Change 'mute' parameter of MediaChannel::SetAudioSend()/SetVideoSend() to 'enable'.

talk/media/webrtc/webrtcvoiceengine.h

 /*
  * libjingle
  * Copyright 2004 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
@@ skipping 177 matching lines @@
   const AudioOptions& options() const { return options_; }
 
   bool SetSendParameters(const AudioSendParameters& params) override;
   bool SetRecvParameters(const AudioRecvParameters& params) override;
   bool SetPlayout(bool playout) override;
   bool PausePlayout();
   bool ResumePlayout();
   bool SetSend(SendFlags send) override;
   bool PauseSend();
   bool ResumeSend();
-  bool SetAudioSend(uint32 ssrc, bool mute, const AudioOptions* options,
+  bool SetAudioSend(uint32 ssrc, bool enable, 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 GetActiveStreams(AudioInfo::StreamList* actives) override;
   int GetOutputLevel() override;
   int GetTimeSinceLastTyping() override;
   void SetTypingDetectionParameters(int time_window,
@@ skipping 26 matching lines @@
 
   bool SendRtcp(const uint8_t* data, size_t len) override {
     rtc::Buffer packet(reinterpret_cast<const uint8_t*>(data), len,
                        kMaxRtpPacketLen);
     return VoiceMediaChannel::SendRtcp(&packet);
   }
 
   bool FindSsrc(int channel_num, uint32* ssrc);
   void OnError(uint32 ssrc, int error);
 
-  bool sending() const { return send_ != SEND_NOTHING; }
   int GetReceiveChannelNum(uint32 ssrc) const;
   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);
@@ skipping 93 matching lines @@
   std::vector<webrtc::RtpExtension> recv_rtp_extensions_;
 
   // Do not lock this on the VoE media processor thread; potential for deadlock
   // exists.
   mutable rtc::CriticalSection receive_channels_cs_;
 };
 
 }  // namespace cricket
 
 #endif  // TALK_MEDIA_WEBRTCVOICEENGINE_H_