsctp/tx/mock_send_queue.h b/net/dcsctp/tx/mock_send_queue.h
index 3511403..e31c99a 100644
--- a/net/dcsctp/tx/mock_send_queue.h
+++ b/net/dcsctp/tx/mock_send_queue.h
@@ -11,9 +11,9 @@
#define NET_DCSCTP_TX_MOCK_SEND_QUEUE_H_
#include <cstdint>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/units/timestamp.h"
#include "net/dcsctp/tx/send_queue.h"
@@ -26,11 +26,11 @@
MockSendQueue() {
ON_CALL(*this, Produce)
.WillByDefault([](webrtc::Timestamp now, size_t max_size) {
- return absl::nullopt;
+ return std::nullopt;
});
}
- MOCK_METHOD(absl::optional<SendQueue::DataToSend>,
+ MOCK_METHOD(std::optional<SendQueue::DataToSend>,
Produce,
(webrtc::Timestamp now, size_t max_size),
(override));
diff --git a/net/dcsctp/tx/outstanding_data.cc b/net/dcsctp/tx/outstanding_data.cc
index ca639ab..b42c77e 100644
--- a/net/dcsctp/tx/outstanding_data.cc
+++ b/net/dcsctp/tx/outstanding_data.cc
@@ -412,7 +412,7 @@
outstanding_data_.size());
}
-absl::optional<UnwrappedTSN> OutstandingData::Insert(
+std::optional<UnwrappedTSN> OutstandingData::Insert(
OutgoingMessageId message_id,
const Data& data,
Timestamp time_sent,
@@ -436,7 +436,7 @@
<< " as expired";
AbandonAllFor(item);
RTC_DCHECK(IsConsistent());
- return absl::nullopt;
+ return std::nullopt;
}
RTC_DCHECK(IsConsistent());
diff --git a/net/dcsctp/tx/outstanding_data.h b/net/dcsctp/tx/outstanding_data.h
index 2a21497..018b02f 100644
--- a/net/dcsctp/tx/outstanding_data.h
+++ b/net/dcsctp/tx/outstanding_data.h
@@ -12,11 +12,11 @@
#include <deque>
#include <map>
+#include <optional>
#include <set>
#include <utility>
#include <vector>
-#include "absl/types/optional.h"
#include "api/units/timestamp.h"
#include "net/dcsctp/common/internal_types.h"
#include "net/dcsctp/common/sequence_numbers.h"
@@ -132,8 +132,8 @@
// Schedules `data` to be sent, with the provided partial reliability
// parameters. Returns the TSN if the item was actually added and scheduled to
- // be sent, and absl::nullopt if it shouldn't be sent.
- absl::optional<UnwrappedTSN> Insert(
+ // be sent, and std::nullopt if it shouldn't be sent.
+ std::optional<UnwrappedTSN> Insert(
OutgoingMessageId message_id,
const Data& data,
webrtc::Timestamp time_sent,
diff --git a/net/dcsctp/tx/outstanding_data_test.cc b/net/dcsctp/tx/outstanding_data_test.cc
index e4bdb7c..6048dbe 100644
--- a/net/dcsctp/tx/outstanding_data_test.cc
+++ b/net/dcsctp/tx/outstanding_data_test.cc
@@ -9,9 +9,9 @@
*/
#include "net/dcsctp/tx/outstanding_data.h"
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "net/dcsctp/common/internal_types.h"
#include "net/dcsctp/common/math.h"
#include "net/dcsctp/common/sequence_numbers.h"
diff --git a/net/dcsctp/tx/retransmission_error_counter.h b/net/dcsctp/tx/retransmission_error_counter.h
index 7078c78..589e66b 100644
--- a/net/dcsctp/tx/retransmission_error_counter.h
+++ b/net/dcsctp/tx/retransmission_error_counter.h
@@ -42,7 +42,7 @@
private:
const absl::string_view log_prefix_;
- const absl::optional<int> limit_;
+ const std::optional<int> limit_;
int counter_ = 0;
};
} // namespace dcsctp
diff --git a/net/dcsctp/tx/retransmission_error_counter_test.cc b/net/dcsctp/tx/retransmission_error_counter_test.cc
index 67bbc0b..dc53740 100644
--- a/net/dcsctp/tx/retransmission_error_counter_test.cc
+++ b/net/dcsctp/tx/retransmission_error_counter_test.cc
@@ -74,7 +74,7 @@
TEST(RetransmissionErrorCounterTest, CanBeLimitless) {
DcSctpOptions options;
- options.max_retransmissions = absl::nullopt;
+ options.max_retransmissions = std::nullopt;
RetransmissionErrorCounter counter("log: ", options);
for (int i = 0; i < 100; ++i) {
EXPECT_TRUE(counter.Increment("test"));
diff --git a/net/dcsctp/tx/retransmission_queue.cc b/net/dcsctp/tx/retransmission_queue.cc
index 9a24617..604139c 100644
--- a/net/dcsctp/tx/retransmission_queue.cc
+++ b/net/dcsctp/tx/retransmission_queue.cc
@@ -14,6 +14,7 @@
#include <functional>
#include <iterator>
#include <map>
+#include <optional>
#include <set>
#include <string>
#include <utility>
@@ -21,7 +22,6 @@
#include "absl/algorithm/container.h"
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "net/dcsctp/common/math.h"
#include "net/dcsctp/common/sequence_numbers.h"
@@ -106,7 +106,7 @@
RTC_DLOG(LS_VERBOSE) << log_prefix_
<< "exit_point=" << *fast_recovery_exit_tsn_->Wrap()
<< " reached - exiting fast recovery";
- fast_recovery_exit_tsn_ = absl::nullopt;
+ fast_recovery_exit_tsn_ = std::nullopt;
}
}
@@ -475,7 +475,7 @@
while (max_bytes > data_chunk_header_size_) {
RTC_DCHECK(IsDivisibleBy4(max_bytes));
- absl::optional<SendQueue::DataToSend> chunk_opt =
+ std::optional<SendQueue::DataToSend> chunk_opt =
send_queue_.Produce(now, max_bytes - data_chunk_header_size_);
if (!chunk_opt.has_value()) {
break;
@@ -485,7 +485,7 @@
max_bytes -= chunk_size;
rwnd_ -= chunk_size;
- absl::optional<UnwrappedTSN> tsn = outstanding_data_.Insert(
+ std::optional<UnwrappedTSN> tsn = outstanding_data_.Insert(
chunk_opt->message_id, chunk_opt->data, now,
partial_reliability_ ? chunk_opt->max_retransmissions
: MaxRetransmits::NoLimit(),
diff --git a/net/dcsctp/tx/retransmission_queue.h b/net/dcsctp/tx/retransmission_queue.h
index 1572064..5767c88 100644
--- a/net/dcsctp/tx/retransmission_queue.h
+++ b/net/dcsctp/tx/retransmission_queue.h
@@ -13,13 +13,13 @@
#include <cstdint>
#include <functional>
#include <map>
+#include <optional>
#include <set>
#include <string>
#include <utility>
#include <vector>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "net/dcsctp/common/sequence_numbers.h"
#include "net/dcsctp/packet/chunk/forward_tsn_chunk.h"
@@ -243,7 +243,7 @@
// If set, fast recovery is enabled until this TSN has been cumulative
// acked.
- absl::optional<UnwrappedTSN> fast_recovery_exit_tsn_ = absl::nullopt;
+ std::optional<UnwrappedTSN> fast_recovery_exit_tsn_ = std::nullopt;
// The send queue.
SendQueue& send_queue_;
diff --git a/net/dcsctp/tx/retransmission_queue_test.cc b/net/dcsctp/tx/retransmission_queue_test.cc
index 4c802e4..cc68a9b 100644
--- a/net/dcsctp/tx/retransmission_queue_test.cc
+++ b/net/dcsctp/tx/retransmission_queue_test.cc
@@ -13,10 +13,10 @@
#include <cstdint>
#include <functional>
#include <memory>
+#include <optional>
#include <utility>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/task_queue/task_queue_base.h"
#include "net/dcsctp/common/handover_testing.h"
@@ -148,7 +148,7 @@
RetransmissionQueue queue = CreateQueue();
EXPECT_CALL(producer_, Produce)
.WillOnce(CreateChunk(OutgoingMessageId(0)))
- .WillRepeatedly([](Timestamp, size_t) { return absl::nullopt; });
+ .WillRepeatedly([](Timestamp, size_t) { return std::nullopt; });
EXPECT_THAT(GetSentPacketTSNs(queue), testing::ElementsAre(TSN(10)));
@@ -161,7 +161,7 @@
RetransmissionQueue queue = CreateQueue();
EXPECT_CALL(producer_, Produce)
.WillOnce(CreateChunk(OutgoingMessageId(0)))
- .WillRepeatedly([](Timestamp, size_t) { return absl::nullopt; });
+ .WillRepeatedly([](Timestamp, size_t) { return std::nullopt; });
EXPECT_THAT(GetSentPacketTSNs(queue), testing::ElementsAre(TSN(10)));
@@ -177,7 +177,7 @@
.WillOnce(CreateChunk(OutgoingMessageId(0)))
.WillOnce(CreateChunk(OutgoingMessageId(1)))
.WillOnce(CreateChunk(OutgoingMessageId(2)))
- .WillRepeatedly([](Timestamp, size_t) { return absl::nullopt; });
+ .WillRepeatedly([](Timestamp, size_t) { return std::nullopt; });
EXPECT_THAT(GetSentPacketTSNs(queue),
testing::ElementsAre(TSN(10), TSN(11), TSN(12)));
@@ -200,7 +200,7 @@
.WillOnce(CreateChunk(OutgoingMessageId(5)))
.WillOnce(CreateChunk(OutgoingMessageId(6)))
.WillOnce(CreateChunk(OutgoingMessageId(7)))
- .WillRepeatedly([](Timestamp, size_t) { return absl::nullopt; });
+ .WillRepeatedly([](Timestamp, size_t) { return std::nullopt; });
EXPECT_THAT(GetSentPacketTSNs(queue),
testing::ElementsAre(TSN(10), TSN(11), TSN(12), TSN(13), TSN(14),
@@ -231,7 +231,7 @@
.WillOnce(CreateChunk(OutgoingMessageId(5)))
.WillOnce(CreateChunk(OutgoingMessageId(6)))
.WillOnce(CreateChunk(OutgoingMessageId(7)))
- .WillRepeatedly([](Timestamp, size_t) { return absl::nullopt; });
+ .WillRepeatedly([](Timestamp, size_t) { return std::nullopt; });
EXPECT_THAT(GetSentPacketTSNs(queue),
testing::ElementsAre(TSN(10), TSN(11), TSN(12), TSN(13), TSN(14),
@@ -243,7 +243,7 @@
// Send 18
EXPECT_CALL(producer_, Produce)
.WillOnce(CreateChunk(OutgoingMessageId(8)))
- .WillRepeatedly([](Timestamp, size_t) { return absl::nullopt; });
+ .WillRepeatedly([](Timestamp, size_t) { return std::nullopt; });
EXPECT_THAT(GetSentPacketTSNs(queue), testing::ElementsAre(TSN(18)));
// Ack 12, 14-15, 17-18
@@ -264,7 +264,7 @@
// Send 19
EXPECT_CALL(producer_, Produce)
.WillOnce(CreateChunk(OutgoingMessageId(9)))
- .WillRepeatedly([](Timestamp, size_t) { return absl::nullopt; });
+ .WillRepeatedly([](Timestamp, size_t) { return std::nullopt; });
EXPECT_THAT(GetSentPacketTSNs(queue), testing::ElementsAre(TSN(19)));
// Ack 12, 14-15, 17-19
@@ -276,7 +276,7 @@
// Send 20
EXPECT_CALL(producer_, Produce)
.WillOnce(CreateChunk(OutgoingMessageId(10)))
- .WillRepeatedly([](Timestamp, size_t) { return absl::nullopt; });
+ .WillRepeatedly([](Timestamp, size_t) { return std::nullopt; });
EXPECT_THAT(GetSentPacketTSNs(queue), testing::ElementsAre(TSN(20)));
// Ack 12, 14-15, 17-20
@@ -323,7 +323,7 @@
.WillOnce(CreateChunk(OutgoingMessageId(0)))
.WillOnce(CreateChunk(OutgoingMessageId(1)))
.WillOnce(CreateChunk(OutgoingMessageId(2)))
- .WillRepeatedly([](Timestamp, size_t) { return absl::nullopt; });
+ .WillRepeatedly([](Timestamp, size_t) { return std::nullopt; });
static constexpr Timestamp kStartTime = Timestamp::Seconds(100);
now_ = kStartTime;
@@ -344,7 +344,7 @@
// Send 13
EXPECT_CALL(producer_, Produce)
.WillOnce(CreateChunk(OutgoingMessageId(3)))
- .WillRepeatedly([](Timestamp, size_t) { return absl::nullopt; });
+ .WillRepeatedly([](Timestamp, size_t) { return std::nullopt; });
EXPECT_THAT(GetSentPacketTSNs(queue), testing::ElementsAre(TSN(13)));
// Ack 10, 12-13, after 100ms.
@@ -355,7 +355,7 @@
// Send 14
EXPECT_CALL(producer_, Produce)
.WillOnce(CreateChunk(OutgoingMessageId(4)))
- .WillRepeatedly([](Timestamp, size_t) { return absl::nullopt; });
+ .WillRepeatedly([](Timestamp, size_t) { return std::nullopt; });
EXPECT_THAT(GetSentPacketTSNs(queue), testing::ElementsAre(TSN(14)));
// Ack 10, 12-14, after 100 ms.
@@ -407,7 +407,7 @@
return SendQueue::DataToSend(OutgoingMessageId(1),
gen_.Ordered({1, 2, 3, 4}, "BE"));
})
- .WillRepeatedly([](Timestamp, size_t) { return absl::nullopt; });
+ .WillRepeatedly([](Timestamp, size_t) { return std::nullopt; });
std::vector<std::pair<TSN, Data>> chunks_to_send =
queue.GetChunksToSend(now_, 1000);
@@ -426,7 +426,7 @@
return SendQueue::DataToSend(OutgoingMessageId(0),
gen_.Ordered({1, 2, 3, 4}, "BE"));
})
- .WillRepeatedly([](Timestamp, size_t) { return absl::nullopt; });
+ .WillRepeatedly([](Timestamp, size_t) { return std::nullopt; });
EXPECT_FALSE(queue.ShouldSendForwardTsn(now_));
std::vector<std::pair<TSN, Data>> chunks_to_send =
@@ -466,7 +466,7 @@
dts.max_retransmissions = MaxRetransmits(0);
return dts;
})
- .WillRepeatedly([](Timestamp, size_t) { return absl::nullopt; });
+ .WillRepeatedly([](Timestamp, size_t) { return std::nullopt; });
EXPECT_FALSE(queue.ShouldSendForwardTsn(now_));
std::vector<std::pair<TSN, Data>> chunks_to_send =
@@ -495,7 +495,7 @@
dts.max_retransmissions = MaxRetransmits(0);
return dts;
})
- .WillRepeatedly([](Timestamp, size_t) { return absl::nullopt; });
+ .WillRepeatedly([](Timestamp, size_t) { return std::nullopt; });
EXPECT_FALSE(queue.ShouldSendForwardTsn(now_));
std::vector<std::pair<TSN, Data>> chunks_to_send =
@@ -536,7 +536,7 @@
dts.max_retransmissions = MaxRetransmits(3);
return dts;
})
- .WillRepeatedly([](Timestamp, size_t) { return absl::nullopt; });
+ .WillRepeatedly([](Timestamp, size_t) { return std::nullopt; });
EXPECT_FALSE(queue.ShouldSendForwardTsn(now_));
std::vector<std::pair<TSN, Data>> chunks_to_send =
@@ -588,7 +588,7 @@
return SendQueue::DataToSend(OutgoingMessageId(0),
gen_.Ordered(payload, "BE"));
})
- .WillRepeatedly([](Timestamp, size_t) { return absl::nullopt; });
+ .WillRepeatedly([](Timestamp, size_t) { return std::nullopt; });
std::vector<std::pair<TSN, Data>> chunks_to_send =
queue.GetChunksToSend(now_, 1500);
@@ -637,7 +637,7 @@
dts.max_retransmissions = MaxRetransmits(0);
return dts;
})
- .WillRepeatedly([](Timestamp, size_t) { return absl::nullopt; });
+ .WillRepeatedly([](Timestamp, size_t) { return std::nullopt; });
// Send and ack first chunk (TSN 10)
std::vector<std::pair<TSN, Data>> chunks_to_send =
@@ -693,7 +693,7 @@
dts.max_retransmissions = MaxRetransmits(0);
return dts;
})
- .WillRepeatedly([](Timestamp, size_t) { return absl::nullopt; });
+ .WillRepeatedly([](Timestamp, size_t) { return std::nullopt; });
// Send and ack first chunk (TSN 10)
std::vector<std::pair<TSN, Data>> chunks_to_send =
@@ -763,7 +763,7 @@
dts.max_retransmissions = MaxRetransmits(0);
return dts;
})
- .WillRepeatedly([](Timestamp, size_t) { return absl::nullopt; });
+ .WillRepeatedly([](Timestamp, size_t) { return std::nullopt; });
std::vector<std::pair<TSN, Data>> chunks_to_send =
queue.GetChunksToSend(now_, 1000);
@@ -858,7 +858,7 @@
dts.max_retransmissions = MaxRetransmits(0);
return dts;
})
- .WillRepeatedly([](Timestamp, size_t) { return absl::nullopt; });
+ .WillRepeatedly([](Timestamp, size_t) { return std::nullopt; });
std::vector<std::pair<TSN, Data>> chunks_to_send =
queue.GetChunksToSend(now_, 1000);
@@ -890,7 +890,7 @@
.WillOnce(CreateChunk(OutgoingMessageId(5)))
.WillOnce(CreateChunk(OutgoingMessageId(6)))
.WillOnce(CreateChunk(OutgoingMessageId(7)))
- .WillRepeatedly([](Timestamp, size_t) { return absl::nullopt; });
+ .WillRepeatedly([](Timestamp, size_t) { return std::nullopt; });
EXPECT_THAT(GetSentPacketTSNs(queue),
testing::ElementsAre(TSN(10), TSN(11), TSN(12), TSN(13), TSN(14),
@@ -920,7 +920,7 @@
.WillOnce(CreateChunk(OutgoingMessageId(5)))
.WillOnce(CreateChunk(OutgoingMessageId(6)))
.WillOnce(CreateChunk(OutgoingMessageId(7)))
- .WillRepeatedly([](Timestamp, size_t) { return absl::nullopt; });
+ .WillRepeatedly([](Timestamp, size_t) { return std::nullopt; });
EXPECT_THAT(GetSentPacketTSNs(queue),
testing::ElementsAre(TSN(10), TSN(11), TSN(12), TSN(13), TSN(14),
@@ -967,7 +967,7 @@
.WillOnce(CreateChunk(OutgoingMessageId(5)))
.WillOnce(CreateChunk(OutgoingMessageId(6)))
.WillOnce(CreateChunk(OutgoingMessageId(7)))
- .WillRepeatedly([](Timestamp, size_t) { return absl::nullopt; });
+ .WillRepeatedly([](Timestamp, size_t) { return std::nullopt; });
EXPECT_THAT(GetSentPacketTSNs(queue),
testing::ElementsAre(TSN(10), TSN(11), TSN(12), TSN(13), TSN(14),
@@ -1036,7 +1036,7 @@
dts.max_retransmissions = MaxRetransmits(0);
return dts;
})
- .WillRepeatedly([](Timestamp, size_t) { return absl::nullopt; });
+ .WillRepeatedly([](Timestamp, size_t) { return std::nullopt; });
// Send and ack first chunk (TSN 10)
std::vector<std::pair<TSN, Data>> chunks_to_send =
@@ -1098,7 +1098,7 @@
dts.expires_at = Timestamp(test_start + TimeDelta::Millis(10));
return dts;
})
- .WillRepeatedly([](Timestamp, size_t) { return absl::nullopt; });
+ .WillRepeatedly([](Timestamp, size_t) { return std::nullopt; });
std::vector<std::pair<TSN, Data>> chunks_to_send =
queue.GetChunksToSend(now_, 24);
@@ -1151,7 +1151,7 @@
dts.expires_at = Timestamp(test_start + TimeDelta::Millis(10));
return dts;
})
- .WillRepeatedly([](Timestamp, size_t) { return absl::nullopt; });
+ .WillRepeatedly([](Timestamp, size_t) { return std::nullopt; });
EXPECT_CALL(producer_, Discard(StreamID(1), OutgoingMessageId(44)))
.WillOnce(Return(true));
@@ -1186,7 +1186,7 @@
.WillOnce(CreateChunk(OutgoingMessageId(0)))
.WillOnce(CreateChunk(OutgoingMessageId(1)))
.WillOnce(CreateChunk(OutgoingMessageId(2)))
- .WillRepeatedly([](Timestamp, size_t) { return absl::nullopt; });
+ .WillRepeatedly([](Timestamp, size_t) { return std::nullopt; });
EXPECT_FALSE(queue.ShouldSendForwardTsn(now_));
@@ -1262,7 +1262,7 @@
.WillOnce(CreateChunk(OutgoingMessageId(6)))
.WillOnce(CreateChunk(OutgoingMessageId(7)))
.WillOnce(CreateChunk(OutgoingMessageId(8)))
- .WillRepeatedly([](Timestamp, size_t) { return absl::nullopt; });
+ .WillRepeatedly([](Timestamp, size_t) { return std::nullopt; });
EXPECT_FALSE(queue.ShouldSendForwardTsn(now_));
@@ -1392,7 +1392,7 @@
return SendQueue::DataToSend(OutgoingMessageId(0),
gen_.Ordered(payload, "BE"));
})
- .WillRepeatedly([](Timestamp, size_t) { return absl::nullopt; });
+ .WillRepeatedly([](Timestamp, size_t) { return std::nullopt; });
std::vector<std::pair<TSN, Data>> chunks_to_send =
queue.GetChunksToSend(now_, 1500);
@@ -1409,7 +1409,7 @@
RetransmissionQueue queue = CreateQueue();
EXPECT_CALL(producer_, Produce)
.WillOnce(CreateChunk(OutgoingMessageId(0)))
- .WillRepeatedly([](Timestamp, size_t) { return absl::nullopt; });
+ .WillRepeatedly([](Timestamp, size_t) { return std::nullopt; });
EXPECT_THAT(GetSentPacketTSNs(queue), SizeIs(1));
EXPECT_EQ(
@@ -1432,7 +1432,7 @@
.WillOnce(CreateChunk(OutgoingMessageId(5)))
.WillOnce(CreateChunk(OutgoingMessageId(6)))
.WillOnce(CreateChunk(OutgoingMessageId(7)))
- .WillRepeatedly([](Timestamp, size_t) { return absl::nullopt; });
+ .WillRepeatedly([](Timestamp, size_t) { return std::nullopt; });
EXPECT_THAT(GetSentPacketTSNs(queue), SizeIs(8));
EXPECT_EQ(
queue.GetHandoverReadiness(),
@@ -1445,7 +1445,7 @@
// Send 18
EXPECT_CALL(producer_, Produce)
.WillOnce(CreateChunk(OutgoingMessageId(8)))
- .WillRepeatedly([](Timestamp, size_t) { return absl::nullopt; });
+ .WillRepeatedly([](Timestamp, size_t) { return std::nullopt; });
EXPECT_THAT(GetSentPacketTSNs(queue), SizeIs(1));
// Ack 12, 14-15, 17-18
@@ -1457,7 +1457,7 @@
// Send 19
EXPECT_CALL(producer_, Produce)
.WillOnce(CreateChunk(OutgoingMessageId(9)))
- .WillRepeatedly([](Timestamp, size_t) { return absl::nullopt; });
+ .WillRepeatedly([](Timestamp, size_t) { return std::nullopt; });
EXPECT_THAT(GetSentPacketTSNs(queue), SizeIs(1));
// Ack 12, 14-15, 17-19
@@ -1469,7 +1469,7 @@
// Send 20
EXPECT_CALL(producer_, Produce)
.WillOnce(CreateChunk(OutgoingMessageId(10)))
- .WillRepeatedly([](Timestamp, size_t) { return absl::nullopt; });
+ .WillRepeatedly([](Timestamp, size_t) { return std::nullopt; });
EXPECT_THAT(GetSentPacketTSNs(queue), SizeIs(1));
// Ack 12, 14-15, 17-20
@@ -1505,7 +1505,7 @@
EXPECT_CALL(producer_, Produce)
.WillOnce(CreateChunk(OutgoingMessageId(0)))
.WillOnce(CreateChunk(OutgoingMessageId(1)))
- .WillRepeatedly([](Timestamp, size_t) { return absl::nullopt; });
+ .WillRepeatedly([](Timestamp, size_t) { return std::nullopt; });
EXPECT_THAT(GetSentPacketTSNs(queue), SizeIs(2));
queue.HandleSack(now_, SackChunk(TSN(11), kArwnd, {}, {}));
@@ -1516,7 +1516,7 @@
.WillOnce(CreateChunk(OutgoingMessageId(2)))
.WillOnce(CreateChunk(OutgoingMessageId(3)))
.WillOnce(CreateChunk(OutgoingMessageId(4)))
- .WillRepeatedly([](Timestamp, size_t) { return absl::nullopt; });
+ .WillRepeatedly([](Timestamp, size_t) { return std::nullopt; });
EXPECT_THAT(GetSentPacketTSNs(*handedover_queue),
testing::ElementsAre(TSN(12), TSN(13), TSN(14)));
@@ -1554,7 +1554,7 @@
return SendQueue::DataToSend(OutgoingMessageId(0),
gen_.Ordered(payload, "E"));
})
- .WillRepeatedly([](Timestamp, size_t) { return absl::nullopt; });
+ .WillRepeatedly([](Timestamp, size_t) { return std::nullopt; });
// Produce all chunks and put them in the retransmission queue.
std::vector<std::pair<TSN, Data>> chunks_to_send =
diff --git a/net/dcsctp/tx/rr_send_queue.cc b/net/dcsctp/tx/rr_send_queue.cc
index 2193880..2016ede 100644
--- a/net/dcsctp/tx/rr_send_queue.cc
+++ b/net/dcsctp/tx/rr_send_queue.cc
@@ -13,12 +13,12 @@
#include <deque>
#include <limits>
#include <map>
+#include <optional>
#include <set>
#include <utility>
#include <vector>
#include "absl/algorithm/container.h"
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "net/dcsctp/common/internal_types.h"
#include "net/dcsctp/packet/data.h"
@@ -136,7 +136,7 @@
RTC_DCHECK(IsConsistent());
}
-absl::optional<SendQueue::DataToSend> RRSendQueue::OutgoingStream::Produce(
+std::optional<SendQueue::DataToSend> RRSendQueue::OutgoingStream::Produce(
Timestamp now,
size_t max_size) {
RTC_DCHECK(pause_state_ != PauseState::kPaused &&
@@ -218,7 +218,7 @@
return chunk;
}
RTC_DCHECK(IsConsistent());
- return absl::nullopt;
+ return std::nullopt;
}
void RRSendQueue::OutgoingStream::HandleMessageExpired(
@@ -331,8 +331,8 @@
item.remaining_size);
item.remaining_offset = 0;
item.remaining_size = item.message.payload().size();
- item.mid = absl::nullopt;
- item.ssn = absl::nullopt;
+ item.mid = std::nullopt;
+ item.ssn = std::nullopt;
item.current_fsn = FSN(0);
if (old_pause_state == PauseState::kPaused ||
old_pause_state == PauseState::kResetting) {
@@ -381,8 +381,8 @@
return total_buffered_amount() == 0;
}
-absl::optional<SendQueue::DataToSend> RRSendQueue::Produce(Timestamp now,
- size_t max_size) {
+std::optional<SendQueue::DataToSend> RRSendQueue::Produce(Timestamp now,
+ size_t max_size) {
return scheduler_.Produce(now, max_size);
}
diff --git a/net/dcsctp/tx/rr_send_queue.h b/net/dcsctp/tx/rr_send_queue.h
index 1a370a2..19c30e1 100644
--- a/net/dcsctp/tx/rr_send_queue.h
+++ b/net/dcsctp/tx/rr_send_queue.h
@@ -14,13 +14,13 @@
#include <deque>
#include <map>
#include <memory>
+#include <optional>
#include <string>
#include <utility>
#include <vector>
#include "absl/algorithm/container.h"
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "net/dcsctp/common/internal_types.h"
#include "net/dcsctp/public/dcsctp_message.h"
@@ -75,8 +75,8 @@
const SendOptions& send_options = {});
// Implementation of `SendQueue`.
- absl::optional<DataToSend> Produce(webrtc::Timestamp now,
- size_t max_size) override;
+ std::optional<DataToSend> Produce(webrtc::Timestamp now,
+ size_t max_size) override;
bool Discard(StreamID stream_id, OutgoingMessageId message_id) override;
void PrepareResetStream(StreamID streams) override;
bool HasStreamsReadyToBeReset() const override;
@@ -154,8 +154,8 @@
void Add(DcSctpMessage message, MessageAttributes attributes);
// Implementing `StreamScheduler::StreamProducer`.
- absl::optional<SendQueue::DataToSend> Produce(webrtc::Timestamp now,
- size_t max_size) override;
+ std::optional<SendQueue::DataToSend> Produce(webrtc::Timestamp now,
+ size_t max_size) override;
size_t bytes_to_send_in_next_message() const override;
const ThresholdWatcher& buffered_amount() const { return buffered_amount_; }
@@ -235,8 +235,8 @@
size_t remaining_size;
// If set, an allocated Message ID and SSN. Will be allocated when the
// first fragment is sent.
- absl::optional<MID> mid = absl::nullopt;
- absl::optional<SSN> ssn = absl::nullopt;
+ std::optional<MID> mid = std::nullopt;
+ std::optional<SSN> ssn = std::nullopt;
// The current Fragment Sequence Number, incremented for each fragment.
FSN current_fsn = FSN(0);
};
@@ -263,7 +263,7 @@
bool IsConsistent() const;
OutgoingStream& GetOrCreateStreamInfo(StreamID stream_id);
- absl::optional<DataToSend> Produce(
+ std::optional<DataToSend> Produce(
std::map<StreamID, OutgoingStream>::iterator it,
webrtc::Timestamp now,
size_t max_size);
diff --git a/net/dcsctp/tx/rr_send_queue_test.cc b/net/dcsctp/tx/rr_send_queue_test.cc
index 9beba95..04f718c 100644
--- a/net/dcsctp/tx/rr_send_queue_test.cc
+++ b/net/dcsctp/tx/rr_send_queue_test.cc
@@ -65,7 +65,7 @@
buf_.Add(kNow, DcSctpMessage(kStreamID, kPPID, {1, 2, 4, 5, 6}));
EXPECT_FALSE(buf_.IsEmpty());
- absl::optional<SendQueue::DataToSend> chunk_opt =
+ std::optional<SendQueue::DataToSend> chunk_opt =
buf_.Produce(kNow, kOneFragmentPacketSize);
ASSERT_TRUE(chunk_opt.has_value());
EXPECT_TRUE(chunk_opt->data.is_beginning);
@@ -76,19 +76,19 @@
std::vector<uint8_t> payload(60);
buf_.Add(kNow, DcSctpMessage(kStreamID, kPPID, payload));
- absl::optional<SendQueue::DataToSend> chunk_beg =
+ std::optional<SendQueue::DataToSend> chunk_beg =
buf_.Produce(kNow, /*max_size=*/20);
ASSERT_TRUE(chunk_beg.has_value());
EXPECT_TRUE(chunk_beg->data.is_beginning);
EXPECT_FALSE(chunk_beg->data.is_end);
- absl::optional<SendQueue::DataToSend> chunk_mid =
+ std::optional<SendQueue::DataToSend> chunk_mid =
buf_.Produce(kNow, /*max_size=*/20);
ASSERT_TRUE(chunk_mid.has_value());
EXPECT_FALSE(chunk_mid->data.is_beginning);
EXPECT_FALSE(chunk_mid->data.is_end);
- absl::optional<SendQueue::DataToSend> chunk_end =
+ std::optional<SendQueue::DataToSend> chunk_end =
buf_.Produce(kNow, /*max_size=*/20);
ASSERT_TRUE(chunk_end.has_value());
EXPECT_FALSE(chunk_end->data.is_beginning);
@@ -102,7 +102,7 @@
buf_.Add(kNow, DcSctpMessage(kStreamID, kPPID, payload));
buf_.Add(kNow, DcSctpMessage(StreamID(3), PPID(54), payload));
- absl::optional<SendQueue::DataToSend> chunk_one =
+ std::optional<SendQueue::DataToSend> chunk_one =
buf_.Produce(kNow, kOneFragmentPacketSize);
ASSERT_TRUE(chunk_one.has_value());
EXPECT_EQ(chunk_one->data.stream_id, kStreamID);
@@ -110,7 +110,7 @@
EXPECT_TRUE(chunk_one->data.is_beginning);
EXPECT_TRUE(chunk_one->data.is_end);
- absl::optional<SendQueue::DataToSend> chunk_two =
+ std::optional<SendQueue::DataToSend> chunk_two =
buf_.Produce(kNow, kOneFragmentPacketSize);
ASSERT_TRUE(chunk_two.has_value());
EXPECT_EQ(chunk_two->data.stream_id, StreamID(3));
@@ -132,14 +132,14 @@
buf_.Add(kNow, DcSctpMessage(StreamID(5), PPID(55), payload));
EXPECT_GE(buf_.total_buffered_amount(), 1000u);
- absl::optional<SendQueue::DataToSend> chunk_one = buf_.Produce(kNow, 1000);
+ std::optional<SendQueue::DataToSend> chunk_one = buf_.Produce(kNow, 1000);
ASSERT_TRUE(chunk_one.has_value());
EXPECT_EQ(chunk_one->data.stream_id, kStreamID);
EXPECT_EQ(chunk_one->data.ppid, kPPID);
EXPECT_GE(buf_.total_buffered_amount(), 1000u);
- absl::optional<SendQueue::DataToSend> chunk_two = buf_.Produce(kNow, 1000);
+ std::optional<SendQueue::DataToSend> chunk_two = buf_.Produce(kNow, 1000);
ASSERT_TRUE(chunk_two.has_value());
EXPECT_EQ(chunk_two->data.stream_id, StreamID(3));
EXPECT_EQ(chunk_two->data.ppid, PPID(54));
@@ -147,7 +147,7 @@
EXPECT_LT(buf_.total_buffered_amount(), 1000u);
EXPECT_FALSE(buf_.IsEmpty());
- absl::optional<SendQueue::DataToSend> chunk_three = buf_.Produce(kNow, 1000);
+ std::optional<SendQueue::DataToSend> chunk_three = buf_.Produce(kNow, 1000);
ASSERT_TRUE(chunk_three.has_value());
EXPECT_EQ(chunk_three->data.stream_id, StreamID(5));
EXPECT_EQ(chunk_three->data.ppid, PPID(55));
@@ -161,7 +161,7 @@
// Default is ordered
buf_.Add(kNow, DcSctpMessage(kStreamID, kPPID, payload));
- absl::optional<SendQueue::DataToSend> chunk_one =
+ std::optional<SendQueue::DataToSend> chunk_one =
buf_.Produce(kNow, kOneFragmentPacketSize);
ASSERT_TRUE(chunk_one.has_value());
EXPECT_FALSE(chunk_one->data.is_unordered);
@@ -170,7 +170,7 @@
SendOptions opts;
opts.unordered = IsUnordered(true);
buf_.Add(kNow, DcSctpMessage(kStreamID, kPPID, payload), opts);
- absl::optional<SendQueue::DataToSend> chunk_two =
+ std::optional<SendQueue::DataToSend> chunk_two =
buf_.Produce(kNow, kOneFragmentPacketSize);
ASSERT_TRUE(chunk_two.has_value());
EXPECT_TRUE(chunk_two->data.is_unordered);
@@ -222,20 +222,20 @@
buf_.Add(kNow, DcSctpMessage(kStreamID, kPPID, payload));
buf_.Add(kNow, DcSctpMessage(StreamID(2), PPID(54), payload));
- absl::optional<SendQueue::DataToSend> chunk_one =
+ std::optional<SendQueue::DataToSend> chunk_one =
buf_.Produce(kNow, kOneFragmentPacketSize);
ASSERT_TRUE(chunk_one.has_value());
EXPECT_FALSE(chunk_one->data.is_end);
EXPECT_EQ(chunk_one->data.stream_id, kStreamID);
buf_.Discard(chunk_one->data.stream_id, chunk_one->message_id);
- absl::optional<SendQueue::DataToSend> chunk_two =
+ std::optional<SendQueue::DataToSend> chunk_two =
buf_.Produce(kNow, kOneFragmentPacketSize);
ASSERT_TRUE(chunk_two.has_value());
EXPECT_FALSE(chunk_two->data.is_end);
EXPECT_EQ(chunk_two->data.stream_id, StreamID(2));
- absl::optional<SendQueue::DataToSend> chunk_three =
+ std::optional<SendQueue::DataToSend> chunk_three =
buf_.Produce(kNow, kOneFragmentPacketSize);
ASSERT_TRUE(chunk_three.has_value());
EXPECT_TRUE(chunk_three->data.is_end);
@@ -268,7 +268,7 @@
buf_.Add(kNow, DcSctpMessage(kStreamID, kPPID, payload));
buf_.Add(kNow, DcSctpMessage(kStreamID, kPPID, payload));
- absl::optional<SendQueue::DataToSend> chunk_one = buf_.Produce(kNow, 50);
+ std::optional<SendQueue::DataToSend> chunk_one = buf_.Produce(kNow, 50);
ASSERT_TRUE(chunk_one.has_value());
EXPECT_EQ(chunk_one->data.stream_id, kStreamID);
EXPECT_EQ(buf_.total_buffered_amount(), 2 * payload.size() - 50);
@@ -297,7 +297,7 @@
buf_.CommitResetStreams();
EXPECT_EQ(buf_.total_buffered_amount(), payload.size());
- absl::optional<SendQueue::DataToSend> chunk_one = buf_.Produce(kNow, 50);
+ std::optional<SendQueue::DataToSend> chunk_one = buf_.Produce(kNow, 50);
ASSERT_TRUE(chunk_one.has_value());
EXPECT_EQ(chunk_one->data.stream_id, kStreamID);
EXPECT_EQ(buf_.total_buffered_amount(), 0u);
@@ -311,7 +311,7 @@
buf_.Add(kNow, DcSctpMessage(kStreamID, kPPID, payload));
buf_.Add(kNow, DcSctpMessage(kStreamID, kPPID, payload));
- absl::optional<SendQueue::DataToSend> chunk_one =
+ std::optional<SendQueue::DataToSend> chunk_one =
buf_.Produce(kNow, kFragmentSize);
ASSERT_TRUE(chunk_one.has_value());
EXPECT_EQ(chunk_one->data.stream_id, kStreamID);
@@ -322,7 +322,7 @@
EXPECT_EQ(buf_.total_buffered_amount(), 1 * kPayloadSize - kFragmentSize);
// Should still produce fragments until end of message.
- absl::optional<SendQueue::DataToSend> chunk_two =
+ std::optional<SendQueue::DataToSend> chunk_two =
buf_.Produce(kNow, kFragmentSize);
ASSERT_TRUE(chunk_two.has_value());
EXPECT_EQ(chunk_two->data.stream_id, kStreamID);
@@ -338,12 +338,12 @@
buf_.Add(kNow, DcSctpMessage(kStreamID, kPPID, payload));
buf_.Add(kNow, DcSctpMessage(kStreamID, kPPID, payload));
- absl::optional<SendQueue::DataToSend> chunk_one =
+ std::optional<SendQueue::DataToSend> chunk_one =
buf_.Produce(kNow, kOneFragmentPacketSize);
ASSERT_TRUE(chunk_one.has_value());
EXPECT_EQ(chunk_one->data.ssn, SSN(0));
- absl::optional<SendQueue::DataToSend> chunk_two =
+ std::optional<SendQueue::DataToSend> chunk_two =
buf_.Produce(kNow, kOneFragmentPacketSize);
ASSERT_TRUE(chunk_two.has_value());
EXPECT_EQ(chunk_two->data.ssn, SSN(1));
@@ -358,7 +358,7 @@
UnorderedElementsAre(StreamID(1)));
buf_.CommitResetStreams();
- absl::optional<SendQueue::DataToSend> chunk_three =
+ std::optional<SendQueue::DataToSend> chunk_three =
buf_.Produce(kNow, kOneFragmentPacketSize);
ASSERT_TRUE(chunk_three.has_value());
EXPECT_EQ(chunk_three->data.ssn, SSN(0));
@@ -396,13 +396,13 @@
buf_.Add(kNow, DcSctpMessage(StreamID(1), kPPID, payload));
buf_.Add(kNow, DcSctpMessage(StreamID(3), kPPID, payload));
- absl::optional<SendQueue::DataToSend> chunk_one =
+ std::optional<SendQueue::DataToSend> chunk_one =
buf_.Produce(kNow, kOneFragmentPacketSize);
ASSERT_TRUE(chunk_one.has_value());
EXPECT_EQ(chunk_one->data.stream_id, StreamID(1));
EXPECT_EQ(chunk_one->data.ssn, SSN(0));
- absl::optional<SendQueue::DataToSend> chunk_two =
+ std::optional<SendQueue::DataToSend> chunk_two =
buf_.Produce(kNow, kOneFragmentPacketSize);
ASSERT_TRUE(chunk_two.has_value());
EXPECT_EQ(chunk_two->data.stream_id, StreamID(3));
@@ -420,13 +420,13 @@
buf_.CommitResetStreams();
- absl::optional<SendQueue::DataToSend> chunk_three =
+ std::optional<SendQueue::DataToSend> chunk_three =
buf_.Produce(kNow, kOneFragmentPacketSize);
ASSERT_TRUE(chunk_three.has_value());
EXPECT_EQ(chunk_three->data.stream_id, StreamID(1));
EXPECT_EQ(chunk_three->data.ssn, SSN(1));
- absl::optional<SendQueue::DataToSend> chunk_four =
+ std::optional<SendQueue::DataToSend> chunk_four =
buf_.Produce(kNow, kOneFragmentPacketSize);
ASSERT_TRUE(chunk_four.has_value());
EXPECT_EQ(chunk_four->data.stream_id, StreamID(3));
@@ -439,12 +439,12 @@
buf_.Add(kNow, DcSctpMessage(kStreamID, kPPID, payload));
buf_.Add(kNow, DcSctpMessage(kStreamID, kPPID, payload));
- absl::optional<SendQueue::DataToSend> chunk_one =
+ std::optional<SendQueue::DataToSend> chunk_one =
buf_.Produce(kNow, kOneFragmentPacketSize);
ASSERT_TRUE(chunk_one.has_value());
EXPECT_EQ(chunk_one->data.ssn, SSN(0));
- absl::optional<SendQueue::DataToSend> chunk_two =
+ std::optional<SendQueue::DataToSend> chunk_two =
buf_.Produce(kNow, kOneFragmentPacketSize);
ASSERT_TRUE(chunk_two.has_value());
EXPECT_EQ(chunk_two->data.ssn, SSN(1));
@@ -459,7 +459,7 @@
UnorderedElementsAre(StreamID(1)));
buf_.RollbackResetStreams();
- absl::optional<SendQueue::DataToSend> chunk_three =
+ std::optional<SendQueue::DataToSend> chunk_three =
buf_.Produce(kNow, kOneFragmentPacketSize);
ASSERT_TRUE(chunk_three.has_value());
EXPECT_EQ(chunk_three->data.ssn, SSN(2));
@@ -744,7 +744,7 @@
// Drain it a bit - will trigger.
EXPECT_CALL(callbacks_, OnTotalBufferedAmountLow).Times(1);
- absl::optional<SendQueue::DataToSend> chunk_two =
+ std::optional<SendQueue::DataToSend> chunk_two =
buf_.Produce(kNow, kOneFragmentPacketSize);
}
@@ -858,7 +858,7 @@
buf_.Add(kNow, DcSctpMessage(kStreamID, kPPID, payload),
SendOptions{.lifecycle_id = LifecycleId(2)});
- absl::optional<SendQueue::DataToSend> chunk_one = buf_.Produce(kNow, 50);
+ std::optional<SendQueue::DataToSend> chunk_one = buf_.Produce(kNow, 50);
ASSERT_TRUE(chunk_one.has_value());
EXPECT_EQ(chunk_one->data.stream_id, kStreamID);
EXPECT_EQ(buf_.total_buffered_amount(), 2 * payload.size() - 50);
@@ -876,7 +876,7 @@
buf_.Add(kNow, DcSctpMessage(kStreamID, kPPID, payload),
SendOptions{.lifecycle_id = LifecycleId(1)});
- absl::optional<SendQueue::DataToSend> chunk_one =
+ std::optional<SendQueue::DataToSend> chunk_one =
buf_.Produce(kNow, kOneFragmentPacketSize);
ASSERT_TRUE(chunk_one.has_value());
EXPECT_FALSE(chunk_one->data.is_end);
diff --git a/net/dcsctp/tx/send_queue.h b/net/dcsctp/tx/send_queue.h
index d0d834c..5f7ca2a 100644
--- a/net/dcsctp/tx/send_queue.h
+++ b/net/dcsctp/tx/send_queue.h
@@ -12,10 +12,10 @@
#include <cstdint>
#include <limits>
+#include <optional>
#include <utility>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/units/timestamp.h"
#include "net/dcsctp/common/internal_types.h"
@@ -56,8 +56,8 @@
//
// `max_size` refers to how many payload bytes that may be produced, not
// including any headers.
- virtual absl::optional<DataToSend> Produce(webrtc::Timestamp now,
- size_t max_size) = 0;
+ virtual std::optional<DataToSend> Produce(webrtc::Timestamp now,
+ size_t max_size) = 0;
// Discards a partially sent message identified by the parameters
// `stream_id` and `message_id`. The `message_id` comes from the returned
diff --git a/net/dcsctp/tx/stream_scheduler.cc b/net/dcsctp/tx/stream_scheduler.cc
index 66c4457..c8984b8 100644
--- a/net/dcsctp/tx/stream_scheduler.cc
+++ b/net/dcsctp/tx/stream_scheduler.cc
@@ -10,9 +10,9 @@
#include "net/dcsctp/tx/stream_scheduler.h"
#include <algorithm>
+#include <optional>
#include "absl/algorithm/container.h"
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "net/dcsctp/packet/data.h"
#include "net/dcsctp/public/dcsctp_message.h"
@@ -30,7 +30,7 @@
inverse_weight_ = InverseWeight(priority);
}
-absl::optional<SendQueue::DataToSend> StreamScheduler::Produce(
+std::optional<SendQueue::DataToSend> StreamScheduler::Produce(
webrtc::Timestamp now,
size_t max_size) {
// For non-interleaved streams, avoid rescheduling while still sending a
@@ -50,7 +50,7 @@
RTC_DCHECK(rescheduling || current_stream_ != nullptr);
- absl::optional<SendQueue::DataToSend> data;
+ std::optional<SendQueue::DataToSend> data;
while (!data.has_value() && !active_streams_.empty()) {
if (rescheduling) {
auto it = active_streams_.begin();
@@ -77,7 +77,7 @@
<< "There is no stream with data; Can't produce any data.";
RTC_DCHECK(IsConsistent());
- return absl::nullopt;
+ return std::nullopt;
}
RTC_DCHECK(data->data.stream_id == current_stream_->stream_id());
@@ -126,10 +126,10 @@
return VirtualTime(*current_virtual_time_ + 1);
}
-absl::optional<SendQueue::DataToSend> StreamScheduler::Stream::Produce(
+std::optional<SendQueue::DataToSend> StreamScheduler::Stream::Produce(
webrtc::Timestamp now,
size_t max_size) {
- absl::optional<SendQueue::DataToSend> data = producer_.Produce(now, max_size);
+ std::optional<SendQueue::DataToSend> data = producer_.Produce(now, max_size);
if (data.has_value()) {
VirtualTime new_current = CalculateFinishTime(data->data.payload.size());
diff --git a/net/dcsctp/tx/stream_scheduler.h b/net/dcsctp/tx/stream_scheduler.h
index 9d76fc6..f5377fa 100644
--- a/net/dcsctp/tx/stream_scheduler.h
+++ b/net/dcsctp/tx/stream_scheduler.h
@@ -15,6 +15,7 @@
#include <deque>
#include <map>
#include <memory>
+#include <optional>
#include <queue>
#include <set>
#include <string>
@@ -23,7 +24,6 @@
#include "absl/algorithm/container.h"
#include "absl/memory/memory.h"
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "net/dcsctp/packet/chunk/idata_chunk.h"
#include "net/dcsctp/packet/sctp_packet.h"
@@ -86,9 +86,9 @@
// as `now` and should be used to skip chunks with expired limited lifetime.
// The parameter `max_size` specifies the maximum amount of actual payload
// that may be returned. If these constraints prevents the stream from
- // sending some data, `absl::nullopt` should be returned.
- virtual absl::optional<SendQueue::DataToSend> Produce(webrtc::Timestamp now,
- size_t max_size) = 0;
+ // sending some data, `std::nullopt` should be returned.
+ virtual std::optional<SendQueue::DataToSend> Produce(webrtc::Timestamp now,
+ size_t max_size) = 0;
// Returns the number of payload bytes that is scheduled to be sent in the
// next enqueued message, or zero if there are no enqueued messages or if
@@ -132,8 +132,8 @@
// Produces a message from this stream. This will only be called on streams
// that have data.
- absl::optional<SendQueue::DataToSend> Produce(webrtc::Timestamp now,
- size_t max_size);
+ std::optional<SendQueue::DataToSend> Produce(webrtc::Timestamp now,
+ size_t max_size);
void MakeActive(size_t bytes_to_send_next);
void ForceMarkInactive();
@@ -180,9 +180,9 @@
// Produces a fragment of data to send. The current wall time is specified as
// `now` and will be used to skip chunks with expired limited lifetime. The
// parameter `max_size` specifies the maximum amount of actual payload that
- // may be returned. If no data can be produced, `absl::nullopt` is returned.
- absl::optional<SendQueue::DataToSend> Produce(webrtc::Timestamp now,
- size_t max_size);
+ // may be returned. If no data can be produced, `std::nullopt` is returned.
+ std::optional<SendQueue::DataToSend> Produce(webrtc::Timestamp now,
+ size_t max_size);
std::set<StreamID> ActiveStreamsForTesting() const;
diff --git a/net/dcsctp/tx/stream_scheduler_test.cc b/net/dcsctp/tx/stream_scheduler_test.cc
index 42d0b3c..a9b4612 100644
--- a/net/dcsctp/tx/stream_scheduler_test.cc
+++ b/net/dcsctp/tx/stream_scheduler_test.cc
@@ -40,7 +40,7 @@
return true;
}
-std::function<absl::optional<SendQueue::DataToSend>(Timestamp, size_t)>
+std::function<std::optional<SendQueue::DataToSend>(Timestamp, size_t)>
CreateChunk(OutgoingMessageId message_id,
StreamID sid,
MID mid,
@@ -58,7 +58,7 @@
size_t packets_to_generate) {
std::map<StreamID, size_t> packet_counts;
for (size_t i = 0; i < packets_to_generate; ++i) {
- absl::optional<SendQueue::DataToSend> data = scheduler.Produce(kNow, kMtu);
+ std::optional<SendQueue::DataToSend> data = scheduler.Produce(kNow, kMtu);
if (data.has_value()) {
++packet_counts[data->data.stream_id];
}
@@ -68,7 +68,7 @@
class MockStreamProducer : public StreamScheduler::StreamProducer {
public:
- MOCK_METHOD(absl::optional<SendQueue::DataToSend>,
+ MOCK_METHOD(std::optional<SendQueue::DataToSend>,
Produce,
(Timestamp, size_t),
(override));
@@ -101,7 +101,7 @@
TEST(StreamSchedulerTest, HasNoActiveStreams) {
StreamScheduler scheduler("", kMtu);
- EXPECT_EQ(scheduler.Produce(kNow, kMtu), absl::nullopt);
+ EXPECT_EQ(scheduler.Produce(kNow, kMtu), std::nullopt);
}
// Stream properties can be set and retrieved
@@ -134,7 +134,7 @@
stream->MaybeMakeActive();
EXPECT_THAT(scheduler.Produce(kNow, kMtu), HasDataWithMid(MID(0)));
- EXPECT_EQ(scheduler.Produce(kNow, kMtu), absl::nullopt);
+ EXPECT_EQ(scheduler.Produce(kNow, kMtu), std::nullopt);
}
// Switches between two streams after every packet.
@@ -175,7 +175,7 @@
EXPECT_THAT(scheduler.Produce(kNow, kMtu), HasDataWithMid(MID(201)));
EXPECT_THAT(scheduler.Produce(kNow, kMtu), HasDataWithMid(MID(102)));
EXPECT_THAT(scheduler.Produce(kNow, kMtu), HasDataWithMid(MID(202)));
- EXPECT_EQ(scheduler.Produce(kNow, kMtu), absl::nullopt);
+ EXPECT_EQ(scheduler.Produce(kNow, kMtu), std::nullopt);
}
// Switches between two streams after every packet, but keeps producing from the
@@ -241,7 +241,7 @@
EXPECT_THAT(scheduler.Produce(kNow, kMtu), HasDataWithMid(MID(201)));
EXPECT_THAT(scheduler.Produce(kNow, kMtu), HasDataWithMid(MID(102)));
EXPECT_THAT(scheduler.Produce(kNow, kMtu), HasDataWithMid(MID(202)));
- EXPECT_EQ(scheduler.Produce(kNow, kMtu), absl::nullopt);
+ EXPECT_EQ(scheduler.Produce(kNow, kMtu), std::nullopt);
}
// Deactivates a stream before it has finished producing all packets.
@@ -265,7 +265,7 @@
// ... but the stream is made inactive before it can be produced.
stream1->MakeInactive();
- EXPECT_EQ(scheduler.Produce(kNow, kMtu), absl::nullopt);
+ EXPECT_EQ(scheduler.Produce(kNow, kMtu), std::nullopt);
}
// Resumes a paused stream - makes a stream active after inactivating it.
@@ -292,10 +292,10 @@
EXPECT_THAT(scheduler.Produce(kNow, kMtu), HasDataWithMid(MID(101)));
stream1->MakeInactive();
- EXPECT_EQ(scheduler.Produce(kNow, kMtu), absl::nullopt);
+ EXPECT_EQ(scheduler.Produce(kNow, kMtu), std::nullopt);
stream1->MaybeMakeActive();
EXPECT_THAT(scheduler.Produce(kNow, kMtu), HasDataWithMid(MID(102)));
- EXPECT_EQ(scheduler.Produce(kNow, kMtu), absl::nullopt);
+ EXPECT_EQ(scheduler.Produce(kNow, kMtu), std::nullopt);
}
// Iterates between streams, where one is suddenly paused and later resumed.
@@ -339,7 +339,7 @@
stream1->MaybeMakeActive();
EXPECT_THAT(scheduler.Produce(kNow, kMtu), HasDataWithMid(MID(101)));
EXPECT_THAT(scheduler.Produce(kNow, kMtu), HasDataWithMid(MID(102)));
- EXPECT_EQ(scheduler.Produce(kNow, kMtu), absl::nullopt);
+ EXPECT_EQ(scheduler.Produce(kNow, kMtu), std::nullopt);
}
// Verifies that packet counts are evenly distributed in round robin scheduling.
@@ -439,7 +439,7 @@
EXPECT_THAT(scheduler.Produce(kNow, kMtu), HasDataWithMid(MID(201)));
// t = 210
EXPECT_THAT(scheduler.Produce(kNow, kMtu), HasDataWithMid(MID(202)));
- EXPECT_EQ(scheduler.Produce(kNow, kMtu), absl::nullopt);
+ EXPECT_EQ(scheduler.Produce(kNow, kMtu), std::nullopt);
}
// Will do weighted fair queuing with three streams having different priority.
@@ -510,7 +510,7 @@
EXPECT_THAT(scheduler.Produce(kNow, kMtu), HasDataWithMid(MID(101)));
// t ~= 240
EXPECT_THAT(scheduler.Produce(kNow, kMtu), HasDataWithMid(MID(102)));
- EXPECT_EQ(scheduler.Produce(kNow, kMtu), absl::nullopt);
+ EXPECT_EQ(scheduler.Produce(kNow, kMtu), std::nullopt);
}
// Will do weighted fair queuing with three streams having different priority
@@ -604,7 +604,7 @@
EXPECT_THAT(scheduler.Produce(kNow, kMtu), HasDataWithMid(MID(202)));
// t ~= 11200
EXPECT_THAT(scheduler.Produce(kNow, kMtu), HasDataWithMid(MID(102)));
- EXPECT_EQ(scheduler.Produce(kNow, kMtu), absl::nullopt);
+ EXPECT_EQ(scheduler.Produce(kNow, kMtu), std::nullopt);
}
TEST(StreamSchedulerTest, WillDistributeWFQPacketsInTwoStreamsByPriority) {
// A simple test with two streams of different priority, but sending packets
@@ -728,7 +728,7 @@
EXPECT_THAT(scheduler.Produce(kNow, kMtu), HasDataWithMid(MID(1)));
EXPECT_THAT(scheduler.Produce(kNow, kMtu), HasDataWithMid(MID(1)));
EXPECT_THAT(scheduler.Produce(kNow, kMtu), HasDataWithMid(MID(0)));
- EXPECT_EQ(scheduler.Produce(kNow, kMtu), absl::nullopt);
+ EXPECT_EQ(scheduler.Produce(kNow, kMtu), std::nullopt);
}
// Sending large messages with large MTU will not fragment messages and will
@@ -759,7 +759,7 @@
stream2->MaybeMakeActive();
EXPECT_THAT(scheduler.Produce(kNow, kMtu), HasDataWithMid(MID(1)));
EXPECT_THAT(scheduler.Produce(kNow, kMtu), HasDataWithMid(MID(0)));
- EXPECT_EQ(scheduler.Produce(kNow, kMtu), absl::nullopt);
+ EXPECT_EQ(scheduler.Produce(kNow, kMtu), std::nullopt);
}
} // namespace
diff --git a/p2p/BUILD.gn b/p2p/BUILD.gn
index 73d5809..5802599 100644
--- a/p2p/BUILD.gn
+++ b/p2p/BUILD.gn
@@ -159,7 +159,6 @@
"//third_party/abseil-cpp/absl/functional:any_invocable",
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -180,7 +179,6 @@
":ice_transport_internal",
":transport_description",
"../api:array_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -360,7 +358,6 @@
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/functional:any_invocable",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -372,7 +369,6 @@
deps = [
"../api:candidate",
"../api/units:timestamp",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -472,7 +468,6 @@
":ice_transport_internal",
"../rtc_base:checks",
"../rtc_base/system:rtc_export",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -524,7 +519,6 @@
"../rtc_base/system:rtc_export",
"../rtc_base/third_party/sigslot",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -598,13 +592,11 @@
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
rtc_source_set("p2p_transport_channel_ice_field_trials") {
sources = [ "base/p2p_transport_channel_ice_field_trials.h" ]
- deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
}
rtc_library("packet_transport_internal") {
@@ -624,7 +616,6 @@
"../rtc_base/system:rtc_export",
"../rtc_base/third_party/sigslot",
"//third_party/abseil-cpp/absl/functional:any_invocable",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -679,7 +670,6 @@
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -706,7 +696,6 @@
"../rtc_base/system:rtc_export",
"../rtc_base/third_party/sigslot",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -725,7 +714,6 @@
"../rtc_base:callback_list",
"../rtc_base:socket_address",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -862,7 +850,6 @@
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -926,7 +913,6 @@
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -975,7 +961,6 @@
"../rtc_base:macromagic",
"../rtc_base:threading",
"../rtc_base:timeutils",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -996,7 +981,6 @@
"../rtc_base/network:received_packet",
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -1186,7 +1170,6 @@
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
}
diff --git a/p2p/base/active_ice_controller_interface.h b/p2p/base/active_ice_controller_interface.h
index e54838e..30de264 100644
--- a/p2p/base/active_ice_controller_interface.h
+++ b/p2p/base/active_ice_controller_interface.h
@@ -11,7 +11,8 @@
#ifndef P2P_BASE_ACTIVE_ICE_CONTROLLER_INTERFACE_H_
#define P2P_BASE_ACTIVE_ICE_CONTROLLER_INTERFACE_H_
-#include "absl/types/optional.h"
+#include <optional>
+
#include "api/array_view.h"
#include "p2p/base/connection.h"
#include "p2p/base/ice_switch_reason.h"
diff --git a/p2p/base/basic_ice_controller.cc b/p2p/base/basic_ice_controller.cc
index f5b3eca..64c6971 100644
--- a/p2p/base/basic_ice_controller.cc
+++ b/p2p/base/basic_ice_controller.cc
@@ -32,7 +32,7 @@
bool LocalCandidateUsesPreferredNetwork(
const cricket::Connection* conn,
- absl::optional<rtc::AdapterType> network_preference) {
+ std::optional<rtc::AdapterType> network_preference) {
rtc::AdapterType network_type = conn->network()->type();
return network_preference.has_value() && (network_type == network_preference);
}
@@ -40,7 +40,7 @@
int CompareCandidatePairsByNetworkPreference(
const cricket::Connection* a,
const cricket::Connection* b,
- absl::optional<rtc::AdapterType> network_preference) {
+ std::optional<rtc::AdapterType> network_preference) {
bool a_uses_preferred_network =
LocalCandidateUsesPreferredNetwork(a, network_preference);
bool b_uses_preferred_network =
@@ -420,7 +420,7 @@
if (!field_trials_->initial_select_dampening.has_value() &&
!field_trials_->initial_select_dampening_ping_received.has_value()) {
// experiment not enabled => select connection.
- return {new_connection, absl::nullopt};
+ return {new_connection, std::nullopt};
}
int64_t now = rtc::TimeMillis();
@@ -441,7 +441,7 @@
<< initial_select_timestamp_ms_
<< " selection delayed by: " << (now - start_wait) << "ms";
initial_select_timestamp_ms_ = 0;
- return {new_connection, absl::nullopt};
+ return {new_connection, std::nullopt};
}
// We are not yet ready to select first connection...
@@ -464,7 +464,7 @@
}
RTC_LOG(LS_INFO) << "delay initial selection up to " << min_delay << "ms";
- return {.connection = absl::nullopt,
+ return {.connection = std::nullopt,
.recheck_event = IceRecheckEvent(
IceSwitchReason::ICE_CONTROLLER_RECHECK, min_delay)};
}
@@ -473,7 +473,7 @@
IceSwitchReason reason,
const Connection* new_connection) {
if (!ReadyToSend(new_connection) || selected_connection_ == new_connection) {
- return {absl::nullopt, absl::nullopt};
+ return {std::nullopt, std::nullopt};
}
if (selected_connection_ == nullptr) {
@@ -486,17 +486,17 @@
int compare_a_b_by_networks = CompareCandidatePairNetworks(
new_connection, selected_connection_, config_.network_preference);
if (compare_a_b_by_networks == b_is_better && !new_connection->receiving()) {
- return {absl::nullopt, absl::nullopt};
+ return {std::nullopt, std::nullopt};
}
bool missed_receiving_unchanged_threshold = false;
- absl::optional<int64_t> receiving_unchanged_threshold(
+ std::optional<int64_t> receiving_unchanged_threshold(
rtc::TimeMillis() - config_.receiving_switching_delay_or_default());
int cmp = CompareConnections(selected_connection_, new_connection,
receiving_unchanged_threshold,
&missed_receiving_unchanged_threshold);
- absl::optional<IceRecheckEvent> recheck_event;
+ std::optional<IceRecheckEvent> recheck_event;
if (missed_receiving_unchanged_threshold &&
config_.receiving_switching_delay_or_default()) {
// If we do not switch to the connection because it missed the receiving
@@ -508,18 +508,18 @@
}
if (cmp < 0) {
- return {new_connection, absl::nullopt};
+ return {new_connection, std::nullopt};
} else if (cmp > 0) {
- return {absl::nullopt, recheck_event};
+ return {std::nullopt, recheck_event};
}
// If everything else is the same, switch only if rtt has improved by
// a margin.
if (new_connection->rtt() <= selected_connection_->rtt() - kMinImprovement) {
- return {new_connection, absl::nullopt};
+ return {new_connection, std::nullopt};
}
- return {absl::nullopt, recheck_event};
+ return {std::nullopt, recheck_event};
}
IceControllerInterface::SwitchResult
@@ -531,7 +531,7 @@
// TODO(honghaiz): Don't sort; Just use std::max_element in the right places.
absl::c_stable_sort(
connections_, [this](const Connection* a, const Connection* b) {
- int cmp = CompareConnections(a, b, absl::nullopt, nullptr);
+ int cmp = CompareConnections(a, b, std::nullopt, nullptr);
if (cmp != 0) {
return cmp > 0;
}
@@ -575,7 +575,7 @@
int BasicIceController::CompareConnectionStates(
const Connection* a,
const Connection* b,
- absl::optional<int64_t> receiving_unchanged_threshold,
+ std::optional<int64_t> receiving_unchanged_threshold,
bool* missed_receiving_unchanged_threshold) const {
// First, prefer a connection that's writable or presumed writable over
// one that's not writable.
@@ -693,7 +693,7 @@
int BasicIceController::CompareConnections(
const Connection* a,
const Connection* b,
- absl::optional<int64_t> receiving_unchanged_threshold,
+ std::optional<int64_t> receiving_unchanged_threshold,
bool* missed_receiving_unchanged_threshold) const {
RTC_CHECK(a != nullptr);
RTC_CHECK(b != nullptr);
@@ -732,7 +732,7 @@
int BasicIceController::CompareCandidatePairNetworks(
const Connection* a,
const Connection* b,
- absl::optional<rtc::AdapterType> network_preference) const {
+ std::optional<rtc::AdapterType> network_preference) const {
int compare_a_b_by_network_preference =
CompareCandidatePairsByNetworkPreference(a, b,
config_.network_preference);
diff --git a/p2p/base/basic_ice_controller.h b/p2p/base/basic_ice_controller.h
index f9f4d45..dfb99ea 100644
--- a/p2p/base/basic_ice_controller.h
+++ b/p2p/base/basic_ice_controller.h
@@ -112,7 +112,7 @@
int CompareCandidatePairNetworks(
const Connection* a,
const Connection* b,
- absl::optional<rtc::AdapterType> network_preference) const;
+ std::optional<rtc::AdapterType> network_preference) const;
// The methods below return a positive value if `a` is preferable to `b`,
// a negative value if `b` is preferable, and 0 if they're equally preferable.
@@ -124,7 +124,7 @@
int CompareConnectionStates(
const Connection* a,
const Connection* b,
- absl::optional<int64_t> receiving_unchanged_threshold,
+ std::optional<int64_t> receiving_unchanged_threshold,
bool* missed_receiving_unchanged_threshold) const;
int CompareConnectionCandidates(const Connection* a,
const Connection* b) const;
@@ -135,7 +135,7 @@
// Returns a positive value if `a` is better than `b`.
int CompareConnections(const Connection* a,
const Connection* b,
- absl::optional<int64_t> receiving_unchanged_threshold,
+ std::optional<int64_t> receiving_unchanged_threshold,
bool* missed_receiving_unchanged_threshold) const;
SwitchResult HandleInitialSelectDampening(IceSwitchReason reason,
diff --git a/p2p/base/connection.cc b/p2p/base/connection.cc
index 762b692..bb79096 100644
--- a/p2p/base/connection.cc
+++ b/p2p/base/connection.cc
@@ -15,12 +15,12 @@
#include <algorithm>
#include <cstdint>
#include <memory>
+#include <optional>
#include <utility>
#include <vector>
#include "absl/algorithm/container.h"
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/units/timestamp.h"
#include "p2p/base/p2p_constants.h"
@@ -387,7 +387,7 @@
return unwritable_timeout_.value_or(CONNECTION_WRITE_CONNECT_TIMEOUT);
}
-void Connection::set_unwritable_timeout(const absl::optional<int>& value_ms) {
+void Connection::set_unwritable_timeout(const std::optional<int>& value_ms) {
RTC_DCHECK_RUN_ON(network_thread_);
unwritable_timeout_ = value_ms;
}
@@ -397,7 +397,7 @@
return unwritable_min_checks_.value_or(CONNECTION_WRITE_CONNECT_FAILURES);
}
-void Connection::set_unwritable_min_checks(const absl::optional<int>& value) {
+void Connection::set_unwritable_min_checks(const std::optional<int>& value) {
RTC_DCHECK_RUN_ON(network_thread_);
unwritable_min_checks_ = value;
}
@@ -407,7 +407,7 @@
return inactive_timeout_.value_or(CONNECTION_WRITE_TIMEOUT);
}
-void Connection::set_inactive_timeout(const absl::optional<int>& value) {
+void Connection::set_inactive_timeout(const std::optional<int>& value) {
RTC_DCHECK_RUN_ON(network_thread_);
inactive_timeout_ = value;
}
@@ -418,7 +418,7 @@
}
void Connection::set_receiving_timeout(
- absl::optional<int> receiving_timeout_ms) {
+ std::optional<int> receiving_timeout_ms) {
RTC_DCHECK_RUN_ON(network_thread_);
receiving_timeout_ = receiving_timeout_ms;
}
@@ -1095,7 +1095,7 @@
return last_ping_response_received_;
}
-const absl::optional<std::string>& Connection::last_ping_id_received() const {
+const std::optional<std::string>& Connection::last_ping_id_received() const {
RTC_DCHECK_RUN_ON(network_thread_);
return last_ping_id_received_;
}
@@ -1113,7 +1113,7 @@
return last_ping_received_;
}
-void Connection::ReceivedPing(const absl::optional<std::string>& request_id) {
+void Connection::ReceivedPing(const std::optional<std::string>& request_id) {
RTC_DCHECK_RUN_ON(network_thread_);
last_ping_received_ = rtc::TimeMillis();
last_ping_id_received_ = request_id;
@@ -1156,7 +1156,7 @@
void Connection::ReceivedPingResponse(
int rtt,
absl::string_view request_id,
- const absl::optional<uint32_t>& nomination) {
+ const std::optional<uint32_t>& nomination) {
RTC_DCHECK_RUN_ON(network_thread_);
RTC_DCHECK_GE(rtt, 0);
// We've already validated that this is a STUN binding response with
@@ -1420,7 +1420,7 @@
", rtt="
<< rtt << ", pings_since_last_response=" << pings;
}
- absl::optional<uint32_t> nomination;
+ std::optional<uint32_t> nomination;
const std::string request_id = request->id();
auto iter = absl::c_find_if(
pings_since_last_response_,
@@ -1563,7 +1563,7 @@
}
// TODO(deadbeef): A value of '0' for the generation is used for both
// generation 0 and "generation unknown". It should be changed to an
- // absl::optional to fix this.
+ // std::optional to fix this.
if (remote_candidate_.username() == ice_params.ufrag &&
remote_candidate_.password() == ice_params.pwd &&
remote_candidate_.generation() == 0) {
@@ -1725,7 +1725,7 @@
}
bool Connection::TooManyOutstandingPings(
- const absl::optional<int>& max_outstanding_pings) const {
+ const std::optional<int>& max_outstanding_pings) const {
RTC_DCHECK_RUN_ON(network_thread_);
if (!max_outstanding_pings.has_value()) {
return false;
diff --git a/p2p/base/connection.h b/p2p/base/connection.h
index 583e62e..8079a56 100644
--- a/p2p/base/connection.h
+++ b/p2p/base/connection.h
@@ -16,6 +16,7 @@
#include <cstdint>
#include <functional>
#include <memory>
+#include <optional>
#include <string>
#include <type_traits>
#include <utility>
@@ -23,7 +24,6 @@
#include "absl/functional/any_invocable.h"
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/candidate.h"
#include "api/rtc_error.h"
#include "api/sequence_checker.h"
@@ -121,11 +121,11 @@
int rtt() const;
int unwritable_timeout() const;
- void set_unwritable_timeout(const absl::optional<int>& value_ms);
+ void set_unwritable_timeout(const std::optional<int>& value_ms);
int unwritable_min_checks() const;
- void set_unwritable_min_checks(const absl::optional<int>& value);
+ void set_unwritable_min_checks(const std::optional<int>& value);
int inactive_timeout() const;
- void set_inactive_timeout(const absl::optional<int>& value);
+ void set_inactive_timeout(const std::optional<int>& value);
// Gets the `ConnectionInfo` stats, where `best_connection` has not been
// populated (default value false).
@@ -186,7 +186,7 @@
bool nominated() const;
int receiving_timeout() const;
- void set_receiving_timeout(absl::optional<int> receiving_timeout_ms);
+ void set_receiving_timeout(std::optional<int> receiving_timeout_ms);
// Deletes a `Connection` instance is by calling the `DestroyConnection`
// method in `Port`.
@@ -219,13 +219,13 @@
void ReceivedPingResponse(
int rtt,
absl::string_view request_id,
- const absl::optional<uint32_t>& nomination = absl::nullopt);
+ const std::optional<uint32_t>& nomination = std::nullopt);
std::unique_ptr<IceMessage> BuildPingRequest(
std::unique_ptr<StunByteStringAttribute> delta)
RTC_RUN_ON(network_thread_);
int64_t last_ping_response_received() const;
- const absl::optional<std::string>& last_ping_id_received() const;
+ const std::optional<std::string>& last_ping_id_received() const;
// Used to check if any STUN ping response has been received.
int rtt_samples() const;
@@ -235,7 +235,7 @@
int64_t last_ping_received() const;
void ReceivedPing(
- const absl::optional<std::string>& request_id = absl::nullopt);
+ const std::optional<std::string>& request_id = std::nullopt);
// Handles the binding request; sends a response if this is a valid request.
void HandleStunBindingOrGoogPingRequest(IceMessage* msg);
// Handles the piggyback acknowledgement of the lastest connectivity check
@@ -297,7 +297,7 @@
bool stable(int64_t now) const;
// Check if we sent `val` pings without receving a response.
- bool TooManyOutstandingPings(const absl::optional<int>& val) const;
+ bool TooManyOutstandingPings(const std::optional<int>& val) const;
// Called by Port when the network cost changes.
void SetLocalCandidateNetworkCost(uint16_t cost);
@@ -355,8 +355,8 @@
}
void ClearStunDictConsumer() {
- goog_delta_consumer_ = absl::nullopt;
- goog_delta_ack_consumer_ = absl::nullopt;
+ goog_delta_consumer_ = std::nullopt;
+ goog_delta_ack_consumer_ = std::nullopt;
}
protected:
@@ -459,7 +459,7 @@
// https://w3c.github.io/webrtc-stats/#dom-rtcicecandidatepairstats-totalroundtriptime
uint64_t total_round_trip_time_ms_ RTC_GUARDED_BY(network_thread_) = 0;
// https://w3c.github.io/webrtc-stats/#dom-rtcicecandidatepairstats-currentroundtriptime
- absl::optional<uint32_t> current_round_trip_time_ms_
+ std::optional<uint32_t> current_round_trip_time_ms_
RTC_GUARDED_BY(network_thread_);
int64_t last_ping_sent_ RTC_GUARDED_BY(
network_thread_); // last time we sent a ping to the other side
@@ -473,21 +473,21 @@
RTC_GUARDED_BY(network_thread_);
// Transaction ID of the last connectivity check received. Null if having not
// received a ping yet.
- absl::optional<std::string> last_ping_id_received_
+ std::optional<std::string> last_ping_id_received_
RTC_GUARDED_BY(network_thread_);
- absl::optional<int> unwritable_timeout_ RTC_GUARDED_BY(network_thread_);
- absl::optional<int> unwritable_min_checks_ RTC_GUARDED_BY(network_thread_);
- absl::optional<int> inactive_timeout_ RTC_GUARDED_BY(network_thread_);
+ std::optional<int> unwritable_timeout_ RTC_GUARDED_BY(network_thread_);
+ std::optional<int> unwritable_min_checks_ RTC_GUARDED_BY(network_thread_);
+ std::optional<int> inactive_timeout_ RTC_GUARDED_BY(network_thread_);
IceCandidatePairState state_ RTC_GUARDED_BY(network_thread_);
// Time duration to switch from receiving to not receiving.
- absl::optional<int> receiving_timeout_ RTC_GUARDED_BY(network_thread_);
+ std::optional<int> receiving_timeout_ RTC_GUARDED_BY(network_thread_);
const int64_t time_created_ms_ RTC_GUARDED_BY(network_thread_);
const int64_t delta_internal_unix_epoch_ms_ RTC_GUARDED_BY(network_thread_);
int num_pings_sent_ RTC_GUARDED_BY(network_thread_) = 0;
- absl::optional<webrtc::IceCandidatePairDescription> log_description_
+ std::optional<webrtc::IceCandidatePairDescription> log_description_
RTC_GUARDED_BY(network_thread_);
webrtc::IceEventLog* ice_event_log_ RTC_GUARDED_BY(network_thread_) = nullptr;
@@ -495,8 +495,7 @@
// if configured via field trial, the remote peer supports it (signaled
// in STUN_BINDING) and if the last STUN BINDING is identical to the one
// that is about to be sent.
- absl::optional<bool> remote_support_goog_ping_
- RTC_GUARDED_BY(network_thread_);
+ std::optional<bool> remote_support_goog_ping_ RTC_GUARDED_BY(network_thread_);
std::unique_ptr<StunMessage> cached_stun_binding_
RTC_GUARDED_BY(network_thread_);
@@ -504,10 +503,10 @@
rtc::EventBasedExponentialMovingAverage rtt_estimate_
RTC_GUARDED_BY(network_thread_);
- absl::optional<std::function<std::unique_ptr<StunAttribute>(
+ std::optional<std::function<std::unique_ptr<StunAttribute>(
const StunByteStringAttribute*)>>
goog_delta_consumer_;
- absl::optional<
+ std::optional<
std::function<void(webrtc::RTCErrorOr<const StunUInt64Attribute*>)>>
goog_delta_ack_consumer_;
absl::AnyInvocable<void(Connection*, const rtc::ReceivedPacket&)>
diff --git a/p2p/base/connection_info.h b/p2p/base/connection_info.h
index e7ed1b4..333c9b1 100644
--- a/p2p/base/connection_info.h
+++ b/p2p/base/connection_info.h
@@ -11,9 +11,9 @@
#ifndef P2P_BASE_CONNECTION_INFO_H_
#define P2P_BASE_CONNECTION_INFO_H_
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/candidate.h"
#include "api/units/timestamp.h"
@@ -72,11 +72,11 @@
// https://w3c.github.io/webrtc-stats/#dom-rtcicecandidatepairstats-totalroundtriptime
uint64_t total_round_trip_time_ms;
// https://w3c.github.io/webrtc-stats/#dom-rtcicecandidatepairstats-currentroundtriptime
- absl::optional<uint32_t> current_round_trip_time_ms;
+ std::optional<uint32_t> current_round_trip_time_ms;
// https://w3c.github.io/webrtc-stats/#dom-rtcicecandidatepairstats-lastpacketreceivedtimestamp
- absl::optional<webrtc::Timestamp> last_data_received;
- absl::optional<webrtc::Timestamp> last_data_sent;
+ std::optional<webrtc::Timestamp> last_data_received;
+ std::optional<webrtc::Timestamp> last_data_sent;
};
// Information about all the candidate pairs of a channel.
diff --git a/p2p/base/dtls_transport.cc b/p2p/base/dtls_transport.cc
index 11ebaf7..27d1849 100644
--- a/p2p/base/dtls_transport.cc
+++ b/p2p/base/dtls_transport.cc
@@ -237,7 +237,7 @@
absl::string_view digest_alg,
const uint8_t* digest,
size_t digest_len,
- absl::optional<rtc::SSLRole> role) {
+ std::optional<rtc::SSLRole> role) {
rtc::Buffer remote_fingerprint_value(digest, digest_len);
bool is_dtls_restart =
dtls_active_ && remote_fingerprint_value_ != remote_fingerprint_value;
@@ -506,7 +506,7 @@
return ice_transport_->GetError();
}
-absl::optional<rtc::NetworkRoute> DtlsTransport::network_route() const {
+std::optional<rtc::NetworkRoute> DtlsTransport::network_route() const {
return ice_transport_->network_route();
}
@@ -759,7 +759,7 @@
}
void DtlsTransport::OnNetworkRouteChanged(
- absl::optional<rtc::NetworkRoute> network_route) {
+ std::optional<rtc::NetworkRoute> network_route) {
RTC_DCHECK_RUN_ON(&thread_checker_);
SignalNetworkRouteChanged(network_route);
}
@@ -872,7 +872,7 @@
void DtlsTransport::ConfigureHandshakeTimeout() {
RTC_DCHECK(dtls_);
- absl::optional<int> rtt = ice_transport_->GetRttEstimate();
+ std::optional<int> rtt = ice_transport_->GetRttEstimate();
if (rtt) {
// Limit the timeout to a reasonable range in case the ICE RTT takes
// extreme values.
diff --git a/p2p/base/dtls_transport.h b/p2p/base/dtls_transport.h
index ceeaa84..02249c2 100644
--- a/p2p/base/dtls_transport.h
+++ b/p2p/base/dtls_transport.h
@@ -143,7 +143,7 @@
absl::string_view digest_alg,
const uint8_t* digest,
size_t digest_len,
- absl::optional<rtc::SSLRole> role) override;
+ std::optional<rtc::SSLRole> role) override;
// Called to send a packet (via DTLS, if turned on).
int SendPacket(const char* data,
@@ -198,7 +198,7 @@
int GetError() override;
- absl::optional<rtc::NetworkRoute> network_route() const override;
+ std::optional<rtc::NetworkRoute> network_route() const override;
int SetOption(rtc::Socket::Option opt, int value) override;
@@ -222,7 +222,7 @@
void OnReadyToSend(rtc::PacketTransportInternal* transport);
void OnReceivingState(rtc::PacketTransportInternal* transport);
void OnDtlsEvent(int sig, int err);
- void OnNetworkRouteChanged(absl::optional<rtc::NetworkRoute> network_route);
+ void OnNetworkRouteChanged(std::optional<rtc::NetworkRoute> network_route);
bool SetupDtls();
void MaybeStartDtls();
bool HandleDtlsPacket(rtc::ArrayView<const uint8_t> payload);
@@ -246,7 +246,7 @@
const std::vector<int> srtp_ciphers_; // SRTP ciphers to use with DTLS.
bool dtls_active_ = false;
rtc::scoped_refptr<rtc::RTCCertificate> local_certificate_;
- absl::optional<rtc::SSLRole> dtls_role_;
+ std::optional<rtc::SSLRole> dtls_role_;
const rtc::SSLProtocolVersion ssl_max_version_;
rtc::Buffer remote_fingerprint_value_;
std::string remote_fingerprint_algorithm_;
diff --git a/p2p/base/dtls_transport_internal.h b/p2p/base/dtls_transport_internal.h
index cbcd2bd..548f59d 100644
--- a/p2p/base/dtls_transport_internal.h
+++ b/p2p/base/dtls_transport_internal.h
@@ -106,7 +106,7 @@
absl::string_view digest_alg,
const uint8_t* digest,
size_t digest_len,
- absl::optional<rtc::SSLRole> role) = 0;
+ std::optional<rtc::SSLRole> role) = 0;
ABSL_DEPRECATED("Set the max version via construction.")
bool SetSslMaxProtocolVersion(rtc::SSLProtocolVersion version) {
diff --git a/p2p/base/dtls_transport_unittest.cc b/p2p/base/dtls_transport_unittest.cc
index 9e2d069..767fc4d 100644
--- a/p2p/base/dtls_transport_unittest.cc
+++ b/p2p/base/dtls_transport_unittest.cc
@@ -66,7 +66,7 @@
->SetRemoteParameters(
fingerprint->algorithm,
reinterpret_cast<const uint8_t*>(fingerprint->digest.data()),
- fingerprint->digest.size(), absl::nullopt)
+ fingerprint->digest.size(), std::nullopt)
.ok());
}
diff --git a/p2p/base/fake_dtls_transport.h b/p2p/base/fake_dtls_transport.h
index a088030..bbcbe43 100644
--- a/p2p/base/fake_dtls_transport.h
+++ b/p2p/base/fake_dtls_transport.h
@@ -152,7 +152,7 @@
webrtc::RTCError SetRemoteParameters(absl::string_view alg,
const uint8_t* digest,
size_t digest_len,
- absl::optional<rtc::SSLRole> role) {
+ std::optional<rtc::SSLRole> role) {
if (role) {
SetDtlsRole(*role);
}
@@ -210,7 +210,7 @@
}
return false;
}
- void SetSslCipherSuite(absl::optional<int> cipher_suite) {
+ void SetSslCipherSuite(std::optional<int> cipher_suite) {
ssl_cipher_suite_ = cipher_suite;
}
uint16_t GetSslPeerSignatureAlgorithm() const override { return 0; }
@@ -266,7 +266,7 @@
}
int GetError() override { return ice_transport_->GetError(); }
- absl::optional<rtc::NetworkRoute> network_route() const override {
+ std::optional<rtc::NetworkRoute> network_route() const override {
return ice_transport_->network_route();
}
@@ -295,7 +295,7 @@
SignalWritableState(this);
}
- void OnNetworkRouteChanged(absl::optional<rtc::NetworkRoute> network_route) {
+ void OnNetworkRouteChanged(std::optional<rtc::NetworkRoute> network_route) {
SignalNetworkRouteChanged(network_route);
}
@@ -309,9 +309,9 @@
bool do_dtls_ = false;
rtc::SSLProtocolVersion ssl_max_version_ = rtc::SSL_PROTOCOL_DTLS_12;
rtc::SSLFingerprint dtls_fingerprint_;
- absl::optional<rtc::SSLRole> dtls_role_;
+ std::optional<rtc::SSLRole> dtls_role_;
int crypto_suite_ = rtc::kSrtpAes128CmSha1_80;
- absl::optional<int> ssl_cipher_suite_;
+ std::optional<int> ssl_cipher_suite_;
webrtc::DtlsTransportState dtls_state_ = webrtc::DtlsTransportState::kNew;
diff --git a/p2p/base/fake_ice_transport.h b/p2p/base/fake_ice_transport.h
index 788299b..cc7ccc1 100644
--- a/p2p/base/fake_ice_transport.h
+++ b/p2p/base/fake_ice_transport.h
@@ -13,12 +13,12 @@
#include <map>
#include <memory>
+#include <optional>
#include <string>
#include <utility>
#include "absl/algorithm/container.h"
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/ice_transport_interface.h"
#include "api/task_queue/pending_task_safety_flag.h"
#include "api/units/time_delta.h"
@@ -270,12 +270,11 @@
return true;
}
- absl::optional<int> GetRttEstimate() override { return absl::nullopt; }
+ std::optional<int> GetRttEstimate() override { return std::nullopt; }
const Connection* selected_connection() const override { return nullptr; }
- absl::optional<const CandidatePair> GetSelectedCandidatePair()
- const override {
- return absl::nullopt;
+ std::optional<const CandidatePair> GetSelectedCandidatePair() const override {
+ return std::nullopt;
}
// Fake PacketTransportInternal implementation.
@@ -345,11 +344,11 @@
return last_sent_packet_;
}
- absl::optional<rtc::NetworkRoute> network_route() const override {
+ std::optional<rtc::NetworkRoute> network_route() const override {
RTC_DCHECK_RUN_ON(network_thread_);
return network_route_;
}
- void SetNetworkRoute(absl::optional<rtc::NetworkRoute> network_route) {
+ void SetNetworkRoute(std::optional<rtc::NetworkRoute> network_route) {
RTC_DCHECK_RUN_ON(network_thread_);
network_route_ = network_route;
SendTask(network_thread_, [this] {
@@ -402,9 +401,9 @@
IceParameters remote_ice_parameters_ RTC_GUARDED_BY(network_thread_);
IceMode remote_ice_mode_ RTC_GUARDED_BY(network_thread_) = ICEMODE_FULL;
size_t connection_count_ RTC_GUARDED_BY(network_thread_) = 0;
- absl::optional<webrtc::IceTransportState> transport_state_
+ std::optional<webrtc::IceTransportState> transport_state_
RTC_GUARDED_BY(network_thread_);
- absl::optional<IceTransportState> legacy_transport_state_
+ std::optional<IceTransportState> legacy_transport_state_
RTC_GUARDED_BY(network_thread_);
IceGatheringState gathering_state_ RTC_GUARDED_BY(network_thread_) =
kIceGatheringNew;
@@ -413,7 +412,7 @@
bool receiving_ RTC_GUARDED_BY(network_thread_) = false;
bool combine_outgoing_packets_ RTC_GUARDED_BY(network_thread_) = false;
rtc::CopyOnWriteBuffer send_packet_ RTC_GUARDED_BY(network_thread_);
- absl::optional<rtc::NetworkRoute> network_route_
+ std::optional<rtc::NetworkRoute> network_route_
RTC_GUARDED_BY(network_thread_);
std::map<rtc::Socket::Option, int> socket_options_
RTC_GUARDED_BY(network_thread_);
diff --git a/p2p/base/fake_packet_transport.h b/p2p/base/fake_packet_transport.h
index a7388f5..e747af5 100644
--- a/p2p/base/fake_packet_transport.h
+++ b/p2p/base/fake_packet_transport.h
@@ -91,10 +91,10 @@
const CopyOnWriteBuffer* last_sent_packet() { return &last_sent_packet_; }
- absl::optional<NetworkRoute> network_route() const override {
+ std::optional<NetworkRoute> network_route() const override {
return network_route_;
}
- void SetNetworkRoute(absl::optional<NetworkRoute> network_route) {
+ void SetNetworkRoute(std::optional<NetworkRoute> network_route) {
network_route_ = network_route;
SignalNetworkRouteChanged(network_route);
}
@@ -141,7 +141,7 @@
std::map<Socket::Option, int> options_;
int error_ = 0;
- absl::optional<NetworkRoute> network_route_;
+ std::optional<NetworkRoute> network_route_;
};
} // namespace rtc
diff --git a/p2p/base/ice_controller_interface.h b/p2p/base/ice_controller_interface.h
index fb421cf..011ca85 100644
--- a/p2p/base/ice_controller_interface.h
+++ b/p2p/base/ice_controller_interface.h
@@ -11,11 +11,11 @@
#ifndef P2P_BASE_ICE_CONTROLLER_INTERFACE_H_
#define P2P_BASE_ICE_CONTROLLER_INTERFACE_H_
+#include <optional>
#include <string>
#include <utility>
#include <vector>
-#include "absl/types/optional.h"
#include "p2p/base/connection.h"
#include "p2p/base/ice_switch_reason.h"
#include "p2p/base/ice_transport_internal.h"
@@ -64,10 +64,10 @@
// This represents the result of a switch call.
struct SwitchResult {
// Connection that we should (optionally) switch to.
- absl::optional<const Connection*> connection;
+ std::optional<const Connection*> connection;
// An optional recheck event for when a Switch() should be attempted again.
- absl::optional<IceRecheckEvent> recheck_event;
+ std::optional<IceRecheckEvent> recheck_event;
// A vector with connection to run ForgetLearnedState on.
std::vector<const Connection*> connections_to_forget_state_on;
@@ -76,12 +76,12 @@
// This represents the result of a call to SelectConnectionToPing.
struct PingResult {
PingResult(const Connection* conn, int _recheck_delay_ms)
- : connection(conn ? absl::optional<const Connection*>(conn)
- : absl::nullopt),
+ : connection(conn ? std::optional<const Connection*>(conn)
+ : std::nullopt),
recheck_delay_ms(_recheck_delay_ms) {}
// Connection that we should (optionally) ping.
- const absl::optional<const Connection*> connection;
+ const std::optional<const Connection*> connection;
// The delay before P2PTransportChannel shall call SelectConnectionToPing()
// again.
diff --git a/p2p/base/ice_transport_internal.h b/p2p/base/ice_transport_internal.h
index b2a4690..ade567c 100644
--- a/p2p/base/ice_transport_internal.h
+++ b/p2p/base/ice_transport_internal.h
@@ -13,12 +13,12 @@
#include <stdint.h>
+#include <optional>
#include <string>
#include <utility>
#include <vector>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/candidate.h"
#include "api/rtc_error.h"
#include "api/transport/enums.h"
@@ -103,16 +103,16 @@
webrtc::RTCError VerifyCandidates(const Candidates& candidates);
// Information about ICE configuration.
-// TODO(deadbeef): Use absl::optional to represent unset values, instead of
+// TODO(deadbeef): Use std::optional to represent unset values, instead of
// -1.
//
// TODO(bugs.webrtc.org/15609): Define a public API for this.
struct RTC_EXPORT IceConfig {
// The ICE connection receiving timeout value in milliseconds.
- absl::optional<int> receiving_timeout;
+ std::optional<int> receiving_timeout;
// Time interval in milliseconds to ping a backup connection when the ICE
// channel is strongly connected.
- absl::optional<int> backup_connection_ping_interval;
+ std::optional<int> backup_connection_ping_interval;
ContinualGatheringPolicy continual_gathering_policy = GATHER_ONCE;
@@ -125,7 +125,7 @@
bool prioritize_most_likely_candidate_pairs = false;
// Writable connections are pinged at a slower rate once stablized.
- absl::optional<int> stable_writable_connection_ping_interval;
+ std::optional<int> stable_writable_connection_ping_interval;
// If set to true, this means the ICE transport should presume TURN-to-TURN
// candidate pairs will succeed, even before a binding response is received.
@@ -140,12 +140,12 @@
// Interval to check on all networks and to perform ICE regathering on any
// active network having no connection on it.
- absl::optional<int> regather_on_failed_networks_interval;
+ std::optional<int> regather_on_failed_networks_interval;
// The time period in which we will not switch the selected connection
// when a new connection becomes receiving but the selected connection is not
// in case that the selected connection may become receiving soon.
- absl::optional<int> receiving_switching_delay;
+ std::optional<int> receiving_switching_delay;
// TODO(honghaiz): Change the default to regular nomination.
// Default nomination mode if the remote does not support renomination.
@@ -155,12 +155,12 @@
// for a candidate pair when it is both writable and receiving (strong
// connectivity). This parameter overrides the default value given by
// `STRONG_PING_INTERVAL` in p2ptransport.h if set.
- absl::optional<int> ice_check_interval_strong_connectivity;
+ std::optional<int> ice_check_interval_strong_connectivity;
// The interval in milliseconds at which ICE checks (STUN pings) will be sent
// for a candidate pair when it is either not writable or not receiving (weak
// connectivity). This parameter overrides the default value given by
// `WEAK_PING_INTERVAL` in p2ptransport.h if set.
- absl::optional<int> ice_check_interval_weak_connectivity;
+ std::optional<int> ice_check_interval_weak_connectivity;
// ICE checks (STUN pings) will not be sent at higher rate (lower interval)
// than this, no matter what other settings there are.
// Measure in milliseconds.
@@ -168,30 +168,30 @@
// Note that this parameter overrides both the above check intervals for
// candidate pairs with strong or weak connectivity, if either of the above
// interval is shorter than the min interval.
- absl::optional<int> ice_check_min_interval;
+ std::optional<int> ice_check_min_interval;
// The min time period for which a candidate pair must wait for response to
// connectivity checks before it becomes unwritable. This parameter
// overrides the default value given by `CONNECTION_WRITE_CONNECT_TIMEOUT`
// in port.h if set, when determining the writability of a candidate pair.
- absl::optional<int> ice_unwritable_timeout;
+ std::optional<int> ice_unwritable_timeout;
// The min number of connectivity checks that a candidate pair must sent
// without receiving response before it becomes unwritable. This parameter
// overrides the default value given by `CONNECTION_WRITE_CONNECT_FAILURES` in
// port.h if set, when determining the writability of a candidate pair.
- absl::optional<int> ice_unwritable_min_checks;
+ std::optional<int> ice_unwritable_min_checks;
// The min time period for which a candidate pair must wait for response to
// connectivity checks it becomes inactive. This parameter overrides the
// default value given by `CONNECTION_WRITE_TIMEOUT` in port.h if set, when
// determining the writability of a candidate pair.
- absl::optional<int> ice_inactive_timeout;
+ std::optional<int> ice_inactive_timeout;
// The interval in milliseconds at which STUN candidates will resend STUN
// binding requests to keep NAT bindings open.
- absl::optional<int> stun_keepalive_interval;
+ std::optional<int> stun_keepalive_interval;
- absl::optional<rtc::AdapterType> network_preference;
+ std::optional<rtc::AdapterType> network_preference;
webrtc::VpnPreference vpn_preference = webrtc::VpnPreference::kDefault;
@@ -208,7 +208,7 @@
// Helper getters for parameters with implementation-specific default value.
// By convention, parameters with default value are represented by
- // absl::optional and setting a parameter to null restores its default value.
+ // std::optional and setting a parameter to null restores its default value.
int receiving_timeout_or_default() const;
int backup_connection_ping_interval_or_default() const;
int stable_writable_connection_ping_interval_or_default() const;
@@ -291,20 +291,20 @@
virtual bool GetStats(IceTransportStats* ice_transport_stats) = 0;
// Returns RTT estimate over the currently active connection, or an empty
- // absl::optional if there is none.
- virtual absl::optional<int> GetRttEstimate() = 0;
+ // std::optional if there is none.
+ virtual std::optional<int> GetRttEstimate() = 0;
// TODO(qingsi): Remove this method once Chrome does not depend on it anymore.
virtual const Connection* selected_connection() const = 0;
- // Returns the selected candidate pair, or an empty absl::optional if there is
+ // Returns the selected candidate pair, or an empty std::optional if there is
// none.
- virtual absl::optional<const CandidatePair> GetSelectedCandidatePair()
+ virtual std::optional<const CandidatePair> GetSelectedCandidatePair()
const = 0;
- virtual absl::optional<std::reference_wrapper<StunDictionaryWriter>>
+ virtual std::optional<std::reference_wrapper<StunDictionaryWriter>>
GetDictionaryWriter() {
- return absl::nullopt;
+ return std::nullopt;
}
void AddGatheringStateCallback(
diff --git a/p2p/base/mock_ice_transport.h b/p2p/base/mock_ice_transport.h
index b7b986e..a9046e1 100644
--- a/p2p/base/mock_ice_transport.h
+++ b/p2p/base/mock_ice_transport.h
@@ -63,11 +63,10 @@
void SetRemoteIceParameters(const IceParameters& ice_params) override {}
void SetRemoteIceMode(IceMode mode) override {}
void SetIceConfig(const IceConfig& config) override {}
- absl::optional<int> GetRttEstimate() override { return absl::nullopt; }
+ std::optional<int> GetRttEstimate() override { return std::nullopt; }
const Connection* selected_connection() const override { return nullptr; }
- absl::optional<const CandidatePair> GetSelectedCandidatePair()
- const override {
- return absl::nullopt;
+ std::optional<const CandidatePair> GetSelectedCandidatePair() const override {
+ return std::nullopt;
}
void MaybeStartGathering() override {}
void AddRemoteCandidate(const Candidate& candidate) override {}
diff --git a/p2p/base/p2p_transport_channel.cc b/p2p/base/p2p_transport_channel.cc
index 2116a3f..e194b9f 100644
--- a/p2p/base/p2p_transport_channel.cc
+++ b/p2p/base/p2p_transport_channel.cc
@@ -350,21 +350,21 @@
return gathering_state_;
}
-absl::optional<int> P2PTransportChannel::GetRttEstimate() {
+std::optional<int> P2PTransportChannel::GetRttEstimate() {
RTC_DCHECK_RUN_ON(network_thread_);
if (selected_connection_ != nullptr &&
selected_connection_->rtt_samples() > 0) {
return selected_connection_->rtt();
} else {
- return absl::nullopt;
+ return std::nullopt;
}
}
-absl::optional<const CandidatePair>
+std::optional<const CandidatePair>
P2PTransportChannel::GetSelectedCandidatePair() const {
RTC_DCHECK_RUN_ON(network_thread_);
if (selected_connection_ == nullptr) {
- return absl::nullopt;
+ return std::nullopt;
}
CandidatePair pair;
@@ -492,9 +492,9 @@
remote_ice_mode_ = mode;
}
-// TODO(qingsi): We apply the convention that setting a absl::optional parameter
+// TODO(qingsi): We apply the convention that setting a std::optional parameter
// to null restores its default value in the implementation. However, some
-// absl::optional parameters are only processed below if non-null, e.g.,
+// std::optional parameters are only processed below if non-null, e.g.,
// regather_on_failed_networks_interval, and thus there is no way to restore the
// defaults. Fix this issue later for consistency.
void P2PTransportChannel::SetIceConfig(const IceConfig& config) {
@@ -1631,7 +1631,7 @@
return true;
}
-absl::optional<rtc::NetworkRoute> P2PTransportChannel::network_route() const {
+std::optional<rtc::NetworkRoute> P2PTransportChannel::network_route() const {
RTC_DCHECK_RUN_ON(network_thread_);
return network_route_;
}
diff --git a/p2p/base/p2p_transport_channel.h b/p2p/base/p2p_transport_channel.h
index 2750af1..e1f5066 100644
--- a/p2p/base/p2p_transport_channel.h
+++ b/p2p/base/p2p_transport_channel.h
@@ -26,6 +26,7 @@
#include <algorithm>
#include <map>
#include <memory>
+#include <optional>
#include <set>
#include <string>
#include <utility>
@@ -33,7 +34,6 @@
#include "absl/base/attributes.h"
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/async_dns_resolver.h"
#include "api/candidate.h"
@@ -144,7 +144,7 @@
// only update the parameter if it is considered set in `config`. For example,
// a negative value of receiving_timeout will be considered "not set" and we
// will not use it to update the respective parameter in `config_`.
- // TODO(deadbeef): Use absl::optional instead of negative values.
+ // TODO(deadbeef): Use std::optional instead of negative values.
void SetIceConfig(const IceConfig& config) override;
const IceConfig& config() const;
static webrtc::RTCError ValidateIceConfig(const IceConfig& config);
@@ -158,9 +158,9 @@
bool GetOption(rtc::Socket::Option opt, int* value) override;
int GetError() override;
bool GetStats(IceTransportStats* ice_transport_stats) override;
- absl::optional<int> GetRttEstimate() override;
+ std::optional<int> GetRttEstimate() override;
const Connection* selected_connection() const override;
- absl::optional<const CandidatePair> GetSelectedCandidatePair() const override;
+ std::optional<const CandidatePair> GetSelectedCandidatePair() const override;
// From IceAgentInterface
void OnStartedPinging() override;
@@ -205,7 +205,7 @@
void PruneAllPorts();
int check_receiving_interval() const;
- absl::optional<rtc::NetworkRoute> network_route() const override;
+ std::optional<rtc::NetworkRoute> network_route() const override;
void RemoveConnection(Connection* connection);
@@ -245,7 +245,7 @@
return ss.Release();
}
- absl::optional<std::reference_wrapper<StunDictionaryWriter>>
+ std::optional<std::reference_wrapper<StunDictionaryWriter>>
GetDictionaryWriter() override {
return stun_dict_writer_;
}
@@ -460,7 +460,7 @@
bool has_been_writable_ RTC_GUARDED_BY(network_thread_) =
false; // if writable_ has ever been true
- absl::optional<rtc::NetworkRoute> network_route_
+ std::optional<rtc::NetworkRoute> network_route_
RTC_GUARDED_BY(network_thread_);
webrtc::IceEventLog ice_event_log_ RTC_GUARDED_BY(network_thread_);
diff --git a/p2p/base/p2p_transport_channel_ice_field_trials.h b/p2p/base/p2p_transport_channel_ice_field_trials.h
index 96a7756..f5573b1 100644
--- a/p2p/base/p2p_transport_channel_ice_field_trials.h
+++ b/p2p/base/p2p_transport_channel_ice_field_trials.h
@@ -11,7 +11,7 @@
#ifndef P2P_BASE_P2P_TRANSPORT_CHANNEL_ICE_FIELD_TRIALS_H_
#define P2P_BASE_P2P_TRANSPORT_CHANNEL_ICE_FIELD_TRIALS_H_
-#include "absl/types/optional.h"
+#include <optional>
namespace cricket {
@@ -23,17 +23,17 @@
// TODO(jonaso) : Consider how members of this struct can be made const.
bool skip_relay_to_non_relay_connections = false;
- absl::optional<int> max_outstanding_pings;
+ std::optional<int> max_outstanding_pings;
// Wait X ms before selecting a connection when having none.
// This will make media slower, but will give us chance to find
// a better connection before starting.
- absl::optional<int> initial_select_dampening;
+ std::optional<int> initial_select_dampening;
// If the connection has recevied a ping-request, delay by
// maximum this delay. This will make media slower, but will
// give us chance to find a better connection before starting.
- absl::optional<int> initial_select_dampening_ping_received;
+ std::optional<int> initial_select_dampening_ping_received;
// Announce GOOG_PING support in STUN_BINDING_RESPONSE if requested
// by peer.
@@ -66,7 +66,7 @@
bool stop_gather_on_strongly_connected = true;
// DSCP taging.
- absl::optional<int> override_dscp;
+ std::optional<int> override_dscp;
bool piggyback_ice_check_acknowledgement = false;
bool extra_ice_ping = false;
diff --git a/p2p/base/p2p_transport_channel_unittest.cc b/p2p/base/p2p_transport_channel_unittest.cc
index cad0a08..fb0eae3 100644
--- a/p2p/base/p2p_transport_channel_unittest.cc
+++ b/p2p/base/p2p_transport_channel_unittest.cc
@@ -137,7 +137,7 @@
cricket::IceConfig CreateIceConfig(
int receiving_timeout,
cricket::ContinualGatheringPolicy continual_gathering_policy,
- absl::optional<int> backup_ping_interval = absl::nullopt) {
+ std::optional<int> backup_ping_interval = std::nullopt) {
cricket::IceConfig config;
config.receiving_timeout = receiving_timeout;
config.continual_gathering_policy = continual_gathering_policy;
@@ -541,8 +541,8 @@
const SocketAddress& addr,
absl::string_view ifname,
rtc::AdapterType adapter_type,
- absl::optional<rtc::AdapterType> underlying_vpn_adapter_type =
- absl::nullopt) {
+ std::optional<rtc::AdapterType> underlying_vpn_adapter_type =
+ std::nullopt) {
GetEndpoint(endpoint)->network_manager_.AddInterface(
addr, ifname, adapter_type, underlying_vpn_adapter_type);
}
@@ -793,7 +793,7 @@
}
}
- void OnNetworkRouteChanged(absl::optional<rtc::NetworkRoute> network_route) {
+ void OnNetworkRouteChanged(std::optional<rtc::NetworkRoute> network_route) {
// If the `network_route` is unset, don't count. This is used in the case
// when the network on remote side is down, the signal will be fired with an
// unset network route and it shouldn't trigger a connection switch.
@@ -3392,7 +3392,7 @@
conn->SignalNominated(conn);
}
- void OnNetworkRouteChanged(absl::optional<rtc::NetworkRoute> network_route) {
+ void OnNetworkRouteChanged(std::optional<rtc::NetworkRoute> network_route) {
last_network_route_ = network_route;
if (last_network_route_) {
last_sent_packet_id_ = last_network_route_->last_sent_packet_id;
@@ -3405,7 +3405,7 @@
absl::string_view remote_ufrag,
int priority,
uint32_t nomination,
- const absl::optional<std::string>& piggyback_ping_id) {
+ const std::optional<std::string>& piggyback_ping_id) {
IceMessage msg(STUN_BINDING_REQUEST);
msg.AddAttribute(std::make_unique<StunByteStringAttribute>(
STUN_ATTR_USERNAME,
@@ -3434,7 +3434,7 @@
int priority,
uint32_t nomination = 0) {
ReceivePingOnConnection(conn, remote_ufrag, priority, nomination,
- absl::nullopt);
+ std::nullopt);
}
void OnReadyToSend(rtc::PacketTransportInternal* transport) {
@@ -3509,9 +3509,9 @@
int selected_candidate_pair_switches_ = 0;
int last_sent_packet_id_ = -1;
bool channel_ready_to_send_ = false;
- absl::optional<CandidatePairChangeEvent> last_candidate_change_event_;
+ std::optional<CandidatePairChangeEvent> last_candidate_change_event_;
IceTransportState channel_state_ = IceTransportState::STATE_INIT;
- absl::optional<rtc::NetworkRoute> last_network_route_;
+ std::optional<rtc::NetworkRoute> last_network_route_;
};
TEST_F(P2PTransportChannelPingTest, TestTriggeredChecks) {
diff --git a/p2p/base/packet_transport_internal.cc b/p2p/base/packet_transport_internal.cc
index 36f797a..f1053b3 100644
--- a/p2p/base/packet_transport_internal.cc
+++ b/p2p/base/packet_transport_internal.cc
@@ -23,8 +23,8 @@
return false;
}
-absl::optional<NetworkRoute> PacketTransportInternal::network_route() const {
- return absl::optional<NetworkRoute>();
+std::optional<NetworkRoute> PacketTransportInternal::network_route() const {
+ return std::optional<NetworkRoute>();
}
void PacketTransportInternal::RegisterReceivedPacketCallback(
diff --git a/p2p/base/packet_transport_internal.h b/p2p/base/packet_transport_internal.h
index 54d7f07..3a9a871 100644
--- a/p2p/base/packet_transport_internal.h
+++ b/p2p/base/packet_transport_internal.h
@@ -11,12 +11,12 @@
#ifndef P2P_BASE_PACKET_TRANSPORT_INTERNAL_H_
#define P2P_BASE_PACKET_TRANSPORT_INTERNAL_H_
+#include <optional>
#include <string>
#include <utility>
#include <vector>
#include "absl/functional/any_invocable.h"
-#include "absl/types/optional.h"
#include "p2p/base/port.h"
#include "rtc_base/async_packet_socket.h"
#include "rtc_base/callback_list.h"
@@ -67,7 +67,7 @@
// Returns the current network route with transport overhead.
// TODO(zhihuang): Make it pure virtual once the Chrome/remoting is updated.
- virtual absl::optional<NetworkRoute> network_route() const;
+ virtual std::optional<NetworkRoute> network_route() const;
// Emitted when the writable state, represented by `writable()`, changes.
sigslot::signal1<PacketTransportInternal*> SignalWritableState;
@@ -95,7 +95,7 @@
SignalSentPacket;
// Signalled when the current network route has changed.
- sigslot::signal1<absl::optional<rtc::NetworkRoute>> SignalNetworkRouteChanged;
+ sigslot::signal1<std::optional<rtc::NetworkRoute>> SignalNetworkRouteChanged;
// Signalled when the transport is closed.
void SetOnCloseCallback(absl::AnyInvocable<void() &&> callback);
diff --git a/p2p/base/packet_transport_internal_unittest.cc b/p2p/base/packet_transport_internal_unittest.cc
index 25492ea..178c273 100644
--- a/p2p/base/packet_transport_internal_unittest.cc
+++ b/p2p/base/packet_transport_internal_unittest.cc
@@ -35,7 +35,7 @@
rtc::ReceivedPacket::kDtlsDecrypted);
});
packet_transport.NotifyPacketReceived(rtc::ReceivedPacket(
- {}, rtc::SocketAddress(), absl::nullopt, rtc::EcnMarking::kNotEct,
+ {}, rtc::SocketAddress(), std::nullopt, rtc::EcnMarking::kNotEct,
rtc::ReceivedPacket::kDtlsDecrypted));
packet_transport.DeregisterReceivedPacketCallback(&receiver);
diff --git a/p2p/base/port.cc b/p2p/base/port.cc
index 25105dc..d7ba32f 100644
--- a/p2p/base/port.cc
+++ b/p2p/base/port.cc
@@ -80,13 +80,13 @@
return PROTO_NAMES[proto];
}
-absl::optional<ProtocolType> StringToProto(absl::string_view proto_name) {
+std::optional<ProtocolType> StringToProto(absl::string_view proto_name) {
for (size_t i = 0; i <= PROTO_LAST; ++i) {
if (absl::EqualsIgnoreCase(PROTO_NAMES[i], proto_name)) {
return static_cast<ProtocolType>(i);
}
}
- return absl::nullopt;
+ return std::nullopt;
}
// RFC 6544, TCP candidate encoding rules.
diff --git a/p2p/base/port.h b/p2p/base/port.h
index 2df1d35..f4c87c6 100644
--- a/p2p/base/port.h
+++ b/p2p/base/port.h
@@ -17,6 +17,7 @@
#include <functional>
#include <map>
#include <memory>
+#include <optional>
#include <set>
#include <string>
#include <type_traits>
@@ -25,7 +26,6 @@
#include "absl/base/attributes.h"
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/candidate.h"
#include "api/field_trials_view.h"
#include "api/packet_socket_factory.h"
@@ -102,7 +102,7 @@
CandidateStats(const CandidateStats&) = default;
CandidateStats(CandidateStats&&) = default;
CandidateStats(Candidate candidate,
- absl::optional<StunStats> stats = absl::nullopt)
+ std::optional<StunStats> stats = std::nullopt)
: candidate_(std::move(candidate)), stun_stats_(std::move(stats)) {}
~CandidateStats() = default;
@@ -110,18 +110,18 @@
const Candidate& candidate() const { return candidate_; }
- const absl::optional<StunStats>& stun_stats() const { return stun_stats_; }
+ const std::optional<StunStats>& stun_stats() const { return stun_stats_; }
private:
Candidate candidate_;
// STUN port stats if this candidate is a STUN candidate.
- absl::optional<StunStats> stun_stats_;
+ std::optional<StunStats> stun_stats_;
};
typedef std::vector<CandidateStats> CandidateStatsList;
const char* ProtoToString(ProtocolType proto);
-absl::optional<ProtocolType> StringToProto(absl::string_view proto_name);
+std::optional<ProtocolType> StringToProto(absl::string_view proto_name);
struct ProtocolAddress {
rtc::SocketAddress address;
@@ -363,7 +363,7 @@
int16_t network_cost() const override { return network_cost_; }
- void GetStunStats(absl::optional<StunStats>* stats) override {}
+ void GetStunStats(std::optional<StunStats>* stats) override {}
protected:
void UpdateNetworkCost() override;
diff --git a/p2p/base/port_allocator.cc b/p2p/base/port_allocator.cc
index 5694007..6eab1d3 100644
--- a/p2p/base/port_allocator.cc
+++ b/p2p/base/port_allocator.cc
@@ -126,7 +126,7 @@
int candidate_pool_size,
bool prune_turn_ports,
webrtc::TurnCustomizer* turn_customizer,
- const absl::optional<int>& stun_candidate_keepalive_interval) {
+ const std::optional<int>& stun_candidate_keepalive_interval) {
webrtc::PortPrunePolicy turn_port_prune_policy =
prune_turn_ports ? webrtc::PRUNE_BASED_ON_PRIORITY : webrtc::NO_PRUNE;
return SetConfiguration(stun_servers, turn_servers, candidate_pool_size,
@@ -140,7 +140,7 @@
int candidate_pool_size,
webrtc::PortPrunePolicy turn_port_prune_policy,
webrtc::TurnCustomizer* turn_customizer,
- const absl::optional<int>& stun_candidate_keepalive_interval) {
+ const std::optional<int>& stun_candidate_keepalive_interval) {
RTC_DCHECK_GE(candidate_pool_size, 0);
RTC_DCHECK_LE(candidate_pool_size, static_cast<int>(UINT16_MAX));
CheckRunOnValidThreadIfInitialized();
diff --git a/p2p/base/port_allocator.h b/p2p/base/port_allocator.h
index 5aba5f2..880d3ae 100644
--- a/p2p/base/port_allocator.h
+++ b/p2p/base/port_allocator.h
@@ -15,11 +15,11 @@
#include <deque>
#include <memory>
+#include <optional>
#include <string>
#include <vector>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/candidate.h"
#include "api/sequence_checker.h"
#include "api/transport/enums.h"
@@ -254,7 +254,7 @@
// The default value of the interval in implementation is restored if a null
// optional value is passed.
virtual void SetStunKeepaliveIntervalForReadyPorts(
- const absl::optional<int>& stun_keepalive_interval) {}
+ const std::optional<int>& stun_keepalive_interval) {}
// Another way of getting the information provided by the signals below.
//
// Ports and candidates are not guaranteed to be in the same order as the
@@ -368,15 +368,15 @@
int candidate_pool_size,
bool prune_turn_ports,
webrtc::TurnCustomizer* turn_customizer = nullptr,
- const absl::optional<int>&
- stun_candidate_keepalive_interval = absl::nullopt);
+ const std::optional<int>&
+ stun_candidate_keepalive_interval = std::nullopt);
bool SetConfiguration(const ServerAddresses& stun_servers,
const std::vector<RelayServerConfig>& turn_servers,
int candidate_pool_size,
webrtc::PortPrunePolicy turn_port_prune_policy,
webrtc::TurnCustomizer* turn_customizer = nullptr,
- const absl::optional<int>&
- stun_candidate_keepalive_interval = absl::nullopt);
+ const std::optional<int>&
+ stun_candidate_keepalive_interval = std::nullopt);
const ServerAddresses& stun_servers() const {
CheckRunOnValidThreadIfInitialized();
@@ -393,7 +393,7 @@
return candidate_pool_size_;
}
- const absl::optional<int>& stun_candidate_keepalive_interval() const {
+ const std::optional<int>& stun_candidate_keepalive_interval() const {
CheckRunOnValidThreadIfInitialized();
return stun_candidate_keepalive_interval_;
}
@@ -630,7 +630,7 @@
// all TurnPort(s) created.
webrtc::TurnCustomizer* turn_customizer_ = nullptr;
- absl::optional<int> stun_candidate_keepalive_interval_;
+ std::optional<int> stun_candidate_keepalive_interval_;
// If true, TakePooledSession() will only return sessions that has same ice
// credentials as requested.
diff --git a/p2p/base/port_interface.h b/p2p/base/port_interface.h
index 34f835d..0d89a9e 100644
--- a/p2p/base/port_interface.h
+++ b/p2p/base/port_interface.h
@@ -12,12 +12,12 @@
#define P2P_BASE_PORT_INTERFACE_H_
#include <memory>
+#include <optional>
#include <string>
#include <utility>
#include <vector>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/candidate.h"
#include "api/field_trials_view.h"
#include "api/packet_socket_factory.h"
@@ -139,7 +139,7 @@
virtual std::string ToString() const = 0;
- virtual void GetStunStats(absl::optional<StunStats>* stats) = 0;
+ virtual void GetStunStats(std::optional<StunStats>* stats) = 0;
// Removes and deletes a connection object. `DestroyConnection` will
// delete the connection object directly whereas `DestroyConnectionAsync`
diff --git a/p2p/base/port_unittest.cc b/p2p/base/port_unittest.cc
index 48d52da..9b05282 100644
--- a/p2p/base/port_unittest.cc
+++ b/p2p/base/port_unittest.cc
@@ -16,13 +16,13 @@
#include <limits>
#include <list>
#include <memory>
+#include <optional>
#include <string>
#include <utility>
#include <vector>
#include "absl/memory/memory.h"
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/candidate.h"
#include "api/packet_socket_factory.h"
@@ -260,13 +260,13 @@
ASSERT_TRUE_WAIT(lport->last_stun_msg(), kDefaultTimeout);
ASSERT_GT(lport->last_stun_buf().size(), 0u);
rconn->OnReadPacket(rtc::ReceivedPacket(lport->last_stun_buf(),
- rtc::SocketAddress(), absl::nullopt));
+ rtc::SocketAddress(), std::nullopt));
clock->AdvanceTime(webrtc::TimeDelta::Millis(ms));
ASSERT_TRUE_WAIT(rport->last_stun_msg(), kDefaultTimeout);
ASSERT_GT(rport->last_stun_buf().size(), 0u);
lconn->OnReadPacket(rtc::ReceivedPacket(rport->last_stun_buf(),
- rtc::SocketAddress(), absl::nullopt));
+ rtc::SocketAddress(), std::nullopt));
}
class TestChannel : public sigslot::has_slots<> {
@@ -541,7 +541,7 @@
.ice_username_fragment = username_,
.ice_password = password_,
.field_trials = &field_trials_},
- 0, 0, true, absl::nullopt);
+ 0, 0, true, std::nullopt);
port->SetIceTiebreaker(kTiebreakerDefault);
return port;
}
@@ -557,7 +557,7 @@
.ice_username_fragment = username_,
.ice_password = password_,
.field_trials = &field_trials_},
- 0, 0, true, absl::nullopt);
+ 0, 0, true, std::nullopt);
port->SetIceTiebreaker(kTiebreakerDefault);
return port;
}
@@ -587,7 +587,7 @@
.ice_username_fragment = username_,
.ice_password = password_,
.field_trials = &field_trials_},
- 0, 0, stun_servers, absl::nullopt);
+ 0, 0, stun_servers, std::nullopt);
port->SetIceTiebreaker(kTiebreakerDefault);
return port;
}
@@ -1097,7 +1097,7 @@
private:
AsyncPacketSocket* next_udp_socket_;
AsyncListenSocket* next_server_tcp_socket_;
- absl::optional<AsyncPacketSocket*> next_client_tcp_socket_;
+ std::optional<AsyncPacketSocket*> next_client_tcp_socket_;
};
class FakeAsyncPacketSocket : public AsyncPacketSocket {
@@ -1513,7 +1513,7 @@
IceMessage* msg = lport->last_stun_msg();
EXPECT_EQ(STUN_BINDING_REQUEST, msg->type());
conn->OnReadPacket(rtc::ReceivedPacket(lport->last_stun_buf(),
- rtc::SocketAddress(), absl::nullopt));
+ rtc::SocketAddress(), std::nullopt));
ASSERT_TRUE_WAIT(lport->last_stun_msg() != NULL, kDefaultTimeout);
msg = lport->last_stun_msg();
EXPECT_EQ(STUN_BINDING_RESPONSE, msg->type());
@@ -1582,7 +1582,7 @@
EXPECT_EQ(STUN_BINDING_REQUEST, msg->type());
// Send rport binding request to lport.
lconn->OnReadPacket(rtc::ReceivedPacket(rport->last_stun_buf(),
- rtc::SocketAddress(), absl::nullopt));
+ rtc::SocketAddress(), std::nullopt));
ASSERT_TRUE_WAIT(lport->last_stun_msg() != NULL, kDefaultTimeout);
EXPECT_EQ(STUN_BINDING_RESPONSE, lport->last_stun_msg()->type());
@@ -1915,13 +1915,13 @@
// Receive the BINDING-REQUEST and respond with BINDING-RESPONSE.
rconn->OnReadPacket(rtc::ReceivedPacket(lport->last_stun_buf(),
- rtc::SocketAddress(), absl::nullopt));
+ rtc::SocketAddress(), std::nullopt));
msg = rport->last_stun_msg();
ASSERT_TRUE(msg != NULL);
EXPECT_EQ(STUN_BINDING_RESPONSE, msg->type());
// Received a BINDING-RESPONSE.
lconn->OnReadPacket(rtc::ReceivedPacket(rport->last_stun_buf(),
- rtc::SocketAddress(), absl::nullopt));
+ rtc::SocketAddress(), std::nullopt));
// Verify the STUN Stats.
EXPECT_EQ(1U, lconn->stats().sent_ping_requests_total);
@@ -2002,11 +2002,11 @@
// Respond with a BINDING-RESPONSE.
request = CopyStunMessage(*msg);
lconn->OnReadPacket(rtc::ReceivedPacket(rport->last_stun_buf(),
- rtc::SocketAddress(), absl::nullopt));
+ rtc::SocketAddress(), std::nullopt));
msg = lport->last_stun_msg();
// Receive the BINDING-RESPONSE.
rconn->OnReadPacket(rtc::ReceivedPacket(lport->last_stun_buf(),
- rtc::SocketAddress(), absl::nullopt));
+ rtc::SocketAddress(), std::nullopt));
// Verify the Stun ping stats.
EXPECT_EQ(3U, rconn->stats().sent_ping_requests_total);
@@ -2058,7 +2058,7 @@
ASSERT_TRUE_WAIT(lport->last_stun_msg(), kDefaultTimeout);
ASSERT_GT(lport->last_stun_buf().size(), 0u);
rconn->OnReadPacket(rtc::ReceivedPacket(lport->last_stun_buf(),
- rtc::SocketAddress(), absl::nullopt));
+ rtc::SocketAddress(), std::nullopt));
EXPECT_EQ(nomination, rconn->remote_nomination());
EXPECT_FALSE(lconn->nominated());
@@ -2071,7 +2071,7 @@
ASSERT_TRUE_WAIT(rport->last_stun_msg(), kDefaultTimeout);
ASSERT_GT(rport->last_stun_buf().size(), 0u);
lconn->OnReadPacket(rtc::ReceivedPacket(rport->last_stun_buf(),
- rtc::SocketAddress(), absl::nullopt));
+ rtc::SocketAddress(), std::nullopt));
EXPECT_EQ(nomination, lconn->acked_nomination());
EXPECT_TRUE(lconn->nominated());
@@ -2199,7 +2199,7 @@
EXPECT_EQ(STUN_BINDING_REQUEST, msg->type());
// Pass the binding request to rport.
rconn->OnReadPacket(rtc::ReceivedPacket(lport->last_stun_buf(),
- rtc::SocketAddress(), absl::nullopt));
+ rtc::SocketAddress(), std::nullopt));
// Wait until rport sends the response and then check the remote network cost.
ASSERT_TRUE_WAIT(rport->last_stun_msg() != NULL, kDefaultTimeout);
@@ -2530,7 +2530,7 @@
lconn->Ping(0);
ASSERT_TRUE_WAIT(lport->last_stun_msg() != NULL, kDefaultTimeout);
rconn->OnReadPacket(rtc::ReceivedPacket(lport->last_stun_buf(),
- rtc::SocketAddress(), absl::nullopt));
+ rtc::SocketAddress(), std::nullopt));
// Intercept request and add comprehension required attribute.
ASSERT_TRUE_WAIT(rport->last_stun_msg() != NULL, kDefaultTimeout);
@@ -2618,7 +2618,7 @@
EXPECT_EQ(STUN_BINDING_REQUEST, msg->type());
// Send rport binding request to lport.
lconn->OnReadPacket(rtc::ReceivedPacket(rport->last_stun_buf(),
- rtc::SocketAddress(), absl::nullopt));
+ rtc::SocketAddress(), std::nullopt));
ASSERT_TRUE_WAIT(lport->last_stun_msg() != NULL, kDefaultTimeout);
EXPECT_EQ(STUN_BINDING_RESPONSE, lport->last_stun_msg()->type());
@@ -3128,7 +3128,7 @@
// Feeding the respone message from litemode to the full mode connection.
ch1.conn()->OnReadPacket(rtc::ReceivedPacket(
- ice_lite_port->last_stun_buf(), rtc::SocketAddress(), absl::nullopt));
+ ice_lite_port->last_stun_buf(), rtc::SocketAddress(), std::nullopt));
// Verifying full mode connection becomes writable from the response.
EXPECT_EQ_WAIT(Connection::STATE_WRITABLE, ch1.conn()->write_state(),
@@ -3149,17 +3149,17 @@
namespace {
// Utility function for testing goog ping.
-absl::optional<int> GetSupportedGoogPingVersion(const StunMessage* msg) {
+std::optional<int> GetSupportedGoogPingVersion(const StunMessage* msg) {
auto goog_misc = msg->GetUInt16List(STUN_ATTR_GOOG_MISC_INFO);
if (goog_misc == nullptr) {
- return absl::nullopt;
+ return std::nullopt;
}
if (msg->type() == STUN_BINDING_REQUEST) {
if (goog_misc->Size() <
static_cast<int>(cricket::IceGoogMiscInfoBindingRequestAttributeIndex::
SUPPORT_GOOG_PING_VERSION)) {
- return absl::nullopt;
+ return std::nullopt;
}
return goog_misc->GetType(
@@ -3171,14 +3171,14 @@
if (goog_misc->Size() <
static_cast<int>(cricket::IceGoogMiscInfoBindingResponseAttributeIndex::
SUPPORT_GOOG_PING_VERSION)) {
- return absl::nullopt;
+ return std::nullopt;
}
return goog_misc->GetType(
static_cast<int>(cricket::IceGoogMiscInfoBindingResponseAttributeIndex::
SUPPORT_GOOG_PING_VERSION));
}
- return absl::nullopt;
+ return std::nullopt;
}
} // namespace
@@ -3246,7 +3246,7 @@
// Feeding the respone message back.
ch1.conn()->OnReadPacket(rtc::ReceivedPacket(
- port2->last_stun_buf(), rtc::SocketAddress(), absl::nullopt));
+ port2->last_stun_buf(), rtc::SocketAddress(), std::nullopt));
port1->Reset();
port2->Reset();
@@ -3503,7 +3503,7 @@
// Feeding the respone message back.
ch1.conn()->OnReadPacket(rtc::ReceivedPacket(
- port2->last_stun_buf(), rtc::SocketAddress(), absl::nullopt));
+ port2->last_stun_buf(), rtc::SocketAddress(), std::nullopt));
port1->Reset();
port2->Reset();
@@ -3588,7 +3588,7 @@
// Feeding the respone message back.
ch1.conn()->OnReadPacket(rtc::ReceivedPacket(
- port2->last_stun_buf(), rtc::SocketAddress(), absl::nullopt));
+ port2->last_stun_buf(), rtc::SocketAddress(), std::nullopt));
port1->Reset();
port2->Reset();
@@ -3854,7 +3854,7 @@
ASSERT_TRUE_WAIT(lport->last_stun_msg(), kDefaultTimeout);
ASSERT_GT(lport->last_stun_buf().size(), 0u);
rconn->OnReadPacket(rtc::ReceivedPacket(
- lport->last_stun_buf(), rtc::SocketAddress(), absl::nullopt));
+ lport->last_stun_buf(), rtc::SocketAddress(), std::nullopt));
clock_.AdvanceTime(webrtc::TimeDelta::Millis(ms));
ASSERT_TRUE_WAIT(rport->last_stun_msg(), kDefaultTimeout);
@@ -3869,7 +3869,7 @@
SendPingAndCaptureReply(lconn, rconn, ms, &reply);
lconn->OnReadPacket(
- rtc::ReceivedPacket(reply, rtc::SocketAddress(), absl::nullopt));
+ rtc::ReceivedPacket(reply, rtc::SocketAddress(), std::nullopt));
}
void OnConnectionStateChange(Connection* connection) { num_state_changes_++; }
@@ -3931,7 +3931,7 @@
EXPECT_FALSE(lconn->receiving());
lconn->OnReadPacket(
- rtc::ReceivedPacket(reply, rtc::SocketAddress(), absl::nullopt));
+ rtc::ReceivedPacket(reply, rtc::SocketAddress(), std::nullopt));
// That reply was discarded due to the ForgetLearnedState() while it was
// outstanding.
@@ -4004,14 +4004,14 @@
ASSERT_TRUE_WAIT(lport_->last_stun_msg(), kDefaultTimeout);
ASSERT_GT(lport_->last_stun_buf().size(), 0u);
rconn->OnReadPacket(rtc::ReceivedPacket(lport_->last_stun_buf(),
- rtc::SocketAddress(), absl::nullopt));
+ rtc::SocketAddress(), std::nullopt));
EXPECT_TRUE(received_goog_delta);
clock_.AdvanceTime(webrtc::TimeDelta::Millis(ms));
ASSERT_TRUE_WAIT(rport_->last_stun_msg(), kDefaultTimeout);
ASSERT_GT(rport_->last_stun_buf().size(), 0u);
lconn->OnReadPacket(rtc::ReceivedPacket(rport_->last_stun_buf(),
- rtc::SocketAddress(), absl::nullopt));
+ rtc::SocketAddress(), std::nullopt));
EXPECT_TRUE(received_goog_delta_ack);
}
@@ -4042,13 +4042,13 @@
ASSERT_TRUE_WAIT(lport_->last_stun_msg(), kDefaultTimeout);
ASSERT_GT(lport_->last_stun_buf().size(), 0u);
rconn->OnReadPacket(rtc::ReceivedPacket(lport_->last_stun_buf(),
- rtc::SocketAddress(), absl::nullopt));
+ rtc::SocketAddress(), std::nullopt));
clock_.AdvanceTime(webrtc::TimeDelta::Millis(ms));
ASSERT_TRUE_WAIT(rport_->last_stun_msg(), kDefaultTimeout);
ASSERT_GT(rport_->last_stun_buf().size(), 0u);
lconn->OnReadPacket(rtc::ReceivedPacket(rport_->last_stun_buf(),
- rtc::SocketAddress(), absl::nullopt));
+ rtc::SocketAddress(), std::nullopt));
EXPECT_TRUE(received_goog_delta_ack_error);
}
diff --git a/p2p/base/regathering_controller.h b/p2p/base/regathering_controller.h
index a0dfb80..8f704c1 100644
--- a/p2p/base/regathering_controller.h
+++ b/p2p/base/regathering_controller.h
@@ -74,7 +74,7 @@
void OnIceTransportStateChanged(cricket::IceTransportInternal*) {}
void OnIceTransportWritableState(rtc::PacketTransportInternal*) {}
void OnIceTransportReceivingState(rtc::PacketTransportInternal*) {}
- void OnIceTransportNetworkRouteChanged(absl::optional<rtc::NetworkRoute>) {}
+ void OnIceTransportNetworkRouteChanged(std::optional<rtc::NetworkRoute>) {}
// Schedules delayed and repeated regathering of local candidates on failed
// networks, where the delay in milliseconds is given by the config. Each
// repetition is separated by the same delay. When scheduled, all previous
diff --git a/p2p/base/stun_dictionary.cc b/p2p/base/stun_dictionary.cc
index 4ace826..b234f98 100644
--- a/p2p/base/stun_dictionary.cc
+++ b/p2p/base/stun_dictionary.cc
@@ -87,7 +87,7 @@
const StunAttribute* StunDictionaryView::GetOrNull(
int key,
- absl::optional<StunAttributeValueType> type) const {
+ std::optional<StunAttributeValueType> type) const {
const auto it = attrs_.find(key);
if (it == attrs_.end()) {
return nullptr;
diff --git a/p2p/base/stun_dictionary.h b/p2p/base/stun_dictionary.h
index f93a1f1..3442a91 100644
--- a/p2p/base/stun_dictionary.h
+++ b/p2p/base/stun_dictionary.h
@@ -81,7 +81,7 @@
const StunAttribute* GetOrNull(
int key,
- absl::optional<StunAttributeValueType> = absl::nullopt) const;
+ std::optional<StunAttributeValueType> = std::nullopt) const;
size_t GetLength(int key) const;
static webrtc::RTCErrorOr<
std::pair<uint64_t, std::deque<std::unique_ptr<StunAttribute>>>>
diff --git a/p2p/base/stun_port.cc b/p2p/base/stun_port.cc
index bdd8830..5a9906d 100644
--- a/p2p/base/stun_port.cc
+++ b/p2p/base/stun_port.cc
@@ -341,11 +341,11 @@
return PROTO_UDP;
}
-void UDPPort::GetStunStats(absl::optional<StunStats>* stats) {
+void UDPPort::GetStunStats(std::optional<StunStats>* stats) {
*stats = stats_;
}
-void UDPPort::set_stun_keepalive_delay(const absl::optional<int>& delay) {
+void UDPPort::set_stun_keepalive_delay(const std::optional<int>& delay) {
stun_keepalive_delay_ = delay.value_or(STUN_KEEPALIVE_INTERVAL);
}
@@ -614,7 +614,7 @@
uint16_t min_port,
uint16_t max_port,
const ServerAddresses& servers,
- absl::optional<int> stun_keepalive_interval) {
+ std::optional<int> stun_keepalive_interval) {
// Using `new` to access a non-public constructor.
auto port = absl::WrapUnique(new StunPort(args, min_port, max_port, servers));
port->set_stun_keepalive_delay(stun_keepalive_interval);
@@ -633,7 +633,7 @@
absl::string_view username,
absl::string_view password,
const ServerAddresses& servers,
- absl::optional<int> stun_keepalive_interval,
+ std::optional<int> stun_keepalive_interval,
const webrtc::FieldTrialsView* field_trials) {
return Create({.network_thread = thread,
.socket_factory = factory,
diff --git a/p2p/base/stun_port.h b/p2p/base/stun_port.h
index 3133a41..89617f0 100644
--- a/p2p/base/stun_port.h
+++ b/p2p/base/stun_port.h
@@ -39,7 +39,7 @@
const PortParametersRef& args,
rtc::AsyncPacketSocket* socket,
bool emit_local_for_anyaddress,
- absl::optional<int> stun_keepalive_interval) {
+ std::optional<int> stun_keepalive_interval) {
// Using `new` to access a non-public constructor.
auto port =
absl::WrapUnique(new UDPPort(args, webrtc::IceCandidateType::kHost,
@@ -59,7 +59,7 @@
absl::string_view username,
absl::string_view password,
bool emit_local_for_anyaddress,
- absl::optional<int> stun_keepalive_interval,
+ std::optional<int> stun_keepalive_interval,
const webrtc::FieldTrialsView* field_trials = nullptr) {
return Create({.network_thread = thread,
.socket_factory = factory,
@@ -75,7 +75,7 @@
uint16_t min_port,
uint16_t max_port,
bool emit_local_for_anyaddress,
- absl::optional<int> stun_keepalive_interval) {
+ std::optional<int> stun_keepalive_interval) {
// Using `new` to access a non-public constructor.
auto port = absl::WrapUnique(
new UDPPort(args, webrtc::IceCandidateType::kHost, min_port, max_port,
@@ -96,7 +96,7 @@
absl::string_view username,
absl::string_view password,
bool emit_local_for_anyaddress,
- absl::optional<int> stun_keepalive_interval,
+ std::optional<int> stun_keepalive_interval,
const webrtc::FieldTrialsView* field_trials = nullptr) {
return Create({.network_thread = thread,
.socket_factory = factory,
@@ -133,9 +133,9 @@
bool SupportsProtocol(absl::string_view protocol) const override;
ProtocolType GetProtocol() const override;
- void GetStunStats(absl::optional<StunStats>* stats) override;
+ void GetStunStats(std::optional<StunStats>* stats) override;
- void set_stun_keepalive_delay(const absl::optional<int>& delay);
+ void set_stun_keepalive_delay(const std::optional<int>& delay);
int stun_keepalive_delay() const { return stun_keepalive_delay_; }
// Visible for testing.
