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..2cc55ec3a2e840a4ee67aec5e617d07d914780ed 100644 |
--- a/webrtc/voice_engine/channel_proxy.h |
+++ b/webrtc/voice_engine/channel_proxy.h |
@@ -83,6 +83,9 @@ 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. |
+ virtual double GetTotalOutputEnergy() const; |
+ virtual double GetTotalOutputDuration() const; |
virtual uint32_t GetDelayEstimate() const; |
virtual bool SetSendTelephoneEventPayloadType(int payload_type, |
int payload_frequency); |