Chromium Code Reviews| Index: talk/media/webrtc/webrtcvoiceengine.h |
| diff --git a/talk/media/webrtc/webrtcvoiceengine.h b/talk/media/webrtc/webrtcvoiceengine.h |
| index e8ae1575964ca3650cdf6e2b93d74948df8507c3..aa7d7dafff0fde40c22f89a35673df1179835201 100644 |
| --- a/talk/media/webrtc/webrtcvoiceengine.h |
| +++ b/talk/media/webrtc/webrtcvoiceengine.h |
| @@ -282,8 +282,9 @@ class WebRtcVoiceEngine |
| class WebRtcVoiceMediaChannel : public VoiceMediaChannel, |
| public webrtc::Transport { |
| public: |
| - explicit WebRtcVoiceMediaChannel(WebRtcVoiceEngine* engine, |
| - webrtc::Call* call); |
| + WebRtcVoiceMediaChannel(WebRtcVoiceEngine* engine, |
| + const AudioOptions& options, |
|
pbos-webrtc
2015/09/17 12:13:53
if these are initial as well, please initial_optio
the sun
2015/09/17 13:37:48
Here also, I don't think that is necessary.
|
| + webrtc::Call* call); |
| ~WebRtcVoiceMediaChannel() override; |
| int voe_channel() const { return voe_channel_; } |
| @@ -292,13 +293,6 @@ 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(); |
| @@ -332,7 +326,6 @@ 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. |
| @@ -360,8 +353,17 @@ class WebRtcVoiceMediaChannel : public VoiceMediaChannel, |
| int GetSendChannelNum(uint32 ssrc) const; |
| 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); |