move the APM to chrome.

content/renderer/media/webrtc_audio_capturer.h

Index: content/renderer/media/webrtc_audio_capturer.h
diff --git a/content/renderer/media/webrtc_audio_capturer.h b/content/renderer/media/webrtc_audio_capturer.h
index 370cfe509cb38336361628ee1dd1d1f30654c9b3..4fb0ec75dd19c70a2765dd3a3c5d23e35141c8c4 100644
--- a/content/renderer/media/webrtc_audio_capturer.h
+++ b/content/renderer/media/webrtc_audio_capturer.h
@@ -13,7 +13,6 @@
 #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 "media/audio/audio_input_device.h"
 #include "media/base/audio_capturer_source.h"
 
@@ -68,11 +67,11 @@ class CONTENT_EXPORT WebRtcAudioCapturer
   // Called on the main render thread or libjingle working thread.
   void RemoveTrack(WebRtcLocalAudioTrack* track);
 
-  // SetCapturerSource() is called if the client on the source side desires to
-  // provide their own captured audio data. Client is responsible for calling
-  // Start() on its own source to have the ball rolling.
+  // InitializeCapturerSource() is called if the client on the source side
+  // desires to provide their own captured audio data. Client is responsible
+  // for calling Start() on its own source to have the ball rolling.
   // Called on the main render thread.
-  void SetCapturerSource(
+  void InitializeCapturerSource(
       const scoped_refptr<media::AudioCapturerSource>& source,
       media::ChannelLayout channel_layout,
       float sample_rate);
@@ -111,13 +110,15 @@ class CONTENT_EXPORT WebRtcAudioCapturer
   const std::string& device_id() const { return device_id_; }
   int session_id() const { return session_id_; }
 
-  WebKit::WebAudioSourceProvider* audio_source_provider() const {
-    return source_provider_.get();
-  }
-
   // Stops recording audio.
   void Stop();
 
+  // 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_|.
+  // TODO(xians): Remove after moving APM from WebRtc to Chrome.
+  void GetAudioProcessingParams(int* delay_ms, int* volume, bool* key_pressed);
+
  protected:
   friend class base::RefCountedThreadSafe<WebRtcAudioCapturer>;
   WebRtcAudioCapturer();
@@ -144,8 +145,6 @@ class CONTENT_EXPORT WebRtcAudioCapturer
   // thread. It should NOT be called under |lock_|.
   void Start();
 
-
-
   // Helper function to get the buffer size based on |peer_connection_mode_|
   // and sample rate;
   int GetBufferSize(int sample_rate) const;
@@ -154,7 +153,7 @@ class CONTENT_EXPORT WebRtcAudioCapturer
   base::ThreadChecker thread_checker_;
 
   // Protects |source_|, |audio_tracks_|, |running_|, |loopback_fifo_|,
-  // |params_|, |buffering_| and |agc_is_enabled_|.
+  // |source_params_|, |buffering_| and |agc_is_enabled_|.
   mutable base::Lock lock_;
 
   // A list of audio tracks that the audio data is fed to.
@@ -188,18 +187,16 @@ class CONTENT_EXPORT WebRtcAudioCapturer
   // Range is [0, 255].
   int volume_;
 
-  // The source provider to feed the capture data to other clients like
-  // WebAudio.
-  // TODO(xians): Move the source provider to track once we don't need to feed
-  // delay, volume, key_pressed information to WebAudioCapturerSource.
-  const scoped_ptr<WebRtcLocalAudioSourceProvider> source_provider_;
-
   // Flag which affects the buffer size used by the capturer.
   bool peer_connection_mode_;
 
   int output_sample_rate_;
   int output_frames_per_buffer_;
 
+  // Cache value for the audio processing params.
+  int audio_delay_ms_;
+  bool key_pressed_;
+
   DISALLOW_COPY_AND_ASSIGN(WebRtcAudioCapturer);
 };