move the APM to chrome.

content/renderer/media/webrtc_audio_device_impl.h

 // Copyright 2013 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_DEVICE_IMPL_H_
 #define CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_
 
 #include <string>
 #include <vector>
 
@@ skipping 192 matching lines @@
 };
 
 class WebRtcAudioCapturerSink {
  public:
   // Callback to deliver the captured interleaved data.
   // |channels| contains a vector of WebRtc VoE channels.
   // |audio_data| is the pointer to the audio data.
   // |sample_rate| is the sample frequency of audio data.
   // |number_of_channels| is the number of channels reflecting the order of
   // surround sound channels.
-  // |audio_delay_milliseconds| is recording delay value.
-  // |current_volume| is current microphone volume, in range of |0, 255].
-  // |need_audio_processing| indicates if the audio needs WebRtc AEC/NS/AGC
-  // audio processing.
-  // The return value is the new microphone volume, in the range of |0, 255].
-  // When the volume does not need to be updated, it returns 0.
-  virtual int CaptureData(const std::vector<int>& channels,
-                          const int16* audio_data,
-                          int sample_rate,
-                          int number_of_channels,
-                          int number_of_frames,
-                          int audio_delay_milliseconds,
-                          int current_volume,
-                          bool need_audio_processing,
-                          bool key_pressed) = 0;
+  virtual void CaptureData(const std::vector<int>& channels,
+                           const int16* audio_data,
+                           int sample_rate,
+                           int number_of_channels,
+                           int number_of_frames) = 0;
 
   // Set the format for the capture audio parameters.
   virtual void SetCaptureFormat(const media::AudioParameters& params) = 0;
 
  protected:
   virtual ~WebRtcAudioCapturerSink() {}
 };
 
 // Note that this class inherits from webrtc::AudioDeviceModule but due to
 // the high number of non-implemented methods, we move the cruft over to the
@@ skipping 88 matching lines @@
 
  private:
   typedef std::list<scoped_refptr<WebRtcAudioCapturer> > CapturerList;
 
   // Make destructor private to ensure that we can only be deleted by Release().
   virtual ~WebRtcAudioDeviceImpl();
 
   // WebRtcAudioCapturerSink implementation.
 
   // Called on the AudioInputDevice worker thread.
-  virtual int CaptureData(const std::vector<int>& channels,
-                          const int16* audio_data,
-                          int sample_rate,
-                          int number_of_channels,
-                          int number_of_frames,
-                          int audio_delay_milliseconds,
-                          int current_volume,
-                          bool need_audio_processing,
-                          bool key_pressed) OVERRIDE;
+  virtual void CaptureData(const std::vector<int>& channels,
+                           const int16* audio_data,
+                           int sample_rate,
+                           int number_of_channels,
+                           int number_of_frames) OVERRIDE;
 
   // Called on the main render thread.
   virtual void SetCaptureFormat(const media::AudioParameters& params) OVERRIDE;
 
   // WebRtcAudioRendererSource implementation.
 
   // Called on the AudioInputDevice worker thread.
   virtual void RenderData(uint8* audio_data,
                           int number_of_channels,
                           int number_of_frames,
@@ skipping 47 matching lines @@
   // Stores latest microphone volume received in a CaptureData() callback.
   // Range is [0, 255].
   uint32_t microphone_volume_;
 
   DISALLOW_COPY_AND_ASSIGN(WebRtcAudioDeviceImpl);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_