| 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);
|
|
|