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

webrtc/media/engine/webrtcvoiceengine.cc

Index: webrtc/media/engine/webrtcvoiceengine.cc
diff --git a/webrtc/media/engine/webrtcvoiceengine.cc b/webrtc/media/engine/webrtcvoiceengine.cc
index 1f197a33e8d4b7261bf52531e686b4e5598e93d9..ac50a61af1790d78944429f4149cc458a5ba3ed7 100644
--- a/webrtc/media/engine/webrtcvoiceengine.cc
+++ b/webrtc/media/engine/webrtcvoiceengine.cc
@@ -2251,6 +2251,8 @@ bool WebRtcVoiceMediaChannel::GetStats(VoiceMediaInfo* info) {
     rinfo.jitter_buffer_preferred_ms = stats.jitter_buffer_preferred_ms;
     rinfo.delay_estimate_ms = stats.delay_estimate_ms;
     rinfo.audio_level = stats.audio_level;
+    rinfo.total_output_energy = stats.total_output_energy;
+    rinfo.total_output_duration = stats.total_output_duration;
     rinfo.expand_rate = stats.expand_rate;
     rinfo.speech_expand_rate = stats.speech_expand_rate;
     rinfo.secondary_decoded_rate = stats.secondary_decoded_rate;