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"
 #include "base/time/time.h"
 #include "content/common/content_export.h"
+#include "content/renderer/media/webrtc_audio_device_impl.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 WebKit {
 class WebAudioSourceProviderClient;
 }
 
 namespace content {
 
 // WebRtcLocalAudioSourceProvider provides a bridge between classes:
 //     WebRtcAudioCapturer ---> WebKit::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 WebKit::WebAudioSourceProvider) {
+      NON_EXPORTED_BASE(public WebKit::WebAudioSourceProvider),
+      NON_EXPORTED_BASE(public WebRtcAudioCapturerSink) {
  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);
+  // 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;
 
-  // 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);
-
-  // 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);
+  virtual void SetCaptureFormat(const media::AudioParameters& params) OVERRIDE;
 
   // 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;
 
   // 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,
                               base::TimeDelta buffer_delay) OVERRIDE;
 
   // Method to allow the unittests to inject its own sink parameters to avoid
   // query the hardware.
   // TODO(xians,tommi): Remove and instead offer a way to inject the sink
   // parameters so that the implementation doesn't rely on the global default
   // hardware config but instead gets the parameters directly from the sink
   // (WebAudio in this case). Ideally the unit test should be able to use that
   // same mechanism to inject the sink parameters for testing.
   void SetSinkParamsForTesting(const media::AudioParameters& sink_params);
 
  private:
   // Used to DCHECK that we are called on the correct thread.
   base::ThreadChecker thread_checker_;
 
   scoped_ptr<media::AudioConverter> audio_converter_;
   scoped_ptr<media::AudioFifo> fifo_;
-  scoped_ptr<media::AudioBus> bus_wrapper_;
-  int audio_delay_ms_;
-  int volume_;
-  bool key_pressed_;
+  scoped_ptr<media::AudioBus> input_wrapper_;
+  scoped_ptr<media::AudioBus> output_wrapper_;
   bool is_enabled_;
   media::AudioParameters source_params_;
   media::AudioParameters sink_params_;
 
   // Protects all the member variables above.
   base::Lock lock_;
 
   // 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_