Replace rtc::Optional with absl::optional in modules/audio_coding
This is a no-op change because rtc::Optional is an alias to absl::optional
This CL generated by running script with parameter 'modules/audio_coding'
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: Ic980ee605148fdb160666d4aa03cc87175e48fe8
Reviewed-on: https://webrtc-review.googlesource.com/84130
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23659}diff --git a/modules/audio_coding/BUILD.gn b/modules/audio_coding/BUILD.gn
index f5e8495..1dd3939 100644
--- a/modules/audio_coding/BUILD.gn
+++ b/modules/audio_coding/BUILD.gn
@@ -50,11 +50,11 @@
deps = [
"../..:webrtc_common",
"../../api:array_view",
- "../../api:optional",
"../../api/audio_codecs:audio_codecs_api",
"../../rtc_base:checks",
"../../rtc_base:rtc_base_approved",
"../../rtc_base:sanitizer",
+ "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -70,7 +70,7 @@
"../..:typedefs",
"../../rtc_base:checks",
"../../api:array_view",
- "../../api:optional",
+ "//third_party/abseil-cpp/absl/types:optional",
"../../api/audio_codecs:audio_codecs_api",
"../..:webrtc_common",
"../../rtc_base:protobuf_utils",
@@ -150,7 +150,7 @@
":rent_a_codec",
"../../rtc_base:audio_format_to_string",
"../../rtc_base:rtc_base_approved",
- "../../api:optional",
+ "//third_party/abseil-cpp/absl/types:optional",
"../../logging:rtc_event_log_api",
]
defines = audio_coding_defines
@@ -503,10 +503,10 @@
":isac_bwinfo",
"../..:typedefs",
"../..:webrtc_common",
- "../../api:optional",
"../../api/audio_codecs:audio_codecs_api",
"../../rtc_base:checks",
"../../rtc_base:rtc_base_approved",
+ "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -892,7 +892,6 @@
deps = [
":audio_network_adaptor",
"../..:webrtc_common",
- "../../api:optional",
"../../api/audio_codecs:audio_codecs_api",
"../../api/audio_codecs/opus:audio_encoder_opus_config",
"../../common_audio",
@@ -901,6 +900,7 @@
"../../rtc_base:rtc_numerics",
"../../rtc_base:safe_minmax",
"../../system_wrappers:field_trial_api",
+ "//third_party/abseil-cpp/absl/types:optional",
]
public_deps = [
":webrtc_opus_c",
@@ -968,7 +968,7 @@
"audio_network_adaptor/include/audio_network_adaptor_config.h",
]
deps = [
- "../../api:optional",
+ "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -1007,7 +1007,6 @@
deps = [
"../..:webrtc_common",
- "../../api:optional",
"../../api/audio_codecs:audio_codecs_api",
"../../common_audio",
"../../logging:rtc_event_audio",
@@ -1018,6 +1017,7 @@
"../../rtc_base/system:file_wrapper",
"../../system_wrappers",
"../../system_wrappers:field_trial_api",
+ "//third_party/abseil-cpp/absl/types:optional",
]
if (rtc_enable_protobuf) {
@@ -1039,9 +1039,9 @@
"neteq/neteq_decoder_enum.h",
]
deps = [
- "../../api:optional",
"../../api/audio_codecs:audio_codecs_api",
"../../rtc_base:rtc_base_approved",
+ "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -1129,7 +1129,6 @@
"../..:typedefs",
"../..:webrtc_common",
"../../api:libjingle_peerconnection_api",
- "../../api:optional",
"../../api/audio:audio_frame_api",
"../../api/audio_codecs:audio_codecs_api",
"../../common_audio",
@@ -1143,6 +1142,7 @@
"../../rtc_base/system:fallthrough",
"../../system_wrappers:field_trial_api",
"../../system_wrappers:metrics_api",
+ "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -1177,13 +1177,13 @@
"../..:typedefs",
"../..:webrtc_common",
"../../api:libjingle_peerconnection_api",
- "../../api:optional",
"../../api/audio:audio_frame_api",
"../../api/audio_codecs:audio_codecs_api",
"../../api/audio_codecs:builtin_audio_decoder_factory",
"../../rtc_base:checks",
"../../rtc_base:rtc_base_approved",
"../rtp_rtcp",
+ "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -1270,12 +1270,12 @@
"../..:typedefs",
"../..:webrtc_common",
"../../api:array_view",
- "../../api:optional",
"../../api/audio_codecs:audio_codecs_api",
"../../rtc_base:checks",
"../../rtc_base:rtc_base_approved",
"../rtp_rtcp",
"../rtp_rtcp:rtp_rtcp_format",
+ "//third_party/abseil-cpp/absl/types:optional",
]
public_deps = [
@@ -1422,7 +1422,6 @@
"..:module_api",
"../..:typedefs",
"../..:webrtc_common",
- "../../api:optional",
"../../api/audio:audio_frame_api",
"../../api/audio_codecs:builtin_audio_decoder_factory",
"../../rtc_base:checks",
@@ -1431,6 +1430,7 @@
"../../system_wrappers",
"../../test:fileutils",
"../../test:test_support",
+ "//third_party/abseil-cpp/absl/types:optional",
]
defines = audio_coding_defines
if (is_win) {
@@ -1537,7 +1537,6 @@
"..:module_api",
"../..:typedefs",
"../../:webrtc_common",
- "../../api:optional",
"../../api/audio:audio_frame_api",
"../../api/audio_codecs:builtin_audio_decoder_factory",
"../../rtc_base:checks",
@@ -1548,6 +1547,7 @@
"../../test:test_support",
"../rtp_rtcp",
"//testing/gtest",
+ "//third_party/abseil-cpp/absl/types:optional",
]
} # delay_test
diff --git a/modules/audio_coding/acm2/acm_receiver.cc b/modules/audio_coding/acm2/acm_receiver.cc
index 84efc5c..b61099c 100644
--- a/modules/audio_coding/acm2/acm_receiver.cc
+++ b/modules/audio_coding/acm2/acm_receiver.cc
@@ -64,7 +64,7 @@
return neteq_->LeastRequiredDelayMs();
}
-rtc::Optional<int> AcmReceiver::last_packet_sample_rate_hz() const {
+absl::optional<int> AcmReceiver::last_packet_sample_rate_hz() const {
rtc::CritScope lock(&crit_sect_);
return last_packet_sample_rate_hz_;
}
@@ -86,7 +86,7 @@
{
rtc::CritScope lock(&crit_sect_);
- const rtc::Optional<CodecInst> ci =
+ const absl::optional<CodecInst> ci =
RtpHeaderToDecoder(*header, incoming_payload[0]);
if (!ci) {
RTC_LOG_F(LS_ERROR) << "Payload-type "
@@ -202,15 +202,15 @@
const auto neteq_decoder = [acm_codec_id, channels]() -> NetEqDecoder {
if (acm_codec_id == -1)
return NetEqDecoder::kDecoderArbitrary; // External decoder.
- const rtc::Optional<RentACodec::CodecId> cid =
+ const absl::optional<RentACodec::CodecId> cid =
RentACodec::CodecIdFromIndex(acm_codec_id);
RTC_DCHECK(cid) << "Invalid codec index: " << acm_codec_id;
- const rtc::Optional<NetEqDecoder> ned =
+ const absl::optional<NetEqDecoder> ned =
RentACodec::NetEqDecoderFromCodecId(*cid, channels);
RTC_DCHECK(ned) << "Invalid codec ID: " << static_cast<int>(*cid);
return *ned;
}();
- const rtc::Optional<SdpAudioFormat> new_format =
+ const absl::optional<SdpAudioFormat> new_format =
NetEqDecoderToSdpAudioFormat(neteq_decoder);
rtc::CritScope lock(&crit_sect_);
@@ -276,9 +276,9 @@
void AcmReceiver::RemoveAllCodecs() {
rtc::CritScope lock(&crit_sect_);
neteq_->RemoveAllPayloadTypes();
- last_audio_decoder_ = rtc::nullopt;
- last_audio_format_ = rtc::nullopt;
- last_packet_sample_rate_hz_ = rtc::nullopt;
+ last_audio_decoder_ = absl::nullopt;
+ last_audio_format_ = absl::nullopt;
+ last_packet_sample_rate_hz_ = absl::nullopt;
}
int AcmReceiver::RemoveCodec(uint8_t payload_type) {
@@ -289,14 +289,14 @@
return -1;
}
if (last_audio_decoder_ && payload_type == last_audio_decoder_->pltype) {
- last_audio_decoder_ = rtc::nullopt;
- last_audio_format_ = rtc::nullopt;
- last_packet_sample_rate_hz_ = rtc::nullopt;
+ last_audio_decoder_ = absl::nullopt;
+ last_audio_format_ = absl::nullopt;
+ last_packet_sample_rate_hz_ = absl::nullopt;
}
return 0;
}
-rtc::Optional<uint32_t> AcmReceiver::GetPlayoutTimestamp() {
+absl::optional<uint32_t> AcmReceiver::GetPlayoutTimestamp() {
return neteq_->GetPlayoutTimestamp();
}
@@ -317,7 +317,7 @@
return 0;
}
-rtc::Optional<SdpAudioFormat> AcmReceiver::LastAudioFormat() const {
+absl::optional<SdpAudioFormat> AcmReceiver::LastAudioFormat() const {
rtc::CritScope lock(&crit_sect_);
return last_audio_format_;
}
@@ -354,7 +354,7 @@
int AcmReceiver::DecoderByPayloadType(uint8_t payload_type,
CodecInst* codec) const {
rtc::CritScope lock(&crit_sect_);
- const rtc::Optional<CodecInst> ci = neteq_->GetDecoder(payload_type);
+ const absl::optional<CodecInst> ci = neteq_->GetDecoder(payload_type);
if (ci) {
*codec = *ci;
return 0;
@@ -384,10 +384,10 @@
// TODO(turajs): Should NetEq Buffer be flushed?
}
-const rtc::Optional<CodecInst> AcmReceiver::RtpHeaderToDecoder(
+const absl::optional<CodecInst> AcmReceiver::RtpHeaderToDecoder(
const RTPHeader& rtp_header,
uint8_t first_payload_byte) const {
- const rtc::Optional<CodecInst> ci =
+ const absl::optional<CodecInst> ci =
neteq_->GetDecoder(rtp_header.payloadType);
if (ci && STR_CASE_CMP(ci->plname, "red") == 0) {
// This is a RED packet. Get the payload of the audio codec.
diff --git a/modules/audio_coding/acm2/acm_receiver.h b/modules/audio_coding/acm2/acm_receiver.h
index ce1e1f2..c0afbb1 100644
--- a/modules/audio_coding/acm2/acm_receiver.h
+++ b/modules/audio_coding/acm2/acm_receiver.h
@@ -16,9 +16,9 @@
#include <string>
#include <vector>
-#include "api/audio/audio_frame.h"
+#include "absl/types/optional.h"
#include "api/array_view.h"
-#include "api/optional.h"
+#include "api/audio/audio_frame.h"
#include "common_audio/vad/include/webrtc_vad.h"
#include "modules/audio_coding/acm2/acm_resampler.h"
#include "modules/audio_coding/acm2/call_statistics.h"
@@ -159,7 +159,7 @@
// packet. If no packet of a registered non-CNG codec has been received, the
// return value is empty. Also, if the decoder was unregistered since the last
// packet was inserted, the return value is empty.
- rtc::Optional<int> last_packet_sample_rate_hz() const;
+ absl::optional<int> last_packet_sample_rate_hz() const;
// Returns last_output_sample_rate_hz from the NetEq instance.
int last_output_sample_rate_hz() const;
@@ -195,7 +195,7 @@
// Returns the RTP timestamp for the last sample delivered by GetAudio().
// The return value will be empty if no valid timestamp is available.
- rtc::Optional<uint32_t> GetPlayoutTimestamp();
+ absl::optional<uint32_t> GetPlayoutTimestamp();
// Returns the current total delay from NetEq (packet buffer and sync buffer)
// in ms, with smoothing applied to even out short-time fluctuations due to
@@ -215,7 +215,7 @@
//
int LastAudioCodec(CodecInst* codec) const;
- rtc::Optional<SdpAudioFormat> LastAudioFormat() const;
+ absl::optional<SdpAudioFormat> LastAudioFormat() const;
//
// Get a decoder given its registered payload-type.
@@ -273,22 +273,23 @@
int sample_rate_hz;
};
- const rtc::Optional<CodecInst> RtpHeaderToDecoder(const RTPHeader& rtp_header,
- uint8_t first_payload_byte)
- const RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
+ const absl::optional<CodecInst> RtpHeaderToDecoder(
+ const RTPHeader& rtp_header,
+ uint8_t first_payload_byte) const
+ RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
uint32_t NowInTimestamp(int decoder_sampling_rate) const;
rtc::CriticalSection crit_sect_;
- rtc::Optional<CodecInst> last_audio_decoder_ RTC_GUARDED_BY(crit_sect_);
- rtc::Optional<SdpAudioFormat> last_audio_format_ RTC_GUARDED_BY(crit_sect_);
+ absl::optional<CodecInst> last_audio_decoder_ RTC_GUARDED_BY(crit_sect_);
+ absl::optional<SdpAudioFormat> last_audio_format_ RTC_GUARDED_BY(crit_sect_);
ACMResampler resampler_ RTC_GUARDED_BY(crit_sect_);
std::unique_ptr<int16_t[]> last_audio_buffer_ RTC_GUARDED_BY(crit_sect_);
CallStatistics call_stats_ RTC_GUARDED_BY(crit_sect_);
const std::unique_ptr<NetEq> neteq_; // NetEq is thread-safe; no lock needed.
const Clock* const clock_;
bool resampled_last_output_frame_ RTC_GUARDED_BY(crit_sect_);
- rtc::Optional<int> last_packet_sample_rate_hz_ RTC_GUARDED_BY(crit_sect_);
+ absl::optional<int> last_packet_sample_rate_hz_ RTC_GUARDED_BY(crit_sect_);
};
} // namespace acm2
diff --git a/modules/audio_coding/acm2/audio_coding_module.cc b/modules/audio_coding/acm2/audio_coding_module.cc
index 4545810..2d8827c 100644
--- a/modules/audio_coding/acm2/audio_coding_module.cc
+++ b/modules/audio_coding/acm2/audio_coding_module.cc
@@ -54,7 +54,7 @@
rtc::FunctionView<void(const AudioEncoder*)> query) override;
// Get current send codec.
- rtc::Optional<CodecInst> SendCodec() const override;
+ absl::optional<CodecInst> SendCodec() const override;
// Get current send frequency.
int SendFrequency() const override;
@@ -142,7 +142,7 @@
// Get current received codec.
int ReceiveCodec(CodecInst* current_codec) const override;
- rtc::Optional<SdpAudioFormat> ReceiveFormat() const override;
+ absl::optional<SdpAudioFormat> ReceiveFormat() const override;
// Incoming packet from network parsed and ready for decode.
int IncomingPacket(const uint8_t* incoming_payload,
@@ -160,7 +160,7 @@
RTC_DEPRECATED int32_t PlayoutTimestamp(uint32_t* timestamp) override;
- rtc::Optional<uint32_t> PlayoutTimestamp() override;
+ absl::optional<uint32_t> PlayoutTimestamp() override;
int FilteredCurrentDelayMs() const override;
@@ -603,7 +603,7 @@
}
// Get current send codec.
-rtc::Optional<CodecInst> AudioCodingModuleImpl::SendCodec() const {
+absl::optional<CodecInst> AudioCodingModuleImpl::SendCodec() const {
rtc::CritScope lock(&acm_crit_sect_);
if (encoder_factory_) {
auto* ci = encoder_factory_->codec_manager.GetCodecInst();
@@ -616,12 +616,12 @@
if (enc) {
return acm2::CodecManager::ForgeCodecInst(enc.get());
}
- return rtc::nullopt;
+ return absl::nullopt;
} else {
return encoder_stack_
- ? rtc::Optional<CodecInst>(
+ ? absl::optional<CodecInst>(
acm2::CodecManager::ForgeCodecInst(encoder_stack_.get()))
- : rtc::nullopt;
+ : absl::nullopt;
}
}
@@ -640,7 +640,7 @@
void AudioCodingModuleImpl::SetBitRate(int bitrate_bps) {
rtc::CritScope lock(&acm_crit_sect_);
if (encoder_stack_) {
- encoder_stack_->OnReceivedUplinkBandwidth(bitrate_bps, rtc::nullopt);
+ encoder_stack_->OnReceivedUplinkBandwidth(bitrate_bps, absl::nullopt);
}
}
@@ -1062,7 +1062,7 @@
return receiver_.LastAudioCodec(current_codec);
}
-rtc::Optional<SdpAudioFormat> AudioCodingModuleImpl::ReceiveFormat() const {
+absl::optional<SdpAudioFormat> AudioCodingModuleImpl::ReceiveFormat() const {
rtc::CritScope lock(&acm_crit_sect_);
return receiver_.LastAudioFormat();
}
@@ -1180,14 +1180,14 @@
}
int32_t AudioCodingModuleImpl::PlayoutTimestamp(uint32_t* timestamp) {
- rtc::Optional<uint32_t> ts = PlayoutTimestamp();
+ absl::optional<uint32_t> ts = PlayoutTimestamp();
if (!ts)
return -1;
*timestamp = *ts;
return 0;
}
-rtc::Optional<uint32_t> AudioCodingModuleImpl::PlayoutTimestamp() {
+absl::optional<uint32_t> AudioCodingModuleImpl::PlayoutTimestamp() {
return receiver_.GetPlayoutTimestamp();
}
@@ -1279,7 +1279,7 @@
CodecInst* codec,
int sampling_freq_hz,
size_t channels) {
- rtc::Optional<CodecInst> ci = acm2::RentACodec::CodecInstByParams(
+ absl::optional<CodecInst> ci = acm2::RentACodec::CodecInstByParams(
payload_name, sampling_freq_hz, channels);
if (ci) {
*codec = *ci;
@@ -1299,12 +1299,12 @@
int AudioCodingModule::Codec(const char* payload_name,
int sampling_freq_hz,
size_t channels) {
- rtc::Optional<acm2::RentACodec::CodecId> ci =
+ absl::optional<acm2::RentACodec::CodecId> ci =
acm2::RentACodec::CodecIdByParams(payload_name, sampling_freq_hz,
channels);
if (!ci)
return -1;
- rtc::Optional<int> i = acm2::RentACodec::CodecIndexFromId(*ci);
+ absl::optional<int> i = acm2::RentACodec::CodecIndexFromId(*ci);
return i ? *i : -1;
}
diff --git a/modules/audio_coding/acm2/audio_coding_module_unittest.cc b/modules/audio_coding/acm2/audio_coding_module_unittest.cc
index e4926b6..e16d54a 100644
--- a/modules/audio_coding/acm2/audio_coding_module_unittest.cc
+++ b/modules/audio_coding/acm2/audio_coding_module_unittest.cc
@@ -241,7 +241,7 @@
// These two have to be kept in sync for now. In the future, we'll be able to
// eliminate the CodecInst and keep only the SdpAudioFormat.
