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

content/renderer/media/webrtc_local_audio_source_provider.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_SOURCE_PROVIDER_H_
 #define CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_SOURCE_PROVIDER_H_
 
+#include <vector>
+
 #include "base/memory/scoped_ptr.h"
 #include "base/synchronization/lock.h"
 #include "base/threading/thread_checker.h"
 #include "base/time/time.h"
 #include "content/common/content_export.h"
-#include "content/renderer/media/webrtc_audio_device_impl.h"
+#include "content/public/renderer/media_stream_audio_sink.h"
 #include "media/base/audio_converter.h"
 #include "third_party/WebKit/public/platform/WebAudioSourceProvider.h"
 #include "third_party/WebKit/public/platform/WebVector.h"
 
 namespace media {
 class AudioBus;
 class AudioConverter;
 class AudioFifo;
 class AudioParameters;
 }
 
 namespace blink {
 class WebAudioSourceProviderClient;
 }
 
 namespace content {
 
 // WebRtcLocalAudioSourceProvider provides a bridge between classes:
 //     WebRtcAudioCapturer ---> blink::WebAudioSourceProvider
 //
 // WebRtcLocalAudioSourceProvider works as a sink to the WebRtcAudiocapturer
 // and store the capture data to a FIFO. When the media stream is connected to
 // WebAudio as a source provider, WebAudio will periodically call
 // provideInput() to get the data from the FIFO.
 //
 // All calls are protected by a lock.
 class CONTENT_EXPORT WebRtcLocalAudioSourceProvider
-    : NON_EXPORTED_BASE(public media::AudioConverter::InputCallback),
-      NON_EXPORTED_BASE(public blink::WebAudioSourceProvider),
-      NON_EXPORTED_BASE(public WebRtcAudioCapturerSink) {
+    :  NON_EXPORTED_BASE(public blink::WebAudioSourceProvider),
+       NON_EXPORTED_BASE(public media::AudioConverter::InputCallback),
+       NON_EXPORTED_BASE(public MediaStreamAudioSink) {
  public:
   static const size_t kWebAudioRenderBufferSize;
 
   WebRtcLocalAudioSourceProvider();
   virtual ~WebRtcLocalAudioSourceProvider();
 
-  // WebRtcAudioCapturerSink implementation.
-  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 SetCaptureFormat(const media::AudioParameters& params) OVERRIDE;
+  // MediaStreamAudioSink implementation.
+  virtual void OnData(const int16* audio_data,
+                      int sample_rate,
+                      int number_of_channels,
+                      int number_of_frames) OVERRIDE;
+  virtual void OnSetFormat(const media::AudioParameters& params) OVERRIDE;
 
   // blink::WebAudioSourceProvider implementation.
   virtual void setClient(blink::WebAudioSourceProviderClient* client) OVERRIDE;
   virtual void provideInput(const blink::WebVector<float*>& audio_data,
                             size_t number_of_frames) OVERRIDE;
 
   // media::AudioConverter::Inputcallback implementation.
   // This function is triggered by provideInput()on the WebAudio audio thread,
   // so it has been under the protection of |lock_|.
   virtual double ProvideInput(media::AudioBus* audio_bus,
@@ skipping 25 matching lines @@
 
   // Used to report the correct delay to |webaudio_source_|.
   base::TimeTicks last_fill_;
 
   DISALLOW_COPY_AND_ASSIGN(WebRtcLocalAudioSourceProvider);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_SOURCE_PROVIDER_H_