Adds Analog Gain Control (AGC) to the WebRTC client.

content/renderer/media/webrtc_audio_device_impl.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_DEVICE_IMPL_H_
 #define CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_
 #pragma once
 
 #include <string>
 #include <vector>
@@ skipping 102 matching lines @@
 
   // AudioDevice::RenderCallback implementation.
   virtual size_t Render(const std::vector<float*>& audio_data,
                         size_t number_of_frames,
                         size_t audio_delay_milliseconds) OVERRIDE;
   virtual void OnRenderError() OVERRIDE;
 
   // AudioInputDevice::CaptureCallback implementation.
   virtual void Capture(const std::vector<float*>& audio_data,
                        size_t number_of_frames,
-                       size_t audio_delay_milliseconds) OVERRIDE;
+                       size_t audio_delay_milliseconds,
+                       double volume) OVERRIDE;
   virtual void OnCaptureError() OVERRIDE;
 
   // AudioInputDevice::CaptureEventHandler implementation.
   virtual void OnDeviceStarted(const std::string& device_id) OVERRIDE;
   virtual void OnDeviceStopped() OVERRIDE;
 
   // webrtc::Module implementation.
   virtual int32_t ChangeUniqueId(const int32_t id) OVERRIDE;
   virtual int32_t TimeUntilNextProcess() OVERRIDE;
   virtual int32_t Process() OVERRIDE;
@@ skipping 189 matching lines @@
 
   // Protects |recording_|, |output_delay_ms_|, |input_delay_ms_|.
   mutable base::Lock lock_;
 
   int bytes_per_sample_;
 
   bool initialized_;
   bool playing_;
   bool recording_;
 
+  // Local copy of the current Automatic Gain Control state.
+  bool agc_is_enabled_;
+
   DISALLOW_COPY_AND_ASSIGN(WebRtcAudioDeviceImpl);
 };
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_