- rtc::Optional<SdpAudioFormat> audio_format_;
+ absl::optional<SdpAudioFormat> audio_format_;
CodecInst codec_;
Clock* clock_;
@@ -1058,7 +1058,7 @@
}
std::unique_ptr<AudioDecoder> MakeAudioDecoder(
const SdpAudioFormat& format,
- rtc::Optional<AudioCodecPairId> codec_pair_id) override {
+ absl::optional<AudioCodecPairId> codec_pair_id) override {
return format.name == "MockPCMu"
? std::move(mock_decoder_)
: fact_->MakeAudioDecoder(format, codec_pair_id);
diff --git a/modules/audio_coding/acm2/codec_manager.h b/modules/audio_coding/acm2/codec_manager.h
index 7485426..ffbad96 100644
--- a/modules/audio_coding/acm2/codec_manager.h
+++ b/modules/audio_coding/acm2/codec_manager.h
@@ -13,7 +13,7 @@
#include <map>
-#include "api/optional.h"
+#include "absl/types/optional.h"
#include "common_types.h" // NOLINT(build/include)
#include "modules/audio_coding/acm2/rent_a_codec.h"
#include "modules/audio_coding/include/audio_coding_module.h"
@@ -43,7 +43,7 @@
return send_codec_inst_ ? &*send_codec_inst_ : nullptr;
}
- void UnsetCodecInst() { send_codec_inst_ = rtc::nullopt; }
+ void UnsetCodecInst() { send_codec_inst_ = absl::nullopt; }
const RentACodec::StackParameters* GetStackParams() const {
return &codec_stack_params_;
@@ -63,7 +63,7 @@
private:
rtc::ThreadChecker thread_checker_;
- rtc::Optional<CodecInst> send_codec_inst_;
+ absl::optional<CodecInst> send_codec_inst_;
RentACodec::StackParameters codec_stack_params_;
bool recreate_encoder_ = true; // Need to recreate encoder?
diff --git a/modules/audio_coding/acm2/rent_a_codec.cc b/modules/audio_coding/acm2/rent_a_codec.cc
index 78db38d..818e17f 100644
--- a/modules/audio_coding/acm2/rent_a_codec.cc
+++ b/modules/audio_coding/acm2/rent_a_codec.cc
@@ -44,7 +44,7 @@
namespace webrtc {
namespace acm2 {
-rtc::Optional<RentACodec::CodecId> RentACodec::CodecIdByParams(
+absl::optional<RentACodec::CodecId> RentACodec::CodecIdByParams(
const char* payload_name,
int sampling_freq_hz,
size_t channels) {
@@ -52,25 +52,25 @@
ACMCodecDB::CodecId(payload_name, sampling_freq_hz, channels));
}
-rtc::Optional<CodecInst> RentACodec::CodecInstById(CodecId codec_id) {
- rtc::Optional<int> mi = CodecIndexFromId(codec_id);
- return mi ? rtc::Optional<CodecInst>(Database()[*mi])
- : rtc::nullopt;
+absl::optional<CodecInst> RentACodec::CodecInstById(CodecId codec_id) {
+ absl::optional<int> mi = CodecIndexFromId(codec_id);
+ return mi ? absl::optional<CodecInst>(Database()[*mi]) : absl::nullopt;
}
-rtc::Optional<RentACodec::CodecId> RentACodec::CodecIdByInst(
+absl::optional<RentACodec::CodecId> RentACodec::CodecIdByInst(
const CodecInst& codec_inst) {
return CodecIdFromIndex(ACMCodecDB::CodecNumber(codec_inst));
}
-rtc::Optional<CodecInst> RentACodec::CodecInstByParams(const char* payload_name,
- int sampling_freq_hz,
- size_t channels) {
- rtc::Optional<CodecId> codec_id =
+absl::optional<CodecInst> RentACodec::CodecInstByParams(
+ const char* payload_name,
+ int sampling_freq_hz,
+ size_t channels) {
+ absl::optional<CodecId> codec_id =
CodecIdByParams(payload_name, sampling_freq_hz, channels);
if (!codec_id)
- return rtc::nullopt;
- rtc::Optional<CodecInst> ci = CodecInstById(*codec_id);
+ return absl::nullopt;
+ absl::optional<CodecInst> ci = CodecInstById(*codec_id);
RTC_DCHECK(ci);
// Keep the number of channels from the function call. For most codecs it
@@ -84,13 +84,13 @@
return ACMCodecDB::CodecNumber(codec_inst) >= 0;
}
-rtc::Optional<bool> RentACodec::IsSupportedNumChannels(CodecId codec_id,
- size_t num_channels) {
+absl::optional<bool> RentACodec::IsSupportedNumChannels(CodecId codec_id,
+ size_t num_channels) {
auto i = CodecIndexFromId(codec_id);
- return i ? rtc::Optional<bool>(
+ return i ? absl::optional<bool>(
ACMCodecDB::codec_settings_[*i].channel_support >=
num_channels)
- : rtc::nullopt;
+ : absl::nullopt;
}
rtc::ArrayView<const CodecInst> RentACodec::Database() {
@@ -98,12 +98,12 @@
NumberOfCodecs());
}
-rtc::Optional<NetEqDecoder> RentACodec::NetEqDecoderFromCodecId(
+absl::optional<NetEqDecoder> RentACodec::NetEqDecoderFromCodecId(
CodecId codec_id,
size_t num_channels) {
- rtc::Optional<int> i = CodecIndexFromId(codec_id);
+ absl::optional<int> i = CodecIndexFromId(codec_id);
if (!i)
- return rtc::nullopt;
+ return absl::nullopt;
const NetEqDecoder ned = ACMCodecDB::neteq_decoders_[*i];
return (ned == NetEqDecoder::kDecoderOpus && num_channels == 2)
? NetEqDecoder::kDecoderOpus_2ch
@@ -276,8 +276,7 @@
auto pt = [¶m](const std::map<int, int>& m) {
auto it = m.find(param->speech_encoder->SampleRateHz());
- return it == m.end() ? rtc::nullopt
- : rtc::Optional<int>(it->second);
+ return it == m.end() ? absl::nullopt : absl::optional<int>(it->second);
};
auto cng_pt = pt(param->cng_payload_types);
param->use_cng =
diff --git a/modules/audio_coding/acm2/rent_a_codec.h b/modules/audio_coding/acm2/rent_a_codec.h
index f8fac4c..02f9d03 100644
--- a/modules/audio_coding/acm2/rent_a_codec.h
+++ b/modules/audio_coding/acm2/rent_a_codec.h
@@ -15,10 +15,10 @@
#include <map>
#include <memory>
+#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/audio_codecs/audio_decoder.h"
#include "api/audio_codecs/audio_encoder.h"
-#include "api/optional.h"
#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
#include "modules/audio_coding/neteq/neteq_decoder_enum.h"
#include "rtc_base/constructormagic.h"
@@ -107,28 +107,28 @@
return static_cast<size_t>(CodecId::kNumCodecs);
}
- static inline rtc::Optional<int> CodecIndexFromId(CodecId codec_id) {
+ static inline absl::optional<int> CodecIndexFromId(CodecId codec_id) {
const int i = static_cast<int>(codec_id);
return i >= 0 && i < static_cast<int>(NumberOfCodecs())
- ? rtc::Optional<int>(i)
- : rtc::nullopt;
+ ? absl::optional<int>(i)
+ : absl::nullopt;
}
- static inline rtc::Optional<CodecId> CodecIdFromIndex(int codec_index) {
+ static inline absl::optional<CodecId> CodecIdFromIndex(int codec_index) {
return static_cast<size_t>(codec_index) < NumberOfCodecs()
- ? rtc::Optional<RentACodec::CodecId>(
+ ? absl::optional<RentACodec::CodecId>(
static_cast<RentACodec::CodecId>(codec_index))
- : rtc::nullopt;
+ : absl::nullopt;
}
- static rtc::Optional<CodecId> CodecIdByParams(const char* payload_name,
- int sampling_freq_hz,
- size_t channels);
- static rtc::Optional<CodecInst> CodecInstById(CodecId codec_id);
- static rtc::Optional<CodecId> CodecIdByInst(const CodecInst& codec_inst);
- static rtc::Optional<CodecInst> CodecInstByParams(const char* payload_name,
- int sampling_freq_hz,
- size_t channels);
+ static absl::optional<CodecId> CodecIdByParams(const char* payload_name,
+ int sampling_freq_hz,
+ size_t channels);
+ static absl::optional<CodecInst> CodecInstById(CodecId codec_id);
+ static absl::optional<CodecId> CodecIdByInst(const CodecInst& codec_inst);
+ static absl::optional<CodecInst> CodecInstByParams(const char* payload_name,
+ int sampling_freq_hz,
+ size_t channels);
static bool IsCodecValid(const CodecInst& codec_inst);
static inline bool IsPayloadTypeValid(int payload_type) {
@@ -137,10 +137,10 @@
static rtc::ArrayView<const CodecInst> Database();
- static rtc::Optional<bool> IsSupportedNumChannels(CodecId codec_id,
- size_t num_channels);
+ static absl::optional<bool> IsSupportedNumChannels(CodecId codec_id,
+ size_t num_channels);
- static rtc::Optional<NetEqDecoder> NetEqDecoderFromCodecId(
+ static absl::optional<NetEqDecoder> NetEqDecoderFromCodecId(
CodecId codec_id,
size_t num_channels);
diff --git a/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.cc b/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.cc
index 55e5309..c4832a3 100644
--- a/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.cc
+++ b/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.cc
@@ -123,7 +123,7 @@
controller->MakeDecision(&config);
// Update ANA stats.
- auto increment_opt = [](rtc::Optional<uint32_t>& a) {
+ auto increment_opt = [](absl::optional<uint32_t>& a) {
a = a.value_or(0) + 1;
};
if (prev_config_) {
diff --git a/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h b/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h
index 14000fe..e208ed2 100644
--- a/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h
+++ b/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h
@@ -75,7 +75,7 @@
Controller::NetworkMetrics last_metrics_;
- rtc::Optional<AudioEncoderRuntimeConfig> prev_config_;
+ absl::optional<AudioEncoderRuntimeConfig> prev_config_;
ANAStats stats_;
diff --git a/modules/audio_coding/audio_network_adaptor/bitrate_controller.h b/modules/audio_coding/audio_network_adaptor/bitrate_controller.h
index 601f794..282f599 100644
--- a/modules/audio_coding/audio_network_adaptor/bitrate_controller.h
+++ b/modules/audio_coding/audio_network_adaptor/bitrate_controller.h
@@ -43,8 +43,8 @@
const Config config_;
int bitrate_bps_;
int frame_length_ms_;
- rtc::Optional<int> target_audio_bitrate_bps_;
- rtc::Optional<size_t> overhead_bytes_per_packet_;
+ absl::optional<int> target_audio_bitrate_bps_;
+ absl::optional<size_t> overhead_bytes_per_packet_;
RTC_DISALLOW_COPY_AND_ASSIGN(BitrateController);
};
diff --git a/modules/audio_coding/audio_network_adaptor/bitrate_controller_unittest.cc b/modules/audio_coding/audio_network_adaptor/bitrate_controller_unittest.cc
index 9864511..98abede 100644
--- a/modules/audio_coding/audio_network_adaptor/bitrate_controller_unittest.cc
+++ b/modules/audio_coding/audio_network_adaptor/bitrate_controller_unittest.cc
@@ -20,8 +20,8 @@
void UpdateNetworkMetrics(
BitrateController* controller,
- const rtc::Optional<int>& target_audio_bitrate_bps,
- const rtc::Optional<size_t>& overhead_bytes_per_packet) {
+ const absl::optional<int>& target_audio_bitrate_bps,
+ const absl::optional<size_t>& overhead_bytes_per_packet) {
// UpdateNetworkMetrics can accept multiple network metric updates at once.
// However, currently, the most used case is to update one metric at a time.
// To reflect this fact, we separate the calls.
@@ -38,7 +38,7 @@
}
void CheckDecision(BitrateController* controller,
- const rtc::Optional<int>& frame_length_ms,
+ const absl::optional<int>& frame_length_ms,
int expected_bitrate_bps) {
AudioEncoderRuntimeConfig config;
config.frame_length_ms = frame_length_ms;
@@ -58,7 +58,7 @@
constexpr size_t kOverheadBytesPerPacket = 64;
BitrateController controller(BitrateController::Config(
kInitialBitrateBps, kInitialFrameLengthMs, 0, 0));
- UpdateNetworkMetrics(&controller, rtc::nullopt, kOverheadBytesPerPacket);
+ UpdateNetworkMetrics(&controller, absl::nullopt, kOverheadBytesPerPacket);
CheckDecision(&controller, kInitialFrameLengthMs * 2, kInitialBitrateBps);
}
@@ -68,7 +68,7 @@
constexpr int kTargetBitrateBps = 48000;
BitrateController controller(BitrateController::Config(
kInitialBitrateBps, kInitialFrameLengthMs, 0, 0));
- UpdateNetworkMetrics(&controller, kTargetBitrateBps, rtc::nullopt);
+ UpdateNetworkMetrics(&controller, kTargetBitrateBps, absl::nullopt);
CheckDecision(&controller, kInitialFrameLengthMs * 2, kInitialBitrateBps);
}
@@ -126,7 +126,7 @@
kTargetBitrateBps -
kOverheadBytesPerPacket * 8 * 1000 / kInitialFrameLengthMs;
UpdateNetworkMetrics(&controller, kTargetBitrateBps, kOverheadBytesPerPacket);
- CheckDecision(&controller, rtc::nullopt, kBitrateBps);
+ CheckDecision(&controller, absl::nullopt, kBitrateBps);
}
TEST(AnaBitrateControllerTest, IncreaseBitrateOnFrameLengthIncreased) {
@@ -142,7 +142,7 @@
kTargetBitrateBps -
kOverheadBytesPerPacket * 8 * 1000 / kInitialFrameLengthMs;
UpdateNetworkMetrics(&controller, kTargetBitrateBps, kOverheadBytesPerPacket);
- CheckDecision(&controller, rtc::nullopt, kBitrateBps);
+ CheckDecision(&controller, absl::nullopt, kBitrateBps);
constexpr int kFrameLengthMs = 60;
constexpr size_t kPacketOverheadRateDiff =
@@ -166,7 +166,7 @@
kTargetBitrateBps -
kOverheadBytesPerPacket * 8 * 1000 / kInitialFrameLengthMs;
UpdateNetworkMetrics(&controller, kTargetBitrateBps, kOverheadBytesPerPacket);
- CheckDecision(&controller, rtc::nullopt, kBitrateBps);
+ CheckDecision(&controller, absl::nullopt, kBitrateBps);
constexpr int kFrameLengthMs = 20;
constexpr size_t kPacketOverheadRateDiff =
diff --git a/modules/audio_coding/audio_network_adaptor/channel_controller.h b/modules/audio_coding/audio_network_adaptor/channel_controller.h
index f53ddd6..23cbef6 100644
--- a/modules/audio_coding/audio_network_adaptor/channel_controller.h
+++ b/modules/audio_coding/audio_network_adaptor/channel_controller.h
@@ -44,7 +44,7 @@
private:
const Config config_;
size_t channels_to_encode_;
- rtc::Optional<int> uplink_bandwidth_bps_;
+ absl::optional<int> uplink_bandwidth_bps_;
RTC_DISALLOW_COPY_AND_ASSIGN(ChannelController);
};
diff --git a/modules/audio_coding/audio_network_adaptor/channel_controller_unittest.cc b/modules/audio_coding/audio_network_adaptor/channel_controller_unittest.cc
index 64e5dae..bfa6f01 100644
--- a/modules/audio_coding/audio_network_adaptor/channel_controller_unittest.cc
+++ b/modules/audio_coding/audio_network_adaptor/channel_controller_unittest.cc
@@ -32,7 +32,7 @@
}
void CheckDecision(ChannelController* controller,
- const rtc::Optional<int>& uplink_bandwidth_bps,
+ const absl::optional<int>& uplink_bandwidth_bps,
size_t expected_num_channels) {
if (uplink_bandwidth_bps) {
Controller::NetworkMetrics network_metrics;
@@ -49,7 +49,7 @@
TEST(ChannelControllerTest, OutputInitValueWhenUplinkBandwidthUnknown) {
constexpr int kInitChannels = 2;
auto controller = CreateChannelController(kInitChannels);
- CheckDecision(controller.get(), rtc::nullopt, kInitChannels);
+ CheckDecision(controller.get(), absl::nullopt, kInitChannels);
}
TEST(ChannelControllerTest, SwitchTo2ChannelsOnHighUplinkBandwidth) {
diff --git a/modules/audio_coding/audio_network_adaptor/controller.h b/modules/audio_coding/audio_network_adaptor/controller.h
index af2f569..19d8599 100644
--- a/modules/audio_coding/audio_network_adaptor/controller.h
+++ b/modules/audio_coding/audio_network_adaptor/controller.h
@@ -11,7 +11,7 @@
#ifndef MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_CONTROLLER_H_
#define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_CONTROLLER_H_
-#include "api/optional.h"
+#include "absl/types/optional.h"
#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
namespace webrtc {
@@ -21,12 +21,12 @@
struct NetworkMetrics {
NetworkMetrics();
~NetworkMetrics();
- rtc::Optional<int> uplink_bandwidth_bps;
- rtc::Optional<float> uplink_packet_loss_fraction;
- rtc::Optional<float> uplink_recoverable_packet_loss_fraction;
- rtc::Optional<int> target_audio_bitrate_bps;
- rtc::Optional<int> rtt_ms;
- rtc::Optional<size_t> overhead_bytes_per_packet;
+ absl::optional<int> uplink_bandwidth_bps;
+ absl::optional<float> uplink_packet_loss_fraction;
+ absl::optional<float> uplink_recoverable_packet_loss_fraction;
+ absl::optional<int> target_audio_bitrate_bps;
+ absl::optional<int> rtt_ms;
+ absl::optional<size_t> overhead_bytes_per_packet;
};
virtual ~Controller() = default;
diff --git a/modules/audio_coding/audio_network_adaptor/controller_manager.cc b/modules/audio_coding/audio_network_adaptor/controller_manager.cc
index 313aa62..80255ea 100644
--- a/modules/audio_coding/audio_network_adaptor/controller_manager.cc
+++ b/modules/audio_coding/audio_network_adaptor/controller_manager.cc
@@ -327,7 +327,7 @@
const std::map<const Controller*, std::pair<int, float>>& scoring_points)
: config_(config),
controllers_(std::move(controllers)),
- last_reordering_time_ms_(rtc::nullopt),
+ last_reordering_time_ms_(absl::nullopt),
last_scoring_point_(0, 0.0) {
for (auto& controller : controllers_)
default_sorted_controllers_.push_back(controller.get());
diff --git a/modules/audio_coding/audio_network_adaptor/controller_manager.h b/modules/audio_coding/audio_network_adaptor/controller_manager.h
index 5c63f2f..1ff9bbf 100644
--- a/modules/audio_coding/audio_network_adaptor/controller_manager.h
+++ b/modules/audio_coding/audio_network_adaptor/controller_manager.h
@@ -104,7 +104,7 @@
std::vector<std::unique_ptr<Controller>> controllers_;
- rtc::Optional<int64_t> last_reordering_time_ms_;
+ absl::optional<int64_t> last_reordering_time_ms_;
ScoringPoint last_scoring_point_;
std::vector<Controller*> default_sorted_controllers_;
diff --git a/modules/audio_coding/audio_network_adaptor/controller_manager_unittest.cc b/modules/audio_coding/audio_network_adaptor/controller_manager_unittest.cc
index 4cab339..061e4aa 100644
--- a/modules/audio_coding/audio_network_adaptor/controller_manager_unittest.cc
+++ b/modules/audio_coding/audio_network_adaptor/controller_manager_unittest.cc
@@ -87,8 +87,8 @@
// exists in the vector.
void CheckControllersOrder(
ControllerManagerStates* states,
- const rtc::Optional<int>& uplink_bandwidth_bps,
- const rtc::Optional<float>& uplink_packet_loss_fraction,
+ const absl::optional<int>& uplink_bandwidth_bps,
+ const absl::optional<float>& uplink_packet_loss_fraction,
const std::vector<int>& expected_order) {
RTC_DCHECK_EQ(kNumControllers, expected_order.size());
Controller::NetworkMetrics metrics;
@@ -124,8 +124,7 @@
auto states = CreateControllerManager();
// |network_metrics| are empty, and the controllers are supposed to follow the
// default order.
