Replace rtc::Optional with absl::optional in api
This is a no-op change because rtc::Optional is an alias to absl::optional
This CL generated by running script with parameter 'api'
Then undo changes to optional target itself and optional_unittests
find $@ -type f \( -name \*.h -o -name \*.cc \) \
-exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+
find $@ -type f -name BUILD.gn \
-exec sed -r -i 's|"[\./api]*:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+;
git cl format
Bug: webrtc:9078
Change-Id: I44093da213369d6a502e33792c694f620f53b779
Reviewed-on: https://webrtc-review.googlesource.com/84621
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23707}diff --git a/api/BUILD.gn b/api/BUILD.gn
index 6e79b27..41e1ad8 100644
--- a/api/BUILD.gn
+++ b/api/BUILD.gn
@@ -101,13 +101,13 @@
":callfactory_api",
":fec_controller_api",
":libjingle_logging_api",
- ":optional",
":rtc_stats_api",
"audio:audio_mixer_api",
"audio_codecs:audio_codecs_api",
"transport:bitrate_settings",
"transport:network_control",
"video:video_frame",
+ "//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
@@ -160,9 +160,9 @@
# libjingle_peerconnection_api.
deps = [
":libjingle_peerconnection_api",
- ":optional",
"..:webrtc_common",
"../rtc_base:rtc_base",
+ "//third_party/abseil-cpp/absl/types:optional",
]
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
@@ -194,8 +194,8 @@
]
deps = [
- ":optional",
"../rtc_base:rtc_base_approved",
+ "//third_party/abseil-cpp/absl/types:optional",
]
}
diff --git a/api/audio/audio_frame.h b/api/audio/audio_frame.h
index d80868a..4c9aa9c 100644
--- a/api/audio/audio_frame.h
+++ b/api/audio/audio_frame.h
@@ -110,7 +110,7 @@
// Monotonically increasing timestamp intended for profiling of audio frames.
// 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. Also, rtc::Optional is not used since it will cause a "complex
+ // by design. Also, absl::optional is not used since it will cause a "complex
// class/struct needs an explicit out-of-line destructor" build error.
int64_t profile_timestamp_ms_ = 0;
diff --git a/api/audio_codecs/BUILD.gn b/api/audio_codecs/BUILD.gn
index 3206a74..7895a93 100644
--- a/api/audio_codecs/BUILD.gn
+++ b/api/audio_codecs/BUILD.gn
@@ -30,13 +30,13 @@
]
deps = [
"..:array_view",
- "..:optional",
"../..:webrtc_common",
"../../:typedefs",
"../../rtc_base:checks",
"../../rtc_base:deprecation",
"../../rtc_base:rtc_base_approved",
"../../rtc_base:sanitizer",
+ "//third_party/abseil-cpp/absl/types:optional",
]
}
diff --git a/api/audio_codecs/L16/BUILD.gn b/api/audio_codecs/L16/BUILD.gn
index 01554aa..043d659 100644
--- a/api/audio_codecs/L16/BUILD.gn
+++ b/api/audio_codecs/L16/BUILD.gn
@@ -21,10 +21,10 @@
]
deps = [
"..:audio_codecs_api",
- "../..:optional",
"../../..:webrtc_common",
"../../../modules/audio_coding:pcm16b",
"../../../rtc_base:rtc_base_approved",
+ "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -37,9 +37,9 @@
]
deps = [
"..:audio_codecs_api",
- "../..:optional",
"../../..:webrtc_common",
"../../../modules/audio_coding:pcm16b",
"../../../rtc_base:rtc_base_approved",
+ "//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 7c6a9ee..a17dc58 100644
--- a/api/audio_codecs/L16/audio_decoder_L16.cc
+++ b/api/audio_codecs/L16/audio_decoder_L16.cc
@@ -18,14 +18,14 @@
namespace webrtc {
-rtc::Optional<AudioDecoderL16::Config> AudioDecoderL16::SdpToConfig(
+absl::optional<AudioDecoderL16::Config> AudioDecoderL16::SdpToConfig(
const SdpAudioFormat& format) {
Config config;
config.sample_rate_hz = format.clockrate_hz;
config.num_channels = rtc::checked_cast<int>(format.num_channels);
return STR_CASE_CMP(format.name.c_str(), "L16") == 0 && config.IsOk()
- ? rtc::Optional<Config>(config)
- : rtc::nullopt;
+ ? absl::optional<Config>(config)
+ : absl::nullopt;
}
void AudioDecoderL16::AppendSupportedDecoders(
@@ -35,7 +35,7 @@
std::unique_ptr<AudioDecoder> AudioDecoderL16::MakeAudioDecoder(
const Config& config,
- rtc::Optional<AudioCodecPairId> /*codec_pair_id*/) {
+ absl::optional<AudioCodecPairId> /*codec_pair_id*/) {
return config.IsOk() ? rtc::MakeUnique<AudioDecoderPcm16B>(
config.sample_rate_hz, config.num_channels)
: nullptr;
diff --git a/api/audio_codecs/L16/audio_decoder_L16.h b/api/audio_codecs/L16/audio_decoder_L16.h
index deef909..184ec24 100644
--- a/api/audio_codecs/L16/audio_decoder_L16.h
+++ b/api/audio_codecs/L16/audio_decoder_L16.h
@@ -14,10 +14,10 @@
#include <memory>
#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"
-#include "api/optional.h"
namespace webrtc {
@@ -35,11 +35,11 @@
int sample_rate_hz = 8000;
int num_channels = 1;
};
- static rtc::Optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
+ static absl::optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
static void AppendSupportedDecoders(std::vector<AudioCodecSpec>* specs);
static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
const Config& config,
- rtc::Optional<AudioCodecPairId> codec_pair_id = rtc::nullopt);
+ absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt);
};
} // namespace webrtc
diff --git a/api/audio_codecs/L16/audio_encoder_L16.cc b/api/audio_codecs/L16/audio_encoder_L16.cc
index 5022993..d80e6bf 100644
--- a/api/audio_codecs/L16/audio_encoder_L16.cc
+++ b/api/audio_codecs/L16/audio_encoder_L16.cc
@@ -18,17 +18,17 @@
namespace webrtc {
-rtc::Optional<AudioEncoderL16::Config> AudioEncoderL16::SdpToConfig(
+absl::optional<AudioEncoderL16::Config> AudioEncoderL16::SdpToConfig(
const SdpAudioFormat& format) {
if (!rtc::IsValueInRangeForNumericType<int>(format.num_channels)) {
- return rtc::nullopt;
+ return absl::nullopt;
}
Config config;
config.sample_rate_hz = format.clockrate_hz;
config.num_channels = rtc::dchecked_cast<int>(format.num_channels);
return STR_CASE_CMP(format.name.c_str(), "L16") == 0 && config.IsOk()
- ? rtc::Optional<Config>(config)
- : rtc::nullopt;
+ ? absl::optional<Config>(config)
+ : absl::nullopt;
}
void AudioEncoderL16::AppendSupportedEncoders(
@@ -47,7 +47,7 @@
std::unique_ptr<AudioEncoder> AudioEncoderL16::MakeAudioEncoder(
const AudioEncoderL16::Config& config,
int payload_type,
- rtc::Optional<AudioCodecPairId> /*codec_pair_id*/) {
+ absl::optional<AudioCodecPairId> /*codec_pair_id*/) {
RTC_DCHECK(config.IsOk());
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 08d7ef1..340e3af 100644
--- a/api/audio_codecs/L16/audio_encoder_L16.h
+++ b/api/audio_codecs/L16/audio_encoder_L16.h
@@ -14,10 +14,10 @@
#include <memory>
#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"
-#include "api/optional.h"
namespace webrtc {
@@ -37,13 +37,13 @@
int num_channels = 1;
int frame_size_ms = 10;
};
- static rtc::Optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
+ static absl::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,
- rtc::Optional<AudioCodecPairId> codec_pair_id = rtc::nullopt);
+ absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt);
};
} // namespace webrtc
diff --git a/api/audio_codecs/audio_decoder.cc b/api/audio_codecs/audio_decoder.cc
index 4903fb6..00e45d9 100644
--- a/api/audio_codecs/audio_decoder.cc
+++ b/api/audio_codecs/audio_decoder.cc
@@ -33,14 +33,14 @@
return ret < 0 ? 0 : static_cast<size_t>(ret);
}
- rtc::Optional<DecodeResult> Decode(
+ absl::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 ? rtc::nullopt
- : rtc::Optional<DecodeResult>(
+ return ret < 0 ? absl::nullopt
+ : absl::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 021288f..4852ad7 100644
--- a/api/audio_codecs/audio_decoder.h
+++ b/api/audio_codecs/audio_decoder.h
@@ -14,8 +14,8 @@
#include <memory>
#include <vector>
+#include "absl/types/optional.h"
#include "api/array_view.h"
-#include "api/optional.h"
#include "rtc_base/buffer.h"
#include "rtc_base/constructormagic.h"
#include "typedefs.h" // NOLINT(build/include)
@@ -53,11 +53,11 @@
// 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 rtc::Optional containing the
+ // call to Duration() . On success, returns an absl::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
- // rtc::Optional. Decode may be called at most once per frame object.
- virtual rtc::Optional<DecodeResult> Decode(
+ // absl::optional. Decode may be called at most once per frame object.
+ virtual absl::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 fb1c965..90f93f0 100644
--- a/api/audio_codecs/audio_decoder_factory.h
+++ b/api/audio_codecs/audio_decoder_factory.h
@@ -14,10 +14,10 @@
#include <memory>
#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"
-#include "api/optional.h"
#include "rtc_base/refcount.h"
namespace webrtc {
@@ -41,7 +41,7 @@
// work.
virtual std::unique_ptr<AudioDecoder> MakeAudioDecoder(
const SdpAudioFormat& format,
- rtc::Optional<AudioCodecPairId> codec_pair_id) = 0;
+ absl::optional<AudioCodecPairId> codec_pair_id) = 0;
};
} // namespace webrtc
diff --git a/api/audio_codecs/audio_decoder_factory_template.h b/api/audio_codecs/audio_decoder_factory_template.h
index 4adac21..cdbe8bd 100644
--- a/api/audio_codecs/audio_decoder_factory_template.h
+++ b/api/audio_codecs/audio_decoder_factory_template.h
@@ -32,7 +32,7 @@
static bool IsSupportedDecoder(const SdpAudioFormat& format) { return false; }
static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
const SdpAudioFormat& format,
- rtc::Optional<AudioCodecPairId> codec_pair_id) {
+ absl::optional<AudioCodecPairId> codec_pair_id) {
return nullptr;
}
};
@@ -48,14 +48,14 @@
static bool IsSupportedDecoder(const SdpAudioFormat& format) {
auto opt_config = T::SdpToConfig(format);
static_assert(std::is_same<decltype(opt_config),
- rtc::Optional<typename T::Config>>::value,
+ absl::optional<typename T::Config>>::value,
"T::SdpToConfig() must return a value of type "
- "rtc::Optional<T::Config>");
+ "absl::optional<T::Config>");
return opt_config ? true : Helper<Ts...>::IsSupportedDecoder(format);
}
static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
const SdpAudioFormat& format,
- rtc::Optional<AudioCodecPairId> codec_pair_id) {
+ absl::optional<AudioCodecPairId> codec_pair_id) {
auto opt_config = T::SdpToConfig(format);
return opt_config ? T::MakeAudioDecoder(*opt_config, codec_pair_id)
: Helper<Ts...>::MakeAudioDecoder(format, codec_pair_id);
@@ -77,7 +77,7 @@
std::unique_ptr<AudioDecoder> MakeAudioDecoder(
const SdpAudioFormat& format,
- rtc::Optional<AudioCodecPairId> codec_pair_id) override {
+ absl::optional<AudioCodecPairId> codec_pair_id) override {
return Helper<Ts...>::MakeAudioDecoder(format, codec_pair_id);
}
};
@@ -92,7 +92,7 @@
// // Converts |audio_format| to a ConfigType instance. Returns an empty
// // optional if |audio_format| doesn't correctly specify an decoder of our
// // type.
-// rtc::Optional<ConfigType> SdpToConfig(const SdpAudioFormat& audio_format);
+// absl::optional<ConfigType> SdpToConfig(const SdpAudioFormat& audio_format);
//
// // Appends zero or more AudioCodecSpecs to the list that will be returned
// // by AudioDecoderFactory::GetSupportedDecoders().
@@ -102,7 +102,7 @@
// // AudioDecoderFactory::MakeAudioDecoder().
// std::unique_ptr<AudioDecoder> MakeAudioDecoder(
// const ConfigType& config,
-// rtc::Optional<AudioCodecPairId> codec_pair_id);
+// absl::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 4f9b9f0..595c111 100644
--- a/api/audio_codecs/audio_encoder.cc
+++ b/api/audio_codecs/audio_encoder.cc
@@ -85,12 +85,12 @@
float uplink_recoverable_packet_loss_fraction) {}
void AudioEncoder::OnReceivedTargetAudioBitrate(int target_audio_bitrate_bps) {
- OnReceivedUplinkBandwidth(target_audio_bitrate_bps, rtc::nullopt);
+ OnReceivedUplinkBandwidth(target_audio_bitrate_bps, absl::nullopt);
}
void AudioEncoder::OnReceivedUplinkBandwidth(
int target_audio_bitrate_bps,
- rtc::Optional<int64_t> bwe_period_ms) {}
+ absl::optional<int64_t> bwe_period_ms) {}
void AudioEncoder::OnReceivedRtt(int rtt_ms) {}
diff --git a/api/audio_codecs/audio_encoder.h b/api/audio_codecs/audio_encoder.h
index d277a19..2509401 100644
--- a/api/audio_codecs/audio_encoder.h
+++ b/api/audio_codecs/audio_encoder.h
@@ -16,8 +16,8 @@
#include <string>
#include <vector>
+#include "absl/types/optional.h"
#include "api/array_view.h"
-#include "api/optional.h"
#include "rtc_base/buffer.h"
#include "rtc_base/deprecation.h"
#include "typedefs.h" // NOLINT(build/include)
@@ -34,30 +34,30 @@
// 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.
- rtc::Optional<uint32_t> bitrate_action_counter;
+ absl::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.
- rtc::Optional<uint32_t> channel_action_counter;
+ absl::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.
- rtc::Optional<uint32_t> dtx_action_counter;
+ absl::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.
- rtc::Optional<uint32_t> fec_action_counter;
+ absl::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.
- rtc::Optional<uint32_t> frame_length_increase_counter;
+ absl::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.
- rtc::Optional<uint32_t> frame_length_decrease_counter;
+ absl::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.
