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

talk/media/webrtc/webrtcvideoengine2.h

 /*
  * libjingle
  * Copyright 2014 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 175 matching lines @@
   // VideoMediaChannel implementation
   void DetachVoiceChannel() override;
   bool SetSendParameters(const VideoSendParameters& params) override;
   bool SetRecvParameters(const VideoRecvParameters& params) override;
   bool SetRecvCodecs(const std::vector<VideoCodec>& codecs) override;
   bool SetSendCodecs(const std::vector<VideoCodec>& codecs) override;
   bool GetSendCodec(VideoCodec* send_codec) override;
   bool SetSendStreamFormat(uint32 ssrc, const VideoFormat& format) override;
   bool SetRender(bool render) override;
   bool SetSend(bool send) override;
-
+  bool SetVideoSend(uint32 ssrc, bool mute,
+                    const VideoOptions* options) override;
   bool AddSendStream(const StreamParams& sp) override;
   bool RemoveSendStream(uint32 ssrc) override;
   bool AddRecvStream(const StreamParams& sp) override;
   bool AddRecvStream(const StreamParams& sp, bool default_stream);
   bool RemoveRecvStream(uint32 ssrc) override;
   bool SetRenderer(uint32 ssrc, VideoRenderer* renderer) override;
   bool GetStats(VideoMediaInfo* info) override;
   bool SetCapturer(uint32 ssrc, VideoCapturer* capturer) override;
   bool SendIntraFrame() override;
   bool RequestIntraFrame() override;
 
   void OnPacketReceived(rtc::Buffer* packet,
                         const rtc::PacketTime& packet_time) override;
   void OnRtcpReceived(rtc::Buffer* packet,
                       const rtc::PacketTime& packet_time) override;
   void OnReadyToSend(bool ready) override;
-  bool MuteStream(uint32 ssrc, bool mute) override;
 
   // Set send/receive RTP header extensions. This must be done before creating
   // streams as it only has effect on future streams.
   bool SetRecvRtpHeaderExtensions(
       const std::vector<RtpHeaderExtension>& extensions) override;
   bool SetSendRtpHeaderExtensions(
       const std::vector<RtpHeaderExtension>& extensions) override;
   bool SetMaxSendBandwidth(int bps) override;
   bool SetOptions(const VideoOptions& options) override;
   void SetInterface(NetworkInterface* iface) override;
   void UpdateAspectRatio(int ratio_w, int ratio_h) override;
 
   void OnMessage(rtc::Message* msg) override;
 
   void OnLoadUpdate(Load load) override;
 
   // Implemented for VideoMediaChannelTest.
   bool sending() const { return sending_; }
   uint32 GetDefaultSendChannelSsrc() { return default_send_ssrc_; }
   bool GetRenderer(uint32 ssrc, VideoRenderer** renderer);
 
  private:
+  bool MuteStream(uint32 ssrc, bool mute);
   class WebRtcVideoReceiveStream;
   void ConfigureReceiverRtp(webrtc::VideoReceiveStream::Config* config,
                             const StreamParams& sp) const;
   bool CodecIsExternallySupported(const std::string& name) const;
   bool ValidateSendSsrcAvailability(const StreamParams& sp) const
       EXCLUSIVE_LOCKS_REQUIRED(stream_crit_);
   bool ValidateReceiveSsrcAvailability(const StreamParams& sp) const
       EXCLUSIVE_LOCKS_REQUIRED(stream_crit_);
   void DeleteReceiveStream(WebRtcVideoReceiveStream* stream)
       EXCLUSIVE_LOCKS_REQUIRED(stream_crit_);
@@ skipping 293 matching lines @@
   WebRtcVideoDecoderFactory* const external_decoder_factory_;
   std::vector<VideoCodecSettings> recv_codecs_;
   std::vector<webrtc::RtpExtension> recv_rtp_extensions_;
   webrtc::Call::Config::BitrateConfig bitrate_config_;
   VideoOptions options_;
 };
 
 }  // namespace cricket
 
 #endif  // TALK_MEDIA_WEBRTC_WEBRTCVIDEOENGINE2_H_