- CheckControllersOrder(&states, rtc::nullopt, rtc::nullopt,
- {0, 1, 2, 3});
+ CheckControllersOrder(&states, absl::nullopt, absl::nullopt, {0, 1, 2, 3});
}
TEST(ControllerManagerTest, ControllersWithoutCharPointAtEndAndInDefaultOrder) {
diff --git a/modules/audio_coding/audio_network_adaptor/dtx_controller.h b/modules/audio_coding/audio_network_adaptor/dtx_controller.h
index 8a2427e..fb40db2 100644
--- a/modules/audio_coding/audio_network_adaptor/dtx_controller.h
+++ b/modules/audio_coding/audio_network_adaptor/dtx_controller.h
@@ -40,7 +40,7 @@
private:
const Config config_;
bool dtx_enabled_;
- rtc::Optional<int> uplink_bandwidth_bps_;
+ absl::optional<int> uplink_bandwidth_bps_;
RTC_DISALLOW_COPY_AND_ASSIGN(DtxController);
};
diff --git a/modules/audio_coding/audio_network_adaptor/dtx_controller_unittest.cc b/modules/audio_coding/audio_network_adaptor/dtx_controller_unittest.cc
index e38e65d..67bf9e5 100644
--- a/modules/audio_coding/audio_network_adaptor/dtx_controller_unittest.cc
+++ b/modules/audio_coding/audio_network_adaptor/dtx_controller_unittest.cc
@@ -30,7 +30,7 @@
}
void CheckDecision(DtxController* controller,
- const rtc::Optional<int>& uplink_bandwidth_bps,
+ const absl::optional<int>& uplink_bandwidth_bps,
bool expected_dtx_enabled) {
if (uplink_bandwidth_bps) {
Controller::NetworkMetrics network_metrics;
@@ -47,7 +47,7 @@
TEST(DtxControllerTest, OutputInitValueWhenUplinkBandwidthUnknown) {
constexpr bool kInitialDtxEnabled = true;
auto controller = CreateController(kInitialDtxEnabled);
- CheckDecision(controller.get(), rtc::nullopt, kInitialDtxEnabled);
+ CheckDecision(controller.get(), absl::nullopt, kInitialDtxEnabled);
}
TEST(DtxControllerTest, TurnOnDtxForLowUplinkBandwidth) {
diff --git a/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.cc b/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.cc
index 62f356d..7409721 100644
--- a/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.cc
+++ b/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.cc
@@ -23,7 +23,7 @@
public:
void AddSample(float sample) override { last_sample_ = sample; }
- rtc::Optional<float> GetAverage() override { return last_sample_; }
+ absl::optional<float> GetAverage() override { return last_sample_; }
bool SetTimeConstantMs(int time_constant_ms) override {
RTC_NOTREACHED();
@@ -31,7 +31,7 @@
}
private:
- rtc::Optional<float> last_sample_;
+ absl::optional<float> last_sample_;
};
}
@@ -89,7 +89,7 @@
}
bool FecControllerPlrBased::FecEnablingDecision(
- const rtc::Optional<float>& packet_loss) const {
+ const absl::optional<float>& packet_loss) const {
if (!uplink_bandwidth_bps_ || !packet_loss) {
return false;
} else {
@@ -100,7 +100,7 @@
}
bool FecControllerPlrBased::FecDisablingDecision(
- const rtc::Optional<float>& packet_loss) const {
+ const absl::optional<float>& packet_loss) const {
if (!uplink_bandwidth_bps_ || !packet_loss) {
return false;
} else {
diff --git a/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.h b/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.h
index c273537..b66883e 100644
--- a/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.h
+++ b/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.h
@@ -56,12 +56,12 @@
void MakeDecision(AudioEncoderRuntimeConfig* config) override;
private:
- bool FecEnablingDecision(const rtc::Optional<float>& packet_loss) const;
- bool FecDisablingDecision(const rtc::Optional<float>& packet_loss) const;
+ bool FecEnablingDecision(const absl::optional<float>& packet_loss) const;
+ bool FecDisablingDecision(const absl::optional<float>& packet_loss) const;
const Config config_;
bool fec_enabled_;
- rtc::Optional<int> uplink_bandwidth_bps_;
+ absl::optional<int> uplink_bandwidth_bps_;
const std::unique_ptr<SmoothingFilter> packet_loss_smoother_;
RTC_DISALLOW_COPY_AND_ASSIGN(FecControllerPlrBased);
diff --git a/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based_unittest.cc b/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based_unittest.cc
index 8636aa9..de66717 100644
--- a/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based_unittest.cc
+++ b/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based_unittest.cc
@@ -78,8 +78,8 @@
}
void UpdateNetworkMetrics(FecControllerPlrBasedTestStates* states,
- const rtc::Optional<int>& uplink_bandwidth_bps,
- const rtc::Optional<float>& uplink_packet_loss) {
+ const absl::optional<int>& uplink_bandwidth_bps,
+ const absl::optional<float>& uplink_packet_loss) {
// UpdateNetworkMetrics can accept multiple network metric updates at once.
// However, currently, the most used case is to update one metric at a time.
// To reflect this fact, we separate the calls.
@@ -131,7 +131,7 @@
kEnablingPacketLossAtLowBw - kEpsilon, kEnablingPacketLossAtLowBw,
kEnablingPacketLossAtLowBw + kEpsilon}) {
auto states = CreateFecControllerPlrBased(initial_fec_enabled);
- UpdateNetworkMetrics(&states, rtc::nullopt, packet_loss);
+ UpdateNetworkMetrics(&states, absl::nullopt, packet_loss);
CheckDecision(&states, initial_fec_enabled, packet_loss);
}
}
@@ -146,7 +146,7 @@
kDisablingBandwidthLow + 1, kEnablingBandwidthLow - 1,
kEnablingBandwidthLow, kEnablingBandwidthLow + 1}) {
auto states = CreateFecControllerPlrBased(initial_fec_enabled);
- UpdateNetworkMetrics(&states, bandwidth, rtc::nullopt);
+ UpdateNetworkMetrics(&states, bandwidth, absl::nullopt);
CheckDecision(&states, initial_fec_enabled, 0.0);
}
}
diff --git a/modules/audio_coding/audio_network_adaptor/fec_controller_rplr_based.h b/modules/audio_coding/audio_network_adaptor/fec_controller_rplr_based.h
index ade55ae..9a3c37c 100644
--- a/modules/audio_coding/audio_network_adaptor/fec_controller_rplr_based.h
+++ b/modules/audio_coding/audio_network_adaptor/fec_controller_rplr_based.h
@@ -55,8 +55,8 @@
const Config config_;
bool fec_enabled_;
- rtc::Optional<int> uplink_bandwidth_bps_;
- rtc::Optional<float> uplink_recoverable_packet_loss_;
+ absl::optional<int> uplink_bandwidth_bps_;
+ absl::optional<float> uplink_recoverable_packet_loss_;
RTC_DISALLOW_COPY_AND_ASSIGN(FecControllerRplrBased);
};
diff --git a/modules/audio_coding/audio_network_adaptor/fec_controller_rplr_based_unittest.cc b/modules/audio_coding/audio_network_adaptor/fec_controller_rplr_based_unittest.cc
index 0fc003b..538a3e0 100644
--- a/modules/audio_coding/audio_network_adaptor/fec_controller_rplr_based_unittest.cc
+++ b/modules/audio_coding/audio_network_adaptor/fec_controller_rplr_based_unittest.cc
@@ -44,14 +44,14 @@
constexpr float kEpsilon = 1e-5f;
-rtc::Optional<float> GetRandomProbabilityOrUnknown() {
+absl::optional<float> GetRandomProbabilityOrUnknown() {
std::random_device rd;
std::mt19937 generator(rd());
std::uniform_real_distribution<> distribution(0, 1);
return (distribution(generator) < 0.2)
- ? rtc::nullopt
- : rtc::Optional<float>(distribution(generator));
+ ? absl::nullopt
+ : absl::optional<float>(distribution(generator));
}
std::unique_ptr<FecControllerRplrBased> CreateFecControllerRplrBased(
@@ -70,9 +70,9 @@
void UpdateNetworkMetrics(
FecControllerRplrBased* controller,
- const rtc::Optional<int>& uplink_bandwidth_bps,
- const rtc::Optional<float>& uplink_packet_loss,
- const rtc::Optional<float>& uplink_recoveralbe_packet_loss) {
+ const absl::optional<int>& uplink_bandwidth_bps,
+ const absl::optional<float>& uplink_packet_loss,
+ const absl::optional<float>& uplink_recoveralbe_packet_loss) {
// UpdateNetworkMetrics can accept multiple network metric updates at once.
// However, currently, the most used case is to update one metric at a time.
// To reflect this fact, we separate the calls.
@@ -96,8 +96,8 @@
void UpdateNetworkMetrics(
FecControllerRplrBased* controller,
- const rtc::Optional<int>& uplink_bandwidth_bps,
- const rtc::Optional<float>& uplink_recoveralbe_packet_loss) {
+ const absl::optional<int>& uplink_bandwidth_bps,
+ const absl::optional<float>& uplink_recoveralbe_packet_loss) {
// FecControllerRplrBased doesn't currently use the PLR (general packet-loss
// rate) at all. (This might be changed in the future.) The unit-tests will
// use a random value (including unknown), to show this does not interfere.
@@ -148,7 +148,7 @@
kEnablingRecoverablePacketLossAtHighBw,
kEnablingRecoverablePacketLossAtHighBw + kEpsilon}) {
auto controller = CreateFecControllerRplrBased(initial_fec_enabled);
- UpdateNetworkMetrics(controller.get(), rtc::nullopt,
+ UpdateNetworkMetrics(controller.get(), absl::nullopt,
recoverable_packet_loss);
CheckDecision(controller.get(), initial_fec_enabled,
recoverable_packet_loss);
@@ -165,7 +165,7 @@
kDisablingBandwidthLow + 1, kEnablingBandwidthLow - 1,
kEnablingBandwidthLow, kEnablingBandwidthLow + 1}) {
auto controller = CreateFecControllerRplrBased(initial_fec_enabled);
- UpdateNetworkMetrics(controller.get(), bandwidth, rtc::nullopt);
+ UpdateNetworkMetrics(controller.get(), bandwidth, absl::nullopt);
CheckDecision(controller.get(), initial_fec_enabled, 0.0);
}
}
diff --git a/modules/audio_coding/audio_network_adaptor/frame_length_controller.h b/modules/audio_coding/audio_network_adaptor/frame_length_controller.h
index c254b3d..f084fd0 100644
--- a/modules/audio_coding/audio_network_adaptor/frame_length_controller.h
+++ b/modules/audio_coding/audio_network_adaptor/frame_length_controller.h
@@ -73,11 +73,11 @@
std::vector<int>::const_iterator frame_length_ms_;
- rtc::Optional<int> uplink_bandwidth_bps_;
+ absl::optional<int> uplink_bandwidth_bps_;
- rtc::Optional<float> uplink_packet_loss_fraction_;
+ absl::optional<float> uplink_packet_loss_fraction_;
- rtc::Optional<size_t> overhead_bytes_per_packet_;
+ absl::optional<size_t> overhead_bytes_per_packet_;
// True if the previous frame length decision was an increase, otherwise
// false.
diff --git a/modules/audio_coding/audio_network_adaptor/frame_length_controller_unittest.cc b/modules/audio_coding/audio_network_adaptor/frame_length_controller_unittest.cc
index 1f98447..f97ad4f 100644
--- a/modules/audio_coding/audio_network_adaptor/frame_length_controller_unittest.cc
+++ b/modules/audio_coding/audio_network_adaptor/frame_length_controller_unittest.cc
@@ -78,9 +78,9 @@
void UpdateNetworkMetrics(
FrameLengthController* controller,
- const rtc::Optional<int>& uplink_bandwidth_bps,
- const rtc::Optional<float>& uplink_packet_loss_fraction,
- const rtc::Optional<size_t>& overhead_bytes_per_packet) {
+ const absl::optional<int>& uplink_bandwidth_bps,
+ const absl::optional<float>& uplink_packet_loss_fraction,
+ const absl::optional<size_t>& overhead_bytes_per_packet) {
// UpdateNetworkMetrics can accept multiple network metric updates at once.
// However, currently, the most used case is to update one metric at a time.
// To reflect this fact, we separate the calls.
@@ -114,14 +114,14 @@
auto controller =
CreateController(CreateChangeCriteriaFor20msAnd60ms(), {20, 60}, 60);
UpdateNetworkMetrics(controller.get(), kFl60msTo20msBandwidthBps,
- rtc::nullopt, kOverheadBytesPerPacket);
+ absl::nullopt, kOverheadBytesPerPacket);
CheckDecision(controller.get(), 20);
}
TEST(FrameLengthControllerTest, DecreaseTo20MsOnHighUplinkPacketLossFraction) {
auto controller =
CreateController(CreateChangeCriteriaFor20msAnd60ms(), {20, 60}, 60);
- UpdateNetworkMetrics(controller.get(), rtc::nullopt,
+ UpdateNetworkMetrics(controller.get(), absl::nullopt,
kFlDecreasingPacketLossFraction,
kOverheadBytesPerPacket);
CheckDecision(controller.get(), 20);
@@ -252,16 +252,12 @@
auto controller = CreateController(CreateChangeCriteriaFor20ms60msAnd120ms(),
{20, 60, 120}, 120);
// It takes two steps for frame length to go from 120ms to 20ms.
- UpdateNetworkMetrics(controller.get(),
- kFl60msTo20msBandwidthBps,
- rtc::nullopt,
- kOverheadBytesPerPacket);
+ UpdateNetworkMetrics(controller.get(), kFl60msTo20msBandwidthBps,
+ absl::nullopt, kOverheadBytesPerPacket);
CheckDecision(controller.get(), 60);
- UpdateNetworkMetrics(controller.get(),
- kFl60msTo20msBandwidthBps,
- rtc::nullopt,
- kOverheadBytesPerPacket);
+ UpdateNetworkMetrics(controller.get(), kFl60msTo20msBandwidthBps,
+ absl::nullopt, kOverheadBytesPerPacket);
CheckDecision(controller.get(), 20);
}
@@ -269,12 +265,12 @@
auto controller = CreateController(CreateChangeCriteriaFor20ms60msAnd120ms(),
{20, 60, 120}, 120);
// It takes two steps for frame length to go from 120ms to 20ms.
- UpdateNetworkMetrics(controller.get(), rtc::nullopt,
+ UpdateNetworkMetrics(controller.get(), absl::nullopt,
kFlDecreasingPacketLossFraction,
kOverheadBytesPerPacket);
CheckDecision(controller.get(), 60);
- UpdateNetworkMetrics(controller.get(), rtc::nullopt,
+ UpdateNetworkMetrics(controller.get(), absl::nullopt,
kFlDecreasingPacketLossFraction,
kOverheadBytesPerPacket);
CheckDecision(controller.get(), 20);
diff --git a/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h b/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h
index 7687446..c11279e 100644
--- a/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h
+++ b/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h
@@ -11,8 +11,8 @@
#ifndef MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_INCLUDE_AUDIO_NETWORK_ADAPTOR_H_
#define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_INCLUDE_AUDIO_NETWORK_ADAPTOR_H_
+#include "absl/types/optional.h"
#include "api/audio_codecs/audio_encoder.h"
-#include "api/optional.h"
#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h"
namespace webrtc {
diff --git a/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h b/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h
index 874fc97..257a79a 100644
--- a/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h
+++ b/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h
@@ -11,7 +11,7 @@
#ifndef MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_INCLUDE_AUDIO_NETWORK_ADAPTOR_CONFIG_H_
#define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_INCLUDE_AUDIO_NETWORK_ADAPTOR_CONFIG_H_
-#include "api/optional.h"
+#include "absl/types/optional.h"
namespace webrtc {
@@ -21,18 +21,18 @@
~AudioEncoderRuntimeConfig();
AudioEncoderRuntimeConfig& operator=(const AudioEncoderRuntimeConfig& other);
bool operator==(const AudioEncoderRuntimeConfig& other) const;
- rtc::Optional<int> bitrate_bps;
- rtc::Optional<int> frame_length_ms;
+ absl::optional<int> bitrate_bps;
+ absl::optional<int> frame_length_ms;
// Note: This is what we tell the encoder. It doesn't have to reflect
// the actual NetworkMetrics; it's subject to our decision.
- rtc::Optional<float> uplink_packet_loss_fraction;
- rtc::Optional<bool> enable_fec;
- rtc::Optional<bool> enable_dtx;
+ absl::optional<float> uplink_packet_loss_fraction;
+ absl::optional<bool> enable_fec;
+ absl::optional<bool> enable_dtx;
// Some encoders can encode fewer channels than the actual input to make
// better use of the bandwidth. |num_channels| sets the number of channels
// to encode.
- rtc::Optional<size_t> num_channels;
+ absl::optional<size_t> num_channels;
// This is true if the last frame length change was an increase, and otherwise
// false.
diff --git a/modules/audio_coding/codecs/audio_format_conversion.cc b/modules/audio_coding/codecs/audio_format_conversion.cc
index a99a28c..e38aa33 100644
--- a/modules/audio_coding/codecs/audio_format_conversion.cc
+++ b/modules/audio_coding/codecs/audio_format_conversion.cc
@@ -12,8 +12,8 @@
#include <string.h>
+#include "absl/types/optional.h"
#include "api/array_view.h"
-#include "api/optional.h"
#include "rtc_base/checks.h"
#include "rtc_base/numerics/safe_conversions.h"
#include "rtc_base/sanitizer.h"
diff --git a/modules/audio_coding/codecs/builtin_audio_decoder_factory_unittest.cc b/modules/audio_coding/codecs/builtin_audio_decoder_factory_unittest.cc
index 158a58b..9b36dfd 100644
--- a/modules/audio_coding/codecs/builtin_audio_decoder_factory_unittest.cc
+++ b/modules/audio_coding/codecs/builtin_audio_decoder_factory_unittest.cc
@@ -20,7 +20,7 @@
CreateBuiltinAudioDecoderFactory();
ASSERT_TRUE(adf);
EXPECT_FALSE(
- adf->MakeAudioDecoder(SdpAudioFormat("rey", 8000, 1), rtc::nullopt));
+ adf->MakeAudioDecoder(SdpAudioFormat("rey", 8000, 1), absl::nullopt));
}
TEST(AudioDecoderFactoryTest, CreatePcmu) {
@@ -29,15 +29,15 @@
ASSERT_TRUE(adf);
// PCMu supports 8 kHz, and any number of channels.
EXPECT_FALSE(
- adf->MakeAudioDecoder(SdpAudioFormat("pcmu", 8000, 0), rtc::nullopt));
+ adf->MakeAudioDecoder(SdpAudioFormat("pcmu", 8000, 0), absl::nullopt));
EXPECT_TRUE(
- adf->MakeAudioDecoder(SdpAudioFormat("pcmu", 8000, 1), rtc::nullopt));
+ adf->MakeAudioDecoder(SdpAudioFormat("pcmu", 8000, 1), absl::nullopt));
EXPECT_TRUE(
- adf->MakeAudioDecoder(SdpAudioFormat("pcmu", 8000, 2), rtc::nullopt));
+ adf->MakeAudioDecoder(SdpAudioFormat("pcmu", 8000, 2), absl::nullopt));
EXPECT_TRUE(
- adf->MakeAudioDecoder(SdpAudioFormat("pcmu", 8000, 3), rtc::nullopt));
+ adf->MakeAudioDecoder(SdpAudioFormat("pcmu", 8000, 3), absl::nullopt));
EXPECT_FALSE(
- adf->MakeAudioDecoder(SdpAudioFormat("pcmu", 16000, 1), rtc::nullopt));
+ adf->MakeAudioDecoder(SdpAudioFormat("pcmu", 16000, 1), absl::nullopt));
}
TEST(AudioDecoderFactoryTest, CreatePcma) {
@@ -46,15 +46,15 @@
ASSERT_TRUE(adf);
// PCMa supports 8 kHz, and any number of channels.
EXPECT_FALSE(
- adf->MakeAudioDecoder(SdpAudioFormat("pcma", 8000, 0), rtc::nullopt));
+ adf->MakeAudioDecoder(SdpAudioFormat("pcma", 8000, 0), absl::nullopt));
EXPECT_TRUE(
- adf->MakeAudioDecoder(SdpAudioFormat("pcma", 8000, 1), rtc::nullopt));
+ adf->MakeAudioDecoder(SdpAudioFormat("pcma", 8000, 1), absl::nullopt));
EXPECT_TRUE(
- adf->MakeAudioDecoder(SdpAudioFormat("pcma", 8000, 2), rtc::nullopt));
+ adf->MakeAudioDecoder(SdpAudioFormat("pcma", 8000, 2), absl::nullopt));
EXPECT_TRUE(
- adf->MakeAudioDecoder(SdpAudioFormat("pcma", 8000, 3), rtc::nullopt));
+ adf->MakeAudioDecoder(SdpAudioFormat("pcma", 8000, 3), absl::nullopt));
EXPECT_FALSE(
- adf->MakeAudioDecoder(SdpAudioFormat("pcma", 16000, 1), rtc::nullopt));
+ adf->MakeAudioDecoder(SdpAudioFormat("pcma", 16000, 1), absl::nullopt));
}
TEST(AudioDecoderFactoryTest, CreateIlbc) {
@@ -63,15 +63,15 @@
ASSERT_TRUE(adf);
// iLBC supports 8 kHz, 1 channel.
EXPECT_FALSE(
- adf->MakeAudioDecoder(SdpAudioFormat("ilbc", 8000, 0), rtc::nullopt));
+ adf->MakeAudioDecoder(SdpAudioFormat("ilbc", 8000, 0), absl::nullopt));
#ifdef WEBRTC_CODEC_ILBC
EXPECT_TRUE(
- adf->MakeAudioDecoder(SdpAudioFormat("ilbc", 8000, 1), rtc::nullopt));
+ adf->MakeAudioDecoder(SdpAudioFormat("ilbc", 8000, 1), absl::nullopt));
#endif
EXPECT_FALSE(
- adf->MakeAudioDecoder(SdpAudioFormat("ilbc", 8000, 2), rtc::nullopt));
+ adf->MakeAudioDecoder(SdpAudioFormat("ilbc", 8000, 2), absl::nullopt));
EXPECT_FALSE(
- adf->MakeAudioDecoder(SdpAudioFormat("ilbc", 16000, 1), rtc::nullopt));
+ adf->MakeAudioDecoder(SdpAudioFormat("ilbc", 16000, 1), absl::nullopt));
}
TEST(AudioDecoderFactoryTest, CreateIsac) {
@@ -81,21 +81,21 @@
// iSAC supports 16 kHz, 1 channel. The float implementation additionally
// supports 32 kHz, 1 channel.
