Adding stats that can be used to compute output audio levels.

webrtc/pc/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 356 matching lines @@
           timestamp_us,
           RTCMediaStreamTrackKind::kAudio));
   SetMediaStreamTrackStatsFromMediaStreamTrackInterface(
       audio_track, audio_track_stats.get());
   audio_track_stats->remote_source = false;
   audio_track_stats->detached = false;
   if (voice_sender_info.audio_level >= 0) {
     audio_track_stats->audio_level = DoubleAudioLevelFromIntAudioLevel(
         voice_sender_info.audio_level);
   }
+  audio_track_stats->total_audio_energy = voice_sender_info.total_input_energy;
+  audio_track_stats->total_samples_duration =
+      voice_sender_info.total_input_duration;
   if (voice_sender_info.echo_return_loss != -100) {
     audio_track_stats->echo_return_loss = static_cast<double>(
         voice_sender_info.echo_return_loss);
   }
   if (voice_sender_info.echo_return_loss_enhancement != -100) {
     audio_track_stats->echo_return_loss_enhancement = static_cast<double>(
         voice_sender_info.echo_return_loss_enhancement);
   }
   return audio_track_stats;
 }
@@ skipping 11 matching lines @@
           timestamp_us,
           RTCMediaStreamTrackKind::kAudio));
   SetMediaStreamTrackStatsFromMediaStreamTrackInterface(
       audio_track, audio_track_stats.get());
   audio_track_stats->remote_source = true;
   audio_track_stats->detached = false;
   if (voice_receiver_info.audio_level >= 0) {
     audio_track_stats->audio_level = DoubleAudioLevelFromIntAudioLevel(
         voice_receiver_info.audio_level);
   }
+  audio_track_stats->total_audio_energy =
+      voice_receiver_info.total_output_energy;
+  audio_track_stats->total_samples_duration =
+      voice_receiver_info.total_output_duration;
   return audio_track_stats;
 }
 
 std::unique_ptr<RTCMediaStreamTrackStats>
 ProduceMediaStreamTrackStatsFromVideoSenderInfo(
     int64_t timestamp_us,
     const VideoTrackInterface& video_track,
     const cricket::VideoSenderInfo& video_sender_info) {
   std::unique_ptr<RTCMediaStreamTrackStats> video_track_stats(
       new RTCMediaStreamTrackStats(
@@ skipping 830 matching lines @@
     const std::string& type) {
   return CandidateTypeToRTCIceCandidateType(type);
 }
 
 const char* DataStateToRTCDataChannelStateForTesting(
     DataChannelInterface::DataState state) {
   return DataStateToRTCDataChannelState(state);
 }
 
 }  // namespace webrtc