- rtc::Optional<float> uplink_packet_loss_fraction;
+ absl::optional<float> uplink_packet_loss_fraction;
};
// This is the interface class for encoders in AudioCoding module. Each codec
@@ -221,7 +221,7 @@
// 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,
- rtc::Optional<int64_t> bwe_period_ms);
+ absl::optional<int64_t> bwe_period_ms);
// Provides RTT to this encoder to allow it to adapt.
virtual void OnReceivedRtt(int rtt_ms);
diff --git a/api/audio_codecs/audio_encoder_factory.h b/api/audio_codecs/audio_encoder_factory.h
index 7825953..fb4e23f 100644
--- a/api/audio_codecs/audio_encoder_factory.h
+++ b/api/audio_codecs/audio_encoder_factory.h
@@ -14,10 +14,10 @@
#include <memory>
#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"
-#include "api/optional.h"
#include "rtc_base/refcount.h"
namespace webrtc {
@@ -32,7 +32,7 @@
// 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 rtc::Optional<AudioCodecInfo> QueryAudioEncoder(
+ virtual absl::optional<AudioCodecInfo> QueryAudioEncoder(
const SdpAudioFormat& format) = 0;
// Creates an AudioEncoder for the specified format. The encoder will tags
@@ -50,7 +50,7 @@
virtual std::unique_ptr<AudioEncoder> MakeAudioEncoder(
int payload_type,
const SdpAudioFormat& format,
- rtc::Optional<AudioCodecPairId> codec_pair_id) = 0;
+ absl::optional<AudioCodecPairId> codec_pair_id) = 0;
};
} // namespace webrtc
diff --git a/api/audio_codecs/audio_encoder_factory_template.h b/api/audio_codecs/audio_encoder_factory_template.h
index f76677d..376b39e 100644
--- a/api/audio_codecs/audio_encoder_factory_template.h
+++ b/api/audio_codecs/audio_encoder_factory_template.h
@@ -29,14 +29,14 @@
template <>
struct Helper<> {
static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs) {}
- static rtc::Optional<AudioCodecInfo> QueryAudioEncoder(
+ static absl::optional<AudioCodecInfo> QueryAudioEncoder(
const SdpAudioFormat& format) {
- return rtc::nullopt;
+ return absl::nullopt;
}
static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
int payload_type,
const SdpAudioFormat& format,
- rtc::Optional<AudioCodecPairId> codec_pair_id) {
+ absl::optional<AudioCodecPairId> codec_pair_id) {
return nullptr;
}
};
@@ -49,21 +49,21 @@
T::AppendSupportedEncoders(specs);
Helper<Ts...>::AppendSupportedEncoders(specs);
}
- static rtc::Optional<AudioCodecInfo> QueryAudioEncoder(
+ static absl::optional<AudioCodecInfo> QueryAudioEncoder(
const SdpAudioFormat& format) {
auto opt_config = T::SdpToConfig(format);
static_assert(std::is_same<decltype(opt_config),
- rtc::Optional<typename T::Config>>::value,
+ absl::optional<typename T::Config>>::value,
"T::SdpToConfig() must return a value of type "
- "rtc::Optional<T::Config>");
- return opt_config ? rtc::Optional<AudioCodecInfo>(
+ "absl::optional<T::Config>");
+ return opt_config ? absl::optional<AudioCodecInfo>(
T::QueryAudioEncoder(*opt_config))
: Helper<Ts...>::QueryAudioEncoder(format);
}
static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
int payload_type,
const SdpAudioFormat& format,
- rtc::Optional<AudioCodecPairId> codec_pair_id) {
+ absl::optional<AudioCodecPairId> codec_pair_id) {
auto opt_config = T::SdpToConfig(format);
if (opt_config) {
return T::MakeAudioEncoder(*opt_config, payload_type, codec_pair_id);
@@ -83,7 +83,7 @@
return specs;
}
- rtc::Optional<AudioCodecInfo> QueryAudioEncoder(
+ absl::optional<AudioCodecInfo> QueryAudioEncoder(
const SdpAudioFormat& format) override {
return Helper<Ts...>::QueryAudioEncoder(format);
}
@@ -91,7 +91,7 @@
std::unique_ptr<AudioEncoder> MakeAudioEncoder(
int payload_type,
const SdpAudioFormat& format,
- rtc::Optional<AudioCodecPairId> codec_pair_id) override {
+ absl::optional<AudioCodecPairId> codec_pair_id) override {
return Helper<Ts...>::MakeAudioEncoder(payload_type, format, codec_pair_id);
}
};
@@ -106,7 +106,7 @@
// // Converts |audio_format| to a ConfigType instance. Returns an empty
// // optional if |audio_format| doesn't correctly specify an encoder of our
// // type.
-// rtc::Optional<ConfigType> SdpToConfig(const SdpAudioFormat& audio_format);
+// absl::optional<ConfigType> SdpToConfig(const SdpAudioFormat& audio_format);
//
// // Appends zero or more AudioCodecSpecs to the list that will be returned
// // by AudioEncoderFactory::GetSupportedEncoders().
@@ -121,7 +121,7 @@
// std::unique_ptr<AudioDecoder> MakeAudioEncoder(
// const ConfigType& config,
// int payload_type,
-// rtc::Optional<AudioCodecPairId> codec_pair_id);
+// absl::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/audio_format.h b/api/audio_codecs/audio_format.h
index 553ab8f..d132067d 100644
--- a/api/audio_codecs/audio_format.h
+++ b/api/audio_codecs/audio_format.h
@@ -15,7 +15,7 @@
#include <string>
#include <utility>
-#include "api/optional.h"
+#include "absl/types/optional.h"
#include "rtc_base/checks.h"
namespace webrtc {
diff --git a/api/audio_codecs/builtin_audio_decoder_factory.cc b/api/audio_codecs/builtin_audio_decoder_factory.cc
index c3e5d50..e3ca1b0 100644
--- a/api/audio_codecs/builtin_audio_decoder_factory.cc
+++ b/api/audio_codecs/builtin_audio_decoder_factory.cc
@@ -33,7 +33,8 @@
template <typename T>
struct NotAdvertised {
using Config = typename T::Config;
- static rtc::Optional<Config> SdpToConfig(const SdpAudioFormat& audio_format) {
+ static absl::optional<Config> SdpToConfig(
+ const SdpAudioFormat& audio_format) {
return T::SdpToConfig(audio_format);
}
static void AppendSupportedDecoders(std::vector<AudioCodecSpec>* specs) {
@@ -41,7 +42,7 @@
}
static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
const Config& config,
- rtc::Optional<AudioCodecPairId> codec_pair_id = rtc::nullopt) {
+ absl::optional<AudioCodecPairId> codec_pair_id = absl::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 5395404..c0caff4 100644
--- a/api/audio_codecs/builtin_audio_encoder_factory.cc
+++ b/api/audio_codecs/builtin_audio_encoder_factory.cc
@@ -33,7 +33,8 @@
template <typename T>
struct NotAdvertised {
using Config = typename T::Config;
- static rtc::Optional<Config> SdpToConfig(const SdpAudioFormat& audio_format) {
+ static absl::optional<Config> SdpToConfig(
+ const SdpAudioFormat& audio_format) {
return T::SdpToConfig(audio_format);
}
static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs) {
@@ -45,7 +46,7 @@
static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
const Config& config,
int payload_type,
- rtc::Optional<AudioCodecPairId> codec_pair_id = rtc::nullopt) {
+ absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt) {
return T::MakeAudioEncoder(config, payload_type, codec_pair_id);
}
};
diff --git a/api/audio_codecs/g711/BUILD.gn b/api/audio_codecs/g711/BUILD.gn
index 7026abb..52e1ee9 100644
--- a/api/audio_codecs/g711/BUILD.gn
+++ b/api/audio_codecs/g711/BUILD.gn
@@ -21,11 +21,11 @@
]
deps = [
"..:audio_codecs_api",
- "../..:optional",
"../../..:webrtc_common",
"../../../modules/audio_coding:g711",
"../../../rtc_base:rtc_base_approved",
"../../../rtc_base:safe_minmax",
+ "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -38,9 +38,9 @@
]
deps = [
"..:audio_codecs_api",
- "../..:optional",
"../../..:webrtc_common",
"../../../modules/audio_coding:g711",
"../../../rtc_base:rtc_base_approved",
+ "//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 c715e80..e8afa60 100644
--- a/api/audio_codecs/g711/audio_decoder_g711.cc
+++ b/api/audio_codecs/g711/audio_decoder_g711.cc
@@ -20,7 +20,7 @@
namespace webrtc {
-rtc::Optional<AudioDecoderG711::Config> AudioDecoderG711::SdpToConfig(
+absl::optional<AudioDecoderG711::Config> AudioDecoderG711::SdpToConfig(
const SdpAudioFormat& format) {
const bool is_pcmu = STR_CASE_CMP(format.name.c_str(), "PCMU") == 0;
const bool is_pcma = STR_CASE_CMP(format.name.c_str(), "PCMA") == 0;
@@ -32,7 +32,7 @@
RTC_DCHECK(config.IsOk());
return config;
} else {
- return rtc::nullopt;
+ return absl::nullopt;
}
}
@@ -45,7 +45,7 @@
std::unique_ptr<AudioDecoder> AudioDecoderG711::MakeAudioDecoder(
const Config& config,
- rtc::Optional<AudioCodecPairId> /*codec_pair_id*/) {
+ absl::optional<AudioCodecPairId> /*codec_pair_id*/) {
RTC_DCHECK(config.IsOk());
switch (config.type) {
case Config::Type::kPcmU:
diff --git a/api/audio_codecs/g711/audio_decoder_g711.h b/api/audio_codecs/g711/audio_decoder_g711.h
index 5085283..8275a8c 100644
--- a/api/audio_codecs/g711/audio_decoder_g711.h
+++ b/api/audio_codecs/g711/audio_decoder_g711.h
@@ -14,10 +14,10 @@
#include <memory>
#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"
-#include "api/optional.h"
namespace webrtc {
@@ -34,11 +34,11 @@
Type type;
int num_channels;
};
- static rtc::Optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
+ static absl::optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
static void AppendSupportedDecoders(std::vector<AudioCodecSpec>* specs);
static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
const Config& config,
- rtc::Optional<AudioCodecPairId> codec_pair_id = rtc::nullopt);
+ absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt);
};
} // namespace webrtc
diff --git a/api/audio_codecs/g711/audio_encoder_g711.cc b/api/audio_codecs/g711/audio_encoder_g711.cc
index e5abc33..95595fa 100644
--- a/api/audio_codecs/g711/audio_encoder_g711.cc
+++ b/api/audio_codecs/g711/audio_encoder_g711.cc
@@ -22,7 +22,7 @@
namespace webrtc {
-rtc::Optional<AudioEncoderG711::Config> AudioEncoderG711::SdpToConfig(
+absl::optional<AudioEncoderG711::Config> AudioEncoderG711::SdpToConfig(
const SdpAudioFormat& format) {
const bool is_pcmu = STR_CASE_CMP(format.name.c_str(), "PCMU") == 0;
const bool is_pcma = STR_CASE_CMP(format.name.c_str(), "PCMA") == 0;
@@ -42,7 +42,7 @@
RTC_DCHECK(config.IsOk());
return config;
} else {
- return rtc::nullopt;
+ return absl::nullopt;
}
}
@@ -62,7 +62,7 @@
std::unique_ptr<AudioEncoder> AudioEncoderG711::MakeAudioEncoder(
const Config& config,
int payload_type,
- rtc::Optional<AudioCodecPairId> /*codec_pair_id*/) {
+ absl::optional<AudioCodecPairId> /*codec_pair_id*/) {
RTC_DCHECK(config.IsOk());
switch (config.type) {
case Config::Type::kPcmU: {
diff --git a/api/audio_codecs/g711/audio_encoder_g711.h b/api/audio_codecs/g711/audio_encoder_g711.h
index 22a74b4..6b6eb5f 100644
--- a/api/audio_codecs/g711/audio_encoder_g711.h
+++ b/api/audio_codecs/g711/audio_encoder_g711.h
@@ -14,10 +14,10 @@
#include <memory>
#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"
-#include "api/optional.h"
namespace webrtc {
@@ -36,14 +36,14 @@
int num_channels = 1;
int frame_size_ms = 20;
};
- static rtc::Optional<AudioEncoderG711::Config> SdpToConfig(
+ static absl::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,
- rtc::Optional<AudioCodecPairId> codec_pair_id = rtc::nullopt);
+ absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt);
};
} // namespace webrtc
diff --git a/api/audio_codecs/g722/BUILD.gn b/api/audio_codecs/g722/BUILD.gn
index 7078aa4..85a8274 100644
--- a/api/audio_codecs/g722/BUILD.gn
+++ b/api/audio_codecs/g722/BUILD.gn
@@ -29,11 +29,11 @@
deps = [
":audio_encoder_g722_config",
"..:audio_codecs_api",
- "../..:optional",
"../../..:webrtc_common",
"../../../modules/audio_coding:g722",
"../../../rtc_base:rtc_base_approved",
"../../../rtc_base:safe_minmax",
+ "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -46,9 +46,9 @@
]
deps = [
"..:audio_codecs_api",
- "../..:optional",
"../../..:webrtc_common",
"../../../modules/audio_coding:g722",
"../../../rtc_base:rtc_base_approved",
+ "//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 6f72037..04a0a4c 100644
--- a/api/audio_codecs/g722/audio_decoder_g722.cc
+++ b/api/audio_codecs/g722/audio_decoder_g722.cc
@@ -20,14 +20,14 @@
namespace webrtc {
-rtc::Optional<AudioDecoderG722::Config> AudioDecoderG722::SdpToConfig(
+absl::optional<AudioDecoderG722::Config> AudioDecoderG722::SdpToConfig(
const SdpAudioFormat& format) {
return STR_CASE_CMP(format.name.c_str(), "G722") == 0 &&
format.clockrate_hz == 8000 &&
(format.num_channels == 1 || format.num_channels == 2)
- ? rtc::Optional<Config>(
+ ? absl::optional<Config>(
Config{rtc::dchecked_cast<int>(format.num_channels)})
- : rtc::nullopt;
+ : absl::nullopt;
}
void AudioDecoderG722::AppendSupportedDecoders(
@@ -37,7 +37,7 @@
std::unique_ptr<AudioDecoder> AudioDecoderG722::MakeAudioDecoder(
Config config,
- rtc::Optional<AudioCodecPairId> /*codec_pair_id*/) {
+ absl::optional<AudioCodecPairId> /*codec_pair_id*/) {
switch (config.num_channels) {
case 1:
return rtc::MakeUnique<AudioDecoderG722Impl>();
diff --git a/api/audio_codecs/g722/audio_decoder_g722.h b/api/audio_codecs/g722/audio_decoder_g722.h
index 34235dc..b7bb089 100644
--- a/api/audio_codecs/g722/audio_decoder_g722.h
+++ b/api/audio_codecs/g722/audio_decoder_g722.h
@@ -14,10 +14,10 @@
#include <memory>
#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"
-#include "api/optional.h"
namespace webrtc {
@@ -30,11 +30,11 @@
bool IsOk() const { return num_channels == 1 || num_channels == 2; }
int num_channels;
};
- static rtc::Optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
+ static absl::optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
static void AppendSupportedDecoders(std::vector<AudioCodecSpec>* specs);
static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
Config config,
- rtc::Optional<AudioCodecPairId> codec_pair_id = rtc::nullopt);
+ absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt);
};
} // namespace webrtc
diff --git a/api/audio_codecs/g722/audio_encoder_g722.cc b/api/audio_codecs/g722/audio_encoder_g722.cc
index 04074b1..d1f5258 100644
--- a/api/audio_codecs/g722/audio_encoder_g722.cc
+++ b/api/audio_codecs/g722/audio_encoder_g722.cc
@@ -22,11 +22,11 @@
namespace webrtc {
-rtc::Optional<AudioEncoderG722Config> AudioEncoderG722::SdpToConfig(
+absl::optional<AudioEncoderG722Config> AudioEncoderG722::SdpToConfig(
const SdpAudioFormat& format) {
if (STR_CASE_CMP(format.name.c_str(), "g722") != 0 ||
format.clockrate_hz != 8000) {
- return rtc::nullopt;
+ return absl::nullopt;
}
AudioEncoderG722Config config;
@@ -39,8 +39,8 @@
config.frame_size_ms = rtc::SafeClamp<int>(whole_packets * 10, 10, 60);
}
}
- return config.IsOk() ? rtc::Optional<AudioEncoderG722Config>(config)
- : rtc::nullopt;
+ return config.IsOk() ? absl::optional<AudioEncoderG722Config>(config)
+ : absl::nullopt;
}
void AudioEncoderG722::AppendSupportedEncoders(
@@ -60,7 +60,7 @@
std::unique_ptr<AudioEncoder> AudioEncoderG722::MakeAudioEncoder(
const AudioEncoderG722Config& config,
int payload_type,
- rtc::Optional<AudioCodecPairId> /*codec_pair_id*/) {
+ absl::optional<AudioCodecPairId> /*codec_pair_id*/) {
RTC_DCHECK(config.IsOk());
return rtc::MakeUnique<AudioEncoderG722Impl>(config, payload_type);
}
diff --git a/api/audio_codecs/g722/audio_encoder_g722.h b/api/audio_codecs/g722/audio_encoder_g722.h
index 08cd304..b97fe1b 100644
--- a/api/audio_codecs/g722/audio_encoder_g722.h
+++ b/api/audio_codecs/g722/audio_encoder_g722.h
@@ -14,11 +14,11 @@
#include <memory>
#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"
#include "api/audio_codecs/g722/audio_encoder_g722_config.h"
-#include "api/optional.h"
namespace webrtc {
@@ -28,14 +28,14 @@
// NOTE: This struct is still under development and may change without notice.