EXPECT_FALSE(
- adf->MakeAudioDecoder(SdpAudioFormat("isac", 16000, 0), rtc::nullopt));
+ adf->MakeAudioDecoder(SdpAudioFormat("isac", 16000, 0), absl::nullopt));
EXPECT_TRUE(
- adf->MakeAudioDecoder(SdpAudioFormat("isac", 16000, 1), rtc::nullopt));
+ adf->MakeAudioDecoder(SdpAudioFormat("isac", 16000, 1), absl::nullopt));
EXPECT_FALSE(
- adf->MakeAudioDecoder(SdpAudioFormat("isac", 16000, 2), rtc::nullopt));
+ adf->MakeAudioDecoder(SdpAudioFormat("isac", 16000, 2), absl::nullopt));
EXPECT_FALSE(
- adf->MakeAudioDecoder(SdpAudioFormat("isac", 8000, 1), rtc::nullopt));
+ adf->MakeAudioDecoder(SdpAudioFormat("isac", 8000, 1), absl::nullopt));
EXPECT_FALSE(
- adf->MakeAudioDecoder(SdpAudioFormat("isac", 48000, 1), rtc::nullopt));
+ adf->MakeAudioDecoder(SdpAudioFormat("isac", 48000, 1), absl::nullopt));
#ifdef WEBRTC_ARCH_ARM
EXPECT_FALSE(
- adf->MakeAudioDecoder(SdpAudioFormat("isac", 32000, 1), rtc::nullopt));
+ adf->MakeAudioDecoder(SdpAudioFormat("isac", 32000, 1), absl::nullopt));
#else
EXPECT_TRUE(
- adf->MakeAudioDecoder(SdpAudioFormat("isac", 32000, 1), rtc::nullopt));
+ adf->MakeAudioDecoder(SdpAudioFormat("isac", 32000, 1), absl::nullopt));
#endif
}
@@ -108,10 +108,10 @@
const int num_channels[] = {1, 2, 3, 4711};
for (int clockrate : clockrates) {
EXPECT_FALSE(adf->MakeAudioDecoder(SdpAudioFormat("l16", clockrate, 0),
- rtc::nullopt));
+ absl::nullopt));
for (int channels : num_channels) {
EXPECT_TRUE(adf->MakeAudioDecoder(
- SdpAudioFormat("l16", clockrate, channels), rtc::nullopt));
+ SdpAudioFormat("l16", clockrate, channels), absl::nullopt));
}
}
}
@@ -122,21 +122,21 @@
ASSERT_TRUE(adf);
// g722 supports 8 kHz, 1-2 channels.
EXPECT_FALSE(
- adf->MakeAudioDecoder(SdpAudioFormat("g722", 8000, 0), rtc::nullopt));
+ adf->MakeAudioDecoder(SdpAudioFormat("g722", 8000, 0), absl::nullopt));
EXPECT_TRUE(
- adf->MakeAudioDecoder(SdpAudioFormat("g722", 8000, 1), rtc::nullopt));
+ adf->MakeAudioDecoder(SdpAudioFormat("g722", 8000, 1), absl::nullopt));
EXPECT_TRUE(
- adf->MakeAudioDecoder(SdpAudioFormat("g722", 8000, 2), rtc::nullopt));
+ adf->MakeAudioDecoder(SdpAudioFormat("g722", 8000, 2), absl::nullopt));
EXPECT_FALSE(
- adf->MakeAudioDecoder(SdpAudioFormat("g722", 8000, 3), rtc::nullopt));
+ adf->MakeAudioDecoder(SdpAudioFormat("g722", 8000, 3), absl::nullopt));
EXPECT_FALSE(
- adf->MakeAudioDecoder(SdpAudioFormat("g722", 16000, 1), rtc::nullopt));
+ adf->MakeAudioDecoder(SdpAudioFormat("g722", 16000, 1), absl::nullopt));
EXPECT_FALSE(
- adf->MakeAudioDecoder(SdpAudioFormat("g722", 32000, 1), rtc::nullopt));
+ adf->MakeAudioDecoder(SdpAudioFormat("g722", 32000, 1), absl::nullopt));
// g722 actually uses a 16 kHz sample rate instead of the nominal 8 kHz.
std::unique_ptr<AudioDecoder> dec =
- adf->MakeAudioDecoder(SdpAudioFormat("g722", 8000, 1), rtc::nullopt);
+ adf->MakeAudioDecoder(SdpAudioFormat("g722", 8000, 1), absl::nullopt);
EXPECT_EQ(16000, dec->SampleRateHz());
}
@@ -158,7 +158,7 @@
EXPECT_EQ(good,
static_cast<bool>(adf->MakeAudioDecoder(
SdpAudioFormat("opus", hz, channels, std::move(params)),
- rtc::nullopt)));
+ absl::nullopt)));
}
}
}
diff --git a/modules/audio_coding/codecs/builtin_audio_encoder_factory_unittest.cc b/modules/audio_coding/codecs/builtin_audio_encoder_factory_unittest.cc
index d371149..64f0159 100644
--- a/modules/audio_coding/codecs/builtin_audio_encoder_factory_unittest.cc
+++ b/modules/audio_coding/codecs/builtin_audio_encoder_factory_unittest.cc
@@ -43,7 +43,7 @@
auto supported_encoders = factory->GetSupportedEncoders();
for (const auto& spec : supported_encoders) {
auto info = factory->QueryAudioEncoder(spec.format);
- auto encoder = factory->MakeAudioEncoder(127, spec.format, rtc::nullopt);
+ auto encoder = factory->MakeAudioEncoder(127, spec.format, absl::nullopt);
EXPECT_TRUE(encoder);
EXPECT_EQ(encoder->SampleRateHz(), info->sample_rate_hz);
EXPECT_EQ(encoder->NumChannels(), info->num_channels);
@@ -57,7 +57,7 @@
auto supported_encoders = factory->GetSupportedEncoders();
for (const auto& spec : supported_encoders) {
auto encoder =
- factory->MakeAudioEncoder(kTestPayloadType, spec.format, rtc::nullopt);
+ factory->MakeAudioEncoder(kTestPayloadType, spec.format, absl::nullopt);
EXPECT_TRUE(encoder);
encoder->Reset();
const int num_samples = rtc::checked_cast<int>(
diff --git a/modules/audio_coding/codecs/cng/audio_encoder_cng.cc b/modules/audio_coding/codecs/cng/audio_encoder_cng.cc
index dc4c21e..91c07a9 100644
--- a/modules/audio_coding/codecs/cng/audio_encoder_cng.cc
+++ b/modules/audio_coding/codecs/cng/audio_encoder_cng.cc
@@ -198,7 +198,7 @@
void AudioEncoderCng::OnReceivedUplinkBandwidth(
int target_audio_bitrate_bps,
- rtc::Optional<int64_t> bwe_period_ms) {
+ absl::optional<int64_t> bwe_period_ms) {
speech_encoder_->OnReceivedUplinkBandwidth(target_audio_bitrate_bps,
bwe_period_ms);
}
diff --git a/modules/audio_coding/codecs/cng/audio_encoder_cng.h b/modules/audio_coding/codecs/cng/audio_encoder_cng.h
index 4491289..e4c6507 100644
--- a/modules/audio_coding/codecs/cng/audio_encoder_cng.h
+++ b/modules/audio_coding/codecs/cng/audio_encoder_cng.h
@@ -69,7 +69,7 @@
float uplink_recoverable_packet_loss_fraction) override;
void OnReceivedUplinkBandwidth(
int target_audio_bitrate_bps,
- rtc::Optional<int64_t> bwe_period_ms) override;
+ absl::optional<int64_t> bwe_period_ms) override;
private:
EncodedInfo EncodePassive(size_t frames_to_encode,
diff --git a/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc b/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc
index f85abe2..c582b44 100644
--- a/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc
+++ b/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc
@@ -220,8 +220,8 @@
TEST_F(AudioEncoderCngTest, CheckTargetAudioBitratePropagation) {
CreateCng(MakeCngConfig());
EXPECT_CALL(*mock_encoder_,
- OnReceivedUplinkBandwidth(4711, rtc::Optional<int64_t>()));
- cng_->OnReceivedUplinkBandwidth(4711, rtc::nullopt);
+ OnReceivedUplinkBandwidth(4711, absl::optional<int64_t>()));
+ cng_->OnReceivedUplinkBandwidth(4711, absl::nullopt);
}
TEST_F(AudioEncoderCngTest, CheckPacketLossFractionPropagation) {
diff --git a/modules/audio_coding/codecs/isac/audio_decoder_isac_t.h b/modules/audio_coding/codecs/isac/audio_decoder_isac_t.h
index eda1cfa..f840ffa 100644
--- a/modules/audio_coding/codecs/isac/audio_decoder_isac_t.h
+++ b/modules/audio_coding/codecs/isac/audio_decoder_isac_t.h
@@ -13,8 +13,8 @@
#include <vector>
+#include "absl/types/optional.h"
#include "api/audio_codecs/audio_decoder.h"
-#include "api/optional.h"
#include "modules/audio_coding/codecs/isac/locked_bandwidth_info.h"
#include "rtc_base/constructormagic.h"
#include "rtc_base/scoped_ref_ptr.h"
diff --git a/modules/audio_coding/codecs/legacy_encoded_audio_frame.cc b/modules/audio_coding/codecs/legacy_encoded_audio_frame.cc
index 341336e..6d322a8 100644
--- a/modules/audio_coding/codecs/legacy_encoded_audio_frame.cc
+++ b/modules/audio_coding/codecs/legacy_encoded_audio_frame.cc
@@ -27,7 +27,7 @@
return (ret < 0) ? 0 : static_cast<size_t>(ret);
}
-rtc::Optional<AudioDecoder::EncodedAudioFrame::DecodeResult>
+absl::optional<AudioDecoder::EncodedAudioFrame::DecodeResult>
LegacyEncodedAudioFrame::Decode(rtc::ArrayView<int16_t> decoded) const {
AudioDecoder::SpeechType speech_type = AudioDecoder::kSpeech;
const int ret = decoder_->Decode(
@@ -35,7 +35,7 @@
decoded.size() * sizeof(int16_t), decoded.data(), &speech_type);
if (ret < 0)
- return rtc::nullopt;
+ return absl::nullopt;
return DecodeResult{static_cast<size_t>(ret), speech_type};
}
diff --git a/modules/audio_coding/codecs/legacy_encoded_audio_frame.h b/modules/audio_coding/codecs/legacy_encoded_audio_frame.h
index 275576e..05d4fe4 100644
--- a/modules/audio_coding/codecs/legacy_encoded_audio_frame.h
+++ b/modules/audio_coding/codecs/legacy_encoded_audio_frame.h
@@ -32,7 +32,7 @@
size_t Duration() const override;
- rtc::Optional<DecodeResult> Decode(
+ absl::optional<DecodeResult> Decode(
rtc::ArrayView<int16_t> decoded) const override;
// For testing:
diff --git a/modules/audio_coding/codecs/opus/audio_decoder_opus.cc b/modules/audio_coding/codecs/opus/audio_decoder_opus.cc
index c784a68..302b714 100644
--- a/modules/audio_coding/codecs/opus/audio_decoder_opus.cc
+++ b/modules/audio_coding/codecs/opus/audio_decoder_opus.cc
@@ -38,7 +38,7 @@
bool IsDtxPacket() const override { return payload_.size() <= 2; }
- rtc::Optional<DecodeResult> Decode(
+ absl::optional<DecodeResult> Decode(
rtc::ArrayView<int16_t> decoded) const override {
AudioDecoder::SpeechType speech_type = AudioDecoder::kSpeech;
int ret;
@@ -53,7 +53,7 @@
}
if (ret < 0)
- return rtc::nullopt;
+ return absl::nullopt;
return DecodeResult{static_cast<size_t>(ret), speech_type};
}
diff --git a/modules/audio_coding/codecs/opus/audio_encoder_opus.cc b/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
index caac4ae..fc6d544 100644
--- a/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
+++ b/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
@@ -105,18 +105,18 @@
}
}
-rtc::Optional<std::string> GetFormatParameter(const SdpAudioFormat& format,
- const std::string& param) {
+absl::optional<std::string> GetFormatParameter(const SdpAudioFormat& format,
+ const std::string& param) {
auto it = format.parameters.find(param);
if (it == format.parameters.end())
- return rtc::nullopt;
+ return absl::nullopt;
return it->second;
}
template <typename T>
-rtc::Optional<T> GetFormatParameter(const SdpAudioFormat& format,
- const std::string& param) {
+absl::optional<T> GetFormatParameter(const SdpAudioFormat& format,
+ const std::string& param) {
return rtc::StringToNumber<T>(GetFormatParameter(format, param).value_or(""));
}
@@ -139,7 +139,7 @@
// out how invalid it is and accurately log invalid values.
int CalculateBitrate(int max_playback_rate_hz,
size_t num_channels,
- rtc::Optional<std::string> bitrate_param) {
+ absl::optional<std::string> bitrate_param) {
const int default_bitrate =
CalculateDefaultBitrate(max_playback_rate_hz, num_channels);
@@ -242,7 +242,7 @@
return rtc::MakeUnique<AudioEncoderOpusImpl>(config, payload_type);
}
-rtc::Optional<AudioCodecInfo> AudioEncoderOpusImpl::QueryAudioEncoder(
+absl::optional<AudioCodecInfo> AudioEncoderOpusImpl::QueryAudioEncoder(
const SdpAudioFormat& format) {
if (STR_CASE_CMP(format.name.c_str(), GetPayloadName()) == 0 &&
format.clockrate_hz == 48000 && format.num_channels == 2) {
@@ -258,7 +258,7 @@
return info;
}
- return rtc::nullopt;
+ return absl::nullopt;
}
AudioEncoderOpusConfig AudioEncoderOpusImpl::CreateConfig(
@@ -274,11 +274,11 @@
return config;
}
-rtc::Optional<AudioEncoderOpusConfig> AudioEncoderOpusImpl::SdpToConfig(
+absl::optional<AudioEncoderOpusConfig> AudioEncoderOpusImpl::SdpToConfig(
const SdpAudioFormat& format) {
if (STR_CASE_CMP(format.name.c_str(), "opus") != 0 ||
format.clockrate_hz != 48000 || format.num_channels != 2) {
- return rtc::nullopt;
+ return absl::nullopt;
}
AudioEncoderOpusConfig config;
@@ -313,7 +313,7 @@
return config;
}
-rtc::Optional<int> AudioEncoderOpusImpl::GetNewComplexity(
+absl::optional<int> AudioEncoderOpusImpl::GetNewComplexity(
const AudioEncoderOpusConfig& config) {
RTC_DCHECK(config.IsOk());
const int bitrate_bps = GetBitrateBps(config);
@@ -322,7 +322,7 @@
bitrate_bps <= config.complexity_threshold_bps +
config.complexity_threshold_window_bps) {
// Within the hysteresis window; make no change.
- return rtc::nullopt;
+ return absl::nullopt;
} else {
return bitrate_bps <= config.complexity_threshold_bps
? config.low_rate_complexity
@@ -330,7 +330,7 @@
}
}
-rtc::Optional<int> AudioEncoderOpusImpl::GetNewBandwidth(
+absl::optional<int> AudioEncoderOpusImpl::GetNewBandwidth(
const AudioEncoderOpusConfig& config,
OpusEncInst* inst) {
constexpr int kMinWidebandBitrate = 8000;
@@ -339,17 +339,17 @@
RTC_DCHECK(config.IsOk());
const int bitrate = GetBitrateBps(config);
if (bitrate > kAutomaticThreshold) {
- return rtc::Optional<int>(OPUS_AUTO);
+ return absl::optional<int>(OPUS_AUTO);
}
const int bandwidth = WebRtcOpus_GetBandwidth(inst);
RTC_DCHECK_GE(bandwidth, 0);
if (bitrate > kMaxNarrowbandBitrate && bandwidth < OPUS_BANDWIDTH_WIDEBAND) {
- return rtc::Optional<int>(OPUS_BANDWIDTH_WIDEBAND);
+ return absl::optional<int>(OPUS_BANDWIDTH_WIDEBAND);
} else if (bitrate < kMinWidebandBitrate &&
bandwidth > OPUS_BANDWIDTH_NARROWBAND) {
- return rtc::Optional<int>(OPUS_BANDWIDTH_NARROWBAND);
+ return absl::optional<int>(OPUS_BANDWIDTH_NARROWBAND);
}
- return rtc::Optional<int>();
+ return absl::optional<int>();
}
class AudioEncoderOpusImpl::PacketLossFractionSmoother {
@@ -529,7 +529,7 @@
void AudioEncoderOpusImpl::OnReceivedUplinkBandwidth(
int target_audio_bitrate_bps,
- rtc::Optional<int64_t> bwe_period_ms) {
+ absl::optional<int64_t> bwe_period_ms) {
if (audio_network_adaptor_) {
audio_network_adaptor_->SetTargetAudioBitrate(target_audio_bitrate_bps);
// We give smoothed bitrate allocation to audio network adaptor as
@@ -801,7 +801,7 @@
if (!bitrate_smoother_last_update_time_ ||
now_ms - *bitrate_smoother_last_update_time_ >=
config_.uplink_bandwidth_update_interval_ms) {
- rtc::Optional<float> smoothed_bitrate = bitrate_smoother_->GetAverage();
+ absl::optional<float> smoothed_bitrate = bitrate_smoother_->GetAverage();
if (smoothed_bitrate)
audio_network_adaptor_->SetUplinkBandwidth(*smoothed_bitrate);
bitrate_smoother_last_update_time_ = now_ms;
diff --git a/modules/audio_coding/codecs/opus/audio_encoder_opus.h b/modules/audio_coding/codecs/opus/audio_encoder_opus.h
index 49c5207..ea4b265 100644
--- a/modules/audio_coding/codecs/opus/audio_encoder_opus.h
+++ b/modules/audio_coding/codecs/opus/audio_encoder_opus.h
@@ -16,10 +16,10 @@
#include <string>
#include <vector>
+#include "absl/types/optional.h"
#include "api/audio_codecs/audio_encoder.h"
#include "api/audio_codecs/audio_format.h"
#include "api/audio_codecs/opus/audio_encoder_opus_config.h"
-#include "api/optional.h"
#include "common_audio/smoothing_filter.h"
#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
#include "modules/audio_coding/codecs/opus/opus_interface.h"
@@ -40,13 +40,13 @@
// defined by complexity_threshold_bps +/- complexity_threshold_window_bps.
// Otherwise, returns the current complexity depending on whether the
// current bitrate is above or below complexity_threshold_bps.
- static rtc::Optional<int> GetNewComplexity(
+ static absl::optional<int> GetNewComplexity(
const AudioEncoderOpusConfig& config);
// Returns OPUS_AUTO if the the current bitrate is above wideband threshold.
// Returns empty if it is below, but bandwidth coincides with the desired one.
// Otherwise returns the desired bandwidth.