@@ -290,7 +290,7 @@
uint16_t min_port,
uint16_t max_port,
const ServerAddresses& servers,
- absl::optional<int> stun_keepalive_interval);
+ std::optional<int> stun_keepalive_interval);
[[deprecated("Pass arguments using PortParametersRef")]] static std::
unique_ptr<StunPort>
Create(webrtc::TaskQueueBase* thread,
@@ -301,7 +301,7 @@
absl::string_view username,
absl::string_view password,
const ServerAddresses& servers,
- absl::optional<int> stun_keepalive_interval,
+ std::optional<int> stun_keepalive_interval,
const webrtc::FieldTrialsView* field_trials);
void PrepareAddress() override;
diff --git a/p2p/base/stun_port_unittest.cc b/p2p/base/stun_port_unittest.cc
index 08c224a..0f8ecde 100644
--- a/p2p/base/stun_port_unittest.cc
+++ b/p2p/base/stun_port_unittest.cc
@@ -144,7 +144,7 @@
.ice_username_fragment = rtc::CreateRandomString(16),
.ice_password = rtc::CreateRandomString(22),
.field_trials = field_trials},
- 0, 0, stun_servers, absl::nullopt);
+ 0, 0, stun_servers, std::nullopt);
stun_port_->SetIceTiebreaker(kTiebreakerDefault);
stun_port_->set_stun_keepalive_delay(stun_keepalive_delay_);
// If `stun_keepalive_lifetime_` is negative, let the stun port
@@ -181,7 +181,7 @@
.ice_username_fragment = rtc::CreateRandomString(16),
.ice_password = rtc::CreateRandomString(22),
.field_trials = field_trials},
- socket_.get(), false, absl::nullopt);
+ socket_.get(), false, std::nullopt);
ASSERT_TRUE(stun_port_ != NULL);
stun_port_->SetIceTiebreaker(kTiebreakerDefault);
ServerAddresses stun_servers;
diff --git a/p2p/base/transport_description.cc b/p2p/base/transport_description.cc
index f3b1fbb..8f4ec01 100644
--- a/p2p/base/transport_description.cc
+++ b/p2p/base/transport_description.cc
@@ -109,7 +109,7 @@
return RTCError::OK();
}
-absl::optional<ConnectionRole> StringToConnectionRole(
+std::optional<ConnectionRole> StringToConnectionRole(
absl::string_view role_str) {
const char* const roles[] = {
CONNECTIONROLE_ACTIVE_STR, CONNECTIONROLE_PASSIVE_STR,
@@ -120,7 +120,7 @@
return static_cast<ConnectionRole>(CONNECTIONROLE_ACTIVE + i);
}
}
- return absl::nullopt;
+ return std::nullopt;
}
bool ConnectionRoleToString(const ConnectionRole& role, std::string* role_str) {
diff --git a/p2p/base/transport_description.h b/p2p/base/transport_description.h
index 53a1804..d1c161e 100644
--- a/p2p/base/transport_description.h
+++ b/p2p/base/transport_description.h
@@ -12,12 +12,12 @@
#define P2P_BASE_TRANSPORT_DESCRIPTION_H_
#include <memory>
+#include <optional>
#include <string>
#include <vector>
#include "absl/algorithm/container.h"
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/rtc_error.h"
#include "p2p/base/p2p_constants.h"
#include "rtc_base/ssl_fingerprint.h"
@@ -87,7 +87,7 @@
constexpr auto* ICE_OPTION_TRICKLE = "trickle";
constexpr auto* ICE_OPTION_RENOMINATION = "renomination";
-absl::optional<ConnectionRole> StringToConnectionRole(
+std::optional<ConnectionRole> StringToConnectionRole(
absl::string_view role_str);
bool ConnectionRoleToString(const ConnectionRole& role, std::string* role_str);
diff --git a/p2p/base/turn_port.cc b/p2p/base/turn_port.cc
index 4dba2a5..f52e1f2 100644
--- a/p2p/base/turn_port.cc
+++ b/p2p/base/turn_port.cc
@@ -13,13 +13,13 @@
#include <cstdint>
#include <functional>
#include <memory>
+#include <optional>
#include <utility>
#include <vector>
#include "absl/algorithm/container.h"
#include "absl/strings/match.h"
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/task_queue/pending_task_safety_flag.h"
#include "api/transport/stun.h"
#include "api/turn_customizer.h"
diff --git a/p2p/base/turn_port_unittest.cc b/p2p/base/turn_port_unittest.cc
index b0f3124..6c16a9d 100644
--- a/p2p/base/turn_port_unittest.cc
+++ b/p2p/base/turn_port_unittest.cc
@@ -19,10 +19,10 @@
#include <list>
#include <memory>
+#include <optional>
#include <utility>
#include <vector>
-#include "absl/types/optional.h"
#include "api/units/time_delta.h"
#include "p2p/base/basic_packet_socket_factory.h"
#include "p2p/base/connection.h"
@@ -372,7 +372,7 @@
.ice_username_fragment = kIceUfrag2,
.ice_password = kIcePwd2,
.field_trials = &field_trials_},
- 0, 0, false, absl::nullopt);
+ 0, 0, false, std::nullopt);
// UDP port will be controlled.
udp_port_->SetIceRole(ICEROLE_CONTROLLED);
udp_port_->SetIceTiebreaker(kTiebreakerDefault);
diff --git a/p2p/base/wrapping_active_ice_controller.h b/p2p/base/wrapping_active_ice_controller.h
index 449c0f0..bf5b8ea 100644
--- a/p2p/base/wrapping_active_ice_controller.h
+++ b/p2p/base/wrapping_active_ice_controller.h
@@ -12,8 +12,8 @@
#define P2P_BASE_WRAPPING_ACTIVE_ICE_CONTROLLER_H_
#include <memory>
+#include <optional>
-#include "absl/types/optional.h"
#include "api/task_queue/pending_task_safety_flag.h"
#include "p2p/base/active_ice_controller_interface.h"
#include "p2p/base/connection.h"
diff --git a/p2p/client/basic_port_allocator.cc b/p2p/client/basic_port_allocator.cc
index c8b293a..3e8ddf4 100644
--- a/p2p/client/basic_port_allocator.cc
+++ b/p2p/client/basic_port_allocator.cc
@@ -494,7 +494,7 @@
for (auto* port : ports) {
auto candidates = port->Candidates();
for (const auto& candidate : candidates) {
- absl::optional<StunStats> stun_stats;
+ std::optional<StunStats> stun_stats;
port->GetStunStats(&stun_stats);
CandidateStats candidate_stats(allocator_->SanitizeCandidate(candidate),
std::move(stun_stats));
@@ -504,7 +504,7 @@
}
void BasicPortAllocatorSession::SetStunKeepaliveIntervalForReadyPorts(
- const absl::optional<int>& stun_keepalive_interval) {
+ const std::optional<int>& stun_keepalive_interval) {
RTC_DCHECK_RUN_ON(network_thread_);
auto ports = ReadyPorts();
for (PortInterface* port : ports) {
diff --git a/p2p/client/basic_port_allocator.h b/p2p/client/basic_port_allocator.h
index 25201fd..3c275c9 100644
--- a/p2p/client/basic_port_allocator.h
+++ b/p2p/client/basic_port_allocator.h
@@ -151,7 +151,7 @@
void GetCandidateStatsFromReadyPorts(
CandidateStatsList* candidate_stats_list) const override;
void SetStunKeepaliveIntervalForReadyPorts(
- const absl::optional<int>& stun_keepalive_interval) override;
+ const std::optional<int>& stun_keepalive_interval) override;
void PruneAllPorts() override;
static std::vector<const rtc::Network*> SelectIPv6Networks(
std::vector<const rtc::Network*>& all_ipv6_networks,
diff --git a/pc/BUILD.gn b/pc/BUILD.gn
index fd94772..f16e2c1 100644
--- a/pc/BUILD.gn
+++ b/pc/BUILD.gn
@@ -108,7 +108,6 @@
"../rtc_base/third_party/sigslot",
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -142,7 +141,6 @@
"../rtc_base:checks",
"../rtc_base:logging",
"../rtc_base:ssl_adapter",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -169,7 +167,6 @@
"../rtc_base:ssl_adapter",
"../rtc_base:threading",
"../rtc_base/synchronization:mutex",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -251,7 +248,6 @@
"../rtc_base:ssl_adapter",
"../rtc_base:stringutils",
"../rtc_base:threading",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -339,7 +335,6 @@
"../rtc_base/third_party/sigslot",
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/functional:any_invocable",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -394,7 +389,6 @@
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -485,7 +479,6 @@
"../api:scoped_refptr",
"../api/crypto:frame_decryptor_interface",
"../api/transport/rtp:rtp_source",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -528,7 +521,6 @@
"../rtc_base/network:received_packet",
"../rtc_base/network:sent_packet",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -571,7 +563,6 @@
"../rtc_base:macromagic",
"../rtc_base:threading",
"../rtc_base/third_party/sigslot",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -595,7 +586,6 @@
"../rtc_base:copy_on_write_buffer",
"../rtc_base:logging",
"../rtc_base:ssl_adapter",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
rtc_source_set("srtp_session") {
@@ -659,7 +649,6 @@
"../rtc_base:zero_memory",
"../rtc_base/third_party/base64",
"//third_party/abseil-cpp/absl/strings",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -707,7 +696,6 @@
"../api/video:recordable_encoded_frame",
"../api/video:video_frame",
"../rtc_base:threading",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -802,7 +790,6 @@
"../rtc_base/containers:flat_set",
"../rtc_base/system:no_unique_address",
"../rtc_base/system:unused",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -881,7 +868,6 @@
"../rtc_base:threading",
"../rtc_base:weak_ptr",
"//third_party/abseil-cpp/absl/algorithm:container",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -898,7 +884,6 @@
"../api/audio:audio_device",
"../call:call_interfaces",
"../modules/audio_device",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -970,7 +955,6 @@
"../rtc_base/synchronization:mutex",
"//third_party/abseil-cpp/absl/functional:bind_front",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -984,7 +968,6 @@
"../api:rtc_stats_api",
"../api:scoped_refptr",
"../rtc_base:checks",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -1073,7 +1056,6 @@
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
rtc_source_set("jsep_ice_candidate") {
@@ -1195,7 +1177,6 @@
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -1215,7 +1196,6 @@
"../rtc_base:stringutils",
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
rtc_source_set("sdp_utils") {
@@ -1281,7 +1261,6 @@
"../rtc_base:threading",
"../rtc_base:timeutils",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
rtc_source_set("stream_collection") {
@@ -1307,7 +1286,6 @@
"../rtc_base:checks",
"../rtc_base:refcount",
"../rtc_base:threading",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
rtc_source_set("webrtc_sdp") {
@@ -1360,7 +1338,6 @@
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
rtc_source_set("webrtc_session_description_factory") {
@@ -1392,7 +1369,6 @@
"../rtc_base:weak_ptr",
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/functional:any_invocable",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -1580,7 +1556,6 @@
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -1619,7 +1594,6 @@
"../rtc_base:ssl",
"../rtc_base:threading",
"../rtc_base:weak_ptr",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -1640,7 +1614,6 @@
"../rtc_base:checks",
"../rtc_base:macromagic",
"../rtc_base/system:no_unique_address",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -1666,7 +1639,6 @@
"../rtc_base:checks",
"../rtc_base:logging",
"../rtc_base:threading",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -1698,7 +1670,6 @@
"../rtc_base:macromagic",
"../rtc_base:threading",
"../rtc_base/system:no_unique_address",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -1732,7 +1703,6 @@
"../rtc_base:macromagic",
"../rtc_base:threading",
"../rtc_base/system:no_unique_address",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -1787,7 +1757,6 @@
"../rtc_base:macromagic",
"../rtc_base:threading",
"../rtc_base/system:no_unique_address",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -1810,7 +1779,6 @@
"../rtc_base:safe_conversions",
"../rtc_base:safe_minmax",
"../rtc_base/system:no_unique_address",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -1835,7 +1803,6 @@
"../rtc_base:stringutils",
"../rtc_base/synchronization:mutex",
"//third_party/abseil-cpp/absl/algorithm:container",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -1873,7 +1840,6 @@
"../rtc_base/synchronization:mutex",
"../rtc_base/third_party/sigslot",
"//third_party/abseil-cpp/absl/algorithm:container",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -1896,7 +1862,6 @@
"../rtc_base:checks",
"../rtc_base:logging",
"../rtc_base:stringutils",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -1952,7 +1917,6 @@
"../rtc_base:macromagic",
"../rtc_base/system:no_unique_address",
"../rtc_base/system:rtc_export",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -2111,7 +2075,6 @@
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
include_dirs = []
@@ -2185,7 +2148,6 @@
"../rtc_base:threading",
"../system_wrappers",
"../test:test_support",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -2210,7 +2172,6 @@
"../rtc_base:gunit_helpers",
"../rtc_base:logging",
"../test:test_support",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -2240,7 +2201,6 @@
"../test/time_controller:time_controller",
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/strings",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -2466,7 +2426,6 @@
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/strings",
- "//third_party/abseil-cpp/absl/types:optional",
]
if (is_android) {
@@ -2681,7 +2640,6 @@
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -2819,7 +2777,6 @@
"../test:test_support",
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/strings",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
diff --git a/pc/audio_rtp_receiver.cc b/pc/audio_rtp_receiver.cc
index 0f395a7..705ec1b 100644
--- a/pc/audio_rtp_receiver.cc
+++ b/pc/audio_rtp_receiver.cc
@@ -165,7 +165,7 @@
track_->internal()->set_ended();
}
-void AudioRtpReceiver::RestartMediaChannel(absl::optional<uint32_t> ssrc) {
+void AudioRtpReceiver::RestartMediaChannel(std::optional<uint32_t> ssrc) {
RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
bool enabled = track_->internal()->enabled();
MediaSourceInterface::SourceState state = source_->state();
@@ -177,7 +177,7 @@
}
void AudioRtpReceiver::RestartMediaChannel_w(
- absl::optional<uint32_t> ssrc,
+ std::optional<uint32_t> ssrc,
bool track_enabled,
MediaSourceInterface::SourceState state) {
RTC_DCHECK_RUN_ON(worker_thread_);
@@ -212,10 +212,10 @@
void AudioRtpReceiver::SetupUnsignaledMediaChannel() {
RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
- RestartMediaChannel(absl::nullopt);
+ RestartMediaChannel(std::nullopt);
}
-absl::optional<uint32_t> AudioRtpReceiver::ssrc() const {
+std::optional<uint32_t> AudioRtpReceiver::ssrc() const {
RTC_DCHECK_RUN_ON(worker_thread_);
if (!signaled_ssrc_.has_value() && media_channel_) {
return media_channel_->GetUnsignaledSsrc();
@@ -314,7 +314,7 @@
}
void AudioRtpReceiver::SetJitterBufferMinimumDelay(
- absl::optional<double> delay_seconds) {
+ std::optional<double> delay_seconds) {
RTC_DCHECK_RUN_ON(worker_thread_);
delay_.Set(delay_seconds);
if (media_channel_ && signaled_ssrc_)
diff --git a/pc/audio_rtp_receiver.h b/pc/audio_rtp_receiver.h
index d0433c3..628690f 100644
--- a/pc/audio_rtp_receiver.h
+++ b/pc/audio_rtp_receiver.h
@@ -13,10 +13,10 @@
#include <stdint.h>
+#include <optional>
#include <string>
#include <vector>
-#include "absl/types/optional.h"
#include "api/crypto/frame_decryptor_interface.h"
#include "api/dtls_transport_interface.h"
#include "api/frame_transformer_interface.h"
@@ -100,7 +100,7 @@
void Stop() override;
void SetupMediaChannel(uint32_t ssrc) override;
void SetupUnsignaledMediaChannel() override;
- absl::optional<uint32_t> ssrc() const override;
+ std::optional<uint32_t> ssrc() const override;
void NotifyFirstPacketReceived() override;
void set_stream_ids(std::vector<std::string> stream_ids) override;
void set_transport(
@@ -110,7 +110,7 @@
void SetObserver(RtpReceiverObserverInterface* observer) override;
void SetJitterBufferMinimumDelay(
- absl::optional<double> delay_seconds) override;
+ std::optional<double> delay_seconds) override;
void SetMediaChannel(
cricket::MediaReceiveChannelInterface* media_channel) override;
@@ -121,9 +121,9 @@
rtc::scoped_refptr<FrameTransformerInterface> frame_transformer) override;
private:
- void RestartMediaChannel(absl::optional<uint32_t> ssrc)
+ void RestartMediaChannel(std::optional<uint32_t> ssrc)
RTC_RUN_ON(&signaling_thread_checker_);
- void RestartMediaChannel_w(absl::optional<uint32_t> ssrc,
+ void RestartMediaChannel_w(std::optional<uint32_t> ssrc,
bool track_enabled,
MediaSourceInterface::SourceState state)
RTC_RUN_ON(worker_thread_);
@@ -137,7 +137,7 @@
const rtc::scoped_refptr<AudioTrackProxyWithInternal<AudioTrack>> track_;
cricket::VoiceMediaReceiveChannelInterface* media_channel_
RTC_GUARDED_BY(worker_thread_) = nullptr;
- absl::optional<uint32_t> signaled_ssrc_ RTC_GUARDED_BY(worker_thread_);
+ std::optional<uint32_t> signaled_ssrc_ RTC_GUARDED_BY(worker_thread_);
std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams_
RTC_GUARDED_BY(&signaling_thread_checker_);
bool cached_track_enabled_ RTC_GUARDED_BY(&signaling_thread_checker_);
diff --git a/pc/channel.cc b/pc/channel.cc
index 483b2ce..af130f8 100644
--- a/pc/channel.cc
+++ b/pc/channel.cc
@@ -170,7 +170,7 @@
rtp_transport_->SubscribeReadyToSend(
this, [this](bool ready) { OnTransportReadyToSend(ready); });
rtp_transport_->SubscribeNetworkRouteChanged(
- this, [this](absl::optional<rtc::NetworkRoute> route) {
+ this, [this](std::optional<rtc::NetworkRoute> route) {
OnNetworkRouteChanged(route);
});
rtp_transport_->SubscribeWritableState(
@@ -335,7 +335,7 @@
}
void BaseChannel::OnNetworkRouteChanged(
- absl::optional<rtc::NetworkRoute> network_route) {
+ std::optional<rtc::NetworkRoute> network_route) {
RTC_DCHECK_RUN_ON(network_thread());
RTC_DCHECK(network_initialized());
@@ -449,7 +449,7 @@
bool BaseChannel::MaybeUpdateDemuxerAndRtpExtensions_w(
bool update_demuxer,
- absl::optional<RtpHeaderExtensions> extensions,
+ std::optional<RtpHeaderExtensions> extensions,
std::string& error_desc) {
if (extensions) {
if (rtp_header_extensions_ == extensions) {
@@ -911,8 +911,8 @@
bool success = MaybeUpdateDemuxerAndRtpExtensions_w(
criteria_modified,
update_header_extensions
- ? absl::optional<RtpHeaderExtensions>(std::move(header_extensions))
- : absl::nullopt,
+ ? std::optional<RtpHeaderExtensions>(std::move(header_extensions))
+ : std::nullopt,
error_desc);
RTC_DCHECK_BLOCK_COUNT_NO_MORE_THAN(1);
@@ -1056,7 +1056,7 @@
}
if (may_ignore_packetization) {
- send_codec.packetization = absl::nullopt;
+ send_codec.packetization = std::nullopt;
needs_send_params_update = true;
} else if (!has_matching_packetization) {
error_desc = StringFormat(
@@ -1113,8 +1113,8 @@
bool success = MaybeUpdateDemuxerAndRtpExtensions_w(
criteria_modified,
update_header_extensions
- ? absl::optional<RtpHeaderExtensions>(std::move(header_extensions))
- : absl::nullopt,
+ ? std::optional<RtpHeaderExtensions>(std::move(header_extensions))
+ : std::nullopt,
error_desc);
RTC_DCHECK_BLOCK_COUNT_NO_MORE_THAN(1);
@@ -1170,7 +1170,7 @@
}
if (may_ignore_packetization) {
- recv_codec.packetization = absl::nullopt;
+ recv_codec.packetization = std::nullopt;
needs_recv_params_update = true;
} else if (!has_matching_packetization) {
error_desc = StringFormat(
diff --git a/pc/channel.h b/pc/channel.h
index c933091..9a1b0a7 100644
--- a/pc/channel.h
+++ b/pc/channel.h
@@ -15,12 +15,12 @@
#include <functional>
#include <memory>
+#include <optional>
#include <string>
#include <utility>
#include <vector>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/crypto/crypto_options.h"
#include "api/jsep.h"
#include "api/media_types.h"
@@ -228,7 +228,7 @@
// From RtpTransportInternal
void OnWritableState(bool writable);
- void OnNetworkRouteChanged(absl::optional<rtc::NetworkRoute> network_route);
+ void OnNetworkRouteChanged(std::optional<rtc::NetworkRoute> network_route);
bool SendPacket(bool rtcp,
rtc::CopyOnWriteBuffer* packet,
@@ -295,7 +295,7 @@
// failed, which needs to be treated as an error.
bool MaybeUpdateDemuxerAndRtpExtensions_w(
bool update_demuxer,
- absl::optional<RtpHeaderExtensions> extensions,
+ std::optional<RtpHeaderExtensions> extensions,
std::string& error_desc) RTC_RUN_ON(worker_thread());
bool RegisterRtpDemuxerSink_w() RTC_RUN_ON(worker_thread());
diff --git a/pc/channel_unittest.cc b/pc/channel_unittest.cc
index a651391..e4ab8fd 100644
--- a/pc/channel_unittest.cc
+++ b/pc/channel_unittest.cc
@@ -936,7 +936,7 @@
rtc::NetworkRoute network_route;
// The transport channel becomes disconnected.
fake_rtp_dtls_transport1_->ice_transport()->SignalNetworkRouteChanged(
- absl::optional<rtc::NetworkRoute>(network_route));
+ std::optional<rtc::NetworkRoute>(network_route));
});
WaitForThreads();
EXPECT_EQ(1, media_send_channel1_impl->num_network_route_changes());
@@ -955,7 +955,7 @@
// The transport channel becomes connected.
fake_rtp_dtls_transport1_->ice_transport()->SignalNetworkRouteChanged(
- absl::optional<rtc::NetworkRoute>(network_route));
+ std::optional<rtc::NetworkRoute>(network_route));
});
WaitForThreads();
EXPECT_EQ(1, media_send_channel1_impl->num_network_route_changes());
@@ -1331,7 +1331,7 @@
return channel1_->SetRemoteContent(&content, SdpType::kOffer, NULL);
}
- webrtc::RtpParameters BitrateLimitedParameters(absl::optional<int> limit) {
+ webrtc::RtpParameters BitrateLimitedParameters(std::optional<int> limit) {
webrtc::RtpParameters parameters;
webrtc::RtpEncodingParameters encoding;
encoding.max_bitrate_bps = limit;
@@ -1340,7 +1340,7 @@
}
void VerifyMaxBitrate(const webrtc::RtpParameters& parameters,
- absl::optional<int> expected_bitrate) {
+ std::optional<int> expected_bitrate) {
EXPECT_EQ(1UL, parameters.encodings.size());
EXPECT_EQ(expected_bitrate, parameters.encodings[0].max_bitrate_bps);
}
@@ -1352,7 +1352,7 @@
SdpType::kOffer, err));
EXPECT_EQ(media_send_channel1_impl()->max_bps(), -1);
VerifyMaxBitrate(media_send_channel1()->GetRtpSendParameters(kSsrc1),
- absl::nullopt);
+ std::nullopt);
}
// Test that when a channel gets new RtpTransport with a call to
@@ -2113,7 +2113,7 @@
EXPECT_TRUE(
media_receive_channel1_impl()->recv_codecs()[0].Matches(kVp8Codec));
EXPECT_EQ(media_receive_channel1_impl()->recv_codecs()[0].packetization,
- absl::nullopt);
+ std::nullopt);
EXPECT_TRUE(
media_receive_channel1_impl()->recv_codecs()[1].Matches(vp9_codec));
EXPECT_EQ(media_receive_channel1_impl()->recv_codecs()[1].packetization,
@@ -2136,7 +2136,7 @@
ASSERT_THAT(media_send_channel1_impl()->send_codecs(), testing::SizeIs(2));
EXPECT_TRUE(media_send_channel1_impl()->send_codecs()[0].Matches(kVp8Codec));
EXPECT_EQ(media_send_channel1_impl()->send_codecs()[0].packetization,
- absl::nullopt);
+ std::nullopt);
EXPECT_TRUE(media_send_channel1_impl()->send_codecs()[1].Matches(vp9_codec));
EXPECT_EQ(media_send_channel1_impl()->send_codecs()[1].packetization,
cricket::kPacketizationParamRaw);
@@ -2160,7 +2160,7 @@
EXPECT_TRUE(
media_receive_channel1_impl()->recv_codecs()[0].Matches(kVp8Codec));
EXPECT_EQ(media_receive_channel1_impl()->recv_codecs()[0].packetization,
- absl::nullopt);
+ std::nullopt);
EXPECT_TRUE(
media_receive_channel1_impl()->recv_codecs()[1].Matches(vp9_codec));
EXPECT_EQ(media_receive_channel1_impl()->recv_codecs()[1].packetization,
@@ -2168,7 +2168,7 @@
EXPECT_THAT(media_send_channel1_impl()->send_codecs(), testing::SizeIs(2));
EXPECT_TRUE(media_send_channel1_impl()->send_codecs()[0].Matches(kVp8Codec));
EXPECT_EQ(media_send_channel1_impl()->send_codecs()[0].packetization,
- absl::nullopt);
+ std::nullopt);
EXPECT_TRUE(media_send_channel1_impl()->send_codecs()[1].Matches(vp9_codec));
EXPECT_EQ(media_send_channel1_impl()->send_codecs()[1].packetization,
cricket::kPacketizationParamRaw);
@@ -2190,10 +2190,10 @@
EXPECT_TRUE(channel1_->SetLocalContent(&local_video, SdpType::kAnswer, err));
ASSERT_THAT(media_receive_channel1_impl()->recv_codecs(), testing::SizeIs(1));
EXPECT_EQ(media_receive_channel1_impl()->recv_codecs()[0].packetization,
- absl::nullopt);
+ std::nullopt);
ASSERT_THAT(media_send_channel1_impl()->send_codecs(), testing::SizeIs(1));
EXPECT_EQ(media_send_channel1_impl()->send_codecs()[0].packetization,
- absl::nullopt);
+ std::nullopt);
}
TEST_F(VideoChannelSingleThreadTest, TestSetRemoteAnswerWithoutPacketization) {
@@ -2213,10 +2213,10 @@
channel1_->SetRemoteContent(&remote_video, SdpType::kAnswer, err));
ASSERT_THAT(media_receive_channel1_impl()->recv_codecs(), testing::SizeIs(1));
EXPECT_EQ(media_receive_channel1_impl()->recv_codecs()[0].packetization,
- absl::nullopt);
+ std::nullopt);
ASSERT_THAT(media_send_channel1_impl()->send_codecs(), testing::SizeIs(1));
EXPECT_EQ(media_send_channel1_impl()->send_codecs()[0].packetization,
- absl::nullopt);
+ std::nullopt);
}
TEST_F(VideoChannelSingleThreadTest,
@@ -2264,7 +2264,7 @@
EXPECT_THAT(media_receive_channel1_impl()->recv_codecs(), testing::IsEmpty());
ASSERT_THAT(media_send_channel1_impl()->send_codecs(), testing::SizeIs(1));
EXPECT_EQ(media_send_channel1_impl()->send_codecs()[0].packetization,
- absl::nullopt);
+ std::nullopt);
}
TEST_F(VideoChannelSingleThreadTest,
@@ -2294,13 +2294,13 @@
AllOf(Field(&cricket::Codec::id, 97),
Field(&cricket::Codec::packetization, "bar")),
AllOf(Field(&cricket::Codec::id, 99),
- Field(&cricket::Codec::packetization, absl::nullopt))));
+ Field(&cricket::Codec::packetization, std::nullopt))));
EXPECT_THAT(
media_send_channel1_impl()->send_codecs(),
ElementsAre(AllOf(Field(&cricket::Codec::id, 96),
Field(&cricket::Codec::packetization, "foo")),
AllOf(Field(&cricket::Codec::id, 98),
- Field(&cricket::Codec::packetization, absl::nullopt))));
+ Field(&cricket::Codec::packetization, std::nullopt))));
}
TEST_F(VideoChannelSingleThreadTest,
@@ -2328,7 +2328,7 @@
ElementsAre(AllOf(Field(&cricket::Codec::id, 96),
Field(&cricket::Codec::packetization, "foo")),
AllOf(Field(&cricket::Codec::id, 98),
- Field(&cricket::Codec::packetization, absl::nullopt))));
+ Field(&cricket::Codec::packetization, std::nullopt))));
EXPECT_THAT(
media_send_channel1_impl()->send_codecs(),
ElementsAre(AllOf(Field(&cricket::Codec::id, 96),
@@ -2336,7 +2336,7 @@
AllOf(Field(&cricket::Codec::id, 97),
Field(&cricket::Codec::packetization, "bar")),
AllOf(Field(&cricket::Codec::id, 99),
- Field(&cricket::Codec::packetization, absl::nullopt))));
+ Field(&cricket::Codec::packetization, std::nullopt))));
}
TEST_F(VideoChannelSingleThreadTest,
@@ -2358,7 +2358,7 @@
EXPECT_THAT(
media_receive_channel1_impl()->recv_codecs(),
ElementsAre(AllOf(Field(&cricket::Codec::id, 96),
- Field(&cricket::Codec::packetization, absl::nullopt)),
+ Field(&cricket::Codec::packetization, std::nullopt)),
AllOf(Field(&cricket::Codec::id, 97),
Field(&cricket::Codec::packetization,
cricket::kPacketizationParamRaw))));
@@ -2368,7 +2368,7 @@
Field(&cricket::Codec::packetization,
cricket::kPacketizationParamRaw)),
AllOf(Field(&cricket::Codec::id, 96),
- Field(&cricket::Codec::packetization, absl::nullopt))));
+ Field(&cricket::Codec::packetization, std::nullopt))));
}
TEST_F(VideoChannelSingleThreadTest,
@@ -2393,11 +2393,11 @@
Field(&cricket::Codec::packetization,
cricket::kPacketizationParamRaw)),
AllOf(Field(&cricket::Codec::id, 96),
- Field(&cricket::Codec::packetization, absl::nullopt))));
+ Field(&cricket::Codec::packetization, std::nullopt))));
EXPECT_THAT(
media_send_channel1_impl()->send_codecs(),
ElementsAre(AllOf(Field(&cricket::Codec::id, 96),
- Field(&cricket::Codec::packetization, absl::nullopt)),
+ Field(&cricket::Codec::packetization, std::nullopt)),
AllOf(Field(&cricket::Codec::id, 97),
Field(&cricket::Codec::packetization,
cricket::kPacketizationParamRaw))));
diff --git a/pc/data_channel_controller.cc b/pc/data_channel_controller.cc
index e252e8f..3445e89 100644
--- a/pc/data_channel_controller.cc
+++ b/pc/data_channel_controller.cc
@@ -11,10 +11,10 @@
#include "pc/data_channel_controller.h"
#include <cstdint>
+#include <optional>
#include <utility>
#include "absl/algorithm/container.h"
-#include "absl/types/optional.h"
#include "api/peer_connection_interface.h"
#include "api/priority.h"
#include "api/rtc_error.h"
@@ -300,8 +300,8 @@
// RTC_RUN_ON(network_thread())
RTCError DataChannelController::ReserveOrAllocateSid(
- absl::optional<StreamId>& sid,
- absl::optional<rtc::SSLRole> fallback_ssl_role) {
+ std::optional<StreamId>& sid,
+ std::optional<rtc::SSLRole> fallback_ssl_role) {
if (sid.has_value()) {
return sid_allocator_.ReserveSid(*sid)
? RTCError::OK()
@@ -309,7 +309,7 @@
}
// Attempt to allocate an ID based on the negotiated role.
- absl::optional<rtc::SSLRole> role = pc_->GetSctpSslRole_n();
+ std::optional<rtc::SSLRole> role = pc_->GetSctpSslRole_n();
if (!role)
role = fallback_ssl_role;
if (role) {
@@ -327,7 +327,7 @@
RTCErrorOr<rtc::scoped_refptr<SctpDataChannel>>
DataChannelController::CreateDataChannel(const std::string& label,
InternalDataChannelInit& config) {
- absl::optional<StreamId> sid = absl::nullopt;
+ std::optional<StreamId> sid = std::nullopt;
if (config.id != -1) {
if (config.id < 0 || config.id > cricket::kMaxSctpSid) {
return RTCError(RTCErrorType::INVALID_RANGE, "StreamId out of range.");
@@ -413,7 +413,7 @@
for (auto it = sctp_data_channels_n_.begin();
it != sctp_data_channels_n_.end();) {
if (!(*it)->sid_n().has_value()) {
- absl::optional<StreamId> sid = sid_allocator_.AllocateSid(role);
+ std::optional<StreamId> sid = sid_allocator_.AllocateSid(role);
if (sid.has_value()) {
(*it)->SetSctpSid_n(*sid);
AddSctpDataStream(*sid, (*it)->priority());
diff --git a/pc/data_channel_controller.h b/pc/data_channel_controller.h
index 74e00e0..916eb990 100644
--- a/pc/data_channel_controller.h
+++ b/pc/data_channel_controller.h
@@ -130,8 +130,8 @@
// will still be unassigned upon return, but will be assigned later.
// If the pool has been exhausted or a sid has already been reserved, an
// error will be returned.
- RTCError ReserveOrAllocateSid(absl::optional<StreamId>& sid,
- absl::optional<rtc::SSLRole> fallback_ssl_role)
+ RTCError ReserveOrAllocateSid(std::optional<StreamId>& sid,
+ std::optional<rtc::SSLRole> fallback_ssl_role)
RTC_RUN_ON(network_thread());
// Called when all data channels need to be notified of a transport channel
diff --git a/pc/data_channel_controller_unittest.cc b/pc/data_channel_controller_unittest.cc
index f885079..78f914a 100644
--- a/pc/data_channel_controller_unittest.cc
+++ b/pc/data_channel_controller_unittest.cc
@@ -159,8 +159,8 @@
int channel_id = 0;
ON_CALL(*pc_, GetSctpSslRole_n).WillByDefault([&]() {
- return absl::optional<rtc::SSLRole>((channel_id & 1) ? rtc::SSL_SERVER
- : rtc::SSL_CLIENT);
+ return std::optional<rtc::SSLRole>((channel_id & 1) ? rtc::SSL_SERVER
+ : rtc::SSL_CLIENT);
});
DataChannelControllerForTest dcc(pc_.get(), &transport);
diff --git a/pc/data_channel_integrationtest.cc b/pc/data_channel_integrationtest.cc
index 1a23a17..d779b23 100644
--- a/pc/data_channel_integrationtest.cc
+++ b/pc/data_channel_integrationtest.cc
@@ -12,12 +12,12 @@
#include <cstdlib>
#include <iterator>
+#include <optional>
#include <string>
#include <tuple>
#include <vector>
#include "absl/algorithm/container.h"
-#include "absl/types/optional.h"
#include "api/data_channel_interface.h"
#include "api/dtls_transport_interface.h"
#include "api/peer_connection_interface.h"
diff --git a/pc/data_channel_unittest.cc b/pc/data_channel_unittest.cc
index c5f6525..f16ed42 100644
--- a/pc/data_channel_unittest.cc
+++ b/pc/data_channel_unittest.cc
@@ -172,7 +172,7 @@
// Check the non-const part of the configuration.
EXPECT_EQ(channel_->id(), init_.id);
network_thread_.BlockingCall(
- [&]() { EXPECT_EQ(inner_channel_->sid_n(), absl::nullopt); });
+ [&]() { EXPECT_EQ(inner_channel_->sid_n(), std::nullopt); });
SetChannelReady();
EXPECT_EQ(channel_->id(), 0);
@@ -188,7 +188,7 @@
EXPECT_TRUE(controller_->IsConnected(dc.get()));
// The sid is not set yet, so it should not have added the streams.
- absl::optional<StreamId> sid =
+ std::optional<StreamId> sid =
network_thread_.BlockingCall([&]() { return dc->sid_n(); });
EXPECT_FALSE(sid.has_value());
@@ -636,7 +636,7 @@
StreamId old_id(1);
EXPECT_TRUE(allocator_.ReserveSid(old_id));
- absl::optional<StreamId> new_id = allocator_.AllocateSid(rtc::SSL_SERVER);
+ std::optional<StreamId> new_id = allocator_.AllocateSid(rtc::SSL_SERVER);
EXPECT_TRUE(new_id.has_value());
EXPECT_NE(old_id, new_id);
@@ -654,7 +654,7 @@
EXPECT_TRUE(allocator_.ReserveSid(odd_id));
EXPECT_TRUE(allocator_.ReserveSid(even_id));
- absl::optional<StreamId> allocated_id =
+ std::optional<StreamId> allocated_id =
allocator_.AllocateSid(rtc::SSL_SERVER);
EXPECT_EQ(odd_id.stream_id_int() + 2, allocated_id->stream_id_int());
diff --git a/pc/dtls_srtp_transport.h b/pc/dtls_srtp_transport.h
index 654eb2f..12c2ffc 100644
--- a/pc/dtls_srtp_transport.h
+++ b/pc/dtls_srtp_transport.h
@@ -12,10 +12,10 @@
#define PC_DTLS_SRTP_TRANSPORT_H_
#include <functional>
+#include <optional>
#include <string>
#include <vector>
-#include "absl/types/optional.h"
#include "api/dtls_transport_interface.h"
#include "api/rtc_error.h"
#include "p2p/base/dtls_transport_internal.h"
@@ -83,8 +83,8 @@
cricket::DtlsTransportInternal* rtcp_dtls_transport_ = nullptr;
// The encrypted header extension IDs.
- absl::optional<std::vector<int>> send_extension_ids_;
- absl::optional<std::vector<int>> recv_extension_ids_;
+ std::optional<std::vector<int>> send_extension_ids_;
+ std::optional<std::vector<int>> recv_extension_ids_;
bool active_reset_srtp_params_ = false;
std::function<void(void)> on_dtls_state_change_;
diff --git a/pc/dtls_transport.cc b/pc/dtls_transport.cc
index 4888d9f..f14f51f 100644
--- a/pc/dtls_transport.cc
+++ b/pc/dtls_transport.cc
@@ -10,9 +10,9 @@
#include "pc/dtls_transport.h"
+#include <optional>
#include <utility>
-#include "absl/types/optional.h"
#include "api/dtls_transport_interface.h"
#include "api/make_ref_counted.h"
#include "api/sequence_checker.h"
@@ -109,7 +109,7 @@
DtlsTransportState::kConnected) {
bool success = true;
rtc::SSLRole internal_role;
- absl::optional<DtlsTransportTlsRole> role;
+ std::optional<DtlsTransportTlsRole> role;
int ssl_cipher_suite;
int tls_version;
int srtp_cipher;
@@ -136,8 +136,8 @@
RTC_LOG(LS_ERROR) << "DtlsTransport in connected state has incomplete "
"TLS information";
set_info(DtlsTransportInformation(
- internal_dtls_transport_->dtls_state(), role, absl::nullopt,
- absl::nullopt, absl::nullopt,
+ internal_dtls_transport_->dtls_state(), role, std::nullopt,
+ std::nullopt, std::nullopt,
internal_dtls_transport_->GetRemoteSSLCertChain()));
}
} else {
diff --git a/pc/dtls_transport_unittest.cc b/pc/dtls_transport_unittest.cc
index c234176..7d71f1e 100644
--- a/pc/dtls_transport_unittest.cc
+++ b/pc/dtls_transport_unittest.cc
@@ -10,10 +10,10 @@
#include "pc/dtls_transport.h"
+#include <optional>
#include <utility>
#include <vector>
-#include "absl/types/optional.h"
#include "api/make_ref_counted.h"
#include "api/rtc_error.h"
#include "p2p/base/fake_dtls_transport.h"
diff --git a/pc/ice_server_parsing.cc b/pc/ice_server_parsing.cc
index 896305c..dc8122f 100644
--- a/pc/ice_server_parsing.cc
+++ b/pc/ice_server_parsing.cc
@@ -89,7 +89,7 @@
return {ServiceType::INVALID, ""};
}
-absl::optional<int> ParsePort(absl::string_view in_str) {
+std::optional<int> ParsePort(absl::string_view in_str) {
// Make sure port only contains digits. StringToNumber doesn't check this.
for (const char& c : in_str) {
if (!std::isdigit(static_cast<unsigned char>(c))) {
@@ -123,7 +123,7 @@
}
auto colonpos = in_str.find(':', closebracket);
if (absl::string_view::npos != colonpos) {
- if (absl::optional<int> opt_port =
+ if (std::optional<int> opt_port =
ParsePort(in_str.substr(closebracket + 2))) {
port = *opt_port;
} else {
@@ -135,7 +135,7 @@
// IPv4address or reg-name syntax
auto colonpos = in_str.find(':');
if (absl::string_view::npos != colonpos) {
- if (absl::optional<int> opt_port =
+ if (std::optional<int> opt_port =
ParsePort(in_str.substr(colonpos + 1))) {
port = *opt_port;
} else {
@@ -196,7 +196,7 @@
"ICE server parsing failed: Transport parameter missing value.");
}
- absl::optional<cricket::ProtocolType> proto =
+ std::optional<cricket::ProtocolType> proto =
cricket::StringToProto(transport_tokens[1]);
if (!proto ||
(*proto != cricket::PROTO_UDP && *proto != cricket::PROTO_TCP)) {
diff --git a/pc/jitter_buffer_delay.cc b/pc/jitter_buffer_delay.cc
index f22b065..a747588 100644
--- a/pc/jitter_buffer_delay.cc
+++ b/pc/jitter_buffer_delay.cc
@@ -22,7 +22,7 @@
namespace webrtc {
-void JitterBufferDelay::Set(absl::optional<double> delay_seconds) {
+void JitterBufferDelay::Set(std::optional<double> delay_seconds) {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
cached_delay_seconds_ = delay_seconds;
}
diff --git a/pc/jitter_buffer_delay.h b/pc/jitter_buffer_delay.h
index caf713b..bc506fe 100644
--- a/pc/jitter_buffer_delay.h
+++ b/pc/jitter_buffer_delay.h
@@ -13,7 +13,8 @@
#include <stdint.h>
-#include "absl/types/optional.h"
+#include <optional>
+
#include "api/sequence_checker.h"
#include "rtc_base/system/no_unique_address.h"
#include "rtc_base/thread_annotations.h"
@@ -27,13 +28,13 @@
public:
JitterBufferDelay() = default;
- void Set(absl::optional<double> delay_seconds);
+ void Set(std::optional<double> delay_seconds);
int GetMs() const;
private:
RTC_NO_UNIQUE_ADDRESS SequenceChecker worker_thread_checker_{
SequenceChecker::kDetached};
- absl::optional<double> cached_delay_seconds_
+ std::optional<double> cached_delay_seconds_
RTC_GUARDED_BY(&worker_thread_checker_);
};
diff --git a/pc/jsep_session_description.cc b/pc/jsep_session_description.cc
index 7fae445..a2e3ab8 100644
--- a/pc/jsep_session_description.cc
+++ b/pc/jsep_session_description.cc
@@ -11,9 +11,9 @@
#include "api/jsep_session_description.h"
#include <memory>
+#include <optional>
#include <utility>
-#include "absl/types/optional.h"
#include "p2p/base/p2p_constants.h"
#include "p2p/base/port.h"
#include "p2p/base/transport_description.h"
@@ -101,7 +101,7 @@
// TODO(steveanton): Remove this default implementation once Chromium has been
// updated.
SdpType SessionDescriptionInterface::GetType() const {
- absl::optional<SdpType> maybe_type = SdpTypeFromString(type());
+ std::optional<SdpType> maybe_type = SdpTypeFromString(type());
if (maybe_type) {
return *maybe_type;
} else {
@@ -116,7 +116,7 @@
SessionDescriptionInterface* CreateSessionDescription(const std::string& type,
const std::string& sdp,
SdpParseError* error) {
- absl::optional<SdpType> maybe_type = SdpTypeFromString(type);
+ std::optional<SdpType> maybe_type = SdpTypeFromString(type);
if (!maybe_type) {
return nullptr;
}
@@ -158,7 +158,7 @@
JsepSessionDescription::JsepSessionDescription(SdpType type) : type_(type) {}
JsepSessionDescription::JsepSessionDescription(const std::string& type) {
- absl::optional<SdpType> maybe_type = SdpTypeFromString(type);
+ std::optional<SdpType> maybe_type = SdpTypeFromString(type);
if (maybe_type) {
type_ = *maybe_type;
} else {
diff --git a/pc/jsep_transport.cc b/pc/jsep_transport.cc
index 5b2f265..95f7aed 100644
--- a/pc/jsep_transport.cc
+++ b/pc/jsep_transport.cc
@@ -305,16 +305,16 @@
}
}
-absl::optional<rtc::SSLRole> JsepTransport::GetDtlsRole() const {
+std::optional<rtc::SSLRole> JsepTransport::GetDtlsRole() const {
RTC_DCHECK_RUN_ON(network_thread_);
RTC_DCHECK(rtp_dtls_transport_);
RTC_DCHECK(rtp_dtls_transport_->internal());
rtc::SSLRole dtls_role;
if (!rtp_dtls_transport_->internal()->GetDtlsRole(&dtls_role)) {
- return absl::optional<rtc::SSLRole>();
+ return std::optional<rtc::SSLRole>();
}
- return absl::optional<rtc::SSLRole>(dtls_role);
+ return std::optional<rtc::SSLRole>(dtls_role);
}
bool JsepTransport::GetStats(TransportStats* stats) {
@@ -404,7 +404,7 @@
webrtc::RTCError JsepTransport::SetNegotiatedDtlsParameters(
DtlsTransportInternal* dtls_transport,
- absl::optional<rtc::SSLRole> dtls_role,
+ std::optional<rtc::SSLRole> dtls_role,
rtc::SSLFingerprint* remote_fingerprint) {
RTC_DCHECK(dtls_transport);
return dtls_transport->SetRemoteParameters(
@@ -475,7 +475,7 @@
"without applying any offer.");
}
std::unique_ptr<rtc::SSLFingerprint> remote_fingerprint;
- absl::optional<rtc::SSLRole> negotiated_dtls_role;
+ std::optional<rtc::SSLRole> negotiated_dtls_role;
rtc::SSLFingerprint* local_fp =
local_description_->transport_desc.identity_fingerprint.get();
@@ -523,7 +523,7 @@
SdpType local_description_type,
ConnectionRole local_connection_role,
ConnectionRole remote_connection_role,
- absl::optional<rtc::SSLRole>* negotiated_dtls_role) {
+ std::optional<rtc::SSLRole>* negotiated_dtls_role) {
// From RFC 4145, section-4.1, The following are the values that the
// 'setup' attribute can take in an offer/answer exchange:
// Offer Answer
diff --git a/pc/jsep_transport.h b/pc/jsep_transport.h
index 3c14f13..005e606 100644
--- a/pc/jsep_transport.h
+++ b/pc/jsep_transport.h
@@ -14,10 +14,10 @@
#include <functional>
#include <map>
#include <memory>
+#include <optional>
#include <string>
#include <vector>
-#include "absl/types/optional.h"
#include "api/candidate.h"
#include "api/ice_transport_interface.h"
#include "api/jsep.h"
@@ -149,9 +149,9 @@
return needs_ice_restart_;
}
- // Returns role if negotiated, or empty absl::optional if it hasn't been
+ // Returns role if negotiated, or empty std::optional if it hasn't been
// negotiated yet.
- absl::optional<rtc::SSLRole> GetDtlsRole() const;
+ std::optional<rtc::SSLRole> GetDtlsRole() const;
// TODO(deadbeef): Make this const. See comment in transportcontroller.h.
bool GetStats(TransportStats* stats);
@@ -274,7 +274,7 @@
webrtc::SdpType local_description_type,
ConnectionRole local_connection_role,
ConnectionRole remote_connection_role,
- absl::optional<rtc::SSLRole>* negotiated_dtls_role);
+ std::optional<rtc::SSLRole>* negotiated_dtls_role);
// Pushes down the ICE parameters from the remote description.
void SetRemoteIceParameters(const IceParameters& ice_parameters,
@@ -283,7 +283,7 @@
// Pushes down the DTLS parameters obtained via negotiation.
static webrtc::RTCError SetNegotiatedDtlsParameters(
DtlsTransportInternal* dtls_transport,
- absl::optional<rtc::SSLRole> dtls_role,
+ std::optional<rtc::SSLRole> dtls_role,
rtc::SSLFingerprint* remote_fingerprint);
bool GetTransportStats(DtlsTransportInternal* dtls_transport,
@@ -324,9 +324,9 @@
RtcpMuxFilter rtcp_mux_negotiator_ RTC_GUARDED_BY(network_thread_);
// Cache the encrypted header extension IDs for SDES negoitation.
- absl::optional<std::vector<int>> send_extension_ids_
+ std::optional<std::vector<int>> send_extension_ids_
RTC_GUARDED_BY(network_thread_);
- absl::optional<std::vector<int>> recv_extension_ids_
+ std::optional<std::vector<int>> recv_extension_ids_
RTC_GUARDED_BY(network_thread_);
// This is invoked when RTCP-mux becomes active and
diff --git a/pc/jsep_transport_controller.cc b/pc/jsep_transport_controller.cc
index 05988ec..2780196 100644
--- a/pc/jsep_transport_controller.cc
+++ b/pc/jsep_transport_controller.cc
@@ -204,7 +204,7 @@
return transport->needs_ice_restart();
}
-absl::optional<rtc::SSLRole> JsepTransportController::GetDtlsRole(
+std::optional<rtc::SSLRole> JsepTransportController::GetDtlsRole(
const std::string& mid) const {
// TODO(tommi): Remove this hop. Currently it's called from the signaling
// thread during negotiations, potentially multiple times.
@@ -217,7 +217,7 @@
const cricket::JsepTransport* t = GetJsepTransportForMid(mid);
if (!t) {
- return absl::optional<rtc::SSLRole>();
+ return std::optional<rtc::SSLRole>();
}
return t->GetDtlsRole();
}
diff --git a/pc/jsep_transport_controller.h b/pc/jsep_transport_controller.h
index c4275bd..d18a375 100644
--- a/pc/jsep_transport_controller.h
+++ b/pc/jsep_transport_controller.h
@@ -16,13 +16,13 @@
#include <functional>
#include <map>
#include <memory>
+#include <optional>
#include <string>
#include <type_traits>
#include <utility>
#include <vector>
#include "absl/functional/any_invocable.h"
-#include "absl/types/optional.h"
#include "api/async_dns_resolver.h"
#include "api/candidate.h"
#include "api/crypto/crypto_options.h"
@@ -233,7 +233,7 @@
std::unique_ptr<rtc::SSLCertChain> GetRemoteSSLCertChain(
const std::string& mid) const;
// Get negotiated role, if one has been negotiated.
- absl::optional<rtc::SSLRole> GetDtlsRole(const std::string& mid) const;
+ std::optional<rtc::SSLRole> GetDtlsRole(const std::string& mid) const;
// Suggest a payload type for a given codec on a given media section.
// Media section is indicated by MID.
@@ -511,7 +511,7 @@
const Config config_;
bool active_reset_srtp_params_ RTC_GUARDED_BY(network_thread_);
- absl::optional<bool> initial_offerer_;
+ std::optional<bool> initial_offerer_;
cricket::IceConfig ice_config_;
cricket::IceRole ice_role_ = cricket::ICEROLE_CONTROLLING;
diff --git a/pc/jsep_transport_controller_unittest.cc b/pc/jsep_transport_controller_unittest.cc
index 9e5d58b..0dcc3b2 100644
--- a/pc/jsep_transport_controller_unittest.cc
+++ b/pc/jsep_transport_controller_unittest.cc
@@ -644,7 +644,7 @@
->SetLocalDescription(SdpType::kOffer, offer_desc.get(), nullptr)
.ok());
- absl::optional<rtc::SSLRole> role =
+ std::optional<rtc::SSLRole> role =
transport_controller_->GetDtlsRole(kAudioMid1);
// The DTLS role is not decided yet.
EXPECT_FALSE(role);
diff --git a/pc/jsep_transport_unittest.cc b/pc/jsep_transport_unittest.cc
index 074f8e5..d185b27 100644
--- a/pc/jsep_transport_unittest.cc
+++ b/pc/jsep_transport_unittest.cc
@@ -802,7 +802,7 @@
.ok());
// Sanity check that role was actually negotiated.
- absl::optional<rtc::SSLRole> role = jsep_transport_->GetDtlsRole();
+ std::optional<rtc::SSLRole> role = jsep_transport_->GetDtlsRole();
ASSERT_TRUE(role);
EXPECT_EQ(rtc::SSL_CLIENT, *role);
@@ -847,7 +847,7 @@
.ok());
// Sanity check that role was actually negotiated.
- absl::optional<rtc::SSLRole> role = jsep_transport_->GetDtlsRole();
+ std::optional<rtc::SSLRole> role = jsep_transport_->GetDtlsRole();
ASSERT_TRUE(role);
EXPECT_EQ(rtc::SSL_CLIENT, *role);
@@ -898,7 +898,7 @@
.ok());
// Sanity check that role was actually negotiated.
- absl::optional<rtc::SSLRole> role = jsep_transport_->GetDtlsRole();
+ std::optional<rtc::SSLRole> role = jsep_transport_->GetDtlsRole();
ASSERT_TRUE(role);
EXPECT_EQ(rtc::SSL_CLIENT, *role);
@@ -947,7 +947,7 @@
->SetRemoteJsepTransportDescription(remote_desc, SdpType::kAnswer)
.ok());
- absl::optional<rtc::SSLRole> role = jsep_transport_->GetDtlsRole();
+ std::optional<rtc::SSLRole> role = jsep_transport_->GetDtlsRole();
ASSERT_TRUE(role);
// Since legacy answer omitted setup atribute, and we offered actpass, we
// should act as passive (server).
diff --git a/pc/legacy_stats_collector.cc b/pc/legacy_stats_collector.cc
index 800f63c..5665fb7 100644
--- a/pc/legacy_stats_collector.cc
+++ b/pc/legacy_stats_collector.cc
@@ -16,12 +16,12 @@
#include <algorithm>
#include <cmath>
#include <list>
+#include <optional>
#include <set>
#include <utility>
#include <vector>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/audio/audio_processing_statistics.h"
#include "api/audio_codecs/audio_encoder.h"
#include "api/candidate.h"
@@ -902,8 +902,8 @@
LegacyStatsCollector::SessionStats LegacyStatsCollector::ExtractSessionInfo_n(
const std::vector<rtc::scoped_refptr<
RtpTransceiverProxyWithInternal<RtpTransceiver>>>& transceivers,
- absl::optional<std::string> sctp_transport_name,
- absl::optional<std::string> sctp_mid) {
+ std::optional<std::string> sctp_transport_name,
+ std::optional<std::string> sctp_mid) {
TRACE_EVENT0("webrtc", "LegacyStatsCollector::ExtractSessionInfo_n");
RTC_DCHECK_RUN_ON(pc_->network_thread());
rtc::Thread::ScopedDisallowBlockingCalls no_blocking_calls;
diff --git a/pc/legacy_stats_collector.h b/pc/legacy_stats_collector.h
index e037163..8980bfd 100644
--- a/pc/legacy_stats_collector.h
+++ b/pc/legacy_stats_collector.h
@@ -20,12 +20,12 @@
#include <cstdint>
#include <map>
#include <memory>
+#include <optional>
#include <string>
#include <type_traits>
#include <utility>
#include <vector>
-#include "absl/types/optional.h"
#include "api/candidate.h"
#include "api/field_trials_view.h"
#include "api/legacy_stats_types.h"
@@ -194,8 +194,8 @@
SessionStats ExtractSessionInfo_n(
const std::vector<rtc::scoped_refptr<
RtpTransceiverProxyWithInternal<RtpTransceiver>>>& transceivers,
- absl::optional<std::string> sctp_transport_name,
- absl::optional<std::string> sctp_mid);
+ std::optional<std::string> sctp_transport_name,
+ std::optional<std::string> sctp_mid);
void ExtractSessionInfo_s(SessionStats& session_stats);
// A collection for all of our stats reports.
diff --git a/pc/legacy_stats_collector_unittest.cc b/pc/legacy_stats_collector_unittest.cc
index f372dca..a792f7a 100644
--- a/pc/legacy_stats_collector_unittest.cc
+++ b/pc/legacy_stats_collector_unittest.cc
@@ -13,9 +13,9 @@
#include <stdio.h>
#include <cstdint>
+#include <optional>
#include "absl/algorithm/container.h"
-#include "absl/types/optional.h"
#include "api/audio/audio_processing_statistics.h"
#include "api/audio_codecs/audio_encoder.h"
#include "api/candidate.h"
@@ -223,18 +223,18 @@
// `n` starts from 1 for finding the first report.
// If either the `n`-th report is not found, or the stat is not present in that
// report, then nullopt is returned.
-absl::optional<std::string> GetValueInNthReportByType(
+std::optional<std::string> GetValueInNthReportByType(
const StatsReports& reports,
StatsReport::StatsType type,
StatsReport::StatsValueName name,
int n) {
const StatsReport* report = FindNthReportByType(reports, type, n);
if (!report) {
- return absl::nullopt;
+ return std::nullopt;
}
std::string value;
if (!GetValue(report, name, &value)) {
- return absl::nullopt;
+ return std::nullopt;
}
return value;
}
@@ -1889,7 +1889,7 @@
// The SSRC in each SSRC report is different and correspond to the sender
// SSRC.
- std::vector<absl::optional<std::string>> ssrcs = {
+ std::vector<std::optional<std::string>> ssrcs = {
GetValueInNthReportByType(reports, StatsReport::kStatsReportTypeSsrc,
StatsReport::kStatsValueNameSsrc, 1),
GetValueInNthReportByType(reports, StatsReport::kStatsReportTypeSsrc,
diff --git a/pc/local_audio_source_unittest.cc b/pc/local_audio_source_unittest.cc
index 76d3b36..72cc20b 100644
--- a/pc/local_audio_source_unittest.cc
+++ b/pc/local_audio_source_unittest.cc
@@ -10,7 +10,8 @@
#include "pc/local_audio_source.h"
-#include "absl/types/optional.h"
+#include <optional>
+
#include "test/gtest.h"
using webrtc::LocalAudioSource;
@@ -26,5 +27,5 @@
TEST(LocalAudioSourceTest, InitWithNoOptions) {
rtc::scoped_refptr<LocalAudioSource> source =
LocalAudioSource::Create(nullptr);
- EXPECT_EQ(absl::nullopt, source->options().highpass_filter);
+ EXPECT_EQ(std::nullopt, source->options().highpass_filter);
}
diff --git a/pc/media_session.cc b/pc/media_session.cc
index 9cd1e9b..9287155 100644
--- a/pc/media_session.cc
+++ b/pc/media_session.cc
@@ -15,6 +15,7 @@
#include <algorithm>
#include <map>
#include <memory>
+#include <optional>
#include <string>
#include <unordered_map>
#include <utility>
@@ -23,7 +24,6 @@
#include "absl/algorithm/container.h"
#include "absl/strings/match.h"
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/field_trials_view.h"
#include "api/media_types.h"
#include "api/rtc_error.h"
@@ -452,7 +452,7 @@
negotiated_codec->packetization =
(local_codec.packetization == remote_codec.packetization)
? local_codec.packetization
- : absl::nullopt;
+ : std::nullopt;
}
#ifdef RTC_ENABLE_H265
@@ -461,16 +461,16 @@
Codec* negotiated_codec) {
negotiated_codec->tx_mode = (local_codec.tx_mode == remote_codec.tx_mode)
? local_codec.tx_mode
- : absl::nullopt;
+ : std::nullopt;
}
#endif
// Finds a codec in `codecs2` that matches `codec_to_match`, which is
// a member of `codecs1`. If `codec_to_match` is an RED or RTX codec, both
// the codecs themselves and their associated codecs must match.
-absl::optional<Codec> FindMatchingCodec(const std::vector<Codec>& codecs1,
- const std::vector<Codec>& codecs2,
- const Codec& codec_to_match) {
+std::optional<Codec> FindMatchingCodec(const std::vector<Codec>& codecs1,
+ const std::vector<Codec>& codecs2,
+ const Codec& codec_to_match) {
// `codec_to_match` should be a member of `codecs1`, in order to look up
// RED/RTX codecs' associated codecs correctly. If not, that's a programming
// error.
@@ -546,7 +546,7 @@
return potential_match;
}
}
- return absl::nullopt;
+ return std::nullopt;
}
void NegotiateCodecs(const std::vector<Codec>& local_codecs,
@@ -554,7 +554,7 @@
std::vector<Codec>* negotiated_codecs,
bool keep_offer_order) {
for (const Codec& ours : local_codecs) {
- absl::optional<Codec> theirs =
+ std::optional<Codec> theirs =
FindMatchingCodec(local_codecs, offered_codecs, ours);
// Note that we intentionally only find one matching codec for each of our
// local codecs, in case the remote offer contains duplicate codecs.
@@ -715,7 +715,7 @@
}
// Find a codec in the offered list that matches the reference codec.
// Its payload type may be different than the reference codec.
- absl::optional<Codec> matching_codec = FindMatchingCodec(
+ std::optional<Codec> matching_codec = FindMatchingCodec(
reference_codecs, *offered_codecs, *associated_codec);
if (!matching_codec) {
RTC_LOG(LS_WARNING)
@@ -735,7 +735,7 @@
const Codec* associated_codec =
GetAssociatedCodecForRed(reference_codecs, red_codec);
if (associated_codec) {
- absl::optional<Codec> matching_codec = FindMatchingCodec(
+ std::optional<Codec> matching_codec = FindMatchingCodec(
reference_codecs, *offered_codecs, *associated_codec);
if (!matching_codec) {
RTC_LOG(LS_WARNING) << "Couldn't find matching "
@@ -787,7 +787,7 @@
});
if (found_codec != supported_codecs.end()) {
- absl::optional<Codec> found_codec_with_correct_pt =
+ std::optional<Codec> found_codec_with_correct_pt =
FindMatchingCodec(supported_codecs, codecs, *found_codec);
if (found_codec_with_correct_pt) {
// RED may already have been added if its primary codec is before RED
@@ -1198,7 +1198,7 @@
}
// Add other supported codecs.
for (const Codec& codec : supported_codecs) {
- absl::optional<Codec> found_codec =
+ std::optional<Codec> found_codec =
FindMatchingCodec(supported_codecs, codecs, codec);
if (found_codec &&
!FindMatchingCodec(supported_codecs, filtered_codecs, codec)) {
@@ -1214,7 +1214,7 @@
RTC_DCHECK(referenced_codec);
// Find the codec we should be referencing and point to it.
- absl::optional<Codec> changed_referenced_codec = FindMatchingCodec(
+ std::optional<Codec> changed_referenced_codec = FindMatchingCodec(
supported_codecs, filtered_codecs, *referenced_codec);
if (changed_referenced_codec) {
found_codec->SetParam(kCodecParamAssociatedPayloadType,
@@ -1617,7 +1617,7 @@
IsMediaContentOfType(offer_content, media_description_options.type));
RTC_DCHECK(media_description_options.mid == offer_content->name);
// Get the index of the BUNDLE group that this MID belongs to, if any.
- absl::optional<size_t> bundle_index;
+ std::optional<size_t> bundle_index;
for (size_t i = 0; i < offer_bundles.size(); ++i) {
if (offer_bundles[i]->HasContentName(media_description_options.mid)) {
bundle_index = i;
diff --git a/pc/media_session_unittest.cc b/pc/media_session_unittest.cc
index c99526f..4ac9a5e 100644
--- a/pc/media_session_unittest.cc
+++ b/pc/media_session_unittest.cc
@@ -16,6 +16,7 @@
#include <cstdint>
#include <map>
#include <memory>
+#include <optional>
#include <string>
#include <tuple>
#include <vector>
@@ -23,7 +24,6 @@
#include "absl/algorithm/container.h"
#include "absl/strings/match.h"
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/candidate.h"
#include "api/rtp_parameters.h"
#include "media/base/codec.h"
@@ -4142,7 +4142,7 @@
ASSERT_EQ(1u, answer->contents().size());
vcd1 = answer->contents()[0].media_description()->as_video();
ASSERT_EQ(1u, vcd1->codecs().size());
- EXPECT_EQ(vcd1->codecs()[0].tx_mode, absl::nullopt);
+ EXPECT_EQ(vcd1->codecs()[0].tx_mode, std::nullopt);
}
#endif
@@ -4215,7 +4215,7 @@
ASSERT_EQ(1u, answer->contents().size());
vcd1 = answer->contents()[0].media_description()->as_video();
ASSERT_EQ(1u, vcd1->codecs().size());
- EXPECT_EQ(vcd1->codecs()[0].packetization, absl::nullopt);
+ EXPECT_EQ(vcd1->codecs()[0].packetization, std::nullopt);
}
// Test that the codec preference order per media section is respected in
diff --git a/pc/peer_connection.cc b/pc/peer_connection.cc
index 083d49b..861ad3f 100644
--- a/pc/peer_connection.cc
+++ b/pc/peer_connection.cc
@@ -15,6 +15,7 @@
#include <algorithm>
#include <memory>
+#include <optional>
#include <set>
#include <string>
#include <utility>
@@ -22,7 +23,6 @@
#include "absl/algorithm/container.h"
#include "absl/strings/match.h"
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/environment/environment.h"
#include "api/jsep_ice_candidate.h"
#include "api/media_types.h"
@@ -175,11 +175,11 @@
return kIceCandidatePairMax;
}
-absl::optional<int> RTCConfigurationToIceConfigOptionalInt(
+std::optional<int> RTCConfigurationToIceConfigOptionalInt(
int rtc_configuration_parameter) {
if (rtc_configuration_parameter ==
PeerConnectionInterface::RTCConfiguration::kUndefined) {
- return absl::nullopt;
+ return std::nullopt;
}
return rtc_configuration_parameter;
}
@@ -261,7 +261,7 @@
// set, which is done via SetLocalDescription.
RTCError ValidateIceCandidatePoolSize(
int ice_candidate_pool_size,
- absl::optional<int> previous_ice_candidate_pool_size) {
+ std::optional<int> previous_ice_candidate_pool_size) {
// Note that this isn't possible through chromium, since it's an unsigned
// short in WebIDL.
if (ice_candidate_pool_size < 0 ||
@@ -407,7 +407,7 @@
bool disable_ipv6_on_wifi;
int max_ipv6_networks;
bool disable_link_local_networks;
- absl::optional<int> screencast_min_bitrate;
+ std::optional<int> screencast_min_bitrate;
TcpCandidatePolicy tcp_candidate_policy;
CandidateNetworkPolicy candidate_network_policy;
int audio_jitter_buffer_max_packets;
@@ -424,27 +424,27 @@
bool enable_ice_renomination;
bool redetermine_role_on_ice_restart;
bool surface_ice_candidates_on_ice_transport_type_changed;
- absl::optional<int> ice_check_interval_strong_connectivity;
- absl::optional<int> ice_check_interval_weak_connectivity;
- absl::optional<int> ice_check_min_interval;
- absl::optional<int> ice_unwritable_timeout;
- absl::optional<int> ice_unwritable_min_checks;
- absl::optional<int> ice_inactive_timeout;
- absl::optional<int> stun_candidate_keepalive_interval;
+ std::optional<int> ice_check_interval_strong_connectivity;
+ std::optional<int> ice_check_interval_weak_connectivity;
+ std::optional<int> ice_check_min_interval;
+ std::optional<int> ice_unwritable_timeout;
+ std::optional<int> ice_unwritable_min_checks;
+ std::optional<int> ice_inactive_timeout;
+ std::optional<int> stun_candidate_keepalive_interval;
TurnCustomizer* turn_customizer;
SdpSemantics sdp_semantics;
- absl::optional<rtc::AdapterType> network_preference;
+ std::optional<rtc::AdapterType> network_preference;
bool active_reset_srtp_params;
- absl::optional<CryptoOptions> crypto_options;
+ std::optional<CryptoOptions> crypto_options;
bool offer_extmap_allow_mixed;
std::string turn_logging_id;
bool enable_implicit_rollback;
- absl::optional<int> report_usage_pattern_delay_ms;
- absl::optional<int> stable_writable_connection_ping_interval_ms;
+ std::optional<int> report_usage_pattern_delay_ms;
+ std::optional<int> stable_writable_connection_ping_interval_ms;
VpnPreference vpn_preference;
std::vector<rtc::NetworkMask> vpn_list;
PortAllocatorConfig port_allocator_config;
- absl::optional<TimeDelta> pacer_burst_interval;
+ std::optional<TimeDelta> pacer_burst_interval;
};
static_assert(sizeof(stuff_being_tested_for_equality) == sizeof(*this),
"Did you add something to RTCConfiguration and forget to "
@@ -1152,7 +1152,7 @@
}
auto result =
- cricket::CheckRtpParametersValues(parameters, codecs, absl::nullopt);
+ cricket::CheckRtpParametersValues(parameters, codecs, std::nullopt);
if (!result.ok()) {
if (result.type() == RTCErrorType::INVALID_MODIFICATION) {
result.set_type(RTCErrorType::UNSUPPORTED_OPERATION);
@@ -1411,18 +1411,18 @@
return ice_gathering_state_;
}
-absl::optional<bool> PeerConnection::can_trickle_ice_candidates() {
+std::optional<bool> PeerConnection::can_trickle_ice_candidates() {
RTC_DCHECK_RUN_ON(signaling_thread());
const SessionDescriptionInterface* description = current_remote_description();
if (!description) {
description = pending_remote_description();
}
if (!description) {
- return absl::nullopt;
+ return std::nullopt;
}
// TODO(bugs.webrtc.org/7443): Change to retrieve from session-level option.
if (description->description()->transport_infos().size() < 1) {
- return absl::nullopt;
+ return std::nullopt;
}
return description->description()->transport_infos()[0].description.HasOption(
"trickle");
@@ -1542,8 +1542,8 @@
RTCError validate_error = ValidateIceCandidatePoolSize(
configuration.ice_candidate_pool_size,
has_local_description
- ? absl::optional<int>(configuration_.ice_candidate_pool_size)
- : absl::nullopt);
+ ? std::optional<int>(configuration_.ice_candidate_pool_size)
+ : std::nullopt);
if (!validate_error.ok()) {
return validate_error;
}
@@ -2079,7 +2079,7 @@
RTC_DCHECK(!sctp_mid().has_value() || mid == sctp_mid().value());
RTC_LOG(LS_INFO) << "Creating data channel, mid=" << mid;
- absl::optional<std::string> transport_name =
+ std::optional<std::string> transport_name =
network_thread()->BlockingCall([&] {
RTC_DCHECK_RUN_ON(network_thread());
return SetupDataChannelTransport_n(mid);
@@ -2180,7 +2180,7 @@
int candidate_pool_size,
PortPrunePolicy turn_port_prune_policy,
TurnCustomizer* turn_customizer,
- absl::optional<int> stun_candidate_keepalive_interval,
+ std::optional<int> stun_candidate_keepalive_interval,
bool have_local_description) {
RTC_DCHECK_RUN_ON(network_thread());
port_allocator_->SetCandidateFilter(
@@ -2213,10 +2213,10 @@
env_.event_log().StopLogging();
}
-absl::optional<rtc::SSLRole> PeerConnection::GetSctpSslRole_n() {
+std::optional<rtc::SSLRole> PeerConnection::GetSctpSslRole_n() {
RTC_DCHECK_RUN_ON(network_thread());
return sctp_mid_n_ ? transport_controller_->GetDtlsRole(*sctp_mid_n_)
- : absl::nullopt;
+ : std::nullopt;
}
bool PeerConnection::GetSslRole(const std::string& content_name,
@@ -2261,11 +2261,11 @@
return data_channel_controller_.GetDataChannelStats();
}
-absl::optional<std::string> PeerConnection::sctp_transport_name() const {
+std::optional<std::string> PeerConnection::sctp_transport_name() const {
RTC_DCHECK_RUN_ON(signaling_thread());
if (sctp_mid_s_ && transport_controller_copy_)
return sctp_transport_name_s_;
- return absl::optional<std::string>();
+ return std::optional<std::string>();
}
void PeerConnection::SetSctpTransportName(std::string sctp_transport_name) {
@@ -2274,7 +2274,7 @@
ClearStatsCache();
}
-absl::optional<std::string> PeerConnection::sctp_mid() const {
+std::optional<std::string> PeerConnection::sctp_mid() const {
RTC_DCHECK_RUN_ON(signaling_thread());
return sctp_mid_s_;
}
@@ -2489,14 +2489,14 @@
}
}
-absl::optional<AudioDeviceModule::Stats> PeerConnection::GetAudioDeviceStats() {
+std::optional<AudioDeviceModule::Stats> PeerConnection::GetAudioDeviceStats() {
if (context_->media_engine()) {
return context_->media_engine()->voice().GetAudioDeviceStats();
}
- return absl::nullopt;
+ return std::nullopt;
}
-absl::optional<std::string> PeerConnection::SetupDataChannelTransport_n(
+std::optional<std::string> PeerConnection::SetupDataChannelTransport_n(
absl::string_view mid) {
sctp_mid_n_ = std::string(mid);
DataChannelTransportInterface* transport =
@@ -2505,11 +2505,11 @@
RTC_LOG(LS_ERROR)
<< "Data channel transport is not available for data channels, mid="
<< mid;
- sctp_mid_n_ = absl::nullopt;
- return absl::nullopt;
+ sctp_mid_n_ = std::nullopt;
+ return std::nullopt;
}
- absl::optional<std::string> transport_name;
+ std::optional<std::string> transport_name;
cricket::DtlsTransportInternal* dtls_transport =
transport_controller_->GetDtlsTransport(*sctp_mid_n_);
if (dtls_transport) {
diff --git a/pc/peer_connection.h b/pc/peer_connection.h
index dcf6eb6..16e43c7 100644
--- a/pc/peer_connection.h
+++ b/pc/peer_connection.h
@@ -16,11 +16,11 @@
#include <functional>
#include <map>
#include <memory>
+#include <optional>
#include <set>
#include <string>
#include <vector>
-#include "absl/types/optional.h"
#include "api/adaptation/resource.h"
#include "api/async_dns_resolver.h"
#include "api/candidate.h"
@@ -185,7 +185,7 @@
IceConnectionState standardized_ice_connection_state() override;
PeerConnectionState peer_connection_state() override;
IceGatheringState ice_gathering_state() override;
- absl::optional<bool> can_trickle_ice_candidates() override;
+ std::optional<bool> can_trickle_ice_candidates() override;
const SessionDescriptionInterface* local_description() const override;
const SessionDescriptionInterface* remote_description() const override;
@@ -286,15 +286,15 @@
std::vector<DataChannelStats> GetDataChannelStats() const override;
- absl::optional<std::string> sctp_transport_name() const override;
- absl::optional<std::string> sctp_mid() const override;
+ std::optional<std::string> sctp_transport_name() const override;
+ std::optional<std::string> sctp_mid() const override;
cricket::CandidateStatsList GetPooledCandidateStats() const override;
std::map<std::string, cricket::TransportStats> GetTransportStatsByNames(
const std::set<std::string>& transport_names) override;
Call::Stats GetCallStats() override;
- absl::optional<AudioDeviceModule::Stats> GetAudioDeviceStats() override;
+ std::optional<AudioDeviceModule::Stats> GetAudioDeviceStats() override;
bool GetLocalCertificate(
const std::string& transport_name,
@@ -315,7 +315,7 @@
sdp_handler_->signaling_state() == PeerConnectionInterface::kClosed;
}
// Get current SSL role used by SCTP's underlying transport.
- absl::optional<rtc::SSLRole> GetSctpSslRole_n() override;
+ std::optional<rtc::SSLRole> GetSctpSslRole_n() override;
void OnSctpDataChannelStateChanged(
int channel_id,
@@ -423,7 +423,7 @@
// this session.
bool SrtpRequired() const override;
- absl::optional<std::string> SetupDataChannelTransport_n(absl::string_view mid)
+ std::optional<std::string> SetupDataChannelTransport_n(absl::string_view mid)
RTC_RUN_ON(network_thread());
void TeardownDataChannelTransport_n(RTCError error)
RTC_RUN_ON(network_thread());
@@ -524,7 +524,7 @@
int candidate_pool_size,
PortPrunePolicy turn_port_prune_policy,
TurnCustomizer* turn_customizer,
- absl::optional<int> stun_candidate_keepalive_interval,
+ std::optional<int> stun_candidate_keepalive_interval,
bool have_local_description);
// Starts output of an RTC event log to the given output object.
@@ -680,8 +680,8 @@
// There is one copy on the signaling thread and another copy on the
// networking thread. Changes are always initiated from the signaling
// thread, but applied first on the networking thread via an invoke().
- absl::optional<std::string> sctp_mid_s_ RTC_GUARDED_BY(signaling_thread());
- absl::optional<std::string> sctp_mid_n_ RTC_GUARDED_BY(network_thread());
+ std::optional<std::string> sctp_mid_s_ RTC_GUARDED_BY(signaling_thread());
+ std::optional<std::string> sctp_mid_n_ RTC_GUARDED_BY(network_thread());
std::string sctp_transport_name_s_ RTC_GUARDED_BY(signaling_thread());
// The machinery for handling offers and answers. Const after initialization.
diff --git a/pc/peer_connection_adaptation_integrationtest.cc b/pc/peer_connection_adaptation_integrationtest.cc
index 882fa36..93aa1eb 100644
--- a/pc/peer_connection_adaptation_integrationtest.cc
+++ b/pc/peer_connection_adaptation_integrationtest.cc
@@ -11,9 +11,9 @@
#include <stdint.h>
#include <memory>
+#include <optional>
#include <string>
-#include "absl/types/optional.h"
#include "api/adaptation/resource.h"
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
diff --git a/pc/peer_connection_crypto_unittest.cc b/pc/peer_connection_crypto_unittest.cc
index 9ba20ca..c808b0a 100644
--- a/pc/peer_connection_crypto_unittest.cc
+++ b/pc/peer_connection_crypto_unittest.cc
@@ -11,6 +11,7 @@
#include <stddef.h>
#include <memory>
+#include <optional>
#include <ostream>
#include <string>
#include <tuple>
@@ -18,7 +19,6 @@
#include <utility>
#include <vector>
-#include "absl/types/optional.h"
#include "api/audio/audio_device.h"
#include "api/audio/audio_mixer.h"
#include "api/audio/audio_processing.h"
diff --git a/pc/peer_connection_data_channel_unittest.cc b/pc/peer_connection_data_channel_unittest.cc
index 8682cfa..0368d82 100644
--- a/pc/peer_connection_data_channel_unittest.cc
+++ b/pc/peer_connection_data_channel_unittest.cc
@@ -9,12 +9,12 @@
*/
#include <memory>
+#include <optional>
#include <string>
#include <type_traits>
#include <utility>
#include <vector>
-#include "absl/types/optional.h"
#include "api/jsep.h"
#include "api/media_types.h"
#include "api/peer_connection_interface.h"
@@ -82,11 +82,11 @@
sctp_transport_factory_ = sctp_transport_factory;
}
- absl::optional<std::string> sctp_mid() {
+ std::optional<std::string> sctp_mid() {
return GetInternalPeerConnection()->sctp_mid();
}
- absl::optional<std::string> sctp_transport_name() {
+ std::optional<std::string> sctp_transport_name() {
return GetInternalPeerConnection()->sctp_transport_name();
}
diff --git a/pc/peer_connection_encodings_integrationtest.cc b/pc/peer_connection_encodings_integrationtest.cc
index fcbc9fc..9b4bfba 100644
--- a/pc/peer_connection_encodings_integrationtest.cc
+++ b/pc/peer_connection_encodings_integrationtest.cc
@@ -12,6 +12,7 @@
#include <cstddef>
#include <cstdint>
#include <memory>
+#include <optional>
#include <set>
#include <string>
#include <string_view>
@@ -19,7 +20,6 @@
#include "absl/strings/match.h"
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
#include "api/audio_options.h"
@@ -233,14 +233,14 @@
size_t index,
const std::string& codec_id) {
return IsCodecIdDifferentWithScalabilityMode(pc_wrapper, index, codec_id,
- absl::nullopt);
+ std::nullopt);
}
bool IsCodecIdDifferentWithScalabilityMode(
rtc::scoped_refptr<PeerConnectionTestWrapper> pc_wrapper,
size_t index,
const std::string& codec_id,
- absl::optional<std::string> wanted_scalability_mode) {
+ std::optional<std::string> wanted_scalability_mode) {
rtc::scoped_refptr<const RTCStatsReport> report = GetStats(pc_wrapper);
std::vector<const RTCOutboundRtpStreamStats*> outbound_rtps =
report->GetStatsOfType<RTCOutboundRtpStreamStats>();
@@ -513,7 +513,7 @@
// `scalability_mode` remains unset because SetParameters() failed.
parameters = sender->GetParameters();
ASSERT_EQ(parameters.encodings.size(), 1u);
- EXPECT_THAT(parameters.encodings[0].scalability_mode, Eq(absl::nullopt));
+ EXPECT_THAT(parameters.encodings[0].scalability_mode, Eq(std::nullopt));
local_pc_wrapper->WaitForConnection();
remote_pc_wrapper->WaitForConnection();
@@ -532,7 +532,7 @@
// GetParameters() confirms `scalability_mode` is still not set.
parameters = sender->GetParameters();
ASSERT_EQ(parameters.encodings.size(), 1u);
- EXPECT_THAT(parameters.encodings[0].scalability_mode, Eq(absl::nullopt));
+ EXPECT_THAT(parameters.encodings[0].scalability_mode, Eq(std::nullopt));
}
TEST_F(PeerConnectionEncodingsIntegrationTest,
@@ -626,7 +626,7 @@
local_pc_wrapper->WaitForConnection();
remote_pc_wrapper->WaitForConnection();
// `scalaiblity_mode` is assigned the fallback value "L1T2" which is different
- // than the default of absl::nullopt.
+ // than the default of std::nullopt.
parameters = sender->GetParameters();
ASSERT_EQ(parameters.encodings.size(), 1u);
EXPECT_THAT(parameters.encodings[0].scalability_mode,
@@ -838,9 +838,9 @@
parameters.encodings[0].scalability_mode = "L2T2_KEY";
parameters.encodings[0].scale_resolution_down_by = 2.0;
parameters.encodings[1].active = false;
- parameters.encodings[1].scalability_mode = absl::nullopt;
+ parameters.encodings[1].scalability_mode = std::nullopt;
parameters.encodings[2].active = false;
- parameters.encodings[2].scalability_mode = absl::nullopt;
+ parameters.encodings[2].scalability_mode = std::nullopt;
sender->SetParameters(parameters);
// Since the standard API is configuring simulcast we get three outbound-rtps,
@@ -890,7 +890,7 @@
parameters.encodings[1].scalability_mode = "L1T1";
parameters.encodings[1].scale_resolution_down_by = 2.0;
parameters.encodings[2].active = false;
- parameters.encodings[2].scalability_mode = absl::nullopt;
+ parameters.encodings[2].scalability_mode = std::nullopt;
EXPECT_TRUE(sender->SetParameters(parameters).ok());
// The original negotiation triggers legacy SVC because we didn't specify
@@ -920,7 +920,7 @@
Optional(StrEq("L1T3")));
EXPECT_THAT(parameters.encodings[1].scalability_mode,
Optional(StrEq("L1T1")));
- EXPECT_THAT(parameters.encodings[2].scalability_mode, Eq(absl::nullopt));
+ EXPECT_THAT(parameters.encodings[2].scalability_mode, Eq(std::nullopt));
}
TEST_F(PeerConnectionEncodingsIntegrationTest,
@@ -976,7 +976,7 @@
// GetParameters() does not report any fallback.
parameters = sender->GetParameters();
ASSERT_THAT(parameters.encodings, SizeIs(3));
- EXPECT_THAT(parameters.encodings[0].scalability_mode, Eq(absl::nullopt));
+ EXPECT_THAT(parameters.encodings[0].scalability_mode, Eq(std::nullopt));
EXPECT_THAT(parameters.encodings[1].scalability_mode,
Optional(StrEq("L1T1")));
EXPECT_THAT(parameters.encodings[2].scalability_mode,
@@ -984,14 +984,14 @@
// Switch to legacy SVC mode.
parameters.encodings[0].active = true;
- parameters.encodings[0].scalability_mode = absl::nullopt;
- parameters.encodings[0].scale_resolution_down_by = absl::nullopt;
+ parameters.encodings[0].scalability_mode = std::nullopt;
+ parameters.encodings[0].scale_resolution_down_by = std::nullopt;
parameters.encodings[1].active = true;
- parameters.encodings[1].scalability_mode = absl::nullopt;
- parameters.encodings[1].scale_resolution_down_by = absl::nullopt;
+ parameters.encodings[1].scalability_mode = std::nullopt;
+ parameters.encodings[1].scale_resolution_down_by = std::nullopt;
parameters.encodings[2].active = false;
- parameters.encodings[2].scalability_mode = absl::nullopt;
- parameters.encodings[2].scale_resolution_down_by = absl::nullopt;
+ parameters.encodings[2].scalability_mode = std::nullopt;
+ parameters.encodings[2].scale_resolution_down_by = std::nullopt;
EXPECT_TRUE(sender->SetParameters(parameters).ok());
// Ensure that we are getting VGA at L1T3 from the "f" rid.
@@ -1284,7 +1284,7 @@
/*audio=*/true, {}, /*video=*/false, {});
rtc::scoped_refptr<AudioTrackInterface> track = stream->GetAudioTracks()[0];
- absl::optional<RtpCodecCapability> pcmu =
+ std::optional<RtpCodecCapability> pcmu =
local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_AUDIO,
"pcmu");
ASSERT_TRUE(pcmu);
@@ -1325,7 +1325,7 @@
/*audio=*/false, {}, /*video=*/true, {.width = 1280, .height = 720});
rtc::scoped_refptr<VideoTrackInterface> track = stream->GetVideoTracks()[0];
- absl::optional<RtpCodecCapability> vp9 =
+ std::optional<RtpCodecCapability> vp9 =
local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_VIDEO,
"vp9");
ASSERT_TRUE(vp9);
@@ -1372,7 +1372,7 @@
/*audio=*/true, {}, /*video=*/false, {});
rtc::scoped_refptr<AudioTrackInterface> track = stream->GetAudioTracks()[0];
- absl::optional<RtpCodecCapability> pcmu =
+ std::optional<RtpCodecCapability> pcmu =
local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_AUDIO,
"pcmu");
@@ -1409,7 +1409,7 @@
/*audio=*/true, {}, /*video=*/false, {});
rtc::scoped_refptr<AudioTrackInterface> track = stream->GetAudioTracks()[0];
- absl::optional<RtpCodecCapability> pcmu =
+ std::optional<RtpCodecCapability> pcmu =
local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_AUDIO,
"pcmu");
@@ -1457,7 +1457,7 @@
/*audio=*/false, {}, /*video=*/true, {.width = 1280, .height = 720});
rtc::scoped_refptr<VideoTrackInterface> track = stream->GetVideoTracks()[0];
- absl::optional<RtpCodecCapability> vp9 =
+ std::optional<RtpCodecCapability> vp9 =
local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_VIDEO,
"vp9");
@@ -1500,7 +1500,7 @@
/*audio=*/false, {}, /*video=*/true, {.width = 1280, .height = 720});
rtc::scoped_refptr<VideoTrackInterface> track = stream->GetVideoTracks()[0];
- absl::optional<RtpCodecCapability> vp9 =
+ std::optional<RtpCodecCapability> vp9 =
local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_VIDEO,
"vp9");
@@ -1635,7 +1635,7 @@
rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc();
ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper);
- absl::optional<RtpCodecCapability> opus =
+ std::optional<RtpCodecCapability> opus =
local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_AUDIO,
"opus");
ASSERT_TRUE(opus);
@@ -1674,7 +1674,7 @@
rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc();
ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper);
- absl::optional<RtpCodecCapability> opus =
+ std::optional<RtpCodecCapability> opus =
local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_AUDIO,
"opus");
ASSERT_TRUE(opus);
@@ -1736,7 +1736,7 @@
rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc();
ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper);
- absl::optional<RtpCodecCapability> vp8 =
+ std::optional<RtpCodecCapability> vp8 =
local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_VIDEO,
"vp8");
ASSERT_TRUE(vp8);
@@ -1775,7 +1775,7 @@
rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc();
ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper);
- absl::optional<RtpCodecCapability> vp8 =
+ std::optional<RtpCodecCapability> vp8 =
local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_VIDEO,
"vp8");
ASSERT_TRUE(vp8);
@@ -1837,7 +1837,7 @@
rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc();
ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper);
- absl::optional<RtpCodecCapability> opus =
+ std::optional<RtpCodecCapability> opus =
local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_AUDIO,
"opus");
ASSERT_TRUE(opus);
@@ -1890,12 +1890,12 @@
->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_AUDIO)
.codecs;
- absl::optional<RtpCodecCapability> opus =
+ std::optional<RtpCodecCapability> opus =
local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_AUDIO,
"opus");
ASSERT_TRUE(opus);
- absl::optional<RtpCodecCapability> red =
+ std::optional<RtpCodecCapability> red =
local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_AUDIO,
"red");
ASSERT_TRUE(red);
@@ -1943,7 +1943,7 @@
SetParametersRejectsScalabilityModeForSelectedCodec) {
rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc();
- absl::optional<RtpCodecCapability> vp8 =
+ std::optional<RtpCodecCapability> vp8 =
local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_VIDEO,
"vp8");
ASSERT_TRUE(vp8);
@@ -1973,7 +1973,7 @@
rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc();
ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper);
- absl::optional<RtpCodecCapability> vp8 =
+ std::optional<RtpCodecCapability> vp8 =
local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_VIDEO,
"vp8");
ASSERT_TRUE(vp8);
@@ -2031,11 +2031,11 @@
rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc();
ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper);
- absl::optional<RtpCodecCapability> vp8 =
+ std::optional<RtpCodecCapability> vp8 =
local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_VIDEO,
"vp8");
ASSERT_TRUE(vp8);
- absl::optional<RtpCodecCapability> vp9 =
+ std::optional<RtpCodecCapability> vp9 =
local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_VIDEO,
"vp9");
diff --git a/pc/peer_connection_end_to_end_unittest.cc b/pc/peer_connection_end_to_end_unittest.cc
index 1fc3d7e..cc57c7b 100644
--- a/pc/peer_connection_end_to_end_unittest.cc
+++ b/pc/peer_connection_end_to_end_unittest.cc
@@ -13,13 +13,13 @@
#include <cstddef>
#include <limits>
#include <memory>
+#include <optional>
#include <string>
#include <type_traits>
#include <utility>
#include <vector>
#include "absl/strings/match.h"
-#include "absl/types/optional.h"
#include "api/audio_codecs/L16/audio_decoder_L16.h"
#include "api/audio_codecs/L16/audio_encoder_L16.h"
#include "api/audio_codecs/audio_codec_pair_id.h"
@@ -305,7 +305,7 @@
[real_decoder_factory](
const webrtc::Environment& env,
const webrtc::SdpAudioFormat& format,
- absl::optional<webrtc::AudioCodecPairId> codec_pair_id) {
+ std::optional<webrtc::AudioCodecPairId> codec_pair_id) {
auto real_decoder =
real_decoder_factory->Create(env, format, codec_pair_id);
return real_decoder
@@ -317,7 +317,7 @@
struct AudioEncoderUnicornSparklesRainbow {
using Config = webrtc::AudioEncoderL16::Config;
- static absl::optional<Config> SdpToConfig(webrtc::SdpAudioFormat format) {
+ static std::optional<Config> SdpToConfig(webrtc::SdpAudioFormat format) {
if (absl::EqualsIgnoreCase(format.name, "UnicornSparklesRainbow")) {
const webrtc::CodecParameterMap expected_params = {{"num_horns", "1"}};
EXPECT_EQ(expected_params, format.parameters);
@@ -325,7 +325,7 @@
format.name = "L16";
return webrtc::AudioEncoderL16::SdpToConfig(format);
} else {
- return absl::nullopt;
+ return std::nullopt;
}
}
static void AppendSupportedEncoders(
@@ -345,7 +345,7 @@
static std::unique_ptr<webrtc::AudioEncoder> MakeAudioEncoder(
const Config& config,
int payload_type,
- absl::optional<webrtc::AudioCodecPairId> codec_pair_id = absl::nullopt) {
+ std::optional<webrtc::AudioCodecPairId> codec_pair_id = std::nullopt) {
return webrtc::AudioEncoderL16::MakeAudioEncoder(config, payload_type,
codec_pair_id);
}
@@ -353,7 +353,7 @@
struct AudioDecoderUnicornSparklesRainbow {
using Config = webrtc::AudioDecoderL16::Config;
- static absl::optional<Config> SdpToConfig(webrtc::SdpAudioFormat format) {
+ static std::optional<Config> SdpToConfig(webrtc::SdpAudioFormat format) {
if (absl::EqualsIgnoreCase(format.name, "UnicornSparklesRainbow")) {
const webrtc::CodecParameterMap expected_params = {{"num_horns", "1"}};
EXPECT_EQ(expected_params, format.parameters);
@@ -361,7 +361,7 @@
format.name = "L16";
return webrtc::AudioDecoderL16::SdpToConfig(format);
} else {
- return absl::nullopt;
+ return std::nullopt;
}
}
static void AppendSupportedDecoders(
@@ -377,7 +377,7 @@
}
static std::unique_ptr<webrtc::AudioDecoder> MakeAudioDecoder(
const Config& config,
- absl::optional<webrtc::AudioCodecPairId> codec_pair_id = absl::nullopt) {
+ std::optional<webrtc::AudioCodecPairId> codec_pair_id = std::nullopt) {
return webrtc::AudioDecoderL16::MakeAudioDecoder(config, codec_pair_id);
}
};
@@ -415,7 +415,7 @@
std::vector<webrtc::AudioCodecSpec> GetSupportedEncoders() override {
return fact_->GetSupportedEncoders();
}
- absl::optional<webrtc::AudioCodecInfo> QueryAudioEncoder(
+ std::optional<webrtc::AudioCodecInfo> QueryAudioEncoder(
const webrtc::SdpAudioFormat& format) override {
return fact_->QueryAudioEncoder(format);
}
@@ -448,7 +448,7 @@
std::unique_ptr<webrtc::AudioDecoder> Create(
const Environment& env,
const webrtc::SdpAudioFormat& format,
- absl::optional<webrtc::AudioCodecPairId> codec_pair_id) override {
+ std::optional<webrtc::AudioCodecPairId> codec_pair_id) override {
EXPECT_TRUE(codec_pair_id.has_value());
codec_ids_->push_back(*codec_pair_id);
return fact_->Create(env, format, codec_pair_id);
diff --git a/pc/peer_connection_header_extension_unittest.cc b/pc/peer_connection_header_extension_unittest.cc
index 15b1ae6..61805a7 100644
--- a/pc/peer_connection_header_extension_unittest.cc
+++ b/pc/peer_connection_header_extension_unittest.cc
@@ -9,13 +9,13 @@
*/
#include <memory>
+#include <optional>
#include <string>
#include <tuple>
#include <utility>
#include <vector>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/jsep.h"
#include "api/media_types.h"
#include "api/peer_connection_interface.h"
@@ -76,7 +76,7 @@
std::unique_ptr<PeerConnectionWrapper> CreatePeerConnection(
cricket::MediaType media_type,
- absl::optional<SdpSemantics> semantics) {
+ std::optional<SdpSemantics> semantics) {
auto media_engine = std::make_unique<cricket::FakeMediaEngine>();
if (media_type == cricket::MediaType::MEDIA_TYPE_AUDIO)
media_engine->fake_voice_engine()->SetRtpHeaderExtensions(extensions_);
diff --git a/pc/peer_connection_histogram_unittest.cc b/pc/peer_connection_histogram_unittest.cc
index 365f58a..0697c67 100644
--- a/pc/peer_connection_histogram_unittest.cc
+++ b/pc/peer_connection_histogram_unittest.cc
@@ -9,12 +9,12 @@
*/
#include <memory>
+#include <optional>
#include <set>
#include <string>
#include <utility>
#include <vector>
-#include "absl/types/optional.h"
#include "api/jsep.h"
#include "api/jsep_session_description.h"
#include "api/peer_connection_interface.h"
@@ -104,18 +104,18 @@
bool HaveDataChannel() { return last_datachannel_ != nullptr; }
- absl::optional<int> interesting_usage_detected() {
+ std::optional<int> interesting_usage_detected() {
return interesting_usage_detected_;
}
void ClearInterestingUsageDetector() {
- interesting_usage_detected_ = absl::optional<int>();
+ interesting_usage_detected_ = std::optional<int>();
}
bool candidate_gathered() const { return candidate_gathered_; }
private:
- absl::optional<int> interesting_usage_detected_;
+ std::optional<int> interesting_usage_detected_;
bool candidate_gathered_ = false;
RawWrapperPtr candidate_target_; // Note: Not thread-safe against deletions.
};
@@ -728,7 +728,7 @@
(expected_fingerprint |
static_cast<int>(UsageEvent::PRIVATE_CANDIDATE_COLLECTED)) ==
ObservedFingerprint());
- EXPECT_METRIC_EQ(absl::make_optional(ObservedFingerprint()),
+ EXPECT_METRIC_EQ(std::make_optional(ObservedFingerprint()),
caller->observer()->interesting_usage_detected());
}
@@ -748,7 +748,7 @@
(expected_fingerprint |
static_cast<int>(UsageEvent::PRIVATE_CANDIDATE_COLLECTED)) ==
ObservedFingerprint());
- EXPECT_METRIC_EQ(absl::make_optional(ObservedFingerprint()),
+ EXPECT_METRIC_EQ(std::make_optional(ObservedFingerprint()),
caller->observer()->interesting_usage_detected());
}
diff --git a/pc/peer_connection_ice_unittest.cc b/pc/peer_connection_ice_unittest.cc
index 59541e5..ecf7da6 100644
--- a/pc/peer_connection_ice_unittest.cc
+++ b/pc/peer_connection_ice_unittest.cc
@@ -12,13 +12,13 @@
#include <stdint.h>
#include <memory>
+#include <optional>
#include <string>
#include <tuple>
#include <type_traits>
#include <utility>
#include <vector>
-#include "absl/types/optional.h"
#include "api/audio/audio_device.h"
#include "api/audio/audio_mixer.h"
#include "api/audio/audio_processing.h"
@@ -1473,7 +1473,7 @@
config.ice_candidate_pool_size = 1;
CreatePeerConnection(config);
ASSERT_NE(port_allocator_, nullptr);
- absl::optional<int> actual_stun_keepalive_interval =
+ std::optional<int> actual_stun_keepalive_interval =
port_allocator_->stun_candidate_keepalive_interval();
EXPECT_EQ(actual_stun_keepalive_interval.value_or(-1), 123);
config.stun_candidate_keepalive_interval = 321;
diff --git a/pc/peer_connection_integrationtest.cc b/pc/peer_connection_integrationtest.cc
index 7bf6b6d..774e08b 100644
--- a/pc/peer_connection_integrationtest.cc
+++ b/pc/peer_connection_integrationtest.cc
@@ -19,6 +19,7 @@
#include <algorithm>
#include <memory>
+#include <optional>
#include <string>
#include <tuple>
#include <utility>
@@ -27,7 +28,6 @@
#include "absl/algorithm/container.h"
#include "absl/memory/memory.h"
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/candidate.h"
#include "api/crypto/crypto_options.h"
#include "api/dtmf_sender_interface.h"
@@ -2353,7 +2353,7 @@
ASSERT_EQ(first_candidate_stats.size(), 0u);
// Add a "fake" candidate.
- absl::optional<RTCError> result;
+ std::optional<RTCError> result;
caller()->pc()->AddIceCandidate(
absl::WrapUnique(CreateIceCandidate(
"", 0,
diff --git a/pc/peer_connection_interface_unittest.cc b/pc/peer_connection_interface_unittest.cc
index 8c17a4a..539b952 100644
--- a/pc/peer_connection_interface_unittest.cc
+++ b/pc/peer_connection_interface_unittest.cc
@@ -13,12 +13,12 @@
#include <limits.h>
#include <stdint.h>
+#include <optional>
#include <string>
#include <utility>
#include <vector>
#include "absl/strings/str_replace.h"
-#include "absl/types/optional.h"
#include "api/audio/audio_device.h"
#include "api/audio/audio_mixer.h"
#include "api/audio/audio_processing.h"
@@ -1955,7 +1955,7 @@
EXPECT_FALSE(channel.value()->reliable());
EXPECT_FALSE(observer_.renegotiation_needed_);
- config.maxRetransmits = absl::nullopt;
+ config.maxRetransmits = std::nullopt;
config.maxRetransmitTime = 0;
channel = pc_->CreateDataChannelOrError("4", &config);
EXPECT_TRUE(channel.ok());
@@ -2223,7 +2223,7 @@
// require a very complex set of mocks.
TEST_P(PeerConnectionInterfaceTest, SetConfigurationChangesIceCheckInterval) {
PeerConnectionInterface::RTCConfiguration config;
- config.ice_check_min_interval = absl::nullopt;
+ config.ice_check_min_interval = std::nullopt;
CreatePeerConnection(config);
config = pc_->GetConfiguration();
config.ice_check_min_interval = 100;
diff --git a/pc/peer_connection_internal.h b/pc/peer_connection_internal.h
index 75e18f2..f3c4707 100644
--- a/pc/peer_connection_internal.h
+++ b/pc/peer_connection_internal.h
@@ -13,11 +13,11 @@
#include <map>
#include <memory>
+#include <optional>
#include <set>
#include <string>
#include <vector>
-#include "absl/types/optional.h"
#include "api/audio/audio_device.h"
#include "api/peer_connection_interface.h"
#include "call/call.h"
@@ -47,7 +47,7 @@
// bundling, returns false.
virtual bool NeedsIceRestart(const std::string& content_name) const = 0;
- virtual absl::optional<std::string> sctp_mid() const = 0;
+ virtual std::optional<std::string> sctp_mid() const = 0;
// Functions below this comment are known to only be accessed
// from SdpOfferAnswerHandler.
@@ -76,7 +76,7 @@
virtual LegacyStatsCollector* legacy_stats() = 0;
// Returns the observer. Will crash on CHECK if the observer is removed.
virtual PeerConnectionObserver* Observer() const = 0;
- virtual absl::optional<rtc::SSLRole> GetSctpSslRole_n() = 0;
+ virtual std::optional<rtc::SSLRole> GetSctpSslRole_n() = 0;
virtual PeerConnectionInterface::IceConnectionState
ice_connection_state_internal() = 0;
virtual void SetIceConnectionState(
@@ -153,7 +153,7 @@
return {};
}
- virtual absl::optional<std::string> sctp_transport_name() const = 0;
+ virtual std::optional<std::string> sctp_transport_name() const = 0;
virtual cricket::CandidateStatsList GetPooledCandidateStats() const = 0;
@@ -165,7 +165,7 @@
virtual Call::Stats GetCallStats() = 0;
- virtual absl::optional<AudioDeviceModule::Stats> GetAudioDeviceStats() = 0;
+ virtual std::optional<AudioDeviceModule::Stats> GetAudioDeviceStats() = 0;
virtual bool GetLocalCertificate(
const std::string& transport_name,
diff --git a/pc/peer_connection_jsep_unittest.cc b/pc/peer_connection_jsep_unittest.cc
index fa033ae..cd1689a 100644
--- a/pc/peer_connection_jsep_unittest.cc
+++ b/pc/peer_connection_jsep_unittest.cc
@@ -13,13 +13,13 @@
#include <algorithm>
#include <map>
#include <memory>
+#include <optional>
#include <string>
#include <tuple>
#include <utility>
#include <vector>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/audio/audio_device.h"
#include "api/enable_media_with_defaults.h"
#include "api/field_trials_view.h"
@@ -331,7 +331,7 @@
auto transceivers = callee->pc()->GetTransceivers();
ASSERT_EQ(2u, transceivers.size());
- EXPECT_EQ(absl::nullopt, transceivers[0]->mid());
+ EXPECT_EQ(std::nullopt, transceivers[0]->mid());
EXPECT_EQ(caller_audio->mid(), transceivers[1]->mid());
}
@@ -349,7 +349,7 @@
auto transceivers = callee->pc()->GetTransceivers();
ASSERT_EQ(2u, transceivers.size());
- EXPECT_EQ(absl::nullopt, transceivers[0]->mid());
+ EXPECT_EQ(std::nullopt, transceivers[0]->mid());
EXPECT_EQ(caller->pc()->GetTransceivers()[0]->mid(), transceivers[1]->mid());
EXPECT_EQ(MediaStreamTrackInterface::kAudioKind,
transceivers[1]->receiver()->track()->kind());
@@ -368,7 +368,7 @@
auto transceivers = callee->pc()->GetTransceivers();
ASSERT_EQ(2u, transceivers.size());
- EXPECT_EQ(absl::nullopt, transceivers[0]->mid());
+ EXPECT_EQ(std::nullopt, transceivers[0]->mid());
EXPECT_EQ(caller->pc()->GetTransceivers()[0]->mid(), transceivers[1]->mid());
EXPECT_EQ(MediaStreamTrackInterface::kAudioKind,
transceivers[1]->receiver()->track()->kind());
@@ -676,7 +676,7 @@
caller->SetRemoteDescription(callee->CreateAnswerAndSetAsLocal()));
EXPECT_TRUE(first_transceiver->stopped());
// First transceivers are dissociated on caller side.
- ASSERT_EQ(absl::nullopt, first_transceiver->mid());
+ ASSERT_EQ(std::nullopt, first_transceiver->mid());
// They are disassociated on callee side.
ASSERT_EQ(0u, callee->pc()->GetTransceivers().size());
@@ -695,7 +695,7 @@
// associate the new transceivers.
ASSERT_TRUE(
caller->SetLocalDescription(CloneSessionDescription(offer.get())));
- EXPECT_EQ(absl::nullopt, first_transceiver->mid());
+ EXPECT_EQ(std::nullopt, first_transceiver->mid());
ASSERT_EQ(1u, caller->pc()->GetTransceivers().size());
EXPECT_EQ(second_mid, caller->pc()->GetTransceivers()[0]->mid());
ASSERT_TRUE(callee->SetRemoteDescription(std::move(offer)));
@@ -814,7 +814,7 @@
// the MID for the new transceiver.
ASSERT_TRUE(
caller->SetLocalDescription(CloneSessionDescription(offer.get())));
- EXPECT_EQ(absl::nullopt, first_transceiver->mid());
+ EXPECT_EQ(std::nullopt, first_transceiver->mid());
EXPECT_EQ(second_mid, second_transceiver->mid());
// Setting the remote offer will dissociate the previous transceiver and
@@ -877,7 +877,7 @@
// the MID for the new transceiver.
ASSERT_TRUE(
caller->SetLocalDescription(CloneSessionDescription(offer.get())));
- EXPECT_EQ(absl::nullopt, caller_first_transceiver->mid());
+ EXPECT_EQ(std::nullopt, caller_first_transceiver->mid());
EXPECT_EQ(second_mid, caller_second_transceiver->mid());
// Setting the remote offer will dissociate the previous transceiver and
@@ -940,7 +940,7 @@
// the MID for the new transceiver.
ASSERT_TRUE(
callee->SetLocalDescription(CloneSessionDescription(offer.get())));
- EXPECT_EQ(absl::nullopt, callee_first_transceiver->mid());
+ EXPECT_EQ(std::nullopt, callee_first_transceiver->mid());
EXPECT_EQ(second_mid, callee_second_transceiver->mid());
// Setting the remote offer will dissociate the previous transceiver and
@@ -1945,7 +1945,7 @@
// Transceiver can't be removed as track was added to it.
EXPECT_EQ(callee->pc()->GetTransceivers().size(), 1u);
// Mid got cleared to make it reusable.
- EXPECT_EQ(callee->pc()->GetTransceivers()[0]->mid(), absl::nullopt);
+ EXPECT_EQ(callee->pc()->GetTransceivers()[0]->mid(), std::nullopt);
// Transceiver should be counted as addTrack-created after rollback.
EXPECT_TRUE(callee->SetRemoteDescription(caller->CreateOffer()));
EXPECT_EQ(callee->pc()->GetTransceivers().size(), 1u);
@@ -1973,7 +1973,7 @@
// Transceiver can't be removed as track was added to it.
EXPECT_EQ(callee->pc()->GetTransceivers().size(), 1u);
// Mid got cleared to make it reusable.
- EXPECT_EQ(callee->pc()->GetTransceivers()[0]->mid(), absl::nullopt);
+ EXPECT_EQ(callee->pc()->GetTransceivers()[0]->mid(), std::nullopt);
// Transceiver should be counted as addTrack-created after rollback.
EXPECT_TRUE(callee->SetRemoteDescription(caller->CreateOffer()));
EXPECT_EQ(callee->pc()->GetTransceivers().size(), 1u);
@@ -1988,9 +1988,9 @@
auto offer = callee->CreateOffer();
EXPECT_TRUE(callee->SetRemoteDescription(caller->CreateOffer()));
EXPECT_EQ(callee->pc()->GetTransceivers().size(), 1u);
- EXPECT_NE(callee->pc()->GetTransceivers()[0]->mid(), absl::nullopt);
+ EXPECT_NE(callee->pc()->GetTransceivers()[0]->mid(), std::nullopt);
EXPECT_TRUE(callee->SetRemoteDescription(caller->CreateRollback()));
- EXPECT_EQ(callee->pc()->GetTransceivers()[0]->mid(), absl::nullopt);
+ EXPECT_EQ(callee->pc()->GetTransceivers()[0]->mid(), std::nullopt);
EXPECT_TRUE(callee->SetLocalDescription(std::move(offer)));
}
@@ -2113,7 +2113,7 @@
auto initial_mid = callee->pc()->GetTransceivers()[0]->mid();
EXPECT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
EXPECT_EQ(callee->pc()->GetTransceivers().size(), 2u);
- EXPECT_EQ(callee->pc()->GetTransceivers()[0]->mid(), absl::nullopt);
+ EXPECT_EQ(callee->pc()->GetTransceivers()[0]->mid(), std::nullopt);
EXPECT_EQ(callee->pc()->GetTransceivers()[1]->mid(),
caller->pc()->GetTransceivers()[0]->mid());
EXPECT_TRUE(callee->CreateAnswerAndSetAsLocal()); // Go to stable.
@@ -2293,8 +2293,8 @@
EXPECT_TRUE(callee->observer()->legacy_renegotiation_needed());
EXPECT_TRUE(callee->observer()->has_negotiation_needed_event());
EXPECT_EQ(callee->pc()->GetTransceivers().size(), 2u);
- EXPECT_EQ(callee->pc()->GetTransceivers()[0]->mid(), absl::nullopt);
- EXPECT_EQ(callee->pc()->GetTransceivers()[1]->mid(), absl::nullopt);
+ EXPECT_EQ(callee->pc()->GetTransceivers()[0]->mid(), std::nullopt);
+ EXPECT_EQ(callee->pc()->GetTransceivers()[1]->mid(), std::nullopt);
}
TEST_F(PeerConnectionJsepTest, NoRollbackNeeded) {
diff --git a/pc/peer_connection_media_unittest.cc b/pc/peer_connection_media_unittest.cc
index 7009624..d9093f4 100644
--- a/pc/peer_connection_media_unittest.cc
+++ b/pc/peer_connection_media_unittest.cc
@@ -17,6 +17,7 @@
#include <iterator>
#include <map>
#include <memory>
+#include <optional>
#include <set>
#include <string>
#include <tuple>
@@ -25,7 +26,6 @@
#include <vector>
#include "absl/algorithm/container.h"
-#include "absl/types/optional.h"
#include "api/audio_options.h"
#include "api/jsep.h"
#include "api/media_types.h"
@@ -614,7 +614,7 @@
auto* offer_description =
cricket::GetFirstVideoContentDescription(offer->description());
for (const auto& codec : offer_description->codecs()) {
- EXPECT_EQ(codec.packetization, absl::nullopt);
+ EXPECT_EQ(codec.packetization, std::nullopt);
}
}
@@ -691,7 +691,7 @@
auto* answer_description =
cricket::GetFirstVideoContentDescription(answer->description());
for (const auto& codec : answer_description->codecs()) {
- EXPECT_EQ(codec.packetization, absl::nullopt);
+ EXPECT_EQ(codec.packetization, std::nullopt);
}
ASSERT_TRUE(caller->SetRemoteDescription(std::move(answer)));
diff --git a/pc/peer_connection_proxy.h b/pc/peer_connection_proxy.h
index 898f24d..b97a62e 100644
--- a/pc/peer_connection_proxy.h
+++ b/pc/peer_connection_proxy.h
@@ -157,7 +157,7 @@
PROXY_METHOD0(IceConnectionState, standardized_ice_connection_state)
PROXY_METHOD0(PeerConnectionState, peer_connection_state)
PROXY_METHOD0(IceGatheringState, ice_gathering_state)
-PROXY_METHOD0(absl::optional<bool>, can_trickle_ice_candidates)
+PROXY_METHOD0(std::optional<bool>, can_trickle_ice_candidates)
PROXY_METHOD1(void, AddAdaptationResource, rtc::scoped_refptr<Resource>)
PROXY_METHOD2(bool,
StartRtcEventLog,
diff --git a/pc/peer_connection_rampup_tests.cc b/pc/peer_connection_rampup_tests.cc
index e51e5aa..4bea516 100644
--- a/pc/peer_connection_rampup_tests.cc
+++ b/pc/peer_connection_rampup_tests.cc
@@ -9,11 +9,11 @@
*/
#include <memory>
+#include <optional>
#include <string>
#include <utility>
#include <vector>
-#include "absl/types/optional.h"
#include "api/audio/audio_device.h"
#include "api/audio/audio_mixer.h"
#include "api/audio/audio_processing.h"
diff --git a/pc/peer_connection_rtp_unittest.cc b/pc/peer_connection_rtp_unittest.cc
index bde8e18..ec9f215 100644
--- a/pc/peer_connection_rtp_unittest.cc
+++ b/pc/peer_connection_rtp_unittest.cc
@@ -12,11 +12,11 @@
#include <cstdint>
#include <memory>
+#include <optional>
#include <string>
#include <utility>
#include <vector>
-#include "absl/types/optional.h"
#include "api/audio/audio_device.h"
#include "api/audio/audio_mixer.h"
#include "api/audio/audio_processing.h"
@@ -959,10 +959,10 @@
auto caller = CreatePeerConnection();
auto transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
- EXPECT_EQ(absl::nullopt, transceiver->mid());
+ EXPECT_EQ(std::nullopt, transceiver->mid());
EXPECT_FALSE(transceiver->stopped());
EXPECT_EQ(RtpTransceiverDirection::kSendRecv, transceiver->direction());
- EXPECT_EQ(absl::nullopt, transceiver->current_direction());
+ EXPECT_EQ(std::nullopt, transceiver->current_direction());
}
// Test that adding a transceiver with the audio kind creates an audio sender
diff --git a/pc/peer_connection_signaling_unittest.cc b/pc/peer_connection_signaling_unittest.cc
index b6ad190..2b85a72 100644
--- a/pc/peer_connection_signaling_unittest.cc
+++ b/pc/peer_connection_signaling_unittest.cc
@@ -16,6 +16,7 @@
#include <functional>
#include <map>
#include <memory>
+#include <optional>
#include <set>
#include <string>
#include <tuple>
@@ -23,7 +24,6 @@
#include <utility>
#include <vector>
-#include "absl/types/optional.h"
#include "api/audio/audio_device.h"
#include "api/audio/audio_mixer.h"
#include "api/audio/audio_processing.h"
diff --git a/pc/peer_connection_wrapper.cc b/pc/peer_connection_wrapper.cc
index 557d0c8..b4842c1 100644
--- a/pc/peer_connection_wrapper.cc
+++ b/pc/peer_connection_wrapper.cc
@@ -12,10 +12,10 @@
#include <stdint.h>
+#include <optional>
#include <utility>
#include <vector>
-#include "absl/types/optional.h"
#include "api/function_view.h"
#include "api/set_remote_description_observer_interface.h"
#include "pc/sdp_utils.h"
@@ -314,7 +314,7 @@
rtc::scoped_refptr<DataChannelInterface>
PeerConnectionWrapper::CreateDataChannel(
const std::string& label,
- const absl::optional<DataChannelInit>& config) {
+ const std::optional<DataChannelInit>& config) {
const DataChannelInit* config_ptr = config.has_value() ? &(*config) : nullptr;
auto result = pc()->CreateDataChannelOrError(label, config_ptr);
if (!result.ok()) {
diff --git a/pc/peer_connection_wrapper.h b/pc/peer_connection_wrapper.h
index bf40bbc..de8dc47 100644
--- a/pc/peer_connection_wrapper.h
+++ b/pc/peer_connection_wrapper.h
@@ -12,10 +12,10 @@
#define PC_PEER_CONNECTION_WRAPPER_H_
#include <memory>
+#include <optional>
#include <string>
#include <vector>
-#include "absl/types/optional.h"
#include "api/data_channel_interface.h"
#include "api/function_view.h"
#include "api/jsep.h"
@@ -171,7 +171,7 @@
// initialization parameters.
rtc::scoped_refptr<DataChannelInterface> CreateDataChannel(
const std::string& label,
- const absl::optional<DataChannelInit>& config = absl::nullopt);
+ const std::optional<DataChannelInit>& config = std::nullopt);
// Returns the signaling state of the underlying PeerConnection.
PeerConnectionInterface::SignalingState signaling_state();
diff --git a/pc/remote_audio_source.cc b/pc/remote_audio_source.cc
index a516c57..87095c1 100644
--- a/pc/remote_audio_source.cc
+++ b/pc/remote_audio_source.cc
@@ -72,7 +72,7 @@
void RemoteAudioSource::Start(
cricket::VoiceMediaReceiveChannelInterface* media_channel,
- absl::optional<uint32_t> ssrc) {
+ std::optional<uint32_t> ssrc) {
RTC_DCHECK_RUN_ON(worker_thread_);
// Register for callbacks immediately before AddSink so that we always get
@@ -87,7 +87,7 @@
void RemoteAudioSource::Stop(
cricket::VoiceMediaReceiveChannelInterface* media_channel,
- absl::optional<uint32_t> ssrc) {
+ std::optional<uint32_t> ssrc) {
RTC_DCHECK_RUN_ON(worker_thread_);
RTC_DCHECK(media_channel);
ssrc ? media_channel->SetRawAudioSink(*ssrc, nullptr)
@@ -159,7 +159,7 @@
// absolute capture timestamp.
sink->OnData(audio.data, 16, audio.sample_rate, audio.channels,
audio.samples_per_channel,
- /*absolute_capture_timestamp_ms=*/absl::nullopt);
+ /*absolute_capture_timestamp_ms=*/std::nullopt);
}
}
diff --git a/pc/remote_audio_source.h b/pc/remote_audio_source.h
index 0fac606..ac4d222 100644
--- a/pc/remote_audio_source.h
+++ b/pc/remote_audio_source.h
@@ -14,9 +14,9 @@
#include <stdint.h>
#include <list>
+#include <optional>
#include <string>
-#include "absl/types/optional.h"
#include "api/call/audio_sink.h"
#include "api/media_stream_interface.h"
#include "api/notifier.h"
@@ -50,9 +50,9 @@
// Register and unregister remote audio source with the underlying media
// engine.
void Start(cricket::VoiceMediaReceiveChannelInterface* media_channel,
- absl::optional<uint32_t> ssrc);
+ std::optional<uint32_t> ssrc);
void Stop(cricket::VoiceMediaReceiveChannelInterface* media_channel,
- absl::optional<uint32_t> ssrc);
+ std::optional<uint32_t> ssrc);
void SetState(SourceState new_state);
// MediaSourceInterface implementation.
diff --git a/pc/rtc_stats_collector.cc b/pc/rtc_stats_collector.cc
index 3f6088a..cd9beb2 100644
--- a/pc/rtc_stats_collector.cc
+++ b/pc/rtc_stats_collector.cc
@@ -1053,7 +1053,7 @@
// Filter mode: RTCStatsCollector::RequestInfo::kReceiverSelector
if (receiver_selector) {
// Find the inbound-rtp of the receiver using ssrc lookup.
- absl::optional<uint32_t> ssrc;
+ std::optional<uint32_t> ssrc;
worker_thread_->BlockingCall([&] { ssrc = receiver_selector->ssrc(); });
if (ssrc.has_value()) {
for (const auto* inbound_rtp :
@@ -1250,7 +1250,7 @@
void RTCStatsCollector::ProducePartialResultsOnNetworkThread(
Timestamp timestamp,
- absl::optional<std::string> sctp_transport_name) {
+ std::optional<std::string> sctp_transport_name) {
TRACE_EVENT0("webrtc",
"RTCStatsCollector::ProducePartialResultsOnNetworkThread");
RTC_DCHECK_RUN_ON(network_thread_);
@@ -2136,8 +2136,8 @@
bool has_audio_receiver = false;
for (auto& stats : transceiver_stats_infos_) {
auto transceiver = stats.transceiver;
- absl::optional<cricket::VoiceMediaInfo> voice_media_info;
- absl::optional<cricket::VideoMediaInfo> video_media_info;
+ std::optional<cricket::VoiceMediaInfo> voice_media_info;
+ std::optional<cricket::VideoMediaInfo> video_media_info;
auto channel = transceiver->channel();
if (channel) {
cricket::MediaType media_type = transceiver->media_type();
@@ -2175,7 +2175,7 @@
call_stats_ = pc_->GetCallStats();
audio_device_stats_ =
- has_audio_receiver ? pc_->GetAudioDeviceStats() : absl::nullopt;
+ has_audio_receiver ? pc_->GetAudioDeviceStats() : std::nullopt;
});
for (auto& stats : transceiver_stats_infos_) {
diff --git a/pc/rtc_stats_collector.h b/pc/rtc_stats_collector.h
index 1dc84b0..1beef4c 100644
--- a/pc/rtc_stats_collector.h
+++ b/pc/rtc_stats_collector.h
@@ -16,10 +16,10 @@
#include <cstdint>
#include <map>
#include <memory>
+#include <optional>
#include <string>
#include <vector>
-#include "absl/types/optional.h"
#include "api/audio/audio_device.h"
#include "api/data_channel_interface.h"
#include "api/media_types.h"
@@ -170,10 +170,10 @@
struct RtpTransceiverStatsInfo {
rtc::scoped_refptr<RtpTransceiver> transceiver;
cricket::MediaType media_type;
- absl::optional<std::string> mid;
- absl::optional<std::string> transport_name;
+ std::optional<std::string> mid;
+ std::optional<std::string> transport_name;
TrackMediaInfoMap track_media_info_map;
- absl::optional<RtpTransceiverDirection> current_direction;
+ std::optional<RtpTransceiverDirection> current_direction;
};
void DeliverCachedReport(
@@ -240,7 +240,7 @@
void ProducePartialResultsOnSignalingThread(Timestamp timestamp);
void ProducePartialResultsOnNetworkThread(
Timestamp timestamp,
- absl::optional<std::string> sctp_transport_name);
+ std::optional<std::string> sctp_transport_name);
// Merges `network_report_` into `partial_report_` and completes the request.
// This is a NO-OP if `network_report_` is null.
void MergeNetworkReport_s();
@@ -293,7 +293,7 @@
Call::Stats call_stats_;
- absl::optional<AudioDeviceModule::Stats> audio_device_stats_;
+ std::optional<AudioDeviceModule::Stats> audio_device_stats_;
// A timestamp, in microseconds, that is based on a timer that is
// monotonically increasing. That is, even if the system clock is modified the
diff --git a/pc/rtc_stats_collector_unittest.cc b/pc/rtc_stats_collector_unittest.cc
index d43ccb4..b0fec81 100644
--- a/pc/rtc_stats_collector_unittest.cc
+++ b/pc/rtc_stats_collector_unittest.cc
@@ -16,6 +16,7 @@
#include <algorithm>
#include <initializer_list>
#include <memory>
+#include <optional>
#include <ostream>
#include <string>
#include <type_traits>
@@ -23,7 +24,6 @@
#include <vector>
#include "absl/strings/str_replace.h"
-#include "absl/types/optional.h"
#include "api/audio/audio_device.h"
#include "api/audio/audio_processing_statistics.h"
#include "api/candidate.h"
@@ -289,7 +289,7 @@
// VideoTrackSourceInterface
bool is_screencast() const override { return false; }
- absl::optional<bool> needs_denoising() const override { return false; }
+ std::optional<bool> needs_denoising() const override { return false; }
bool GetStats(VideoTrackSourceInterface::Stats* stats) override {
stats->input_width = input_width_;
stats->input_height = input_height_;
@@ -1932,7 +1932,7 @@
connection_info.sent_total_bytes = 42;
connection_info.recv_total_bytes = 1234;
connection_info.total_round_trip_time_ms = 0;
- connection_info.current_round_trip_time_ms = absl::nullopt;
+ connection_info.current_round_trip_time_ms = std::nullopt;
connection_info.recv_ping_requests = 2020;
connection_info.sent_ping_requests_total = 2222;
connection_info.sent_ping_requests_before_first_response = 2000;
@@ -2200,7 +2200,7 @@
voice_media_info.receivers[0].relative_packet_arrival_delay_seconds = 16;
voice_media_info.receivers[0].interruption_count = 7788;
voice_media_info.receivers[0].total_interruption_duration_ms = 778899;
- voice_media_info.receivers[0].last_packet_received = absl::nullopt;
+ voice_media_info.receivers[0].last_packet_received = std::nullopt;
voice_media_info.receivers[0].total_processing_delay_seconds = 0.123;
RtpCodecParameters codec_parameters;
@@ -2342,7 +2342,7 @@
video_media_info.receivers[0].frames_decoded = 9;
video_media_info.receivers[0].key_frames_decoded = 3;
video_media_info.receivers[0].frames_dropped = 13;
- video_media_info.receivers[0].qp_sum = absl::nullopt;
+ video_media_info.receivers[0].qp_sum = std::nullopt;
video_media_info.receivers[0].total_decode_time = TimeDelta::Seconds(9);
video_media_info.receivers[0].total_processing_delay = TimeDelta::Millis(600);
video_media_info.receivers[0].total_assembly_time = TimeDelta::Millis(500);
@@ -2358,11 +2358,11 @@
video_media_info.receivers[0].jitter_buffer_target_delay_seconds = 1.1;
video_media_info.receivers[0].jitter_buffer_minimum_delay_seconds = 0.999;
video_media_info.receivers[0].jitter_buffer_emitted_count = 13;
- video_media_info.receivers[0].last_packet_received = absl::nullopt;
+ video_media_info.receivers[0].last_packet_received = std::nullopt;
video_media_info.receivers[0].content_type = VideoContentType::UNSPECIFIED;
video_media_info.receivers[0].estimated_playout_ntp_timestamp_ms =
- absl::nullopt;
- video_media_info.receivers[0].decoder_implementation_name = absl::nullopt;
+ std::nullopt;
+ video_media_info.receivers[0].decoder_implementation_name = std::nullopt;
video_media_info.receivers[0].min_playout_delay_ms = 50;
video_media_info.receivers[0].power_efficient_decoder = false;
video_media_info.receivers[0].retransmitted_packets_received = 17;
@@ -2629,9 +2629,9 @@
.quality_limitation_durations_ms[QualityLimitationReason::kBandwidth] =
300;
video_media_info.senders[0].quality_limitation_resolution_changes = 56u;
- video_media_info.senders[0].qp_sum = absl::nullopt;
+ video_media_info.senders[0].qp_sum = std::nullopt;
video_media_info.senders[0].content_type = VideoContentType::UNSPECIFIED;
- video_media_info.senders[0].encoder_implementation_name = absl::nullopt;
+ video_media_info.senders[0].encoder_implementation_name = std::nullopt;
video_media_info.senders[0].power_efficient_encoder = false;
video_media_info.senders[0].send_frame_width = 200;
video_media_info.senders[0].send_frame_height = 100;
@@ -3242,7 +3242,7 @@
void AddSenderInfoAndMediaChannel(
std::string transport_name,
const std::vector<ReportBlockData>& report_block_datas,
- absl::optional<RtpCodecParameters> codec) {
+ std::optional<RtpCodecParameters> codec) {
switch (media_type_) {
case cricket::MEDIA_TYPE_AUDIO: {
cricket::VoiceMediaInfo voice_media_info;
@@ -3319,7 +3319,7 @@
report_block_datas.push_back(report_block_data);
}
AddSenderInfoAndMediaChannel("TransportName", report_block_datas,
- absl::nullopt);
+ std::nullopt);
rtc::scoped_refptr<const RTCStatsReport> report = stats_->GetStatsReport();
for (auto ssrc : ssrcs) {
@@ -3370,7 +3370,7 @@
report_block_data.SetReportBlock(0, report_block, kReportBlockTimestampUtc);
AddSenderInfoAndMediaChannel("TransportName", {report_block_data},
- absl::nullopt);
+ std::nullopt);
rtc::scoped_refptr<const RTCStatsReport> report = stats_->GetStatsReport();
@@ -3397,7 +3397,7 @@
report_block_data.SetReportBlock(0, report_block, kReportBlockTimestampUtc);
AddSenderInfoAndMediaChannel("TransportName", {report_block_data},
- absl::nullopt);
+ std::nullopt);
// Advance time, it should be OK to have fresher reports than report blocks.
fake_clock_.AdvanceTime(TimeDelta::Micros(1234));
@@ -3474,7 +3474,7 @@
pc_->SetTransportStats("TransportName", {rtp_transport_channel_stats,
rtcp_transport_channel_stats});
AddSenderInfoAndMediaChannel("TransportName", {report_block_data},
- absl::nullopt);
+ std::nullopt);
rtc::scoped_refptr<const RTCStatsReport> report = stats_->GetStatsReport();
@@ -3732,7 +3732,7 @@
RTCTestStats(const std::string& id, Timestamp timestamp)
: RTCStats(id, timestamp) {}
- absl::optional<int32_t> dummy_stat;
+ std::optional<int32_t> dummy_stat;
};
WEBRTC_RTCSTATS_IMPL(RTCTestStats,
diff --git a/pc/rtc_stats_integrationtest.cc b/pc/rtc_stats_integrationtest.cc
index f8b0899..779f06b 100644
--- a/pc/rtc_stats_integrationtest.cc
+++ b/pc/rtc_stats_integrationtest.cc
@@ -12,13 +12,13 @@
#include <string.h>
#include <memory>
+#include <optional>
#include <set>
#include <string>
#include <vector>
#include "absl/algorithm/container.h"
#include "absl/strings/match.h"
-#include "absl/types/optional.h"
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
#include "api/audio_options.h"
@@ -160,14 +160,14 @@
}
template <typename T>
- void MarkAttributeTested(const absl::optional<T>& field,
+ void MarkAttributeTested(const std::optional<T>& field,
bool test_successful) {
untested_attribute_names_.erase(stats_->GetAttribute(field).name());
all_tests_successful_ &= test_successful;
}
template <typename T>
- void TestAttributeIsDefined(const absl::optional<T>& field) {
+ void TestAttributeIsDefined(const std::optional<T>& field) {
EXPECT_TRUE(field.has_value())
<< stats_->type() << "." << stats_->GetAttribute(field).name() << "["
<< stats_->id() << "] was undefined.";
@@ -175,7 +175,7 @@
}
template <typename T>
- void TestAttributeIsUndefined(const absl::optional<T>& field) {
+ void TestAttributeIsUndefined(const std::optional<T>& field) {
Attribute attribute = stats_->GetAttribute(field);
EXPECT_FALSE(field.has_value())
<< stats_->type() << "." << attribute.name() << "[" << stats_->id()
@@ -184,7 +184,7 @@
}
template <typename T>
- void TestAttributeIsPositive(const absl::optional<T>& field) {
+ void TestAttributeIsPositive(const std::optional<T>& field) {
Attribute attribute = stats_->GetAttribute(field);
EXPECT_TRUE(field.has_value()) << stats_->type() << "." << attribute.name()
<< "[" << stats_->id() << "] was undefined.";
@@ -200,7 +200,7 @@
}
template <typename T>
- void TestAttributeIsNonNegative(const absl::optional<T>& field) {
+ void TestAttributeIsNonNegative(const std::optional<T>& field) {
Attribute attribute = stats_->GetAttribute(field);
EXPECT_TRUE(field.has_value()) << stats_->type() << "." << attribute.name()
<< "[" << stats_->id() << "] was undefined.";
@@ -216,13 +216,13 @@
}
template <typename T>
- void TestAttributeIsIDReference(const absl::optional<T>& field,
+ void TestAttributeIsIDReference(const std::optional<T>& field,
const char* expected_type) {
TestAttributeIsIDReference(field, expected_type, false);
}
template <typename T>
- void TestAttributeIsOptionalIDReference(const absl::optional<T>& field,
+ void TestAttributeIsOptionalIDReference(const std::optional<T>& field,
const char* expected_type) {
TestAttributeIsIDReference(field, expected_type, true);
}
@@ -239,7 +239,7 @@
private:
template <typename T>
- void TestAttributeIsIDReference(const absl::optional<T>& field,
+ void TestAttributeIsIDReference(const std::optional<T>& field,
const char* expected_type,
bool optional) {
if (optional && !field.has_value()) {
diff --git a/pc/rtc_stats_traversal.cc b/pc/rtc_stats_traversal.cc
index e9d11b5..aeaf244 100644
--- a/pc/rtc_stats_traversal.cc
+++ b/pc/rtc_stats_traversal.cc
@@ -11,11 +11,11 @@
#include "pc/rtc_stats_traversal.h"
#include <memory>
+#include <optional>
#include <string>
#include <utility>
#include <vector>
-#include "absl/types/optional.h"
#include "api/stats/rtcstats_objects.h"
#include "rtc_base/checks.h"
@@ -43,7 +43,7 @@
}
}
-void AddIdIfDefined(const absl::optional<std::string>& id,
+void AddIdIfDefined(const std::optional<std::string>& id,
std::vector<const std::string*>* neighbor_ids) {
if (id.has_value())
neighbor_ids->push_back(&(*id));
diff --git a/pc/rtp_parameters_conversion.cc b/pc/rtp_parameters_conversion.cc
index 2463cef..1c0c3ee 100644
--- a/pc/rtp_parameters_conversion.cc
+++ b/pc/rtp_parameters_conversion.cc
@@ -196,7 +196,7 @@
return std::move(cricket_streams);
}
-absl::optional<RtcpFeedback> ToRtcpFeedback(
+std::optional<RtcpFeedback> ToRtcpFeedback(
const cricket::FeedbackParam& cricket_feedback) {
if (cricket_feedback.id() == cricket::kRtcpFbParamCcm) {
if (cricket_feedback.param() == cricket::kRtcpFbCcmParamFir) {
@@ -204,7 +204,7 @@
} else {
RTC_LOG(LS_WARNING) << "Unsupported parameter for CCM RTCP feedback: "
<< cricket_feedback.param();
- return absl::nullopt;
+ return std::nullopt;
}
} else if (cricket_feedback.id() == cricket::kRtcpFbParamLntf) {
if (cricket_feedback.param().empty()) {
@@ -212,7 +212,7 @@
} else {
RTC_LOG(LS_WARNING) << "Unsupported parameter for LNTF RTCP feedback: "
<< cricket_feedback.param();
- return absl::nullopt;
+ return std::nullopt;
}
} else if (cricket_feedback.id() == cricket::kRtcpFbParamNack) {
if (cricket_feedback.param().empty()) {
@@ -223,13 +223,13 @@
} else {
RTC_LOG(LS_WARNING) << "Unsupported parameter for NACK RTCP feedback: "
<< cricket_feedback.param();
- return absl::nullopt;
+ return std::nullopt;
}
} else if (cricket_feedback.id() == cricket::kRtcpFbParamRemb) {
if (!cricket_feedback.param().empty()) {
RTC_LOG(LS_WARNING) << "Unsupported parameter for REMB RTCP feedback: "
<< cricket_feedback.param();
- return absl::nullopt;
+ return std::nullopt;
} else {
return RtcpFeedback(RtcpFeedbackType::REMB);
}
@@ -238,14 +238,14 @@
RTC_LOG(LS_WARNING)
<< "Unsupported parameter for transport-cc RTCP feedback: "
<< cricket_feedback.param();
- return absl::nullopt;
+ return std::nullopt;
} else {
return RtcpFeedback(RtcpFeedbackType::TRANSPORT_CC);
}
}
RTC_LOG(LS_WARNING) << "Unsupported RTCP feedback type: "
<< cricket_feedback.id();
- return absl::nullopt;
+ return std::nullopt;
}
std::vector<RtpEncodingParameters> ToRtpEncodings(
@@ -269,7 +269,7 @@
codec.preferred_payload_type.emplace(cricket_codec.id);
for (const cricket::FeedbackParam& cricket_feedback :
cricket_codec.feedback_params.params()) {
- absl::optional<RtcpFeedback> feedback = ToRtcpFeedback(cricket_feedback);
+ std::optional<RtcpFeedback> feedback = ToRtcpFeedback(cricket_feedback);
if (feedback) {
codec.rtcp_feedback.push_back(feedback.value());
}
@@ -297,7 +297,7 @@
codec_param.payload_type = cricket_codec.id;
for (const cricket::FeedbackParam& cricket_feedback :
cricket_codec.feedback_params.params()) {
- absl::optional<RtcpFeedback> feedback = ToRtcpFeedback(cricket_feedback);
+ std::optional<RtcpFeedback> feedback = ToRtcpFeedback(cricket_feedback);
if (feedback) {
codec_param.rtcp_feedback.push_back(feedback.value());
}
diff --git a/pc/rtp_parameters_conversion.h b/pc/rtp_parameters_conversion.h
index 2cc39dd..eab77f4 100644
--- a/pc/rtp_parameters_conversion.h
+++ b/pc/rtp_parameters_conversion.h
@@ -11,9 +11,9 @@
#ifndef PC_RTP_PARAMETERS_CONVERSION_H_
#define PC_RTP_PARAMETERS_CONVERSION_H_
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/rtc_error.h"
#include "api/rtp_parameters.h"
#include "media/base/codec.h"
@@ -74,7 +74,7 @@
// Returns empty value if `cricket_feedback` is a feedback type not
// supported/recognized.
-absl::optional<RtcpFeedback> ToRtcpFeedback(
+std::optional<RtcpFeedback> ToRtcpFeedback(
const cricket::FeedbackParam& cricket_feedback);
std::vector<RtpEncodingParameters> ToRtpEncodings(
diff --git a/pc/rtp_parameters_conversion_unittest.cc b/pc/rtp_parameters_conversion_unittest.cc
index 0c68b0a..5bf5419 100644
--- a/pc/rtp_parameters_conversion_unittest.cc
+++ b/pc/rtp_parameters_conversion_unittest.cc
@@ -344,7 +344,7 @@
}
TEST(RtpParametersConversionTest, ToRtcpFeedback) {
- absl::optional<RtcpFeedback> result = ToRtcpFeedback({"ccm", "fir"});
+ std::optional<RtcpFeedback> result = ToRtcpFeedback({"ccm", "fir"});
EXPECT_EQ(RtcpFeedback(RtcpFeedbackType::CCM, RtcpFeedbackMessageType::FIR),
*result);
@@ -369,7 +369,7 @@
TEST(RtpParametersConversionTest, ToRtcpFeedbackErrors) {
// CCM with missing or invalid message type.
- absl::optional<RtcpFeedback> result = ToRtcpFeedback({"ccm", "pli"});
+ std::optional<RtcpFeedback> result = ToRtcpFeedback({"ccm", "pli"});
EXPECT_FALSE(result);
result = ToRtcpFeedback(cricket::FeedbackParam("ccm"));
diff --git a/pc/rtp_receiver.h b/pc/rtp_receiver.h
index 53451a3..14e102f 100644
--- a/pc/rtp_receiver.h
+++ b/pc/rtp_receiver.h
@@ -17,10 +17,10 @@
#include <stdint.h>
+#include <optional>
#include <string>
#include <vector>
-#include "absl/types/optional.h"
#include "api/dtls_transport_interface.h"
#include "api/media_stream_interface.h"
#include "api/rtp_receiver_interface.h"
@@ -59,7 +59,7 @@
rtc::scoped_refptr<DtlsTransportInterface> dtls_transport) = 0;
// This SSRC is used as an identifier for the receiver between the API layer
// and the WebRtcVideoEngine, WebRtcVoiceEngine layer.