struct AudioEncoderG722 {
using Config = AudioEncoderG722Config;
- static rtc::Optional<AudioEncoderG722Config> SdpToConfig(
+ static absl::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,
- rtc::Optional<AudioCodecPairId> codec_pair_id = rtc::nullopt);
+ absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt);
};
} // namespace webrtc
diff --git a/api/audio_codecs/ilbc/BUILD.gn b/api/audio_codecs/ilbc/BUILD.gn
index 52dac5f..e1b2731 100644
--- a/api/audio_codecs/ilbc/BUILD.gn
+++ b/api/audio_codecs/ilbc/BUILD.gn
@@ -29,11 +29,11 @@
deps = [
":audio_encoder_ilbc_config",
"..:audio_codecs_api",
- "../..:optional",
"../../..:webrtc_common",
"../../../modules/audio_coding:ilbc",
"../../../rtc_base:rtc_base_approved",
"../../../rtc_base:safe_minmax",
+ "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -46,9 +46,9 @@
]
deps = [
"..:audio_codecs_api",
- "../..:optional",
"../../..:webrtc_common",
"../../../modules/audio_coding:ilbc",
"../../../rtc_base:rtc_base_approved",
+ "//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 88a2471..f1ecbdc 100644
--- a/api/audio_codecs/ilbc/audio_decoder_ilbc.cc
+++ b/api/audio_codecs/ilbc/audio_decoder_ilbc.cc
@@ -19,12 +19,12 @@
namespace webrtc {
-rtc::Optional<AudioDecoderIlbc::Config> AudioDecoderIlbc::SdpToConfig(
+absl::optional<AudioDecoderIlbc::Config> AudioDecoderIlbc::SdpToConfig(
const SdpAudioFormat& format) {
return STR_CASE_CMP(format.name.c_str(), "ILBC") == 0 &&
format.clockrate_hz == 8000 && format.num_channels == 1
- ? rtc::Optional<Config>(Config())
- : rtc::nullopt;
+ ? absl::optional<Config>(Config())
+ : absl::nullopt;
}
void AudioDecoderIlbc::AppendSupportedDecoders(
@@ -34,7 +34,7 @@
std::unique_ptr<AudioDecoder> AudioDecoderIlbc::MakeAudioDecoder(
Config config,
- rtc::Optional<AudioCodecPairId> /*codec_pair_id*/) {
+ absl::optional<AudioCodecPairId> /*codec_pair_id*/) {
return rtc::MakeUnique<AudioDecoderIlbcImpl>();
}
diff --git a/api/audio_codecs/ilbc/audio_decoder_ilbc.h b/api/audio_codecs/ilbc/audio_decoder_ilbc.h
index c233c4b..20f6ffd 100644
--- a/api/audio_codecs/ilbc/audio_decoder_ilbc.h
+++ b/api/audio_codecs/ilbc/audio_decoder_ilbc.h
@@ -14,10 +14,10 @@
#include <memory>
#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"
-#include "api/optional.h"
namespace webrtc {
@@ -27,11 +27,11 @@
// NOTE: This struct is still under development and may change without notice.
struct AudioDecoderIlbc {
struct Config {}; // Empty---no config values needed!
- static rtc::Optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
+ static absl::optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
static void AppendSupportedDecoders(std::vector<AudioCodecSpec>* specs);
static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
Config config,
- rtc::Optional<AudioCodecPairId> codec_pair_id = rtc::nullopt);
+ absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt);
};
} // namespace webrtc
diff --git a/api/audio_codecs/ilbc/audio_encoder_ilbc.cc b/api/audio_codecs/ilbc/audio_encoder_ilbc.cc
index 53dfdd3..59a16b5 100644
--- a/api/audio_codecs/ilbc/audio_encoder_ilbc.cc
+++ b/api/audio_codecs/ilbc/audio_encoder_ilbc.cc
@@ -38,11 +38,11 @@
}
} // namespace
-rtc::Optional<AudioEncoderIlbcConfig> AudioEncoderIlbc::SdpToConfig(
+absl::optional<AudioEncoderIlbcConfig> AudioEncoderIlbc::SdpToConfig(
const SdpAudioFormat& format) {
if (STR_CASE_CMP(format.name.c_str(), "ILBC") != 0 ||
format.clockrate_hz != 8000 || format.num_channels != 1) {
- return rtc::nullopt;
+ return absl::nullopt;
}
AudioEncoderIlbcConfig config;
@@ -54,8 +54,8 @@
config.frame_size_ms = rtc::SafeClamp<int>(whole_packets * 10, 20, 60);
}
}
- return config.IsOk() ? rtc::Optional<AudioEncoderIlbcConfig>(config)
- : rtc::nullopt;
+ return config.IsOk() ? absl::optional<AudioEncoderIlbcConfig>(config)
+ : absl::nullopt;
}
void AudioEncoderIlbc::AppendSupportedEncoders(
@@ -74,7 +74,7 @@
std::unique_ptr<AudioEncoder> AudioEncoderIlbc::MakeAudioEncoder(
const AudioEncoderIlbcConfig& config,
int payload_type,
- rtc::Optional<AudioCodecPairId> /*codec_pair_id*/) {
+ absl::optional<AudioCodecPairId> /*codec_pair_id*/) {
RTC_DCHECK(config.IsOk());
return rtc::MakeUnique<AudioEncoderIlbcImpl>(config, payload_type);
}
diff --git a/api/audio_codecs/ilbc/audio_encoder_ilbc.h b/api/audio_codecs/ilbc/audio_encoder_ilbc.h
index 85cdab0..0a86b16 100644
--- a/api/audio_codecs/ilbc/audio_encoder_ilbc.h
+++ b/api/audio_codecs/ilbc/audio_encoder_ilbc.h
@@ -14,11 +14,11 @@
#include <memory>
#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"
#include "api/audio_codecs/ilbc/audio_encoder_ilbc_config.h"
-#include "api/optional.h"
namespace webrtc {
@@ -28,14 +28,14 @@
// NOTE: This struct is still under development and may change without notice.
struct AudioEncoderIlbc {
using Config = AudioEncoderIlbcConfig;
- static rtc::Optional<AudioEncoderIlbcConfig> SdpToConfig(
+ static absl::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,
- rtc::Optional<AudioCodecPairId> codec_pair_id = rtc::nullopt);
+ absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt);
};
} // namespace webrtc
diff --git a/api/audio_codecs/isac/BUILD.gn b/api/audio_codecs/isac/BUILD.gn
index 08cabc5..e8e6d23 100644
--- a/api/audio_codecs/isac/BUILD.gn
+++ b/api/audio_codecs/isac/BUILD.gn
@@ -77,10 +77,10 @@
]
deps = [
"..:audio_codecs_api",
- "../..:optional",
"../../..:webrtc_common",
"../../../modules/audio_coding:isac_fix",
"../../../rtc_base:rtc_base_approved",
+ "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -93,10 +93,10 @@
]
deps = [
"..:audio_codecs_api",
- "../..:optional",
"../../..:webrtc_common",
"../../../modules/audio_coding:isac_fix",
"../../../rtc_base:rtc_base_approved",
+ "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -109,10 +109,10 @@
]
deps = [
"..:audio_codecs_api",
- "../..:optional",
"../../..:webrtc_common",
"../../../modules/audio_coding:isac",
"../../../rtc_base:rtc_base_approved",
+ "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -125,9 +125,9 @@
]
deps = [
"..:audio_codecs_api",
- "../..:optional",
"../../..:webrtc_common",
"../../../modules/audio_coding:isac",
"../../../rtc_base:rtc_base_approved",
+ "//third_party/abseil-cpp/absl/types:optional",
]
}
diff --git a/api/audio_codecs/isac/audio_decoder_isac_fix.cc b/api/audio_codecs/isac/audio_decoder_isac_fix.cc
index ab220f0..8435d05 100644
--- a/api/audio_codecs/isac/audio_decoder_isac_fix.cc
+++ b/api/audio_codecs/isac/audio_decoder_isac_fix.cc
@@ -16,12 +16,12 @@
namespace webrtc {
-rtc::Optional<AudioDecoderIsacFix::Config> AudioDecoderIsacFix::SdpToConfig(
+absl::optional<AudioDecoderIsacFix::Config> AudioDecoderIsacFix::SdpToConfig(
const SdpAudioFormat& format) {
return STR_CASE_CMP(format.name.c_str(), "ISAC") == 0 &&
format.clockrate_hz == 16000 && format.num_channels == 1
- ? rtc::Optional<Config>(Config())
- : rtc::nullopt;
+ ? absl::optional<Config>(Config())
+ : absl::nullopt;
}
void AudioDecoderIsacFix::AppendSupportedDecoders(
@@ -31,7 +31,7 @@
std::unique_ptr<AudioDecoder> AudioDecoderIsacFix::MakeAudioDecoder(
Config config,
- rtc::Optional<AudioCodecPairId> /*codec_pair_id*/) {
+ absl::optional<AudioCodecPairId> /*codec_pair_id*/) {
return rtc::MakeUnique<AudioDecoderIsacFixImpl>(16000);
}
diff --git a/api/audio_codecs/isac/audio_decoder_isac_fix.h b/api/audio_codecs/isac/audio_decoder_isac_fix.h
index 115486c..a4ce685 100644
--- a/api/audio_codecs/isac/audio_decoder_isac_fix.h
+++ b/api/audio_codecs/isac/audio_decoder_isac_fix.h
@@ -14,10 +14,10 @@
#include <memory>
#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"
-#include "api/optional.h"
namespace webrtc {
@@ -27,11 +27,11 @@
// NOTE: This struct is still under development and may change without notice.
struct AudioDecoderIsacFix {
struct Config {}; // Empty---no config values needed!
