Skip RTCMediaStreamTrackStats.echoReturnLoss[Enhancement] default value.

webrtc/api/rtcstatscollector.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 299 matching lines @@
       int signal_level;
       if (audio_track->GetSignalLevel(&signal_level)) {
         // Convert signal level from [0,32767] int to [0,1] double.
         RTC_DCHECK_GE(signal_level, 0);
         RTC_DCHECK_LE(signal_level, 32767);
         audio_track_stats->audio_level = signal_level / 32767.0;
       }
       if (audio_track->GetAudioProcessor()) {
         AudioProcessorInterface::AudioProcessorStats audio_processor_stats;
         audio_track->GetAudioProcessor()->GetStats(&audio_processor_stats);
-        audio_track_stats->echo_return_loss = static_cast<double>(
-            audio_processor_stats.echo_return_loss);
-        audio_track_stats->echo_return_loss_enhancement = static_cast<double>(
-            audio_processor_stats.echo_return_loss_enhancement);
+        if (audio_processor_stats.echo_return_loss != -100) {
+          audio_track_stats->echo_return_loss = static_cast<double>(
+              audio_processor_stats.echo_return_loss);
+        }
+        if (audio_processor_stats.echo_return_loss_enhancement != -100) {
+          audio_track_stats->echo_return_loss_enhancement = static_cast<double>(
+              audio_processor_stats.echo_return_loss_enhancement);
+        }
       }
       report->AddStats(std::move(audio_track_stats));
     }
     // Video Tracks
     for (const rtc::scoped_refptr<VideoTrackInterface>& video_track :
          stream->GetVideoTracks()) {
       std::string id = RTCMediaStreamTrackStatsIDFromMediaStreamTrackInterface(
           *video_track.get());
       if (report->Get(id)) {
         // Skip track, stats already exist for it.
@@ skipping 588 matching lines @@
     const std::string& type) {
   return CandidateTypeToRTCIceCandidateType(type);
 }
 
 const char* DataStateToRTCDataChannelStateForTesting(
     DataChannelInterface::DataState state) {
   return DataStateToRTCDataChannelState(state);
 }
 
 }  // namespace webrtc