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

content/browser/renderer_host/media/audio_renderer_host.h

Index: content/browser/renderer_host/media/audio_renderer_host.h
diff --git a/content/browser/renderer_host/media/audio_renderer_host.h b/content/browser/renderer_host/media/audio_renderer_host.h
index eb3c90f21a602e0586a41fc0e2d01bdb8f87a99d..740225affe7397506afbf26a96458b2035f2b050 100644
--- a/content/browser/renderer_host/media/audio_renderer_host.h
+++ b/content/browser/renderer_host/media/audio_renderer_host.h
@@ -80,11 +80,6 @@ class CONTENT_EXPORT AudioRendererHost
                     MediaStreamManager* media_stream_manager,
                     const std::string& salt);
 
-  // Calls |callback| with the list of AudioOutputControllers for this object.
-  void GetOutputControllers(
-      const RenderProcessHost::GetAudioOutputControllersCallback&
-          callback) const;
-
   // BrowserMessageFilter implementation.
   void OnChannelClosing() override;
   void OnDestruct() const override;
@@ -180,8 +175,6 @@ class CONTENT_EXPORT AudioRendererHost
   // the number of playing streams.
   void StreamStateChanged(int stream_id, bool is_playing);
 
-  RenderProcessHost::AudioOutputControllerList DoGetOutputControllers() const;
-
   // Send an error message to the renderer.
   void SendErrorMessage(int stream_id);