Revert "Fix echo return loss stats and add to RTCAudioSourceStats."

This reverts commit a27cfbffdfa0bf359628d2164db5b9d6321f9c9c.

Reason for revert: WebRtcBrowserTest.RunsAudioVideoWebRTCCallInTwoTabsGetStatsPromise failing.

Original change's description:
> Fix echo return loss stats and add to RTCAudioSourceStats.
>
> This solves two problems:
> * Echo return loss stats weren't being gathered in Chrome, because they
>   need to be taken from the audio processor attached to the track
>   rather than the audio send stream.
> * The standardized location is in RTCAudioSourceStats, not
>   RTCMediaStreamTrackStats. For now, will populate the stats in both
>   locations.
>
> Bug: webrtc:12770
> Change-Id: I47eaf7f2b50b914a1be84156aa831e27497d07e3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223182
> Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#34344}

TBR=deadbeef@webrtc.org,hbos@webrtc.org,hbos@chromium.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com

Change-Id: I6b2587d762f005adef67c0d5121f1b58c3b76688
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:12770
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223068
Reviewed-by: Evan Shrubsole <eshr@google.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#34352}

api/stats/rtcstats_objects.h

@@ -629,8 +629,6 @@ class RTC_EXPORT RTCAudioSourceStats final : public RTCMediaSourceStats {
   RTCStatsMember<double> audio_level;
   RTCStatsMember<double> total_audio_energy;
   RTCStatsMember<double> total_samples_duration;
-  RTCStatsMember<double> echo_return_loss;
-  RTCStatsMember<double> echo_return_loss_enhancement;
 };
 
 // https://w3c.github.io/webrtc-stats/#dom-rtcvideosourcestats

pc/rtc_stats_collector.cc

@@ -743,22 +743,10 @@ const std::string& ProduceIceCandidateStats(int64_t timestamp_us,
   return stats->id();
 }
 
-template <typename StatsType>
-void SetAudioProcessingStats(StatsType* stats,
-                             const AudioProcessingStats& apm_stats) {
-  if (apm_stats.echo_return_loss) {
-    stats->echo_return_loss = *apm_stats.echo_return_loss;
-  }
-  if (apm_stats.echo_return_loss_enhancement) {
-    stats->echo_return_loss_enhancement =
-        *apm_stats.echo_return_loss_enhancement;
-  }
-}
-
 std::unique_ptr<RTCMediaStreamTrackStats>
 ProduceMediaStreamTrackStatsFromVoiceSenderInfo(
     int64_t timestamp_us,
-    AudioTrackInterface& audio_track,
+    const AudioTrackInterface& audio_track,
     const cricket::VoiceSenderInfo& voice_sender_info,
     int attachment_id) {
   std::unique_ptr<RTCMediaStreamTrackStats> audio_track_stats(
@@ -773,17 +761,13 @@ ProduceMediaStreamTrackStatsFromVoiceSenderInfo(
                                                  attachment_id);
   audio_track_stats->remote_source = false;
   audio_track_stats->detached = false;
-  // Audio processor may be attached to either the track or the send
+  if (voice_sender_info.apm_statistics.echo_return_loss) {
-  // stream, so look in both places.
+    audio_track_stats->echo_return_loss =
-  SetAudioProcessingStats(audio_track_stats.get(),
+        *voice_sender_info.apm_statistics.echo_return_loss;
-                          voice_sender_info.apm_statistics);
+  }
-  auto audio_processor(audio_track.GetAudioProcessor());
+  if (voice_sender_info.apm_statistics.echo_return_loss_enhancement) {
-  if (audio_processor.get()) {
+    audio_track_stats->echo_return_loss_enhancement =
-    // The |has_remote_tracks| argument is obsolete; makes no difference if it's
+        *voice_sender_info.apm_statistics.echo_return_loss_enhancement;
-    // set to true or false.
-    AudioProcessorInterface::AudioProcessorStatistics ap_stats =
-        audio_processor->GetStats(/*has_remote_tracks=*/false);
-    SetAudioProcessingStats(audio_track_stats.get(), ap_stats.apm_statistics);
   }
   return audio_track_stats;
 }
@@ -1673,8 +1657,6 @@ void RTCStatsCollector::ProduceMediaSourceStats_s(
       // create separate media source stats objects on a per-attachment basis.
       std::unique_ptr<RTCMediaSourceStats> media_source_stats;
       if (track->kind() == MediaStreamTrackInterface::kAudioKind) {
-        AudioTrackInterface* audio_track =
-            static_cast<AudioTrackInterface*>(track.get());
         auto audio_source_stats = std::make_unique<RTCAudioSourceStats>(
             RTCMediaSourceStatsIDFromKindAndAttachment(
                 cricket::MEDIA_TYPE_AUDIO, sender_internal->AttachmentId()),
@@ -1695,21 +1677,8 @@ void RTCStatsCollector::ProduceMediaSourceStats_s(
                 voice_sender_info->total_input_energy;
             audio_source_stats->total_samples_duration =
                 voice_sender_info->total_input_duration;
-            SetAudioProcessingStats(audio_source_stats.get(),
-                                    voice_sender_info->apm_statistics);
           }
         }
-        // Audio processor may be attached to either the track or the send
-        // stream, so look in both places.
-        auto audio_processor(audio_track->GetAudioProcessor());
-        if (audio_processor.get()) {
-          // The |has_remote_tracks| argument is obsolete; makes no difference
-          // if it's set to true or false.
-          AudioProcessorInterface::AudioProcessorStatistics ap_stats =
-              audio_processor->GetStats(/*has_remote_tracks=*/false);
-          SetAudioProcessingStats(audio_source_stats.get(),
-                                  ap_stats.apm_statistics);
-        }
         media_source_stats = std::move(audio_source_stats);
       } else {
         RTC_DCHECK_EQ(MediaStreamTrackInterface::kVideoKind, track->kind());

