Migrate webkit/renderer/media/ to content/renderer/media/.

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 "base/callback.h"
 #include "base/memory/ref_counted.h"
 #include "base/synchronization/lock.h"
 #include "base/threading/thread_checker.h"
 #include "content/common/content_export.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 "webkit/renderer/media/media_stream_audio_renderer.h"
 
 namespace media {
 class AudioBus;
 class AudioFifo;
 class AudioOutputDevice;
 class AudioParameters;
 }
 
 namespace content {
 
 class WebRtcAudioCapturer;
 
-// WebRtcLocalAudioRenderer is a webkit_media::MediaStreamAudioRenderer
-// designed for rendering local audio media stream tracks,
+// 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
 // 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 webkit_media::MediaStreamAudioRenderer),
+    : NON_EXPORTED_BASE(public MediaStreamAudioRenderer),
       NON_EXPORTED_BASE(public media::AudioRendererSink::RenderCallback),
       NON_EXPORTED_BASE(public WebRtcAudioCapturerSink) {
  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,
                            int source_render_view_id);
 
-  // webkit_media::MediaStreamAudioRenderer implementation.
+  // 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:
-  // content::WebRtcAudioCapturerSink implementation.
+  // WebRtcAudioCapturerSink implementation.
 
   // Called on the AudioInputDevice worker thread.
   virtual void CaptureData(const int16* audio_data,
                            int number_of_channels,
                            int number_of_frames,
                            int audio_delay_milliseconds,
                            double volume) OVERRIDE;
 
   // Can be called on different user thread.
   virtual void SetCaptureFormat(const media::AudioParameters& params) OVERRIDE;
@@ skipping 41 matching lines @@
 
   // Protects |loopback_fifo_|, |playing_| and |sink_|.
   mutable base::Lock thread_lock_;
 
   DISALLOW_COPY_AND_ASSIGN(WebRtcLocalAudioRenderer);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_RENDERER_H_