Default enable sending transport sequence numbers on audio packets.

This enables send side bandwidth estimation for audio and removes field
trial "WebRTC-Audio-SendSideBwe" which this was controlled through.

Transport-cc extension still needs to be negotiated.

Bug: webrtc:12222
Change-Id: Ie2268fad13703eeb0f0d38fcf484baaa29715b7c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/194142
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32681}

audio/audio_send_stream.cc

@@ -143,7 +143,6 @@ AudioSendStream::AudioSendStream(
     std::unique_ptr<voe::ChannelSendInterface> 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) ||
+      TransportSeqNumId(config_) == TransportSeqNumId(new_config) &&
-       !audio_send_side_bwe_) &&
       config_.audio_network_adaptor_config ==
           new_config.audio_network_adaptor_config) {
     return;

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_;

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;

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();

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);
 }

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<int> 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<int> loss_rates = {0, 0, 0, 0};
   RampUpDownUpTester test(0, 1, 0, kStartBitrateBps,
                           RtpExtension::kTransportSequenceNumberUri, true,

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 {
+TEST_P(WebRtcVoiceEngineTestFake,
- public:
-  WebRtcVoiceEngineWithSendSideBweTest()
-      : WebRtcVoiceEngineTestFake("WebRTC-Audio-SendSideBwe/Enabled/") {}
-};
-
-INSTANTIATE_TEST_SUITE_P(UnusedParameter,
-                         WebRtcVoiceEngineWithSendSideBweTest,
-                         ::testing::Values(true));
-
-TEST_P(WebRtcVoiceEngineWithSendSideBweTest,
        SupportsTransportSequenceNumberHeaderExtension) {
   const std::vector<webrtc::RtpExtension> header_extensions =
       GetDefaultEnabledRtpHeaderExtensions(*engine_);
@@ -2530,9 +2518,7 @@ class WebRtcVoiceEngineWithSendSideBweWithOverheadTest
  public:
   WebRtcVoiceEngineWithSendSideBweWithOverheadTest()
       : WebRtcVoiceEngineTestFake(
-            "WebRTC-Audio-SendSideBwe/Enabled/WebRTC-Audio-Allocation/"
+            "WebRTC-Audio-Allocation/min:6000bps,max:32000bps/") {}
-            "min:6000bps,max:32000bps/WebRTC-SendSideBwe-WithOverhead/"
-            "Enabled/") {}
 };
 
 // Test that we can set the outgoing SSRC properly.

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(
+  std::string audio_twcc_trials("WebRTC-Audio-AlrProbing/Disabled/");
-      "WebRTC-Audio-SendSideBwe/Enabled/"         //
-      "WebRTC-SendSideBwe-WithOverhead/Enabled/"  //
-      "WebRTC-Audio-AlrProbing/Disabled/");
   std::string separate_audio_video(
       "WebRTC-Bwe-SeparateAudioPackets/"
       "enabled:true,packet_threshold:15,time_threshold:1000ms/");

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;

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: