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"
 
 namespace media {
 class AudioBus;
 }
 
 namespace content {
 
 class WebRtcLocalAudioRenderer;
@@ skipping 34 matching lines @@
   // 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);
 
   // 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();
 
   // Volume APIs used by WebRtcAudioDeviceImpl.
@@ skipping 18 matching lines @@
 
   // Gets information about the paired output device. Returns true if such a
   // device exists.
   bool GetPairedOutputParameters(int* session_id,
                                  int* output_sample_rate,
                                  int* output_frames_per_buffer) const;
 
   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();
   virtual ~WebRtcAudioCapturer();
 
  private:
   class TrackOwner;
   typedef std::list<scoped_refptr<TrackOwner> > TrackList;
 
   // 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);
 
   // 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();
 
-
-
   // 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_;
@@ skipping 13 matching lines @@
   // the browser.
   int session_id_;
 
   // The device this capturer is given permission to use.
   std::string device_id_;
 
   // Stores latest microphone volume received in a CaptureData() callback.
   // 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);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_CAPTURER_H_