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

webrtc/audio/audio_send_stream.cc

Index: webrtc/audio/audio_send_stream.cc
diff --git a/webrtc/audio/audio_send_stream.cc b/webrtc/audio/audio_send_stream.cc
index f8ee3ab565d52f91cdce7ea8523fc2ccc00493d5..e2f780161e1eb0fc50d4f3f64f24b6481ee77a17 100644
--- a/webrtc/audio/audio_send_stream.cc
+++ b/webrtc/audio/audio_send_stream.cc
@@ -279,6 +279,9 @@ webrtc::AudioSendStream::Stats AudioSendStream::GetStats() const {
   stats.audio_level = base->transmit_mixer()->AudioLevelFullRange();
   RTC_DCHECK_LE(0, stats.audio_level);
 
+  stats.total_input_energy = base->transmit_mixer()->GetTotalInputEnergy();
+  stats.total_input_duration = base->transmit_mixer()->GetTotalInputDuration();
+
   RTC_DCHECK(audio_state_->audio_processing());
   auto audio_processing_stats =
       audio_state_->audio_processing()->GetStatistics();