Revert "Always offer transport sequence number header extension for audio"

This reverts commit fd965c008c7bc395bb276f260262ac11ccd25406.

Reason for revert: Cause test failure.

Original change's description:
> Always offer transport sequence number header extension for audio
> 
> If the extension is negotiated, it will only be used if
> the field trial WebRTC-Audio-SendSideBwe is enabled.
> This allows simpler experimentation if it should be used or not.
> 
> Bug: webrtc:10309 webrtc:10286
> Change-Id: I797e6f14c06d46189e40f6d09805c2e09afc015b
> Reviewed-on: https://webrtc-review.googlesource.com/c/122542
> Commit-Queue: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26689}

TBR=ossu@webrtc.org,sprang@webrtc.org,srte@webrtc.org,perkj@webrtc.org

Change-Id: I1b7d3fa5c282a5bf049ca54695ad16c8278a2698
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10309 webrtc:10286
Reviewed-on: https://webrtc-review.googlesource.com/c/123182
Reviewed-by: Ying Wang <yinwa@webrtc.org>
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26703}

audio/BUILD.gn

@@ -149,7 +149,6 @@ if (rtc_include_tests) {
       "../logging:rtc_event_log_api",
       "../modules/audio_device:mock_audio_device",
       "../rtc_base:rtc_base_tests_utils",
-      "../test:field_trial",
 
       # For TestAudioDeviceModule
       "../modules/audio_device:audio_device_impl",

audio/audio_send_stream.cc

@@ -264,8 +264,8 @@ void AudioSendStream::ConfigureStream(
 
     RtcpBandwidthObserver* bandwidth_observer = nullptr;
 
-    if (stream->allocation_settings_.ShouldSendTransportSequenceNumber(
+    if (stream->allocation_settings_.IncludeAudioInFeedback(
-            new_ids.transport_sequence_number)) {
+            new_ids.transport_sequence_number != 0)) {
       channel_send->EnableSendTransportSequenceNumber(
           new_ids.transport_sequence_number);
       // Probing in application limited region is only used in combination with

audio/audio_send_stream_tests.cc

@@ -9,8 +9,6 @@
  */
 
 #include "test/call_test.h"
-#include "test/constants.h"
-#include "test/field_trial.h"
 #include "test/gtest.h"
 #include "test/rtcp_packet_parser.h"
 
@@ -139,54 +137,42 @@ TEST_F(AudioSendStreamCallTest, SupportsAudioLevel) {
   RunBaseTest(&test);
 }
 
-class TransportWideSequenceNumberObserver : public AudioSendTest {
+TEST_F(AudioSendStreamCallTest, SupportsTransportWideSequenceNumbers) {
- public:
+  static const uint8_t kExtensionId = test::kTransportSequenceNumberExtensionId;
-  explicit TransportWideSequenceNumberObserver(bool expect_sequence_number)
+  class TransportWideSequenceNumberObserver : public AudioSendTest {
-      : AudioSendTest(), expect_sequence_number_(expect_sequence_number) {
+   public:
-    EXPECT_TRUE(parser_->RegisterRtpHeaderExtension(
+    TransportWideSequenceNumberObserver() : AudioSendTest() {
-        kRtpExtensionTransportSequenceNumber,
+      EXPECT_TRUE(parser_->RegisterRtpHeaderExtension(
-        kTransportSequenceNumberExtensionId));
+          kRtpExtensionTransportSequenceNumber, kExtensionId));
-  }
+    }
 
- private:
+   private:
-  Action OnSendRtp(const uint8_t* packet, size_t length) override {
+    Action OnSendRtp(const uint8_t* packet, size_t length) override {
-    RTPHeader header;
+      RTPHeader header;
-    EXPECT_TRUE(parser_->Parse(packet, length, &header));
+      EXPECT_TRUE(parser_->Parse(packet, length, &header));
 
-    EXPECT_EQ(header.extension.hasTransportSequenceNumber,
+      EXPECT_TRUE(header.extension.hasTransportSequenceNumber);
-              expect_sequence_number_);
+      EXPECT_FALSE(header.extension.hasTransmissionTimeOffset);
-    EXPECT_FALSE(header.extension.hasTransmissionTimeOffset);
+      EXPECT_FALSE(header.extension.hasAbsoluteSendTime);
-    EXPECT_FALSE(header.extension.hasAbsoluteSendTime);
 
-    observation_complete_.Set();
+      observation_complete_.Set();
 
-    return SEND_PACKET;
+      return SEND_PACKET;
-  }
+    }
 
-  void ModifyAudioConfigs(
+    void ModifyAudioConfigs(
-      AudioSendStream::Config* send_config,
+        AudioSendStream::Config* send_config,
-      std::vector<AudioReceiveStream::Config>* receive_configs) override {
+        std::vector<AudioReceiveStream::Config>* receive_configs) override {
-    send_config->rtp.extensions.clear();
+      send_config->rtp.extensions.clear();
-    send_config->rtp.extensions.push_back(
+      send_config->rtp.extensions.push_back(RtpExtension(
-        RtpExtension(RtpExtension::kTransportSequenceNumberUri,
+          RtpExtension::kTransportSequenceNumberUri, kExtensionId));
-                     kTransportSequenceNumberExtensionId));
+    }
-  }
 
