Add generic interfaces for the sinks of the media stream audio track

content/renderer/media/webrtc_local_audio_track.h

 // Copyright 2013 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 CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_TRACK_H_
 #define CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_TRACK_H_
 
 #include <list>
 #include <string>
 
 #include "base/synchronization/lock.h"
 #include "base/threading/thread_checker.h"
-#include "content/renderer/media/webrtc_audio_device_impl.h"
 #include "content/renderer/media/webrtc_local_audio_source_provider.h"
 #include "third_party/libjingle/source/talk/app/webrtc/mediaconstraintsinterface.h"
 #include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h"
 #include "third_party/libjingle/source/talk/app/webrtc/mediastreamtrack.h"
 #include "third_party/libjingle/source/talk/media/base/audiorenderer.h"
 
 namespace cricket {
 class AudioRenderer;
 }  // namespace cricket
 
 namespace media {
 class AudioBus;
 }  // namespace media
 
 namespace content {
 
+class MediaStreamAudioSink;
+class MediaStreamAudioSinkOwner;
+class PeerConnectionAudioSink;
 class WebAudioCapturerSource;
 class WebRtcAudioCapturer;
-class WebRtcAudioCapturerSinkOwner;
 
 // A WebRtcLocalAudioTrack instance contains the implementations of
-// MediaStreamTrack and WebRtcAudioCapturerSink.
+// MediaStreamTrack and MediaStreamAudioSink.
 // When an instance is created, it will register itself as a track to the
 // WebRtcAudioCapturer to get the captured data, and forward the data to
 // its |sinks_|. The data flow can be stopped by disabling the audio track.
 class CONTENT_EXPORT WebRtcLocalAudioTrack
     : NON_EXPORTED_BASE(public cricket::AudioRenderer),
       NON_EXPORTED_BASE(
           public webrtc::MediaStreamTrack<webrtc::AudioTrackInterface>) {
  public:
   static scoped_refptr<WebRtcLocalAudioTrack> Create(
       const std::string& id,
       const scoped_refptr<WebRtcAudioCapturer>& capturer,
       WebAudioCapturerSource* webaudio_source,
       webrtc::AudioSourceInterface* track_source,
       const webrtc::MediaConstraintsInterface* constraints);
 
-  // Add a sink to the track. This function will trigger a SetCaptureFormat()
+  // Add a sink to the track. This function will trigger a OnSetFormat()
   // call on the |sink|.
   // Called on the main render thread.
-  void AddSink(WebRtcAudioCapturerSink* sink);
+  void AddSink(MediaStreamAudioSink* sink);
 
   // Remove a sink from the track.
   // Called on the main render thread.
-  void RemoveSink(WebRtcAudioCapturerSink* sink);
+  void RemoveSink(MediaStreamAudioSink* sink);
+
+  // Add/remove PeerConnection sink to/from the track.
+  // TODO(xians): Remove these two methods after PeerConnection can use the
+  // same sink interface as MediaStreamAudioSink.
+  void AddSink(PeerConnectionAudioSink* sink);
+  void RemoveSink(PeerConnectionAudioSink* sink);
 
   // Starts the local audio track. Called on the main render thread and
   // should be called only once when audio track is created.
   void Start();
 
   // Stops the local audio track. Called on the main render thread and
   // should be called only once when audio track going away.
   void Stop();
 
   // Method called by the capturer to deliver the capture data.
   // Call on the capture audio thread.
   void Capture(media::AudioBus* audio_source,
                int audio_delay_milliseconds,
                int volume,
                bool key_pressed);
 
   // Method called by the capturer to set the audio parameters used by source
   // of the capture data..
   // Call on the capture audio thread.
-  void SetCaptureFormat(const media::AudioParameters& params);
+  void OnSetFormat(const media::AudioParameters& params);
 
   blink::WebAudioSourceProvider* audio_source_provider() const {
     return source_provider_.get();
   }
 
  protected:
   WebRtcLocalAudioTrack(
       const std::string& label,
       const scoped_refptr<WebRtcAudioCapturer>& capturer,
       WebAudioCapturerSource* webaudio_source,
       webrtc::AudioSourceInterface* track_source,
       const webrtc::MediaConstraintsInterface* constraints);
 
   virtual ~WebRtcLocalAudioTrack();
 
  private:
-  typedef std::list<scoped_refptr<WebRtcAudioCapturerSinkOwner> > SinkList;
+  typedef std::list<scoped_refptr<MediaStreamAudioSinkOwner> > SinkList;
 
   // cricket::AudioCapturer implementation.
   virtual void AddChannel(int channel_id) OVERRIDE;
   virtual void RemoveChannel(int channel_id) OVERRIDE;
 
   // webrtc::AudioTrackInterface implementation.
   virtual webrtc::AudioSourceInterface* GetSource() const OVERRIDE;
   virtual cricket::AudioRenderer* GetRenderer() OVERRIDE;
 
   // webrtc::MediaStreamTrack implementation.
@@ skipping 40 matching lines @@
   // The source provider to feed the track data to other clients like
   // WebAudio.
   scoped_ptr<WebRtcLocalAudioSourceProvider> source_provider_;
 
   DISALLOW_COPY_AND_ASSIGN(WebRtcLocalAudioTrack);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_TRACK_H_