pc/rtc_stats_collector_unittest.cc

@@ -200,34 +200,14 @@ std::unique_ptr<cricket::Candidate> CreateFakeCandidate(
   return candidate;
 }
 
-class FakeAudioProcessor : public AudioProcessorInterface {
- public:
-  FakeAudioProcessor() {}
-  ~FakeAudioProcessor() {}
-
- private:
-  AudioProcessorInterface::AudioProcessorStatistics GetStats(
-      bool has_recv_streams) override {
-    AudioProcessorStatistics stats;
-    stats.apm_statistics.echo_return_loss = 2.0;
-    stats.apm_statistics.echo_return_loss_enhancement = 3.0;
-    return stats;
-  }
-};
-
 class FakeAudioTrackForStats : public MediaStreamTrack<AudioTrackInterface> {
  public:
   static rtc::scoped_refptr<FakeAudioTrackForStats> Create(
       const std::string& id,
-      MediaStreamTrackInterface::TrackState state,
+      MediaStreamTrackInterface::TrackState state) {
-      bool create_fake_audio_processor) {
     rtc::scoped_refptr<FakeAudioTrackForStats> audio_track_stats(
         new rtc::RefCountedObject<FakeAudioTrackForStats>(id));
     audio_track_stats->set_state(state);
-    if (create_fake_audio_processor) {
-      audio_track_stats->processor_ =
-          rtc::make_ref_counted<FakeAudioProcessor>();
-    }
     return audio_track_stats;
   }
 
@@ -242,11 +222,8 @@ class FakeAudioTrackForStats : public MediaStreamTrack<AudioTrackInterface> {
   void RemoveSink(webrtc::AudioTrackSinkInterface* sink) override {}
   bool GetSignalLevel(int* level) override { return false; }
   rtc::scoped_refptr<AudioProcessorInterface> GetAudioProcessor() override {
-    return processor_;
+    return nullptr;
   }
-
- private:
-  rtc::scoped_refptr<FakeAudioProcessor> processor_;
 };
 
