move the APM to chrome.

content/renderer/media/webrtc_local_audio_track.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_TRACK_H_
 #define CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_TRACK_H_
 
 #include <list>
 #include <string>
 
 #include "base/synchronization/lock.h"
 #include "base/threading/thread_checker.h"
 #include "content/renderer/media/webrtc_audio_device_impl.h"
+#include "content/renderer/media/webrtc_local_audio_source_provider.h"
 #include "third_party/libjingle/source/talk/app/webrtc/mediaconstraintsinterface.h"
 #include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h"
 #include "third_party/libjingle/source/talk/app/webrtc/mediastreamtrack.h"
 #include "third_party/libjingle/source/talk/media/base/audiorenderer.h"
 
 namespace cricket {
 class AudioRenderer;
 }  // namespace cricket
 
 namespace media {
 class AudioBus;
 }  // namespace media
 
 namespace content {
 
 class WebAudioCapturerSource;
 class WebRtcAudioCapturer;
 class WebRtcAudioCapturerSinkOwner;
+class WebRtcAudioProcessor;
 
 // A WebRtcLocalAudioTrack instance contains the implementations of
 // MediaStreamTrack and WebRtcAudioCapturerSink.
 // When an instance is created, it will register itself as a track to the
 // WebRtcAudioCapturer to get the captured data, and forward the data to
 // its |sinks_|. The data flow can be stopped by disabling the audio track.
 class CONTENT_EXPORT WebRtcLocalAudioTrack
     : NON_EXPORTED_BASE(public cricket::AudioRenderer),
       NON_EXPORTED_BASE(
-          public webrtc::MediaStreamTrack<webrtc::AudioTrackInterface>) {
+          public webrtc::MediaStreamTrack<webrtc::AudioTrackInterface>),
+      NON_EXPORTED_BASE(public WebRtcAudioRenderDataObserver) {
  public:
   static scoped_refptr<WebRtcLocalAudioTrack> Create(
       const std::string& id,
       const scoped_refptr<WebRtcAudioCapturer>& capturer,
       WebAudioCapturerSource* webaudio_source,
       webrtc::AudioSourceInterface* track_source,
-      const webrtc::MediaConstraintsInterface* constraints);
+      const webrtc::MediaConstraintsInterface* constraints,
+      WebRtcAudioDeviceImpl* audio_device);
 
   // Add a sink to the track. This function will trigger a SetCaptureFormat()
   // call on the |sink|.
   // Called on the main render thread.
   void AddSink(WebRtcAudioCapturerSink* sink);
 
   // Remove a sink from the track.
   // Called on the main render thread.
   void RemoveSink(WebRtcAudioCapturerSink* sink);
 
   // Starts the local audio track. Called on the main render thread and
   // should be called only once when audio track is created.
   void Start();
 
-  // Stops the local audio track. Called on the main render thread and
-  // should be called only once when audio track going away.
+  // Stops the local audio track. Called on the main render thread by the
+  // capturer and should be called only once when audio track going away.
   void Stop();
 
   // Method called by the capturer to deliver the capture data.
   void Capture(media::AudioBus* audio_source,
                int audio_delay_milliseconds,
                int volume,
                bool key_pressed);
 
   // Method called by the capturer to set the audio parameters used by source
   // of the capture data..
   // Can be called on different user threads.
   void SetCaptureFormat(const media::AudioParameters& params);
 
+  WebKit::WebAudioSourceProvider* audio_source_provider() const {
+    return source_provider_.get();
+  }
+
  protected:
   WebRtcLocalAudioTrack(
       const std::string& label,
       const scoped_refptr<WebRtcAudioCapturer>& capturer,
       WebAudioCapturerSource* webaudio_source,
       webrtc::AudioSourceInterface* track_source,
-      const webrtc::MediaConstraintsInterface* constraints);
+      const webrtc::MediaConstraintsInterface* constraints,
+       WebRtcAudioDeviceImpl* audio_device);
 
   virtual ~WebRtcLocalAudioTrack();
 
  private:
   typedef std::list<scoped_refptr<WebRtcAudioCapturerSinkOwner> > SinkList;
 
   // cricket::AudioCapturer implementation.
   virtual void AddChannel(int channel_id) OVERRIDE;
   virtual void RemoveChannel(int channel_id) OVERRIDE;
 
   // webrtc::AudioTrackInterface implementation.
   virtual webrtc::AudioSourceInterface* GetSource() const OVERRIDE;
   virtual cricket::AudioRenderer* GetRenderer() OVERRIDE;
 
   // webrtc::MediaStreamTrack implementation.
   virtual std::string kind() const OVERRIDE;
 
+  // WebRtcAudioRenderDataObserver implementation.
+  virtual void OnRenderData(const int16* render_audio,
+                            int sample_rate,
+                            int number_of_channels,
+                            int number_of_frames,
+                            int render_delay_ms) OVERRIDE;
+  virtual void OnRenderClosing() OVERRIDE;
+
   // The provider of captured data to render.
   // The WebRtcAudioCapturer is today created by WebRtcAudioDeviceImpl.
   scoped_refptr<WebRtcAudioCapturer> capturer_;
 
   // The source of the audio track which is used by WebAudio, which provides
   // data to the audio track when hooking up with WebAudio.
   scoped_refptr<WebAudioCapturerSource> webaudio_source_;
 
   // The source of the audio track which handles the audio constraints.
   // TODO(xians): merge |track_source_| to |capturer_|.
   talk_base::scoped_refptr<webrtc::AudioSourceInterface> track_source_;
 
   // A list of sinks that the audio data is fed to.
   SinkList sinks_;
 
   // Used to DCHECK that we are called on the correct thread.
   base::ThreadChecker thread_checker_;
 
   // Protects |params_| and |sinks_|.
   mutable base::Lock lock_;
 
   // A vector of WebRtc VoE channels that the capturer sends datat to.
   std::vector<int> voe_channels_;
 
   bool need_audio_processing_;
 
-  // Buffers used for temporary storage during capture callbacks.
-  // Allocated during initialization.
-  class ConfiguredBuffer;
-  scoped_refptr<ConfiguredBuffer> buffer_;
+  // Cached audio parameters for the source.
+  media::AudioParameters source_params_;
+
+  // Cached audio parameters for the sinks.
+  media::AudioParameters sink_params_;
+
+  scoped_ptr<WebRtcAudioProcessor> audio_processor_;
+
+  // The source provider to feed the track data to other clients like
+  // WebAudio.
+  const scoped_ptr<WebRtcLocalAudioSourceProvider> source_provider_;
+
+  WebRtcAudioDeviceImpl* audio_device_;
 
   DISALLOW_COPY_AND_ASSIGN(WebRtcLocalAudioTrack);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_TRACK_H_