WebRTC Audio private API: removing WebRtcAudioPrivate(Set/Get)ActiveSinkFunction

media/audio/audio_output_delegate.h

 // Copyright 2016 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #ifndef MEDIA_AUDIO_AUDIO_OUTPUT_DELEGATE_H_
 #define MEDIA_AUDIO_AUDIO_OUTPUT_DELEGATE_H_
 
 #include <memory>
 #include <string>
 
@@ skipping 25 matching lines @@
                                  base::SharedMemory* shared_memory,
                                  base::CancelableSyncSocket* socket) = 0;
 
     // Called if stream encounters an error and has become unusable.
     virtual void OnStreamError(int stream_id) = 0;
   };
 
   AudioOutputDelegate();
   virtual ~AudioOutputDelegate();
 
-  // TODO(maxmorin): Remove GetController() when crbug.com/647185 is closed.
-  // This function is used to provide control of the audio stream to
-  // WebrtcAudioPrivateGetActiveSinkFunction and others in the webrtc extension
-  // API. Since the controller is shared, this means that it might outlive the
-  // AudioOutputDelegate. In this case, it is still safe to call functions on
-  // the controller, but it will not do anything. The controller is also shared
-  // with AudioStreamMonitor.
-  virtual scoped_refptr<AudioOutputController> GetController() const = 0;
   virtual int GetStreamId() const = 0;
 
   // Stream control:
   virtual void OnPlayStream() = 0;
   virtual void OnPauseStream() = 0;
   virtual void OnSetVolume(double volume) = 0;
 };
 
 }  // namespace media
 
 #endif  // MEDIA_AUDIO_AUDIO_OUTPUT_DELEGATE_H_