- static rtc::Optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
+ static absl::optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
static void AppendSupportedDecoders(std::vector<AudioCodecSpec>* specs);
static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
Config config,
- rtc::Optional<AudioCodecPairId> codec_pair_id = rtc::nullopt);
+ absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt);
};
} // namespace webrtc
diff --git a/api/audio_codecs/isac/audio_decoder_isac_float.cc b/api/audio_codecs/isac/audio_decoder_isac_float.cc
index e568f07..2e08e55 100644
--- a/api/audio_codecs/isac/audio_decoder_isac_float.cc
+++ b/api/audio_codecs/isac/audio_decoder_isac_float.cc
@@ -16,8 +16,8 @@
namespace webrtc {
-rtc::Optional<AudioDecoderIsacFloat::Config> AudioDecoderIsacFloat::SdpToConfig(
- const SdpAudioFormat& format) {
+absl::optional<AudioDecoderIsacFloat::Config>
+AudioDecoderIsacFloat::SdpToConfig(const SdpAudioFormat& format) {
if (STR_CASE_CMP(format.name.c_str(), "ISAC") == 0 &&
(format.clockrate_hz == 16000 || format.clockrate_hz == 32000) &&
format.num_channels == 1) {
@@ -25,7 +25,7 @@
config.sample_rate_hz = format.clockrate_hz;
return config;
} else {
- return rtc::nullopt;
+ return absl::nullopt;
}
}
@@ -37,7 +37,7 @@
std::unique_ptr<AudioDecoder> AudioDecoderIsacFloat::MakeAudioDecoder(
Config config,
- rtc::Optional<AudioCodecPairId> /*codec_pair_id*/) {
+ absl::optional<AudioCodecPairId> /*codec_pair_id*/) {
RTC_DCHECK(config.IsOk());
return rtc::MakeUnique<AudioDecoderIsacFloatImpl>(config.sample_rate_hz);
}
diff --git a/api/audio_codecs/isac/audio_decoder_isac_float.h b/api/audio_codecs/isac/audio_decoder_isac_float.h
index 47c2c60..cc13963 100644
--- a/api/audio_codecs/isac/audio_decoder_isac_float.h
+++ b/api/audio_codecs/isac/audio_decoder_isac_float.h
@@ -14,10 +14,10 @@
#include <memory>
#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"
-#include "api/optional.h"
namespace webrtc {
@@ -32,11 +32,11 @@
}
int sample_rate_hz = 16000;
};
- static rtc::Optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
+ static absl::optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
static void AppendSupportedDecoders(std::vector<AudioCodecSpec>* specs);
static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
Config config,
- rtc::Optional<AudioCodecPairId> codec_pair_id = rtc::nullopt);
+ absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt);
};
} // namespace webrtc
diff --git a/api/audio_codecs/isac/audio_encoder_isac_fix.cc b/api/audio_codecs/isac/audio_encoder_isac_fix.cc
index e2c7958..cb41214 100644
--- a/api/audio_codecs/isac/audio_encoder_isac_fix.cc
+++ b/api/audio_codecs/isac/audio_encoder_isac_fix.cc
@@ -17,7 +17,7 @@
namespace webrtc {
-rtc::Optional<AudioEncoderIsacFix::Config> AudioEncoderIsacFix::SdpToConfig(
+absl::optional<AudioEncoderIsacFix::Config> AudioEncoderIsacFix::SdpToConfig(
const SdpAudioFormat& format) {
if (STR_CASE_CMP(format.name.c_str(), "ISAC") == 0 &&
format.clockrate_hz == 16000 && format.num_channels == 1) {
@@ -31,7 +31,7 @@
}
return config;
} else {
- return rtc::nullopt;
+ return absl::nullopt;
}
}
@@ -51,7 +51,7 @@
std::unique_ptr<AudioEncoder> AudioEncoderIsacFix::MakeAudioEncoder(
AudioEncoderIsacFix::Config config,
int payload_type,
- rtc::Optional<AudioCodecPairId> /*codec_pair_id*/) {
+ absl::optional<AudioCodecPairId> /*codec_pair_id*/) {
RTC_DCHECK(config.IsOk());
AudioEncoderIsacFixImpl::Config c;
c.frame_size_ms = config.frame_size_ms;
diff --git a/api/audio_codecs/isac/audio_encoder_isac_fix.h b/api/audio_codecs/isac/audio_encoder_isac_fix.h
index 7f2743c..731e48d0 100644
--- a/api/audio_codecs/isac/audio_encoder_isac_fix.h
+++ b/api/audio_codecs/isac/audio_encoder_isac_fix.h
@@ -14,10 +14,10 @@
#include <memory>
#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"
-#include "api/optional.h"
namespace webrtc {
@@ -30,13 +30,13 @@
bool IsOk() const { return frame_size_ms == 30 || frame_size_ms == 60; }
int frame_size_ms = 30;
};
- static rtc::Optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
+ static absl::optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs);
static AudioCodecInfo QueryAudioEncoder(Config config);
static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
Config config,
int payload_type,
- rtc::Optional<AudioCodecPairId> codec_pair_id = rtc::nullopt);
+ absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt);
};
} // namespace webrtc
diff --git a/api/audio_codecs/isac/audio_encoder_isac_float.cc b/api/audio_codecs/isac/audio_encoder_isac_float.cc
index 5fbbe2b..510244c 100644
--- a/api/audio_codecs/isac/audio_encoder_isac_float.cc
+++ b/api/audio_codecs/isac/audio_encoder_isac_float.cc
@@ -17,8 +17,8 @@
namespace webrtc {
-rtc::Optional<AudioEncoderIsacFloat::Config> AudioEncoderIsacFloat::SdpToConfig(
- const SdpAudioFormat& format) {
+absl::optional<AudioEncoderIsacFloat::Config>
+AudioEncoderIsacFloat::SdpToConfig(const SdpAudioFormat& format) {
if (STR_CASE_CMP(format.name.c_str(), "ISAC") == 0 &&
(format.clockrate_hz == 16000 || format.clockrate_hz == 32000) &&
format.num_channels == 1) {
@@ -37,7 +37,7 @@
}
return config;
} else {
- return rtc::nullopt;
+ return absl::nullopt;
}
}
@@ -62,7 +62,7 @@
std::unique_ptr<AudioEncoder> AudioEncoderIsacFloat::MakeAudioEncoder(
const AudioEncoderIsacFloat::Config& config,
int payload_type,
- rtc::Optional<AudioCodecPairId> /*codec_pair_id*/) {
+ absl::optional<AudioCodecPairId> /*codec_pair_id*/) {
RTC_DCHECK(config.IsOk());
AudioEncoderIsacFloatImpl::Config c;
c.sample_rate_hz = config.sample_rate_hz;
diff --git a/api/audio_codecs/isac/audio_encoder_isac_float.h b/api/audio_codecs/isac/audio_encoder_isac_float.h
index b6043f2..6d98bf9 100644
--- a/api/audio_codecs/isac/audio_encoder_isac_float.h
+++ b/api/audio_codecs/isac/audio_encoder_isac_float.h
@@ -14,10 +14,10 @@
#include <memory>
#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"
-#include "api/optional.h"
namespace webrtc {
@@ -35,13 +35,13 @@
int sample_rate_hz = 16000;
int frame_size_ms = 30;
};
- static rtc::Optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
+ static absl::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,
- rtc::Optional<AudioCodecPairId> codec_pair_id = rtc::nullopt);
+ absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt);
};
} // namespace webrtc
diff --git a/api/audio_codecs/opus/BUILD.gn b/api/audio_codecs/opus/BUILD.gn
index d235d1a..953482e 100644
--- a/api/audio_codecs/opus/BUILD.gn
+++ b/api/audio_codecs/opus/BUILD.gn
@@ -19,8 +19,8 @@
"audio_encoder_opus_config.h",
]
deps = [
- "../..:optional",
"../../../rtc_base:rtc_base_approved",
+ "//third_party/abseil-cpp/absl/types:optional",
]
defines = []
if (rtc_opus_variable_complexity) {
@@ -42,9 +42,9 @@
deps = [
":audio_encoder_opus_config",
"..:audio_codecs_api",
- "../..:optional",
"../../../modules/audio_coding:webrtc_opus",
"../../../rtc_base:rtc_base_approved",
+ "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -57,9 +57,9 @@
]
deps = [
"..:audio_codecs_api",
- "../..:optional",
"../../..:webrtc_common",
"../../../modules/audio_coding:webrtc_opus",
"../../../rtc_base:rtc_base_approved",
+ "//third_party/abseil-cpp/absl/types:optional",
]
}
diff --git a/api/audio_codecs/opus/audio_decoder_opus.cc b/api/audio_codecs/opus/audio_decoder_opus.cc
index 73a0a3f..81bee77 100644
--- a/api/audio_codecs/opus/audio_decoder_opus.cc
+++ b/api/audio_codecs/opus/audio_decoder_opus.cc
@@ -20,9 +20,9 @@
namespace webrtc {
-rtc::Optional<AudioDecoderOpus::Config> AudioDecoderOpus::SdpToConfig(
+absl::optional<AudioDecoderOpus::Config> AudioDecoderOpus::SdpToConfig(
const SdpAudioFormat& format) {
- const auto num_channels = [&]() -> rtc::Optional<int> {
+ const auto num_channels = [&]() -> absl::optional<int> {
auto stereo = format.parameters.find("stereo");
if (stereo != format.parameters.end()) {
if (stereo->second == "0") {
@@ -30,7 +30,7 @@
} else if (stereo->second == "1") {
return 2;
} else {
- return rtc::nullopt; // Bad stereo parameter.
+ return absl::nullopt; // Bad stereo parameter.
}
}
return 1; // Default to mono.
@@ -40,7 +40,7 @@
num_channels) {
return Config{*num_channels};
} else {
- return rtc::nullopt;
+ return absl::nullopt;
}
}
@@ -56,7 +56,7 @@
std::unique_ptr<AudioDecoder> AudioDecoderOpus::MakeAudioDecoder(
Config config,
- rtc::Optional<AudioCodecPairId> /*codec_pair_id*/) {
+ absl::optional<AudioCodecPairId> /*codec_pair_id*/) {
return rtc::MakeUnique<AudioDecoderOpusImpl>(config.num_channels);
}
diff --git a/api/audio_codecs/opus/audio_decoder_opus.h b/api/audio_codecs/opus/audio_decoder_opus.h
index f76d244..de26026 100644
--- a/api/audio_codecs/opus/audio_decoder_opus.h
+++ b/api/audio_codecs/opus/audio_decoder_opus.h
@@ -14,10 +14,10 @@
#include <memory>
#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"
-#include "api/optional.h"
namespace webrtc {
@@ -29,11 +29,11 @@
struct Config {
int num_channels;
};
- static rtc::Optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
+ static absl::optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
static void AppendSupportedDecoders(std::vector<AudioCodecSpec>* specs);
static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
Config config,
- rtc::Optional<AudioCodecPairId> codec_pair_id = rtc::nullopt);
+ absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt);
};
} // namespace webrtc
diff --git a/api/audio_codecs/opus/audio_encoder_opus.cc b/api/audio_codecs/opus/audio_encoder_opus.cc
index 8ba66fb..36d82b3 100644
--- a/api/audio_codecs/opus/audio_encoder_opus.cc
+++ b/api/audio_codecs/opus/audio_encoder_opus.cc
@@ -14,7 +14,7 @@
namespace webrtc {
-rtc::Optional<AudioEncoderOpusConfig> AudioEncoderOpus::SdpToConfig(
+absl::optional<AudioEncoderOpusConfig> AudioEncoderOpus::SdpToConfig(
const SdpAudioFormat& format) {
return AudioEncoderOpusImpl::SdpToConfig(format);
}
@@ -32,7 +32,7 @@
std::unique_ptr<AudioEncoder> AudioEncoderOpus::MakeAudioEncoder(
const AudioEncoderOpusConfig& config,
int payload_type,
- rtc::Optional<AudioCodecPairId> /*codec_pair_id*/) {
+ absl::optional<AudioCodecPairId> /*codec_pair_id*/) {
return AudioEncoderOpusImpl::MakeAudioEncoder(config, payload_type);
}
diff --git a/api/audio_codecs/opus/audio_encoder_opus.h b/api/audio_codecs/opus/audio_encoder_opus.h
index 6325269..20aaaf7 100644
--- a/api/audio_codecs/opus/audio_encoder_opus.h
+++ b/api/audio_codecs/opus/audio_encoder_opus.h
@@ -14,11 +14,11 @@
#include <memory>
#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"
#include "api/audio_codecs/opus/audio_encoder_opus_config.h"
-#include "api/optional.h"
namespace webrtc {
@@ -28,14 +28,14 @@
// NOTE: This struct is still under development and may change without notice.
struct AudioEncoderOpus {
using Config = AudioEncoderOpusConfig;
- static rtc::Optional<AudioEncoderOpusConfig> SdpToConfig(
+ static absl::optional<AudioEncoderOpusConfig> SdpToConfig(
const SdpAudioFormat& audio_format);
static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs);
static AudioCodecInfo QueryAudioEncoder(const AudioEncoderOpusConfig& config);
static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
const AudioEncoderOpusConfig& config,
int payload_type,
- rtc::Optional<AudioCodecPairId> codec_pair_id = rtc::nullopt);
+ absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt);
};
} // namespace webrtc
diff --git a/api/audio_codecs/opus/audio_encoder_opus_config.h b/api/audio_codecs/opus/audio_encoder_opus_config.h
index d586592..c7067bb 100644
--- a/api/audio_codecs/opus/audio_encoder_opus_config.h
+++ b/api/audio_codecs/opus/audio_encoder_opus_config.h
@@ -15,7 +15,7 @@
#include <vector>
-#include "api/optional.h"
+#include "absl/types/optional.h"
namespace webrtc {
@@ -42,7 +42,7 @@
// NOTE: This member must always be set.
// TODO(kwiberg): Turn it into just an int.
