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); |