- static rtc::Optional<int> GetNewBandwidth(
+ static absl::optional<int> GetNewBandwidth(
const AudioEncoderOpusConfig& config,
OpusEncInst* inst);
@@ -69,7 +69,7 @@
// Static interface for use by BuiltinAudioEncoderFactory.
static constexpr const char* GetPayloadName() { return "opus"; }
- static rtc::Optional<AudioCodecInfo> QueryAudioEncoder(
+ static absl::optional<AudioCodecInfo> QueryAudioEncoder(
const SdpAudioFormat& format);
int SampleRateHz() const override;
@@ -98,7 +98,7 @@
float uplink_recoverable_packet_loss_fraction) override;
void OnReceivedUplinkBandwidth(
int target_audio_bitrate_bps,
- rtc::Optional<int64_t> bwe_period_ms) override;
+ absl::optional<int64_t> bwe_period_ms) override;
void OnReceivedRtt(int rtt_ms) override;
void OnReceivedOverhead(size_t overhead_bytes_per_packet) override;
void SetReceiverFrameLengthRange(int min_frame_length_ms,
@@ -125,7 +125,7 @@
private:
class PacketLossFractionSmoother;
- static rtc::Optional<AudioEncoderOpusConfig> SdpToConfig(
+ static absl::optional<AudioEncoderOpusConfig> SdpToConfig(
const SdpAudioFormat& format);
static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs);
static AudioCodecInfo QueryAudioEncoder(const AudioEncoderOpusConfig& config);
@@ -167,9 +167,9 @@
std::unique_ptr<PacketLossFractionSmoother> packet_loss_fraction_smoother_;
const AudioNetworkAdaptorCreator audio_network_adaptor_creator_;
std::unique_ptr<AudioNetworkAdaptor> audio_network_adaptor_;
- rtc::Optional<size_t> overhead_bytes_per_packet_;
+ absl::optional<size_t> overhead_bytes_per_packet_;
const std::unique_ptr<SmoothingFilter> bitrate_smoother_;
- rtc::Optional<int64_t> bitrate_smoother_last_update_time_;
+ absl::optional<int64_t> bitrate_smoother_last_update_time_;
int consecutive_dtx_frames_;
friend struct AudioEncoderOpus;
diff --git a/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc b/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
index 3ee734e..c4d37da 100644
--- a/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
+++ b/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
@@ -198,20 +198,20 @@
const int kMinBitrateBps = 6000;
const int kMaxBitrateBps = 510000;
// Set a too low bitrate.
- states->encoder->OnReceivedUplinkBandwidth(kMinBitrateBps - 1, rtc::nullopt);
+ states->encoder->OnReceivedUplinkBandwidth(kMinBitrateBps - 1, absl::nullopt);
EXPECT_EQ(kMinBitrateBps, states->encoder->GetTargetBitrate());
// Set a too high bitrate.
- states->encoder->OnReceivedUplinkBandwidth(kMaxBitrateBps + 1, rtc::nullopt);
+ states->encoder->OnReceivedUplinkBandwidth(kMaxBitrateBps + 1, absl::nullopt);
EXPECT_EQ(kMaxBitrateBps, states->encoder->GetTargetBitrate());
// Set the minimum rate.
- states->encoder->OnReceivedUplinkBandwidth(kMinBitrateBps, rtc::nullopt);
+ states->encoder->OnReceivedUplinkBandwidth(kMinBitrateBps, absl::nullopt);
EXPECT_EQ(kMinBitrateBps, states->encoder->GetTargetBitrate());
// Set the maximum rate.
- states->encoder->OnReceivedUplinkBandwidth(kMaxBitrateBps, rtc::nullopt);
+ states->encoder->OnReceivedUplinkBandwidth(kMaxBitrateBps, absl::nullopt);
EXPECT_EQ(kMaxBitrateBps, states->encoder->GetTargetBitrate());
// Set rates from kMaxBitrateBps up to 32000 bps.
for (int rate = kMinBitrateBps; rate <= 32000; rate += 1000) {
- states->encoder->OnReceivedUplinkBandwidth(rate, rtc::nullopt);
+ states->encoder->OnReceivedUplinkBandwidth(rate, absl::nullopt);
EXPECT_EQ(rate, states->encoder->GetTargetBitrate());
}
}
@@ -392,7 +392,7 @@
auto states = CreateCodec(2);
states->encoder->OnReceivedUplinkBandwidth(kDefaultOpusSettings.rate * 2,
- rtc::nullopt);
+ absl::nullopt);
// Since |OnReceivedOverhead| has not been called, the codec bitrate should
// not change.
@@ -409,7 +409,7 @@
states->encoder->OnReceivedOverhead(kOverheadBytesPerPacket);
constexpr int kTargetBitrateBps = 40000;
- states->encoder->OnReceivedUplinkBandwidth(kTargetBitrateBps, rtc::nullopt);
+ states->encoder->OnReceivedUplinkBandwidth(kTargetBitrateBps, absl::nullopt);
int packet_rate = rtc::CheckedDivExact(48000, kDefaultOpusSettings.pacsize);
EXPECT_EQ(kTargetBitrateBps -
@@ -435,14 +435,14 @@
// subtracted. The eventual codec rate should be bounded by |kMinBitrateBps|.
int target_bitrate =
kOverheadBytesPerPacket * 8 * packet_rate + kMinBitrateBps - 1;
- states->encoder->OnReceivedUplinkBandwidth(target_bitrate, rtc::nullopt);
+ states->encoder->OnReceivedUplinkBandwidth(target_bitrate, absl::nullopt);
EXPECT_EQ(kMinBitrateBps, states->encoder->GetTargetBitrate());
// Set a target rate that is greater than |kMaxBitrateBps| when overhead is
// subtracted. The eventual codec rate should be bounded by |kMaxBitrateBps|.
target_bitrate =
kOverheadBytesPerPacket * 8 * packet_rate + kMaxBitrateBps + 1;
- states->encoder->OnReceivedUplinkBandwidth(target_bitrate, rtc::nullopt);
+ states->encoder->OnReceivedUplinkBandwidth(target_bitrate, absl::nullopt);
EXPECT_EQ(kMaxBitrateBps, states->encoder->GetTargetBitrate());
}
@@ -454,7 +454,7 @@
// Bitrate within hysteresis window. Expect empty output.
config.bitrate_bps = 12500;
- EXPECT_EQ(rtc::nullopt, AudioEncoderOpusImpl::GetNewComplexity(config));
+ EXPECT_EQ(absl::nullopt, AudioEncoderOpusImpl::GetNewComplexity(config));
// Bitrate below hysteresis window. Expect higher complexity.
config.bitrate_bps = 10999;
@@ -462,7 +462,7 @@
// Bitrate within hysteresis window. Expect empty output.
config.bitrate_bps = 12500;
- EXPECT_EQ(rtc::nullopt, AudioEncoderOpusImpl::GetNewComplexity(config));
+ EXPECT_EQ(absl::nullopt, AudioEncoderOpusImpl::GetNewComplexity(config));
// Bitrate above hysteresis window. Expect lower complexity.
config.bitrate_bps = 14001;
@@ -488,9 +488,9 @@
: 1));
// Bitrate below minmum wideband. Expect narrowband.
- config.bitrate_bps = rtc::Optional<int>(7999);
+ config.bitrate_bps = absl::optional<int>(7999);
auto bandwidth = AudioEncoderOpusImpl::GetNewBandwidth(config, inst);
- EXPECT_EQ(rtc::Optional<int>(OPUS_BANDWIDTH_NARROWBAND), bandwidth);
+ EXPECT_EQ(absl::optional<int>(OPUS_BANDWIDTH_NARROWBAND), bandwidth);
WebRtcOpus_SetBandwidth(inst, *bandwidth);
// It is necessary to encode here because Opus has some logic in the encoder
// that goes from the user-set bandwidth to the used and returned one.
@@ -499,14 +499,14 @@
kMaxBytes, bitstream);
// Bitrate not yet above maximum narrowband. Expect empty.
- config.bitrate_bps = rtc::Optional<int>(9000);
+ config.bitrate_bps = absl::optional<int>(9000);
bandwidth = AudioEncoderOpusImpl::GetNewBandwidth(config, inst);
- EXPECT_EQ(rtc::Optional<int>(), bandwidth);
+ EXPECT_EQ(absl::optional<int>(), bandwidth);
// Bitrate above maximum narrowband. Expect wideband.
- config.bitrate_bps = rtc::Optional<int>(9001);
+ config.bitrate_bps = absl::optional<int>(9001);
bandwidth = AudioEncoderOpusImpl::GetNewBandwidth(config, inst);
- EXPECT_EQ(rtc::Optional<int>(OPUS_BANDWIDTH_WIDEBAND), bandwidth);
+ EXPECT_EQ(absl::optional<int>(OPUS_BANDWIDTH_WIDEBAND), bandwidth);
WebRtcOpus_SetBandwidth(inst, *bandwidth);
// It is necessary to encode here because Opus has some logic in the encoder
// that goes from the user-set bandwidth to the used and returned one.
@@ -515,14 +515,14 @@
kMaxBytes, bitstream);
// Bitrate not yet below minimum wideband. Expect empty.
- config.bitrate_bps = rtc::Optional<int>(8000);
+ config.bitrate_bps = absl::optional<int>(8000);
bandwidth = AudioEncoderOpusImpl::GetNewBandwidth(config, inst);
- EXPECT_EQ(rtc::Optional<int>(), bandwidth);
+ EXPECT_EQ(absl::optional<int>(), bandwidth);
// Bitrate above automatic threshold. Expect automatic.
- config.bitrate_bps = rtc::Optional<int>(12001);
+ config.bitrate_bps = absl::optional<int>(12001);
bandwidth = AudioEncoderOpusImpl::GetNewBandwidth(config, inst);
- EXPECT_EQ(rtc::Optional<int>(OPUS_AUTO), bandwidth);
+ EXPECT_EQ(absl::optional<int>(OPUS_AUTO), bandwidth);
EXPECT_EQ(0, WebRtcOpus_EncoderFree(inst));
}
@@ -586,7 +586,7 @@
rtc::Buffer encoded;
uint32_t rtp_timestamp = 12345; // Just a number not important to this test.
- states->encoder->OnReceivedUplinkBandwidth(0, rtc::nullopt);
+ states->encoder->OnReceivedUplinkBandwidth(0, absl::nullopt);
for (int packet_index = 0; packet_index < kNumPacketsToEncode;
packet_index++) {
// Make sure we are not encoding before we have enough data for
diff --git a/modules/audio_coding/codecs/opus/opus_bandwidth_unittest.cc b/modules/audio_coding/codecs/opus/opus_bandwidth_unittest.cc
index 3c4cb02..7f09c2a 100644
--- a/modules/audio_coding/codecs/opus/opus_bandwidth_unittest.cc
+++ b/modules/audio_coding/codecs/opus/opus_bandwidth_unittest.cc
@@ -110,7 +110,7 @@
// Create encoder.
AudioEncoderOpusConfig enc_config;
- enc_config.bitrate_bps = rtc::Optional<int>(7999);
+ enc_config.bitrate_bps = absl::optional<int>(7999);
enc_config.num_channels = kNumChannels;
constexpr int payload_type = 17;
auto encoder = AudioEncoderOpus::MakeAudioEncoder(enc_config, payload_type);
diff --git a/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc b/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc
index 4b9df6e..cd62069 100644
--- a/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc
+++ b/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc
@@ -134,7 +134,7 @@
void AudioEncoderCopyRed::OnReceivedUplinkBandwidth(
int target_audio_bitrate_bps,
- rtc::Optional<int64_t> bwe_period_ms) {
+ absl::optional<int64_t> bwe_period_ms) {
speech_encoder_->OnReceivedUplinkBandwidth(target_audio_bitrate_bps,
bwe_period_ms);
}
diff --git a/modules/audio_coding/codecs/red/audio_encoder_copy_red.h b/modules/audio_coding/codecs/red/audio_encoder_copy_red.h
index e625c50..492ee3a 100644
--- a/modules/audio_coding/codecs/red/audio_encoder_copy_red.h
+++ b/modules/audio_coding/codecs/red/audio_encoder_copy_red.h
@@ -57,7 +57,7 @@
float uplink_recoverable_packet_loss_fraction) override;
void OnReceivedUplinkBandwidth(
int target_audio_bitrate_bps,
- rtc::Optional<int64_t> bwe_period_ms) override;
+ absl::optional<int64_t> bwe_period_ms) override;
protected:
EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
diff --git a/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc b/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc
index 64bafd2..890ac22 100644
--- a/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc
+++ b/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc
@@ -99,8 +99,8 @@
TEST_F(AudioEncoderCopyRedTest, CheckTargetAudioBitratePropagation) {
EXPECT_CALL(*mock_encoder_,
- OnReceivedUplinkBandwidth(4711, rtc::Optional<int64_t>()));
- red_->OnReceivedUplinkBandwidth(4711, rtc::nullopt);
+ OnReceivedUplinkBandwidth(4711, absl::optional<int64_t>()));
+ red_->OnReceivedUplinkBandwidth(4711, absl::nullopt);
}
TEST_F(AudioEncoderCopyRedTest, CheckPacketLossFractionPropagation) {
diff --git a/modules/audio_coding/include/audio_coding_module.h b/modules/audio_coding/include/audio_coding_module.h
index 3c193a4..dfbe459 100644
--- a/modules/audio_coding/include/audio_coding_module.h
+++ b/modules/audio_coding/include/audio_coding_module.h
@@ -15,9 +15,9 @@
#include <string>
#include <vector>
+#include "absl/types/optional.h"
#include "api/audio_codecs/audio_decoder_factory.h"
#include "api/audio_codecs/audio_encoder.h"
-#include "api/optional.h"
#include "common_types.h" // NOLINT(build/include)
#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
#include "modules/audio_coding/neteq/include/neteq.h"
@@ -228,7 +228,7 @@
// Return value:
// The send codec, or nothing if we don't have one
//
- virtual rtc::Optional<CodecInst> SendCodec() const = 0;
+ virtual absl::optional<CodecInst> SendCodec() const = 0;
///////////////////////////////////////////////////////////////////////////
// int32_t SendFrequency()
@@ -546,7 +546,7 @@
virtual int32_t ReceiveCodec(CodecInst* curr_receive_codec) const = 0;
///////////////////////////////////////////////////////////////////////////
- // rtc::Optional<SdpAudioFormat> ReceiveFormat()
+ // absl::optional<SdpAudioFormat> ReceiveFormat()
// Get the format associated with last received payload.
//
// Return value:
@@ -554,7 +554,7 @@
// received payload.
// An empty Optional if no payload has yet been received.
//
- virtual rtc::Optional<SdpAudioFormat> ReceiveFormat() const = 0;
+ virtual absl::optional<SdpAudioFormat> ReceiveFormat() const = 0;
///////////////////////////////////////////////////////////////////////////
// int32_t IncomingPacket()
@@ -631,7 +631,7 @@
// the latest audio obtained by calling PlayoutData10ms(), or empty if no
// valid timestamp is available.
//
- virtual rtc::Optional<uint32_t> PlayoutTimestamp() = 0;
+ virtual absl::optional<uint32_t> PlayoutTimestamp() = 0;
///////////////////////////////////////////////////////////////////////////
// int FilteredCurrentDelayMs()
diff --git a/modules/audio_coding/neteq/audio_decoder_unittest.cc b/modules/audio_coding/neteq/audio_decoder_unittest.cc
index 3181d6f..e8f7a4a 100644
--- a/modules/audio_coding/neteq/audio_decoder_unittest.cc
+++ b/modules/audio_coding/neteq/audio_decoder_unittest.cc
@@ -462,7 +462,7 @@
namespace {
int SetAndGetTargetBitrate(AudioEncoder* audio_encoder, int rate) {
- audio_encoder->OnReceivedUplinkBandwidth(rate, rtc::nullopt);
+ audio_encoder->OnReceivedUplinkBandwidth(rate, absl::nullopt);
return audio_encoder->GetTargetBitrate();
}
void TestSetAndGetTargetBitratesWithFixedCodec(AudioEncoder* audio_encoder,
diff --git a/modules/audio_coding/neteq/decision_logic_unittest.cc b/modules/audio_coding/neteq/decision_logic_unittest.cc
index be1b854..5f0e5c2 100644
--- a/modules/audio_coding/neteq/decision_logic_unittest.cc
+++ b/modules/audio_coding/neteq/decision_logic_unittest.cc
@@ -26,7 +26,7 @@
int fs_hz = 8000;
int output_size_samples = fs_hz / 100; // Samples per 10 ms.
DecoderDatabase decoder_database(
- new rtc::RefCountedObject<MockAudioDecoderFactory>, rtc::nullopt);
+ new rtc::RefCountedObject<MockAudioDecoderFactory>, absl::nullopt);
TickTimer tick_timer;
PacketBuffer packet_buffer(10, &tick_timer);
DelayPeakDetector delay_peak_detector(&tick_timer);
diff --git a/modules/audio_coding/neteq/decoder_database.cc b/modules/audio_coding/neteq/decoder_database.cc
index 5b940ae..72c0376 100644
--- a/modules/audio_coding/neteq/decoder_database.cc
+++ b/modules/audio_coding/neteq/decoder_database.cc
@@ -21,7 +21,7 @@
DecoderDatabase::DecoderDatabase(
const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory,
- rtc::Optional<AudioCodecPairId> codec_pair_id)
+ absl::optional<AudioCodecPairId> codec_pair_id)
: active_decoder_type_(-1),
active_cng_decoder_type_(-1),
decoder_factory_(decoder_factory),
@@ -31,7 +31,7 @@
DecoderDatabase::DecoderInfo::DecoderInfo(
const SdpAudioFormat& audio_format,
- rtc::Optional<AudioCodecPairId> codec_pair_id,
+ absl::optional<AudioCodecPairId> codec_pair_id,
AudioDecoderFactory* factory,
const std::string& codec_name)
: name_(codec_name),
@@ -44,13 +44,13 @@
DecoderDatabase::DecoderInfo::DecoderInfo(
const SdpAudioFormat& audio_format,
- rtc::Optional<AudioCodecPairId> codec_pair_id,
+ absl::optional<AudioCodecPairId> codec_pair_id,
AudioDecoderFactory* factory)
: DecoderInfo(audio_format, codec_pair_id, factory, audio_format.name) {}
DecoderDatabase::DecoderInfo::DecoderInfo(
NetEqDecoder ct,
- rtc::Optional<AudioCodecPairId> codec_pair_id,
+ absl::optional<AudioCodecPairId> codec_pair_id,
AudioDecoderFactory* factory)
: DecoderInfo(*NetEqDecoderToSdpAudioFormat(ct), codec_pair_id, factory) {}
@@ -59,7 +59,7 @@
const std::string& codec_name)
: name_(codec_name),
audio_format_(audio_format),
- codec_pair_id_(rtc::nullopt),
+ codec_pair_id_(absl::nullopt),
factory_(nullptr),
external_decoder_(ext_dec),
subtype_(Subtype::kNormal) {
@@ -108,7 +108,7 @@
return IsType(name.c_str());
}
-rtc::Optional<DecoderDatabase::DecoderInfo::CngDecoder>
+absl::optional<DecoderDatabase::DecoderInfo::CngDecoder>
DecoderDatabase::DecoderInfo::CngDecoder::Create(const SdpAudioFormat& format) {
if (STR_CASE_CMP(format.name.c_str(), "CN") == 0) {
// CN has a 1:1 RTP clock rate to sample rate ratio.