- virtual absl::optional<uint32_t> ssrc() const = 0;
+ virtual std::optional<uint32_t> ssrc() const = 0;
// Call this to notify the RtpReceiver when the first packet has been received
// on the corresponding channel.
diff --git a/pc/rtp_receiver_proxy.h b/pc/rtp_receiver_proxy.h
index 0cdf6ff..feb569b 100644
--- a/pc/rtp_receiver_proxy.h
+++ b/pc/rtp_receiver_proxy.h
@@ -11,10 +11,10 @@
#ifndef PC_RTP_RECEIVER_PROXY_H_
#define PC_RTP_RECEIVER_PROXY_H_
+#include <optional>
#include <string>
#include <vector>
-#include "absl/types/optional.h"
#include "api/crypto/frame_decryptor_interface.h"
#include "api/dtls_transport_interface.h"
#include "api/frame_transformer_interface.h"
@@ -44,7 +44,7 @@
PROXY_METHOD1(void, SetObserver, RtpReceiverObserverInterface*)
PROXY_SECONDARY_METHOD1(void,
SetJitterBufferMinimumDelay,
- absl::optional<double>)
+ std::optional<double>)
PROXY_SECONDARY_CONSTMETHOD0(std::vector<RtpSource>, GetSources)
// TODO(bugs.webrtc.org/12772): Remove.
PROXY_SECONDARY_METHOD1(void,
diff --git a/pc/rtp_sender.cc b/pc/rtp_sender.cc
index 6e28734..cc74716 100644
--- a/pc/rtp_sender.cc
+++ b/pc/rtp_sender.cc
@@ -248,7 +248,7 @@
}
if (!media_channel_ || !ssrc_) {
auto result = cricket::CheckRtpParametersInvalidModificationAndValues(
- init_parameters_, parameters, send_codecs_, absl::nullopt);
+ init_parameters_, parameters, send_codecs_, std::nullopt);
if (result.ok()) {
init_parameters_ = parameters;
}
@@ -299,7 +299,7 @@
}
if (!media_channel_ || !ssrc_) {
auto result = cricket::CheckRtpParametersInvalidModificationAndValues(
- init_parameters_, parameters, send_codecs_, absl::nullopt);
+ init_parameters_, parameters, send_codecs_, std::nullopt);
if (result.ok()) {
init_parameters_ = parameters;
}
@@ -339,11 +339,11 @@
}
RTCError RtpSenderBase::CheckCodecParameters(const RtpParameters& parameters) {
- absl::optional<cricket::Codec> send_codec = media_channel_->GetSendCodec();
+ std::optional<cricket::Codec> send_codec = media_channel_->GetSendCodec();
// Match the currently used codec against the codec preferences to gather
// the SVC capabilities.
- absl::optional<cricket::Codec> send_codec_with_svc_info;
+ std::optional<cricket::Codec> send_codec_with_svc_info;
if (send_codec && send_codec->type == cricket::Codec::Type::kVideo) {
auto codec_match = absl::c_find_if(
send_codecs_, [&](auto& codec) { return send_codec->Matches(codec); });
@@ -498,7 +498,7 @@
init_parameters_.degradation_preference;
media_channel_->SetRtpSendParameters(ssrc_, current_parameters, nullptr);
init_parameters_.encodings.clear();
- init_parameters_.degradation_preference = absl::nullopt;
+ init_parameters_.degradation_preference = std::nullopt;
});
}
// Attempt to attach the frame decryptor to the current media channel.
@@ -606,7 +606,7 @@
int sample_rate,
size_t number_of_channels,
size_t number_of_frames,
- absl::optional<int64_t> absolute_capture_timestamp_ms) {
+ std::optional<int64_t> absolute_capture_timestamp_ms) {
TRACE_EVENT2("webrtc", "LocalAudioSinkAdapter::OnData", "sample_rate",
sample_rate, "number_of_frames", number_of_frames);
MutexLock lock(&lock_);
diff --git a/pc/rtp_sender.h b/pc/rtp_sender.h
index bd2cf9d..ec8f923 100644
--- a/pc/rtp_sender.h
+++ b/pc/rtp_sender.h
@@ -19,10 +19,10 @@
#include <stdint.h>
#include <memory>
+#include <optional>
#include <string>
#include <vector>
-#include "absl/types/optional.h"
#include "api/crypto/frame_encryptor_interface.h"
#include "api/dtls_transport_interface.h"
#include "api/dtmf_sender_interface.h"
@@ -276,7 +276,7 @@
// As such, it is used for internal verification and is not observable by the
// the client. It is marked as mutable to enable `GetParameters` to be a
// const method.
- mutable absl::optional<std::string> last_transaction_id_;
+ mutable std::optional<std::string> last_transaction_id_;
std::vector<std::string> disabled_rids_;
SetStreamsObserver* set_streams_observer_ = nullptr;
@@ -303,7 +303,7 @@
int sample_rate,
size_t number_of_channels,
size_t number_of_frames,
- absl::optional<int64_t> absolute_capture_timestamp_ms) override;
+ std::optional<int64_t> absolute_capture_timestamp_ms) override;
// AudioSinkInterface implementation.
void OnData(const void* audio_data,
@@ -313,7 +313,7 @@
size_t number_of_frames) override {
OnData(audio_data, bits_per_sample, sample_rate, number_of_channels,
number_of_frames,
- /*absolute_capture_timestamp_ms=*/absl::nullopt);
+ /*absolute_capture_timestamp_ms=*/std::nullopt);
}
// AudioSinkInterface implementation.
diff --git a/pc/rtp_sender_receiver_unittest.cc b/pc/rtp_sender_receiver_unittest.cc
index b73edf5..99a113a 100644
--- a/pc/rtp_sender_receiver_unittest.cc
+++ b/pc/rtp_sender_receiver_unittest.cc
@@ -13,12 +13,12 @@
#include <cstdint>
#include <iterator>
#include <memory>
+#include <optional>
#include <string>
#include <utility>
#include <vector>
#include "absl/algorithm/container.h"
-#include "absl/types/optional.h"
#include "api/audio_codecs/audio_codec_pair_id.h"
#include "api/audio_options.h"
#include "api/crypto/crypto_options.h"
@@ -491,7 +491,7 @@
RtpReceiverInterface* receiver,
uint32_t ssrc) {
receiver->SetJitterBufferMinimumDelay(/*delay_seconds=*/0.5);
- absl::optional<int> delay_ms =
+ std::optional<int> delay_ms =
media_channel->GetBaseMinimumPlayoutDelayMs(ssrc); // In milliseconds.
EXPECT_DOUBLE_EQ(0.5, delay_ms.value_or(0) / 1000.0);
}
@@ -888,7 +888,7 @@
RtpParameters params = audio_rtp_sender_->GetParameters();
EXPECT_EQ(1u, params.encodings.size());
- absl::optional<RTCError> result;
+ std::optional<RTCError> result;
audio_rtp_sender_->SetParametersAsync(
params, [&result](RTCError error) { result = error; });
run_loop_.Flush();
@@ -918,7 +918,7 @@
audio_rtp_sender_ =
AudioRtpSender::Create(worker_thread_, /*id=*/"", nullptr, nullptr);
- absl::optional<RTCError> result;
+ std::optional<RTCError> result;
RtpParameters params = audio_rtp_sender_->GetParameters();
ASSERT_EQ(1u, params.encodings.size());
params.encodings[0].max_bitrate_bps = 90000;
@@ -1016,7 +1016,7 @@
RtpParameters params = audio_rtp_sender_->GetParameters();
EXPECT_EQ(1u, params.encodings.size());
- absl::optional<RTCError> result;
+ std::optional<RTCError> result;
audio_rtp_sender_->SetParametersAsync(
params, [&result](RTCError error) { result = error; });
run_loop_.Flush();
@@ -1139,7 +1139,7 @@
RtpParameters params = video_rtp_sender_->GetParameters();
EXPECT_EQ(1u, params.encodings.size());
- absl::optional<RTCError> result;
+ std::optional<RTCError> result;
video_rtp_sender_->SetParametersAsync(
params, [&result](RTCError error) { result = error; });
run_loop_.Flush();
@@ -1169,7 +1169,7 @@
video_rtp_sender_ =
VideoRtpSender::Create(worker_thread_, /*id=*/"", nullptr);
- absl::optional<RTCError> result;
+ std::optional<RTCError> result;
RtpParameters params = video_rtp_sender_->GetParameters();
ASSERT_EQ(1u, params.encodings.size());
params.encodings[0].max_bitrate_bps = 90000;
@@ -1349,7 +1349,7 @@
RtpParameters params = video_rtp_sender_->GetParameters();
EXPECT_EQ(1u, params.encodings.size());
- absl::optional<RTCError> result;
+ std::optional<RTCError> result;
video_rtp_sender_->SetParametersAsync(
params, [&result](RTCError error) { result = error; });
run_loop_.Flush();
@@ -1723,7 +1723,7 @@
video_track_->set_enabled(true);
// Sender is not ready to send (no SSRC) so no option should have been set.
- EXPECT_EQ(absl::nullopt, video_media_send_channel()->options().is_screencast);
+ EXPECT_EQ(std::nullopt, video_media_send_channel()->options().is_screencast);
// Verify that the content hint is accounted for when video_rtp_sender_ does
// get enabled.
diff --git a/pc/rtp_transceiver.cc b/pc/rtp_transceiver.cc
index f0a09ce..c275504 100644
--- a/pc/rtp_transceiver.cc
+++ b/pc/rtp_transceiver.cc
@@ -17,6 +17,7 @@
#include <functional>
#include <iterator>
#include <memory>
+#include <optional>
#include <set>
#include <string>
#include <utility>
@@ -24,7 +25,6 @@
#include "absl/algorithm/container.h"
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/audio_codecs/audio_codec_pair_id.h"
#include "api/audio_options.h"
@@ -513,7 +513,7 @@
return media_type_;
}
-absl::optional<std::string> RtpTransceiver::mid() const {
+std::optional<std::string> RtpTransceiver::mid() const {
return mid_;
}
@@ -550,7 +550,7 @@
}
void RtpTransceiver::set_fired_direction(
- absl::optional<RtpTransceiverDirection> direction) {
+ std::optional<RtpTransceiverDirection> direction) {
fired_direction_ = direction;
}
@@ -591,7 +591,7 @@
return RTCError::OK();
}
-absl::optional<RtpTransceiverDirection> RtpTransceiver::current_direction()
+std::optional<RtpTransceiverDirection> RtpTransceiver::current_direction()
const {
if (unified_plan_ && stopped())
return RtpTransceiverDirection::kStopped;
@@ -599,8 +599,7 @@
return current_direction_;
}
-absl::optional<RtpTransceiverDirection> RtpTransceiver::fired_direction()
- const {
+std::optional<RtpTransceiverDirection> RtpTransceiver::fired_direction() const {
return fired_direction_;
}
@@ -684,7 +683,7 @@
// 3. Set transceiver.[[Receptive]] to false.
// 4. Set transceiver.[[CurrentDirection]] to null.
- current_direction_ = absl::nullopt;
+ current_direction_ = std::nullopt;
}
RTCError RtpTransceiver::SetCodecPreferences(
diff --git a/pc/rtp_transceiver.h b/pc/rtp_transceiver.h
index 88febb9..610b842 100644
--- a/pc/rtp_transceiver.h
+++ b/pc/rtp_transceiver.h
@@ -15,11 +15,11 @@
#include <functional>
#include <memory>
+#include <optional>
#include <string>
#include <vector>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/audio_options.h"
#include "api/crypto/crypto_options.h"
@@ -203,15 +203,15 @@
// when setting a local offer we need a way to remember which transceiver was
// used to create which media section in the offer. Storing the mline index
// in CreateOffer is specified in JSEP to allow us to do that.
- absl::optional<size_t> mline_index() const { return mline_index_; }
- void set_mline_index(absl::optional<size_t> mline_index) {
+ std::optional<size_t> mline_index() const { return mline_index_; }
+ void set_mline_index(std::optional<size_t> mline_index) {
mline_index_ = mline_index;
}
// Sets the MID for this transceiver. If the MID is not null, then the
// transceiver is considered "associated" with the media section that has the
// same MID.
- void set_mid(const absl::optional<std::string>& mid) { mid_ = mid; }
+ void set_mid(const std::optional<std::string>& mid) { mid_ = mid; }
// Sets the intended direction for this transceiver. Intended to be used
// internally over SetDirection since this does not trigger a negotiation
@@ -228,7 +228,7 @@
// Sets the fired direction for this transceiver. The fired direction is null
// until SetRemoteDescription is called or an answer is set (either local or
// remote) after which the only valid reason to go back to null is rollback.
- void set_fired_direction(absl::optional<RtpTransceiverDirection> direction);
+ void set_fired_direction(std::optional<RtpTransceiverDirection> direction);
// According to JSEP rules for SetRemoteDescription, RtpTransceivers can be
// reused only if they were added by AddTrack.
@@ -261,7 +261,7 @@
// RtpTransceiverInterface implementation.
cricket::MediaType media_type() const override;
- absl::optional<std::string> mid() const override;
+ std::optional<std::string> mid() const override;
rtc::scoped_refptr<RtpSenderInterface> sender() const override;
rtc::scoped_refptr<RtpReceiverInterface> receiver() const override;
bool stopped() const override;
@@ -269,8 +269,8 @@
RtpTransceiverDirection direction() const override;
RTCError SetDirectionWithError(
RtpTransceiverDirection new_direction) override;
- absl::optional<RtpTransceiverDirection> current_direction() const override;
- absl::optional<RtpTransceiverDirection> fired_direction() const override;
+ std::optional<RtpTransceiverDirection> current_direction() const override;
+ std::optional<RtpTransceiverDirection> fired_direction() const override;
RTCError StopStandard() override;
void StopInternal() override;
RTCError SetCodecPreferences(
@@ -323,10 +323,10 @@
bool stopping_ RTC_GUARDED_BY(thread_) = false;
bool is_pc_closed_ = false;
RtpTransceiverDirection direction_ = RtpTransceiverDirection::kInactive;
- absl::optional<RtpTransceiverDirection> current_direction_;
- absl::optional<RtpTransceiverDirection> fired_direction_;
- absl::optional<std::string> mid_;
- absl::optional<size_t> mline_index_;
+ std::optional<RtpTransceiverDirection> current_direction_;
+ std::optional<RtpTransceiverDirection> fired_direction_;
+ std::optional<std::string> mid_;
+ std::optional<size_t> mline_index_;
bool created_by_addtrack_ = false;
bool reused_for_addtrack_ = false;
bool has_ever_been_used_to_send_ = false;
@@ -352,15 +352,15 @@
PROXY_PRIMARY_THREAD_DESTRUCTOR()
BYPASS_PROXY_CONSTMETHOD0(cricket::MediaType, media_type)
-PROXY_CONSTMETHOD0(absl::optional<std::string>, mid)
+PROXY_CONSTMETHOD0(std::optional<std::string>, mid)
PROXY_CONSTMETHOD0(rtc::scoped_refptr<RtpSenderInterface>, sender)
PROXY_CONSTMETHOD0(rtc::scoped_refptr<RtpReceiverInterface>, receiver)
PROXY_CONSTMETHOD0(bool, stopped)
PROXY_CONSTMETHOD0(bool, stopping)
PROXY_CONSTMETHOD0(RtpTransceiverDirection, direction)
PROXY_METHOD1(RTCError, SetDirectionWithError, RtpTransceiverDirection)
-PROXY_CONSTMETHOD0(absl::optional<RtpTransceiverDirection>, current_direction)
-PROXY_CONSTMETHOD0(absl::optional<RtpTransceiverDirection>, fired_direction)
+PROXY_CONSTMETHOD0(std::optional<RtpTransceiverDirection>, current_direction)
+PROXY_CONSTMETHOD0(std::optional<RtpTransceiverDirection>, fired_direction)
PROXY_METHOD0(RTCError, StopStandard)
PROXY_METHOD0(void, StopInternal)
PROXY_METHOD1(RTCError, SetCodecPreferences, rtc::ArrayView<RtpCodecCapability>)
diff --git a/pc/rtp_transceiver_unittest.cc b/pc/rtp_transceiver_unittest.cc
index b6dc7b2..9979154 100644
--- a/pc/rtp_transceiver_unittest.cc
+++ b/pc/rtp_transceiver_unittest.cc
@@ -13,10 +13,10 @@
#include "pc/rtp_transceiver.h"
#include <memory>
+#include <optional>
#include <utility>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/environment/environment_factory.h"
#include "api/peer_connection_interface.h"
#include "api/rtp_parameters.h"
diff --git a/pc/rtp_transmission_manager.cc b/pc/rtp_transmission_manager.cc
index ab9c58a..d42f8fc 100644
--- a/pc/rtp_transmission_manager.cc
+++ b/pc/rtp_transmission_manager.cc
@@ -10,10 +10,10 @@
#include "pc/rtp_transmission_manager.h"
+#include <optional>
#include <type_traits>
#include <utility>
-#include "absl/types/optional.h"
#include "api/peer_connection_interface.h"
#include "api/rtp_transceiver_direction.h"
#include "pc/audio_rtp_receiver.h"
@@ -520,8 +520,8 @@
video_receiver->SetupMediaChannel(
remote_sender_info.sender_id == kDefaultVideoSenderId
- ? absl::nullopt
- : absl::optional<uint32_t>(remote_sender_info.first_ssrc),
+ ? std::nullopt
+ : std::optional<uint32_t>(remote_sender_info.first_ssrc),
video_media_receive_channel());
auto receiver = RtpReceiverProxyWithInternal<RtpReceiverInternal>::Create(
diff --git a/pc/rtp_transport.cc b/pc/rtp_transport.cc
index b2fc9b21..b2124003 100644
--- a/pc/rtp_transport.cc
+++ b/pc/rtp_transport.cc
@@ -58,7 +58,7 @@
rtp_packet_transport_->SignalWritableState.disconnect(this);
rtp_packet_transport_->SignalSentPacket.disconnect(this);
// Reset the network route of the old transport.
- SendNetworkRouteChanged(absl::optional<rtc::NetworkRoute>());
+ SendNetworkRouteChanged(std::optional<rtc::NetworkRoute>());
}
if (new_packet_transport) {
new_packet_transport->SignalReadyToSend.connect(
@@ -95,7 +95,7 @@
rtcp_packet_transport_->SignalWritableState.disconnect(this);
rtcp_packet_transport_->SignalSentPacket.disconnect(this);
// Reset the network route of the old transport.
- SendNetworkRouteChanged(absl::optional<rtc::NetworkRoute>());
+ SendNetworkRouteChanged(std::optional<rtc::NetworkRoute>());
}
if (new_packet_transport) {
new_packet_transport->SignalReadyToSend.connect(
@@ -222,7 +222,7 @@
}
void RtpTransport::OnNetworkRouteChanged(
- absl::optional<rtc::NetworkRoute> network_route) {
+ std::optional<rtc::NetworkRoute> network_route) {
SendNetworkRouteChanged(network_route);
}
diff --git a/pc/rtp_transport.h b/pc/rtp_transport.h
index ebb7aa6..284bd30 100644
--- a/pc/rtp_transport.h
+++ b/pc/rtp_transport.h
@@ -14,9 +14,9 @@
#include <stddef.h>
#include <stdint.h>
+#include <optional>
#include <string>
-#include "absl/types/optional.h"
#include "api/field_trials_view.h"
#include "api/task_queue/pending_task_safety_flag.h"
#include "api/units/timestamp.h"
@@ -107,7 +107,7 @@
// Overridden by SrtpTransport.
virtual void OnNetworkRouteChanged(
- absl::optional<rtc::NetworkRoute> network_route);
+ std::optional<rtc::NetworkRoute> network_route);
virtual void OnRtpPacketReceived(const rtc::ReceivedPacket& packet);
virtual void OnRtcpPacketReceived(const rtc::ReceivedPacket& packet);
// Overridden by SrtpTransport and DtlsSrtpTransport.
diff --git a/pc/rtp_transport_internal.h b/pc/rtp_transport_internal.h
index 483a1ce..90537dd 100644
--- a/pc/rtp_transport_internal.h
+++ b/pc/rtp_transport_internal.h
@@ -80,7 +80,7 @@
// The argument is an optional network route.
void SubscribeNetworkRouteChanged(
const void* tag,
- absl::AnyInvocable<void(absl::optional<rtc::NetworkRoute>)> callback) {
+ absl::AnyInvocable<void(std::optional<rtc::NetworkRoute>)> callback) {
callback_list_network_route_changed_.AddReceiver(tag, std::move(callback));
}
void UnsubscribeNetworkRouteChanged(const void* tag) {
@@ -146,7 +146,7 @@
void NotifyUnDemuxableRtpPacketReceived(RtpPacketReceived& packet) {
callback_undemuxable_rtp_packet_received_(packet);
}
- void SendNetworkRouteChanged(absl::optional<rtc::NetworkRoute> route) {
+ void SendNetworkRouteChanged(std::optional<rtc::NetworkRoute> route) {
callback_list_network_route_changed_.Send(route);
}
void SendWritableState(bool state) {
@@ -163,7 +163,7 @@
absl::AnyInvocable<void(RtpPacketReceived&)>
callback_undemuxable_rtp_packet_received_ =
[](RtpPacketReceived& packet) {};
- CallbackList<absl::optional<rtc::NetworkRoute>>
+ CallbackList<std::optional<rtc::NetworkRoute>>
callback_list_network_route_changed_;
CallbackList<bool> callback_list_writable_state_;
CallbackList<const rtc::SentPacket&> callback_list_sent_packet_;
diff --git a/pc/rtp_transport_unittest.cc b/pc/rtp_transport_unittest.cc
index a3ab768..d4a0188 100644
--- a/pc/rtp_transport_unittest.cc
+++ b/pc/rtp_transport_unittest.cc
@@ -40,7 +40,7 @@
transport->SubscribeReadyToSend(
this, [this](bool ready) { OnReadyToSend(ready); });
transport->SubscribeNetworkRouteChanged(
- this, [this](absl::optional<rtc::NetworkRoute> route) {
+ this, [this](std::optional<rtc::NetworkRoute> route) {
OnNetworkRouteChanged(route);
});
if (transport->rtp_packet_transport()) {
@@ -57,8 +57,8 @@
bool ready() const { return ready_; }
void OnReadyToSend(bool ready) { ready_ = ready; }
- absl::optional<rtc::NetworkRoute> network_route() { return network_route_; }
- void OnNetworkRouteChanged(absl::optional<rtc::NetworkRoute> network_route) {
+ std::optional<rtc::NetworkRoute> network_route() { return network_route_; }
+ void OnNetworkRouteChanged(std::optional<rtc::NetworkRoute> network_route) {
network_route_ = network_route;
}
@@ -81,7 +81,7 @@
int rtcp_transport_sent_count_ = 0;
RtpTransport* transport_ = nullptr;
bool ready_ = false;
- absl::optional<rtc::NetworkRoute> network_route_;
+ std::optional<rtc::NetworkRoute> network_route_;
};
TEST(RtpTransportTest, SettingRtcpAndRtpSignalsReady) {
@@ -164,7 +164,7 @@
network_route.remote = rtc::RouteEndpoint::CreateWithNetworkId(kRemoteNetId);
network_route.last_sent_packet_id = kLastPacketId;
network_route.packet_overhead = kTransportOverheadPerPacket;
- fake_rtp.SetNetworkRoute(absl::optional<rtc::NetworkRoute>(network_route));
+ fake_rtp.SetNetworkRoute(std::optional<rtc::NetworkRoute>(network_route));
transport.SetRtpPacketTransport(&fake_rtp);
ASSERT_TRUE(observer.network_route());
EXPECT_TRUE(observer.network_route()->connected);
@@ -193,7 +193,7 @@
network_route.remote = rtc::RouteEndpoint::CreateWithNetworkId(kRemoteNetId);
network_route.last_sent_packet_id = kLastPacketId;
network_route.packet_overhead = kTransportOverheadPerPacket;
- fake_rtcp.SetNetworkRoute(absl::optional<rtc::NetworkRoute>(network_route));
+ fake_rtcp.SetNetworkRoute(std::optional<rtc::NetworkRoute>(network_route));
transport.SetRtcpPacketTransport(&fake_rtcp);
ASSERT_TRUE(observer.network_route());
EXPECT_TRUE(observer.network_route()->connected);
diff --git a/pc/sctp_data_channel.cc b/pc/sctp_data_channel.cc
index 6c2fb0b..bd8d3fa 100644
--- a/pc/sctp_data_channel.cc
+++ b/pc/sctp_data_channel.cc
@@ -45,8 +45,8 @@
BYPASS_PROXY_CONSTMETHOD0(bool, ordered)
BYPASS_PROXY_CONSTMETHOD0(uint16_t, maxRetransmitTime)
BYPASS_PROXY_CONSTMETHOD0(uint16_t, maxRetransmits)
-BYPASS_PROXY_CONSTMETHOD0(absl::optional<int>, maxRetransmitsOpt)
-BYPASS_PROXY_CONSTMETHOD0(absl::optional<int>, maxPacketLifeTime)
+BYPASS_PROXY_CONSTMETHOD0(std::optional<int>, maxRetransmitsOpt)
+BYPASS_PROXY_CONSTMETHOD0(std::optional<int>, maxPacketLifeTime)
BYPASS_PROXY_CONSTMETHOD0(std::string, protocol)
BYPASS_PROXY_CONSTMETHOD0(bool, negotiated)
// Can't bypass the proxy since the id may change.
@@ -85,7 +85,7 @@
if (*maxRetransmits < 0) {
RTC_LOG(LS_ERROR)
<< "Accepting maxRetransmits < 0 for backwards compatibility";
- maxRetransmits = absl::nullopt;
+ maxRetransmits = std::nullopt;
} else if (*maxRetransmits > std::numeric_limits<uint16_t>::max()) {
maxRetransmits = std::numeric_limits<uint16_t>::max();
}
@@ -95,7 +95,7 @@
if (*maxRetransmitTime < 0) {
RTC_LOG(LS_ERROR)
<< "Accepting maxRetransmitTime < 0 for backwards compatibility";
- maxRetransmitTime = absl::nullopt;
+ maxRetransmitTime = std::nullopt;
} else if (*maxRetransmitTime > std::numeric_limits<uint16_t>::max()) {
maxRetransmitTime = std::numeric_limits<uint16_t>::max();
}
@@ -119,7 +119,7 @@
return true;
}
-absl::optional<StreamId> SctpSidAllocator::AllocateSid(rtc::SSLRole role) {
+std::optional<StreamId> SctpSidAllocator::AllocateSid(rtc::SSLRole role) {
RTC_DCHECK_RUN_ON(&sequence_checker_);
int potential_sid = (role == rtc::SSL_CLIENT) ? 0 : 1;
while (potential_sid <= static_cast<int>(cricket::kMaxSctpSid)) {
@@ -129,7 +129,7 @@
potential_sid += 2;
}
RTC_LOG(LS_ERROR) << "SCTP sid allocation pool exhausted.";
- return absl::nullopt;
+ return std::nullopt;
}
bool SctpSidAllocator::ReserveSid(StreamId sid) {
@@ -315,7 +315,7 @@
rtc::Thread* network_thread)
: signaling_thread_(signaling_thread),
network_thread_(network_thread),
- id_n_(config.id == -1 ? absl::nullopt : absl::make_optional(config.id)),
+ id_n_(config.id == -1 ? std::nullopt : std::make_optional(config.id)),
internal_id_(GenerateUniqueId()),
label_(label),
protocol_(config.protocol),
@@ -459,11 +459,11 @@
return max_retransmits_ ? *max_retransmits_ : static_cast<uint16_t>(-1);
}
-absl::optional<int> SctpDataChannel::maxPacketLifeTime() const {
+std::optional<int> SctpDataChannel::maxPacketLifeTime() const {
return max_retransmit_time_;
}
-absl::optional<int> SctpDataChannel::maxRetransmitsOpt() const {
+std::optional<int> SctpDataChannel::maxRetransmitsOpt() const {
return max_retransmits_;
}
diff --git a/pc/sctp_data_channel.h b/pc/sctp_data_channel.h
index ea8e255..108f640 100644
--- a/pc/sctp_data_channel.h
+++ b/pc/sctp_data_channel.h
@@ -14,10 +14,10 @@
#include <stdint.h>
#include <memory>
+#include <optional>
#include <set>
#include <string>
-#include "absl/types/optional.h"
#include "api/data_channel_interface.h"
#include "api/priority.h"
#include "api/rtc_error.h"
@@ -78,7 +78,7 @@
// stream ids in situations where we cannot determine the SSL role from the
// transport for purposes of generating a stream ID.
// See: https://www.rfc-editor.org/rfc/rfc8832.html#name-protocol-overview
- absl::optional<rtc::SSLRole> fallback_ssl_role;
+ std::optional<rtc::SSLRole> fallback_ssl_role;
};
// Helper class to allocate unique IDs for SCTP DataChannels.
@@ -88,8 +88,8 @@
// Gets the first unused odd/even id based on the DTLS role. If `role` is
// SSL_CLIENT, the allocated id starts from 0 and takes even numbers;
// otherwise, the id starts from 1 and takes odd numbers.
- // If a `StreamId` cannot be allocated, `absl::nullopt` is returned.
- absl::optional<StreamId> AllocateSid(rtc::SSLRole role);
+ // If a `StreamId` cannot be allocated, `std::nullopt` is returned.
+ std::optional<StreamId> AllocateSid(rtc::SSLRole role);
// Attempts to reserve a specific sid. Returns false if it's unavailable.
bool ReserveSid(StreamId sid);
@@ -159,8 +159,8 @@
uint16_t maxRetransmitTime() const override;
uint16_t maxRetransmits() const override;
- absl::optional<int> maxPacketLifeTime() const override;
- absl::optional<int> maxRetransmitsOpt() const override;
+ std::optional<int> maxPacketLifeTime() const override;
+ std::optional<int> maxRetransmitsOpt() const override;
std::string protocol() const override;
bool negotiated() const override;
int id() const override;
@@ -221,7 +221,7 @@
// stats purposes (see also `GetStats()`).
int internal_id() const { return internal_id_; }
- absl::optional<StreamId> sid_n() const {
+ std::optional<StreamId> sid_n() const {
RTC_DCHECK_RUN_ON(network_thread_);
return id_n_;
}
@@ -270,14 +270,13 @@
rtc::Thread* const signaling_thread_;
rtc::Thread* const network_thread_;
- absl::optional<StreamId> id_n_ RTC_GUARDED_BY(network_thread_) =
- absl::nullopt;
+ std::optional<StreamId> id_n_ RTC_GUARDED_BY(network_thread_) = std::nullopt;
const int internal_id_;
const std::string label_;
const std::string protocol_;
- const absl::optional<int> max_retransmit_time_;
- const absl::optional<int> max_retransmits_;
- const absl::optional<PriorityValue> priority_;
+ const std::optional<int> max_retransmit_time_;
+ const std::optional<int> max_retransmits_;
+ const std::optional<PriorityValue> priority_;
const bool negotiated_;
const bool ordered_;
// See the body of `MaybeSendOnBufferedAmountChanged`.
diff --git a/pc/sctp_transport.cc b/pc/sctp_transport.cc
index 6473bf4..f26d7ca 100644
--- a/pc/sctp_transport.cc
+++ b/pc/sctp_transport.cc
@@ -11,9 +11,9 @@
#include "pc/sctp_transport.h"
#include <algorithm>
+#include <optional>
#include <utility>
-#include "absl/types/optional.h"
#include "api/dtls_transport_interface.h"
#include "api/priority.h"
#include "api/sequence_checker.h"
@@ -28,8 +28,8 @@
: owner_thread_(rtc::Thread::Current()),
info_(SctpTransportState::kConnecting,
dtls_transport,
- /*max_message_size=*/absl::nullopt,
- /*max_channels=*/absl::nullopt),
+ /*max_message_size=*/std::nullopt,
+ /*max_channels=*/std::nullopt),
internal_sctp_transport_(std::move(internal)),
dtls_transport_(dtls_transport) {
RTC_DCHECK(internal_sctp_transport_.get());
diff --git a/pc/sctp_transport_unittest.cc b/pc/sctp_transport_unittest.cc
index 049cf41..f70bd70 100644
--- a/pc/sctp_transport_unittest.cc
+++ b/pc/sctp_transport_unittest.cc
@@ -10,11 +10,11 @@
#include "pc/sctp_transport.h"
+#include <optional>
#include <utility>
#include <vector>
#include "absl/memory/memory.h"
-#include "absl/types/optional.h"
#include "api/dtls_transport_interface.h"
#include "api/priority.h"
#include "api/transport/data_channel_transport_interface.h"
@@ -58,10 +58,10 @@
bool ReadyToSendData() override { return true; }
void set_debug_name_for_testing(const char* debug_name) override {}
int max_message_size() const override { return 0; }
- absl::optional<int> max_outbound_streams() const override {
+ std::optional<int> max_outbound_streams() const override {
return max_outbound_streams_;
}
- absl::optional<int> max_inbound_streams() const override {
+ std::optional<int> max_inbound_streams() const override {
return max_inbound_streams_;
}
size_t buffered_amount(int sid) const override { return 0; }
@@ -79,8 +79,8 @@
void set_max_inbound_streams(int streams) { max_inbound_streams_ = streams; }
private:
- absl::optional<int> max_outbound_streams_;
- absl::optional<int> max_inbound_streams_;
+ std::optional<int> max_outbound_streams_;
+ std::optional<int> max_inbound_streams_;
std::function<void()> on_connected_callback_;
};
diff --git a/pc/sctp_utils.cc b/pc/sctp_utils.cc
index df1da4d..2d81534 100644
--- a/pc/sctp_utils.cc
+++ b/pc/sctp_utils.cc
@@ -13,8 +13,8 @@
#include <stddef.h>
#include <cstdint>
+#include <optional>
-#include "absl/types/optional.h"
#include "api/priority.h"
#include "rtc_base/byte_buffer.h"
#include "rtc_base/copy_on_write_buffer.h"
@@ -122,8 +122,8 @@
config->ordered = false;
}
- config->maxRetransmits = absl::nullopt;
- config->maxRetransmitTime = absl::nullopt;
+ config->maxRetransmits = std::nullopt;
+ config->maxRetransmitTime = std::nullopt;
switch (channel_type) {
case DCOMCT_ORDERED_PARTIAL_RTXS:
case DCOMCT_UNORDERED_PARTIAL_RTXS:
@@ -162,10 +162,10 @@
bool WriteDataChannelOpenMessage(const std::string& label,
const std::string& protocol,
- absl::optional<PriorityValue> opt_priority,
+ std::optional<PriorityValue> opt_priority,
bool ordered,
- absl::optional<int> max_retransmits,
- absl::optional<int> max_retransmit_time,
+ std::optional<int> max_retransmits,
+ std::optional<int> max_retransmit_time,
rtc::CopyOnWriteBuffer* payload) {
// Format defined at
// http://tools.ietf.org/html/draft-ietf-rtcweb-data-protocol-09#section-5.1
diff --git a/pc/sctp_utils.h b/pc/sctp_utils.h
index 2f4b1f0..16759a1 100644
--- a/pc/sctp_utils.h
+++ b/pc/sctp_utils.h
@@ -64,10 +64,10 @@
bool WriteDataChannelOpenMessage(const std::string& label,
const std::string& protocol,
- absl::optional<PriorityValue> priority,
+ std::optional<PriorityValue> priority,
bool ordered,
- absl::optional<int> max_retransmits,
- absl::optional<int> max_retransmit_time,
+ std::optional<int> max_retransmits,
+ std::optional<int> max_retransmit_time,
rtc::CopyOnWriteBuffer* payload);
bool WriteDataChannelOpenMessage(const std::string& label,
const DataChannelInit& config,
diff --git a/pc/sctp_utils_unittest.cc b/pc/sctp_utils_unittest.cc
index 0907b90..9f4c4a1 100644
--- a/pc/sctp_utils_unittest.cc
+++ b/pc/sctp_utils_unittest.cc
@@ -13,8 +13,8 @@
#include <stdint.h>
#include <limits>
+#include <optional>
-#include "absl/types/optional.h"
#include "api/priority.h"
#include "media/sctp/sctp_transport_internal.h"
#include "rtc_base/byte_buffer.h"
diff --git a/pc/sdp_offer_answer.cc b/pc/sdp_offer_answer.cc
index 199d7ec..5a50867 100644
--- a/pc/sdp_offer_answer.cc
+++ b/pc/sdp_offer_answer.cc
@@ -17,6 +17,7 @@
#include <iterator>
#include <map>
#include <memory>
+#include <optional>
#include <queue>
#include <set>
#include <string>
@@ -27,7 +28,6 @@
#include "absl/memory/memory.h"
#include "absl/strings/match.h"
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/candidate.h"
#include "api/crypto/crypto_options.h"
@@ -3241,7 +3241,7 @@
}
}
-absl::optional<bool> SdpOfferAnswerHandler::is_caller() const {
+std::optional<bool> SdpOfferAnswerHandler::is_caller() const {
RTC_DCHECK_RUN_ON(signaling_thread());
return is_caller_;
}
@@ -3263,7 +3263,7 @@
return pc_->NeedsIceRestart(content_name);
}
-absl::optional<rtc::SSLRole> SdpOfferAnswerHandler::GetDtlsRole(
+std::optional<rtc::SSLRole> SdpOfferAnswerHandler::GetDtlsRole(
const std::string& mid) const {
RTC_DCHECK_RUN_ON(signaling_thread());
return transport_controller_s()->GetDtlsRole(mid);
@@ -3341,11 +3341,11 @@
return;
}
- absl::optional<rtc::SSLRole> guessed_role = GuessSslRole();
+ std::optional<rtc::SSLRole> guessed_role = GuessSslRole();
network_thread()->BlockingCall(
[&, data_channel_controller = data_channel_controller()] {
RTC_DCHECK_RUN_ON(network_thread());
- absl::optional<rtc::SSLRole> role = pc_->GetSctpSslRole_n();
+ std::optional<rtc::SSLRole> role = pc_->GetSctpSslRole_n();
if (!role)
role = guessed_role;
if (role)
@@ -3353,10 +3353,10 @@
});
}
-absl::optional<rtc::SSLRole> SdpOfferAnswerHandler::GuessSslRole() const {
+std::optional<rtc::SSLRole> SdpOfferAnswerHandler::GuessSslRole() const {
RTC_DCHECK_RUN_ON(signaling_thread());
if (!pc_->sctp_mid())
- return absl::nullopt;
+ return std::nullopt;
// TODO(bugs.webrtc.org/13668): This guesswork is guessing wrong (returning
// SSL_CLIENT = ACTIVE) if remote offer has role ACTIVE, but we'll be able
@@ -4188,9 +4188,9 @@
(offer_answer_options.offer_to_receive_video > 0);
}
}
- absl::optional<size_t> audio_index;
- absl::optional<size_t> video_index;
- absl::optional<size_t> data_index;
+ std::optional<size_t> audio_index;
+ std::optional<size_t> video_index;
+ std::optional<size_t> data_index;
// If a current description exists, generate m= sections in the same order,
// using the first audio/video/data section that appears and rejecting
// extraneous ones.
@@ -4454,9 +4454,9 @@
}
}
- absl::optional<size_t> audio_index;
- absl::optional<size_t> video_index;
- absl::optional<size_t> data_index;
+ std::optional<size_t> audio_index;
+ std::optional<size_t> video_index;
+ std::optional<size_t> data_index;
// Generate m= sections that match those in the offer.
// Note that mediasession.cc will handle intersection our preferred
@@ -4979,8 +4979,8 @@
(remote_content && remote_content->rejected)) {
RTC_LOG(LS_INFO) << "Dissociating transceiver"
" since the media section is being recycled.";
- transceiver->internal()->set_mid(absl::nullopt);
- transceiver->internal()->set_mline_index(absl::nullopt);
+ transceiver->internal()->set_mid(std::nullopt);
+ transceiver->internal()->set_mline_index(std::nullopt);
} else if (!local_content && !remote_content) {
// TODO(bugs.webrtc.org/11973): Consider if this should be removed already
// See https://github.com/w3c/webrtc-pc/issues/2576
@@ -5241,9 +5241,9 @@
const SessionDescriptionInterface* session_desc,
RtpTransceiverDirection audio_direction,
RtpTransceiverDirection video_direction,
- absl::optional<size_t>* audio_index,
- absl::optional<size_t>* video_index,
- absl::optional<size_t>* data_index,
+ std::optional<size_t>* audio_index,
+ std::optional<size_t>* video_index,
+ std::optional<size_t>* data_index,
cricket::MediaSessionOptions* session_options) {
RTC_DCHECK_RUN_ON(signaling_thread());
for (const cricket::ContentInfo& content :
diff --git a/pc/sdp_offer_answer.h b/pc/sdp_offer_answer.h
index 8b845f4..13492b6 100644
--- a/pc/sdp_offer_answer.h
+++ b/pc/sdp_offer_answer.h
@@ -17,11 +17,11 @@
#include <functional>
#include <map>
#include <memory>
+#include <optional>
#include <set>
#include <string>
#include <vector>
-#include "absl/types/optional.h"
#include "api/audio_options.h"
#include "api/candidate.h"
#include "api/jsep.h"
@@ -111,7 +111,7 @@
bool NeedsIceRestart(const std::string& content_name) const override;
bool IceRestartPending(const std::string& content_name) const override;
- absl::optional<rtc::SSLRole> GetDtlsRole(
+ std::optional<rtc::SSLRole> GetDtlsRole(
const std::string& mid) const override;
void RestartIce();
@@ -155,7 +155,7 @@
bool AddStream(MediaStreamInterface* local_stream);
void RemoveStream(MediaStreamInterface* local_stream);
- absl::optional<bool> is_caller() const;
+ std::optional<bool> is_caller() const;
bool HasNewIceCredentials();
void UpdateNegotiationNeeded();
void AllocateSctpSids();
@@ -163,7 +163,7 @@
// directly getting the information from the transport.
// This is used for allocating stream ids for data channels.
// See also `InternalDataChannelInit::fallback_ssl_role`.
- absl::optional<rtc::SSLRole> GuessSslRole() const;
+ std::optional<rtc::SSLRole> GuessSslRole() const;
// Destroys all media BaseChannels.
void DestroyMediaChannels();
@@ -531,9 +531,9 @@
const SessionDescriptionInterface* session_desc,
RtpTransceiverDirection audio_direction,
RtpTransceiverDirection video_direction,
- absl::optional<size_t>* audio_index,
- absl::optional<size_t>* video_index,
- absl::optional<size_t>* data_index,
+ std::optional<size_t>* audio_index,
+ std::optional<size_t>* video_index,
+ std::optional<size_t>* data_index,
cricket::MediaSessionOptions* session_options);
// Generates the active MediaDescriptionOptions for the local data channel
@@ -605,7 +605,7 @@
RTC_GUARDED_BY(signaling_thread()) = PeerConnectionInterface::kStable;
// Whether this peer is the caller. Set when the local description is applied.
- absl::optional<bool> is_caller_ RTC_GUARDED_BY(signaling_thread());
+ std::optional<bool> is_caller_ RTC_GUARDED_BY(signaling_thread());
// Streams added via AddStream.
const rtc::scoped_refptr<StreamCollection> local_streams_
@@ -678,7 +678,7 @@
// Whether we are the initial offerer on the association. This
// determines the SSL role.
- absl::optional<bool> initial_offerer_ RTC_GUARDED_BY(signaling_thread());
+ std::optional<bool> initial_offerer_ RTC_GUARDED_BY(signaling_thread());
rtc::WeakPtrFactory<SdpOfferAnswerHandler> weak_ptr_factory_
RTC_GUARDED_BY(signaling_thread());
diff --git a/pc/sdp_state_provider.h b/pc/sdp_state_provider.h
index 23ffc91..8fe5bf5 100644
--- a/pc/sdp_state_provider.h
+++ b/pc/sdp_state_provider.h
@@ -45,7 +45,7 @@
// Whether an ICE restart was indicated in the remote offer.
// Used in CreateAnswer.
virtual bool IceRestartPending(const std::string& content_name) const = 0;
- virtual absl::optional<rtc::SSLRole> GetDtlsRole(
+ virtual std::optional<rtc::SSLRole> GetDtlsRole(
const std::string& mid) const = 0;
};
diff --git a/pc/simulcast_sdp_serializer.cc b/pc/simulcast_sdp_serializer.cc
index ceb2881..225c733 100644
--- a/pc/simulcast_sdp_serializer.cc
+++ b/pc/simulcast_sdp_serializer.cc
@@ -11,6 +11,7 @@
#include "pc/simulcast_sdp_serializer.h"
#include <map>
+#include <optional>
#include <string>
#include <type_traits>
#include <utility>
@@ -18,7 +19,6 @@
#include "absl/algorithm/container.h"
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "rtc_base/checks.h"
#include "rtc_base/string_encode.h"
@@ -164,7 +164,7 @@
}
for (const std::string& payload_type : string_payloads) {
- absl::optional<int> value = rtc::StringToNumber<int>(payload_type);
+ std::optional<int> value = rtc::StringToNumber<int>(payload_type);
if (!value.has_value()) {
return ParseError("Invalid payload type: " + payload_type);
}
diff --git a/pc/slow_peer_connection_integration_test.cc b/pc/slow_peer_connection_integration_test.cc
index 9e49291..fae81b5 100644
--- a/pc/slow_peer_connection_integration_test.cc
+++ b/pc/slow_peer_connection_integration_test.cc
@@ -14,13 +14,13 @@
#include <stdint.h>
#include <memory>
+#include <optional>
#include <string>
#include <utility>
#include <vector>
#include "absl/algorithm/container.h"
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/dtmf_sender_interface.h"
#include "api/peer_connection_interface.h"
#include "api/rtp_receiver_interface.h"
diff --git a/pc/srtp_transport.cc b/pc/srtp_transport.cc
index dd4fb14..aaf7283 100644
--- a/pc/srtp_transport.cc
+++ b/pc/srtp_transport.cc
@@ -173,7 +173,7 @@
}
void SrtpTransport::OnNetworkRouteChanged(
- absl::optional<rtc::NetworkRoute> network_route) {
+ std::optional<rtc::NetworkRoute> network_route) {
// Only append the SRTP overhead when there is a selected network route.
if (network_route) {
int srtp_overhead = 0;
diff --git a/pc/srtp_transport.h b/pc/srtp_transport.h
index e435e4a..ff8b728 100644
--- a/pc/srtp_transport.h
+++ b/pc/srtp_transport.h
@@ -15,10 +15,10 @@
#include <cstdint>
#include <memory>
+#include <optional>
#include <string>
#include <vector>
-#include "absl/types/optional.h"
#include "api/field_trials_view.h"
#include "api/rtc_error.h"
#include "p2p/base/packet_transport_internal.h"
@@ -120,7 +120,7 @@
void OnRtpPacketReceived(const rtc::ReceivedPacket& packet) override;
void OnRtcpPacketReceived(const rtc::ReceivedPacket& packet) override;
void OnNetworkRouteChanged(
- absl::optional<rtc::NetworkRoute> network_route) override;
+ std::optional<rtc::NetworkRoute> network_route) override;
// Override the RtpTransport::OnWritableState.
void OnWritableState(rtc::PacketTransportInternal* packet_transport) override;
@@ -151,8 +151,8 @@
std::unique_ptr<cricket::SrtpSession> send_rtcp_session_;
std::unique_ptr<cricket::SrtpSession> recv_rtcp_session_;
- absl::optional<int> send_crypto_suite_;
- absl::optional<int> recv_crypto_suite_;
+ std::optional<int> send_crypto_suite_;
+ std::optional<int> recv_crypto_suite_;
rtc::ZeroOnFreeBuffer<uint8_t> send_key_;
rtc::ZeroOnFreeBuffer<uint8_t> recv_key_;
diff --git a/pc/test/fake_audio_capture_module.h b/pc/test/fake_audio_capture_module.h
index f25d151..ac3fe61 100644
--- a/pc/test/fake_audio_capture_module.h
+++ b/pc/test/fake_audio_capture_module.h
@@ -137,7 +137,7 @@
int32_t GetPlayoutUnderrunCount() const override { return -1; }
- absl::optional<webrtc::AudioDeviceModule::Stats> GetStats() const override {
+ std::optional<webrtc::AudioDeviceModule::Stats> GetStats() const override {
return webrtc::AudioDeviceModule::Stats();
}
#if defined(WEBRTC_IOS)
diff --git a/pc/test/fake_peer_connection_base.h b/pc/test/fake_peer_connection_base.h
index c69abe5..640f741 100644
--- a/pc/test/fake_peer_connection_base.h
+++ b/pc/test/fake_peer_connection_base.h
@@ -13,11 +13,11 @@
#include <map>
#include <memory>
+#include <optional>
#include <set>
#include <string>
#include <vector>
-#include "absl/types/optional.h"
#include "api/field_trials_view.h"
#include "api/sctp_transport_interface.h"
#include "pc/peer_connection_internal.h"
@@ -229,7 +229,7 @@
return IceGatheringState::kIceGatheringNew;
}
- absl::optional<bool> can_trickle_ice_candidates() { return absl::nullopt; }
+ std::optional<bool> can_trickle_ice_candidates() { return std::nullopt; }
void AddAdaptationResource(rtc::scoped_refptr<Resource> resource) {}
@@ -262,13 +262,11 @@
return {};
}
- absl::optional<std::string> sctp_transport_name() const override {
- return absl::nullopt;
+ std::optional<std::string> sctp_transport_name() const override {
+ return std::nullopt;
}
- absl::optional<std::string> sctp_mid() const override {
- return absl::nullopt;
- }
+ std::optional<std::string> sctp_mid() const override { return std::nullopt; }
std::map<std::string, cricket::TransportStats> GetTransportStatsByNames(
const std::set<std::string>& transport_names) override {
@@ -277,8 +275,8 @@
Call::Stats GetCallStats() override { return Call::Stats(); }
- absl::optional<AudioDeviceModule::Stats> GetAudioDeviceStats() override {
- return absl::nullopt;
+ std::optional<AudioDeviceModule::Stats> GetAudioDeviceStats() override {
+ return std::nullopt;
}
bool GetLocalCertificate(
@@ -327,8 +325,8 @@
cricket::PortAllocator* port_allocator() override { return nullptr; }
LegacyStatsCollector* legacy_stats() override { return nullptr; }
PeerConnectionObserver* Observer() const override { return nullptr; }
- absl::optional<rtc::SSLRole> GetSctpSslRole_n() override {
- return absl::nullopt;
+ std::optional<rtc::SSLRole> GetSctpSslRole_n() override {
+ return std::nullopt;
}
PeerConnectionInterface::IceConnectionState ice_connection_state_internal()
override {
diff --git a/pc/test/fake_peer_connection_for_stats.h b/pc/test/fake_peer_connection_for_stats.h
index 3e45899..e953eff 100644
--- a/pc/test/fake_peer_connection_for_stats.h
+++ b/pc/test/fake_peer_connection_for_stats.h
@@ -52,7 +52,7 @@
}
private:
- absl::optional<cricket::VoiceMediaSendInfo> send_stats_;
+ std::optional<cricket::VoiceMediaSendInfo> send_stats_;
};
class FakeVoiceMediaReceiveChannelForStats
@@ -80,7 +80,7 @@
}
private:
- absl::optional<cricket::VoiceMediaReceiveInfo> receive_stats_;
+ std::optional<cricket::VoiceMediaReceiveInfo> receive_stats_;
};
// Fake VideoMediaChannel where the result of GetStats can be configured.
@@ -108,7 +108,7 @@
}
private:
- absl::optional<cricket::VideoMediaSendInfo> send_stats_;
+ std::optional<cricket::VideoMediaSendInfo> send_stats_;
};
class FakeVideoMediaReceiveChannelForStats
@@ -134,7 +134,7 @@
}
private:
- absl::optional<cricket::VideoMediaReceiveInfo> receive_stats_;
+ std::optional<cricket::VideoMediaReceiveInfo> receive_stats_;
};
constexpr bool kDefaultRtcpMuxRequired = true;
@@ -387,7 +387,7 @@
void SetCallStats(const Call::Stats& call_stats) { call_stats_ = call_stats; }
void SetAudioDeviceStats(
- absl::optional<AudioDeviceModule::Stats> audio_device_stats) {
+ std::optional<AudioDeviceModule::Stats> audio_device_stats) {
audio_device_stats_ = audio_device_stats;
}
@@ -473,7 +473,7 @@
Call::Stats GetCallStats() override { return call_stats_; }
- absl::optional<AudioDeviceModule::Stats> GetAudioDeviceStats() override {
+ std::optional<AudioDeviceModule::Stats> GetAudioDeviceStats() override {
return audio_device_stats_;
}
@@ -558,7 +558,7 @@
Call::Stats call_stats_;
- absl::optional<AudioDeviceModule::Stats> audio_device_stats_;
+ std::optional<AudioDeviceModule::Stats> audio_device_stats_;
std::map<std::string, rtc::scoped_refptr<rtc::RTCCertificate>>
local_certificates_by_transport_;
diff --git a/pc/test/fake_rtc_certificate_generator.h b/pc/test/fake_rtc_certificate_generator.h
index 61da26a..5d60529 100644
--- a/pc/test/fake_rtc_certificate_generator.h
+++ b/pc/test/fake_rtc_certificate_generator.h
@@ -11,10 +11,10 @@
#ifndef PC_TEST_FAKE_RTC_CERTIFICATE_GENERATOR_H_
#define PC_TEST_FAKE_RTC_CERTIFICATE_GENERATOR_H_
+#include <optional>
#include <string>
#include <utility>
-#include "absl/types/optional.h"
#include "api/peer_connection_interface.h"
#include "api/task_queue/task_queue_base.h"
#include "api/units/time_delta.h"
@@ -136,7 +136,7 @@
int generated_failures() { return generated_failures_; }
void GenerateCertificateAsync(const rtc::KeyParams& key_params,
- const absl::optional<uint64_t>& expires_ms,
+ const std::optional<uint64_t>& expires_ms,
Callback callback) override {
// The certificates are created from constant PEM strings and use its coded
// expiration time, we do not support modifying it.
diff --git a/pc/test/frame_generator_capturer_video_track_source.h b/pc/test/frame_generator_capturer_video_track_source.h
index 79a5b34..700a64a 100644
--- a/pc/test/frame_generator_capturer_video_track_source.h
+++ b/pc/test/frame_generator_capturer_video_track_source.h
@@ -49,7 +49,7 @@
video_capturer_ = std::make_unique<test::FrameGeneratorCapturer>(
clock,
test::CreateSquareFrameGenerator(config.width, config.height,
- absl::nullopt,
+ std::nullopt,
config.num_squares_generated),
config.frames_per_second, *task_queue_factory_);
video_capturer_->Init();
diff --git a/pc/test/integration_test_helpers.h b/pc/test/integration_test_helpers.h
index 5b9d632..8938ac2 100644
--- a/pc/test/integration_test_helpers.h
+++ b/pc/test/integration_test_helpers.h
@@ -21,6 +21,7 @@
#include <list>
#include <map>
#include <memory>
+#include <optional>
#include <set>
#include <string>
#include <utility>
@@ -29,7 +30,6 @@
#include "absl/algorithm/container.h"
#include "absl/memory/memory.h"
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/audio/audio_device.h"
#include "api/audio/audio_processing.h"
#include "api/audio_options.h"
@@ -915,7 +915,7 @@
int sdp_mline_index,
const std::string& msg) override {
RTC_LOG(LS_INFO) << debug_name_ << ": ReceiveIceMessage";
- absl::optional<RTCError> result;
+ std::optional<RTCError> result;
pc()->AddIceCandidate(absl::WrapUnique(CreateIceCandidate(
sdp_mid, sdp_mline_index, msg, nullptr)),
[&result](RTCError r) { result = r; });
@@ -1256,7 +1256,7 @@
public:
PeerConnectionIntegrationBaseTest(
SdpSemantics sdp_semantics,
- absl::optional<std::string> field_trials = absl::nullopt)
+ std::optional<std::string> field_trials = std::nullopt)
: sdp_semantics_(sdp_semantics),
ss_(new rtc::VirtualSocketServer()),
fss_(new rtc::FirewallSocketServer(ss_.get())),
diff --git a/pc/test/mock_peer_connection_internal.h b/pc/test/mock_peer_connection_internal.h
index 64d0962..09a11dc 100644
--- a/pc/test/mock_peer_connection_internal.h
+++ b/pc/test/mock_peer_connection_internal.h
@@ -200,7 +200,7 @@
AddAdaptationResource,
(rtc::scoped_refptr<Resource>),
(override));
- MOCK_METHOD(absl::optional<bool>, can_trickle_ice_candidates, (), (override));
+ MOCK_METHOD(std::optional<bool>, can_trickle_ice_candidates, (), (override));
MOCK_METHOD(bool,
StartRtcEventLog,
(std::unique_ptr<RtcEventLogOutput>, int64_t),
@@ -216,7 +216,7 @@
// PeerConnectionSdpMethods
MOCK_METHOD(std::string, session_id, (), (const, override));
MOCK_METHOD(bool, NeedsIceRestart, (const std::string&), (const, override));
- MOCK_METHOD(absl::optional<std::string>, sctp_mid, (), (const, override));
+ MOCK_METHOD(std::optional<std::string>, sctp_mid, (), (const, override));
MOCK_METHOD(PeerConnectionInterface::RTCConfiguration*,
configuration,
(),
@@ -243,7 +243,7 @@
MOCK_METHOD(cricket::PortAllocator*, port_allocator, (), (override));
MOCK_METHOD(LegacyStatsCollector*, legacy_stats, (), (override));
MOCK_METHOD(PeerConnectionObserver*, Observer, (), (const, override));
- MOCK_METHOD(absl::optional<rtc::SSLRole>, GetSctpSslRole_n, (), (override));
+ MOCK_METHOD(std::optional<rtc::SSLRole>, GetSctpSslRole_n, (), (override));
MOCK_METHOD(PeerConnectionInterface::IceConnectionState,
ice_connection_state_internal,
(),
@@ -295,7 +295,7 @@
GetDataChannelStats,
(),
(const, override));
- MOCK_METHOD(absl::optional<std::string>,
+ MOCK_METHOD(std::optional<std::string>,
sctp_transport_name,
(),
(const, override));
@@ -308,7 +308,7 @@
(const std::set<std::string>&),
(override));
MOCK_METHOD(Call::Stats, GetCallStats, (), (override));
- MOCK_METHOD(absl::optional<AudioDeviceModule::Stats>,
+ MOCK_METHOD(std::optional<AudioDeviceModule::Stats>,
GetAudioDeviceStats,
(),
(override));
diff --git a/pc/test/mock_peer_connection_observers.h b/pc/test/mock_peer_connection_observers.h
index 6222ef7..063e0d6 100644
--- a/pc/test/mock_peer_connection_observers.h
+++ b/pc/test/mock_peer_connection_observers.h
@@ -227,7 +227,7 @@
return latest_negotiation_needed_event_.value_or(0u);
}
void clear_latest_negotiation_needed_event() {
- latest_negotiation_needed_event_ = absl::nullopt;
+ latest_negotiation_needed_event_ = std::nullopt;
}
rtc::scoped_refptr<PeerConnectionInterface> pc_;
@@ -236,7 +236,7 @@
rtc::scoped_refptr<DataChannelInterface> last_datachannel_;
rtc::scoped_refptr<StreamCollection> remote_streams_;
bool renegotiation_needed_ = false;
- absl::optional<uint32_t> latest_negotiation_needed_event_;
+ std::optional<uint32_t> latest_negotiation_needed_event_;
bool ice_gathering_complete_ = false;
bool ice_connected_ = false;
bool callback_triggered_ = false;
@@ -352,7 +352,7 @@
private:
// Set on complete, on success this is set to an RTCError::OK() error.
- absl::optional<RTCError> error_;
+ std::optional<RTCError> error_;
};
class FakeSetRemoteDescriptionObserver
@@ -371,7 +371,7 @@
private:
// Set on complete, on success this is set to an RTCError::OK() error.
- absl::optional<RTCError> error_;
+ std::optional<RTCError> error_;
};
class MockDataChannelObserver : public DataChannelObserver {
diff --git a/pc/test/mock_rtp_receiver_internal.h b/pc/test/mock_rtp_receiver_internal.h
index e76b567..c429a9b 100644
--- a/pc/test/mock_rtp_receiver_internal.h
+++ b/pc/test/mock_rtp_receiver_internal.h
@@ -11,10 +11,10 @@
#ifndef PC_TEST_MOCK_RTP_RECEIVER_INTERNAL_H_
#define PC_TEST_MOCK_RTP_RECEIVER_INTERNAL_H_
+#include <optional>
#include <string>
#include <vector>
-#include "absl/types/optional.h"
#include "pc/rtp_receiver.h"
#include "test/gmock.h"
@@ -43,7 +43,7 @@
MOCK_METHOD(void, SetObserver, (RtpReceiverObserverInterface*), (override));
MOCK_METHOD(void,
SetJitterBufferMinimumDelay,
- (absl::optional<double>),
+ (std::optional<double>),
(override));
MOCK_METHOD(std::vector<RtpSource>, GetSources, (), (const, override));
MOCK_METHOD(void,
@@ -63,7 +63,7 @@
(override));
MOCK_METHOD(void, SetupMediaChannel, (uint32_t), (override));
MOCK_METHOD(void, SetupUnsignaledMediaChannel, (), (override));
- MOCK_METHOD(absl::optional<uint32_t>, ssrc, (), (const, override));
+ MOCK_METHOD(std::optional<uint32_t>, ssrc, (), (const, override));
MOCK_METHOD(void, NotifyFirstPacketReceived, (), (override));
MOCK_METHOD(void, set_stream_ids, (std::vector<std::string>), (override));
MOCK_METHOD(void,
diff --git a/pc/test/mock_voice_media_receive_channel_interface.h b/pc/test/mock_voice_media_receive_channel_interface.h
index 8c0f837..a86420f 100644
--- a/pc/test/mock_voice_media_receive_channel_interface.h
+++ b/pc/test/mock_voice_media_receive_channel_interface.h
@@ -92,7 +92,7 @@
OnPacketReceived,
(const webrtc::RtpPacketReceived& packet),
(override));
- MOCK_METHOD(absl::optional<uint32_t>,
+ MOCK_METHOD(std::optional<uint32_t>,
GetUnsignaledSsrc,
(),
(const, override));
@@ -118,7 +118,7 @@
SetBaseMinimumPlayoutDelayMs,
(uint32_t ssrc, int delay_ms),
(override));
- MOCK_METHOD(absl::optional<int>,
+ MOCK_METHOD(std::optional<int>,
GetBaseMinimumPlayoutDelayMs,
(uint32_t ssrc),
(const, override));
diff --git a/pc/test/peer_connection_test_wrapper.cc b/pc/test/peer_connection_test_wrapper.cc
index e5c3811..dc926c7 100644
--- a/pc/test/peer_connection_test_wrapper.cc
+++ b/pc/test/peer_connection_test_wrapper.cc
@@ -13,12 +13,12 @@
#include <stddef.h>
#include <memory>
+#include <optional>
#include <string>
#include <utility>
#include <vector>
#include "absl/strings/match.h"
-#include "absl/types/optional.h"
#include "api/audio/audio_device.h"
#include "api/audio/audio_mixer.h"
#include "api/audio/audio_processing.h"
@@ -81,7 +81,7 @@
std::unique_ptr<webrtc::VideoEncoder> Create(
const Environment& env,
const webrtc::SdpVideoFormat& format) override {
- if (absl::optional<webrtc::SdpVideoFormat> original_format =
+ if (std::optional<webrtc::SdpVideoFormat> original_format =
webrtc::FuzzyMatchSdpVideoFormat(factory_.GetSupportedFormats(),
format)) {
return std::make_unique<webrtc::SimulcastEncoderAdapter>(
@@ -93,7 +93,7 @@
CodecSupport QueryCodecSupport(
const webrtc::SdpVideoFormat& format,
- absl::optional<std::string> scalability_mode) const override {
+ std::optional<std::string> scalability_mode) const override {
return factory_.QueryCodecSupport(format, scalability_mode);
}
@@ -207,7 +207,7 @@
return result.MoveValue();
}
-absl::optional<webrtc::RtpCodecCapability>
+std::optional<webrtc::RtpCodecCapability>
PeerConnectionTestWrapper::FindFirstSendCodecWithName(
cricket::MediaType media_type,
const std::string& name) const {
@@ -218,7 +218,7 @@
return codec;
}
}
- return absl::nullopt;
+ return std::nullopt;
}
void PeerConnectionTestWrapper::WaitForNegotiation() {
diff --git a/pc/test/peer_connection_test_wrapper.h b/pc/test/peer_connection_test_wrapper.h
index 751c946..966bee0 100644
--- a/pc/test/peer_connection_test_wrapper.h
+++ b/pc/test/peer_connection_test_wrapper.h
@@ -65,7 +65,7 @@
const std::string& label,
const webrtc::DataChannelInit& init);
- absl::optional<webrtc::RtpCodecCapability> FindFirstSendCodecWithName(
+ std::optional<webrtc::RtpCodecCapability> FindFirstSendCodecWithName(
cricket::MediaType media_type,
const std::string& name) const;
diff --git a/pc/test/svc_e2e_tests.cc b/pc/test/svc_e2e_tests.cc
index 757e983..332e553 100644
--- a/pc/test/svc_e2e_tests.cc
+++ b/pc/test/svc_e2e_tests.cc
@@ -95,7 +95,7 @@
struct SvcTestParameters {
static SvcTestParameters Create(const std::string& codec_name,
const std::string& scalability_mode_str) {
- absl::optional<ScalabilityMode> scalability_mode =
+ std::optional<ScalabilityMode> scalability_mode =
ScalabilityModeFromString(scalability_mode_str);
RTC_CHECK(scalability_mode.has_value())
<< "Unsupported scalability mode: " << scalability_mode_str;
@@ -177,8 +177,8 @@
const EncodedImage& encoded_image,
const EncoderStats& stats,
bool discarded) override {
- absl::optional<int> spatial_id = encoded_image.SpatialIndex();
- absl::optional<int> temporal_id = encoded_image.TemporalIndex();
+ std::optional<int> spatial_id = encoded_image.SpatialIndex();
+ std::optional<int> temporal_id = encoded_image.TemporalIndex();
encoder_layers_seen_[spatial_id.value_or(0)][temporal_id.value_or(0)]++;
DefaultVideoQualityAnalyzer::OnFrameEncoded(
peer_name, frame_id, encoded_image, stats, discarded);
@@ -187,8 +187,8 @@
void OnFramePreDecode(absl::string_view peer_name,
uint16_t frame_id,
const EncodedImage& input_image) override {
- absl::optional<int> spatial_id = input_image.SpatialIndex();
- absl::optional<int> temporal_id = input_image.TemporalIndex();
+ std::optional<int> spatial_id = input_image.SpatialIndex();
+ std::optional<int> temporal_id = input_image.TemporalIndex();
if (!spatial_id) {
decoder_layers_seen_[0][temporal_id.value_or(0)]++;
} else {
@@ -223,14 +223,14 @@
const SpatialTemporalLayerCounts& decoder_layers_seen() const {
return decoder_layers_seen_;
}
- const absl::optional<std::string> reported_scalability_mode() const {
+ const std::optional<std::string> reported_scalability_mode() const {
return reported_scalability_mode_;
}
private:
SpatialTemporalLayerCounts encoder_layers_seen_;
SpatialTemporalLayerCounts decoder_layers_seen_;
- absl::optional<std::string> reported_scalability_mode_;
+ std::optional<std::string> reported_scalability_mode_;
};
MATCHER_P2(HasSpatialAndTemporalLayers,
diff --git a/pc/track_media_info_map.cc b/pc/track_media_info_map.cc
index ac24d07..7d440b8 100644
--- a/pc/track_media_info_map.cc
+++ b/pc/track_media_info_map.cc
@@ -107,8 +107,8 @@
TrackMediaInfoMap::TrackMediaInfoMap() = default;
void TrackMediaInfoMap::Initialize(
- absl::optional<cricket::VoiceMediaInfo> voice_media_info,
- absl::optional<cricket::VideoMediaInfo> video_media_info,
+ std::optional<cricket::VoiceMediaInfo> voice_media_info,
+ std::optional<cricket::VideoMediaInfo> video_media_info,
rtc::ArrayView<rtc::scoped_refptr<RtpSenderInternal>> rtp_senders,
rtc::ArrayView<rtc::scoped_refptr<RtpReceiverInternal>> rtp_receivers) {
rtc::Thread::ScopedDisallowBlockingCalls no_blocking_calls;
@@ -262,12 +262,12 @@
return FindValueOrNull(video_track_by_receiver_info_, &video_receiver_info);
}
-absl::optional<int> TrackMediaInfoMap::GetAttachmentIdByTrack(
+std::optional<int> TrackMediaInfoMap::GetAttachmentIdByTrack(
const MediaStreamTrackInterface* track) const {
RTC_DCHECK(is_initialized_);
auto it = attachment_id_by_track_.find(track);
- return it != attachment_id_by_track_.end() ? absl::optional<int>(it->second)
- : absl::nullopt;
+ return it != attachment_id_by_track_.end() ? std::optional<int>(it->second)
+ : std::nullopt;
}
} // namespace webrtc
diff --git a/pc/track_media_info_map.h b/pc/track_media_info_map.h
index 98f8548..a3780d1 100644
--- a/pc/track_media_info_map.h
+++ b/pc/track_media_info_map.h
@@ -15,10 +15,10 @@
#include <map>
#include <memory>
+#include <optional>
#include <string>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/media_stream_interface.h"
#include "api/scoped_refptr.h"
@@ -41,16 +41,16 @@
// or receivers, but TrackMediaInfoMap will keep their associated tracks alive
// through reference counting until the map is destroyed.
void Initialize(
- absl::optional<cricket::VoiceMediaInfo> voice_media_info,
- absl::optional<cricket::VideoMediaInfo> video_media_info,
+ std::optional<cricket::VoiceMediaInfo> voice_media_info,
+ std::optional<cricket::VideoMediaInfo> video_media_info,
rtc::ArrayView<rtc::scoped_refptr<RtpSenderInternal>> rtp_senders,
rtc::ArrayView<rtc::scoped_refptr<RtpReceiverInternal>> rtp_receivers);
- const absl::optional<cricket::VoiceMediaInfo>& voice_media_info() const {
+ const std::optional<cricket::VoiceMediaInfo>& voice_media_info() const {
RTC_DCHECK(is_initialized_);
return voice_media_info_;
}
- const absl::optional<cricket::VideoMediaInfo>& video_media_info() const {
+ const std::optional<cricket::VideoMediaInfo>& video_media_info() const {
RTC_DCHECK(is_initialized_);
return video_media_info_;
}
@@ -75,13 +75,13 @@
// It is not going to work if a track is attached multiple times, and
// it is not going to work if a received track is attached as a sending
// track (loopback).
- absl::optional<int> GetAttachmentIdByTrack(
+ std::optional<int> GetAttachmentIdByTrack(
const MediaStreamTrackInterface* track) const;
private:
bool is_initialized_ = false;
- absl::optional<cricket::VoiceMediaInfo> voice_media_info_;
- absl::optional<cricket::VideoMediaInfo> video_media_info_;
+ std::optional<cricket::VoiceMediaInfo> voice_media_info_;
+ std::optional<cricket::VideoMediaInfo> video_media_info_;
// These maps map info objects to their corresponding tracks. They are always
// the inverse of the maps above. One info object always maps to only one
// track. The use of scoped_refptr<> here ensures the tracks outlive
diff --git a/pc/track_media_info_map_unittest.cc b/pc/track_media_info_map_unittest.cc
index bffa3eb..43a97b8 100644
--- a/pc/track_media_info_map_unittest.cc
+++ b/pc/track_media_info_map_unittest.cc
@@ -325,7 +325,7 @@
InitializeMap();
EXPECT_EQ(rtp_senders_[0]->AttachmentId(),
map_.GetAttachmentIdByTrack(local_audio_track_.get()));
- EXPECT_EQ(absl::nullopt,
+ EXPECT_EQ(std::nullopt,
map_.GetAttachmentIdByTrack(local_video_track_.get()));
}
diff --git a/pc/transceiver_list.cc b/pc/transceiver_list.cc
index 250dfbc..e9059f9 100644
--- a/pc/transceiver_list.cc
+++ b/pc/transceiver_list.cc
@@ -22,8 +22,8 @@
}
void TransceiverStableState::SetMSectionIfUnset(
- absl::optional<std::string> mid,
- absl::optional<size_t> mline_index) {
+ std::optional<std::string> mid,
+ std::optional<size_t> mline_index) {
if (!has_m_section_) {
mid_ = mid;
mline_index_ = mline_index;
diff --git a/pc/transceiver_list.h b/pc/transceiver_list.h
index 848ccc2..9e03171 100644
--- a/pc/transceiver_list.h
+++ b/pc/transceiver_list.h
@@ -15,10 +15,10 @@
#include <algorithm>
#include <map>
+#include <optional>
#include <string>
#include <vector>
-#include "absl/types/optional.h"
#include "api/media_types.h"
#include "api/rtc_error.h"
#include "api/rtp_parameters.h"
@@ -41,21 +41,21 @@
public:
TransceiverStableState() {}
void set_newly_created();
- void SetMSectionIfUnset(absl::optional<std::string> mid,
- absl::optional<size_t> mline_index);
+ void SetMSectionIfUnset(std::optional<std::string> mid,
+ std::optional<size_t> mline_index);
void SetRemoteStreamIds(const std::vector<std::string>& ids);
void SetInitSendEncodings(
const std::vector<RtpEncodingParameters>& encodings);
void SetFiredDirection(
- absl::optional<RtpTransceiverDirection> fired_direction) {
+ std::optional<RtpTransceiverDirection> fired_direction) {
fired_direction_ = fired_direction;
}
- absl::optional<std::string> mid() const { return mid_; }
- absl::optional<size_t> mline_index() const { return mline_index_; }
- absl::optional<std::vector<std::string>> remote_stream_ids() const {
+ std::optional<std::string> mid() const { return mid_; }
+ std::optional<size_t> mline_index() const { return mline_index_; }
+ std::optional<std::vector<std::string>> remote_stream_ids() const {
return remote_stream_ids_;
}
- absl::optional<std::vector<RtpEncodingParameters>> init_send_encodings()
+ std::optional<std::vector<RtpEncodingParameters>> init_send_encodings()
const {
return init_send_encodings_;
}
@@ -64,16 +64,16 @@
bool did_set_fired_direction() const { return fired_direction_.has_value(); }
// Because fired_direction() is nullable, did_set_fired_direction() is used to
// distinguish beteen "no value" and "null value".
- absl::optional<RtpTransceiverDirection> fired_direction() const {
+ std::optional<RtpTransceiverDirection> fired_direction() const {
RTC_DCHECK(did_set_fired_direction());
return fired_direction_.value();
}
private:
- absl::optional<std::string> mid_;
- absl::optional<size_t> mline_index_;
- absl::optional<std::vector<std::string>> remote_stream_ids_;
- absl::optional<std::vector<RtpEncodingParameters>> init_send_encodings_;
+ std::optional<std::string> mid_;
+ std::optional<size_t> mline_index_;
+ std::optional<std::vector<std::string>> remote_stream_ids_;
+ std::optional<std::vector<RtpEncodingParameters>> init_send_encodings_;
// Indicates that mid value from stable state has been captured and
// that rollback has to restore the transceiver. Also protects against
// subsequent overwrites.
@@ -84,7 +84,7 @@
bool newly_created_ = false;
// `fired_direction_` is nullable, so an optional of an optional is used to
// distinguish between null and not set (sorry if this hurts your eyes).
- absl::optional<absl::optional<RtpTransceiverDirection>> fired_direction_;
+ std::optional<std::optional<RtpTransceiverDirection>> fired_direction_;
};
// This class encapsulates the active list of transceivers on a
diff --git a/pc/transport_stats.h b/pc/transport_stats.h
index 46dccc9..db98813 100644
--- a/pc/transport_stats.h
+++ b/pc/transport_stats.h
@@ -31,7 +31,7 @@
int ssl_version_bytes = 0;
int srtp_crypto_suite = rtc::kSrtpInvalidCryptoSuite;
int ssl_cipher_suite = rtc::kTlsNullWithNullNull;
- absl::optional<rtc::SSLRole> dtls_role;
+ std::optional<rtc::SSLRole> dtls_role;
webrtc::DtlsTransportState dtls_state = webrtc::DtlsTransportState::kNew;
IceTransportStats ice_transport_stats;
uint16_t ssl_peer_signature_algorithm = rtc::kSslSignatureAlgorithmUnknown;
diff --git a/pc/video_rtp_receiver.cc b/pc/video_rtp_receiver.cc
index 6694d3e..c3aeec9 100644
--- a/pc/video_rtp_receiver.cc
+++ b/pc/video_rtp_receiver.cc
@@ -114,7 +114,7 @@
track_->internal()->set_ended();
}
-void VideoRtpReceiver::RestartMediaChannel(absl::optional<uint32_t> ssrc) {
+void VideoRtpReceiver::RestartMediaChannel(std::optional<uint32_t> ssrc) {
RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
MediaSourceInterface::SourceState state = source_->state();
// TODO(tommi): Can we restart the media channel without blocking?
@@ -126,7 +126,7 @@
}
void VideoRtpReceiver::RestartMediaChannel_w(
- absl::optional<uint32_t> ssrc,
+ std::optional<uint32_t> ssrc,
MediaSourceInterface::SourceState state) {
RTC_DCHECK_RUN_ON(worker_thread_);
if (!media_channel_) {
@@ -184,10 +184,10 @@
void VideoRtpReceiver::SetupUnsignaledMediaChannel() {
RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
- RestartMediaChannel(absl::nullopt);
+ RestartMediaChannel(std::nullopt);
}
-absl::optional<uint32_t> VideoRtpReceiver::ssrc() const {
+std::optional<uint32_t> VideoRtpReceiver::ssrc() const {
RTC_DCHECK_RUN_ON(worker_thread_);
if (!signaled_ssrc_.has_value() && media_channel_) {
return media_channel_->GetUnsignaledSsrc();
@@ -250,7 +250,7 @@
}
void VideoRtpReceiver::SetJitterBufferMinimumDelay(
- absl::optional<double> delay_seconds) {
+ std::optional<double> delay_seconds) {
RTC_DCHECK_RUN_ON(worker_thread_);
delay_.Set(delay_seconds);
if (media_channel_ && signaled_ssrc_)
@@ -326,7 +326,7 @@
}
void VideoRtpReceiver::SetupMediaChannel(
- absl::optional<uint32_t> ssrc,
+ std::optional<uint32_t> ssrc,
cricket::MediaReceiveChannelInterface* media_channel) {
RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
RTC_DCHECK(media_channel);
diff --git a/pc/video_rtp_receiver.h b/pc/video_rtp_receiver.h
index dfa491c..bc42ad3 100644
--- a/pc/video_rtp_receiver.h
+++ b/pc/video_rtp_receiver.h
@@ -13,10 +13,10 @@
#include <stdint.h>
+#include <optional>
#include <string>
#include <vector>
-#include "absl/types/optional.h"
#include "api/crypto/frame_decryptor_interface.h"
#include "api/dtls_transport_interface.h"
#include "api/frame_transformer_interface.h"
@@ -89,7 +89,7 @@
void Stop() override;
void SetupMediaChannel(uint32_t ssrc) override;
void SetupUnsignaledMediaChannel() override;
- absl::optional<uint32_t> ssrc() const override;
+ std::optional<uint32_t> ssrc() const override;
void NotifyFirstPacketReceived() override;
void set_stream_ids(std::vector<std::string> stream_ids) override;
void set_transport(
@@ -100,7 +100,7 @@
void SetObserver(RtpReceiverObserverInterface* observer) override;
void SetJitterBufferMinimumDelay(
- absl::optional<double> delay_seconds) override;
+ std::optional<double> delay_seconds) override;
void SetMediaChannel(
cricket::MediaReceiveChannelInterface* media_channel) override;
@@ -111,13 +111,13 @@
// Combines SetMediaChannel, SetupMediaChannel and
// SetupUnsignaledMediaChannel.
- void SetupMediaChannel(absl::optional<uint32_t> ssrc,
+ void SetupMediaChannel(std::optional<uint32_t> ssrc,
cricket::MediaReceiveChannelInterface* media_channel);
private:
- void RestartMediaChannel(absl::optional<uint32_t> ssrc)
+ void RestartMediaChannel(std::optional<uint32_t> ssrc)
RTC_RUN_ON(&signaling_thread_checker_);
- void RestartMediaChannel_w(absl::optional<uint32_t> ssrc,
+ void RestartMediaChannel_w(std::optional<uint32_t> ssrc,
MediaSourceInterface::SourceState state)
RTC_RUN_ON(worker_thread_);
void SetSink(rtc::VideoSinkInterface<VideoFrame>* sink)
@@ -151,7 +151,7 @@
const std::string id_;
cricket::VideoMediaReceiveChannelInterface* media_channel_
RTC_GUARDED_BY(worker_thread_) = nullptr;
- absl::optional<uint32_t> signaled_ssrc_ RTC_GUARDED_BY(worker_thread_);
+ std::optional<uint32_t> signaled_ssrc_ RTC_GUARDED_BY(worker_thread_);
// `source_` is held here to be able to change the state of the source when
// the VideoRtpReceiver is stopped.
const rtc::scoped_refptr<VideoRtpTrackSource> source_;
diff --git a/pc/video_rtp_track_source_unittest.cc b/pc/video_rtp_track_source_unittest.cc
index 55632ce..62d33f5 100644
--- a/pc/video_rtp_track_source_unittest.cc
+++ b/pc/video_rtp_track_source_unittest.cc
@@ -10,7 +10,8 @@
#include "pc/video_rtp_track_source.h"
-#include "absl/types/optional.h"
+#include <optional>
+
#include "api/scoped_refptr.h"
#include "api/units/timestamp.h"
#include "api/video/color_space.h"
@@ -113,8 +114,8 @@
const override {
return nullptr;
}
- absl::optional<ColorSpace> color_space() const override {
- return absl::nullopt;
+ std::optional<ColorSpace> color_space() const override {
+ return std::nullopt;
}
VideoCodecType codec() const override { return kVideoCodecGeneric; }
bool is_key_frame() const override { return false; }
diff --git a/pc/video_track.h b/pc/video_track.h
index e504182..6733ad0 100644
--- a/pc/video_track.h
+++ b/pc/video_track.h
@@ -11,9 +11,9 @@
#ifndef PC_VIDEO_TRACK_H_
#define PC_VIDEO_TRACK_H_
+#include <optional>
#include <string>
-#include "absl/types/optional.h"
#include "api/media_stream_interface.h"
#include "api/media_stream_track.h"
#include "api/scoped_refptr.h"
diff --git a/pc/video_track_source.h b/pc/video_track_source.h
index 6aae178..75ea4ba 100644
--- a/pc/video_track_source.h
+++ b/pc/video_track_source.h
@@ -11,7 +11,8 @@
#ifndef PC_VIDEO_TRACK_SOURCE_H_
#define PC_VIDEO_TRACK_SOURCE_H_
-#include "absl/types/optional.h"
+#include <optional>
+
#include "api/media_stream_interface.h"
#include "api/notifier.h"
#include "api/sequence_checker.h"
@@ -41,9 +42,7 @@
bool remote() const override { return remote_; }
bool is_screencast() const override { return false; }
- absl::optional<bool> needs_denoising() const override {
- return absl::nullopt;
- }
+ std::optional<bool> needs_denoising() const override { return std::nullopt; }
bool GetStats(Stats* stats) override { return false; }
diff --git a/pc/video_track_source_proxy.h b/pc/video_track_source_proxy.h
index 40d2423..d0ca34b 100644
--- a/pc/video_track_source_proxy.h
+++ b/pc/video_track_source_proxy.h
@@ -11,7 +11,8 @@
#ifndef PC_VIDEO_TRACK_SOURCE_PROXY_H_
#define PC_VIDEO_TRACK_SOURCE_PROXY_H_
-#include "absl/types/optional.h"
+#include <optional>
+
#include "api/media_stream_interface.h"
#include "api/video/recordable_encoded_frame.h"
#include "api/video/video_frame.h"
@@ -32,7 +33,7 @@
PROXY_CONSTMETHOD0(SourceState, state)
BYPASS_PROXY_CONSTMETHOD0(bool, remote)
BYPASS_PROXY_CONSTMETHOD0(bool, is_screencast)
-PROXY_CONSTMETHOD0(absl::optional<bool>, needs_denoising)
+PROXY_CONSTMETHOD0(std::optional<bool>, needs_denoising)
PROXY_METHOD1(bool, GetStats, Stats*)
PROXY_SECONDARY_METHOD2(void,
AddOrUpdateSink,
diff --git a/pc/webrtc_sdp.cc b/pc/webrtc_sdp.cc
index 3e638b0..cab245a 100644
--- a/pc/webrtc_sdp.cc
+++ b/pc/webrtc_sdp.cc
@@ -18,6 +18,7 @@
#include <cstdint>
#include <map>
#include <memory>
+#include <optional>
#include <set>
#include <string>
#include <type_traits>
@@ -28,13 +29,11 @@
#include "absl/algorithm/container.h"
#include "absl/strings/ascii.h"
#include "absl/strings/match.h"
+#include "absl/strings/string_view.h"
#include "api/candidate.h"
#include "api/jsep_ice_candidate.h"
#include "api/jsep_session_description.h"
#include "api/media_types.h"
-// for RtpExtension
-#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/rtc_error.h"
#include "api/rtp_parameters.h"
#include "api/rtp_transceiver_direction.h"
@@ -503,11 +502,11 @@
// Gets line of `message` starting at `pos`, and checks overall SDP syntax. On
// success, advances `pos` to the next line.
-static absl::optional<absl::string_view> GetLine(absl::string_view message,
- size_t* pos) {
+static std::optional<absl::string_view> GetLine(absl::string_view message,
+ size_t* pos) {
size_t line_end = message.find(kNewLineChar, *pos);
if (line_end == absl::string_view::npos) {
- return absl::nullopt;
+ return std::nullopt;
}
absl::string_view line =
TrimReturnChar(message.substr(*pos, line_end - *pos));
@@ -528,7 +527,7 @@
if (line.length() < 3 || !islower(static_cast<unsigned char>(line[0])) ||
line[1] != kSdpDelimiterEqualChar ||
(line[0] != kLineTypeSessionName && line[2] == kSdpDelimiterSpaceChar)) {
- return absl::nullopt;
+ return std::nullopt;
}
*pos = line_end + 1;
return line;
@@ -571,12 +570,12 @@
return IsLineType(line, type, 0);
}
-static absl::optional<absl::string_view>
+static std::optional<absl::string_view>
GetLineWithType(absl::string_view message, size_t* pos, const char type) {
if (IsLineType(message, type, *pos)) {
return GetLine(message, pos);
}
- return absl::nullopt;
+ return std::nullopt;
}
static bool HasAttribute(absl::string_view line, absl::string_view attribute) {
@@ -1102,7 +1101,7 @@
}
SocketAddress address(connection_address, port);
- absl::optional<cricket::ProtocolType> protocol =
+ std::optional<cricket::ProtocolType> protocol =
cricket::StringToProto(transport);
if (!protocol) {
return ParseFailed(first_line, "Unsupported transport type.", error);
@@ -2096,7 +2095,7 @@
rtc::SocketAddress* connection_addr,
cricket::SessionDescription* desc,
SdpParseError* error) {
- absl::optional<absl::string_view> line;
+ std::optional<absl::string_view> line;
desc->set_msid_signaling(cricket::kMsidSignalingNotUsed);
desc->set_extmap_allow_mixed(false);
@@ -2153,7 +2152,7 @@
// RFC 4566
// c=* (connection information -- not required if included in
// all media)
- if (absl::optional<absl::string_view> cline =
+ if (std::optional<absl::string_view> cline =
GetLineWithType(message, pos, kLineTypeConnection);
cline.has_value()) {
if (!ParseConnectionData(*cline, connection_addr, error)) {
@@ -2198,7 +2197,7 @@
// RFC 4566
// a=* (zero or more session attribute lines)
- while (absl::optional<absl::string_view> aline =
+ while (std::optional<absl::string_view> aline =
GetLineWithType(message, pos, kLineTypeAttributes)) {
if (HasAttribute(*aline, kAttributeGroup)) {
if (!ParseGroupAttribute(*aline, desc, error)) {
@@ -2320,7 +2319,7 @@
if (fields.size() != expected_fields) {
return ParseFailedExpectFieldNum(line, expected_fields, error);
}
- if (absl::optional<cricket::ConnectionRole> role =
+ if (std::optional<cricket::ConnectionRole> role =
cricket::StringToConnectionRole(fields[1]);
role.has_value()) {
*role_ptr = *role;
@@ -2702,7 +2701,7 @@
// Zero or more media descriptions
// RFC 4566
// m=<media> <port> <proto> <fmt>
- while (absl::optional<absl::string_view> mline =
+ while (std::optional<absl::string_view> mline =
GetLineWithType(message, pos, kLineTypeMedia)) {
++mline_index;
@@ -2984,7 +2983,7 @@
AddOrReplaceCodec(desc, codec);
}
-absl::optional<cricket::Codec> PopWildcardCodec(
+std::optional<cricket::Codec> PopWildcardCodec(
std::vector<cricket::Codec>* codecs) {
RTC_DCHECK(codecs);
for (auto iter = codecs->begin(); iter != codecs->end(); ++iter) {
@@ -2994,13 +2993,13 @@
return wildcard_codec;
}
}
- return absl::nullopt;
+ return std::nullopt;
}
void UpdateFromWildcardCodecs(cricket::MediaContentDescription* desc) {
RTC_DCHECK(desc);
auto codecs = desc->codecs();
- absl::optional<cricket::Codec> wildcard_codec = PopWildcardCodec(&codecs);
+ std::optional<cricket::Codec> wildcard_codec = PopWildcardCodec(&codecs);
if (!wildcard_codec) {
return;
}
@@ -3063,7 +3062,7 @@
// Loop until the next m line
while (!IsLineType(message, kLineTypeMedia, *pos)) {
- absl::optional<absl::string_view> line = GetLine(message, pos);
+ std::optional<absl::string_view> line = GetLine(message, pos);
if (!line.has_value()) {
if (*pos >= message.size()) {
break; // Done parsing
diff --git a/pc/webrtc_sdp_unittest.cc b/pc/webrtc_sdp_unittest.cc
index 71f6d04..6c08bf4 100644
--- a/pc/webrtc_sdp_unittest.cc
+++ b/pc/webrtc_sdp_unittest.cc
@@ -14,6 +14,7 @@
#include <cstdint>
#include <map>
#include <memory>
+#include <optional>
#include <sstream>
#include <string>
#include <utility>
@@ -23,7 +24,6 @@
#include "absl/memory/memory.h"
#include "absl/strings/str_replace.h"
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/jsep_session_description.h"
#include "api/media_types.h"
@@ -3352,11 +3352,11 @@
cricket::Codec vp9 = vcd->codecs()[1];
EXPECT_EQ(vp9.name, "VP9");
EXPECT_EQ(vp9.id, 121);
- EXPECT_EQ(vp9.packetization, absl::nullopt);
+ EXPECT_EQ(vp9.packetization, std::nullopt);
cricket::Codec h264 = vcd->codecs()[2];
EXPECT_EQ(h264.name, "H264");
EXPECT_EQ(h264.id, 122);
- EXPECT_EQ(h264.packetization, absl::nullopt);
+ EXPECT_EQ(h264.packetization, std::nullopt);
}
TEST_F(WebRtcSdpTest, SerializeAudioFmtpWithUnknownParameter) {
diff --git a/pc/webrtc_session_description_factory.cc b/pc/webrtc_session_description_factory.cc
index 9919260..236ec74 100644
--- a/pc/webrtc_session_description_factory.cc
+++ b/pc/webrtc_session_description_factory.cc
@@ -12,6 +12,7 @@
#include <stddef.h>
+#include <optional>
#include <queue>
#include <string>
#include <type_traits>
@@ -19,7 +20,6 @@
#include <vector>
#include "absl/algorithm/container.h"
-#include "absl/types/optional.h"
#include "api/jsep.h"
#include "api/jsep_session_description.h"
#include "api/rtc_error.h"
@@ -163,7 +163,7 @@
<< key_params.type() << ").";
// Request certificate. This happens asynchronously on a different thread.
- cert_generator_->GenerateCertificateAsync(key_params, absl::nullopt,
+ cert_generator_->GenerateCertificateAsync(key_params, std::nullopt,
std::move(callback));
}
@@ -318,7 +318,7 @@
sdp_info_->IceRestartPending(options.mid);
// We should pass the current DTLS role to the transport description
// factory, if there is already an existing ongoing session.
- absl::optional<rtc::SSLRole> dtls_role =
+ std::optional<rtc::SSLRole> dtls_role =
sdp_info_->GetDtlsRole(options.mid);
if (dtls_role) {
options.transport_options.prefer_passive_role =
diff --git a/rtc_base/BUILD.gn b/rtc_base/BUILD.gn
index 4e06cdd..bbe25fd 100644
--- a/rtc_base/BUILD.gn
+++ b/rtc_base/BUILD.gn
@@ -91,10 +91,7 @@
rtc_source_set("moving_max_counter") {
visibility = [ "*" ]
sources = [ "numerics/moving_max_counter.h" ]
- deps = [
- ":checks",
- "//third_party/abseil-cpp/absl/types:optional",
- ]
+ deps = [ ":checks" ]
}
rtc_source_set("one_time_event") {
@@ -220,10 +217,7 @@
"numerics/histogram_percentile_counter.cc",
"numerics/histogram_percentile_counter.h",
]
- deps = [
- ":checks",
- "//third_party/abseil-cpp/absl/types:optional",
- ]
+ deps = [ ":checks" ]
}
rtc_library("race_checker") {
@@ -264,7 +258,6 @@
"../api/units:time_delta",
"../api/units:timestamp",
"system:rtc_export",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -280,7 +273,6 @@
"../api/units:time_delta",
"../api/units:timestamp",
"system:rtc_export",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -297,7 +289,6 @@
":logging",
":safe_conversions",
"system:rtc_export",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -322,7 +313,6 @@
deps = [
":checks",
":safe_conversions",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -404,7 +394,6 @@
":timeutils",
"../api:sequence_checker",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -431,7 +420,6 @@
"../api/units:time_delta",
"synchronization:yield_policy",
"system:warn_current_thread_is_deadlocked",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
}
@@ -454,7 +442,6 @@
"//third_party/abseil-cpp/absl/base:core_headers",
"//third_party/abseil-cpp/absl/meta:type_traits",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
if (build_with_chromium) {
@@ -523,7 +510,6 @@
":rate_statistics",
"../system_wrappers",
"synchronization:mutex",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -614,7 +600,6 @@
":safe_minmax",
"../api:array_view",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -648,7 +633,6 @@
"../api:scoped_refptr",
"../api:sequence_checker",
"system:no_unique_address",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -722,7 +706,6 @@
"synchronization:mutex",
"//third_party/abseil-cpp/absl/functional:any_invocable",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
}
@@ -798,7 +781,6 @@
deps = [
":checks",
":mod_ops",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -1082,7 +1064,6 @@
"./network:ecn_marking",
"system:rtc_export",
"third_party/sigslot",
- "//third_party/abseil-cpp/absl/types:optional",
]
if (is_win) {
deps += [ ":win32" ]
@@ -1322,7 +1303,6 @@
"network:received_packet",
"network:sent_packet",
"system:no_unique_address",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -1388,7 +1368,6 @@
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -1460,7 +1439,6 @@
"../api:scoped_refptr",
"system:rtc_export",
"//third_party/abseil-cpp/absl/functional:any_invocable",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -1634,7 +1612,6 @@
"//third_party/abseil-cpp/absl/functional:any_invocable",
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/strings",
- "//third_party/abseil-cpp/absl/types:optional",
]
# If we are building the SSL library ourselves, we know it's BoringSSL.
@@ -1773,7 +1750,6 @@
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
if (is_fuchsia) {
deps += [ "//third_party/fuchsia-sdk/sdk/pkg/zx" ]
@@ -2007,7 +1983,6 @@
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/numeric:bits",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
if (is_win) {
@@ -2067,7 +2042,6 @@
"../test:test_main",
"../test:test_support",
"//third_party/abseil-cpp/absl/algorithm:container",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -2172,7 +2146,6 @@
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
if (rtc_enable_google_benchmarks) {
deps += [ "synchronization:synchronization_unittests" ]
diff --git a/rtc_base/async_udp_socket.cc b/rtc_base/async_udp_socket.cc
index e9c16dc..f6e2c51 100644
--- a/rtc_base/async_udp_socket.cc
+++ b/rtc_base/async_udp_socket.cc
@@ -10,7 +10,8 @@
#include "rtc_base/async_udp_socket.h"
-#include "absl/types/optional.h"
+#include <optional>
+
#include "api/units/time_delta.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
diff --git a/rtc_base/async_udp_socket.h b/rtc_base/async_udp_socket.h
index af361b9..5eb4b4c 100644
--- a/rtc_base/async_udp_socket.h
+++ b/rtc_base/async_udp_socket.h
@@ -15,8 +15,8 @@
#include <cstdint>
#include <memory>
+#include <optional>
-#include "absl/types/optional.h"
#include "api/sequence_checker.h"
#include "api/units/time_delta.h"
#include "rtc_base/async_packet_socket.h"
@@ -70,7 +70,7 @@
RTC_NO_UNIQUE_ADDRESS webrtc::SequenceChecker sequence_checker_;
std::unique_ptr<Socket> socket_;
rtc::Buffer buffer_ RTC_GUARDED_BY(sequence_checker_);
- absl::optional<webrtc::TimeDelta> socket_time_offset_
+ std::optional<webrtc::TimeDelta> socket_time_offset_
RTC_GUARDED_BY(sequence_checker_);
};
diff --git a/rtc_base/bitrate_tracker.cc b/rtc_base/bitrate_tracker.cc
index 340e444..0146d3c 100644
--- a/rtc_base/bitrate_tracker.cc
+++ b/rtc_base/bitrate_tracker.cc
@@ -10,7 +10,8 @@
#include "rtc_base/bitrate_tracker.h"
-#include "absl/types/optional.h"
+#include <optional>
+
#include "api/units/data_rate.h"
#include "api/units/timestamp.h"
#include "rtc_base/rate_statistics.h"
@@ -20,11 +21,11 @@
BitrateTracker::BitrateTracker(TimeDelta max_window_size)
: impl_(max_window_size.ms(), RateStatistics::kBpsScale) {}
-absl::optional<DataRate> BitrateTracker::Rate(Timestamp now) const {
- if (absl::optional<int64_t> rate = impl_.Rate(now.ms())) {
+std::optional<DataRate> BitrateTracker::Rate(Timestamp now) const {
+ if (std::optional<int64_t> rate = impl_.Rate(now.ms())) {
return DataRate::BitsPerSec(*rate);
}
- return absl::nullopt;
+ return std::nullopt;
}
bool BitrateTracker::SetWindowSize(TimeDelta window_size, Timestamp now) {
diff --git a/rtc_base/bitrate_tracker.h b/rtc_base/bitrate_tracker.h
index a54bd9a..9570cd6 100644
--- a/rtc_base/bitrate_tracker.h
+++ b/rtc_base/bitrate_tracker.h
@@ -14,7 +14,8 @@
#include <stddef.h>
#include <stdint.h>
-#include "absl/types/optional.h"
+#include <optional>
+
#include "api/units/data_rate.h"
#include "api/units/data_size.h"
#include "api/units/time_delta.h"
@@ -50,7 +51,7 @@
// Returns bitrate, moving averaging window as needed.
// Returns nullopt when bitrate can't be measured.
- absl::optional<DataRate> Rate(Timestamp now) const;
+ std::optional<DataRate> Rate(Timestamp now) const;
// Update the size of the averaging window. The maximum allowed value for
// `window_size` is `max_window_size` as supplied in the constructor.
diff --git a/rtc_base/bitrate_tracker_unittest.cc b/rtc_base/bitrate_tracker_unittest.cc
index c9e3d1e..61f569d 100644
--- a/rtc_base/bitrate_tracker_unittest.cc
+++ b/rtc_base/bitrate_tracker_unittest.cc
@@ -12,8 +12,8 @@
#include <cstdlib>
#include <limits>
+#include <optional>
-#include "absl/types/optional.h"
#include "api/units/data_rate.h"
#include "api/units/time_delta.h"
#include "api/units/timestamp.h"
@@ -34,7 +34,7 @@
Timestamp now = Timestamp::Seconds(12'345);
BitrateTracker stats(kWindow);
- EXPECT_EQ(stats.Rate(now), absl::nullopt);
+ EXPECT_EQ(stats.Rate(now), std::nullopt);
}
TEST(BitrateTrackerTest, ReturnsNulloptAfterSingleDataPoint) {
@@ -44,7 +44,7 @@
stats.Update(1'500, now);
now += TimeDelta::Millis(10);
- EXPECT_EQ(stats.Rate(now), absl::nullopt);
+ EXPECT_EQ(stats.Rate(now), std::nullopt);
}
TEST(BitrateTrackerTest, ReturnsRateAfterTwoMeasurements) {
@@ -82,7 +82,7 @@
// Until window is full, bitrate is measured over a smaller window and might
// look larger than the constant rate.
- absl::optional<DataRate> bitrate = stats.Rate(now);
+ std::optional<DataRate> bitrate = stats.Rate(now);
ASSERT_THAT(bitrate,
AllOf(Ge(kConstantRate), Le(total_size / (now - start))));
@@ -115,7 +115,7 @@
now += kLargeInterval;
stats.Update(kLargePacketSize, now);
}
- absl::optional<DataRate> last_bitrate = stats.Rate(now);
+ std::optional<DataRate> last_bitrate = stats.Rate(now);
EXPECT_EQ(last_bitrate, kLargePacketSize / kLargeInterval);
// Decrease bitrate with smaller measurments.
@@ -124,7 +124,7 @@
now += kLargeInterval;
stats.Update(kSmallPacketSize, now);
- absl::optional<DataRate> bitrate = stats.Rate(now);
+ std::optional<DataRate> bitrate = stats.Rate(now);
EXPECT_LT(bitrate, last_bitrate);
last_bitrate = bitrate;
@@ -137,7 +137,7 @@
now += kSmallInterval;
stats.Update(kSmallPacketSize, now);
- absl::optional<DataRate> bitrate = stats.Rate(now);
+ std::optional<DataRate> bitrate = stats.Rate(now);
EXPECT_GE(bitrate, last_bitrate);
last_bitrate = bitrate;
@@ -162,17 +162,17 @@
now += kWindow + kEpsilon;
// Silence over window size should trigger auto reset for coming sample.
- EXPECT_EQ(stats.Rate(now), absl::nullopt);
+ EXPECT_EQ(stats.Rate(now), std::nullopt);
stats.Update(kPacketSize, now);
// Single measurment after reset is not enough to estimate the rate.
- EXPECT_EQ(stats.Rate(now), absl::nullopt);
+ EXPECT_EQ(stats.Rate(now), std::nullopt);
// Manual reset, add the same check again.
stats.Reset();
- EXPECT_EQ(stats.Rate(now), absl::nullopt);
+ EXPECT_EQ(stats.Rate(now), std::nullopt);
now += kInterval;
stats.Update(kPacketSize, now);
- EXPECT_EQ(stats.Rate(now), absl::nullopt);
+ EXPECT_EQ(stats.Rate(now), std::nullopt);
}
TEST(BitrateTrackerTest, HandlesChangingWindowSize) {
@@ -223,17 +223,17 @@
BitrateTracker stats(kWindow);
stats.Update(kPacketSize, now);
- ASSERT_EQ(stats.Rate(now), absl::nullopt);
+ ASSERT_EQ(stats.Rate(now), std::nullopt);
now += kInterval;
stats.Update(0, now);
- absl::optional<DataRate> last_bitrate = stats.Rate(now);
+ std::optional<DataRate> last_bitrate = stats.Rate(now);
EXPECT_GT(last_bitrate, DataRate::Zero());
now += kInterval;
while (now < start + kWindow) {
SCOPED_TRACE(ToString(now - start));
stats.Update(0, now);
- absl::optional<DataRate> bitrate = stats.Rate(now);
+ std::optional<DataRate> bitrate = stats.Rate(now);
EXPECT_GT(bitrate, DataRate::Zero());
// As window expands, average bitrate decreases.
EXPECT_LT(bitrate, last_bitrate);
@@ -260,7 +260,7 @@
now += kEpsilon;
stats.Update(very_large_number, now);
- EXPECT_EQ(stats.Rate(now), absl::nullopt);
+ EXPECT_EQ(stats.Rate(now), std::nullopt);
}
} // namespace
diff --git a/rtc_base/bitstream_reader_unittest.cc b/rtc_base/bitstream_reader_unittest.cc
index 46309b2..b9b51a2 100644
--- a/rtc_base/bitstream_reader_unittest.cc
+++ b/rtc_base/bitstream_reader_unittest.cc
@@ -15,9 +15,9 @@
#include <array>
#include <limits>
+#include <optional>
#include "absl/numeric/bits.h"
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "rtc_base/checks.h"
#include "test/gmock.h"
@@ -28,26 +28,26 @@
TEST(BitstreamReaderTest, InDebugModeRequiresToCheckOkStatusBeforeDestruction) {
const uint8_t bytes[32] = {};
- absl::optional<BitstreamReader> reader(absl::in_place, bytes);
+ std::optional<BitstreamReader> reader(absl::in_place, bytes);
EXPECT_GE(reader->ReadBits(7), 0u);
#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(OS_ANDROID)
- EXPECT_DEATH(reader = absl::nullopt, "");
+ EXPECT_DEATH(reader = std::nullopt, "");
#endif
EXPECT_TRUE(reader->Ok());
- reader = absl::nullopt;
+ reader = std::nullopt;
}
TEST(BitstreamReaderTest, InDebugModeMayCheckRemainingBitsInsteadOfOkStatus) {
const uint8_t bytes[32] = {};
- absl::optional<BitstreamReader> reader(absl::in_place, bytes);
+ std::optional<BitstreamReader> reader(absl::in_place, bytes);
EXPECT_GE(reader->ReadBit(), 0);
#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(OS_ANDROID)
- EXPECT_DEATH(reader = absl::nullopt, "");
+ EXPECT_DEATH(reader = std::nullopt, "");
#endif
EXPECT_GE(reader->RemainingBitCount(), 0);
- reader = absl::nullopt;
+ reader = std::nullopt;
}
TEST(BitstreamReaderTest, ConsumeBits) {
diff --git a/rtc_base/event.cc b/rtc_base/event.cc
index c2f6f8a..37d3f63 100644
--- a/rtc_base/event.cc
+++ b/rtc_base/event.cc
@@ -21,7 +21,8 @@
#error "Must define either WEBRTC_WIN or WEBRTC_POSIX."
#endif
-#include "absl/types/optional.h"
+#include <optional>
+
#include "rtc_base/checks.h"
#include "rtc_base/synchronization/yield_policy.h"
#include "rtc_base/system/warn_current_thread_is_deadlocked.h"
@@ -148,27 +149,26 @@
// Instant when we'll log a warning message (because we've been waiting so
// long it might be a bug), but not yet give up waiting. nullopt if we
// shouldn't log a warning.
- const absl::optional<timespec> warn_ts =
- warn_after >= give_up_after
- ? absl::nullopt
- : absl::make_optional(GetTimespec(warn_after));
+ const std::optional<timespec> warn_ts =
+ warn_after >= give_up_after ? std::nullopt
+ : std::make_optional(GetTimespec(warn_after));
// Instant when we'll stop waiting and return an error. nullopt if we should
// never give up.
- const absl::optional<timespec> give_up_ts =
+ const std::optional<timespec> give_up_ts =
give_up_after.IsPlusInfinity()
- ? absl::nullopt
- : absl::make_optional(GetTimespec(give_up_after));
+ ? std::nullopt
+ : std::make_optional(GetTimespec(give_up_after));
ScopedYieldPolicy::YieldExecution();
pthread_mutex_lock(&event_mutex_);
// Wait for `event_cond_` to trigger and `event_status_` to be set, with the
// given timeout (or without a timeout if none is given).
- const auto wait = [&](const absl::optional<timespec> timeout_ts) {
+ const auto wait = [&](const std::optional<timespec> timeout_ts) {
int error = 0;
while (!event_status_ && error == 0) {
- if (timeout_ts == absl::nullopt) {
+ if (timeout_ts == std::nullopt) {
error = pthread_cond_wait(&event_cond_, &event_mutex_);
} else {
#if USE_PTHREAD_COND_TIMEDWAIT_MONOTONIC_NP
@@ -184,7 +184,7 @@
};
int error;
- if (warn_ts == absl::nullopt) {
+ if (warn_ts == std::nullopt) {
error = wait(give_up_ts);
} else {
error = wait(warn_ts);
diff --git a/rtc_base/experiments/BUILD.gn b/rtc_base/experiments/BUILD.gn
index dc6f8b2..8b21a61 100644
--- a/rtc_base/experiments/BUILD.gn
+++ b/rtc_base/experiments/BUILD.gn
@@ -17,7 +17,6 @@
"..:logging",
"../../api:field_trials_view",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -42,7 +41,6 @@
"../../rtc_base:stringutils",
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -55,7 +53,6 @@
":field_trial_parser",
"..:logging",
"../../api:field_trials_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -70,7 +67,6 @@
"../../api/transport:field_trial_based_config",
"../../api/video_codecs:video_codecs_api",
"//third_party/abseil-cpp/absl/strings",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -82,7 +78,6 @@
deps = [
"..:logging",
"../../api:field_trials_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -97,7 +92,6 @@
"../../api:field_trials_view",
"../../api/video_codecs:video_codecs_api",
"../../system_wrappers:field_trial",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -112,7 +106,6 @@
"../../api:field_trials_view",
"../../api/video_codecs:video_codecs_api",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -130,7 +123,6 @@
"../../api/video_codecs:video_codecs_api",
"../../video/config:encoder_config",
"//third_party/abseil-cpp/absl/strings",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -142,7 +134,6 @@
deps = [
":field_trial_parser",
"../../api:field_trials_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -170,7 +161,6 @@
"../../api/video:video_frame",
"../../rtc_base:checks",
"../../rtc_base:logging",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -217,7 +207,6 @@
"../../test:test_support",
"../../video/config:encoder_config",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
}
diff --git a/rtc_base/experiments/alr_experiment.cc b/rtc_base/experiments/alr_experiment.cc
index 5370de5..1c5ca14 100644
--- a/rtc_base/experiments/alr_experiment.cc
+++ b/rtc_base/experiments/alr_experiment.cc
@@ -34,11 +34,11 @@
key_value_config.Lookup(kScreenshareProbingBweExperimentName).empty();
}
-absl::optional<AlrExperimentSettings>
+std::optional<AlrExperimentSettings>
AlrExperimentSettings::CreateFromFieldTrial(
const FieldTrialsView& key_value_config,
absl::string_view experiment_name) {
- absl::optional<AlrExperimentSettings> ret;
+ std::optional<AlrExperimentSettings> ret;
std::string group_name = key_value_config.Lookup(experiment_name);
const std::string kIgnoredSuffix = "_Dogfood";
diff --git a/rtc_base/experiments/alr_experiment.h b/rtc_base/experiments/alr_experiment.h
index 9914828..bc4514e 100644
--- a/rtc_base/experiments/alr_experiment.h
+++ b/rtc_base/experiments/alr_experiment.h
@@ -13,8 +13,9 @@
#include <stdint.h>
+#include <optional>
+
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/field_trials_view.h"
namespace webrtc {
@@ -35,7 +36,7 @@
static constexpr absl::string_view kStrictPacingAndProbingExperimentName =
"WebRTC-StrictPacingAndProbing";
- static absl::optional<AlrExperimentSettings> CreateFromFieldTrial(
+ static std::optional<AlrExperimentSettings> CreateFromFieldTrial(
const FieldTrialsView& key_value_config,
absl::string_view experiment_name);
static bool MaxOneFieldTrialEnabled(const FieldTrialsView& key_value_config);
diff --git a/rtc_base/experiments/balanced_degradation_settings.cc b/rtc_base/experiments/balanced_degradation_settings.cc
index 1a269b4..8e0b513 100644
--- a/rtc_base/experiments/balanced_degradation_settings.cc
+++ b/rtc_base/experiments/balanced_degradation_settings.cc
@@ -144,11 +144,11 @@
return DefaultConfigs();
}
-absl::optional<VideoEncoder::QpThresholds> GetThresholds(
+std::optional<VideoEncoder::QpThresholds> GetThresholds(
VideoCodecType type,
const BalancedDegradationSettings::Config& config) {
- absl::optional<int> low;
- absl::optional<int> high;
+ std::optional<int> low;
+ std::optional<int> high;
switch (type) {
case kVideoCodecVP8:
@@ -179,19 +179,19 @@
if (low && high) {
RTC_LOG(LS_INFO) << "QP thresholds: low: " << *low << ", high: " << *high;
- return absl::optional<VideoEncoder::QpThresholds>(
+ return std::optional<VideoEncoder::QpThresholds>(
VideoEncoder::QpThresholds(*low, *high));
}
- return absl::nullopt;
+ return std::nullopt;
}
int GetFps(VideoCodecType type,
- const absl::optional<BalancedDegradationSettings::Config>& config) {
+ const std::optional<BalancedDegradationSettings::Config>& config) {
if (!config.has_value()) {
return std::numeric_limits<int>::max();
}
- absl::optional<int> fps;
+ std::optional<int> fps;
switch (type) {
case kVideoCodecVP8:
fps = config->vp8.GetFps();
@@ -219,13 +219,13 @@
return (framerate == kMaxFps) ? std::numeric_limits<int>::max() : framerate;
}
-absl::optional<int> GetKbps(
+std::optional<int> GetKbps(
VideoCodecType type,
- const absl::optional<BalancedDegradationSettings::Config>& config) {
+ const std::optional<BalancedDegradationSettings::Config>& config) {
if (!config.has_value())
- return absl::nullopt;
+ return std::nullopt;
- absl::optional<int> kbps;
+ std::optional<int> kbps;
switch (type) {
case kVideoCodecVP8:
kbps = config->vp8.GetKbps();
@@ -251,16 +251,16 @@
if (kbps.has_value())
return kbps;
- return config->kbps > 0 ? absl::optional<int>(config->kbps) : absl::nullopt;
+ return config->kbps > 0 ? std::optional<int>(config->kbps) : std::nullopt;
}
-absl::optional<int> GetKbpsRes(
+std::optional<int> GetKbpsRes(
VideoCodecType type,
- const absl::optional<BalancedDegradationSettings::Config>& config) {
+ const std::optional<BalancedDegradationSettings::Config>& config) {
if (!config.has_value())
- return absl::nullopt;
+ return std::nullopt;
- absl::optional<int> kbps_res;
+ std::optional<int> kbps_res;
switch (type) {
case kVideoCodecVP8:
kbps_res = config->vp8.GetKbpsRes();
@@ -286,34 +286,34 @@
if (kbps_res.has_value())
return kbps_res;
- return config->kbps_res > 0 ? absl::optional<int>(config->kbps_res)
- : absl::nullopt;
+ return config->kbps_res > 0 ? std::optional<int>(config->kbps_res)
+ : std::nullopt;
}
} // namespace
-absl::optional<int> BalancedDegradationSettings::CodecTypeSpecific::GetQpLow()
+std::optional<int> BalancedDegradationSettings::CodecTypeSpecific::GetQpLow()
const {
- return (qp_low > 0) ? absl::optional<int>(qp_low) : absl::nullopt;
+ return (qp_low > 0) ? std::optional<int>(qp_low) : std::nullopt;
}
-absl::optional<int> BalancedDegradationSettings::CodecTypeSpecific::GetQpHigh()
+std::optional<int> BalancedDegradationSettings::CodecTypeSpecific::GetQpHigh()
const {
- return (qp_high > 0) ? absl::optional<int>(qp_high) : absl::nullopt;
+ return (qp_high > 0) ? std::optional<int>(qp_high) : std::nullopt;
}
-absl::optional<int> BalancedDegradationSettings::CodecTypeSpecific::GetFps()
+std::optional<int> BalancedDegradationSettings::CodecTypeSpecific::GetFps()
const {
- return (fps > 0) ? absl::optional<int>(fps) : absl::nullopt;
+ return (fps > 0) ? std::optional<int>(fps) : std::nullopt;
}
-absl::optional<int> BalancedDegradationSettings::CodecTypeSpecific::GetKbps()
+std::optional<int> BalancedDegradationSettings::CodecTypeSpecific::GetKbps()
const {
- return (kbps > 0) ? absl::optional<int>(kbps) : absl::nullopt;
+ return (kbps > 0) ? std::optional<int>(kbps) : std::nullopt;
}
-absl::optional<int> BalancedDegradationSettings::CodecTypeSpecific::GetKbpsRes()
+std::optional<int> BalancedDegradationSettings::CodecTypeSpecific::GetKbpsRes()
const {
- return (kbps_res > 0) ? absl::optional<int>(kbps_res) : absl::nullopt;
+ return (kbps_res > 0) ? std::optional<int>(kbps_res) : std::nullopt;
}
BalancedDegradationSettings::Config::Config() = default;
@@ -415,32 +415,32 @@
return GetFps(type, GetMinFpsConfig(pixels));
}
-absl::optional<BalancedDegradationSettings::Config>
+std::optional<BalancedDegradationSettings::Config>
BalancedDegradationSettings::GetMinFpsConfig(int pixels) const {
for (const auto& config : configs_) {
if (pixels <= config.pixels)
return config;
}
- return absl::nullopt;
+ return std::nullopt;
}
int BalancedDegradationSettings::MaxFps(VideoCodecType type, int pixels) const {
return GetFps(type, GetMaxFpsConfig(pixels));
}
-absl::optional<BalancedDegradationSettings::Config>
+std::optional<BalancedDegradationSettings::Config>
BalancedDegradationSettings::GetMaxFpsConfig(int pixels) const {
for (size_t i = 0; i < configs_.size() - 1; ++i) {
if (pixels <= configs_[i].pixels)
return configs_[i + 1];
}
- return absl::nullopt;
+ return std::nullopt;
}
bool BalancedDegradationSettings::CanAdaptUp(VideoCodecType type,
int pixels,
uint32_t bitrate_bps) const {
- absl::optional<int> min_kbps = GetKbps(type, GetMaxFpsConfig(pixels));
+ std::optional<int> min_kbps = GetKbps(type, GetMaxFpsConfig(pixels));
if (!min_kbps.has_value() || bitrate_bps == 0) {
return true; // No limit configured or bitrate provided.
}
@@ -451,25 +451,25 @@
VideoCodecType type,
int pixels,
uint32_t bitrate_bps) const {
- absl::optional<int> min_kbps = GetKbpsRes(type, GetMaxFpsConfig(pixels));
+ std::optional<int> min_kbps = GetKbpsRes(type, GetMaxFpsConfig(pixels));
if (!min_kbps.has_value() || bitrate_bps == 0) {
return true; // No limit configured or bitrate provided.
}
return bitrate_bps >= static_cast<uint32_t>(min_kbps.value() * 1000);
}
-absl::optional<int> BalancedDegradationSettings::MinFpsDiff(int pixels) const {
+std::optional<int> BalancedDegradationSettings::MinFpsDiff(int pixels) const {
for (const auto& config : configs_) {
if (pixels <= config.pixels) {
return (config.fps_diff > kNoFpsDiff)
- ? absl::optional<int>(config.fps_diff)
- : absl::nullopt;
+ ? std::optional<int>(config.fps_diff)
+ : std::nullopt;
}
}
- return absl::nullopt;
+ return std::nullopt;
}
-absl::optional<VideoEncoder::QpThresholds>
+std::optional<VideoEncoder::QpThresholds>
BalancedDegradationSettings::GetQpThresholds(VideoCodecType type,
int pixels) const {
return GetThresholds(type, GetConfig(pixels));
diff --git a/rtc_base/experiments/balanced_degradation_settings.h b/rtc_base/experiments/balanced_degradation_settings.h
index 2bca73d..6993681 100644
--- a/rtc_base/experiments/balanced_degradation_settings.h
+++ b/rtc_base/experiments/balanced_degradation_settings.h
@@ -11,9 +11,9 @@
#ifndef RTC_BASE_EXPERIMENTS_BALANCED_DEGRADATION_SETTINGS_H_
#define RTC_BASE_EXPERIMENTS_BALANCED_DEGRADATION_SETTINGS_H_
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/field_trials_view.h"
#include "api/video_codecs/video_encoder.h"
@@ -40,11 +40,11 @@
kbps == o.kbps && kbps_res == o.kbps_res;
}
- absl::optional<int> GetQpLow() const;
- absl::optional<int> GetQpHigh() const;
- absl::optional<int> GetFps() const;
- absl::optional<int> GetKbps() const;
- absl::optional<int> GetKbpsRes() const;
+ std::optional<int> GetQpLow() const;
+ std::optional<int> GetQpHigh() const;
+ std::optional<int> GetFps() const;
+ std::optional<int> GetKbps() const;
+ std::optional<int> GetKbpsRes() const;
// Optional settings.
int qp_low = 0;
@@ -123,16 +123,15 @@
uint32_t bitrate_bps) const;
// Gets the min framerate diff from `configs_` based on `pixels`.
- absl::optional<int> MinFpsDiff(int pixels) const;
+ std::optional<int> MinFpsDiff(int pixels) const;
// Gets QpThresholds for the codec `type` based on `pixels`.
- absl::optional<VideoEncoder::QpThresholds> GetQpThresholds(
- VideoCodecType type,
- int pixels) const;
+ std::optional<VideoEncoder::QpThresholds> GetQpThresholds(VideoCodecType type,
+ int pixels) const;
private:
- absl::optional<Config> GetMinFpsConfig(int pixels) const;
- absl::optional<Config> GetMaxFpsConfig(int pixels) const;
+ std::optional<Config> GetMinFpsConfig(int pixels) const;
+ std::optional<Config> GetMaxFpsConfig(int pixels) const;
Config GetConfig(int pixels) const;
std::vector<Config> configs_;
diff --git a/rtc_base/experiments/encoder_info_settings.cc b/rtc_base/experiments/encoder_info_settings.cc
index e9229df..82480ea 100644
--- a/rtc_base/experiments/encoder_info_settings.cc
+++ b/rtc_base/experiments/encoder_info_settings.cc
@@ -73,7 +73,7 @@
{1280 * 720, 900000, 30000, 2500000}};
}
-absl::optional<VideoEncoder::ResolutionBitrateLimits>
+std::optional<VideoEncoder::ResolutionBitrateLimits>
EncoderInfoSettings::GetDefaultSinglecastBitrateLimitsForResolution(
VideoCodecType codec_type,
int frame_size_pixels) {
@@ -101,13 +101,13 @@
// Through linear interpolation, return the bitrate limit corresponding to the
// specified |frame_size_pixels|.
-absl::optional<VideoEncoder::ResolutionBitrateLimits>
+std::optional<VideoEncoder::ResolutionBitrateLimits>
EncoderInfoSettings::GetSinglecastBitrateLimitForResolutionWhenQpIsUntrusted(
- absl::optional<int> frame_size_pixels,
+ std::optional<int> frame_size_pixels,
const std::vector<VideoEncoder::ResolutionBitrateLimits>&
resolution_bitrate_limits) {
if (!frame_size_pixels.has_value() || frame_size_pixels.value() <= 0) {
- return absl::nullopt;
+ return std::nullopt;
}
std::vector<VideoEncoder::ResolutionBitrateLimits> bitrate_limits =
@@ -121,7 +121,7 @@
});
if (bitrate_limits.empty()) {
- return absl::nullopt;
+ return std::nullopt;
}
int interpolation_index = -1;
@@ -171,7 +171,7 @@
<< " min_start_bitrate_bps = " << min_start_bitrate_bps
<< " min_bitrate_bps = " << kDefaultMinBitratebps
<< " max_bitrate_bps = " << max_bitrate_bps;
- return absl::nullopt;
+ return std::nullopt;
}
}
@@ -208,12 +208,12 @@
resolution_bitrate_limits_ = ToResolutionBitrateLimits(bitrate_limits.Get());
}
-absl::optional<uint32_t> EncoderInfoSettings::requested_resolution_alignment()
+std::optional<uint32_t> EncoderInfoSettings::requested_resolution_alignment()
const {
if (requested_resolution_alignment_ &&
requested_resolution_alignment_.Value() < 1) {
RTC_LOG(LS_WARNING) << "Unsupported alignment value, ignored.";
- return absl::nullopt;
+ return std::nullopt;
}
return requested_resolution_alignment_.GetOptional();
}
diff --git a/rtc_base/experiments/encoder_info_settings.h b/rtc_base/experiments/encoder_info_settings.h
index b200238..b534659 100644
--- a/rtc_base/experiments/encoder_info_settings.h
+++ b/rtc_base/experiments/encoder_info_settings.h
@@ -11,11 +11,11 @@
#ifndef RTC_BASE_EXPERIMENTS_ENCODER_INFO_SETTINGS_H_
#define RTC_BASE_EXPERIMENTS_ENCODER_INFO_SETTINGS_H_
+#include <optional>
#include <string>
#include <vector>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/field_trials_view.h"
#include "api/video_codecs/video_encoder.h"
#include "rtc_base/experiments/field_trial_parser.h"
@@ -34,7 +34,7 @@
int max_bitrate_bps = 0; // The maximum bitrate.
};
- absl::optional<uint32_t> requested_resolution_alignment() const;
+ std::optional<uint32_t> requested_resolution_alignment() const;
bool apply_alignment_to_all_simulcast_layers() const {
return apply_alignment_to_all_simulcast_layers_.Get();
}
@@ -46,16 +46,16 @@
static std::vector<VideoEncoder::ResolutionBitrateLimits>
GetDefaultSinglecastBitrateLimits(VideoCodecType codec_type);
- static absl::optional<VideoEncoder::ResolutionBitrateLimits>
+ static std::optional<VideoEncoder::ResolutionBitrateLimits>
GetDefaultSinglecastBitrateLimitsForResolution(VideoCodecType codec_type,
int frame_size_pixels);
static std::vector<VideoEncoder::ResolutionBitrateLimits>
GetDefaultSinglecastBitrateLimitsWhenQpIsUntrusted();
- static absl::optional<VideoEncoder::ResolutionBitrateLimits>
+ static std::optional<VideoEncoder::ResolutionBitrateLimits>
GetSinglecastBitrateLimitForResolutionWhenQpIsUntrusted(
- absl::optional<int> frame_size_pixels,
+ std::optional<int> frame_size_pixels,
const std::vector<VideoEncoder::ResolutionBitrateLimits>&
resolution_bitrate_limits);
diff --git a/rtc_base/experiments/encoder_info_settings_unittest.cc b/rtc_base/experiments/encoder_info_settings_unittest.cc
index 5739673..916f980 100644
--- a/rtc_base/experiments/encoder_info_settings_unittest.cc
+++ b/rtc_base/experiments/encoder_info_settings_unittest.cc
@@ -22,7 +22,7 @@
ExplicitKeyValueConfig field_trials("");
SimulcastEncoderAdapterEncoderInfoSettings settings(field_trials);
- EXPECT_EQ(absl::nullopt, settings.requested_resolution_alignment());
+ EXPECT_EQ(std::nullopt, settings.requested_resolution_alignment());
EXPECT_FALSE(settings.apply_alignment_to_all_simulcast_layers());
EXPECT_TRUE(settings.resolution_bitrate_limits().empty());
}
@@ -33,7 +33,7 @@
"requested_resolution_alignment:0/");
SimulcastEncoderAdapterEncoderInfoSettings settings(field_trials);
- EXPECT_EQ(absl::nullopt, settings.requested_resolution_alignment());
+ EXPECT_EQ(std::nullopt, settings.requested_resolution_alignment());
}
TEST(SimulcastEncoderAdapterSettingsTest, GetResolutionAlignment) {
@@ -68,7 +68,7 @@
"max_bitrate_bps:77000/");
SimulcastEncoderAdapterEncoderInfoSettings settings(field_trials);
- EXPECT_EQ(absl::nullopt, settings.requested_resolution_alignment());
+ EXPECT_EQ(std::nullopt, settings.requested_resolution_alignment());
EXPECT_FALSE(settings.apply_alignment_to_all_simulcast_layers());
EXPECT_THAT(settings.resolution_bitrate_limits(),
::testing::ElementsAre(VideoEncoder::ResolutionBitrateLimits{
diff --git a/rtc_base/experiments/field_trial_list.cc b/rtc_base/experiments/field_trial_list.cc
index 72cd79f..26c17b5 100644
--- a/rtc_base/experiments/field_trial_list.cc
+++ b/rtc_base/experiments/field_trial_list.cc
@@ -35,7 +35,7 @@
return GetList()->Used();
}
-bool FieldTrialStructListBase::Parse(absl::optional<std::string> str_value) {
+bool FieldTrialStructListBase::Parse(std::optional<std::string> str_value) {
RTC_DCHECK_NOTREACHED();
return true;
}
diff --git a/rtc_base/experiments/field_trial_list.h b/rtc_base/experiments/field_trial_list.h
index 63403cc..0c90f94 100644
--- a/rtc_base/experiments/field_trial_list.h
+++ b/rtc_base/experiments/field_trial_list.h
@@ -65,7 +65,7 @@
const std::vector<T>* operator->() const { return &values_; }
protected:
- bool Parse(absl::optional<std::string> str_value) override {
+ bool Parse(std::optional<std::string> str_value) override {
parse_got_called_ = true;
if (!str_value) {
@@ -76,7 +76,7 @@
std::vector<T> new_values_;
for (const absl::string_view token : rtc::split(str_value.value(), '|')) {
- absl::optional<T> value = ParseTypedParameter<T>(token);
+ std::optional<T> value = ParseTypedParameter<T>(token);
if (value) {
new_values_.push_back(*value);
} else {
@@ -180,7 +180,7 @@
// user-supplied values, we return -1.
int ValidateAndGetLength();
- bool Parse(absl::optional<std::string> str_value) override;
+ bool Parse(std::optional<std::string> str_value) override;
std::vector<std::unique_ptr<FieldTrialListWrapper>> sub_lists_;
};
diff --git a/rtc_base/experiments/field_trial_parser.cc b/rtc_base/experiments/field_trial_parser.cc
index 78d5489..d27fbba 100644
--- a/rtc_base/experiments/field_trial_parser.cc
+++ b/rtc_base/experiments/field_trial_parser.cc
@@ -60,7 +60,7 @@
while (!tail.empty()) {
size_t key_end = tail.find_first_of(",:");
absl::string_view key = tail.substr(0, key_end);
- absl::optional<std::string> opt_value;
+ std::optional<std::string> opt_value;
if (key_end == absl::string_view::npos) {
tail = "";
} else if (tail[key_end] == ':') {
@@ -112,17 +112,17 @@
}
template <>
-absl::optional<bool> ParseTypedParameter<bool>(absl::string_view str) {
+std::optional<bool> ParseTypedParameter<bool>(absl::string_view str) {
if (str == "true" || str == "1") {
return true;
} else if (str == "false" || str == "0") {
return false;
}
- return absl::nullopt;
+ return std::nullopt;
}
template <>
-absl::optional<double> ParseTypedParameter<double>(absl::string_view str) {
+std::optional<double> ParseTypedParameter<double>(absl::string_view str) {
double value;
char unit[2]{0, 0};
if (sscanf(std::string(str).c_str(), "%lf%1s", &value, unit) >= 1) {
@@ -130,56 +130,56 @@
return value / 100;
return value;
} else {
- return absl::nullopt;
+ return std::nullopt;
}
}
template <>
-absl::optional<int> ParseTypedParameter<int>(absl::string_view str) {
+std::optional<int> ParseTypedParameter<int>(absl::string_view str) {
int64_t value;
if (sscanf(std::string(str).c_str(), "%" SCNd64, &value) == 1) {
if (rtc::IsValueInRangeForNumericType<int, int64_t>(value)) {
return static_cast<int>(value);
}
}
- return absl::nullopt;
+ return std::nullopt;
}
template <>
-absl::optional<unsigned> ParseTypedParameter<unsigned>(absl::string_view str) {
+std::optional<unsigned> ParseTypedParameter<unsigned>(absl::string_view str) {
int64_t value;
if (sscanf(std::string(str).c_str(), "%" SCNd64, &value) == 1) {
if (rtc::IsValueInRangeForNumericType<unsigned, int64_t>(value)) {
return static_cast<unsigned>(value);
}
}
- return absl::nullopt;
+ return std::nullopt;
}
template <>
-absl::optional<std::string> ParseTypedParameter<std::string>(
+std::optional<std::string> ParseTypedParameter<std::string>(
absl::string_view str) {
return std::string(str);
}
template <>
-absl::optional<absl::optional<bool>> ParseTypedParameter<absl::optional<bool>>(
+std::optional<std::optional<bool>> ParseTypedParameter<std::optional<bool>>(
absl::string_view str) {
return ParseOptionalParameter<bool>(str);
}
template <>
-absl::optional<absl::optional<int>> ParseTypedParameter<absl::optional<int>>(
+std::optional<std::optional<int>> ParseTypedParameter<std::optional<int>>(
absl::string_view str) {
return ParseOptionalParameter<int>(str);
}
template <>
-absl::optional<absl::optional<unsigned>>
-ParseTypedParameter<absl::optional<unsigned>>(absl::string_view str) {
+std::optional<std::optional<unsigned>>
+ParseTypedParameter<std::optional<unsigned>>(absl::string_view str) {
return ParseOptionalParameter<unsigned>(str);
}
template <>
-absl::optional<absl::optional<double>>
-ParseTypedParameter<absl::optional<double>>(absl::string_view str) {
+std::optional<std::optional<double>> ParseTypedParameter<std::optional<double>>(
+ absl::string_view str) {
return ParseOptionalParameter<double>(str);
}
@@ -197,10 +197,10 @@
return value_;
}
-bool FieldTrialFlag::Parse(absl::optional<std::string> str_value) {
+bool FieldTrialFlag::Parse(std::optional<std::string> str_value) {
// Only set the flag if there is no argument provided.
if (str_value) {
- absl::optional<bool> opt_value = ParseTypedParameter<bool>(*str_value);
+ std::optional<bool> opt_value = ParseTypedParameter<bool>(*str_value);
if (!opt_value)
return false;
value_ = *opt_value;
@@ -224,14 +224,14 @@
default;
AbstractFieldTrialEnum::~AbstractFieldTrialEnum() = default;
-bool AbstractFieldTrialEnum::Parse(absl::optional<std::string> str_value) {
+bool AbstractFieldTrialEnum::Parse(std::optional<std::string> str_value) {
if (str_value) {
auto it = enum_mapping_.find(*str_value);
if (it != enum_mapping_.end()) {
value_ = it->second;
return true;
}
- absl::optional<int> value = ParseTypedParameter<int>(*str_value);
+ std::optional<int> value = ParseTypedParameter<int>(*str_value);
if (value.has_value() &&
(valid_values_.find(*value) != valid_values_.end())) {
value_ = *value;
diff --git a/rtc_base/experiments/field_trial_parser.h b/rtc_base/experiments/field_trial_parser.h
index 822895e..890da80 100644
--- a/rtc_base/experiments/field_trial_parser.h
+++ b/rtc_base/experiments/field_trial_parser.h
@@ -14,12 +14,12 @@
#include <initializer_list>
#include <map>
+#include <optional>
#include <set>
#include <string>
#include <vector>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
// Field trial parser functionality. Provides funcitonality to parse field trial
// argument strings in key:value format. Each parameter is described using
@@ -51,7 +51,7 @@
std::initializer_list<FieldTrialParameterInterface*> fields,
absl::string_view trial_string);
void MarkAsUsed() { used_ = true; }
- virtual bool Parse(absl::optional<std::string> str_value) = 0;
+ virtual bool Parse(std::optional<std::string> str_value) = 0;
virtual void ParseDone() {}
@@ -68,10 +68,10 @@
std::initializer_list<FieldTrialParameterInterface*> fields,
absl::string_view trial_string);
-// Specialize this in code file for custom types. Should return absl::nullopt if
+// Specialize this in code file for custom types. Should return std::nullopt if
// the given string cannot be properly parsed.
template <typename T>
-absl::optional<T> ParseTypedParameter(absl::string_view);
+std::optional<T> ParseTypedParameter(absl::string_view);
// This class uses the ParseTypedParameter function to implement a parameter
// implementation with an enforced default value.
@@ -87,9 +87,9 @@
void SetForTest(T value) { value_ = value; }
protected:
- bool Parse(absl::optional<std::string> str_value) override {
+ bool Parse(std::optional<std::string> str_value) override {
if (str_value) {
- absl::optional<T> value = ParseTypedParameter<T>(*str_value);
+ std::optional<T> value = ParseTypedParameter<T>(*str_value);
if (value.has_value()) {
value_ = value.value();
return true;
@@ -110,8 +110,8 @@
public:
FieldTrialConstrained(absl::string_view key,
T default_value,
- absl::optional<T> lower_limit,
- absl::optional<T> upper_limit)
+ std::optional<T> lower_limit,
+ std::optional<T> upper_limit)
: FieldTrialParameterInterface(key),
value_(default_value),
lower_limit_(lower_limit),
@@ -121,9 +121,9 @@
const T* operator->() const { return &value_; }
protected:
- bool Parse(absl::optional<std::string> str_value) override {
+ bool Parse(std::optional<std::string> str_value) override {
if (str_value) {
- absl::optional<T> value = ParseTypedParameter<T>(*str_value);
+ std::optional<T> value = ParseTypedParameter<T>(*str_value);
if (value && (!lower_limit_ || *value >= *lower_limit_) &&
(!upper_limit_ || *value <= *upper_limit_)) {
value_ = *value;
@@ -135,8 +135,8 @@
private:
T value_;
- absl::optional<T> lower_limit_;
- absl::optional<T> upper_limit_;
+ std::optional<T> lower_limit_;
+ std::optional<T> upper_limit_;
};
class AbstractFieldTrialEnum : public FieldTrialParameterInterface {
@@ -148,7 +148,7 @@
AbstractFieldTrialEnum(const AbstractFieldTrialEnum&);
protected:
- bool Parse(absl::optional<std::string> str_value) override;
+ bool Parse(std::optional<std::string> str_value) override;
protected:
int value_;
@@ -181,35 +181,35 @@
};
// This class uses the ParseTypedParameter function to implement an optional
-// parameter implementation that can default to absl::nullopt.
