Remove MediaChannel::SetRemoteRenderer().

talk/app/webrtc/mediastreamprovider.h

 /*
  * libjingle
  * Copyright 2012 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 37 matching lines @@
 
 // TODO(deadbeef): These interfaces are effectively just a way for the
 // RtpSenders/Receivers to get to the BaseChannels. These interfaces should be
 // refactored away eventually, as the classes converge.
 
 // This interface is called by AudioTrackHandler classes in mediastreamhandler.h
 // to change the settings of an audio track connected to certain PeerConnection.
 class AudioProviderInterface {
  public:
   // Enable/disable the audio playout of a remote audio track with |ssrc|.
-  virtual void SetAudioPlayout(uint32_t ssrc,
-                               bool enable,
-                               cricket::AudioRenderer* renderer) = 0;
+  virtual void SetAudioPlayout(uint32_t ssrc, bool enable) = 0;
   // Enable/disable sending audio on the local audio track with |ssrc|.
   // When |enable| is true |options| should be applied to the audio track.
   virtual void SetAudioSend(uint32_t ssrc,
                             bool enable,
                             const cricket::AudioOptions& options,
                             cricket::AudioRenderer* renderer) = 0;
 
   // Sets the audio playout volume of a remote audio track with |ssrc|.
   // |volume| is in the range of [0, 10].
   virtual void SetAudioPlayoutVolume(uint32_t ssrc, double volume) = 0;
@@ skipping 18 matching lines @@
                             bool enable,
                             const cricket::VideoOptions* options) = 0;
 
  protected:
   virtual ~VideoProviderInterface() {}
 };
 
 }  // namespace webrtc
 
 #endif  // TALK_APP_WEBRTC_MEDIASTREAMPROVIDER_H_