@@ -117,7 +117,7 @@
sample_rate_hz == 32000 || sample_rate_hz == 48000);
return DecoderDatabase::DecoderInfo::CngDecoder{sample_rate_hz};
} else {
- return rtc::nullopt;
+ return absl::nullopt;
}
}
diff --git a/modules/audio_coding/neteq/decoder_database.h b/modules/audio_coding/neteq/decoder_database.h
index f769e39..6b388dd 100644
--- a/modules/audio_coding/neteq/decoder_database.h
+++ b/modules/audio_coding/neteq/decoder_database.h
@@ -43,14 +43,14 @@
class DecoderInfo {
public:
DecoderInfo(const SdpAudioFormat& audio_format,
- rtc::Optional<AudioCodecPairId> codec_pair_id,
+ absl::optional<AudioCodecPairId> codec_pair_id,
AudioDecoderFactory* factory,
const std::string& codec_name);
explicit DecoderInfo(const SdpAudioFormat& audio_format,
- rtc::Optional<AudioCodecPairId> codec_pair_id,
+ absl::optional<AudioCodecPairId> codec_pair_id,
AudioDecoderFactory* factory = nullptr);
explicit DecoderInfo(NetEqDecoder ct,
- rtc::Optional<AudioCodecPairId> codec_pair_id,
+ absl::optional<AudioCodecPairId> codec_pair_id,
AudioDecoderFactory* factory = nullptr);
DecoderInfo(const SdpAudioFormat& audio_format,
AudioDecoder* ext_dec,
@@ -111,7 +111,7 @@
const std::string name_;
const SdpAudioFormat audio_format_;
- const rtc::Optional<AudioCodecPairId> codec_pair_id_;
+ const absl::optional<AudioCodecPairId> codec_pair_id_;
AudioDecoderFactory* const factory_;
mutable std::unique_ptr<AudioDecoder> decoder_;
@@ -120,10 +120,10 @@
// Set iff this is a comfort noise decoder.
struct CngDecoder {
- static rtc::Optional<CngDecoder> Create(const SdpAudioFormat& format);
+ static absl::optional<CngDecoder> Create(const SdpAudioFormat& format);
int sample_rate_hz;
};
- const rtc::Optional<CngDecoder> cng_decoder_;
+ const absl::optional<CngDecoder> cng_decoder_;
enum class Subtype : int8_t {
kNormal,
@@ -143,7 +143,7 @@
DecoderDatabase(
const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory,
- rtc::Optional<AudioCodecPairId> codec_pair_id);
+ absl::optional<AudioCodecPairId> codec_pair_id);
virtual ~DecoderDatabase();
@@ -247,7 +247,7 @@
int active_cng_decoder_type_;
mutable std::unique_ptr<ComfortNoiseDecoder> active_cng_decoder_;
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_;
- const rtc::Optional<AudioCodecPairId> codec_pair_id_;
+ const absl::optional<AudioCodecPairId> codec_pair_id_;
RTC_DISALLOW_COPY_AND_ASSIGN(DecoderDatabase);
};
diff --git a/modules/audio_coding/neteq/decoder_database_unittest.cc b/modules/audio_coding/neteq/decoder_database_unittest.cc
index a6b9689..afd10ae 100644
--- a/modules/audio_coding/neteq/decoder_database_unittest.cc
+++ b/modules/audio_coding/neteq/decoder_database_unittest.cc
@@ -29,7 +29,7 @@
TEST(DecoderDatabase, CreateAndDestroy) {
DecoderDatabase db(new rtc::RefCountedObject<MockAudioDecoderFactory>,
- rtc::nullopt);
+ absl::nullopt);
EXPECT_EQ(0, db.Size());
EXPECT_TRUE(db.Empty());
}
@@ -42,7 +42,7 @@
EXPECT_EQ("pcmu", format.name);
return true;
}));
- DecoderDatabase db(factory, rtc::nullopt);
+ DecoderDatabase db(factory, absl::nullopt);
const uint8_t kPayloadType = 0;
const std::string kCodecName = "Robert\'); DROP TABLE Students;";
EXPECT_EQ(
@@ -67,7 +67,7 @@
EXPECT_EQ("pcma", format.name);
return true;
}));
- DecoderDatabase db(factory, rtc::nullopt);
+ DecoderDatabase db(factory, absl::nullopt);
const std::string kCodecName1 = "Robert\'); DROP TABLE Students;";
const std::string kCodecName2 = "https://xkcd.com/327/";
EXPECT_EQ(DecoderDatabase::kOK,
@@ -92,12 +92,12 @@
auto* decoder = new MockAudioDecoder;
EXPECT_CALL(*factory, MakeAudioDecoderMock(_, _, _))
.WillOnce(Invoke([decoder](const SdpAudioFormat& format,
- rtc::Optional<AudioCodecPairId> codec_pair_id,
+ absl::optional<AudioCodecPairId> codec_pair_id,
std::unique_ptr<AudioDecoder>* dec) {
EXPECT_EQ("pcmu", format.name);
dec->reset(decoder);
}));
- DecoderDatabase db(factory, rtc::nullopt);
+ DecoderDatabase db(factory, absl::nullopt);
const uint8_t kPayloadType = 0;
const std::string kCodecName = "Robert\'); DROP TABLE Students;";
EXPECT_EQ(
@@ -114,7 +114,7 @@
}
TEST(DecoderDatabase, GetDecoder) {
- DecoderDatabase db(CreateBuiltinAudioDecoderFactory(), rtc::nullopt);
+ DecoderDatabase db(CreateBuiltinAudioDecoderFactory(), absl::nullopt);
const uint8_t kPayloadType = 0;
const std::string kCodecName = "Robert\'); DROP TABLE Students;";
EXPECT_EQ(DecoderDatabase::kOK,
@@ -132,7 +132,7 @@
EXPECT_EQ("pcmu", format.name);
return true;
}));
- DecoderDatabase db(factory, rtc::nullopt);
+ DecoderDatabase db(factory, absl::nullopt);
const uint8_t kPayloadTypePcmU = 0;
const uint8_t kPayloadTypeCng = 13;
const uint8_t kPayloadTypeDtmf = 100;
@@ -168,7 +168,7 @@
TEST(DecoderDatabase, ExternalDecoder) {
DecoderDatabase db(new rtc::RefCountedObject<MockAudioDecoderFactory>,
- rtc::nullopt);
+ absl::nullopt);
const uint8_t kPayloadType = 0;
const std::string kCodecName = "Robert\'); DROP TABLE Students;";
MockAudioDecoder decoder;
@@ -205,7 +205,7 @@
EXPECT_EQ("pcmu", format.name);
return true;
}));
- DecoderDatabase db(factory, rtc::nullopt);
+ DecoderDatabase db(factory, absl::nullopt);
// Load a number of payloads into the database. Payload types are 0, 1, ...,
// while the decoder type is the same for all payload types (this does not
// matter for the test).
@@ -245,7 +245,7 @@
// Test the methods for setting and getting active speech and CNG decoders.
TEST(DecoderDatabase, IF_ISAC(ActiveDecoders)) {
- DecoderDatabase db(CreateBuiltinAudioDecoderFactory(), rtc::nullopt);
+ DecoderDatabase db(CreateBuiltinAudioDecoderFactory(), absl::nullopt);
// Load payload types.
ASSERT_EQ(DecoderDatabase::kOK,
db.RegisterPayload(0, NetEqDecoder::kDecoderPCMu, "pcmu"));
diff --git a/modules/audio_coding/neteq/expand_uma_logger.cc b/modules/audio_coding/neteq/expand_uma_logger.cc
index c656eed..01c2dab 100644
--- a/modules/audio_coding/neteq/expand_uma_logger.cc
+++ b/modules/audio_coding/neteq/expand_uma_logger.cc
@@ -45,7 +45,7 @@
last_value_ = samples;
sample_rate_hz_ = sample_rate_hz;
if (!last_logged_value_) {
- last_logged_value_ = rtc::Optional<uint64_t>(samples);
+ last_logged_value_ = absl::optional<uint64_t>(samples);
}
if (!timer_->Finished()) {
@@ -56,7 +56,7 @@
RTC_DCHECK(last_logged_value_);
RTC_DCHECK_GE(last_value_, *last_logged_value_);
const uint64_t diff = last_value_ - *last_logged_value_;
- last_logged_value_ = rtc::Optional<uint64_t>(last_value_);
+ last_logged_value_ = absl::optional<uint64_t>(last_value_);
// Calculate rate in percent.
RTC_DCHECK_GT(sample_rate_hz, 0);
const int rate = (100 * diff) / (sample_rate_hz * logging_period_s_);
diff --git a/modules/audio_coding/neteq/expand_uma_logger.h b/modules/audio_coding/neteq/expand_uma_logger.h
index 70af39b..00907d4 100644
--- a/modules/audio_coding/neteq/expand_uma_logger.h
+++ b/modules/audio_coding/neteq/expand_uma_logger.h
@@ -13,7 +13,7 @@
#include <memory>
#include <string>
-#include "api/optional.h"
+#include "absl/types/optional.h"
#include "modules/audio_coding/neteq/tick_timer.h"
#include "rtc_base/constructormagic.h"
@@ -43,7 +43,7 @@
const int logging_period_s_;
const TickTimer& tick_timer_;
std::unique_ptr<TickTimer::Countdown> timer_;
- rtc::Optional<uint64_t> last_logged_value_;
+ absl::optional<uint64_t> last_logged_value_;
uint64_t last_value_ = 0;
int sample_rate_hz_ = 0;
diff --git a/modules/audio_coding/neteq/include/neteq.h b/modules/audio_coding/neteq/include/neteq.h
index 310a227..6288aeb 100644
--- a/modules/audio_coding/neteq/include/neteq.h
+++ b/modules/audio_coding/neteq/include/neteq.h
@@ -16,9 +16,9 @@
#include <string>
#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/optional.h"
#include "api/rtp_headers.h"
#include "common_types.h" // NOLINT(build/include)
#include "modules/audio_coding/neteq/neteq_decoder_enum.h"
@@ -101,7 +101,7 @@
NetEqPlayoutMode playout_mode = kPlayoutOn;
bool enable_fast_accelerate = false;
bool enable_muted_state = false;
- rtc::Optional<AudioCodecPairId> codec_pair_id;
+ absl::optional<AudioCodecPairId> codec_pair_id;
};
enum ReturnCodes {
@@ -247,7 +247,7 @@
// Returns the RTP timestamp for the last sample delivered by GetAudio().
// The return value will be empty if no valid timestamp is available.
- virtual rtc::Optional<uint32_t> GetPlayoutTimestamp() const = 0;
+ virtual absl::optional<uint32_t> GetPlayoutTimestamp() const = 0;
// Returns the sample rate in Hz of the audio produced in the last GetAudio
// call. If GetAudio has not been called yet, the configured sample rate
@@ -256,11 +256,11 @@
// Returns info about the decoder for the given payload type, or an empty
// value if we have no decoder for that payload type.
- virtual rtc::Optional<CodecInst> GetDecoder(int payload_type) const = 0;
+ virtual absl::optional<CodecInst> GetDecoder(int payload_type) const = 0;
// Returns the decoder format for the given payload type. Returns empty if no
// such payload type was registered.
- virtual rtc::Optional<SdpAudioFormat> GetDecoderFormat(
+ virtual absl::optional<SdpAudioFormat> GetDecoderFormat(
int payload_type) const = 0;
// Not implemented.
diff --git a/modules/audio_coding/neteq/mock/mock_decoder_database.h b/modules/audio_coding/neteq/mock/mock_decoder_database.h
index a4240ce..3d57edd 100644
--- a/modules/audio_coding/neteq/mock/mock_decoder_database.h
+++ b/modules/audio_coding/neteq/mock/mock_decoder_database.h
@@ -23,7 +23,7 @@
public:
explicit MockDecoderDatabase(
rtc::scoped_refptr<AudioDecoderFactory> factory = nullptr)
- : DecoderDatabase(factory, rtc::nullopt) {}
+ : DecoderDatabase(factory, absl::nullopt) {}
virtual ~MockDecoderDatabase() { Die(); }
MOCK_METHOD0(Die, void());
MOCK_CONST_METHOD0(Empty,
diff --git a/modules/audio_coding/neteq/mock/mock_packet_buffer.h b/modules/audio_coding/neteq/mock/mock_packet_buffer.h
index ac7d9b7..b477b1a 100644
--- a/modules/audio_coding/neteq/mock/mock_packet_buffer.h
+++ b/modules/audio_coding/neteq/mock/mock_packet_buffer.h
@@ -38,8 +38,8 @@
MOCK_METHOD5(InsertPacketList,
int(PacketList* packet_list,
const DecoderDatabase& decoder_database,
- rtc::Optional<uint8_t>* current_rtp_payload_type,
- rtc::Optional<uint8_t>* current_cng_rtp_payload_type,
+ absl::optional<uint8_t>* current_rtp_payload_type,
+ absl::optional<uint8_t>* current_cng_rtp_payload_type,
StatisticsCalculator* stats));
MOCK_CONST_METHOD1(NextTimestamp,
int(uint32_t* next_timestamp));
@@ -47,8 +47,7 @@
int(uint32_t timestamp, uint32_t* next_timestamp));
MOCK_CONST_METHOD0(PeekNextPacket,
const Packet*());
- MOCK_METHOD0(GetNextPacket,
- rtc::Optional<Packet>());
+ MOCK_METHOD0(GetNextPacket, absl::optional<Packet>());
MOCK_METHOD1(DiscardNextPacket, int(StatisticsCalculator* stats));
MOCK_METHOD3(DiscardOldPackets,
void(uint32_t timestamp_limit,
diff --git a/modules/audio_coding/neteq/neteq_decoder_enum.cc b/modules/audio_coding/neteq/neteq_decoder_enum.cc
index 8d66c2a..e3b633e 100644
--- a/modules/audio_coding/neteq/neteq_decoder_enum.cc
+++ b/modules/audio_coding/neteq/neteq_decoder_enum.cc
@@ -15,7 +15,7 @@
namespace webrtc {
-rtc::Optional<SdpAudioFormat> NetEqDecoderToSdpAudioFormat(NetEqDecoder nd) {
+absl::optional<SdpAudioFormat> NetEqDecoderToSdpAudioFormat(NetEqDecoder nd) {
switch (nd) {
case NetEqDecoder::kDecoderPCMu:
return SdpAudioFormat("pcmu", 8000, 1);
@@ -78,7 +78,7 @@
case NetEqDecoder::kDecoderCNGswb48kHz:
return SdpAudioFormat("cn", 48000, 1);
default:
- return rtc::nullopt;
+ return absl::nullopt;
}
}
diff --git a/modules/audio_coding/neteq/neteq_decoder_enum.h b/modules/audio_coding/neteq/neteq_decoder_enum.h
index 024f03c..00629bc 100644
--- a/modules/audio_coding/neteq/neteq_decoder_enum.h
+++ b/modules/audio_coding/neteq/neteq_decoder_enum.h
@@ -11,8 +11,8 @@
#ifndef MODULES_AUDIO_CODING_NETEQ_NETEQ_DECODER_ENUM_H_
#define MODULES_AUDIO_CODING_NETEQ_NETEQ_DECODER_ENUM_H_
+#include "absl/types/optional.h"
#include "api/audio_codecs/audio_format.h"
-#include "api/optional.h"
namespace webrtc {
@@ -49,7 +49,7 @@
kDecoderOpus_2ch,
};
-rtc::Optional<SdpAudioFormat> NetEqDecoderToSdpAudioFormat(NetEqDecoder nd);
+absl::optional<SdpAudioFormat> NetEqDecoderToSdpAudioFormat(NetEqDecoder nd);
} // namespace webrtc
diff --git a/modules/audio_coding/neteq/neteq_impl.cc b/modules/audio_coding/neteq/neteq_impl.cc
index 688162b..40eae1b 100644
--- a/modules/audio_coding/neteq/neteq_impl.cc
+++ b/modules/audio_coding/neteq/neteq_impl.cc
@@ -422,14 +422,14 @@
vad_->Disable();
}
-rtc::Optional<uint32_t> NetEqImpl::GetPlayoutTimestamp() const {
+absl::optional<uint32_t> NetEqImpl::GetPlayoutTimestamp() const {
rtc::CritScope lock(&crit_sect_);
if (first_packet_ || last_mode_ == kModeRfc3389Cng ||
last_mode_ == kModeCodecInternalCng) {
// We don't have a valid RTP timestamp until we have decoded our first
// RTP packet. Also, the RTP timestamp is not accurate while playing CNG,
// which is indicated by returning an empty value.
- return rtc::nullopt;
+ return absl::nullopt;
}
return timestamp_scaler_->ToExternal(playout_timestamp_);
}
@@ -439,12 +439,12 @@
return last_output_sample_rate_hz_;
}
-rtc::Optional<CodecInst> NetEqImpl::GetDecoder(int payload_type) const {
+absl::optional<CodecInst> NetEqImpl::GetDecoder(int payload_type) const {
rtc::CritScope lock(&crit_sect_);
const DecoderDatabase::DecoderInfo* di =
decoder_database_->GetDecoderInfo(payload_type);
if (!di) {
- return rtc::nullopt;
+ return absl::nullopt;
}
// Create a CodecInst with some fields set. The remaining fields are zeroed,
@@ -460,13 +460,13 @@
return ci;
}
-rtc::Optional<SdpAudioFormat> NetEqImpl::GetDecoderFormat(
+absl::optional<SdpAudioFormat> NetEqImpl::GetDecoderFormat(
int payload_type) const {
rtc::CritScope lock(&crit_sect_);
const DecoderDatabase::DecoderInfo* const di =
decoder_database_->GetDecoderInfo(payload_type);
if (!di) {
- return rtc::nullopt; // Payload type not registered.
+ return absl::nullopt; // Payload type not registered.
}
return di->GetFormat();
}
@@ -1957,7 +1957,7 @@
// Packet extraction loop.
do {
timestamp_ = next_packet->timestamp;
- rtc::Optional<Packet> packet = packet_buffer_->GetNextPacket();
+ absl::optional<Packet> packet = packet_buffer_->GetNextPacket();
// |next_packet| may be invalid after the |packet_buffer_| operation.
next_packet = nullptr;
if (!packet) {
@@ -2009,7 +2009,7 @@
stats_.JitterBufferDelay(extracted_samples, waiting_time_ms);
packet_list->push_back(std::move(*packet)); // Store packet in list.
- packet = rtc::nullopt; // Ensure it's never used after the move.
+ packet = absl::nullopt; // Ensure it's never used after the move.
// Check what packet is available next.
next_packet = packet_buffer_->PeekNextPacket();
diff --git a/modules/audio_coding/neteq/neteq_impl.h b/modules/audio_coding/neteq/neteq_impl.h
index 17589aa..6f69680 100644
--- a/modules/audio_coding/neteq/neteq_impl.h
+++ b/modules/audio_coding/neteq/neteq_impl.h
@@ -14,7 +14,7 @@
#include <memory>
#include <string>
-#include "api/optional.h"
+#include "absl/types/optional.h"
#include "api/audio/audio_frame.h"
#include "modules/audio_coding/neteq/audio_multi_vector.h"
#include "modules/audio_coding/neteq/defines.h"
@@ -198,13 +198,13 @@
// Disables post-decode VAD.
void DisableVad() override;
- rtc::Optional<uint32_t> GetPlayoutTimestamp() const override;
+ absl::optional<uint32_t> GetPlayoutTimestamp() const override;
int last_output_sample_rate_hz() const override;
- rtc::Optional<CodecInst> GetDecoder(int payload_type) const override;
+ absl::optional<CodecInst> GetDecoder(int payload_type) const override;
- rtc::Optional<SdpAudioFormat> GetDecoderFormat(
+ absl::optional<SdpAudioFormat> GetDecoderFormat(
int payload_type) const override;
int SetTargetNumberOfChannels() override;
@@ -424,8 +424,8 @@
bool new_codec_ RTC_GUARDED_BY(crit_sect_);
uint32_t timestamp_ RTC_GUARDED_BY(crit_sect_);
bool reset_decoder_ RTC_GUARDED_BY(crit_sect_);
- rtc::Optional<uint8_t> current_rtp_payload_type_ RTC_GUARDED_BY(crit_sect_);
- rtc::Optional<uint8_t> current_cng_rtp_payload_type_
+ absl::optional<uint8_t> current_rtp_payload_type_ RTC_GUARDED_BY(crit_sect_);
+ absl::optional<uint8_t> current_cng_rtp_payload_type_
RTC_GUARDED_BY(crit_sect_);
uint32_t ssrc_ RTC_GUARDED_BY(crit_sect_);
bool first_packet_ RTC_GUARDED_BY(crit_sect_);
diff --git a/modules/audio_coding/neteq/neteq_impl_unittest.cc b/modules/audio_coding/neteq/neteq_impl_unittest.cc
index de24cda..c8fd91a 100644
--- a/modules/audio_coding/neteq/neteq_impl_unittest.cc
+++ b/modules/audio_coding/neteq/neteq_impl_unittest.cc
@@ -314,7 +314,7 @@
new rtc::RefCountedObject<MockAudioDecoderFactory>);
EXPECT_CALL(*mock_decoder_factory, MakeAudioDecoderMock(_, _, _))
.WillOnce(Invoke([&](const SdpAudioFormat& format,
- rtc::Optional<AudioCodecPairId> codec_pair_id,
+ absl::optional<AudioCodecPairId> codec_pair_id,
std::unique_ptr<AudioDecoder>* dec) {
EXPECT_EQ("pcmu", format.name);
@@ -334,7 +334,7 @@
*dec = std::move(mock_decoder);
}));
- DecoderDatabase::DecoderInfo info(NetEqDecoder::kDecoderPCMu, rtc::nullopt,
+ DecoderDatabase::DecoderInfo info(NetEqDecoder::kDecoderPCMu, absl::nullopt,
mock_decoder_factory);
// Expectations for decoder database.
@@ -518,10 +518,10 @@
// The value of the last of the output samples is the same as the number of
// samples played from the decoded packet. Thus, this number + the RTP
// timestamp should match the playout timestamp.
- // Wrap the expected value in an rtc::Optional to compare them as such.
+ // Wrap the expected value in an absl::optional to compare them as such.
