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

webrtc/stats/rtcstats_objects.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 355 matching lines @@
     &full_frames_lost,
     &audio_level,
     &echo_return_loss,
     &echo_return_loss_enhancement);
 
 RTCMediaStreamTrackStats::RTCMediaStreamTrackStats(
     const std::string& id, int64_t timestamp_us, const char* kind)
     : RTCMediaStreamTrackStats(std::string(id), timestamp_us, kind) {
 }
 
-RTCMediaStreamTrackStats::RTCMediaStreamTrackStats(
-    std::string&& id, int64_t timestamp_us, const char* kind)
+RTCMediaStreamTrackStats::RTCMediaStreamTrackStats(std::string&& id,
+                                                   int64_t timestamp_us,
+                                                   const char* kind)
     : RTCStats(std::move(id), timestamp_us),
       track_identifier("trackIdentifier"),
       remote_source("remoteSource"),
       ended("ended"),
       detached("detached"),
       kind("kind", kind),
       frame_width("frameWidth"),
       frame_height("frameHeight"),
       frames_per_second("framesPerSecond"),
       frames_sent("framesSent"),
       frames_received("framesReceived"),
       frames_decoded("framesDecoded"),
       frames_dropped("framesDropped"),
       frames_corrupted("framesCorrupted"),
       partial_frames_lost("partialFramesLost"),
       full_frames_lost("fullFramesLost"),
       audio_level("audioLevel"),
+      total_audio_energy("totalAudioEnergy"),
+      total_samples_duration("totalSamplesDuration"),
       echo_return_loss("echoReturnLoss"),
       echo_return_loss_enhancement("echoReturnLossEnhancement") {
   RTC_DCHECK(kind == RTCMediaStreamTrackKind::kAudio ||
              kind == RTCMediaStreamTrackKind::kVideo);
 }
 
 RTCMediaStreamTrackStats::RTCMediaStreamTrackStats(
     const RTCMediaStreamTrackStats& other)
     : RTCStats(other.id(), other.timestamp_us()),
       track_identifier(other.track_identifier),
       remote_source(other.remote_source),
       ended(other.ended),
       detached(other.detached),
       kind(other.kind),
       frame_width(other.frame_width),
       frame_height(other.frame_height),
       frames_per_second(other.frames_per_second),
       frames_sent(other.frames_sent),
       frames_received(other.frames_received),
       frames_decoded(other.frames_decoded),
       frames_dropped(other.frames_dropped),
       frames_corrupted(other.frames_corrupted),
       partial_frames_lost(other.partial_frames_lost),
       full_frames_lost(other.full_frames_lost),
       audio_level(other.audio_level),
+      total_audio_energy(other.total_audio_energy),
+      total_samples_duration(other.total_samples_duration),
       echo_return_loss(other.echo_return_loss),
-      echo_return_loss_enhancement(other.echo_return_loss_enhancement) {
-}
+      echo_return_loss_enhancement(other.echo_return_loss_enhancement) {}
 
 RTCMediaStreamTrackStats::~RTCMediaStreamTrackStats() {
 }
 
 WEBRTC_RTCSTATS_IMPL(RTCPeerConnectionStats, RTCStats, "peer-connection",
     &data_channels_opened,
     &data_channels_closed);
 
 RTCPeerConnectionStats::RTCPeerConnectionStats(
     const std::string& id, int64_t timestamp_us)
@@ skipping 213 matching lines @@
       dtls_state(other.dtls_state),
       selected_candidate_pair_id(other.selected_candidate_pair_id),
       local_certificate_id(other.local_certificate_id),
       remote_certificate_id(other.remote_certificate_id) {
 }
 
 RTCTransportStats::~RTCTransportStats() {
 }
 
 }  // namespace webrtc