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

webrtc/voice_engine/channel.cc

Index: webrtc/voice_engine/channel.cc
diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc
index 16122709c31fe976ddde3c63a477d78957f72070..87553048cd938c14fd574fb1d6aef978f4c4e2f1 100644
--- a/webrtc/voice_engine/channel.cc
+++ b/webrtc/voice_engine/channel.cc
@@ -696,7 +696,17 @@ MixerParticipant::AudioFrameInfo Channel::GetAudioFrameWithMuted(
 
   // Measure audio level (0-9)
   // TODO(henrik.lundin) Use the |muted| information here too.
+  // TODO(deadbeef): Use RmsLevel for |_outputAudioLevel| as well.
   _outputAudioLevel.ComputeLevel(*audioFrame);
+  // See the description for "totalAudioEnergy" in the WebRTC stats spec for an
+  // explanation of these formulas. In short, we need a value that can be used
+  // to compute RMS audio levels over different time intervals, by taking the
+  // difference between the results from two getStats calls. To do this, the
+  // value needs to be of units "squared sample value * time".
+  double additional_energy = _outputAudioLevel.LevelFullRange() / SHRT_MAX;
+  additional_energy *= additional_energy;
+  _totalOutputEnergy += additional_energy * 0.01;
+  _totalOutputDuration += 0.01;
 
   if (capture_start_rtp_time_stamp_ < 0 && audioFrame->timestamp_ != 0) {
     // The first frame with a valid rtp timestamp.
@@ -2370,6 +2380,14 @@ int Channel::GetSpeechOutputLevelFullRange() const {
   return _outputAudioLevel.LevelFullRange();
 }
 
+double Channel::GetTotalOutputEnergy() const {
+  return _totalOutputEnergy;
+}
+
+double Channel::GetTotalOutputDuration() const {
+  return _totalOutputDuration;
+}
+
 void Channel::SetInputMute(bool enable) {
   rtc::CritScope cs(&volume_settings_critsect_);
   input_mute_ = enable;