EXPECT_EQ(
- rtc::Optional<uint32_t>(rtp_header.timestamp +
- output.data()[output.samples_per_channel_ - 1]),
+ absl::optional<uint32_t>(rtp_header.timestamp +
+ output.data()[output.samples_per_channel_ - 1]),
neteq_->GetPlayoutTimestamp());
// Check the timestamp for the last value in the sync buffer. This should
@@ -783,12 +783,12 @@
bool muted;
EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
- rtc::Optional<uint32_t> last_timestamp = neteq_->GetPlayoutTimestamp();
+ absl::optional<uint32_t> last_timestamp = neteq_->GetPlayoutTimestamp();
ASSERT_TRUE(last_timestamp);
// Lambda for verifying the timestamps.
auto verify_timestamp = [&last_timestamp, &expected_timestamp_increment](
- rtc::Optional<uint32_t> ts, size_t i) {
+ absl::optional<uint32_t> ts, size_t i) {
if (expected_timestamp_increment[i] == -1) {
// Expect to get an empty timestamp value during CNG and PLC.
EXPECT_FALSE(ts) << "i = " << i;
diff --git a/modules/audio_coding/neteq/neteq_network_stats_unittest.cc b/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
index b317099..585fd8f 100644
--- a/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
+++ b/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
@@ -46,7 +46,7 @@
size_t Duration() const override { return kPacketDuration; }
- rtc::Optional<DecodeResult> Decode(
+ absl::optional<DecodeResult> Decode(
rtc::ArrayView<int16_t> decoded) const override {
const size_t output_size =
sizeof(int16_t) * kPacketDuration * num_channels_;
@@ -57,7 +57,7 @@
} else {
ADD_FAILURE() << "Expected decoded.size() to be >= output_size ("
<< decoded.size() << " vs. " << output_size << ")";
- return rtc::nullopt;
+ return absl::nullopt;
}
}
diff --git a/modules/audio_coding/neteq/neteq_unittest.cc b/modules/audio_coding/neteq/neteq_unittest.cc
index dff4568..6239985 100644
--- a/modules/audio_coding/neteq/neteq_unittest.cc
+++ b/modules/audio_coding/neteq/neteq_unittest.cc
@@ -698,7 +698,7 @@
}
EXPECT_EQ(AudioFrame::kNormalSpeech, out_frame_.speech_type_);
- rtc::Optional<uint32_t> playout_timestamp = neteq_->GetPlayoutTimestamp();
+ absl::optional<uint32_t> playout_timestamp = neteq_->GetPlayoutTimestamp();
ASSERT_TRUE(playout_timestamp);
int32_t delay_before = timestamp - *playout_timestamp;
@@ -1121,7 +1121,7 @@
ASSERT_EQ(1u, output.num_channels_);
// Expect delay (in samples) to be less than 2 packets.
- rtc::Optional<uint32_t> playout_timestamp = neteq_->GetPlayoutTimestamp();
+ absl::optional<uint32_t> playout_timestamp = neteq_->GetPlayoutTimestamp();
ASSERT_TRUE(playout_timestamp);
EXPECT_LE(timestamp - *playout_timestamp,
static_cast<uint32_t>(kSamples * 2));
@@ -1233,7 +1233,7 @@
ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
EXPECT_EQ(AudioFrame::kNormalSpeech, out_frame_.speech_type_);
- rtc::Optional<uint32_t> playout_timestamp = neteq_->GetPlayoutTimestamp();
+ absl::optional<uint32_t> playout_timestamp = neteq_->GetPlayoutTimestamp();
ASSERT_TRUE(playout_timestamp);
EXPECT_EQ(timestamp + kSamples - algorithmic_delay_samples,
*playout_timestamp);
diff --git a/modules/audio_coding/neteq/packet_buffer.cc b/modules/audio_coding/neteq/packet_buffer.cc
index c08c447..f7b622d 100644
--- a/modules/audio_coding/neteq/packet_buffer.cc
+++ b/modules/audio_coding/neteq/packet_buffer.cc
@@ -129,8 +129,8 @@
int PacketBuffer::InsertPacketList(
PacketList* packet_list,
const DecoderDatabase& decoder_database,
- rtc::Optional<uint8_t>* current_rtp_payload_type,
- rtc::Optional<uint8_t>* current_cng_rtp_payload_type,
+ absl::optional<uint8_t>* current_rtp_payload_type,
+ absl::optional<uint8_t>* current_cng_rtp_payload_type,
StatisticsCalculator* stats) {
RTC_DCHECK(stats);
bool flushed = false;
@@ -139,7 +139,7 @@
if (*current_cng_rtp_payload_type &&
**current_cng_rtp_payload_type != packet.payload_type) {
// New CNG payload type implies new codec type.
- *current_rtp_payload_type = rtc::nullopt;
+ *current_rtp_payload_type = absl::nullopt;
Flush();
flushed = true;
}
@@ -152,7 +152,7 @@
!EqualSampleRates(packet.payload_type,
**current_cng_rtp_payload_type,
decoder_database))) {
- *current_cng_rtp_payload_type = rtc::nullopt;
+ *current_cng_rtp_payload_type = absl::nullopt;
Flush();
flushed = true;
}
@@ -206,13 +206,13 @@
return buffer_.empty() ? nullptr : &buffer_.front();
}
-rtc::Optional<Packet> PacketBuffer::GetNextPacket() {
+absl::optional<Packet> PacketBuffer::GetNextPacket() {
if (Empty()) {
// Buffer is empty.
- return rtc::nullopt;
+ return absl::nullopt;
}
- rtc::Optional<Packet> packet(std::move(buffer_.front()));
+ absl::optional<Packet> packet(std::move(buffer_.front()));
// Assert that the packet sanity checks in InsertPacket method works.
RTC_DCHECK(!packet->empty());
buffer_.pop_front();
diff --git a/modules/audio_coding/neteq/packet_buffer.h b/modules/audio_coding/neteq/packet_buffer.h
index c646626..74f36a5 100644
--- a/modules/audio_coding/neteq/packet_buffer.h
+++ b/modules/audio_coding/neteq/packet_buffer.h
@@ -11,7 +11,7 @@
#ifndef MODULES_AUDIO_CODING_NETEQ_PACKET_BUFFER_H_
#define MODULES_AUDIO_CODING_NETEQ_PACKET_BUFFER_H_
-#include "api/optional.h"
+#include "absl/types/optional.h"
#include "modules/audio_coding/neteq/decoder_database.h"
#include "modules/audio_coding/neteq/packet.h"
#include "modules/include/module_common_types.h"
@@ -66,8 +66,8 @@
virtual int InsertPacketList(
PacketList* packet_list,
const DecoderDatabase& decoder_database,
- rtc::Optional<uint8_t>* current_rtp_payload_type,
- rtc::Optional<uint8_t>* current_cng_rtp_payload_type,
+ absl::optional<uint8_t>* current_rtp_payload_type,
+ absl::optional<uint8_t>* current_cng_rtp_payload_type,
StatisticsCalculator* stats);
// Gets the timestamp for the first packet in the buffer and writes it to the
@@ -90,7 +90,7 @@
// Extracts the first packet in the buffer and returns it.
// Returns an empty optional if the buffer is empty.
- virtual rtc::Optional<Packet> GetNextPacket();
+ virtual absl::optional<Packet> GetNextPacket();
// Discards the first packet in the buffer. The packet is deleted.
// Returns PacketBuffer::kBufferEmpty if the buffer is empty,
diff --git a/modules/audio_coding/neteq/packet_buffer_unittest.cc b/modules/audio_coding/neteq/packet_buffer_unittest.cc
index 1aaed8b..cb40d7d 100644
--- a/modules/audio_coding/neteq/packet_buffer_unittest.cc
+++ b/modules/audio_coding/neteq/packet_buffer_unittest.cc
@@ -176,21 +176,21 @@
MockDecoderDatabase decoder_database;
auto factory = CreateBuiltinAudioDecoderFactory();
const DecoderDatabase::DecoderInfo info(NetEqDecoder::kDecoderPCMu,
- rtc::nullopt, factory);
+ absl::nullopt, factory);
EXPECT_CALL(decoder_database, GetDecoderInfo(0))
.WillRepeatedly(Return(&info));
StrictMock<MockStatisticsCalculator> mock_stats;
- rtc::Optional<uint8_t> current_pt;
- rtc::Optional<uint8_t> current_cng_pt;
+ absl::optional<uint8_t> current_pt;
+ absl::optional<uint8_t> current_cng_pt;
EXPECT_EQ(PacketBuffer::kOK,
buffer.InsertPacketList(&list, decoder_database, ¤t_pt,
¤t_cng_pt, &mock_stats));
EXPECT_TRUE(list.empty()); // The PacketBuffer should have depleted the list.
EXPECT_EQ(10u, buffer.NumPacketsInBuffer());
EXPECT_EQ(0, current_pt); // Current payload type changed to 0.
- EXPECT_EQ(rtc::nullopt, current_cng_pt); // CNG payload type not changed.
+ EXPECT_EQ(absl::nullopt, current_cng_pt); // CNG payload type not changed.
buffer.Flush(); // Clean up.
@@ -221,25 +221,25 @@
MockDecoderDatabase decoder_database;
auto factory = CreateBuiltinAudioDecoderFactory();
const DecoderDatabase::DecoderInfo info0(NetEqDecoder::kDecoderPCMu,
- rtc::nullopt, factory);
+ absl::nullopt, factory);
EXPECT_CALL(decoder_database, GetDecoderInfo(0))
.WillRepeatedly(Return(&info0));
const DecoderDatabase::DecoderInfo info1(NetEqDecoder::kDecoderPCMa,
- rtc::nullopt, factory);
+ absl::nullopt, factory);
EXPECT_CALL(decoder_database, GetDecoderInfo(1))
.WillRepeatedly(Return(&info1));
StrictMock<MockStatisticsCalculator> mock_stats;
- rtc::Optional<uint8_t> current_pt;
- rtc::Optional<uint8_t> current_cng_pt;
+ absl::optional<uint8_t> current_pt;
+ absl::optional<uint8_t> current_cng_pt;
EXPECT_EQ(PacketBuffer::kFlushed,
buffer.InsertPacketList(&list, decoder_database, ¤t_pt,
¤t_cng_pt, &mock_stats));
EXPECT_TRUE(list.empty()); // The PacketBuffer should have depleted the list.
EXPECT_EQ(1u, buffer.NumPacketsInBuffer()); // Only the last packet.
EXPECT_EQ(1, current_pt); // Current payload type changed to 1.
- EXPECT_EQ(rtc::nullopt, current_cng_pt); // CNG payload type not changed.
+ EXPECT_EQ(absl::nullopt, current_cng_pt); // CNG payload type not changed.
buffer.Flush(); // Clean up.
@@ -313,7 +313,7 @@
EXPECT_EQ(kExpectPacketsInBuffer, buffer.NumPacketsInBuffer());
for (size_t i = 0; i < kExpectPacketsInBuffer; ++i) {
- const rtc::Optional<Packet> packet = buffer.GetNextPacket();
+ const absl::optional<Packet> packet = buffer.GetNextPacket();
EXPECT_EQ(packet, expect_order[i]); // Compare contents.
}
EXPECT_TRUE(buffer.Empty());
@@ -408,11 +408,11 @@
MockDecoderDatabase decoder_database;
auto factory = CreateBuiltinAudioDecoderFactory();
const DecoderDatabase::DecoderInfo info(NetEqDecoder::kDecoderPCMu,
- rtc::nullopt, factory);
+ absl::nullopt, factory);
EXPECT_CALL(decoder_database, GetDecoderInfo(0))
.WillRepeatedly(Return(&info));
- rtc::Optional<uint8_t> current_pt;
- rtc::Optional<uint8_t> current_cng_pt;
+ absl::optional<uint8_t> current_pt;
+ absl::optional<uint8_t> current_cng_pt;
StrictMock<MockStatisticsCalculator> mock_stats;
@@ -424,7 +424,7 @@
// Extract them and make sure that come out in the right order.
uint32_t current_ts = start_ts;
for (int i = 0; i < 10; ++i) {
- const rtc::Optional<Packet> packet = buffer.GetNextPacket();
+ const absl::optional<Packet> packet = buffer.GetNextPacket();
ASSERT_TRUE(packet);
EXPECT_EQ(current_ts, packet->timestamp);
current_ts += ts_increment;
@@ -448,11 +448,11 @@
MockDecoderDatabase decoder_database;
auto factory = CreateBuiltinAudioDecoderFactory();
const DecoderDatabase::DecoderInfo info_cng(NetEqDecoder::kDecoderCNGnb,
- rtc::nullopt, factory);
+ absl::nullopt, factory);
EXPECT_CALL(decoder_database, GetDecoderInfo(kCngPt))
.WillRepeatedly(Return(&info_cng));
const DecoderDatabase::DecoderInfo info_speech(NetEqDecoder::kDecoderPCM16Bwb,
- rtc::nullopt, factory);
+ absl::nullopt, factory);
EXPECT_CALL(decoder_database, GetDecoderInfo(kSpeechPt))
.WillRepeatedly(Return(&info_speech));
@@ -460,8 +460,8 @@
PacketGenerator gen(0, 0, kCngPt, 10);
PacketList list;
list.push_back(gen.NextPacket(kPayloadLen));
- rtc::Optional<uint8_t> current_pt;
- rtc::Optional<uint8_t> current_cng_pt;
+ absl::optional<uint8_t> current_pt;
+ absl::optional<uint8_t> current_cng_pt;
StrictMock<MockStatisticsCalculator> mock_stats;
@@ -472,7 +472,7 @@
EXPECT_EQ(1u, buffer.NumPacketsInBuffer());
ASSERT_TRUE(buffer.PeekNextPacket());
EXPECT_EQ(kCngPt, buffer.PeekNextPacket()->payload_type);
- EXPECT_EQ(current_pt, rtc::nullopt); // Current payload type not set.
+ EXPECT_EQ(current_pt, absl::nullopt); // Current payload type not set.
EXPECT_EQ(kCngPt, current_cng_pt); // CNG payload type set.
// Insert second packet, which is wide-band speech.
@@ -492,7 +492,7 @@
EXPECT_EQ(kSpeechPt, buffer.PeekNextPacket()->payload_type);
EXPECT_EQ(kSpeechPt, current_pt); // Current payload type set.
- EXPECT_EQ(rtc::nullopt, current_cng_pt); // CNG payload type reset.
+ EXPECT_EQ(absl::nullopt, current_cng_pt); // CNG payload type reset.
buffer.Flush(); // Clean up.
EXPECT_CALL(decoder_database, Die()); // Called when object is deleted.
@@ -550,11 +550,11 @@
MockDecoderDatabase decoder_database;
auto factory = CreateBuiltinAudioDecoderFactory();
const DecoderDatabase::DecoderInfo info(NetEqDecoder::kDecoderPCMu,
- rtc::nullopt, factory);
+ absl::nullopt, factory);
EXPECT_CALL(decoder_database, GetDecoderInfo(0))
.WillRepeatedly(Return(&info));
- rtc::Optional<uint8_t> current_pt;
- rtc::Optional<uint8_t> current_cng_pt;
+ absl::optional<uint8_t> current_pt;
+ absl::optional<uint8_t> current_cng_pt;
EXPECT_EQ(PacketBuffer::kInvalidPacket,
buffer->InsertPacketList(&list, decoder_database, ¤t_pt,
¤t_cng_pt, &mock_stats));
diff --git a/modules/audio_coding/neteq/red_payload_splitter_unittest.cc b/modules/audio_coding/neteq/red_payload_splitter_unittest.cc
index 077c8ea..c3d9f33 100644
--- a/modules/audio_coding/neteq/red_payload_splitter_unittest.cc
+++ b/modules/audio_coding/neteq/red_payload_splitter_unittest.cc
@@ -299,7 +299,7 @@
// easier to just register the payload types and let the actual implementation
// do its job.
DecoderDatabase decoder_database(
- new rtc::RefCountedObject<MockAudioDecoderFactory>, rtc::nullopt);
+ new rtc::RefCountedObject<MockAudioDecoderFactory>, absl::nullopt);
decoder_database.RegisterPayload(0, NetEqDecoder::kDecoderCNGnb, "cng-nb");
decoder_database.RegisterPayload(1, NetEqDecoder::kDecoderPCMu, "pcmu");
decoder_database.RegisterPayload(2, NetEqDecoder::kDecoderAVT, "avt");
diff --git a/modules/audio_coding/neteq/timestamp_scaler_unittest.cc b/modules/audio_coding/neteq/timestamp_scaler_unittest.cc
index 1a7b71a..eeaf772 100644
--- a/modules/audio_coding/neteq/timestamp_scaler_unittest.cc
+++ b/modules/audio_coding/neteq/timestamp_scaler_unittest.cc
@@ -26,7 +26,7 @@
auto factory = CreateBuiltinAudioDecoderFactory();
// Use PCMu, because it doesn't use scaled timestamps.
const DecoderDatabase::DecoderInfo info(NetEqDecoder::kDecoderPCMu,
- rtc::nullopt, factory);
+ absl::nullopt, factory);
static const uint8_t kRtpPayloadType = 0;
EXPECT_CALL(db, GetDecoderInfo(kRtpPayloadType))
.WillRepeatedly(Return(&info));
@@ -48,7 +48,7 @@
auto factory = CreateBuiltinAudioDecoderFactory();
// Use PCMu, because it doesn't use scaled timestamps.
const DecoderDatabase::DecoderInfo info(NetEqDecoder::kDecoderPCMu,
- rtc::nullopt, factory);
+ absl::nullopt, factory);
static const uint8_t kRtpPayloadType = 0;
EXPECT_CALL(db, GetDecoderInfo(kRtpPayloadType))
.WillRepeatedly(Return(&info));
@@ -75,7 +75,7 @@
auto factory = CreateBuiltinAudioDecoderFactory();
// Use G722, which has a factor 2 scaling.
const DecoderDatabase::DecoderInfo info(NetEqDecoder::kDecoderG722,
- rtc::nullopt, factory);
+ absl::nullopt, factory);
static const uint8_t kRtpPayloadType = 17;
EXPECT_CALL(db, GetDecoderInfo(kRtpPayloadType))
.WillRepeatedly(Return(&info));
@@ -101,7 +101,7 @@
auto factory = CreateBuiltinAudioDecoderFactory();
// Use G722, which has a factor 2 scaling.
const DecoderDatabase::DecoderInfo info(NetEqDecoder::kDecoderG722,
- rtc::nullopt, factory);
+ absl::nullopt, factory);
static const uint8_t kRtpPayloadType = 17;
EXPECT_CALL(db, GetDecoderInfo(kRtpPayloadType))
.WillRepeatedly(Return(&info));
@@ -131,9 +131,9 @@
auto factory = CreateBuiltinAudioDecoderFactory();
// Use G722, which has a factor 2 scaling.
const DecoderDatabase::DecoderInfo info_g722(NetEqDecoder::kDecoderG722,
- rtc::nullopt, factory);
+ absl::nullopt, factory);
const DecoderDatabase::DecoderInfo info_cng(NetEqDecoder::kDecoderCNGwb,
- rtc::nullopt, factory);
+ absl::nullopt, factory);
static const uint8_t kRtpPayloadTypeG722 = 17;
static const uint8_t kRtpPayloadTypeCng = 13;
EXPECT_CALL(db, GetDecoderInfo(kRtpPayloadTypeG722))
@@ -175,7 +175,7 @@
auto factory = CreateBuiltinAudioDecoderFactory();
// Use G722, which has a factor 2 scaling.
const DecoderDatabase::DecoderInfo info(NetEqDecoder::kDecoderG722,
- rtc::nullopt, factory);
+ absl::nullopt, factory);
static const uint8_t kRtpPayloadType = 17;
EXPECT_CALL(db, GetDecoderInfo(kRtpPayloadType))
.WillRepeatedly(Return(&info));
@@ -205,7 +205,7 @@
auto factory = CreateBuiltinAudioDecoderFactory();
// Use G722, which has a factor 2 scaling.
const DecoderDatabase::DecoderInfo info(NetEqDecoder::kDecoderG722,
- rtc::nullopt, factory);
+ absl::nullopt, factory);
static const uint8_t kRtpPayloadType = 17;
EXPECT_CALL(db, GetDecoderInfo(kRtpPayloadType))
.WillRepeatedly(Return(&info));
@@ -239,7 +239,7 @@
auto factory = CreateBuiltinAudioDecoderFactory();
// Use G722, which has a factor 2 scaling.
const DecoderDatabase::DecoderInfo info(NetEqDecoder::kDecoderG722,
- rtc::nullopt, factory);
+ absl::nullopt, factory);
static const uint8_t kRtpPayloadType = 17;
EXPECT_CALL(db, GetDecoderInfo(kRtpPayloadType))
.WillRepeatedly(Return(&info));
@@ -280,7 +280,7 @@
MockDecoderDatabase db;
auto factory = CreateBuiltinAudioDecoderFactory();
const DecoderDatabase::DecoderInfo info(NetEqDecoder::kDecoderOpus,
- rtc::nullopt, factory);
+ absl::nullopt, factory);
static const uint8_t kRtpPayloadType = 17;
EXPECT_CALL(db, GetDecoderInfo(kRtpPayloadType))
.WillRepeatedly(Return(&info));
diff --git a/modules/audio_coding/neteq/tools/encode_neteq_input.cc b/modules/audio_coding/neteq/tools/encode_neteq_input.cc
index a8d1bdf..f0d5b6f 100644
--- a/modules/audio_coding/neteq/tools/encode_neteq_input.cc
+++ b/modules/audio_coding/neteq/tools/encode_neteq_input.cc
@@ -27,12 +27,12 @@
CreatePacket();
}
-rtc::Optional<int64_t> EncodeNetEqInput::NextPacketTime() const {
+absl::optional<int64_t> EncodeNetEqInput::NextPacketTime() const {
RTC_DCHECK(packet_data_);
return static_cast<int64_t>(packet_data_->time_ms);
}
-rtc::Optional<int64_t> EncodeNetEqInput::NextOutputEventTime() const {
+absl::optional<int64_t> EncodeNetEqInput::NextOutputEventTime() const {
return next_output_event_ms_;
}
@@ -50,7 +50,7 @@
next_output_event_ms_ += kOutputPeriodMs;
}
-rtc::Optional<RTPHeader> EncodeNetEqInput::NextHeader() const {
+absl::optional<RTPHeader> EncodeNetEqInput::NextHeader() const {
RTC_DCHECK(packet_data_);
return packet_data_->header;
}
diff --git a/modules/audio_coding/neteq/tools/encode_neteq_input.h b/modules/audio_coding/neteq/tools/encode_neteq_input.h
index 13b39b3..a75262b 100644
--- a/modules/audio_coding/neteq/tools/encode_neteq_input.h
+++ b/modules/audio_coding/neteq/tools/encode_neteq_input.h
@@ -36,9 +36,9 @@
std::unique_ptr<AudioEncoder> encoder,
int64_t input_duration_ms);
- rtc::Optional<int64_t> NextPacketTime() const override;
+ absl::optional<int64_t> NextPacketTime() const override;
- rtc::Optional<int64_t> NextOutputEventTime() const override;
+ absl::optional<int64_t> NextOutputEventTime() const override;
std::unique_ptr<PacketData> PopPacket() override;
@@ -48,7 +48,7 @@
return next_output_event_ms_ <= input_duration_ms_;
}
- rtc::Optional<RTPHeader> NextHeader() const override;
+ absl::optional<RTPHeader> NextHeader() const override;
private:
static constexpr int64_t kOutputPeriodMs = 10;
diff --git a/modules/audio_coding/neteq/tools/fake_decode_from_file.h b/modules/audio_coding/neteq/tools/fake_decode_from_file.h
index 7aa8e6e..01d3ac0 100644
--- a/modules/audio_coding/neteq/tools/fake_decode_from_file.h
+++ b/modules/audio_coding/neteq/tools/fake_decode_from_file.h
@@ -13,9 +13,9 @@
#include <memory>
+#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/audio_codecs/audio_decoder.h"
-#include "api/optional.h"
#include "modules/audio_coding/neteq/tools/input_audio_file.h"
namespace webrtc {
@@ -61,7 +61,7 @@
private:
std::unique_ptr<InputAudioFile> input_;
- rtc::Optional<uint32_t> next_timestamp_from_input_;
+ absl::optional<uint32_t> next_timestamp_from_input_;
const int sample_rate_hz_;
const bool stereo_;
size_t last_decoded_length_ = 0;
diff --git a/modules/audio_coding/neteq/tools/neteq_delay_analyzer.cc b/modules/audio_coding/neteq/tools/neteq_delay_analyzer.cc
index ba0b217..2b21520 100644
--- a/modules/audio_coding/neteq/tools/neteq_delay_analyzer.cc
+++ b/modules/audio_coding/neteq/tools/neteq_delay_analyzer.cc
@@ -101,8 +101,8 @@
std::vector<float>* send_time_s,
std::vector<float>* arrival_delay_ms,
std::vector<float>* corrected_arrival_delay_ms,
- std::vector<rtc::Optional<float>>* playout_delay_ms,
- std::vector<rtc::Optional<float>>* target_delay_ms) const {
+ std::vector<absl::optional<float>>* playout_delay_ms,
+ std::vector<absl::optional<float>>* target_delay_ms) const {
if (get_audio_time_ms_.empty()) {
return;
}
@@ -167,8 +167,8 @@
target_delay_ms->push_back(target);
} else {
// This packet was never decoded. Mark target and playout delays as empty.
- playout_delay_ms->push_back(rtc::nullopt);
- target_delay_ms->push_back(rtc::nullopt);
+ playout_delay_ms->push_back(absl::nullopt);
+ target_delay_ms->push_back(absl::nullopt);
}
}
RTC_DCHECK(data_it == data_.end());
@@ -182,8 +182,8 @@
std::vector<float> send_time_s;
std::vector<float> arrival_delay_ms;
std::vector<float> corrected_arrival_delay_ms;
- std::vector<rtc::Optional<float>> playout_delay_ms;
- std::vector<rtc::Optional<float>> target_delay_ms;
+ std::vector<absl::optional<float>> playout_delay_ms;
+ std::vector<absl::optional<float>> target_delay_ms;
CreateGraphs(&send_time_s, &arrival_delay_ms, &corrected_arrival_delay_ms,
&playout_delay_ms, &target_delay_ms);
@@ -258,8 +258,8 @@
std::vector<float> send_time_s;
std::vector<float> arrival_delay_ms;
std::vector<float> corrected_arrival_delay_ms;
- std::vector<rtc::Optional<float>> playout_delay_ms;
- std::vector<rtc::Optional<float>> target_delay_ms;
+ std::vector<absl::optional<float>> playout_delay_ms;
+ std::vector<absl::optional<float>> target_delay_ms;
CreateGraphs(&send_time_s, &arrival_delay_ms, &corrected_arrival_delay_ms,
&playout_delay_ms, &target_delay_ms);
diff --git a/modules/audio_coding/neteq/tools/neteq_delay_analyzer.h b/modules/audio_coding/neteq/tools/neteq_delay_analyzer.h
index e6d6913..b74ff77 100644
--- a/modules/audio_coding/neteq/tools/neteq_delay_analyzer.h
+++ b/modules/audio_coding/neteq/tools/neteq_delay_analyzer.h
@@ -16,7 +16,7 @@
#include <string>
#include <vector>
-#include "api/optional.h"
+#include "absl/types/optional.h"
#include "modules/audio_coding/neteq/tools/neteq_input.h"
#include "modules/audio_coding/neteq/tools/neteq_test.h"
#include "typedefs.h" // NOLINT(build/include)
@@ -40,8 +40,8 @@
void CreateGraphs(std::vector<float>* send_times_s,
std::vector<float>* arrival_delay_ms,
std::vector<float>* corrected_arrival_delay_ms,
- std::vector<rtc::Optional<float>>* playout_delay_ms,
- std::vector<rtc::Optional<float>>* target_delay_ms) const;
+ std::vector<absl::optional<float>>* playout_delay_ms,
+ std::vector<absl::optional<float>>* target_delay_ms) const;
// Creates a matlab script with file name script_name. When executed in
// Matlab, the script will generate graphs with the same timing information
@@ -57,10 +57,10 @@
struct TimingData {
explicit TimingData(double at) : arrival_time_ms(at) {}
double arrival_time_ms;
- rtc::Optional<int64_t> decode_get_audio_count;
- rtc::Optional<int64_t> sync_delay_ms;
- rtc::Optional<int> target_delay_ms;
- rtc::Optional<int> current_delay_ms;
+ absl::optional<int64_t> decode_get_audio_count;
+ absl::optional<int64_t> sync_delay_ms;
+ absl::optional<int> target_delay_ms;
+ absl::optional<int> current_delay_ms;
};
std::map<uint32_t, TimingData> data_;
std::vector<int64_t> get_audio_time_ms_;
diff --git a/modules/audio_coding/neteq/tools/neteq_input.h b/modules/audio_coding/neteq/tools/neteq_input.h
index 68c076e..5e2cbd2 100644
--- a/modules/audio_coding/neteq/tools/neteq_input.h
+++ b/modules/audio_coding/neteq/tools/neteq_input.h
@@ -15,7 +15,7 @@
#include <memory>
#include <string>
-#include "api/optional.h"
+#include "absl/types/optional.h"
#include "common_types.h" // NOLINT(build/include)
#include "modules/audio_coding/neteq/tools/packet.h"
#include "modules/audio_coding/neteq/tools/packet_source.h"
@@ -39,22 +39,22 @@
// Returns at what time (in ms) NetEq::InsertPacket should be called next, or
// empty if the source is out of packets.
- virtual rtc::Optional<int64_t> NextPacketTime() const = 0;
+ virtual absl::optional<int64_t> NextPacketTime() const = 0;
// Returns at what time (in ms) NetEq::GetAudio should be called next, or
// empty if no more output events are available.
- virtual rtc::Optional<int64_t> NextOutputEventTime() const = 0;
+ virtual absl::optional<int64_t> NextOutputEventTime() const = 0;
// Returns the time (in ms) for the next event from either NextPacketTime()
// or NextOutputEventTime(), or empty if both are out of events.
- rtc::Optional<int64_t> NextEventTime() const {
+ absl::optional<int64_t> NextEventTime() const {
const auto a = NextPacketTime();
const auto b = NextOutputEventTime();
// Return the minimum of non-empty |a| and |b|, or empty if both are empty.
if (a) {
return b ? std::min(*a, *b) : a;
}
- return b ? b : rtc::nullopt;
+ return b ? b : absl::nullopt;
}
// Returns the next packet to be inserted into NetEq. The packet following the
@@ -75,7 +75,7 @@
// Returns the RTP header for the next packet, i.e., the packet that will be
// delivered next by PopPacket().
- virtual rtc::Optional<RTPHeader> NextHeader() const = 0;
+ virtual absl::optional<RTPHeader> NextHeader() const = 0;
};
} // namespace test
diff --git a/modules/audio_coding/neteq/tools/neteq_packet_source_input.cc b/modules/audio_coding/neteq/tools/neteq_packet_source_input.cc
index 79a939d..05ee806 100644
--- a/modules/audio_coding/neteq/tools/neteq_packet_source_input.cc
+++ b/modules/audio_coding/neteq/tools/neteq_packet_source_input.cc
@@ -22,15 +22,14 @@
NetEqPacketSourceInput::NetEqPacketSourceInput() : next_output_event_ms_(0) {}
-rtc::Optional<int64_t> NetEqPacketSourceInput::NextPacketTime() const {
+absl::optional<int64_t> NetEqPacketSourceInput::NextPacketTime() const {
return packet_
- ? rtc::Optional<int64_t>(static_cast<int64_t>(packet_->time_ms()))
- : rtc::nullopt;
+ ? absl::optional<int64_t>(static_cast<int64_t>(packet_->time_ms()))
+ : absl::nullopt;
}
-rtc::Optional<RTPHeader> NetEqPacketSourceInput::NextHeader() const {
- return packet_ ? rtc::Optional<RTPHeader>(packet_->header())
- : rtc::nullopt;
+absl::optional<RTPHeader> NetEqPacketSourceInput::NextHeader() const {
+ return packet_ ? absl::optional<RTPHeader>(packet_->header()) : absl::nullopt;
}
void NetEqPacketSourceInput::LoadNextPacket() {
@@ -75,7 +74,7 @@
LoadNextPacket();
}
-rtc::Optional<int64_t> NetEqRtpDumpInput::NextOutputEventTime() const {
+absl::optional<int64_t> NetEqRtpDumpInput::NextOutputEventTime() const {
return next_output_event_ms_;
}
@@ -84,7 +83,7 @@
*next_output_event_ms_ += kOutputPeriodMs;
}
if (!NextPacketTime()) {
- next_output_event_ms_ = rtc::nullopt;
+ next_output_event_ms_ = absl::nullopt;
}
}
@@ -102,14 +101,14 @@
AdvanceOutputEvent();
}
-rtc::Optional<int64_t> NetEqEventLogInput::NextOutputEventTime() const {
+absl::optional<int64_t> NetEqEventLogInput::NextOutputEventTime() const {
return next_output_event_ms_;
}
void NetEqEventLogInput::AdvanceOutputEvent() {
next_output_event_ms_ = source_->NextAudioOutputEventMs();
if (*next_output_event_ms_ == std::numeric_limits<int64_t>::max()) {
- next_output_event_ms_ = rtc::nullopt;
+ next_output_event_ms_ = absl::nullopt;
}
}
diff --git a/modules/audio_coding/neteq/tools/neteq_packet_source_input.h b/modules/audio_coding/neteq/tools/neteq_packet_source_input.h
index 70fe111..302a4e1 100644
--- a/modules/audio_coding/neteq/tools/neteq_packet_source_input.h
+++ b/modules/audio_coding/neteq/tools/neteq_packet_source_input.h
@@ -29,9 +29,9 @@
using RtpHeaderExtensionMap = std::map<int, webrtc::RTPExtensionType>;
NetEqPacketSourceInput();
- rtc::Optional<int64_t> NextPacketTime() const override;
+ absl::optional<int64_t> NextPacketTime() const override;
std::unique_ptr<PacketData> PopPacket() override;
- rtc::Optional<RTPHeader> NextHeader() const override;
+ absl::optional<RTPHeader> NextHeader() const override;
bool ended() const override { return !next_output_event_ms_; }
void SelectSsrc(uint32_t);
@@ -39,7 +39,7 @@
virtual PacketSource* source() = 0;
void LoadNextPacket();
- rtc::Optional<int64_t> next_output_event_ms_;
+ absl::optional<int64_t> next_output_event_ms_;
private:
std::unique_ptr<Packet> packet_;
@@ -51,7 +51,7 @@
NetEqRtpDumpInput(const std::string& file_name,
const RtpHeaderExtensionMap& hdr_ext_map);
- rtc::Optional<int64_t> NextOutputEventTime() const override;
+ absl::optional<int64_t> NextOutputEventTime() const override;
void AdvanceOutputEvent() override;
protected:
@@ -70,7 +70,7 @@
NetEqEventLogInput(const std::string& file_name,
const RtpHeaderExtensionMap& hdr_ext_map);
- rtc::Optional<int64_t> NextOutputEventTime() const override;
+ absl::optional<int64_t> NextOutputEventTime() const override;
void AdvanceOutputEvent() override;
protected:
diff --git a/modules/audio_coding/neteq/tools/neteq_replacement_input.cc b/modules/audio_coding/neteq/tools/neteq_replacement_input.cc
index 6c846c0..6aa7581 100644
--- a/modules/audio_coding/neteq/tools/neteq_replacement_input.cc
+++ b/modules/audio_coding/neteq/tools/neteq_replacement_input.cc
@@ -31,13 +31,13 @@
RTC_CHECK(packet_);
}
-rtc::Optional<int64_t> NetEqReplacementInput::NextPacketTime() const {
+absl::optional<int64_t> NetEqReplacementInput::NextPacketTime() const {
return packet_
- ? rtc::Optional<int64_t>(static_cast<int64_t>(packet_->time_ms))
- : rtc::nullopt;
+ ? absl::optional<int64_t>(static_cast<int64_t>(packet_->time_ms))
+ : absl::nullopt;
}
-rtc::Optional<int64_t> NetEqReplacementInput::NextOutputEventTime() const {
+absl::optional<int64_t> NetEqReplacementInput::NextOutputEventTime() const {
return source_->NextOutputEventTime();
}
@@ -56,7 +56,7 @@
return source_->ended();
}
-rtc::Optional<RTPHeader> NetEqReplacementInput::NextHeader() const {
+absl::optional<RTPHeader> NetEqReplacementInput::NextHeader() const {
return source_->NextHeader();
}
@@ -82,7 +82,7 @@
return;
}
- rtc::Optional<RTPHeader> next_hdr = source_->NextHeader();
+ absl::optional<RTPHeader> next_hdr = source_->NextHeader();
RTC_DCHECK(next_hdr);
uint8_t payload[12];
RTC_DCHECK_LE(last_frame_size_timestamps_, 120 * 48);
diff --git a/modules/audio_coding/neteq/tools/neteq_replacement_input.h b/modules/audio_coding/neteq/tools/neteq_replacement_input.h
index 3a89399..1113001 100644
--- a/modules/audio_coding/neteq/tools/neteq_replacement_input.h
+++ b/modules/audio_coding/neteq/tools/neteq_replacement_input.h
@@ -28,12 +28,12 @@
const std::set<uint8_t>& comfort_noise_types,
const std::set<uint8_t>& forbidden_types);
- rtc::Optional<int64_t> NextPacketTime() const override;
- rtc::Optional<int64_t> NextOutputEventTime() const override;
+ absl::optional<int64_t> NextPacketTime() const override;
+ absl::optional<int64_t> NextOutputEventTime() const override;
std::unique_ptr<PacketData> PopPacket() override;
void AdvanceOutputEvent() override;
bool ended() const override;
- rtc::Optional<RTPHeader> NextHeader() const override;
+ absl::optional<RTPHeader> NextHeader() const override;
private:
void ReplacePacket();
diff --git a/modules/audio_coding/neteq/tools/neteq_rtpplay.cc b/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
index 4d3bd0f..d69b1a7 100644
--- a/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
+++ b/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
@@ -206,7 +206,7 @@
PrintCodecMappingEntry(NetEqDecoder::kDecoderCNGswb48kHz, FLAG_cn_swb48);
}
-rtc::Optional<int> CodecSampleRate(uint8_t payload_type) {
+absl::optional<int> CodecSampleRate(uint8_t payload_type) {
if (payload_type == FLAG_pcmu || payload_type == FLAG_pcma ||
payload_type == FLAG_ilbc || payload_type == FLAG_pcm16b ||
payload_type == FLAG_cn_nb || payload_type == FLAG_avt)
@@ -223,7 +223,7 @@
return 48000;
if (payload_type == FLAG_red)
return 0;
- return rtc::nullopt;
+ return absl::nullopt;
}
// A callback class which prints whenver the inserted packet stream changes
@@ -253,7 +253,7 @@
private:
NetEqPostInsertPacket* other_callback_;
- rtc::Optional<uint32_t> last_ssrc_;
+ absl::optional<uint32_t> last_ssrc_;
};
int RunTest(int argc, char* argv[]) {
@@ -340,9 +340,9 @@
}
// Check the sample rate.
- rtc::Optional<int> sample_rate_hz;
+ absl::optional<int> sample_rate_hz;
std::set<std::pair<int, uint32_t>> discarded_pt_and_ssrc;
- while (rtc::Optional<RTPHeader> first_rtp_header = input->NextHeader()) {
+ while (absl::optional<RTPHeader> first_rtp_header = input->NextHeader()) {
RTC_DCHECK(first_rtp_header);
sample_rate_hz = CodecSampleRate(first_rtp_header->payloadType);
if (sample_rate_hz) {
diff --git a/modules/audio_coding/test/PCMFile.h b/modules/audio_coding/test/PCMFile.h
index 84386dc..05b9828 100644
--- a/modules/audio_coding/test/PCMFile.h
+++ b/modules/audio_coding/test/PCMFile.h
@@ -16,8 +16,8 @@
#include <string>
+#include "absl/types/optional.h"
#include "api/audio/audio_frame.h"
-#include "api/optional.h"
#include "typedefs.h" // NOLINT(build/include)
namespace webrtc {
@@ -67,7 +67,7 @@
uint32_t timestamp_;
bool read_stereo_;
bool save_stereo_;
- rtc::Optional<int> num_10ms_blocks_to_read_;
+ absl::optional<int> num_10ms_blocks_to_read_;
int blocks_read_ = 0;
};
diff --git a/modules/audio_coding/test/delay_test.cc b/modules/audio_coding/test/delay_test.cc
index e1e9d71..532a8eb 100644
--- a/modules/audio_coding/test/delay_test.cc
+++ b/modules/audio_coding/test/delay_test.cc
@@ -217,7 +217,7 @@
audio_frame.data(),
audio_frame.samples_per_channel_ * audio_frame.num_channels_);
received_ts = channel_a2b_->LastInTimestamp();
- rtc::Optional<uint32_t> playout_timestamp = acm_b_->PlayoutTimestamp();
+ absl::optional<uint32_t> playout_timestamp = acm_b_->PlayoutTimestamp();
ASSERT_TRUE(playout_timestamp);
inst_delay_sec = static_cast<uint32_t>(received_ts - *playout_timestamp) /
static_cast<double>(encoding_sample_rate_hz_);