+// parameter implementation that can default to std::nullopt.
template <typename T>
class FieldTrialOptional : public FieldTrialParameterInterface {
public:
explicit FieldTrialOptional(absl::string_view key)
: FieldTrialParameterInterface(key) {}
- FieldTrialOptional(absl::string_view key, absl::optional<T> default_value)
+ FieldTrialOptional(absl::string_view key, std::optional<T> default_value)
: FieldTrialParameterInterface(key), value_(default_value) {}
- absl::optional<T> GetOptional() const { return value_; }
+ std::optional<T> GetOptional() const { return value_; }
const T& Value() const { return value_.value(); }
const T& operator*() const { return value_.value(); }
const T* operator->() const { return &value_.value(); }
explicit operator bool() const { return value_.has_value(); }
protected:
- bool Parse(absl::optional<std::string> str_value) override {
+ bool Parse(std::optional<std::string> str_value) override {
if (str_value) {
- absl::optional<T> value = ParseTypedParameter<T>(*str_value);
+ std::optional<T> value = ParseTypedParameter<T>(*str_value);
if (!value.has_value())
return false;
value_ = value.value();
} else {
- value_ = absl::nullopt;
+ value_ = std::nullopt;
}
return true;
}
private:
- absl::optional<T> value_;
+ std::optional<T> value_;
};
// Equivalent to a FieldTrialParameter<bool> in the case that both key and value
@@ -223,47 +223,46 @@
explicit operator bool() const;
protected:
- bool Parse(absl::optional<std::string> str_value) override;
+ bool Parse(std::optional<std::string> str_value) override;
private:
bool value_;
};
template <typename T>
-absl::optional<absl::optional<T>> ParseOptionalParameter(
- absl::string_view str) {
+std::optional<std::optional<T>> ParseOptionalParameter(absl::string_view str) {
if (str.empty())
- return absl::optional<T>();
+ return std::optional<T>();
auto parsed = ParseTypedParameter<T>(str);
if (parsed.has_value())
return parsed;
- return absl::nullopt;
+ return std::nullopt;
}
template <>
-absl::optional<bool> ParseTypedParameter<bool>(absl::string_view str);
+std::optional<bool> ParseTypedParameter<bool>(absl::string_view str);
template <>
-absl::optional<double> ParseTypedParameter<double>(absl::string_view str);
+std::optional<double> ParseTypedParameter<double>(absl::string_view str);
template <>
-absl::optional<int> ParseTypedParameter<int>(absl::string_view str);
+std::optional<int> ParseTypedParameter<int>(absl::string_view str);
template <>
-absl::optional<unsigned> ParseTypedParameter<unsigned>(absl::string_view str);
+std::optional<unsigned> ParseTypedParameter<unsigned>(absl::string_view str);
template <>
-absl::optional<std::string> ParseTypedParameter<std::string>(
+std::optional<std::string> ParseTypedParameter<std::string>(
absl::string_view str);
template <>
-absl::optional<absl::optional<bool>> ParseTypedParameter<absl::optional<bool>>(
+std::optional<std::optional<bool>> ParseTypedParameter<std::optional<bool>>(
absl::string_view str);
template <>
-absl::optional<absl::optional<int>> ParseTypedParameter<absl::optional<int>>(
+std::optional<std::optional<int>> ParseTypedParameter<std::optional<int>>(
absl::string_view str);
template <>
-absl::optional<absl::optional<unsigned>>
-ParseTypedParameter<absl::optional<unsigned>>(absl::string_view str);
+std::optional<std::optional<unsigned>>
+ParseTypedParameter<std::optional<unsigned>>(absl::string_view str);
template <>
-absl::optional<absl::optional<double>>
-ParseTypedParameter<absl::optional<double>>(absl::string_view str);
+std::optional<std::optional<double>> ParseTypedParameter<std::optional<double>>(
+ absl::string_view str);
// Accepts true, false, else parsed with sscanf %i, true if != 0.
extern template class FieldTrialParameter<bool>;
diff --git a/rtc_base/experiments/field_trial_parser_unittest.cc b/rtc_base/experiments/field_trial_parser_unittest.cc
index 73d1153..9968350 100644
--- a/rtc_base/experiments/field_trial_parser_unittest.cc
+++ b/rtc_base/experiments/field_trial_parser_unittest.cc
@@ -116,8 +116,8 @@
EXPECT_EQ(exp.hash.Get(), "a80");
}
TEST(FieldTrialParserTest, IgnoresOutOfRange) {
- FieldTrialConstrained<double> low("low", 10, absl::nullopt, 100);
- FieldTrialConstrained<double> high("high", 10, 5, absl::nullopt);
+ FieldTrialConstrained<double> low("low", 10, std::nullopt, 100);
+ FieldTrialConstrained<double> high("high", 10, 5, std::nullopt);
ParseFieldTrial({&low, &high}, "low:1000,high:0");
EXPECT_EQ(low.Get(), 10);
EXPECT_EQ(high.Get(), 10);
@@ -141,7 +141,7 @@
EXPECT_EQ(req.Get(), 30);
}
TEST(FieldTrialParserTest, ParsesOptionalParameters) {
- FieldTrialOptional<int> max_count("c", absl::nullopt);
+ FieldTrialOptional<int> max_count("c", std::nullopt);
ParseFieldTrial({&max_count}, "");
EXPECT_FALSE(max_count.GetOptional().has_value());
ParseFieldTrial({&max_count}, "c:10");
@@ -153,7 +153,7 @@
ParseFieldTrial({&max_count}, "c:");
EXPECT_EQ(max_count.GetOptional().value(), 20);
- FieldTrialOptional<unsigned> max_size("c", absl::nullopt);
+ FieldTrialOptional<unsigned> max_size("c", std::nullopt);
ParseFieldTrial({&max_size}, "");
EXPECT_FALSE(max_size.GetOptional().has_value());
ParseFieldTrial({&max_size}, "c:10");
diff --git a/rtc_base/experiments/field_trial_units.cc b/rtc_base/experiments/field_trial_units.cc
index 92af46a..ddb954a 100644
--- a/rtc_base/experiments/field_trial_units.cc
+++ b/rtc_base/experiments/field_trial_units.cc
@@ -12,10 +12,10 @@
#include <stdio.h>
#include <limits>
+#include <optional>
#include <string>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
// Large enough to fit "seconds", the longest supported unit name.
#define RTC_TRIAL_UNIT_LENGTH_STR "7"
@@ -29,7 +29,7 @@
std::string unit;
};
-absl::optional<ValueWithUnit> ParseValueWithUnit(absl::string_view str) {
+std::optional<ValueWithUnit> ParseValueWithUnit(absl::string_view str) {
if (str == "inf") {
return ValueWithUnit{std::numeric_limits<double>::infinity(), ""};
} else if (str == "-inf") {
@@ -43,13 +43,13 @@
return ValueWithUnit{double_val, unit_char};
}
}
- return absl::nullopt;
+ return std::nullopt;
}
} // namespace
template <>
-absl::optional<DataRate> ParseTypedParameter<DataRate>(absl::string_view str) {
- absl::optional<ValueWithUnit> result = ParseValueWithUnit(str);
+std::optional<DataRate> ParseTypedParameter<DataRate>(absl::string_view str) {
+ std::optional<ValueWithUnit> result = ParseValueWithUnit(str);
if (result) {
if (result->unit.empty() || result->unit == "kbps") {
return DataRate::KilobitsPerSec(result->value);
@@ -57,23 +57,22 @@
return DataRate::BitsPerSec(result->value);
}
}
- return absl::nullopt;
+ return std::nullopt;
}
template <>
-absl::optional<DataSize> ParseTypedParameter<DataSize>(absl::string_view str) {
- absl::optional<ValueWithUnit> result = ParseValueWithUnit(str);
+std::optional<DataSize> ParseTypedParameter<DataSize>(absl::string_view str) {
+ std::optional<ValueWithUnit> result = ParseValueWithUnit(str);
if (result) {
if (result->unit.empty() || result->unit == "bytes")
return DataSize::Bytes(result->value);
}
- return absl::nullopt;
+ return std::nullopt;
}
template <>
-absl::optional<TimeDelta> ParseTypedParameter<TimeDelta>(
- absl::string_view str) {
- absl::optional<ValueWithUnit> result = ParseValueWithUnit(str);
+std::optional<TimeDelta> ParseTypedParameter<TimeDelta>(absl::string_view str) {
+ std::optional<ValueWithUnit> result = ParseValueWithUnit(str);
if (result) {
if (result->unit == "s" || result->unit == "seconds") {
return TimeDelta::Seconds(result->value);
@@ -83,22 +82,22 @@
return TimeDelta::Millis(result->value);
}
}
- return absl::nullopt;
+ return std::nullopt;
}
template <>
-absl::optional<absl::optional<DataRate>>
-ParseTypedParameter<absl::optional<DataRate>>(absl::string_view str) {
+std::optional<std::optional<DataRate>>
+ParseTypedParameter<std::optional<DataRate>>(absl::string_view str) {
return ParseOptionalParameter<DataRate>(str);
}
template <>
-absl::optional<absl::optional<DataSize>>
-ParseTypedParameter<absl::optional<DataSize>>(absl::string_view str) {
+std::optional<std::optional<DataSize>>
+ParseTypedParameter<std::optional<DataSize>>(absl::string_view str) {
return ParseOptionalParameter<DataSize>(str);
}
template <>
-absl::optional<absl::optional<TimeDelta>>
-ParseTypedParameter<absl::optional<TimeDelta>>(absl::string_view str) {
+std::optional<std::optional<TimeDelta>>
+ParseTypedParameter<std::optional<TimeDelta>>(absl::string_view str) {
return ParseOptionalParameter<TimeDelta>(str);
}
diff --git a/rtc_base/experiments/field_trial_units.h b/rtc_base/experiments/field_trial_units.h
index 408367c..0bab5fe 100644
--- a/rtc_base/experiments/field_trial_units.h
+++ b/rtc_base/experiments/field_trial_units.h
@@ -19,11 +19,11 @@
namespace webrtc {
template <>
-absl::optional<DataRate> ParseTypedParameter<DataRate>(absl::string_view str);
+std::optional<DataRate> ParseTypedParameter<DataRate>(absl::string_view str);
template <>
-absl::optional<DataSize> ParseTypedParameter<DataSize>(absl::string_view str);
+std::optional<DataSize> ParseTypedParameter<DataSize>(absl::string_view str);
template <>
-absl::optional<TimeDelta> ParseTypedParameter<TimeDelta>(absl::string_view str);
+std::optional<TimeDelta> ParseTypedParameter<TimeDelta>(absl::string_view str);
extern template class FieldTrialParameter<DataRate>;
extern template class FieldTrialParameter<DataSize>;
diff --git a/rtc_base/experiments/field_trial_units_unittest.cc b/rtc_base/experiments/field_trial_units_unittest.cc
index 8996663..82ef580 100644
--- a/rtc_base/experiments/field_trial_units_unittest.cc
+++ b/rtc_base/experiments/field_trial_units_unittest.cc
@@ -9,10 +9,10 @@
*/
#include "rtc_base/experiments/field_trial_units.h"
+#include <optional>
#include <string>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "rtc_base/experiments/field_trial_parser.h"
#include "test/gtest.h"
@@ -24,7 +24,7 @@
FieldTrialParameter<TimeDelta> period =
FieldTrialParameter<TimeDelta>("p", TimeDelta::Millis(100));
FieldTrialOptional<DataSize> max_buffer =
- FieldTrialOptional<DataSize>("b", absl::nullopt);
+ FieldTrialOptional<DataSize>("b", std::nullopt);
explicit DummyExperiment(absl::string_view field_trial) {
ParseFieldTrial({&target_rate, &max_buffer, &period}, field_trial);
diff --git a/rtc_base/experiments/keyframe_interval_settings.cc b/rtc_base/experiments/keyframe_interval_settings.cc
index df31d29..760870b 100644
--- a/rtc_base/experiments/keyframe_interval_settings.cc
+++ b/rtc_base/experiments/keyframe_interval_settings.cc
@@ -27,8 +27,7 @@
key_value_config.Lookup(kFieldTrialName));
}
-absl::optional<int> KeyframeIntervalSettings::MinKeyframeSendIntervalMs()
- const {
+std::optional<int> KeyframeIntervalSettings::MinKeyframeSendIntervalMs() const {
return min_keyframe_send_interval_ms_.GetOptional();
}
} // namespace webrtc
diff --git a/rtc_base/experiments/keyframe_interval_settings.h b/rtc_base/experiments/keyframe_interval_settings.h
index df1f679..a9c1971 100644
--- a/rtc_base/experiments/keyframe_interval_settings.h
+++ b/rtc_base/experiments/keyframe_interval_settings.h
@@ -11,7 +11,8 @@
#ifndef RTC_BASE_EXPERIMENTS_KEYFRAME_INTERVAL_SETTINGS_H_
#define RTC_BASE_EXPERIMENTS_KEYFRAME_INTERVAL_SETTINGS_H_
-#include "absl/types/optional.h"
+#include <optional>
+
#include "api/field_trials_view.h"
#include "rtc_base/experiments/field_trial_parser.h"
@@ -26,7 +27,7 @@
// Sender side.
// The encoded keyframe send rate is <= 1/MinKeyframeSendIntervalMs().
- absl::optional<int> MinKeyframeSendIntervalMs() const;
+ std::optional<int> MinKeyframeSendIntervalMs() const;
private:
FieldTrialOptional<int> min_keyframe_send_interval_ms_;
diff --git a/rtc_base/experiments/min_video_bitrate_experiment.cc b/rtc_base/experiments/min_video_bitrate_experiment.cc
index 179d4db..dd216b8 100644
--- a/rtc_base/experiments/min_video_bitrate_experiment.cc
+++ b/rtc_base/experiments/min_video_bitrate_experiment.cc
@@ -26,20 +26,20 @@
"WebRTC-VP8-Forced-Fallback-Encoder-v2";
const char kMinVideoBitrateExperiment[] = "WebRTC-Video-MinVideoBitrate";
-absl::optional<int> GetFallbackMinBpsFromFieldTrial(
+std::optional<int> GetFallbackMinBpsFromFieldTrial(
const FieldTrialsView& field_trials,
VideoCodecType type) {
if (type != kVideoCodecVP8) {
- return absl::nullopt;
+ return std::nullopt;
}
if (!field_trials.IsEnabled(kForcedFallbackFieldTrial)) {
- return absl::nullopt;
+ return std::nullopt;
}
const std::string group = field_trials.Lookup(kForcedFallbackFieldTrial);
if (group.empty()) {
- return absl::nullopt;
+ return std::nullopt;
}
int min_pixels; // Ignored.
@@ -47,21 +47,21 @@
int min_bps;
if (sscanf(group.c_str(), "Enabled-%d,%d,%d", &min_pixels, &max_pixels,
&min_bps) != 3) {
- return absl::nullopt;
+ return std::nullopt;
}
if (min_bps <= 0) {
- return absl::nullopt;
+ return std::nullopt;
}
return min_bps;
}
} // namespace
-absl::optional<DataRate> GetExperimentalMinVideoBitrate(
+std::optional<DataRate> GetExperimentalMinVideoBitrate(
const FieldTrialsView& field_trials,
VideoCodecType type) {
- const absl::optional<int> fallback_min_bitrate_bps =
+ const std::optional<int> fallback_min_bitrate_bps =
GetFallbackMinBpsFromFieldTrial(field_trials, type);
if (fallback_min_bitrate_bps) {
return DataRate::BitsPerSec(*fallback_min_bitrate_bps);
@@ -106,13 +106,13 @@
case kVideoCodecH264:
return min_bitrate_h264.GetOptional();
case kVideoCodecGeneric:
- return absl::nullopt;
+ return std::nullopt;
}
RTC_DCHECK_NOTREACHED();
}
- return absl::nullopt;
+ return std::nullopt;
}
} // namespace webrtc
diff --git a/rtc_base/experiments/min_video_bitrate_experiment.h b/rtc_base/experiments/min_video_bitrate_experiment.h
index ac1335f..af6acb3 100644
--- a/rtc_base/experiments/min_video_bitrate_experiment.h
+++ b/rtc_base/experiments/min_video_bitrate_experiment.h
@@ -11,7 +11,8 @@
#ifndef RTC_BASE_EXPERIMENTS_MIN_VIDEO_BITRATE_EXPERIMENT_H_
#define RTC_BASE_EXPERIMENTS_MIN_VIDEO_BITRATE_EXPERIMENT_H_
-#include "absl/types/optional.h"
+#include <optional>
+
#include "api/field_trials_view.h"
#include "api/units/data_rate.h"
#include "api/video/video_codec_type.h"
@@ -22,7 +23,7 @@
// Return the experiment-driven minimum video bitrate.
// If no experiment is effective, returns nullopt.
-absl::optional<DataRate> GetExperimentalMinVideoBitrate(
+std::optional<DataRate> GetExperimentalMinVideoBitrate(
const FieldTrialsView& field_trials,
VideoCodecType type);
diff --git a/rtc_base/experiments/min_video_bitrate_experiment_unittest.cc b/rtc_base/experiments/min_video_bitrate_experiment_unittest.cc
index b8a5809..182669d 100644
--- a/rtc_base/experiments/min_video_bitrate_experiment_unittest.cc
+++ b/rtc_base/experiments/min_video_bitrate_experiment_unittest.cc
@@ -10,7 +10,8 @@
#include "rtc_base/experiments/min_video_bitrate_experiment.h"
-#include "absl/types/optional.h"
+#include <optional>
+
#include "api/units/data_rate.h"
#include "api/video/video_codec_type.h"
#include "test/explicit_key_value_config.h"
@@ -26,13 +27,13 @@
ExplicitKeyValueConfig field_trials("");
EXPECT_EQ(GetExperimentalMinVideoBitrate(field_trials, kVideoCodecGeneric),
- absl::nullopt);
+ std::nullopt);
EXPECT_EQ(GetExperimentalMinVideoBitrate(field_trials, kVideoCodecVP8),
- absl::nullopt);
+ std::nullopt);
EXPECT_EQ(GetExperimentalMinVideoBitrate(field_trials, kVideoCodecVP9),
- absl::nullopt);
+ std::nullopt);
EXPECT_EQ(GetExperimentalMinVideoBitrate(field_trials, kVideoCodecH264),
- absl::nullopt);
+ std::nullopt);
}
TEST(GetExperimentalMinVideoBitrateTest,
@@ -41,13 +42,13 @@
"WebRTC-Video-MinVideoBitrate/Disabled,br:123kbps/");
EXPECT_EQ(GetExperimentalMinVideoBitrate(field_trials, kVideoCodecGeneric),
- absl::nullopt);
+ std::nullopt);
EXPECT_EQ(GetExperimentalMinVideoBitrate(field_trials, kVideoCodecVP8),
- absl::nullopt);
+ std::nullopt);
EXPECT_EQ(GetExperimentalMinVideoBitrate(field_trials, kVideoCodecVP9),
- absl::nullopt);
+ std::nullopt);
EXPECT_EQ(GetExperimentalMinVideoBitrate(field_trials, kVideoCodecH264),
- absl::nullopt);
+ std::nullopt);
}
TEST(GetExperimentalMinVideoBitrateTest, BrForAllCodecsIfDefined) {
@@ -86,13 +87,13 @@
"Disabled,vp8_br:100kbps,vp9_br:200kbps,h264_br:300kbps/");
EXPECT_EQ(GetExperimentalMinVideoBitrate(field_trials, kVideoCodecGeneric),
- absl::nullopt);
+ std::nullopt);
EXPECT_EQ(GetExperimentalMinVideoBitrate(field_trials, kVideoCodecVP8),
- absl::nullopt);
+ std::nullopt);
EXPECT_EQ(GetExperimentalMinVideoBitrate(field_trials, kVideoCodecVP9),
- absl::nullopt);
+ std::nullopt);
EXPECT_EQ(GetExperimentalMinVideoBitrate(field_trials, kVideoCodecH264),
- absl::nullopt);
+ std::nullopt);
}
TEST(GetExperimentalMinVideoBitrateTest, SpecificCodecConfigsUsedIfExpEnabled) {
@@ -101,7 +102,7 @@
"Enabled,vp8_br:100kbps,vp9_br:200kbps,h264_br:300kbps/");
EXPECT_EQ(GetExperimentalMinVideoBitrate(field_trials, kVideoCodecGeneric),
- absl::nullopt);
+ std::nullopt);
EXPECT_EQ(GetExperimentalMinVideoBitrate(field_trials, kVideoCodecVP8),
DataRate::KilobitsPerSec(100));
EXPECT_EQ(GetExperimentalMinVideoBitrate(field_trials, kVideoCodecVP9),
@@ -131,7 +132,7 @@
"Enabled-444444,555555,666666/");
EXPECT_EQ(GetExperimentalMinVideoBitrate(field_trials, kVideoCodecGeneric),
- absl::nullopt);
+ std::nullopt);
EXPECT_EQ(GetExperimentalMinVideoBitrate(field_trials, kVideoCodecVP9),
DataRate::KilobitsPerSec(200));
EXPECT_EQ(GetExperimentalMinVideoBitrate(field_trials, kVideoCodecH264),
diff --git a/rtc_base/experiments/normalize_simulcast_size_experiment.cc b/rtc_base/experiments/normalize_simulcast_size_experiment.cc
index 6fe881a..0e54357 100644
--- a/rtc_base/experiments/normalize_simulcast_size_experiment.cc
+++ b/rtc_base/experiments/normalize_simulcast_size_experiment.cc
@@ -24,27 +24,27 @@
constexpr int kMaxSetting = 5;
} // namespace
-absl::optional<int> NormalizeSimulcastSizeExperiment::GetBase2Exponent(
+std::optional<int> NormalizeSimulcastSizeExperiment::GetBase2Exponent(
const FieldTrialsView& field_trials) {
if (!field_trials.IsEnabled(kFieldTrial))
- return absl::nullopt;
+ return std::nullopt;
const std::string group = field_trials.Lookup(kFieldTrial);
if (group.empty())
- return absl::nullopt;
+ return std::nullopt;
int exponent;
if (sscanf(group.c_str(), "Enabled-%d", &exponent) != 1) {
RTC_LOG(LS_WARNING) << "No parameter provided.";
- return absl::nullopt;
+ return std::nullopt;
}
if (exponent < kMinSetting || exponent > kMaxSetting) {
RTC_LOG(LS_WARNING) << "Unsupported exp value provided, value ignored.";
- return absl::nullopt;
+ return std::nullopt;
}
- return absl::optional<int>(exponent);
+ return std::optional<int>(exponent);
}
} // namespace webrtc
diff --git a/rtc_base/experiments/normalize_simulcast_size_experiment.h b/rtc_base/experiments/normalize_simulcast_size_experiment.h
index 3c187f9..79eada1 100644
--- a/rtc_base/experiments/normalize_simulcast_size_experiment.h
+++ b/rtc_base/experiments/normalize_simulcast_size_experiment.h
@@ -11,14 +11,15 @@
#ifndef RTC_BASE_EXPERIMENTS_NORMALIZE_SIMULCAST_SIZE_EXPERIMENT_H_
#define RTC_BASE_EXPERIMENTS_NORMALIZE_SIMULCAST_SIZE_EXPERIMENT_H_
-#include "absl/types/optional.h"
+#include <optional>
+
#include "api/field_trials_view.h"
namespace webrtc {
class NormalizeSimulcastSizeExperiment {
public:
// Returns the base two exponent from field trial.
- static absl::optional<int> GetBase2Exponent(
+ static std::optional<int> GetBase2Exponent(
const FieldTrialsView& field_trials);
};
diff --git a/rtc_base/experiments/quality_scaler_settings.cc b/rtc_base/experiments/quality_scaler_settings.cc
index 24da211..0193a69 100644
--- a/rtc_base/experiments/quality_scaler_settings.cc
+++ b/rtc_base/experiments/quality_scaler_settings.cc
@@ -34,61 +34,61 @@
field_trials.Lookup("WebRTC-Video-QualityScalerSettings"));
}
-absl::optional<int> QualityScalerSettings::SamplingPeriodMs() const {
+std::optional<int> QualityScalerSettings::SamplingPeriodMs() const {
if (sampling_period_ms_ && sampling_period_ms_.Value() <= 0) {
RTC_LOG(LS_WARNING) << "Unsupported sampling_period_ms value, ignored.";
- return absl::nullopt;
+ return std::nullopt;
}
return sampling_period_ms_.GetOptional();
}
-absl::optional<int> QualityScalerSettings::AverageQpWindow() const {
+std::optional<int> QualityScalerSettings::AverageQpWindow() const {
if (average_qp_window_ && average_qp_window_.Value() <= 0) {
RTC_LOG(LS_WARNING) << "Unsupported average_qp_window value, ignored.";
- return absl::nullopt;
+ return std::nullopt;
}
return average_qp_window_.GetOptional();
}
-absl::optional<int> QualityScalerSettings::MinFrames() const {
+std::optional<int> QualityScalerSettings::MinFrames() const {
if (min_frames_ && min_frames_.Value() < kMinFrames) {
RTC_LOG(LS_WARNING) << "Unsupported min_frames value, ignored.";
- return absl::nullopt;
+ return std::nullopt;
}
return min_frames_.GetOptional();
}
-absl::optional<double> QualityScalerSettings::InitialScaleFactor() const {
+std::optional<double> QualityScalerSettings::InitialScaleFactor() const {
if (initial_scale_factor_ &&
initial_scale_factor_.Value() < kMinScaleFactor) {
RTC_LOG(LS_WARNING) << "Unsupported initial_scale_factor value, ignored.";
- return absl::nullopt;
+ return std::nullopt;
}
return initial_scale_factor_.GetOptional();
}
-absl::optional<double> QualityScalerSettings::ScaleFactor() const {
+std::optional<double> QualityScalerSettings::ScaleFactor() const {
if (scale_factor_ && scale_factor_.Value() < kMinScaleFactor) {
RTC_LOG(LS_WARNING) << "Unsupported scale_factor value, ignored.";
- return absl::nullopt;
+ return std::nullopt;
}
return scale_factor_.GetOptional();
}
-absl::optional<int> QualityScalerSettings::InitialBitrateIntervalMs() const {
+std::optional<int> QualityScalerSettings::InitialBitrateIntervalMs() const {
if (initial_bitrate_interval_ms_ &&
initial_bitrate_interval_ms_.Value() < 0) {
RTC_LOG(LS_WARNING) << "Unsupported bitrate_interval value, ignored.";
- return absl::nullopt;
+ return std::nullopt;
}
return initial_bitrate_interval_ms_.GetOptional();
}
-absl::optional<double> QualityScalerSettings::InitialBitrateFactor() const {
+std::optional<double> QualityScalerSettings::InitialBitrateFactor() const {
if (initial_bitrate_factor_ &&
initial_bitrate_factor_.Value() < kMinScaleFactor) {
RTC_LOG(LS_WARNING) << "Unsupported initial_bitrate_factor value, ignored.";
- return absl::nullopt;
+ return std::nullopt;
}
return initial_bitrate_factor_.GetOptional();
}
diff --git a/rtc_base/experiments/quality_scaler_settings.h b/rtc_base/experiments/quality_scaler_settings.h
index 1085816..428f18f 100644
--- a/rtc_base/experiments/quality_scaler_settings.h
+++ b/rtc_base/experiments/quality_scaler_settings.h
@@ -11,7 +11,8 @@
#ifndef RTC_BASE_EXPERIMENTS_QUALITY_SCALER_SETTINGS_H_
#define RTC_BASE_EXPERIMENTS_QUALITY_SCALER_SETTINGS_H_
-#include "absl/types/optional.h"
+#include <optional>
+
#include "api/field_trials_view.h"
#include "rtc_base/experiments/field_trial_parser.h"
@@ -21,13 +22,13 @@
public:
explicit QualityScalerSettings(const FieldTrialsView& field_trials);
- absl::optional<int> SamplingPeriodMs() const;
- absl::optional<int> AverageQpWindow() const;
- absl::optional<int> MinFrames() const;
- absl::optional<double> InitialScaleFactor() const;
- absl::optional<double> ScaleFactor() const;
- absl::optional<int> InitialBitrateIntervalMs() const;
- absl::optional<double> InitialBitrateFactor() const;
+ std::optional<int> SamplingPeriodMs() const;
+ std::optional<int> AverageQpWindow() const;
+ std::optional<int> MinFrames() const;
+ std::optional<double> InitialScaleFactor() const;
+ std::optional<double> ScaleFactor() const;
+ std::optional<int> InitialBitrateIntervalMs() const;
+ std::optional<double> InitialBitrateFactor() const;
private:
FieldTrialOptional<int> sampling_period_ms_;
diff --git a/rtc_base/experiments/quality_scaling_experiment.cc b/rtc_base/experiments/quality_scaling_experiment.cc
index ee3d7c0..a5b909e 100644
--- a/rtc_base/experiments/quality_scaling_experiment.cc
+++ b/rtc_base/experiments/quality_scaling_experiment.cc
@@ -32,14 +32,14 @@
"Enabled-29,95,149,205,24,37,26,36,0.9995,0.9999,1";
#endif
-absl::optional<VideoEncoder::QpThresholds> GetThresholds(int low,
- int high,
- int max) {
+std::optional<VideoEncoder::QpThresholds> GetThresholds(int low,
+ int high,
+ int max) {
if (low < kMinQp || high > max || high < low)
- return absl::nullopt;
+ return std::nullopt;
RTC_LOG(LS_INFO) << "QP thresholds: low: " << low << ", high: " << high;
- return absl::optional<VideoEncoder::QpThresholds>(
+ return std::optional<VideoEncoder::QpThresholds>(
VideoEncoder::QpThresholds(low, high));
}
} // namespace
@@ -52,7 +52,7 @@
#endif
}
-absl::optional<QualityScalingExperiment::Settings>
+std::optional<QualityScalingExperiment::Settings>
QualityScalingExperiment::ParseSettings(const FieldTrialsView& field_trials) {
std::string group = field_trials.Lookup(kFieldTrial);
// TODO(http://crbug.com/webrtc/12401): Completely remove the experiment code
@@ -67,17 +67,17 @@
&s.h264_high, &s.generic_low, &s.generic_high, &s.alpha_high,
&s.alpha_low, &s.drop) != 11) {
RTC_LOG(LS_WARNING) << "Invalid number of parameters provided.";
- return absl::nullopt;
+ return std::nullopt;
}
return s;
}
-absl::optional<VideoEncoder::QpThresholds>
+std::optional<VideoEncoder::QpThresholds>
QualityScalingExperiment::GetQpThresholds(VideoCodecType codec_type,
const FieldTrialsView& field_trials) {
const auto settings = ParseSettings(field_trials);
if (!settings)
- return absl::nullopt;
+ return std::nullopt;
switch (codec_type) {
case kVideoCodecVP8:
@@ -92,7 +92,7 @@
return GetThresholds(settings->generic_low, settings->generic_high,
kMaxGenericQp);
default:
- return absl::nullopt;
+ return std::nullopt;
}
}
diff --git a/rtc_base/experiments/quality_scaling_experiment.h b/rtc_base/experiments/quality_scaling_experiment.h
index bd24c06..0d197c1 100644
--- a/rtc_base/experiments/quality_scaling_experiment.h
+++ b/rtc_base/experiments/quality_scaling_experiment.h
@@ -10,7 +10,8 @@
#ifndef RTC_BASE_EXPERIMENTS_QUALITY_SCALING_EXPERIMENT_H_
#define RTC_BASE_EXPERIMENTS_QUALITY_SCALING_EXPERIMENT_H_
-#include "absl/types/optional.h"
+#include <optional>
+
#include "api/field_trials_view.h"
#include "api/video_codecs/video_encoder.h"
@@ -44,11 +45,11 @@
static bool Enabled(const FieldTrialsView& field_trials);
// Returns settings from field trial.
- static absl::optional<Settings> ParseSettings(
+ static std::optional<Settings> ParseSettings(
const FieldTrialsView& field_trials);
// Returns QpThresholds for the `codec_type`.
- static absl::optional<VideoEncoder::QpThresholds> GetQpThresholds(
+ static std::optional<VideoEncoder::QpThresholds> GetQpThresholds(
VideoCodecType codec_type,
const FieldTrialsView& field_trials);
diff --git a/rtc_base/experiments/rate_control_settings.cc b/rtc_base/experiments/rate_control_settings.cc
index 597bc73..2aaf0e1 100644
--- a/rtc_base/experiments/rate_control_settings.cc
+++ b/rtc_base/experiments/rate_control_settings.cc
@@ -110,12 +110,12 @@
kDefaultMinPushbackTargetBitrateBps);
}
-absl::optional<DataSize>
-RateControlSettings::CongestionWindowInitialDataWindow() const {
+std::optional<DataSize> RateControlSettings::CongestionWindowInitialDataWindow()
+ const {
return congestion_window_config_.initial_data_window;
}
-absl::optional<double> RateControlSettings::GetPacingFactor() const {
+std::optional<double> RateControlSettings::GetPacingFactor() const {
return video_config_.pacing_factor;
}
@@ -123,18 +123,18 @@
return video_config_.alr_probing;
}
-absl::optional<int> RateControlSettings::LibvpxVp8QpMax() const {
+std::optional<int> RateControlSettings::LibvpxVp8QpMax() const {
if (video_config_.vp8_qp_max &&
(*video_config_.vp8_qp_max < 0 || *video_config_.vp8_qp_max > 63)) {
RTC_LOG(LS_WARNING) << "Unsupported vp8_qp_max_ value, ignored.";
- return absl::nullopt;
+ return std::nullopt;
}
return video_config_.vp8_qp_max;
}
-absl::optional<int> RateControlSettings::LibvpxVp8MinPixels() const {
+std::optional<int> RateControlSettings::LibvpxVp8MinPixels() const {
if (video_config_.vp8_min_pixels && *video_config_.vp8_min_pixels < 1) {
- return absl::nullopt;
+ return std::nullopt;
}
return video_config_.vp8_min_pixels;
}
diff --git a/rtc_base/experiments/rate_control_settings.h b/rtc_base/experiments/rate_control_settings.h
index 0dd4a14..d48da1c 100644
--- a/rtc_base/experiments/rate_control_settings.h
+++ b/rtc_base/experiments/rate_control_settings.h
@@ -11,7 +11,8 @@
#ifndef RTC_BASE_EXPERIMENTS_RATE_CONTROL_SETTINGS_H_
#define RTC_BASE_EXPERIMENTS_RATE_CONTROL_SETTINGS_H_
-#include "absl/types/optional.h"
+#include <optional>
+
#include "api/field_trials_view.h"
#include "api/units/data_size.h"
#include "api/video_codecs/video_codec.h"
@@ -22,9 +23,9 @@
struct CongestionWindowConfig {
static constexpr char kKey[] = "WebRTC-CongestionWindow";
- absl::optional<int> queue_size_ms;
- absl::optional<int> min_bitrate_bps;
- absl::optional<DataSize> initial_data_window;
+ std::optional<int> queue_size_ms;
+ std::optional<int> min_bitrate_bps;
+ std::optional<DataSize> initial_data_window;
bool drop_frame_only = false;
std::unique_ptr<StructParametersParser> Parser();
static CongestionWindowConfig Parse(absl::string_view config);
@@ -32,10 +33,10 @@
struct VideoRateControlConfig {
static constexpr char kKey[] = "WebRTC-VideoRateControl";
- absl::optional<double> pacing_factor;
+ std::optional<double> pacing_factor;
bool alr_probing = false;
- absl::optional<int> vp8_qp_max;
- absl::optional<int> vp8_min_pixels;
+ std::optional<int> vp8_qp_max;
+ std::optional<int> vp8_min_pixels;
bool trust_vp8 = true;
bool trust_vp9 = true;
bool bitrate_adjuster = true;
@@ -60,13 +61,13 @@
bool UseCongestionWindowPushback() const;
bool UseCongestionWindowDropFrameOnly() const;
uint32_t CongestionWindowMinPushbackTargetBitrateBps() const;
- absl::optional<DataSize> CongestionWindowInitialDataWindow() const;
+ std::optional<DataSize> CongestionWindowInitialDataWindow() const;
- absl::optional<double> GetPacingFactor() const;
+ std::optional<double> GetPacingFactor() const;
bool UseAlrProbing() const;
- absl::optional<int> LibvpxVp8QpMax() const;
- absl::optional<int> LibvpxVp8MinPixels() const;
+ std::optional<int> LibvpxVp8QpMax() const;
+ std::optional<int> LibvpxVp8MinPixels() const;
bool LibvpxVp8TrustedRateController() const;
bool Vp8BoostBaseLayerQuality() const;
bool Vp8DynamicRateSettings() const;
diff --git a/rtc_base/experiments/struct_parameters_parser.cc b/rtc_base/experiments/struct_parameters_parser.cc
index 011df3e..ffcd0a4 100644
--- a/rtc_base/experiments/struct_parameters_parser.cc
+++ b/rtc_base/experiments/struct_parameters_parser.cc
@@ -48,7 +48,7 @@
}
template <typename T>
-inline void StringEncode(std::string* sb, absl::optional<T> val) {
+inline void StringEncode(std::string* sb, std::optional<T> val) {
if (val)
StringEncode(sb, *val);
}
@@ -69,16 +69,16 @@
template class TypedParser<double>;
template class TypedParser<int>;
template class TypedParser<unsigned>;
-template class TypedParser<absl::optional<double>>;
-template class TypedParser<absl::optional<int>>;
-template class TypedParser<absl::optional<unsigned>>;
+template class TypedParser<std::optional<double>>;
+template class TypedParser<std::optional<int>>;
+template class TypedParser<std::optional<unsigned>>;
template class TypedParser<DataRate>;
template class TypedParser<DataSize>;
template class TypedParser<TimeDelta>;
-template class TypedParser<absl::optional<DataRate>>;
-template class TypedParser<absl::optional<DataSize>>;
-template class TypedParser<absl::optional<TimeDelta>>;
+template class TypedParser<std::optional<DataRate>>;
+template class TypedParser<std::optional<DataSize>>;
+template class TypedParser<std::optional<TimeDelta>>;
} // namespace struct_parser_impl
StructParametersParser::StructParametersParser(
diff --git a/rtc_base/experiments/struct_parameters_parser.h b/rtc_base/experiments/struct_parameters_parser.h
index f5f8340..3d0b05b 100644
--- a/rtc_base/experiments/struct_parameters_parser.h
+++ b/rtc_base/experiments/struct_parameters_parser.h
@@ -13,13 +13,13 @@
#include <functional>
#include <map>
#include <memory>
+#include <optional>
#include <string>
#include <utility>
#include <vector>
#include "absl/memory/memory.h"
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "rtc_base/experiments/field_trial_parser.h"
#include "rtc_base/experiments/field_trial_units.h"
#include "rtc_base/string_encode.h"
@@ -54,16 +54,16 @@
extern template class TypedParser<double>;
extern template class TypedParser<int>;
extern template class TypedParser<unsigned>;
-extern template class TypedParser<absl::optional<double>>;
-extern template class TypedParser<absl::optional<int>>;
-extern template class TypedParser<absl::optional<unsigned>>;
+extern template class TypedParser<std::optional<double>>;
+extern template class TypedParser<std::optional<int>>;
+extern template class TypedParser<std::optional<unsigned>>;
extern template class TypedParser<DataRate>;
extern template class TypedParser<DataSize>;
extern template class TypedParser<TimeDelta>;
-extern template class TypedParser<absl::optional<DataRate>>;
-extern template class TypedParser<absl::optional<DataSize>>;
-extern template class TypedParser<absl::optional<TimeDelta>>;
+extern template class TypedParser<std::optional<DataRate>>;
+extern template class TypedParser<std::optional<DataSize>>;
+extern template class TypedParser<std::optional<TimeDelta>>;
template <typename T>
void AddMembers(MemberParameter* out, const char* key, T* member) {
diff --git a/rtc_base/experiments/struct_parameters_parser_unittest.cc b/rtc_base/experiments/struct_parameters_parser_unittest.cc
index 0824bd3..918bcb6 100644
--- a/rtc_base/experiments/struct_parameters_parser_unittest.cc
+++ b/rtc_base/experiments/struct_parameters_parser_unittest.cc
@@ -19,8 +19,8 @@
int retries = 5;
unsigned size = 3;
bool ping = 0;
- absl::optional<TimeDelta> duration;
- absl::optional<TimeDelta> latency = TimeDelta::Millis(100);
+ std::optional<TimeDelta> duration;
+ std::optional<TimeDelta> latency = TimeDelta::Millis(100);
std::unique_ptr<StructParametersParser> Parser();
};
diff --git a/rtc_base/fake_network.h b/rtc_base/fake_network.h
index bc03306..b050198 100644
--- a/rtc_base/fake_network.h
+++ b/rtc_base/fake_network.h
@@ -36,7 +36,7 @@
struct Iface {
SocketAddress socket_address;
AdapterType adapter_type;
- absl::optional<AdapterType> underlying_vpn_adapter_type;
+ std::optional<AdapterType> underlying_vpn_adapter_type;
};
typedef std::vector<Iface> IfaceList;
@@ -53,7 +53,7 @@
const SocketAddress& iface,
absl::string_view if_name,
AdapterType type,
- absl::optional<AdapterType> underlying_vpn_adapter_type = absl::nullopt) {
+ std::optional<AdapterType> underlying_vpn_adapter_type = std::nullopt) {
SocketAddress address(if_name, 0);
address.SetResolvedIP(iface.ipaddr());
ifaces_.push_back({address, type, underlying_vpn_adapter_type});
diff --git a/rtc_base/file_rotating_stream.cc b/rtc_base/file_rotating_stream.cc
index c56396f..c1d3bfc 100644
--- a/rtc_base/file_rotating_stream.cc
+++ b/rtc_base/file_rotating_stream.cc
@@ -26,9 +26,10 @@
#include <unistd.h>
#endif // WEBRTC_WIN
+#include <optional>
+
#include "absl/algorithm/container.h"
#include "absl/strings/match.h"
-#include "absl/types/optional.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/strings/string_builder.h"
@@ -52,7 +53,7 @@
bool MoveFile(absl::string_view old_file, absl::string_view new_file);
bool IsFile(absl::string_view file);
bool IsFolder(absl::string_view file);
-absl::optional<size_t> GetFileSize(absl::string_view file);
+std::optional<size_t> GetFileSize(absl::string_view file);
#if defined(WEBRTC_WIN)
@@ -110,11 +111,11 @@
FILE_ATTRIBUTE_DIRECTORY;
}
-absl::optional<size_t> GetFileSize(absl::string_view file) {
+std::optional<size_t> GetFileSize(absl::string_view file) {
WIN32_FILE_ATTRIBUTE_DATA data = {0};
if (::GetFileAttributesExW(ToUtf16(file).c_str(), GetFileExInfoStandard,
&data) == 0)
- return absl::nullopt;
+ return std::nullopt;
return data.nFileSizeLow;
}
@@ -168,10 +169,10 @@
return res == 0 && S_ISDIR(st.st_mode);
}
-absl::optional<size_t> GetFileSize(absl::string_view file) {
+std::optional<size_t> GetFileSize(absl::string_view file) {
struct stat st;
if (::stat(std::string(file).c_str(), &st) != 0)
- return absl::nullopt;
+ return std::nullopt;
return st.st_size;
}
diff --git a/rtc_base/frequency_tracker.cc b/rtc_base/frequency_tracker.cc
index c3be30e..d99c67b 100644
--- a/rtc_base/frequency_tracker.cc
+++ b/rtc_base/frequency_tracker.cc
@@ -10,7 +10,8 @@
#include "rtc_base/frequency_tracker.h"
-#include "absl/types/optional.h"
+#include <optional>
+
#include "api/units/frequency.h"
#include "api/units/time_delta.h"
#include "api/units/timestamp.h"
@@ -21,11 +22,11 @@
FrequencyTracker::FrequencyTracker(TimeDelta max_window_size)
: impl_(max_window_size.ms(), 1'000'000) {}
-absl::optional<Frequency> FrequencyTracker::Rate(Timestamp now) const {
- if (absl::optional<int64_t> rate = impl_.Rate(now.ms())) {
+std::optional<Frequency> FrequencyTracker::Rate(Timestamp now) const {
+ if (std::optional<int64_t> rate = impl_.Rate(now.ms())) {
return Frequency::MilliHertz(*rate);
}
- return absl::nullopt;
+ return std::nullopt;
}
void FrequencyTracker::Update(int64_t count, Timestamp now) {
diff --git a/rtc_base/frequency_tracker.h b/rtc_base/frequency_tracker.h
index 3ee2ab0..6039c53 100644
--- a/rtc_base/frequency_tracker.h
+++ b/rtc_base/frequency_tracker.h
@@ -14,7 +14,8 @@
#include <stddef.h>
#include <stdint.h>
-#include "absl/types/optional.h"
+#include <optional>
+
#include "api/units/frequency.h"
#include "api/units/time_delta.h"
#include "api/units/timestamp.h"
@@ -46,7 +47,7 @@
// Returns rate, moving averaging window as needed.
// Returns nullopt when rate can't be measured.
- absl::optional<Frequency> Rate(Timestamp now) const;
+ std::optional<Frequency> Rate(Timestamp now) const;
private:
RateStatistics impl_;
diff --git a/rtc_base/frequency_tracker_unittest.cc b/rtc_base/frequency_tracker_unittest.cc
index 7f0b3d5..af9eeff 100644
--- a/rtc_base/frequency_tracker_unittest.cc
+++ b/rtc_base/frequency_tracker_unittest.cc
@@ -12,8 +12,8 @@
#include <cstdlib>
#include <limits>
+#include <optional>
-#include "absl/types/optional.h"
#include "api/units/frequency.h"
#include "api/units/time_delta.h"
#include "api/units/timestamp.h"
@@ -34,7 +34,7 @@
Timestamp now = Timestamp::Seconds(12'345);
FrequencyTracker stats(kWindow);
- EXPECT_EQ(stats.Rate(now), absl::nullopt);
+ EXPECT_EQ(stats.Rate(now), std::nullopt);
}
TEST(FrequencyTrackerTest, ReturnsNulloptAfterSingleDataPoint) {
@@ -44,7 +44,7 @@
stats.Update(now);
now += TimeDelta::Millis(10);
- EXPECT_EQ(stats.Rate(now), absl::nullopt);
+ EXPECT_EQ(stats.Rate(now), std::nullopt);
}
TEST(FrequencyTrackerTest, ReturnsRateAfterTwoMeasurements) {
@@ -76,7 +76,7 @@
// Until window is full, rate is measured over a smaller window and might
// look larger than the constant rate.
- absl::optional<Frequency> rate = stats.Rate(now);
+ std::optional<Frequency> rate = stats.Rate(now);
ASSERT_GE(rate, kConstantRate);
// Expect the estimation error to decrease as the window is extended.
@@ -127,7 +127,7 @@
now += kLargeInterval;
stats.Update(kLargeSize, now);
}
- absl::optional<Frequency> last_rate = stats.Rate(now);
+ std::optional<Frequency> last_rate = stats.Rate(now);
EXPECT_EQ(last_rate, kLargeSize / kLargeInterval);
// Decrease rate with smaller measurments.
@@ -136,7 +136,7 @@
now += kLargeInterval;
stats.Update(kSmallSize, now);
- absl::optional<Frequency> rate = stats.Rate(now);
+ std::optional<Frequency> rate = stats.Rate(now);
EXPECT_LT(rate, last_rate);
last_rate = rate;
@@ -149,7 +149,7 @@
now += kSmallInterval;
stats.Update(kSmallSize, now);
- absl::optional<Frequency> rate = stats.Rate(now);
+ std::optional<Frequency> rate = stats.Rate(now);
EXPECT_GE(rate, last_rate);
last_rate = rate;
@@ -174,17 +174,17 @@
now += kWindow + kEpsilon;
// Silence over window size should trigger auto reset for coming sample.
- EXPECT_EQ(pixel_rate.Rate(now), absl::nullopt);
+ EXPECT_EQ(pixel_rate.Rate(now), std::nullopt);
pixel_rate.Update(kPixels, now);
// Single measurment after reset is not enough to estimate the rate.
- EXPECT_EQ(pixel_rate.Rate(now), absl::nullopt);
+ EXPECT_EQ(pixel_rate.Rate(now), std::nullopt);
// Manual reset, add the same check again.
pixel_rate.Reset();
- EXPECT_EQ(pixel_rate.Rate(now), absl::nullopt);
+ EXPECT_EQ(pixel_rate.Rate(now), std::nullopt);
now += kInterval;
pixel_rate.Update(kPixels, now);
- EXPECT_EQ(pixel_rate.Rate(now), absl::nullopt);
+ EXPECT_EQ(pixel_rate.Rate(now), std::nullopt);
}
TEST(FrequencyTrackerTest, ReturnsNulloptWhenOverflows) {
@@ -196,7 +196,7 @@
now += kEpsilon;
stats.Update(very_large_number, now);
- EXPECT_EQ(stats.Rate(now), absl::nullopt);
+ EXPECT_EQ(stats.Rate(now), std::nullopt);
}
} // namespace
diff --git a/rtc_base/logging.h b/rtc_base/logging.h
index b171cfe..3d27d6c 100644
--- a/rtc_base/logging.h
+++ b/rtc_base/logging.h
@@ -51,6 +51,7 @@
#include <errno.h>
#include <atomic>
+#include <optional>
#include <sstream> // no-presubmit-check TODO(webrtc:8982)
#include <string>
#include <type_traits>
@@ -59,7 +60,6 @@
#include "absl/base/attributes.h"
#include "absl/meta/type_traits.h"
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/units/timestamp.h"
#include "rtc_base/platform_thread_types.h"
#include "rtc_base/strings/string_builder.h"
@@ -118,7 +118,7 @@
absl::string_view message() const { return message_; }
absl::string_view filename() const { return filename_; }
int line() const { return line_; }
- absl::optional<PlatformThreadId> thread_id() const { return thread_id_; }
+ std::optional<PlatformThreadId> thread_id() const { return thread_id_; }
webrtc::Timestamp timestamp() const { return timestamp_; }
absl::string_view tag() const { return tag_; }
LoggingSeverity severity() const { return severity_; }
@@ -134,7 +134,7 @@
void set_message(std::string message) { message_ = std::move(message); }
void set_filename(absl::string_view filename) { filename_ = filename; }
void set_line(int line) { line_ = line; }
- void set_thread_id(absl::optional<PlatformThreadId> thread_id) {
+ void set_thread_id(std::optional<PlatformThreadId> thread_id) {
thread_id_ = thread_id;
}
void set_timestamp(webrtc::Timestamp timestamp) { timestamp_ = timestamp; }
@@ -144,7 +144,7 @@
std::string message_;
absl::string_view filename_;
int line_ = 0;
- absl::optional<PlatformThreadId> thread_id_;
+ std::optional<PlatformThreadId> thread_id_;
webrtc::Timestamp timestamp_ = webrtc::Timestamp::MinusInfinity();
// The default Android debug output tag.
absl::string_view tag_ = "libjingle";
diff --git a/rtc_base/network/BUILD.gn b/rtc_base/network/BUILD.gn
index c1e52cd..d4933f2 100644
--- a/rtc_base/network/BUILD.gn
+++ b/rtc_base/network/BUILD.gn
@@ -13,10 +13,7 @@
"sent_packet.cc",
"sent_packet.h",
]
- deps = [
- "../system:rtc_export",
- "//third_party/abseil-cpp/absl/types:optional",
- ]
+ deps = [ "../system:rtc_export" ]
}
rtc_source_set("ecn_marking") {
@@ -36,6 +33,5 @@
"../../api:array_view",
"../../api/units:timestamp",
"../system:rtc_export",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
diff --git a/rtc_base/network/received_packet.cc b/rtc_base/network/received_packet.cc
index 9588e37..346a17f 100644
--- a/rtc_base/network/received_packet.cc
+++ b/rtc_base/network/received_packet.cc
@@ -10,16 +10,16 @@
#include "rtc_base/network/received_packet.h"
+#include <optional>
#include <utility>
-#include "absl/types/optional.h"
#include "rtc_base/socket_address.h"
namespace rtc {
ReceivedPacket::ReceivedPacket(rtc::ArrayView<const uint8_t> payload,
const SocketAddress& source_address,
- absl::optional<webrtc::Timestamp> arrival_time,
+ std::optional<webrtc::Timestamp> arrival_time,
EcnMarking ecn,
DecryptionInfo decryption)
: payload_(payload),
@@ -43,9 +43,9 @@
RTC_DCHECK(packet_time_us == -1 || packet_time_us >= 0);
return ReceivedPacket(rtc::MakeArrayView(data, size), source_address,
(packet_time_us >= 0)
- ? absl::optional<webrtc::Timestamp>(
+ ? std::optional<webrtc::Timestamp>(
webrtc::Timestamp::Micros(packet_time_us))
- : absl::nullopt);
+ : std::nullopt);
}
} // namespace rtc
diff --git a/rtc_base/network/received_packet.h b/rtc_base/network/received_packet.h
index 68dd33b..ec0b0bf 100644
--- a/rtc_base/network/received_packet.h
+++ b/rtc_base/network/received_packet.h
@@ -11,8 +11,8 @@
#define RTC_BASE_NETWORK_RECEIVED_PACKET_H_
#include <cstdint>
+#include <optional>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/units/timestamp.h"
#include "rtc_base/network/ecn_marking.h"
@@ -38,7 +38,7 @@
// lifetime of this ReceivedPacket.
ReceivedPacket(rtc::ArrayView<const uint8_t> payload,
const SocketAddress& source_address,
- absl::optional<webrtc::Timestamp> arrival_time = absl::nullopt,
+ std::optional<webrtc::Timestamp> arrival_time = std::nullopt,
EcnMarking ecn = EcnMarking::kNotEct,
DecryptionInfo decryption = kNotDecrypted);
@@ -50,7 +50,7 @@
// Timestamp when this packet was received. Not available on all socket
// implementations.
- absl::optional<webrtc::Timestamp> arrival_time() const {
+ std::optional<webrtc::Timestamp> arrival_time() const {
return arrival_time_;
}
@@ -76,7 +76,7 @@
private:
rtc::ArrayView<const uint8_t> payload_;
- absl::optional<webrtc::Timestamp> arrival_time_;
+ std::optional<webrtc::Timestamp> arrival_time_;
const SocketAddress& source_address_;
EcnMarking ecn_;
DecryptionInfo decryption_info_;
diff --git a/rtc_base/network/sent_packet.h b/rtc_base/network/sent_packet.h
index 457fb96..3e6f0d0 100644
--- a/rtc_base/network/sent_packet.h
+++ b/rtc_base/network/sent_packet.h
@@ -14,7 +14,8 @@
#include <stddef.h>
#include <stdint.h>
-#include "absl/types/optional.h"
+#include <optional>
+
#include "rtc_base/system/rtc_export.h"
namespace rtc {
@@ -45,8 +46,8 @@
bool included_in_allocation = false;
PacketType packet_type = PacketType::kUnknown;
PacketInfoProtocolType protocol = PacketInfoProtocolType::kUnknown;
- // A unique id assigned by the network manager, and absl::nullopt if not set.
- absl::optional<uint16_t> network_id;
+ // A unique id assigned by the network manager, and std::nullopt if not set.
+ std::optional<uint16_t> network_id;
size_t packet_size_bytes = 0;
size_t turn_overhead_bytes = 0;
size_t ip_overhead_bytes = 0;
diff --git a/rtc_base/numerics/event_based_exponential_moving_average.h b/rtc_base/numerics/event_based_exponential_moving_average.h
index 69f4e61..7055a4d 100644
--- a/rtc_base/numerics/event_based_exponential_moving_average.h
+++ b/rtc_base/numerics/event_based_exponential_moving_average.h
@@ -14,8 +14,7 @@
#include <cmath>
#include <cstdint>
#include <limits>
-
-#include "absl/types/optional.h"
+#include <optional>
namespace rtc {
@@ -63,7 +62,7 @@
double sample_variance_ = std::numeric_limits<double>::infinity();
// This is the ratio between variance of the estimate and variance of samples.
double estimator_variance_ = 1;
- absl::optional<int64_t> last_observation_timestamp_;
+ std::optional<int64_t> last_observation_timestamp_;
};
} // namespace rtc
diff --git a/rtc_base/numerics/histogram_percentile_counter.cc b/rtc_base/numerics/histogram_percentile_counter.cc
index 498bb4b..28f9174 100644
--- a/rtc_base/numerics/histogram_percentile_counter.cc
+++ b/rtc_base/numerics/histogram_percentile_counter.cc
@@ -14,8 +14,8 @@
#include <cmath>
#include <cstddef>
#include <cstdint>
+#include <optional>
-#include "absl/types/optional.h"
#include "rtc_base/checks.h"
namespace rtc {
@@ -51,12 +51,12 @@
Add(value, 1);
}
-absl::optional<uint32_t> HistogramPercentileCounter::GetPercentile(
+std::optional<uint32_t> HistogramPercentileCounter::GetPercentile(
float fraction) {
RTC_CHECK_LE(fraction, 1.0);
RTC_CHECK_GE(fraction, 0.0);
if (total_elements_ == 0)
- return absl::nullopt;
+ return std::nullopt;
size_t elements_to_skip = static_cast<size_t>(
std::max(0.0f, std::ceil(total_elements_ * fraction) - 1));
if (elements_to_skip >= total_elements_)
@@ -76,7 +76,7 @@
}
}
RTC_DCHECK_NOTREACHED();
- return absl::nullopt;
+ return std::nullopt;
}
} // namespace rtc
diff --git a/rtc_base/numerics/histogram_percentile_counter.h b/rtc_base/numerics/histogram_percentile_counter.h
index 4787f2e..327ce0d 100644
--- a/rtc_base/numerics/histogram_percentile_counter.h
+++ b/rtc_base/numerics/histogram_percentile_counter.h
@@ -15,10 +15,9 @@
#include <stdint.h>
#include <map>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
-
namespace rtc {
// Calculates percentiles on the stream of data. Use `Add` methods to add new
// values. Use `GetPercentile` to get percentile of the currently added values.
@@ -32,7 +31,7 @@
void Add(uint32_t value, size_t count);
void Add(const HistogramPercentileCounter& other);
// Argument should be from 0 to 1.
- absl::optional<uint32_t> GetPercentile(float fraction);
+ std::optional<uint32_t> GetPercentile(float fraction);
private:
std::vector<size_t> histogram_low_;
diff --git a/rtc_base/numerics/moving_average.cc b/rtc_base/numerics/moving_average.cc
index 126223e..f91926f 100644
--- a/rtc_base/numerics/moving_average.cc
+++ b/rtc_base/numerics/moving_average.cc
@@ -13,8 +13,8 @@
#include <algorithm>
#include <cstddef>
#include <cstdint>
+#include <optional>
-#include "absl/types/optional.h"
#include "rtc_base/checks.h"
namespace rtc {
@@ -34,21 +34,21 @@
history_[index] = sample;
}
-absl::optional<int> MovingAverage::GetAverageRoundedDown() const {
+std::optional<int> MovingAverage::GetAverageRoundedDown() const {
if (count_ == 0)
- return absl::nullopt;
+ return std::nullopt;
return sum_ / Size();
}
-absl::optional<int> MovingAverage::GetAverageRoundedToClosest() const {
+std::optional<int> MovingAverage::GetAverageRoundedToClosest() const {
if (count_ == 0)
- return absl::nullopt;
+ return std::nullopt;
return (sum_ + Size() / 2) / Size();
}
-absl::optional<double> MovingAverage::GetUnroundedAverage() const {
+std::optional<double> MovingAverage::GetUnroundedAverage() const {
if (count_ == 0)
- return absl::nullopt;
+ return std::nullopt;
return sum_ / static_cast<double>(Size());
}
diff --git a/rtc_base/numerics/moving_average.h b/rtc_base/numerics/moving_average.h
index 41ce603..f9aa8ee 100644
--- a/rtc_base/numerics/moving_average.h
+++ b/rtc_base/numerics/moving_average.h
@@ -14,10 +14,9 @@
#include <stddef.h>
#include <stdint.h>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
-
namespace rtc {
// Calculates average over fixed size window. If there are less than window
@@ -38,13 +37,13 @@
// Returns rounded down average of last `window_size` elements or all
// elements if there are not enough of them. Returns nullopt if there were
// no elements added.
- absl::optional<int> GetAverageRoundedDown() const;
+ std::optional<int> GetAverageRoundedDown() const;
// Same as above but rounded to the closest integer.
- absl::optional<int> GetAverageRoundedToClosest() const;
+ std::optional<int> GetAverageRoundedToClosest() const;
// Returns unrounded average over the window.
- absl::optional<double> GetUnroundedAverage() const;
+ std::optional<double> GetUnroundedAverage() const;
// Resets to the initial state before any elements were added.
void Reset();
diff --git a/rtc_base/numerics/moving_average_unittest.cc b/rtc_base/numerics/moving_average_unittest.cc
index 5993401..163a778 100644
--- a/rtc_base/numerics/moving_average_unittest.cc
+++ b/rtc_base/numerics/moving_average_unittest.cc
@@ -10,7 +10,8 @@
#include "rtc_base/numerics/moving_average.h"
-#include "absl/types/optional.h"
+#include <optional>
+
#include "test/gtest.h"
namespace test {
@@ -18,7 +19,7 @@
TEST(MovingAverageTest, EmptyAverage) {
rtc::MovingAverage moving_average(1);
EXPECT_EQ(0u, moving_average.Size());
- EXPECT_EQ(absl::nullopt, moving_average.GetAverageRoundedDown());
+ EXPECT_EQ(std::nullopt, moving_average.GetAverageRoundedDown());
}
// Test single value.
diff --git a/rtc_base/numerics/moving_max_counter.h b/rtc_base/numerics/moving_max_counter.h
index 5eb45d39..f35d360 100644
--- a/rtc_base/numerics/moving_max_counter.h
+++ b/rtc_base/numerics/moving_max_counter.h
@@ -15,9 +15,9 @@
#include <deque>
#include <limits>
+#include <optional>
#include <utility>
-#include "absl/types/optional.h"
#include "rtc_base/checks.h"
namespace rtc {
@@ -43,7 +43,7 @@
// Advances the current time, and returns the maximum sample in the time
// window ending at the current time. The new current time must be at least as
// large as the old current time.
- absl::optional<T> Max(int64_t current_time_ms);
+ std::optional<T> Max(int64_t current_time_ms);
void Reset();
private:
@@ -85,9 +85,9 @@
}
template <class T>
-absl::optional<T> MovingMaxCounter<T>::Max(int64_t current_time_ms) {
+std::optional<T> MovingMaxCounter<T>::Max(int64_t current_time_ms) {
RollWindow(current_time_ms);
- absl::optional<T> res;
+ std::optional<T> res;
if (!samples_.empty()) {
res.emplace(samples_.front().second);
}
diff --git a/rtc_base/numerics/running_statistics.h b/rtc_base/numerics/running_statistics.h
index 134b895..c747c79 100644
--- a/rtc_base/numerics/running_statistics.h
+++ b/rtc_base/numerics/running_statistics.h
@@ -14,8 +14,8 @@
#include <algorithm>
#include <cmath>
#include <cstdint>
+#include <optional>
-#include "absl/types/optional.h"
#include "rtc_base/checks.h"
#include "rtc_base/numerics/math_utils.h"
@@ -33,7 +33,7 @@
// If you want a full-fledged moving window over N last samples,
// please use webrtc::RollingAccumulator.
//
-// The measures return absl::nullopt if no samples were fed (Size() == 0),
+// The measures return std::nullopt if no samples were fed (Size() == 0),
// otherwise the returned optional is guaranteed to contain a value.
//
// [1]
@@ -108,50 +108,50 @@
// Returns minimum among all seen samples, in O(1) time.
// This isn't affected by RemoveSample().
- absl::optional<T> GetMin() const {
+ std::optional<T> GetMin() const {
if (size_ == 0) {
- return absl::nullopt;
+ return std::nullopt;
}
return min_;
}
// Returns maximum among all seen samples, in O(1) time.
// This isn't affected by RemoveSample().
- absl::optional<T> GetMax() const {
+ std::optional<T> GetMax() const {
if (size_ == 0) {
- return absl::nullopt;
+ return std::nullopt;
}
return max_;
}
// Returns sum in O(1) time.
- absl::optional<double> GetSum() const {
+ std::optional<double> GetSum() const {
if (size_ == 0) {
- return absl::nullopt;
+ return std::nullopt;
}
return sum_;
}
// Returns mean in O(1) time.
- absl::optional<double> GetMean() const {
+ std::optional<double> GetMean() const {
if (size_ == 0) {
- return absl::nullopt;
+ return std::nullopt;
}
return mean_;
}
// Returns unbiased sample variance in O(1) time.
- absl::optional<double> GetVariance() const {
+ std::optional<double> GetVariance() const {
if (size_ == 0) {
- return absl::nullopt;
+ return std::nullopt;
}
return cumul_ / size_;
}
// Returns unbiased standard deviation in O(1) time.
- absl::optional<double> GetStandardDeviation() const {
+ std::optional<double> GetStandardDeviation() const {
if (size_ == 0) {
- return absl::nullopt;
+ return std::nullopt;
}
return std::sqrt(*GetVariance());
}
diff --git a/rtc_base/numerics/sample_counter.cc b/rtc_base/numerics/sample_counter.cc
index 5e63b97..50e9f2a 100644
--- a/rtc_base/numerics/sample_counter.cc
+++ b/rtc_base/numerics/sample_counter.cc
@@ -12,8 +12,8 @@
#include <cstdint>
#include <limits>
+#include <optional>
-#include "absl/types/optional.h"
#include "rtc_base/checks.h"
#include "rtc_base/numerics/safe_conversions.h"
@@ -54,25 +54,25 @@
min_ = other.min_;
}
-absl::optional<int> SampleCounter::Avg(int64_t min_required_samples) const {
+std::optional<int> SampleCounter::Avg(int64_t min_required_samples) const {
RTC_DCHECK_GT(min_required_samples, 0);
if (num_samples_ < min_required_samples)
- return absl::nullopt;
+ return std::nullopt;
return rtc::dchecked_cast<int>(sum_ / num_samples_);
}
-absl::optional<int> SampleCounter::Max() const {
+std::optional<int> SampleCounter::Max() const {
return max_;
}
-absl::optional<int> SampleCounter::Min() const {
+std::optional<int> SampleCounter::Min() const {
return min_;
}
-absl::optional<int64_t> SampleCounter::Sum(int64_t min_required_samples) const {
+std::optional<int64_t> SampleCounter::Sum(int64_t min_required_samples) const {
RTC_DCHECK_GT(min_required_samples, 0);
if (num_samples_ < min_required_samples)
- return absl::nullopt;
+ return std::nullopt;
return sum_;
}
@@ -87,11 +87,11 @@
SampleCounterWithVariance::SampleCounterWithVariance() = default;
SampleCounterWithVariance::~SampleCounterWithVariance() = default;
-absl::optional<int64_t> SampleCounterWithVariance::Variance(
+std::optional<int64_t> SampleCounterWithVariance::Variance(
int64_t min_required_samples) const {
RTC_DCHECK_GT(min_required_samples, 0);
if (num_samples_ < min_required_samples)
- return absl::nullopt;
+ return std::nullopt;
// E[(x-mean)^2] = E[x^2] - mean^2
int64_t mean = sum_ / num_samples_;
return sum_squared_ / num_samples_ - mean * mean;
diff --git a/rtc_base/numerics/sample_counter.h b/rtc_base/numerics/sample_counter.h
index 2b41f95..fa58ce1 100644
--- a/rtc_base/numerics/sample_counter.h
+++ b/rtc_base/numerics/sample_counter.h
@@ -13,7 +13,7 @@
#include <stdint.h>
-#include "absl/types/optional.h"
+#include <optional>
namespace rtc {
@@ -24,10 +24,10 @@
SampleCounter();
~SampleCounter();
void Add(int sample);
- absl::optional<int> Avg(int64_t min_required_samples) const;
- absl::optional<int> Max() const;
- absl::optional<int> Min() const;
- absl::optional<int64_t> Sum(int64_t min_required_samples) const;
+ std::optional<int> Avg(int64_t min_required_samples) const;
+ std::optional<int> Max() const;
+ std::optional<int> Min() const;
+ std::optional<int64_t> Sum(int64_t min_required_samples) const;
int64_t NumSamples() const;
void Reset();
// Adds all the samples from the `other` SampleCounter as if they were all
@@ -37,8 +37,8 @@
protected:
int64_t sum_ = 0;
int64_t num_samples_ = 0;
- absl::optional<int> max_;
- absl::optional<int> min_;
+ std::optional<int> max_;
+ std::optional<int> min_;
};
class SampleCounterWithVariance : public SampleCounter {
@@ -46,7 +46,7 @@
SampleCounterWithVariance();
~SampleCounterWithVariance();
void Add(int sample);
- absl::optional<int64_t> Variance(int64_t min_required_samples) const;
+ std::optional<int64_t> Variance(int64_t min_required_samples) const;
void Reset();
// Adds all the samples from the `other` SampleCounter as if they were all
// individually added using `Add(int)` method.
diff --git a/rtc_base/numerics/sample_counter_unittest.cc b/rtc_base/numerics/sample_counter_unittest.cc
index 6877f5e..82e61fd 100644
--- a/rtc_base/numerics/sample_counter_unittest.cc
+++ b/rtc_base/numerics/sample_counter_unittest.cc
@@ -11,8 +11,8 @@
#include "rtc_base/numerics/sample_counter.h"
#include <initializer_list>
+#include <optional>
-#include "absl/types/optional.h"
#include "test/gmock.h"
#include "test/gtest.h"
@@ -23,9 +23,9 @@
TEST(SampleCounterTest, ProcessesNoSamples) {
constexpr int kMinSamples = 1;
SampleCounter counter;
- EXPECT_THAT(counter.Avg(kMinSamples), Eq(absl::nullopt));
- EXPECT_THAT(counter.Max(), Eq(absl::nullopt));
- EXPECT_THAT(counter.Min(), Eq(absl::nullopt));
+ EXPECT_THAT(counter.Avg(kMinSamples), Eq(std::nullopt));
+ EXPECT_THAT(counter.Max(), Eq(std::nullopt));
+ EXPECT_THAT(counter.Min(), Eq(std::nullopt));
}
TEST(SampleCounterTest, NotEnoughSamples) {
@@ -34,8 +34,8 @@
for (int value : {1, 2, 3, 4, 5}) {
counter.Add(value);
}
- EXPECT_THAT(counter.Avg(kMinSamples), Eq(absl::nullopt));
- EXPECT_THAT(counter.Sum(kMinSamples), Eq(absl::nullopt));
+ EXPECT_THAT(counter.Avg(kMinSamples), Eq(std::nullopt));
+ EXPECT_THAT(counter.Sum(kMinSamples), Eq(std::nullopt));
EXPECT_THAT(counter.Max(), Eq(5));
EXPECT_THAT(counter.Min(), Eq(1));
}
@@ -72,8 +72,8 @@
counter2.Add(value);
}
// Before aggregation there is not enough samples.
- EXPECT_THAT(counter1.Avg(kMinSamples), Eq(absl::nullopt));
- EXPECT_THAT(counter1.Variance(kMinSamples), Eq(absl::nullopt));
+ EXPECT_THAT(counter1.Avg(kMinSamples), Eq(std::nullopt));
+ EXPECT_THAT(counter1.Variance(kMinSamples), Eq(std::nullopt));
// Aggregate counter2 in counter1.
counter1.Add(counter2);
EXPECT_THAT(counter1.Avg(kMinSamples), Eq(3));
diff --git a/rtc_base/numerics/sequence_number_unwrapper.h b/rtc_base/numerics/sequence_number_unwrapper.h
index fb21283..1def581 100644
--- a/rtc_base/numerics/sequence_number_unwrapper.h
+++ b/rtc_base/numerics/sequence_number_unwrapper.h
@@ -14,9 +14,9 @@
#include <stdint.h>
#include <limits>
+#include <optional>
#include <type_traits>
-#include "absl/types/optional.h"
#include "rtc_base/numerics/sequence_number_util.h"
namespace webrtc {
@@ -70,7 +70,7 @@
}
int64_t last_unwrapped_ = 0;
- absl::optional<T> last_value_;
+ std::optional<T> last_value_;
};
using RtpTimestampUnwrapper = SeqNumUnwrapper<uint32_t>;
diff --git a/rtc_base/openssl_stream_adapter.h b/rtc_base/openssl_stream_adapter.h
index 2116b2d..92e8c10 100644
--- a/rtc_base/openssl_stream_adapter.h
+++ b/rtc_base/openssl_stream_adapter.h
@@ -16,12 +16,12 @@
#include <stdint.h>
#include <memory>
+#include <optional>
#include <string>
#include <vector>
#include "absl/functional/any_invocable.h"
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "rtc_base/buffer.h"
#ifdef OPENSSL_IS_BORINGSSL
#include "rtc_base/boringssl_identity.h"
diff --git a/rtc_base/operations_chain.h b/rtc_base/operations_chain.h
index 0e8c068..012f3dd 100644
--- a/rtc_base/operations_chain.h
+++ b/rtc_base/operations_chain.h
@@ -13,12 +13,12 @@
#include <functional>
#include <memory>
+#include <optional>
#include <queue>
#include <set>
#include <type_traits>
#include <utility>
-#include "absl/types/optional.h"
#include "api/ref_counted_base.h"
#include "api/scoped_refptr.h"
#include "api/sequence_checker.h"
@@ -194,7 +194,7 @@
// to it.
std::queue<std::unique_ptr<rtc_operations_chain_internal::Operation>>
chained_operations_ RTC_GUARDED_BY(sequence_checker_);
- absl::optional<std::function<void()>> on_chain_empty_callback_
+ std::optional<std::function<void()>> on_chain_empty_callback_
RTC_GUARDED_BY(sequence_checker_);
};
diff --git a/rtc_base/platform_thread.cc b/rtc_base/platform_thread.cc
index 6433323..6af1f3f 100644
--- a/rtc_base/platform_thread.cc
+++ b/rtc_base/platform_thread.cc
@@ -113,14 +113,14 @@
PlatformThread::PlatformThread(PlatformThread&& rhs)
: handle_(rhs.handle_), joinable_(rhs.joinable_) {
- rhs.handle_ = absl::nullopt;
+ rhs.handle_ = std::nullopt;
}
PlatformThread& PlatformThread::operator=(PlatformThread&& rhs) {
Finalize();
handle_ = rhs.handle_;
joinable_ = rhs.joinable_;
- rhs.handle_ = absl::nullopt;
+ rhs.handle_ = std::nullopt;
return *this;
}
@@ -144,7 +144,7 @@
/*joinable=*/false);
}
-absl::optional<PlatformThread::Handle> PlatformThread::GetHandle() const {
+std::optional<PlatformThread::Handle> PlatformThread::GetHandle() const {
return handle_;
}
@@ -167,7 +167,7 @@
if (joinable_)
RTC_CHECK_EQ(0, pthread_join(*handle_, nullptr));
#endif
- handle_ = absl::nullopt;
+ handle_ = std::nullopt;
}
PlatformThread PlatformThread::SpawnThread(
diff --git a/rtc_base/platform_thread.h b/rtc_base/platform_thread.h
index befd618..cfc9d17 100644
--- a/rtc_base/platform_thread.h
+++ b/rtc_base/platform_thread.h
@@ -17,8 +17,9 @@
#include <pthread.h>
#endif
+#include <optional>
+
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "rtc_base/platform_thread_types.h"
namespace rtc {
@@ -97,7 +98,7 @@
ThreadAttributes attributes = ThreadAttributes());
// Returns the base platform thread handle of this thread.
- absl::optional<Handle> GetHandle() const;
+ std::optional<Handle> GetHandle() const;
#if defined(WEBRTC_WIN)
// Queue a Windows APC function that runs when the thread is alertable.
@@ -111,7 +112,7 @@
ThreadAttributes attributes,
bool joinable);
- absl::optional<Handle> handle_;
+ std::optional<Handle> handle_;
bool joinable_ = false;
};
diff --git a/rtc_base/platform_thread_unittest.cc b/rtc_base/platform_thread_unittest.cc
index 97b25e0..be4417a 100644
--- a/rtc_base/platform_thread_unittest.cc
+++ b/rtc_base/platform_thread_unittest.cc
@@ -10,7 +10,8 @@
#include "rtc_base/platform_thread.h"
-#include "absl/types/optional.h"
+#include <optional>
+
#include "rtc_base/event.h"
#include "system_wrappers/include/sleep.h"
#include "test/gmock.h"
@@ -19,13 +20,13 @@
TEST(PlatformThreadTest, DefaultConstructedIsEmpty) {
PlatformThread thread;
- EXPECT_EQ(thread.GetHandle(), absl::nullopt);
+ EXPECT_EQ(thread.GetHandle(), std::nullopt);
EXPECT_TRUE(thread.empty());
}
TEST(PlatformThreadTest, StartFinalize) {
PlatformThread thread = PlatformThread::SpawnJoinable([] {}, "1");
- EXPECT_NE(thread.GetHandle(), absl::nullopt);
+ EXPECT_NE(thread.GetHandle(), std::nullopt);
EXPECT_FALSE(thread.empty());
thread.Finalize();
EXPECT_TRUE(thread.empty());
diff --git a/rtc_base/rate_limiter.cc b/rtc_base/rate_limiter.cc
index 0f3f343..26e1c30 100644
--- a/rtc_base/rate_limiter.cc
+++ b/rtc_base/rate_limiter.cc
@@ -11,8 +11,8 @@
#include "rtc_base/rate_limiter.h"
#include <limits>
+#include <optional>
-#include "absl/types/optional.h"
#include "system_wrappers/include/clock.h"
namespace webrtc {
@@ -33,7 +33,7 @@
bool RateLimiter::TryUseRate(size_t packet_size_bytes) {
MutexLock lock(&lock_);
int64_t now_ms = clock_->TimeInMilliseconds();
- absl::optional<uint32_t> current_rate = current_rate_.Rate(now_ms);
+ std::optional<uint32_t> current_rate = current_rate_.Rate(now_ms);
if (current_rate) {
// If there is a current rate, check if adding bytes would cause maximum
// bitrate target to be exceeded. If there is NOT a valid current rate,
diff --git a/rtc_base/rate_statistics.cc b/rtc_base/rate_statistics.cc
index 5c83796..401ca24 100644
--- a/rtc_base/rate_statistics.cc
+++ b/rtc_base/rate_statistics.cc
@@ -84,7 +84,7 @@
++num_samples_;
}
-absl::optional<int64_t> RateStatistics::Rate(int64_t now_ms) const {
+std::optional<int64_t> RateStatistics::Rate(int64_t now_ms) const {
// Yeah, this const_cast ain't pretty, but the alternative is to declare most
// of the members as mutable...
const_cast<RateStatistics*>(this)->EraseOld(now_ms);
@@ -109,7 +109,7 @@
(num_samples_ <= 1 &&
rtc::SafeLt(active_window_size, current_window_size_ms_)) ||
overflow_) {
- return absl::nullopt;
+ return std::nullopt;
}
float scale = static_cast<float>(scale_) / active_window_size;
@@ -117,7 +117,7 @@
// Better return unavailable rate than garbage value (undefined behavior).
if (result > static_cast<float>(std::numeric_limits<int64_t>::max())) {
- return absl::nullopt;
+ return std::nullopt;
}
return rtc::dchecked_cast<int64_t>(result);
}
diff --git a/rtc_base/rate_statistics.h b/rtc_base/rate_statistics.h
index e7ce8ad..7bcb001 100644
--- a/rtc_base/rate_statistics.h
+++ b/rtc_base/rate_statistics.h
@@ -16,8 +16,8 @@
#include <deque>
#include <memory>
+#include <optional>
-#include "absl/types/optional.h"
#include "rtc_base/system/rtc_export.h"
namespace webrtc {
@@ -62,7 +62,7 @@
// from a monotonic clock. Ie, it doesn't matter if this call moves the
// window, since any subsequent call to Update or Rate would still have moved
// the window as much or more.
- absl::optional<int64_t> Rate(int64_t now_ms) const;
+ std::optional<int64_t> Rate(int64_t now_ms) const;
// Update the size of the averaging window. The maximum allowed value for
// window_size_ms is max_window_size_ms as supplied in the constructor.
diff --git a/rtc_base/rate_statistics_unittest.cc b/rtc_base/rate_statistics_unittest.cc
index 8f1a838..5f9707a 100644
--- a/rtc_base/rate_statistics_unittest.cc
+++ b/rtc_base/rate_statistics_unittest.cc
@@ -54,7 +54,7 @@
// Approximately 1200 kbps expected. Not exact since when packets
// are removed we will jump 10 ms to the next packet.
if (i > kInterval) {
- absl::optional<uint32_t> rate = stats_.Rate(now_ms);
+ std::optional<uint32_t> rate = stats_.Rate(now_ms);
EXPECT_TRUE(static_cast<bool>(rate));
uint32_t samples = i / kInterval + 1;
uint64_t total_bits = samples * kPacketSize * 8;
@@ -79,7 +79,7 @@
const uint32_t kExpectedBitrate = 8000000;
// 1000 bytes per millisecond until plateau is reached.
int prev_error = kExpectedBitrate;
- absl::optional<uint32_t> bitrate;
+ std::optional<uint32_t> bitrate;
while (++now_ms < 10000) {
stats_.Update(1000, now_ms);
bitrate = stats_.Rate(now_ms);
@@ -103,7 +103,7 @@
// Zero bytes per millisecond until 0 is reached.
while (++now_ms < 20000) {
stats_.Update(0, now_ms);
- absl::optional<uint32_t> new_bitrate = stats_.Rate(now_ms);
+ std::optional<uint32_t> new_bitrate = stats_.Rate(now_ms);
if (static_cast<bool>(new_bitrate) && *new_bitrate != *bitrate) {
// New bitrate must be lower than previous one.
EXPECT_LT(*new_bitrate, *bitrate);
@@ -131,7 +131,7 @@
const uint32_t kExpectedBitrate = 8000000;
// 1000 bytes per millisecond until the window has been filled.
int prev_error = kExpectedBitrate;
- absl::optional<uint32_t> bitrate;
+ std::optional<uint32_t> bitrate;
while (++now_ms < 10000) {
stats_.Update(1000, now_ms);
bitrate = stats_.Rate(now_ms);
@@ -215,7 +215,7 @@
// Window size should be full, and the single data point should be accepted.
++now_ms;
- absl::optional<uint32_t> bitrate = stats_.Rate(now_ms);
+ std::optional<uint32_t> bitrate = stats_.Rate(now_ms);
EXPECT_TRUE(static_cast<bool>(bitrate));
EXPECT_EQ(1000 * 8u, *bitrate);
@@ -241,7 +241,7 @@
stats_.Update(kWindowMs, now_ms);
now_ms += kWindowMs - 1;
stats_.Update(0, now_ms);
- absl::optional<uint32_t> bitrate = stats_.Rate(now_ms);
+ std::optional<uint32_t> bitrate = stats_.Rate(now_ms);
EXPECT_TRUE(static_cast<bool>(bitrate));
EXPECT_EQ(1000 * 8u, *bitrate);
@@ -264,7 +264,7 @@
stats_.Update(0, now_ms);
now_ms += kWindowMs - 1;
- absl::optional<uint32_t> bitrate = stats_.Rate(now_ms);
+ std::optional<uint32_t> bitrate = stats_.Rate(now_ms);
EXPECT_TRUE(static_cast<bool>(bitrate));
EXPECT_EQ(0u, *bitrate);
diff --git a/rtc_base/rtc_certificate_generator.cc b/rtc_base/rtc_certificate_generator.cc
index ffc51aa..a0c6f5b 100644
--- a/rtc_base/rtc_certificate_generator.cc
+++ b/rtc_base/rtc_certificate_generator.cc
@@ -32,7 +32,7 @@
// static
scoped_refptr<RTCCertificate> RTCCertificateGenerator::GenerateCertificate(
const KeyParams& key_params,
- const absl::optional<uint64_t>& expires_ms) {
+ const std::optional<uint64_t>& expires_ms) {
if (!key_params.IsValid()) {
return nullptr;
}
@@ -68,7 +68,7 @@
void RTCCertificateGenerator::GenerateCertificateAsync(
const KeyParams& key_params,
- const absl::optional<uint64_t>& expires_ms,
+ const std::optional<uint64_t>& expires_ms,
RTCCertificateGenerator::Callback callback) {
RTC_DCHECK(signaling_thread_->IsCurrent());
RTC_DCHECK(callback);
diff --git a/rtc_base/rtc_certificate_generator.h b/rtc_base/rtc_certificate_generator.h
index a881f1a..34c05b8 100644
--- a/rtc_base/rtc_certificate_generator.h
+++ b/rtc_base/rtc_certificate_generator.h
@@ -13,8 +13,9 @@
#include <stdint.h>
+#include <optional>
+
#include "absl/functional/any_invocable.h"
-#include "absl/types/optional.h"
#include "api/scoped_refptr.h"
#include "rtc_base/rtc_certificate.h"
#include "rtc_base/ssl_identity.h"
@@ -40,7 +41,7 @@
// its own restrictions on the expiration time.
virtual void GenerateCertificateAsync(
const KeyParams& key_params,
- const absl::optional<uint64_t>& expires_ms,
+ const std::optional<uint64_t>& expires_ms,
Callback callback) = 0;
};
@@ -58,7 +59,7 @@
// specified, a default expiration time is used.
static scoped_refptr<RTCCertificate> GenerateCertificate(
const KeyParams& key_params,
- const absl::optional<uint64_t>& expires_ms);
+ const std::optional<uint64_t>& expires_ms);
RTCCertificateGenerator(Thread* signaling_thread, Thread* worker_thread);
~RTCCertificateGenerator() override {}
@@ -69,7 +70,7 @@
// larger value than that is clamped down to a year. If `expires_ms` is not
// specified, a default expiration time is used.
void GenerateCertificateAsync(const KeyParams& key_params,
- const absl::optional<uint64_t>& expires_ms,
+ const std::optional<uint64_t>& expires_ms,
Callback callback) override;
private:
diff --git a/rtc_base/rtc_certificate_generator_unittest.cc b/rtc_base/rtc_certificate_generator_unittest.cc
index fb7ec91..84a15d4 100644
--- a/rtc_base/rtc_certificate_generator_unittest.cc
+++ b/rtc_base/rtc_certificate_generator_unittest.cc
@@ -11,8 +11,8 @@
#include "rtc_base/rtc_certificate_generator.h"
#include <memory>
+#include <optional>
-#include "absl/types/optional.h"
#include "api/make_ref_counted.h"
#include "rtc_base/checks.h"
#include "rtc_base/gunit.h"
@@ -73,18 +73,18 @@
TEST_F(RTCCertificateGeneratorTest, GenerateECDSA) {
EXPECT_TRUE(RTCCertificateGenerator::GenerateCertificate(KeyParams::ECDSA(),
- absl::nullopt));
+ std::nullopt));
}
TEST_F(RTCCertificateGeneratorTest, GenerateRSA) {
EXPECT_TRUE(RTCCertificateGenerator::GenerateCertificate(KeyParams::RSA(),
- absl::nullopt));
+ std::nullopt));
}
TEST_F(RTCCertificateGeneratorTest, GenerateAsyncECDSA) {
EXPECT_FALSE(fixture_.certificate());
fixture_.generator()->GenerateCertificateAsync(
- KeyParams::ECDSA(), absl::nullopt, fixture_.OnGenerated());
+ KeyParams::ECDSA(), std::nullopt, fixture_.OnGenerated());
// Until generation has completed, the certificate is null. Since this is an
// async call, generation must not have completed until we process messages
// posted to this thread (which is done by `EXPECT_TRUE_WAIT`).
@@ -126,9 +126,9 @@
EXPECT_FALSE(invalid_params.IsValid());
EXPECT_FALSE(RTCCertificateGenerator::GenerateCertificate(invalid_params,
- absl::nullopt));
+ std::nullopt));
- fixture_.generator()->GenerateCertificateAsync(invalid_params, absl::nullopt,
+ fixture_.generator()->GenerateCertificateAsync(invalid_params, std::nullopt,
fixture_.OnGenerated());
EXPECT_TRUE_WAIT(fixture_.GenerateAsyncCompleted(), kGenerationTimeoutMs);
EXPECT_FALSE(fixture_.certificate());
diff --git a/rtc_base/socket.h b/rtc_base/socket.h
index 275294e..aeb4ad7 100644
--- a/rtc_base/socket.h
+++ b/rtc_base/socket.h
@@ -13,7 +13,8 @@
#include <errno.h>
-#include "absl/types/optional.h"
+#include <optional>
+
#include "rtc_base/checks.h"
#if defined(WEBRTC_POSIX)
@@ -90,7 +91,7 @@
struct ReceiveBuffer {
ReceiveBuffer(Buffer& payload) : payload(payload) {}
- absl::optional<webrtc::Timestamp> arrival_time;
+ std::optional<webrtc::Timestamp> arrival_time;
SocketAddress source_address;
EcnMarking ecn = EcnMarking::kNotEct;
Buffer& payload;
diff --git a/rtc_base/string_encode.h b/rtc_base/string_encode.h
index 82a9dfd..a28559d 100644
--- a/rtc_base/string_encode.h
+++ b/rtc_base/string_encode.h
@@ -13,12 +13,12 @@
#include <stddef.h>
+#include <optional>
#include <string>
#include <type_traits>
#include <vector>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "rtc_base/checks.h"
#include "rtc_base/string_to_number.h"
@@ -91,7 +91,7 @@
int>::type = 0>
static bool FromString(absl::string_view s, T* t) {
RTC_DCHECK(t);
- absl::optional<T> result = StringToNumber<T>(s);
+ std::optional<T> result = StringToNumber<T>(s);
if (result)
*t = *result;
diff --git a/rtc_base/string_to_number.cc b/rtc_base/string_to_number.cc
index 1209ece..b02288b 100644
--- a/rtc_base/string_to_number.cc
+++ b/rtc_base/string_to_number.cc
@@ -20,9 +20,9 @@
namespace rtc {
namespace string_to_number_internal {
-absl::optional<signed_type> ParseSigned(absl::string_view str, int base) {
+std::optional<signed_type> ParseSigned(absl::string_view str, int base) {
if (str.empty())
- return absl::nullopt;
+ return std::nullopt;
if (isdigit(static_cast<unsigned char>(str[0])) || str[0] == '-') {
std::string str_str(str);
@@ -35,12 +35,12 @@
return value;
}
}
- return absl::nullopt;
+ return std::nullopt;
}
-absl::optional<unsigned_type> ParseUnsigned(absl::string_view str, int base) {
+std::optional<unsigned_type> ParseUnsigned(absl::string_view str, int base) {
if (str.empty())
- return absl::nullopt;
+ return std::nullopt;
if (isdigit(static_cast<unsigned char>(str[0])) || str[0] == '-') {
std::string str_str(str);
@@ -58,7 +58,7 @@
return value;
}
}
- return absl::nullopt;
+ return std::nullopt;
}
template <typename T>
@@ -80,12 +80,12 @@
}
template <typename T>
-absl::optional<T> ParseFloatingPoint(absl::string_view str) {
+std::optional<T> ParseFloatingPoint(absl::string_view str) {
if (str.empty())
- return absl::nullopt;
+ return std::nullopt;
if (str[0] == '\0')
- return absl::nullopt;
+ return std::nullopt;
std::string str_str(str);
char* end = nullptr;
errno = 0;
@@ -93,12 +93,12 @@
if (end == str_str.c_str() + str_str.size() && errno == 0) {
return value;
}
- return absl::nullopt;
+ return std::nullopt;
}
-template absl::optional<float> ParseFloatingPoint(absl::string_view str);
-template absl::optional<double> ParseFloatingPoint(absl::string_view str);
-template absl::optional<long double> ParseFloatingPoint(absl::string_view str);
+template std::optional<float> ParseFloatingPoint(absl::string_view str);
+template std::optional<double> ParseFloatingPoint(absl::string_view str);
+template std::optional<long double> ParseFloatingPoint(absl::string_view str);
} // namespace string_to_number_internal
} // namespace rtc
diff --git a/rtc_base/string_to_number.h b/rtc_base/string_to_number.h
index 1d704ee..5e8aea5 100644
--- a/rtc_base/string_to_number.h
+++ b/rtc_base/string_to_number.h
@@ -12,11 +12,11 @@
#define RTC_BASE_STRING_TO_NUMBER_H_
#include <limits>
+#include <optional>
#include <string>
#include <type_traits>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
namespace rtc {
@@ -27,7 +27,7 @@
// are disabled in WebRTC.
//
// Integers are parsed using:
-// absl::optional<int-type> StringToNumber(absl::string_view str,
+// std::optional<int-type> StringToNumber(absl::string_view str,
// int base = 10);
//
// These functions parse a value from the beginning of a string into one of the
@@ -44,16 +44,16 @@
using unsigned_type = unsigned long long; // NOLINT(runtime/int)
using signed_type = long long; // NOLINT(runtime/int)
-absl::optional<signed_type> ParseSigned(absl::string_view str, int base);
-absl::optional<unsigned_type> ParseUnsigned(absl::string_view str, int base);
+std::optional<signed_type> ParseSigned(absl::string_view str, int base);
+std::optional<unsigned_type> ParseUnsigned(absl::string_view str, int base);
template <typename T>
-absl::optional<T> ParseFloatingPoint(absl::string_view str);
+std::optional<T> ParseFloatingPoint(absl::string_view str);
} // namespace string_to_number_internal
template <typename T>
typename std::enable_if<std::is_integral<T>::value && std::is_signed<T>::value,
- absl::optional<T>>::type
+ std::optional<T>>::type
StringToNumber(absl::string_view str, int base = 10) {
using string_to_number_internal::signed_type;
static_assert(
@@ -62,36 +62,36 @@
std::numeric_limits<T>::lowest() >=
std::numeric_limits<signed_type>::lowest(),
"StringToNumber only supports signed integers as large as long long int");
- absl::optional<signed_type> value =
+ std::optional<signed_type> value =
string_to_number_internal::ParseSigned(str, base);
if (value && *value >= std::numeric_limits<T>::lowest() &&
*value <= std::numeric_limits<T>::max()) {
return static_cast<T>(*value);
}
- return absl::nullopt;
+ return std::nullopt;
}
template <typename T>
typename std::enable_if<std::is_integral<T>::value &&
std::is_unsigned<T>::value,
- absl::optional<T>>::type
+ std::optional<T>>::type
StringToNumber(absl::string_view str, int base = 10) {
using string_to_number_internal::unsigned_type;
static_assert(std::numeric_limits<T>::max() <=
std::numeric_limits<unsigned_type>::max(),
"StringToNumber only supports unsigned integers as large as "
"unsigned long long int");
- absl::optional<unsigned_type> value =
+ std::optional<unsigned_type> value =
string_to_number_internal::ParseUnsigned(str, base);
if (value && *value <= std::numeric_limits<T>::max()) {
return static_cast<T>(*value);
}
- return absl::nullopt;
+ return std::nullopt;
}
template <typename T>
typename std::enable_if<std::is_floating_point<T>::value,
- absl::optional<T>>::type
+ std::optional<T>>::type
StringToNumber(absl::string_view str, int base = 10) {
static_assert(
std::numeric_limits<T>::max() <= std::numeric_limits<long double>::max(),
diff --git a/rtc_base/string_to_number_unittest.cc b/rtc_base/string_to_number_unittest.cc
index edfdbf6..0c6164d 100644
--- a/rtc_base/string_to_number_unittest.cc
+++ b/rtc_base/string_to_number_unittest.cc
@@ -13,10 +13,10 @@
#include <stdint.h>
#include <limits>
+#include <optional>
#include <string>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "test/gtest.h"
namespace rtc {
@@ -71,10 +71,10 @@
(min_value == 0) ? "-2" : (std::to_string(min_value) + "1");
// Make the large value approximately ten times larger than the maximum.
const std::string too_large_string = std::to_string(max_value) + "1";
- EXPECT_EQ(absl::nullopt, StringToNumber<T>(too_low_string));
- EXPECT_EQ(absl::nullopt, StringToNumber<T>(too_low_string.c_str()));
- EXPECT_EQ(absl::nullopt, StringToNumber<T>(too_large_string));
- EXPECT_EQ(absl::nullopt, StringToNumber<T>(too_large_string.c_str()));
+ EXPECT_EQ(std::nullopt, StringToNumber<T>(too_low_string));
+ EXPECT_EQ(std::nullopt, StringToNumber<T>(too_low_string.c_str()));
+ EXPECT_EQ(std::nullopt, StringToNumber<T>(too_large_string));
+ EXPECT_EQ(std::nullopt, StringToNumber<T>(too_large_string.c_str()));
}
TYPED_TEST_P(BasicNumberTest, TestInvalidInputs) {
@@ -86,33 +86,33 @@
const char kBeginningEmbeddedNul[] = {'\0', '1', '2', '3', '4'};
const char kTrailingEmbeddedNul[] = {'1', '2', '3', '4', '\0'};
- EXPECT_EQ(absl::nullopt, StringToNumber<T>(kInvalidCharArray));
- EXPECT_EQ(absl::nullopt, StringToNumber<T>(std::string(kInvalidCharArray)));
- EXPECT_EQ(absl::nullopt, StringToNumber<T>(kPlusMinusCharArray));
- EXPECT_EQ(absl::nullopt, StringToNumber<T>(std::string(kPlusMinusCharArray)));
- EXPECT_EQ(absl::nullopt, StringToNumber<T>(kNumberFollowedByCruft));
- EXPECT_EQ(absl::nullopt,
+ EXPECT_EQ(std::nullopt, StringToNumber<T>(kInvalidCharArray));
+ EXPECT_EQ(std::nullopt, StringToNumber<T>(std::string(kInvalidCharArray)));
+ EXPECT_EQ(std::nullopt, StringToNumber<T>(kPlusMinusCharArray));
+ EXPECT_EQ(std::nullopt, StringToNumber<T>(std::string(kPlusMinusCharArray)));
+ EXPECT_EQ(std::nullopt, StringToNumber<T>(kNumberFollowedByCruft));
+ EXPECT_EQ(std::nullopt,
StringToNumber<T>(std::string(kNumberFollowedByCruft)));
- EXPECT_EQ(absl::nullopt, StringToNumber<T>(" 5"));
- EXPECT_EQ(absl::nullopt, StringToNumber<T>(" - 5"));
- EXPECT_EQ(absl::nullopt, StringToNumber<T>("- 5"));
- EXPECT_EQ(absl::nullopt, StringToNumber<T>(" -5"));
- EXPECT_EQ(absl::nullopt, StringToNumber<T>("5 "));
+ EXPECT_EQ(std::nullopt, StringToNumber<T>(" 5"));
+ EXPECT_EQ(std::nullopt, StringToNumber<T>(" - 5"));
+ EXPECT_EQ(std::nullopt, StringToNumber<T>("- 5"));
+ EXPECT_EQ(std::nullopt, StringToNumber<T>(" -5"));
+ EXPECT_EQ(std::nullopt, StringToNumber<T>("5 "));
// Test various types of empty inputs
- EXPECT_EQ(absl::nullopt, StringToNumber<T>({nullptr, 0}));
- EXPECT_EQ(absl::nullopt, StringToNumber<T>(""));
- EXPECT_EQ(absl::nullopt, StringToNumber<T>(std::string()));
- EXPECT_EQ(absl::nullopt, StringToNumber<T>(std::string("")));
- EXPECT_EQ(absl::nullopt, StringToNumber<T>(absl::string_view()));
- EXPECT_EQ(absl::nullopt, StringToNumber<T>(absl::string_view(nullptr, 0)));
- EXPECT_EQ(absl::nullopt, StringToNumber<T>(absl::string_view("")));
+ EXPECT_EQ(std::nullopt, StringToNumber<T>({nullptr, 0}));
+ EXPECT_EQ(std::nullopt, StringToNumber<T>(""));
+ EXPECT_EQ(std::nullopt, StringToNumber<T>(std::string()));
+ EXPECT_EQ(std::nullopt, StringToNumber<T>(std::string("")));
+ EXPECT_EQ(std::nullopt, StringToNumber<T>(absl::string_view()));
+ EXPECT_EQ(std::nullopt, StringToNumber<T>(absl::string_view(nullptr, 0)));
+ EXPECT_EQ(std::nullopt, StringToNumber<T>(absl::string_view("")));
// Test strings with embedded nuls.
- EXPECT_EQ(absl::nullopt, StringToNumber<T>(absl::string_view(
- kEmbeddedNul, sizeof(kEmbeddedNul))));
- EXPECT_EQ(absl::nullopt,
+ EXPECT_EQ(std::nullopt, StringToNumber<T>(absl::string_view(
+ kEmbeddedNul, sizeof(kEmbeddedNul))));
+ EXPECT_EQ(std::nullopt,
StringToNumber<T>(absl::string_view(
kBeginningEmbeddedNul, sizeof(kBeginningEmbeddedNul))));
- EXPECT_EQ(absl::nullopt,
+ EXPECT_EQ(std::nullopt,
StringToNumber<T>(absl::string_view(kTrailingEmbeddedNul,
sizeof(kTrailingEmbeddedNul))));
}
@@ -129,10 +129,10 @@
IntegerTypes);
TEST(StringToNumberTest, TestSpecificValues) {
- EXPECT_EQ(absl::nullopt, StringToNumber<uint8_t>("256"));
- EXPECT_EQ(absl::nullopt, StringToNumber<uint8_t>("-256"));
- EXPECT_EQ(absl::nullopt, StringToNumber<int8_t>("256"));
- EXPECT_EQ(absl::nullopt, StringToNumber<int8_t>("-256"));
+ EXPECT_EQ(std::nullopt, StringToNumber<uint8_t>("256"));
+ EXPECT_EQ(std::nullopt, StringToNumber<uint8_t>("-256"));
+ EXPECT_EQ(std::nullopt, StringToNumber<int8_t>("256"));
+ EXPECT_EQ(std::nullopt, StringToNumber<int8_t>("-256"));
}
} // namespace rtc
diff --git a/rtc_base/synchronization/mutex_pthread.h b/rtc_base/synchronization/mutex_pthread.h
index c749a20..a532bd2 100644
--- a/rtc_base/synchronization/mutex_pthread.h
+++ b/rtc_base/synchronization/mutex_pthread.h
@@ -76,8 +76,8 @@
}
private:
- // Use two separate primitive types, rather than absl::optional, since the
- // data race described below might invalidate absl::optional invariants.
+ // Use two separate primitive types, rather than std::optional, since the
+ // data race described below might invalidate std::optional invariants.
bool is_owned_ = false;
pthread_t latest_owner_ = pthread_self();
#endif
diff --git a/rtc_base/system/BUILD.gn b/rtc_base/system/BUILD.gn
index eef2754..b18114f 100644
--- a/rtc_base/system/BUILD.gn
+++ b/rtc_base/system/BUILD.gn
@@ -30,7 +30,6 @@
"..:criticalsection",
"..:safe_conversions",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
diff --git a/rtc_base/system/file_wrapper.cc b/rtc_base/system/file_wrapper.cc
index b768ad0..12c27a5 100644
--- a/rtc_base/system/file_wrapper.cc
+++ b/rtc_base/system/file_wrapper.cc
@@ -14,10 +14,10 @@
#include <cerrno>
#include <cstdint>
+#include <optional>
#include <string>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "rtc_base/checks.h"
#include "rtc_base/numerics/safe_conversions.h"
@@ -82,19 +82,19 @@
return fseek(file_, rtc::checked_cast<long>(position), SEEK_SET) == 0;
}
-absl::optional<size_t> FileWrapper::FileSize() {
+std::optional<size_t> FileWrapper::FileSize() {
if (file_ == nullptr)
- return absl::nullopt;
+ return std::nullopt;
long original_position = ftell(file_);
if (original_position < 0)
- return absl::nullopt;
+ return std::nullopt;
int seek_error = fseek(file_, 0, SEEK_END);
if (seek_error)
- return absl::nullopt;
+ return std::nullopt;
long file_size = ftell(file_);
seek_error = fseek(file_, original_position, SEEK_SET);
if (seek_error)
- return absl::nullopt;
+ return std::nullopt;
return rtc::checked_cast<size_t>(file_size);
}
diff --git a/rtc_base/system/file_wrapper.h b/rtc_base/system/file_wrapper.h
index 882393b..a1553ff 100644
--- a/rtc_base/system/file_wrapper.h
+++ b/rtc_base/system/file_wrapper.h
@@ -15,10 +15,10 @@
#include <stdint.h>
#include <stdio.h>
+#include <optional>
#include <string>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
// Implementation that can read (exclusive) or write from/to a file.
@@ -90,7 +90,7 @@
// Returns the file size or -1 if a size could not be determined.
// (A file size might not exists for non-seekable files or file-like
// objects, for example /dev/tty on unix.)
- absl::optional<size_t> FileSize();
+ std::optional<size_t> FileSize();
// Returns number of bytes read. Short count indicates EOF or error.
size_t Read(void* buf, size_t length);
diff --git a/rtc_base/task_queue_win.cc b/rtc_base/task_queue_win.cc
index 7e46d58..513ccd7 100644
--- a/rtc_base/task_queue_win.cc
+++ b/rtc_base/task_queue_win.cc
@@ -26,12 +26,12 @@
#include <algorithm>
#include <functional>
#include <memory>
+#include <optional>
#include <queue>
#include <utility>
#include "absl/functional/any_invocable.h"
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/task_queue/task_queue_base.h"
#include "api/units/time_delta.h"
#include "api/units/timestamp.h"
@@ -210,7 +210,7 @@
void TaskQueueWin::Delete() {
RTC_DCHECK(!IsCurrent());
- RTC_CHECK(thread_.GetHandle() != absl::nullopt);
+ RTC_CHECK(thread_.GetHandle() != std::nullopt);
while (
!::PostThreadMessage(GetThreadId(*thread_.GetHandle()), WM_QUIT, 0, 0)) {
RTC_CHECK_EQ(ERROR_NOT_ENOUGH_QUOTA, ::GetLastError());
@@ -239,7 +239,7 @@
}
auto* task_info = new DelayedTaskInfo(delay, std::move(task));
- RTC_CHECK(thread_.GetHandle() != absl::nullopt);
+ RTC_CHECK(thread_.GetHandle() != std::nullopt);
if (!::PostThreadMessage(GetThreadId(*thread_.GetHandle()),
WM_QUEUE_DELAYED_TASK, 0,
reinterpret_cast<LPARAM>(task_info))) {
diff --git a/rtc_base/task_utils/BUILD.gn b/rtc_base/task_utils/BUILD.gn
index 073d545..ee6abc3 100644
--- a/rtc_base/task_utils/BUILD.gn
+++ b/rtc_base/task_utils/BUILD.gn
@@ -41,7 +41,6 @@
"../../system_wrappers:system_wrappers",
"../../test:test_support",
"//third_party/abseil-cpp/absl/functional:any_invocable",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
}
diff --git a/rtc_base/task_utils/repeating_task_unittest.cc b/rtc_base/task_utils/repeating_task_unittest.cc
index de6cfdb..202a6f5 100644
--- a/rtc_base/task_utils/repeating_task_unittest.cc
+++ b/rtc_base/task_utils/repeating_task_unittest.cc
@@ -12,9 +12,9 @@
#include <atomic>
#include <memory>
+#include <optional>
#include "absl/functional/any_invocable.h"
-#include "absl/types/optional.h"
#include "api/task_queue/task_queue_base.h"
#include "api/task_queue/test/mock_task_queue_base.h"
#include "api/units/time_delta.h"
@@ -63,7 +63,7 @@
const PostTaskTraits& /*traits*/,
const Location& /*location*/) override {
last_task_ = std::move(task);
- last_precision_ = absl::nullopt;
+ last_precision_ = std::nullopt;
last_delay_ = TimeDelta::Zero();
}
@@ -95,7 +95,7 @@
return last_delay_;
}
- absl::optional<TaskQueueBase::DelayPrecision> last_precision() const {
+ std::optional<TaskQueueBase::DelayPrecision> last_precision() const {
return last_precision_;
}
@@ -104,7 +104,7 @@
SimulatedClock* clock_;
absl::AnyInvocable<void() &&> last_task_;
TimeDelta last_delay_ = TimeDelta::MinusInfinity();
- absl::optional<TaskQueueBase::DelayPrecision> last_precision_;
+ std::optional<TaskQueueBase::DelayPrecision> last_precision_;
};
// NOTE: Since this utility class holds a raw pointer to a variable that likely
diff --git a/rtc_base/test_client.cc b/rtc_base/test_client.cc
index 87c9465..3bcc509 100644
--- a/rtc_base/test_client.cc
+++ b/rtc_base/test_client.cc
@@ -113,7 +113,7 @@
}
bool TestClient::CheckTimestamp(
- absl::optional<webrtc::Timestamp> packet_timestamp) {
+ std::optional<webrtc::Timestamp> packet_timestamp) {
bool res = true;
if (!packet_timestamp) {
res = false;
diff --git a/rtc_base/test_client.h b/rtc_base/test_client.h
index 6fe6fd5..56c4b7b 100644
--- a/rtc_base/test_client.h
+++ b/rtc_base/test_client.h
@@ -34,7 +34,7 @@
SocketAddress addr;
Buffer buf;
- absl::optional<webrtc::Timestamp> packet_time;
+ std::optional<webrtc::Timestamp> packet_time;
};
// Default timeout for NextPacket reads.
@@ -98,7 +98,7 @@
void OnPacket(AsyncPacketSocket* socket,
const rtc::ReceivedPacket& received_packet);
void OnReadyToSend(AsyncPacketSocket* socket);
- bool CheckTimestamp(absl::optional<webrtc::Timestamp> packet_timestamp);
+ bool CheckTimestamp(std::optional<webrtc::Timestamp> packet_timestamp);
void AdvanceTime(int ms);
ThreadProcessingFakeClock* fake_clock_ = nullptr;
@@ -106,7 +106,7 @@
std::unique_ptr<AsyncPacketSocket> socket_;
std::vector<std::unique_ptr<Packet>> packets_;
int ready_to_send_count_ = 0;
- absl::optional<webrtc::Timestamp> prev_packet_timestamp_;
+ std::optional<webrtc::Timestamp> prev_packet_timestamp_;
};
} // namespace rtc
diff --git a/rtc_base/unique_id_generator.cc b/rtc_base/unique_id_generator.cc
index c80a843..b29f4c6 100644
--- a/rtc_base/unique_id_generator.cc
+++ b/rtc_base/unique_id_generator.cc
@@ -59,7 +59,7 @@
bool UniqueStringGenerator::AddKnownId(absl::string_view value) {
// TODO(webrtc:13579): remove string copy here once absl::string_view version
// of StringToNumber is available.
- absl::optional<uint32_t> int_value =
+ std::optional<uint32_t> int_value =
StringToNumber<uint32_t>(std::string(value));
// The underlying generator works for uint32_t values, so if the provided
// value is not a uint32_t it will never be generated anyway.
diff --git a/rtc_base/virtual_socket_server.cc b/rtc_base/virtual_socket_server.cc
index bec7d97..dd02281 100644
--- a/rtc_base/virtual_socket_server.cc
+++ b/rtc_base/virtual_socket_server.cc
@@ -160,7 +160,7 @@
for (const SocketAddress& remote_addr : *listen_queue_) {
server->Disconnect(remote_addr);
}
- listen_queue_ = absl::nullopt;
+ listen_queue_ = std::nullopt;
}
// Cancel potential connects
diff --git a/rtc_base/virtual_socket_server.h b/rtc_base/virtual_socket_server.h
index 93ef288..7704abf 100644
--- a/rtc_base/virtual_socket_server.h
+++ b/rtc_base/virtual_socket_server.h
@@ -13,9 +13,9 @@
#include <deque>
#include <map>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/make_ref_counted.h"
#include "api/ref_counted_base.h"
#include "api/scoped_refptr.h"
@@ -151,7 +151,7 @@
std::list<std::unique_ptr<Packet>> recv_buffer_ RTC_GUARDED_BY(mutex_);
// Pending sockets which can be Accepted
- absl::optional<std::deque<SocketAddress>> listen_queue_
+ std::optional<std::deque<SocketAddress>> listen_queue_
RTC_GUARDED_BY(mutex_);
};
diff --git a/rtc_tools/BUILD.gn b/rtc_tools/BUILD.gn
index 50a42f6..bfd8085 100644
--- a/rtc_tools/BUILD.gn
+++ b/rtc_tools/BUILD.gn
@@ -63,7 +63,6 @@
"../rtc_base:refcount",
"../rtc_base:stringutils",
"//third_party/abseil-cpp/absl/strings",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -109,7 +108,6 @@
"../common_video",
"../rtc_base:checks",
"../rtc_base:logging",
- "//third_party/abseil-cpp/absl/types:optional",
"//third_party/libyuv",
]
}
@@ -411,7 +409,6 @@
"//third_party/abseil-cpp/absl/base:core_headers",
"//third_party/abseil-cpp/absl/functional:bind_front",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -444,7 +441,6 @@
"//rtc_base/system:file_wrapper",
"//test:fileutils",
"//test:test_support",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
}
@@ -512,7 +508,6 @@
"//third_party/abseil-cpp/absl/flags:parse",
"//third_party/abseil-cpp/absl/flags:usage",
"//third_party/abseil-cpp/absl/strings",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
diff --git a/rtc_tools/frame_analyzer/linear_least_squares.h b/rtc_tools/frame_analyzer/linear_least_squares.h
index 7006db1..5b5a783 100644
--- a/rtc_tools/frame_analyzer/linear_least_squares.h
+++ b/rtc_tools/frame_analyzer/linear_least_squares.h
@@ -13,11 +13,10 @@
#include <stdint.h>
+#include <optional>
#include <valarray>
#include <vector>
-#include "absl/types/optional.h"
-
namespace webrtc {
namespace test {
@@ -45,9 +44,9 @@
private:
// Running sum of x^T * x.
- absl::optional<std::valarray<std::valarray<uint64_t>>> sum_xx;
+ std::optional<std::valarray<std::valarray<uint64_t>>> sum_xx;
// Running sum of x^T * y.
- absl::optional<std::valarray<std::valarray<uint64_t>>> sum_xy;
+ std::optional<std::valarray<std::valarray<uint64_t>>> sum_xy;
};
} // namespace test
diff --git a/rtc_tools/network_tester/BUILD.gn b/rtc_tools/network_tester/BUILD.gn
index ab622cc..995c3d0 100644
--- a/rtc_tools/network_tester/BUILD.gn
+++ b/rtc_tools/network_tester/BUILD.gn
@@ -60,7 +60,6 @@
"../../rtc_base/synchronization:mutex",
"../../rtc_base/system:no_unique_address",
"//third_party/abseil-cpp/absl/functional:any_invocable",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
diff --git a/rtc_tools/network_tester/config_reader.cc b/rtc_tools/network_tester/config_reader.cc
index 16ae458..5ee1676 100644
--- a/rtc_tools/network_tester/config_reader.cc
+++ b/rtc_tools/network_tester/config_reader.cc
@@ -33,10 +33,10 @@
ConfigReader::~ConfigReader() = default;
-absl::optional<ConfigReader::Config> ConfigReader::GetNextConfig() {
+std::optional<ConfigReader::Config> ConfigReader::GetNextConfig() {
#ifdef WEBRTC_NETWORK_TESTER_PROTO
if (proto_config_index_ >= proto_all_configs_.configs_size())
- return absl::nullopt;
+ return std::nullopt;
auto proto_config = proto_all_configs_.configs(proto_config_index_++);
RTC_DCHECK(proto_config.has_packet_send_interval_ms());
RTC_DCHECK(proto_config.has_packet_size());
@@ -47,7 +47,7 @@
config.execution_time_ms = proto_config.execution_time_ms();
return config;
#else
- return absl::nullopt;
+ return std::nullopt;
#endif // WEBRTC_NETWORK_TESTER_PROTO
}
diff --git a/rtc_tools/network_tester/config_reader.h b/rtc_tools/network_tester/config_reader.h
index 3903613..936da82 100644
--- a/rtc_tools/network_tester/config_reader.h
+++ b/rtc_tools/network_tester/config_reader.h
@@ -12,10 +12,9 @@
#define RTC_TOOLS_NETWORK_TESTER_CONFIG_READER_H_
#include <fstream>
+#include <optional>
#include <string>
-#include "absl/types/optional.h"
-
#ifdef WEBRTC_NETWORK_TESTER_PROTO
#include "rtc_tools/network_tester/network_tester_config.pb.h"
using webrtc::network_tester::config::NetworkTesterAllConfigs;
@@ -38,7 +37,7 @@
ConfigReader(const ConfigReader&) = delete;
ConfigReader& operator=(const ConfigReader&) = delete;
- absl::optional<Config> GetNextConfig();
+ std::optional<Config> GetNextConfig();
private:
NetworkTesterAllConfigs proto_all_configs_;
diff --git a/rtc_tools/network_tester/packet_sender.cc b/rtc_tools/network_tester/packet_sender.cc
index c991737..eed185b 100644
--- a/rtc_tools/network_tester/packet_sender.cc
+++ b/rtc_tools/network_tester/packet_sender.cc
@@ -54,7 +54,7 @@
if (!task_safety_flag->alive()) {
return;
}
- if (absl::optional<ConfigReader::Config> config =
+ if (std::optional<ConfigReader::Config> config =
config_reader->GetNextConfig()) {
packet_sender->UpdateTestSetting(config->packet_size,
config->packet_send_interval_ms);
diff --git a/rtc_tools/network_tester/test_controller.cc b/rtc_tools/network_tester/test_controller.cc
index f8641aa..7b948df 100644
--- a/rtc_tools/network_tester/test_controller.cc
+++ b/rtc_tools/network_tester/test_controller.cc
@@ -11,8 +11,8 @@
#include "rtc_tools/network_tester/test_controller.h"
#include <limits>
+#include <optional>
-#include "absl/types/optional.h"
#include "api/units/timestamp.h"
#include "rtc_base/checks.h"
#include "rtc_base/internal/default_socket_server.h"
@@ -64,14 +64,14 @@
remote_address_ = rtc::SocketAddress(hostname, port);
NetworkTesterPacket packet;
packet.set_type(NetworkTesterPacket::HAND_SHAKING);
- SendData(packet, absl::nullopt);
+ SendData(packet, std::nullopt);
MutexLock scoped_lock(&test_done_lock_);
local_test_done_ = false;
remote_test_done_ = false;
}
void TestController::SendData(const NetworkTesterPacket& packet,
- absl::optional<size_t> data_size) {
+ std::optional<size_t> data_size) {
if (!packet_sender_thread_->IsCurrent()) {
packet_sender_thread_->PostTask(SafeTask(
task_safety_flag_,
@@ -96,7 +96,7 @@
RTC_DCHECK_RUN_ON(packet_sender_thread_.get());
NetworkTesterPacket packet;
packet.set_type(NetworkTesterPacket::TEST_DONE);
- SendData(packet, absl::nullopt);
+ SendData(packet, std::nullopt);
MutexLock scoped_lock(&test_done_lock_);
local_test_done_ = true;
}
@@ -123,7 +123,7 @@
NetworkTesterPacket packet;
packet.set_type(NetworkTesterPacket::TEST_START);
remote_address_ = received_packet.source_address();
- SendData(packet, absl::nullopt);
+ SendData(packet, std::nullopt);
packet_sender_.reset(new PacketSender(this, packet_sender_thread_.get(),
task_safety_flag_,
config_file_path_));
diff --git a/rtc_tools/network_tester/test_controller.h b/rtc_tools/network_tester/test_controller.h
index 423eb08..ce4d793 100644
--- a/rtc_tools/network_tester/test_controller.h
+++ b/rtc_tools/network_tester/test_controller.h
@@ -16,9 +16,9 @@
#include <array>
#include <memory>
+#include <optional>
#include <string>
-#include "absl/types/optional.h"
#include "api/sequence_checker.h"
#include "p2p/base/basic_packet_socket_factory.h"
#include "rtc_base/async_packet_socket.h"
@@ -57,7 +57,7 @@
void SendConnectTo(const std::string& hostname, int port);
void SendData(const NetworkTesterPacket& packet,
- absl::optional<size_t> data_size);
+ std::optional<size_t> data_size);
void OnTestDone();
diff --git a/rtc_tools/rtc_event_log_visualizer/alerts.cc b/rtc_tools/rtc_event_log_visualizer/alerts.cc
index 98ffbd3..9137d0b 100644
--- a/rtc_tools/rtc_event_log_visualizer/alerts.cc
+++ b/rtc_tools/rtc_event_log_visualizer/alerts.cc
@@ -17,9 +17,9 @@
#include <cstdlib>
#include <functional>
#include <map>
+#include <optional>
#include <string>
-#include "absl/types/optional.h"
#include "api/units/timestamp.h"
#include "logging/rtc_event_log/events/logged_rtp_rtcp.h"
#include "logging/rtc_event_log/rtc_event_log_parser.h"
@@ -132,7 +132,7 @@
for (const LoggedRtpPacket& rtp_packet : stream.packet_view)
rtp_in_direction.emplace(rtp_packet.log_time_us(), &rtp_packet);
}
- absl::optional<int64_t> last_rtp_time;
+ std::optional<int64_t> last_rtp_time;
for (const auto& kv : rtp_in_direction) {
int64_t timestamp = kv.first;
if (timestamp > segment_end_us) {
@@ -148,7 +148,7 @@
last_rtp_time.emplace(timestamp);
}
- absl::optional<int64_t> last_rtcp_time;
+ std::optional<int64_t> last_rtcp_time;
if (direction == kIncomingPacket) {
for (const auto& rtcp : parsed_log.incoming_rtcp_packets()) {
if (rtcp.log_time_us() > segment_end_us) {
diff --git a/rtc_tools/rtc_event_log_visualizer/analyze_audio.cc b/rtc_tools/rtc_event_log_visualizer/analyze_audio.cc
index cba8998..4fbe9a6 100644
--- a/rtc_tools/rtc_event_log_visualizer/analyze_audio.cc
+++ b/rtc_tools/rtc_event_log_visualizer/analyze_audio.cc
@@ -13,13 +13,13 @@
#include <cstdint>
#include <map>
#include <memory>
+#include <optional>
#include <set>
#include <string>
#include <utility>
#include <vector>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/audio_codecs/audio_codec_pair_id.h"
#include "api/audio_codecs/audio_decoder.h"
#include "api/audio_codecs/audio_decoder_factory.h"
@@ -53,11 +53,11 @@
TimeSeries time_series("Audio encoder target bitrate", LineStyle::kLine,
PointStyle::kHighlight);
auto GetAnaBitrateBps = [](const LoggedAudioNetworkAdaptationEvent& ana_event)
- -> absl::optional<float> {
+ -> std::optional<float> {
if (ana_event.config.bitrate_bps)
- return absl::optional<float>(
+ return std::optional<float>(
static_cast<float>(*ana_event.config.bitrate_bps));
- return absl::nullopt;
+ return std::nullopt;
};
auto ToCallTime = [config](const LoggedAudioNetworkAdaptationEvent& packet) {
return config.GetCallTimeSec(packet.log_time());
@@ -80,9 +80,9 @@
auto GetAnaFrameLengthMs =
[](const LoggedAudioNetworkAdaptationEvent& ana_event) {
if (ana_event.config.frame_length_ms)
- return absl::optional<float>(
+ return std::optional<float>(
static_cast<float>(*ana_event.config.frame_length_ms));
- return absl::optional<float>();
+ return std::optional<float>();
};
auto ToCallTime = [config](const LoggedAudioNetworkAdaptationEvent& packet) {
return config.GetCallTimeSec(packet.log_time());
@@ -105,9 +105,9 @@
auto GetAnaPacketLoss =
[](const LoggedAudioNetworkAdaptationEvent& ana_event) {
if (ana_event.config.uplink_packet_loss_fraction)
- return absl::optional<float>(static_cast<float>(
+ return std::optional<float>(static_cast<float>(
*ana_event.config.uplink_packet_loss_fraction));
- return absl::optional<float>();
+ return std::optional<float>();
};
auto ToCallTime = [config](const LoggedAudioNetworkAdaptationEvent& packet) {
return config.GetCallTimeSec(packet.log_time());
@@ -131,9 +131,9 @@
auto GetAnaFecEnabled =
[](const LoggedAudioNetworkAdaptationEvent& ana_event) {
if (ana_event.config.enable_fec)
- return absl::optional<float>(
+ return std::optional<float>(
static_cast<float>(*ana_event.config.enable_fec));
- return absl::optional<float>();
+ return std::optional<float>();
};
auto ToCallTime = [config](const LoggedAudioNetworkAdaptationEvent& packet) {
return config.GetCallTimeSec(packet.log_time());
@@ -156,9 +156,9 @@
auto GetAnaDtxEnabled =
[](const LoggedAudioNetworkAdaptationEvent& ana_event) {
if (ana_event.config.enable_dtx)
- return absl::optional<float>(
+ return std::optional<float>(
static_cast<float>(*ana_event.config.enable_dtx));
- return absl::optional<float>();
+ return std::optional<float>();
};
auto ToCallTime = [config](const LoggedAudioNetworkAdaptationEvent& packet) {
return config.GetCallTimeSec(packet.log_time());
@@ -181,9 +181,9 @@
auto GetAnaNumChannels =
[](const LoggedAudioNetworkAdaptationEvent& ana_event) {
if (ana_event.config.num_channels)
- return absl::optional<float>(
+ return std::optional<float>(
static_cast<float>(*ana_event.config.num_channels));
- return absl::optional<float>();
+ return std::optional<float>();
};
auto ToCallTime = [config](const LoggedAudioNetworkAdaptationEvent& packet) {
return config.GetCallTimeSec(packet.log_time());
@@ -221,7 +221,7 @@
std::unique_ptr<AudioDecoder> MakeAudioDecoder(
const SdpAudioFormat& format,
- absl::optional<AudioCodecPairId> codec_pair_id) override {
+ std::optional<AudioCodecPairId> codec_pair_id) override {
auto replacement_file = std::make_unique<test::ResampleInputAudioFile>(
replacement_file_name_, file_sample_rate_hz_);
replacement_file->set_output_rate_hz(48000);
diff --git a/rtc_tools/rtc_event_log_visualizer/analyzer.cc b/rtc_tools/rtc_event_log_visualizer/analyzer.cc
index 62868ff8..d481f3a 100644
--- a/rtc_tools/rtc_event_log_visualizer/analyzer.cc
+++ b/rtc_tools/rtc_event_log_visualizer/analyzer.cc
@@ -18,6 +18,7 @@
#include <limits>
#include <map>
#include <memory>
+#include <optional>
#include <string>
#include <tuple>
#include <unordered_set>
@@ -27,7 +28,6 @@
#include "absl/algorithm/container.h"
#include "absl/functional/bind_front.h"
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/candidate.h"
#include "api/dtls_transport_interface.h"
#include "api/environment/environment_factory.h"
@@ -136,7 +136,7 @@
// This is much more reliable for outgoing streams than for incoming streams.
template <typename RtpPacketContainer>
-absl::optional<uint32_t> EstimateRtpClockFrequency(
+std::optional<uint32_t> EstimateRtpClockFrequency(
const RtpPacketContainer& packets,
int64_t end_time_us) {
RTC_CHECK(packets.size() >= 2);
@@ -157,7 +157,7 @@
<< "Failed to estimate RTP clock frequency: Stream too short. ("
<< packets.size() << " packets, "
<< last_log_timestamp - first_log_timestamp << " us)";
- return absl::nullopt;
+ return std::nullopt;
}
double duration =
static_cast<double>(last_log_timestamp - first_log_timestamp) /
@@ -174,10 +174,10 @@
<< " not close to any standard RTP frequency."
<< " Last timestamp " << last_rtp_timestamp
<< " first timestamp " << first_rtp_timestamp;
- return absl::nullopt;
+ return std::nullopt;
}
-absl::optional<double> NetworkDelayDiff_AbsSendTime(
+std::optional<double> NetworkDelayDiff_AbsSendTime(
const LoggedRtpPacketIncoming& old_packet,
const LoggedRtpPacketIncoming& new_packet) {
if (old_packet.rtp.header.extension.hasAbsoluteSendTime &&
@@ -191,11 +191,11 @@
recv_time_diff - AbsSendTimeToMicroseconds(send_time_diff);
return delay_change_us / 1000;
} else {
- return absl::nullopt;
+ return std::nullopt;
}
}
-absl::optional<double> NetworkDelayDiff_CaptureTime(
+std::optional<double> NetworkDelayDiff_CaptureTime(
const LoggedRtpPacketIncoming& old_packet,
const LoggedRtpPacketIncoming& new_packet,
const double sample_rate) {
@@ -716,7 +716,7 @@
TimeSeries time_series(GetStreamName(parsed_log_, direction, stream.ssrc),
LineStyle::kBar);
auto GetPacketSize = [](const LoggedRtpPacket& packet) {
- return absl::optional<float>(packet.total_length);
+ return std::optional<float>(packet.total_length);
};
auto ToCallTime = [this](const LoggedRtpPacket& packet) {
return this->config_.GetCallTimeSec(packet.timestamp);
@@ -902,7 +902,7 @@
uint32_t ssrc = playout_stream.first;
if (!MatchingSsrc(ssrc, desired_ssrc_))
continue;
- absl::optional<int64_t> last_playout_ms;
+ std::optional<int64_t> last_playout_ms;
TimeSeries time_series(SsrcToString(ssrc), LineStyle::kBar);
for (const auto& playout_event : playout_stream.second) {
float x = config_.GetCallTimeSec(playout_event.log_time());
@@ -1084,7 +1084,7 @@
continue;
}
int64_t segment_end_us = parsed_log_.first_log_segment().stop_time_us();
- absl::optional<uint32_t> estimated_frequency =
+ std::optional<uint32_t> estimated_frequency =
EstimateRtpClockFrequency(packets, segment_end_us);
if (!estimated_frequency)
continue;
@@ -1462,7 +1462,7 @@
std::map<LayerDescription, TimeSeries> time_series;
const auto& xr_list = parsed_log_.extended_reports(direction);
for (const auto& rtcp : xr_list) {
- const absl::optional<rtcp::TargetBitrate>& target_bitrate =
+ const std::optional<rtcp::TargetBitrate>& target_bitrate =
rtcp.xr.target_bitrate();
if (!target_bitrate.has_value())
continue;
@@ -1795,20 +1795,20 @@
raw_acked_bitrate.Update(packet.sent_packet.size.bytes(),
packet.receive_time.ms());
}
- absl::optional<uint32_t> raw_bitrate_bps =
+ std::optional<uint32_t> raw_bitrate_bps =
raw_acked_bitrate.Rate(feedback.back().receive_time.ms());
float x = config_.GetCallTimeSec(clock.CurrentTime());
if (raw_bitrate_bps) {
float y = raw_bitrate_bps.value() / 1000;
acked_time_series.points.emplace_back(x, y);
}
- absl::optional<DataRate> robust_estimate =
+ std::optional<DataRate> robust_estimate =
robust_throughput_estimator->bitrate();
if (robust_estimate) {
float y = robust_estimate.value().kbps();
robust_time_series.points.emplace_back(x, y);
}
- absl::optional<DataRate> acked_estimate =
+ std::optional<DataRate> acked_estimate =
acknowledged_bitrate_estimator->bitrate();
if (acked_estimate) {
float y = acked_estimate.value().kbps();
@@ -1905,7 +1905,7 @@
rscc.OnReceivedPacket(rtp_packet, MediaType::VIDEO);
int64_t arrival_time_ms = packet.rtp.log_time().ms();
acked_bitrate.Update(packet.rtp.total_length, arrival_time_ms);
- absl::optional<uint32_t> bitrate_bps = acked_bitrate.Rate(arrival_time_ms);
+ std::optional<uint32_t> bitrate_bps = acked_bitrate.Rate(arrival_time_ms);
if (bitrate_bps) {
uint32_t y = *bitrate_bps / 1000;
float x = config_.GetCallTimeSec(clock.CurrentTime());
@@ -1987,7 +1987,7 @@
continue;
}
int64_t segment_end_us = parsed_log_.first_log_segment().stop_time_us();
- absl::optional<uint32_t> estimated_frequency =
+ std::optional<uint32_t> estimated_frequency =
EstimateRtpClockFrequency(packets, segment_end_us);
if (!estimated_frequency)
continue;
diff --git a/rtc_tools/rtc_event_log_visualizer/analyzer_bindings_unittest.cc b/rtc_tools/rtc_event_log_visualizer/analyzer_bindings_unittest.cc
index 5d61845..cb9f76a 100644
--- a/rtc_tools/rtc_event_log_visualizer/analyzer_bindings_unittest.cc
+++ b/rtc_tools/rtc_event_log_visualizer/analyzer_bindings_unittest.cc
@@ -13,10 +13,10 @@
#include <cstddef>
#include <cstdint>
#include <cstring>
+#include <optional>
#include <string>
#include <vector>
-#include "absl/types/optional.h"
#include "rtc_base/protobuf_utils.h"
#include "rtc_base/system/file_wrapper.h"
#include "test/gtest.h"
@@ -36,7 +36,7 @@
webrtc::FileWrapper file = webrtc::FileWrapper::OpenReadOnly(file_name);
ASSERT_TRUE(file.is_open());
- absl::optional<size_t> file_size = file.FileSize();
+ std::optional<size_t> file_size = file.FileSize();
ASSERT_TRUE(file_size.has_value());
constexpr size_t kMaxFileSize = 1'000'000;
ASSERT_GT(*file_size, 0u);
diff --git a/rtc_tools/rtc_event_log_visualizer/analyzer_common.h b/rtc_tools/rtc_event_log_visualizer/analyzer_common.h
index 5f778cb..1513ace 100644
--- a/rtc_tools/rtc_event_log_visualizer/analyzer_common.h
+++ b/rtc_tools/rtc_event_log_visualizer/analyzer_common.h
@@ -13,9 +13,9 @@
#include <cstddef>
#include <cstdint>
+#include <optional>
#include <string>
-#include "absl/types/optional.h"
#include "api/function_view.h"
#include "api/units/time_delta.h"
#include "api/units/timestamp.h"
@@ -104,13 +104,13 @@
// store the result in a TimeSeries.
template <typename DataType, typename IterableType>
void ProcessPoints(rtc::FunctionView<float(const DataType&)> fx,
- rtc::FunctionView<absl::optional<float>(const DataType&)> fy,
+ rtc::FunctionView<std::optional<float>(const DataType&)> fy,
const IterableType& data_view,
TimeSeries* result) {
for (size_t i = 0; i < data_view.size(); i++) {
const DataType& elem = data_view[i];
float x = fx(elem);
- absl::optional<float> y = fy(elem);
+ std::optional<float> y = fy(elem);
if (y)
result->points.emplace_back(x, *y);
}
@@ -122,13 +122,13 @@
template <typename DataType, typename ResultType, typename IterableType>
void ProcessPairs(
rtc::FunctionView<float(const DataType&)> fx,
- rtc::FunctionView<absl::optional<ResultType>(const DataType&,
- const DataType&)> fy,
+ rtc::FunctionView<std::optional<ResultType>(const DataType&,
+ const DataType&)> fy,
const IterableType& data,
TimeSeries* result) {
for (size_t i = 1; i < data.size(); i++) {
float x = fx(data[i]);
- absl::optional<ResultType> y = fy(data[i - 1], data[i]);
+ std::optional<ResultType> y = fy(data[i - 1], data[i]);
if (y)
result->points.emplace_back(x, static_cast<float>(*y));
}
@@ -140,14 +140,14 @@
template <typename DataType, typename ResultType, typename IterableType>
void AccumulatePairs(
rtc::FunctionView<float(const DataType&)> fx,
- rtc::FunctionView<absl::optional<ResultType>(const DataType&,
- const DataType&)> fy,
+ rtc::FunctionView<std::optional<ResultType>(const DataType&,
+ const DataType&)> fy,
const IterableType& data,
TimeSeries* result) {
ResultType sum = 0;
for (size_t i = 1; i < data.size(); i++) {
float x = fx(data[i]);
- absl::optional<ResultType> y = fy(data[i - 1], data[i]);
+ std::optional<ResultType> y = fy(data[i - 1], data[i]);
if (y) {
sum += *y;
result->points.emplace_back(x, static_cast<float>(sum));
@@ -161,7 +161,7 @@
// during the preceding `window_duration_us` microseconds.
template <typename DataType, typename ResultType, typename IterableType>
void MovingAverage(
- rtc::FunctionView<absl::optional<ResultType>(const DataType&)> fy,
+ rtc::FunctionView<std::optional<ResultType>(const DataType&)> fy,
const IterableType& data_view,
AnalyzerConfig config,
TimeSeries* result) {
@@ -173,7 +173,7 @@
t += config.step_) {
while (window_index_end < data_view.size() &&
data_view[window_index_end].log_time() < t) {
- absl::optional<ResultType> value = fy(data_view[window_index_end]);
+ std::optional<ResultType> value = fy(data_view[window_index_end]);
if (value)
sum_in_window += *value;
++window_index_end;
@@ -181,7 +181,7 @@
while (window_index_begin < data_view.size() &&
data_view[window_index_begin].log_time() <
t - config.window_duration_) {
- absl::optional<ResultType> value = fy(data_view[window_index_begin]);
+ std::optional<ResultType> value = fy(data_view[window_index_begin]);
if (value)
sum_in_window -= *value;
++window_index_begin;
diff --git a/rtc_tools/rtc_event_log_visualizer/main.cc b/rtc_tools/rtc_event_log_visualizer/main.cc
index feded78..5ddaff8 100644
--- a/rtc_tools/rtc_event_log_visualizer/main.cc
+++ b/rtc_tools/rtc_event_log_visualizer/main.cc
@@ -16,6 +16,7 @@
#include <iostream>
#include <map>
#include <memory>
+#include <optional>
#include <string>
#include <utility>
#include <vector>
@@ -27,7 +28,6 @@
#include "absl/flags/usage_config.h"
#include "absl/strings/match.h"
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/neteq/neteq.h"
#include "api/units/time_delta.h"
#include "logging/rtc_event_log/rtc_event_log_parser.h"
@@ -324,7 +324,7 @@
// * cache the simulation results between different plots
// * open and read files
// * dont have a 1-to-1 mapping between IDs and charts.
- absl::optional<webrtc::NetEqStatsGetterMap> neteq_stats;
+ std::optional<webrtc::NetEqStatsGetterMap> neteq_stats;
if (absl::c_find(plot_names, "simulated_neteq_expand_rate") !=
plot_names.end()) {
if (!neteq_stats) {
diff --git a/rtc_tools/rtc_event_log_visualizer/plot_base.h b/rtc_tools/rtc_event_log_visualizer/plot_base.h
index aacffa4..722a79d 100644
--- a/rtc_tools/rtc_event_log_visualizer/plot_base.h
+++ b/rtc_tools/rtc_event_log_visualizer/plot_base.h
@@ -12,13 +12,13 @@
#include <cstdint>
#include <memory>
+#include <optional>
#include <string>
#include <utility>
#include <vector>
#include "absl/base/attributes.h"
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
// Generated at build-time by the protobuf compiler.
#include "rtc_tools/rtc_event_log_visualizer/proto/chart.pb.h"
@@ -213,7 +213,7 @@
protected:
std::vector<std::unique_ptr<Plot>> plots_;
- absl::optional<int64_t> calltime_to_utc_ms_;
+ std::optional<int64_t> calltime_to_utc_ms_;
};
} // namespace webrtc
diff --git a/rtc_tools/rtp_generator/main.cc b/rtc_tools/rtp_generator/main.cc
index df49576..7c4fcdf 100644
--- a/rtc_tools/rtp_generator/main.cc
+++ b/rtc_tools/rtp_generator/main.cc
@@ -35,7 +35,7 @@
return EXIT_FAILURE;
}
- absl::optional<webrtc::RtpGeneratorOptions> options =
+ std::optional<webrtc::RtpGeneratorOptions> options =
webrtc::ParseRtpGeneratorOptionsFromFile(config_path);
if (!options.has_value()) {
return EXIT_FAILURE;
diff --git a/rtc_tools/rtp_generator/rtp_generator.cc b/rtc_tools/rtp_generator/rtp_generator.cc
index ade169a..66ab032 100644
--- a/rtc_tools/rtp_generator/rtp_generator.cc
+++ b/rtc_tools/rtp_generator/rtp_generator.cc
@@ -69,7 +69,7 @@
}
// Creates a single VideoSendStream configuration.