- rtc::Optional<int> bitrate_bps;
+ absl::optional<int> bitrate_bps;
bool fec_enabled;
bool cbr_enabled;
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 5fa4344..e4f09d4 100644
--- a/api/audio_codecs/test/audio_decoder_factory_template_unittest.cc
+++ b/api/audio_codecs/test/audio_decoder_factory_template_unittest.cc
@@ -43,12 +43,13 @@
SdpAudioFormat audio_format;
};
- static rtc::Optional<Config> SdpToConfig(const SdpAudioFormat& audio_format) {
+ static absl::optional<Config> SdpToConfig(
+ const SdpAudioFormat& audio_format) {
if (Params::AudioFormat() == audio_format) {
Config config = {audio_format};
return config;
} else {
- return rtc::nullopt;
+ return absl::nullopt;
}
}
@@ -62,7 +63,7 @@
static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
const Config&,
- rtc::Optional<AudioCodecPairId> /*codec_pair_id*/ = rtc::nullopt) {
+ absl::optional<AudioCodecPairId> /*codec_pair_id*/ = absl::nullopt) {
auto dec = rtc::MakeUnique<testing::StrictMock<MockAudioDecoder>>();
EXPECT_CALL(*dec, SampleRateHz())
.WillOnce(testing::Return(Params::CodecInfo().sample_rate_hz));
@@ -80,7 +81,7 @@
EXPECT_THAT(factory->GetSupportedDecoders(), testing::IsEmpty());
EXPECT_FALSE(factory->IsSupportedDecoder({"foo", 8000, 1}));
EXPECT_EQ(nullptr,
- factory->MakeAudioDecoder({"bar", 16000, 1}, rtc::nullopt));
+ factory->MakeAudioDecoder({"bar", 16000, 1}, absl::nullopt));
}
TEST(AudioDecoderFactoryTemplateTest, OneDecoderType) {
@@ -91,8 +92,8 @@
EXPECT_FALSE(factory->IsSupportedDecoder({"foo", 8000, 1}));
EXPECT_TRUE(factory->IsSupportedDecoder({"bogus", 8000, 1}));
EXPECT_EQ(nullptr,
- factory->MakeAudioDecoder({"bar", 16000, 1}, rtc::nullopt));
- auto dec = factory->MakeAudioDecoder({"bogus", 8000, 1}, rtc::nullopt);
+ factory->MakeAudioDecoder({"bar", 16000, 1}, absl::nullopt));
+ auto dec = factory->MakeAudioDecoder({"bogus", 8000, 1}, absl::nullopt);
ASSERT_NE(nullptr, dec);
EXPECT_EQ(8000, dec->SampleRateHz());
}
@@ -110,14 +111,14 @@
EXPECT_TRUE(
factory->IsSupportedDecoder({"sham", 16000, 2, {{"param", "value"}}}));
EXPECT_EQ(nullptr,
- factory->MakeAudioDecoder({"bar", 16000, 1}, rtc::nullopt));
- auto dec1 = factory->MakeAudioDecoder({"bogus", 8000, 1}, rtc::nullopt);
+ factory->MakeAudioDecoder({"bar", 16000, 1}, absl::nullopt));
+ auto dec1 = factory->MakeAudioDecoder({"bogus", 8000, 1}, absl::nullopt);
ASSERT_NE(nullptr, dec1);
EXPECT_EQ(8000, dec1->SampleRateHz());
EXPECT_EQ(nullptr,
- factory->MakeAudioDecoder({"sham", 16000, 2}, rtc::nullopt));
+ factory->MakeAudioDecoder({"sham", 16000, 2}, absl::nullopt));
auto dec2 = factory->MakeAudioDecoder(
- {"sham", 16000, 2, {{"param", "value"}}}, rtc::nullopt);
+ {"sham", 16000, 2, {{"param", "value"}}}, absl::nullopt);
ASSERT_NE(nullptr, dec2);
EXPECT_EQ(16000, dec2->SampleRateHz());
}
@@ -132,11 +133,11 @@
EXPECT_TRUE(factory->IsSupportedDecoder({"PCMU", 8000, 1}));
EXPECT_TRUE(factory->IsSupportedDecoder({"pcma", 8000, 1}));
EXPECT_EQ(nullptr,
- factory->MakeAudioDecoder({"pcmu", 16000, 1}, rtc::nullopt));
- auto dec1 = factory->MakeAudioDecoder({"pcmu", 8000, 1}, rtc::nullopt);
+ factory->MakeAudioDecoder({"pcmu", 16000, 1}, absl::nullopt));
+ auto dec1 = factory->MakeAudioDecoder({"pcmu", 8000, 1}, absl::nullopt);
ASSERT_NE(nullptr, dec1);
EXPECT_EQ(8000, dec1->SampleRateHz());
- auto dec2 = factory->MakeAudioDecoder({"PCMA", 8000, 1}, rtc::nullopt);
+ auto dec2 = factory->MakeAudioDecoder({"PCMA", 8000, 1}, absl::nullopt);
ASSERT_NE(nullptr, dec2);
EXPECT_EQ(8000, dec2->SampleRateHz());
}
@@ -149,16 +150,16 @@
EXPECT_FALSE(factory->IsSupportedDecoder({"foo", 8000, 1}));
EXPECT_TRUE(factory->IsSupportedDecoder({"G722", 8000, 1}));
EXPECT_EQ(nullptr,
- factory->MakeAudioDecoder({"bar", 16000, 1}, rtc::nullopt));
- auto dec1 = factory->MakeAudioDecoder({"G722", 8000, 1}, rtc::nullopt);
+ factory->MakeAudioDecoder({"bar", 16000, 1}, absl::nullopt));
+ auto dec1 = factory->MakeAudioDecoder({"G722", 8000, 1}, absl::nullopt);
ASSERT_NE(nullptr, dec1);
EXPECT_EQ(16000, dec1->SampleRateHz());
EXPECT_EQ(1u, dec1->Channels());
- auto dec2 = factory->MakeAudioDecoder({"G722", 8000, 2}, rtc::nullopt);
+ auto dec2 = factory->MakeAudioDecoder({"G722", 8000, 2}, absl::nullopt);
ASSERT_NE(nullptr, dec2);
EXPECT_EQ(16000, dec2->SampleRateHz());
EXPECT_EQ(2u, dec2->Channels());
- auto dec3 = factory->MakeAudioDecoder({"G722", 8000, 3}, rtc::nullopt);
+ auto dec3 = factory->MakeAudioDecoder({"G722", 8000, 3}, absl::nullopt);
ASSERT_EQ(nullptr, dec3);
}
@@ -169,8 +170,9 @@
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->MakeAudioDecoder({"bar", 8000, 1}, rtc::nullopt));
- auto dec = factory->MakeAudioDecoder({"ilbc", 8000, 1}, rtc::nullopt);
+ EXPECT_EQ(nullptr,
+ factory->MakeAudioDecoder({"bar", 8000, 1}, absl::nullopt));
+ auto dec = factory->MakeAudioDecoder({"ilbc", 8000, 1}, absl::nullopt);
ASSERT_NE(nullptr, dec);
EXPECT_EQ(8000, dec->SampleRateHz());
}
@@ -184,8 +186,8 @@
EXPECT_TRUE(factory->IsSupportedDecoder({"isac", 16000, 1}));
EXPECT_FALSE(factory->IsSupportedDecoder({"isac", 32000, 1}));
EXPECT_EQ(nullptr,
- factory->MakeAudioDecoder({"isac", 8000, 1}, rtc::nullopt));
- auto dec = factory->MakeAudioDecoder({"isac", 16000, 1}, rtc::nullopt);
+ factory->MakeAudioDecoder({"isac", 8000, 1}, absl::nullopt));
+ auto dec = factory->MakeAudioDecoder({"isac", 16000, 1}, absl::nullopt);
ASSERT_NE(nullptr, dec);
EXPECT_EQ(16000, dec->SampleRateHz());
}
@@ -201,11 +203,11 @@
EXPECT_TRUE(factory->IsSupportedDecoder({"isac", 16000, 1}));
EXPECT_TRUE(factory->IsSupportedDecoder({"isac", 32000, 1}));
EXPECT_EQ(nullptr,
- factory->MakeAudioDecoder({"isac", 8000, 1}, rtc::nullopt));
- auto dec1 = factory->MakeAudioDecoder({"isac", 16000, 1}, rtc::nullopt);
+ factory->MakeAudioDecoder({"isac", 8000, 1}, absl::nullopt));
+ auto dec1 = factory->MakeAudioDecoder({"isac", 16000, 1}, absl::nullopt);
ASSERT_NE(nullptr, dec1);
EXPECT_EQ(16000, dec1->SampleRateHz());
- auto dec2 = factory->MakeAudioDecoder({"isac", 32000, 1}, rtc::nullopt);
+ auto dec2 = factory->MakeAudioDecoder({"isac", 32000, 1}, absl::nullopt);
ASSERT_NE(nullptr, dec2);
EXPECT_EQ(32000, dec2->SampleRateHz());
}
@@ -224,8 +226,9 @@
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->MakeAudioDecoder({"L16", 8000, 0}, rtc::nullopt));
- auto dec = factory->MakeAudioDecoder({"L16", 48000, 2}, rtc::nullopt);
+ EXPECT_EQ(nullptr,
+ factory->MakeAudioDecoder({"L16", 8000, 0}, absl::nullopt));
+ auto dec = factory->MakeAudioDecoder({"L16", 48000, 2}, absl::nullopt);
ASSERT_NE(nullptr, dec);
EXPECT_EQ(48000, dec->SampleRateHz());
}
@@ -242,8 +245,8 @@
EXPECT_FALSE(factory->IsSupportedDecoder({"opus", 48000, 1}));
EXPECT_TRUE(factory->IsSupportedDecoder({"opus", 48000, 2}));
EXPECT_EQ(nullptr,
- factory->MakeAudioDecoder({"bar", 16000, 1}, rtc::nullopt));
- auto dec = factory->MakeAudioDecoder({"opus", 48000, 2}, rtc::nullopt);
+ factory->MakeAudioDecoder({"bar", 16000, 1}, absl::nullopt));
+ auto dec = factory->MakeAudioDecoder({"opus", 48000, 2}, absl::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 3da588d..46781ce 100644
--- a/api/audio_codecs/test/audio_encoder_factory_template_unittest.cc
+++ b/api/audio_codecs/test/audio_encoder_factory_template_unittest.cc
@@ -43,12 +43,13 @@
SdpAudioFormat audio_format;
};
- static rtc::Optional<Config> SdpToConfig(const SdpAudioFormat& audio_format) {
+ static absl::optional<Config> SdpToConfig(
+ const SdpAudioFormat& audio_format) {
if (Params::AudioFormat() == audio_format) {
Config config = {audio_format};
return config;
} else {
- return rtc::nullopt;
+ return absl::nullopt;
}
}
@@ -63,7 +64,7 @@
static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
const Config&,
int payload_type,
- rtc::Optional<AudioCodecPairId> /*codec_pair_id*/ = rtc::nullopt) {
+ absl::optional<AudioCodecPairId> /*codec_pair_id*/ = absl::nullopt) {
auto enc = rtc::MakeUnique<testing::StrictMock<MockAudioEncoder>>();
EXPECT_CALL(*enc, SampleRateHz())
.WillOnce(testing::Return(Params::CodecInfo().sample_rate_hz));
@@ -78,9 +79,9 @@
new rtc::RefCountedObject<
audio_encoder_factory_template_impl::AudioEncoderFactoryT<>>());
EXPECT_THAT(factory->GetSupportedEncoders(), testing::IsEmpty());
- EXPECT_EQ(rtc::nullopt, factory->QueryAudioEncoder({"foo", 8000, 1}));
+ EXPECT_EQ(absl::nullopt, factory->QueryAudioEncoder({"foo", 8000, 1}));
EXPECT_EQ(nullptr,
- factory->MakeAudioEncoder(17, {"bar", 16000, 1}, rtc::nullopt));
+ factory->MakeAudioEncoder(17, {"bar", 16000, 1}, absl::nullopt));
}
TEST(AudioEncoderFactoryTemplateTest, OneEncoderType) {
@@ -88,12 +89,12 @@
EXPECT_THAT(factory->GetSupportedEncoders(),
testing::ElementsAre(
AudioCodecSpec{{"bogus", 8000, 1}, {8000, 1, 12345}}));
- EXPECT_EQ(rtc::nullopt, factory->QueryAudioEncoder({"foo", 8000, 1}));
+ EXPECT_EQ(absl::nullopt, factory->QueryAudioEncoder({"foo", 8000, 1}));
EXPECT_EQ(AudioCodecInfo(8000, 1, 12345),
factory->QueryAudioEncoder({"bogus", 8000, 1}));
EXPECT_EQ(nullptr,
- factory->MakeAudioEncoder(17, {"bar", 16000, 1}, rtc::nullopt));
- auto enc = factory->MakeAudioEncoder(17, {"bogus", 8000, 1}, rtc::nullopt);
+ factory->MakeAudioEncoder(17, {"bar", 16000, 1}, absl::nullopt));
+ auto enc = factory->MakeAudioEncoder(17, {"bogus", 8000, 1}, absl::nullopt);
ASSERT_NE(nullptr, enc);
EXPECT_EQ(8000, enc->SampleRateHz());
}
@@ -106,21 +107,21 @@
AudioCodecSpec{{"bogus", 8000, 1}, {8000, 1, 12345}},
AudioCodecSpec{{"sham", 16000, 2, {{"param", "value"}}},
{16000, 2, 23456}}));
- EXPECT_EQ(rtc::nullopt, factory->QueryAudioEncoder({"foo", 8000, 1}));
+ EXPECT_EQ(absl::nullopt, factory->QueryAudioEncoder({"foo", 8000, 1}));
EXPECT_EQ(AudioCodecInfo(8000, 1, 12345),
factory->QueryAudioEncoder({"bogus", 8000, 1}));
EXPECT_EQ(
AudioCodecInfo(16000, 2, 23456),
factory->QueryAudioEncoder({"sham", 16000, 2, {{"param", "value"}}}));
EXPECT_EQ(nullptr,
- factory->MakeAudioEncoder(17, {"bar", 16000, 1}, rtc::nullopt));
- auto enc1 = factory->MakeAudioEncoder(17, {"bogus", 8000, 1}, rtc::nullopt);
+ factory->MakeAudioEncoder(17, {"bar", 16000, 1}, absl::nullopt));
+ auto enc1 = factory->MakeAudioEncoder(17, {"bogus", 8000, 1}, absl::nullopt);
ASSERT_NE(nullptr, enc1);
EXPECT_EQ(8000, enc1->SampleRateHz());
EXPECT_EQ(nullptr,
- factory->MakeAudioEncoder(17, {"sham", 16000, 2}, rtc::nullopt));
+ factory->MakeAudioEncoder(17, {"sham", 16000, 2}, absl::nullopt));
auto enc2 = factory->MakeAudioEncoder(
- 17, {"sham", 16000, 2, {{"param", "value"}}}, rtc::nullopt);
+ 17, {"sham", 16000, 2, {{"param", "value"}}}, absl::nullopt);
ASSERT_NE(nullptr, enc2);
EXPECT_EQ(16000, enc2->SampleRateHz());
}
@@ -131,15 +132,15 @@
testing::ElementsAre(
AudioCodecSpec{{"PCMU", 8000, 1}, {8000, 1, 64000}},
AudioCodecSpec{{"PCMA", 8000, 1}, {8000, 1, 64000}}));
- EXPECT_EQ(rtc::nullopt, factory->QueryAudioEncoder({"PCMA", 16000, 1}));
+ EXPECT_EQ(absl::nullopt, factory->QueryAudioEncoder({"PCMA", 16000, 1}));
EXPECT_EQ(AudioCodecInfo(8000, 1, 64000),
factory->QueryAudioEncoder({"PCMA", 8000, 1}));
EXPECT_EQ(nullptr,
- factory->MakeAudioEncoder(17, {"PCMU", 16000, 1}, rtc::nullopt));
- auto enc1 = factory->MakeAudioEncoder(17, {"PCMU", 8000, 1}, rtc::nullopt);
+ factory->MakeAudioEncoder(17, {"PCMU", 16000, 1}, absl::nullopt));
+ auto enc1 = factory->MakeAudioEncoder(17, {"PCMU", 8000, 1}, absl::nullopt);
ASSERT_NE(nullptr, enc1);
EXPECT_EQ(8000, enc1->SampleRateHz());
- auto enc2 = factory->MakeAudioEncoder(17, {"PCMA", 8000, 1}, rtc::nullopt);
+ auto enc2 = factory->MakeAudioEncoder(17, {"PCMA", 8000, 1}, absl::nullopt);
ASSERT_NE(nullptr, enc2);
EXPECT_EQ(8000, enc2->SampleRateHz());
}
@@ -149,12 +150,12 @@
