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); |