 class FakeVideoTrackSourceForStats : public VideoTrackSourceInterface {
@@ -331,11 +308,9 @@ class FakeVideoTrackForStats : public MediaStreamTrack<VideoTrackInterface> {
 rtc::scoped_refptr<MediaStreamTrackInterface> CreateFakeTrack(
     cricket::MediaType media_type,
     const std::string& track_id,
-    MediaStreamTrackInterface::TrackState track_state,
+    MediaStreamTrackInterface::TrackState track_state) {
-    bool create_fake_audio_processor = false) {
   if (media_type == cricket::MEDIA_TYPE_AUDIO) {
-    return FakeAudioTrackForStats::Create(track_id, track_state,
+    return FakeAudioTrackForStats::Create(track_id, track_state);
-                                          create_fake_audio_processor);
   } else {
     RTC_DCHECK_EQ(media_type, cricket::MEDIA_TYPE_VIDEO);
     return FakeVideoTrackForStats::Create(track_id, track_state, nullptr);
@@ -2605,9 +2580,6 @@ TEST_F(RTCStatsCollectorTest, RTCAudioSourceStatsCollectedForSenderWithTrack) {
   voice_media_info.senders[0].audio_level = 32767;  // [0,32767]
   voice_media_info.senders[0].total_input_energy = 2.0;
   voice_media_info.senders[0].total_input_duration = 3.0;
-  voice_media_info.senders[0].apm_statistics.echo_return_loss = 42.0;
-  voice_media_info.senders[0].apm_statistics.echo_return_loss_enhancement =
-      52.0;
   auto* voice_media_channel = pc_->AddVoiceChannel("AudioMid", "TransportName");
   voice_media_channel->SetStats(voice_media_info);
   stats_->SetupLocalTrackAndSender(cricket::MEDIA_TYPE_AUDIO,
@@ -2623,8 +2595,6 @@ TEST_F(RTCStatsCollectorTest, RTCAudioSourceStatsCollectedForSenderWithTrack) {
   expected_audio.audio_level = 1.0;  // [0,1]
   expected_audio.total_audio_energy = 2.0;
   expected_audio.total_samples_duration = 3.0;
-  expected_audio.echo_return_loss = 42.0;
-  expected_audio.echo_return_loss_enhancement = 52.0;
 
   ASSERT_TRUE(report->Get(expected_audio.id()));
   EXPECT_EQ(report->Get(expected_audio.id())->cast_to<RTCAudioSourceStats>(),
@@ -3086,64 +3056,6 @@ TEST_F(RTCStatsCollectorTest,
   EXPECT_FALSE(report->Get("RTCVideoSource_42"));
 }
 
-// Test collecting echo return loss stats from the audio processor attached to
-// the track, rather than the voice sender info.
-TEST_F(RTCStatsCollectorTest, CollectEchoReturnLossFromTrackAudioProcessor) {
-  rtc::scoped_refptr<MediaStream> local_stream =
-      MediaStream::Create("LocalStreamId");
-  pc_->mutable_local_streams()->AddStream(local_stream);
-
-  // Local audio track
-  rtc::scoped_refptr<MediaStreamTrackInterface> local_audio_track =
-      CreateFakeTrack(cricket::MEDIA_TYPE_AUDIO, "LocalAudioTrackID",
-                      MediaStreamTrackInterface::kEnded,
-                      /*create_fake_audio_processor=*/true);
-  local_stream->AddTrack(
-      static_cast<AudioTrackInterface*>(local_audio_track.get()));
-
-  cricket::VoiceSenderInfo voice_sender_info_ssrc1;
-  voice_sender_info_ssrc1.local_stats.push_back(cricket::SsrcSenderInfo());
-  voice_sender_info_ssrc1.local_stats[0].ssrc = 1;
-
-  stats_->CreateMockRtpSendersReceiversAndChannels(
-      {std::make_pair(local_audio_track.get(), voice_sender_info_ssrc1)}, {},
-      {}, {}, {local_stream->id()}, {});
-
-  rtc::scoped_refptr<const RTCStatsReport> report = stats_->GetStatsReport();
-
-  RTCMediaStreamTrackStats expected_local_audio_track_ssrc1(
-      IdForType<RTCMediaStreamTrackStats>(report), report->timestamp_us(),
-      RTCMediaStreamTrackKind::kAudio);
-  expected_local_audio_track_ssrc1.track_identifier = local_audio_track->id();
-  expected_local_audio_track_ssrc1.media_source_id =
-      "RTCAudioSource_11";  // Attachment ID = SSRC + 10
-  expected_local_audio_track_ssrc1.remote_source = false;
-  expected_local_audio_track_ssrc1.ended = true;
-  expected_local_audio_track_ssrc1.detached = false;
-  expected_local_audio_track_ssrc1.echo_return_loss = 2.0;
-  expected_local_audio_track_ssrc1.echo_return_loss_enhancement = 3.0;
-  ASSERT_TRUE(report->Get(expected_local_audio_track_ssrc1.id()))
-      << "Did not find " << expected_local_audio_track_ssrc1.id() << " in "
-      << report->ToJson();
-  EXPECT_EQ(expected_local_audio_track_ssrc1,
-            report->Get(expected_local_audio_track_ssrc1.id())
-                ->cast_to<RTCMediaStreamTrackStats>());
-
-  RTCAudioSourceStats expected_audio("RTCAudioSource_11",
-                                     report->timestamp_us());
-  expected_audio.track_identifier = "LocalAudioTrackID";
-  expected_audio.kind = "audio";
-  expected_audio.audio_level = 0;
-  expected_audio.total_audio_energy = 0;
-  expected_audio.total_samples_duration = 0;
-  expected_audio.echo_return_loss = 2.0;
-  expected_audio.echo_return_loss_enhancement = 3.0;
-
-  ASSERT_TRUE(report->Get(expected_audio.id()));
-  EXPECT_EQ(report->Get(expected_audio.id())->cast_to<RTCAudioSourceStats>(),
-            expected_audio);
-}
-
 TEST_F(RTCStatsCollectorTest, GetStatsWithSenderSelector) {
   ExampleStatsGraph graph = SetupExampleStatsGraphForSelectorTests();
   // Expected stats graph when filtered by sender:

