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

media/audio/fake_audio_input_stream.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.
 //
 // A fake implementation of AudioInputStream, useful for testing purpose.
 
 #ifndef MEDIA_AUDIO_FAKE_AUDIO_INPUT_STREAM_H_
 #define MEDIA_AUDIO_FAKE_AUDIO_INOUT_STREAM_H_
 
 #include <vector>
@@ skipping 12 matching lines @@
   static AudioInputStream* MakeFakeStream(AudioManagerBase* manager,
                                           const AudioParameters& params);
 
   virtual bool Open() OVERRIDE;
   virtual void Start(AudioInputCallback* callback) OVERRIDE;
   virtual void Stop() OVERRIDE;
   virtual void Close() OVERRIDE;
   virtual double GetMaxVolume() OVERRIDE;
   virtual void SetVolume(double volume) OVERRIDE;
   virtual double GetVolume() OVERRIDE;
+  virtual void SetAutomaticGainControl(bool enabled) OVERRIDE;
+  virtual bool GetAutomaticGainControl() OVERRIDE;
 
  private:
   FakeAudioInputStream(AudioManagerBase* manager,
                        const AudioParameters& params);
 
   virtual ~FakeAudioInputStream();
 
   void DoCallback();
 
   AudioManagerBase* audio_manager_;
   AudioInputCallback* callback_;
   scoped_array<uint8> buffer_;
   int buffer_size_;
   base::Thread thread_;
   base::Time last_callback_time_;
   base::TimeDelta callback_interval_;
 
   DISALLOW_COPY_AND_ASSIGN(FakeAudioInputStream);
 };
 
 #endif  // MEDIA_AUDIO_FAKE_AUDIO_INPUT_STREAM_H_