| Index: talk/media/webrtc/webrtcvoiceengine.h
|
| diff --git a/talk/media/webrtc/webrtcvoiceengine.h b/talk/media/webrtc/webrtcvoiceengine.h
|
| index 2c19df7c63d8aff492df7946b3d303dc3108f023..43239f4565185d73d7b3b510ed145cc341e38ee1 100644
|
| --- a/talk/media/webrtc/webrtcvoiceengine.h
|
| +++ b/talk/media/webrtc/webrtcvoiceengine.h
|
| @@ -288,17 +288,10 @@ class WebRtcVoiceMediaChannel : public VoiceMediaChannel,
|
|
|
| bool SetSendParameters(const AudioSendParameters& params) override;
|
| bool SetRecvParameters(const AudioRecvParameters& params) override;
|
| - bool SetOptions(const AudioOptions& options) override;
|
| bool GetOptions(AudioOptions* options) const override {
|
| *options = options_;
|
| return true;
|
| }
|
| - bool SetRecvCodecs(const std::vector<AudioCodec>& codecs) override;
|
| - bool SetSendCodecs(const std::vector<AudioCodec>& codecs) override;
|
| - bool SetRecvRtpHeaderExtensions(
|
| - const std::vector<RtpHeaderExtension>& extensions) override;
|
| - bool SetSendRtpHeaderExtensions(
|
| - const std::vector<RtpHeaderExtension>& extensions) override;
|
| bool SetPlayout(bool playout) override;
|
| bool PausePlayout();
|
| bool ResumePlayout();
|
| @@ -332,8 +325,7 @@ class WebRtcVoiceMediaChannel : public VoiceMediaChannel,
|
| void OnRtcpReceived(rtc::Buffer* packet,
|
| const rtc::PacketTime& packet_time) override;
|
| void OnReadyToSend(bool ready) override {}
|
| - bool MuteStream(uint32 ssrc, bool on) override;
|
| - bool SetMaxSendBandwidth(int bps) override;
|
| + bool MuteStream(uint32 ssrc, bool mute, const AudioOptions* options) override;
|
| bool GetStats(VoiceMediaInfo* info) override;
|
| // Gets last reported error from WebRtc voice engine. This should be only
|
| // called in response a failure.
|
| @@ -363,6 +355,15 @@ class WebRtcVoiceMediaChannel : public VoiceMediaChannel,
|
| void SetCall(webrtc::Call* call);
|
|
|
| private:
|
| + bool SetSendCodecs(const std::vector<AudioCodec>& codecs);
|
| + bool SetSendRtpHeaderExtensions(
|
| + const std::vector<RtpHeaderExtension>& extensions);
|
| + bool SetOptions(const AudioOptions& options);
|
| + bool SetMaxSendBandwidth(int bps);
|
| + bool SetRecvCodecs(const std::vector<AudioCodec>& codecs);
|
| + bool SetRecvRtpHeaderExtensions(
|
| + const std::vector<RtpHeaderExtension>& extensions);
|
| +
|
| WebRtcVoiceEngine* engine() { return engine_; }
|
| int GetLastEngineError() { return engine()->GetLastEngineError(); }
|
| int GetOutputLevel(int channel);
|
|
|