-absl::optional<RtpGeneratorOptions::VideoSendStreamConfig>
+std::optional<RtpGeneratorOptions::VideoSendStreamConfig>
ParseVideoSendStreamConfig(const Json::Value& json) {
RtpGeneratorOptions::VideoSendStreamConfig config;
@@ -100,16 +100,16 @@
Json::Value rtp_json;
if (!rtc::GetValueFromJsonObject(json, "rtp", &rtp_json)) {
RTC_LOG(LS_ERROR) << "video_streams must have an rtp section";
- return absl::nullopt;
+ return std::nullopt;
}
if (!rtc::GetStringFromJsonObject(rtp_json, "payload_name",
&config.rtp.payload_name)) {
RTC_LOG(LS_ERROR) << "rtp.payload_name must be specified";
- return absl::nullopt;
+ return std::nullopt;
}
if (!IsValidCodecType(config.rtp.payload_name)) {
RTC_LOG(LS_ERROR) << "rtp.payload_name must be VP8,VP9 or H264";
- return absl::nullopt;
+ return std::nullopt;
}
config.rtp.payload_type =
@@ -126,11 +126,11 @@
} // namespace
-absl::optional<RtpGeneratorOptions> ParseRtpGeneratorOptionsFromFile(
+std::optional<RtpGeneratorOptions> ParseRtpGeneratorOptionsFromFile(
const std::string& options_file) {
if (!test::FileExists(options_file)) {
RTC_LOG(LS_ERROR) << " configuration file does not exist";
- return absl::nullopt;
+ return std::nullopt;
}
// Read the configuration file from disk.
@@ -140,7 +140,7 @@
config_file.Read(raw_json_buffer.data(), raw_json_buffer.size() - 1);
if (bytes_read == 0) {
RTC_LOG(LS_ERROR) << "Unable to read the configuration file.";
- return absl::nullopt;
+ return std::nullopt;
}
// Parse the file as JSON
@@ -153,16 +153,16 @@
&json, &error_message)) {
RTC_LOG(LS_ERROR) << "Unable to parse the corpus config json file. Error:"
<< error_message;
- return absl::nullopt;
+ return std::nullopt;
}
RtpGeneratorOptions gen_options;
for (const auto& video_stream_json : json["video_streams"]) {
- absl::optional<RtpGeneratorOptions::VideoSendStreamConfig>
+ std::optional<RtpGeneratorOptions::VideoSendStreamConfig>
video_stream_config = ParseVideoSendStreamConfig(video_stream_json);
if (!video_stream_config.has_value()) {
RTC_LOG(LS_ERROR) << "Unable to parse the corpus config json file";
- return absl::nullopt;
+ return std::nullopt;
}
gen_options.video_streams.push_back(*video_stream_config);
}
@@ -243,7 +243,7 @@
&env_.clock(),
test::CreateSquareFrameGenerator(send_config.video_width,
send_config.video_height,
- absl::nullopt, absl::nullopt),
+ std::nullopt, std::nullopt),
send_config.video_fps, env_.task_queue_factory());
frame_generator->Init();
diff --git a/rtc_tools/rtp_generator/rtp_generator.h b/rtc_tools/rtp_generator/rtp_generator.h
index 2f178d3..9a37f39 100644
--- a/rtc_tools/rtp_generator/rtp_generator.h
+++ b/rtc_tools/rtp_generator/rtp_generator.h
@@ -54,8 +54,8 @@
};
// Attempts to parse RtpGeneratorOptions from a JSON file. Any failures
-// will result in absl::nullopt.
-absl::optional<RtpGeneratorOptions> ParseRtpGeneratorOptionsFromFile(
+// will result in std::nullopt.
+std::optional<RtpGeneratorOptions> ParseRtpGeneratorOptionsFromFile(
const std::string& options_file);
// The RtpGenerator allows generating of corpus material intended to be
diff --git a/rtc_tools/video_encoder/encoded_image_file_writer.cc b/rtc_tools/video_encoder/encoded_image_file_writer.cc
index 624bce3..e495ca3 100644
--- a/rtc_tools/video_encoder/encoded_image_file_writer.cc
+++ b/rtc_tools/video_encoder/encoded_image_file_writer.cc
@@ -22,7 +22,7 @@
CodecTypeToPayloadString(video_codec_setting.codecType);
// Retrieve scalability mode information.
- absl::optional<ScalabilityMode> scalability_mode =
+ std::optional<ScalabilityMode> scalability_mode =
video_codec_setting.GetScalabilityMode();
RTC_CHECK(scalability_mode);
spatial_layers_ = ScalabilityModeToNumSpatialLayers(*scalability_mode);
diff --git a/rtc_tools/video_encoder/video_encoder.cc b/rtc_tools/video_encoder/video_encoder.cc
index 7e8a3d6..a6317d3 100644
--- a/rtc_tools/video_encoder/video_encoder.cc
+++ b/rtc_tools/video_encoder/video_encoder.cc
@@ -377,7 +377,7 @@
RTC_CHECK_NE(codec_type, kVideoCodecGeneric);
// Retrieve scalability mode information.
- absl::optional<ScalabilityMode> scalability_mode =
+ std::optional<ScalabilityMode> scalability_mode =
ScalabilityModeFromString(scalability_mode_string);
RTC_CHECK(scalability_mode);
@@ -641,7 +641,7 @@
frame_buffer_generator = webrtc::test::CreateSquareFrameGenerator(
width, height,
webrtc::test::FrameGeneratorInterface::OutputType::kI420,
- absl::nullopt);
+ std::nullopt);
RTC_CHECK(frame_buffer_generator);
RTC_LOG(LS_INFO) << "CreateSquareFrameGenerator: " << width << "x"
diff --git a/rtc_tools/video_file_reader.cc b/rtc_tools/video_file_reader.cc
index 66fed42..5ec2bfd 100644
--- a/rtc_tools/video_file_reader.cc
+++ b/rtc_tools/video_file_reader.cc
@@ -11,11 +11,11 @@
#include "rtc_tools/video_file_reader.h"
#include <cstdio>
+#include <optional>
#include <string>
#include <vector>
#include "absl/strings/match.h"
-#include "absl/types/optional.h"
#include "api/make_ref_counted.h"
#include "api/video/i420_buffer.h"
#include "rtc_base/checks.h"
@@ -144,9 +144,9 @@
header_line.push_back(static_cast<char>(c));
}
- absl::optional<int> width;
- absl::optional<int> height;
- absl::optional<float> fps;
+ std::optional<int> width;
+ std::optional<int> height;
+ std::optional<float> fps;
std::vector<std::string> fields;
rtc::tokenize(header_line, ' ', &fields);
@@ -173,9 +173,9 @@
std::vector<std::string> fraction;
rtc::tokenize(suffix, ':', &fraction);
if (fraction.size() == 2) {
- const absl::optional<int> numerator =
+ const std::optional<int> numerator =
rtc::StringToNumber<int>(fraction[0]);
- const absl::optional<int> denominator =
+ const std::optional<int> denominator =
rtc::StringToNumber<int>(fraction[1]);
if (numerator && denominator && *denominator != 0)
fps = *numerator / static_cast<float>(*denominator);
diff --git a/sdk/BUILD.gn b/sdk/BUILD.gn
index a03d737..6a48cbf 100644
--- a/sdk/BUILD.gn
+++ b/sdk/BUILD.gn
@@ -36,7 +36,6 @@
deps = [
"../api:audio_options_api",
"../api:libjingle_peerconnection_api",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -453,7 +452,6 @@
"../media:rtc_media_base",
"../rtc_base:checks",
"../rtc_base:logging",
- "//third_party/abseil-cpp/absl/types:optional",
]
configs += [
diff --git a/sdk/android/BUILD.gn b/sdk/android/BUILD.gn
index 9add4ae..f95c749 100644
--- a/sdk/android/BUILD.gn
+++ b/sdk/android/BUILD.gn
@@ -588,7 +588,6 @@
"../../system_wrappers:field_trial",
"../../system_wrappers:metrics",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
"//third_party/jni_zero",
]
}
@@ -690,7 +689,6 @@
"../../rtc_base:timeutils",
"../../rtc_base/synchronization:mutex",
"//third_party/abseil-cpp/absl/memory",
- "//third_party/abseil-cpp/absl/types:optional",
"//third_party/jni_zero",
"//third_party/libyuv",
]
@@ -811,7 +809,6 @@
"../../stats:rtc_stats",
"../../system_wrappers:field_trial",
"//third_party/abseil-cpp/absl/memory",
- "//third_party/abseil-cpp/absl/types:optional",
"//third_party/jni_zero",
]
}
@@ -959,7 +956,6 @@
"../../api:sequence_checker",
"//api:array_view",
"//rtc_base:checks",
- "//third_party/abseil-cpp/absl/types:optional",
"//third_party/jni_zero",
]
}
@@ -1190,7 +1186,6 @@
"../../rtc_base:checks",
"../../rtc_base:logging",
"../../system_wrappers:metrics",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -1217,7 +1212,6 @@
"../../rtc_base:timeutils",
"../../system_wrappers:field_trial",
"../../system_wrappers:metrics",
- "//third_party/abseil-cpp/absl/types:optional",
"//third_party/jni_zero",
]
}
@@ -1278,7 +1272,6 @@
"../../rtc_base:macromagic",
"../../rtc_base:platform_thread",
"../../rtc_base:timeutils",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
diff --git a/sdk/android/native_api/jni/java_types.cc b/sdk/android/native_api/jni/java_types.cc
index 540963f..ab730c4 100644
--- a/sdk/android/native_api/jni/java_types.cc
+++ b/sdk/android/native_api/jni/java_types.cc
@@ -132,27 +132,27 @@
return JNI_Long::Java_Long_longValue(env, j_long);
}
-absl::optional<bool> JavaToNativeOptionalBool(
+std::optional<bool> JavaToNativeOptionalBool(
JNIEnv* jni,
const jni_zero::JavaRef<jobject>& boolean) {
if (IsNull(jni, boolean))
- return absl::nullopt;
+ return std::nullopt;
return JNI_Boolean::Java_Boolean_booleanValue(jni, boolean);
}
-absl::optional<double> JavaToNativeOptionalDouble(
+std::optional<double> JavaToNativeOptionalDouble(
JNIEnv* jni,
const jni_zero::JavaRef<jobject>& j_double) {
if (IsNull(jni, j_double))
- return absl::nullopt;
+ return std::nullopt;
return JNI_Double::Java_Double_doubleValue(jni, j_double);
}
-absl::optional<int32_t> JavaToNativeOptionalInt(
+std::optional<int32_t> JavaToNativeOptionalInt(
JNIEnv* jni,
const jni_zero::JavaRef<jobject>& integer) {
if (IsNull(jni, integer))
- return absl::nullopt;
+ return std::nullopt;
return JNI_Integer::Java_Integer_intValue(jni, integer);
}
@@ -213,19 +213,19 @@
ScopedJavaLocalRef<jobject> NativeToJavaDouble(
JNIEnv* jni,
- const absl::optional<double>& optional_double) {
+ const std::optional<double>& optional_double) {
return optional_double ? NativeToJavaDouble(jni, *optional_double) : nullptr;
}
ScopedJavaLocalRef<jobject> NativeToJavaInteger(
JNIEnv* jni,
- const absl::optional<int32_t>& optional_int) {
+ const std::optional<int32_t>& optional_int) {
return optional_int ? NativeToJavaInteger(jni, *optional_int) : nullptr;
}
ScopedJavaLocalRef<jstring> NativeToJavaString(
JNIEnv* jni,
- const absl::optional<std::string>& str) {
+ const std::optional<std::string>& str) {
return str ? NativeToJavaString(jni, *str) : nullptr;
}
diff --git a/sdk/android/native_api/jni/java_types.h b/sdk/android/native_api/jni/java_types.h
index a4deda1..13dc6ae 100644
--- a/sdk/android/native_api/jni/java_types.h
+++ b/sdk/android/native_api/jni/java_types.h
@@ -21,10 +21,10 @@
#include <map>
#include <memory>
+#include <optional>
#include <string>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/sequence_checker.h"
#include "rtc_base/checks.h"
@@ -134,13 +134,13 @@
int64_t JavaToNativeLong(JNIEnv* env, const jni_zero::JavaRef<jobject>& j_long);
-absl::optional<bool> JavaToNativeOptionalBool(
+std::optional<bool> JavaToNativeOptionalBool(
JNIEnv* jni,
const jni_zero::JavaRef<jobject>& boolean);
-absl::optional<double> JavaToNativeOptionalDouble(
+std::optional<double> JavaToNativeOptionalDouble(
JNIEnv* jni,
const jni_zero::JavaRef<jobject>& j_double);
-absl::optional<int32_t> JavaToNativeOptionalInt(
+std::optional<int32_t> JavaToNativeOptionalInt(
JNIEnv* jni,
const jni_zero::JavaRef<jobject>& integer);
@@ -211,13 +211,13 @@
ScopedJavaLocalRef<jobject> NativeToJavaDouble(
JNIEnv* jni,
- const absl::optional<double>& optional_double);
+ const std::optional<double>& optional_double);
ScopedJavaLocalRef<jobject> NativeToJavaInteger(
JNIEnv* jni,
- const absl::optional<int32_t>& optional_int);
+ const std::optional<int32_t>& optional_int);
ScopedJavaLocalRef<jstring> NativeToJavaString(
JNIEnv* jni,
- const absl::optional<std::string>& str);
+ const std::optional<std::string>& str);
// Helper function for converting std::vector<T> into a Java array.
template <typename T, typename Convert>
diff --git a/sdk/android/native_api/video/video_source.cc b/sdk/android/native_api/video/video_source.cc
index e967c2a..6f66bd1 100644
--- a/sdk/android/native_api/video/video_source.cc
+++ b/sdk/android/native_api/video/video_source.cc
@@ -76,7 +76,7 @@
return android_video_track_source_->is_screencast();
}
- absl::optional<bool> needs_denoising() const override {
+ std::optional<bool> needs_denoising() const override {
return android_video_track_source_->needs_denoising();
}
diff --git a/sdk/android/src/jni/android_network_monitor.cc b/sdk/android/src/jni/android_network_monitor.cc
index a656a34..f7a9f69 100644
--- a/sdk/android/src/jni/android_network_monitor.cc
+++ b/sdk/android/src/jni/android_network_monitor.cc
@@ -317,7 +317,7 @@
return rtc::NetworkBindingResult::NOT_IMPLEMENTED;
}
- absl::optional<NetworkHandle> network_handle =
+ std::optional<NetworkHandle> network_handle =
FindNetworkHandleFromAddressOrName(address, if_name);
if (!network_handle) {
RTC_LOG(LS_WARNING)
@@ -445,7 +445,7 @@
InvokeNetworksChangedCallback();
}
-absl::optional<NetworkHandle>
+std::optional<NetworkHandle>
AndroidNetworkMonitor::FindNetworkHandleFromAddressOrName(
const rtc::IPAddress& ip_address,
absl::string_view if_name) const {
@@ -458,21 +458,20 @@
return AddressMatch(ip_address, address);
});
if (address_it != addresses.end()) {
- return absl::make_optional(iter.first);
+ return std::make_optional(iter.first);
}
}
} else {
auto iter = network_handle_by_address_.find(ip_address);
if (iter != network_handle_by_address_.end()) {
- return absl::make_optional(iter->second);
+ return std::make_optional(iter->second);
}
}
return FindNetworkHandleFromIfname(if_name);
}
-absl::optional<NetworkHandle>
-AndroidNetworkMonitor::FindNetworkHandleFromIfname(
+std::optional<NetworkHandle> AndroidNetworkMonitor::FindNetworkHandleFromIfname(
absl::string_view if_name) const {
RTC_DCHECK_RUN_ON(network_thread_);
@@ -486,12 +485,12 @@
// Use substring match so that e.g if_name="v4-wlan0" is matched
// agains iter="wlan0"
if (if_name.find(iter.first) != absl::string_view::npos) {
- return absl::make_optional(iter.second);
+ return std::make_optional(iter.second);
}
}
}
- return absl::nullopt;
+ return std::nullopt;
}
void AndroidNetworkMonitor::OnNetworkDisconnected_n(NetworkHandle handle) {
diff --git a/sdk/android/src/jni/android_network_monitor.h b/sdk/android/src/jni/android_network_monitor.h
index f94650b..0d1374a 100644
--- a/sdk/android/src/jni/android_network_monitor.h
+++ b/sdk/android/src/jni/android_network_monitor.h
@@ -14,11 +14,11 @@
#include <stdint.h>
#include <map>
+#include <optional>
#include <string>
#include <vector>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/field_trials_view.h"
#include "api/task_queue/pending_task_safety_flag.h"
#include "rtc_base/network_monitor.h"
@@ -118,7 +118,7 @@
void OnNetworkConnected_n(const NetworkInformation& network_info);
// Visible for testing.
- absl::optional<NetworkHandle> FindNetworkHandleFromAddressOrName(
+ std::optional<NetworkHandle> FindNetworkHandleFromAddressOrName(
const rtc::IPAddress& address,
absl::string_view ifname) const;
@@ -129,7 +129,7 @@
rtc::NetworkPreference preference);
rtc::NetworkPreference GetNetworkPreference(rtc::AdapterType) const;
- absl::optional<NetworkHandle> FindNetworkHandleFromIfname(
+ std::optional<NetworkHandle> FindNetworkHandleFromIfname(
absl::string_view ifname) const;
const int android_sdk_int_;
diff --git a/sdk/android/src/jni/android_video_track_source.cc b/sdk/android/src/jni/android_video_track_source.cc
index e7bf5b9..c009ca8 100644
--- a/sdk/android/src/jni/android_video_track_source.cc
+++ b/sdk/android/src/jni/android_video_track_source.cc
@@ -29,11 +29,11 @@
return static_cast<VideoRotation>(rotation);
}
-absl::optional<std::pair<int, int>> OptionalAspectRatio(jint j_width,
- jint j_height) {
+std::optional<std::pair<int, int>> OptionalAspectRatio(jint j_width,
+ jint j_height) {
if (j_width > 0 && j_height > 0)
return std::pair<int, int>(j_width, j_height);
- return absl::nullopt;
+ return std::nullopt;
}
} // namespace
@@ -54,7 +54,7 @@
return is_screencast_.load();
}
-absl::optional<bool> AndroidVideoTrackSource::needs_denoising() const {
+std::optional<bool> AndroidVideoTrackSource::needs_denoising() const {
return false;
}
diff --git a/sdk/android/src/jni/android_video_track_source.h b/sdk/android/src/jni/android_video_track_source.h
index 9099ce6..b69a23a 100644
--- a/sdk/android/src/jni/android_video_track_source.h
+++ b/sdk/android/src/jni/android_video_track_source.h
@@ -41,7 +41,7 @@
// Indicates that the encoder should denoise video before encoding it.
// If it is not set, the default configuration is used which is different
// depending on video codec.
- absl::optional<bool> needs_denoising() const override;
+ std::optional<bool> needs_denoising() const override;
void SetState(SourceState state);
diff --git a/sdk/android/src/jni/audio_device/aaudio_player.cc b/sdk/android/src/jni/audio_device/aaudio_player.cc
index 2b745b3..c7bae74 100644
--- a/sdk/android/src/jni/audio_device/aaudio_player.cc
+++ b/sdk/android/src/jni/audio_device/aaudio_player.cc
@@ -133,16 +133,16 @@
return -1;
}
-absl::optional<uint32_t> AAudioPlayer::SpeakerVolume() const {
- return absl::nullopt;
+std::optional<uint32_t> AAudioPlayer::SpeakerVolume() const {
+ return std::nullopt;
}
-absl::optional<uint32_t> AAudioPlayer::MaxSpeakerVolume() const {
- return absl::nullopt;
+std::optional<uint32_t> AAudioPlayer::MaxSpeakerVolume() const {
+ return std::nullopt;
}
-absl::optional<uint32_t> AAudioPlayer::MinSpeakerVolume() const {
- return absl::nullopt;
+std::optional<uint32_t> AAudioPlayer::MinSpeakerVolume() const {
+ return std::nullopt;
}
void AAudioPlayer::OnErrorCallback(aaudio_result_t error) {
diff --git a/sdk/android/src/jni/audio_device/aaudio_player.h b/sdk/android/src/jni/audio_device/aaudio_player.h
index ddf948c..1fe581b 100644
--- a/sdk/android/src/jni/audio_device/aaudio_player.h
+++ b/sdk/android/src/jni/audio_device/aaudio_player.h
@@ -14,8 +14,8 @@
#include <aaudio/AAudio.h>
#include <memory>
+#include <optional>
-#include "absl/types/optional.h"
#include "api/audio/audio_device_defines.h"
#include "api/sequence_checker.h"
#include "api/task_queue/task_queue_base.h"
@@ -71,9 +71,9 @@
// Not implemented in AAudio.
bool SpeakerVolumeIsAvailable() override;
int SetSpeakerVolume(uint32_t volume) override;
- absl::optional<uint32_t> SpeakerVolume() const override;
- absl::optional<uint32_t> MaxSpeakerVolume() const override;
- absl::optional<uint32_t> MinSpeakerVolume() const override;
+ std::optional<uint32_t> SpeakerVolume() const override;
+ std::optional<uint32_t> MaxSpeakerVolume() const override;
+ std::optional<uint32_t> MinSpeakerVolume() const override;
protected:
// AAudioObserverInterface implementation.
diff --git a/sdk/android/src/jni/audio_device/audio_device_module.cc b/sdk/android/src/jni/audio_device/audio_device_module.cc
index 3c910b8..eb1c150 100644
--- a/sdk/android/src/jni/audio_device/audio_device_module.cc
+++ b/sdk/android/src/jni/audio_device/audio_device_module.cc
@@ -353,7 +353,7 @@
RTC_DLOG(LS_INFO) << __FUNCTION__;
if (!initialized_)
return -1;
- absl::optional<uint32_t> volume = output_->SpeakerVolume();
+ std::optional<uint32_t> volume = output_->SpeakerVolume();
if (!volume)
return -1;
*output_volume = *volume;
@@ -365,7 +365,7 @@
RTC_DLOG(LS_INFO) << __FUNCTION__;
if (!initialized_)
return -1;
- absl::optional<uint32_t> max_volume = output_->MaxSpeakerVolume();
+ std::optional<uint32_t> max_volume = output_->MaxSpeakerVolume();
if (!max_volume)
return -1;
*output_max_volume = *max_volume;
@@ -376,7 +376,7 @@
RTC_DLOG(LS_INFO) << __FUNCTION__;
if (!initialized_)
return -1;
- absl::optional<uint32_t> min_volume = output_->MinSpeakerVolume();
+ std::optional<uint32_t> min_volume = output_->MinSpeakerVolume();
if (!min_volume)
return -1;
*output_min_volume = *min_volume;
@@ -575,9 +575,9 @@
return output_->GetPlayoutUnderrunCount();
}
- absl::optional<Stats> GetStats() const override {
+ std::optional<Stats> GetStats() const override {
if (!initialized_)
- return absl::nullopt;
+ return std::nullopt;
return output_->GetStats();
}
diff --git a/sdk/android/src/jni/audio_device/audio_device_module.h b/sdk/android/src/jni/audio_device/audio_device_module.h
index 92dbb21..c06991f 100644
--- a/sdk/android/src/jni/audio_device/audio_device_module.h
+++ b/sdk/android/src/jni/audio_device/audio_device_module.h
@@ -12,8 +12,8 @@
#define SDK_ANDROID_SRC_JNI_AUDIO_DEVICE_AUDIO_DEVICE_MODULE_H_
#include <memory>
+#include <optional>
-#include "absl/types/optional.h"
#include "api/audio/audio_device.h"
#include "sdk/android/native_api/jni/scoped_java_ref.h"
@@ -61,13 +61,13 @@
virtual bool Playing() const = 0;
virtual bool SpeakerVolumeIsAvailable() = 0;
virtual int SetSpeakerVolume(uint32_t volume) = 0;
- virtual absl::optional<uint32_t> SpeakerVolume() const = 0;
- virtual absl::optional<uint32_t> MaxSpeakerVolume() const = 0;
- virtual absl::optional<uint32_t> MinSpeakerVolume() const = 0;
+ virtual std::optional<uint32_t> SpeakerVolume() const = 0;
+ virtual std::optional<uint32_t> MaxSpeakerVolume() const = 0;
+ virtual std::optional<uint32_t> MinSpeakerVolume() const = 0;
virtual void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) = 0;
virtual int GetPlayoutUnderrunCount() = 0;
- virtual absl::optional<AudioDeviceModule::Stats> GetStats() const {
- return absl::nullopt;
+ virtual std::optional<AudioDeviceModule::Stats> GetStats() const {
+ return std::nullopt;
}
};
diff --git a/sdk/android/src/jni/audio_device/audio_track_jni.cc b/sdk/android/src/jni/audio_device/audio_track_jni.cc
index 4554b11..464805f 100644
--- a/sdk/android/src/jni/audio_device/audio_track_jni.cc
+++ b/sdk/android/src/jni/audio_device/audio_track_jni.cc
@@ -194,17 +194,17 @@
: -1;
}
-absl::optional<uint32_t> AudioTrackJni::MaxSpeakerVolume() const {
+std::optional<uint32_t> AudioTrackJni::MaxSpeakerVolume() const {
RTC_DCHECK(thread_checker_.IsCurrent());
return Java_WebRtcAudioTrack_getStreamMaxVolume(env_, j_audio_track_);
}
-absl::optional<uint32_t> AudioTrackJni::MinSpeakerVolume() const {
+std::optional<uint32_t> AudioTrackJni::MinSpeakerVolume() const {
RTC_DCHECK(thread_checker_.IsCurrent());
return 0;
}
-absl::optional<uint32_t> AudioTrackJni::SpeakerVolume() const {
+std::optional<uint32_t> AudioTrackJni::SpeakerVolume() const {
RTC_DCHECK(thread_checker_.IsCurrent());
const uint32_t volume =
Java_WebRtcAudioTrack_getStreamVolume(env_, j_audio_track_);
diff --git a/sdk/android/src/jni/audio_device/audio_track_jni.h b/sdk/android/src/jni/audio_device/audio_track_jni.h
index 41fbb0a..fee50ef 100644
--- a/sdk/android/src/jni/audio_device/audio_track_jni.h
+++ b/sdk/android/src/jni/audio_device/audio_track_jni.h
@@ -14,8 +14,8 @@
#include <jni.h>
#include <memory>
+#include <optional>
-#include "absl/types/optional.h"
#include "api/audio/audio_device_defines.h"
#include "api/sequence_checker.h"
#include "modules/audio_device/audio_device_buffer.h"
@@ -63,9 +63,9 @@
bool SpeakerVolumeIsAvailable() override;
int SetSpeakerVolume(uint32_t volume) override;
- absl::optional<uint32_t> SpeakerVolume() const override;
- absl::optional<uint32_t> MaxSpeakerVolume() const override;
- absl::optional<uint32_t> MinSpeakerVolume() const override;
+ std::optional<uint32_t> SpeakerVolume() const override;
+ std::optional<uint32_t> MaxSpeakerVolume() const override;
+ std::optional<uint32_t> MinSpeakerVolume() const override;
int GetPlayoutUnderrunCount() override;
void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) override;
diff --git a/sdk/android/src/jni/audio_device/opensles_player.cc b/sdk/android/src/jni/audio_device/opensles_player.cc
index 6300a3a..588178d 100644
--- a/sdk/android/src/jni/audio_device/opensles_player.cc
+++ b/sdk/android/src/jni/audio_device/opensles_player.cc
@@ -182,16 +182,16 @@
return -1;
}
-absl::optional<uint32_t> OpenSLESPlayer::SpeakerVolume() const {
- return absl::nullopt;
+std::optional<uint32_t> OpenSLESPlayer::SpeakerVolume() const {
+ return std::nullopt;
}
-absl::optional<uint32_t> OpenSLESPlayer::MaxSpeakerVolume() const {
- return absl::nullopt;
+std::optional<uint32_t> OpenSLESPlayer::MaxSpeakerVolume() const {
+ return std::nullopt;
}
-absl::optional<uint32_t> OpenSLESPlayer::MinSpeakerVolume() const {
- return absl::nullopt;
+std::optional<uint32_t> OpenSLESPlayer::MinSpeakerVolume() const {
+ return std::nullopt;
}
void OpenSLESPlayer::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {
diff --git a/sdk/android/src/jni/audio_device/opensles_player.h b/sdk/android/src/jni/audio_device/opensles_player.h
index 3d63bd7..8d6c2ce 100644
--- a/sdk/android/src/jni/audio_device/opensles_player.h
+++ b/sdk/android/src/jni/audio_device/opensles_player.h
@@ -16,8 +16,8 @@
#include <SLES/OpenSLES_AndroidConfiguration.h>
#include <memory>
+#include <optional>
-#include "absl/types/optional.h"
#include "api/audio/audio_device_defines.h"
#include "api/scoped_refptr.h"
#include "api/sequence_checker.h"
@@ -77,9 +77,9 @@
bool SpeakerVolumeIsAvailable() override;
int SetSpeakerVolume(uint32_t volume) override;
- absl::optional<uint32_t> SpeakerVolume() const override;
- absl::optional<uint32_t> MaxSpeakerVolume() const override;
- absl::optional<uint32_t> MinSpeakerVolume() const override;
+ std::optional<uint32_t> SpeakerVolume() const override;
+ std::optional<uint32_t> MaxSpeakerVolume() const override;
+ std::optional<uint32_t> MinSpeakerVolume() const override;
void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) override;
diff --git a/sdk/android/src/jni/pc/crypto_options.cc b/sdk/android/src/jni/pc/crypto_options.cc
index af5f195..f37ebc1 100644
--- a/sdk/android/src/jni/pc/crypto_options.cc
+++ b/sdk/android/src/jni/pc/crypto_options.cc
@@ -15,11 +15,11 @@
namespace webrtc {
namespace jni {
-absl::optional<CryptoOptions> JavaToNativeOptionalCryptoOptions(
+std::optional<CryptoOptions> JavaToNativeOptionalCryptoOptions(
JNIEnv* jni,
const JavaRef<jobject>& j_crypto_options) {
if (j_crypto_options.is_null()) {
- return absl::nullopt;
+ return std::nullopt;
}
ScopedJavaLocalRef<jobject> j_srtp =
@@ -36,7 +36,7 @@
Java_Srtp_getEnableEncryptedRtpHeaderExtensions(jni, j_srtp);
native_crypto_options.sframe.require_frame_encryption =
Java_SFrame_getRequireFrameEncryption(jni, j_sframe);
- return absl::optional<CryptoOptions>(native_crypto_options);
+ return std::optional<CryptoOptions>(native_crypto_options);
}
} // namespace jni
diff --git a/sdk/android/src/jni/pc/crypto_options.h b/sdk/android/src/jni/pc/crypto_options.h
index a9c8f26..f82f218 100644
--- a/sdk/android/src/jni/pc/crypto_options.h
+++ b/sdk/android/src/jni/pc/crypto_options.h
@@ -13,14 +13,15 @@
#include <jni.h>
-#include "absl/types/optional.h"
+#include <optional>
+
#include "api/crypto/crypto_options.h"
#include "sdk/android/native_api/jni/scoped_java_ref.h"
namespace webrtc {
namespace jni {
-absl::optional<CryptoOptions> JavaToNativeOptionalCryptoOptions(
+std::optional<CryptoOptions> JavaToNativeOptionalCryptoOptions(
JNIEnv* jni,
const JavaRef<jobject>& j_crypto_options);
diff --git a/sdk/android/src/jni/pc/ice_candidate.cc b/sdk/android/src/jni/pc/ice_candidate.cc
index af92ff8..fdfe1ec 100644
--- a/sdk/android/src/jni/pc/ice_candidate.cc
+++ b/sdk/android/src/jni/pc/ice_candidate.cc
@@ -228,13 +228,13 @@
return PeerConnectionInterface::kTlsCertPolicySecure;
}
-absl::optional<rtc::AdapterType> JavaToNativeNetworkPreference(
+std::optional<rtc::AdapterType> JavaToNativeNetworkPreference(
JNIEnv* jni,
const JavaRef<jobject>& j_network_preference) {
std::string enum_name = GetJavaEnumName(jni, j_network_preference);
if (enum_name == "UNKNOWN")
- return absl::nullopt;
+ return std::nullopt;
if (enum_name == "ETHERNET")
return rtc::ADAPTER_TYPE_ETHERNET;
@@ -252,7 +252,7 @@
return rtc::ADAPTER_TYPE_LOOPBACK;
RTC_CHECK(false) << "Unexpected NetworkPreference enum_name " << enum_name;
- return absl::nullopt;
+ return std::nullopt;
}
} // namespace jni
diff --git a/sdk/android/src/jni/pc/ice_candidate.h b/sdk/android/src/jni/pc/ice_candidate.h
index 4bdeea6..4126294 100644
--- a/sdk/android/src/jni/pc/ice_candidate.h
+++ b/sdk/android/src/jni/pc/ice_candidate.h
@@ -79,7 +79,7 @@
JNIEnv* jni,
const JavaRef<jobject>& j_ice_server_tls_cert_policy);
-absl::optional<rtc::AdapterType> JavaToNativeNetworkPreference(
+std::optional<rtc::AdapterType> JavaToNativeNetworkPreference(
JNIEnv* jni,
const JavaRef<jobject>& j_network_preference);
diff --git a/sdk/android/src/jni/pc/peer_connection_factory.cc b/sdk/android/src/jni/pc/peer_connection_factory.cc
index e98ac6f..2619140 100644
--- a/sdk/android/src/jni/pc/peer_connection_factory.cc
+++ b/sdk/android/src/jni/pc/peer_connection_factory.cc
@@ -76,12 +76,12 @@
});
}
-absl::optional<PeerConnectionFactoryInterface::Options>
+std::optional<PeerConnectionFactoryInterface::Options>
JavaToNativePeerConnectionFactoryOptions(
JNIEnv* jni,
const jni_zero::JavaRef<jobject>& j_options) {
if (j_options.is_null())
- return absl::nullopt;
+ return std::nullopt;
PeerConnectionFactoryInterface::Options native_options;
@@ -262,7 +262,7 @@
signaling_thread->SetName("signaling_thread", NULL);
RTC_CHECK(signaling_thread->Start()) << "Failed to start thread";
- const absl::optional<PeerConnectionFactoryInterface::Options> options =
+ const std::optional<PeerConnectionFactoryInterface::Options> options =
JavaToNativePeerConnectionFactoryOptions(jni, joptions);
PeerConnectionFactoryDependencies dependencies;
@@ -447,7 +447,7 @@
if (key_type != rtc::KT_DEFAULT) {
rtc::scoped_refptr<rtc::RTCCertificate> certificate =
rtc::RTCCertificateGenerator::GenerateCertificate(
- rtc::KeyParams(key_type), absl::nullopt);
+ rtc::KeyParams(key_type), std::nullopt);
if (!certificate) {
RTC_LOG(LS_ERROR) << "Failed to generate certificate. KeyType: "
<< key_type;
diff --git a/sdk/android/src/jni/pc/rtp_transceiver.cc b/sdk/android/src/jni/pc/rtp_transceiver.cc
index 3746b11..c2b4920 100644
--- a/sdk/android/src/jni/pc/rtp_transceiver.cc
+++ b/sdk/android/src/jni/pc/rtp_transceiver.cc
@@ -96,7 +96,7 @@
ScopedJavaLocalRef<jstring> JNI_RtpTransceiver_GetMid(
JNIEnv* jni,
jlong j_rtp_transceiver_pointer) {
- absl::optional<std::string> mid =
+ std::optional<std::string> mid =
reinterpret_cast<RtpTransceiverInterface*>(j_rtp_transceiver_pointer)
->mid();
return NativeToJavaString(jni, mid);
@@ -135,7 +135,7 @@
ScopedJavaLocalRef<jobject> JNI_RtpTransceiver_CurrentDirection(
JNIEnv* jni,
jlong j_rtp_transceiver_pointer) {
- absl::optional<RtpTransceiverDirection> direction =
+ std::optional<RtpTransceiverDirection> direction =
reinterpret_cast<RtpTransceiverInterface*>(j_rtp_transceiver_pointer)
->current_direction();
return direction ? NativeToJavaRtpTransceiverDirection(jni, *direction)
diff --git a/sdk/android/src/jni/pc/session_description.cc b/sdk/android/src/jni/pc/session_description.cc
index bbac721..64b2dae 100644
--- a/sdk/android/src/jni/pc/session_description.cc
+++ b/sdk/android/src/jni/pc/session_description.cc
@@ -27,7 +27,7 @@
jni, Java_SessionDescription_getTypeInCanonicalForm(jni, j_sdp));
std::string std_description =
JavaToStdString(jni, Java_SessionDescription_getDescription(jni, j_sdp));
- absl::optional<SdpType> sdp_type_maybe = SdpTypeFromString(std_type);
+ std::optional<SdpType> sdp_type_maybe = SdpTypeFromString(std_type);
if (!sdp_type_maybe) {
RTC_LOG(LS_ERROR) << "Unexpected SDP type: " << std_type;
return nullptr;
diff --git a/sdk/android/src/jni/video_decoder_wrapper.cc b/sdk/android/src/jni/video_decoder_wrapper.cc
index eaeeb2d..dae1c08 100644
--- a/sdk/android/src/jni/video_decoder_wrapper.cc
+++ b/sdk/android/src/jni/video_decoder_wrapper.cc
@@ -36,9 +36,9 @@
const int64_t kNumRtpTicksPerMillisec = 90000 / rtc::kNumMillisecsPerSec;
template <typename Dst, typename Src>
-inline absl::optional<Dst> cast_optional(const absl::optional<Src>& value) {
- return value ? absl::optional<Dst>(rtc::dchecked_cast<Dst, Src>(*value))
- : absl::nullopt;
+inline std::optional<Dst> cast_optional(const std::optional<Src>& value) {
+ return value ? std::optional<Dst>(rtc::dchecked_cast<Dst, Src>(*value))
+ : std::nullopt;
}
} // namespace
@@ -110,7 +110,7 @@
frame_extra_info.timestamp_rtp = input_image.RtpTimestamp();
frame_extra_info.timestamp_ntp = input_image.ntp_time_ms_;
frame_extra_info.qp =
- qp_parsing_enabled_ ? ParseQP(input_image) : absl::nullopt;
+ qp_parsing_enabled_ ? ParseQP(input_image) : std::nullopt;
{
MutexLock lock(&frame_extra_infos_lock_);
frame_extra_infos_.push_back(frame_extra_info);
@@ -188,10 +188,10 @@
JavaToNativeFrame(env, j_frame, frame_extra_info.timestamp_rtp);
frame.set_ntp_time_ms(frame_extra_info.timestamp_ntp);
- absl::optional<int32_t> decoding_time_ms =
+ std::optional<int32_t> decoding_time_ms =
JavaToNativeOptionalInt(env, j_decode_time_ms);
- absl::optional<uint8_t> decoder_qp =
+ std::optional<uint8_t> decoder_qp =
cast_optional<uint8_t, int32_t>(JavaToNativeOptionalInt(env, j_qp));
// If the decoder provides QP values itself, no need to parse the bitstream.
// Enable QP parsing if decoder does not provide QP values itself.
@@ -230,13 +230,13 @@
return WEBRTC_VIDEO_CODEC_FALLBACK_SOFTWARE;
}
-absl::optional<uint8_t> VideoDecoderWrapper::ParseQP(
+std::optional<uint8_t> VideoDecoderWrapper::ParseQP(
const EncodedImage& input_image) {
if (input_image.qp_ != -1) {
return input_image.qp_;
}
- absl::optional<uint8_t> qp;
+ std::optional<uint8_t> qp;
switch (decoder_settings_.codec_type()) {
case kVideoCodecVP8: {
int qp_int;
diff --git a/sdk/android/src/jni/video_decoder_wrapper.h b/sdk/android/src/jni/video_decoder_wrapper.h
index 5e397f1..ff2ae29 100644
--- a/sdk/android/src/jni/video_decoder_wrapper.h
+++ b/sdk/android/src/jni/video_decoder_wrapper.h
@@ -65,7 +65,7 @@
uint32_t timestamp_rtp;
int64_t timestamp_ntp;
- absl::optional<uint8_t> qp;
+ std::optional<uint8_t> qp;
FrameExtraInfo();
FrameExtraInfo(const FrameExtraInfo&);
@@ -81,7 +81,7 @@
const char* method_name)
RTC_RUN_ON(decoder_thread_checker_);
- absl::optional<uint8_t> ParseQP(const EncodedImage& input_image)
+ std::optional<uint8_t> ParseQP(const EncodedImage& input_image)
RTC_RUN_ON(decoder_thread_checker_);
const ScopedJavaGlobalRef<jobject> decoder_;
diff --git a/sdk/android/src/jni/video_encoder_factory_wrapper.cc b/sdk/android/src/jni/video_encoder_factory_wrapper.cc
index 78aa145..709fa58 100644
--- a/sdk/android/src/jni/video_encoder_factory_wrapper.cc
+++ b/sdk/android/src/jni/video_encoder_factory_wrapper.cc
@@ -37,36 +37,36 @@
j_codec_info);
}
- absl::optional<SdpVideoFormat> OnAvailableBitrate(
+ std::optional<SdpVideoFormat> OnAvailableBitrate(
const DataRate& rate) override {
JNIEnv* jni = AttachCurrentThreadIfNeeded();
ScopedJavaLocalRef<jobject> codec_info =
Java_VideoEncoderSelector_onAvailableBitrate(jni, encoder_selector_,
rate.kbps<int>());
if (codec_info.is_null()) {
- return absl::nullopt;
+ return std::nullopt;
}
return VideoCodecInfoToSdpVideoFormat(jni, codec_info);
}
- absl::optional<SdpVideoFormat> OnResolutionChange(
+ std::optional<SdpVideoFormat> OnResolutionChange(
const RenderResolution& resolution) override {
JNIEnv* jni = AttachCurrentThreadIfNeeded();
ScopedJavaLocalRef<jobject> codec_info =
Java_VideoEncoderSelector_onResolutionChange(
jni, encoder_selector_, resolution.Width(), resolution.Height());
if (codec_info.is_null()) {
- return absl::nullopt;
+ return std::nullopt;
}
return VideoCodecInfoToSdpVideoFormat(jni, codec_info);
}
- absl::optional<SdpVideoFormat> OnEncoderBroken() override {
+ std::optional<SdpVideoFormat> OnEncoderBroken() override {
JNIEnv* jni = AttachCurrentThreadIfNeeded();
ScopedJavaLocalRef<jobject> codec_info =
Java_VideoEncoderSelector_onEncoderBroken(jni, encoder_selector_);
if (codec_info.is_null()) {
- return absl::nullopt;
+ return std::nullopt;
}
return VideoCodecInfoToSdpVideoFormat(jni, codec_info);
}
diff --git a/sdk/android/src/jni/video_encoder_wrapper.cc b/sdk/android/src/jni/video_encoder_wrapper.cc
index 92e6c80..08112f7 100644
--- a/sdk/android/src/jni/video_encoder_wrapper.cc
+++ b/sdk/android/src/jni/video_encoder_wrapper.cc
@@ -201,10 +201,10 @@
if (!isOn)
return ScalingSettings::kOff;
- absl::optional<int> low = JavaToNativeOptionalInt(
+ std::optional<int> low = JavaToNativeOptionalInt(
jni,
Java_VideoEncoderWrapper_getScalingSettingsLow(jni, j_scaling_settings));
- absl::optional<int> high = JavaToNativeOptionalInt(
+ std::optional<int> high = JavaToNativeOptionalInt(
jni,
Java_VideoEncoderWrapper_getScalingSettingsHigh(jni, j_scaling_settings));
diff --git a/sdk/android/src/jni/video_encoder_wrapper.h b/sdk/android/src/jni/video_encoder_wrapper.h
index d0b3893..651f162 100644
--- a/sdk/android/src/jni/video_encoder_wrapper.h
+++ b/sdk/android/src/jni/video_encoder_wrapper.h
@@ -15,10 +15,10 @@
#include <deque>
#include <memory>
+#include <optional>
#include <string>
#include <vector>
-#include "absl/types/optional.h"
#include "api/video_codecs/video_encoder.h"
#include "common_video/h264/h264_bitstream_parser.h"
#ifdef RTC_ENABLE_H265
@@ -101,7 +101,7 @@
EncodedImageCallback* callback_;
bool initialized_;
int num_resets_;
- absl::optional<VideoEncoder::Capabilities> capabilities_;
+ std::optional<VideoEncoder::Capabilities> capabilities_;
int number_of_cores_;
VideoCodec codec_settings_;
EncoderInfo encoder_info_;
diff --git a/sdk/media_constraints.cc b/sdk/media_constraints.cc
index 88261e7..1281353 100644
--- a/sdk/media_constraints.cc
+++ b/sdk/media_constraints.cc
@@ -10,7 +10,8 @@
#include "sdk/media_constraints.h"
-#include "absl/types/optional.h"
+#include <optional>
+
#include "api/peer_connection_interface.h"
namespace webrtc {
@@ -74,11 +75,11 @@
}
// Converts a constraint (mandatory takes precedence over optional) to an
-// absl::optional.
+// std::optional.
template <typename T>
void ConstraintToOptional(const MediaConstraints* constraints,
const std::string& key,
- absl::optional<T>* value_out) {
+ std::optional<T>* value_out) {
T value;
bool present = FindConstraint<T>(constraints, key, &value, nullptr);
if (present) {
diff --git a/sdk/objc/api/peerconnection/RTCCertificate.mm b/sdk/objc/api/peerconnection/RTCCertificate.mm
index f6817a6..c3e5d7f 100644
--- a/sdk/objc/api/peerconnection/RTCCertificate.mm
+++ b/sdk/objc/api/peerconnection/RTCCertificate.mm
@@ -51,7 +51,7 @@
expirationTimestamp);
} else {
cc_certificate =
- rtc::RTCCertificateGenerator::GenerateCertificate(rtc::KeyParams(keyType), absl::nullopt);
+ rtc::RTCCertificateGenerator::GenerateCertificate(rtc::KeyParams(keyType), std::nullopt);
}
if (!cc_certificate) {
RTCLogError(@"Failed to generate certificate.");
diff --git a/sdk/objc/api/peerconnection/RTCConfiguration.mm b/sdk/objc/api/peerconnection/RTCConfiguration.mm
index 7635ac0..d53a948 100644
--- a/sdk/objc/api/peerconnection/RTCConfiguration.mm
+++ b/sdk/objc/api/peerconnection/RTCConfiguration.mm
@@ -248,7 +248,7 @@
if (keyType != rtc::KT_DEFAULT) {
rtc::scoped_refptr<rtc::RTCCertificate> certificate =
rtc::RTCCertificateGenerator::GenerateCertificate(rtc::KeyParams(keyType),
- absl::optional<uint64_t>());
+ std::optional<uint64_t>());
if (!certificate) {
RTCLogError(@"Failed to generate certificate.");
return nullptr;
@@ -263,7 +263,7 @@
nativeConfig->surface_ice_candidates_on_ice_transport_type_changed =
_shouldSurfaceIceCandidatesOnIceTransportTypeChanged ? true : false;
if (_iceCheckMinInterval != nil) {
- nativeConfig->ice_check_min_interval = absl::optional<int>(_iceCheckMinInterval.intValue);
+ nativeConfig->ice_check_min_interval = std::optional<int>(_iceCheckMinInterval.intValue);
}
nativeConfig->sdp_semantics = [[self class] nativeSdpSemanticsForSdpSemantics:_sdpSemantics];
if (_turnCustomizer) {
@@ -280,7 +280,7 @@
_cryptoOptions.srtpEnableEncryptedRtpHeaderExtensions ? true : false;
nativeCryptoOptions.sframe.require_frame_encryption =
_cryptoOptions.sframeRequireFrameEncryption ? true : false;
- nativeConfig->crypto_options = absl::optional<webrtc::CryptoOptions>(nativeCryptoOptions);
+ nativeConfig->crypto_options = std::optional<webrtc::CryptoOptions>(nativeCryptoOptions);
}
nativeConfig->turn_logging_id = [_turnLoggingId UTF8String];
nativeConfig->set_audio_rtcp_report_interval_ms(_rtcpAudioReportIntervalMs);
@@ -289,20 +289,20 @@
nativeConfig->offer_extmap_allow_mixed = _offerExtmapAllowMixed;
if (_iceCheckIntervalStrongConnectivity != nil) {
nativeConfig->ice_check_interval_strong_connectivity =
- absl::optional<int>(_iceCheckIntervalStrongConnectivity.intValue);
+ std::optional<int>(_iceCheckIntervalStrongConnectivity.intValue);
}
if (_iceCheckIntervalWeakConnectivity != nil) {
nativeConfig->ice_check_interval_weak_connectivity =
- absl::optional<int>(_iceCheckIntervalWeakConnectivity.intValue);
+ std::optional<int>(_iceCheckIntervalWeakConnectivity.intValue);
}
if (_iceUnwritableTimeout != nil) {
- nativeConfig->ice_unwritable_timeout = absl::optional<int>(_iceUnwritableTimeout.intValue);
+ nativeConfig->ice_unwritable_timeout = std::optional<int>(_iceUnwritableTimeout.intValue);
}
if (_iceUnwritableMinChecks != nil) {
- nativeConfig->ice_unwritable_min_checks = absl::optional<int>(_iceUnwritableMinChecks.intValue);
+ nativeConfig->ice_unwritable_min_checks = std::optional<int>(_iceUnwritableMinChecks.intValue);
}
if (_iceInactiveTimeout != nil) {
- nativeConfig->ice_inactive_timeout = absl::optional<int>(_iceInactiveTimeout.intValue);
+ nativeConfig->ice_inactive_timeout = std::optional<int>(_iceInactiveTimeout.intValue);
}
return nativeConfig.release();
}
diff --git a/sdk/objc/api/peerconnection/RTCPeerConnection.mm b/sdk/objc/api/peerconnection/RTCPeerConnection.mm
index 62e640e..5a9f14a 100644
--- a/sdk/objc/api/peerconnection/RTCPeerConnection.mm
+++ b/sdk/objc/api/peerconnection/RTCPeerConnection.mm
@@ -622,13 +622,13 @@
maxBitrateBps:(nullable NSNumber *)maxBitrateBps {
webrtc::BitrateSettings params;
if (minBitrateBps != nil) {
- params.min_bitrate_bps = absl::optional<int>(minBitrateBps.intValue);
+ params.min_bitrate_bps = std::optional<int>(minBitrateBps.intValue);
}
if (currentBitrateBps != nil) {
- params.start_bitrate_bps = absl::optional<int>(currentBitrateBps.intValue);
+ params.start_bitrate_bps = std::optional<int>(currentBitrateBps.intValue);
}
if (maxBitrateBps != nil) {
- params.max_bitrate_bps = absl::optional<int>(maxBitrateBps.intValue);
+ params.max_bitrate_bps = std::optional<int>(maxBitrateBps.intValue);
}
return _peerConnection->SetBitrate(params).ok();
}
diff --git a/sdk/objc/api/peerconnection/RTCRtpCodecCapability.mm b/sdk/objc/api/peerconnection/RTCRtpCodecCapability.mm
index 9c38bb7..cfce5e3 100644
--- a/sdk/objc/api/peerconnection/RTCRtpCodecCapability.mm
+++ b/sdk/objc/api/peerconnection/RTCRtpCodecCapability.mm
@@ -88,7 +88,7 @@
- (webrtc::RtpCodecCapability)nativeRtpCodecCapability {
webrtc::RtpCodecCapability rtpCodecCapability;
if (_preferredPayloadType != nil) {
- rtpCodecCapability.preferred_payload_type = absl::optional<int>(_preferredPayloadType.intValue);
+ rtpCodecCapability.preferred_payload_type = std::optional<int>(_preferredPayloadType.intValue);
}
rtpCodecCapability.name = [NSString stdStringForString:_name];
// NSString pointer comparison is safe here since "kind" is readonly and only
@@ -101,10 +101,10 @@
RTC_DCHECK_NOTREACHED();
}
if (_clockRate != nil) {
- rtpCodecCapability.clock_rate = absl::optional<int>(_clockRate.intValue);
+ rtpCodecCapability.clock_rate = std::optional<int>(_clockRate.intValue);
}
if (_numChannels != nil) {
- rtpCodecCapability.num_channels = absl::optional<int>(_numChannels.intValue);
+ rtpCodecCapability.num_channels = std::optional<int>(_numChannels.intValue);
}
for (NSString *paramKey in _parameters.allKeys) {
std::string key = [NSString stdStringForString:paramKey];
diff --git a/sdk/objc/api/peerconnection/RTCRtpCodecParameters.mm b/sdk/objc/api/peerconnection/RTCRtpCodecParameters.mm
index e273219..f01d367 100644
--- a/sdk/objc/api/peerconnection/RTCRtpCodecParameters.mm
+++ b/sdk/objc/api/peerconnection/RTCRtpCodecParameters.mm
@@ -97,10 +97,10 @@
RTC_DCHECK_NOTREACHED();
}
if (_clockRate != nil) {
- parameters.clock_rate = absl::optional<int>(_clockRate.intValue);
+ parameters.clock_rate = std::optional<int>(_clockRate.intValue);
}
if (_numChannels != nil) {
- parameters.num_channels = absl::optional<int>(_numChannels.intValue);
+ parameters.num_channels = std::optional<int>(_numChannels.intValue);
}
for (NSString *paramKey in _parameters.allKeys) {
std::string key = [NSString stdStringForString:paramKey];
diff --git a/sdk/objc/api/peerconnection/RTCRtpEncodingParameters.mm b/sdk/objc/api/peerconnection/RTCRtpEncodingParameters.mm
index 8bb2f43..69f8885 100644
--- a/sdk/objc/api/peerconnection/RTCRtpEncodingParameters.mm
+++ b/sdk/objc/api/peerconnection/RTCRtpEncodingParameters.mm
@@ -75,23 +75,22 @@
}
parameters.active = _isActive;
if (_maxBitrateBps != nil) {
- parameters.max_bitrate_bps = absl::optional<int>(_maxBitrateBps.intValue);
+ parameters.max_bitrate_bps = std::optional<int>(_maxBitrateBps.intValue);
}
if (_minBitrateBps != nil) {
- parameters.min_bitrate_bps = absl::optional<int>(_minBitrateBps.intValue);
+ parameters.min_bitrate_bps = std::optional<int>(_minBitrateBps.intValue);
}
if (_maxFramerate != nil) {
- parameters.max_framerate = absl::optional<int>(_maxFramerate.intValue);
+ parameters.max_framerate = std::optional<int>(_maxFramerate.intValue);
}
if (_numTemporalLayers != nil) {
- parameters.num_temporal_layers = absl::optional<int>(_numTemporalLayers.intValue);
+ parameters.num_temporal_layers = std::optional<int>(_numTemporalLayers.intValue);
}
if (_scaleResolutionDownBy != nil) {
- parameters.scale_resolution_down_by =
- absl::optional<double>(_scaleResolutionDownBy.doubleValue);
+ parameters.scale_resolution_down_by = std::optional<double>(_scaleResolutionDownBy.doubleValue);
}
if (_ssrc != nil) {
- parameters.ssrc = absl::optional<uint32_t>(_ssrc.unsignedLongValue);
+ parameters.ssrc = std::optional<uint32_t>(_ssrc.unsignedLongValue);
}
parameters.bitrate_priority = _bitratePriority;
parameters.network_priority =
diff --git a/sdk/objc/api/peerconnection/RTCRtpHeaderExtensionCapability.mm b/sdk/objc/api/peerconnection/RTCRtpHeaderExtensionCapability.mm
index c5cbda1..22e5338 100644
--- a/sdk/objc/api/peerconnection/RTCRtpHeaderExtensionCapability.mm
+++ b/sdk/objc/api/peerconnection/RTCRtpHeaderExtensionCapability.mm
@@ -48,7 +48,7 @@
webrtc::RtpHeaderExtensionCapability rtpHeaderExtensionCapability;
rtpHeaderExtensionCapability.uri = [NSString stdStringForString:_uri];
if (_preferredId != nil) {
- rtpHeaderExtensionCapability.preferred_id = absl::optional<int>(_preferredId.intValue);
+ rtpHeaderExtensionCapability.preferred_id = std::optional<int>(_preferredId.intValue);
}
rtpHeaderExtensionCapability.preferred_encrypt = _preferredEncrypted;
return rtpHeaderExtensionCapability;
diff --git a/sdk/objc/api/peerconnection/RTCRtpParameters.mm b/sdk/objc/api/peerconnection/RTCRtpParameters.mm
index d725f6a..1c2a94f 100644
--- a/sdk/objc/api/peerconnection/RTCRtpParameters.mm
+++ b/sdk/objc/api/peerconnection/RTCRtpParameters.mm
@@ -102,7 +102,7 @@
}
+ (NSNumber *)degradationPreferenceFromNativeDegradationPreference:
- (absl::optional<webrtc::DegradationPreference>)nativeDegradationPreference {
+ (std::optional<webrtc::DegradationPreference>)nativeDegradationPreference {
if (!nativeDegradationPreference.has_value()) {
return nil;
}
diff --git a/sdk/objc/api/peerconnection/RTCRtpSource.mm b/sdk/objc/api/peerconnection/RTCRtpSource.mm
index 5f7a678..52b0e86 100644
--- a/sdk/objc/api/peerconnection/RTCRtpSource.mm
+++ b/sdk/objc/api/peerconnection/RTCRtpSource.mm
@@ -41,7 +41,7 @@
}
- (NSNumber *)audioLevel {
- absl::optional<uint8_t> level = _nativeRtpSource.value().audio_level();
+ std::optional<uint8_t> level = _nativeRtpSource.value().audio_level();
if (!level.has_value()) {
return nil;
}
diff --git a/sdk/objc/api/peerconnection/RTCVideoCodecInfo+Private.mm b/sdk/objc/api/peerconnection/RTCVideoCodecInfo+Private.mm
index ea01a5f..dd45ef0 100644
--- a/sdk/objc/api/peerconnection/RTCVideoCodecInfo+Private.mm
+++ b/sdk/objc/api/peerconnection/RTCVideoCodecInfo+Private.mm
@@ -44,7 +44,7 @@
absl::InlinedVector<webrtc::ScalabilityMode, webrtc::kScalabilityModeCount> scalability_modes;
for (NSString* mode_name in self.scalabilityModes) {
- absl::optional<webrtc::ScalabilityMode> mode =
+ std::optional<webrtc::ScalabilityMode> mode =
webrtc::ScalabilityModeStringToEnum([NSString stdStringForString:mode_name]);
if (mode.has_value()) {
scalability_modes.push_back(*mode);
diff --git a/sdk/objc/components/renderer/opengl/RTCDefaultShader.mm b/sdk/objc/components/renderer/opengl/RTCDefaultShader.mm
index 9d686f6..ca0a844 100644
--- a/sdk/objc/components/renderer/opengl/RTCDefaultShader.mm
+++ b/sdk/objc/components/renderer/opengl/RTCDefaultShader.mm
@@ -16,7 +16,7 @@
#import "RTCShader.h"
#import "base/RTCLogging.h"
-#include "absl/types/optional.h"
+#include <optional>
static const int kYTextureUnit = 0;
static const int kUTextureUnit = 1;
@@ -69,7 +69,7 @@
GLuint _vertexBuffer;
GLuint _vertexArray;
// Store current rotation and only upload new vertex data when rotation changes.
- absl::optional<RTCVideoRotation> _currentRotation;
+ std::optional<RTCVideoRotation> _currentRotation;
GLuint _i420Program;
GLuint _nv12Program;
@@ -138,7 +138,7 @@
glBindBuffer(GL_ARRAY_BUFFER, _vertexBuffer);
if (!_currentRotation || rotation != *_currentRotation) {
- _currentRotation = absl::optional<RTCVideoRotation>(rotation);
+ _currentRotation = std::optional<RTCVideoRotation>(rotation);
RTCSetVertexData(*_currentRotation);
}
return YES;
diff --git a/sdk/objc/components/video_codec/RTCH264ProfileLevelId.mm b/sdk/objc/components/video_codec/RTCH264ProfileLevelId.mm
index 028f005..400bc78 100644
--- a/sdk/objc/components/video_codec/RTCH264ProfileLevelId.mm
+++ b/sdk/objc/components/video_codec/RTCH264ProfileLevelId.mm
@@ -39,10 +39,10 @@
#if defined(WEBRTC_IOS)
NSString *MaxSupportedLevelForProfile(webrtc::H264Profile profile) {
- const absl::optional<webrtc::H264ProfileLevelId> profileLevelId =
+ const std::optional<webrtc::H264ProfileLevelId> profileLevelId =
[UIDevice maxSupportedH264Profile];
if (profileLevelId && profileLevelId->profile >= profile) {
- const absl::optional<std::string> profileString =
+ const std::optional<std::string> profileString =
H264ProfileLevelIdToString(webrtc::H264ProfileLevelId(profile, profileLevelId->level));
if (profileString) {
return [NSString stringForStdString:*profileString];
@@ -94,7 +94,7 @@
if (self) {
self.hexString = hexString;
- absl::optional<webrtc::H264ProfileLevelId> profile_level_id =
+ std::optional<webrtc::H264ProfileLevelId> profile_level_id =
webrtc::ParseH264ProfileLevelId([hexString cStringUsingEncoding:NSUTF8StringEncoding]);
if (profile_level_id.has_value()) {
self.profile = static_cast<RTCH264Profile>(profile_level_id->profile);
@@ -110,7 +110,7 @@
self.profile = profile;
self.level = level;
- absl::optional<std::string> hex_string =
+ std::optional<std::string> hex_string =
webrtc::H264ProfileLevelIdToString(webrtc::H264ProfileLevelId(
static_cast<webrtc::H264Profile>(profile), static_cast<webrtc::H264Level>(level)));
self.hexString =
diff --git a/sdk/objc/components/video_codec/RTCVideoEncoderH264.mm b/sdk/objc/components/video_codec/RTCVideoEncoderH264.mm
index 27ccd6d..ec76f0a 100644
--- a/sdk/objc/components/video_codec/RTCVideoEncoderH264.mm
+++ b/sdk/objc/components/video_codec/RTCVideoEncoderH264.mm
@@ -325,7 +325,7 @@
uint32_t _encoderFrameRate;
uint32_t _maxAllowedFrameRate;
RTCH264PacketizationMode _packetizationMode;
- absl::optional<webrtc::H264ProfileLevelId> _profile_level_id;
+ std::optional<webrtc::H264ProfileLevelId> _profile_level_id;
RTCVideoEncoderCallback _callback;
int32_t _width;
int32_t _height;
diff --git a/sdk/objc/components/video_codec/UIDevice+H264Profile.h b/sdk/objc/components/video_codec/UIDevice+H264Profile.h
index a51debb..722f6f4 100644
--- a/sdk/objc/components/video_codec/UIDevice+H264Profile.h
+++ b/sdk/objc/components/video_codec/UIDevice+H264Profile.h
@@ -14,6 +14,6 @@
@interface UIDevice (H264Profile)
-+ (absl::optional<webrtc::H264ProfileLevelId>)maxSupportedH264Profile;
++ (std::optional<webrtc::H264ProfileLevelId>)maxSupportedH264Profile;
@end
diff --git a/sdk/objc/components/video_codec/UIDevice+H264Profile.mm b/sdk/objc/components/video_codec/UIDevice+H264Profile.mm
index cfb2b97..88edc52 100644
--- a/sdk/objc/components/video_codec/UIDevice+H264Profile.mm
+++ b/sdk/objc/components/video_codec/UIDevice+H264Profile.mm
@@ -261,7 +261,7 @@
{H264Profile::kProfileHigh, H264Level::kLevel4_1}},
};
-absl::optional<H264ProfileLevelId> FindMaxSupportedProfileForDevice(NSString* machineName) {
+std::optional<H264ProfileLevelId> FindMaxSupportedProfileForDevice(NSString* machineName) {
const auto* result =
std::find_if(std::begin(kH264MaxSupportedProfiles),
std::end(kH264MaxSupportedProfiles),
@@ -279,14 +279,14 @@
H264ProfileLevelId fallbackProfile{H264Profile::kProfileMain, H264Level::kLevel4_1};
return fallbackProfile;
}
- return absl::nullopt;
+ return std::nullopt;
}
} // namespace
@implementation UIDevice (H264Profile)
-+ (absl::optional<webrtc::H264ProfileLevelId>)maxSupportedH264Profile {
++ (std::optional<webrtc::H264ProfileLevelId>)maxSupportedH264Profile {
return FindMaxSupportedProfileForDevice([UIDevice machineName]);
}
diff --git a/sdk/objc/native/src/objc_video_encoder_factory.mm b/sdk/objc/native/src/objc_video_encoder_factory.mm
index 1085cb8..e5fb3da 100644
--- a/sdk/objc/native/src/objc_video_encoder_factory.mm
+++ b/sdk/objc/native/src/objc_video_encoder_factory.mm
@@ -120,22 +120,22 @@
[[RTC_OBJC_TYPE(RTCVideoCodecInfo) alloc] initWithNativeSdpVideoFormat:format];
[selector_ registerCurrentEncoderInfo:info];
}
- absl::optional<SdpVideoFormat> OnEncoderBroken() override {
+ std::optional<SdpVideoFormat> OnEncoderBroken() override {
RTC_OBJC_TYPE(RTCVideoCodecInfo) *info = [selector_ encoderForBrokenEncoder];
if (info) {
return [info nativeSdpVideoFormat];
}
- return absl::nullopt;
+ return std::nullopt;
}
- absl::optional<SdpVideoFormat> OnAvailableBitrate(const DataRate &rate) override {
+ std::optional<SdpVideoFormat> OnAvailableBitrate(const DataRate &rate) override {
RTC_OBJC_TYPE(RTCVideoCodecInfo) *info = [selector_ encoderForBitrate:rate.kbps<NSInteger>()];
if (info) {
return [info nativeSdpVideoFormat];
}
- return absl::nullopt;
+ return std::nullopt;
}
- absl::optional<SdpVideoFormat> OnResolutionChange(const RenderResolution &resolution) override {
+ std::optional<SdpVideoFormat> OnResolutionChange(const RenderResolution &resolution) override {
if ([selector_ respondsToSelector:@selector(encoderForResolutionChangeBySize:)]) {
RTC_OBJC_TYPE(RTCVideoCodecInfo) *info = [selector_
encoderForResolutionChangeBySize:CGSizeMake(resolution.Width(), resolution.Height())];
@@ -143,7 +143,7 @@
return [info nativeSdpVideoFormat];
}
}
- return absl::nullopt;
+ return std::nullopt;
}
private:
diff --git a/sdk/objc/native/src/objc_video_track_source.h b/sdk/objc/native/src/objc_video_track_source.h
index 19a3d6d..242dc2a 100644
--- a/sdk/objc/native/src/objc_video_track_source.h
+++ b/sdk/objc/native/src/objc_video_track_source.h
@@ -35,7 +35,7 @@
// Indicates that the encoder should denoise video before encoding it.
// If it is not set, the default configuration is used which is different
// depending on video codec.
- absl::optional<bool> needs_denoising() const override;
+ std::optional<bool> needs_denoising() const override;
SourceState state() const override;
diff --git a/sdk/objc/native/src/objc_video_track_source.mm b/sdk/objc/native/src/objc_video_track_source.mm
index 7937e90..9da0d9a 100644
--- a/sdk/objc/native/src/objc_video_track_source.mm
+++ b/sdk/objc/native/src/objc_video_track_source.mm
@@ -48,7 +48,7 @@
return is_screencast_;
}
-absl::optional<bool> ObjCVideoTrackSource::needs_denoising() const {
+std::optional<bool> ObjCVideoTrackSource::needs_denoising() const {
return false;
}
diff --git a/stats/BUILD.gn b/stats/BUILD.gn
index 3024f39..ab4be89 100644
--- a/stats/BUILD.gn
+++ b/stats/BUILD.gn
@@ -44,7 +44,6 @@
"../api:rtc_stats_api",
"../rtc_base:checks",
"../rtc_base/system:rtc_export",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -65,7 +64,6 @@
"../rtc_base:rtc_json",
"../test:test_main",
"../test:test_support",
- "//third_party/abseil-cpp/absl/types:optional",
]
if (is_android) {
diff --git a/stats/attribute.cc b/stats/attribute.cc
index cf49cb2..52cdbc2 100644
--- a/stats/attribute.cc
+++ b/stats/attribute.cc
@@ -25,12 +25,12 @@
struct VisitIsSequence {
// Any type of vector is a sequence.
template <typename T>
- bool operator()(const absl::optional<std::vector<T>>* attribute) {
+ bool operator()(const std::optional<std::vector<T>>* attribute) {
return true;
}
// Any other type is not.
template <typename T>
- bool operator()(const absl::optional<T>* attribute) {
+ bool operator()(const std::optional<T>* attribute) {
return false;
}
};
@@ -62,7 +62,7 @@
// Vector attributes.
template <typename T>
- std::string operator()(const absl::optional<std::vector<T>>* attribute) {
+ std::string operator()(const std::optional<std::vector<T>>* attribute) {
rtc::StringBuilder sb;
sb << "[";
const char* separator = "";
@@ -84,7 +84,7 @@
// Map attributes.
template <typename T>
std::string operator()(
- const absl::optional<std::map<std::string, T>>* attribute) {
+ const std::optional<std::map<std::string, T>>* attribute) {
rtc::StringBuilder sb;
sb << "{";
const char* separator = "";
@@ -106,14 +106,14 @@
}
// Simple attributes.
template <typename T>
- std::string operator()(const absl::optional<T>* attribute) {
+ std::string operator()(const std::optional<T>* attribute) {
return ValueToString(attribute->value());
}
};
struct VisitIsEqual {
template <typename T>
- bool operator()(const absl::optional<T>* attribute) {
+ bool operator()(const std::optional<T>* attribute) {
if (!other.holds_alternative<T>()) {
return false;
}
@@ -143,8 +143,7 @@
}
bool Attribute::is_string() const {
- return absl::holds_alternative<const absl::optional<std::string>*>(
- attribute_);
+ return absl::holds_alternative<const std::optional<std::string>*>(attribute_);
}
std::string Attribute::ToString() const {
diff --git a/stats/rtc_stats_report_unittest.cc b/stats/rtc_stats_report_unittest.cc
index f11ff52..250433f 100644
--- a/stats/rtc_stats_report_unittest.cc
+++ b/stats/rtc_stats_report_unittest.cc
@@ -10,7 +10,8 @@
#include "api/stats/rtc_stats_report.h"
-#include "absl/types/optional.h"
+#include <optional>
+
#include "api/stats/attribute.h"
#include "api/stats/rtc_stats.h"
#include "rtc_base/checks.h"
@@ -25,7 +26,7 @@
RTCTestStats1(const std::string& id, Timestamp timestamp)
: RTCStats(id, timestamp) {}
- absl::optional<int32_t> integer;
+ std::optional<int32_t> integer;
};
WEBRTC_RTCSTATS_IMPL(RTCTestStats1,
@@ -40,7 +41,7 @@
RTCTestStats2(const std::string& id, Timestamp timestamp)
: RTCStats(id, timestamp) {}
- absl::optional<double> number;
+ std::optional<double> number;
};
WEBRTC_RTCSTATS_IMPL(RTCTestStats2,
@@ -55,7 +56,7 @@
RTCTestStats3(const std::string& id, Timestamp timestamp)
: RTCStats(id, timestamp) {}
- absl::optional<std::string> string;
+ std::optional<std::string> string;
};
WEBRTC_RTCSTATS_IMPL(RTCTestStats3,
diff --git a/stats/rtc_stats_unittest.cc b/stats/rtc_stats_unittest.cc
index 555360f..19f3a3f 100644
--- a/stats/rtc_stats_unittest.cc
+++ b/stats/rtc_stats_unittest.cc
@@ -14,8 +14,8 @@
#include <cstdint>
#include <cstring>
#include <iostream>
+#include <optional>
-#include "absl/types/optional.h"
#include "rtc_base/checks.h"
#include "rtc_base/strings/json.h"
#include "stats/test/rtc_test_stats.h"
@@ -47,7 +47,7 @@
RTCChildStats(const std::string& id, Timestamp timestamp)
: RTCStats(id, timestamp) {}
- absl::optional<int32_t> child_int;
+ std::optional<int32_t> child_int;
};
WEBRTC_RTCSTATS_IMPL(RTCChildStats,
@@ -62,7 +62,7 @@
RTCGrandChildStats(const std::string& id, Timestamp timestamp)
: RTCChildStats(id, timestamp) {}
- absl::optional<int32_t> grandchild_int;
+ std::optional<int32_t> grandchild_int;
};
WEBRTC_RTCSTATS_IMPL(RTCGrandChildStats,
diff --git a/stats/test/rtc_test_stats.h b/stats/test/rtc_test_stats.h
index ff1a06f..52bcd9c 100644
--- a/stats/test/rtc_test_stats.h
+++ b/stats/test/rtc_test_stats.h
@@ -13,10 +13,10 @@
#include <cstdint>
#include <map>
+#include <optional>
#include <string>
#include <vector>
-#include "absl/types/optional.h"
#include "api/stats/rtc_stats.h"
#include "rtc_base/system/rtc_export.h"
@@ -28,22 +28,22 @@
RTCTestStats(const std::string& id, Timestamp timestamp);
~RTCTestStats() override;
- absl::optional<bool> m_bool;
- absl::optional<int32_t> m_int32;
- absl::optional<uint32_t> m_uint32;
- absl::optional<int64_t> m_int64;
- absl::optional<uint64_t> m_uint64;
- absl::optional<double> m_double;
- absl::optional<std::string> m_string;
- absl::optional<std::vector<bool>> m_sequence_bool;
- absl::optional<std::vector<int32_t>> m_sequence_int32;
- absl::optional<std::vector<uint32_t>> m_sequence_uint32;
- absl::optional<std::vector<int64_t>> m_sequence_int64;
- absl::optional<std::vector<uint64_t>> m_sequence_uint64;
- absl::optional<std::vector<double>> m_sequence_double;
- absl::optional<std::vector<std::string>> m_sequence_string;
- absl::optional<std::map<std::string, uint64_t>> m_map_string_uint64;
- absl::optional<std::map<std::string, double>> m_map_string_double;
+ std::optional<bool> m_bool;
+ std::optional<int32_t> m_int32;
+ std::optional<uint32_t> m_uint32;
+ std::optional<int64_t> m_int64;
+ std::optional<uint64_t> m_uint64;
+ std::optional<double> m_double;
+ std::optional<std::string> m_string;
+ std::optional<std::vector<bool>> m_sequence_bool;
+ std::optional<std::vector<int32_t>> m_sequence_int32;
+ std::optional<std::vector<uint32_t>> m_sequence_uint32;
+ std::optional<std::vector<int64_t>> m_sequence_int64;
+ std::optional<std::vector<uint64_t>> m_sequence_uint64;
+ std::optional<std::vector<double>> m_sequence_double;
+ std::optional<std::vector<std::string>> m_sequence_string;
+ std::optional<std::map<std::string, uint64_t>> m_map_string_uint64;
+ std::optional<std::map<std::string, double>> m_map_string_double;
};
} // namespace webrtc
diff --git a/system_wrappers/BUILD.gn b/system_wrappers/BUILD.gn
index 43bae7f..5e66d4b 100644
--- a/system_wrappers/BUILD.gn
+++ b/system_wrappers/BUILD.gn
@@ -41,7 +41,6 @@
"../rtc_base/synchronization:mutex",
"../rtc_base/system:arch",
"../rtc_base/system:rtc_export",
- "//third_party/abseil-cpp/absl/types:optional",
]
if (is_android) {
diff --git a/system_wrappers/include/rtp_to_ntp_estimator.h b/system_wrappers/include/rtp_to_ntp_estimator.h
index a5c7a15..2f36981 100644
--- a/system_wrappers/include/rtp_to_ntp_estimator.h
+++ b/system_wrappers/include/rtp_to_ntp_estimator.h
@@ -14,8 +14,8 @@
#include <stdint.h>
#include <list>
+#include <optional>
-#include "absl/types/optional.h"
#include "rtc_base/checks.h"
#include "rtc_base/numerics/sequence_number_unwrapper.h"
#include "system_wrappers/include/ntp_time.h"
@@ -64,7 +64,7 @@
int consecutive_invalid_samples_ = 0;
std::list<RtcpMeasurement> measurements_;
- absl::optional<Parameters> params_;
+ std::optional<Parameters> params_;
mutable RtpTimestampUnwrapper unwrapper_;
};
} // namespace webrtc
diff --git a/system_wrappers/source/rtp_to_ntp_estimator.cc b/system_wrappers/source/rtp_to_ntp_estimator.cc
index ef5d9a7..3df4b77 100644
--- a/system_wrappers/source/rtp_to_ntp_estimator.cc
+++ b/system_wrappers/source/rtp_to_ntp_estimator.cc
@@ -116,7 +116,7 @@
RTC_LOG(LS_WARNING) << "Multiple consecutively invalid RTCP SR reports, "
"clearing measurements.";
measurements_.clear();
- params_ = absl::nullopt;
+ params_ = std::nullopt;
}
consecutive_invalid_samples_ = 0;
diff --git a/test/BUILD.gn b/test/BUILD.gn
index bec49f0..42217274 100644
--- a/test/BUILD.gn
+++ b/test/BUILD.gn
@@ -76,7 +76,6 @@
"../rtc_base/system:file_wrapper",
"../system_wrappers",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -94,7 +93,6 @@
"../api/video:video_frame",
"../rtc_base:checks",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -144,7 +142,6 @@
"../rtc_base:checks",
"../system_wrappers",
"//third_party/abseil-cpp/absl/strings",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -168,7 +165,6 @@
"../rtc_base/synchronization:mutex",
"../rtc_base/task_utils:repeating_task",
"../system_wrappers",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -310,7 +306,6 @@
"../rtc_base/synchronization:mutex",
"../rtc_base/system:arch",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -414,7 +409,6 @@
"../api/test/metrics:print_result_proxy_metrics_exporter",
"../api/test/metrics:stdout_metrics_exporter",
"../sdk:helpers_objc",
- "//third_party/abseil-cpp/absl/types:optional",
]
configs += [ ":test_support_objc_config" ]
}
@@ -497,7 +491,6 @@
"../api/video:video_frame",
"../rtc_base:checks",
"../system_wrappers",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -563,7 +556,6 @@
"../rtc_base:rtc_event",
"../rtc_base:stringutils",
"../rtc_base/system:file_wrapper",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -619,7 +611,6 @@
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
"//third_party/perfetto/include/perfetto/tracing",
]
@@ -769,7 +760,6 @@
"time_controller:time_controller_unittests",
"//third_party/abseil-cpp/absl/flags:flag",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
"//third_party/libyuv",
]
sources = [
@@ -864,7 +854,6 @@
"../rtc_base:stringutils",
"//third_party/abseil-cpp/absl/base:core_headers",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
if (is_ios) {
deps += [ ":fileutils_ios_objc" ]
@@ -895,7 +884,6 @@
"../rtc_base:macromagic",
"../rtc_base:stringutils",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
if (is_ios) {
deps += [ ":fileutils_ios_objc" ]
@@ -948,7 +936,6 @@
"../rtc_base:checks",
"../rtc_base:crypto_random",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -1016,7 +1003,6 @@
"../rtc_base/synchronization:mutex",
"../system_wrappers",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -1202,7 +1188,6 @@
"../common_audio",
"../modules/audio_device:test_audio_device_module",
"../rtc_base:buffer",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -1295,7 +1280,6 @@
"../system_wrappers:field_trial",
"../video/config:encoder_config",
"//third_party/abseil-cpp/absl/flags:flag",
- "//third_party/abseil-cpp/absl/types:optional",
]
if (!is_android && !build_with_chromium) {
deps += [ "../modules/video_capture:video_capture_internal_impl" ]
@@ -1395,7 +1379,6 @@
"../test:fileutils",
"../video/config:streams_config",
"//third_party/abseil-cpp/absl/strings",
- "//third_party/abseil-cpp/absl/types:optional",
"//third_party/libyuv",
]
}
diff --git a/test/audio_decoder_proxy_factory.h b/test/audio_decoder_proxy_factory.h
index 95606d6..192b417 100644
--- a/test/audio_decoder_proxy_factory.h
+++ b/test/audio_decoder_proxy_factory.h
@@ -41,7 +41,7 @@
std::unique_ptr<AudioDecoder> MakeAudioDecoder(
const SdpAudioFormat& /* format */,
- absl::optional<AudioCodecPairId> /* codec_pair_id */) override {
+ std::optional<AudioCodecPairId> /* codec_pair_id */) override {
return std::make_unique<DecoderProxy>(decoder_);
}
diff --git a/test/call_test.cc b/test/call_test.cc
index 543bf93..bee23d6 100644
--- a/test/call_test.cc
+++ b/test/call_test.cc
@@ -374,15 +374,15 @@
const VideoSendStream::Config& video_send_config,
Transport* rtcp_send_transport) {
CreateMatchingVideoReceiveConfigs(video_send_config, rtcp_send_transport,
- &fake_decoder_factory_, absl::nullopt,
- false, 0);
+ &fake_decoder_factory_, std::nullopt, false,
+ 0);
}
void CallTest::CreateMatchingVideoReceiveConfigs(
const VideoSendStream::Config& video_send_config,
Transport* rtcp_send_transport,
VideoDecoderFactory* decoder_factory,
- absl::optional<size_t> decode_sub_stream,
+ std::optional<size_t> decode_sub_stream,
bool receiver_reference_time_report,
int rtp_history_ms) {
AddMatchingVideoReceiveConfigs(
@@ -396,7 +396,7 @@
const VideoSendStream::Config& video_send_config,
Transport* rtcp_send_transport,
VideoDecoderFactory* decoder_factory,
- absl::optional<size_t> decode_sub_stream,
+ std::optional<size_t> decode_sub_stream,
bool receiver_reference_time_report,
int rtp_history_ms) {
RTC_DCHECK(!video_send_config.rtp.ssrcs.empty());
@@ -502,8 +502,8 @@
auto frame_generator_capturer =
std::make_unique<test::FrameGeneratorCapturer>(
clock,
- test::CreateSquareFrameGenerator(width, height, absl::nullopt,
- absl::nullopt),
+ test::CreateSquareFrameGenerator(width, height, std::nullopt,
+ std::nullopt),
framerate * speed, env_.task_queue_factory());
frame_generator_capturer_ = frame_generator_capturer.get();
frame_generator_capturer->Init();
@@ -518,8 +518,8 @@
auto frame_generator_capturer =
std::make_unique<test::FrameGeneratorCapturer>(
&env_.clock(),
- test::CreateSquareFrameGenerator(width, height, absl::nullopt,
- absl::nullopt),
+ test::CreateSquareFrameGenerator(width, height, std::nullopt,
+ std::nullopt),
framerate, env_.task_queue_factory());
frame_generator_capturer_ = frame_generator_capturer.get();
frame_generator_capturer->Init();
@@ -744,20 +744,20 @@
flexfec_recv_stream->OnRtpPacket(packet);
}
-absl::optional<RtpExtension> CallTest::GetRtpExtensionByUri(
+std::optional<RtpExtension> CallTest::GetRtpExtensionByUri(
const std::string& uri) const {
for (const auto& extension : rtp_extensions_) {
if (extension.uri == uri) {
return extension;
}
}
- return absl::nullopt;
+ return std::nullopt;
}
void CallTest::AddRtpExtensionByUri(
const std::string& uri,
std::vector<RtpExtension>* extensions) const {
- const absl::optional<RtpExtension> extension = GetRtpExtensionByUri(uri);
+ const std::optional<RtpExtension> extension = GetRtpExtensionByUri(uri);
if (extension) {
extensions->push_back(*extension);
}
diff --git a/test/call_test.h b/test/call_test.h
index 2446a0f..8c97848 100644
--- a/test/call_test.h
+++ b/test/call_test.h
@@ -12,10 +12,10 @@
#include <map>
#include <memory>
+#include <optional>
#include <string>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/audio/audio_device.h"
#include "api/environment/environment.h"
@@ -115,7 +115,7 @@
const VideoSendStream::Config& video_send_config,
Transport* rtcp_send_transport,
VideoDecoderFactory* decoder_factory,
- absl::optional<size_t> decode_sub_stream,
+ std::optional<size_t> decode_sub_stream,
bool receiver_reference_time_report,
int rtp_history_ms);
void AddMatchingVideoReceiveConfigs(
@@ -123,7 +123,7 @@
const VideoSendStream::Config& video_send_config,
Transport* rtcp_send_transport,
VideoDecoderFactory* decoder_factory,
- absl::optional<size_t> decode_sub_stream,
+ std::optional<size_t> decode_sub_stream,
bool receiver_reference_time_report,
int rtp_history_ms);
@@ -241,7 +241,7 @@
test::FakeVideoRenderer fake_renderer_;
private:
- absl::optional<RtpExtension> GetRtpExtensionByUri(
+ std::optional<RtpExtension> GetRtpExtensionByUri(
const std::string& uri) const;
void AddRtpExtensionByUri(const std::string& uri,
diff --git a/test/configurable_frame_size_encoder.h b/test/configurable_frame_size_encoder.h
index 747fc09..1fa032a 100644
--- a/test/configurable_frame_size_encoder.h
+++ b/test/configurable_frame_size_encoder.h
@@ -16,9 +16,9 @@
#include <functional>
#include <memory>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/video/video_bitrate_allocation.h"
#include "api/video/video_frame.h"
#include "api/video_codecs/video_codec.h"
@@ -60,7 +60,7 @@
private:
EncodedImageCallback* callback_;
- absl::optional<std::function<void(void)>> post_encode_callback_;
+ std::optional<std::function<void(void)>> post_encode_callback_;
size_t current_frame_size_;
VideoCodecType codec_type_;
diff --git a/test/create_frame_generator_capturer.h b/test/create_frame_generator_capturer.h
index 0d8ec71..de12d73 100644
--- a/test/create_frame_generator_capturer.h
+++ b/test/create_frame_generator_capturer.h
@@ -11,10 +11,10 @@
#define TEST_CREATE_FRAME_GENERATOR_CAPTURER_H_
#include <memory>
+#include <optional>
#include <string>
#include <vector>
-#include "absl/types/optional.h"
#include "api/task_queue/task_queue_factory.h"
#include "api/test/frame_generator_interface.h"
#include "api/units/time_delta.h"
@@ -26,13 +26,13 @@
namespace frame_gen_cap_impl {
template <typename T>
-class AutoOpt : public absl::optional<T> {
+class AutoOpt : public std::optional<T> {
public:
- using absl::optional<T>::optional;
+ using std::optional<T>::optional;
T* operator->() {
- if (!absl::optional<T>::has_value())
+ if (!std::optional<T>::has_value())
this->emplace(T());
- return absl::optional<T>::operator->();
+ return std::optional<T>::operator->();
}
};
} // namespace frame_gen_cap_impl
@@ -67,8 +67,8 @@
TimeDelta change_interval = TimeDelta::Seconds(10);
struct Crop {
TimeDelta scroll_duration = TimeDelta::Seconds(0);
- absl::optional<int> width;
- absl::optional<int> height;
+ std::optional<int> width;
+ std::optional<int> height;
} crop;
int width = 1850;
int height = 1110;
diff --git a/test/direct_transport.cc b/test/direct_transport.cc
index 3aa8508..6b2c5f8 100644
--- a/test/direct_transport.cc
+++ b/test/direct_transport.cc
@@ -123,9 +123,9 @@
}
void DirectTransport::ProcessPackets() {
- absl::optional<int64_t> initial_delay_ms =
+ std::optional<int64_t> initial_delay_ms =
fake_network_->TimeUntilNextProcess();
- if (initial_delay_ms == absl::nullopt)
+ if (initial_delay_ms == std::nullopt)
return;
next_process_task_ = RepeatingTaskHandle::DelayedStart(
diff --git a/test/fake_encoded_frame.h b/test/fake_encoded_frame.h
index 36d4781..cfac3d9 100644
--- a/test/fake_encoded_frame.h
+++ b/test/fake_encoded_frame.h
@@ -65,15 +65,15 @@
std::unique_ptr<FakeEncodedFrame> Build();
private:
- absl::optional<uint32_t> rtp_timestamp_;
- absl::optional<int64_t> frame_id_;
- absl::optional<VideoPlayoutDelay> playout_delay_;
- absl::optional<int> spatial_layer_;
- absl::optional<Timestamp> received_time_;
- absl::optional<int> payload_type_;
- absl::optional<Timestamp> ntp_time_;
- absl::optional<VideoRotation> rotation_;
- absl::optional<RtpPacketInfos> packet_infos_;
+ std::optional<uint32_t> rtp_timestamp_;
+ std::optional<int64_t> frame_id_;
+ std::optional<VideoPlayoutDelay> playout_delay_;
+ std::optional<int> spatial_layer_;
+ std::optional<Timestamp> received_time_;
+ std::optional<int> payload_type_;
+ std::optional<Timestamp> ntp_time_;
+ std::optional<VideoRotation> rotation_;
+ std::optional<RtpPacketInfos> packet_infos_;
std::vector<int64_t> references_;
bool last_spatial_layer_ = false;
size_t size_ = 10;
diff --git a/test/fake_encoder.cc b/test/fake_encoder.cc
index 4ff0926..a2f49a0 100644
--- a/test/fake_encoder.cc
+++ b/test/fake_encoder.cc
@@ -105,7 +105,7 @@
RateControlParameters rates;
bool keyframe;
uint32_t counter;
- absl::optional<int> qp;
+ std::optional<int> qp;
{
MutexLock lock(&mutex_);
max_framerate = config_.maxFramerate;
diff --git a/test/fake_encoder.h b/test/fake_encoder.h
index 203e67c..d595a8c 100644
--- a/test/fake_encoder.h
+++ b/test/fake_encoder.h
@@ -15,10 +15,10 @@
#include <stdint.h>
#include <memory>
+#include <optional>
#include <vector>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/environment/environment.h"
#include "api/fec_controller_override.h"
#include "api/sequence_checker.h"
@@ -112,8 +112,8 @@
uint32_t counter_ RTC_GUARDED_BY(mutex_);
mutable Mutex mutex_;
bool used_layers_[kMaxSimulcastStreams];
- absl::optional<int> qp_ RTC_GUARDED_BY(mutex_);
- absl::optional<std::string> implementation_name_ RTC_GUARDED_BY(mutex_);
+ std::optional<int> qp_ RTC_GUARDED_BY(mutex_);
+ std::optional<std::string> implementation_name_ RTC_GUARDED_BY(mutex_);
// Current byte debt to be payed over a number of frames.
// The debt is acquired by keyframes overshooting the bitrate target.
diff --git a/test/fake_vp8_decoder.cc b/test/fake_vp8_decoder.cc
index 8f29b74..3a47939 100644
--- a/test/fake_vp8_decoder.cc
+++ b/test/fake_vp8_decoder.cc
@@ -12,7 +12,8 @@
#include <stddef.h>
-#include "absl/types/optional.h"
+#include <optional>
+
#include "api/scoped_refptr.h"
#include "api/video/i420_buffer.h"
#include "api/video/video_frame.h"
@@ -60,8 +61,8 @@
frame.set_rtp_timestamp(input.RtpTimestamp());
frame.set_ntp_time_ms(input.ntp_time_ms_);
- callback_->Decoded(frame, /*decode_time_ms=*/absl::nullopt,
- /*qp=*/absl::nullopt);
+ callback_->Decoded(frame, /*decode_time_ms=*/std::nullopt,
+ /*qp=*/std::nullopt);
return WEBRTC_VIDEO_CODEC_OK;
}
diff --git a/test/fake_vp8_encoder.cc b/test/fake_vp8_encoder.cc
index 50a3009..a3e77ea 100644
--- a/test/fake_vp8_encoder.cc
+++ b/test/fake_vp8_encoder.cc
@@ -11,8 +11,8 @@
#include "test/fake_vp8_encoder.h"
#include <algorithm>
+#include <optional>
-#include "absl/types/optional.h"
#include "api/video_codecs/video_encoder.h"
#include "api/video_codecs/vp8_temporal_layers.h"
#include "api/video_codecs/vp8_temporal_layers_factory.h"
diff --git a/test/frame_generator.cc b/test/frame_generator.cc
index d2fa953..7ef563a 100644
--- a/test/frame_generator.cc
+++ b/test/frame_generator.cc
@@ -100,7 +100,7 @@
buffer = NV12Buffer::Copy(*buffer->ToI420());
}
- return VideoFrameData(buffer, absl::nullopt);
+ return VideoFrameData(buffer, std::nullopt);
}
SquareGenerator::Square::Square(int width, int height, int seed)
@@ -298,7 +298,7 @@
if (++current_display_count_ >= frame_display_count_)
current_display_count_ = 0;
- return VideoFrameData(buffer_, absl::nullopt);
+ return VideoFrameData(buffer_, std::nullopt);
}
FrameGeneratorInterface::Resolution SlideGenerator::GetResolution() const {
@@ -363,8 +363,8 @@
target_height_(static_cast<int>(target_height)),
current_frame_num_(num_frames_ - 1),
prev_frame_not_scrolled_(false),
- current_source_frame_(nullptr, absl::nullopt),
- current_frame_(nullptr, absl::nullopt),
+ current_source_frame_(nullptr, std::nullopt),
+ current_frame_(nullptr, std::nullopt),
file_generator_(files, source_width, source_height, 1) {
RTC_DCHECK(clock_ != nullptr);
RTC_DCHECK_GT(num_frames_, 0);
diff --git a/test/frame_generator.h b/test/frame_generator.h
index 76f195d..471ce52 100644
--- a/test/frame_generator.h
+++ b/test/frame_generator.h
@@ -40,7 +40,7 @@
VideoFrameData NextFrame() override;
Resolution GetResolution() const override;
- absl::optional<int> fps() const override { return absl::nullopt; }
+ std::optional<int> fps() const override { return std::nullopt; }
private:
rtc::scoped_refptr<I420Buffer> CreateI420Buffer(int width, int height);
@@ -84,7 +84,7 @@
}
Resolution GetResolution() const override;
- absl::optional<int> fps() const override { return absl::nullopt; }
+ std::optional<int> fps() const override { return std::nullopt; }
private:
// Returns true if the new frame was loaded.
@@ -119,7 +119,7 @@
}
Resolution GetResolution() const override;
- absl::optional<int> fps() const override { return absl::nullopt; }
+ std::optional<int> fps() const override { return std::nullopt; }
private:
// Returns true if the new frame was loaded.
@@ -151,7 +151,7 @@
}
Resolution GetResolution() const override;
- absl::optional<int> fps() const override { return absl::nullopt; }
+ std::optional<int> fps() const override { return std::nullopt; }
private:
// Generates some randomly sized and colored squares scattered
@@ -185,7 +185,7 @@
}
Resolution GetResolution() const override;
- absl::optional<int> fps() const override { return absl::nullopt; }
+ std::optional<int> fps() const override { return std::nullopt; }
private:
void UpdateSourceFrame(size_t frame_num);
diff --git a/test/frame_generator_capturer.cc b/test/frame_generator_capturer.cc
index 4641f4e..c96b056 100644
--- a/test/frame_generator_capturer.cc
+++ b/test/frame_generator_capturer.cc
@@ -14,9 +14,9 @@
#include <cmath>
#include <cstddef>
#include <memory>
+#include <optional>
#include <utility>
-#include "absl/types/optional.h"
#include "api/task_queue/task_queue_base.h"
#include "api/task_queue/task_queue_factory.h"
#include "api/test/frame_generator_interface.h"
@@ -67,7 +67,7 @@
}
void FrameGeneratorCapturer::SetFakeColorSpace(
- absl::optional<ColorSpace> color_space) {
+ std::optional<ColorSpace> color_space) {
MutexLock lock(&lock_);
fake_color_space_ = color_space;
}
@@ -112,7 +112,7 @@
}
}
-absl::optional<FrameGeneratorCapturer::Resolution>
+std::optional<FrameGeneratorCapturer::Resolution>
FrameGeneratorCapturer::GetResolution() const {
FrameGeneratorInterface::Resolution resolution =
frame_generator_->GetResolution();
@@ -175,7 +175,7 @@
void FrameGeneratorCapturer::OnOutputFormatRequest(
int width,
int height,
- const absl::optional<int>& max_fps) {
+ const std::optional<int>& max_fps) {
TestVideoCapturer::OnOutputFormatRequest(width, height, max_fps);
}
diff --git a/test/frame_generator_capturer.h b/test/frame_generator_capturer.h
index c1d0130..078bcec 100644
--- a/test/frame_generator_capturer.h
+++ b/test/frame_generator_capturer.h
@@ -13,8 +13,8 @@
#include <cstddef>
#include <cstdint>
#include <memory>
+#include <optional>
-#include "absl/types/optional.h"
#include "api/task_queue/task_queue_base.h"
#include "api/task_queue/task_queue_factory.h"
#include "api/test/frame_generator_interface.h"
@@ -64,11 +64,11 @@
int width;
int height;
};
- absl::optional<Resolution> GetResolution() const;
+ std::optional<Resolution> GetResolution() const;
void OnOutputFormatRequest(int width,
int height,
- const absl::optional<int>& max_fps);
+ const std::optional<int>& max_fps);
void SetSinkWantsObserver(SinkWantsObserver* observer);
@@ -78,7 +78,7 @@
void ForceFrame();
void SetFakeRotation(VideoRotation rotation);
- void SetFakeColorSpace(absl::optional<ColorSpace> color_space);
+ void SetFakeColorSpace(std::optional<ColorSpace> color_space);
bool Init();
@@ -98,7 +98,7 @@
int source_fps_ RTC_GUARDED_BY(&lock_);
int target_capture_fps_ RTC_GUARDED_BY(&lock_);
VideoRotation fake_rotation_ = kVideoRotation_0;
- absl::optional<ColorSpace> fake_color_space_ RTC_GUARDED_BY(&lock_);
+ std::optional<ColorSpace> fake_color_space_ RTC_GUARDED_BY(&lock_);
std::unique_ptr<TaskQueueBase, TaskQueueDeleter> task_queue_;
};
diff --git a/test/function_audio_decoder_factory.h b/test/function_audio_decoder_factory.h
index 2848ce8..c314249 100644
--- a/test/function_audio_decoder_factory.h
+++ b/test/function_audio_decoder_factory.h
@@ -32,14 +32,14 @@
std::function<std::unique_ptr<AudioDecoder>()> create)
: create_([create](const Environment&,
const SdpAudioFormat&,
- absl::optional<AudioCodecPairId> codec_pair_id) {
+ std::optional<AudioCodecPairId> codec_pair_id) {
return create();
}) {}
explicit FunctionAudioDecoderFactory(
std::function<std::unique_ptr<AudioDecoder>(
const Environment&,
const SdpAudioFormat&,
- absl::optional<AudioCodecPairId> codec_pair_id)> create)
+ std::optional<AudioCodecPairId> codec_pair_id)> create)
: create_(std::move(create)) {}
// Unused by tests.
@@ -55,7 +55,7 @@
std::unique_ptr<AudioDecoder> Create(
const Environment& env,
const SdpAudioFormat& format,
- absl::optional<AudioCodecPairId> codec_pair_id) override {
+ std::optional<AudioCodecPairId> codec_pair_id) override {
return create_(env, format, codec_pair_id);
}
@@ -63,7 +63,7 @@
const std::function<std::unique_ptr<AudioDecoder>(
const Environment&,
const SdpAudioFormat&,
- absl::optional<AudioCodecPairId> codec_pair_id)>
+ std::optional<AudioCodecPairId> codec_pair_id)>
create_;
};
diff --git a/test/fuzzers/BUILD.gn b/test/fuzzers/BUILD.gn
index 68c4d11..6d28c7b 100644
--- a/test/fuzzers/BUILD.gn
+++ b/test/fuzzers/BUILD.gn
@@ -238,7 +238,6 @@
deps = [
"../../common_video:common_video",
"../../modules/rtp_rtcp:rtp_rtcp_format",
- "//third_party/abseil-cpp/absl/types:optional",
]
seed_corpus = "corpora/rtp-corpus"
}
@@ -307,7 +306,6 @@
"../../api/audio_codecs:audio_codecs_api",
"../../modules/rtp_rtcp:rtp_rtcp_format",
"../../rtc_base:checks",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -590,7 +588,6 @@
"../../api/audio:audio_processing",
"../../modules/audio_processing:audio_buffer",
"../../modules/audio_processing/aec3",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
diff --git a/test/fuzzers/aec3_fuzzer.cc b/test/fuzzers/aec3_fuzzer.cc
index 2e6d12e..71a316d 100644
--- a/test/fuzzers/aec3_fuzzer.cc
+++ b/test/fuzzers/aec3_fuzzer.cc
@@ -8,7 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "absl/types/optional.h"
+#include <optional>
+
#include "api/audio/audio_processing.h"
#include "modules/audio_processing/aec3/echo_canceller3.h"
#include "modules/audio_processing/audio_buffer.h"
@@ -53,7 +54,7 @@
1 + fuzz_data.ReadOrDefaultValue<uint8_t>(0) % (kMaxNumChannels - 1);
EchoCanceller3 aec3(EchoCanceller3Config(),
- /*multichannel_config=*/absl::nullopt, sample_rate_hz,
+ /*multichannel_config=*/std::nullopt, sample_rate_hz,
num_render_channels, num_capture_channels);
AudioBuffer capture_audio(sample_rate_hz, num_capture_channels,
diff --git a/test/fuzzers/audio_decoder_fuzzer.cc b/test/fuzzers/audio_decoder_fuzzer.cc
index 1db332e..ccfd604 100644
--- a/test/fuzzers/audio_decoder_fuzzer.cc
+++ b/test/fuzzers/audio_decoder_fuzzer.cc
@@ -11,8 +11,8 @@
#include "test/fuzzers/audio_decoder_fuzzer.h"
#include <limits>
+#include <optional>
-#include "absl/types/optional.h"
#include "api/audio_codecs/audio_decoder.h"
#include "modules/rtp_rtcp/source/byte_io.h"
#include "rtc_base/checks.h"
diff --git a/test/fuzzers/dcsctp_packet_fuzzer.cc b/test/fuzzers/dcsctp_packet_fuzzer.cc
index 2fc3fe1..339f7a5 100644
--- a/test/fuzzers/dcsctp_packet_fuzzer.cc
+++ b/test/fuzzers/dcsctp_packet_fuzzer.cc
@@ -14,7 +14,7 @@
using dcsctp::SctpPacket;
void FuzzOneInput(const uint8_t* data, size_t size) {
- absl::optional<SctpPacket> c =
+ std::optional<SctpPacket> c =
SctpPacket::Parse(rtc::ArrayView<const uint8_t>(data, size),
/*disable_checksum_verification=*/true);
diff --git a/test/fuzzers/neteq_rtp_fuzzer.cc b/test/fuzzers/neteq_rtp_fuzzer.cc
index 3caa5fe..e6933e3 100644
--- a/test/fuzzers/neteq_rtp_fuzzer.cc
+++ b/test/fuzzers/neteq_rtp_fuzzer.cc
@@ -69,15 +69,15 @@
MaybeFuzzPayload();
}
- absl::optional<int64_t> NextPacketTime() const override {
+ std::optional<int64_t> NextPacketTime() const override {
return packet_->time_ms;
}
- absl::optional<int64_t> NextOutputEventTime() const override {
+ std::optional<int64_t> NextOutputEventTime() const override {
return input_->NextOutputEventTime();
}
- absl::optional<SetMinimumDelayInfo> NextSetMinimumDelayInfo() const override {
+ std::optional<SetMinimumDelayInfo> NextSetMinimumDelayInfo() const override {
return input_->NextSetMinimumDelayInfo();
}
@@ -98,7 +98,7 @@
bool ended() const override { return ended_; }
- absl::optional<RTPHeader> NextHeader() const override {
+ std::optional<RTPHeader> NextHeader() const override {
RTC_DCHECK(packet_);
return packet_->header;
}
diff --git a/test/fuzzers/neteq_signal_fuzzer.cc b/test/fuzzers/neteq_signal_fuzzer.cc
index 3b1f70c..189f1dd 100644
--- a/test/fuzzers/neteq_signal_fuzzer.cc
+++ b/test/fuzzers/neteq_signal_fuzzer.cc
@@ -89,15 +89,15 @@
output_event_period_ms_ = fuzz_data_.SelectOneOf(output_event_periods);
}
- absl::optional<int64_t> NextPacketTime() const override {
+ std::optional<int64_t> NextPacketTime() const override {
return packet_->time_ms;
}
- absl::optional<int64_t> NextOutputEventTime() const override {
+ std::optional<int64_t> NextOutputEventTime() const override {
return next_output_event_ms_;
}
- absl::optional<SetMinimumDelayInfo> NextSetMinimumDelayInfo() const override {
+ std::optional<SetMinimumDelayInfo> NextSetMinimumDelayInfo() const override {
return input_->NextSetMinimumDelayInfo();
}
@@ -133,7 +133,7 @@
bool ended() const override { return ended_; }
- absl::optional<RTPHeader> NextHeader() const override {
+ std::optional<RTPHeader> NextHeader() const override {
RTC_DCHECK(packet_);
return packet_->header;
}
diff --git a/test/fuzzers/rtp_frame_reference_finder_fuzzer.cc b/test/fuzzers/rtp_frame_reference_finder_fuzzer.cc
index f1c4904..5be9837c 100644
--- a/test/fuzzers/rtp_frame_reference_finder_fuzzer.cc
+++ b/test/fuzzers/rtp_frame_reference_finder_fuzzer.cc
@@ -68,9 +68,9 @@
return result;
}
-absl::optional<RTPVideoHeader::GenericDescriptorInfo>
+std::optional<RTPVideoHeader::GenericDescriptorInfo>
GenerateGenericFrameDependencies(DataReader* reader) {
- absl::optional<RTPVideoHeader::GenericDescriptorInfo> result;
+ std::optional<RTPVideoHeader::GenericDescriptorInfo> result;
uint8_t flags = reader->GetNum<uint8_t>();
if (flags & 0b1000'0000) {
// i.e. with 50% chance there are no generic dependencies.
@@ -156,7 +156,7 @@
kVideoRotation_0,
VideoContentType::UNSPECIFIED,
video_header,
- /*color_space=*/absl::nullopt,
+ /*color_space=*/std::nullopt,
RtpPacketInfos(),
EncodedImageBuffer::Create(/*size=*/0));
// clang-format on
diff --git a/test/fuzzers/rtp_packet_fuzzer.cc b/test/fuzzers/rtp_packet_fuzzer.cc
index 5d7c31d..eaa423e 100644
--- a/test/fuzzers/rtp_packet_fuzzer.cc
+++ b/test/fuzzers/rtp_packet_fuzzer.cc
@@ -9,9 +9,9 @@
*/
#include <bitset>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "common_video/corruption_detection_message.h"
#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
#include "modules/rtp_rtcp/source/corruption_detection_extension.h"
@@ -103,7 +103,7 @@
break;
case kRtpExtensionTransportSequenceNumber02: {
uint16_t seqnum;
- absl::optional<FeedbackRequest> feedback_request;
+ std::optional<FeedbackRequest> feedback_request;
packet.GetExtension<TransportSequenceNumberV2>(&seqnum,
&feedback_request);
break;
@@ -148,7 +148,7 @@
break;
}
case kRtpExtensionInbandComfortNoise: {
- absl::optional<uint8_t> noise_level;
+ std::optional<uint8_t> noise_level;
packet.GetExtension<InbandComfortNoiseExtension>(&noise_level);
break;
}
diff --git a/test/ios/test_support.h b/test/ios/test_support.h
index 5ac7313..1f8f962 100644
--- a/test/ios/test_support.h
+++ b/test/ios/test_support.h
@@ -11,11 +11,10 @@
#ifndef TEST_IOS_TEST_SUPPORT_H_
#define TEST_IOS_TEST_SUPPORT_H_
+#include <optional>
#include <string>
#include <vector>
-#include "absl/types/optional.h"
-
namespace rtc {
namespace test {
// Launches an iOS app that serves as a host for a test suite.
@@ -28,7 +27,7 @@
bool save_chartjson_result,
bool export_perf_results_new_api,
std::string webrtc_test_metrics_output_path,
- absl::optional<std::vector<std::string>> metrics_to_plot);
+ std::optional<std::vector<std::string>> metrics_to_plot);
// Returns true if unittests should be run by the XCTest runnner.
bool ShouldRunIOSUnittestsWithXCTest();
diff --git a/test/ios/test_support.mm b/test/ios/test_support.mm
index d3c9ee0..80208d5 100644
--- a/test/ios/test_support.mm
+++ b/test/ios/test_support.mm
@@ -46,8 +46,8 @@
static bool g_write_perf_output;
static bool g_export_perf_results_new_api;
static std::string g_webrtc_test_metrics_output_path;
-static absl::optional<bool> g_is_xctest;
-static absl::optional<std::vector<std::string>> g_metrics_to_plot;
+static std::optional<bool> g_is_xctest;
+static std::optional<std::vector<std::string>> g_metrics_to_plot;
@interface UIApplication (Testing)
- (void)_terminateWithStatus:(int)status;
@@ -180,7 +180,7 @@
bool write_perf_output,
bool export_perf_results_new_api,
std::string webrtc_test_metrics_output_path,
- absl::optional<std::vector<std::string>> metrics_to_plot) {
+ std::optional<std::vector<std::string>> metrics_to_plot) {
g_test_suite = test_suite;
g_argc = argc;
g_argv = argv;
@@ -204,12 +204,12 @@
char **argv = g_argv;
while (*argv != nullptr) {
if (strstr(*argv, kEnableRunIOSUnittestsWithXCTest) != nullptr) {
- g_is_xctest = absl::optional<bool>(true);
+ g_is_xctest = std::optional<bool>(true);
return true;
}
argv++;
}
- g_is_xctest = absl::optional<bool>(false);
+ g_is_xctest = std::optional<bool>(false);
return false;
}
diff --git a/test/jitter/BUILD.gn b/test/jitter/BUILD.gn
index 393ef28..c14c94c 100644
--- a/test/jitter/BUILD.gn
+++ b/test/jitter/BUILD.gn
@@ -24,7 +24,6 @@
"../../api/video:video_frame_type",
"../../rtc_base:logging",
"../../rtc_base:rtc_numerics",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
diff --git a/test/jitter/delay_variation_calculator.cc b/test/jitter/delay_variation_calculator.cc
index 092bd7c..2d409f8 100644
--- a/test/jitter/delay_variation_calculator.cc
+++ b/test/jitter/delay_variation_calculator.cc
@@ -10,9 +10,9 @@
#include "test/jitter/delay_variation_calculator.h"
+#include <optional>
#include <string>
-#include "absl/types/optional.h"
#include "api/units/frequency.h"
#include "api/units/time_delta.h"
#include "api/units/timestamp.h"
@@ -29,9 +29,9 @@
uint32_t rtp_timestamp,
Timestamp arrival_time,
DataSize size,
- absl::optional<int> spatial_layer,
- absl::optional<int> temporal_layer,
- absl::optional<VideoFrameType> frame_type) {
+ std::optional<int> spatial_layer,
+ std::optional<int> temporal_layer,
+ std::optional<VideoFrameType> frame_type) {
Frame frame{.rtp_timestamp = rtp_timestamp,
.unwrapped_rtp_timestamp = unwrapper_.Unwrap(rtp_timestamp),
.arrival_time = arrival_time,
diff --git a/test/jitter/delay_variation_calculator.h b/test/jitter/delay_variation_calculator.h
index 6400f82..6b06afe 100644
--- a/test/jitter/delay_variation_calculator.h
+++ b/test/jitter/delay_variation_calculator.h
@@ -14,9 +14,9 @@
#include <stdint.h>
#include <map>
+#include <optional>
#include <string>
-#include "absl/types/optional.h"
#include "api/numerics/samples_stats_counter.h"
#include "api/test/metrics/metrics_logger.h"
#include "api/units/data_size.h"
@@ -56,9 +56,9 @@
void Insert(uint32_t rtp_timestamp,
Timestamp arrival_time,
DataSize size,
- absl::optional<int> spatial_layer = absl::nullopt,
- absl::optional<int> temporal_layer = absl::nullopt,
- absl::optional<VideoFrameType> frame_type = absl::nullopt);
+ std::optional<int> spatial_layer = std::nullopt,
+ std::optional<int> temporal_layer = std::nullopt,
+ std::optional<VideoFrameType> frame_type = std::nullopt);
const TimeSeries& time_series() const { return time_series_; }
@@ -68,9 +68,9 @@
int64_t unwrapped_rtp_timestamp;
Timestamp arrival_time;
DataSize size;
- absl::optional<int> spatial_layer;
- absl::optional<int> temporal_layer;
- absl::optional<VideoFrameType> frame_type;
+ std::optional<int> spatial_layer;
+ std::optional<int> temporal_layer;
+ std::optional<VideoFrameType> frame_type;
};
using MetadataT = std::map<std::string, std::string>;
@@ -84,7 +84,7 @@
MetadataT BuildMetadata(const Frame& frame);
RtpTimestampUnwrapper unwrapper_;
- absl::optional<Frame> prev_frame_ = absl::nullopt;
+ std::optional<Frame> prev_frame_ = std::nullopt;
TimeSeries time_series_;
};
diff --git a/test/jitter/logging_delay_variation_calculator.cc b/test/jitter/logging_delay_variation_calculator.cc
index aa58b0f..dc08a17 100644
--- a/test/jitter/logging_delay_variation_calculator.cc
+++ b/test/jitter/logging_delay_variation_calculator.cc
@@ -20,9 +20,9 @@
uint32_t rtp_timestamp,
Timestamp arrival_time,
DataSize size,
- absl::optional<int> spatial_layer,
- absl::optional<int> temporal_layer,
- absl::optional<VideoFrameType> frame_type) {
+ std::optional<int> spatial_layer,
+ std::optional<int> temporal_layer,
+ std::optional<VideoFrameType> frame_type) {
calc_.Insert(rtp_timestamp, arrival_time, size, spatial_layer, temporal_layer,
frame_type);
}
diff --git a/test/jitter/logging_delay_variation_calculator.h b/test/jitter/logging_delay_variation_calculator.h
index f15d2ae..a3a67af 100644
--- a/test/jitter/logging_delay_variation_calculator.h
+++ b/test/jitter/logging_delay_variation_calculator.h
@@ -35,9 +35,9 @@
void Insert(uint32_t rtp_timestamp,
Timestamp arrival_time,
DataSize size,
- absl::optional<int> spatial_layer = absl::nullopt,
- absl::optional<int> temporal_layer = absl::nullopt,
- absl::optional<VideoFrameType> frame_type = absl::nullopt);
+ std::optional<int> spatial_layer = std::nullopt,
+ std::optional<int> temporal_layer = std::nullopt,
+ std::optional<VideoFrameType> frame_type = std::nullopt);
private:
void LogMetrics() const;
diff --git a/test/mock_audio_decoder_factory.h b/test/mock_audio_decoder_factory.h
index a577fdaf..d14a7dc 100644
--- a/test/mock_audio_decoder_factory.h
+++ b/test/mock_audio_decoder_factory.h
@@ -51,7 +51,7 @@
Create,
(const Environment&,
const SdpAudioFormat&,
- absl::optional<AudioCodecPairId>),
+ std::optional<AudioCodecPairId>),
(override));
};
diff --git a/test/mock_audio_encoder.h b/test/mock_audio_encoder.h
index 30518e8..19c7d9c 100644
--- a/test/mock_audio_encoder.h
+++ b/test/mock_audio_encoder.h
@@ -29,11 +29,11 @@
MOCK_METHOD(size_t, Num10MsFramesInNextPacket, (), (const, override));
MOCK_METHOD(size_t, Max10MsFramesInAPacket, (), (const, override));
MOCK_METHOD(int, GetTargetBitrate, (), (const, override));
- MOCK_METHOD((absl::optional<std::pair<TimeDelta, TimeDelta>>),
+ MOCK_METHOD((std::optional<std::pair<TimeDelta, TimeDelta>>),
GetFrameLengthRange,
(),
(const, override));
- MOCK_METHOD((absl::optional<std::pair<DataRate, DataRate>>),
+ MOCK_METHOD((std::optional<std::pair<DataRate, DataRate>>),
GetBitrateRange,
(),
(const, override));
@@ -46,7 +46,7 @@
MOCK_METHOD(void,
OnReceivedUplinkBandwidth,
(int target_audio_bitrate_bps,
- absl::optional<int64_t> probing_interval_ms),
+ std::optional<int64_t> probing_interval_ms),
(override));
MOCK_METHOD(void,
OnReceivedUplinkPacketLossFraction,
diff --git a/test/mock_audio_encoder_factory.h b/test/mock_audio_encoder_factory.h
index 9f1b0ce..0d3c96a 100644
--- a/test/mock_audio_encoder_factory.h
+++ b/test/mock_audio_encoder_factory.h
@@ -29,7 +29,7 @@
GetSupportedEncoders,
(),
(override));
- MOCK_METHOD(absl::optional<AudioCodecInfo>,
+ MOCK_METHOD(std::optional<AudioCodecInfo>,
QueryAudioEncoder,
(const SdpAudioFormat& format),
(override));
diff --git a/test/network/BUILD.gn b/test/network/BUILD.gn
index ab92d88..8de643e 100644
--- a/test/network/BUILD.gn
+++ b/test/network/BUILD.gn
@@ -89,7 +89,6 @@
"//third_party/abseil-cpp/absl/base:nullability",
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -138,7 +137,6 @@
"../../rtc_base:rtc_event",
"../time_controller",
"//third_party/abseil-cpp/absl/memory",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -223,7 +221,6 @@
"../../rtc_base:race_checker",
"../../rtc_base:random",
"../../rtc_base/synchronization:mutex",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
diff --git a/test/network/cross_traffic.cc b/test/network/cross_traffic.cc
index dd34bd0..f5a0359 100644
--- a/test/network/cross_traffic.cc
+++ b/test/network/cross_traffic.cc
@@ -12,10 +12,10 @@
#include <math.h>
+#include <optional>
#include <utility>
#include "absl/memory/memory.h"
-#include "absl/types/optional.h"
#include "cross_traffic.h"
#include "rtc_base/logging.h"
#include "rtc_base/numerics/safe_minmax.h"
diff --git a/test/network/cross_traffic_unittest.cc b/test/network/cross_traffic_unittest.cc
index 5382f66..80f4d02 100644
--- a/test/network/cross_traffic_unittest.cc
+++ b/test/network/cross_traffic_unittest.cc
@@ -12,11 +12,11 @@
#include <atomic>
#include <memory>
+#include <optional>
#include <utility>
#include <vector>
#include "absl/memory/memory.h"
-#include "absl/types/optional.h"
#include "api/test/network_emulation_manager.h"
#include "api/test/simulated_network.h"
#include "api/units/data_rate.h"
diff --git a/test/network/fake_network_socket_server.cc b/test/network/fake_network_socket_server.cc
index 63828e5..8dcca34 100644
--- a/test/network/fake_network_socket_server.cc
+++ b/test/network/fake_network_socket_server.cc
@@ -75,7 +75,7 @@
int error_ RTC_GUARDED_BY(&thread_);
std::map<Option, int> options_map_ RTC_GUARDED_BY(&thread_);
- absl::optional<EmulatedIpPacket> pending_ RTC_GUARDED_BY(thread_);
+ std::optional<EmulatedIpPacket> pending_ RTC_GUARDED_BY(thread_);
rtc::scoped_refptr<PendingTaskSafetyFlag> alive_;
};
@@ -135,7 +135,7 @@
error_ = EADDRNOTAVAIL;
return 2;
}
- absl::optional<uint16_t> port =
+ std::optional<uint16_t> port =
endpoint_->BindReceiver(local_addr_.port(), this);
if (!port) {
local_addr_.Clear();
diff --git a/test/network/network_emulation.cc b/test/network/network_emulation.cc
index 28c7637..5a1d1dd 100644
--- a/test/network/network_emulation.cc
+++ b/test/network/network_emulation.cc
@@ -17,12 +17,12 @@
#include <limits>
#include <map>
#include <memory>
+#include <optional>
#include <string>
#include <utility>
#include <vector>
#include "absl/base/nullability.h"
-#include "absl/types/optional.h"
#include "api/numerics/samples_stats_counter.h"
#include "api/sequence_checker.h"
#include "api/task_queue/task_queue_base.h"
@@ -423,8 +423,7 @@
if (process_task_.Running()) {
process_task_.Stop();
};
- absl::optional<int64_t> next_time_us =
- network_behavior_->NextDeliveryTimeUs();
+ std::optional<int64_t> next_time_us = network_behavior_->NextDeliveryTimeUs();
if (!next_time_us)
return;
Timestamp current_time = clock_->CurrentTime();
@@ -436,7 +435,7 @@
RTC_DCHECK_RUN_ON(task_queue_);
Timestamp current_time = clock_->CurrentTime();
Process(current_time);
- absl::optional<int64_t> next_time_us =
+ std::optional<int64_t> next_time_us =
network_behavior_->NextDeliveryTimeUs();
if (!next_time_us) {
process_task_.Stop();
@@ -502,7 +501,7 @@
void NetworkRouterNode::RemoveDefaultReceiver() {
RTC_DCHECK_RUN_ON(task_queue_);
- default_receiver_ = absl::nullopt;
+ default_receiver_ = std::nullopt;
}
void NetworkRouterNode::SetWatcher(
@@ -636,19 +635,19 @@
});
}
-absl::optional<uint16_t> EmulatedEndpointImpl::BindReceiver(
+std::optional<uint16_t> EmulatedEndpointImpl::BindReceiver(
uint16_t desired_port,
EmulatedNetworkReceiverInterface* receiver) {
return BindReceiverInternal(desired_port, receiver, /*is_one_shot=*/false);
}
-absl::optional<uint16_t> EmulatedEndpointImpl::BindOneShotReceiver(
+std::optional<uint16_t> EmulatedEndpointImpl::BindOneShotReceiver(
uint16_t desired_port,
EmulatedNetworkReceiverInterface* receiver) {
return BindReceiverInternal(desired_port, receiver, /*is_one_shot=*/true);
}
-absl::optional<uint16_t> EmulatedEndpointImpl::BindReceiverInternal(
+std::optional<uint16_t> EmulatedEndpointImpl::BindReceiverInternal(
uint16_t desired_port,
EmulatedNetworkReceiverInterface* receiver,
bool is_one_shot) {
@@ -675,7 +674,7 @@
RTC_LOG(LS_INFO) << "Can't bind receiver to used port " << desired_port
<< " in endpoint " << options_.log_name
<< "; id=" << options_.id;
- return absl::nullopt;
+ return std::nullopt;
}
RTC_LOG(LS_INFO) << "New receiver is binded to endpoint " << options_.log_name
<< "; id=" << options_.id << " on port " << port;
@@ -715,7 +714,7 @@
MutexLock lock(&receiver_lock_);
RTC_LOG(LS_INFO) << "Default receiver is removed from endpoint "
<< options_.log_name << "; id=" << options_.id;
- default_receiver_ = absl::nullopt;
+ default_receiver_ = std::nullopt;
}
rtc::IPAddress EmulatedEndpointImpl::GetPeerLocalAddress() const {
diff --git a/test/network/network_emulation.h b/test/network/network_emulation.h
index 2feaa35..6dec782 100644
--- a/test/network/network_emulation.h
+++ b/test/network/network_emulation.h
@@ -16,12 +16,12 @@
#include <deque>
#include <map>
#include <memory>
+#include <optional>
#include <string>
#include <utility>
#include <vector>
#include "absl/base/nullability.h"
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/numerics/samples_stats_counter.h"
#include "api/sequence_checker.h"
@@ -203,7 +203,7 @@
private:
const absl::Nonnull<TaskQueueBase*> task_queue_;
- absl::optional<EmulatedNetworkReceiverInterface*> default_receiver_
+ std::optional<EmulatedNetworkReceiverInterface*> default_receiver_
RTC_GUARDED_BY(task_queue_);
std::map<rtc::IPAddress, EmulatedNetworkReceiverInterface*> routing_
RTC_GUARDED_BY(task_queue_);
@@ -299,12 +299,12 @@
rtc::CopyOnWriteBuffer packet_data,
uint16_t application_overhead = 0) override;
- absl::optional<uint16_t> BindReceiver(
+ std::optional<uint16_t> BindReceiver(
uint16_t desired_port,
EmulatedNetworkReceiverInterface* receiver) override;
// Binds a receiver, and automatically removes the binding after first call to
// OnPacketReceived.
- absl::optional<uint16_t> BindOneShotReceiver(
+ std::optional<uint16_t> BindOneShotReceiver(
uint16_t desired_port,
EmulatedNetworkReceiverInterface* receiver);
void UnbindReceiver(uint16_t port) override;
@@ -330,7 +330,7 @@
bool is_one_shot;
};
- absl::optional<uint16_t> BindReceiverInternal(
+ std::optional<uint16_t> BindReceiverInternal(
uint16_t desired_port,
EmulatedNetworkReceiverInterface* receiver,
bool is_one_shot);
@@ -349,7 +349,7 @@
NetworkRouterNode router_;
uint16_t next_port_ RTC_GUARDED_BY(receiver_lock_);
- absl::optional<EmulatedNetworkReceiverInterface*> default_receiver_
+ std::optional<EmulatedNetworkReceiverInterface*> default_receiver_
RTC_GUARDED_BY(receiver_lock_);
std::map<uint16_t, ReceiverBinding> port_to_receiver_
RTC_GUARDED_BY(receiver_lock_);
diff --git a/test/network/network_emulation_manager.cc b/test/network/network_emulation_manager.cc
index 6a583a3..47de184 100644
--- a/test/network/network_emulation_manager.cc
+++ b/test/network/network_emulation_manager.cc
@@ -99,7 +99,7 @@
EmulatedEndpointImpl* NetworkEmulationManagerImpl::CreateEndpoint(
EmulatedEndpointConfig config) {
- absl::optional<rtc::IPAddress> ip = config.ip;
+ std::optional<rtc::IPAddress> ip = config.ip;
if (!ip) {
switch (config.generated_ip_family) {
case EmulatedEndpointConfig::IpAddressFamily::kIpv4:
@@ -341,7 +341,7 @@
});
}
-absl::optional<rtc::IPAddress>
+std::optional<rtc::IPAddress>
NetworkEmulationManagerImpl::GetNextIPv4Address() {
uint32_t addresses_count = kMaxIPv4Address - kMinIPv4Address;
for (uint32_t i = 0; i < addresses_count; i++) {
@@ -355,7 +355,7 @@
return ip;
}
}
- return absl::nullopt;
+ return std::nullopt;
}
Timestamp NetworkEmulationManagerImpl::Now() const {
diff --git a/test/network/network_emulation_manager.h b/test/network/network_emulation_manager.h
index 1c54241..154bd4b 100644
--- a/test/network/network_emulation_manager.h
+++ b/test/network/network_emulation_manager.h
@@ -99,7 +99,7 @@
using CrossTrafficSource =
std::pair<std::unique_ptr<CrossTrafficGenerator>, RepeatingTaskHandle>;
- absl::optional<rtc::IPAddress> GetNextIPv4Address();
+ std::optional<rtc::IPAddress> GetNextIPv4Address();
const TimeMode time_mode_;
const EmulatedNetworkStatsGatheringMode stats_gathering_mode_;
diff --git a/test/network/network_emulation_unittest.cc b/test/network/network_emulation_unittest.cc
index 805a1bf..b644361 100644
--- a/test/network/network_emulation_unittest.cc
+++ b/test/network/network_emulation_unittest.cc
@@ -88,10 +88,7 @@
DequeueDeliverablePackets,
(int64_t),
(override));
- MOCK_METHOD(absl::optional<int64_t>,
- NextDeliveryTimeUs,
- (),
- (const override));
+ MOCK_METHOD(std::optional<int64_t>, NextDeliveryTimeUs, (), (const override));
MOCK_METHOD(void,
RegisterDeliveryTimeChangedCallback,
(absl::AnyInvocable<void()>),
diff --git a/test/network/simulated_network.cc b/test/network/simulated_network.cc
index 5fe68b7..0663c0d 100644
--- a/test/network/simulated_network.cc
+++ b/test/network/simulated_network.cc
@@ -13,9 +13,9 @@
#include <algorithm>
#include <cmath>
#include <cstdint>
+#include <optional>
#include <utility>
-#include "absl/types/optional.h"
#include "api/test/simulated_network.h"
#include "api/units/data_rate.h"
#include "api/units/data_size.h"
@@ -188,12 +188,12 @@
return true;
}
-absl::optional<int64_t> SimulatedNetwork::NextDeliveryTimeUs() const {
+std::optional<int64_t> SimulatedNetwork::NextDeliveryTimeUs() const {
RTC_DCHECK_RUNS_SERIALIZED(&process_checker_);
if (next_process_time_.IsFinite()) {
return next_process_time_.us();
}
- return absl::nullopt;
+ return std::nullopt;
}
void SimulatedNetwork::UpdateCapacityQueue(ConfigState state,
diff --git a/test/network/simulated_network.h b/test/network/simulated_network.h
index ae68824..4322aaf 100644
--- a/test/network/simulated_network.h
+++ b/test/network/simulated_network.h
@@ -14,10 +14,10 @@
#include <cstdint>
#include <deque>
+#include <optional>
#include <queue>
#include <vector>
-#include "absl/types/optional.h"
#include "api/sequence_checker.h"
#include "api/test/simulated_network.h"
#include "api/units/timestamp.h"
@@ -70,7 +70,7 @@
std::vector<PacketDeliveryInfo> DequeueDeliverablePackets(
int64_t receive_time_us) override;
- absl::optional<int64_t> NextDeliveryTimeUs() const override;
+ std::optional<int64_t> NextDeliveryTimeUs() const override;
void RegisterDeliveryTimeChangedCallback(
absl::AnyInvocable<void()> callback) override;
diff --git a/test/network/simulated_network_unittest.cc b/test/network/simulated_network_unittest.cc
index 536c1e2..21d70ec 100644
--- a/test/network/simulated_network_unittest.cc
+++ b/test/network/simulated_network_unittest.cc
@@ -37,7 +37,7 @@
TEST(SimulatedNetworkTest, NextDeliveryTimeIsUnknownOnEmptyNetwork) {
SimulatedNetwork network = SimulatedNetwork({});
- EXPECT_EQ(network.NextDeliveryTimeUs(), absl::nullopt);
+ EXPECT_EQ(network.NextDeliveryTimeUs(), std::nullopt);
}
TEST(SimulatedNetworkTest, EnqueueFirstPacketOnNetworkWithInfiniteCapacity) {
@@ -329,7 +329,7 @@
EXPECT_EQ(delivered_packets.size(), 1ul);
// ... leaves the network empty.
- EXPECT_EQ(network.NextDeliveryTimeUs(), absl::nullopt);
+ EXPECT_EQ(network.NextDeliveryTimeUs(), std::nullopt);
}
TEST(SimulatedNetworkTest, DequeueDeliverablePacketsOnLateCall) {
diff --git a/test/network/traffic_route.cc b/test/network/traffic_route.cc
index 94db5ae..5e39eb1 100644
--- a/test/network/traffic_route.cc
+++ b/test/network/traffic_route.cc
@@ -12,9 +12,9 @@
#include <algorithm>
#include <memory>
+#include <optional>
#include <utility>
-#include "absl/types/optional.h"
#include "rtc_base/logging.h"
#include "rtc_base/numerics/safe_minmax.h"
@@ -46,7 +46,7 @@
EmulatedEndpointImpl* endpoint)
: clock_(clock), receiver_(receiver), endpoint_(endpoint) {
null_receiver_ = std::make_unique<NullReceiver>();
- absl::optional<uint16_t> port =
+ std::optional<uint16_t> port =
endpoint_->BindReceiver(0, null_receiver_.get());
RTC_DCHECK(port);
null_receiver_port_ = port.value();
@@ -65,7 +65,7 @@
auto action_receiver = std::make_unique<ActionReceiver>(action);
// BindOneShotReceiver arranges to free the port in the endpoint after the
// action is done.
- absl::optional<uint16_t> port =
+ std::optional<uint16_t> port =
endpoint_->BindOneShotReceiver(0, action_receiver.get());
RTC_DCHECK(port);
actions_.push_back(std::move(action_receiver));
diff --git a/test/pc/e2e/BUILD.gn b/test/pc/e2e/BUILD.gn
index f9d1f47..01ae598 100644
--- a/test/pc/e2e/BUILD.gn
+++ b/test/pc/e2e/BUILD.gn
@@ -80,7 +80,6 @@
"../../../rtc_base/synchronization:mutex",
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -126,7 +125,6 @@
"../../../api:sequence_checker",
"../../../api/test/video:test_video_track_source",
"../../../api/video:video_frame",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -201,7 +199,6 @@
"../../../rtc_base/task_utils:repeating_task",
"../../../system_wrappers",
"//third_party/abseil-cpp/absl/memory",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -371,7 +368,6 @@
"../../../api/test/pclf:peer_configurer",
"../../../api/units:time_delta",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -440,7 +436,6 @@
"../../../api:track_id_stream_info_map",
"../../../rtc_base:macromagic",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -553,7 +548,6 @@
"../../../rtc_base/synchronization:mutex",
"../../../system_wrappers:field_trial",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -578,7 +572,6 @@
"../../../rtc_base:stringutils",
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
}
diff --git a/test/pc/e2e/analyzer/video/BUILD.gn b/test/pc/e2e/analyzer/video/BUILD.gn
index 33e358a..e055703 100644
--- a/test/pc/e2e/analyzer/video/BUILD.gn
+++ b/test/pc/e2e/analyzer/video/BUILD.gn
@@ -76,10 +76,7 @@
testonly = true
sources = [ "encoded_image_data_injector.h" ]
- deps = [
- "../../../../../api/video:encoded_image",
- "//third_party/abseil-cpp/absl/types:optional",
- ]
+ deps = [ "../../../../../api/video:encoded_image" ]
}
rtc_library("single_process_encoded_image_data_injector") {
@@ -140,7 +137,6 @@
"../../../../../rtc_base:logging",
"../../../../../rtc_base/synchronization:mutex",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -177,7 +173,6 @@
"../../../../../rtc_base:macromagic",
"../../../../../rtc_base/synchronization:mutex",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -249,17 +244,13 @@
"../../../../../system_wrappers",
"dvqa:pausable_state",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
rtc_library("multi_reader_queue") {
testonly = true
sources = [ "multi_reader_queue.h" ]
- deps = [
- "../../../../../rtc_base:checks",
- "//third_party/abseil-cpp/absl/types:optional",
- ]
+ deps = [ "../../../../../rtc_base:checks" ]
}
rtc_library("video_quality_metrics_reporter") {
@@ -330,7 +321,6 @@
"../../../../../api/units:timestamp",
"../../../../../rtc_base:checks",
"../../../../../rtc_base:stringutils",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -362,7 +352,6 @@
"../../../../../system_wrappers",
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -434,7 +423,6 @@
"../../../../../system_wrappers",
"../../../../time_controller",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -445,7 +433,6 @@
":analyzing_video_sinks_helper",
"../../../..:test_support",
"../../../../../api/test/pclf:media_configuration",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -469,7 +456,6 @@
deps = [
":default_video_quality_analyzer_internal",
"../../../..:test_support",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -479,7 +465,6 @@
deps = [
":multi_reader_queue",
"../../../..:test_support",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -548,7 +533,6 @@
"../../../../../api:scoped_refptr",
"../../../../../api/video:video_frame",
"../../../../../rtc_base:random",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
diff --git a/test/pc/e2e/analyzer/video/analyzing_video_sink.cc b/test/pc/e2e/analyzer/video/analyzing_video_sink.cc
index 2392483..20e3966 100644
--- a/test/pc/e2e/analyzer/video/analyzing_video_sink.cc
+++ b/test/pc/e2e/analyzer/video/analyzing_video_sink.cc
@@ -10,12 +10,12 @@
#include "test/pc/e2e/analyzer/video/analyzing_video_sink.h"
#include <memory>
+#include <optional>
#include <set>
#include <utility>
#include "absl/memory/memory.h"
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/test/metrics/metric.h"
#include "api/test/metrics/metrics_logger.h"
#include "api/test/pclf/media_configuration.h"
@@ -58,7 +58,7 @@
MutexLock lock(&mutex_);
subscription_ = subscription;
for (auto it = stream_sinks_.cbegin(); it != stream_sinks_.cend();) {
- absl::optional<VideoResolution> new_requested_resolution =
+ std::optional<VideoResolution> new_requested_resolution =
subscription_.GetResolutionForPeer(it->second.sender_peer_name);
if (!new_requested_resolution.has_value() ||
(*new_requested_resolution != it->second.resolution)) {
@@ -191,14 +191,14 @@
}
// Slow pass: we need to create and save sinks
- absl::optional<std::pair<std::string, VideoConfig>> peer_and_config =
+ std::optional<std::pair<std::string, VideoConfig>> peer_and_config =
sinks_helper_->GetPeerAndConfig(stream_label);
RTC_CHECK(peer_and_config.has_value())
<< "No video config for stream " << stream_label;
const std::string& sender_peer_name = peer_and_config->first;
const VideoConfig& config = peer_and_config->second;
- absl::optional<VideoResolution> resolution =
+ std::optional<VideoResolution> resolution =
subscription_.GetResolutionForPeer(sender_peer_name);
if (!resolution.has_value()) {
RTC_LOG(LS_ERROR) << peer_name_ << " received stream " << stream_label
diff --git a/test/pc/e2e/analyzer/video/analyzing_video_sink_test.cc b/test/pc/e2e/analyzer/video/analyzing_video_sink_test.cc
index 6cd8955..b2f8ef8 100644
--- a/test/pc/e2e/analyzer/video/analyzing_video_sink_test.cc
+++ b/test/pc/e2e/analyzer/video/analyzing_video_sink_test.cc
@@ -11,11 +11,11 @@
#include <stdio.h>
+#include <optional>
#include <string>
#include <vector>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/scoped_refptr.h"
#include "api/test/create_frame_generator.h"
#include "api/test/frame_generator_interface.h"
@@ -45,7 +45,7 @@
// Remove files and directories in a directory non-recursively.
void CleanDir(absl::string_view dir, size_t expected_output_files_count) {
- absl::optional<std::vector<std::string>> dir_content =
+ std::optional<std::vector<std::string>> dir_content =
test::ReadDirectory(dir);
if (expected_output_files_count == 0) {
ASSERT_TRUE(!dir_content.has_value() || dir_content->empty())
@@ -81,8 +81,8 @@
size_t width,
size_t height) {
return test::CreateSquareFrameGenerator(width, height,
- /*type=*/absl::nullopt,
- /*num_squares=*/absl::nullopt);
+ /*type=*/std::nullopt,
+ /*num_squares=*/std::nullopt);
}
void AssertFrameIdsAre(const std::string& filename,
diff --git a/test/pc/e2e/analyzer/video/analyzing_video_sinks_helper.cc b/test/pc/e2e/analyzer/video/analyzing_video_sinks_helper.cc
index 70dc4b0..cf4b1bb 100644
--- a/test/pc/e2e/analyzer/video/analyzing_video_sinks_helper.cc
+++ b/test/pc/e2e/analyzer/video/analyzing_video_sinks_helper.cc
@@ -36,12 +36,12 @@
}
}
-absl::optional<std::pair<std::string, VideoConfig>>
+std::optional<std::pair<std::string, VideoConfig>>
AnalyzingVideoSinksHelper::GetPeerAndConfig(absl::string_view stream_label) {
MutexLock lock(&mutex_);
auto it = video_configs_.find(std::string(stream_label));
if (it == video_configs_.end()) {
- return absl::nullopt;
+ return std::nullopt;
}
return it->second;
}
diff --git a/test/pc/e2e/analyzer/video/analyzing_video_sinks_helper.h b/test/pc/e2e/analyzer/video/analyzing_video_sinks_helper.h
index 5f38c5a..a4a92af 100644
--- a/test/pc/e2e/analyzer/video/analyzing_video_sinks_helper.h
+++ b/test/pc/e2e/analyzer/video/analyzing_video_sinks_helper.h
@@ -14,12 +14,12 @@
#include <list>
#include <map>
#include <memory>
+#include <optional>
#include <set>
#include <string>
#include <utility>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/test/pclf/media_configuration.h"
#include "api/test/video/video_frame_writer.h"
#include "rtc_base/synchronization/mutex.h"
@@ -35,7 +35,7 @@
// Adds config in the registry. If config with such stream label was
// registered before, the new value will override the old one.
void AddConfig(absl::string_view sender_peer_name, VideoConfig config);
- absl::optional<std::pair<std::string, VideoConfig>> GetPeerAndConfig(
+ std::optional<std::pair<std::string, VideoConfig>> GetPeerAndConfig(
absl::string_view stream_label);
// Removes video config for specified stream label. If there are no know video
// config for such stream label - does nothing.
diff --git a/test/pc/e2e/analyzer/video/analyzing_video_sinks_helper_test.cc b/test/pc/e2e/analyzer/video/analyzing_video_sinks_helper_test.cc
index 1a820a5..792f44f 100644
--- a/test/pc/e2e/analyzer/video/analyzing_video_sinks_helper_test.cc
+++ b/test/pc/e2e/analyzer/video/analyzing_video_sinks_helper_test.cc
@@ -10,10 +10,10 @@
#include "test/pc/e2e/analyzer/video/analyzing_video_sinks_helper.h"
#include <memory>
+#include <optional>
#include <string>
#include <utility>
-#include "absl/types/optional.h"
#include "api/test/pclf/media_configuration.h"
#include "test/gmock.h"
#include "test/gtest.h"
@@ -40,7 +40,7 @@
AnalyzingVideoSinksHelper helper;
helper.AddConfig("alice", config);
- absl::optional<std::pair<std::string, VideoConfig>> registred_config =
+ std::optional<std::pair<std::string, VideoConfig>> registred_config =
helper.GetPeerAndConfig("alice_video");
ASSERT_TRUE(registred_config.has_value());
EXPECT_THAT(registred_config->first, Eq("alice"));
@@ -56,7 +56,7 @@
AnalyzingVideoSinksHelper helper;
helper.AddConfig("alice", config_before);
- absl::optional<std::pair<std::string, VideoConfig>> registred_config =
+ std::optional<std::pair<std::string, VideoConfig>> registred_config =
helper.GetPeerAndConfig("alice_video");
ASSERT_TRUE(registred_config.has_value());
EXPECT_THAT(registred_config->first, Eq("alice"));
diff --git a/test/pc/e2e/analyzer/video/default_video_quality_analyzer.cc b/test/pc/e2e/analyzer/video/default_video_quality_analyzer.cc
index 6c52cc5..5b1d558 100644
--- a/test/pc/e2e/analyzer/video/default_video_quality_analyzer.cc
+++ b/test/pc/e2e/analyzer/video/default_video_quality_analyzer.cc
@@ -111,10 +111,10 @@
}
template <typename T>
-absl::optional<T> MaybeGetValue(const std::map<size_t, T>& map, size_t key) {
+std::optional<T> MaybeGetValue(const std::map<size_t, T>& map, size_t key) {
auto it = map.find(key);
if (it == map.end()) {
- return absl::nullopt;
+ return std::nullopt;
}
return it->second;
}
@@ -211,7 +211,7 @@
}
StreamState* state = &stream_states_.at(stream_index);
state->PushBack(frame_id);
- absl::optional<TimeDelta> time_between_captured_frames = absl::nullopt;
+ std::optional<TimeDelta> time_between_captured_frames = std::nullopt;
if (state->last_captured_frame_time().has_value()) {
time_between_captured_frames =
captured_time - *state->last_captured_frame_time();
@@ -243,8 +243,8 @@
StatsSample(captured_frames_in_flight_.size(), Now()));
frames_comparator_.AddComparison(
InternalStatsKey(stream_index, peer_index, i),
- /*captured=*/absl::nullopt,
- /*rendered=*/absl::nullopt, FrameComparisonType::kDroppedFrame,
+ /*captured=*/std::nullopt,
+ /*rendered=*/std::nullopt, FrameComparisonType::kDroppedFrame,
it->second.GetStatsForPeer(i));
}
@@ -349,7 +349,7 @@
}
Timestamp now = Now();
StreamState& state = stream_states_.at(frame_in_flight.stream());
- absl::optional<TimeDelta> time_between_encoded_frames = absl::nullopt;
+ std::optional<TimeDelta> time_between_encoded_frames = std::nullopt;
if (state.last_encoded_frame_time().has_value()) {
time_between_encoded_frames = now - *state.last_encoded_frame_time();
}
@@ -527,7 +527,7 @@
// Find corresponding captured frame.
FrameInFlight* frame_in_flight = &frame_it->second;
- absl::optional<VideoFrame> captured_frame = frames_storage_.Get(frame.id());
+ std::optional<VideoFrame> captured_frame = frames_storage_.Get(frame.id());
const size_t stream_index = frame_in_flight->stream();
StreamState* state = &stream_states_.at(stream_index);
@@ -699,7 +699,7 @@
absl::string_view peer_name) {
MutexLock lock(&mutex_);
RTC_CHECK(peers_->HasName(peer_name));
- absl::optional<size_t> peer_index = peers_->RemoveIfPresent(peer_name);
+ std::optional<size_t> peer_index = peers_->RemoveIfPresent(peer_name);
RTC_CHECK(peer_index.has_value());
for (auto& [stream_index, stream_state] : stream_states_) {
@@ -964,8 +964,8 @@
RTC_DCHECK(it != captured_frames_in_flight_.end());
FrameInFlight& frame = it->second;
- frames_comparator_.AddComparison(stats_key, /*captured=*/absl::nullopt,
- /*rendered=*/absl::nullopt,
+ frames_comparator_.AddComparison(stats_key, /*captured=*/std::nullopt,
+ /*rendered=*/std::nullopt,
FrameComparisonType::kFrameInFlight,
frame.GetStatsForPeer(peer_index));
}
@@ -1053,8 +1053,8 @@
analyzer_stats_.frames_in_flight_left_count.AddSample(
StatsSample(captured_frames_in_flight_.size(), Now()));
- frames_comparator_.AddComparison(stats_key, /*captured=*/absl::nullopt,
- /*rendered=*/absl::nullopt,
+ frames_comparator_.AddComparison(stats_key, /*captured=*/std::nullopt,
+ /*rendered=*/std::nullopt,
FrameComparisonType::kDroppedFrame,
next_frame.GetStatsForPeer(peer_index));
} else {
diff --git a/test/pc/e2e/analyzer/video/default_video_quality_analyzer_frame_in_flight.cc b/test/pc/e2e/analyzer/video/default_video_quality_analyzer_frame_in_flight.cc
index 6d7aa32..fb96ee6e 100644
--- a/test/pc/e2e/analyzer/video/default_video_quality_analyzer_frame_in_flight.cc
+++ b/test/pc/e2e/analyzer/video/default_video_quality_analyzer_frame_in_flight.cc
@@ -10,11 +10,11 @@
#include "test/pc/e2e/analyzer/video/default_video_quality_analyzer_frame_in_flight.h"
+#include <optional>
#include <unordered_map>
#include <utility>
#include <vector>
-#include "absl/types/optional.h"
#include "api/units/data_size.h"
#include "api/units/time_delta.h"
#include "api/units/timestamp.h"
@@ -26,11 +26,11 @@
namespace {
template <typename T>
-absl::optional<T> MaybeGetValue(const std::unordered_map<size_t, T>& map,
- size_t key) {
+std::optional<T> MaybeGetValue(const std::unordered_map<size_t, T>& map,
+ size_t key) {
auto it = map.find(key);
if (it == map.end()) {
- return absl::nullopt;
+ return std::nullopt;
}
return it->second;
}
@@ -41,7 +41,7 @@
size_t stream,
uint16_t frame_id,
Timestamp captured_time,
- absl::optional<TimeDelta> time_between_captured_frames,
+ std::optional<TimeDelta> time_between_captured_frames,
std::set<size_t> expected_receivers)
: stream_(stream),
expected_receivers_(std::move(expected_receivers)),
@@ -79,7 +79,7 @@
void FrameInFlight::OnFrameEncoded(
webrtc::Timestamp time,
- absl::optional<TimeDelta> time_between_encoded_frames,
+ std::optional<TimeDelta> time_between_encoded_frames,
VideoFrameType frame_type,
DataSize encoded_image_size,
uint32_t target_encode_bitrate,
@@ -136,7 +136,7 @@
int width,
int height,
const StreamCodecInfo& used_decoder,
- const absl::optional<uint8_t> qp) {
+ const std::optional<uint8_t> qp) {
receiver_stats_[peer].decode_end_time = time;
receiver_stats_[peer].used_decoder = used_decoder;
receiver_stats_[peer].decoded_frame_width = width;
@@ -194,7 +194,7 @@
stats.used_encoder = used_encoder_;
stats.spatial_layers_qp = stream_layers_qp_;
- absl::optional<ReceiverFrameStats> receiver_stats =
+ std::optional<ReceiverFrameStats> receiver_stats =
MaybeGetValue<ReceiverFrameStats>(receiver_stats_, peer);
if (receiver_stats.has_value()) {
stats.received_time = receiver_stats->received_time;
diff --git a/test/pc/e2e/analyzer/video/default_video_quality_analyzer_frame_in_flight.h b/test/pc/e2e/analyzer/video/default_video_quality_analyzer_frame_in_flight.h
index 669330a..287aa6f 100644
--- a/test/pc/e2e/analyzer/video/default_video_quality_analyzer_frame_in_flight.h
+++ b/test/pc/e2e/analyzer/video/default_video_quality_analyzer_frame_in_flight.h
@@ -12,12 +12,12 @@
#define TEST_PC_E2E_ANALYZER_VIDEO_DEFAULT_VIDEO_QUALITY_ANALYZER_FRAME_IN_FLIGHT_H_
#include <map>
+#include <optional>
#include <set>
#include <unordered_map>
#include <utility>
#include <vector>
-#include "absl/types/optional.h"
#include "api/numerics/samples_stats_counter.h"
#include "api/units/data_size.h"
#include "api/units/time_delta.h"
@@ -36,20 +36,20 @@
Timestamp rendered_time = Timestamp::MinusInfinity();
// Will be set if there is frame rendered before this one.
- absl::optional<Timestamp> prev_frame_rendered_time = absl::nullopt;
- absl::optional<TimeDelta> time_between_rendered_frames = absl::nullopt;
+ std::optional<Timestamp> prev_frame_rendered_time = std::nullopt;
+ std::optional<TimeDelta> time_between_rendered_frames = std::nullopt;
// Type and encoded size of received frame.
VideoFrameType frame_type = VideoFrameType::kEmptyFrame;
DataSize encoded_image_size = DataSize::Bytes(0);
- absl::optional<int> decoded_frame_width = absl::nullopt;
- absl::optional<int> decoded_frame_height = absl::nullopt;
+ std::optional<int> decoded_frame_width = std::nullopt;
+ std::optional<int> decoded_frame_height = std::nullopt;
- absl::optional<uint8_t> decoded_frame_qp = absl::nullopt;
+ std::optional<uint8_t> decoded_frame_qp = std::nullopt;
// Can be not set if frame was dropped in the network.
- absl::optional<StreamCodecInfo> used_decoder = absl::nullopt;
+ std::optional<StreamCodecInfo> used_decoder = std::nullopt;
bool dropped = false;
bool decoder_failed = false;
@@ -71,7 +71,7 @@
FrameInFlight(size_t stream,
uint16_t frame_id,
Timestamp captured_time,
- absl::optional<TimeDelta> time_between_captured_frames,
+ std::optional<TimeDelta> time_between_captured_frames,
std::set<size_t> expected_receivers);
size_t stream() const { return stream_; }
@@ -93,7 +93,7 @@
void SetPreEncodeTime(Timestamp time) { pre_encode_time_ = time; }
void OnFrameEncoded(Timestamp time,
- absl::optional<TimeDelta> time_between_encoded_frames,
+ std::optional<TimeDelta> time_between_encoded_frames,
VideoFrameType frame_type,
DataSize encoded_image_size,
uint32_t target_encode_bitrate,
@@ -116,7 +116,7 @@
int width,
int height,
const StreamCodecInfo& used_decoder,
- const absl::optional<uint8_t> qp);
+ const std::optional<uint8_t> qp);
void OnDecoderError(size_t peer, const StreamCodecInfo& used_decoder);
@@ -170,8 +170,8 @@
Timestamp pre_encode_time_ = Timestamp::MinusInfinity();
Timestamp encoded_time_ = Timestamp::MinusInfinity();
- absl::optional<TimeDelta> time_between_captured_frames_ = absl::nullopt;
- absl::optional<TimeDelta> time_between_encoded_frames_ = absl::nullopt;
+ std::optional<TimeDelta> time_between_captured_frames_ = std::nullopt;
+ std::optional<TimeDelta> time_between_encoded_frames_ = std::nullopt;
// Type and encoded size of sent frame.
VideoFrameType frame_type_ = VideoFrameType::kEmptyFrame;
@@ -181,7 +181,7 @@
// spatial or simulcast index is set in `EncodedImage`, 0 is used.
std::map<int, SamplesStatsCounter> stream_layers_qp_;
// Can be not set if frame was dropped by encoder.
- absl::optional<StreamCodecInfo> used_encoder_ = absl::nullopt;
+ std::optional<StreamCodecInfo> used_encoder_ = std::nullopt;
// Map from the receiver peer's index to frame stats for that peer.
std::unordered_map<size_t, ReceiverFrameStats> receiver_stats_;
};
diff --git a/test/pc/e2e/analyzer/video/default_video_quality_analyzer_frames_comparator.cc b/test/pc/e2e/analyzer/video/default_video_quality_analyzer_frames_comparator.cc
index 9db4057..96ba330 100644
--- a/test/pc/e2e/analyzer/video/default_video_quality_analyzer_frames_comparator.cc
+++ b/test/pc/e2e/analyzer/video/default_video_quality_analyzer_frames_comparator.cc
@@ -12,11 +12,11 @@
#include <algorithm>
#include <map>
+#include <optional>
#include <string>
#include <utility>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/scoped_refptr.h"
#include "api/video/i420_buffer.h"
@@ -317,8 +317,8 @@
void DefaultVideoQualityAnalyzerFramesComparator::AddComparison(
InternalStatsKey stats_key,
- absl::optional<VideoFrame> captured,
- absl::optional<VideoFrame> rendered,
+ std::optional<VideoFrame> captured,
+ std::optional<VideoFrame> rendered,
FrameComparisonType type,
FrameStats frame_stats) {
MutexLock lock(&mutex_);
@@ -331,8 +331,8 @@
void DefaultVideoQualityAnalyzerFramesComparator::AddComparison(
InternalStatsKey stats_key,
int skipped_between_rendered,
- absl::optional<VideoFrame> captured,
- absl::optional<VideoFrame> rendered,
+ std::optional<VideoFrame> captured,
+ std::optional<VideoFrame> rendered,
FrameComparisonType type,
FrameStats frame_stats) {
MutexLock lock(&mutex_);
@@ -349,8 +349,8 @@
void DefaultVideoQualityAnalyzerFramesComparator::AddComparisonInternal(
InternalStatsKey stats_key,
- absl::optional<VideoFrame> captured,
- absl::optional<VideoFrame> rendered,
+ std::optional<VideoFrame> captured,
+ std::optional<VideoFrame> rendered,
FrameComparisonType type,
FrameStats frame_stats) {
cpu_measurer_.StartExcludingCpuThreadTime();
@@ -360,9 +360,9 @@
// frames itself to make future computations lighter.
if (comparisons_.size() >= kMaxActiveComparisons) {
comparisons_.emplace_back(ValidateFrameComparison(
- FrameComparison(std::move(stats_key), /*captured=*/absl::nullopt,
- /*rendered=*/absl::nullopt, type,
- std::move(frame_stats), OverloadReason::kCpu)));
+ FrameComparison(std::move(stats_key), /*captured=*/std::nullopt,
+ /*rendered=*/std::nullopt, type, std::move(frame_stats),
+ OverloadReason::kCpu)));
} else {
OverloadReason overload_reason = OverloadReason::kNone;
if (!captured && type == FrameComparisonType::kRegular) {
@@ -379,7 +379,7 @@
void DefaultVideoQualityAnalyzerFramesComparator::ProcessComparisons() {
while (true) {
// Try to pick next comparison to perform from the queue.
- absl::optional<FrameComparison> comparison = absl::nullopt;
+ std::optional<FrameComparison> comparison = std::nullopt;
bool more_new_comparisons_expected;
{
MutexLock lock(&mutex_);
diff --git a/test/pc/e2e/analyzer/video/default_video_quality_analyzer_frames_comparator.h b/test/pc/e2e/analyzer/video/default_video_quality_analyzer_frames_comparator.h
index 006c3eb..b41268d 100644
--- a/test/pc/e2e/analyzer/video/default_video_quality_analyzer_frames_comparator.h
+++ b/test/pc/e2e/analyzer/video/default_video_quality_analyzer_frames_comparator.h
@@ -94,23 +94,23 @@
// `captured` - video frame captured by sender to use for PSNR/SSIM
// computation. If `type` is `FrameComparisonType::kRegular` and
- // `captured` is `absl::nullopt` comparison is assumed to be overloaded
+ // `captured` is `std::nullopt` comparison is assumed to be overloaded
// due to memory constraints.
// `rendered` - video frame rendered by receiver to use for PSNR/SSIM
// computation. Required only if `type` is
// `FrameComparisonType::kRegular`, but can still be omitted if
- // `captured` is `absl::nullopt`.
+ // `captured` is `std::nullopt`.
void AddComparison(InternalStatsKey stats_key,
- absl::optional<VideoFrame> captured,
- absl::optional<VideoFrame> rendered,
+ std::optional<VideoFrame> captured,
+ std::optional<VideoFrame> rendered,
FrameComparisonType type,
FrameStats frame_stats);
// `skipped_between_rendered` - amount of frames dropped on this stream before
// last received frame and current frame.
void AddComparison(InternalStatsKey stats_key,
int skipped_between_rendered,
- absl::optional<VideoFrame> captured,
- absl::optional<VideoFrame> rendered,
+ std::optional<VideoFrame> captured,
+ std::optional<VideoFrame> rendered,
FrameComparisonType type,
FrameStats frame_stats);
@@ -127,8 +127,8 @@
enum State { kNew, kActive, kStopped };
void AddComparisonInternal(InternalStatsKey stats_key,
- absl::optional<VideoFrame> captured,
- absl::optional<VideoFrame> rendered,
+ std::optional<VideoFrame> captured,
+ std::optional<VideoFrame> rendered,
FrameComparisonType type,
FrameStats frame_stats)
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
diff --git a/test/pc/e2e/analyzer/video/default_video_quality_analyzer_frames_comparator_test.cc b/test/pc/e2e/analyzer/video/default_video_quality_analyzer_frames_comparator_test.cc
index ecd2a54..ca39eb7 100644
--- a/test/pc/e2e/analyzer/video/default_video_quality_analyzer_frames_comparator_test.cc
+++ b/test/pc/e2e/analyzer/video/default_video_quality_analyzer_frames_comparator_test.cc
@@ -50,8 +50,8 @@
Timestamp timestamp) {
std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
test::CreateSquareFrameGenerator(width, height,
- /*type=*/absl::nullopt,
- /*num_squares=*/absl::nullopt);
+ /*type=*/std::nullopt,
+ /*num_squares=*/std::nullopt);
test::FrameGeneratorInterface::VideoFrameData frame_data =
frame_generator->NextFrame();
return VideoFrame::Builder()
@@ -177,8 +177,8 @@
comparator.EnsureStatsForStream(stream, sender, peers_count,
stream_start_time, stream_start_time);
comparator.AddComparison(stats_key,
- /*captured=*/absl::nullopt,
- /*rendered=*/absl::nullopt,
+ /*captured=*/std::nullopt,
+ /*rendered=*/std::nullopt,
FrameComparisonType::kRegular, frame_stats);
comparator.Stop(/*last_rendered_frame_times=*/{});
@@ -223,12 +223,12 @@
comparator.EnsureStatsForStream(stream, sender, peers_count,
stream_start_time, stream_start_time);
comparator.AddComparison(stats_key,
- /*captured=*/absl::nullopt,
- /*rendered=*/absl::nullopt,
+ /*captured=*/std::nullopt,
+ /*rendered=*/std::nullopt,
FrameComparisonType::kRegular, frame_stats1);
comparator.AddComparison(stats_key,
- /*captured=*/absl::nullopt,
- /*rendered=*/absl::nullopt,
+ /*captured=*/std::nullopt,
+ /*rendered=*/std::nullopt,
FrameComparisonType::kRegular, frame_stats2);
comparator.Stop(/*last_rendered_frame_times=*/{});
@@ -311,13 +311,13 @@
stream_start_time, stream_start_time);
for (size_t i = 0; i < stats.size() - 1; ++i) {
comparator.AddComparison(stats_key,
- /*captured=*/absl::nullopt,
- /*rendered=*/absl::nullopt,
+ /*captured=*/std::nullopt,
+ /*rendered=*/std::nullopt,
FrameComparisonType::kFrameInFlight, stats[i]);
}
comparator.AddComparison(stats_key,
- /*captured=*/absl::nullopt,
- /*rendered=*/absl::nullopt,
+ /*captured=*/std::nullopt,
+ /*rendered=*/std::nullopt,
FrameComparisonType::kRegular,
stats[stats.size() - 1]);
comparator.Stop(/*last_rendered_frame_times=*/{});
@@ -378,8 +378,8 @@
comparator.EnsureStatsForStream(stream, sender, /*peers_count=*/2,
captured_time, captured_time);
comparator.AddComparison(stats_key,
- /*captured=*/absl::nullopt,
- /*rendered=*/absl::nullopt,
+ /*captured=*/std::nullopt,
+ /*rendered=*/std::nullopt,
FrameComparisonType::kFrameInFlight, frame_stats);
comparator.Stop(/*last_rendered_frame_times=*/{});
@@ -439,8 +439,8 @@
comparator.EnsureStatsForStream(stream, sender, /*peers_count=*/2,
captured_time, captured_time);
comparator.AddComparison(stats_key,
- /*captured=*/absl::nullopt,
- /*rendered=*/absl::nullopt,
+ /*captured=*/std::nullopt,
+ /*rendered=*/std::nullopt,
FrameComparisonType::kFrameInFlight, frame_stats);
comparator.Stop(/*last_rendered_frame_times=*/{});
@@ -512,8 +512,8 @@
comparator.EnsureStatsForStream(stream, sender, /*peers_count=*/2,
captured_time, captured_time);
comparator.AddComparison(stats_key,
- /*captured=*/absl::nullopt,
- /*rendered=*/absl::nullopt,
+ /*captured=*/std::nullopt,
+ /*rendered=*/std::nullopt,
FrameComparisonType::kFrameInFlight, frame_stats);
comparator.Stop(/*last_rendered_frame_times=*/{});
@@ -589,8 +589,8 @@
comparator.EnsureStatsForStream(stream, sender, /*peers_count=*/2,
captured_time, captured_time);
comparator.AddComparison(stats_key,
- /*captured=*/absl::nullopt,
- /*rendered=*/absl::nullopt,
+ /*captured=*/std::nullopt,
+ /*rendered=*/std::nullopt,
FrameComparisonType::kFrameInFlight, frame_stats);
comparator.Stop(/*last_rendered_frame_times=*/{});
@@ -671,8 +671,8 @@
comparator.EnsureStatsForStream(stream, sender, /*peers_count=*/2,
captured_time, captured_time);
comparator.AddComparison(stats_key,
- /*captured=*/absl::nullopt,
- /*rendered=*/absl::nullopt,
+ /*captured=*/std::nullopt,
+ /*rendered=*/std::nullopt,
FrameComparisonType::kFrameInFlight, frame_stats);
comparator.Stop(/*last_rendered_frame_times=*/{});
@@ -763,8 +763,8 @@
comparator.EnsureStatsForStream(stream, sender, /*peers_count=*/2,
captured_time, captured_time);
comparator.AddComparison(stats_key,
- /*captured=*/absl::nullopt,
- /*rendered=*/absl::nullopt,
+ /*captured=*/std::nullopt,
+ /*rendered=*/std::nullopt,
FrameComparisonType::kFrameInFlight, frame_stats);
comparator.Stop(/*last_rendered_frame_times=*/{});
@@ -854,8 +854,8 @@
comparator.EnsureStatsForStream(stream, sender, /*peers_count=*/2,
captured_time, captured_time);
comparator.AddComparison(stats_key,
- /*captured=*/absl::nullopt,
- /*rendered=*/absl::nullopt,
+ /*captured=*/std::nullopt,
+ /*rendered=*/std::nullopt,
FrameComparisonType::kFrameInFlight, frame_stats);
comparator.Stop(/*last_rendered_frame_times=*/{});
@@ -923,8 +923,8 @@
comparator.EnsureStatsForStream(stream, sender, /*peers_count=*/2,
captured_time, captured_time);
comparator.AddComparison(stats_key,
- /*captured=*/absl::nullopt,
- /*rendered=*/absl::nullopt,
+ /*captured=*/std::nullopt,
+ /*rendered=*/std::nullopt,
FrameComparisonType::kDroppedFrame, frame_stats);
comparator.Stop(/*last_rendered_frame_times=*/{});
@@ -984,8 +984,8 @@
comparator.EnsureStatsForStream(stream, sender, /*peers_count=*/2,
captured_time, captured_time);
comparator.AddComparison(stats_key,
- /*captured=*/absl::nullopt,
- /*rendered=*/absl::nullopt,
+ /*captured=*/std::nullopt,
+ /*rendered=*/std::nullopt,
FrameComparisonType::kDroppedFrame, frame_stats);
comparator.Stop(/*last_rendered_frame_times=*/{});
@@ -1057,8 +1057,8 @@
comparator.EnsureStatsForStream(stream, sender, /*peers_count=*/2,
captured_time, captured_time);
comparator.AddComparison(stats_key,
- /*captured=*/absl::nullopt,
- /*rendered=*/absl::nullopt,
+ /*captured=*/std::nullopt,
+ /*rendered=*/std::nullopt,
FrameComparisonType::kDroppedFrame, frame_stats);
comparator.Stop(/*last_rendered_frame_times=*/{});
@@ -1134,8 +1134,8 @@
comparator.EnsureStatsForStream(stream, sender, /*peers_count=*/2,
captured_time, captured_time);
comparator.AddComparison(stats_key,
- /*captured=*/absl::nullopt,
- /*rendered=*/absl::nullopt,
+ /*captured=*/std::nullopt,
+ /*rendered=*/std::nullopt,
FrameComparisonType::kDroppedFrame, frame_stats);
comparator.Stop(/*last_rendered_frame_times=*/{});
@@ -1212,8 +1212,8 @@
comparator.EnsureStatsForStream(stream, sender, /*peers_count=*/2,
captured_time, captured_time);
comparator.AddComparison(stats_key,
- /*captured=*/absl::nullopt,
- /*rendered=*/absl::nullopt,
+ /*captured=*/std::nullopt,
+ /*rendered=*/std::nullopt,
FrameComparisonType::kDroppedFrame, frame_stats);
comparator.Stop(/*last_rendered_frame_times=*/{});
@@ -1301,8 +1301,8 @@
comparator.EnsureStatsForStream(stream, sender, /*peers_count=*/2,
captured_time, captured_time);
comparator.AddComparison(stats_key,
- /*captured=*/absl::nullopt,
- /*rendered=*/absl::nullopt,
+ /*captured=*/std::nullopt,
+ /*rendered=*/std::nullopt,
FrameComparisonType::kDroppedFrame, frame_stats);
comparator.Stop(/*last_rendered_frame_times=*/{});
@@ -1389,8 +1389,8 @@
comparator.EnsureStatsForStream(stream, sender, /*peers_count=*/2,
captured_time, captured_time);
comparator.AddComparison(stats_key,
- /*captured=*/absl::nullopt,
- /*rendered=*/absl::nullopt,
+ /*captured=*/std::nullopt,
+ /*rendered=*/std::nullopt,
FrameComparisonType::kDroppedFrame, frame_stats);
comparator.Stop(/*last_rendered_frame_times=*/{});
@@ -1628,21 +1628,21 @@
// Add 5 frames which were rendered with 30 fps (~30ms between frames)
// Frame ids are in [1..5] and last frame is with 120ms offset from first.
- absl::optional<Timestamp> prev_frame_rendered_time = absl::nullopt;
+ std::optional<Timestamp> prev_frame_rendered_time = std::nullopt;
for (int i = 0; i < 5; ++i) {
FrameStats frame_stats = FrameStatsWith10msDeltaBetweenPhasesAnd10x10Frame(
/*frame_id=*/i + 1, stream_start_time + TimeDelta::Millis(30 * i));
frame_stats.prev_frame_rendered_time = prev_frame_rendered_time;
frame_stats.time_between_rendered_frames =
prev_frame_rendered_time.has_value()
- ? absl::optional<TimeDelta>(frame_stats.rendered_time -
- *prev_frame_rendered_time)
- : absl::nullopt;
+ ? std::optional<TimeDelta>(frame_stats.rendered_time -
+ *prev_frame_rendered_time)
+ : std::nullopt;
prev_frame_rendered_time = frame_stats.rendered_time;
comparator.AddComparison(stats_key,
- /*captured=*/absl::nullopt,
- /*rendered=*/absl::nullopt,
+ /*captured=*/std::nullopt,
+ /*rendered=*/std::nullopt,
FrameComparisonType::kRegular, frame_stats);
}
@@ -1657,8 +1657,8 @@
comparator.AddComparison(stats_key,
/*skipped_between_rendered=*/4,
- /*captured=*/absl::nullopt,
- /*rendered=*/absl::nullopt,
+ /*captured=*/std::nullopt,
+ /*rendered=*/std::nullopt,
FrameComparisonType::kRegular, freeze_frame_stats);
comparator.Stop(/*last_rendered_frame_times=*/{});
diff --git a/test/pc/e2e/analyzer/video/default_video_quality_analyzer_internal_shared_objects.cc b/test/pc/e2e/analyzer/video/default_video_quality_analyzer_internal_shared_objects.cc
index 16f49ef..49e0f42 100644
--- a/test/pc/e2e/analyzer/video/default_video_quality_analyzer_internal_shared_objects.cc
+++ b/test/pc/e2e/analyzer/video/default_video_quality_analyzer_internal_shared_objects.cc
@@ -37,8 +37,8 @@
}
FrameComparison::FrameComparison(InternalStatsKey stats_key,
- absl::optional<VideoFrame> captured,
- absl::optional<VideoFrame> rendered,
+ std::optional<VideoFrame> captured,
+ std::optional<VideoFrame> rendered,
FrameComparisonType type,
FrameStats frame_stats,
OverloadReason overload_reason)
diff --git a/test/pc/e2e/analyzer/video/default_video_quality_analyzer_internal_shared_objects.h b/test/pc/e2e/analyzer/video/default_video_quality_analyzer_internal_shared_objects.h
index 7a2cd71..c688748 100644
--- a/test/pc/e2e/analyzer/video/default_video_quality_analyzer_internal_shared_objects.h
+++ b/test/pc/e2e/analyzer/video/default_video_quality_analyzer_internal_shared_objects.h
@@ -12,11 +12,11 @@
#define TEST_PC_E2E_ANALYZER_VIDEO_DEFAULT_VIDEO_QUALITY_ANALYZER_INTERNAL_SHARED_OBJECTS_H_
#include <memory>
+#include <optional>
#include <string>
#include <utility>
#include <vector>
-#include "absl/types/optional.h"
#include "api/numerics/samples_stats_counter.h"
#include "api/units/data_size.h"
#include "api/units/timestamp.h"
@@ -59,10 +59,10 @@
Timestamp rendered_time = Timestamp::MinusInfinity();
// Next timings are set if and only if previous frame exist.
- absl::optional<Timestamp> prev_frame_rendered_time = absl::nullopt;
- absl::optional<TimeDelta> time_between_captured_frames = absl::nullopt;
- absl::optional<TimeDelta> time_between_encoded_frames = absl::nullopt;
- absl::optional<TimeDelta> time_between_rendered_frames = absl::nullopt;
+ std::optional<Timestamp> prev_frame_rendered_time = std::nullopt;
+ std::optional<TimeDelta> time_between_captured_frames = std::nullopt;
+ std::optional<TimeDelta> time_between_encoded_frames = std::nullopt;
+ std::optional<TimeDelta> time_between_rendered_frames = std::nullopt;
VideoFrameType encoded_frame_type = VideoFrameType::kEmptyFrame;
DataSize encoded_image_size = DataSize::Bytes(0);
@@ -76,15 +76,15 @@
// time index. The QP value here corresponds to one of the encoded spatial
// layer's QP given in `spatial_layers_qp`, i.e. to the one that corresponds
// to the rendered frame.
- absl::optional<uint8_t> decoded_frame_qp = absl::nullopt;
+ std::optional<uint8_t> decoded_frame_qp = std::nullopt;
- absl::optional<int> decoded_frame_width = absl::nullopt;
- absl::optional<int> decoded_frame_height = absl::nullopt;
+ std::optional<int> decoded_frame_width = std::nullopt;
+ std::optional<int> decoded_frame_height = std::nullopt;
// Can be not set if frame was dropped by encoder.
- absl::optional<StreamCodecInfo> used_encoder = absl::nullopt;
+ std::optional<StreamCodecInfo> used_encoder = std::nullopt;
// Can be not set if frame was dropped in the network.
- absl::optional<StreamCodecInfo> used_decoder = absl::nullopt;
+ std::optional<StreamCodecInfo> used_decoder = std::nullopt;
bool decoder_failed = false;
};
@@ -121,8 +121,8 @@
// true or false showing was frame dropped or not.
struct FrameComparison {
FrameComparison(InternalStatsKey stats_key,
- absl::optional<VideoFrame> captured,
- absl::optional<VideoFrame> rendered,
+ std::optional<VideoFrame> captured,
+ std::optional<VideoFrame> rendered,
FrameComparisonType type,
FrameStats frame_stats,
OverloadReason overload_reason);
@@ -130,8 +130,8 @@
InternalStatsKey stats_key;
// Frames can be omitted if there too many computations waiting in the
// queue.
- absl::optional<VideoFrame> captured;
- absl::optional<VideoFrame> rendered;
+ std::optional<VideoFrame> captured;
+ std::optional<VideoFrame> rendered;
FrameComparisonType type;
FrameStats frame_stats;
OverloadReason overload_reason;
diff --git a/test/pc/e2e/analyzer/video/default_video_quality_analyzer_metric_names_test.cc b/test/pc/e2e/analyzer/video/default_video_quality_analyzer_metric_names_test.cc
index 59ce998..906db56 100644
--- a/test/pc/e2e/analyzer/video/default_video_quality_analyzer_metric_names_test.cc
+++ b/test/pc/e2e/analyzer/video/default_video_quality_analyzer_metric_names_test.cc
@@ -160,8 +160,8 @@
TEST(DefaultVideoQualityAnalyzerMetricNamesTest, MetricNamesForP2PAreCorrect) {
std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
test::CreateSquareFrameGenerator(kFrameWidth, kFrameHeight,
- /*type=*/absl::nullopt,
- /*num_squares=*/absl::nullopt);
+ /*type=*/std::nullopt,
+ /*num_squares=*/std::nullopt);
DefaultVideoQualityAnalyzerOptions options = AnalyzerOptionsForTest();
DefaultMetricsLogger metrics_logger(Clock::GetRealTimeClock());
@@ -336,8 +336,8 @@
MetricNamesFor3PeersAreCorrect) {
std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
test::CreateSquareFrameGenerator(kFrameWidth, kFrameHeight,
- /*type=*/absl::nullopt,
- /*num_squares=*/absl::nullopt);
+ /*type=*/std::nullopt,
+ /*num_squares=*/std::nullopt);
DefaultVideoQualityAnalyzerOptions options = AnalyzerOptionsForTest();
DefaultMetricsLogger metrics_logger(Clock::GetRealTimeClock());
@@ -662,8 +662,8 @@
TestCaseFor3PeerIsTheSameAfterAllPeersLeft) {
std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
test::CreateSquareFrameGenerator(kFrameWidth, kFrameHeight,
- /*type=*/absl::nullopt,
- /*num_squares=*/absl::nullopt);
+ /*type=*/std::nullopt,
+ /*num_squares=*/std::nullopt);
DefaultVideoQualityAnalyzerOptions options = AnalyzerOptionsForTest();
DefaultMetricsLogger metrics_logger(Clock::GetRealTimeClock());
diff --git a/test/pc/e2e/analyzer/video/default_video_quality_analyzer_shared_objects.cc b/test/pc/e2e/analyzer/video/default_video_quality_analyzer_shared_objects.cc
index 79b9286..7915df3 100644
--- a/test/pc/e2e/analyzer/video/default_video_quality_analyzer_shared_objects.cc
+++ b/test/pc/e2e/analyzer/video/default_video_quality_analyzer_shared_objects.cc
@@ -151,11 +151,11 @@
return it->second;
}
-absl::optional<std::string> VideoStreamsInfo::GetSender(
+std::optional<std::string> VideoStreamsInfo::GetSender(
absl::string_view stream_label) const {
auto it = stream_to_sender_.find(std::string(stream_label));
if (it == stream_to_sender_.end()) {
- return absl::nullopt;
+ return std::nullopt;
}
return it->second;
}
diff --git a/test/pc/e2e/analyzer/video/default_video_quality_analyzer_shared_objects.h b/test/pc/e2e/analyzer/video/default_video_quality_analyzer_shared_objects.h
index e5ab235..a2dbd5c 100644
--- a/test/pc/e2e/analyzer/video/default_video_quality_analyzer_shared_objects.h
+++ b/test/pc/e2e/analyzer/video/default_video_quality_analyzer_shared_objects.h
@@ -14,13 +14,13 @@
#include <cstdint>
#include <map>
#include <memory>
+#include <optional>
#include <ostream>
#include <set>
#include <string>
#include <utility>
#include <vector>
-#include "absl/types/optional.h"
#include "api/numerics/samples_stats_counter.h"
#include "api/units/time_delta.h"
#include "api/units/timestamp.h"
@@ -238,9 +238,9 @@
// empty set will be returned.
std::set<std::string> GetStreams(absl::string_view sender_name) const;
- // Returns sender name for specified `stream_label`. Returns `absl::nullopt`
+ // Returns sender name for specified `stream_label`. Returns `std::nullopt`
// if provided `stream_label` isn't known to the video analyzer.
- absl::optional<std::string> GetSender(absl::string_view stream_label) const;
+ std::optional<std::string> GetSender(absl::string_view stream_label) const;
// Returns set of the receivers for specified `stream_label`. If stream wasn't
// received by any peer or `stream_label` isn't known to the video analyzer
diff --git a/test/pc/e2e/analyzer/video/default_video_quality_analyzer_stream_state.cc b/test/pc/e2e/analyzer/video/default_video_quality_analyzer_stream_state.cc
index bcdf16d..4b7b063 100644
--- a/test/pc/e2e/analyzer/video/default_video_quality_analyzer_stream_state.cc
+++ b/test/pc/e2e/analyzer/video/default_video_quality_analyzer_stream_state.cc
@@ -10,10 +10,10 @@
#include "test/pc/e2e/analyzer/video/default_video_quality_analyzer_stream_state.h"
+#include <optional>
#include <set>
#include <unordered_map>
-#include "absl/types/optional.h"
#include "api/units/timestamp.h"
#include "rtc_base/checks.h"
#include "system_wrappers/include/clock.h"
@@ -23,11 +23,11 @@
namespace {
template <typename T>
-absl::optional<T> MaybeGetValue(const std::unordered_map<size_t, T>& map,
- size_t key) {
+std::optional<T> MaybeGetValue(const std::unordered_map<size_t, T>& map,
+ size_t key) {
auto it = map.find(key);
if (it == map.end()) {
- return absl::nullopt;
+ return std::nullopt;
}
return it->second;
}
@@ -53,7 +53,7 @@
uint16_t StreamState::PopFront(size_t peer) {
RTC_CHECK_NE(peer, kAliveFramesQueueIndex);
- absl::optional<uint16_t> frame_id = frame_ids_.PopFront(peer);
+ std::optional<uint16_t> frame_id = frame_ids_.PopFront(peer);
RTC_DCHECK(frame_id.has_value());
// If alive's frame queue is longer than all others, than also pop frame from
@@ -62,7 +62,7 @@
size_t other_size = GetLongestReceiverQueue();
// Pops frame from alive queue if alive's queue is the longest one.
if (alive_size > other_size) {
- absl::optional<uint16_t> alive_frame_id =
+ std::optional<uint16_t> alive_frame_id =
frame_ids_.PopFront(kAliveFramesQueueIndex);
RTC_DCHECK(alive_frame_id.has_value());
RTC_DCHECK_EQ(frame_id.value(), alive_frame_id.value());
@@ -110,7 +110,7 @@
}
}
-absl::optional<Timestamp> StreamState::last_rendered_frame_time(
+std::optional<Timestamp> StreamState::last_rendered_frame_time(
size_t peer) const {
return MaybeGetValue(last_rendered_frame_time_, peer);
}
diff --git a/test/pc/e2e/analyzer/video/default_video_quality_analyzer_stream_state.h b/test/pc/e2e/analyzer/video/default_video_quality_analyzer_stream_state.h
index 22fbfd4..bfc28e0 100644
--- a/test/pc/e2e/analyzer/video/default_video_quality_analyzer_stream_state.h
+++ b/test/pc/e2e/analyzer/video/default_video_quality_analyzer_stream_state.h
@@ -12,10 +12,10 @@
#define TEST_PC_E2E_ANALYZER_VIDEO_DEFAULT_VIDEO_QUALITY_ANALYZER_STREAM_STATE_H_
#include <limits>
+#include <optional>
#include <set>
#include <unordered_map>
-#include "absl/types/optional.h"
#include "api/units/timestamp.h"
#include "system_wrappers/include/clock.h"
#include "test/pc/e2e/analyzer/video/dvqa/pausable_state.h"
@@ -73,19 +73,19 @@
void SetLastCapturedFrameTime(Timestamp time) {
last_captured_frame_time_ = time;
}
- absl::optional<Timestamp> last_captured_frame_time() const {
+ std::optional<Timestamp> last_captured_frame_time() const {
return last_captured_frame_time_;
}
void SetLastEncodedFrameTime(Timestamp time) {
last_encoded_frame_time_ = time;
}
- absl::optional<Timestamp> last_encoded_frame_time() const {
+ std::optional<Timestamp> last_encoded_frame_time() const {
return last_encoded_frame_time_;
}
void SetLastRenderedFrameTime(size_t peer, Timestamp time);
- absl::optional<Timestamp> last_rendered_frame_time(size_t peer) const;
+ std::optional<Timestamp> last_rendered_frame_time(size_t peer) const;
private:
// Index of the `frame_ids_` queue which is used to track alive frames for
@@ -113,8 +113,8 @@
// frame_id2 and consider those frames as dropped and then compare received
// frame with the one from `FrameInFlight` with id frame_id3.
MultiReaderQueue<uint16_t> frame_ids_;
- absl::optional<Timestamp> last_captured_frame_time_ = absl::nullopt;
- absl::optional<Timestamp> last_encoded_frame_time_ = absl::nullopt;
+ std::optional<Timestamp> last_captured_frame_time_ = std::nullopt;
+ std::optional<Timestamp> last_encoded_frame_time_ = std::nullopt;
std::unordered_map<size_t, Timestamp> last_rendered_frame_time_;
// Mapping from peer's index to pausable state for this receiver.
std::unordered_map<size_t, PausableState> pausable_state_;
diff --git a/test/pc/e2e/analyzer/video/default_video_quality_analyzer_test.cc b/test/pc/e2e/analyzer/video/default_video_quality_analyzer_test.cc
index 97c0496..93b13f1 100644
--- a/test/pc/e2e/analyzer/video/default_video_quality_analyzer_test.cc
+++ b/test/pc/e2e/analyzer/video/default_video_quality_analyzer_test.cc
@@ -211,8 +211,8 @@
TEST(DefaultVideoQualityAnalyzerTest, NormalScenario) {
std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
test::CreateSquareFrameGenerator(kFrameWidth, kFrameHeight,
- /*type=*/absl::nullopt,
- /*num_squares=*/absl::nullopt);
+ /*type=*/std::nullopt,
+ /*num_squares=*/std::nullopt);
DefaultVideoQualityAnalyzer analyzer(Clock::GetRealTimeClock(),
test::GetGlobalMetricsLogger(),
@@ -271,8 +271,8 @@
TEST(DefaultVideoQualityAnalyzerTest, OneFrameReceivedTwice) {
std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
test::CreateSquareFrameGenerator(kFrameWidth, kFrameHeight,
- /*type=*/absl::nullopt,
- /*num_squares=*/absl::nullopt);
+ /*type=*/std::nullopt,
+ /*num_squares=*/std::nullopt);
DefaultVideoQualityAnalyzer analyzer(Clock::GetRealTimeClock(),
test::GetGlobalMetricsLogger(),
@@ -324,8 +324,8 @@
TEST(DefaultVideoQualityAnalyzerTest, NormalScenario2Receivers) {
std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
test::CreateSquareFrameGenerator(kFrameWidth, kFrameHeight,
- /*type=*/absl::nullopt,
- /*num_squares=*/absl::nullopt);
+ /*type=*/std::nullopt,
+ /*num_squares=*/std::nullopt);
constexpr char kAlice[] = "alice";
constexpr char kBob[] = "bob";
@@ -456,8 +456,8 @@
OneFrameReceivedTwiceBySamePeerWith2Receivers) {
std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
test::CreateSquareFrameGenerator(kFrameWidth, kFrameHeight,
- /*type=*/absl::nullopt,
- /*num_squares=*/absl::nullopt);
+ /*type=*/std::nullopt,
+ /*num_squares=*/std::nullopt);
constexpr char kAlice[] = "alice";
constexpr char kBob[] = "bob";
@@ -514,8 +514,8 @@
TEST(DefaultVideoQualityAnalyzerTest, HeavyQualityMetricsFromEqualFrames) {
std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
test::CreateSquareFrameGenerator(kFrameWidth, kFrameHeight,
- /*type=*/absl::nullopt,
- /*num_squares=*/absl::nullopt);
+ /*type=*/std::nullopt,
+ /*num_squares=*/std::nullopt);
DefaultVideoQualityAnalyzerOptions analyzer_options;
analyzer_options.compute_psnr = true;
@@ -573,8 +573,8 @@
HeavyQualityMetricsFromShiftedFramesWithAdjustment) {
std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
test::CreateSquareFrameGenerator(kFrameWidth, kFrameHeight,
- /*type=*/absl::nullopt,
- /*num_squares=*/absl::nullopt);
+ /*type=*/std::nullopt,
+ /*num_squares=*/std::nullopt);
DefaultVideoQualityAnalyzerOptions analyzer_options;
analyzer_options.compute_psnr = true;
@@ -637,8 +637,8 @@
TEST(DefaultVideoQualityAnalyzerTest, CpuUsage) {
std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
test::CreateSquareFrameGenerator(kFrameWidth, kFrameHeight,
- /*type=*/absl::nullopt,
- /*num_squares=*/absl::nullopt);
+ /*type=*/std::nullopt,
+ /*num_squares=*/std::nullopt);
DefaultVideoQualityAnalyzer analyzer(Clock::GetRealTimeClock(),
test::GetGlobalMetricsLogger(),
@@ -693,8 +693,8 @@
TEST(DefaultVideoQualityAnalyzerTest, RuntimeParticipantsAdding) {
std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
test::CreateSquareFrameGenerator(kFrameWidth, kFrameHeight,
- /*type=*/absl::nullopt,
- /*num_squares=*/absl::nullopt);
+ /*type=*/std::nullopt,
+ /*num_squares=*/std::nullopt);
constexpr char kAlice[] = "alice";
constexpr char kBob[] = "bob";
@@ -849,8 +849,8 @@
SimulcastFrameWasFullyReceivedByAllPeersBeforeEncodeFinish) {
std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
test::CreateSquareFrameGenerator(kFrameWidth, kFrameHeight,
- /*type=*/absl::nullopt,
- /*num_squares=*/absl::nullopt);
+ /*type=*/std::nullopt,
+ /*num_squares=*/std::nullopt);
DefaultVideoQualityAnalyzer analyzer(Clock::GetRealTimeClock(),
test::GetGlobalMetricsLogger(),
@@ -911,8 +911,8 @@
FrameCanBeReceivedBySenderAfterItWasReceivedByReceiver) {
std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
test::CreateSquareFrameGenerator(kFrameWidth, kFrameHeight,
- /*type=*/absl::nullopt,
- /*num_squares=*/absl::nullopt);
+ /*type=*/std::nullopt,
+ /*num_squares=*/std::nullopt);
DefaultVideoQualityAnalyzerOptions options = AnalyzerOptionsForTest();
options.enable_receive_own_stream = true;
@@ -1008,8 +1008,8 @@
FrameCanBeReceivedByReceiverAfterItWasReceivedBySender) {
std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
test::CreateSquareFrameGenerator(kFrameWidth, kFrameHeight,
- /*type=*/absl::nullopt,
- /*num_squares=*/absl::nullopt);
+ /*type=*/std::nullopt,
+ /*num_squares=*/std::nullopt);
DefaultVideoQualityAnalyzerOptions options = AnalyzerOptionsForTest();
options.enable_receive_own_stream = true;
@@ -1104,8 +1104,8 @@
TEST(DefaultVideoQualityAnalyzerTest, CodecTrackedCorrectly) {
std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
test::CreateSquareFrameGenerator(kFrameWidth, kFrameHeight,
- /*type=*/absl::nullopt,
- /*num_squares=*/absl::nullopt);
+ /*type=*/std::nullopt,
+ /*num_squares=*/std::nullopt);
DefaultVideoQualityAnalyzer analyzer(Clock::GetRealTimeClock(),
test::GetGlobalMetricsLogger(),
@@ -1175,8 +1175,8 @@
FramesInFlightAreCorrectlySentToTheComparatorAfterStop) {
std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
test::CreateSquareFrameGenerator(kFrameWidth, kFrameHeight,
- /*type=*/absl::nullopt,
- /*num_squares=*/absl::nullopt);
+ /*type=*/std::nullopt,
+ /*num_squares=*/std::nullopt);
DefaultVideoQualityAnalyzerOptions options = AnalyzerOptionsForTest();
DefaultVideoQualityAnalyzer analyzer(Clock::GetRealTimeClock(),
@@ -1267,8 +1267,8 @@
FramesInFlightAreCorrectlySentToTheComparatorAfterStopForSenderAndReceiver) {
std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
test::CreateSquareFrameGenerator(kFrameWidth, kFrameHeight,
- /*type=*/absl::nullopt,
- /*num_squares=*/absl::nullopt);
+ /*type=*/std::nullopt,
+ /*num_squares=*/std::nullopt);
DefaultVideoQualityAnalyzerOptions options = AnalyzerOptionsForTest();
options.enable_receive_own_stream = true;
@@ -1373,8 +1373,8 @@
TEST(DefaultVideoQualityAnalyzerTest, GetStreamFrames) {
std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
test::CreateSquareFrameGenerator(kFrameWidth, kFrameHeight,
- /*type=*/absl::nullopt,
- /*num_squares=*/absl::nullopt);
+ /*type=*/std::nullopt,
+ /*num_squares=*/std::nullopt);
DefaultVideoQualityAnalyzerOptions options = AnalyzerOptionsForTest();
DefaultVideoQualityAnalyzer analyzer(Clock::GetRealTimeClock(),
@@ -1419,8 +1419,8 @@
TEST(DefaultVideoQualityAnalyzerTest, ReceiverReceivedFramesWhenSenderRemoved) {
std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
test::CreateSquareFrameGenerator(kFrameWidth, kFrameHeight,
- /*type=*/absl::nullopt,
- /*num_squares=*/absl::nullopt);
+ /*type=*/std::nullopt,
+ /*num_squares=*/std::nullopt);
DefaultVideoQualityAnalyzerOptions options = AnalyzerOptionsForTest();
DefaultVideoQualityAnalyzer analyzer(Clock::GetRealTimeClock(),
@@ -1462,8 +1462,8 @@
ReceiverReceivedFramesWhenSenderRemovedWithSelfview) {
std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
test::CreateSquareFrameGenerator(kFrameWidth, kFrameHeight,
- /*type=*/absl::nullopt,
- /*num_squares=*/absl::nullopt);
+ /*type=*/std::nullopt,
+ /*num_squares=*/std::nullopt);
DefaultVideoQualityAnalyzerOptions options = AnalyzerOptionsForTest();
options.enable_receive_own_stream = true;
@@ -1506,8 +1506,8 @@
SenderReceivedFramesWhenReceiverRemovedWithSelfview) {
std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
test::CreateSquareFrameGenerator(kFrameWidth, kFrameHeight,
- /*type=*/absl::nullopt,
- /*num_squares=*/absl::nullopt);
+ /*type=*/std::nullopt,
+ /*num_squares=*/std::nullopt);
DefaultVideoQualityAnalyzerOptions options = AnalyzerOptionsForTest();
options.enable_receive_own_stream = true;
@@ -1550,8 +1550,8 @@
SenderAndReceiverReceivedFramesWhenReceiverRemovedWithSelfview) {
std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
test::CreateSquareFrameGenerator(kFrameWidth, kFrameHeight,
- /*type=*/absl::nullopt,
- /*num_squares=*/absl::nullopt);
+ /*type=*/std::nullopt,
+ /*num_squares=*/std::nullopt);
DefaultVideoQualityAnalyzerOptions options = AnalyzerOptionsForTest();
options.enable_receive_own_stream = true;
@@ -1607,8 +1607,8 @@
TEST(DefaultVideoQualityAnalyzerTest, ReceiverRemovedBeforeCapturing2ndFrame) {
std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
test::CreateSquareFrameGenerator(kFrameWidth, kFrameHeight,
- /*type=*/absl::nullopt,
- /*num_squares=*/absl::nullopt);
+ /*type=*/std::nullopt,
+ /*num_squares=*/std::nullopt);
DefaultVideoQualityAnalyzerOptions options = AnalyzerOptionsForTest();
DefaultVideoQualityAnalyzer analyzer(Clock::GetRealTimeClock(),
@@ -1648,8 +1648,8 @@
TEST(DefaultVideoQualityAnalyzerTest, ReceiverRemovedBeforePreEncoded) {
std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
test::CreateSquareFrameGenerator(kFrameWidth, kFrameHeight,
- /*type=*/absl::nullopt,
- /*num_squares=*/absl::nullopt);
+ /*type=*/std::nullopt,
+ /*num_squares=*/std::nullopt);
DefaultVideoQualityAnalyzerOptions options = AnalyzerOptionsForTest();
DefaultVideoQualityAnalyzer analyzer(Clock::GetRealTimeClock(),
@@ -1691,8 +1691,8 @@
TEST(DefaultVideoQualityAnalyzerTest, ReceiverRemovedBeforeEncoded) {
std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
test::CreateSquareFrameGenerator(kFrameWidth, kFrameHeight,
- /*type=*/absl::nullopt,
- /*num_squares=*/absl::nullopt);
+ /*type=*/std::nullopt,
+ /*num_squares=*/std::nullopt);
DefaultVideoQualityAnalyzerOptions options = AnalyzerOptionsForTest();
DefaultVideoQualityAnalyzer analyzer(Clock::GetRealTimeClock(),
@@ -1735,8 +1735,8 @@
ReceiverRemovedBetweenSimulcastLayersEncoded) {
std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
test::CreateSquareFrameGenerator(kFrameWidth, kFrameHeight,
- /*type=*/absl::nullopt,
- /*num_squares=*/absl::nullopt);
+ /*type=*/std::nullopt,
+ /*num_squares=*/std::nullopt);
DefaultVideoQualityAnalyzerOptions options = AnalyzerOptionsForTest();
DefaultVideoQualityAnalyzer analyzer(Clock::GetRealTimeClock(),
@@ -1782,8 +1782,8 @@
TEST(DefaultVideoQualityAnalyzerTest, UnregisterOneAndRegisterAnother) {
std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
test::CreateSquareFrameGenerator(kFrameWidth, kFrameHeight,
- /*type=*/absl::nullopt,
- /*num_squares=*/absl::nullopt);
+ /*type=*/std::nullopt,
+ /*num_squares=*/std::nullopt);
DefaultVideoQualityAnalyzerOptions options = AnalyzerOptionsForTest();
DefaultVideoQualityAnalyzer analyzer(Clock::GetRealTimeClock(),
@@ -1844,8 +1844,8 @@
UnregisterOneAndRegisterAnotherRegisterBack) {
std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
test::CreateSquareFrameGenerator(kFrameWidth, kFrameHeight,
- /*type=*/absl::nullopt,
- /*num_squares=*/absl::nullopt);
+ /*type=*/std::nullopt,
+ /*num_squares=*/std::nullopt);
DefaultVideoQualityAnalyzerOptions options = AnalyzerOptionsForTest();
DefaultVideoQualityAnalyzer analyzer(Clock::GetRealTimeClock(),
@@ -1900,8 +1900,8 @@
FramesInFlightAreAccountedForUnregisterPeers) {
std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
test::CreateSquareFrameGenerator(kFrameWidth, kFrameHeight,
- /*type=*/absl::nullopt,
- /*num_squares=*/absl::nullopt);
+ /*type=*/std::nullopt,
+ /*num_squares=*/std::nullopt);
DefaultVideoQualityAnalyzerOptions options = AnalyzerOptionsForTest();
DefaultVideoQualityAnalyzer analyzer(Clock::GetRealTimeClock(),
@@ -1934,8 +1934,8 @@
TEST(DefaultVideoQualityAnalyzerTest, InfraMetricsAreReportedWhenRequested) {
std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
test::CreateSquareFrameGenerator(kFrameWidth, kFrameHeight,
- /*type=*/absl::nullopt,
- /*num_squares=*/absl::nullopt);
+ /*type=*/std::nullopt,
+ /*num_squares=*/std::nullopt);
DefaultVideoQualityAnalyzerOptions options = AnalyzerOptionsForTest();
options.report_infra_metrics = true;
@@ -1966,8 +1966,8 @@
TEST(DefaultVideoQualityAnalyzerTest, InfraMetricsNotCollectedByDefault) {
std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
test::CreateSquareFrameGenerator(kFrameWidth, kFrameHeight,
- /*type=*/absl::nullopt,
- /*num_squares=*/absl::nullopt);
+ /*type=*/std::nullopt,
+ /*num_squares=*/std::nullopt);
DefaultVideoQualityAnalyzerOptions options = AnalyzerOptionsForTest();
options.report_infra_metrics = false;
@@ -1999,8 +1999,8 @@
FrameDroppedByDecoderIsAccountedCorrectly) {
std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
test::CreateSquareFrameGenerator(kFrameWidth, kFrameHeight,
- /*type=*/absl::nullopt,
- /*num_squares=*/absl::nullopt);
+ /*type=*/std::nullopt,
+ /*num_squares=*/std::nullopt);
DefaultVideoQualityAnalyzerOptions options = AnalyzerOptionsForTest();
options.report_infra_metrics = false;
@@ -2041,8 +2041,8 @@
TimeBetweenFreezesIsEqualToStreamDurationWhenThereAreNoFeeezes) {
std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
test::CreateSquareFrameGenerator(kFrameWidth, kFrameHeight,
- /*type=*/absl::nullopt,
- /*num_squares=*/absl::nullopt);
+ /*type=*/std::nullopt,
+ /*num_squares=*/std::nullopt);
DefaultVideoQualityAnalyzerOptions options = AnalyzerOptionsForTest();
DefaultVideoQualityAnalyzer analyzer(Clock::GetRealTimeClock(),
@@ -2094,8 +2094,8 @@
PausedAndResumedStreamIsAccountedInStatsCorrectly) {
std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
test::CreateSquareFrameGenerator(kFrameWidth, kFrameHeight,
- /*type=*/absl::nullopt,
- /*num_squares=*/absl::nullopt);
+ /*type=*/std::nullopt,
+ /*num_squares=*/std::nullopt);
DefaultVideoQualityAnalyzerOptions options = AnalyzerOptionsForTest();
options.report_infra_metrics = false;
@@ -2172,8 +2172,8 @@
PausedAndResumedTwoStreamsAreAccountedInStatsCorrectly) {
std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
test::CreateSquareFrameGenerator(kFrameWidth, kFrameHeight,
- /*type=*/absl::nullopt,
- /*num_squares=*/absl::nullopt);
+ /*type=*/std::nullopt,
+ /*num_squares=*/std::nullopt);
DefaultVideoQualityAnalyzer analyzer(
GetClock(), test::GetGlobalMetricsLogger(), AnalyzerOptionsForTest());
@@ -2250,8 +2250,8 @@
PausedStreamIsAccountedInStatsCorrectly) {
std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
test::CreateSquareFrameGenerator(kFrameWidth, kFrameHeight,
- /*type=*/absl::nullopt,
- /*num_squares=*/absl::nullopt);
+ /*type=*/std::nullopt,
+ /*num_squares=*/std::nullopt);
DefaultVideoQualityAnalyzerOptions options = AnalyzerOptionsForTest();
options.report_infra_metrics = false;
@@ -2320,8 +2320,8 @@
MemoryOverloadedAndThenAllFramesReceived) {
std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
test::CreateSquareFrameGenerator(kFrameWidth, kFrameHeight,
- /*type=*/absl::nullopt,
- /*num_squares=*/absl::nullopt);
+ /*type=*/std::nullopt,
+ /*num_squares=*/std::nullopt);
DefaultVideoQualityAnalyzer analyzer(
GetClock(), test::GetGlobalMetricsLogger(), AnalyzerOptionsForTest());
@@ -2382,8 +2382,8 @@
constexpr char kBobStreamLabel[] = "bob-video";
std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
test::CreateSquareFrameGenerator(kFrameWidth, kFrameHeight,
- /*type=*/absl::nullopt,
- /*num_squares=*/absl::nullopt);
+ /*type=*/std::nullopt,
+ /*num_squares=*/std::nullopt);
DefaultVideoQualityAnalyzer analyzer(Clock::GetRealTimeClock(),
test::GetGlobalMetricsLogger(),
AnalyzerOptionsForTest());
diff --git a/test/pc/e2e/analyzer/video/dvqa/BUILD.gn b/test/pc/e2e/analyzer/video/dvqa/BUILD.gn
index aae1852..fd250f4 100644
--- a/test/pc/e2e/analyzer/video/dvqa/BUILD.gn
+++ b/test/pc/e2e/analyzer/video/dvqa/BUILD.gn
@@ -72,7 +72,6 @@
"../../../../../../api/units:timestamp",
"../../../../../../api/video:video_frame",
"../../../../../../system_wrappers",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
diff --git a/test/pc/e2e/analyzer/video/dvqa/frames_storage.cc b/test/pc/e2e/analyzer/video/dvqa/frames_storage.cc
index 2a70468..369295f 100644
--- a/test/pc/e2e/analyzer/video/dvqa/frames_storage.cc
+++ b/test/pc/e2e/analyzer/video/dvqa/frames_storage.cc
@@ -11,9 +11,9 @@
#include "test/pc/e2e/analyzer/video/dvqa/frames_storage.h"
#include <cstdint>
+#include <optional>
#include <utility>
-#include "absl/types/optional.h"
#include "api/units/timestamp.h"
#include "api/video/video_frame.h"
@@ -26,10 +26,10 @@
RemoveTooOldFrames();
}
-absl::optional<VideoFrame> FramesStorage::Get(uint16_t frame_id) {
+std::optional<VideoFrame> FramesStorage::Get(uint16_t frame_id) {
auto it = frame_id_index_.find(frame_id);
if (it == frame_id_index_.end()) {
- return absl::nullopt;
+ return std::nullopt;
}
return heap_[it->second].frame;
diff --git a/test/pc/e2e/analyzer/video/dvqa/frames_storage.h b/test/pc/e2e/analyzer/video/dvqa/frames_storage.h
index d3c6bd4..05012e6 100644
--- a/test/pc/e2e/analyzer/video/dvqa/frames_storage.h
+++ b/test/pc/e2e/analyzer/video/dvqa/frames_storage.h
@@ -12,10 +12,10 @@
#define TEST_PC_E2E_ANALYZER_VIDEO_DVQA_FRAMES_STORAGE_H_
#include <cstdint>
+#include <optional>
#include <unordered_map>
#include <vector>
-#include "absl/types/optional.h"
#include "api/units/time_delta.h"
#include "api/units/timestamp.h"
#include "api/video/video_frame.h"
@@ -41,7 +41,7 @@
void Add(const VideoFrame& frame, Timestamp captured_time);
// Complexity: O(1)
- absl::optional<VideoFrame> Get(uint16_t frame_id);
+ std::optional<VideoFrame> Get(uint16_t frame_id);
// Removes the frame identified by `frame_id` from the storage. No error
// happens in case there isn't a frame identified by `frame_id`.
diff --git a/test/pc/e2e/analyzer/video/dvqa/frames_storage_test.cc b/test/pc/e2e/analyzer/video/dvqa/frames_storage_test.cc
index 88a172e..ae6e95d 100644
--- a/test/pc/e2e/analyzer/video/dvqa/frames_storage_test.cc
+++ b/test/pc/e2e/analyzer/video/dvqa/frames_storage_test.cc
@@ -40,7 +40,7 @@
}
void AssertHasFrame(FramesStorage& storage, uint16_t frame_id) {
- absl::optional<VideoFrame> frame = storage.Get(frame_id);
+ std::optional<VideoFrame> frame = storage.Get(frame_id);
ASSERT_TRUE(frame.has_value()) << "Frame " << frame_id << " wasn't found";
EXPECT_EQ(frame->id(), frame_id);
}
diff --git a/test/pc/e2e/analyzer/video/encoded_image_data_injector.h b/test/pc/e2e/analyzer/video/encoded_image_data_injector.h
index 384e901..3074701 100644
--- a/test/pc/e2e/analyzer/video/encoded_image_data_injector.h
+++ b/test/pc/e2e/analyzer/video/encoded_image_data_injector.h
@@ -12,9 +12,9 @@
#define TEST_PC_E2E_ANALYZER_VIDEO_ENCODED_IMAGE_DATA_INJECTOR_H_
#include <cstdint>
+#include <optional>
#include <utility>
-#include "absl/types/optional.h"
#include "api/video/encoded_image.h"
namespace webrtc {
@@ -35,7 +35,7 @@
};
struct EncodedImageExtractionResult {
- absl::optional<uint16_t> id;
+ std::optional<uint16_t> id;
EncodedImage image;
// Is true if encoded image should be discarded. It is used to filter out
// unnecessary spatial layers and simulcast streams.
diff --git a/test/pc/e2e/analyzer/video/multi_reader_queue.h b/test/pc/e2e/analyzer/video/multi_reader_queue.h
index 39d26b4..99c53d0 100644
--- a/test/pc/e2e/analyzer/video/multi_reader_queue.h
+++ b/test/pc/e2e/analyzer/video/multi_reader_queue.h
@@ -13,10 +13,10 @@
#include <deque>
#include <memory>
+#include <optional>
#include <set>
#include <unordered_map>
-#include "absl/types/optional.h"
#include "rtc_base/checks.h"
namespace webrtc {
@@ -90,10 +90,10 @@
}
// Extract element from specified head. Complexity O(1).
- absl::optional<T> PopFront(size_t reader) {
+ std::optional<T> PopFront(size_t reader) {
size_t pos = GetHeadPositionOrDie(reader);
if (pos >= queue_.size()) {
- return absl::nullopt;
+ return std::nullopt;
}
T out = queue_[pos];
@@ -109,10 +109,10 @@
}
// Returns element at specified head. Complexity O(1).
- absl::optional<T> Front(size_t reader) const {
+ std::optional<T> Front(size_t reader) const {
size_t pos = GetHeadPositionOrDie(reader);
if (pos >= queue_.size()) {
- return absl::nullopt;
+ return std::nullopt;
}
return queue_[pos];
}
diff --git a/test/pc/e2e/analyzer/video/multi_reader_queue_test.cc b/test/pc/e2e/analyzer/video/multi_reader_queue_test.cc
index ea6aa0a..a035f44 100644
--- a/test/pc/e2e/analyzer/video/multi_reader_queue_test.cc
+++ b/test/pc/e2e/analyzer/video/multi_reader_queue_test.cc
@@ -10,7 +10,8 @@
#include "test/pc/e2e/analyzer/video/multi_reader_queue.h"
-#include "absl/types/optional.h"
+#include <optional>
+
#include "test/gtest.h"
namespace webrtc {
@@ -48,7 +49,7 @@
// Removing elements from queue #0
for (int i = 0; i < 5; ++i) {
EXPECT_EQ(queue.size(/*reader=*/0), static_cast<size_t>(5 - i));
- EXPECT_EQ(queue.PopFront(/*reader=*/0), absl::optional<int>(i));
+ EXPECT_EQ(queue.PopFront(/*reader=*/0), std::optional<int>(i));
for (int j = 1; j < 10; ++j) {
EXPECT_EQ(queue.size(/*reader=*/j), 5lu);
}
@@ -63,7 +64,7 @@
EXPECT_EQ(queue.size(), 1lu);
EXPECT_TRUE(queue.Front(/*reader=*/0).has_value());
EXPECT_EQ(queue.Front(/*reader=*/0).value(), 1);
- absl::optional<int> value = queue.PopFront(/*reader=*/0);
+ std::optional<int> value = queue.PopFront(/*reader=*/0);
EXPECT_TRUE(value.has_value());
EXPECT_EQ(value.value(), 1);
EXPECT_EQ(queue.size(), 0lu);
@@ -78,8 +79,8 @@
EXPECT_EQ(queue.size(), i + 1);
}
for (size_t i = 0; i < 10; ++i) {
- EXPECT_EQ(queue.Front(/*reader=*/0), absl::optional<size_t>(i));
- EXPECT_EQ(queue.PopFront(/*reader=*/0), absl::optional<size_t>(i));
+ EXPECT_EQ(queue.Front(/*reader=*/0), std::optional<size_t>(i));
+ EXPECT_EQ(queue.PopFront(/*reader=*/0), std::optional<size_t>(i));
EXPECT_EQ(queue.size(), 10 - i - 1);
}
}
@@ -92,19 +93,19 @@
EXPECT_EQ(queue.size(), i + 1);
}
for (size_t i = 0; i < 10; ++i) {
- absl::optional<size_t> value = queue.PopFront(/*reader=*/0);
+ std::optional<size_t> value = queue.PopFront(/*reader=*/0);
EXPECT_EQ(queue.size(), 10lu);
ASSERT_TRUE(value.has_value());
EXPECT_EQ(value.value(), i);
}
for (size_t i = 0; i < 10; ++i) {
- absl::optional<size_t> value = queue.PopFront(/*reader=*/1);
+ std::optional<size_t> value = queue.PopFront(/*reader=*/1);
EXPECT_EQ(queue.size(), 10lu);
ASSERT_TRUE(value.has_value());
EXPECT_EQ(value.value(), i);
}
for (size_t i = 0; i < 10; ++i) {
- absl::optional<size_t> value = queue.PopFront(/*reader=*/2);
+ std::optional<size_t> value = queue.PopFront(/*reader=*/2);
EXPECT_EQ(queue.size(), 10 - i - 1);
ASSERT_TRUE(value.has_value());
EXPECT_EQ(value.value(), i);
@@ -119,9 +120,9 @@
EXPECT_EQ(queue.size(), i + 1);
}
for (size_t i = 0; i < 10; ++i) {
- absl::optional<size_t> value1 = queue.PopFront(/*reader=*/0);
- absl::optional<size_t> value2 = queue.PopFront(/*reader=*/1);
- absl::optional<size_t> value3 = queue.PopFront(/*reader=*/2);
+ std::optional<size_t> value1 = queue.PopFront(/*reader=*/0);
+ std::optional<size_t> value2 = queue.PopFront(/*reader=*/1);
+ std::optional<size_t> value3 = queue.PopFront(/*reader=*/2);
EXPECT_EQ(queue.size(), 10 - i - 1);
ASSERT_TRUE(value1.has_value());
ASSERT_TRUE(value2.has_value());
@@ -146,7 +147,7 @@
EXPECT_EQ(queue.readers_count(), 3lu);
for (size_t i = 5; i < 10; ++i) {
- absl::optional<size_t> value = queue.PopFront(/*reader=*/2);
+ std::optional<size_t> value = queue.PopFront(/*reader=*/2);
EXPECT_EQ(queue.size(/*reader=*/2), 10 - i - 1);
ASSERT_TRUE(value.has_value());
EXPECT_EQ(value.value(), i);
@@ -167,7 +168,7 @@
EXPECT_EQ(queue.readers_count(), 3lu);
for (size_t i = 0; i < 10; ++i) {
- absl::optional<size_t> value = queue.PopFront(/*reader=*/2);
+ std::optional<size_t> value = queue.PopFront(/*reader=*/2);
EXPECT_EQ(queue.size(/*reader=*/2), 10 - i - 1);
ASSERT_TRUE(value.has_value());
EXPECT_EQ(value.value(), i);
diff --git a/test/pc/e2e/analyzer/video/names_collection.cc b/test/pc/e2e/analyzer/video/names_collection.cc
index 3ccab620..33dbc93 100644
--- a/test/pc/e2e/analyzer/video/names_collection.cc
+++ b/test/pc/e2e/analyzer/video/names_collection.cc
@@ -10,10 +10,10 @@
#include "test/pc/e2e/analyzer/video/names_collection.h"
+#include <optional>
#include <set>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
namespace webrtc {
@@ -65,15 +65,14 @@
return out;
}
-absl::optional<size_t> NamesCollection::RemoveIfPresent(
- absl::string_view name) {
+std::optional<size_t> NamesCollection::RemoveIfPresent(absl::string_view name) {
auto it = index_.find(name);
if (it == index_.end()) {
- return absl::nullopt;
+ return std::nullopt;
}
size_t index = it->second;
if (removed_[index]) {
- return absl::nullopt;
+ return std::nullopt;
}
removed_[index] = true;
size_--;
diff --git a/test/pc/e2e/analyzer/video/names_collection.h b/test/pc/e2e/analyzer/video/names_collection.h
index f9a13a2..a306ca7 100644
--- a/test/pc/e2e/analyzer/video/names_collection.h
+++ b/test/pc/e2e/analyzer/video/names_collection.h
@@ -12,12 +12,12 @@
#define TEST_PC_E2E_ANALYZER_VIDEO_NAMES_COLLECTION_H_
#include <map>
+#include <optional>
#include <set>
#include <string>
#include <vector>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/array_view.h"
namespace webrtc {
@@ -70,9 +70,9 @@
// will continue to return previously known index for `index(name)` and return
// `name` for `name(index(name))`.
//
- // Returns the index of the removed value or absl::nullopt if no such `name`
+ // Returns the index of the removed value or std::nullopt if no such `name`
// registered in the collection.
- absl::optional<size_t> RemoveIfPresent(absl::string_view name);
+ std::optional<size_t> RemoveIfPresent(absl::string_view name);
// Returns a set of indexes for all currently present names in the
// collection.
diff --git a/test/pc/e2e/analyzer/video/names_collection_test.cc b/test/pc/e2e/analyzer/video/names_collection_test.cc
index 6c52f96..e67c23f 100644
--- a/test/pc/e2e/analyzer/video/names_collection_test.cc
+++ b/test/pc/e2e/analyzer/video/names_collection_test.cc
@@ -10,10 +10,10 @@
#include "test/pc/e2e/analyzer/video/names_collection.h"
+#include <optional>
#include <string>
#include <vector>
-#include "absl/types/optional.h"
#include "test/gmock.h"
#include "test/gtest.h"
@@ -76,7 +76,7 @@
EXPECT_THAT(collection.size(), Eq(static_cast<size_t>(2)));
EXPECT_THAT(collection.RemoveIfPresent("bob"),
- Eq(absl::optional<size_t>(bob_index)));
+ Eq(std::optional<size_t>(bob_index)));
EXPECT_THAT(collection.size(), Eq(static_cast<size_t>(1)));
EXPECT_FALSE(collection.HasName("bob"));
@@ -94,7 +94,7 @@
EXPECT_THAT(collection.size(), Eq(static_cast<size_t>(2)));
EXPECT_THAT(collection.RemoveIfPresent("alice"),
- Eq(absl::optional<size_t>(alice_index)));
+ Eq(std::optional<size_t>(alice_index)));
EXPECT_THAT(collection.size(), Eq(static_cast<size_t>(1)));
EXPECT_THAT(collection.index("bob"), Eq(bob_index));
@@ -108,17 +108,17 @@
EXPECT_THAT(collection.size(), Eq(static_cast<size_t>(1)));
EXPECT_THAT(collection.RemoveIfPresent("bob"),
- Eq(absl::optional<size_t>(bob_index)));
+ Eq(std::optional<size_t>(bob_index)));
EXPECT_THAT(collection.size(), Eq(static_cast<size_t>(0)));
- EXPECT_THAT(collection.RemoveIfPresent("bob"), Eq(absl::nullopt));
+ EXPECT_THAT(collection.RemoveIfPresent("bob"), Eq(std::nullopt));
}
TEST(NamesCollectionTest, RemoveOfNotExistingHasNoEffect) {
NamesCollection collection(std::vector<std::string>{"bob"});
EXPECT_THAT(collection.size(), Eq(static_cast<size_t>(1)));
- EXPECT_THAT(collection.RemoveIfPresent("alice"), Eq(absl::nullopt));
+ EXPECT_THAT(collection.RemoveIfPresent("alice"), Eq(std::nullopt));
EXPECT_THAT(collection.size(), Eq(static_cast<size_t>(1)));
}
@@ -129,7 +129,7 @@
EXPECT_THAT(collection.size(), Eq(static_cast<size_t>(1)));
EXPECT_THAT(collection.RemoveIfPresent("alice"),
- Eq(absl::optional<size_t>(alice_index)));
+ Eq(std::optional<size_t>(alice_index)));
EXPECT_THAT(collection.size(), Eq(static_cast<size_t>(0)));
EXPECT_THAT(collection.AddIfAbsent("alice"), Eq(alice_index));
@@ -144,7 +144,7 @@
EXPECT_THAT(collection.GetKnownSize(), Eq(static_cast<size_t>(1)));
EXPECT_THAT(collection.RemoveIfPresent("alice"),
- Eq(absl::optional<size_t>(alice_index)));
+ Eq(std::optional<size_t>(alice_index)));
EXPECT_THAT(collection.GetKnownSize(), Eq(static_cast<size_t>(1)));
}
diff --git a/test/pc/e2e/analyzer/video/quality_analyzing_video_decoder.cc b/test/pc/e2e/analyzer/video/quality_analyzing_video_decoder.cc
index ccddd39..ac99b15 100644
--- a/test/pc/e2e/analyzer/video/quality_analyzing_video_decoder.cc
+++ b/test/pc/e2e/analyzer/video/quality_analyzing_video_decoder.cc
@@ -13,10 +13,10 @@
#include <cstdint>
#include <cstring>
#include <memory>
+#include <optional>
#include <utility>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/video/i420_buffer.h"
#include "api/video/video_frame.h"
#include "modules/video_coding/include/video_error_codes.h"
@@ -156,8 +156,8 @@
// method on `delegate_callback_`, as was called on `this` callback.
int32_t QualityAnalyzingVideoDecoder::DecoderCallback::Decoded(
VideoFrame& decodedImage) {
- decoder_->OnFrameDecoded(&decodedImage, /*decode_time_ms=*/absl::nullopt,
- /*qp=*/absl::nullopt);
+ decoder_->OnFrameDecoded(&decodedImage, /*decode_time_ms=*/std::nullopt,
+ /*qp=*/std::nullopt);
MutexLock lock(&callback_mutex_);
RTC_DCHECK(delegate_callback_);
@@ -167,7 +167,7 @@
int32_t QualityAnalyzingVideoDecoder::DecoderCallback::Decoded(
VideoFrame& decodedImage,
int64_t decode_time_ms) {
- decoder_->OnFrameDecoded(&decodedImage, decode_time_ms, /*qp=*/absl::nullopt);
+ decoder_->OnFrameDecoded(&decodedImage, decode_time_ms, /*qp=*/std::nullopt);
MutexLock lock(&callback_mutex_);
RTC_DCHECK(delegate_callback_);
@@ -176,8 +176,8 @@
void QualityAnalyzingVideoDecoder::DecoderCallback::Decoded(
VideoFrame& decodedImage,
- absl::optional<int32_t> decode_time_ms,
- absl::optional<uint8_t> qp) {
+ std::optional<int32_t> decode_time_ms,
+ std::optional<uint8_t> qp) {
decoder_->OnFrameDecoded(&decodedImage, decode_time_ms, qp);
MutexLock lock(&callback_mutex_);
@@ -197,7 +197,7 @@
.build();
MutexLock lock(&callback_mutex_);
RTC_DCHECK(delegate_callback_);
- delegate_callback_->Decoded(dummy_frame, absl::nullopt, absl::nullopt);
+ delegate_callback_->Decoded(dummy_frame, std::nullopt, std::nullopt);
return WEBRTC_VIDEO_CODEC_OK;
}
@@ -212,9 +212,9 @@
void QualityAnalyzingVideoDecoder::OnFrameDecoded(
VideoFrame* frame,
- absl::optional<int32_t> decode_time_ms,
- absl::optional<uint8_t> qp) {
- absl::optional<uint16_t> frame_id;
+ std::optional<int32_t> decode_time_ms,
+ std::optional<uint8_t> qp) {
+ std::optional<uint16_t> frame_id;
std::string codec_name;
{
MutexLock lock(&mutex_);
diff --git a/test/pc/e2e/analyzer/video/quality_analyzing_video_decoder.h b/test/pc/e2e/analyzer/video/quality_analyzing_video_decoder.h
index daa919d..d70b75f5 100644
--- a/test/pc/e2e/analyzer/video/quality_analyzing_video_decoder.h
+++ b/test/pc/e2e/analyzer/video/quality_analyzing_video_decoder.h
@@ -13,11 +13,11 @@
#include <map>
#include <memory>
+#include <optional>
#include <string>
#include <vector>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/environment/environment.h"
#include "api/test/video_quality_analyzer_interface.h"
#include "api/video/encoded_image.h"
@@ -79,8 +79,8 @@
int32_t Decoded(VideoFrame& decodedImage) override;
int32_t Decoded(VideoFrame& decodedImage, int64_t decode_time_ms) override;
void Decoded(VideoFrame& decodedImage,
- absl::optional<int32_t> decode_time_ms,
- absl::optional<uint8_t> qp) override;
+ std::optional<int32_t> decode_time_ms,
+ std::optional<uint8_t> qp) override;
int32_t IrrelevantSimulcastStreamDecoded(uint16_t frame_id,
uint32_t timestamp_ms);
@@ -97,8 +97,8 @@
};
void OnFrameDecoded(VideoFrame* frame,
- absl::optional<int32_t> decode_time_ms,
- absl::optional<uint8_t> qp);
+ std::optional<int32_t> decode_time_ms,
+ std::optional<uint8_t> qp);
const std::string peer_name_;
const std::string implementation_name_;
@@ -114,7 +114,7 @@
// Name of the video codec type used. Ex: VP8, VP9, H264 etc.
std::string codec_name_ RTC_GUARDED_BY(mutex_);
- std::map<uint32_t, absl::optional<uint16_t>> timestamp_to_frame_id_
+ std::map<uint32_t, std::optional<uint16_t>> timestamp_to_frame_id_
RTC_GUARDED_BY(mutex_);
// Stores currently being decoded images by timestamp. Because
// EncodedImageDataExtractor can create new copy on EncodedImage we need to
diff --git a/test/pc/e2e/analyzer/video/quality_analyzing_video_encoder.cc b/test/pc/e2e/analyzer/video/quality_analyzing_video_encoder.cc
index 7ecfa70..b8790e9 100644
--- a/test/pc/e2e/analyzer/video/quality_analyzing_video_encoder.cc
+++ b/test/pc/e2e/analyzer/video/quality_analyzing_video_encoder.cc
@@ -85,7 +85,7 @@
MutexLock lock(&mutex_);
codec_settings_ = *codec_settings;
mode_ = SimulcastMode::kNormal;
- absl::optional<InterLayerPredMode> inter_layer_pred_mode;
+ std::optional<InterLayerPredMode> inter_layer_pred_mode;
if (codec_settings->GetScalabilityMode().has_value()) {
inter_layer_pred_mode = ScalabilityModeToInterLayerPredMode(
*codec_settings->GetScalabilityMode());
@@ -399,7 +399,7 @@
VideoEncoderFactory::CodecSupport
QualityAnalyzingVideoEncoderFactory::QueryCodecSupport(
const SdpVideoFormat& format,
- absl::optional<std::string> scalability_mode) const {
+ std::optional<std::string> scalability_mode) const {
return delegate_->QueryCodecSupport(format, scalability_mode);
}
diff --git a/test/pc/e2e/analyzer/video/quality_analyzing_video_encoder.h b/test/pc/e2e/analyzer/video/quality_analyzing_video_encoder.h
index 1fb6302..02159d7 100644
--- a/test/pc/e2e/analyzer/video/quality_analyzing_video_encoder.h
+++ b/test/pc/e2e/analyzer/video/quality_analyzing_video_encoder.h
@@ -52,7 +52,7 @@
public EncodedImageCallback {
public:
using EmulatedSFUConfigMap =
- std::map<std::string, absl::optional<EmulatedSFUConfig>>;
+ std::map<std::string, std::optional<EmulatedSFUConfig>>;
QualityAnalyzingVideoEncoder(absl::string_view peer_name,
std::unique_ptr<VideoEncoder> delegate,
@@ -138,7 +138,7 @@
const double bitrate_multiplier_;
// Contains mapping from stream label to optional spatial index.
// If we have stream label "Foo" and mapping contains
- // 1. `absl::nullopt` means all streams are required
+ // 1. `std::nullopt` means all streams are required
// 2. Concrete value means that particular simulcast/SVC stream have to be
// analyzed.
EmulatedSFUConfigMap stream_to_sfu_config_;
@@ -176,7 +176,7 @@
std::vector<SdpVideoFormat> GetSupportedFormats() const override;
VideoEncoderFactory::CodecSupport QueryCodecSupport(
const SdpVideoFormat& format,
- absl::optional<std::string> scalability_mode) const override;
+ std::optional<std::string> scalability_mode) const override;
std::unique_ptr<VideoEncoder> Create(const Environment& env,
const SdpVideoFormat& format) override;
diff --git a/test/pc/e2e/analyzer/video/single_process_encoded_image_data_injector.cc b/test/pc/e2e/analyzer/video/single_process_encoded_image_data_injector.cc
index ccd2f03..3bc40a5 100644
--- a/test/pc/e2e/analyzer/video/single_process_encoded_image_data_injector.cc
+++ b/test/pc/e2e/analyzer/video/single_process_encoded_image_data_injector.cc
@@ -100,7 +100,7 @@
}
size_t prev_frames_size = 0;
- absl::optional<uint16_t> id = absl::nullopt;
+ std::optional<uint16_t> id = std::nullopt;
bool discard = true;
std::vector<ExtractionInfo> extraction_infos;
for (size_t frame_size : frame_sizes) {
diff --git a/test/pc/e2e/analyzer/video/video_dumping_test.cc b/test/pc/e2e/analyzer/video/video_dumping_test.cc
index 5dd4021..21f1b2c 100644
--- a/test/pc/e2e/analyzer/video/video_dumping_test.cc
+++ b/test/pc/e2e/analyzer/video/video_dumping_test.cc
@@ -12,10 +12,10 @@
#include <stdio.h>
#include <memory>
+#include <optional>
#include <string>
#include <vector>
-#include "absl/types/optional.h"
#include "api/scoped_refptr.h"
#include "api/video/i420_buffer.h"
#include "api/video/video_frame.h"
diff --git a/test/pc/e2e/analyzer/video/video_quality_metrics_reporter.h b/test/pc/e2e/analyzer/video/video_quality_metrics_reporter.h
index d3d9763..cc31c48 100644
--- a/test/pc/e2e/analyzer/video/video_quality_metrics_reporter.h
+++ b/test/pc/e2e/analyzer/video/video_quality_metrics_reporter.h
@@ -64,7 +64,7 @@
test::MetricsLogger* const metrics_logger_;
std::string test_case_name_;
- absl::optional<Timestamp> start_time_;
+ std::optional<Timestamp> start_time_;
Mutex video_bwe_stats_lock_;
// Map between a peer connection label (provided by the framework) and
diff --git a/test/pc/e2e/analyzer_helper.cc b/test/pc/e2e/analyzer_helper.cc
index 76cd9a7..0987f8c 100644
--- a/test/pc/e2e/analyzer_helper.cc
+++ b/test/pc/e2e/analyzer_helper.cc
@@ -24,7 +24,7 @@
absl::string_view track_id,
absl::string_view receiver_peer,
absl::string_view stream_label,
- absl::optional<std::string> sync_group) {
+ std::optional<std::string> sync_group) {
RTC_DCHECK_RUN_ON(&signaling_sequence_checker_);
track_to_stream_map_.insert(
{std::string(track_id),
diff --git a/test/pc/e2e/analyzer_helper.h b/test/pc/e2e/analyzer_helper.h
index d0b47c4..008cb3b 100644
--- a/test/pc/e2e/analyzer_helper.h
+++ b/test/pc/e2e/analyzer_helper.h
@@ -12,10 +12,10 @@
#define TEST_PC_E2E_ANALYZER_HELPER_H_
#include <map>
+#include <optional>
#include <string>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/sequence_checker.h"
#include "api/test/track_id_stream_info_map.h"
#include "rtc_base/thread_annotations.h"
@@ -40,7 +40,7 @@
void AddTrackToStreamMapping(absl::string_view track_id,
absl::string_view receiver_peer,
absl::string_view stream_label,
- absl::optional<std::string> sync_group);
+ std::optional<std::string> sync_group);
void AddTrackToStreamMapping(std::string track_id, std::string stream_label);
void AddTrackToStreamMapping(std::string track_id,
std::string stream_label,
diff --git a/test/pc/e2e/cross_media_metrics_reporter.h b/test/pc/e2e/cross_media_metrics_reporter.h
index 2d51ebb..c031b06 100644
--- a/test/pc/e2e/cross_media_metrics_reporter.h
+++ b/test/pc/e2e/cross_media_metrics_reporter.h
@@ -12,10 +12,10 @@
#define TEST_PC_E2E_CROSS_MEDIA_METRICS_REPORTER_H_
#include <map>
+#include <optional>
#include <string>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/numerics/samples_stats_counter.h"
#include "api/test/metrics/metrics_logger.h"
#include "api/test/peerconnection_quality_test_fixture.h"
diff --git a/test/pc/e2e/media/test_video_capturer_video_track_source.h b/test/pc/e2e/media/test_video_capturer_video_track_source.h
index abcdc6c..cbfffd0 100644
--- a/test/pc/e2e/media/test_video_capturer_video_track_source.h
+++ b/test/pc/e2e/media/test_video_capturer_video_track_source.h
@@ -12,9 +12,9 @@
#define TEST_PC_E2E_MEDIA_TEST_VIDEO_CAPTURER_VIDEO_TRACK_SOURCE_H_
#include <memory>
+#include <optional>
#include <utility>
-#include "absl/types/optional.h"
#include "api/sequence_checker.h"
#include "api/test/video/test_video_track_source.h"
#include "api/video/video_frame.h"
@@ -29,7 +29,7 @@
TestVideoCapturerVideoTrackSource(
std::unique_ptr<test::TestVideoCapturer> video_capturer,
bool is_screencast,
- absl::optional<std::string> stream_label = absl::nullopt)
+ std::optional<std::string> stream_label = std::nullopt)
: TestVideoTrackSource(/*remote=*/false, std::move(stream_label)),
video_capturer_(std::move(video_capturer)),
is_screencast_(is_screencast) {
@@ -76,7 +76,7 @@
void OnOutputFormatRequest(int width,
int height,
- const absl::optional<int>& max_fps) override {
+ const std::optional<int>& max_fps) override {
video_capturer_->OnOutputFormatRequest(width, height, max_fps);
}
diff --git a/test/pc/e2e/peer_connection_quality_test.cc b/test/pc/e2e/peer_connection_quality_test.cc
index 6c58115..fac96e2 100644
--- a/test/pc/e2e/peer_connection_quality_test.cc
+++ b/test/pc/e2e/peer_connection_quality_test.cc
@@ -180,7 +180,7 @@
void PeerConnectionE2EQualityTest::ExecuteAt(
TimeDelta target_time_since_start,
std::function<void(TimeDelta)> func) {
- executor_->ScheduleActivity(target_time_since_start, absl::nullopt, func);
+ executor_->ScheduleActivity(target_time_since_start, std::nullopt, func);
}
void PeerConnectionE2EQualityTest::ExecuteEvery(
@@ -256,9 +256,9 @@
// Audio streams are intercepted in AudioDeviceModule, so if it is required to
// catch output of Alice's stream, Alice's output_dump_file_name should be
// passed to Bob's TestPeer setup as audio output file name.
- absl::optional<RemotePeerAudioConfig> alice_remote_audio_config =
+ std::optional<RemotePeerAudioConfig> alice_remote_audio_config =
RemotePeerAudioConfig::Create(bob_configurer->params()->audio_config);
- absl::optional<RemotePeerAudioConfig> bob_remote_audio_config =
+ std::optional<RemotePeerAudioConfig> bob_remote_audio_config =
RemotePeerAudioConfig::Create(alice_configurer->params()->audio_config);
// Copy Alice and Bob video configs, subscriptions and names to correctly pass
// them into lambdas.
diff --git a/test/pc/e2e/peer_connection_quality_test_test.cc b/test/pc/e2e/peer_connection_quality_test_test.cc
index a63bc29..87e0c3f 100644
--- a/test/pc/e2e/peer_connection_quality_test_test.cc
+++ b/test/pc/e2e/peer_connection_quality_test_test.cc
@@ -40,7 +40,7 @@
// Remove files and directories in a directory non-recursively.
void CleanDir(absl::string_view dir, size_t expected_output_files_count) {
- absl::optional<std::vector<std::string>> dir_content =
+ std::optional<std::vector<std::string>> dir_content =
test::ReadDirectory(dir);
if (expected_output_files_count == 0) {
ASSERT_FALSE(dir_content.has_value()) << "Empty directory is expected";
diff --git a/test/pc/e2e/peer_params_preprocessor.cc b/test/pc/e2e/peer_params_preprocessor.cc
index 3647c01..7c820a0 100644
--- a/test/pc/e2e/peer_params_preprocessor.cc
+++ b/test/pc/e2e/peer_params_preprocessor.cc
@@ -52,7 +52,7 @@
rtc::ArrayView<const absl::string_view> default_names = {})
: prefix_(prefix), default_names_(default_names) {}
- void MaybeSetName(absl::optional<std::string>& name) {
+ void MaybeSetName(std::optional<std::string>& name) {
if (name.has_value()) {
known_names_.insert(name.value());
} else {
@@ -168,11 +168,11 @@
if (!encoding_param.scalability_mode)
continue;
- absl::optional<ScalabilityMode> scalability_mode =
+ std::optional<ScalabilityMode> scalability_mode =
ScalabilityModeFromString(*encoding_param.scalability_mode);
RTC_CHECK(scalability_mode) << "Unknown scalability_mode requested";
- absl::optional<ScalableVideoController::StreamLayersConfig>
+ std::optional<ScalableVideoController::StreamLayersConfig>
stream_layers_config =
ScalabilityStructureConfig(*scalability_mode);
is_svc |= stream_layers_config->num_spatial_layers > 1;
diff --git a/test/pc/e2e/sdp/sdp_changer.h b/test/pc/e2e/sdp/sdp_changer.h
index 6f68d03..018150c 100644
--- a/test/pc/e2e/sdp/sdp_changer.h
+++ b/test/pc/e2e/sdp/sdp_changer.h
@@ -12,11 +12,11 @@
#define TEST_PC_E2E_SDP_SDP_CHANGER_H_
#include <map>
+#include <optional>
#include <string>
#include <vector>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/jsep.h"
#include "api/rtp_parameters.h"
diff --git a/test/pc/e2e/stats_based_network_quality_metrics_reporter_test.cc b/test/pc/e2e/stats_based_network_quality_metrics_reporter_test.cc
index a4ed112..1a169dc 100644
--- a/test/pc/e2e/stats_based_network_quality_metrics_reporter_test.cc
+++ b/test/pc/e2e/stats_based_network_quality_metrics_reporter_test.cc
@@ -12,11 +12,11 @@
#include <map>
#include <memory>
+#include <optional>
#include <string>
#include <vector>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/test/create_network_emulation_manager.h"
#include "api/test/create_peer_connection_quality_test_frame_generator.h"
@@ -65,14 +65,14 @@
fixture.AddPeer(std::move(peer));
}
-absl::optional<Metric> FindMeetricByName(absl::string_view name,
- rtc::ArrayView<const Metric> metrics) {
+std::optional<Metric> FindMeetricByName(absl::string_view name,
+ rtc::ArrayView<const Metric> metrics) {
for (const Metric& metric : metrics) {
if (metric.name == name) {
return metric;
}
}
- return absl::nullopt;
+ return std::nullopt;
}
TEST(StatsBasedNetworkQualityMetricsReporterTest, DebugStatsAreCollected) {
@@ -128,20 +128,20 @@
fixture.Run(RunParams(TimeDelta::Seconds(4)));
std::vector<Metric> metrics = metrics_logger.GetCollectedMetrics();
- absl::optional<Metric> uplink_packet_transport_time =
+ std::optional<Metric> uplink_packet_transport_time =
FindMeetricByName("uplink_packet_transport_time", metrics);
ASSERT_TRUE(uplink_packet_transport_time.has_value());
ASSERT_FALSE(uplink_packet_transport_time->time_series.samples.empty());
- absl::optional<Metric> uplink_size_to_packet_transport_time =
+ std::optional<Metric> uplink_size_to_packet_transport_time =
FindMeetricByName("uplink_size_to_packet_transport_time", metrics);
ASSERT_TRUE(uplink_size_to_packet_transport_time.has_value());
ASSERT_FALSE(
uplink_size_to_packet_transport_time->time_series.samples.empty());
- absl::optional<Metric> downlink_packet_transport_time =
+ std::optional<Metric> downlink_packet_transport_time =
FindMeetricByName("downlink_packet_transport_time", metrics);
ASSERT_TRUE(downlink_packet_transport_time.has_value());
ASSERT_FALSE(downlink_packet_transport_time->time_series.samples.empty());
- absl::optional<Metric> downlink_size_to_packet_transport_time =
+ std::optional<Metric> downlink_size_to_packet_transport_time =
FindMeetricByName("downlink_size_to_packet_transport_time", metrics);
ASSERT_TRUE(downlink_size_to_packet_transport_time.has_value());
ASSERT_FALSE(
diff --git a/test/pc/e2e/test_activities_executor.cc b/test/pc/e2e/test_activities_executor.cc
index 7bcf7dd..cad4f63 100644
--- a/test/pc/e2e/test_activities_executor.cc
+++ b/test/pc/e2e/test_activities_executor.cc
@@ -48,7 +48,7 @@
void TestActivitiesExecutor::ScheduleActivity(
TimeDelta initial_delay_since_start,
- absl::optional<TimeDelta> interval,
+ std::optional<TimeDelta> interval,
std::function<void(TimeDelta)> func) {
RTC_CHECK(initial_delay_since_start.IsFinite() &&
initial_delay_since_start >= TimeDelta::Zero());
@@ -112,7 +112,7 @@
TestActivitiesExecutor::ScheduledActivity::ScheduledActivity(
TimeDelta initial_delay_since_start,
- absl::optional<TimeDelta> interval,
+ std::optional<TimeDelta> interval,
std::function<void(TimeDelta)> func)
: initial_delay_since_start(initial_delay_since_start),
interval(interval),
diff --git a/test/pc/e2e/test_activities_executor.h b/test/pc/e2e/test_activities_executor.h
index 2469ac7..1d9d7ca 100644
--- a/test/pc/e2e/test_activities_executor.h
+++ b/test/pc/e2e/test_activities_executor.h
@@ -11,10 +11,10 @@
#ifndef TEST_PC_E2E_TEST_ACTIVITIES_EXECUTOR_H_
#define TEST_PC_E2E_TEST_ACTIVITIES_EXECUTOR_H_
+#include <optional>
#include <queue>
#include <vector>
-#include "absl/types/optional.h"
#include "api/task_queue/task_queue_base.h"
#include "api/units/time_delta.h"
#include "api/units/timestamp.h"
@@ -43,17 +43,17 @@
// If test is started, then it will be executed immediately according to its
// schedule.
void ScheduleActivity(TimeDelta initial_delay_since_start,
- absl::optional<TimeDelta> interval,
+ std::optional<TimeDelta> interval,
std::function<void(TimeDelta)> func);
private:
struct ScheduledActivity {
ScheduledActivity(TimeDelta initial_delay_since_start,
- absl::optional<TimeDelta> interval,
+ std::optional<TimeDelta> interval,
std::function<void(TimeDelta)> func);
TimeDelta initial_delay_since_start;
- absl::optional<TimeDelta> interval;
+ std::optional<TimeDelta> interval;
std::function<void(TimeDelta)> func;
};
diff --git a/test/pc/e2e/test_peer.h b/test/pc/e2e/test_peer.h
index 1ce2acb..0956e05 100644
--- a/test/pc/e2e/test_peer.h
+++ b/test/pc/e2e/test_peer.h
@@ -12,11 +12,11 @@
#define TEST_PC_E2E_TEST_PEER_H_
#include <memory>
+#include <optional>
#include <vector>
#include "absl/memory/memory.h"
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/function_view.h"
#include "api/scoped_refptr.h"
#include "api/sequence_checker.h"
@@ -111,7 +111,7 @@
rtc::scoped_refptr<DataChannelInterface> CreateDataChannel(
const std::string& label,
- const absl::optional<DataChannelInit>& config = absl::nullopt) {
+ const std::optional<DataChannelInit>& config = std::nullopt) {
RTC_CHECK(wrapper_) << "TestPeer is already closed";
return wrapper_->CreateDataChannel(label, config);
}
diff --git a/test/pc/e2e/test_peer_factory.cc b/test/pc/e2e/test_peer_factory.cc
index a184c5d..f0ffe9f 100644
--- a/test/pc/e2e/test_peer_factory.cc
+++ b/test/pc/e2e/test_peer_factory.cc
@@ -60,7 +60,7 @@
// Returns mapping from stream label to optional spatial index.
// If we have stream label "Foo" and mapping contains
-// 1. `absl::nullopt` means all simulcast/SVC streams are required
+// 1. `std::nullopt` means all simulcast/SVC streams are required
// 2. Concrete value means that particular simulcast/SVC stream have to be
// analyzed.
EmulatedSFUConfigMap CalculateRequiredSpatialIndexPerStream(
@@ -80,7 +80,7 @@
}
std::unique_ptr<TestAudioDeviceModule::Renderer> CreateAudioRenderer(
- const absl::optional<RemotePeerAudioConfig>& config) {
+ const std::optional<RemotePeerAudioConfig>& config) {
if (!config) {
// Return default renderer because we always require some renderer.
return TestAudioDeviceModule::CreateDiscardRenderer(
@@ -95,7 +95,7 @@
}
std::unique_ptr<TestAudioDeviceModule::Capturer> CreateAudioCapturer(
- const absl::optional<AudioConfig>& audio_config) {
+ const std::optional<AudioConfig>& audio_config) {
if (!audio_config) {
// If we have no audio config we still need to provide some audio device.
// In such case use generated capturer. Despite of we provided audio here,
@@ -113,9 +113,9 @@
}
rtc::scoped_refptr<AudioDeviceModule> CreateAudioDeviceModule(
- absl::optional<AudioConfig> audio_config,
- absl::optional<RemotePeerAudioConfig> remote_audio_config,
- absl::optional<EchoEmulationConfig> echo_emulation_config,
+ std::optional<AudioConfig> audio_config,
+ std::optional<RemotePeerAudioConfig> remote_audio_config,
+ std::optional<EchoEmulationConfig> echo_emulation_config,
TaskQueueFactory* task_queue_factory) {
std::unique_ptr<TestAudioDeviceModule::Renderer> renderer =
CreateAudioRenderer(remote_audio_config);
@@ -259,10 +259,10 @@
} // namespace
-absl::optional<RemotePeerAudioConfig> RemotePeerAudioConfig::Create(
- absl::optional<AudioConfig> config) {
+std::optional<RemotePeerAudioConfig> RemotePeerAudioConfig::Create(
+ std::optional<AudioConfig> config) {
if (!config) {
- return absl::nullopt;
+ return std::nullopt;
}
return RemotePeerAudioConfig(config.value());
}
@@ -270,8 +270,8 @@
std::unique_ptr<TestPeer> TestPeerFactory::CreateTestPeer(
std::unique_ptr<PeerConfigurer> configurer,
std::unique_ptr<MockPeerConnectionObserver> observer,
- absl::optional<RemotePeerAudioConfig> remote_audio_config,
- absl::optional<EchoEmulationConfig> echo_emulation_config) {
+ std::optional<RemotePeerAudioConfig> remote_audio_config,
+ std::optional<EchoEmulationConfig> echo_emulation_config) {
std::unique_ptr<InjectableComponents> components =
configurer->ReleaseComponents();
std::unique_ptr<Params> params = configurer->ReleaseParams();
diff --git a/test/pc/e2e/test_peer_factory.h b/test/pc/e2e/test_peer_factory.h
index cc61b04..34462c1 100644
--- a/test/pc/e2e/test_peer_factory.h
+++ b/test/pc/e2e/test_peer_factory.h
@@ -35,11 +35,11 @@
: sampling_frequency_in_hz(config.sampling_frequency_in_hz),
output_file_name(config.output_dump_file_name) {}
- static absl::optional<RemotePeerAudioConfig> Create(
- absl::optional<AudioConfig> config);
+ static std::optional<RemotePeerAudioConfig> Create(
+ std::optional<AudioConfig> config);
int sampling_frequency_in_hz;
- absl::optional<std::string> output_file_name;
+ std::optional<std::string> output_file_name;
};
class TestPeerFactory {
@@ -68,8 +68,8 @@
std::unique_ptr<TestPeer> CreateTestPeer(
std::unique_ptr<PeerConfigurer> configurer,
std::unique_ptr<MockPeerConnectionObserver> observer,
- absl::optional<RemotePeerAudioConfig> remote_audio_config,
- absl::optional<EchoEmulationConfig> echo_emulation_config);
+ std::optional<RemotePeerAudioConfig> remote_audio_config,
+ std::optional<EchoEmulationConfig> echo_emulation_config);
private:
rtc::Thread* signaling_thread_;
diff --git a/test/pc/sctp/fake_sctp_transport.h b/test/pc/sctp/fake_sctp_transport.h
index 8aacd3b..b5d0866 100644
--- a/test/pc/sctp/fake_sctp_transport.h
+++ b/test/pc/sctp/fake_sctp_transport.h
@@ -45,11 +45,11 @@
void set_debug_name_for_testing(const char* debug_name) override {}
int max_message_size() const override { return max_message_size_; }
- absl::optional<int> max_outbound_streams() const override {
- return absl::nullopt;
+ std::optional<int> max_outbound_streams() const override {
+ return std::nullopt;
}
- absl::optional<int> max_inbound_streams() const override {
- return absl::nullopt;
+ std::optional<int> max_inbound_streams() const override {
+ return std::nullopt;
}
size_t buffered_amount(int sid) const override { return 0; }
size_t buffered_amount_low_threshold(int sid) const override { return 0; }
@@ -64,8 +64,8 @@
}
private:
- absl::optional<int> local_port_;
- absl::optional<int> remote_port_;
+ std::optional<int> local_port_;
+ std::optional<int> remote_port_;
int max_message_size_;
};
diff --git a/test/peer_scenario/peer_scenario_client.h b/test/peer_scenario/peer_scenario_client.h
index cb025e9..729703a 100644
--- a/test/peer_scenario/peer_scenario_client.h
+++ b/test/peer_scenario/peer_scenario_client.h
@@ -79,7 +79,7 @@
struct PulsedNoise {
double amplitude = 0.1;
};
- absl::optional<PulsedNoise> pulsed_noise = PulsedNoise();
+ std::optional<PulsedNoise> pulsed_noise = PulsedNoise();
} audio;
struct Video {
bool use_fake_codecs = false;
diff --git a/test/peer_scenario/tests/unsignaled_stream_test.cc b/test/peer_scenario/tests/unsignaled_stream_test.cc
index d7565c9..befd116 100644
--- a/test/peer_scenario/tests/unsignaled_stream_test.cc
+++ b/test/peer_scenario/tests/unsignaled_stream_test.cc
@@ -143,7 +143,7 @@
uint32_t first_ssrc = 0;
uint32_t second_ssrc = 0;
- absl::optional<int> mid_header_extension_id = absl::nullopt;
+ std::optional<int> mid_header_extension_id = std::nullopt;
signaling.NegotiateSdp(
/* munge_sdp = */
diff --git a/test/rtp_file_writer.cc b/test/rtp_file_writer.cc
index 22f664a..f992756 100644
--- a/test/rtp_file_writer.cc
+++ b/test/rtp_file_writer.cc
@@ -13,9 +13,9 @@
#include <stdint.h>
#include <stdio.h>
+#include <optional>
#include <string>
-#include "absl/types/optional.h"
#include "rtc_base/checks.h"
namespace webrtc {
@@ -92,7 +92,7 @@
}
FILE* file_;
- absl::optional<uint32_t> first_packet_time_;
+ std::optional<uint32_t> first_packet_time_;
};
RtpFileWriter* RtpFileWriter::Create(FileFormat format,
diff --git a/test/scenario/BUILD.gn b/test/scenario/BUILD.gn
index 672cd91..8f9234a 100644
--- a/test/scenario/BUILD.gn
+++ b/test/scenario/BUILD.gn
@@ -157,7 +157,6 @@
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/strings:string_view",
- "//third_party/abseil-cpp/absl/types:optional",
]
if (is_android) {
deps += [ "../../modules/video_coding:android_codec_factory_helper" ]
diff --git a/test/scenario/audio_stream.cc b/test/scenario/audio_stream.cc
index 232f738..64b6406 100644
--- a/test/scenario/audio_stream.cc
+++ b/test/scenario/audio_stream.cc
@@ -29,7 +29,7 @@
kAbsSendTimeExtensionId
};
-absl::optional<std::string> CreateAdaptationString(
+std::optional<std::string> CreateAdaptationString(
AudioStreamConfig::NetworkAdaptation config) {
#if WEBRTC_ENABLE_PROTOBUF
@@ -61,7 +61,7 @@
RTC_LOG(LS_ERROR) << "audio_network_adaptation is enabled"
" but WEBRTC_ENABLE_PROTOBUF is false.\n"
"Ignoring settings.";
- return absl::nullopt;
+ return std::nullopt;
#endif // WEBRTC_ENABLE_PROTOBUF
}
} // namespace
diff --git a/test/scenario/scenario_config.h b/test/scenario/scenario_config.h
index 50845cd..b47a220 100644
--- a/test/scenario/scenario_config.h
+++ b/test/scenario/scenario_config.h
@@ -12,9 +12,9 @@
#include <stddef.h>
+#include <optional>
#include <string>
-#include "absl/types/optional.h"
#include "api/fec_controller.h"
#include "api/rtp_parameters.h"
#include "api/test/frame_generator_interface.h"
@@ -90,8 +90,8 @@
struct Images {
struct Crop {
TimeDelta scroll_duration = TimeDelta::Seconds(0);
- absl::optional<int> width;
- absl::optional<int> height;
+ std::optional<int> width;
+ std::optional<int> height;
} crop;
int width = 1850;
int height = 1110;
@@ -132,11 +132,11 @@
using Codec = VideoCodecType;
Codec codec = Codec::kVideoCodecGeneric;
- absl::optional<DataRate> max_data_rate;
- absl::optional<DataRate> min_data_rate;
- absl::optional<int> max_framerate;
+ std::optional<DataRate> max_data_rate;
+ std::optional<DataRate> min_data_rate;
+ std::optional<int> max_framerate;
// Counted in frame count.
- absl::optional<int> key_frame_interval = 3000;
+ std::optional<int> key_frame_interval = 3000;
bool frame_dropping = true;
struct SingleLayer {
bool denoising = true;
@@ -199,8 +199,8 @@
bool enable_dtx = false;
DataRate fixed_rate = DataRate::KilobitsPerSec(32);
// Overrides fixed rate.
- absl::optional<DataRate> min_rate;
- absl::optional<DataRate> max_rate;
+ std::optional<DataRate> min_rate;
+ std::optional<DataRate> max_rate;
TimeDelta initial_frame_length = TimeDelta::Millis(20);
} encoder;
struct Stream {
@@ -221,7 +221,7 @@
TimeDelta delay = TimeDelta::Zero();
TimeDelta delay_std_dev = TimeDelta::Zero();
double loss_rate = 0;
- absl::optional<int> packet_queue_length_limit;
+ std::optional<int> packet_queue_length_limit;
DataSize packet_overhead = DataSize::Zero();
};
} // namespace test
diff --git a/test/scenario/stats_collection.h b/test/scenario/stats_collection.h
index 1f5d8da..fcf8f71 100644
--- a/test/scenario/stats_collection.h
+++ b/test/scenario/stats_collection.h
@@ -12,8 +12,8 @@
#include <map>
#include <memory>
+#include <optional>
-#include "absl/types/optional.h"
#include "call/call.h"
#include "rtc_base/thread.h"
#include "test/logging/log_writer.h"
@@ -59,7 +59,7 @@
const VideoQualityAnalyzerConfig config_;
std::map<int, VideoLayerAnalyzer> layer_analyzers_;
const std::unique_ptr<RtcEventLogOutput> writer_;
- absl::optional<VideoQualityStats> cached_;
+ std::optional<VideoQualityStats> cached_;
};
class CallStatsCollector {
diff --git a/test/scenario/video_stream.cc b/test/scenario/video_stream.cc
index eb403f9..db4918b 100644
--- a/test/scenario/video_stream.cc
+++ b/test/scenario/video_stream.cc
@@ -263,7 +263,7 @@
case Capture::kGenerator:
return CreateSquareFrameGenerator(
source.generator.width, source.generator.height,
- source.generator.pixel_format, /*num_squares*/ absl::nullopt);
+ source.generator.pixel_format, /*num_squares*/ std::nullopt);
case Capture::kVideoFile:
RTC_CHECK(source.video_file.width && source.video_file.height);
return CreateFromYuvFileFrameGenerator(
diff --git a/test/test_main_lib.cc b/test/test_main_lib.cc
index 486913b..08f9b4e 100644
--- a/test/test_main_lib.cc
+++ b/test/test_main_lib.cc
@@ -15,6 +15,7 @@
#include <fstream>
#include <ios>
#include <memory>
+#include <optional>
#include <string>
#include <vector>
@@ -22,7 +23,6 @@
#include "absl/memory/memory.h"
#include "absl/strings/match.h"
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/test/metrics/chrome_perf_dashboard_metrics_exporter.h"
#include "api/test/metrics/global_metrics_logger_and_exporter.h"
#include "api/test/metrics/metrics_exporter.h"
@@ -167,11 +167,11 @@
StartTracingCapture(trace_event_path);
}
- absl::optional<std::vector<std::string>> metrics_to_plot =
+ std::optional<std::vector<std::string>> metrics_to_plot =
absl::GetFlag(FLAGS_plot);
if (metrics_to_plot->empty()) {
- metrics_to_plot = absl::nullopt;
+ metrics_to_plot = std::nullopt;
} else {
if (metrics_to_plot->size() == 1 &&
(*metrics_to_plot)[0] == kPlotAllMetrics) {
diff --git a/test/test_video_capturer.cc b/test/test_video_capturer.cc
index 385af12..de4e175 100644
--- a/test/test_video_capturer.cc
+++ b/test/test_video_capturer.cc
@@ -25,10 +25,10 @@
void TestVideoCapturer::OnOutputFormatRequest(
int width,
int height,
- const absl::optional<int>& max_fps) {
- absl::optional<std::pair<int, int>> target_aspect_ratio =
+ const std::optional<int>& max_fps) {
+ std::optional<std::pair<int, int>> target_aspect_ratio =
std::make_pair(width, height);
- absl::optional<int> max_pixel_count = width * height;
+ std::optional<int> max_pixel_count = width * height;
video_adapter_.OnOutputFormatRequest(target_aspect_ratio, max_pixel_count,
max_fps);
}
diff --git a/test/test_video_capturer.h b/test/test_video_capturer.h
index 49660d8..4ac17db 100644
--- a/test/test_video_capturer.h
+++ b/test/test_video_capturer.h
@@ -47,7 +47,7 @@
}
void OnOutputFormatRequest(int width,
int height,
- const absl::optional<int>& max_fps);
+ const std::optional<int>& max_fps);
// Starts or resumes video capturing. Can be called multiple times during
// lifetime of this object.
diff --git a/test/testsupport/copy_to_file_audio_capturer.h b/test/testsupport/copy_to_file_audio_capturer.h
index a410bee..9170cc1 100644
--- a/test/testsupport/copy_to_file_audio_capturer.h
+++ b/test/testsupport/copy_to_file_audio_capturer.h
@@ -12,9 +12,9 @@
#define TEST_TESTSUPPORT_COPY_TO_FILE_AUDIO_CAPTURER_H_
#include <memory>
+#include <optional>
#include <string>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "common_audio/wav_file.h"
#include "modules/audio_device/include/test_audio_device.h"
diff --git a/test/testsupport/file_utils.cc b/test/testsupport/file_utils.cc
index f796361..9451c20 100644
--- a/test/testsupport/file_utils.cc
+++ b/test/testsupport/file_utils.cc
@@ -138,9 +138,9 @@
return filename;
}
-absl::optional<std::vector<std::string>> ReadDirectory(absl::string_view path) {
+std::optional<std::vector<std::string>> ReadDirectory(absl::string_view path) {
if (path.length() == 0)
- return absl::optional<std::vector<std::string>>();
+ return std::optional<std::vector<std::string>>();
std::string path_str(path);
@@ -153,7 +153,7 @@
WIN32_FIND_DATAW data;
HANDLE handle = ::FindFirstFileW(rtc::ToUtf16(path_str + '*').c_str(), &data);
if (handle == INVALID_HANDLE_VALUE)
- return absl::optional<std::vector<std::string>>();
+ return std::optional<std::vector<std::string>>();
// Populate output.
std::vector<std::string> found_entries;
@@ -174,7 +174,7 @@
// Init.
DIR* dir = ::opendir(path_str.c_str());
if (dir == nullptr)
- return absl::optional<std::vector<std::string>>();
+ return std::optional<std::vector<std::string>>();
// Populate output.
std::vector<std::string> found_entries;
@@ -188,7 +188,7 @@
closedir(dir);
#endif
- return absl::optional<std::vector<std::string>>(std::move(found_entries));
+ return std::optional<std::vector<std::string>>(std::move(found_entries));
}
bool CreateDir(absl::string_view directory_name) {
diff --git a/test/testsupport/file_utils.h b/test/testsupport/file_utils.h
index 120c6cb..22d3d11 100644
--- a/test/testsupport/file_utils.h
+++ b/test/testsupport/file_utils.h
@@ -13,12 +13,12 @@
#ifndef TEST_TESTSUPPORT_FILE_UTILS_H_
#define TEST_TESTSUPPORT_FILE_UTILS_H_
+#include <optional>
#include <string>
#include <vector>
#include "absl/base/attributes.h"
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
namespace webrtc {
namespace test {
@@ -80,7 +80,7 @@
// of strings with one element for each found file or directory. Each element is
// a path created by prepending `dir` to the file/directory name. "." and ".."
// are never added in the returned vector.
-absl::optional<std::vector<std::string>> ReadDirectory(absl::string_view path);
+std::optional<std::vector<std::string>> ReadDirectory(absl::string_view path);
// Creates a directory if it not already exists.
// Returns true if successful. Will print an error message to stderr and return
diff --git a/test/testsupport/file_utils_override.cc b/test/testsupport/file_utils_override.cc
index 0140e6c..fe8cdd6 100644
--- a/test/testsupport/file_utils_override.cc
+++ b/test/testsupport/file_utils_override.cc
@@ -40,8 +40,9 @@
#include "test/testsupport/mac_file_utils.h"
#endif
+#include <optional>
+
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "rtc_base/arraysize.h"
#include "rtc_base/checks.h"
#include "rtc_base/string_utils.h"
@@ -80,7 +81,7 @@
// Finds the WebRTC src dir.
// The returned path always ends with a path separator.
-absl::optional<std::string> ProjectRootPath() {
+std::optional<std::string> ProjectRootPath() {
#if defined(WEBRTC_ANDROID)
return std::string(kAndroidChromiumTestsRoot);
#elif defined WEBRTC_IOS
@@ -102,7 +103,7 @@
ssize_t count = ::readlink("/proc/self/exe", buf, arraysize(buf));
if (count <= 0) {
RTC_DCHECK_NOTREACHED() << "Unable to resolve /proc/self/exe.";
- return absl::nullopt;
+ return std::nullopt;
}
// On POSIX, tests execute in out/Whatever, so src is two levels up.
std::string exe_dir = DirName(absl::string_view(buf, count));
@@ -112,7 +113,7 @@
wchar_t buf[MAX_PATH];
buf[0] = 0;
if (GetModuleFileNameW(NULL, buf, MAX_PATH) == 0)
- return absl::nullopt;
+ return std::nullopt;
std::string exe_path = rtc::ToUtf8(std::wstring(buf));
std::string exe_dir = DirName(exe_path);
@@ -128,7 +129,7 @@
#elif defined(WEBRTC_FUCHSIA)
return std::string(kFuchsiaTempWritableDir);
#else
- absl::optional<std::string> path_opt = ProjectRootPath();
+ std::optional<std::string> path_opt = ProjectRootPath();
RTC_DCHECK(path_opt);
std::string path = *path_opt + "out";
if (!CreateDir(path)) {
@@ -156,7 +157,7 @@
#if defined(WEBRTC_IOS)
return IOSResourcePath(name, extension);
#else
- absl::optional<std::string> path_opt = ProjectRootPath();
+ std::optional<std::string> path_opt = ProjectRootPath();
RTC_DCHECK(path_opt);
rtc::StringBuilder os(*path_opt);
os << kResourcesDirName << kPathDelimiter << name << "." << extension;
diff --git a/test/testsupport/file_utils_unittest.cc b/test/testsupport/file_utils_unittest.cc
index 005e12c..fff6563 100644
--- a/test/testsupport/file_utils_unittest.cc
+++ b/test/testsupport/file_utils_unittest.cc
@@ -14,10 +14,10 @@
#include <algorithm>
#include <fstream>
+#include <optional>
#include <string>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "rtc_base/checks.h"
#include "rtc_base/crypto_random.h"
#include "test/gmock.h"
@@ -45,7 +45,7 @@
void CleanDir(absl::string_view dir, size_t* num_deleted_entries) {
RTC_DCHECK(num_deleted_entries);
*num_deleted_entries = 0;
- absl::optional<std::vector<std::string>> dir_content = ReadDirectory(dir);
+ std::optional<std::vector<std::string>> dir_content = ReadDirectory(dir);
EXPECT_TRUE(dir_content);
for (const auto& entry : *dir_content) {
if (DirExists(entry)) {
@@ -271,7 +271,7 @@
EXPECT_TRUE(DirExists(temp_subdir));
// Checks.
- absl::optional<std::vector<std::string>> dir_content =
+ std::optional<std::vector<std::string>> dir_content =
ReadDirectory(temp_directory);
EXPECT_TRUE(dir_content);
EXPECT_EQ(2u, dir_content->size());
diff --git a/test/testsupport/fixed_fps_video_frame_writer_adapter.cc b/test/testsupport/fixed_fps_video_frame_writer_adapter.cc
index 531dade..4b90d21 100644
--- a/test/testsupport/fixed_fps_video_frame_writer_adapter.cc
+++ b/test/testsupport/fixed_fps_video_frame_writer_adapter.cc
@@ -11,9 +11,9 @@
#include "test/testsupport/fixed_fps_video_frame_writer_adapter.h"
#include <cmath>
+#include <optional>
#include <utility>
-#include "absl/types/optional.h"
#include "api/units/time_delta.h"
#include "api/video/video_sink_interface.h"
#include "rtc_base/checks.h"
diff --git a/test/testsupport/fixed_fps_video_frame_writer_adapter.h b/test/testsupport/fixed_fps_video_frame_writer_adapter.h
index d4d95e9..6d966bf 100644
--- a/test/testsupport/fixed_fps_video_frame_writer_adapter.h
+++ b/test/testsupport/fixed_fps_video_frame_writer_adapter.h
@@ -12,8 +12,8 @@
#define TEST_TESTSUPPORT_FIXED_FPS_VIDEO_FRAME_WRITER_ADAPTER_H_
#include <memory>
+#include <optional>
-#include "absl/types/optional.h"
#include "api/test/video/video_frame_writer.h"
#include "api/video/video_sink_interface.h"
#include "system_wrappers/include/clock.h"
@@ -78,7 +78,7 @@
// Expected time slot for the last frame.
Timestamp last_frame_time_ = Timestamp::MinusInfinity();
- absl::optional<VideoFrame> last_frame_ = absl::nullopt;
+ std::optional<VideoFrame> last_frame_ = std::nullopt;
};
} // namespace test
diff --git a/test/testsupport/frame_reader.h b/test/testsupport/frame_reader.h
index 7856476..29f4773 100644
--- a/test/testsupport/frame_reader.h
+++ b/test/testsupport/frame_reader.h
@@ -13,9 +13,9 @@
#include <stdio.h>
+#include <optional>
#include <string>
-#include "absl/types/optional.h"
#include "api/scoped_refptr.h"
#include "api/video/resolution.h"
@@ -104,7 +104,7 @@
int Skip(Ratio framerate_scale);
private:
- absl::optional<int> ticks_;
+ std::optional<int> ticks_;
};
const std::string filepath_;
diff --git a/test/testsupport/ivf_video_frame_generator.cc b/test/testsupport/ivf_video_frame_generator.cc
index 42b9c19..3410205 100644
--- a/test/testsupport/ivf_video_frame_generator.cc
+++ b/test/testsupport/ivf_video_frame_generator.cc
@@ -83,7 +83,7 @@
video_decoder_.reset();
{
MutexLock frame_lock(&frame_decode_lock_);
- next_frame_ = absl::nullopt;
+ next_frame_ = std::nullopt;
// Set event in case another thread is waiting on it.
next_frame_decoded_.Set();
}
@@ -96,7 +96,7 @@
if (!file_reader_->HasMoreFrames()) {
file_reader_->Reset();
}
- absl::optional<EncodedImage> image = file_reader_->NextFrame();
+ std::optional<EncodedImage> image = file_reader_->NextFrame();
RTC_CHECK(image);
// Last parameter is undocumented and there is no usage of it found.
RTC_CHECK_EQ(WEBRTC_VIDEO_CODEC_OK,
@@ -127,7 +127,7 @@
if (!file_reader_->HasMoreFrames()) {
file_reader_->Reset();
}
- absl::optional<EncodedImage> image = file_reader_->NextFrame();
+ std::optional<EncodedImage> image = file_reader_->NextFrame();
RTC_CHECK(image);
// Last parameter is undocumented and there is no usage of it found.
// Frame has to be decoded in case it is a key frame.
@@ -159,8 +159,8 @@
}
void IvfVideoFrameGenerator::DecodedCallback::Decoded(
VideoFrame& decoded_image,
- absl::optional<int32_t> decode_time_ms,
- absl::optional<uint8_t> qp) {
+ std::optional<int32_t> decode_time_ms,
+ std::optional<uint8_t> qp) {
reader_->OnFrameDecoded(decoded_image);
}
diff --git a/test/testsupport/ivf_video_frame_generator.h b/test/testsupport/ivf_video_frame_generator.h
index 41cf2d9..b91037e 100644
--- a/test/testsupport/ivf_video_frame_generator.h
+++ b/test/testsupport/ivf_video_frame_generator.h
@@ -12,10 +12,10 @@
#define TEST_TESTSUPPORT_IVF_VIDEO_FRAME_GENERATOR_H_
#include <memory>
+#include <optional>
#include <string>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/environment/environment.h"
#include "api/sequence_checker.h"
#include "api/test/frame_generator_interface.h"
@@ -40,7 +40,7 @@
void ChangeResolution(size_t width, size_t height) override;
Resolution GetResolution() const override;
- absl::optional<int> fps() const override { return absl::nullopt; }
+ std::optional<int> fps() const override { return std::nullopt; }
private:
class DecodedCallback : public DecodedImageCallback {
@@ -51,8 +51,8 @@
int32_t Decoded(VideoFrame& decoded_image) override;
int32_t Decoded(VideoFrame& decoded_image, int64_t decode_time_ms) override;
void Decoded(VideoFrame& decoded_image,
- absl::optional<int32_t> decode_time_ms,
- absl::optional<uint8_t> qp) override;
+ std::optional<int32_t> decode_time_ms,
+ std::optional<uint8_t> qp) override;
private:
IvfVideoFrameGenerator* const reader_;
@@ -82,7 +82,7 @@
Mutex frame_decode_lock_;
rtc::Event next_frame_decoded_;
- absl::optional<VideoFrame> next_frame_ RTC_GUARDED_BY(frame_decode_lock_);
+ std::optional<VideoFrame> next_frame_ RTC_GUARDED_BY(frame_decode_lock_);
};
} // namespace test
diff --git a/test/testsupport/ivf_video_frame_generator_unittest.cc b/test/testsupport/ivf_video_frame_generator_unittest.cc
index f550890..0c83a93 100644
--- a/test/testsupport/ivf_video_frame_generator_unittest.cc
+++ b/test/testsupport/ivf_video_frame_generator_unittest.cc
@@ -11,9 +11,9 @@
#include "test/testsupport/ivf_video_frame_generator.h"
#include <memory>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/environment/environment.h"
#include "api/environment/environment_factory.h"
#include "api/test/create_frame_generator.h"
@@ -118,7 +118,7 @@
std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
test::CreateSquareFrameGenerator(
kWidth, kHeight, test::FrameGeneratorInterface::OutputType::kI420,
- absl::nullopt);
+ std::nullopt);
VideoCodec codec_settings;
webrtc::test::CodecSettings(video_codec_type, &codec_settings);
@@ -170,7 +170,7 @@
TEST_F(IvfVideoFrameGeneratorTest, DoesNotKnowFps) {
CreateTestVideoFile(VideoCodecType::kVideoCodecVP8, CreateVp8Encoder(env_));
IvfVideoFrameGenerator generator(env_, file_name_);
- EXPECT_EQ(generator.fps(), absl::nullopt);
+ EXPECT_EQ(generator.fps(), std::nullopt);
}
TEST_F(IvfVideoFrameGeneratorTest, Vp8) {
diff --git a/test/testsupport/y4m_frame_generator.h b/test/testsupport/y4m_frame_generator.h
index 16d8485..dfffb4a 100644
--- a/test/testsupport/y4m_frame_generator.h
+++ b/test/testsupport/y4m_frame_generator.h
@@ -13,10 +13,10 @@
#include <cstddef>
#include <memory>
+#include <optional>
#include <string>
#include "absl/strings/string_view.h"
-#include "absl/types/optional.h"
#include "api/test/frame_generator_interface.h"
#include "rtc_base/checks.h"
#include "test/testsupport/frame_reader.h"
@@ -51,7 +51,7 @@
Resolution GetResolution() const override;
- absl::optional<int> fps() const override { return fps_; }
+ std::optional<int> fps() const override { return fps_; }
private:
YuvFrameReaderImpl::RepeatMode ToYuvFrameReaderRepeatMode(
diff --git a/test/video_codec_tester.cc b/test/video_codec_tester.cc
index a51a532..2782a66 100644
--- a/test/video_codec_tester.cc
+++ b/test/video_codec_tester.cc
@@ -162,7 +162,7 @@
// scaling. This value increases by the source frame duration each time a
// frame is read from the source, and decreases by the output frame duration
// each time an output frame is delivered.
- absl::optional<TimeDelta> time_delta_;
+ std::optional<TimeDelta> time_delta_;
};
// Pacer calculates delay necessary to keep frame encode or decode call spaced
@@ -203,8 +203,8 @@
}
PacingSettings settings_;
- absl::optional<Timestamp> prev_timestamp_;
- absl::optional<Timestamp> prev_scheduled_;
+ std::optional<Timestamp> prev_timestamp_;
+ std::optional<Timestamp> prev_scheduled_;
TimeDelta delay_;
};
@@ -361,7 +361,7 @@
}
private:
- absl::optional<VideoCodecStats::Frame> prev_frame_;
+ std::optional<VideoCodecStats::Frame> prev_frame_;
double level_bits_;
};
@@ -457,7 +457,7 @@
void FinishDecode(const VideoFrame& decoded_frame,
int spatial_idx,
- absl::optional<VideoFrame> ref_frame = absl::nullopt) {
+ std::optional<VideoFrame> ref_frame = std::nullopt) {
int64_t decode_finished_us = rtc::TimeMicros();
task_queue_.PostTask([this, timestamp_rtp = decoded_frame.rtp_timestamp(),
spatial_idx, width = decoded_frame.width(),
@@ -759,7 +759,7 @@
}
DataRate GetTargetBitrate(const EncodingSettings& encoding_settings,
- absl::optional<LayerId> layer_id) const {
+ std::optional<LayerId> layer_id) const {
int base_spatial_idx;
if (layer_id.has_value()) {
bool is_svc =
@@ -790,7 +790,7 @@
}
Frequency GetTargetFramerate(const EncodingSettings& encoding_settings,
- absl::optional<LayerId> layer_id) const {
+ std::optional<LayerId> layer_id) const {
if (layer_id.has_value()) {
auto layer_settings = encoding_settings.layers_settings.find(
{.spatial_idx = layer_id->spatial_idx,
@@ -857,7 +857,7 @@
}
void Decode(const EncodedImage& encoded_frame,
- absl::optional<VideoFrame> ref_frame = absl::nullopt) {
+ std::optional<VideoFrame> ref_frame = std::nullopt) {
int spatial_idx = encoded_frame.SpatialIndex().value_or(
encoded_frame.SimulcastIndex().value_or(0));
{
@@ -900,7 +900,7 @@
private:
int Decoded(VideoFrame& decoded_frame) override {
int spatial_idx;
- absl::optional<VideoFrame> ref_frame;
+ std::optional<VideoFrame> ref_frame;
{
MutexLock lock(&mutex_);
spatial_idx = *spatial_idx_;
@@ -930,8 +930,8 @@
LimitedTaskQueue task_queue_;
std::unique_ptr<TesterIvfWriter> ivf_writer_;
std::unique_ptr<TesterY4mWriter> y4m_writer_;
- absl::optional<VideoCodecType> codec_type_;
- absl::optional<int> spatial_idx_ RTC_GUARDED_BY(mutex_);
+ std::optional<VideoCodecType> codec_type_;
+ std::optional<int> spatial_idx_ RTC_GUARDED_BY(mutex_);
std::map<uint32_t, VideoFrame> ref_frames_ RTC_GUARDED_BY(mutex_);
Mutex mutex_;
};
@@ -1260,7 +1260,7 @@
std::unique_ptr<VideoEncoder> encoder_;
VideoCodecAnalyzer* const analyzer_;
Pacer pacer_;
- absl::optional<EncodingSettings> last_encoding_settings_;
+ std::optional<EncodingSettings> last_encoding_settings_;
std::unique_ptr<VideoBitrateAllocator> bitrate_allocator_;
LimitedTaskQueue task_queue_;
std::unique_ptr<TesterY4mWriter> y4m_writer_;
@@ -1268,7 +1268,7 @@
std::map<uint32_t, int> sidx_ RTC_GUARDED_BY(mutex_);
std::map<uint32_t, EncodeCallback> callbacks_ RTC_GUARDED_BY(mutex_);
VideoCodecType codec_type_;
- absl::optional<Superframe> last_superframe_;
+ std::optional<Superframe> last_superframe_;
Mutex mutex_;
};
diff --git a/test/video_codec_tester.h b/test/video_codec_tester.h
index 3789f56..fcec47e 100644
--- a/test/video_codec_tester.h
+++ b/test/video_codec_tester.h
@@ -14,10 +14,10 @@
#include <limits>
#include <map>
#include <memory>
+#include <optional>
#include <string>
#include <vector>
-#include "absl/types/optional.h"
#include "api/environment/environment.h"
#include "api/numerics/samples_stats_counter.h"
#include "api/test/metrics/metric.h"
@@ -68,7 +68,7 @@
struct Filter {
uint32_t min_timestamp_rtp = std::numeric_limits<uint32_t>::min();
uint32_t max_timestamp_rtp = std::numeric_limits<uint32_t>::max();
- absl::optional<LayerId> layer_id;
+ std::optional<LayerId> layer_id;
};
struct Frame {
@@ -80,20 +80,20 @@
int height = 0;
DataSize frame_size = DataSize::Zero();
bool keyframe = false;
- absl::optional<int> qp;
+ std::optional<int> qp;
Timestamp encode_start = Timestamp::Zero();
TimeDelta encode_time = TimeDelta::Zero();
Timestamp decode_start = Timestamp::Zero();
TimeDelta decode_time = TimeDelta::Zero();
- absl::optional<DataRate> target_bitrate;
- absl::optional<Frequency> target_framerate;
+ std::optional<DataRate> target_bitrate;
+ std::optional<Frequency> target_framerate;
struct Psnr {
double y = 0.0;
double u = 0.0;
double v = 0.0;
};
- absl::optional<Psnr> psnr;
+ std::optional<Psnr> psnr;
};
struct Stream {
@@ -166,14 +166,14 @@
struct DecoderSettings {
PacingSettings pacing_settings;
- absl::optional<std::string> decoder_input_base_path;
- absl::optional<std::string> decoder_output_base_path;
+ std::optional<std::string> decoder_input_base_path;
+ std::optional<std::string> decoder_output_base_path;
};
struct EncoderSettings {
PacingSettings pacing_settings;
- absl::optional<std::string> encoder_input_base_path;
- absl::optional<std::string> encoder_output_base_path;
+ std::optional<std::string> encoder_input_base_path;
+ std::optional<std::string> encoder_output_base_path;
};
virtual ~VideoCodecTester() = default;
@@ -183,10 +183,10 @@
public:
virtual ~CodedVideoSource() = default;
- // Returns next frame. Returns `absl::nullopt` if the end-of-stream is
+ // Returns next frame. Returns `std::nullopt` if the end-of-stream is
// reached. Frames should have RTP timestamps representing desired frame
// rate.
