diff --git a/audio/audio_send_stream.cc b/audio/audio_send_stream.cc index 1c0a32f86b..4e21b1f31d 100644 --- a/audio/audio_send_stream.cc +++ b/audio/audio_send_stream.cc @@ -143,7 +143,6 @@ AudioSendStream::AudioSendStream( std::unique_ptr channel_send) : clock_(clock), worker_queue_(rtp_transport->GetWorkerQueue()), - audio_send_side_bwe_(field_trial::IsEnabled("WebRTC-Audio-SendSideBwe")), allocate_audio_without_feedback_( field_trial::IsEnabled("WebRTC-Audio-ABWENoTWCC")), enable_audio_alr_probing_( @@ -289,7 +288,7 @@ void AudioSendStream::ConfigureStream( RtcpBandwidthObserver* bandwidth_observer = nullptr; - if (audio_send_side_bwe_ && !allocate_audio_without_feedback_ && + if (!allocate_audio_without_feedback_ && new_ids.transport_sequence_number != 0) { rtp_rtcp_module_->RegisterRtpHeaderExtension( TransportSequenceNumber::kUri, new_ids.transport_sequence_number); @@ -809,8 +808,7 @@ void AudioSendStream::ReconfigureBitrateObserver( if (config_.min_bitrate_bps == new_config.min_bitrate_bps && config_.max_bitrate_bps == new_config.max_bitrate_bps && config_.bitrate_priority == new_config.bitrate_priority && - (TransportSeqNumId(config_) == TransportSeqNumId(new_config) || - !audio_send_side_bwe_) && + TransportSeqNumId(config_) == TransportSeqNumId(new_config) && config_.audio_network_adaptor_config == new_config.audio_network_adaptor_config) { return; diff --git a/audio/audio_send_stream.h b/audio/audio_send_stream.h index 12fcb9f21a..1e6982e41f 100644 --- a/audio/audio_send_stream.h +++ b/audio/audio_send_stream.h @@ -155,7 +155,6 @@ class AudioSendStream final : public webrtc::AudioSendStream, rtc::RaceChecker audio_capture_race_checker_; rtc::TaskQueue* worker_queue_; - const bool audio_send_side_bwe_; const bool allocate_audio_without_feedback_; const bool force_no_audio_feedback_ = allocate_audio_without_feedback_; const bool enable_audio_alr_probing_; diff --git a/audio/audio_send_stream_tests.cc b/audio/audio_send_stream_tests.cc index d2ea99ce08..e3895039d8 100644 --- a/audio/audio_send_stream_tests.cc +++ b/audio/audio_send_stream_tests.cc @@ -188,17 +188,10 @@ class TransportWideSequenceNumberObserver : public AudioSendTest { }; TEST_F(AudioSendStreamCallTest, SendsTransportWideSequenceNumbersInFieldTrial) { - ScopedFieldTrials field_trials("WebRTC-Audio-SendSideBwe/Enabled/"); TransportWideSequenceNumberObserver test(/*expect_sequence_number=*/true); RunBaseTest(&test); } -TEST_F(AudioSendStreamCallTest, - DoesNotSendTransportWideSequenceNumbersPerDefault) { - TransportWideSequenceNumberObserver test(/*expect_sequence_number=*/false); - RunBaseTest(&test); -} - TEST_F(AudioSendStreamCallTest, SendDtmf) { static const uint8_t kDtmfPayloadType = 120; static const int kDtmfPayloadFrequency = 8000; diff --git a/audio/audio_send_stream_unittest.cc b/audio/audio_send_stream_unittest.cc index bfec59bf92..b91296e212 100644 --- a/audio/audio_send_stream_unittest.cc +++ b/audio/audio_send_stream_unittest.cc @@ -421,7 +421,6 @@ TEST(AudioSendStreamTest, SetMuted) { } TEST(AudioSendStreamTest, AudioBweCorrectObjectsOnChannelProxy) { - ScopedFieldTrials field_trials("WebRTC-Audio-SendSideBwe/Enabled/"); for (bool use_null_audio_processing : {false, true}) { ConfigHelper helper(true, true, use_null_audio_processing); auto send_stream = helper.CreateAudioSendStream(); @@ -523,14 +522,12 @@ TEST(AudioSendStreamTest, GetStatsAudioLevel) { TEST(AudioSendStreamTest, SendCodecAppliesAudioNetworkAdaptor) { for (bool use_null_audio_processing : {false, true}) { - ConfigHelper helper(false, true, use_null_audio_processing); + ConfigHelper helper(true, true, use_null_audio_processing); helper.config().send_codec_spec = AudioSendStream::Config::SendCodecSpec(0, kOpusFormat); const std::string kAnaConfigString = "abcde"; const std::string kAnaReconfigString = "12345"; - helper.config().rtp.extensions.push_back(RtpExtension( - RtpExtension::kTransportSequenceNumberUri, kTransportSequenceNumberId)); helper.config().audio_network_adaptor_config = kAnaConfigString; EXPECT_CALL(helper.mock_encoder_factory(), MakeAudioEncoderMock(_, _, _, _)) @@ -559,12 +556,10 @@ TEST(AudioSendStreamTest, SendCodecAppliesAudioNetworkAdaptor) { TEST(AudioSendStreamTest, AudioNetworkAdaptorReceivesOverhead) { for (bool use_null_audio_processing : {false, true}) { - ConfigHelper helper(false, true, use_null_audio_processing); + ConfigHelper helper(true, true, use_null_audio_processing); helper.config().send_codec_spec = AudioSendStream::Config::SendCodecSpec(0, kOpusFormat); const std::string kAnaConfigString = "abcde"; - helper.config().rtp.extensions.push_back(RtpExtension( - RtpExtension::kTransportSequenceNumberUri, kTransportSequenceNumberId)); EXPECT_CALL(helper.mock_encoder_factory(), MakeAudioEncoderMock(_, _, _, _)) .WillOnce(Invoke( @@ -647,7 +642,6 @@ TEST(AudioSendStreamTest, DoesNotPassHigherBitrateThanMaxBitrate) { } TEST(AudioSendStreamTest, SSBweTargetInRangeRespected) { - ScopedFieldTrials field_trials("WebRTC-Audio-SendSideBwe/Enabled/"); for (bool use_null_audio_processing : {false, true}) { ConfigHelper helper(true, true, use_null_audio_processing); auto send_stream = helper.CreateAudioSendStream(); @@ -666,7 +660,6 @@ TEST(AudioSendStreamTest, SSBweTargetInRangeRespected) { TEST(AudioSendStreamTest, SSBweFieldTrialMinRespected) { ScopedFieldTrials field_trials( - "WebRTC-Audio-SendSideBwe/Enabled/" "WebRTC-Audio-Allocation/min:6kbps,max:64kbps/"); for (bool use_null_audio_processing : {false, true}) { ConfigHelper helper(true, true, use_null_audio_processing); @@ -684,7 +677,6 @@ TEST(AudioSendStreamTest, SSBweFieldTrialMinRespected) { TEST(AudioSendStreamTest, SSBweFieldTrialMaxRespected) { ScopedFieldTrials field_trials( - "WebRTC-Audio-SendSideBwe/Enabled/" "WebRTC-Audio-Allocation/min:6kbps,max:64kbps/"); for (bool use_null_audio_processing : {false, true}) { ConfigHelper helper(true, true, use_null_audio_processing); @@ -702,8 +694,6 @@ TEST(AudioSendStreamTest, SSBweFieldTrialMaxRespected) { TEST(AudioSendStreamTest, SSBweWithOverhead) { ScopedFieldTrials field_trials( - "WebRTC-Audio-SendSideBwe/Enabled/" - "WebRTC-SendSideBwe-WithOverhead/Enabled/" "WebRTC-Audio-LegacyOverhead/Disabled/"); for (bool use_null_audio_processing : {false, true}) { ConfigHelper helper(true, true, use_null_audio_processing); @@ -725,8 +715,6 @@ TEST(AudioSendStreamTest, SSBweWithOverhead) { TEST(AudioSendStreamTest, SSBweWithOverheadMinRespected) { ScopedFieldTrials field_trials( - "WebRTC-Audio-SendSideBwe/Enabled/" - "WebRTC-SendSideBwe-WithOverhead/Enabled/" "WebRTC-Audio-LegacyOverhead/Disabled/" "WebRTC-Audio-Allocation/min:6kbps,max:64kbps/"); for (bool use_null_audio_processing : {false, true}) { @@ -747,8 +735,6 @@ TEST(AudioSendStreamTest, SSBweWithOverheadMinRespected) { TEST(AudioSendStreamTest, SSBweWithOverheadMaxRespected) { ScopedFieldTrials field_trials( - "WebRTC-Audio-SendSideBwe/Enabled/" - "WebRTC-SendSideBwe-WithOverhead/Enabled/" "WebRTC-Audio-LegacyOverhead/Disabled/" "WebRTC-Audio-Allocation/min:6kbps,max:64kbps/"); for (bool use_null_audio_processing : {false, true}) { @@ -808,7 +794,6 @@ TEST(AudioSendStreamTest, DontRecreateEncoder) { } TEST(AudioSendStreamTest, ReconfigureTransportCcResetsFirst) { - ScopedFieldTrials field_trials("WebRTC-Audio-SendSideBwe/Enabled/"); for (bool use_null_audio_processing : {false, true}) { ConfigHelper helper(false, true, use_null_audio_processing); auto send_stream = helper.CreateAudioSendStream(); diff --git a/audio/test/audio_bwe_integration_test.cc b/audio/test/audio_bwe_integration_test.cc index eed7acb8de..f9953955df 100644 --- a/audio/test/audio_bwe_integration_test.cc +++ b/audio/test/audio_bwe_integration_test.cc @@ -160,9 +160,6 @@ using AudioBweIntegrationTest = CallTest; // TODO(tschumim): This test is flaky when run on android and mac. Re-enable the // test for when the issue is fixed. TEST_F(AudioBweIntegrationTest, DISABLED_NoBandwidthDropAfterDtx) { - webrtc::test::ScopedFieldTrials override_field_trials( - "WebRTC-Audio-SendSideBwe/Enabled/" - "WebRTC-SendSideBwe-WithOverhead/Enabled/"); NoBandwidthDropAfterDtx test; RunBaseTest(&test); } diff --git a/call/rampup_tests.cc b/call/rampup_tests.cc index 89fbe3dde7..379f9dcf84 100644 --- a/call/rampup_tests.cc +++ b/call/rampup_tests.cc @@ -663,7 +663,6 @@ TEST_F(RampUpTest, DISABLED_UpDownUpTransportSequenceNumberPacketLoss) { UpDownUpAudioVideoTransportSequenceNumberRtx #endif TEST_F(RampUpTest, MAYBE_UpDownUpAudioVideoTransportSequenceNumberRtx) { - test::ScopedFieldTrials field_trials("WebRTC-Audio-SendSideBwe/Enabled/"); std::vector loss_rates = {0, 0, 0, 0}; RampUpDownUpTester test(3, 1, 0, kStartBitrateBps, RtpExtension::kTransportSequenceNumberUri, true, @@ -672,7 +671,6 @@ TEST_F(RampUpTest, MAYBE_UpDownUpAudioVideoTransportSequenceNumberRtx) { } TEST_F(RampUpTest, UpDownUpAudioTransportSequenceNumberRtx) { - test::ScopedFieldTrials field_trials("WebRTC-Audio-SendSideBwe/Enabled/"); std::vector loss_rates = {0, 0, 0, 0}; RampUpDownUpTester test(0, 1, 0, kStartBitrateBps, RtpExtension::kTransportSequenceNumberUri, true, diff --git a/media/engine/webrtc_voice_engine_unittest.cc b/media/engine/webrtc_voice_engine_unittest.cc index 6497f5e82b..87678be087 100644 --- a/media/engine/webrtc_voice_engine_unittest.cc +++ b/media/engine/webrtc_voice_engine_unittest.cc @@ -1344,8 +1344,6 @@ TEST_P(WebRtcVoiceEngineTestFake, SetAndGetRtpSendParameters) { // Test that max_bitrate_bps in send stream config gets updated correctly when // SetRtpSendParameters is called. TEST_P(WebRtcVoiceEngineTestFake, SetRtpSendParameterUpdatesMaxBitrate) { - webrtc::test::ScopedFieldTrials override_field_trials( - "WebRTC-Audio-SendSideBwe/Enabled/"); EXPECT_TRUE(SetupSendStream()); cricket::AudioSendParameters send_parameters; send_parameters.codecs.push_back(kOpusCodec); @@ -2127,17 +2125,7 @@ TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsCaseInsensitive) { EXPECT_TRUE(channel_->CanInsertDtmf()); } -class