EXPECT_THAT(factory->GetSupportedEncoders(),
testing::ElementsAre(
AudioCodecSpec{{"G722", 8000, 1}, {16000, 1, 64000}}));
- EXPECT_EQ(rtc::nullopt, factory->QueryAudioEncoder({"foo", 8000, 1}));
+ EXPECT_EQ(absl::nullopt, factory->QueryAudioEncoder({"foo", 8000, 1}));
EXPECT_EQ(AudioCodecInfo(16000, 1, 64000),
factory->QueryAudioEncoder({"G722", 8000, 1}));
EXPECT_EQ(nullptr,
- factory->MakeAudioEncoder(17, {"bar", 16000, 1}, rtc::nullopt));
- auto enc = factory->MakeAudioEncoder(17, {"G722", 8000, 1}, rtc::nullopt);
+ factory->MakeAudioEncoder(17, {"bar", 16000, 1}, absl::nullopt));
+ auto enc = factory->MakeAudioEncoder(17, {"G722", 8000, 1}, absl::nullopt);
ASSERT_NE(nullptr, enc);
EXPECT_EQ(16000, enc->SampleRateHz());
}
@@ -164,12 +165,12 @@
EXPECT_THAT(factory->GetSupportedEncoders(),
testing::ElementsAre(
AudioCodecSpec{{"ILBC", 8000, 1}, {8000, 1, 13333}}));
- EXPECT_EQ(rtc::nullopt, factory->QueryAudioEncoder({"foo", 8000, 1}));
+ EXPECT_EQ(absl::nullopt, factory->QueryAudioEncoder({"foo", 8000, 1}));
EXPECT_EQ(AudioCodecInfo(8000, 1, 13333),
factory->QueryAudioEncoder({"ilbc", 8000, 1}));
EXPECT_EQ(nullptr,
- factory->MakeAudioEncoder(17, {"bar", 8000, 1}, rtc::nullopt));
- auto enc = factory->MakeAudioEncoder(17, {"ilbc", 8000, 1}, rtc::nullopt);
+ factory->MakeAudioEncoder(17, {"bar", 8000, 1}, absl::nullopt));
+ auto enc = factory->MakeAudioEncoder(17, {"ilbc", 8000, 1}, absl::nullopt);
ASSERT_NE(nullptr, enc);
EXPECT_EQ(8000, enc->SampleRateHz());
}
@@ -179,18 +180,18 @@
EXPECT_THAT(factory->GetSupportedEncoders(),
testing::ElementsAre(AudioCodecSpec{
{"ISAC", 16000, 1}, {16000, 1, 32000, 10000, 32000}}));
- EXPECT_EQ(rtc::nullopt, factory->QueryAudioEncoder({"isac", 16000, 2}));
+ EXPECT_EQ(absl::nullopt, factory->QueryAudioEncoder({"isac", 16000, 2}));
EXPECT_EQ(AudioCodecInfo(16000, 1, 32000, 10000, 32000),
factory->QueryAudioEncoder({"isac", 16000, 1}));
- EXPECT_EQ(rtc::nullopt, factory->QueryAudioEncoder({"isac", 32000, 1}));
+ EXPECT_EQ(absl::nullopt, factory->QueryAudioEncoder({"isac", 32000, 1}));
EXPECT_EQ(nullptr,
- factory->MakeAudioEncoder(17, {"isac", 8000, 1}, rtc::nullopt));
- auto enc1 = factory->MakeAudioEncoder(17, {"isac", 16000, 1}, rtc::nullopt);
+ factory->MakeAudioEncoder(17, {"isac", 8000, 1}, absl::nullopt));
+ auto enc1 = factory->MakeAudioEncoder(17, {"isac", 16000, 1}, absl::nullopt);
ASSERT_NE(nullptr, enc1);
EXPECT_EQ(16000, enc1->SampleRateHz());
EXPECT_EQ(3u, enc1->Num10MsFramesInNextPacket());
auto enc2 = factory->MakeAudioEncoder(
- 17, {"isac", 16000, 1, {{"ptime", "60"}}}, rtc::nullopt);
+ 17, {"isac", 16000, 1, {{"ptime", "60"}}}, absl::nullopt);
ASSERT_NE(nullptr, enc2);
EXPECT_EQ(6u, enc2->Num10MsFramesInNextPacket());
}
@@ -202,17 +203,17 @@
testing::ElementsAre(
AudioCodecSpec{{"ISAC", 16000, 1}, {16000, 1, 32000, 10000, 32000}},
AudioCodecSpec{{"ISAC", 32000, 1}, {32000, 1, 56000, 10000, 56000}}));
- EXPECT_EQ(rtc::nullopt, factory->QueryAudioEncoder({"isac", 16000, 2}));
+ EXPECT_EQ(absl::nullopt, factory->QueryAudioEncoder({"isac", 16000, 2}));
EXPECT_EQ(AudioCodecInfo(16000, 1, 32000, 10000, 32000),
factory->QueryAudioEncoder({"isac", 16000, 1}));
EXPECT_EQ(AudioCodecInfo(32000, 1, 56000, 10000, 56000),
factory->QueryAudioEncoder({"isac", 32000, 1}));
EXPECT_EQ(nullptr,
- factory->MakeAudioEncoder(17, {"isac", 8000, 1}, rtc::nullopt));
- auto enc1 = factory->MakeAudioEncoder(17, {"isac", 16000, 1}, rtc::nullopt);
+ factory->MakeAudioEncoder(17, {"isac", 8000, 1}, absl::nullopt));
+ auto enc1 = factory->MakeAudioEncoder(17, {"isac", 16000, 1}, absl::nullopt);
ASSERT_NE(nullptr, enc1);
EXPECT_EQ(16000, enc1->SampleRateHz());
- auto enc2 = factory->MakeAudioEncoder(17, {"isac", 32000, 1}, rtc::nullopt);
+ auto enc2 = factory->MakeAudioEncoder(17, {"isac", 32000, 1}, absl::nullopt);
ASSERT_NE(nullptr, enc2);
EXPECT_EQ(32000, enc2->SampleRateHz());
}
@@ -228,12 +229,12 @@
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(rtc::nullopt, factory->QueryAudioEncoder({"L16", 8000, 0}));
+ EXPECT_EQ(absl::nullopt, factory->QueryAudioEncoder({"L16", 8000, 0}));
EXPECT_EQ(AudioCodecInfo(48000, 1, 48000 * 16),
factory->QueryAudioEncoder({"L16", 48000, 1}));
EXPECT_EQ(nullptr,
- factory->MakeAudioEncoder(17, {"L16", 8000, 0}, rtc::nullopt));
- auto enc = factory->MakeAudioEncoder(17, {"L16", 48000, 2}, rtc::nullopt);
+ factory->MakeAudioEncoder(17, {"L16", 8000, 0}, absl::nullopt));
+ auto enc = factory->MakeAudioEncoder(17, {"L16", 48000, 2}, absl::nullopt);
ASSERT_NE(nullptr, enc);
EXPECT_EQ(48000, enc->SampleRateHz());
}
@@ -248,14 +249,14 @@
testing::ElementsAre(AudioCodecSpec{
{"opus", 48000, 2, {{"minptime", "10"}, {"useinbandfec", "1"}}},
info}));
- EXPECT_EQ(rtc::nullopt, factory->QueryAudioEncoder({"foo", 8000, 1}));
+ EXPECT_EQ(absl::nullopt, factory->QueryAudioEncoder({"foo", 8000, 1}));
EXPECT_EQ(
info,
factory->QueryAudioEncoder(
{"opus", 48000, 2, {{"minptime", "10"}, {"useinbandfec", "1"}}}));
EXPECT_EQ(nullptr,
- factory->MakeAudioEncoder(17, {"bar", 16000, 1}, rtc::nullopt));
- auto enc = factory->MakeAudioEncoder(17, {"opus", 48000, 2}, rtc::nullopt);
+ factory->MakeAudioEncoder(17, {"bar", 16000, 1}, absl::nullopt));
+ auto enc = factory->MakeAudioEncoder(17, {"opus", 48000, 2}, absl::nullopt);
ASSERT_NE(nullptr, enc);
EXPECT_EQ(48000, enc->SampleRateHz());
}
diff --git a/api/audio_options.h b/api/audio_options.h
index 28f03b6..df66d36 100644
--- a/api/audio_options.h
+++ b/api/audio_options.h
@@ -13,7 +13,7 @@
#include <string>
-#include "api/optional.h"
+#include "absl/types/optional.h"
#include "rtc_base/stringencode.h"
namespace cricket {
@@ -126,53 +126,53 @@
// Audio processing that attempts to filter away the output signal from
// later inbound pickup.
- rtc::Optional<bool> echo_cancellation;
+ absl::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.
- rtc::Optional<bool> ios_force_software_aec_HACK;
+ absl::optional<bool> ios_force_software_aec_HACK;
#endif
// Audio processing to adjust the sensitivity of the local mic dynamically.
- rtc::Optional<bool> auto_gain_control;
+ absl::optional<bool> auto_gain_control;
// Audio processing to filter out background noise.
- rtc::Optional<bool> noise_suppression;
+ absl::optional<bool> noise_suppression;
// Audio processing to remove background noise of lower frequencies.
- rtc::Optional<bool> highpass_filter;
+ absl::optional<bool> highpass_filter;
// Audio processing to swap the left and right channels.
- rtc::Optional<bool> stereo_swapping;
+ absl::optional<bool> stereo_swapping;
// Audio receiver jitter buffer (NetEq) max capacity in number of packets.
- rtc::Optional<int> audio_jitter_buffer_max_packets;
+ absl::optional<int> audio_jitter_buffer_max_packets;
// Audio receiver jitter buffer (NetEq) fast accelerate mode.
- rtc::Optional<bool> audio_jitter_buffer_fast_accelerate;
+ absl::optional<bool> audio_jitter_buffer_fast_accelerate;
// Audio processing to detect typing.
- rtc::Optional<bool> typing_detection;
- rtc::Optional<bool> aecm_generate_comfort_noise;
- rtc::Optional<bool> experimental_agc;
- rtc::Optional<bool> extended_filter_aec;
- rtc::Optional<bool> delay_agnostic_aec;
- rtc::Optional<bool> experimental_ns;
- rtc::Optional<bool> intelligibility_enhancer;
+ absl::optional<bool> typing_detection;
+ absl::optional<bool> aecm_generate_comfort_noise;
+ absl::optional<bool> experimental_agc;
+ absl::optional<bool> extended_filter_aec;
+ absl::optional<bool> delay_agnostic_aec;
+ absl::optional<bool> experimental_ns;
+ absl::optional<bool> intelligibility_enhancer;
// Note that tx_agc_* only applies to non-experimental AGC.
- rtc::Optional<bool> residual_echo_detector;
- rtc::Optional<uint16_t> tx_agc_target_dbov;
- rtc::Optional<uint16_t> tx_agc_digital_compression_gain;
- rtc::Optional<bool> tx_agc_limiter;
+ absl::optional<bool> residual_echo_detector;
+ absl::optional<uint16_t> tx_agc_target_dbov;
+ absl::optional<uint16_t> tx_agc_digital_compression_gain;
+ absl::optional<bool> tx_agc_limiter;
// Enable combined audio+bandwidth BWE.
// TODO(pthatcher): This flag is set from the
// "googCombinedAudioVideoBwe", but not used anywhere. So delete it,
// and check if any other AudioOptions members are unused.
- rtc::Optional<bool> combined_audio_video_bwe;
+ absl::optional<bool> combined_audio_video_bwe;
// Enable audio network adaptor.
- rtc::Optional<bool> audio_network_adaptor;
+ absl::optional<bool> audio_network_adaptor;
// Config string for audio network adaptor.
- rtc::Optional<std::string> audio_network_adaptor_config;
+ absl::optional<std::string> audio_network_adaptor_config;
private:
template <class T>
static std::string ToStringIfSet(const char* key,
- const rtc::Optional<T>& val) {
+ const absl::optional<T>& val) {
std::string str;
if (val) {
str = key;
@@ -184,7 +184,7 @@
}
template <typename T>
- static void SetFrom(rtc::Optional<T>* s, const rtc::Optional<T>& o) {
+ static void SetFrom(absl::optional<T>* s, const absl::optional<T>& o) {
if (o) {
*s = o;
}
diff --git a/api/datachannelinterface.h b/api/datachannelinterface.h
index 85e6fa6..5cbe717 100644
--- a/api/datachannelinterface.h
+++ b/api/datachannelinterface.h
@@ -23,7 +23,7 @@
namespace webrtc {
// C++ version of: https://www.w3.org/TR/webrtc/#idl-def-rtcdatachannelinit
-// TODO(deadbeef): Use rtc::Optional for the "-1 if unset" things.
+// TODO(deadbeef): Use absl::optional for the "-1 if unset" things.
struct DataChannelInit {
// Deprecated. Reliability is assumed, and channel will be unreliable if
// maxRetransmitTime or MaxRetransmits is set.
diff --git a/api/jsep.h b/api/jsep.h
index 0118490..dbf97f6 100644
--- a/api/jsep.h
+++ b/api/jsep.h
@@ -26,7 +26,7 @@
#include <string>
#include <vector>
-#include "api/optional.h"
+#include "absl/types/optional.h"
#include "api/rtcerror.h"
#include "rtc_base/refcount.h"
@@ -107,7 +107,7 @@
// 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::Optional<SdpType> SdpTypeFromString(const std::string& type_str);
+absl::optional<SdpType> SdpTypeFromString(const std::string& type_str);
// Class representation of an SDP session description.
//
diff --git a/api/mediaconstraintsinterface.cc b/api/mediaconstraintsinterface.cc
index 50a26de..fb4481f 100644
--- a/api/mediaconstraintsinterface.cc
+++ b/api/mediaconstraintsinterface.cc
@@ -59,11 +59,11 @@
}
// Converts a constraint (mandatory takes precedence over optional) to an
-// rtc::Optional.
+// absl::optional.
template <typename T>
void ConstraintToOptional(const webrtc::MediaConstraintsInterface* constraints,
const std::string& key,
- rtc::Optional<T>* value_out) {
+ absl::optional<T>* value_out) {
T value;
bool present = FindConstraint<T>(constraints, key, &value, nullptr);
if (present) {
diff --git a/api/mediaconstraintsinterface.h b/api/mediaconstraintsinterface.h
index 91043d2..54ab706 100644
--- a/api/mediaconstraintsinterface.h
+++ b/api/mediaconstraintsinterface.h
@@ -23,7 +23,7 @@
#include <string>
#include <vector>
-#include "api/optional.h"
+#include "absl/types/optional.h"
#include "api/peerconnectioninterface.h"
namespace webrtc {
diff --git a/api/mediastreaminterface.h b/api/mediastreaminterface.h
index 416073d..b661351 100644
--- a/api/mediastreaminterface.h
+++ b/api/mediastreaminterface.h
@@ -22,7 +22,7 @@
#include <string>
#include <vector>
-#include "api/optional.h"
+#include "absl/types/optional.h"
#include "api/video/video_frame.h"
// TODO(zhihuang): Remove unrelated headers once downstream applications stop
// relying on them; they were previously transitively included by
@@ -132,7 +132,7 @@
// depending on video codec.