- virtual absl::optional<EncodedImage> PullFrame() = 0;
+ virtual std::optional<EncodedImage> PullFrame() = 0;
};
// A helper function that creates `EncodingSettings` from the given
diff --git a/test/video_codec_tester_unittest.cc b/test/video_codec_tester_unittest.cc
index e219995..df2ac23 100644
--- a/test/video_codec_tester_unittest.cc
+++ b/test/video_codec_tester_unittest.cc
@@ -188,7 +188,7 @@
std::string codec_type,
ScalabilityMode scalability_mode,
std::vector<std::vector<Frame>> encoded_frames,
- absl::optional<int> num_source_frames = absl::nullopt) {
+ std::optional<int> num_source_frames = std::nullopt) {
int num_frames = encoded_frames.size();
std::string yuv_path =
CreateYuvFile(kWidth, kHeight, num_source_frames.value_or(num_frames));
@@ -276,9 +276,9 @@
MockCodedVideoSource(int num_frames, Frequency framerate)
: num_frames_(num_frames), frame_num_(0), framerate_(framerate) {}
- absl::optional<EncodedImage> PullFrame() override {
+ std::optional<EncodedImage> PullFrame() override {
if (frame_num_ >= num_frames_) {
- return absl::nullopt;
+ return std::nullopt;
}
uint32_t timestamp_rtp = frame_num_ * k90kHz / framerate_;
++frame_num_;
diff --git a/test/video_encoder_proxy_factory.h b/test/video_encoder_proxy_factory.h
index 6a5ff6f..5560fb8 100644
--- a/test/video_encoder_proxy_factory.h
+++ b/test/video_encoder_proxy_factory.h
@@ -127,17 +127,17 @@
encoder_selector_->OnCurrentEncoder(format);
}
- absl::optional<SdpVideoFormat> OnAvailableBitrate(
+ std::optional<SdpVideoFormat> OnAvailableBitrate(
const DataRate& rate) override {
return encoder_selector_->OnAvailableBitrate(rate);
}
- absl::optional<SdpVideoFormat> OnResolutionChange(
+ std::optional<SdpVideoFormat> OnResolutionChange(
const RenderResolution& resolution) override {
return encoder_selector_->OnResolutionChange(resolution);
}
- absl::optional<SdpVideoFormat> OnEncoderBroken() override {
+ std::optional<SdpVideoFormat> OnEncoderBroken() override {
return encoder_selector_->OnEncoderBroken();
}
diff --git a/video/BUILD.gn b/video/BUILD.gn
index 507e428..f46c29f 100644
--- a/video/BUILD.gn
+++ b/video/BUILD.gn
@@ -170,7 +170,6 @@
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/strings",
- "//third_party/abseil-cpp/absl/types:optional",
]
if (!build_with_mozilla) {
@@ -292,7 +291,6 @@
"../system_wrappers:field_trial",
"//third_party/abseil-cpp/absl/base:core_headers",
"//third_party/abseil-cpp/absl/functional:bind_front",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -302,7 +300,6 @@
":frame_decode_timing",
"../api/units:timestamp",
"//third_party/abseil-cpp/absl/functional:any_invocable",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -335,7 +332,6 @@
"../modules/video_coding/timing:timing_module",
"../rtc_base:logging",
"../system_wrappers",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -371,7 +367,6 @@
"../rtc_base:event_tracer",
"../rtc_base:logging",
"../rtc_base:macromagic",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -469,7 +464,6 @@
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/cleanup",
"//third_party/abseil-cpp/absl/container:inlined_vector",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -605,7 +599,6 @@
"//testing/gtest",
"//third_party/abseil-cpp/absl/flags:flag",
"//third_party/abseil-cpp/absl/flags:parse",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -666,7 +659,6 @@
"//testing/gtest",
"//third_party/abseil-cpp/absl/flags:flag",
"//third_party/abseil-cpp/absl/flags:parse",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -710,7 +702,6 @@
"../test:test_support",
"//third_party/abseil-cpp/absl/flags:flag",
"//third_party/abseil-cpp/absl/flags:parse",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -739,7 +730,6 @@
"//testing/gtest",
"//third_party/abseil-cpp/absl/flags:flag",
"//third_party/abseil-cpp/absl/flags:parse",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
}
@@ -965,7 +955,6 @@
"//third_party/abseil-cpp/absl/functional:any_invocable",
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/strings",
- "//third_party/abseil-cpp/absl/types:optional",
"//third_party/abseil-cpp/absl/types:variant",
]
if (!build_with_mozilla) {
diff --git a/video/adaptation/BUILD.gn b/video/adaptation/BUILD.gn
index 1aa280a..829d1f1 100644
--- a/video/adaptation/BUILD.gn
+++ b/video/adaptation/BUILD.gn
@@ -68,7 +68,6 @@
"../../video/config:encoder_config",
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/base:core_headers",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -112,7 +111,6 @@
"../../test:test_support",
"../../test/time_controller:time_controller",
"//third_party/abseil-cpp/absl/functional:any_invocable",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
}
diff --git a/video/adaptation/balanced_constraint.cc b/video/adaptation/balanced_constraint.cc
index f9ee08a..0e8975c 100644
--- a/video/adaptation/balanced_constraint.cc
+++ b/video/adaptation/balanced_constraint.cc
@@ -20,7 +20,7 @@
BalancedConstraint::BalancedConstraint(
DegradationPreferenceProvider* degradation_preference_provider,
const FieldTrialsView& field_trials)
- : encoder_target_bitrate_bps_(absl::nullopt),
+ : encoder_target_bitrate_bps_(std::nullopt),
balanced_settings_(field_trials),
degradation_preference_provider_(degradation_preference_provider) {
RTC_DCHECK(degradation_preference_provider_);
@@ -28,7 +28,7 @@
}
void BalancedConstraint::OnEncoderTargetBitrateUpdated(
- absl::optional<uint32_t> encoder_target_bitrate_bps) {
+ std::optional<uint32_t> encoder_target_bitrate_bps) {
RTC_DCHECK_RUN_ON(&sequence_checker_);
encoder_target_bitrate_bps_ = std::move(encoder_target_bitrate_bps);
}
diff --git a/video/adaptation/balanced_constraint.h b/video/adaptation/balanced_constraint.h
index 22c7d29..294fe6f 100644
--- a/video/adaptation/balanced_constraint.h
+++ b/video/adaptation/balanced_constraint.h
@@ -11,9 +11,9 @@
#ifndef VIDEO_ADAPTATION_BALANCED_CONSTRAINT_H_
#define VIDEO_ADAPTATION_BALANCED_CONSTRAINT_H_
+#include <optional>
#include <string>
-#include "absl/types/optional.h"
#include "api/field_trials_view.h"
#include "api/sequence_checker.h"
#include "call/adaptation/adaptation_constraint.h"
@@ -31,7 +31,7 @@
~BalancedConstraint() override = default;
void OnEncoderTargetBitrateUpdated(
- absl::optional<uint32_t> encoder_target_bitrate_bps);
+ std::optional<uint32_t> encoder_target_bitrate_bps);
// AdaptationConstraint implementation.
std::string Name() const override { return "BalancedConstraint"; }
@@ -42,7 +42,7 @@
private:
RTC_NO_UNIQUE_ADDRESS SequenceChecker sequence_checker_;
- absl::optional<uint32_t> encoder_target_bitrate_bps_
+ std::optional<uint32_t> encoder_target_bitrate_bps_
RTC_GUARDED_BY(&sequence_checker_);
const BalancedDegradationSettings balanced_settings_;
const DegradationPreferenceProvider* degradation_preference_provider_;
diff --git a/video/adaptation/bandwidth_quality_scaler_resource.h b/video/adaptation/bandwidth_quality_scaler_resource.h
index a57c990..c108039 100644
--- a/video/adaptation/bandwidth_quality_scaler_resource.h
+++ b/video/adaptation/bandwidth_quality_scaler_resource.h
@@ -12,11 +12,11 @@
#define VIDEO_ADAPTATION_BANDWIDTH_QUALITY_SCALER_RESOURCE_H_
#include <memory>
+#include <optional>
#include <queue>
#include <string>
#include <vector>
-#include "absl/types/optional.h"
#include "api/scoped_refptr.h"
#include "api/video/video_adaptation_reason.h"
#include "api/video_codecs/video_encoder.h"
diff --git a/video/adaptation/bitrate_constraint.cc b/video/adaptation/bitrate_constraint.cc
index 2f92095..6eb480b 100644
--- a/video/adaptation/bitrate_constraint.cc
+++ b/video/adaptation/bitrate_constraint.cc
@@ -20,19 +20,19 @@
namespace webrtc {
BitrateConstraint::BitrateConstraint()
- : encoder_settings_(absl::nullopt),
- encoder_target_bitrate_bps_(absl::nullopt) {
+ : encoder_settings_(std::nullopt),
+ encoder_target_bitrate_bps_(std::nullopt) {
sequence_checker_.Detach();
}
void BitrateConstraint::OnEncoderSettingsUpdated(
- absl::optional<EncoderSettings> encoder_settings) {
+ std::optional<EncoderSettings> encoder_settings) {
RTC_DCHECK_RUN_ON(&sequence_checker_);
encoder_settings_ = std::move(encoder_settings);
}
void BitrateConstraint::OnEncoderTargetBitrateUpdated(
- absl::optional<uint32_t> encoder_target_bitrate_bps) {
+ std::optional<uint32_t> encoder_target_bitrate_bps) {
RTC_DCHECK_RUN_ON(&sequence_checker_);
encoder_target_bitrate_bps_ = std::move(encoder_target_bitrate_bps);
}
@@ -64,13 +64,13 @@
return true;
}
- absl::optional<int> current_frame_size_px =
+ std::optional<int> current_frame_size_px =
input_state.single_active_stream_pixels();
if (!current_frame_size_px.has_value()) {
return true;
}
- absl::optional<VideoEncoder::ResolutionBitrateLimits> bitrate_limits =
+ std::optional<VideoEncoder::ResolutionBitrateLimits> bitrate_limits =
encoder_settings_->encoder_info().GetEncoderBitrateLimitsForResolution(
// Need some sort of expected resulting pixels to be used
// instead of unrestricted.
diff --git a/video/adaptation/bitrate_constraint.h b/video/adaptation/bitrate_constraint.h
index a608e5d..8906a16 100644
--- a/video/adaptation/bitrate_constraint.h
+++ b/video/adaptation/bitrate_constraint.h
@@ -11,9 +11,9 @@
#ifndef VIDEO_ADAPTATION_BITRATE_CONSTRAINT_H_
#define VIDEO_ADAPTATION_BITRATE_CONSTRAINT_H_
+#include <optional>
#include <string>
-#include "absl/types/optional.h"
#include "api/sequence_checker.h"
#include "call/adaptation/adaptation_constraint.h"
#include "call/adaptation/encoder_settings.h"
@@ -29,9 +29,9 @@
~BitrateConstraint() override = default;
void OnEncoderSettingsUpdated(
- absl::optional<EncoderSettings> encoder_settings);
+ std::optional<EncoderSettings> encoder_settings);
void OnEncoderTargetBitrateUpdated(
- absl::optional<uint32_t> encoder_target_bitrate_bps);
+ std::optional<uint32_t> encoder_target_bitrate_bps);
// AdaptationConstraint implementation.
std::string Name() const override { return "BitrateConstraint"; }
@@ -42,9 +42,9 @@
private:
RTC_NO_UNIQUE_ADDRESS SequenceChecker sequence_checker_;
- absl::optional<EncoderSettings> encoder_settings_
+ std::optional<EncoderSettings> encoder_settings_
RTC_GUARDED_BY(&sequence_checker_);
- absl::optional<uint32_t> encoder_target_bitrate_bps_
+ std::optional<uint32_t> encoder_target_bitrate_bps_
RTC_GUARDED_BY(&sequence_checker_);
};
diff --git a/video/adaptation/bitrate_constraint_unittest.cc b/video/adaptation/bitrate_constraint_unittest.cc
index 8a416db..29ebb6a 100644
--- a/video/adaptation/bitrate_constraint_unittest.cc
+++ b/video/adaptation/bitrate_constraint_unittest.cc
@@ -36,7 +36,7 @@
struct TestParams {
bool active;
- absl::optional<ScalabilityMode> scalability_mode;
+ std::optional<ScalabilityMode> scalability_mode;
};
void FillCodecConfig(VideoCodec* video_codec,
diff --git a/video/adaptation/encode_usage_resource.cc b/video/adaptation/encode_usage_resource.cc
index 4a97881..24d0e5f 100644
--- a/video/adaptation/encode_usage_resource.cc
+++ b/video/adaptation/encode_usage_resource.cc
@@ -29,7 +29,7 @@
: VideoStreamEncoderResource("EncoderUsageResource"),
overuse_detector_(std::move(overuse_detector)),
is_started_(false),
- target_frame_rate_(absl::nullopt) {
+ target_frame_rate_(std::nullopt) {
RTC_DCHECK(overuse_detector_);
}
@@ -56,7 +56,7 @@
}
void EncodeUsageResource::SetTargetFrameRate(
- absl::optional<double> target_frame_rate) {
+ std::optional<double> target_frame_rate) {
RTC_DCHECK_RUN_ON(encoder_queue());
if (target_frame_rate == target_frame_rate_)
return;
@@ -77,7 +77,7 @@
uint32_t timestamp,
int64_t time_sent_in_us,
int64_t capture_time_us,
- absl::optional<int> encode_duration_us) {
+ std::optional<int> encode_duration_us) {
RTC_DCHECK_RUN_ON(encoder_queue());
// TODO(hbos): Rename FrameSent() to something more appropriate (e.g.
// "OnEncodeCompleted"?).
diff --git a/video/adaptation/encode_usage_resource.h b/video/adaptation/encode_usage_resource.h
index c391132..d12e19d 100644
--- a/video/adaptation/encode_usage_resource.h
+++ b/video/adaptation/encode_usage_resource.h
@@ -12,9 +12,9 @@
#define VIDEO_ADAPTATION_ENCODE_USAGE_RESOURCE_H_
#include <memory>
+#include <optional>
#include <string>
-#include "absl/types/optional.h"
#include "api/scoped_refptr.h"
#include "api/video/video_adaptation_reason.h"
#include "video/adaptation/overuse_frame_detector.h"
@@ -42,13 +42,13 @@
void StartCheckForOveruse(CpuOveruseOptions options);
void StopCheckForOveruse();
- void SetTargetFrameRate(absl::optional<double> target_frame_rate);
+ void SetTargetFrameRate(std::optional<double> target_frame_rate);
void OnEncodeStarted(const VideoFrame& cropped_frame,
int64_t time_when_first_seen_us);
void OnEncodeCompleted(uint32_t timestamp,
int64_t time_sent_in_us,
int64_t capture_time_us,
- absl::optional<int> encode_duration_us);
+ std::optional<int> encode_duration_us);
// OveruseFrameDetectorObserverInterface implementation.
void AdaptUp() override;
@@ -60,7 +60,7 @@
const std::unique_ptr<OveruseFrameDetector> overuse_detector_
RTC_GUARDED_BY(encoder_queue());
bool is_started_ RTC_GUARDED_BY(encoder_queue());
- absl::optional<double> target_frame_rate_ RTC_GUARDED_BY(encoder_queue());
+ std::optional<double> target_frame_rate_ RTC_GUARDED_BY(encoder_queue());
};
} // namespace webrtc
diff --git a/video/adaptation/overuse_frame_detector.cc b/video/adaptation/overuse_frame_detector.cc
index 2397b55..1739f95 100644
--- a/video/adaptation/overuse_frame_detector.cc
+++ b/video/adaptation/overuse_frame_detector.cc
@@ -110,12 +110,12 @@
frame.timestamp_us(), frame.rtp_timestamp(), time_when_first_seen_us));
}
- absl::optional<int> FrameSent(
+ std::optional<int> FrameSent(
uint32_t timestamp,
int64_t time_sent_in_us,
int64_t /* capture_time_us */,
- absl::optional<int> /* encode_duration_us */) override {
- absl::optional<int> encode_duration_us;
+ std::optional<int> /* encode_duration_us */) override {
+ std::optional<int> encode_duration_us;
// Delay before reporting actual encoding time, used to have the ability to
// detect total encoding time when encoding more than one layer. Encoding is
// here assumed to finish within a second (or that we get enough long-time
@@ -242,11 +242,10 @@
int64_t time_when_first_seen_us,
int64_t last_capture_time_us) override {}
- absl::optional<int> FrameSent(
- uint32_t /* timestamp */,
- int64_t /* time_sent_in_us */,
- int64_t capture_time_us,
- absl::optional<int> encode_duration_us) override {
+ std::optional<int> FrameSent(uint32_t /* timestamp */,
+ int64_t /* time_sent_in_us */,
+ int64_t capture_time_us,
+ std::optional<int> encode_duration_us) override {
if (encode_duration_us) {
int duration_per_frame_us =
DurationPerInputFrame(capture_time_us, *encode_duration_us);
@@ -364,13 +363,13 @@
usage_->FrameCaptured(frame, time_when_first_seen_us, last_capture_time_us);
}
- absl::optional<int> FrameSent(
+ std::optional<int> FrameSent(
// These two argument used by old estimator.
uint32_t timestamp,
int64_t time_sent_in_us,
// And these two by the new estimator.
int64_t capture_time_us,
- absl::optional<int> encode_duration_us) override {
+ std::optional<int> encode_duration_us) override {
return usage_->FrameSent(timestamp, time_sent_in_us, capture_time_us,
encode_duration_us);
}
@@ -405,7 +404,7 @@
}
}
- absl::optional<int> overried_usage_value;
+ std::optional<int> overried_usage_value;
switch (state_) {
case State::kNormal:
break;
@@ -473,7 +472,7 @@
: env_(env),
metrics_observer_(metrics_observer),
num_process_times_(0),
- // TODO(bugs.webrtc.org/9078): Use absl::optional
+ // TODO(bugs.webrtc.org/9078): Use std::optional
last_capture_time_us_(-1),
num_pixels_(0),
max_framerate_(kDefaultFrameRate),
@@ -543,7 +542,7 @@
usage_->Reset();
last_capture_time_us_ = -1;
num_process_times_ = 0;
- encode_usage_percent_ = absl::nullopt;
+ encode_usage_percent_ = std::nullopt;
OnTargetFramerateUpdated(max_framerate_);
}
@@ -571,7 +570,7 @@
void OveruseFrameDetector::FrameSent(uint32_t timestamp,
int64_t time_sent_in_us,
int64_t capture_time_us,
- absl::optional<int> encode_duration_us) {
+ std::optional<int> encode_duration_us) {
RTC_DCHECK_RUN_ON(&task_checker_);
encode_duration_us = usage_->FrameSent(timestamp, time_sent_in_us,
capture_time_us, encode_duration_us);
diff --git a/video/adaptation/overuse_frame_detector.h b/video/adaptation/overuse_frame_detector.h
index e1729a0..bfdc8eb 100644
--- a/video/adaptation/overuse_frame_detector.h
+++ b/video/adaptation/overuse_frame_detector.h
@@ -13,8 +13,8 @@
#include <list>
#include <memory>
+#include <optional>
-#include "absl/types/optional.h"
#include "api/environment/environment.h"
#include "api/field_trials_view.h"
#include "api/sequence_checker.h"
@@ -105,7 +105,7 @@
void FrameSent(uint32_t timestamp,
int64_t time_sent_in_us,
int64_t capture_time_us,
- absl::optional<int> encode_duration_us);
+ std::optional<int> encode_duration_us);
// Interface for cpu load estimation. Intended for internal use only.
class ProcessingUsage {
@@ -116,13 +116,13 @@
int64_t time_when_first_seen_us,
int64_t last_capture_time_us) = 0;
// Returns encode_time in us, if there's a new measurement.
- virtual absl::optional<int> FrameSent(
+ virtual std::optional<int> FrameSent(
// These two argument used by old estimator.
uint32_t timestamp,
int64_t time_sent_in_us,
// And these two by the new estimator.
int64_t capture_time_us,
- absl::optional<int> encode_duration_us) = 0;
+ std::optional<int> encode_duration_us) = 0;
virtual int Value() = 0;
virtual ~ProcessingUsage() = default;
@@ -156,7 +156,7 @@
// Stats metrics.
CpuOveruseMetricsObserver* const metrics_observer_;
- absl::optional<int> encode_usage_percent_ RTC_GUARDED_BY(task_checker_);
+ std::optional<int> encode_usage_percent_ RTC_GUARDED_BY(task_checker_);
int64_t num_process_times_ RTC_GUARDED_BY(task_checker_);
diff --git a/video/adaptation/overuse_frame_detector_unittest.cc b/video/adaptation/overuse_frame_detector_unittest.cc
index 3c6ce57..b1abdd3 100644
--- a/video/adaptation/overuse_frame_detector_unittest.cc
+++ b/video/adaptation/overuse_frame_detector_unittest.cc
@@ -176,7 +176,7 @@
clock_.AdvanceTime(TimeDelta::Micros(delay_us));
overuse_detector_->FrameSent(timestamp, rtc::TimeMicros(),
capture_time_us,
- absl::optional<int>(delay_us));
+ std::optional<int>(delay_us));
overuse_detector_->CheckForOveruse(observer_);
// Avoid turning clock backwards.
diff --git a/video/adaptation/pixel_limit_resource.cc b/video/adaptation/pixel_limit_resource.cc
index 872e169..9baed46 100644
--- a/video/adaptation/pixel_limit_resource.cc
+++ b/video/adaptation/pixel_limit_resource.cc
@@ -36,7 +36,7 @@
VideoStreamInputStateProvider* input_state_provider)
: task_queue_(task_queue),
input_state_provider_(input_state_provider),
- max_pixels_(absl::nullopt) {
+ max_pixels_(std::nullopt) {
RTC_DCHECK(task_queue_);
RTC_DCHECK(input_state_provider_);
}
@@ -67,7 +67,7 @@
// No pixel limit configured yet, try again later.
return kResourceUsageCheckIntervalMs;
}
- absl::optional<int> frame_size_pixels =
+ std::optional<int> frame_size_pixels =
input_state_provider_->InputState().frame_size_pixels();
if (!frame_size_pixels.has_value()) {
// We haven't observed a frame yet so we don't know if it's going to be
diff --git a/video/adaptation/pixel_limit_resource.h b/video/adaptation/pixel_limit_resource.h
index b42f924..3e5f9f4 100644
--- a/video/adaptation/pixel_limit_resource.h
+++ b/video/adaptation/pixel_limit_resource.h
@@ -11,9 +11,9 @@
#ifndef VIDEO_ADAPTATION_PIXEL_LIMIT_RESOURCE_H_
#define VIDEO_ADAPTATION_PIXEL_LIMIT_RESOURCE_H_
+#include <optional>
#include <string>
-#include "absl/types/optional.h"
#include "api/adaptation/resource.h"
#include "api/scoped_refptr.h"
#include "call/adaptation/video_stream_input_state_provider.h"
@@ -50,7 +50,7 @@
private:
TaskQueueBase* const task_queue_;
VideoStreamInputStateProvider* const input_state_provider_;
- absl::optional<int> max_pixels_ RTC_GUARDED_BY(task_queue_);
+ std::optional<int> max_pixels_ RTC_GUARDED_BY(task_queue_);
webrtc::ResourceListener* listener_ RTC_GUARDED_BY(task_queue_);
RepeatingTaskHandle repeating_task_ RTC_GUARDED_BY(task_queue_);
};
diff --git a/video/adaptation/quality_scaler_resource.h b/video/adaptation/quality_scaler_resource.h
index 0bdc25e..92e72db 100644
--- a/video/adaptation/quality_scaler_resource.h
+++ b/video/adaptation/quality_scaler_resource.h
@@ -12,10 +12,10 @@
#define VIDEO_ADAPTATION_QUALITY_SCALER_RESOURCE_H_
#include <memory>
+#include <optional>
#include <queue>
#include <string>
-#include "absl/types/optional.h"
#include "api/field_trials_view.h"
#include "api/scoped_refptr.h"
#include "api/video/video_adaptation_reason.h"
diff --git a/video/adaptation/quality_scaler_resource_unittest.cc b/video/adaptation/quality_scaler_resource_unittest.cc
index 70d2975..3b9c523 100644
--- a/video/adaptation/quality_scaler_resource_unittest.cc
+++ b/video/adaptation/quality_scaler_resource_unittest.cc
@@ -11,8 +11,8 @@
#include "video/adaptation/quality_scaler_resource.h"
#include <memory>
+#include <optional>
-#include "absl/types/optional.h"
#include "api/task_queue/task_queue_base.h"
#include "api/video_codecs/video_encoder.h"
#include "call/adaptation/test/mock_resource_listener.h"
diff --git a/video/adaptation/video_stream_encoder_resource.h b/video/adaptation/video_stream_encoder_resource.h
index e10f595..5e93b31 100644
--- a/video/adaptation/video_stream_encoder_resource.h
+++ b/video/adaptation/video_stream_encoder_resource.h
@@ -11,10 +11,10 @@
#ifndef VIDEO_ADAPTATION_VIDEO_STREAM_ENCODER_RESOURCE_H_
#define VIDEO_ADAPTATION_VIDEO_STREAM_ENCODER_RESOURCE_H_
+#include <optional>
#include <string>
#include <vector>
-#include "absl/types/optional.h"
#include "api/adaptation/resource.h"
#include "api/sequence_checker.h"
#include "api/task_queue/task_queue_base.h"
diff --git a/video/adaptation/video_stream_encoder_resource_manager.cc b/video/adaptation/video_stream_encoder_resource_manager.cc
index 2bb6f29..96b7f89 100644
--- a/video/adaptation/video_stream_encoder_resource_manager.cc
+++ b/video/adaptation/video_stream_encoder_resource_manager.cc
@@ -114,15 +114,15 @@
return initial_framedrop_ < kMaxInitialFramedrop;
}
- absl::optional<uint32_t> single_active_stream_pixels() const {
+ std::optional<uint32_t> single_active_stream_pixels() const {
return single_active_stream_pixels_;
}
- absl::optional<uint32_t> UseBandwidthAllocationBps() const {
+ std::optional<uint32_t> UseBandwidthAllocationBps() const {
return (use_bandwidth_allocation_ &&
bandwidth_allocation_ > DataRate::Zero())
- ? absl::optional<uint32_t>(bandwidth_allocation_.bps())
- : absl::nullopt;
+ ? std::optional<uint32_t>(bandwidth_allocation_.bps())
+ : std::nullopt;
}
bool last_stream_configuration_changed() const {
@@ -229,7 +229,7 @@
int64_t set_start_bitrate_time_ms_;
// Counts how many frames we've dropped in the initial framedrop phase.
int initial_framedrop_;
- absl::optional<uint32_t> single_active_stream_pixels_;
+ std::optional<uint32_t> single_active_stream_pixels_;
bool use_bandwidth_allocation_;
DataRate bandwidth_allocation_;
@@ -276,8 +276,8 @@
QualityScalingExperiment::Enabled(field_trials_)),
pixel_limit_resource_experiment_enabled_(
field_trials.IsEnabled(kPixelLimitResourceFieldTrialName)),
- encoder_target_bitrate_bps_(absl::nullopt),
- encoder_settings_(absl::nullopt) {
+ encoder_target_bitrate_bps_(std::nullopt),
+ encoder_settings_(std::nullopt) {
TRACE_EVENT0(
"webrtc",
"VideoStreamEncoderResourceManager::VideoStreamEncoderResourceManager");
@@ -479,7 +479,7 @@
void VideoStreamEncoderResourceManager::OnEncodeCompleted(
const EncodedImage& encoded_image,
int64_t time_sent_in_us,
- absl::optional<int> encode_duration_us,
+ std::optional<int> encode_duration_us,
DataSize frame_size) {
RTC_DCHECK_RUN_ON(encoder_queue_);
// Inform `encode_usage_resource_` of the encode completed event.
@@ -504,13 +504,13 @@
return initial_frame_dropper_->DropInitialFrames();
}
-absl::optional<uint32_t>
+std::optional<uint32_t>
VideoStreamEncoderResourceManager::SingleActiveStreamPixels() const {
RTC_DCHECK_RUN_ON(encoder_queue_);
return initial_frame_dropper_->single_active_stream_pixels();
}
-absl::optional<uint32_t>
+std::optional<uint32_t>
VideoStreamEncoderResourceManager::UseBandwidthAllocationBps() const {
RTC_DCHECK_RUN_ON(encoder_queue_);
return initial_frame_dropper_->UseBandwidthAllocationBps();
@@ -522,7 +522,7 @@
}
void VideoStreamEncoderResourceManager::UpdateQualityScalerSettings(
- absl::optional<VideoEncoder::QpThresholds> qp_thresholds) {
+ std::optional<VideoEncoder::QpThresholds> qp_thresholds) {
RTC_DCHECK_RUN_ON(encoder_queue_);
if (qp_thresholds.has_value()) {
if (quality_scaler_resource_->is_started()) {
@@ -580,7 +580,7 @@
// Quality scaler has not already been configured.
// Use experimental thresholds if available.
- absl::optional<VideoEncoder::QpThresholds> experimental_thresholds;
+ std::optional<VideoEncoder::QpThresholds> experimental_thresholds;
if (quality_scaling_experiment_enabled_) {
experimental_thresholds = QualityScalingExperiment::GetQpThresholds(
GetVideoCodecTypeOrGeneric(encoder_settings_), field_trials_);
@@ -590,13 +590,13 @@
: scaling_settings.thresholds);
}
} else {
- UpdateQualityScalerSettings(absl::nullopt);
+ UpdateQualityScalerSettings(std::nullopt);
}
// Set the qp-thresholds to the balanced settings if balanced mode.
if (degradation_preference_ == DegradationPreference::BALANCED &&
quality_scaler_resource_->is_started()) {
- absl::optional<VideoEncoder::QpThresholds> thresholds =
+ std::optional<VideoEncoder::QpThresholds> thresholds =
balanced_settings_.GetQpThresholds(
GetVideoCodecTypeOrGeneric(encoder_settings_),
LastFrameSizeOrDefault());
@@ -712,16 +712,15 @@
void VideoStreamEncoderResourceManager::MaybeUpdateTargetFrameRate() {
RTC_DCHECK_RUN_ON(encoder_queue_);
- absl::optional<double> codec_max_frame_rate =
+ std::optional<double> codec_max_frame_rate =
encoder_settings_.has_value()
- ? absl::optional<double>(
- encoder_settings_->video_codec().maxFramerate)
- : absl::nullopt;
+ ? std::optional<double>(encoder_settings_->video_codec().maxFramerate)
+ : std::nullopt;
// The current target framerate is the maximum frame rate as specified by
// the current codec configuration or any limit imposed by the adaptation
// module. This is used to make sure overuse detection doesn't needlessly
// trigger in low and/or variable framerate scenarios.
- absl::optional<double> target_frame_rate =
+ std::optional<double> target_frame_rate =
video_source_restrictions_.max_frame_rate();
if (!target_frame_rate.has_value() ||
(codec_max_frame_rate.has_value() &&
@@ -776,7 +775,7 @@
return false;
}
- absl::optional<int> num_spatial_layers;
+ std::optional<int> num_spatial_layers;
if (simulcast_layers[0].scalability_mode.has_value() &&
video_codec.numberOfSimulcastStreams == 1) {
num_spatial_layers = ScalabilityModeToNumSpatialLayers(
diff --git a/video/adaptation/video_stream_encoder_resource_manager.h b/video/adaptation/video_stream_encoder_resource_manager.h
index 38f3f08..32e260d 100644
--- a/video/adaptation/video_stream_encoder_resource_manager.h
+++ b/video/adaptation/video_stream_encoder_resource_manager.h
@@ -14,12 +14,12 @@
#include <atomic>
#include <map>
#include <memory>
+#include <optional>
#include <string>
#include <unordered_map>
#include <utility>
#include <vector>
-#include "absl/types/optional.h"
#include "api/adaptation/resource.h"
#include "api/field_trials_view.h"
#include "api/rtp_parameters.h"
@@ -119,7 +119,7 @@
int64_t time_when_first_seen_us);
void OnEncodeCompleted(const EncodedImage& encoded_image,
int64_t time_sent_in_us,
- absl::optional<int> encode_duration_us,
+ std::optional<int> encode_duration_us,
DataSize frame_size);
void OnFrameDropped(EncodedImageCallback::DropReason reason);
@@ -132,8 +132,8 @@
// If true, the VideoStreamEncoder should execute its logic to maybe drop
// frames based on size and bitrate.
bool DropInitialFrames() const;
- absl::optional<uint32_t> SingleActiveStreamPixels() const;
- absl::optional<uint32_t> UseBandwidthAllocationBps() const;
+ std::optional<uint32_t> SingleActiveStreamPixels() const;
+ std::optional<uint32_t> UseBandwidthAllocationBps() const;
// VideoSourceRestrictionsListener implementation.
// Updates `video_source_restrictions_`.
@@ -166,7 +166,7 @@
// Use nullopt to disable quality scaling.
void UpdateQualityScalerSettings(
- absl::optional<VideoEncoder::QpThresholds> qp_thresholds);
+ std::optional<VideoEncoder::QpThresholds> qp_thresholds);
void UpdateBandwidthQualityScalerSettings(
bool bandwidth_quality_scaling_allowed,
@@ -214,11 +214,11 @@
const bool quality_scaling_experiment_enabled_ RTC_GUARDED_BY(encoder_queue_);
const bool pixel_limit_resource_experiment_enabled_
RTC_GUARDED_BY(encoder_queue_);
- absl::optional<uint32_t> encoder_target_bitrate_bps_
+ std::optional<uint32_t> encoder_target_bitrate_bps_
RTC_GUARDED_BY(encoder_queue_);
- absl::optional<VideoEncoder::RateControlParameters> encoder_rates_
+ std::optional<VideoEncoder::RateControlParameters> encoder_rates_
RTC_GUARDED_BY(encoder_queue_);
- absl::optional<EncoderSettings> encoder_settings_
+ std::optional<EncoderSettings> encoder_settings_
RTC_GUARDED_BY(encoder_queue_);
// Ties a resource to a reason for statistical reporting. This AdaptReason is
diff --git a/video/alignment_adjuster.cc b/video/alignment_adjuster.cc
index 1762bec..de32992 100644
--- a/video/alignment_adjuster.cc
+++ b/video/alignment_adjuster.cc
@@ -67,7 +67,7 @@
int AlignmentAdjuster::GetAlignmentAndMaybeAdjustScaleFactors(
const VideoEncoder::EncoderInfo& encoder_info,
VideoEncoderConfig* config,
- absl::optional<size_t> max_layers) {
+ std::optional<size_t> max_layers) {
const int requested_alignment = encoder_info.requested_resolution_alignment;
if (!encoder_info.apply_alignment_to_all_simulcast_layers) {
return requested_alignment;
diff --git a/video/alignment_adjuster.h b/video/alignment_adjuster.h
index 36ac062..086f633 100644
--- a/video/alignment_adjuster.h
+++ b/video/alignment_adjuster.h
@@ -34,7 +34,7 @@
static int GetAlignmentAndMaybeAdjustScaleFactors(
const VideoEncoder::EncoderInfo& info,
VideoEncoderConfig* config,
- absl::optional<size_t> max_layers);
+ std::optional<size_t> max_layers);
};
} // namespace webrtc
diff --git a/video/alignment_adjuster_unittest.cc b/video/alignment_adjuster_unittest.cc
index 28e4bc0..95a5e13 100644
--- a/video/alignment_adjuster_unittest.cc
+++ b/video/alignment_adjuster_unittest.cc
@@ -125,7 +125,7 @@
VideoEncoder::EncoderInfo info =
GetEncoderInfo(kRequestedAlignment, kApplyAlignmentToAllLayers);
int alignment = AlignmentAdjuster::GetAlignmentAndMaybeAdjustScaleFactors(
- info, &config, absl::nullopt);
+ info, &config, std::nullopt);
EXPECT_EQ(alignment, kAdjustedAlignment);
// Verify adjusted scale factors.
@@ -150,7 +150,7 @@
VideoEncoder::EncoderInfo info =
GetEncoderInfo(kRequestedAlignment, kApplyAlignmentToAllLayers);
int alignment = AlignmentAdjuster::GetAlignmentAndMaybeAdjustScaleFactors(
- info, &config, absl::nullopt);
+ info, &config, std::nullopt);
EXPECT_EQ(alignment, kRequestedAlignment);
// Verify that scale factors are not adjusted.
@@ -175,7 +175,7 @@
VideoEncoder::EncoderInfo info =
GetEncoderInfo(kRequestedAlignment, kApplyAlignmentToAllLayers);
int alignment = AlignmentAdjuster::GetAlignmentAndMaybeAdjustScaleFactors(
- info, &config, absl::optional<size_t>(kMaxLayers));
+ info, &config, std::optional<size_t>(kMaxLayers));
EXPECT_EQ(alignment, kAdjustedAlignment);
// Verify adjusted scale factors.
diff --git a/video/buffered_frame_decryptor_unittest.cc b/video/buffered_frame_decryptor_unittest.cc
index 074777b..5089a0e 100644
--- a/video/buffered_frame_decryptor_unittest.cc
+++ b/video/buffered_frame_decryptor_unittest.cc
@@ -74,7 +74,7 @@
kVideoRotation_0,
VideoContentType::UNSPECIFIED,
rtp_video_header,
- /*color_space=*/absl::nullopt,
+ /*color_space=*/std::nullopt,
RtpPacketInfos(),
EncodedImageBuffer::Create(/*size=*/0));
// clang-format on
diff --git a/video/config/BUILD.gn b/video/config/BUILD.gn
index 9c1c064..17930de 100644
--- a/video/config/BUILD.gn
+++ b/video/config/BUILD.gn
@@ -37,7 +37,6 @@
"../../rtc_base/experiments:rate_control_settings",
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/strings",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -56,7 +55,6 @@
"../../rtc_base:checks",
"../../rtc_base:refcount",
"../../rtc_base:stringutils",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
diff --git a/video/config/encoder_stream_factory.cc b/video/config/encoder_stream_factory.cc
index d80f0d1..fc5254d 100644
--- a/video/config/encoder_stream_factory.cc
+++ b/video/config/encoder_stream_factory.cc
@@ -118,7 +118,7 @@
int size,
size_t simulcast_layers) {
int base2_exponent = static_cast<int>(simulcast_layers) - 1;
- const absl::optional<int> experimental_base2_exponent =
+ const std::optional<int> experimental_base2_exponent =
webrtc::NormalizeSimulcastSizeExperiment::GetBase2Exponent(field_trials);
if (experimental_base2_exponent &&
(size > (1 << *experimental_base2_exponent))) {
@@ -131,7 +131,7 @@
// `encoder_config.simulcast_layers` which come from `RtpEncodingParameters`.
void OverrideStreamSettings(
const webrtc::VideoEncoderConfig& encoder_config,
- const absl::optional<webrtc::DataRate>& experimental_min_bitrate,
+ const std::optional<webrtc::DataRate>& experimental_min_bitrate,
std::vector<webrtc::VideoStream>& layers) {
RTC_DCHECK_LE(layers.size(), encoder_config.simulcast_layers.size());
@@ -248,7 +248,7 @@
EncoderStreamFactory::EncoderStreamFactory(
const webrtc::VideoEncoder::EncoderInfo& encoder_info,
- absl::optional<webrtc::VideoSourceRestrictions> restrictions)
+ std::optional<webrtc::VideoSourceRestrictions> restrictions)
: encoder_info_requested_resolution_alignment_(
encoder_info.requested_resolution_alignment),
restrictions_(restrictions) {}
@@ -262,7 +262,7 @@
RTC_DCHECK_GE(encoder_config.simulcast_layers.size(),
encoder_config.number_of_streams);
- const absl::optional<webrtc::DataRate> experimental_min_bitrate =
+ const std::optional<webrtc::DataRate> experimental_min_bitrate =
GetExperimentalMinVideoBitrate(trials, encoder_config.codec_type);
bool is_simulcast = (encoder_config.number_of_streams > 1);
@@ -306,7 +306,7 @@
int width,
int height,
const webrtc::VideoEncoderConfig& encoder_config,
- const absl::optional<webrtc::DataRate>& experimental_min_bitrate) const {
+ const std::optional<webrtc::DataRate>& experimental_min_bitrate) const {
bool is_screencast = encoder_config.content_type ==
webrtc::VideoEncoderConfig::ContentType::kScreen;
@@ -316,7 +316,7 @@
// - `encoder_config.max_bitrate_bps` comes from SDP; "b=AS" or conditionally
// "x-google-max-bitrate".
// If `api_max_bitrate_bps` has a value then it is positive.
- absl::optional<int> api_max_bitrate_bps;
+ std::optional<int> api_max_bitrate_bps;
if (encoder_config.simulcast_layers[0].max_bitrate_bps > 0) {
api_max_bitrate_bps = encoder_config.simulcast_layers[0].max_bitrate_bps;
}
@@ -450,7 +450,7 @@
int width,
int height,
const webrtc::VideoEncoderConfig& encoder_config,
- const absl::optional<webrtc::DataRate>& experimental_min_bitrate) const {
+ const std::optional<webrtc::DataRate>& experimental_min_bitrate) const {
std::vector<webrtc::Resolution> resolutions =
GetStreamResolutions(trials, width, height, encoder_config);
@@ -475,7 +475,7 @@
VideoAdapter adapter(encoder_info_requested_resolution_alignment_);
adapter.OnOutputFormatRequest(requested_resolution.ToPair(),
requested_resolution.PixelCount(),
- absl::nullopt);
+ std::nullopt);
if (restrictions_) {
rtc::VideoSinkWants wants;
wants.is_active = true;
diff --git a/video/config/encoder_stream_factory.h b/video/config/encoder_stream_factory.h
index 7792a81..3d0705e 100644
--- a/video/config/encoder_stream_factory.h
+++ b/video/config/encoder_stream_factory.h
@@ -25,8 +25,8 @@
: public webrtc::VideoEncoderConfig::VideoStreamFactoryInterface {
public:
EncoderStreamFactory(const webrtc::VideoEncoder::EncoderInfo& encoder_info,
- absl::optional<webrtc::VideoSourceRestrictions>
- restrictions = absl::nullopt);
+ std::optional<webrtc::VideoSourceRestrictions>
+ restrictions = std::nullopt);
std::vector<webrtc::VideoStream> CreateEncoderStreams(
const webrtc::FieldTrialsView& trials,
@@ -39,7 +39,7 @@
int width,
int height,
const webrtc::VideoEncoderConfig& encoder_config,
- const absl::optional<webrtc::DataRate>& experimental_min_bitrate) const;
+ const std::optional<webrtc::DataRate>& experimental_min_bitrate) const;
std::vector<webrtc::VideoStream>
CreateSimulcastOrConferenceModeScreenshareStreams(
@@ -47,7 +47,7 @@
int width,
int height,
const webrtc::VideoEncoderConfig& encoder_config,
- const absl::optional<webrtc::DataRate>& experimental_min_bitrate) const;
+ const std::optional<webrtc::DataRate>& experimental_min_bitrate) const;
webrtc::Resolution GetLayerResolutionFromRequestedResolution(
int in_frame_width,
@@ -61,7 +61,7 @@
const webrtc::VideoEncoderConfig& encoder_config) const;
const int encoder_info_requested_resolution_alignment_;
- const absl::optional<webrtc::VideoSourceRestrictions> restrictions_;
+ const std::optional<webrtc::VideoSourceRestrictions> restrictions_;
};
} // namespace cricket
diff --git a/video/config/encoder_stream_factory_unittest.cc b/video/config/encoder_stream_factory_unittest.cc
index b20adb5..aa33367 100644
--- a/video/config/encoder_stream_factory_unittest.cc
+++ b/video/config/encoder_stream_factory_unittest.cc
@@ -69,7 +69,7 @@
const FieldTrialsView& field_trials,
const Resolution& resolution,
const VideoEncoderConfig& encoder_config,
- absl::optional<VideoSourceRestrictions> restrictions = absl::nullopt) {
+ std::optional<VideoSourceRestrictions> restrictions = std::nullopt) {
VideoEncoder::EncoderInfo encoder_info;
auto factory =
rtc::make_ref_counted<EncoderStreamFactory>(encoder_info, restrictions);
@@ -99,8 +99,8 @@
ExplicitKeyValueConfig field_trials("");
VideoSourceRestrictions restrictions(
/* max_pixels_per_frame= */ (320 * 320),
- /* target_pixels_per_frame= */ absl::nullopt,
- /* max_frame_rate= */ absl::nullopt);
+ /* target_pixels_per_frame= */ std::nullopt,
+ /* max_frame_rate= */ std::nullopt);
VideoEncoderConfig encoder_config;
encoder_config.number_of_streams = 1;
encoder_config.simulcast_layers.resize(1);
diff --git a/video/config/simulcast.cc b/video/config/simulcast.cc
index a655c2e..08246ac 100644
--- a/video/config/simulcast.cc
+++ b/video/config/simulcast.cc
@@ -14,11 +14,11 @@
#include <stdio.h>
#include <algorithm>
+#include <optional>
#include <string>
#include <vector>
#include "absl/strings/match.h"
-#include "absl/types/optional.h"
#include "api/field_trials_view.h"
#include "api/video/video_codec_constants.h"
#include "media/base/media_constants.h"
@@ -198,7 +198,7 @@
SimulcastFormat InterpolateSimulcastFormat(
int width,
int height,
- absl::optional<double> max_roundup_rate,
+ std::optional<double> max_roundup_rate,
bool enable_lowres_bitrate_interpolation,
webrtc::VideoCodecType codec) {
const auto formats =
@@ -247,7 +247,7 @@
layers[s].num_temporal_layers = num_temporal_layers;
SimulcastFormat interpolated_format = InterpolateSimulcastFormat(
- layers[s].width, layers[s].height, /*max_roundup_rate=*/absl::nullopt,
+ layers[s].width, layers[s].height, /*max_roundup_rate=*/std::nullopt,
enable_lowres_bitrate_interpolation, codec);
layers[s].max_bitrate_bps = interpolated_format.max_bitrate.bps();
diff --git a/video/config/video_encoder_config.cc b/video/config/video_encoder_config.cc
index c3e81ec..24b8f2a 100644
--- a/video/config/video_encoder_config.cc
+++ b/video/config/video_encoder_config.cc
@@ -24,7 +24,7 @@
max_bitrate_bps(-1),
scale_resolution_down_by(-1.),
max_qp(-1),
- num_temporal_layers(absl::nullopt),
+ num_temporal_layers(std::nullopt),
active(true) {}
VideoStream::VideoStream(const VideoStream& other) = default;
diff --git a/video/config/video_encoder_config.h b/video/config/video_encoder_config.h
index 3f8f286..bd05ab0 100644
--- a/video/config/video_encoder_config.h
+++ b/video/config/video_encoder_config.h
@@ -13,10 +13,10 @@
#include <stddef.h>
+#include <optional>
#include <string>
#include <vector>
-#include "absl/types/optional.h"
#include "api/field_trials_view.h"
#include "api/scoped_refptr.h"
#include "api/video/resolution.h"
@@ -62,13 +62,13 @@
// (meaning that this field _must_ be set), and for signaling the app-level
// encoder settings (meaning that the field _may_ be set). We should separate
// this and remove this optional instead.
- absl::optional<size_t> num_temporal_layers;
+ std::optional<size_t> num_temporal_layers;
// The priority of this stream, to be used when allocating resources
// between multiple streams.
- absl::optional<double> bitrate_priority;
+ std::optional<double> bitrate_priority;
- absl::optional<ScalabilityMode> scalability_mode;
+ std::optional<ScalabilityMode> scalability_mode;
// If this stream is enabled by the user, or not.
bool active;
@@ -82,7 +82,7 @@
// which can be lower than requested_resolution,
// e.g. if source only provides lower resolution or
// if resource adaptation is active.
- absl::optional<Resolution> requested_resolution;
+ std::optional<Resolution> requested_resolution;
};
class VideoEncoderConfig {
@@ -90,7 +90,7 @@
// These are reference counted to permit copying VideoEncoderConfig and be
// kept alive until all encoder_specific_settings go out of scope.
// TODO(kthelgason): Consider removing the need for copying VideoEncoderConfig
- // and use absl::optional for encoder_specific_settings instead.
+ // and use std::optional for encoder_specific_settings instead.
class EncoderSpecificSettings : public RefCountInterface {
public:
// TODO(pbos): Remove FillEncoderSpecificSettings as soon as VideoCodec is
diff --git a/video/corruption_detection/BUILD.gn b/video/corruption_detection/BUILD.gn
index 4e66857..336e896 100644
--- a/video/corruption_detection/BUILD.gn
+++ b/video/corruption_detection/BUILD.gn
@@ -25,7 +25,6 @@
"../../modules/video_coding:video_coding_utility",
"../../rtc_base:logging",
"//third_party/abseil-cpp/absl/algorithm:container",
- "//third_party/abseil-cpp/absl/types:optional",
"//third_party/abseil-cpp/absl/types:variant",
]
}
@@ -53,7 +52,6 @@
"../../api/video:video_frame",
"../../rtc_base:checks",
"../../rtc_base:logging",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -76,7 +74,6 @@
"../../api/video:video_frame",
"../../api/video:video_frame_type",
"../../test:test_support",
- "//third_party/abseil-cpp/absl/types:optional",
"//third_party/abseil-cpp/absl/types:variant",
]
}
diff --git a/video/corruption_detection/frame_instrumentation_generator.cc b/video/corruption_detection/frame_instrumentation_generator.cc
index ee8b492..d0e6596 100644
--- a/video/corruption_detection/frame_instrumentation_generator.cc
+++ b/video/corruption_detection/frame_instrumentation_generator.cc
@@ -13,10 +13,10 @@
#include <algorithm>
#include <cstdint>
#include <iterator>
+#include <optional>
#include <vector>
#include "absl/algorithm/container.h"
-#include "absl/types/optional.h"
#include "absl/types/variant.h"
#include "api/scoped_refptr.h"
#include "api/video/encoded_image.h"
@@ -34,7 +34,7 @@
namespace webrtc {
namespace {
-absl::optional<FilterSettings> GetCorruptionFilterSettings(
+std::optional<FilterSettings> GetCorruptionFilterSettings(
const EncodedImage& encoded_image,
VideoCodecType video_codec_type,
int layer_id) {
@@ -51,13 +51,13 @@
int qp = encoded_image.qp_;
if (qp == -1) {
- absl::optional<uint32_t> parsed_qp = QpParser().Parse(
+ std::optional<uint32_t> parsed_qp = QpParser().Parse(
video_codec_type, layer_id, encoded_image.data(), encoded_image.size());
if (!parsed_qp.has_value()) {
RTC_LOG(LS_VERBOSE) << "Missing QP for "
<< CodecTypeToPayloadString(video_codec_type)
<< " layer " << layer_id << ".";
- return absl::nullopt;
+ return std::nullopt;
}
qp = *parsed_qp;
}
@@ -75,7 +75,7 @@
captured_frames_.push(frame);
}
-absl::optional<
+std::optional<
absl::variant<FrameInstrumentationSyncData, FrameInstrumentationData>>
FrameInstrumentationGenerator::OnEncodedImage(
const EncodedImage& encoded_image) {
@@ -89,7 +89,7 @@
captured_frames_.front().rtp_timestamp() != rtp_timestamp_encoded_image) {
RTC_LOG(LS_VERBOSE) << "No captured frames for RTC timestamp "
<< rtp_timestamp_encoded_image << ".";
- return absl::nullopt;
+ return std::nullopt;
}
VideoFrame captured_frame = captured_frames_.front();
@@ -116,7 +116,7 @@
} else if (contexts_.find(layer_id) == contexts_.end()) {
RTC_LOG(LS_INFO) << "The first frame of a spatial or simulcast layer is "
"not a key frame.";
- return absl::nullopt;
+ return std::nullopt;
}
int sequence_index = contexts_[layer_id].frame_sampler.GetCurrentIndex();
@@ -128,16 +128,16 @@
/*sample_size=*/13);
if (sample_coordinates.empty()) {
if (!is_key_frame) {
- return absl::nullopt;
+ return std::nullopt;
}
return FrameInstrumentationSyncData{.sequence_index = sequence_index,
.is_key_frame = true};
}
- absl::optional<FilterSettings> filter_settings =
+ std::optional<FilterSettings> filter_settings =
GetCorruptionFilterSettings(encoded_image, video_codec_type_, layer_id);
if (!filter_settings.has_value()) {
- return absl::nullopt;
+ return std::nullopt;
}
scoped_refptr<I420BufferInterface> captured_frame_buffer_as_i420 =
@@ -147,7 +147,7 @@
<< VideoFrameBufferTypeToString(
captured_frame.video_frame_buffer()->type())
<< " image to I420.";
- return absl::nullopt;
+ return std::nullopt;
}
FrameInstrumentationData data = {
diff --git a/video/corruption_detection/frame_instrumentation_generator.h b/video/corruption_detection/frame_instrumentation_generator.h
index a2004e5..2e0f909 100644
--- a/video/corruption_detection/frame_instrumentation_generator.h
+++ b/video/corruption_detection/frame_instrumentation_generator.h
@@ -13,10 +13,10 @@
#include <cstdint>
#include <map>
+#include <optional>
#include <queue>
#include <vector>
-#include "absl/types/optional.h"
#include "absl/types/variant.h"
#include "api/video/encoded_image.h"
#include "api/video/video_codec_type.h"
@@ -52,7 +52,7 @@
~FrameInstrumentationGenerator() = default;
void OnCapturedFrame(VideoFrame frame);
- absl::optional<
+ std::optional<
absl::variant<FrameInstrumentationSyncData, FrameInstrumentationData>>
OnEncodedImage(const EncodedImage& encoded_image);
diff --git a/video/corruption_detection/frame_instrumentation_generator_unittest.cc b/video/corruption_detection/frame_instrumentation_generator_unittest.cc
index 796f929..4af4af8 100644
--- a/video/corruption_detection/frame_instrumentation_generator_unittest.cc
+++ b/video/corruption_detection/frame_instrumentation_generator_unittest.cc
@@ -11,9 +11,9 @@
#include "video/corruption_detection/frame_instrumentation_generator.h"
#include <cstdint>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "absl/types/variant.h"
#include "api/scoped_refptr.h"
#include "api/video/encoded_image.h"
@@ -142,7 +142,7 @@
encoded_image._encodedHeight = kDefaultScaledHeight;
generator.OnCapturedFrame(frame);
- absl::optional<
+ std::optional<
absl::variant<FrameInstrumentationSyncData, FrameInstrumentationData>>
data = generator.OnEncodedImage(encoded_image);
@@ -182,7 +182,7 @@
encoded_image._encodedHeight = kDefaultScaledHeight;
generator.OnCapturedFrame(frame);
- absl::optional<
+ std::optional<
absl::variant<FrameInstrumentationSyncData, FrameInstrumentationData>>
data = generator.OnEncodedImage(encoded_image);
@@ -224,7 +224,7 @@
generator.OnCapturedFrame(frame);
generator.OnEncodedImage(encoded_image1);
- absl::optional<
+ std::optional<
absl::variant<FrameInstrumentationSyncData, FrameInstrumentationData>>
data = generator.OnEncodedImage(encoded_image2);
@@ -307,11 +307,11 @@
generator.OnCapturedFrame(frame1);
generator.OnCapturedFrame(frame2);
- absl::optional<
+ std::optional<
absl::variant<FrameInstrumentationSyncData, FrameInstrumentationData>>
data1 = generator.OnEncodedImage(encoded_image1);
- absl::optional<
+ std::optional<
absl::variant<FrameInstrumentationSyncData, FrameInstrumentationData>>
data2 = generator.OnEncodedImage(encoded_image2);
@@ -357,11 +357,11 @@
generator.OnCapturedFrame(frame);
- absl::optional<
+ std::optional<
absl::variant<FrameInstrumentationSyncData, FrameInstrumentationData>>
data1 = generator.OnEncodedImage(encoded_image1);
- absl::optional<
+ std::optional<
absl::variant<FrameInstrumentationSyncData, FrameInstrumentationData>>
data2 = generator.OnEncodedImage(encoded_image2);
diff --git a/video/corruption_detection/halton_frame_sampler.h b/video/corruption_detection/halton_frame_sampler.h
index 877bc79..9908c56 100644
--- a/video/corruption_detection/halton_frame_sampler.h
+++ b/video/corruption_detection/halton_frame_sampler.h
@@ -12,9 +12,9 @@
#define VIDEO_CORRUPTION_DETECTION_HALTON_FRAME_SAMPLER_H_
#include <cstdint>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/scoped_refptr.h"
#include "api/video/video_frame_buffer.h"
#include "video/corruption_detection/halton_sequence.h"
@@ -56,7 +56,7 @@
Coordinates GetNextSampleCoordinates();
HaltonSequence coordinate_sampler_prng_;
- absl::optional<uint32_t> rtp_timestamp_last_frame_sampled_;
+ std::optional<uint32_t> rtp_timestamp_last_frame_sampled_;
int frames_sampled_ = 0;
int frames_until_next_sample_ = 0;
};
diff --git a/video/decode_synchronizer.cc b/video/decode_synchronizer.cc
index 32702c2..94c4902 100644
--- a/video/decode_synchronizer.cc
+++ b/video/decode_synchronizer.cc
@@ -58,11 +58,11 @@
RTC_DCHECK(stopped_);
}
-absl::optional<uint32_t>
+std::optional<uint32_t>
DecodeSynchronizer::SynchronizedFrameDecodeScheduler::ScheduledRtpTimestamp() {
return next_frame_.has_value()
- ? absl::make_optional(next_frame_->rtp_timestamp())
- : absl::nullopt;
+ ? std::make_optional(next_frame_->rtp_timestamp())
+ : std::nullopt;
}
DecodeSynchronizer::ScheduledFrame
diff --git a/video/decode_synchronizer.h b/video/decode_synchronizer.h
index 1718186..d441fa0 100644
--- a/video/decode_synchronizer.h
+++ b/video/decode_synchronizer.h
@@ -15,10 +15,10 @@
#include <functional>
#include <memory>
+#include <optional>
#include <set>
#include <utility>
-#include "absl/types/optional.h"
#include "api/metronome/metronome.h"
#include "api/sequence_checker.h"
#include "api/task_queue/task_queue_base.h"
@@ -103,7 +103,7 @@
Timestamp LatestDecodeTime();
// FrameDecodeScheduler implementation.
- absl::optional<uint32_t> ScheduledRtpTimestamp() override;
+ std::optional<uint32_t> ScheduledRtpTimestamp() override;
void ScheduleFrame(uint32_t rtp,
FrameDecodeTiming::FrameSchedule schedule,
FrameReleaseCallback cb) override;
@@ -112,7 +112,7 @@
private:
DecodeSynchronizer* sync_;
- absl::optional<ScheduledFrame> next_frame_;
+ std::optional<ScheduledFrame> next_frame_;
bool stopped_ = false;
};
diff --git a/video/encoder_bitrate_adjuster.cc b/video/encoder_bitrate_adjuster.cc
index 8743099..c74ac49 100644
--- a/video/encoder_bitrate_adjuster.cc
+++ b/video/encoder_bitrate_adjuster.cc
@@ -193,11 +193,11 @@
layer_info.media_utilization_factor = 0.0;
for (size_t ti = 0; ti < active_tls[si]; ++ti) {
RTC_DCHECK(overshoot_detectors_[si][ti]);
- const absl::optional<double> ti_link_utilization_factor =
+ const std::optional<double> ti_link_utilization_factor =
overshoot_detectors_[si][ti]->GetNetworkRateUtilizationFactor(
now.ms());
- const absl::optional<double> ti_media_utilization_factor =
+ const std::optional<double> ti_media_utilization_factor =
overshoot_detectors_[si][ti]->GetMediaRateUtilizationFactor(
now.ms());
if (!ti_link_utilization_factor || !ti_media_utilization_factor) {
diff --git a/video/encoder_overshoot_detector.cc b/video/encoder_overshoot_detector.cc
index 8446c135..b9697f5 100644
--- a/video/encoder_overshoot_detector.cc
+++ b/video/encoder_overshoot_detector.cc
@@ -136,13 +136,13 @@
return utilization_factor;
}
-absl::optional<double>
-EncoderOvershootDetector::GetNetworkRateUtilizationFactor(int64_t time_ms) {
+std::optional<double> EncoderOvershootDetector::GetNetworkRateUtilizationFactor(
+ int64_t time_ms) {
CullOldUpdates(time_ms);
// No data points within window, return.
if (utilization_factors_.empty()) {
- return absl::nullopt;
+ return std::nullopt;
}
// TODO(sprang): Consider changing from arithmetic mean to some other
@@ -150,13 +150,13 @@
return sum_network_utilization_factors_ / utilization_factors_.size();
}
-absl::optional<double> EncoderOvershootDetector::GetMediaRateUtilizationFactor(
+std::optional<double> EncoderOvershootDetector::GetMediaRateUtilizationFactor(
int64_t time_ms) {
CullOldUpdates(time_ms);
// No data points within window, return.
if (utilization_factors_.empty()) {
- return absl::nullopt;
+ return std::nullopt;
}
return sum_media_utilization_factors_ / utilization_factors_.size();
diff --git a/video/encoder_overshoot_detector.h b/video/encoder_overshoot_detector.h
index 12c4bba..e78b601 100644
--- a/video/encoder_overshoot_detector.h
+++ b/video/encoder_overshoot_detector.h
@@ -12,8 +12,8 @@
#define VIDEO_ENCODER_OVERSHOOT_DETECTOR_H_
#include <deque>
+#include <optional>
-#include "absl/types/optional.h"
#include "api/units/data_rate.h"
#include "api/video_codecs/video_codec.h"
@@ -34,11 +34,11 @@
// This utilization factor reaches 1.0 only if the encoder produces encoded
// frame in such a way that they can be sent onto the network at
// `target_bitrate` without building growing queues.
- absl::optional<double> GetNetworkRateUtilizationFactor(int64_t time_ms);
+ std::optional<double> GetNetworkRateUtilizationFactor(int64_t time_ms);
// This utilization factor is based just on actual encoded frame sizes in
// relation to ideal sizes. An undershoot may be compensated by an
// overshoot so that the average over time is close to `target_bitrate`.
- absl::optional<double> GetMediaRateUtilizationFactor(int64_t time_ms);
+ std::optional<double> GetMediaRateUtilizationFactor(int64_t time_ms);
void Reset();
private:
diff --git a/video/encoder_overshoot_detector_unittest.cc b/video/encoder_overshoot_detector_unittest.cc
index 27e58ad..95d0fd1 100644
--- a/video/encoder_overshoot_detector_unittest.cc
+++ b/video/encoder_overshoot_detector_unittest.cc
@@ -87,12 +87,12 @@
// At constant utilization, both network and media utilization should be
// close to expected.
- const absl::optional<double> network_utilization_factor =
+ const std::optional<double> network_utilization_factor =
detector_.GetNetworkRateUtilizationFactor(rtc::TimeMillis());
EXPECT_NEAR(network_utilization_factor.value_or(-1),
expected_utilization_factor, allowed_error);
- const absl::optional<double> media_utilization_factor =
+ const std::optional<double> media_utilization_factor =
detector_.GetMediaRateUtilizationFactor(rtc::TimeMillis());
EXPECT_NEAR(media_utilization_factor.value_or(-1),
expected_utilization_factor, allowed_error);
@@ -189,12 +189,12 @@
}
// Expect 5% overshoot for network rate, see above.
- const absl::optional<double> network_utilization_factor =
+ const std::optional<double> network_utilization_factor =
detector_.GetNetworkRateUtilizationFactor(rtc::TimeMillis());
EXPECT_NEAR(network_utilization_factor.value_or(-1), 1.05, 0.01);
// Expect media rate to be on average correct.
- const absl::optional<double> media_utilization_factor =
+ const std::optional<double> media_utilization_factor =
detector_.GetMediaRateUtilizationFactor(rtc::TimeMillis());
EXPECT_NEAR(media_utilization_factor.value_or(-1), 1.00, 0.01);
}
diff --git a/video/encoder_rtcp_feedback.cc b/video/encoder_rtcp_feedback.cc
index d9d9aea..42827e2 100644
--- a/video/encoder_rtcp_feedback.cc
+++ b/video/encoder_rtcp_feedback.cc
@@ -11,9 +11,9 @@
#include "video/encoder_rtcp_feedback.h"
#include <algorithm>
+#include <optional>
#include <utility>
-#include "absl/types/optional.h"
#include "api/environment/environment.h"
#include "api/video_codecs/video_encoder.h"
#include "rtc_base/checks.h"
@@ -124,7 +124,7 @@
loss_notification.dependencies_of_last_received_decodable =
decodability_flag;
loss_notification.last_received_decodable =
- !decodability_flag ? absl::make_optional(false) : absl::nullopt;
+ !decodability_flag ? std::make_optional(false) : std::nullopt;
} else if (!last_received.is_first && last_received.is_last) {
if (decodability_flag) {
// The frame has been received in full, and found to be decodable.
@@ -136,7 +136,7 @@
// It is impossible to tell whether some dependencies were undecodable,
// or whether the frame was unassemblable, but in either case, the frame
// itself was undecodable.
- loss_notification.dependencies_of_last_received_decodable = absl::nullopt;
+ loss_notification.dependencies_of_last_received_decodable = std::nullopt;
loss_notification.last_received_decodable = false;
}
} else { // !last_received.is_first && !last_received.is_last
@@ -146,12 +146,12 @@
// (Messages of this type are not sent by WebRTC at the moment, but are
// theoretically possible, for example for serving as acks.)
loss_notification.dependencies_of_last_received_decodable = true;
- loss_notification.last_received_decodable = absl::nullopt;
+ loss_notification.last_received_decodable = std::nullopt;
} else {
// It is impossible to tell whether some dependencies were undecodable,
// or whether the frame was unassemblable, but in either case, the frame
// itself was undecodable.
- loss_notification.dependencies_of_last_received_decodable = absl::nullopt;
+ loss_notification.dependencies_of_last_received_decodable = std::nullopt;
loss_notification.last_received_decodable = false;
}
}
diff --git a/video/end_to_end_tests/call_operation_tests.cc b/video/end_to_end_tests/call_operation_tests.cc
index fe965f2..f99565d 100644
--- a/video/end_to_end_tests/call_operation_tests.cc
+++ b/video/end_to_end_tests/call_operation_tests.cc
@@ -116,8 +116,8 @@
// Create frames that are smaller than the send width/height, this is
// done to check that the callbacks are done after processing video.
std::unique_ptr<test::FrameGeneratorInterface> frame_generator(
- test::CreateSquareFrameGenerator(kWidth, kHeight, absl::nullopt,
- absl::nullopt));
+ test::CreateSquareFrameGenerator(kWidth, kHeight, std::nullopt,
+ std::nullopt));
GetVideoSendStream()->SetSource(&frame_forwarder,
DegradationPreference::MAINTAIN_FRAMERATE);
@@ -175,7 +175,7 @@
frame_generator = test::CreateSquareFrameGenerator(
test::VideoTestConstants::kDefaultWidth,
- test::VideoTestConstants::kDefaultHeight, absl::nullopt, absl::nullopt);
+ test::VideoTestConstants::kDefaultHeight, std::nullopt, std::nullopt);
GetVideoSendStream()->SetSource(&frame_forwarder,
DegradationPreference::MAINTAIN_FRAMERATE);
test::FrameGeneratorInterface::VideoFrameData frame_data =
diff --git a/video/end_to_end_tests/codec_tests.cc b/video/end_to_end_tests/codec_tests.cc
index e2ceb5e..6460cda 100644
--- a/video/end_to_end_tests/codec_tests.cc
+++ b/video/end_to_end_tests/codec_tests.cc
@@ -9,8 +9,8 @@
*/
#include <memory>
+#include <optional>
-#include "absl/types/optional.h"
#include "api/environment/environment.h"
#include "api/test/video/function_video_encoder_factory.h"
#include "api/video/color_space.h"
@@ -52,7 +52,7 @@
public:
CodecObserver(int no_frames_to_wait_for,
VideoRotation rotation_to_test,
- absl::optional<ColorSpace> color_space_to_test,
+ std::optional<ColorSpace> color_space_to_test,
const std::string& payload_name,
VideoEncoderFactory* encoder_factory,
VideoDecoderFactory* decoder_factory)
@@ -98,8 +98,8 @@
if (expected_color_space_) {
EXPECT_EQ(expected_color_space_,
video_frame.color_space()
- ? absl::make_optional(*video_frame.color_space())
- : absl::nullopt);
+ ? std::make_optional(*video_frame.color_space())
+ : std::nullopt);
}
if (++frame_counter_ == no_frames_to_wait_for_)
observation_complete_.Set();
@@ -114,7 +114,7 @@
private:
int no_frames_to_wait_for_;
VideoRotation expected_rotation_;
- absl::optional<ColorSpace> expected_color_space_;
+ std::optional<ColorSpace> expected_color_space_;
std::string payload_name_;
VideoEncoderFactory* encoder_factory_;
VideoDecoderFactory* decoder_factory_;
@@ -130,8 +130,8 @@
[](const Environment& env, const SdpVideoFormat& format) {
return CreateVp8Decoder(env);
});
- CodecObserver test(5, kVideoRotation_0, absl::nullopt, "VP8",
- &encoder_factory, &decoder_factory);
+ CodecObserver test(5, kVideoRotation_0, std::nullopt, "VP8", &encoder_factory,
+ &decoder_factory);
RunBaseTest(&test);
}
@@ -144,7 +144,7 @@
[](const Environment& env, const SdpVideoFormat& format) {
return CreateVp8Decoder(env);
});
- CodecObserver test(5, kVideoRotation_90, absl::nullopt, "VP8",
+ CodecObserver test(5, kVideoRotation_90, std::nullopt, "VP8",
&encoder_factory, &decoder_factory);
RunBaseTest(&test);
}
@@ -157,7 +157,7 @@
});
test::FunctionVideoDecoderFactory decoder_factory(
[]() { return VP9Decoder::Create(); });
- CodecObserver test(500, kVideoRotation_0, absl::nullopt, "VP9",
+ CodecObserver test(500, kVideoRotation_0, std::nullopt, "VP9",
&encoder_factory, &decoder_factory);
RunBaseTest(&test);
}
@@ -169,7 +169,7 @@
});
test::FunctionVideoDecoderFactory decoder_factory(
[]() { return VP9Decoder::Create(); });
- CodecObserver test(5, kVideoRotation_90, absl::nullopt, "VP9",
+ CodecObserver test(5, kVideoRotation_90, std::nullopt, "VP9",
&encoder_factory, &decoder_factory);
RunBaseTest(&test);
}
@@ -229,7 +229,7 @@
});
test::FunctionVideoDecoderFactory decoder_factory(
[]() { return H264Decoder::Create(); });
- CodecObserver test(500, kVideoRotation_0, absl::nullopt, "H264",
+ CodecObserver test(500, kVideoRotation_0, std::nullopt, "H264",
&encoder_factory, &decoder_factory);
RunBaseTest(&test);
}
@@ -241,7 +241,7 @@
});
test::FunctionVideoDecoderFactory decoder_factory(
[]() { return H264Decoder::Create(); });
- CodecObserver test(5, kVideoRotation_90, absl::nullopt, "H264",
+ CodecObserver test(5, kVideoRotation_90, std::nullopt, "H264",
&encoder_factory, &decoder_factory);
RunBaseTest(&test);
}
@@ -255,7 +255,7 @@
});
test::FunctionVideoDecoderFactory decoder_factory(
[]() { return H264Decoder::Create(); });
- CodecObserver test(500, kVideoRotation_0, absl::nullopt, "H264",
+ CodecObserver test(500, kVideoRotation_0, std::nullopt, "H264",
&encoder_factory, &decoder_factory);
RunBaseTest(&test);
}
@@ -269,7 +269,7 @@
});
test::FunctionVideoDecoderFactory decoder_factory(
[]() { return H264Decoder::Create(); });
- CodecObserver test(500, kVideoRotation_0, absl::nullopt, "H264",
+ CodecObserver test(500, kVideoRotation_0, std::nullopt, "H264",
&encoder_factory, &decoder_factory);
RunBaseTest(&test);
}
diff --git a/video/end_to_end_tests/extended_reports_tests.cc b/video/end_to_end_tests/extended_reports_tests.cc
index 9ce5311..4e6cf0e 100644
--- a/video/end_to_end_tests/extended_reports_tests.cc
+++ b/video/end_to_end_tests/extended_reports_tests.cc
@@ -12,11 +12,11 @@
#include <stdint.h>
#include <memory>
+#include <optional>
#include <string>
#include <utility>
#include <vector>
-#include "absl/types/optional.h"
#include "api/rtp_headers.h"
#include "api/task_queue/task_queue_base.h"
#include "api/test/simulated_network.h"
diff --git a/video/end_to_end_tests/histogram_tests.cc b/video/end_to_end_tests/histogram_tests.cc
index d12242f..1d27f97 100644
--- a/video/end_to_end_tests/histogram_tests.cc
+++ b/video/end_to_end_tests/histogram_tests.cc
@@ -8,7 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "absl/types/optional.h"
+#include <optional>
+
#include "api/test/video/function_video_encoder_factory.h"
#include "modules/video_coding/codecs/vp8/include/vp8.h"
#include "rtc_base/synchronization/mutex.h"
@@ -151,7 +152,7 @@
const bool use_fec_;
const bool screenshare_;
test::FunctionVideoEncoderFactory encoder_factory_;
- absl::optional<int64_t> start_runtime_ms_;
+ std::optional<int64_t> start_runtime_ms_;
int num_frames_received_ RTC_GUARDED_BY(&mutex_);
} test(use_rtx, use_fec, screenshare);
diff --git a/video/end_to_end_tests/multi_codec_receive_tests.cc b/video/end_to_end_tests/multi_codec_receive_tests.cc
index fe223af..e9c8bc4 100644
--- a/video/end_to_end_tests/multi_codec_receive_tests.cc
+++ b/video/end_to_end_tests/multi_codec_receive_tests.cc
@@ -120,8 +120,8 @@
}
Mutex mutex_;
- absl::optional<uint32_t> last_timestamp_; // Only accessed from pacer thread.
- absl::optional<uint8_t> expected_payload_type_ RTC_GUARDED_BY(mutex_);
+ std::optional<uint32_t> last_timestamp_; // Only accessed from pacer thread.
+ std::optional<uint8_t> expected_payload_type_ RTC_GUARDED_BY(mutex_);
int num_sent_frames_ RTC_GUARDED_BY(mutex_) = 0;
int num_rendered_frames_ RTC_GUARDED_BY(mutex_) = 0;
std::vector<uint32_t> sent_timestamps_ RTC_GUARDED_BY(mutex_);
diff --git a/video/end_to_end_tests/multi_stream_tester.cc b/video/end_to_end_tests/multi_stream_tester.cc
index abd3238..7d82085 100644
--- a/video/end_to_end_tests/multi_stream_tester.cc
+++ b/video/end_to_end_tests/multi_stream_tester.cc
@@ -115,8 +115,8 @@
auto* frame_generator = new test::FrameGeneratorCapturer(
&env.clock(),
- test::CreateSquareFrameGenerator(width, height, absl::nullopt,
- absl::nullopt),
+ test::CreateSquareFrameGenerator(width, height, std::nullopt,
+ std::nullopt),
30, env.task_queue_factory());
frame_generators[i] = frame_generator;
send_streams[i]->SetSource(frame_generator,
diff --git a/video/end_to_end_tests/resolution_bitrate_limits_tests.cc b/video/end_to_end_tests/resolution_bitrate_limits_tests.cc
index 7949376..06eb71e 100644
--- a/video/end_to_end_tests/resolution_bitrate_limits_tests.cc
+++ b/video/end_to_end_tests/resolution_bitrate_limits_tests.cc
@@ -103,13 +103,13 @@
public test::FakeEncoder {
public:
struct Bitrate {
- const absl::optional<DataRate> min;
- const absl::optional<DataRate> max;
+ const std::optional<DataRate> min;
+ const std::optional<DataRate> max;
};
struct TestConfig {
const bool active;
const Bitrate bitrate;
- const absl::optional<ScalabilityMode> scalability_mode;
+ const std::optional<ScalabilityMode> scalability_mode;
};
struct Expectation {
const uint32_t pixels = 0;
@@ -252,7 +252,7 @@
.scalability_mode = ScalabilityMode::kL2T1}},
// Expectations:
{{.pixels = 1280 * 720,
- .ne_bitrate = {absl::nullopt, DataRate::KilobitsPerSec(3000)}}});
+ .ne_bitrate = {std::nullopt, DataRate::KilobitsPerSec(3000)}}});
RunBaseTest(&test);
}
TEST_F(ResolutionBitrateLimitsWithScalabilityModeTest,
@@ -369,7 +369,7 @@
}
TEST_P(ResolutionBitrateLimitsTest, DefaultLimitsAppliedMiddleActive) {
- const absl::optional<VideoEncoder::ResolutionBitrateLimits>
+ const std::optional<VideoEncoder::ResolutionBitrateLimits>
kDefaultSinglecastLimits360p =
EncoderInfoSettings::GetDefaultSinglecastBitrateLimitsForResolution(
PayloadStringToCodecType(payload_name_), 640 * 360);
@@ -388,7 +388,7 @@
TEST_F(ResolutionBitrateLimitsWithScalabilityModeTest,
DefaultLimitsAppliedForOneSpatialLayer) {
- const absl::optional<VideoEncoder::ResolutionBitrateLimits>
+ const std::optional<VideoEncoder::ResolutionBitrateLimits>
kDefaultSinglecastLimits720p =
EncoderInfoSettings::GetDefaultSinglecastBitrateLimitsForResolution(
PayloadStringToCodecType("VP9"), 1280 * 720);
diff --git a/video/end_to_end_tests/retransmission_tests.cc b/video/end_to_end_tests/retransmission_tests.cc
index f74af24..afcd8b9 100644
--- a/video/end_to_end_tests/retransmission_tests.cc
+++ b/video/end_to_end_tests/retransmission_tests.cc
@@ -191,7 +191,7 @@
uint32_t local_ssrc_;
uint32_t remote_ssrc_;
Transport* receive_transport_;
- absl::optional<uint16_t> sequence_number_to_retransmit_;
+ std::optional<uint16_t> sequence_number_to_retransmit_;
} test;
RunBaseTest(&test);
diff --git a/video/end_to_end_tests/rtp_rtcp_tests.cc b/video/end_to_end_tests/rtp_rtcp_tests.cc
index 00a46dd..7ea0ed0 100644
--- a/video/end_to_end_tests/rtp_rtcp_tests.cc
+++ b/video/end_to_end_tests/rtp_rtcp_tests.cc
@@ -456,9 +456,9 @@
return SEND_PACKET;
}
- absl::optional<uint16_t> last_observed_sequence_number_
+ std::optional<uint16_t> last_observed_sequence_number_
RTC_GUARDED_BY(mutex_);
- absl::optional<uint32_t> last_observed_timestamp_ RTC_GUARDED_BY(mutex_);
+ std::optional<uint32_t> last_observed_timestamp_ RTC_GUARDED_BY(mutex_);
size_t num_flexfec_packets_sent_ RTC_GUARDED_BY(mutex_);
Mutex mutex_;
} observer;
diff --git a/video/end_to_end_tests/stats_tests.cc b/video/end_to_end_tests/stats_tests.cc
index 226f5fc..4d14fa4 100644
--- a/video/end_to_end_tests/stats_tests.cc
+++ b/video/end_to_end_tests/stats_tests.cc
@@ -9,9 +9,9 @@
*/
#include <memory>
+#include <optional>
#include "absl/algorithm/container.h"
-#include "absl/types/optional.h"
#include "api/task_queue/task_queue_base.h"
#include "api/test/simulated_network.h"
#include "api/test/video/function_video_encoder_factory.h"
@@ -691,7 +691,7 @@
bool dropped_rtp_packet_requested_ RTC_GUARDED_BY(&mutex_) = false;
std::vector<VideoReceiveStreamInterface*> receive_streams_;
VideoSendStream* send_stream_ = nullptr;
- absl::optional<int64_t> start_runtime_ms_;
+ std::optional<int64_t> start_runtime_ms_;
TaskQueueBase* const task_queue_;
rtc::scoped_refptr<PendingTaskSafetyFlag> task_safety_flag_ =
PendingTaskSafetyFlag::CreateDetached();
diff --git a/video/frame_cadence_adapter.cc b/video/frame_cadence_adapter.cc
index 8a7852b..21dec7f 100644
--- a/video/frame_cadence_adapter.cc
+++ b/video/frame_cadence_adapter.cc
@@ -56,7 +56,7 @@
const VideoFrame& frame) = 0;
// Returns the currently estimated input framerate.
- virtual absl::optional<uint32_t> GetInputFrameRateFps() = 0;
+ virtual std::optional<uint32_t> GetInputFrameRateFps() = 0;
// Updates the frame rate.
virtual void UpdateFrameRate(Timestamp frame_timestamp) = 0;
@@ -79,7 +79,7 @@
callback_->OnFrame(post_time, queue_overload, frame);
}
- absl::optional<uint32_t> GetInputFrameRateFps() override {
+ std::optional<uint32_t> GetInputFrameRateFps() override {
RTC_DCHECK_RUN_ON(&sequence_checker_);
return last_frame_rate_;
}
@@ -92,7 +92,7 @@
}
private:
- absl::optional<uint64_t> last_frame_rate_;
+ std::optional<uint64_t> last_frame_rate_;
FrameCadenceAdapterInterface::Callback* const callback_;
RTC_NO_UNIQUE_ADDRESS SequenceChecker sequence_checker_;
// Input frame rate statistics for use when not in zero-hertz mode.
@@ -127,7 +127,7 @@
void OnFrame(Timestamp post_time,
bool queue_overload,
const VideoFrame& frame) override;
- absl::optional<uint32_t> GetInputFrameRateFps() override;
+ std::optional<uint32_t> GetInputFrameRateFps() override;
void UpdateFrameRate(Timestamp frame_timestamp) override {}
// Notified on dropped frames.
@@ -139,7 +139,7 @@
// Updates the restrictions of max frame rate for the video source.
// Always called during construction using latest `restricted_frame_delay_`.
- void UpdateVideoSourceRestrictions(absl::optional<double> max_frame_rate);
+ void UpdateVideoSourceRestrictions(std::optional<double> max_frame_rate);
private:
// The tracking state of each spatial layer. Used for determining when to
@@ -147,7 +147,7 @@
struct SpatialLayerTracker {
// If unset, the layer is disabled. Otherwise carries the quality
// convergence status of the layer.
- absl::optional<bool> quality_converged;
+ std::optional<bool> quality_converged;
};
// The state of a scheduled repeat.
struct ScheduledRepeat {
@@ -201,9 +201,9 @@
// parameter in the OnFrame callback will be true while
// `queue_overload_count_` is larger than zero to allow the client to drop
// frames and thereby mitigate delay buildups.
- // Repeated frames are sent with `post_time` set to absl::nullopt.
- void SendFrameNow(absl::optional<Timestamp> post_time,
- const VideoFrame& frame) RTC_RUN_ON(sequence_checker_);
+ // Repeated frames are sent with `post_time` set to std::nullopt.
+ void SendFrameNow(std::optional<Timestamp> post_time, const VideoFrame& frame)
+ RTC_RUN_ON(sequence_checker_);
// Returns the repeat duration depending on if it's an idle repeat or not.
TimeDelta RepeatDuration(bool idle_repeat) const
RTC_RUN_ON(sequence_checker_);
@@ -239,7 +239,7 @@
// for cancelling deferred repeated frame processing happening.
int current_frame_id_ RTC_GUARDED_BY(sequence_checker_) = 0;
// Has content when we are repeating frames.
- absl::optional<ScheduledRepeat> scheduled_repeat_
+ std::optional<ScheduledRepeat> scheduled_repeat_
RTC_GUARDED_BY(sequence_checker_);
// Convergent state of each of the configured simulcast layers.
std::vector<SpatialLayerTracker> layer_trackers_
@@ -250,7 +250,7 @@
RTC_GUARDED_BY(sequence_checker_);
// Can be set by UpdateVideoSourceRestrictions when the video source restricts
// the max frame rate.
- absl::optional<TimeDelta> restricted_frame_delay_
+ std::optional<TimeDelta> restricted_frame_delay_
RTC_GUARDED_BY(sequence_checker_);
// Set in OnSendFrame to reflect how many future frames will be forwarded with
// the `queue_overload` flag set to true.
@@ -289,7 +289,7 @@
bool queue_overload,
const VideoFrame& frame) override;
- absl::optional<uint32_t> GetInputFrameRateFps() override {
+ std::optional<uint32_t> GetInputFrameRateFps() override {
RTC_DCHECK_RUN_ON(&queue_sequence_checker_);
return last_frame_rate_;
}
@@ -323,7 +323,7 @@
RTC_NO_UNIQUE_ADDRESS SequenceChecker queue_sequence_checker_;
rtc::scoped_refptr<PendingTaskSafetyFlag> queue_safety_flag_;
// Input frame rate statistics for use when not in zero-hertz mode.
- absl::optional<uint64_t> last_frame_rate_
+ std::optional<uint64_t> last_frame_rate_
RTC_GUARDED_BY(queue_sequence_checker_);
RateStatistics input_framerate_ RTC_GUARDED_BY(queue_sequence_checker_){
FrameCadenceAdapterInterface::kFrameRateAveragingWindowSizeMs, 1000};
@@ -354,13 +354,13 @@
// FrameCadenceAdapterInterface overrides.
void Initialize(Callback* callback) override;
void SetZeroHertzModeEnabled(
- absl::optional<ZeroHertzModeParams> params) override;
- absl::optional<uint32_t> GetInputFrameRateFps() override;
+ std::optional<ZeroHertzModeParams> params) override;
+ std::optional<uint32_t> GetInputFrameRateFps() override;
void UpdateLayerQualityConvergence(size_t spatial_index,
bool quality_converged) override;
void UpdateLayerStatus(size_t spatial_index, bool enabled) override;
void UpdateVideoSourceRestrictions(
- absl::optional<double> max_frame_rate) override;
+ std::optional<double> max_frame_rate) override;
void ProcessKeyFrameRequest() override;
// VideoFrameSink overrides.
@@ -400,17 +400,17 @@
// calculating input frame rate.
const bool use_video_frame_timestamp_;
// Used for verifying that timestamps are monotonically increasing.
- absl::optional<Timestamp> last_incoming_frame_timestamp_;
+ std::optional<Timestamp> last_incoming_frame_timestamp_;
bool incoming_frame_timestamp_monotonically_increasing_ = true;
// The three possible modes we're under.
- absl::optional<PassthroughAdapterMode> passthrough_adapter_;
- absl::optional<ZeroHertzAdapterMode> zero_hertz_adapter_;
+ std::optional<PassthroughAdapterMode> passthrough_adapter_;
+ std::optional<ZeroHertzAdapterMode> zero_hertz_adapter_;
// The `vsync_encode_adapter_` must be destroyed on the worker queue since
// VSync metronome needs to happen on worker thread.
std::unique_ptr<VSyncEncodeAdapterMode> vsync_encode_adapter_;
// If set, zero-hertz mode has been enabled.
- absl::optional<ZeroHertzModeParams> zero_hertz_params_;
+ std::optional<ZeroHertzModeParams> zero_hertz_params_;
// Cache for the current adapter mode.
AdapterMode* current_adapter_mode_ = nullptr;
@@ -419,20 +419,20 @@
TaskQueueBase* const worker_queue_;
// Timestamp for statistics reporting.
- absl::optional<Timestamp> zero_hertz_adapter_created_timestamp_
+ std::optional<Timestamp> zero_hertz_adapter_created_timestamp_
RTC_GUARDED_BY(queue_);
// Set up during Initialize.
Callback* callback_ = nullptr;
// The source's constraints.
- absl::optional<VideoTrackSourceConstraints> source_constraints_
+ std::optional<VideoTrackSourceConstraints> source_constraints_
RTC_GUARDED_BY(queue_);
// Stores the latest restriction in max frame rate set by
// UpdateVideoSourceRestrictions. Ensures that a previously set restriction
// can be maintained during reconstructions of the adapter.
- absl::optional<double> restricted_max_frame_rate_ RTC_GUARDED_BY(queue_);
+ std::optional<double> restricted_max_frame_rate_ RTC_GUARDED_BY(queue_);
// Race checker for incoming frames. This is the network thread in chromium,
// but may vary from test contexts.
@@ -501,7 +501,7 @@
layer_trackers_[spatial_index].quality_converged = false;
}
} else {
- layer_trackers_[spatial_index].quality_converged = absl::nullopt;
+ layer_trackers_[spatial_index].quality_converged = std::nullopt;
}
}
@@ -526,7 +526,7 @@
// Store the frame in the queue and schedule deferred processing.
queued_frames_.push_back(frame);
current_frame_id_++;
- scheduled_repeat_ = absl::nullopt;
+ scheduled_repeat_ = std::nullopt;
TimeDelta time_spent_since_post = clock_->CurrentTime() - post_time;
queue_->PostDelayedHighPrecisionTask(
SafeTask(safety_.flag(),
@@ -548,13 +548,13 @@
MaybeStartRefreshFrameRequester();
}
-absl::optional<uint32_t> ZeroHertzAdapterMode::GetInputFrameRateFps() {
+std::optional<uint32_t> ZeroHertzAdapterMode::GetInputFrameRateFps() {
RTC_DCHECK_RUN_ON(&sequence_checker_);
return max_fps_;
}
void ZeroHertzAdapterMode::UpdateVideoSourceRestrictions(
- absl::optional<double> max_frame_rate) {
+ std::optional<double> max_frame_rate) {
RTC_DCHECK_RUN_ON(&sequence_checker_);
TRACE_EVENT_INSTANT1(TRACE_DISABLED_BY_DEFAULT("webrtc"), __func__,
TRACE_EVENT_SCOPE_GLOBAL, "max_frame_rate",
@@ -564,7 +564,7 @@
restricted_frame_delay_ = TimeDelta::Seconds(1) / *max_frame_rate;
} else {
// Source reports that the frame rate is now unrestricted.
- restricted_frame_delay_ = absl::nullopt;
+ restricted_frame_delay_ = std::nullopt;
}
}
@@ -708,10 +708,10 @@
// Schedule another repeat before sending the frame off which could take time.
ScheduleRepeat(frame_id, HasQualityConverged());
- SendFrameNow(absl::nullopt, frame);
+ SendFrameNow(std::nullopt, frame);
}
-void ZeroHertzAdapterMode::SendFrameNow(absl::optional<Timestamp> post_time,
+void ZeroHertzAdapterMode::SendFrameNow(std::optional<Timestamp> post_time,
const VideoFrame& frame) {
RTC_DCHECK_RUN_ON(&sequence_checker_);
TRACE_EVENT0("webrtc", __func__);
@@ -903,14 +903,14 @@
}
void FrameCadenceAdapterImpl::SetZeroHertzModeEnabled(
- absl::optional<ZeroHertzModeParams> params) {
+ std::optional<ZeroHertzModeParams> params) {
RTC_DCHECK_RUN_ON(queue_);
bool was_zero_hertz_enabled = zero_hertz_params_.has_value();
zero_hertz_params_ = params;
MaybeReconfigureAdapters(was_zero_hertz_enabled);
}
-absl::optional<uint32_t> FrameCadenceAdapterImpl::GetInputFrameRateFps() {
+std::optional<uint32_t> FrameCadenceAdapterImpl::GetInputFrameRateFps() {
RTC_DCHECK_RUN_ON(queue_);
return current_adapter_mode_->GetInputFrameRateFps();
}
@@ -945,7 +945,7 @@
}
void FrameCadenceAdapterImpl::UpdateVideoSourceRestrictions(
- absl::optional<double> max_frame_rate) {
+ std::optional<double> max_frame_rate) {
RTC_DCHECK_RUN_ON(queue_);
// Store the restriction to ensure that it can be reapplied in possible
// future adapter creations on configuration changes.
@@ -975,7 +975,7 @@
if (zero_hertz_adapter_created_timestamp_.has_value()) {
TimeDelta time_until_first_frame =
clock_->CurrentTime() - *zero_hertz_adapter_created_timestamp_;
- zero_hertz_adapter_created_timestamp_ = absl::nullopt;
+ zero_hertz_adapter_created_timestamp_ = std::nullopt;
RTC_HISTOGRAM_COUNTS_10000(
"WebRTC.Screenshare.ZeroHz.TimeUntilFirstFrameMs",
time_until_first_frame.ms());
@@ -1073,7 +1073,7 @@
current_adapter_mode_ = &zero_hertz_adapter_.value();
} else {
if (was_zero_hertz_enabled) {
- zero_hertz_adapter_ = absl::nullopt;
+ zero_hertz_adapter_ = std::nullopt;
RTC_LOG(LS_INFO) << "Zero hertz mode disabled.";
}
ConfigureCurrentAdapterWithoutZeroHertz();
diff --git a/video/frame_cadence_adapter.h b/video/frame_cadence_adapter.h
index 0dc2e5a..cdee92b 100644
--- a/video/frame_cadence_adapter.h
+++ b/video/frame_cadence_adapter.h
@@ -93,11 +93,11 @@
// zero-hertz operation. If absl:::nullopt is passed, the cadence adapter will
// switch to passthrough mode.
virtual void SetZeroHertzModeEnabled(
- absl::optional<ZeroHertzModeParams> params) = 0;
+ std::optional<ZeroHertzModeParams> params) = 0;
// Returns the input framerate. This is measured by RateStatistics when
// zero-hertz mode is off, and returns the max framerate in zero-hertz mode.
- virtual absl::optional<uint32_t> GetInputFrameRateFps() = 0;
+ virtual std::optional<uint32_t> GetInputFrameRateFps() = 0;
// Updates quality convergence status for an enabled spatial layer.
// Convergence means QP has dropped to a low-enough level to warrant ceasing
@@ -112,7 +112,7 @@
// The new `max_frame_rate` will only affect the cadence of Callback::OnFrame
// for non-idle (non converged) repeated frames.
virtual void UpdateVideoSourceRestrictions(
- absl::optional<double> max_frame_rate) = 0;
+ std::optional<double> max_frame_rate) = 0;
// Conditionally requests a refresh frame via
// Callback::RequestRefreshFrame.
diff --git a/video/frame_cadence_adapter_unittest.cc b/video/frame_cadence_adapter_unittest.cc
index 237b142..e3bdf14 100644
--- a/video/frame_cadence_adapter_unittest.cc
+++ b/video/frame_cadence_adapter_unittest.cc
@@ -113,7 +113,7 @@
for (int frame = 0; frame != 10; ++frame) {
time_controller.AdvanceTime(TimeDelta::Millis(10));
- absl::optional<int64_t> expected_fps =
+ std::optional<int64_t> expected_fps =
rate.Rate(time_controller.GetClock()->TimeInMilliseconds());
rate.Update(1, time_controller.GetClock()->TimeInMilliseconds());
// FrameCadanceAdapter::OnFrame post the frame to another sequence.
@@ -186,7 +186,7 @@
}
// Turn off zero hertz on the next-last frame; after the last frame we
// should see a value that tracks the rate oracle.
- adapter->SetZeroHertzModeEnabled(absl::nullopt);
+ adapter->SetZeroHertzModeEnabled(std::nullopt);
// Last frame.
time_controller.AdvanceTime(TimeDelta::Millis(10));
adapter->OnFrame(CreateFrameWithTimestamps(&time_controller));
@@ -926,7 +926,7 @@
adapter_->UpdateVideoSourceRestrictions(kRestrictedMaxFps);
});
ScheduleDelayed(5.5 * kMinFrameDelay, [&] {
- adapter_->UpdateVideoSourceRestrictions(absl::nullopt);
+ adapter_->UpdateVideoSourceRestrictions(std::nullopt);
});
ExpectFrameEntriesAtDelaysFromNow({
1 * kMinFrameDelay, // Original frame emitted at non-restricted rate.
@@ -1122,7 +1122,7 @@
std::unique_ptr<FrameCadenceAdapterInterface> adapter;
int frame_counter = 0;
rtc::Event event;
- absl::optional<Timestamp> start_time;
+ std::optional<Timestamp> start_time;
test::ScopedKeyValueConfig no_field_trials;
queue->PostTask([&] {
adapter = CreateAdapter(no_field_trials, clock);
diff --git a/video/frame_decode_scheduler.h b/video/frame_decode_scheduler.h
index 29e27c2..762268a 100644
--- a/video/frame_decode_scheduler.h
+++ b/video/frame_decode_scheduler.h
@@ -13,8 +13,9 @@
#include <stdint.h>
+#include <optional>
+
#include "absl/functional/any_invocable.h"
-#include "absl/types/optional.h"
#include "api/units/timestamp.h"
#include "video/frame_decode_timing.h"
@@ -31,7 +32,7 @@
// Returns the rtp timestamp of the next frame scheduled for release, or
// `nullopt` if no frame is currently scheduled.
- virtual absl::optional<uint32_t> ScheduledRtpTimestamp() = 0;
+ virtual std::optional<uint32_t> ScheduledRtpTimestamp() = 0;
// Schedules a frame for release based on `schedule`. When released,
// `callback` will be invoked with the `rtp` timestamp of the frame and the
diff --git a/video/frame_decode_timing.cc b/video/frame_decode_timing.cc
index 58ecd41..e7401a0 100644
--- a/video/frame_decode_timing.cc
+++ b/video/frame_decode_timing.cc
@@ -11,8 +11,8 @@
#include "video/frame_decode_timing.h"
#include <algorithm>
+#include <optional>
-#include "absl/types/optional.h"
#include "api/units/time_delta.h"
#include "rtc_base/logging.h"
@@ -25,7 +25,7 @@
RTC_DCHECK(timing_);
}
-absl::optional<FrameDecodeTiming::FrameSchedule>
+std::optional<FrameDecodeTiming::FrameSchedule>
FrameDecodeTiming::OnFrameBufferUpdated(uint32_t next_temporal_unit_rtp,
uint32_t last_temporal_unit_rtp,
TimeDelta max_wait_for_frame,
@@ -44,7 +44,7 @@
RTC_DLOG(LS_VERBOSE) << "Fast-forwarded frame " << next_temporal_unit_rtp
<< " render time " << render_time << " with delay "
<< max_wait;
- return absl::nullopt;
+ return std::nullopt;
}
max_wait.Clamp(TimeDelta::Zero(), max_wait_for_frame);
diff --git a/video/frame_decode_timing.h b/video/frame_decode_timing.h
index 6bde470..c5b024f 100644
--- a/video/frame_decode_timing.h
+++ b/video/frame_decode_timing.h
@@ -38,7 +38,7 @@
Timestamp render_time;
};
- absl::optional<FrameSchedule> OnFrameBufferUpdated(
+ std::optional<FrameSchedule> OnFrameBufferUpdated(
uint32_t next_temporal_unit_rtp,
uint32_t last_temporal_unit_rtp,
TimeDelta max_wait_for_frame,
diff --git a/video/frame_decode_timing_unittest.cc b/video/frame_decode_timing_unittest.cc
index 83ea916..cb45d70 100644
--- a/video/frame_decode_timing_unittest.cc
+++ b/video/frame_decode_timing_unittest.cc
@@ -12,7 +12,8 @@
#include <stdint.h>
-#include "absl/types/optional.h"
+#include <optional>
+
#include "api/units/time_delta.h"
#include "modules/video_coding/timing/timing.h"
#include "rtc_base/containers/flat_map.h"
@@ -98,7 +99,7 @@
EXPECT_THAT(frame_decode_scheduler_.OnFrameBufferUpdated(
90000, 180000, kMaxWaitForFrame, false),
- Eq(absl::nullopt));
+ Eq(std::nullopt));
}
TEST_F(FrameDecodeTimingTest, NoFastForwardIfOnlyFrameToDecode) {
diff --git a/video/frame_encode_metadata_writer.cc b/video/frame_encode_metadata_writer.cc
index 4ff4871..1fd06a6 100644
--- a/video/frame_encode_metadata_writer.cc
+++ b/video/frame_encode_metadata_writer.cc
@@ -141,8 +141,8 @@
size_t simulcast_svc_idx,
EncodedImage* encoded_image) {
MutexLock lock(&lock_);
- absl::optional<size_t> outlier_frame_size;
- absl::optional<int64_t> encode_start_ms;
+ std::optional<size_t> outlier_frame_size;
+ std::optional<int64_t> encode_start_ms;
uint8_t timing_flags = VideoSendTiming::kNotTriggered;
int64_t encode_done_ms = rtc::TimeMillis();
@@ -225,11 +225,11 @@
stalled_encoder_logged_messages_ = 0;
}
-absl::optional<int64_t>
+std::optional<int64_t>
FrameEncodeMetadataWriter::ExtractEncodeStartTimeAndFillMetadata(
size_t simulcast_svc_idx,
EncodedImage* encoded_image) {
- absl::optional<int64_t> result;
+ std::optional<int64_t> result;
size_t num_simulcast_svc_streams = timing_frames_info_.size();
if (simulcast_svc_idx < num_simulcast_svc_streams) {
auto metadata_list = &timing_frames_info_[simulcast_svc_idx].frames;
diff --git a/video/frame_encode_metadata_writer.h b/video/frame_encode_metadata_writer.h
index 555bc9b..637d03a 100644
--- a/video/frame_encode_metadata_writer.h
+++ b/video/frame_encode_metadata_writer.h
@@ -12,9 +12,9 @@
#define VIDEO_FRAME_ENCODE_METADATA_WRITER_H_
#include <list>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/video/encoded_image.h"
#include "api/video_codecs/video_codec.h"
#include "api/video_codecs/video_encoder.h"
@@ -45,7 +45,7 @@
private:
// For non-internal-source encoders, returns encode started time and fixes
// capture timestamp for the frame, if corrupted by the encoder.
- absl::optional<int64_t> ExtractEncodeStartTimeAndFillMetadata(
+ std::optional<int64_t> ExtractEncodeStartTimeAndFillMetadata(
size_t simulcast_svc_idx,
EncodedImage* encoded_image) RTC_EXCLUSIVE_LOCKS_REQUIRED(lock_);
@@ -55,7 +55,7 @@
int64_t ntp_time_ms = 0;
int64_t timestamp_us = 0;
VideoRotation rotation = kVideoRotation_0;
- absl::optional<ColorSpace> color_space;
+ std::optional<ColorSpace> color_space;
bool is_steady_state_refresh_frame = false;
RtpPacketInfos packet_infos;
};
diff --git a/video/full_stack_tests.cc b/video/full_stack_tests.cc
index 5e30b86..7e46260 100644
--- a/video/full_stack_tests.cc
+++ b/video/full_stack_tests.cc
@@ -8,13 +8,13 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include <memory>
+#include <optional>
#include <string>
#include <utility>
#include <vector>
#include "absl/flags/flag.h"
#include "absl/flags/parse.h"
-#include "absl/types/optional.h"
#include "api/test/simulated_network.h"
#include "api/test/test_dependency_factory.h"
#include "api/test/video_quality_test_fixture.h"
diff --git a/video/picture_id_tests.cc b/video/picture_id_tests.cc
index e78f7d3..14abcb1 100644
--- a/video/picture_id_tests.cc
+++ b/video/picture_id_tests.cc
@@ -98,7 +98,7 @@
parsed->timestamp = rtp_packet.Timestamp();
parsed->ssrc = rtp_packet.Ssrc();
- absl::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed_payload =
+ std::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed_payload =
depacketizer_->Parse(rtp_packet.PayloadBuffer());
EXPECT_TRUE(parsed_payload);
diff --git a/video/quality_convergence_controller.cc b/video/quality_convergence_controller.cc
index a19cb8c..d897635 100644
--- a/video/quality_convergence_controller.cc
+++ b/video/quality_convergence_controller.cc
@@ -41,7 +41,7 @@
void QualityConvergenceController::Initialize(
int number_of_layers,
- absl::optional<int> static_qp_threshold,
+ std::optional<int> static_qp_threshold,
VideoCodecType codec,
const FieldTrialsView& trials) {
RTC_DCHECK(sequence_checker_.IsCurrent());
diff --git a/video/quality_convergence_controller.h b/video/quality_convergence_controller.h
index a09dc87..16978fd 100644
--- a/video/quality_convergence_controller.h
+++ b/video/quality_convergence_controller.h
@@ -12,9 +12,9 @@
#define VIDEO_QUALITY_CONVERGENCE_CONTROLLER_H_
#include <memory>
+#include <optional>
#include <vector>
-#include "absl/types/optional.h"
#include "api/field_trials_view.h"
#include "api/sequence_checker.h"
#include "api/video/video_codec_type.h"
@@ -25,7 +25,7 @@
class QualityConvergenceController {
public:
void Initialize(int number_of_layers,
- absl::optional<int> static_qp_threshold,
+ std::optional<int> static_qp_threshold,
VideoCodecType codec,
const FieldTrialsView& trials);
diff --git a/video/quality_scaling_tests.cc b/video/quality_scaling_tests.cc
index cd67e68..12b5650 100644
--- a/video/quality_scaling_tests.cc
+++ b/video/quality_scaling_tests.cc
@@ -57,17 +57,17 @@
protected:
const std::string kPrefix = "WebRTC-Video-QualityScaling/Enabled-";
const std::string kEnd = ",0,0,0.9995,0.9999,1/";
- const absl::optional<VideoEncoder::ResolutionBitrateLimits>
+ const std::optional<VideoEncoder::ResolutionBitrateLimits>
kSinglecastLimits720pVp8 =
EncoderInfoSettings::GetDefaultSinglecastBitrateLimitsForResolution(
kVideoCodecVP8,
1280 * 720);
- const absl::optional<VideoEncoder::ResolutionBitrateLimits>
+ const std::optional<VideoEncoder::ResolutionBitrateLimits>
kSinglecastLimits360pVp9 =
EncoderInfoSettings::GetDefaultSinglecastBitrateLimitsForResolution(
kVideoCodecVP9,
640 * 360);
- const absl::optional<VideoEncoder::ResolutionBitrateLimits>
+ const std::optional<VideoEncoder::ResolutionBitrateLimits>
kSinglecastLimits720pVp9 =
EncoderInfoSettings::GetDefaultSinglecastBitrateLimitsForResolution(
kVideoCodecVP9,
@@ -78,7 +78,7 @@
protected:
struct TestParams {
bool active;
- absl::optional<ScalabilityMode> scalability_mode;
+ std::optional<ScalabilityMode> scalability_mode;
};
ScalingObserver(const std::string& payload_name,
const std::vector<TestParams>& test_params,
diff --git a/video/rate_utilization_tracker.cc b/video/rate_utilization_tracker.cc
index 9403ff1..ad7125a 100644
--- a/video/rate_utilization_tracker.cc
+++ b/video/rate_utilization_tracker.cc
@@ -63,10 +63,10 @@
CullOldData(time);
}
-absl::optional<double> RateUtilizationTracker::GetRateUtilizationFactor(
+std::optional<double> RateUtilizationTracker::GetRateUtilizationFactor(
Timestamp time) const {
if (data_points_.empty()) {
- return absl::nullopt;
+ return std::nullopt;
}
RTC_CHECK_GE(time, data_points_.back().time);
@@ -104,7 +104,7 @@
if (allocated_send_data_size.IsZero() && current_rate_.IsZero()) {
// No allocated rate across all of the data points, ignore.
- return absl::nullopt;
+ return std::nullopt;
}
// Calculate the rate past the very last data point until the polling time.
diff --git a/video/rate_utilization_tracker.h b/video/rate_utilization_tracker.h
index 23f4088..619d542 100644
--- a/video/rate_utilization_tracker.h
+++ b/video/rate_utilization_tracker.h
@@ -12,8 +12,8 @@
#define VIDEO_RATE_UTILIZATION_TRACKER_H_
#include <deque>
+#include <optional>
-#include "absl/types/optional.h"
#include "api/units/data_rate.h"
#include "api/units/data_size.h"
#include "api/units/time_delta.h"
@@ -46,7 +46,7 @@
// The timestamps used should never decrease relative the last one.
void OnDataRateChanged(DataRate rate, Timestamp time);
void OnDataProduced(DataSize size, Timestamp time);
- absl::optional<double> GetRateUtilizationFactor(Timestamp time) const;
+ std::optional<double> GetRateUtilizationFactor(Timestamp time) const;
private:
struct RateUsageUpdate {
diff --git a/video/receive_statistics_proxy.cc b/video/receive_statistics_proxy.cc
index 05dafbd..e74968b 100644
--- a/video/receive_statistics_proxy.cc
+++ b/video/receive_statistics_proxy.cc
@@ -96,7 +96,7 @@
}
void ReceiveStatisticsProxy::UpdateHistograms(
- absl::optional<int> fraction_lost,
+ std::optional<int> fraction_lost,
const StreamDataCounters& rtp_stats,
const StreamDataCounters* rtx_stats) {
RTC_DCHECK_RUN_ON(&main_thread_);
@@ -170,7 +170,7 @@
round(render_pixel_tracker_.ComputeTotalRate()));
}
- absl::optional<int> sync_offset_ms =
+ std::optional<int> sync_offset_ms =
sync_offset_counter_.Avg(kMinRequiredSamples);
if (sync_offset_ms) {
RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.AVSyncOffsetInMs",
@@ -197,18 +197,18 @@
<< key_frames_permille << '\n';
}
- absl::optional<int> qp = qp_counters_.vp8.Avg(kMinRequiredSamples);
+ std::optional<int> qp = qp_counters_.vp8.Avg(kMinRequiredSamples);
if (qp) {
RTC_HISTOGRAM_COUNTS_200("WebRTC.Video.Decoded.Vp8.Qp", *qp);
log_stream << "WebRTC.Video.Decoded.Vp8.Qp " << *qp << '\n';
}
- absl::optional<int> decode_ms = decode_time_counter_.Avg(kMinRequiredSamples);
+ std::optional<int> decode_ms = decode_time_counter_.Avg(kMinRequiredSamples);
if (decode_ms) {
RTC_HISTOGRAM_COUNTS_1000("WebRTC.Video.DecodeTimeInMs", *decode_ms);
log_stream << "WebRTC.Video.DecodeTimeInMs " << *decode_ms << '\n';
}
- absl::optional<int> jb_delay_ms =
+ std::optional<int> jb_delay_ms =
jitter_delay_counter_.Avg(kMinRequiredSamples);
if (jb_delay_ms) {
RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.JitterBufferDelayInMs",
@@ -216,21 +216,21 @@
log_stream << "WebRTC.Video.JitterBufferDelayInMs " << *jb_delay_ms << '\n';
}
- absl::optional<int> target_delay_ms =
+ std::optional<int> target_delay_ms =
target_delay_counter_.Avg(kMinRequiredSamples);
if (target_delay_ms) {
RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.TargetDelayInMs",
*target_delay_ms);
log_stream << "WebRTC.Video.TargetDelayInMs " << *target_delay_ms << '\n';
}
- absl::optional<int> current_delay_ms =
+ std::optional<int> current_delay_ms =
current_delay_counter_.Avg(kMinRequiredSamples);
if (current_delay_ms) {
RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.CurrentDelayInMs",
*current_delay_ms);
log_stream << "WebRTC.Video.CurrentDelayInMs " << *current_delay_ms << '\n';
}
- absl::optional<int> delay_ms = oneway_delay_counter_.Avg(kMinRequiredSamples);
+ std::optional<int> delay_ms = oneway_delay_counter_.Avg(kMinRequiredSamples);
if (delay_ms)
RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.OnewayDelayInMs", *delay_ms);
@@ -242,7 +242,7 @@
auto stats = it.second;
std::string uma_prefix = UmaPrefixForContentType(content_type);
- absl::optional<int> e2e_delay_ms =
+ std::optional<int> e2e_delay_ms =
stats.e2e_delay_counter.Avg(kMinRequiredSamples);
if (e2e_delay_ms) {
RTC_HISTOGRAM_COUNTS_SPARSE_10000(uma_prefix + ".EndToEndDelayInMs",
@@ -250,14 +250,14 @@
log_stream << uma_prefix << ".EndToEndDelayInMs"
<< " " << *e2e_delay_ms << '\n';
}
- absl::optional<int> e2e_delay_max_ms = stats.e2e_delay_counter.Max();
+ std::optional<int> e2e_delay_max_ms = stats.e2e_delay_counter.Max();
if (e2e_delay_max_ms && e2e_delay_ms) {
RTC_HISTOGRAM_COUNTS_SPARSE_100000(uma_prefix + ".EndToEndDelayMaxInMs",
*e2e_delay_max_ms);
log_stream << uma_prefix << ".EndToEndDelayMaxInMs"
<< " " << *e2e_delay_max_ms << '\n';
}
- absl::optional<int> interframe_delay_ms =
+ std::optional<int> interframe_delay_ms =
stats.interframe_delay_counter.Avg(kMinRequiredSamples);
if (interframe_delay_ms) {
RTC_HISTOGRAM_COUNTS_SPARSE_10000(uma_prefix + ".InterframeDelayInMs",
@@ -265,7 +265,7 @@
log_stream << uma_prefix << ".InterframeDelayInMs"
<< " " << *interframe_delay_ms << '\n';
}
- absl::optional<int> interframe_delay_max_ms =
+ std::optional<int> interframe_delay_max_ms =
stats.interframe_delay_counter.Max();
if (interframe_delay_max_ms && interframe_delay_ms) {
RTC_HISTOGRAM_COUNTS_SPARSE_10000(uma_prefix + ".InterframeDelayMaxInMs",
@@ -274,7 +274,7 @@
<< " " << *interframe_delay_max_ms << '\n';
}
- absl::optional<uint32_t> interframe_delay_95p_ms =
+ std::optional<uint32_t> interframe_delay_95p_ms =
stats.interframe_delay_percentiles.GetPercentile(0.95f);
if (interframe_delay_95p_ms && interframe_delay_ms != -1) {
RTC_HISTOGRAM_COUNTS_SPARSE_10000(
@@ -284,7 +284,7 @@
<< " " << *interframe_delay_95p_ms << '\n';
}
- absl::optional<int> width = stats.received_width.Avg(kMinRequiredSamples);
+ std::optional<int> width = stats.received_width.Avg(kMinRequiredSamples);
if (width) {
RTC_HISTOGRAM_COUNTS_SPARSE_10000(uma_prefix + ".ReceivedWidthInPixels",
*width);
@@ -292,7 +292,7 @@
<< " " << *width << '\n';
}
- absl::optional<int> height = stats.received_height.Avg(kMinRequiredSamples);
+ std::optional<int> height = stats.received_height.Avg(kMinRequiredSamples);
if (height) {
RTC_HISTOGRAM_COUNTS_SPARSE_10000(uma_prefix + ".ReceivedHeightInPixels",
*height);
@@ -300,7 +300,7 @@
<< " " << *height << '\n';
}
- absl::optional<double> corruption_score = stats.corruption_score.GetMean();
+ std::optional<double> corruption_score = stats.corruption_score.GetMean();
if (corruption_score) {
// Granularity level: 2e-3.
RTC_HISTOGRAM_COUNTS_SPARSE(uma_prefix + ".CorruptionLikelihoodPermille",
@@ -336,7 +336,7 @@
<< " " << key_frames_permille << '\n';
}
- absl::optional<int> qp = stats.qp_counter.Avg(kMinRequiredSamples);
+ std::optional<int> qp = stats.qp_counter.Avg(kMinRequiredSamples);
if (qp) {
RTC_HISTOGRAM_COUNTS_SPARSE_200(uma_prefix + ".Decoded.Vp8.Qp", *qp);
log_stream << uma_prefix << ".Decoded.Vp8.Qp"
@@ -409,13 +409,13 @@
stats_.network_frame_rate = static_cast<int>(framerate);
}
-absl::optional<int64_t>
+std::optional<int64_t>
ReceiveStatisticsProxy::GetCurrentEstimatedPlayoutNtpTimestampMs(
int64_t now_ms) const {
RTC_DCHECK_RUN_ON(&main_thread_);
if (!last_estimated_playout_ntp_timestamp_ms_ ||
!last_estimated_playout_time_ms_) {
- return absl::nullopt;
+ return std::nullopt;
}
int64_t elapsed_ms = now_ms - *last_estimated_playout_time_ms_;
return *last_estimated_playout_ntp_timestamp_ms_ + elapsed_ms;
@@ -590,7 +590,7 @@
}
void ReceiveStatisticsProxy::OnDecodedFrame(const VideoFrame& frame,
- absl::optional<uint8_t> qp,
+ std::optional<uint8_t> qp,
TimeDelta decode_time,
VideoContentType content_type,
VideoFrameType frame_type) {
@@ -626,7 +626,7 @@
void ReceiveStatisticsProxy::OnDecodedFrame(
const VideoFrameMetaData& frame_meta,
- absl::optional<uint8_t> qp,
+ std::optional<uint8_t> qp,
TimeDelta decode_time,
TimeDelta processing_delay,
TimeDelta assembly_time,
diff --git a/video/receive_statistics_proxy.h b/video/receive_statistics_proxy.h
index a678602..7af28fd 100644
--- a/video/receive_statistics_proxy.h
+++ b/video/receive_statistics_proxy.h
@@ -13,10 +13,10 @@
#include <map>
#include <memory>
+#include <optional>
#include <string>
#include <vector>
-#include "absl/types/optional.h"
#include "api/sequence_checker.h"
#include "api/task_queue/pending_task_safety_flag.h"
#include "api/task_queue/task_queue_base.h"
@@ -57,7 +57,7 @@
VideoReceiveStreamInterface::Stats GetStats() const;
void OnDecodedFrame(const VideoFrame& frame,
- absl::optional<uint8_t> qp,
+ std::optional<uint8_t> qp,
TimeDelta decode_time,
VideoContentType content_type,
VideoFrameType frame_type);
@@ -66,7 +66,7 @@
// above OnDecodedFrame method, which is called back on the thread where
// the actual decoding happens.
void OnDecodedFrame(const VideoFrameMetaData& frame_meta,
- absl::optional<uint8_t> qp,
+ std::optional<uint8_t> qp,
TimeDelta decode_time,
TimeDelta processing_delay,
TimeDelta assembly_time,
@@ -123,7 +123,7 @@
// Produce histograms. Must be called after DecoderThreadStopped(), typically
// at the end of the call.
- void UpdateHistograms(absl::optional<int> fraction_lost,
+ void UpdateHistograms(std::optional<int> fraction_lost,
const StreamDataCounters& rtp_stats,
const StreamDataCounters* rtx_stats);
@@ -153,7 +153,7 @@
// Removes info about old frames and then updates the framerate.
void UpdateFramerate(int64_t now_ms) const;
- absl::optional<int64_t> GetCurrentEstimatedPlayoutNtpTimestampMs(
+ std::optional<int64_t> GetCurrentEstimatedPlayoutNtpTimestampMs(
int64_t now_ms) const;
Clock* const clock_;
@@ -186,11 +186,11 @@
mutable std::map<int64_t, size_t> frame_window_ RTC_GUARDED_BY(main_thread_);
VideoContentType last_content_type_ RTC_GUARDED_BY(&main_thread_);
VideoCodecType last_codec_type_ RTC_GUARDED_BY(main_thread_);
- absl::optional<int64_t> first_frame_received_time_ms_
+ std::optional<int64_t> first_frame_received_time_ms_
RTC_GUARDED_BY(main_thread_);
- absl::optional<int64_t> first_decoded_frame_time_ms_
+ std::optional<int64_t> first_decoded_frame_time_ms_
RTC_GUARDED_BY(main_thread_);
- absl::optional<int64_t> last_decoded_frame_time_ms_
+ std::optional<int64_t> last_decoded_frame_time_ms_
RTC_GUARDED_BY(main_thread_);
size_t num_delayed_frames_rendered_ RTC_GUARDED_BY(main_thread_);
int64_t sum_missed_render_deadline_ms_ RTC_GUARDED_BY(main_thread_);
@@ -198,10 +198,10 @@
// called from const GetStats().
mutable rtc::MovingMaxCounter<TimingFrameInfo> timing_frame_info_counter_
RTC_GUARDED_BY(main_thread_);
- absl::optional<int> num_unique_frames_ RTC_GUARDED_BY(main_thread_);
- absl::optional<int64_t> last_estimated_playout_ntp_timestamp_ms_
+ std::optional<int> num_unique_frames_ RTC_GUARDED_BY(main_thread_);
+ std::optional<int64_t> last_estimated_playout_ntp_timestamp_ms_
RTC_GUARDED_BY(main_thread_);
- absl::optional<int64_t> last_estimated_playout_time_ms_
+ std::optional<int64_t> last_estimated_playout_time_ms_
RTC_GUARDED_BY(main_thread_);
// The thread on which this instance is constructed and some of its main
diff --git a/video/receive_statistics_proxy_unittest.cc b/video/receive_statistics_proxy_unittest.cc
index 4b5af07..22bed59 100644
--- a/video/receive_statistics_proxy_unittest.cc
+++ b/video/receive_statistics_proxy_unittest.cc
@@ -12,11 +12,11 @@
#include <limits>
#include <memory>
+#include <optional>
#include <string>
#include <tuple>
#include <utility>
-#include "absl/types/optional.h"
#include "api/scoped_refptr.h"
#include "api/units/frequency.h"
#include "api/units/time_delta.h"
@@ -63,7 +63,7 @@
return statistics_proxy_->GetStats();
}
- void FlushAndUpdateHistograms(absl::optional<int> fraction_lost,
+ void FlushAndUpdateHistograms(std::optional<int> fraction_lost,
const StreamDataCounters& rtp_stats,
const StreamDataCounters* rtx_stats) {
time_controller_.AdvanceTime(TimeDelta::Zero());
@@ -117,7 +117,7 @@
EXPECT_EQ(0u, statistics_proxy_->GetStats().frames_decoded);
webrtc::VideoFrame frame = CreateFrame(kWidth, kHeight);
for (uint32_t i = 1; i <= 3; ++i) {
- statistics_proxy_->OnDecodedFrame(frame, absl::nullopt, TimeDelta::Zero(),
+ statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Zero(),
VideoContentType::UNSPECIFIED,
VideoFrameType::kVideoFrameKey);
EXPECT_EQ(i, FlushAndGetStats().frames_decoded);
@@ -130,12 +130,12 @@
TimeDelta::Seconds(metrics::kMinRunTimeInSeconds) * kFps;
webrtc::VideoFrame frame = CreateFrame(kWidth, kHeight);
for (int i = 0; i < kRequiredSamples; ++i) {
- statistics_proxy_->OnDecodedFrame(frame, absl::nullopt, TimeDelta::Zero(),
+ statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Zero(),
VideoContentType::UNSPECIFIED,
VideoFrameType::kVideoFrameKey);
time_controller_.AdvanceTime(1 / kFps);
}
- FlushAndUpdateHistograms(absl::nullopt, StreamDataCounters(), nullptr);
+ FlushAndUpdateHistograms(std::nullopt, StreamDataCounters(), nullptr);
EXPECT_METRIC_EQ(1,
metrics::NumSamples("WebRTC.Video.DecodedFramesPerSecond"));
EXPECT_METRIC_EQ(1, metrics::NumEvents("WebRTC.Video.DecodedFramesPerSecond",
@@ -148,12 +148,12 @@
TimeDelta::Seconds(metrics::kMinRunTimeInSeconds) * kFps;
webrtc::VideoFrame frame = CreateFrame(kWidth, kHeight);
for (int i = 0; i < kRequiredSamples - 1; ++i) {
- statistics_proxy_->OnDecodedFrame(frame, absl::nullopt, TimeDelta::Zero(),
+ statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Zero(),
VideoContentType::UNSPECIFIED,
VideoFrameType::kVideoFrameKey);
time_controller_.AdvanceTime(1 / kFps);
}
- FlushAndUpdateHistograms(absl::nullopt, StreamDataCounters(), nullptr);
+ FlushAndUpdateHistograms(std::nullopt, StreamDataCounters(), nullptr);
EXPECT_METRIC_EQ(0,
metrics::NumSamples("WebRTC.Video.DecodedFramesPerSecond"));
}
@@ -165,9 +165,9 @@
TimeDelta expected_total_decode_time = TimeDelta::Zero();
unsigned int expected_frames_decoded = 0;
for (uint32_t i = 1; i <= 3; ++i) {
- statistics_proxy_->OnDecodedFrame(
- frame, absl::nullopt, TimeDelta::Millis(1),
- VideoContentType::UNSPECIFIED, VideoFrameType::kVideoFrameKey);
+ statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Millis(1),
+ VideoContentType::UNSPECIFIED,
+ VideoFrameType::kVideoFrameKey);
expected_total_decode_time += TimeDelta::Millis(1);
++expected_frames_decoded;
time_controller_.AdvanceTime(TimeDelta::Zero());
@@ -202,9 +202,9 @@
frame.set_packet_infos(RtpPacketInfos(packet_infos));
for (int i = 1; i <= 3; ++i) {
time_controller_.AdvanceTime(kProcessingDelay);
- statistics_proxy_->OnDecodedFrame(
- frame, absl::nullopt, TimeDelta::Millis(1),
- VideoContentType::UNSPECIFIED, VideoFrameType::kVideoFrameKey);
+ statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Millis(1),
+ VideoContentType::UNSPECIFIED,
+ VideoFrameType::kVideoFrameKey);
expected_total_processing_delay += i * kProcessingDelay;
++expected_frames_decoded;
time_controller_.AdvanceTime(TimeDelta::Zero());
@@ -239,7 +239,7 @@
RtpPacketInfos::vector_type single_packet_frame = {RtpPacketInfo(
/*ssrc=*/{}, /*csrcs=*/{}, /*rtp_timestamp=*/{}, /*receive_time=*/Now())};
frame.set_packet_infos(RtpPacketInfos(single_packet_frame));
- statistics_proxy_->OnDecodedFrame(frame, absl::nullopt, TimeDelta::Millis(1),
+ statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Millis(1),
VideoContentType::UNSPECIFIED,
VideoFrameType::kVideoFrameKey);
++expected_frames_decoded;
@@ -302,7 +302,7 @@
}
TEST_F(ReceiveStatisticsProxyTest, OnDecodedFrameIncreasesQpSum) {
- EXPECT_EQ(absl::nullopt, statistics_proxy_->GetStats().qp_sum);
+ EXPECT_EQ(std::nullopt, statistics_proxy_->GetStats().qp_sum);
webrtc::VideoFrame frame = CreateFrame(kWidth, kHeight);
statistics_proxy_->OnDecodedFrame(frame, 3u, TimeDelta::Zero(),
VideoContentType::UNSPECIFIED,
@@ -315,7 +315,7 @@
}
TEST_F(ReceiveStatisticsProxyTest, OnDecodedFrameIncreasesTotalDecodeTime) {
- EXPECT_EQ(absl::nullopt, statistics_proxy_->GetStats().qp_sum);
+ EXPECT_EQ(std::nullopt, statistics_proxy_->GetStats().qp_sum);
webrtc::VideoFrame frame = CreateFrame(kWidth, kHeight);
statistics_proxy_->OnDecodedFrame(frame, 3u, TimeDelta::Millis(4),
VideoContentType::UNSPECIFIED,
@@ -351,25 +351,25 @@
const TimeDelta kInterframeDelay2 = TimeDelta::Millis(200);
const TimeDelta kInterframeDelay3 = TimeDelta::Millis(100);
EXPECT_EQ(-1, statistics_proxy_->GetStats().interframe_delay_max_ms);
- statistics_proxy_->OnDecodedFrame(frame, absl::nullopt, TimeDelta::Zero(),
+ statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Zero(),
VideoContentType::UNSPECIFIED,
VideoFrameType::kVideoFrameKey);
EXPECT_EQ(-1, FlushAndGetStats().interframe_delay_max_ms);
time_controller_.AdvanceTime(kInterframeDelay1);
- statistics_proxy_->OnDecodedFrame(frame, absl::nullopt, TimeDelta::Zero(),
+ statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Zero(),
VideoContentType::UNSPECIFIED,
VideoFrameType::kVideoFrameDelta);
EXPECT_EQ(kInterframeDelay1.ms(), FlushAndGetStats().interframe_delay_max_ms);
time_controller_.AdvanceTime(kInterframeDelay2);
- statistics_proxy_->OnDecodedFrame(frame, absl::nullopt, TimeDelta::Zero(),
+ statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Zero(),
VideoContentType::UNSPECIFIED,
VideoFrameType::kVideoFrameDelta);
EXPECT_EQ(kInterframeDelay2.ms(), FlushAndGetStats().interframe_delay_max_ms);
time_controller_.AdvanceTime(kInterframeDelay3);
- statistics_proxy_->OnDecodedFrame(frame, absl::nullopt, TimeDelta::Zero(),
+ statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Zero(),
VideoContentType::UNSPECIFIED,
VideoFrameType::kVideoFrameDelta);
// kInterframeDelay3 is smaller than kInterframeDelay2.
@@ -382,26 +382,26 @@
const TimeDelta kInterframeDelay2 = TimeDelta::Millis(750);
const TimeDelta kInterframeDelay3 = TimeDelta::Millis(700);
EXPECT_EQ(-1, statistics_proxy_->GetStats().interframe_delay_max_ms);
- statistics_proxy_->OnDecodedFrame(frame, absl::nullopt, TimeDelta::Zero(),
+ statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Zero(),
VideoContentType::UNSPECIFIED,
VideoFrameType::kVideoFrameKey);
EXPECT_EQ(-1, FlushAndGetStats().interframe_delay_max_ms);
time_controller_.AdvanceTime(kInterframeDelay1);
- statistics_proxy_->OnDecodedFrame(frame, absl::nullopt, TimeDelta::Zero(),
+ statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Zero(),
VideoContentType::UNSPECIFIED,
VideoFrameType::kVideoFrameDelta);
EXPECT_EQ(kInterframeDelay1.ms(), FlushAndGetStats().interframe_delay_max_ms);
time_controller_.AdvanceTime(kInterframeDelay2);
- statistics_proxy_->OnDecodedFrame(frame, absl::nullopt, TimeDelta::Zero(),
+ statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Zero(),
VideoContentType::UNSPECIFIED,
VideoFrameType::kVideoFrameDelta);
// Still first delay is the maximum
EXPECT_EQ(kInterframeDelay1.ms(), FlushAndGetStats().interframe_delay_max_ms);
time_controller_.AdvanceTime(kInterframeDelay3);
- statistics_proxy_->OnDecodedFrame(frame, absl::nullopt, TimeDelta::Zero(),
+ statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Zero(),
VideoContentType::UNSPECIFIED,
VideoFrameType::kVideoFrameDelta);
// Now the first sample is out of the window, so the second is the maximum.
@@ -510,24 +510,24 @@
TEST_F(ReceiveStatisticsProxyTest, OnDecodedFrameWithoutQpQpSumWontExist) {
webrtc::VideoFrame frame = CreateFrame(kWidth, kHeight);
- EXPECT_EQ(absl::nullopt, statistics_proxy_->GetStats().qp_sum);
- statistics_proxy_->OnDecodedFrame(frame, absl::nullopt, TimeDelta::Zero(),
+ EXPECT_EQ(std::nullopt, statistics_proxy_->GetStats().qp_sum);
+ statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Zero(),
VideoContentType::UNSPECIFIED,
VideoFrameType::kVideoFrameKey);
- EXPECT_EQ(absl::nullopt, FlushAndGetStats().qp_sum);
+ EXPECT_EQ(std::nullopt, FlushAndGetStats().qp_sum);
}
TEST_F(ReceiveStatisticsProxyTest, OnDecodedFrameWithoutQpResetsQpSum) {
webrtc::VideoFrame frame = CreateFrame(kWidth, kHeight);
- EXPECT_EQ(absl::nullopt, statistics_proxy_->GetStats().qp_sum);
+ EXPECT_EQ(std::nullopt, statistics_proxy_->GetStats().qp_sum);
statistics_proxy_->OnDecodedFrame(frame, 3u, TimeDelta::Zero(),
VideoContentType::UNSPECIFIED,
VideoFrameType::kVideoFrameKey);
EXPECT_EQ(3u, FlushAndGetStats().qp_sum);
- statistics_proxy_->OnDecodedFrame(frame, absl::nullopt, TimeDelta::Zero(),
+ statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Zero(),
VideoContentType::UNSPECIFIED,
VideoFrameType::kVideoFrameDelta);
- EXPECT_EQ(absl::nullopt, FlushAndGetStats().qp_sum);
+ EXPECT_EQ(std::nullopt, FlushAndGetStats().qp_sum);
}
TEST_F(ReceiveStatisticsProxyTest, OnRenderedFrameIncreasesFramesRendered) {
@@ -540,8 +540,8 @@
}
TEST_F(ReceiveStatisticsProxyTest, GetStatsReportsOnCorruptionScore) {
- EXPECT_EQ(absl::nullopt, statistics_proxy_->GetStats().corruption_score_sum);
- EXPECT_EQ(absl::nullopt,
+ EXPECT_EQ(std::nullopt, statistics_proxy_->GetStats().corruption_score_sum);
+ EXPECT_EQ(std::nullopt,
statistics_proxy_->GetStats().corruption_score_squared_sum);
EXPECT_EQ(0u, statistics_proxy_->GetStats().corruption_score_count);
@@ -576,8 +576,8 @@
TEST_F(ReceiveStatisticsProxyTest, GetStatsReportsDecoderInfo) {
auto init_stats = statistics_proxy_->GetStats();
- EXPECT_EQ(init_stats.decoder_implementation_name, absl::nullopt);
- EXPECT_EQ(init_stats.power_efficient_decoder, absl::nullopt);
+ EXPECT_EQ(init_stats.decoder_implementation_name, std::nullopt);
+ EXPECT_EQ(init_stats.power_efficient_decoder, std::nullopt);
const VideoDecoder::DecoderInfo decoder_info{
.implementation_name = "decoderName", .is_hardware_accelerated = true};
@@ -682,12 +682,12 @@
const int kDeltaFrames = 22;
webrtc::VideoFrame frame = CreateFrame(kWidth, kHeight);
for (int i = 0; i < kKeyFrames; i++) {
- statistics_proxy_->OnDecodedFrame(frame, absl::nullopt, TimeDelta::Zero(),
+ statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Zero(),
VideoContentType::UNSPECIFIED,
VideoFrameType::kVideoFrameKey);
}
for (int i = 0; i < kDeltaFrames; i++) {
- statistics_proxy_->OnDecodedFrame(frame, absl::nullopt, TimeDelta::Zero(),
+ statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Zero(),
VideoContentType::UNSPECIFIED,
VideoFrameType::kVideoFrameDelta);
}
@@ -715,7 +715,7 @@
const int64_t kLongEndToEndDelay = 100;
const uint32_t kExpectedRtpTimestamp = 2;
TimingFrameInfo info;
- absl::optional<TimingFrameInfo> result;
+ std::optional<TimingFrameInfo> result;
info.rtp_timestamp = kExpectedRtpTimestamp - 1;
info.capture_time_ms = 0;
info.decode_finish_ms = kShortEndToEndDelay;
@@ -740,7 +740,7 @@
const uint32_t kExpectedRtpTimestamp = 2;
const TimeDelta kShortDelay = TimeDelta::Seconds(1);
const TimeDelta kLongDelay = TimeDelta::Seconds(10);
- absl::optional<TimingFrameInfo> result;
+ std::optional<TimingFrameInfo> result;
info.rtp_timestamp = kExpectedRtpTimestamp;
info.capture_time_ms = 0;
info.decode_finish_ms = kShortEndToEndDelay;
@@ -762,11 +762,11 @@
webrtc::VideoFrame frame = CreateFrame(kWidth, kHeight);
statistics_proxy_->OnCompleteFrame(true, 1000, VideoContentType::UNSPECIFIED);
statistics_proxy_->OnDecodedFrame(
- frame, absl::nullopt, TimeDelta::Millis(1000),
+ frame, std::nullopt, TimeDelta::Millis(1000),
VideoContentType::UNSPECIFIED, VideoFrameType::kVideoFrameKey);
FlushAndGetStats();
- statistics_proxy_->UpdateHistograms(absl::nullopt, StreamDataCounters(),
+ statistics_proxy_->UpdateHistograms(std::nullopt, StreamDataCounters(),
nullptr);
EXPECT_METRIC_EQ(
1, metrics::NumSamples("WebRTC.Video.ReceiveStreamLifetimeInSeconds"));
@@ -780,7 +780,7 @@
const TimeDelta kLifetime = TimeDelta::Seconds(3);
time_controller_.AdvanceTime(kLifetime);
// No frames received.
- statistics_proxy_->UpdateHistograms(absl::nullopt, StreamDataCounters(),
+ statistics_proxy_->UpdateHistograms(std::nullopt, StreamDataCounters(),
nullptr);
EXPECT_METRIC_EQ(
0, metrics::NumSamples("WebRTC.Video.ReceiveStreamLifetimeInSeconds"));
@@ -808,7 +808,7 @@
const int64_t kVideoNtpMs = 21;
const int64_t kSyncOffsetMs = 22;
const double kFreqKhz = 90.0;
- EXPECT_EQ(absl::nullopt,
+ EXPECT_EQ(std::nullopt,
statistics_proxy_->GetStats().estimated_playout_ntp_timestamp_ms);
statistics_proxy_->OnSyncOffsetUpdated(kVideoNtpMs, kSyncOffsetMs, kFreqKhz);
EXPECT_EQ(kVideoNtpMs, FlushAndGetStats().estimated_playout_ntp_timestamp_ms);
@@ -838,7 +838,7 @@
statistics_proxy_->OnSyncOffsetUpdated(kVideoNtpMs, kSyncOffsetMs,
kFreqKhz);
}
- FlushAndUpdateHistograms(absl::nullopt, StreamDataCounters(), nullptr);
+ FlushAndUpdateHistograms(std::nullopt, StreamDataCounters(), nullptr);
EXPECT_METRIC_EQ(1, metrics::NumSamples("WebRTC.Video.AVSyncOffsetInMs"));
EXPECT_METRIC_EQ(
1, metrics::NumEvents("WebRTC.Video.AVSyncOffsetInMs", kSyncOffsetMs));
@@ -862,7 +862,7 @@
time_controller_.AdvanceTime(kFreqOffsetProcessInterval);
//) Process interval passed, max diff: 4.
statistics_proxy_->OnSyncOffsetUpdated(kVideoNtpMs, kSyncOffsetMs, kFreqKhz);
- FlushAndUpdateHistograms(absl::nullopt, StreamDataCounters(), nullptr);
+ FlushAndUpdateHistograms(std::nullopt, StreamDataCounters(), nullptr);
// Average reported: (2 + 4) / 2 = 3.
