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

media/audio/audio_parameters.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 MEDIA_AUDIO_AUDIO_PARAMETERS_H_
 #define MEDIA_AUDIO_AUDIO_PARAMETERS_H_
 
 #include "base/basictypes.h"
 #include "media/base/channel_layout.h"
 #include "media/base/media_export.h"
 
+struct MEDIA_EXPORT AudioInputBufferParameters {
+  double volume;
+  uint32 size;
+};
+
+// Use a struct-in-struct approach to ensure that we can calculate the required
+// size as sizeof(AudioInputBufferParameters) + #(bytes in audio buffer) without
+// using packing.
+struct MEDIA_EXPORT AudioInputBuffer {
+  AudioInputBufferParameters params;
+  int8 audio[1];
+};
+
 class MEDIA_EXPORT AudioParameters {
  public:
   // Compare is useful when AudioParameters is used as a key in std::map.
   class MEDIA_EXPORT Compare {
    public:
     bool operator()(const AudioParameters& a, const AudioParameters& b) const;
   };
 
   enum Format {
     AUDIO_PCM_LINEAR = 0,     // PCM is 'raw' amplitude samples.
@@ skipping 42 matching lines @@
   ChannelLayout channel_layout_;  // Order of surround sound channels.
   int sample_rate_;               // Sampling frequency/rate.
   int bits_per_sample_;           // Number of bits per sample.
   int frames_per_buffer_;         // Number of frames in a buffer.
 
   int channels_;                  // Number of channels. Value set based on
                                   // |channel_layout|.
 };
 
 #endif  // MEDIA_AUDIO_AUDIO_PARAMETERS_H_