Improved AGC update scheme for the audio backend in 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 "base/basictypes.h"
@@ skipping 104 matching lines @@
 //         AudioInputController::DoSetAutomaticGainControl(true)
 //            AudioInputStream::SetAutomaticGainControl(true)
 //                                |
 // AGC is now enabled in the media layer and streaming starts (details omitted).
 // The figure below illustrates the AGC scheme which is active in combination
 // with the default media flow explained earlier.
 //                                |
 //                            [browser]
 //                                |
 //                AudioInputStream::(Capture thread loop)
-//   AudioInputStreamImpl::QueryAgcVolume() => new volume once per second
+//  AgcAudioStream<AudioInputStream>::GetAgcVolume() => get latest mic volume
 //                 AudioInputData::OnData(..., volume)
 //              AudioInputController::OnData(..., volume)
 //               AudioInputSyncWriter::Write(..., volume)
 //                                |
 //      [volume | size | data] is sent to the renderer         [shared memory]
 //                                |
 //                            [renderer]
 //                                |
 //          AudioInputDevice::AudioThreadCallback::Process()
 //            WebRtcAudioDeviceImpl::Capture(..., volume)
@@ skipping 246 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_