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

webrtc/audio/audio_receive_stream.cc

Index: webrtc/audio/audio_receive_stream.cc
diff --git a/webrtc/audio/audio_receive_stream.cc b/webrtc/audio/audio_receive_stream.cc
index 079e97178619878c02d87852608fe6615dbc6a9a..d994d0c67e2728b07c668f29f04ef3af9d4447f8 100644
--- a/webrtc/audio/audio_receive_stream.cc
+++ b/webrtc/audio/audio_receive_stream.cc
@@ -187,6 +187,8 @@ webrtc::AudioReceiveStream::Stats AudioReceiveStream::GetStats() const {
   }
   stats.delay_estimate_ms = channel_proxy_->GetDelayEstimate();
   stats.audio_level = channel_proxy_->GetSpeechOutputLevelFullRange();
+  stats.total_output_energy = channel_proxy_->GetTotalOutputEnergy();
+  stats.total_output_duration = channel_proxy_->GetTotalOutputDuration();
 
   // Get jitter buffer and total delay (alg + jitter + playout) stats.
   auto ns = channel_proxy_->GetNetworkStatistics();