// TODO(perkj): Remove this once denoising is done by the source, and not by
// the encoder.
- virtual rtc::Optional<bool> needs_denoising() const = 0;
+ virtual absl::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.
diff --git a/api/ortc/mediadescription.h b/api/ortc/mediadescription.h
index 1a6d0e9..5cf1d1a 100644
--- a/api/ortc/mediadescription.h
+++ b/api/ortc/mediadescription.h
@@ -15,8 +15,8 @@
#include <utility>
#include <vector>
+#include "absl/types/optional.h"
#include "api/cryptoparams.h"
-#include "api/optional.h"
namespace webrtc {
@@ -31,7 +31,7 @@
// The mid(media stream identification) is used for identifying media streams
// within a session description.
// https://tools.ietf.org/html/rfc5888#section-6
- rtc::Optional<std::string> mid() const { return mid_; }
+ absl::optional<std::string> mid() const { return mid_; }
void set_mid(std::string mid) { mid_.emplace(std::move(mid)); }
// Security keys and parameters for this media stream. Can be used to
@@ -43,7 +43,7 @@
}
private:
- rtc::Optional<std::string> mid_;
+ absl::optional<std::string> mid_;
std::vector<cricket::CryptoParams> sdes_params_;
};
diff --git a/api/ortc/rtptransportinterface.h b/api/ortc/rtptransportinterface.h
index 8822300..b0d30e8 100644
--- a/api/ortc/rtptransportinterface.h
+++ b/api/ortc/rtptransportinterface.h
@@ -13,7 +13,7 @@
#include <string>
-#include "api/optional.h"
+#include "absl/types/optional.h"
#include "api/ortc/packettransportinterface.h"
#include "api/rtcerror.h"
#include "api/rtp_headers.h"
diff --git a/api/peerconnectioninterface.h b/api/peerconnectioninterface.h
index 683a597..cdc3266 100644
--- a/api/peerconnectioninterface.h
+++ b/api/peerconnectioninterface.h
@@ -361,7 +361,7 @@
// The below fields correspond to constraints from the deprecated
// constraints interface for constructing a PeerConnection.
//
- // rtc::Optional fields can be "missing", in which case the implementation
+ // absl::optional fields can be "missing", in which case the implementation
// default will be used.
//////////////////////////////////////////////////////////////////////////
@@ -396,15 +396,15 @@
// 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.
- rtc::Optional<int> screencast_min_bitrate;
+ absl::optional<int> screencast_min_bitrate;
// Use new combined audio/video bandwidth estimation?
- rtc::Optional<bool> combined_audio_video_bwe;
+ absl::optional<bool> combined_audio_video_bwe;
// Can be used to disable DTLS-SRTP. This should never be done, but can be
// useful for testing purposes, for example in setting up a loopback call
// with a single PeerConnection.
- rtc::Optional<bool> enable_dtls_srtp;
+ absl::optional<bool> enable_dtls_srtp;
/////////////////////////////////////////////////
// The below fields are not part of the standard.
@@ -504,29 +504,29 @@
// 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.
- rtc::Optional<int> ice_check_interval_strong_connectivity;
- rtc::Optional<int> ice_check_interval_weak_connectivity;
- rtc::Optional<int> ice_check_min_interval;
+ absl::optional<int> ice_check_interval_strong_connectivity;
+ absl::optional<int> ice_check_interval_weak_connectivity;
+ absl::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.
- rtc::Optional<int> ice_unwritable_timeout;
+ absl::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.
- rtc::Optional<int> ice_unwritable_min_checks;
+ absl::optional<int> ice_unwritable_min_checks;
// The interval in milliseconds at which STUN candidates will resend STUN
// binding requests to keep NAT bindings open.
- rtc::Optional<int> stun_candidate_keepalive_interval;
+ absl::optional<int> stun_candidate_keepalive_interval;
// ICE Periodic Regathering
// If set, WebRTC will periodically create and propose candidates without
// starting a new ICE generation. The regathering happens continuously with
// interval specified in milliseconds by the uniform distribution [a, b].
- rtc::Optional<rtc::IntervalRange> ice_regather_interval_range;
+ absl::optional<rtc::IntervalRange> ice_regather_interval_range;
// Optional TurnCustomizer.
// With this class one can modify outgoing TURN messages.
@@ -538,7 +538,7 @@
// 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.
- rtc::Optional<rtc::AdapterType> network_preference;
+ absl::optional<rtc::AdapterType> network_preference;
// Configure the SDP semantics used by this PeerConnection. Note that the
// WebRTC 1.0 specification requires kUnifiedPlan semantics. The
@@ -979,9 +979,9 @@
// 0 <= min <= current <= max should hold for set parameters.
struct BitrateParameters {
- rtc::Optional<int> min_bitrate_bps;
- rtc::Optional<int> current_bitrate_bps;
- rtc::Optional<int> max_bitrate_bps;
+ absl::optional<int> min_bitrate_bps;
+ absl::optional<int> current_bitrate_bps;
+ absl::optional<int> max_bitrate_bps;
};
// SetBitrate limits the bandwidth allocated for all RTP streams sent by
diff --git a/api/rtp_headers.h b/api/rtp_headers.h
index 3318e60..ded6c4b 100644
--- a/api/rtp_headers.h
+++ b/api/rtp_headers.h
@@ -16,8 +16,8 @@
#include <string>
#include <vector>
+#include "absl/types/optional.h"
#include "api/array_view.h"
-#include "api/optional.h"
#include "api/video/video_content_type.h"
#include "api/video/video_rotation.h"
#include "api/video/video_timing.h"
@@ -102,7 +102,7 @@
bool hasVideoRotation;
VideoRotation videoRotation;
- // TODO(ilnik): Refactor this and one above to be rtc::Optional() and remove
+ // TODO(ilnik): Refactor this and one above to be absl::optional() and remove
// a corresponding bool flag.
bool hasVideoContentType;
VideoContentType videoContentType;
diff --git a/api/rtpparameters.h b/api/rtpparameters.h
index e2405d3..84da811 100644
--- a/api/rtpparameters.h
+++ b/api/rtpparameters.h
@@ -15,8 +15,8 @@
#include <unordered_map>
#include <vector>
+#include "absl/types/optional.h"
#include "api/mediatypes.h"
-#include "api/optional.h"
namespace webrtc {
@@ -94,7 +94,7 @@
// 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.
- rtc::Optional<RtcpFeedbackMessageType> message_type;
+ absl::optional<RtcpFeedbackMessageType> message_type;
// Constructors for convenience.
RtcpFeedback();
@@ -125,23 +125,23 @@
cricket::MediaType kind = cricket::MEDIA_TYPE_AUDIO;
// Clock rate in Hertz. If unset, the codec is applicable to any clock rate.
- rtc::Optional<int> clock_rate;
+ absl::optional<int> clock_rate;
// Default payload type for this codec. Mainly needed for codecs that use
// that have statically assigned payload types.
- rtc::Optional<int> preferred_payload_type;
+ absl::optional<int> preferred_payload_type;
// Maximum packetization time supported by an RtpReceiver for this codec.
// TODO(deadbeef): Not implemented.
- rtc::Optional<int> max_ptime;
+ absl::optional<int> max_ptime;
// Preferred packetization time for an RtpReceiver or RtpSender of this
// codec.
// TODO(deadbeef): Not implemented.
- rtc::Optional<int> ptime;
+ absl::optional<int> ptime;
// The number of audio channels supported. Unused for video codecs.
- rtc::Optional<int> num_channels;
+ absl::optional<int> num_channels;
// Feedback mechanisms supported for this codec.
std::vector<RtcpFeedback> rtcp_feedback;
@@ -204,7 +204,7 @@
std::string uri;
// Preferred value of ID that goes in the packet.
- rtc::Optional<int> preferred_id;
+ absl::optional<int> preferred_id;
// If true, it's preferred that the value in the header is encrypted.
// TODO(deadbeef): Not implemented.
@@ -313,7 +313,7 @@
struct RtpFecParameters {
// If unset, a value is chosen by the implementation.
// Works just like RtpEncodingParameters::ssrc.
- rtc::Optional<uint32_t> ssrc;
+ absl::optional<uint32_t> ssrc;
FecMechanism mechanism = FecMechanism::RED;
@@ -332,7 +332,7 @@
struct RtpRtxParameters {
// If unset, a value is chosen by the implementation.
// Works just like RtpEncodingParameters::ssrc.
- rtc::Optional<uint32_t> ssrc;
+ absl::optional<uint32_t> ssrc;
// Constructors for convenience.
RtpRtxParameters();
@@ -353,7 +353,7 @@
// 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.
- rtc::Optional<uint32_t> ssrc;
+ absl::optional<uint32_t> ssrc;
// Can be used to reference a codec in the |codecs| member of the
// RtpParameters that contains this RtpEncodingParameters. If unset, the
@@ -361,23 +361,23 @@
// prepare to receive any codec (for a receiver).
// TODO(deadbeef): Not implemented. Implementation of RtpSender will always
// choose the first codec from the list.
- rtc::Optional<int> codec_payload_type;
+ absl::optional<int> codec_payload_type;
// Specifies the FEC mechanism, if set.
// TODO(deadbeef): Not implemented. Current implementation will use whatever
// FEC codecs are available, including red+ulpfec.
- rtc::Optional<RtpFecParameters> fec;
+ absl::optional<RtpFecParameters> fec;
// Specifies the RTX parameters, if set.
// TODO(deadbeef): Not implemented with PeerConnection senders/receivers.
- rtc::Optional<RtpRtxParameters> rtx;
+ absl::optional<RtpRtxParameters> rtx;
// Only used for audio. If set, determines whether or not discontinuous
// transmission will be used, if an available codec supports it. If not
// set, the implementation default setting will be used.
// TODO(deadbeef): Not implemented. Current implementation will use a CN
// codec as long as it's present.
- rtc::Optional<DtxStatus> dtx;
+ absl::optional<DtxStatus> dtx;
// The relative bitrate priority of this encoding. Currently this is
// implemented for the entire rtp sender by using the value of the first
@@ -394,7 +394,7 @@
// creates a ptime for a specific codec, which is later changed in the
// RtpEncodingParameters by the application.
// TODO(bugs.webrtc.org/8819): Not implemented.
- rtc::Optional<int> ptime;
+ absl::optional<int> ptime;
// If set, this represents the Transport Independent Application Specific
// maximum bandwidth defined in RFC3890. If unset, there is no maximum
@@ -407,23 +407,23 @@
// 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.
- rtc::Optional<int> max_bitrate_bps;
+ absl::optional<int> max_bitrate_bps;
// Specifies the minimum bitrate in bps for video.
// TODO(asapersson): Not implemented for ORTC API.
// TODO(asapersson): Not implemented for single layer.
- rtc::Optional<int> min_bitrate_bps;
+ absl::optional<int> min_bitrate_bps;
// TODO(deadbeef): Not implemented.
- rtc::Optional<int> max_framerate;
+ absl::optional<int> max_framerate;
// For video, scale the resolution down by this factor.
// TODO(deadbeef): Not implemented.
- rtc::Optional<double> scale_resolution_down_by;
+ absl::optional<double> scale_resolution_down_by;
// Scale the framerate down by this factor.
// TODO(deadbeef): Not implemented.
- rtc::Optional<double> scale_framerate_down_by;
+ absl::optional<double> scale_framerate_down_by;
// 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
@@ -478,24 +478,24 @@
int payload_type = 0;
// If unset, the implementation default is used.
- rtc::Optional<int> clock_rate;
+ absl::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.
- rtc::Optional<int> num_channels;
+ absl::optional<int> num_channels;
// The maximum packetization time to be used by an RtpSender.
// If |ptime| is also set, this will be ignored.
// TODO(deadbeef): Not implemented.
- rtc::Optional<int> max_ptime;
+ absl::optional<int> max_ptime;
// The packetization time to be used by an RtpSender.
// If unset, will use any time up to max_ptime.
// TODO(deadbeef): Not implemented.
- rtc::Optional<int> ptime;
+ absl::optional<int> ptime;
// Feedback mechanisms to be used for this codec.
// TODO(deadbeef): Not implemented with PeerConnection senders/receivers.
@@ -551,7 +551,7 @@
// 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.
- rtc::Optional<uint32_t> ssrc;
+ absl::optional<uint32_t> ssrc;
// The Canonical Name (CNAME) used by RTCP (e.g. in SDES messages).
//
diff --git a/api/rtpreceiverinterface.h b/api/rtpreceiverinterface.h
index 30eb667..1801dc1 100644
--- a/api/rtpreceiverinterface.h
+++ b/api/rtpreceiverinterface.h
@@ -57,8 +57,8 @@
// The source can be either a contributing source or a synchronization source.
RtpSourceType source_type() const { return source_type_; }
- rtc::Optional<uint8_t> audio_level() const { return audio_level_; }
- void set_audio_level(const rtc::Optional<uint8_t>& level) {
+ absl::optional<uint8_t> audio_level() const { return audio_level_; }
+ void set_audio_level(const absl::optional<uint8_t>& level) {
audio_level_ = level;
}
@@ -71,7 +71,7 @@
int64_t timestamp_ms_;
uint32_t source_id_;
RtpSourceType source_type_;
- rtc::Optional<uint8_t> audio_level_;
+ absl::optional<uint8_t> audio_level_;
};
class RtpReceiverObserverInterface {
diff --git a/api/rtpsenderinterface.h b/api/rtpsenderinterface.h
index 66267c7..6003aa0 100644
--- a/api/rtpsenderinterface.h
+++ b/api/rtpsenderinterface.h
@@ -38,7 +38,7 @@
// Returns primary SSRC used by this sender for sending media.
// Returns 0 if not yet determined.
- // TODO(deadbeef): Change to rtc::Optional.
+ // TODO(deadbeef): Change to absl::optional.
