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

webrtc/voice_engine/transmit_mixer.h

Index: webrtc/voice_engine/transmit_mixer.h
diff --git a/webrtc/voice_engine/transmit_mixer.h b/webrtc/voice_engine/transmit_mixer.h
index a045cf8322855abb0b7217e1b23c5e54eb80a00b..db5d35d20eee63700f5bd13f7dbb4abcc1239f56 100644
--- a/webrtc/voice_engine/transmit_mixer.h
+++ b/webrtc/voice_engine/transmit_mixer.h
@@ -76,6 +76,11 @@ public:
     // 'virtual' to allow mocking.
     virtual int16_t AudioLevelFullRange() const;
 
+    // See description of "totalAudioEnergy" in the WebRTC stats spec.
+    double GetTotalInputEnergy() const;
+
+    double GetTotalInputDuration() const;
+
     bool IsRecordingCall();
 
     bool IsRecordingMic();
@@ -189,6 +194,8 @@ private:
     bool _fileRecording = false;
     bool _fileCallRecording = false;
     voe::AudioLevel _audioLevel;
+    double _totalInputEnergy = 0;

[hlundin-webrtc, 2017/07/03 13:23:03]

0.0 here and below.

[Zach Stein, 2017/07/06 17:41:37]

Done.

+    double _totalInputDuration = 0;
     // protect file instances and their variables in MixedParticipants()
     rtc::CriticalSection _critSect;
     rtc::CriticalSection _callbackCritSect;