- Rename VoiceChannel::MuteStream() -> SetAudioSend() (incl. media channel)

talk/media/webrtc/webrtcvoiceengine.h

Index: talk/media/webrtc/webrtcvoiceengine.h
diff --git a/talk/media/webrtc/webrtcvoiceengine.h b/talk/media/webrtc/webrtcvoiceengine.h
index 313a371f7a977d1d90f6c0b6e7663eae230726ec..e8854833c7da1dec314f2865f62552d9a730d4c4 100644
--- a/talk/media/webrtc/webrtcvoiceengine.h
+++ b/talk/media/webrtc/webrtcvoiceengine.h
@@ -305,12 +305,13 @@ class WebRtcVoiceMediaChannel : public VoiceMediaChannel,
   bool SetSend(SendFlags send) override;
   bool PauseSend();
   bool ResumeSend();
+  bool SetAudioSend(uint32 ssrc, bool mute, const AudioOptions* options,
+                    AudioRenderer* renderer) override;
   bool AddSendStream(const StreamParams& sp) override;
   bool RemoveSendStream(uint32 ssrc) override;
   bool AddRecvStream(const StreamParams& sp) override;
   bool RemoveRecvStream(uint32 ssrc) override;
   bool SetRemoteRenderer(uint32 ssrc, AudioRenderer* renderer) override;
-  bool SetLocalRenderer(uint32 ssrc, AudioRenderer* renderer) override;
   bool GetActiveStreams(AudioInfo::StreamList* actives) override;
   int GetOutputLevel() override;
   int GetTimeSinceLastTyping() override;
@@ -331,7 +332,6 @@ 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 GetStats(VoiceMediaInfo* info) override;
   // Gets last reported error from WebRtc voice engine.  This should be only