move the APM to chrome.

content/renderer/media/webrtc_audio_capturer.h

 // Copyright (c) 2012 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_AUDIO_CAPTURER_H_
 #define CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_CAPTURER_H_
 
 #include <list>
 #include <string>
 
 #include "base/callback.h"
 #include "base/memory/ref_counted.h"
 #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"
+#include "third_party/libjingle/source/talk/app/webrtc/mediaconstraintsinterface.h"
 
 namespace media {
 class AudioBus;
 }
 
 namespace content {
 
+class WebRtcAudioProcessingWrapper;
 class WebRtcLocalAudioRenderer;
 class WebRtcLocalAudioTrack;
 
 // This class manages the capture data flow by getting data from its
 // |source_|, and passing it to its |tracks_|.
 // It allows clients to inject their own capture data source by calling
 // SetCapturerSource().
 // The threading model for this class is rather complex since it will be
 // created on the main render thread, captured data is provided on a dedicated
 // AudioInputDevice thread, and methods can be called either on the Libjingle
@@ skipping 12 matching lines @@
   // consuming audio for capture.  |session_id| is passed to the browser to
   // decide which device to use.  |device_id| is used to identify which device
   // the capturer is created for.  Called on the main render thread.
   bool Initialize(int render_view_id,
                   media::ChannelLayout channel_layout,
                   int sample_rate,
                   int buffer_size,
                   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
   // other clients may call it from other threads. The current implementation
   // does not support multi-thread calling.
   // Called on the main render thread or libjingle working thread.
   void AddTrack(WebRtcLocalAudioTrack* track);
 
   // Remove a audio track from the sinks of the capturer.
   // 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);
   int Volume() const;
   int MaxVolume() const;
 
   // Enables or disables the WebRtc AGC control.
   // Called from a Libjingle working thread.
   void SetAutomaticGainControl(bool enable);
 
@@ skipping 31 matching lines @@
   // AudioCapturerSource::CaptureCallback implementation.
   // Called on the AudioInputDevice audio thread.
   virtual void Capture(media::AudioBus* audio_source,
                        int audio_delay_milliseconds,
                        double volume,
                        bool key_pressed) OVERRIDE;
   virtual void OnCaptureError() OVERRIDE;
 
   // 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
   // thread. It should NOT be called under |lock_|.
   void Start();
 
   // Stops recording audio.
   // Triggered by RemoveSink() on the main render thread or a Libjingle working
   // thread. It should NOT be called under |lock_|.
   void Stop();
 
   // Helper function to get the buffer size based on |peer_connection_mode_|
   // and sample rate;
   int GetBufferSize(int sample_rate) const;
 
   // Used to DCHECK that we are called on the correct thread.
   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.
   TrackList tracks_;
 
   // 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_;
 
   // True when automatic gain control is enabled, false otherwise.
   bool agc_is_enabled_;
 
   int render_view_id_;
 
   // Cached value for the hardware native buffer size, used when
   // |peer_connection_mode_| is set to false.
@@ skipping 15 matching lines @@
   // 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_;
 
+  scoped_ptr<WebRtcAudioProcessingWrapper> audio_processing_;
+
   DISALLOW_COPY_AND_ASSIGN(WebRtcAudioCapturer);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_CAPTURER_H_