// TODO(deadbeef): Remove? With GetParameters this should be redundant.
virtual uint32_t ssrc() const = 0;
diff --git a/api/rtptransceiverinterface.h b/api/rtptransceiverinterface.h
index 7d2a1df..8cb3bd5 100644
--- a/api/rtptransceiverinterface.h
+++ b/api/rtptransceiverinterface.h
@@ -14,8 +14,8 @@
#include <string>
#include <vector>
+#include "absl/types/optional.h"
#include "api/array_view.h"
-#include "api/optional.h"
#include "api/rtpreceiverinterface.h"
#include "api/rtpsenderinterface.h"
#include "rtc_base/refcount.h"
@@ -68,7 +68,7 @@
// 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 rtc::Optional<std::string> mid() const = 0;
+ virtual absl::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,
@@ -105,7 +105,7 @@
// 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 rtc::Optional<RtpTransceiverDirection> current_direction() const = 0;
+ virtual absl::optional<RtpTransceiverDirection> current_direction() const = 0;
// The Stop method irreversibly stops the RtpTransceiver. The sender of this
// transceiver will no longer send, the receiver will no longer receive.
diff --git a/api/transport/BUILD.gn b/api/transport/BUILD.gn
index bd5dcac..f473179 100644
--- a/api/transport/BUILD.gn
+++ b/api/transport/BUILD.gn
@@ -15,7 +15,7 @@
"bitrate_settings.h",
]
deps = [
- "..:optional",
+ "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -27,11 +27,11 @@
]
deps = [
- "..:optional",
"../units:data_rate",
"../units:data_size",
"../units:time_delta",
"../units:timestamp",
+ "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -45,10 +45,10 @@
]
deps = [
":network_control",
- "../:optional",
"../../rtc_base:checks",
"../../rtc_base:rtc_base_approved",
"../../test:test_support",
+ "//third_party/abseil-cpp/absl/types:optional",
]
}
}
diff --git a/api/transport/bitrate_settings.h b/api/transport/bitrate_settings.h
index 1a24d90..77654bc 100644
--- a/api/transport/bitrate_settings.h
+++ b/api/transport/bitrate_settings.h
@@ -11,7 +11,7 @@
#ifndef API_TRANSPORT_BITRATE_SETTINGS_H_
#define API_TRANSPORT_BITRATE_SETTINGS_H_
-#include "api/optional.h"
+#include "absl/types/optional.h"
namespace webrtc {
@@ -25,9 +25,9 @@
~BitrateSettings();
BitrateSettings(const BitrateSettings&);
// 0 <= min <= start <= max should hold for set parameters.
- rtc::Optional<int> min_bitrate_bps;
- rtc::Optional<int> start_bitrate_bps;
- rtc::Optional<int> max_bitrate_bps;
+ absl::optional<int> min_bitrate_bps;
+ absl::optional<int> start_bitrate_bps;
+ absl::optional<int> max_bitrate_bps;
};
} // namespace webrtc
diff --git a/api/transport/network_types.h b/api/transport/network_types.h
index 155d4ec..a389716 100644
--- a/api/transport/network_types.h
+++ b/api/transport/network_types.h
@@ -13,7 +13,7 @@
#include <stdint.h>
#include <vector>
-#include "api/optional.h"
+#include "absl/types/optional.h"
#include "api/units/data_rate.h"
#include "api/units/data_size.h"
#include "api/units/time_delta.h"
@@ -32,10 +32,10 @@
~StreamsConfig();
Timestamp at_time = Timestamp::Infinity();
bool requests_alr_probing = false;
- rtc::Optional<double> pacing_factor;
- rtc::Optional<DataRate> min_pacing_rate;
- rtc::Optional<DataRate> max_padding_rate;
- rtc::Optional<DataRate> max_total_allocated_bitrate;
+ absl::optional<double> pacing_factor;
+ absl::optional<DataRate> min_pacing_rate;
+ absl::optional<DataRate> max_padding_rate;
+ absl::optional<DataRate> max_total_allocated_bitrate;
};
struct TargetRateConstraints {
@@ -43,8 +43,8 @@
TargetRateConstraints(const TargetRateConstraints&);
~TargetRateConstraints();
Timestamp at_time = Timestamp::Infinity();
- rtc::Optional<DataRate> min_data_rate;
- rtc::Optional<DataRate> max_data_rate;
+ absl::optional<DataRate> min_data_rate;
+ absl::optional<DataRate> max_data_rate;
};
// Send side information
@@ -62,7 +62,7 @@
// The TargetRateConstraints are set here so they can be changed synchronously
// when network route changes.
TargetRateConstraints constraints;
- rtc::Optional<DataRate> starting_rate;
+ absl::optional<DataRate> starting_rate;
};
struct PacedPacketInfo {
@@ -121,7 +121,7 @@
PacketResult(const PacketResult&);
~PacketResult();
- rtc::Optional<SentPacket> sent_packet;
+ absl::optional<SentPacket> sent_packet;
Timestamp receive_time = Timestamp::Infinity();
};
@@ -185,10 +185,10 @@
NetworkControlUpdate();
NetworkControlUpdate(const NetworkControlUpdate&);
~NetworkControlUpdate();
- rtc::Optional<DataSize> congestion_window;
- rtc::Optional<PacerConfig> pacer_config;
+ absl::optional<DataSize> congestion_window;
+ absl::optional<PacerConfig> pacer_config;
std::vector<ProbeClusterConfig> probe_cluster_configs;
- rtc::Optional<TargetTransferRate> target_rate;
+ absl::optional<TargetTransferRate> target_rate;
};
// Process control
diff --git a/api/transport/test/network_control_tester.h b/api/transport/test/network_control_tester.h
index 2e9fc02..4dfcc14 100644
--- a/api/transport/test/network_control_tester.h
+++ b/api/transport/test/network_control_tester.h
@@ -15,7 +15,7 @@
#include <functional>
#include <memory>
-#include "api/optional.h"
+#include "absl/types/optional.h"
#include "api/transport/network_control.h"
namespace webrtc {
diff --git a/api/video/BUILD.gn b/api/video/BUILD.gn
index ec07584..b65fa18 100644
--- a/api/video/BUILD.gn
+++ b/api/video/BUILD.gn
@@ -26,9 +26,9 @@
]
deps = [
- "..:optional",
"../../rtc_base:checks",
"../../rtc_base:rtc_base_approved",
+ "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -66,11 +66,11 @@
"video_bitrate_allocation.h",
]
deps = [
- "..:optional",
"../..:typedefs",
"../../rtc_base:checks",
"../../rtc_base:safe_conversions",
"../../rtc_base:stringutils",
+ "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -83,8 +83,8 @@
deps = [
":encoded_frame",
":video_frame",
- "..:optional",
"../video_codecs:video_codecs_api",
+ "//third_party/abseil-cpp/absl/types:optional",
]
}
diff --git a/api/video/video_bitrate_allocation.cc b/api/video/video_bitrate_allocation.cc
index 059eb8f..d5a1db5 100644
--- a/api/video/video_bitrate_allocation.cc
+++ b/api/video/video_bitrate_allocation.cc
@@ -27,7 +27,7 @@
RTC_CHECK_LT(spatial_index, kMaxSpatialLayers);
RTC_CHECK_LT(temporal_index, kMaxTemporalStreams);
int64_t new_bitrate_sum_bps = sum_;
- rtc::Optional<uint32_t>& layer_bitrate =
+ absl::optional<uint32_t>& layer_bitrate =
bitrates_[spatial_index][temporal_index];
if (layer_bitrate) {
RTC_DCHECK_LE(*layer_bitrate, sum_);
diff --git a/api/video/video_bitrate_allocation.h b/api/video/video_bitrate_allocation.h
index b748b67..ab5bfae 100644
--- a/api/video/video_bitrate_allocation.h
+++ b/api/video/video_bitrate_allocation.h
@@ -15,7 +15,7 @@
#include <string>
#include <vector>
-#include "api/optional.h"
+#include "absl/types/optional.h"
#include "typedefs.h" // NOLINT(build/include)
namespace webrtc {
@@ -77,7 +77,7 @@
private:
uint32_t sum_;
- rtc::Optional<uint32_t> bitrates_[kMaxSpatialLayers][kMaxTemporalStreams];
+ absl::optional<uint32_t> bitrates_[kMaxSpatialLayers][kMaxTemporalStreams];
};
} // namespace webrtc
diff --git a/api/video/video_source_interface.h b/api/video/video_source_interface.h
index d4e2d3a..4ee4719 100644
--- a/api/video/video_source_interface.h
+++ b/api/video/video_source_interface.h
@@ -13,7 +13,7 @@
#include <limits>
-#include "api/optional.h"
+#include "absl/types/optional.h"
#include "api/video/video_sink_interface.h"
namespace rtc {
@@ -38,7 +38,7 @@
// 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.
- rtc::Optional<int> target_pixel_count;
+ absl::optional<int> target_pixel_count;
// Tells the source the maximum framerate the sink wants.
int max_framerate_fps = std::numeric_limits<int>::max();
};
diff --git a/api/video/video_stream_decoder.h b/api/video/video_stream_decoder.h
index 1c4c5ff..dff60d8 100644
--- a/api/video/video_stream_decoder.h
+++ b/api/video/video_stream_decoder.h
@@ -37,8 +37,8 @@
// Called with the decoded frame.
virtual void OnDecodedFrame(VideoFrame decodedImage,
- rtc::Optional<int> decode_time_ms,
- rtc::Optional<int> qp) = 0;
+ absl::optional<int> decode_time_ms,
+ absl::optional<int> qp) = 0;
};
virtual ~VideoStreamDecoder() = default;
diff --git a/api/video_codecs/BUILD.gn b/api/video_codecs/BUILD.gn
index 079df98..fa62518 100644
--- a/api/video_codecs/BUILD.gn
+++ b/api/video_codecs/BUILD.gn
@@ -30,13 +30,13 @@
]
deps = [
- "..:optional",
"../..:webrtc_common",
"../../common_video",
"../../rtc_base:checks",
"../../rtc_base:rtc_base_approved",
"../video:video_bitrate_allocation",
"../video:video_frame",
+ "//third_party/abseil-cpp/absl/types:optional",
]
}
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 29f005a..d2d0e2b 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
@@ -178,8 +178,8 @@
return -1;
}
void Decoded(webrtc::VideoFrame& decodedImage,
- rtc::Optional<int32_t> decode_time_ms,
- rtc::Optional<uint8_t> qp) override {
+ absl::optional<int32_t> decode_time_ms,
+ absl::optional<uint8_t> qp) override {
RTC_NOTREACHED();
}
} callback;
diff --git a/api/video_codecs/video_decoder.cc b/api/video_codecs/video_decoder.cc
index 4e8db88..b5fff32 100644
--- a/api/video_codecs/video_decoder.cc
+++ b/api/video_codecs/video_decoder.cc
@@ -19,8 +19,8 @@
}
void DecodedImageCallback::Decoded(VideoFrame& decodedImage,
- rtc::Optional<int32_t> decode_time_ms,
- rtc::Optional<uint8_t> qp) {
+ absl::optional<int32_t> decode_time_ms,
+ absl::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 190d794..7995fcc 100644
--- a/api/video_codecs/video_decoder.h
+++ b/api/video_codecs/video_decoder.h
@@ -39,8 +39,8 @@
// TODO(sakal): Remove other implementations when upstream projects have been
// updated.
virtual void Decoded(VideoFrame& decodedImage,
- rtc::Optional<int32_t> decode_time_ms,
- rtc::Optional<uint8_t> qp);
+ absl::optional<int32_t> decode_time_ms,
+ absl::optional<uint8_t> qp);
virtual int32_t ReceivedDecodedReferenceFrame(const uint64_t pictureId);
diff --git a/api/video_codecs/video_encoder.h b/api/video_codecs/video_encoder.h
index b2f9a39..68d9b44 100644
--- a/api/video_codecs/video_encoder.h
+++ b/api/video_codecs/video_encoder.h
@@ -15,7 +15,7 @@
#include <string>
#include <vector>
-#include "api/optional.h"
+#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"
@@ -90,7 +90,7 @@
public:
// TODO(nisse): Would be nicer if kOff were a constant ScalingSettings
- // rather than a magic value. However, rtc::Optional is not trivially copy
+ // rather than a magic value. However, absl::optional is not trivially copy
// constructible, and hence a constant ScalingSettings needs a static
// initializer, which is strongly discouraged in Chrome. We can hopefully
// fix this when we switch to absl::optional or std::optional.
@@ -102,7 +102,7 @@
ScalingSettings(KOff); // NOLINT(runtime/explicit)
~ScalingSettings();
- const rtc::Optional<QpThresholds> thresholds;
+ const absl::optional<QpThresholds> thresholds;
// We will never ask for a resolution lower than this.
// TODO(kthelgason): Lower this limit when better testing
diff --git a/api/video_codecs/video_encoder_config.h b/api/video_codecs/video_encoder_config.h
index 636c0e7..47ff925 100644
--- a/api/video_codecs/video_encoder_config.h
+++ b/api/video_codecs/video_encoder_config.h
@@ -14,9 +14,9 @@
#include <string>
#include <vector>
-#include "api/optional.h"
-#include "api/video_codecs/video_codec.h"
+#include "absl/types/optional.h"
#include "api/video_codecs/sdp_video_format.h"
+#include "api/video_codecs/video_codec.h"
#include "rtc_base/refcount.h"
#include "rtc_base/scoped_ref_ptr.h"
@@ -37,9 +37,9 @@
int max_bitrate_bps;
int max_qp;
- rtc::Optional<size_t> num_temporal_layers;
+ absl::optional<size_t> num_temporal_layers;
- rtc::Optional<double> bitrate_priority;
+ absl::optional<double> bitrate_priority;
// TODO(bugs.webrtc.org/8653): Support active per-simulcast layer.
bool active;
@@ -50,7 +50,7 @@
// 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 rtc::Optional for encoder_specific_settings instead.
+ // and use absl::optional for encoder_specific_settings instead.
class EncoderSpecificSettings : public rtc::RefCountInterface {
public:
// TODO(pbos): Remove FillEncoderSpecificSettings as soon as VideoCodec is
diff --git a/api/videosourceproxy.h b/api/videosourceproxy.h
index 0bb6d48..dbd9045 100644
--- a/api/videosourceproxy.h
+++ b/api/videosourceproxy.h
@@ -25,7 +25,7 @@
PROXY_CONSTMETHOD0(SourceState, state)
PROXY_CONSTMETHOD0(bool, remote)
PROXY_CONSTMETHOD0(bool, is_screencast)
-PROXY_CONSTMETHOD0(rtc::Optional<bool>, needs_denoising)
+PROXY_CONSTMETHOD0(absl::optional<bool>, needs_denoising)
PROXY_METHOD1(bool, GetStats, Stats*)
PROXY_WORKER_METHOD2(void,
AddOrUpdateSink,
