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

content/renderer/media/webrtc_local_audio_renderer.h

 // Copyright (c) 2012 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_RENDERER_H_
 #define CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_RENDERER_H_
 
 #include <vector>
 
 #include "base/callback.h"
 #include "base/memory/ref_counted.h"
 #include "base/message_loop/message_loop_proxy.h"
 #include "base/synchronization/lock.h"
 #include "base/threading/thread_checker.h"
 #include "content/common/content_export.h"
+#include "content/public/renderer/media_stream_audio_sink.h"
 #include "content/renderer/media/media_stream_audio_renderer.h"
 #include "content/renderer/media/webrtc_audio_device_impl.h"
 #include "content/renderer/media/webrtc_local_audio_track.h"
+#include "third_party/WebKit/public/platform/WebMediaStreamTrack.h"
 
 namespace media {
 class AudioBus;
 class AudioFifo;
 class AudioOutputDevice;
 class AudioParameters;
 }
 
 namespace content {
 
 class WebRtcAudioCapturer;
 
 // WebRtcLocalAudioRenderer is a MediaStreamAudioRenderer designed for rendering
 // local audio media stream tracks,
 // http://dev.w3.org/2011/webrtc/editor/getusermedia.html#mediastreamtrack
 // It also implements media::AudioRendererSink::RenderCallback to render audio
 // data provided from a WebRtcLocalAudioTrack source.
 // When the audio layer in the browser process asks for data to render, this
-// class provides the data by implementing the WebRtcAudioCapturerSink
+// class provides the data by implementing the MediaStreamAudioSink
 // interface, i.e., we are a sink seen from the WebRtcAudioCapturer perspective.
 // TODO(henrika): improve by using similar principles as in RTCVideoRenderer
 // which register itself to the video track when the provider is started and
 // deregisters itself when it is stopped.
 // Tracking this at http://crbug.com/164813.
 class CONTENT_EXPORT WebRtcLocalAudioRenderer
     : NON_EXPORTED_BASE(public MediaStreamAudioRenderer),
-      NON_EXPORTED_BASE(public media::AudioRendererSink::RenderCallback),
-      NON_EXPORTED_BASE(public WebRtcAudioCapturerSink) {
+      NON_EXPORTED_BASE(public MediaStreamAudioSink),
+      NON_EXPORTED_BASE(public media::AudioRendererSink::RenderCallback) {
  public:
   // Creates a local renderer and registers a capturing |source| object.
   // The |source| is owned by the WebRtcAudioDeviceImpl.
   // Called on the main thread.
-  WebRtcLocalAudioRenderer(WebRtcLocalAudioTrack* audio_track,
+  WebRtcLocalAudioRenderer(const blink::WebMediaStreamTrack& audio_track,
                            int source_render_view_id,
                            int session_id,
                            int frames_per_buffer);
 
   // MediaStreamAudioRenderer implementation.
   // Called on the main thread.
   virtual void Start() OVERRIDE;
   virtual void Stop() OVERRIDE;
   virtual void Play() OVERRIDE;
   virtual void Pause() OVERRIDE;
   virtual void SetVolume(float volume) OVERRIDE;
   virtual base::TimeDelta GetCurrentRenderTime() const OVERRIDE;
   virtual bool IsLocalRenderer() const OVERRIDE;
 
   const base::TimeDelta& total_render_time() const {
     return total_render_time_;
   }
 
  protected:
   virtual ~WebRtcLocalAudioRenderer();
 
  private:
-  // WebRtcAudioCapturerSink implementation.
+  // MediaStreamAudioSink implementation.
 
   // Called on the AudioInputDevice worker thread.
-  virtual int CaptureData(const std::vector<int>& channels,
-                          const int16* audio_data,
-                          int sample_rate,
-                          int number_of_channels,
-                          int number_of_frames,
-                          int audio_delay_milliseconds,
-                          int current_volume,
-                          bool need_audio_processing,
-                          bool key_pressed) OVERRIDE;
+  virtual void OnData(const int16* audio_data,
+                      int sample_rate,
+                      int number_of_channels,
+                      int number_of_frames) OVERRIDE;
 
-  // Can be called on different user thread.
-  virtual void SetCaptureFormat(const media::AudioParameters& params) OVERRIDE;
+  // Called on the AudioInputDevice worker thread.
+  virtual void OnSetFormat(const media::AudioParameters& params) OVERRIDE;
 
   // media::AudioRendererSink::RenderCallback implementation.
   // Render() is called on the AudioOutputDevice thread and OnRenderError()
   // on the IO thread.
   virtual int Render(media::AudioBus* audio_bus,
                      int audio_delay_milliseconds) OVERRIDE;
   virtual void OnRenderError() OVERRIDE;
 
   // Initializes and starts the |sink_| if
   //  we have received valid |source_params_| &&
   //  |playing_| has been set to true &&
   //  |volume_| is not zero.
   void MaybeStartSink();
 
   // Sets new |source_params_| and then re-initializes and restarts |sink_|.
   void ReconfigureSink(const media::AudioParameters& params);
 
   // The audio track which provides data to render. Given that this class
   // implements local loopback, the audio track is getting data from a capture
   // instance like a selected microphone and forwards the recorded data to its
   // sinks. The recorded data is stored in a FIFO and consumed
   // by this class when the sink asks for new data.
-  // The WebRtcAudioCapturer is today created by WebRtcAudioDeviceImpl.
-  scoped_refptr<WebRtcLocalAudioTrack> audio_track_;
+  // This class is calling MediaStreamAudioSink::AddToAudioTrack() and
+  // MediaStreamAudioSink::RemoveFromAudioTrack() to connect and disconnect
+  // with the audio track.
+  blink::WebMediaStreamTrack audio_track_;
 
   // The render view in which the audio is rendered into |sink_|.
   const int source_render_view_id_;
   const int session_id_;
 
   // MessageLoop associated with the single thread that performs all control
   // tasks.  Set to the MessageLoop that invoked the ctor.
   const scoped_refptr<base::MessageLoopProxy> message_loop_;
 
   // The sink (destination) for rendered audio.
@@ skipping 37 matching lines @@
 
   // Used to DCHECK that some methods are called on the capture audio thread.
   base::ThreadChecker capture_thread_checker_;
 
   DISALLOW_COPY_AND_ASSIGN(WebRtcLocalAudioRenderer);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_RENDERER_H_