WebRtcVoiceEngineWithSendSideBweTest : public WebRtcVoiceEngineTestFake { - public: - WebRtcVoiceEngineWithSendSideBweTest() - : WebRtcVoiceEngineTestFake("WebRTC-Audio-SendSideBwe/Enabled/") {} -}; - -INSTANTIATE_TEST_SUITE_P(UnusedParameter, - WebRtcVoiceEngineWithSendSideBweTest, - ::testing::Values(true)); - -TEST_P(WebRtcVoiceEngineWithSendSideBweTest, +TEST_P(WebRtcVoiceEngineTestFake, SupportsTransportSequenceNumberHeaderExtension) { const std::vector header_extensions = GetDefaultEnabledRtpHeaderExtensions(*engine_); @@ -2530,9 +2518,7 @@ class WebRtcVoiceEngineWithSendSideBweWithOverheadTest public: WebRtcVoiceEngineWithSendSideBweWithOverheadTest() : WebRtcVoiceEngineTestFake( - "WebRTC-Audio-SendSideBwe/Enabled/WebRTC-Audio-Allocation/" - "min:6000bps,max:32000bps/WebRTC-SendSideBwe-WithOverhead/" - "Enabled/") {} + "WebRTC-Audio-Allocation/min:6000bps,max:32000bps/") {} }; // Test that we can set the outgoing SSRC properly. diff --git a/pc/scenario_tests/goog_cc_test.cc b/pc/scenario_tests/goog_cc_test.cc index fba617dd5c..4a996b8684 100644 --- a/pc/scenario_tests/goog_cc_test.cc +++ b/pc/scenario_tests/goog_cc_test.cc @@ -32,10 +32,7 @@ TEST(GoogCcPeerScenarioTest, MAYBE_NoBweChangeFromVideoUnmute) { // packets sizes. This will create a change in propagation time which might be // detected as an overuse. Using separate overuse detectors for audio and // video avoids the issue. - std::string audio_twcc_trials( - "WebRTC-Audio-SendSideBwe/Enabled/" // - "WebRTC-SendSideBwe-WithOverhead/Enabled/" // - "WebRTC-Audio-AlrProbing/Disabled/"); + std::string audio_twcc_trials("WebRTC-Audio-AlrProbing/Disabled/"); std::string separate_audio_video( "WebRTC-Bwe-SeparateAudioPackets/" "enabled:true,packet_threshold:15,time_threshold:1000ms/"); diff --git a/test/scenario/scenario_unittest.cc b/test/scenario/scenario_unittest.cc index 7c05ea39dd..177ac27373 100644 --- a/test/scenario/scenario_unittest.cc +++ b/test/scenario/scenario_unittest.cc @@ -146,7 +146,7 @@ TEST(ScenarioTest, RetransmitsVideoPacketsInAudioAndVideoCallWithSendSideBweAndLoss) { // Make sure audio packets are included in transport feedback. test::ScopedFieldTrials override_field_trials( - "WebRTC-Audio-SendSideBwe/Enabled/WebRTC-Audio-ABWENoTWCC/Disabled/"); + "WebRTC-Audio-ABWENoTWCC/Disabled/"); Scenario s; CallClientConfig call_client_config; diff --git a/video/end_to_end_tests/transport_feedback_tests.cc b/video/end_to_end_tests/transport_feedback_tests.cc index 9cfa7d14f4..a675d784bc 100644 --- a/video/end_to_end_tests/transport_feedback_tests.cc +++ b/video/end_to_end_tests/transport_feedback_tests.cc @@ -327,7 +327,6 @@ TEST_F(TransportFeedbackEndToEndTest, VideoTransportFeedbackNotConfigured) { } TEST_F(TransportFeedbackEndToEndTest, AudioReceivesTransportFeedback) { - test::ScopedFieldTrials field_trials("WebRTC-Audio-SendSideBwe/Enabled/"); TransportFeedbackTester test(true, 0, 1); RunBaseTest(&test); } @@ -435,7 +434,6 @@ TEST_F(TransportFeedbackEndToEndTest, } TEST_F(TransportFeedbackEndToEndTest, TransportSeqNumOnAudioAndVideo) { - test::ScopedFieldTrials field_trials("WebRTC-Audio-SendSideBwe/Enabled/"); static constexpr size_t kMinPacketsToWaitFor = 50; class TransportSequenceNumberTest : public test::EndToEndTest { public: