move the APM to chrome.

content/renderer/media/webrtc_local_audio_track.h

Index: content/renderer/media/webrtc_local_audio_track.h
diff --git a/content/renderer/media/webrtc_local_audio_track.h b/content/renderer/media/webrtc_local_audio_track.h
index 01b11208021d3e32198eb3061f1895957fb7f29d..64f20fe70b1d9bac3a60823bc84ba9639b31f355 100644
--- a/content/renderer/media/webrtc_local_audio_track.h
+++ b/content/renderer/media/webrtc_local_audio_track.h
@@ -11,6 +11,7 @@
 #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"
@@ -29,6 +30,7 @@ namespace content {
 class WebAudioCapturerSource;
 class WebRtcAudioCapturer;
 class WebRtcAudioCapturerSinkOwner;
+class WebRtcAudioProcessor;
 
 // A WebRtcLocalAudioTrack instance contains the implementations of
 // MediaStreamTrack and WebRtcAudioCapturerSink.
@@ -38,14 +40,16 @@ class WebRtcAudioCapturerSinkOwner;
 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|.
@@ -60,8 +64,8 @@ class CONTENT_EXPORT WebRtcLocalAudioTrack
   // 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.
@@ -75,13 +79,18 @@ class CONTENT_EXPORT WebRtcLocalAudioTrack
   // 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();
 
@@ -99,6 +108,14 @@ class CONTENT_EXPORT WebRtcLocalAudioTrack
   // 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_;
@@ -125,10 +142,19 @@ class CONTENT_EXPORT WebRtcLocalAudioTrack
 
   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);
 };