-  void PerformTest() override {
+    void PerformTest() override {
-    EXPECT_TRUE(Wait()) << "Timed out while waiting for a single RTP packet.";
+      EXPECT_TRUE(Wait()) << "Timed out while waiting for a single RTP packet.";
-  }
+    }
-  const bool expect_sequence_number_;
+  } test;
-};
 
-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);
 }
 

audio/audio_send_stream_unittest.cc

@@ -28,7 +28,6 @@
 #include "modules/rtp_rtcp/mocks/mock_rtcp_rtt_stats.h"
 #include "modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
 #include "rtc_base/task_queue.h"
-#include "test/field_trial.h"
 #include "test/gtest.h"
 #include "test/mock_audio_encoder.h"
 #include "test/mock_audio_encoder_factory.h"
@@ -354,7 +353,6 @@ TEST(AudioSendStreamTest, SetMuted) {
 }
 
 TEST(AudioSendStreamTest, AudioBweCorrectObjectsOnChannelProxy) {
-  ScopedFieldTrials field_trials("WebRTC-Audio-SendSideBwe/Enabled/");
   ConfigHelper helper(true, true);
   auto send_stream = helper.CreateAudioSendStream();
 }
@@ -506,7 +504,6 @@ TEST(AudioSendStreamTest, DontRecreateEncoder) {
 }
 
 TEST(AudioSendStreamTest, ReconfigureTransportCcResetsFirst) {
-  ScopedFieldTrials field_trials("WebRTC-Audio-SendSideBwe/Enabled/");
   ConfigHelper helper(false, true);
   auto send_stream = helper.CreateAudioSendStream();
   auto new_config = helper.config();

media/engine/webrtc_voice_engine.cc

@@ -559,8 +559,11 @@ RtpCapabilities WebRtcVoiceEngine::GetCapabilities() const {
   int id = 1;
   capabilities.header_extensions.push_back(
       webrtc::RtpExtension(webrtc::RtpExtension::kAudioLevelUri, id++));
-  capabilities.header_extensions.push_back(webrtc::RtpExtension(
+  if (allocation_settings_.EnableTransportSequenceNumberExtension()) {
-      webrtc::RtpExtension::kTransportSequenceNumberUri, id++));
+    capabilities.header_extensions.push_back(webrtc::RtpExtension(
+        webrtc::RtpExtension::kTransportSequenceNumberUri, id++));
+  }
+
   return capabilities;
 }
 

rtc_base/experiments/audio_allocation_settings.cc

@@ -65,12 +65,16 @@ bool AudioAllocationSettings::ConfigureRateAllocationRange() const {
   return audio_send_side_bwe_;
 }
 
-bool AudioAllocationSettings::ShouldSendTransportSequenceNumber(
+bool AudioAllocationSettings::EnableTransportSequenceNumberExtension() const {
+  // TODO(srte): Update this to be more accurate.
+  return audio_send_side_bwe_ && !allocate_audio_without_feedback_;
+}
+
+bool AudioAllocationSettings::IncludeAudioInFeedback(
     int transport_seq_num_extension_header_id) const {
   if (force_no_audio_feedback_)
     return false;
-  return audio_send_side_bwe_ && !allocate_audio_without_feedback_ &&
+  return transport_seq_num_extension_header_id != 0;
-         transport_seq_num_extension_header_id != 0;
 }
 
 bool AudioAllocationSettings::UpdateAudioTargetBitrate(

rtc_base/experiments/audio_allocation_settings.h

@@ -27,13 +27,14 @@ class AudioAllocationSettings {
   bool IgnoreSeqNumIdChange() const;
   // Returns true if the bitrate allocation range should be configured.
   bool ConfigureRateAllocationRange() const;
-  // Returns true if sent audio packets should have transport wide sequence
+  // Returns true if the transport sequence number extension should be enabled.
-  // numbers.
+  bool EnableTransportSequenceNumberExtension() const;
+  // Returns true if audio traffic should be included in transport wide feedback
+  // packets.
   // |transport_seq_num_extension_header_id| the extension header id for
   // transport sequence numbers. Set to 0 if not the extension is not
   // configured.
-  bool ShouldSendTransportSequenceNumber(
+  bool IncludeAudioInFeedback(int transport_seq_num_extension_header_id) const;
-      int transport_seq_num_extension_header_id) const;
   // Returns true if target bitrate for audio streams should be updated.
   // |transport_seq_num_extension_header_id| the extension header id for
   // transport sequence numbers. Set to 0 if not the extension is not

video/end_to_end_tests/transport_feedback_tests.cc

@@ -316,7 +316,6 @@ TEST_F(TransportFeedbackEndToEndTest, VideoTransportFeedbackNotConfigured) {
 }
 
 TEST_F(TransportFeedbackEndToEndTest, AudioReceivesTransportFeedback) {
-  test::ScopedFieldTrials field_trials("WebRTC-Audio-SendSideBwe/Enabled/");
   TransportFeedbackTester test(true, 0, 1);
   RunBaseTest(&test);
 }
@@ -425,7 +424,6 @@ TEST_F(TransportFeedbackEndToEndTest,
 }
 
 TEST_F(TransportFeedbackEndToEndTest, TransportSeqNumOnAudioAndVideo) {
-  test::ScopedFieldTrials field_trials("WebRTC-Audio-SendSideBwe/Enabled/");
   static constexpr int kExtensionId = 8;
   static constexpr size_t kMinPacketsToWaitFor = 50;
   class TransportSequenceNumberTest : public test::EndToEndTest {