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 eb1988d0028ae92ef6896c2974067e847ca24125..a7680d5e556a380c237a998d80d4ec1cd82de919 100644
--- a/content/renderer/media/webrtc_audio_capturer.h
+++ b/content/renderer/media/webrtc_audio_capturer.h
@@ -16,6 +16,7 @@
 #include "content/renderer/media/webrtc_local_audio_source_provider.h"
 #include "media/audio/audio_input_device.h"
 #include "media/base/audio_capturer_source.h"
+#include "third_party/libjingle/source/talk/app/webrtc/mediaconstraintsinterface.h"
 
 namespace media {
 class AudioBus;
@@ -23,6 +24,7 @@ class AudioBus;
 
 namespace content {
 
+class WebRtcAudioProcessingWrapper;
 class WebRtcLocalAudioRenderer;
 class WebRtcLocalAudioTrack;
 
@@ -55,7 +57,8 @@ class CONTENT_EXPORT WebRtcAudioCapturer
                   int session_id,
                   const std::string& device_id,
                   int paired_output_sample_rate,
-                  int paired_output_frames_per_buffer);
+                  int paired_output_frames_per_buffer,
+                  const webrtc::MediaConstraintsInterface* constraints);
 
   // Add a audio track to the sinks of the capturer.
   // WebRtcAudioDeviceImpl calls this method on the main render thread but
@@ -68,20 +71,27 @@ 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);
+      float sample_rate,
+      const webrtc::MediaConstraintsInterface* constraints);
 
   // Called when a stream is connecting to a peer connection. This will set
   // up the native buffer size for the stream in order to optimize the
   // performance for peer connection.
   void EnablePeerConnectionMode();
 
+  void FeedRenderDataToAudioProcessing(const int16* render_audio,
+                                       int sample_rate,
+                                       int number_of_channels,
+                                       int number_of_frames,
+                                       int render_delay_ms);
+
   // Volume APIs used by WebRtcAudioDeviceImpl.
   // Called on the AudioInputDevice audio thread.
   void SetVolume(int volume);
@@ -133,7 +143,8 @@ class CONTENT_EXPORT WebRtcAudioCapturer
 
   // Reconfigures the capturer with a new capture parameters.
   // Must be called without holding the lock.
-  void Reconfigure(int sample_rate, media::ChannelLayout channel_layout);
+  void Reconfigure(int sample_rate, media::ChannelLayout channel_layout,
+                   const webrtc::MediaConstraintsInterface* constraints);
 
   // Starts recording audio.
   // Triggered by AddSink() on the main render thread or a Libjingle working
@@ -153,7 +164,7 @@ 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.
@@ -162,8 +173,11 @@ class CONTENT_EXPORT WebRtcAudioCapturer
   // The audio data source from the browser process.
   scoped_refptr<media::AudioCapturerSource> source_;
 
-  // Cached audio parameters for output.
-  media::AudioParameters params_;
+  // Cached audio parameters for the source.
+  media::AudioParameters source_params_;
+
+  // Cached audio parameters for the sinks.
+  media::AudioParameters sink_params_;
 
   bool running_;
 
@@ -199,6 +213,8 @@ class CONTENT_EXPORT WebRtcAudioCapturer
   int output_sample_rate_;
   int output_frames_per_buffer_;
 
+  scoped_ptr<WebRtcAudioProcessingWrapper> audio_processing_;
+
   DISALLOW_COPY_AND_ASSIGN(WebRtcAudioCapturer);
 };