Skip RTCMediaStreamTrackStats.echoReturnLoss[Enhancement] default value.

webrtc/api/rtcstatscollector_unittest.cc

 /*
  *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 1191 matching lines @@
   expected_remote_audio_track.audio_level = 0.0;
   expected_remote_audio_track.echo_return_loss = 13.0;
   expected_remote_audio_track.echo_return_loss_enhancement = 37.0;
   EXPECT_TRUE(report->Get(expected_remote_audio_track.id()));
   EXPECT_EQ(expected_remote_audio_track,
             report->Get(expected_remote_audio_track.id())->cast_to<
                 RTCMediaStreamTrackStats>());
 }
 
 TEST_F(RTCStatsCollectorTest,
+       CollectRTCMediaStreamStatsAndRTCMediaStreamTrackStats_Audio_Defaults) {
+  rtc::scoped_refptr<StreamCollection> local_streams =
+      StreamCollection::Create();
+  EXPECT_CALL(test_->pc(), local_streams())
+      .WillRepeatedly(Return(local_streams));
+
+  rtc::scoped_refptr<MediaStream> local_stream =
+      MediaStream::Create("LocalStreamLabel");
+  local_streams->AddStream(local_stream);
+
+  // Local audio track
+  AudioProcessorInterface::AudioProcessorStats local_audio_processor_stats;
+  local_audio_processor_stats.echo_return_loss = -100;
+  local_audio_processor_stats.echo_return_loss_enhancement = -100;
+  rtc::scoped_refptr<FakeAudioTrackForStats> local_audio_track =
+      FakeAudioTrackForStats::Create(
+          "LocalAudioTrackID",
+          MediaStreamTrackInterface::TrackState::kEnded,
+          32767,
+          new FakeAudioProcessorForStats(local_audio_processor_stats));
+  local_stream->AddTrack(local_audio_track);
+
+  rtc::scoped_refptr<const RTCStatsReport> report = GetStatsReport();
+
+  RTCMediaStreamTrackStats expected_local_audio_track(
+      "RTCMediaStreamTrack_LocalAudioTrackID", report->timestamp_us());
+  expected_local_audio_track.track_identifier = local_audio_track->id();
+  expected_local_audio_track.remote_source = false;
+  expected_local_audio_track.ended = true;
+  expected_local_audio_track.detached = false;
+  expected_local_audio_track.audio_level = 1.0;
+  // Should be undefined: |expected_local_audio_track.echo_return_loss| and
+  // |expected_local_audio_track.echo_return_loss_enhancement|.
+  EXPECT_TRUE(report->Get(expected_local_audio_track.id()));
+  EXPECT_EQ(expected_local_audio_track,
+            report->Get(expected_local_audio_track.id())->cast_to<
+                RTCMediaStreamTrackStats>());
+}
+
+TEST_F(RTCStatsCollectorTest,
        CollectRTCMediaStreamStatsAndRTCMediaStreamTrackStats_Video) {
   rtc::scoped_refptr<StreamCollection> local_streams =
       StreamCollection::Create();
   rtc::scoped_refptr<StreamCollection> remote_streams =
       StreamCollection::Create();
   EXPECT_CALL(test_->pc(), local_streams())
       .WillRepeatedly(Return(local_streams));
   EXPECT_CALL(test_->pc(), remote_streams())
       .WillRepeatedly(Return(remote_streams));
 
@@ skipping 532 matching lines @@
   rtc::scoped_refptr<FakeRTCStatsCollector> collector_;
 };
 
 TEST_F(RTCStatsCollectorTestWithFakeCollector, ThreadUsageAndResultsMerging) {
   collector_->VerifyThreadUsageAndResultsMerging();
 }
 
 }  // namespace
 
 }  // namespace webrtc