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..84bbbfcc4337bf99367898bd0444ecfc6c3ee966 100644
--- a/webrtc/voice_engine/transmit_mixer.h
+++ b/webrtc/voice_engine/transmit_mixer.h
@@ -76,6 +76,14 @@ public:
     // 'virtual' to allow mocking.
     virtual int16_t AudioLevelFullRange() const;
 
+    // See description of "totalAudioEnergy" in the WebRTC stats spec:
+    // https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy
+    // 'virtual' to allow mocking.
+    virtual double GetTotalInputEnergy() const;
+
+    // 'virtual' to allow mocking.
+    virtual double GetTotalInputDuration() const;
+
     bool IsRecordingCall();
 
     bool IsRecordingMic();
@@ -189,6 +197,8 @@ private:
     bool _fileRecording = false;
     bool _fileCallRecording = false;
     voe::AudioLevel _audioLevel;
+    double totalInputEnergy_ = 0.0;
+    double totalInputDuration_ = 0.0;
     // protect file instances and their variables in MixedParticipants()
     rtc::CriticalSection _critSect;
     rtc::CriticalSection _callbackCritSect;