EXPECT_METRIC_EQ(1,
metrics::NumSamples("WebRTC.Video.RtpToNtpFreqOffsetInKhz"));
@@ -876,7 +876,7 @@
for (int i = 0; i < kMinRequiredSamples; ++i)
statistics_proxy_->OnPreDecode(kVideoCodecVP8, kQp);
- FlushAndUpdateHistograms(absl::nullopt, StreamDataCounters(), nullptr);
+ FlushAndUpdateHistograms(std::nullopt, StreamDataCounters(), nullptr);
EXPECT_METRIC_EQ(1, metrics::NumSamples("WebRTC.Video.Decoded.Vp8.Qp"));
EXPECT_METRIC_EQ(1, metrics::NumEvents("WebRTC.Video.Decoded.Vp8.Qp", kQp));
}
@@ -887,7 +887,7 @@
for (int i = 0; i < kMinRequiredSamples - 1; ++i)
statistics_proxy_->OnPreDecode(kVideoCodecVP8, kQp);
- statistics_proxy_->UpdateHistograms(absl::nullopt, StreamDataCounters(),
+ statistics_proxy_->UpdateHistograms(std::nullopt, StreamDataCounters(),
nullptr);
EXPECT_METRIC_EQ(0, metrics::NumSamples("WebRTC.Video.Decoded.Vp8.Qp"));
}
@@ -896,7 +896,7 @@
for (int i = 0; i < kMinRequiredSamples; ++i)
statistics_proxy_->OnPreDecode(kVideoCodecVP8, -1);
- statistics_proxy_->UpdateHistograms(absl::nullopt, StreamDataCounters(),
+ statistics_proxy_->UpdateHistograms(std::nullopt, StreamDataCounters(),
nullptr);
EXPECT_METRIC_EQ(0, metrics::NumSamples("WebRTC.Video.Decoded.Vp8.Qp"));
}
@@ -911,7 +911,7 @@
statistics_proxy_->OnCompleteFrame(kIsKeyFrame, kFrameSizeBytes,
VideoContentType::UNSPECIFIED);
statistics_proxy_->OnDecodedFrame(
- frame, absl::nullopt, TimeDelta::Millis(1000),
+ frame, std::nullopt, TimeDelta::Millis(1000),
VideoContentType::UNSPECIFIED, VideoFrameType::kVideoFrameDelta);
}
FlushAndGetStats();
@@ -920,7 +920,7 @@
EXPECT_EQ(kMinRequiredSamples - 1,
statistics_proxy_->GetStats().frame_counts.delta_frames);
- statistics_proxy_->UpdateHistograms(absl::nullopt, StreamDataCounters(),
+ statistics_proxy_->UpdateHistograms(std::nullopt, StreamDataCounters(),
nullptr);
EXPECT_METRIC_EQ(
0, metrics::NumSamples("WebRTC.Video.KeyFramesReceivedInPermille"));
@@ -936,7 +936,7 @@
statistics_proxy_->OnCompleteFrame(kIsKeyFrame, kFrameSizeBytes,
VideoContentType::UNSPECIFIED);
statistics_proxy_->OnDecodedFrame(
- frame, absl::nullopt, TimeDelta::Millis(1000),
+ frame, std::nullopt, TimeDelta::Millis(1000),
VideoContentType::UNSPECIFIED, VideoFrameType::kVideoFrameDelta);
}
FlushAndGetStats();
@@ -945,7 +945,7 @@
EXPECT_EQ(kMinRequiredSamples,
statistics_proxy_->GetStats().frame_counts.delta_frames);
- statistics_proxy_->UpdateHistograms(absl::nullopt, StreamDataCounters(),
+ statistics_proxy_->UpdateHistograms(std::nullopt, StreamDataCounters(),
nullptr);
EXPECT_METRIC_EQ(
1, metrics::NumSamples("WebRTC.Video.KeyFramesReceivedInPermille"));
@@ -961,14 +961,14 @@
statistics_proxy_->OnCompleteFrame(true, kFrameSizeBytes,
VideoContentType::UNSPECIFIED);
statistics_proxy_->OnDecodedFrame(
- frame, absl::nullopt, TimeDelta::Millis(1000),
+ frame, std::nullopt, TimeDelta::Millis(1000),
VideoContentType::UNSPECIFIED, VideoFrameType::kVideoFrameKey);
}
for (int i = 0; i < kMinRequiredSamples; ++i) {
statistics_proxy_->OnCompleteFrame(false, kFrameSizeBytes,
VideoContentType::UNSPECIFIED);
statistics_proxy_->OnDecodedFrame(
- frame, absl::nullopt, TimeDelta::Millis(1000),
+ frame, std::nullopt, TimeDelta::Millis(1000),
VideoContentType::UNSPECIFIED, VideoFrameType::kVideoFrameDelta);
}
FlushAndGetStats();
@@ -978,7 +978,7 @@
EXPECT_EQ(kMinRequiredSamples,
statistics_proxy_->GetStats().frame_counts.delta_frames);
- statistics_proxy_->UpdateHistograms(absl::nullopt, StreamDataCounters(),
+ statistics_proxy_->UpdateHistograms(std::nullopt, StreamDataCounters(),
nullptr);
EXPECT_METRIC_EQ(
1, metrics::NumSamples("WebRTC.Video.KeyFramesReceivedInPermille"));
@@ -1000,7 +1000,7 @@
kMinPlayoutDelayMs, kRenderDelayMs);
}
- statistics_proxy_->UpdateHistograms(absl::nullopt, StreamDataCounters(),
+ statistics_proxy_->UpdateHistograms(std::nullopt, StreamDataCounters(),
nullptr);
EXPECT_METRIC_EQ(0, metrics::NumSamples("WebRTC.Video.DecodeTimeInMs"));
EXPECT_METRIC_EQ(0,
@@ -1024,7 +1024,7 @@
kMinPlayoutDelayMs, kRenderDelayMs);
}
- FlushAndUpdateHistograms(absl::nullopt, StreamDataCounters(), nullptr);
+ FlushAndUpdateHistograms(std::nullopt, StreamDataCounters(), nullptr);
EXPECT_METRIC_EQ(1,
metrics::NumSamples("WebRTC.Video.JitterBufferDelayInMs"));
EXPECT_METRIC_EQ(1, metrics::NumSamples("WebRTC.Video.TargetDelayInMs"));
@@ -1050,7 +1050,7 @@
// i.e. bad
frame.set_ntp_time_ms(
time_controller_.GetClock()->CurrentNtpInMilliseconds());
- statistics_proxy_->OnDecodedFrame(frame, absl::nullopt, TimeDelta::Zero(),
+ statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Zero(),
VideoContentType::UNSPECIFIED,
VideoFrameType::kVideoFrameKey);
statistics_proxy_->OnRenderedFrame(MetaData(frame));
@@ -1088,7 +1088,7 @@
statistics_proxy_->OnRenderedFrame(MetaData(CreateFrame(kWidth, kHeight)));
}
- statistics_proxy_->UpdateHistograms(absl::nullopt, StreamDataCounters(),
+ statistics_proxy_->UpdateHistograms(std::nullopt, StreamDataCounters(),
nullptr);
EXPECT_METRIC_EQ(0,
metrics::NumSamples("WebRTC.Video.ReceivedWidthInPixels"));
@@ -1105,7 +1105,7 @@
statistics_proxy_->OnRenderedFrame(MetaData(CreateFrame(kWidth, kHeight)));
}
- statistics_proxy_->UpdateHistograms(absl::nullopt, StreamDataCounters(),
+ statistics_proxy_->UpdateHistograms(std::nullopt, StreamDataCounters(),
nullptr);
EXPECT_METRIC_EQ(1,
metrics::NumSamples("WebRTC.Video.ReceivedWidthInPixels"));
@@ -1123,7 +1123,7 @@
TEST_F(ReceiveStatisticsProxyTest, ZeroDelayReportedIfFrameNotDelayed) {
webrtc::VideoFrame frame = CreateFrame(kWidth, kHeight);
- statistics_proxy_->OnDecodedFrame(frame, absl::nullopt, TimeDelta::Zero(),
+ statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Zero(),
VideoContentType::UNSPECIFIED,
VideoFrameType::kVideoFrameKey);
@@ -1134,7 +1134,7 @@
// Min run time has passed.
time_controller_.AdvanceTime(
TimeDelta::Seconds((metrics::kMinRunTimeInSeconds)));
- FlushAndUpdateHistograms(absl::nullopt, StreamDataCounters(), nullptr);
+ FlushAndUpdateHistograms(std::nullopt, StreamDataCounters(), nullptr);
EXPECT_METRIC_EQ(1,
metrics::NumSamples("WebRTC.Video.DelayedFramesToRenderer"));
EXPECT_METRIC_EQ(
@@ -1146,7 +1146,7 @@
TEST_F(ReceiveStatisticsProxyTest,
DelayedFrameHistogramsAreNotUpdatedIfMinRuntimeHasNotPassed) {
webrtc::VideoFrame frame = CreateFrame(kWidth, kHeight);
- statistics_proxy_->OnDecodedFrame(frame, absl::nullopt, TimeDelta::Zero(),
+ statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Zero(),
VideoContentType::UNSPECIFIED,
VideoFrameType::kVideoFrameKey);
@@ -1157,7 +1157,7 @@
// Min run time has not passed.
time_controller_.AdvanceTime(
TimeDelta::Seconds(metrics::kMinRunTimeInSeconds) - TimeDelta::Millis(1));
- statistics_proxy_->UpdateHistograms(absl::nullopt, StreamDataCounters(),
+ statistics_proxy_->UpdateHistograms(std::nullopt, StreamDataCounters(),
nullptr);
EXPECT_METRIC_EQ(0,
metrics::NumSamples("WebRTC.Video.DelayedFramesToRenderer"));
@@ -1168,14 +1168,14 @@
TEST_F(ReceiveStatisticsProxyTest,
DelayedFramesHistogramsAreNotUpdatedIfNoRenderedFrames) {
webrtc::VideoFrame frame = CreateFrame(kWidth, kHeight);
- statistics_proxy_->OnDecodedFrame(frame, absl::nullopt, TimeDelta::Zero(),
+ statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Zero(),
VideoContentType::UNSPECIFIED,
VideoFrameType::kVideoFrameKey);
// Min run time has passed. No rendered frames.
time_controller_.AdvanceTime(
TimeDelta::Seconds((metrics::kMinRunTimeInSeconds)));
- statistics_proxy_->UpdateHistograms(absl::nullopt, StreamDataCounters(),
+ statistics_proxy_->UpdateHistograms(std::nullopt, StreamDataCounters(),
nullptr);
EXPECT_METRIC_EQ(0,
metrics::NumSamples("WebRTC.Video.DelayedFramesToRenderer"));
@@ -1185,7 +1185,7 @@
TEST_F(ReceiveStatisticsProxyTest, DelayReportedIfFrameIsDelayed) {
webrtc::VideoFrame frame = CreateFrame(kWidth, kHeight);
- statistics_proxy_->OnDecodedFrame(frame, absl::nullopt, TimeDelta::Zero(),
+ statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Zero(),
VideoContentType::UNSPECIFIED,
VideoFrameType::kVideoFrameKey);
@@ -1196,7 +1196,7 @@
// Min run time has passed.
time_controller_.AdvanceTime(
TimeDelta::Seconds(metrics::kMinRunTimeInSeconds));
- FlushAndUpdateHistograms(absl::nullopt, StreamDataCounters(), nullptr);
+ FlushAndUpdateHistograms(std::nullopt, StreamDataCounters(), nullptr);
EXPECT_METRIC_EQ(1,
metrics::NumSamples("WebRTC.Video.DelayedFramesToRenderer"));
EXPECT_METRIC_EQ(
@@ -1210,7 +1210,7 @@
TEST_F(ReceiveStatisticsProxyTest, AverageDelayOfDelayedFramesIsReported) {
webrtc::VideoFrame frame = CreateFrame(kWidth, kHeight);
- statistics_proxy_->OnDecodedFrame(frame, absl::nullopt, TimeDelta::Zero(),
+ statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Zero(),
VideoContentType::UNSPECIFIED,
VideoFrameType::kVideoFrameKey);
@@ -1229,7 +1229,7 @@
// Min run time has passed.
time_controller_.AdvanceTime(
TimeDelta::Seconds(metrics::kMinRunTimeInSeconds));
- FlushAndUpdateHistograms(absl::nullopt, StreamDataCounters(), nullptr);
+ FlushAndUpdateHistograms(std::nullopt, StreamDataCounters(), nullptr);
EXPECT_METRIC_EQ(1,
metrics::NumSamples("WebRTC.Video.DelayedFramesToRenderer"));
EXPECT_METRIC_EQ(
@@ -1252,7 +1252,7 @@
RtcpPacketTypeCounter counter;
statistics_proxy_->RtcpPacketTypesCounterUpdated(kRemoteSsrc, counter);
- statistics_proxy_->UpdateHistograms(absl::nullopt, data_counters, nullptr);
+ statistics_proxy_->UpdateHistograms(std::nullopt, data_counters, nullptr);
EXPECT_METRIC_EQ(0,
metrics::NumSamples("WebRTC.Video.FirPacketsSentPerMinute"));
EXPECT_METRIC_EQ(0,
@@ -1277,7 +1277,7 @@
counter.nack_packets = kNackPackets;
statistics_proxy_->RtcpPacketTypesCounterUpdated(kRemoteSsrc, counter);
- statistics_proxy_->UpdateHistograms(absl::nullopt, data_counters, nullptr);
+ statistics_proxy_->UpdateHistograms(std::nullopt, data_counters, nullptr);
EXPECT_METRIC_EQ(1,
metrics::NumSamples("WebRTC.Video.FirPacketsSentPerMinute"));
EXPECT_METRIC_EQ(1,
@@ -1364,18 +1364,18 @@
webrtc::VideoFrame frame = CreateFrame(kWidth, kHeight);
for (int i = 0; i < kMinRequiredSamples; ++i) {
- statistics_proxy_->OnDecodedFrame(frame, absl::nullopt, TimeDelta::Zero(),
+ statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Zero(),
content_type_,
VideoFrameType::kVideoFrameKey);
time_controller_.AdvanceTime(kInterFrameDelay);
}
// One extra with double the interval.
time_controller_.AdvanceTime(kInterFrameDelay);
- statistics_proxy_->OnDecodedFrame(frame, absl::nullopt, TimeDelta::Zero(),
+ statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Zero(),
content_type_,
VideoFrameType::kVideoFrameDelta);
- FlushAndUpdateHistograms(absl::nullopt, StreamDataCounters(), nullptr);
+ FlushAndUpdateHistograms(std::nullopt, StreamDataCounters(), nullptr);
const TimeDelta kExpectedInterFrame =
(kInterFrameDelay * (kMinRequiredSamples - 1) + kInterFrameDelay * 2) /
kMinRequiredSamples;
@@ -1402,24 +1402,24 @@
for (int i = 0; i <= kMinRequiredSamples - kLastFivePercentsSamples; ++i) {
time_controller_.AdvanceTime(kInterFrameDelay);
- statistics_proxy_->OnDecodedFrame(frame, absl::nullopt, TimeDelta::Zero(),
+ statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Zero(),
content_type_,
VideoFrameType::kVideoFrameKey);
}
// Last 5% of intervals are double in size.
for (int i = 0; i < kLastFivePercentsSamples; ++i) {
time_controller_.AdvanceTime(2 * kInterFrameDelay);
- statistics_proxy_->OnDecodedFrame(frame, absl::nullopt, TimeDelta::Zero(),
+ statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Zero(),
content_type_,
VideoFrameType::kVideoFrameKey);
}
// Final sample is outlier and 10 times as big.
time_controller_.AdvanceTime(10 * kInterFrameDelay);
- statistics_proxy_->OnDecodedFrame(frame, absl::nullopt, TimeDelta::Zero(),
+ statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Zero(),
content_type_,
VideoFrameType::kVideoFrameKey);
- FlushAndUpdateHistograms(absl::nullopt, StreamDataCounters(), nullptr);
+ FlushAndUpdateHistograms(std::nullopt, StreamDataCounters(), nullptr);
const TimeDelta kExpectedInterFrame = kInterFrameDelay * 2;
if (videocontenttypehelpers::IsScreenshare(content_type_)) {
EXPECT_METRIC_EQ(
@@ -1439,7 +1439,7 @@
webrtc::VideoFrame frame = CreateFrame(kWidth, kHeight);
for (int i = 0; i < kMinRequiredSamples; ++i) {
- statistics_proxy_->OnDecodedFrame(frame, absl::nullopt, TimeDelta::Zero(),
+ statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Zero(),
content_type_,
VideoFrameType::kVideoFrameKey);
time_controller_.AdvanceTime(kInterFrameDelay);
@@ -1447,7 +1447,7 @@
// `kMinRequiredSamples` samples, and thereby intervals, is required. That
// means we're one frame short of having a valid data set.
- statistics_proxy_->UpdateHistograms(absl::nullopt, StreamDataCounters(),
+ statistics_proxy_->UpdateHistograms(std::nullopt, StreamDataCounters(),
nullptr);
EXPECT_METRIC_EQ(0, metrics::NumSamples("WebRTC.Video.InterframeDelayInMs"));
EXPECT_METRIC_EQ(0,
@@ -1463,7 +1463,7 @@
webrtc::VideoFrame frame = CreateFrame(kWidth, kHeight);
for (int i = 0; i <= kMinRequiredSamples; ++i) {
- statistics_proxy_->OnDecodedFrame(frame, absl::nullopt, TimeDelta::Zero(),
+ statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Zero(),
content_type_,
VideoFrameType::kVideoFrameKey);
time_controller_.AdvanceTime(kInterFrameDelay);
@@ -1475,15 +1475,15 @@
time_controller_.AdvanceTime(TimeDelta::Seconds(5));
// Insert two more frames. The interval during the pause should be
// disregarded in the stats.
- statistics_proxy_->OnDecodedFrame(frame, absl::nullopt, TimeDelta::Zero(),
+ statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Zero(),
content_type_,
VideoFrameType::kVideoFrameKey);
time_controller_.AdvanceTime(kInterFrameDelay);
- statistics_proxy_->OnDecodedFrame(frame, absl::nullopt, TimeDelta::Zero(),
+ statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Zero(),
content_type_,
VideoFrameType::kVideoFrameDelta);
- FlushAndUpdateHistograms(absl::nullopt, StreamDataCounters(), nullptr);
+ FlushAndUpdateHistograms(std::nullopt, StreamDataCounters(), nullptr);
if (videocontenttypehelpers::IsScreenshare(content_type_)) {
EXPECT_METRIC_EQ(
1, metrics::NumSamples("WebRTC.Video.Screenshare.InterframeDelayInMs"));
@@ -1519,7 +1519,7 @@
FlushAndGetStats();
EXPECT_EQ(3u, statistics_proxy_->GetStats().corruption_score_count);
- statistics_proxy_->UpdateHistograms(absl::nullopt, StreamDataCounters(),
+ statistics_proxy_->UpdateHistograms(std::nullopt, StreamDataCounters(),
nullptr);
if (videocontenttypehelpers::IsScreenshare(content_type_)) {
EXPECT_METRIC_EQ(
@@ -1553,7 +1553,7 @@
for (int i = 0; i < kMinRequiredSamples; ++i) {
VideoFrameMetaData meta = MetaData(frame);
statistics_proxy_->OnDecodedFrame(
- meta, absl::nullopt, TimeDelta::Zero(), TimeDelta::Zero(),
+ meta, std::nullopt, TimeDelta::Zero(), TimeDelta::Zero(),
TimeDelta::Zero(), content_type_, VideoFrameType::kVideoFrameKey);
statistics_proxy_->OnRenderedFrame(meta);
time_controller_.AdvanceTime(kInterFrameDelay);
@@ -1562,11 +1562,11 @@
time_controller_.AdvanceTime(kFreezeDelay);
VideoFrameMetaData meta = MetaData(frame);
statistics_proxy_->OnDecodedFrame(
- meta, absl::nullopt, TimeDelta::Zero(), TimeDelta::Zero(),
+ meta, std::nullopt, TimeDelta::Zero(), TimeDelta::Zero(),
TimeDelta::Zero(), content_type_, VideoFrameType::kVideoFrameDelta);
statistics_proxy_->OnRenderedFrame(meta);
- FlushAndUpdateHistograms(absl::nullopt, StreamDataCounters(), nullptr);
+ FlushAndUpdateHistograms(std::nullopt, StreamDataCounters(), nullptr);
const TimeDelta kExpectedTimeBetweenFreezes =
kInterFrameDelay * (kMinRequiredSamples - 1);
const int kExpectedNumberFreezesPerMinute = 60 / kCallDuration.seconds();
@@ -1601,7 +1601,7 @@
for (int i = 0; i < kMinRequiredSamples; ++i) {
time_controller_.AdvanceTime(kFrameDuration);
- statistics_proxy_->OnDecodedFrame(frame, absl::nullopt, TimeDelta::Zero(),
+ statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Zero(),
content_type_,
VideoFrameType::kVideoFrameKey);
statistics_proxy_->OnRenderedFrame(MetaData(frame));
@@ -1610,7 +1610,7 @@
// Freezes and pauses should be included into harmonic frame rate.
// Add freeze.
time_controller_.AdvanceTime(kFreezeDuration);
- statistics_proxy_->OnDecodedFrame(frame, absl::nullopt, TimeDelta::Zero(),
+ statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Zero(),
content_type_,
VideoFrameType::kVideoFrameDelta);
statistics_proxy_->OnRenderedFrame(MetaData(frame));
@@ -1618,12 +1618,12 @@
// Add pause.
time_controller_.AdvanceTime(kPauseDuration);
statistics_proxy_->OnStreamInactive();
- statistics_proxy_->OnDecodedFrame(frame, absl::nullopt, TimeDelta::Zero(),
+ statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Zero(),
content_type_,
VideoFrameType::kVideoFrameDelta);
statistics_proxy_->OnRenderedFrame(MetaData(frame));
- FlushAndUpdateHistograms(absl::nullopt, StreamDataCounters(), nullptr);
+ FlushAndUpdateHistograms(std::nullopt, StreamDataCounters(), nullptr);
double kSumSquaredFrameDurationSecs =
(kMinRequiredSamples - 1) *
(kFrameDuration.seconds<double>() * kFrameDuration.seconds<double>());
@@ -1651,7 +1651,7 @@
for (int i = 0; i <= kMinRequiredSamples; ++i) {
VideoFrameMetaData meta = MetaData(frame);
statistics_proxy_->OnDecodedFrame(
- meta, absl::nullopt, TimeDelta::Zero(), TimeDelta::Zero(),
+ meta, std::nullopt, TimeDelta::Zero(), TimeDelta::Zero(),
TimeDelta::Zero(), content_type_, VideoFrameType::kVideoFrameKey);
statistics_proxy_->OnRenderedFrame(meta);
time_controller_.AdvanceTime(kInterFrameDelay);
@@ -1663,13 +1663,13 @@
for (int i = 0; i <= kMinRequiredSamples * 3; ++i) {
VideoFrameMetaData meta = MetaData(frame);
statistics_proxy_->OnDecodedFrame(
- meta, absl::nullopt, TimeDelta::Zero(), TimeDelta::Zero(),
+ meta, std::nullopt, TimeDelta::Zero(), TimeDelta::Zero(),
TimeDelta::Zero(), content_type_, VideoFrameType::kVideoFrameDelta);
statistics_proxy_->OnRenderedFrame(meta);
time_controller_.AdvanceTime(kInterFrameDelay);
}
- FlushAndUpdateHistograms(absl::nullopt, StreamDataCounters(), nullptr);
+ FlushAndUpdateHistograms(std::nullopt, StreamDataCounters(), nullptr);
// Average of two playback intervals.
const TimeDelta kExpectedTimeBetweenFreezes =
kInterFrameDelay * kMinRequiredSamples * 2;
@@ -1694,19 +1694,19 @@
webrtc::VideoFrame frame = CreateFrame(kWidth, kHeight);
for (int i = 0; i <= kMinRequiredSamples; ++i) {
- statistics_proxy_->OnDecodedFrame(frame, absl::nullopt, TimeDelta::Zero(),
+ statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Zero(),
content_type_,
VideoFrameType::kVideoFrameKey);
time_controller_.AdvanceTime(kInterFrameDelay);
statistics_proxy_->OnStreamInactive();
time_controller_.AdvanceTime(kPauseDuration);
- statistics_proxy_->OnDecodedFrame(frame, absl::nullopt, TimeDelta::Zero(),
+ statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Zero(),
content_type_,
VideoFrameType::kVideoFrameDelta);
time_controller_.AdvanceTime(kInterFrameDelay);
}
- statistics_proxy_->UpdateHistograms(absl::nullopt, StreamDataCounters(),
+ statistics_proxy_->UpdateHistograms(std::nullopt, StreamDataCounters(),
nullptr);
// No freezes should be detected, as all long inter-frame delays were
// pauses.
@@ -1728,7 +1728,7 @@
for (int i = 0; i < kMinRequiredSamples; ++i) {
VideoFrameMetaData meta = MetaData(frame_hd);
statistics_proxy_->OnDecodedFrame(
- meta, absl::nullopt, TimeDelta::Zero(), TimeDelta::Zero(),
+ meta, std::nullopt, TimeDelta::Zero(), TimeDelta::Zero(),
TimeDelta::Zero(), content_type_, VideoFrameType::kVideoFrameKey);
statistics_proxy_->OnRenderedFrame(meta);
time_controller_.AdvanceTime(kInterFrameDelay);
@@ -1737,7 +1737,7 @@
for (int i = 0; i < 2 * kMinRequiredSamples; ++i) {
VideoFrameMetaData meta = MetaData(frame_sd);
statistics_proxy_->OnDecodedFrame(
- meta, absl::nullopt, TimeDelta::Zero(), TimeDelta::Zero(),
+ meta, std::nullopt, TimeDelta::Zero(), TimeDelta::Zero(),
TimeDelta::Zero(), content_type_, VideoFrameType::kVideoFrameKey);
statistics_proxy_->OnRenderedFrame(meta);
time_controller_.AdvanceTime(kInterFrameDelay);
@@ -1745,7 +1745,7 @@
// Extra last frame.
statistics_proxy_->OnRenderedFrame(MetaData(frame_sd));
- statistics_proxy_->UpdateHistograms(absl::nullopt, StreamDataCounters(),
+ statistics_proxy_->UpdateHistograms(std::nullopt, StreamDataCounters(),
nullptr);
const int kExpectedTimeInHdPercents = 33;
if (videocontenttypehelpers::IsScreenshare(content_type_)) {
@@ -1788,7 +1788,7 @@
VideoFrameType::kVideoFrameKey);
statistics_proxy_->OnRenderedFrame(MetaData(frame));
- FlushAndUpdateHistograms(absl::nullopt, StreamDataCounters(), nullptr);
+ FlushAndUpdateHistograms(std::nullopt, StreamDataCounters(), nullptr);
const int kExpectedTimeInHdPercents = 66;
if (videocontenttypehelpers::IsScreenshare(content_type_)) {
EXPECT_METRIC_EQ(
@@ -1811,7 +1811,7 @@
webrtc::VideoFrame frame_ld = CreateFrame(320, 180);
// Call once to pass content type.
- statistics_proxy_->OnDecodedFrame(frame_hd, absl::nullopt, TimeDelta::Zero(),
+ statistics_proxy_->OnDecodedFrame(frame_hd, std::nullopt, TimeDelta::Zero(),
content_type_,
VideoFrameType::kVideoFrameKey);
@@ -1824,7 +1824,7 @@
// Downscale.
statistics_proxy_->OnRenderedFrame(MetaData(frame_ld));
time_controller_.AdvanceTime(kInterFrameDelay);
- statistics_proxy_->UpdateHistograms(absl::nullopt, StreamDataCounters(),
+ statistics_proxy_->UpdateHistograms(std::nullopt, StreamDataCounters(),
nullptr);
const int kExpectedDownscales = 30; // 2 per 4 seconds = 30 per minute.
if (!videocontenttypehelpers::IsScreenshare(content_type_)) {
@@ -1846,7 +1846,7 @@
VideoFrameType::kVideoFrameKey);
time_controller_.AdvanceTime(kInterFrameDelay);
}
- FlushAndUpdateHistograms(absl::nullopt, StreamDataCounters(), nullptr);
+ FlushAndUpdateHistograms(std::nullopt, StreamDataCounters(), nullptr);
EXPECT_METRIC_EQ(
1, metrics::NumEvents("WebRTC.Video.DecodeTimeInMs", kDecodeTime.ms()));
}
diff --git a/video/render/BUILD.gn b/video/render/BUILD.gn
index 7940187..6b23583 100644
--- a/video/render/BUILD.gn
+++ b/video/render/BUILD.gn
@@ -26,7 +26,6 @@
"../../rtc_base:event_tracer",
"../../rtc_base:macromagic",
"../../rtc_base:race_checker",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -44,6 +43,5 @@
"../../rtc_base:logging",
"../../rtc_base:timeutils",
"../../system_wrappers:metrics",
- "//third_party/abseil-cpp/absl/types:optional",
]
}
diff --git a/video/render/incoming_video_stream.cc b/video/render/incoming_video_stream.cc
index 650036d..20495eb 100644
--- a/video/render/incoming_video_stream.cc
+++ b/video/render/incoming_video_stream.cc
@@ -11,9 +11,9 @@
#include "video/render/incoming_video_stream.h"
#include <memory>
+#include <optional>
#include <utility>
-#include "absl/types/optional.h"
#include "api/units/time_delta.h"
#include "rtc_base/checks.h"
#include "rtc_base/trace_event.h"
@@ -58,7 +58,7 @@
void IncomingVideoStream::Dequeue() {
TRACE_EVENT0("webrtc", "IncomingVideoStream::Dequeue");
RTC_DCHECK_RUN_ON(incoming_render_queue_.get());
- absl::optional<VideoFrame> frame_to_render = render_buffers_.FrameToRender();
+ std::optional<VideoFrame> frame_to_render = render_buffers_.FrameToRender();
if (frame_to_render)
callback_->OnFrame(*frame_to_render);
diff --git a/video/render/video_render_frames.cc b/video/render/video_render_frames.cc
index 4b2e703..df17e3c 100644
--- a/video/render/video_render_frames.cc
+++ b/video/render/video_render_frames.cc
@@ -88,8 +88,8 @@
return static_cast<int32_t>(incoming_frames_.size());
}
-absl::optional<VideoFrame> VideoRenderFrames::FrameToRender() {
- absl::optional<VideoFrame> render_frame;
+std::optional<VideoFrame> VideoRenderFrames::FrameToRender() {
+ std::optional<VideoFrame> render_frame;
// Get the newest frame that can be released for rendering.
while (!incoming_frames_.empty() && TimeToNextFrameRelease() <= 0) {
if (render_frame) {
diff --git a/video/render/video_render_frames.h b/video/render/video_render_frames.h
index 7f48eae..c69f9df 100644
--- a/video/render/video_render_frames.h
+++ b/video/render/video_render_frames.h
@@ -15,8 +15,8 @@
#include <stdint.h>
#include <list>
+#include <optional>
-#include "absl/types/optional.h"
#include "api/video/video_frame.h"
namespace webrtc {
@@ -32,7 +32,7 @@
int32_t AddFrame(VideoFrame&& new_frame);
// Get a frame for rendering, or false if it's not time to render.
- absl::optional<VideoFrame> FrameToRender();
+ std::optional<VideoFrame> FrameToRender();
// Returns the number of ms to next frame to render
uint32_t TimeToNextFrameRelease();
diff --git a/video/rtp_streams_synchronizer2.cc b/video/rtp_streams_synchronizer2.cc
index 0fbb391..a38fa9f 100644
--- a/video/rtp_streams_synchronizer2.cc
+++ b/video/rtp_streams_synchronizer2.cc
@@ -10,7 +10,8 @@
#include "video/rtp_streams_synchronizer2.h"
-#include "absl/types/optional.h"
+#include <optional>
+
#include "call/syncable.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
@@ -94,7 +95,7 @@
int64_t last_audio_receive_time_ms =
audio_measurement_.latest_receive_time_ms;
- absl::optional<Syncable::Info> audio_info = syncable_audio_->GetInfo();
+ std::optional<Syncable::Info> audio_info = syncable_audio_->GetInfo();
if (!audio_info || !UpdateMeasurements(&audio_measurement_, *audio_info)) {
return;
}
@@ -105,7 +106,7 @@
}
int64_t last_video_receive_ms = video_measurement_.latest_receive_time_ms;
- absl::optional<Syncable::Info> video_info = syncable_video_->GetInfo();
+ std::optional<Syncable::Info> video_info = syncable_video_->GetInfo();
if (!video_info || !UpdateMeasurements(&video_measurement_, *video_info)) {
return;
}
diff --git a/video/rtp_video_stream_receiver2.cc b/video/rtp_video_stream_receiver2.cc
index 88417de..d6c859f 100644
--- a/video/rtp_video_stream_receiver2.cc
+++ b/video/rtp_video_stream_receiver2.cc
@@ -13,12 +13,12 @@
#include <algorithm>
#include <limits>
#include <memory>
+#include <optional>
#include <utility>
#include <vector>
#include "absl/algorithm/container.h"
#include "absl/memory/memory.h"
-#include "absl/types/optional.h"
#include "api/video/video_codec_type.h"
#include "media/base/media_constants.h"
#include "modules/pacing/packet_router.h"
@@ -186,7 +186,7 @@
RTC_DCHECK(!lntf_state_)
<< "SendLossNotification() called twice in a row with no call to "
"SendBufferedRtcpFeedback() in between.";
- lntf_state_ = absl::make_optional<LossNotificationState>(
+ lntf_state_ = std::make_optional<LossNotificationState>(
last_decoded_seq_num, last_received_seq_num, decodability_flag);
}
@@ -195,7 +195,7 @@
bool request_key_frame = false;
std::vector<uint16_t> nack_sequence_numbers;
- absl::optional<LossNotificationState> lntf_state;
+ std::optional<LossNotificationState> lntf_state;
std::swap(request_key_frame, request_key_frame_);
std::swap(nack_sequence_numbers, nack_sequence_numbers_);
@@ -245,8 +245,8 @@
config_(*config),
packet_router_(packet_router),
ntp_estimator_(&env_.clock()),
- forced_playout_delay_max_ms_("max_ms", absl::nullopt),
- forced_playout_delay_min_ms_("min_ms", absl::nullopt),
+ forced_playout_delay_max_ms_("max_ms", std::nullopt),
+ forced_playout_delay_min_ms_("min_ms", std::nullopt),
rtp_receive_statistics_(rtp_receive_statistics),
ulpfec_receiver_(
MaybeConstructUlpfecReceiver(config->rtp.remote_ssrc,
@@ -369,20 +369,20 @@
h26x_packet_buffer_.reset();
}
-absl::optional<Syncable::Info> RtpVideoStreamReceiver2::GetSyncInfo() const {
+std::optional<Syncable::Info> RtpVideoStreamReceiver2::GetSyncInfo() const {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
Syncable::Info info;
- absl::optional<RtpRtcpInterface::SenderReportStats> last_sr =
+ std::optional<RtpRtcpInterface::SenderReportStats> last_sr =
rtp_rtcp_->GetSenderReportStats();
if (!last_sr.has_value()) {
- return absl::nullopt;
+ return std::nullopt;
}
info.capture_time_ntp_secs = last_sr->last_remote_timestamp.seconds();
info.capture_time_ntp_frac = last_sr->last_remote_timestamp.fractions();
info.capture_time_source_clock = last_sr->last_remote_rtp_timestamp;
if (!last_received_rtp_timestamp_ || !last_received_rtp_system_time_) {
- return absl::nullopt;
+ return std::nullopt;
}
info.latest_received_capture_timestamp = *last_received_rtp_timestamp_;
info.latest_receive_time_ms = last_received_rtp_system_time_->ms();
@@ -548,7 +548,7 @@
if (!video_header.playout_delay.emplace().Set(
TimeDelta::Millis(*forced_playout_delay_min_ms_),
TimeDelta::Millis(*forced_playout_delay_max_ms_))) {
- video_header.playout_delay = absl::nullopt;
+ video_header.playout_delay = std::nullopt;
}
} else {
video_header.playout_delay = rtp_packet.GetExtension<PlayoutDelayLimits>();
@@ -826,7 +826,7 @@
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
RTC_DCHECK(frame);
- const absl::optional<RTPVideoHeader::GenericDescriptorInfo>& descriptor =
+ const std::optional<RTPVideoHeader::GenericDescriptorInfo>& descriptor =
frame->GetRtpVideoHeader().generic;
if (loss_notification_controller_ && descriptor) {
@@ -1013,31 +1013,31 @@
ulpfec_payload_type, this, &env_.clock());
}
-absl::optional<int64_t> RtpVideoStreamReceiver2::LastReceivedPacketMs() const {
+std::optional<int64_t> RtpVideoStreamReceiver2::LastReceivedPacketMs() const {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
if (last_received_rtp_system_time_) {
- return absl::optional<int64_t>(last_received_rtp_system_time_->ms());
+ return std::optional<int64_t>(last_received_rtp_system_time_->ms());
}
- return absl::nullopt;
+ return std::nullopt;
}
-absl::optional<uint32_t>
-RtpVideoStreamReceiver2::LastReceivedFrameRtpTimestamp() const {
+std::optional<uint32_t> RtpVideoStreamReceiver2::LastReceivedFrameRtpTimestamp()
+ const {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
return last_received_rtp_timestamp_;
}
-absl::optional<int64_t> RtpVideoStreamReceiver2::LastReceivedKeyframePacketMs()
+std::optional<int64_t> RtpVideoStreamReceiver2::LastReceivedKeyframePacketMs()
const {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
if (last_received_keyframe_rtp_system_time_) {
- return absl::optional<int64_t>(
+ return std::optional<int64_t>(
last_received_keyframe_rtp_system_time_->ms());
}
- return absl::nullopt;
+ return std::nullopt;
}
-absl::optional<RtpRtcpInterface::SenderReportStats>
+std::optional<RtpRtcpInterface::SenderReportStats>
RtpVideoStreamReceiver2::GetSenderReportStats() const {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
return rtp_rtcp_->GetSenderReportStats();
@@ -1071,9 +1071,9 @@
auto parse_and_insert = [&](const RtpPacketReceived& packet) {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
- absl::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed_payload =
+ std::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed_payload =
type_it->second->Parse(packet.PayloadBuffer());
- if (parsed_payload == absl::nullopt) {
+ if (parsed_payload == std::nullopt) {
RTC_LOG(LS_WARNING) << "Failed parsing payload.";
return false;
}
@@ -1164,13 +1164,13 @@
rtp_rtcp_->IncomingRtcpPacket(
rtc::MakeArrayView(rtcp_packet, rtcp_packet_length));
- absl::optional<TimeDelta> rtt = rtp_rtcp_->LastRtt();
+ std::optional<TimeDelta> rtt = rtp_rtcp_->LastRtt();
if (!rtt.has_value()) {
// Waiting for valid rtt.
return true;
}
- absl::optional<RtpRtcpInterface::SenderReportStats> last_sr =
+ std::optional<RtpRtcpInterface::SenderReportStats> last_sr =
rtp_rtcp_->GetSenderReportStats();
if (!last_sr.has_value()) {
// Waiting for RTCP.
@@ -1182,7 +1182,7 @@
if (time_since_received <= 1) {
ntp_estimator_.UpdateRtcpTimestamp(*rtt, last_sr->last_remote_timestamp,
last_sr->last_remote_rtp_timestamp);
- absl::optional<int64_t> remote_to_local_clock_offset =
+ std::optional<int64_t> remote_to_local_clock_offset =
ntp_estimator_.EstimateRemoteToLocalClockOffset();
if (remote_to_local_clock_offset.has_value()) {
capture_clock_offset_updater_.SetRemoteToLocalClockOffset(
diff --git a/video/rtp_video_stream_receiver2.h b/video/rtp_video_stream_receiver2.h
index 221fc09..141f15c 100644
--- a/video/rtp_video_stream_receiver2.h
+++ b/video/rtp_video_stream_receiver2.h
@@ -13,10 +13,10 @@
#include <map>
#include <memory>
+#include <optional>
#include <string>
#include <vector>
-#include "absl/types/optional.h"
#include "api/crypto/frame_decryptor_interface.h"
#include "api/environment/environment.h"
#include "api/sequence_checker.h"
@@ -112,7 +112,7 @@
void StopReceive();
// Produces the transport-related timestamps; current_delay_ms is left unset.
- absl::optional<Syncable::Info> GetSyncInfo() const;
+ std::optional<Syncable::Info> GetSyncInfo() const;
bool DeliverRtcp(const uint8_t* rtcp_packet, size_t rtcp_packet_length);
@@ -204,11 +204,11 @@
int red_payload_type() const;
void SetProtectionPayloadTypes(int red_payload_type, int ulpfec_payload_type);
- absl::optional<int64_t> LastReceivedPacketMs() const;
- absl::optional<uint32_t> LastReceivedFrameRtpTimestamp() const;
- absl::optional<int64_t> LastReceivedKeyframePacketMs() const;
+ std::optional<int64_t> LastReceivedPacketMs() const;
+ std::optional<uint32_t> LastReceivedFrameRtpTimestamp() const;
+ std::optional<int64_t> LastReceivedKeyframePacketMs() const;
- absl::optional<RtpRtcpInterface::SenderReportStats> GetSenderReportStats()
+ std::optional<RtpRtcpInterface::SenderReportStats> GetSenderReportStats()
const;
private:
@@ -279,7 +279,7 @@
std::vector<uint16_t> nack_sequence_numbers_
RTC_GUARDED_BY(packet_sequence_checker_);
- absl::optional<LossNotificationState> lntf_state_
+ std::optional<LossNotificationState> lntf_state_
RTC_GUARDED_BY(packet_sequence_checker_);
};
enum ParseGenericDependenciesResult {
@@ -353,7 +353,7 @@
const KeyFrameReqMethod keyframe_request_method_;
RtcpFeedbackBuffer rtcp_feedback_buffer_;
- // TODO(tommi): Consider absl::optional<NackRequester> instead of unique_ptr
+ // TODO(tommi): Consider std::optional<NackRequester> instead of unique_ptr
// since nack is usually configured.
std::unique_ptr<NackRequester> nack_module_
RTC_GUARDED_BY(packet_sequence_checker_);
@@ -377,15 +377,15 @@
std::unique_ptr<FrameDependencyStructure> video_structure_
RTC_GUARDED_BY(packet_sequence_checker_);
// Frame id of the last frame with the attached video structure.
- // absl::nullopt when `video_structure_ == nullptr`;
- absl::optional<int64_t> video_structure_frame_id_
+ // std::nullopt when `video_structure_ == nullptr`;
+ std::optional<int64_t> video_structure_frame_id_
RTC_GUARDED_BY(packet_sequence_checker_);
Timestamp last_logged_failed_to_parse_dd_
RTC_GUARDED_BY(packet_sequence_checker_) = Timestamp::MinusInfinity();
std::unique_ptr<RtpFrameReferenceFinder> reference_finder_
RTC_GUARDED_BY(packet_sequence_checker_);
- absl::optional<VideoCodecType> current_codec_
+ std::optional<VideoCodecType> current_codec_
RTC_GUARDED_BY(packet_sequence_checker_);
uint32_t last_assembled_frame_rtp_timestamp_
RTC_GUARDED_BY(packet_sequence_checker_);
@@ -408,13 +408,13 @@
bool has_received_frame_ RTC_GUARDED_BY(packet_sequence_checker_);
- absl::optional<uint32_t> last_received_rtp_timestamp_
+ std::optional<uint32_t> last_received_rtp_timestamp_
RTC_GUARDED_BY(packet_sequence_checker_);
- absl::optional<uint32_t> last_received_keyframe_rtp_timestamp_
+ std::optional<uint32_t> last_received_keyframe_rtp_timestamp_
RTC_GUARDED_BY(packet_sequence_checker_);
- absl::optional<Timestamp> last_received_rtp_system_time_
+ std::optional<Timestamp> last_received_rtp_system_time_
RTC_GUARDED_BY(packet_sequence_checker_);
- absl::optional<Timestamp> last_received_keyframe_rtp_system_time_
+ std::optional<Timestamp> last_received_keyframe_rtp_system_time_
RTC_GUARDED_BY(packet_sequence_checker_);
// Handles incoming encrypted frames and forwards them to the
@@ -422,7 +422,7 @@
std::unique_ptr<BufferedFrameDecryptor> buffered_frame_decryptor_
RTC_PT_GUARDED_BY(packet_sequence_checker_);
bool frames_decryptable_ RTC_GUARDED_BY(worker_task_checker_);
- absl::optional<ColorSpace> last_color_space_;
+ std::optional<ColorSpace> last_color_space_;
AbsoluteCaptureTimeInterpolator absolute_capture_time_interpolator_
RTC_GUARDED_BY(packet_sequence_checker_);
diff --git a/video/rtp_video_stream_receiver2_unittest.cc b/video/rtp_video_stream_receiver2_unittest.cc
index 335bb81..0034f72 100644
--- a/video/rtp_video_stream_receiver2_unittest.cc
+++ b/video/rtp_video_stream_receiver2_unittest.cc
@@ -400,7 +400,7 @@
rtp_packet.SetSsrc(kSsrc);
rtp_packet.SetExtension<AbsoluteCaptureTimeExtension>(
AbsoluteCaptureTime{kAbsoluteCaptureTimestamp,
- /*estimated_capture_clock_offset=*/absl::nullopt});
+ /*estimated_capture_clock_offset=*/std::nullopt});
RTPVideoHeader video_header =
GetGenericVideoHeader(VideoFrameType::kVideoFrameKey);
@@ -433,7 +433,7 @@
rtp_packet.SetSsrc(kSsrc);
rtp_packet.SetExtension<AbsoluteCaptureTimeExtension>(
AbsoluteCaptureTime{kAbsoluteCaptureTimestamp,
- /*estimated_capture_clock_offset=*/absl::nullopt});
+ /*estimated_capture_clock_offset=*/std::nullopt});
RTPVideoHeader video_header =
GetGenericVideoHeader(VideoFrameType::kVideoFrameKey);
diff --git a/video/screenshare_loopback.cc b/video/screenshare_loopback.cc
index 882e58f..c6cade7 100644
--- a/video/screenshare_loopback.cc
+++ b/video/screenshare_loopback.cc
@@ -11,12 +11,12 @@
#include <stdio.h>
#include <memory>
+#include <optional>
#include <string>
#include <vector>
#include "absl/flags/flag.h"
#include "absl/flags/parse.h"
-#include "absl/types/optional.h"
#include "api/test/simulated_network.h"
#include "api/test/video_quality_test_fixture.h"
#include "api/transport/bitrate_settings.h"
diff --git a/video/send_statistics_proxy.cc b/video/send_statistics_proxy.cc
index 324a5d6..8c97a27 100644
--- a/video/send_statistics_proxy.cc
+++ b/video/send_statistics_proxy.cc
@@ -97,38 +97,38 @@
codec_info->codecSpecific.VP8.temporalIdx == kNoTemporalIdx);
}
-absl::optional<int> GetFallbackMaxPixels(const std::string& group) {
+std::optional<int> GetFallbackMaxPixels(const std::string& group) {
if (group.empty())
- return absl::nullopt;
+ return std::nullopt;
int min_pixels;
int max_pixels;
int min_bps;
if (sscanf(group.c_str(), "-%d,%d,%d", &min_pixels, &max_pixels, &min_bps) !=
3) {
- return absl::optional<int>();
+ return std::optional<int>();
}
if (min_pixels <= 0 || max_pixels <= 0 || max_pixels < min_pixels)
- return absl::optional<int>();
+ return std::optional<int>();
- return absl::optional<int>(max_pixels);
+ return std::optional<int>(max_pixels);
}
-absl::optional<int> GetFallbackMaxPixelsIfFieldTrialEnabled(
+std::optional<int> GetFallbackMaxPixelsIfFieldTrialEnabled(
const webrtc::FieldTrialsView& field_trials) {
std::string group = field_trials.Lookup(kVp8ForcedFallbackEncoderFieldTrial);
return (absl::StartsWith(group, "Enabled"))
? GetFallbackMaxPixels(group.substr(7))
- : absl::optional<int>();
+ : std::optional<int>();
}
-absl::optional<int> GetFallbackMaxPixelsIfFieldTrialDisabled(
+std::optional<int> GetFallbackMaxPixelsIfFieldTrialDisabled(
const webrtc::FieldTrialsView& field_trials) {
std::string group = field_trials.Lookup(kVp8ForcedFallbackEncoderFieldTrial);
return (absl::StartsWith(group, "Disabled"))
? GetFallbackMaxPixels(group.substr(8))
- : absl::optional<int>();
+ : std::optional<int>();
}
} // namespace
@@ -978,7 +978,7 @@
stats->total_encode_time_ms += encoded_image.timing_.encode_finish_ms -
encoded_image.timing_.encode_start_ms;
stats->scalability_mode =
- codec_info ? codec_info->scalability_mode : absl::nullopt;
+ codec_info ? codec_info->scalability_mode : std::nullopt;
// Report resolution of the top spatial layer.
bool is_top_spatial_layer =
codec_info == nullptr || codec_info->end_of_picture;
@@ -1040,7 +1040,7 @@
track.encoded_frame_rate.AddSamples(1);
}
- absl::optional<int> downscales =
+ std::optional<int> downscales =
adaptation_limitations_.MaskedQualityCounts().resolution_adaptations;
stats_.bw_limited_resolution |=
(downscales.has_value() && downscales.value() > 0);
@@ -1063,7 +1063,7 @@
// Clear cached scalability mode values, they may no longer be accurate.
for (auto& pair : stats_.substreams) {
VideoSendStream::StreamStats& stream_stats = pair.second;
- stream_stats.scalability_mode = absl::nullopt;
+ stream_stats.scalability_mode = std::nullopt;
}
}
@@ -1271,8 +1271,8 @@
}
void SendStatisticsProxy::TryUpdateInitialQualityResolutionAdaptUp(
- absl::optional<int> old_quality_downscales,
- absl::optional<int> updated_quality_downscales) {
+ std::optional<int> old_quality_downscales,
+ std::optional<int> updated_quality_downscales) {
if (uma_container_->initial_quality_changes_.down == 0)
return;
diff --git a/video/send_statistics_proxy.h b/video/send_statistics_proxy.h
index e9ac243..ab1605c 100644
--- a/video/send_statistics_proxy.h
+++ b/video/send_statistics_proxy.h
@@ -167,7 +167,7 @@
bool is_active = false;
int on_off_events = 0;
int64_t elapsed_ms = 0;
- absl::optional<int64_t> last_update_ms;
+ std::optional<int64_t> last_update_ms;
const int max_frame_diff_ms = 2000;
};
struct FallbackEncoderInfoDisabled {
@@ -218,8 +218,8 @@
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
struct MaskedAdaptationCounts {
- absl::optional<int> resolution_adaptations = absl::nullopt;
- absl::optional<int> num_framerate_reductions = absl::nullopt;
+ std::optional<int> resolution_adaptations = std::nullopt;
+ std::optional<int> num_framerate_reductions = std::nullopt;
};
struct Adaptations {
@@ -275,8 +275,8 @@
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
void UpdateAdaptationStats() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
void TryUpdateInitialQualityResolutionAdaptUp(
- absl::optional<int> old_quality_downscales,
- absl::optional<int> updated_quality_downscales)
+ std::optional<int> old_quality_downscales,
+ std::optional<int> updated_quality_downscales)
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
void UpdateEncoderFallbackStats(const CodecSpecificInfo* codec_info,
@@ -291,8 +291,8 @@
Clock* const clock_;
const std::string payload_name_;
const RtpConfig rtp_config_;
- const absl::optional<int> fallback_max_pixels_;
- const absl::optional<int> fallback_max_pixels_disabled_;
+ const std::optional<int> fallback_max_pixels_;
+ const std::optional<int> fallback_max_pixels_disabled_;
mutable Mutex mutex_;
VideoEncoderConfig::ContentType content_type_ RTC_GUARDED_BY(mutex_);
const int64_t start_ms_;
@@ -305,7 +305,7 @@
// Trackers mapped by ssrc.
std::map<uint32_t, Trackers> trackers_ RTC_GUARDED_BY(mutex_);
- absl::optional<int64_t> last_outlier_timestamp_ RTC_GUARDED_BY(mutex_);
+ std::optional<int64_t> last_outlier_timestamp_ RTC_GUARDED_BY(mutex_);
int last_num_spatial_layers_ RTC_GUARDED_BY(mutex_);
int last_num_simulcast_streams_ RTC_GUARDED_BY(mutex_);
@@ -324,7 +324,7 @@
};
// Stores the last change in encoder implementation in an optional, so that
// the event can be consumed.
- absl::optional<EncoderChangeEvent> encoder_changed_;
+ std::optional<EncoderChangeEvent> encoder_changed_;
// Contains stats used for UMA histograms. These stats will be reset if
// content type changes between real-time video and screenshare, since these
diff --git a/video/send_statistics_proxy_unittest.cc b/video/send_statistics_proxy_unittest.cc
index f743299..01b0349 100644
--- a/video/send_statistics_proxy_unittest.cc
+++ b/video/send_statistics_proxy_unittest.cc
@@ -382,7 +382,7 @@
EncodedImage encoded_image;
CodecSpecificInfo codec_info;
auto ssrc = config_.rtp.ssrcs[0];
- EXPECT_EQ(absl::nullopt,
+ EXPECT_EQ(std::nullopt,
statistics_proxy_->GetStats().substreams[ssrc].qp_sum);
encoded_image.qp_ = 3;
statistics_proxy_->OnSendEncodedImage(encoded_image, &codec_info);
@@ -397,10 +397,10 @@
CodecSpecificInfo codec_info;
auto ssrc = config_.rtp.ssrcs[0];
encoded_image.qp_ = -1;
- EXPECT_EQ(absl::nullopt,
+ EXPECT_EQ(std::nullopt,
statistics_proxy_->GetStats().substreams[ssrc].qp_sum);
statistics_proxy_->OnSendEncodedImage(encoded_image, &codec_info);
- EXPECT_EQ(absl::nullopt,
+ EXPECT_EQ(std::nullopt,
statistics_proxy_->GetStats().substreams[ssrc].qp_sum);
}
@@ -412,16 +412,16 @@
ScalabilityMode layer1_mode = ScalabilityMode::kL1T3;
auto ssrc0 = config_.rtp.ssrcs[0];
auto ssrc1 = config_.rtp.ssrcs[1];
- EXPECT_EQ(absl::nullopt,
+ EXPECT_EQ(std::nullopt,
statistics_proxy_->GetStats().substreams[ssrc0].scalability_mode);
- EXPECT_EQ(absl::nullopt,
+ EXPECT_EQ(std::nullopt,
statistics_proxy_->GetStats().substreams[ssrc1].scalability_mode);
encoded_image.SetSimulcastIndex(0);
codec_info.scalability_mode = layer0_mode;
statistics_proxy_->OnSendEncodedImage(encoded_image, &codec_info);
EXPECT_THAT(statistics_proxy_->GetStats().substreams[ssrc0].scalability_mode,
layer0_mode);
- EXPECT_EQ(absl::nullopt,
+ EXPECT_EQ(std::nullopt,
statistics_proxy_->GetStats().substreams[ssrc1].scalability_mode);
encoded_image.SetSimulcastIndex(1);
codec_info.scalability_mode = layer1_mode;
@@ -2533,7 +2533,7 @@
EXPECT_NE(GetStreamStats(kFirstSsrc).type,
VideoSendStream::StreamStats::StreamType::kRtx);
- EXPECT_EQ(GetStreamStats(kFirstSsrc).referenced_media_ssrc, absl::nullopt);
+ EXPECT_EQ(GetStreamStats(kFirstSsrc).referenced_media_ssrc, std::nullopt);
EXPECT_EQ(GetStreamStats(kFirstRtxSsrc).type,
VideoSendStream::StreamStats::StreamType::kRtx);
EXPECT_EQ(GetStreamStats(kFirstRtxSsrc).referenced_media_ssrc, kFirstSsrc);
@@ -2553,7 +2553,7 @@
EXPECT_NE(GetStreamStats(kFirstSsrc).type,
VideoSendStream::StreamStats::StreamType::kFlexfec);
- EXPECT_EQ(GetStreamStats(kFirstSsrc).referenced_media_ssrc, absl::nullopt);
+ EXPECT_EQ(GetStreamStats(kFirstSsrc).referenced_media_ssrc, std::nullopt);
EXPECT_EQ(GetStreamStats(kFlexFecSsrc).type,
VideoSendStream::StreamStats::StreamType::kFlexfec);
EXPECT_EQ(GetStreamStats(kFlexFecSsrc).referenced_media_ssrc, kFirstSsrc);
diff --git a/video/sv_loopback.cc b/video/sv_loopback.cc
index 86bdaf5..7d22030 100644
--- a/video/sv_loopback.cc
+++ b/video/sv_loopback.cc
@@ -11,12 +11,12 @@
#include <stdio.h>
#include <memory>
+#include <optional>
#include <string>
#include <vector>
#include "absl/flags/flag.h"
#include "absl/flags/parse.h"
-#include "absl/types/optional.h"
#include "api/test/simulated_network.h"
#include "api/test/video_quality_test_fixture.h"
#include "api/transport/bitrate_settings.h"
diff --git a/video/task_queue_frame_decode_scheduler.cc b/video/task_queue_frame_decode_scheduler.cc
index cd109c2..624f8ab 100644
--- a/video/task_queue_frame_decode_scheduler.cc
+++ b/video/task_queue_frame_decode_scheduler.cc
@@ -53,18 +53,17 @@
// this scheduled release should be skipped.
if (scheduled_rtp_ != rtp)
return;
- scheduled_rtp_ = absl::nullopt;
+ scheduled_rtp_ = std::nullopt;
std::move(cb)(rtp, schedule.render_time);
}),
wait);
}
void TaskQueueFrameDecodeScheduler::CancelOutstanding() {
- scheduled_rtp_ = absl::nullopt;
+ scheduled_rtp_ = std::nullopt;
}
-absl::optional<uint32_t>
-TaskQueueFrameDecodeScheduler::ScheduledRtpTimestamp() {
+std::optional<uint32_t> TaskQueueFrameDecodeScheduler::ScheduledRtpTimestamp() {
return scheduled_rtp_;
}
diff --git a/video/task_queue_frame_decode_scheduler.h b/video/task_queue_frame_decode_scheduler.h
index 69c6dae..1d95c3e 100644
--- a/video/task_queue_frame_decode_scheduler.h
+++ b/video/task_queue_frame_decode_scheduler.h
@@ -27,7 +27,7 @@
const TaskQueueFrameDecodeScheduler&) = delete;
// FrameDecodeScheduler implementation.
- absl::optional<uint32_t> ScheduledRtpTimestamp() override;
+ std::optional<uint32_t> ScheduledRtpTimestamp() override;
void ScheduleFrame(uint32_t rtp,
FrameDecodeTiming::FrameSchedule schedule,
FrameReleaseCallback cb) override;
@@ -38,7 +38,7 @@
Clock* const clock_;
TaskQueueBase* const bookkeeping_queue_;
- absl::optional<uint32_t> scheduled_rtp_;
+ std::optional<uint32_t> scheduled_rtp_;
ScopedTaskSafetyDetached task_safety_;
bool stopped_ = false;
};
diff --git a/video/task_queue_frame_decode_scheduler_unittest.cc b/video/task_queue_frame_decode_scheduler_unittest.cc
index 20258c6..38b43e6 100644
--- a/video/task_queue_frame_decode_scheduler_unittest.cc
+++ b/video/task_queue_frame_decode_scheduler_unittest.cc
@@ -13,9 +13,9 @@
#include <stddef.h>
#include <memory>
+#include <optional>
#include <utility>
-#include "absl/types/optional.h"
#include "api/units/time_delta.h"
#include "api/units/timestamp.h"
#include "test/gmock.h"
@@ -93,7 +93,7 @@
time_controller_.AdvanceTime(decode_delay / 2);
EXPECT_THAT(scheduler.ScheduledRtpTimestamp(), Optional(rtp));
scheduler.CancelOutstanding();
- EXPECT_THAT(scheduler.ScheduledRtpTimestamp(), Eq(absl::nullopt));
+ EXPECT_THAT(scheduler.ScheduledRtpTimestamp(), Eq(std::nullopt));
time_controller_.AdvanceTime(decode_delay / 2);
scheduler.Stop();
diff --git a/video/video_analyzer.h b/video/video_analyzer.h
index 8c8100c..ee0cda9 100644
--- a/video/video_analyzer.h
+++ b/video/video_analyzer.h
@@ -110,8 +110,8 @@
int64_t render_time_ms,
size_t encoded_frame_size);
- absl::optional<VideoFrame> reference;
- absl::optional<VideoFrame> render;
+ std::optional<VideoFrame> reference;
+ std::optional<VideoFrame> render;
bool dropped;
int64_t input_time_ms;
int64_t send_time_ms;
@@ -259,7 +259,7 @@
SamplesStatsCounter audio_jitter_buffer_ms_ RTC_GUARDED_BY(comparison_lock_);
SamplesStatsCounter pixels_ RTC_GUARDED_BY(comparison_lock_);
// Rendered frame with worst PSNR is saved for further analysis.
- absl::optional<FrameWithPsnr> worst_frame_ RTC_GUARDED_BY(comparison_lock_);
+ std::optional<FrameWithPsnr> worst_frame_ RTC_GUARDED_BY(comparison_lock_);
// Freeze metrics.
SamplesStatsCounter time_between_freezes_ RTC_GUARDED_BY(comparison_lock_);
uint32_t freeze_count_ RTC_GUARDED_BY(comparison_lock_);
@@ -292,13 +292,13 @@
int64_t wallclock_time_ RTC_GUARDED_BY(cpu_measurement_lock_);
std::deque<VideoFrame> frames_ RTC_GUARDED_BY(lock_);
- absl::optional<VideoFrame> last_rendered_frame_ RTC_GUARDED_BY(lock_);
+ std::optional<VideoFrame> last_rendered_frame_ RTC_GUARDED_BY(lock_);
RtpTimestampUnwrapper wrap_handler_ RTC_GUARDED_BY(lock_);
std::map<int64_t, int64_t> send_times_ RTC_GUARDED_BY(lock_);
std::map<int64_t, int64_t> recv_times_ RTC_GUARDED_BY(lock_);
std::map<int64_t, size_t> encoded_frame_sizes_ RTC_GUARDED_BY(lock_);
- absl::optional<uint32_t> first_encoded_timestamp_ RTC_GUARDED_BY(lock_);
- absl::optional<uint32_t> first_sent_timestamp_ RTC_GUARDED_BY(lock_);
+ std::optional<uint32_t> first_encoded_timestamp_ RTC_GUARDED_BY(lock_);
+ std::optional<uint32_t> first_sent_timestamp_ RTC_GUARDED_BY(lock_);
const double avg_psnr_threshold_;
const double avg_ssim_threshold_;
bool is_quick_test_enabled_;
diff --git a/video/video_loopback.cc b/video/video_loopback.cc
index 6358105..bcd3b35 100644
--- a/video/video_loopback.cc
+++ b/video/video_loopback.cc
@@ -12,12 +12,12 @@
#include <stdio.h>
#include <memory>
+#include <optional>
#include <string>
#include <vector>
#include "absl/flags/flag.h"
#include "absl/flags/parse.h"
-#include "absl/types/optional.h"
#include "api/test/simulated_network.h"
#include "api/test/video_quality_test_fixture.h"
#include "api/transport/bitrate_settings.h"
diff --git a/video/video_quality_observer2.cc b/video/video_quality_observer2.cc
index 0afc2f5..421ed10 100644
--- a/video/video_quality_observer2.cc
+++ b/video/video_quality_observer2.cc
@@ -165,7 +165,7 @@
bool was_freeze = false;
if (render_interframe_delays_.Size() >= kMinFrameSamplesToDetectFreeze) {
- const absl::optional<int64_t> avg_interframe_delay =
+ const std::optional<int64_t> avg_interframe_delay =
render_interframe_delays_.GetAverageRoundedDown();
RTC_DCHECK(avg_interframe_delay);
was_freeze = interframe_delay_ms >=
@@ -231,12 +231,12 @@
}
void VideoQualityObserver::OnDecodedFrame(uint32_t rtp_frame_timestamp,
- absl::optional<uint8_t> qp,
+ std::optional<uint8_t> qp,
VideoCodecType codec) {
if (!qp)
return;
- absl::optional<int> qp_blocky_threshold;
+ std::optional<int> qp_blocky_threshold;
// TODO(ilnik): add other codec types when we have QP for them.
switch (codec) {
case kVideoCodecVP8:
@@ -246,7 +246,7 @@
qp_blocky_threshold = kBlockyQpThresholdVp9;
break;
default:
- qp_blocky_threshold = absl::nullopt;
+ qp_blocky_threshold = std::nullopt;
}
RTC_DCHECK(blocky_frames_.find(rtp_frame_timestamp) == blocky_frames_.end());
diff --git a/video/video_quality_observer2.h b/video/video_quality_observer2.h
index 3587785..00650b5 100644
--- a/video/video_quality_observer2.h
+++ b/video/video_quality_observer2.h
@@ -13,10 +13,10 @@
#include <stdint.h>
+#include <optional>
#include <set>
#include <vector>
-#include "absl/types/optional.h"
#include "api/video/video_codec_type.h"
#include "api/video/video_content_type.h"
#include "rtc_base/numerics/moving_average.h"
@@ -37,7 +37,7 @@
~VideoQualityObserver() = default;
void OnDecodedFrame(uint32_t rtp_frame_timestamp,
- absl::optional<uint8_t> qp,
+ std::optional<uint8_t> qp,
VideoCodecType codec);
void OnRenderedFrame(const VideoFrameMetaData& frame_meta);
diff --git a/video/video_quality_test.cc b/video/video_quality_test.cc
index 571547e..824f088 100644
--- a/video/video_quality_test.cc
+++ b/video/video_quality_test.cc
@@ -805,7 +805,7 @@
decode_all_receive_streams = params_.ss[video_idx].selected_stream ==
params_.ss[video_idx].streams.size();
- absl::optional<int> decode_sub_stream;
+ std::optional<int> decode_sub_stream;
if (!decode_all_receive_streams)
decode_sub_stream = params_.ss[video_idx].selected_stream;
CreateMatchingVideoReceiveConfigs(
@@ -978,10 +978,10 @@
thumbnail_encoder_configs_.push_back(thumbnail_encoder_config.Copy());
thumbnail_send_configs_.push_back(thumbnail_send_config.Copy());
- AddMatchingVideoReceiveConfigs(
- &thumbnail_receive_configs_, thumbnail_send_config, send_transport,
- &video_decoder_factory_, absl::nullopt, false,
- test::VideoTestConstants::kNackRtpHistoryMs);
+ AddMatchingVideoReceiveConfigs(&thumbnail_receive_configs_,
+ thumbnail_send_config, send_transport,
+ &video_decoder_factory_, std::nullopt, false,
+ test::VideoTestConstants::kNackRtpHistoryMs);
}
for (size_t i = 0; i < thumbnail_send_configs_.size(); ++i) {
thumbnail_send_streams_.push_back(receiver_call_->CreateVideoSendStream(
@@ -1019,7 +1019,7 @@
clock_,
test::CreateSquareFrameGenerator(static_cast<int>(thumbnail.width),
static_cast<int>(thumbnail.height),
- absl::nullopt, absl::nullopt),
+ std::nullopt, std::nullopt),
thumbnail.max_framerate, *task_queue_factory_);
EXPECT_TRUE(frame_generator_capturer->Init());
thumbnail_capturers_.push_back(std::move(frame_generator_capturer));
@@ -1086,23 +1086,23 @@
} else if (params_.video[video_idx].clip_path == "Generator") {
frame_generator = test::CreateSquareFrameGenerator(
static_cast<int>(params_.video[video_idx].width),
- static_cast<int>(params_.video[video_idx].height), absl::nullopt,
- absl::nullopt);
+ static_cast<int>(params_.video[video_idx].height), std::nullopt,
+ std::nullopt);
} else if (params_.video[video_idx].clip_path == "GeneratorI420A") {
frame_generator = test::CreateSquareFrameGenerator(
static_cast<int>(params_.video[video_idx].width),
static_cast<int>(params_.video[video_idx].height),
- test::FrameGeneratorInterface::OutputType::kI420A, absl::nullopt);
+ test::FrameGeneratorInterface::OutputType::kI420A, std::nullopt);
} else if (params_.video[video_idx].clip_path == "GeneratorI010") {
frame_generator = test::CreateSquareFrameGenerator(
static_cast<int>(params_.video[video_idx].width),
static_cast<int>(params_.video[video_idx].height),
- test::FrameGeneratorInterface::OutputType::kI010, absl::nullopt);
+ test::FrameGeneratorInterface::OutputType::kI010, std::nullopt);
} else if (params_.video[video_idx].clip_path == "GeneratorNV12") {
frame_generator = test::CreateSquareFrameGenerator(
static_cast<int>(params_.video[video_idx].width),
static_cast<int>(params_.video[video_idx].height),
- test::FrameGeneratorInterface::OutputType::kNV12, absl::nullopt);
+ test::FrameGeneratorInterface::OutputType::kNV12, std::nullopt);
} else if (params_.video[video_idx].clip_path.empty()) {
video_sources_[video_idx] = test::CreateVideoCapturer(
params_.video[video_idx].width, params_.video[video_idx].height,
@@ -1114,8 +1114,8 @@
// Failed to get actual camera, use chroma generator as backup.
frame_generator = test::CreateSquareFrameGenerator(
static_cast<int>(params_.video[video_idx].width),
- static_cast<int>(params_.video[video_idx].height), absl::nullopt,
- absl::nullopt);
+ static_cast<int>(params_.video[video_idx].height), std::nullopt,
+ std::nullopt);
}
} else {
frame_generator = test::CreateFromYuvFileFrameGenerator(
diff --git a/video/video_receive_stream2.cc b/video/video_receive_stream2.cc
index 85a3afa..c6a20ef 100644
--- a/video/video_receive_stream2.cc
+++ b/video/video_receive_stream2.cc
@@ -15,12 +15,12 @@
#include <algorithm>
#include <memory>
+#include <optional>
#include <set>
#include <string>
#include <utility>
#include "absl/algorithm/container.h"
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/crypto/frame_decryptor_interface.h"
#include "api/scoped_refptr.h"
@@ -89,7 +89,7 @@
return buffer_;
}
- absl::optional<webrtc::ColorSpace> color_space() const override {
+ std::optional<webrtc::ColorSpace> color_space() const override {
return color_space_;
}
@@ -109,7 +109,7 @@
VideoCodecType codec_;
bool is_key_frame_;
EncodedResolution resolution_;
- absl::optional<webrtc::ColorSpace> color_space_;
+ std::optional<webrtc::ColorSpace> color_space_;
};
RenderResolution InitialDecoderResolution(const FieldTrialsView& field_trials) {
@@ -157,7 +157,7 @@
frame.EncodedImage()._encodedHeight == 0;
}
-std::string OptionalDelayToLogString(const absl::optional<TimeDelta> opt) {
+std::string OptionalDelayToLogString(const std::optional<TimeDelta> opt) {
return opt.has_value() ? ToLogString(*opt) : "<unset>";
}
@@ -566,7 +566,7 @@
}
}
- absl::optional<RtpRtcpInterface::SenderReportStats> rtcp_sr_stats =
+ std::optional<RtpRtcpInterface::SenderReportStats> rtcp_sr_stats =
rtp_video_stream_receiver_.GetSenderReportStats();
if (rtcp_sr_stats) {
stats.last_sender_report_timestamp_ms =
@@ -583,7 +583,7 @@
void VideoReceiveStream2::UpdateHistograms() {
RTC_DCHECK_RUN_ON(&worker_sequence_checker_);
- absl::optional<int> fraction_lost;
+ std::optional<int> fraction_lost;
StreamDataCounters rtp_stats;
StreamStatistician* statistician =
rtp_receive_statistics_->GetStatistician(remote_ssrc());
@@ -694,7 +694,7 @@
void VideoReceiveStream2::OnCompleteFrame(std::unique_ptr<EncodedFrame> frame) {
RTC_DCHECK_RUN_ON(&worker_sequence_checker_);
- if (absl::optional<VideoPlayoutDelay> playout_delay =
+ if (std::optional<VideoPlayoutDelay> playout_delay =
frame->EncodedImage().PlayoutDelay()) {
frame_minimum_playout_delay_ = playout_delay->min();
frame_maximum_playout_delay_ = playout_delay->max();
@@ -724,13 +724,12 @@
return remote_ssrc();
}
-absl::optional<Syncable::Info> VideoReceiveStream2::GetInfo() const {
+std::optional<Syncable::Info> VideoReceiveStream2::GetInfo() const {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
- absl::optional<Syncable::Info> info =
- rtp_video_stream_receiver_.GetSyncInfo();
+ std::optional<Syncable::Info> info = rtp_video_stream_receiver_.GetSyncInfo();
if (!info)
- return absl::nullopt;
+ return std::nullopt;
info->current_delay_ms = timing_->TargetVideoDelay().ms();
return info;
@@ -810,7 +809,7 @@
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
Timestamp now = env_.clock().CurrentTime();
- absl::optional<int64_t> last_packet_ms =
+ std::optional<int64_t> last_packet_ms =
rtp_video_stream_receiver_.LastReceivedPacketMs();
// To avoid spamming keyframe requests for a stream that is not active we
@@ -825,7 +824,7 @@
if (stream_is_active && !IsReceivingKeyFrame(now) &&
(!config_.crypto_options.sframe.require_frame_encryption ||
rtp_video_stream_receiver_.IsDecryptable())) {
- absl::optional<uint32_t> last_timestamp =
+ std::optional<uint32_t> last_timestamp =
rtp_video_stream_receiver_.LastReceivedFrameRtpTimestamp();
RTC_LOG(LS_WARNING) << "No decodable frame in " << wait
<< " requesting keyframe. Last RTP timestamp "
@@ -850,7 +849,7 @@
RTC_DCHECK_RUN_ON(&decode_sequence_checker_);
bool force_request_key_frame = false;
- absl::optional<int64_t> decoded_frame_picture_id;
+ std::optional<int64_t> decoded_frame_picture_id;
if (!video_receiver_.IsExternalDecoderRegistered(frame->PayloadType())) {
// Look for the decoder with this payload type.
@@ -935,8 +934,7 @@
}
if (encoded_frame_output_enabled) {
- absl::optional<RecordableEncodedFrame::EncodedResolution>
- pending_resolution;
+ std::optional<RecordableEncodedFrame::EncodedResolution> pending_resolution;
{
// Fish out `pending_resolution_` to avoid taking the mutex on every lap
// or dispatching under the mutex in the flush loop.
@@ -994,7 +992,7 @@
bool VideoReceiveStream2::IsReceivingKeyFrame(Timestamp now) const {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
- absl::optional<int64_t> last_keyframe_packet_ms =
+ std::optional<int64_t> last_keyframe_packet_ms =
rtp_video_stream_receiver_.LastReceivedKeyframePacketMs();
// If we recently have been receiving packets belonging to a keyframe then
@@ -1007,13 +1005,13 @@
void VideoReceiveStream2::UpdatePlayoutDelays() const {
RTC_DCHECK_RUN_ON(&worker_sequence_checker_);
- const std::initializer_list<absl::optional<TimeDelta>> min_delays = {
+ const std::initializer_list<std::optional<TimeDelta>> min_delays = {
frame_minimum_playout_delay_, base_minimum_playout_delay_,
syncable_minimum_playout_delay_};
// Since nullopt < anything, this will return the largest of the minumum
// delays, or nullopt if all are nullopt.
- absl::optional<TimeDelta> minimum_delay = std::max(min_delays);
+ std::optional<TimeDelta> minimum_delay = std::max(min_delays);
if (minimum_delay) {
auto num_playout_delays_set =
absl::c_count_if(min_delays, [](auto opt) { return opt.has_value(); });
@@ -1061,7 +1059,7 @@
// Save old state, set the new state.
RecordingState old_state;
- absl::optional<Timestamp> last_keyframe_request;
+ std::optional<Timestamp> last_keyframe_request;
{
// TODO(bugs.webrtc.org/11993): Post this to the network thread.
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
diff --git a/video/video_receive_stream2.h b/video/video_receive_stream2.h
index b6e86b2..f161f63 100644
--- a/video/video_receive_stream2.h
+++ b/video/video_receive_stream2.h
@@ -13,10 +13,10 @@
#include <map>
#include <memory>
+#include <optional>
#include <string>
#include <vector>
-#include "absl/types/optional.h"
#include "api/environment/environment.h"
#include "api/sequence_checker.h"
#include "api/task_queue/pending_task_safety_flag.h"
@@ -178,7 +178,7 @@
// Implements Syncable.
uint32_t id() const override;
- absl::optional<Syncable::Info> GetInfo() const override;
+ std::optional<Syncable::Info> GetInfo() const override;
bool GetPlayoutRtpTimestamp(uint32_t* rtp_timestamp,
int64_t* time_ms) const override;
void SetEstimatedPlayoutNtpTimestampMs(int64_t ntp_timestamp_ms,
@@ -213,7 +213,7 @@
// The picture id of the frame that was decoded, or nullopt if the frame was
// not decoded.
- absl::optional<int64_t> decoded_frame_picture_id;
+ std::optional<int64_t> decoded_frame_picture_id;
// True if the next frame decoded must be a keyframe. This value will set
// the value of `keyframe_required_`, which will force the frame buffer to
@@ -288,7 +288,7 @@
RTC_GUARDED_BY(packet_sequence_checker_);
std::unique_ptr<RtxReceiveStream> rtx_receive_stream_
RTC_GUARDED_BY(packet_sequence_checker_);
- absl::optional<uint32_t> updated_rtx_ssrc_
+ std::optional<uint32_t> updated_rtx_ssrc_
RTC_GUARDED_BY(packet_sequence_checker_);
std::unique_ptr<RtpStreamReceiverInterface> rtx_receiver_
RTC_GUARDED_BY(packet_sequence_checker_);
@@ -300,7 +300,7 @@
// If we have successfully decoded any frame.
bool frame_decoded_ RTC_GUARDED_BY(decode_sequence_checker_) = false;
- absl::optional<Timestamp> last_keyframe_request_
+ std::optional<Timestamp> last_keyframe_request_
RTC_GUARDED_BY(packet_sequence_checker_);
// Keyframe request intervals are configurable through field trials.
@@ -312,17 +312,17 @@
// biggest delay is used. -1 means use default value from the `timing_`.
//
// Minimum delay as decided by the RTP playout delay extension.
- absl::optional<TimeDelta> frame_minimum_playout_delay_
+ std::optional<TimeDelta> frame_minimum_playout_delay_
RTC_GUARDED_BY(worker_sequence_checker_);
// Minimum delay as decided by the setLatency function in "webrtc/api".
- absl::optional<TimeDelta> base_minimum_playout_delay_
+ std::optional<TimeDelta> base_minimum_playout_delay_
RTC_GUARDED_BY(worker_sequence_checker_);
// Minimum delay as decided by the A/V synchronization feature.
- absl::optional<TimeDelta> syncable_minimum_playout_delay_
+ std::optional<TimeDelta> syncable_minimum_playout_delay_
RTC_GUARDED_BY(worker_sequence_checker_);
// Maximum delay as decided by the RTP playout delay extension.
- absl::optional<TimeDelta> frame_maximum_playout_delay_
+ std::optional<TimeDelta> frame_maximum_playout_delay_
RTC_GUARDED_BY(worker_sequence_checker_);
// Function that is triggered with encoded frames, if not empty.
@@ -334,9 +334,9 @@
// Lock to avoid unnecessary per-frame idle wakeups in the code.
webrtc::Mutex pending_resolution_mutex_;
// Signal from decode queue to OnFrame callback to fill pending_resolution_.
- // absl::nullopt - no resolution needed. 0x0 - next OnFrame to fill with
+ // std::nullopt - no resolution needed. 0x0 - next OnFrame to fill with
// received resolution. Not 0x0 - OnFrame has filled a resolution.
- absl::optional<RecordableEncodedFrame::EncodedResolution> pending_resolution_
+ std::optional<RecordableEncodedFrame::EncodedResolution> pending_resolution_
RTC_GUARDED_BY(pending_resolution_mutex_);
// Buffered encoded frames held while waiting for decoded resolution.
std::vector<std::unique_ptr<EncodedFrame>> buffered_encoded_frames_
@@ -351,7 +351,7 @@
// VideoReceiveStream2 destruction.
std::unique_ptr<TaskQueueBase, TaskQueueDeleter> decode_queue_;
- absl::optional<uint32_t> last_decoded_rtp_timestamp_;
+ std::optional<uint32_t> last_decoded_rtp_timestamp_;
};
} // namespace internal
diff --git a/video/video_receive_stream2_unittest.cc b/video/video_receive_stream2_unittest.cc
index d1cee24..ef89290 100644
--- a/video/video_receive_stream2_unittest.cc
+++ b/video/video_receive_stream2_unittest.cc
@@ -15,13 +15,13 @@
#include <deque>
#include <limits>
#include <memory>
+#include <optional>
#include <queue>
#include <tuple>
#include <utility>
#include <vector>
#include "absl/memory/memory.h"
-#include "absl/types/optional.h"
#include "api/environment/environment.h"
#include "api/environment/environment_factory.h"
#include "api/metronome/test/fake_metronome.h"
@@ -84,8 +84,8 @@
auto RenderedFrame() {
return RenderedFrameWith(_);
}
-testing::Matcher<absl::optional<VideoFrame>> DidNotReceiveFrame() {
- return Eq(absl::nullopt);
+testing::Matcher<std::optional<VideoFrame>> DidNotReceiveFrame() {
+ return Eq(std::nullopt);
}
constexpr TimeDelta kDefaultTimeOut = TimeDelta::Millis(50);
@@ -115,14 +115,14 @@
}
// If `advance_time`, then the clock will always advance by `timeout`.
- absl::optional<VideoFrame> WaitForFrame(TimeDelta timeout,
- bool advance_time = false) {
+ std::optional<VideoFrame> WaitForFrame(TimeDelta timeout,
+ bool advance_time = false) {
auto start = time_controller_->GetClock()->CurrentTime();
if (last_frame_.empty()) {
time_controller_->AdvanceTime(TimeDelta::Zero());
time_controller_->Wait([this] { return !last_frame_.empty(); }, timeout);
}
- absl::optional<VideoFrame> ret;
+ std::optional<VideoFrame> ret;
if (!last_frame_.empty()) {
ret = last_frame_.front();
last_frame_.pop_front();
@@ -232,8 +232,8 @@
}
void RecreateReceiveStream(
- absl::optional<VideoReceiveStreamInterface::RecordingState> state =
- absl::nullopt) {
+ std::optional<VideoReceiveStreamInterface::RecordingState> state =
+ std::nullopt) {
if (video_receive_stream_) {
video_receive_stream_->UnregisterFromTransport();
video_receive_stream_ = nullptr;
@@ -379,7 +379,7 @@
.Build();
video_receive_stream_->OnCompleteFrame(std::move(test_frame0));
EXPECT_THAT(timing_->RenderParameters().max_composition_delay_in_frames,
- Eq(absl::nullopt));
+ Eq(std::nullopt));
time_controller_.AdvanceTime(k30FpsDelay);
// Max composition delay not set for playout delay 0,0.
@@ -393,7 +393,7 @@
.Build();
video_receive_stream_->OnCompleteFrame(std::move(test_frame1));
EXPECT_THAT(timing_->RenderParameters().max_composition_delay_in_frames,
- Eq(absl::nullopt));
+ Eq(std::nullopt));
time_controller_.AdvanceTime(k30FpsDelay);
// Max composition delay not set for playout delay X,Y, where X,Y>0.
@@ -407,7 +407,7 @@
.Build();
video_receive_stream_->OnCompleteFrame(std::move(test_frame2));
EXPECT_THAT(timing_->RenderParameters().max_composition_delay_in_frames,
- Eq(absl::nullopt));
+ Eq(std::nullopt));
time_controller_.AdvanceTime(k30FpsDelay);
@@ -1066,7 +1066,7 @@
.AsLast()
.Build());
EXPECT_THAT(fake_renderer_.WaitForFrame(k30FpsDelay, /*advance_time=*/true),
- Eq(absl::nullopt));
+ Eq(std::nullopt));
}
uint32_t current_rtp = RtpTimestampForFrame(id);
Timestamp send_15fps_end_time =
@@ -1084,7 +1084,7 @@
.AsLast()
.Build());
EXPECT_THAT(fake_renderer_.WaitForFrame(k15FpsDelay, /*advance_time=*/true),
- Eq(absl::nullopt));
+ Eq(std::nullopt));
}
++id;
diff --git a/video/video_send_stream_impl.cc b/video/video_send_stream_impl.cc
index 181f146..77f40f9 100644
--- a/video/video_send_stream_impl.cc
+++ b/video/video_send_stream_impl.cc
@@ -15,12 +15,12 @@
#include <cstdint>
#include <map>
#include <memory>
+#include <optional>
#include <string>
#include <utility>
#include <vector>
#include "absl/algorithm/container.h"
-#include "absl/types/optional.h"
#include "api/adaptation/resource.h"
#include "api/call/bitrate_allocation.h"
#include "api/crypto/crypto_options.h"
@@ -181,7 +181,7 @@
return pad_up_to_bitrate_bps;
}
-absl::optional<AlrExperimentSettings> GetAlrSettings(
+std::optional<AlrExperimentSettings> GetAlrSettings(
const FieldTrialsView& field_trials,
VideoEncoderConfig::ContentType content_type) {
if (content_type == VideoEncoderConfig::ContentType::kScreen) {
@@ -208,15 +208,15 @@
// Returns an optional that has value iff TransportSeqNumExtensionConfigured
// is `true` for the given video send stream config.
-absl::optional<float> GetConfiguredPacingFactor(
+std::optional<float> GetConfiguredPacingFactor(
const VideoSendStream::Config& config,
VideoEncoderConfig::ContentType content_type,
const PacingConfig& default_pacing_config,
const FieldTrialsView& field_trials) {
if (!TransportSeqNumExtensionConfigured(config))
- return absl::nullopt;
+ return std::nullopt;
- absl::optional<AlrExperimentSettings> alr_settings =
+ std::optional<AlrExperimentSettings> alr_settings =
GetAlrSettings(field_trials, content_type);
if (alr_settings)
return alr_settings->pacing_factor;
@@ -472,12 +472,12 @@
RTC_CHECK(
AlrExperimentSettings::MaxOneFieldTrialEnabled(env_.field_trials()));
- absl::optional<bool> enable_alr_bw_probing;
+ std::optional<bool> enable_alr_bw_probing;
// If send-side BWE is enabled, check if we should apply updated probing and
// pacing settings.
if (configured_pacing_factor_) {
- absl::optional<AlrExperimentSettings> alr_settings =
+ std::optional<AlrExperimentSettings> alr_settings =
GetAlrSettings(env_.field_trials(), content_type_);
int queue_time_limit_ms;
if (alr_settings) {
@@ -575,7 +575,7 @@
return stats_proxy_.GetStats();
}
-absl::optional<float> VideoSendStreamImpl::GetPacingFactorOverride() const {
+std::optional<float> VideoSendStreamImpl::GetPacingFactorOverride() const {
return configured_pacing_factor_;
}
@@ -782,8 +782,8 @@
!config_.suspend_below_min_bitrate,
encoder_bitrate_priority_,
(content_type_ == VideoEncoderConfig::ContentType::kRealtimeVideo)
- ? absl::optional(TrackRateElasticity::kCanConsumeExtraRate)
- : absl::nullopt};
+ ? std::optional(TrackRateElasticity::kCanConsumeExtraRate)
+ : std::nullopt};
}
void VideoSendStreamImpl::OnEncoderConfigurationChanged(
@@ -802,7 +802,7 @@
const VideoCodecType codec_type =
PayloadStringToCodecType(config_.rtp.payload_name);
- const absl::optional<DataRate> experimental_min_bitrate =
+ const std::optional<DataRate> experimental_min_bitrate =
GetExperimentalMinVideoBitrate(env_.field_trials(), codec_type);
encoder_min_bitrate_bps_ =
experimental_min_bitrate
@@ -947,10 +947,10 @@
return protection_bitrate_bps;
}
-absl::optional<DataRate> VideoSendStreamImpl::GetUsedRate() const {
+std::optional<DataRate> VideoSendStreamImpl::GetUsedRate() const {
// This value is for real-time video. Screenshare may have unused bandwidth
// that can be shared, and this needs to be changed to support that.
- return absl::nullopt;
+ return std::nullopt;
}
} // namespace internal
diff --git a/video/video_send_stream_impl.h b/video/video_send_stream_impl.h
index ea405ff..fd91e00 100644
--- a/video/video_send_stream_impl.h
+++ b/video/video_send_stream_impl.h
@@ -16,10 +16,10 @@
#include <atomic>
#include <map>
#include <memory>
+#include <optional>
#include <string>
#include <vector>
-#include "absl/types/optional.h"
#include "api/environment/environment.h"
#include "api/field_trials_view.h"
#include "api/metronome/metronome.h"
@@ -117,7 +117,7 @@
std::map<uint32_t, RtpPayloadState> GetRtpPayloadStates() const;
- const absl::optional<float>& configured_pacing_factor() const {
+ const std::optional<float>& configured_pacing_factor() const {
return configured_pacing_factor_;
}
@@ -129,7 +129,7 @@
SendDelayStats* send_delay_stats)
: stats_proxy_(*stats_proxy), send_delay_stats_(*send_delay_stats) {}
- void OnSendPacket(absl::optional<uint16_t> packet_id,
+ void OnSendPacket(std::optional<uint16_t> packet_id,
Timestamp capture_time,
uint32_t ssrc) override {
stats_proxy_.OnSendPacket(ssrc, capture_time);
@@ -143,10 +143,10 @@
SendDelayStats& send_delay_stats_;
};
- absl::optional<float> GetPacingFactorOverride() const;
+ std::optional<float> GetPacingFactorOverride() const;
// Implements BitrateAllocatorObserver.
uint32_t OnBitrateUpdated(BitrateAllocationUpdate update) override;
- absl::optional<DataRate> GetUsedRate() const override;
+ std::optional<DataRate> GetUsedRate() const override;
// Implements VideoStreamEncoderInterface::EncoderSink
void OnEncoderConfigurationChanged(
@@ -229,12 +229,12 @@
// throttle sending of similar bitrate allocations.
struct VbaSendContext {
VideoBitrateAllocation last_sent_allocation;
- absl::optional<VideoBitrateAllocation> throttled_allocation;
+ std::optional<VideoBitrateAllocation> throttled_allocation;
int64_t last_send_time_ms;
};
- absl::optional<VbaSendContext> video_bitrate_allocation_context_
+ std::optional<VbaSendContext> video_bitrate_allocation_context_
RTC_GUARDED_BY(thread_checker_);
- const absl::optional<float> configured_pacing_factor_;
+ const std::optional<float> configured_pacing_factor_;
};
} // namespace internal
} // namespace webrtc
diff --git a/video/video_send_stream_impl_unittest.cc b/video/video_send_stream_impl_unittest.cc
index 08ac956..8ce4c8f 100644
--- a/video/video_send_stream_impl_unittest.cc
+++ b/video/video_send_stream_impl_unittest.cc
@@ -15,11 +15,11 @@
#include <cstdint>
#include <map>
#include <memory>
+#include <optional>
#include <string>
#include <utility>
#include <vector>
-#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/call/bitrate_allocation.h"
#include "api/environment/environment.h"
@@ -1233,7 +1233,7 @@
EXPECT_CALL(bitrate_allocator_,
AddObserver(vss_impl.get(),
Field(&MediaStreamAllocationConfig::rate_elasticity,
- absl::nullopt)));
+ std::nullopt)));
vss_impl->Start();
EXPECT_CALL(bitrate_allocator_, RemoveObserver(vss_impl.get()));
vss_impl->Stop();
diff --git a/video/video_send_stream_tests.cc b/video/video_send_stream_tests.cc
index da9ded0..92014e6 100644
--- a/video/video_send_stream_tests.cc
+++ b/video/video_send_stream_tests.cc
@@ -86,7 +86,7 @@
explicit VideoSendStreamPeer(webrtc::VideoSendStream* base_class_stream)
: internal_stream_(
static_cast<internal::VideoSendStreamImpl*>(base_class_stream)) {}
- absl::optional<float> GetPacingFactorOverride() const {
+ std::optional<float> GetPacingFactorOverride() const {
return internal_stream_->GetPacingFactorOverride();
}
@@ -1426,7 +1426,7 @@
TestState test_state_ = kBeforeStopCapture;
Clock* const clock_;
Mutex mutex_;
- absl::optional<int64_t> last_packet_time_ms_ RTC_GUARDED_BY(mutex_);
+ std::optional<int64_t> last_packet_time_ms_ RTC_GUARDED_BY(mutex_);
test::FrameGeneratorCapturer* capturer_ RTC_GUARDED_BY(mutex_);
} test;
@@ -3316,10 +3316,10 @@
}
ScalableVideoController::StreamLayersConfig GetScalabilityConfig() const {
- absl::optional<ScalabilityMode> scalability_mode =
+ std::optional<ScalabilityMode> scalability_mode =
ScalabilityModeFromString(params_.scalability_mode);
EXPECT_TRUE(scalability_mode.has_value());
- absl::optional<ScalableVideoController::StreamLayersConfig> config =
+ std::optional<ScalableVideoController::StreamLayersConfig> config =
ScalabilityStructureConfig(*scalability_mode);
EXPECT_TRUE(config.has_value());
EXPECT_EQ(config->num_spatial_layers, params_.num_spatial_layers);
@@ -3331,8 +3331,8 @@
VideoCodecVP9 vp9_settings_;
webrtc::VideoEncoderConfig encoder_config_;
bool last_packet_marker_ = false;
- absl::optional<uint16_t> last_packet_sequence_number_;
- absl::optional<uint32_t> last_packet_timestamp_;
+ std::optional<uint16_t> last_packet_sequence_number_;
+ std::optional<uint32_t> last_packet_timestamp_;
RTPVideoHeaderVP9 last_vp9_;
std::map<int, int> last_temporal_idx_by_spatial_idx_;
Mutex mutex_;
@@ -3451,7 +3451,7 @@
vp9_settings_.numberOfSpatialLayers = params_.num_spatial_layers;
vp9_settings_.interLayerPred = params_.inter_layer_pred;
} else {
- absl::optional<ScalabilityMode> mode =
+ std::optional<ScalabilityMode> mode =
ScalabilityModeFromString(params_.scalability_mode);
encoder_config->simulcast_layers[0].scalability_mode = mode;
EXPECT_TRUE(mode.has_value());
@@ -3780,7 +3780,7 @@
class PacingFactorObserver : public test::SendTest {
public:
PacingFactorObserver(bool configure_send_side,
- absl::optional<float> expected_pacing_factor)
+ std::optional<float> expected_pacing_factor)
: test::SendTest(test::VideoTestConstants::kDefaultTimeout),
configure_send_side_(configure_send_side),
expected_pacing_factor_(expected_pacing_factor) {}
@@ -3836,7 +3836,7 @@
private:
const bool configure_send_side_;
- const absl::optional<float> expected_pacing_factor_;
+ const std::optional<float> expected_pacing_factor_;
};
std::string GetAlrProbingExperimentString() {
@@ -3857,7 +3857,7 @@
TEST_F(VideoSendStreamTest, AlrNotConfiguredWhenSendSideOff) {
test::ScopedFieldTrials alr_experiment(GetAlrProbingExperimentString());
// Send-side bwe off, use configuration should not be overridden.
- PacingFactorObserver test_without_send_side(false, absl::nullopt);
+ PacingFactorObserver test_without_send_side(false, std::nullopt);
RunBaseTest(&test_without_send_side);
}
@@ -4120,7 +4120,7 @@
parsed.timestamp = rtp_packet.Timestamp();
parsed.ssrc = rtp_packet.Ssrc();
- absl::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed_payload =
+ std::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed_payload =
depacketizer_->Parse(rtp_packet.PayloadBuffer());
EXPECT_TRUE(parsed_payload);
diff --git a/video/video_source_sink_controller.cc b/video/video_source_sink_controller.cc
index 2f7b375..3a49044 100644
--- a/video/video_source_sink_controller.cc
+++ b/video/video_source_sink_controller.cc
@@ -71,13 +71,13 @@
return restrictions_;
}
-absl::optional<size_t> VideoSourceSinkController::pixels_per_frame_upper_limit()
+std::optional<size_t> VideoSourceSinkController::pixels_per_frame_upper_limit()
const {
RTC_DCHECK_RUN_ON(&sequence_checker_);
return pixels_per_frame_upper_limit_;
}
-absl::optional<double> VideoSourceSinkController::frame_rate_upper_limit()
+std::optional<double> VideoSourceSinkController::frame_rate_upper_limit()
const {
RTC_DCHECK_RUN_ON(&sequence_checker_);
return frame_rate_upper_limit_;
@@ -104,7 +104,7 @@
return active_;
}
-absl::optional<rtc::VideoSinkWants::FrameSize>
+std::optional<rtc::VideoSinkWants::FrameSize>
VideoSourceSinkController::requested_resolution() const {
RTC_DCHECK_RUN_ON(&sequence_checker_);
return requested_resolution_;
@@ -117,13 +117,13 @@
}
void VideoSourceSinkController::SetPixelsPerFrameUpperLimit(
- absl::optional<size_t> pixels_per_frame_upper_limit) {
+ std::optional<size_t> pixels_per_frame_upper_limit) {
RTC_DCHECK_RUN_ON(&sequence_checker_);
pixels_per_frame_upper_limit_ = std::move(pixels_per_frame_upper_limit);
}
void VideoSourceSinkController::SetFrameRateUpperLimit(
- absl::optional<double> frame_rate_upper_limit) {
+ std::optional<double> frame_rate_upper_limit) {
RTC_DCHECK_RUN_ON(&sequence_checker_);
frame_rate_upper_limit_ = std::move(frame_rate_upper_limit);
}
@@ -151,7 +151,7 @@
}
void VideoSourceSinkController::SetRequestedResolution(
- absl::optional<rtc::VideoSinkWants::FrameSize> requested_resolution) {
+ std::optional<rtc::VideoSinkWants::FrameSize> requested_resolution) {
RTC_DCHECK_RUN_ON(&sequence_checker_);
requested_resolution_ = std::move(requested_resolution);
}
@@ -167,9 +167,9 @@
std::numeric_limits<int>::max()));
wants.target_pixel_count =
restrictions_.target_pixels_per_frame().has_value()
- ? absl::optional<int>(rtc::dchecked_cast<int>(
+ ? std::optional<int>(rtc::dchecked_cast<int>(
restrictions_.target_pixels_per_frame().value()))
- : absl::nullopt;
+ : std::nullopt;
wants.max_framerate_fps =
restrictions_.max_frame_rate().has_value()
? static_cast<int>(restrictions_.max_frame_rate().value())
diff --git a/video/video_source_sink_controller.h b/video/video_source_sink_controller.h
index 1bb6ef6..9ce0fb5 100644
--- a/video/video_source_sink_controller.h
+++ b/video/video_source_sink_controller.h
@@ -11,10 +11,10 @@
#ifndef VIDEO_VIDEO_SOURCE_SINK_CONTROLLER_H_
#define VIDEO_VIDEO_SOURCE_SINK_CONTROLLER_H_
+#include <optional>
#include <string>
#include <vector>
-#include "absl/types/optional.h"
#include "api/sequence_checker.h"
#include "api/video/video_frame.h"
#include "api/video/video_sink_interface.h"
@@ -46,26 +46,26 @@
void PushSourceSinkSettings();
VideoSourceRestrictions restrictions() const;
- absl::optional<size_t> pixels_per_frame_upper_limit() const;
- absl::optional<double> frame_rate_upper_limit() const;
+ std::optional<size_t> pixels_per_frame_upper_limit() const;
+ std::optional<double> frame_rate_upper_limit() const;
bool rotation_applied() const;
int resolution_alignment() const;
const std::vector<rtc::VideoSinkWants::FrameSize>& resolutions() const;
bool active() const;
- absl::optional<rtc::VideoSinkWants::FrameSize> requested_resolution() const;
+ std::optional<rtc::VideoSinkWants::FrameSize> requested_resolution() const;
// Updates the settings stored internally. In order for these settings to be
// applied to the sink, PushSourceSinkSettings() must subsequently be called.
void SetRestrictions(VideoSourceRestrictions restrictions);
void SetPixelsPerFrameUpperLimit(
- absl::optional<size_t> pixels_per_frame_upper_limit);
- void SetFrameRateUpperLimit(absl::optional<double> frame_rate_upper_limit);
+ std::optional<size_t> pixels_per_frame_upper_limit);
+ void SetFrameRateUpperLimit(std::optional<double> frame_rate_upper_limit);
void SetRotationApplied(bool rotation_applied);
void SetResolutionAlignment(int resolution_alignment);
void SetResolutions(std::vector<rtc::VideoSinkWants::FrameSize> resolutions);
void SetActive(bool active);
void SetRequestedResolution(
- absl::optional<rtc::VideoSinkWants::FrameSize> requested_resolution);
+ std::optional<rtc::VideoSinkWants::FrameSize> requested_resolution);
private:
rtc::VideoSinkWants CurrentSettingsToSinkWants() const
@@ -84,16 +84,16 @@
// Ensures that even if we are not restricted, the sink is never configured
// above this limit. Example: We are not CPU limited (no `restrictions_`) but
// our encoder is capped at 30 fps (= `frame_rate_upper_limit_`).
- absl::optional<size_t> pixels_per_frame_upper_limit_
+ std::optional<size_t> pixels_per_frame_upper_limit_
RTC_GUARDED_BY(&sequence_checker_);
- absl::optional<double> frame_rate_upper_limit_
+ std::optional<double> frame_rate_upper_limit_
RTC_GUARDED_BY(&sequence_checker_);
bool rotation_applied_ RTC_GUARDED_BY(&sequence_checker_) = false;
int resolution_alignment_ RTC_GUARDED_BY(&sequence_checker_) = 1;
std::vector<rtc::VideoSinkWants::FrameSize> resolutions_
RTC_GUARDED_BY(&sequence_checker_);
bool active_ RTC_GUARDED_BY(&sequence_checker_) = true;
- absl::optional<rtc::VideoSinkWants::FrameSize> requested_resolution_
+ std::optional<rtc::VideoSinkWants::FrameSize> requested_resolution_
RTC_GUARDED_BY(&sequence_checker_);
};
diff --git a/video/video_source_sink_controller_unittest.cc b/video/video_source_sink_controller_unittest.cc
index 75cc52b..bddfc0d 100644
--- a/video/video_source_sink_controller_unittest.cc
+++ b/video/video_source_sink_controller_unittest.cc
@@ -70,7 +70,7 @@
const rtc::VideoSinkWants& wants) {
EXPECT_FALSE(wants.rotation_applied);
EXPECT_EQ(wants.max_pixel_count, kIntUnconstrained);
- EXPECT_EQ(wants.target_pixel_count, absl::nullopt);
+ EXPECT_EQ(wants.target_pixel_count, std::nullopt);
EXPECT_EQ(wants.max_framerate_fps, kIntUnconstrained);
EXPECT_EQ(wants.resolution_alignment, 1);
EXPECT_FALSE(wants.requested_resolution.has_value());
diff --git a/video/video_stream_buffer_controller.cc b/video/video_stream_buffer_controller.cc
index b7ed896..83de75a 100644
--- a/video/video_stream_buffer_controller.cc
+++ b/video/video_stream_buffer_controller.cc
@@ -12,11 +12,11 @@
#include <algorithm>
#include <memory>
+#include <optional>
#include <utility>
#include "absl/base/attributes.h"
#include "absl/functional/bind_front.h"
-#include "absl/types/optional.h"
#include "api/sequence_checker.h"
#include "api/task_queue/task_queue_base.h"
#include "api/units/data_size.h"
@@ -65,7 +65,7 @@
const VideoContentType contentType;
const bool delayed_by_retransmission;
const uint32_t rtp_timestamp;
- const absl::optional<Timestamp> receive_time;
+ const std::optional<Timestamp> receive_time;
};
Timestamp MinReceiveTime(const EncodedFrame& frame) {
@@ -79,7 +79,7 @@
}
Timestamp ReceiveTime(const EncodedFrame& frame) {
- absl::optional<Timestamp> ts = frame.ReceivedTimestamp();
+ std::optional<Timestamp> ts = frame.ReceivedTimestamp();
RTC_DCHECK(ts.has_value()) << "Received frame must have a timestamp set!";
return *ts;
}
@@ -148,7 +148,7 @@
frame_decode_scheduler_->CancelOutstanding();
}
-absl::optional<int64_t> VideoStreamBufferController::InsertFrame(
+std::optional<int64_t> VideoStreamBufferController::InsertFrame(
std::unique_ptr<EncodedFrame> frame) {
RTC_DCHECK_RUN_ON(&worker_sequence_checker_);
FrameMetadata metadata(*frame);
@@ -240,7 +240,7 @@
}
if (!superframe_delayed_by_retransmission) {
- absl::optional<TimeDelta> inter_frame_delay_variation =
+ std::optional<TimeDelta> inter_frame_delay_variation =
ifdv_calculator_.Calculate(first_frame.RtpTimestamp(),
max_receive_time);
if (inter_frame_delay_variation) {
@@ -352,7 +352,7 @@
}
void VideoStreamBufferController::UpdateTimingFrameInfo() {
- absl::optional<TimingFrameInfo> info = timing_->GetTimingFrameInfo();
+ std::optional<TimingFrameInfo> info = timing_->GetTimingFrameInfo();
if (info)
stats_proxy_->OnTimingFrameInfoUpdated(*info);
}
@@ -405,7 +405,7 @@
// Ensures the frame is scheduled for decode before the stream times out.
// This is otherwise a race condition.
max_wait = std::max(max_wait - TimeDelta::Millis(1), TimeDelta::Zero());
- absl::optional<FrameDecodeTiming::FrameSchedule> schedule;
+ std::optional<FrameDecodeTiming::FrameSchedule> schedule;
while (decodable_tu_info) {
schedule = decode_timing_.OnFrameBufferUpdated(
decodable_tu_info->next_rtp_timestamp,
diff --git a/video/video_stream_buffer_controller.h b/video/video_stream_buffer_controller.h
index f07b3eb..43d4544 100644
--- a/video/video_stream_buffer_controller.h
+++ b/video/video_stream_buffer_controller.h
@@ -80,7 +80,7 @@
void Stop();
void SetProtectionMode(VCMVideoProtection protection_mode);
void Clear();
- absl::optional<int64_t> InsertFrame(std::unique_ptr<EncodedFrame> frame);
+ std::optional<int64_t> InsertFrame(std::unique_ptr<EncodedFrame> frame);
void UpdateRtt(int64_t max_rtt_ms);
void SetMaxWaits(TimeDelta max_wait_for_keyframe,
TimeDelta max_wait_for_frame);
diff --git a/video/video_stream_buffer_controller_unittest.cc b/video/video_stream_buffer_controller_unittest.cc
index d2f2f73..c97601b 100644
--- a/video/video_stream_buffer_controller_unittest.cc
+++ b/video/video_stream_buffer_controller_unittest.cc
@@ -14,12 +14,12 @@
#include <limits>
#include <memory>
+#include <optional>
#include <string>
#include <tuple>
#include <utility>
#include <vector>
-#include "absl/types/optional.h"
#include "absl/types/variant.h"
#include "api/metronome/test/fake_metronome.h"
#include "api/units/frequency.h"
@@ -183,7 +183,7 @@
using WaitResult =
absl::variant<std::unique_ptr<EncodedFrame>, TimeDelta /*wait_time*/>;
- absl::optional<WaitResult> WaitForFrameOrTimeout(TimeDelta wait) {
+ std::optional<WaitResult> WaitForFrameOrTimeout(TimeDelta wait) {
if (wait_result_) {
return std::move(wait_result_);
}
@@ -245,7 +245,7 @@
}
uint32_t dropped_frames_ = 0;
- absl::optional<WaitResult> wait_result_;
+ std::optional<WaitResult> wait_result_;
};
class VideoStreamBufferControllerTest
@@ -260,7 +260,7 @@
// No new timeout set since receiver has not started new decode.
ResetLastResult();
- EXPECT_THAT(WaitForFrameOrTimeout(kMaxWaitForKeyframe), Eq(absl::nullopt));
+ EXPECT_THAT(WaitForFrameOrTimeout(kMaxWaitForKeyframe), Eq(std::nullopt));
// Now that receiver has asked for new frame, a new timeout can occur.
StartNextDecodeForceKeyframe();
@@ -362,7 +362,7 @@
buffer_->Stop();
// Wait for 2x max wait time. Since we stopped, this should cause no timeouts
// or frame-ready callbacks.
- EXPECT_THAT(WaitForFrameOrTimeout(kMaxWaitForFrame * 2), Eq(absl::nullopt));
+ EXPECT_THAT(WaitForFrameOrTimeout(kMaxWaitForFrame * 2), Eq(std::nullopt));
}
TEST_P(VideoStreamBufferControllerTest, FramesWaitForDecoderToComplete) {
@@ -383,7 +383,7 @@
// Advancing time should not result in a frame since the scheduler has not
// been signalled that we are ready.
- EXPECT_THAT(WaitForFrameOrTimeout(kFps30Delay), Eq(absl::nullopt));
+ EXPECT_THAT(WaitForFrameOrTimeout(kFps30Delay), Eq(std::nullopt));
// Signal ready.
StartNextDecode();
EXPECT_THAT(WaitForFrameOrTimeout(kFps30Delay), Frame(test::WithId(1)));
@@ -566,7 +566,7 @@
.AsLast()
.Build()));
StartNextDecode();
- EXPECT_THAT(WaitForFrameOrTimeout(TimeDelta::Zero()), Eq(absl::nullopt));
+ EXPECT_THAT(WaitForFrameOrTimeout(TimeDelta::Zero()), Eq(std::nullopt));
// Scheduler is waiting to deliver Frame 1 now. Insert Frame 2. Frame 1 should
// be delivered still.
@@ -747,7 +747,7 @@
// Avoid timeout being set while waiting for the frame and before the receiver
// is ready.
ResetLastResult();
- EXPECT_THAT(WaitForFrameOrTimeout(kMaxWaitForFrame), Eq(absl::nullopt));
+ EXPECT_THAT(WaitForFrameOrTimeout(kMaxWaitForFrame), Eq(std::nullopt));
time_controller_.AdvanceTime(kRolloverDelay - kMaxWaitForFrame);
StartNextDecode();
buffer_->InsertFrame(test::FakeFrameBuilder()
@@ -844,7 +844,7 @@
.Build();
buffer_->InsertFrame(std::move(frame));
// Pacing is set to 16ms in the field trial so we should not decode yet.
- EXPECT_THAT(WaitForFrameOrTimeout(TimeDelta::Zero()), Eq(absl::nullopt));
+ EXPECT_THAT(WaitForFrameOrTimeout(TimeDelta::Zero()), Eq(std::nullopt));
time_controller_.AdvanceTime(TimeDelta::Millis(16));
EXPECT_THAT(WaitForFrameOrTimeout(TimeDelta::Zero()), Frame(test::WithId(1)));
}
diff --git a/video/video_stream_decoder2.cc b/video/video_stream_decoder2.cc
index 5640835..df4f3530 100644
--- a/video/video_stream_decoder2.cc
+++ b/video/video_stream_decoder2.cc
@@ -44,7 +44,7 @@
// thread may have held the lock when calling VideoDecoder::Decode, Reset, or
// Release. Acquiring the same lock in the path of decode callback can deadlock.
int32_t VideoStreamDecoder::FrameToRender(VideoFrame& video_frame,
- absl::optional<uint8_t> qp,
+ std::optional<uint8_t> qp,
TimeDelta decode_time,
VideoContentType content_type,
VideoFrameType frame_type) {
diff --git a/video/video_stream_decoder2.h b/video/video_stream_decoder2.h
index 19db810..7847025 100644
--- a/video/video_stream_decoder2.h
+++ b/video/video_stream_decoder2.h
@@ -41,7 +41,7 @@
// Implements VCMReceiveCallback.
int32_t FrameToRender(VideoFrame& video_frame,
- absl::optional<uint8_t> qp,
+ std::optional<uint8_t> qp,
TimeDelta decode_time,
VideoContentType content_type,
VideoFrameType frame_type) override;
diff --git a/video/video_stream_encoder.cc b/video/video_stream_encoder.cc
index 4579702..821a1db 100644
--- a/video/video_stream_encoder.cc
+++ b/video/video_stream_encoder.cc
@@ -15,11 +15,11 @@
#include <limits>
#include <memory>
#include <numeric>
+#include <optional>
#include <utility>
#include "absl/algorithm/container.h"
#include "absl/cleanup/cleanup.h"
-#include "absl/types/optional.h"
#include "api/field_trials_view.h"
#include "api/sequence_checker.h"
#include "api/task_queue/task_queue_base.h"
@@ -91,7 +91,7 @@
}
}
-absl::optional<EncodedImageCallback::DropReason> MaybeConvertDropReason(
+std::optional<EncodedImageCallback::DropReason> MaybeConvertDropReason(
VideoStreamEncoderObserver::DropReason reason) {
switch (reason) {
case VideoStreamEncoderObserver::DropReason::kMediaOptimization:
@@ -99,7 +99,7 @@
case VideoStreamEncoderObserver::DropReason::kEncoder:
return EncodedImageCallback::DropReason::kDroppedByEncoder;
default:
- return absl::nullopt;
+ return std::nullopt;
}
}
@@ -403,18 +403,18 @@
}
// Get bitrate limits for active stream.
- absl::optional<uint32_t> pixels =
+ std::optional<uint32_t> pixels =
VideoStreamAdapter::GetSingleActiveLayerPixels(*codec);
if (!pixels.has_value()) {
return;
}
- absl::optional<VideoEncoder::ResolutionBitrateLimits> bitrate_limits =
+ std::optional<VideoEncoder::ResolutionBitrateLimits> bitrate_limits =
encoder_info.GetEncoderBitrateLimitsForResolution(*pixels);
if (!bitrate_limits.has_value()) {
return;
}
// Index for the active stream.
- absl::optional<size_t> index;
+ std::optional<size_t> index;
for (size_t i = 0; i < encoder_config.simulcast_layers.size(); ++i) {
if (encoder_config.simulcast_layers[i].active)
index = i;
@@ -478,7 +478,7 @@
}
// Get bitrate limits for active stream.
- absl::optional<VideoEncoder::ResolutionBitrateLimits> encoder_bitrate_limits =
+ std::optional<VideoEncoder::ResolutionBitrateLimits> encoder_bitrate_limits =
encoder_info.GetEncoderBitrateLimitsForResolution(
(*streams)[index].width * (*streams)[index].height);
if (!encoder_bitrate_limits) {
@@ -516,19 +516,19 @@
encoder_bitrate_limits->max_bitrate_bps);
}
-absl::optional<int> ParseVp9LowTierCoreCountThreshold(
+std::optional<int> ParseVp9LowTierCoreCountThreshold(
const FieldTrialsView& trials) {
FieldTrialFlag disable_low_tier("Disabled");
FieldTrialParameter<int> max_core_count("max_core_count", 2);
ParseFieldTrial({&disable_low_tier, &max_core_count},
trials.Lookup("WebRTC-VP9-LowTierOptimizations"));
if (disable_low_tier.Get()) {
- return absl::nullopt;
+ return std::nullopt;
}
return max_core_count.Get();
}
-absl::optional<int> ParseEncoderThreadLimit(const FieldTrialsView& trials) {
+std::optional<int> ParseEncoderThreadLimit(const FieldTrialsView& trials) {
FieldTrialOptional<int> encoder_thread_limit("encoder_thread_limit");
ParseFieldTrial({&encoder_thread_limit},
trials.Lookup("WebRTC-VideoEncoderSettings"));
@@ -847,8 +847,8 @@
RTC_DCHECK_RUN_ON(encoder_queue_.get());
RTC_LOG(LS_INFO) << "SetStartBitrate " << start_bitrate_bps;
encoder_target_bitrate_bps_ =
- start_bitrate_bps != 0 ? absl::optional<uint32_t>(start_bitrate_bps)
- : absl::nullopt;
+ start_bitrate_bps != 0 ? std::optional<uint32_t>(start_bitrate_bps)
+ : std::nullopt;
stream_resource_manager_.SetStartBitrate(
DataRate::BitsPerSec(start_bitrate_bps));
});
@@ -870,7 +870,7 @@
// Is any layer active.
bool active = false;
// The max requested_resolution.
- absl::optional<rtc::VideoSinkWants::FrameSize> requested_resolution;
+ std::optional<rtc::VideoSinkWants::FrameSize> requested_resolution;
for (const auto& stream : config.simulcast_layers) {
active |= stream.active;
if (stream.active) {
@@ -900,7 +900,7 @@
if (max_framerate >= 0) {
video_source_sink_controller_.SetFrameRateUpperLimit(max_framerate);
} else {
- video_source_sink_controller_.SetFrameRateUpperLimit(absl::nullopt);
+ video_source_sink_controller_.SetFrameRateUpperLimit(std::nullopt);
}
video_source_sink_controller_.SetActive(active);
video_source_sink_controller_.PushSourceSinkSettings();
@@ -924,7 +924,7 @@
frame_cadence_adapter_->SetZeroHertzModeEnabled(
FrameCadenceAdapterInterface::ZeroHertzModeParams{});
} else {
- frame_cadence_adapter_->SetZeroHertzModeEnabled(absl::nullopt);
+ frame_cadence_adapter_->SetZeroHertzModeEnabled(std::nullopt);
}
pending_encoder_creation_ =
@@ -989,7 +989,7 @@
// Possibly adjusts scale_resolution_down_by in `encoder_config_` to limit the
// alignment value.
AlignmentAdjuster::GetAlignmentAndMaybeAdjustScaleFactors(
- encoder_->GetEncoderInfo(), &encoder_config_, absl::nullopt);
+ encoder_->GetEncoderInfo(), &encoder_config_, std::nullopt);
std::vector<VideoStream> streams;
if (encoder_config_.video_stream_factory) {
@@ -1059,7 +1059,7 @@
// to get a certain bitrate for certain pixel_count. It also doesn't work
// for 960*540 and 640*520, we will nerver be stable at 640*520 due to their
// |target_bitrate_bps| are both 2000Kbps.
- absl::optional<VideoEncoder::ResolutionBitrateLimits>
+ std::optional<VideoEncoder::ResolutionBitrateLimits>
qp_untrusted_bitrate_limit = EncoderInfoSettings::
GetSinglecastBitrateLimitForResolutionWhenQpIsUntrusted(
last_frame_info_->width * last_frame_info_->height,
@@ -1086,7 +1086,7 @@
}
}
} else {
- absl::optional<VideoEncoder::ResolutionBitrateLimits>
+ std::optional<VideoEncoder::ResolutionBitrateLimits>
encoder_bitrate_limits =
encoder_->GetEncoderInfo().GetEncoderBitrateLimitsForResolution(
last_frame_info_->width * last_frame_info_->height);
@@ -1107,7 +1107,7 @@
int max_bitrate_bps;
// The API max bitrate comes from both `encoder_config_.max_bitrate_bps`
// and `encoder_config_.simulcast_layers[0].max_bitrate_bps`.
- absl::optional<int> api_max_bitrate_bps;
+ std::optional<int> api_max_bitrate_bps;
if (encoder_config_.simulcast_layers[0].max_bitrate_bps > 0) {
api_max_bitrate_bps =
encoder_config_.simulcast_layers[0].max_bitrate_bps;
@@ -1172,7 +1172,7 @@
rtc::SimpleStringBuilder log_stream(log_stream_buf);
log_stream << "ReconfigureEncoder: simulcast streams: ";
for (size_t i = 0; i < codec.numberOfSimulcastStreams; ++i) {
- absl::optional<ScalabilityMode> scalability_mode =
+ std::optional<ScalabilityMode> scalability_mode =
codec.simulcastStream[i].GetScalabilityMode();
if (scalability_mode) {
log_stream << "{" << i << ": " << codec.simulcastStream[i].width << "x"
@@ -1294,7 +1294,7 @@
}
frame_encode_metadata_writer_.Reset();
- last_encode_info_ms_ = absl::nullopt;
+ last_encode_info_ms_ = std::nullopt;
was_encode_called_since_last_initialization_ = false;
}
@@ -1446,7 +1446,7 @@
// If encoder selector is available, switch to the encoder it prefers.
// Otherwise try switching to VP8 (default WebRTC codec).
- absl::optional<SdpVideoFormat> preferred_fallback_encoder;
+ std::optional<SdpVideoFormat> preferred_fallback_encoder;
if (is_encoder_selector_available) {
preferred_fallback_encoder = encoder_selector_->OnEncoderBroken();
}
@@ -1644,9 +1644,9 @@
// This method may be called after we cleared out the frame_cadence_adapter_
// reference in Stop(). In such a situation it's probably not important with a
// decent estimate.
- absl::optional<uint32_t> input_fps =
+ std::optional<uint32_t> input_fps =
frame_cadence_adapter_ ? frame_cadence_adapter_->GetInputFrameRateFps()
- : absl::nullopt;
+ : std::nullopt;
if (!input_fps || *input_fps == 0) {
return default_fps;
}
@@ -2337,7 +2337,7 @@
stream_resource_manager_.UseBandwidthAllocationBps().value_or(
encoder_target_bitrate_bps_.value());
- absl::optional<VideoEncoder::ResolutionBitrateLimits> encoder_bitrate_limits =
+ std::optional<VideoEncoder::ResolutionBitrateLimits> encoder_bitrate_limits =
GetEncoderInfoWithBitrateLimitUpdate(
encoder_->GetEncoderInfo(), encoder_config_, default_limits_allowed_)
.GetEncoderBitrateLimitsForResolution(pixel_count);
@@ -2374,7 +2374,7 @@
bool max_pixels_updated =
(latest_restrictions_.has_value()
? latest_restrictions_->max_pixels_per_frame()
- : absl::nullopt) != restrictions.max_pixels_per_frame();
+ : std::nullopt) != restrictions.max_pixels_per_frame();
// TODO(webrtc:14451) Split video_source_sink_controller_
// so that ownership on restrictions/wants is kept on &encoder_queue_
@@ -2412,7 +2412,7 @@
RTC_DCHECK_RUN_ON(encoder_queue_.get());
- absl::optional<int> encode_duration_us;
+ std::optional<int> encode_duration_us;
if (encoded_image.timing_.flags != VideoSendTiming::kInvalid) {
encode_duration_us =
TimeDelta::Millis(encoded_image.timing_.encode_finish_ms -
diff --git a/video/video_stream_encoder.h b/video/video_stream_encoder.h
index e593f91..dd423c3 100644
--- a/video/video_stream_encoder.h
+++ b/video/video_stream_encoder.h
@@ -309,14 +309,13 @@
absl::InlinedVector<SetParametersCallback, 2> encoder_configuration_callbacks_
RTC_GUARDED_BY(encoder_queue_);
- absl::optional<VideoFrameInfo> last_frame_info_
- RTC_GUARDED_BY(encoder_queue_);
+ std::optional<VideoFrameInfo> last_frame_info_ RTC_GUARDED_BY(encoder_queue_);
int crop_width_ RTC_GUARDED_BY(encoder_queue_) = 0;
int crop_height_ RTC_GUARDED_BY(encoder_queue_) = 0;
- absl::optional<uint32_t> encoder_target_bitrate_bps_
+ std::optional<uint32_t> encoder_target_bitrate_bps_
RTC_GUARDED_BY(encoder_queue_);
size_t max_data_payload_length_ RTC_GUARDED_BY(encoder_queue_) = 0;
- absl::optional<EncoderRateSettings> last_encoder_rate_settings_
+ std::optional<EncoderRateSettings> last_encoder_rate_settings_
RTC_GUARDED_BY(encoder_queue_);
bool encoder_paused_and_dropped_frame_ RTC_GUARDED_BY(encoder_queue_) = false;
@@ -336,7 +335,7 @@
int captured_frame_count_ RTC_GUARDED_BY(encoder_queue_) = 0;
int dropped_frame_cwnd_pushback_count_ RTC_GUARDED_BY(encoder_queue_) = 0;
int dropped_frame_encoder_block_count_ RTC_GUARDED_BY(encoder_queue_) = 0;
- absl::optional<VideoFrame> pending_frame_ RTC_GUARDED_BY(encoder_queue_);
+ std::optional<VideoFrame> pending_frame_ RTC_GUARDED_BY(encoder_queue_);
int64_t pending_frame_post_time_us_ RTC_GUARDED_BY(encoder_queue_) = 0;
VideoFrame::UpdateRect accumulated_update_rect_
@@ -345,9 +344,9 @@
FecControllerOverride* fec_controller_override_
RTC_GUARDED_BY(encoder_queue_) = nullptr;
- absl::optional<int64_t> last_parameters_update_ms_
+ std::optional<int64_t> last_parameters_update_ms_
RTC_GUARDED_BY(encoder_queue_);
- absl::optional<int64_t> last_encode_info_ms_ RTC_GUARDED_BY(encoder_queue_);
+ std::optional<int64_t> last_encode_info_ms_ RTC_GUARDED_BY(encoder_queue_);
VideoEncoder::EncoderInfo encoder_info_ RTC_GUARDED_BY(encoder_queue_);
VideoCodec send_codec_ RTC_GUARDED_BY(encoder_queue_);
@@ -366,7 +365,7 @@
// Congestion window frame drop ratio (drop 1 in every
// cwnd_frame_drop_interval_ frames).
- absl::optional<int> cwnd_frame_drop_interval_ RTC_GUARDED_BY(encoder_queue_);
+ std::optional<int> cwnd_frame_drop_interval_ RTC_GUARDED_BY(encoder_queue_);
// Frame counter for congestion window frame drop.
int cwnd_frame_counter_ RTC_GUARDED_BY(encoder_queue_) = 0;
@@ -428,8 +427,8 @@
// Enables encoder switching on initialization failures.
bool switch_encoder_on_init_failures_;
- const absl::optional<int> vp9_low_tier_core_threshold_;
- const absl::optional<int> experimental_encoder_thread_limit_;
+ const std::optional<int> vp9_low_tier_core_threshold_;
+ const std::optional<int> experimental_encoder_thread_limit_;
// This is a copy of restrictions (glorified max_pixel_count) set by
// OnVideoSourceRestrictionsUpdated. It is used to scale down encoding
@@ -438,7 +437,7 @@
// TODO(webrtc:14451) Split video_source_sink_controller_
// so that ownership on restrictions/wants is kept on &encoder_queue_, that
// these extra copies would not be needed.
- absl::optional<VideoSourceRestrictions> latest_restrictions_
+ std::optional<VideoSourceRestrictions> latest_restrictions_
RTC_GUARDED_BY(encoder_queue_);
// Used to cancel any potentially pending tasks to the worker thread.
diff --git a/video/video_stream_encoder_unittest.cc b/video/video_stream_encoder_unittest.cc
index e208501..46df928 100644
--- a/video/video_stream_encoder_unittest.cc
+++ b/video/video_stream_encoder_unittest.cc
@@ -300,7 +300,7 @@
return AllOf(
WantsMaxPixels(Eq(std::numeric_limits<int>::max())),
Field("target_pixel_count", &rtc::VideoSinkWants::target_pixel_count,
- Eq(absl::nullopt)));
+ Eq(std::nullopt)));
}
auto FpsMax() {
@@ -553,8 +553,8 @@
return last_wants_;
}
- absl::optional<int> last_sent_width() const { return last_width_; }
- absl::optional<int> last_sent_height() const { return last_height_; }
+ std::optional<int> last_sent_width() const { return last_width_; }
+ std::optional<int> last_sent_height() const { return last_height_; }
void IncomingCapturedFrame(const VideoFrame& video_frame) override {
RTC_DCHECK(time_controller_->GetMainThread()->IsCurrent());
@@ -591,8 +591,8 @@
last_width_.emplace(adapted_frame.width());
last_height_.emplace(adapted_frame.height());
} else {
- last_width_ = absl::nullopt;
- last_height_ = absl::nullopt;
+ last_width_ = std::nullopt;
+ last_height_ = std::nullopt;
}
} else {
RTC_DLOG(LS_INFO) << "IncomingCapturedFrame: adaptation not enabled";
@@ -603,10 +603,10 @@
}
void OnOutputFormatRequest(int width, int height) {
- absl::optional<std::pair<int, int>> target_aspect_ratio =
+ std::optional<std::pair<int, int>> target_aspect_ratio =
std::make_pair(width, height);
- absl::optional<int> max_pixel_count = width * height;
- absl::optional<int> max_fps;
+ std::optional<int> max_pixel_count = width * height;
+ std::optional<int> max_fps;
adapter_.OnOutputFormatRequest(target_aspect_ratio, max_pixel_count,
max_fps);
}
@@ -632,8 +632,8 @@
cricket::VideoAdapter adapter_;
bool adaptation_enabled_ RTC_GUARDED_BY(mutex_);
rtc::VideoSinkWants last_wants_ RTC_GUARDED_BY(mutex_);
- absl::optional<int> last_width_;
- absl::optional<int> last_height_;
+ std::optional<int> last_width_;
+ std::optional<int> last_height_;
int refresh_frames_requested_{0};
};
@@ -681,7 +681,7 @@
}
mutable Mutex lock_;
- absl::optional<VideoSendStream::Stats> mock_stats_ RTC_GUARDED_BY(lock_);
+ std::optional<VideoSendStream::Stats> mock_stats_ RTC_GUARDED_BY(lock_);
std::function<void(DropReason)> on_frame_dropped_;
};
@@ -797,10 +797,10 @@
MOCK_METHOD(void, Initialize, (Callback * callback), (override));
MOCK_METHOD(void,
SetZeroHertzModeEnabled,
- (absl::optional<ZeroHertzModeParams>),
+ (std::optional<ZeroHertzModeParams>),
(override));
MOCK_METHOD(void, OnFrame, (const VideoFrame&), (override));
- MOCK_METHOD(absl::optional<uint32_t>, GetInputFrameRateFps, (), (override));
+ MOCK_METHOD(std::optional<uint32_t>, GetInputFrameRateFps, (), (override));
MOCK_METHOD(void,
UpdateLayerQualityConvergence,
(size_t spatial_index, bool converged),
@@ -811,7 +811,7 @@
(override));
MOCK_METHOD(void,
UpdateVideoSourceRestrictions,
- (absl::optional<double>),
+ (std::optional<double>),
(override));
MOCK_METHOD(void, ProcessKeyFrameRequest, (), (override));
};
@@ -823,15 +823,15 @@
OnCurrentEncoder,
(const SdpVideoFormat& format),
(override));
- MOCK_METHOD(absl::optional<SdpVideoFormat>,
+ MOCK_METHOD(std::optional<SdpVideoFormat>,
OnAvailableBitrate,
(const DataRate& rate),
(override));
- MOCK_METHOD(absl::optional<SdpVideoFormat>,
+ MOCK_METHOD(std::optional<SdpVideoFormat>,
OnResolutionChange,
(const RenderResolution& resolution),
(override));
- MOCK_METHOD(absl::optional<SdpVideoFormat>, OnEncoderBroken, (), (override));
+ MOCK_METHOD(std::optional<SdpVideoFormat>, OnEncoderBroken, (), (override));
};
class MockVideoSourceInterface : public rtc::VideoSourceInterface<VideoFrame> {
@@ -922,7 +922,7 @@
size_t num_temporal_layers,
unsigned char num_spatial_layers,
bool screenshare,
- absl::optional<int> max_frame_rate = kDefaultFramerate,
+ std::optional<int> max_frame_rate = kDefaultFramerate,
VideoStreamEncoder::BitrateAllocationCallbackType
allocation_callback_type =
VideoStreamEncoder::BitrateAllocationCallbackType::
@@ -1205,7 +1205,7 @@
expect_null_frame_ = true;
}
- absl::optional<VideoEncoder::RateControlParameters>
+ std::optional<VideoEncoder::RateControlParameters>
GetAndResetLastRateControlSettings() {
auto settings = last_rate_control_settings_;
last_rate_control_settings_.reset();
@@ -1222,7 +1222,7 @@
return last_input_height_;
}
- absl::optional<VideoFrameBuffer::Type> GetLastInputPixelFormat() {
+ std::optional<VideoFrameBuffer::Type> GetLastInputPixelFormat() {
MutexLock lock(&local_mutex_);
return last_input_pixel_format_;
}
@@ -1239,7 +1239,7 @@
preferred_pixel_formats_ = std::move(pixel_formats);
}
- void SetIsQpTrusted(absl::optional<bool> trusted) {
+ void SetIsQpTrusted(std::optional<bool> trusted) {
MutexLock lock(&local_mutex_);
is_qp_trusted_ = trusted;
}
@@ -1367,13 +1367,13 @@
RTC_GUARDED_BY(local_mutex_);
std::unique_ptr<Vp8FrameBufferController> frame_buffer_controller_
RTC_GUARDED_BY(local_mutex_);
- absl::optional<bool>
+ std::optional<bool>
temporal_layers_supported_[kMaxSpatialLayers] RTC_GUARDED_BY(
local_mutex_);
bool force_init_encode_failed_ RTC_GUARDED_BY(local_mutex_) = false;
double rate_factor_ RTC_GUARDED_BY(local_mutex_) = 1.0;
uint32_t last_framerate_ RTC_GUARDED_BY(local_mutex_) = 0;
- absl::optional<VideoEncoder::RateControlParameters>
+ std::optional<VideoEncoder::RateControlParameters>
last_rate_control_settings_;
VideoFrame::UpdateRect last_update_rect_ RTC_GUARDED_BY(local_mutex_) = {
0, 0, 0, 0};
@@ -1385,11 +1385,11 @@
std::vector<ResolutionBitrateLimits> resolution_bitrate_limits_
RTC_GUARDED_BY(local_mutex_);
int num_set_rates_ RTC_GUARDED_BY(local_mutex_) = 0;
- absl::optional<VideoFrameBuffer::Type> last_input_pixel_format_
+ std::optional<VideoFrameBuffer::Type> last_input_pixel_format_
RTC_GUARDED_BY(local_mutex_);
absl::InlinedVector<VideoFrameBuffer::Type, kMaxPreferredPixelFormats>
preferred_pixel_formats_ RTC_GUARDED_BY(local_mutex_);
- absl::optional<bool> is_qp_trusted_ RTC_GUARDED_BY(local_mutex_);
+ std::optional<bool> is_qp_trusted_ RTC_GUARDED_BY(local_mutex_);
VideoCodecComplexity last_encoder_complexity_ RTC_GUARDED_BY(local_mutex_){
VideoCodecComplexity::kComplexityNormal};
};
@@ -2300,7 +2300,7 @@
video_stream_encoder_->ConfigureEncoder(config.Copy(), kMaxPayloadLength);
// Default bitrate limits for 270p should be used.
- const absl::optional<VideoEncoder::ResolutionBitrateLimits>
+ const std::optional<VideoEncoder::ResolutionBitrateLimits>
kDefaultLimits270p =
EncoderInfoSettings::GetDefaultSinglecastBitrateLimitsForResolution(
kVideoCodecVP8, 480 * 270);
@@ -2313,7 +2313,7 @@
fake_encoder_.config().simulcastStream[1].maxBitrate * 1000);
// Default bitrate limits for 360p should be used.
- const absl::optional<VideoEncoder::ResolutionBitrateLimits>
+ const std::optional<VideoEncoder::ResolutionBitrateLimits>
kDefaultLimits360p =
EncoderInfoSettings::GetDefaultSinglecastBitrateLimitsForResolution(
kVideoCodecVP8, 640 * 360);
@@ -2334,7 +2334,7 @@
fake_encoder_.config().simulcastStream[1].maxBitrate * 1000);
// Default bitrate limits for 540p should be used.
- const absl::optional<VideoEncoder::ResolutionBitrateLimits>
+ const std::optional<VideoEncoder::ResolutionBitrateLimits>
kDefaultLimits540p =
EncoderInfoSettings::GetDefaultSinglecastBitrateLimitsForResolution(
kVideoCodecVP8, 960 * 540);
@@ -6066,7 +6066,7 @@
kMaxPayloadLength);
// The default bitrate limits for 360p should be used.
- const absl::optional<VideoEncoder::ResolutionBitrateLimits> kLimits360p =
+ const std::optional<VideoEncoder::ResolutionBitrateLimits> kLimits360p =
EncoderInfoSettings::GetDefaultSinglecastBitrateLimitsForResolution(
kVideoCodecVP9, 640 * 360);
video_source_.IncomingCapturedFrame(CreateFrame(1, 1280, 720));
@@ -6083,7 +6083,7 @@
fake_encoder_.config().spatialLayers[0].maxBitrate * 1000);
// The default bitrate limits for 270p should be used.
- const absl::optional<VideoEncoder::ResolutionBitrateLimits> kLimits270p =
+ const std::optional<VideoEncoder::ResolutionBitrateLimits> kLimits270p =
EncoderInfoSettings::GetDefaultSinglecastBitrateLimitsForResolution(
kVideoCodecVP9, 480 * 270);
video_source_.IncomingCapturedFrame(CreateFrame(2, 960, 540));
@@ -6131,7 +6131,7 @@
kMaxPayloadLength);
// The default bitrate limits for 360p should not be used.
- const absl::optional<VideoEncoder::ResolutionBitrateLimits> kLimits360p =
+ const std::optional<VideoEncoder::ResolutionBitrateLimits> kLimits360p =
EncoderInfoSettings::GetDefaultSinglecastBitrateLimitsForResolution(
kVideoCodecVP9, 640 * 360);
video_source_.IncomingCapturedFrame(CreateFrame(1, 1280, 720));
@@ -6153,7 +6153,7 @@
/*num_spatial_layers=*/1, /*screenshare=*/false);
// The default singlecast bitrate limits for 720p should not be used.
- const absl::optional<VideoEncoder::ResolutionBitrateLimits> kLimits720p =
+ const std::optional<VideoEncoder::ResolutionBitrateLimits> kLimits720p =
EncoderInfoSettings::GetDefaultSinglecastBitrateLimitsForResolution(
kVideoCodecVP9, 1280 * 720);
video_source_.IncomingCapturedFrame(CreateFrame(1, 1280, 720));
@@ -7820,7 +7820,7 @@
ConfigureEncoder(video_encoder_config_.Copy());
EXPECT_CALL(encoder_selector, OnResolutionChange(RenderResolution(640, 480)))
- .WillOnce(Return(absl::nullopt));
+ .WillOnce(Return(std::nullopt));
EXPECT_CALL(encoder_selector, OnResolutionChange(RenderResolution(320, 240)))
.WillOnce(Return(SdpVideoFormat("AV1")));
EXPECT_CALL(switch_callback,
@@ -8542,7 +8542,7 @@
// Set QP not trusted in encoder info.
fake_encoder_.SetIsQpTrusted(false);
- absl::optional<VideoEncoder::ResolutionBitrateLimits> suitable_bitrate_limit =
+ std::optional<VideoEncoder::ResolutionBitrateLimits> suitable_bitrate_limit =
EncoderInfoSettings::
GetSinglecastBitrateLimitForResolutionWhenQpIsUntrusted(
codec_width_ * codec_height_,
@@ -8583,7 +8583,7 @@
// Set QP not trusted in encoder info.
fake_encoder_.SetIsQpTrusted(false);
- absl::optional<VideoEncoder::ResolutionBitrateLimits> suitable_bitrate_limit =
+ std::optional<VideoEncoder::ResolutionBitrateLimits> suitable_bitrate_limit =
EncoderInfoSettings::
GetSinglecastBitrateLimitForResolutionWhenQpIsUntrusted(
codec_width_ * codec_height_,
@@ -9243,7 +9243,7 @@
stream.scale_resolution_down_by = 1.0;
stream.num_temporal_layers = 1;
stream.bitrate_priority = 1.0;
- stream.scalability_mode = absl::nullopt;
+ stream.scalability_mode = std::nullopt;
return stream;
}
@@ -9418,7 +9418,7 @@
Mock::VerifyAndClearExpectations(adapter_ptr);
// Expect a disabled zero-hertz mode after passing realtime video.
- EXPECT_CALL(*adapter_ptr, SetZeroHertzModeEnabled(Eq(absl::nullopt)));
+ EXPECT_CALL(*adapter_ptr, SetZeroHertzModeEnabled(Eq(std::nullopt)));
VideoEncoderConfig config2;
test::FillEncoderConfiguration(kVideoCodecVP8, 1, &config2);
config2.content_type = VideoEncoderConfig::ContentType::kRealtimeVideo;
@@ -9715,7 +9715,7 @@
TEST_F(VideoStreamEncoderFrameCadenceRestrictionTest,
UpdatesVideoSourceRestrictionsUnRestricted) {
- EXPECT_CALL(*adapter_ptr_, UpdateVideoSourceRestrictions(Eq(absl::nullopt)));
+ EXPECT_CALL(*adapter_ptr_, UpdateVideoSourceRestrictions(Eq(std::nullopt)));
UpdateVideoSourceRestrictions(VideoSourceRestrictions());
}
@@ -9732,7 +9732,7 @@
// FPS is unlimited.
restrictions_.set_max_pixels_per_frame(99);
restrictions_.set_target_pixels_per_frame(101);
- EXPECT_CALL(*adapter_ptr_, UpdateVideoSourceRestrictions(Eq(absl::nullopt)));
+ EXPECT_CALL(*adapter_ptr_, UpdateVideoSourceRestrictions(Eq(std::nullopt)));
UpdateVideoSourceRestrictions(restrictions_);
}
