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

webrtc/media/base/mediachannel.h

Index: webrtc/media/base/mediachannel.h
diff --git a/webrtc/media/base/mediachannel.h b/webrtc/media/base/mediachannel.h
index 1206cccbea527c5af2b9aa5c186e1a3a5f1834fb..7dc2239bf735f360b1ccc1f3b8bd34dd2a74f454 100644
--- a/webrtc/media/base/mediachannel.h
+++ b/webrtc/media/base/mediachannel.h
@@ -675,6 +675,9 @@ struct VoiceReceiverInfo : public MediaReceiverInfo {
   int jitter_buffer_preferred_ms;
   int delay_estimate_ms;
   int audio_level;
+  // See description of "totalAudioEnergy" in the WebRTC stats spec.
+  double total_output_energy = 0.0;
+  double total_output_duration = 0.0;
   // fraction of synthesized audio inserted through expansion.
   float expand_rate;
   // fraction of synthesized speech inserted through expansion.