Chromium Code Reviews| 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. |
|
hbos
2017/07/10 09:58:24
Ditto link.
Zach Stein
2017/07/10 18:35:20
Done.
|
| + virtual double GetTotalOutputEnergy() const; |
| + virtual double GetTotalOutputDuration() const; |
| virtual uint32_t GetDelayEstimate() const; |
| virtual bool SetSendTelephoneEventPayloadType(int payload_type, |
| int payload_frequency); |