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

media/audio/audio_parameters.cc

 // Copyright (c) 2011 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.
 
 #include "media/audio/audio_parameters.h"
 
 #include "media/base/limits.h"
 
+const int AudioInputBuffer::kSizeOfStructMinusArr = sizeof(AudioInputBuffer) -
+    sizeof(reinterpret_cast<AudioInputBuffer*>(0)->audio);
+
 AudioParameters::AudioParameters()
     : format(AUDIO_PCM_LINEAR),
       channel_layout(CHANNEL_LAYOUT_NONE),
       sample_rate(0),
       bits_per_sample(0),
       samples_per_packet(0),
       channels(0) {
 }
 
 AudioParameters::AudioParameters(Format format, ChannelLayout channel_layout,
@@ skipping 39 matching lines @@
   if (a.sample_rate < b.sample_rate)
     return true;
   if (a.sample_rate > b.sample_rate)
     return false;
   if (a.bits_per_sample < b.bits_per_sample)
     return true;
   if (a.bits_per_sample > b.bits_per_sample)
     return false;
   return a.samples_per_packet < b.samples_per_packet;
 }