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 cb90a68a0f72e6898fe11726d31d2b99c9770c98..da028c0277c9e9c9edd40387f08cc0cec9b6738e 100644
--- a/webrtc/audio/audio_receive_stream.cc
+++ b/webrtc/audio/audio_receive_stream.cc
@@ -181,6 +181,8 @@ 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();