move the APM to chrome.

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 "base/memory/scoped_ptr.h"
 #include "base/synchronization/lock.h"
 #include "base/threading/thread_checker.h"
@@ skipping 31 matching lines @@
  public:
   static const size_t kWebAudioRenderBufferSize;
 
   WebRtcLocalAudioSourceProvider();
   virtual ~WebRtcLocalAudioSourceProvider();
 
   // Initialize function for the souce provider. This can be called multiple
   // times if the source format has changed.
   void Initialize(const media::AudioParameters& source_params);
 
+  // TODO(xians): Merge with the sink interface.
   // Called by the WebRtcAudioCapturer to deliever captured data into fifo on
   // the capture audio thread.
-  void DeliverData(media::AudioBus* audio_source,
-                   int audio_delay_milliseconds,
-                   int volume,
-                   bool key_pressed);
+  void DeliverData(const int16* data,
+                   int sample_rate,
+                   int number_of_channels,
+                   int number_of_frames);
 
   // Called by the WebAudioCapturerSource to get the audio processing params.
   // This function is triggered by provideInput() on the WebAudio audio thread,
   // so it has been under the protection of |lock_|.
   void GetAudioProcessingParams(int* delay_ms, int* volume, bool* key_pressed);
 
   // WebKit::WebAudioSourceProvider implementation.
   virtual void setClient(WebKit::WebAudioSourceProviderClient* client) OVERRIDE;
   virtual void provideInput(const WebKit::WebVector<float*>& audio_data,
                             size_t number_of_frames) OVERRIDE;
@@ skipping 32 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_