pc/rtc_stats_integrationtest.cc

@@ -1082,12 +1082,6 @@ class RTCStatsReportVerifier {
     verifier.TestMemberIsNonNegative<double>(audio_source.audio_level);
     verifier.TestMemberIsPositive<double>(audio_source.total_audio_energy);
     verifier.TestMemberIsPositive<double>(audio_source.total_samples_duration);
-    // TODO(hbos): |echo_return_loss| and |echo_return_loss_enhancement| are
-    // flaky on msan bot (sometimes defined, sometimes undefined). Should the
-    // test run until available or is there a way to have it always be
-    // defined? crbug.com/627816
-    verifier.MarkMemberTested(audio_source.echo_return_loss, true);
-    verifier.MarkMemberTested(audio_source.echo_return_loss_enhancement, true);
     return verifier.ExpectAllMembersSuccessfullyTested();
   }
 

stats/rtcstats_objects.cc

@@ -987,9 +987,7 @@ RTCMediaSourceStats::~RTCMediaSourceStats() {}
 WEBRTC_RTCSTATS_IMPL(RTCAudioSourceStats, RTCMediaSourceStats, "media-source",
     &audio_level,
     &total_audio_energy,
-    &total_samples_duration,
+    &total_samples_duration)
-    &echo_return_loss,
-    &echo_return_loss_enhancement)
 // clang-format on
 
 RTCAudioSourceStats::RTCAudioSourceStats(const std::string& id,
@@ -1000,17 +998,13 @@ RTCAudioSourceStats::RTCAudioSourceStats(std::string&& id, int64_t timestamp_us)
     : RTCMediaSourceStats(std::move(id), timestamp_us),
       audio_level("audioLevel"),
       total_audio_energy("totalAudioEnergy"),
-      total_samples_duration("totalSamplesDuration"),
+      total_samples_duration("totalSamplesDuration") {}
-      echo_return_loss("echoReturnLoss"),
-      echo_return_loss_enhancement("echoReturnLossEnhancement") {}
 
 RTCAudioSourceStats::RTCAudioSourceStats(const RTCAudioSourceStats& other)
     : RTCMediaSourceStats(other),
       audio_level(other.audio_level),
       total_audio_energy(other.total_audio_energy),
-      total_samples_duration(other.total_samples_duration),
+      total_samples_duration(other.total_samples_duration) {}
-      echo_return_loss(other.echo_return_loss),
-      echo_return_loss_enhancement(other.echo_return_loss_enhancement) {}
 
 RTCAudioSourceStats::~RTCAudioSourceStats() {}