| Index: talk/media/webrtc/webrtcvoiceengine.h
|
| diff --git a/talk/media/webrtc/webrtcvoiceengine.h b/talk/media/webrtc/webrtcvoiceengine.h
|
| index 951c878161585b2ef4738457929fc7aa0ba38ea1..2332b4c36db1fb44935a99db70f233841bd8dccf 100644
|
| --- a/talk/media/webrtc/webrtcvoiceengine.h
|
| +++ b/talk/media/webrtc/webrtcvoiceengine.h
|
| @@ -280,7 +280,8 @@ class WebRtcVoiceEngine
|
| class WebRtcVoiceMediaChannel : public VoiceMediaChannel,
|
| public webrtc::Transport {
|
| public:
|
| - explicit WebRtcVoiceMediaChannel(WebRtcVoiceEngine *engine);
|
| + WebRtcVoiceMediaChannel(WebRtcVoiceEngine *engine,
|
| + const AudioOptions& options);
|
| ~WebRtcVoiceMediaChannel() override;
|
|
|
| int voe_channel() const { return voe_channel_; }
|
| @@ -289,13 +290,6 @@ class WebRtcVoiceMediaChannel : public VoiceMediaChannel,
|
|
|
| bool SetSendParameters(const AudioSendParameters& params) override;
|
| bool SetRecvParameters(const AudioRecvParameters& params) override;
|
| - bool SetOptions(const AudioOptions& options) override;
|
| - 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();
|
| @@ -329,7 +323,6 @@ class WebRtcVoiceMediaChannel : public VoiceMediaChannel,
|
| void OnRtcpReceived(rtc::Buffer* packet,
|
| const rtc::PacketTime& packet_time) override;
|
| void OnReadyToSend(bool ready) override {}
|
| - bool SetMaxSendBandwidth(int bps) override;
|
| bool GetStats(VoiceMediaInfo* info) override;
|
| // Gets last reported error from WebRtc voice engine. This should be only
|
| // called in response a failure.
|
| @@ -359,8 +352,17 @@ 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);
|
| bool SetLocalRenderer(uint32 ssrc, AudioRenderer* renderer);
|
| bool MuteStream(uint32 ssrc, bool mute);
|
| +
|
| WebRtcVoiceEngine* engine() { return engine_; }
|
| int GetLastEngineError() { return engine()->GetLastEngineError(); }
|
| int GetOutputLevel(int channel);
|
|
|