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

webrtc/voice_engine/channel_proxy.h

Index: webrtc/voice_engine/channel_proxy.h
diff --git a/webrtc/voice_engine/channel_proxy.h b/webrtc/voice_engine/channel_proxy.h
index f5417f254a1da0e75c82ff28cb992316b738b9cf..b2d8c96e69ad348f0cb9e11900401238533d921a 100644
--- a/webrtc/voice_engine/channel_proxy.h
+++ b/webrtc/voice_engine/channel_proxy.h
@@ -83,6 +83,10 @@ class ChannelProxy : public RtpPacketSinkInterface {
   virtual AudioDecodingCallStats GetDecodingCallStatistics() const;
   virtual int GetSpeechOutputLevel() const;
   virtual int GetSpeechOutputLevelFullRange() const;
+  // See description of "totalAudioEnergy" in the WebRTC stats spec:
+  // https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy
+  virtual double GetTotalOutputEnergy() const;
+  virtual double GetTotalOutputDuration() const;
   virtual uint32_t GetDelayEstimate() const;
   virtual bool SetSendTelephoneEventPayloadType(int payload_type,
                                                 int payload_frequency);