Adds support for 16kHz input sample rate and mono channel config. in WebRTC.

content/renderer/media/audio_hardware.h

 // 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.
 //
 #ifndef CONTENT_RENDERER_MEDIA_AUDIO_HARDWARE_H_
 #define CONTENT_RENDERER_MEDIA_AUDIO_HARDWARE_H_
 #pragma once
 
 #include "base/basictypes.h"
 #include "content/common/content_export.h"
 
 // Contains static methods to query hardware properties from the browser
 // process. Values are cached to avoid unnecessary round trips, but the cache
 // can be cleared if needed (currently only used by tests).
 namespace audio_hardware {
 
 // Fetch the sample rate of the default audio output end point device.
 // Must be called from RenderThreadImpl::current().
 CONTENT_EXPORT double GetOutputSampleRate();
 
 // Fetch the sample rate of the default audio input end point device.
 // Must be called from RenderThreadImpl::current().
 CONTENT_EXPORT double GetInputSampleRate();
 
 // Fetch the buffer size we use for the default output device.
 // Must be called from RenderThreadImpl::current().
 CONTENT_EXPORT size_t GetOutputBufferSize();
 
+// Fetch the number of audio channels for the default input device.
+// Must be called from RenderThreadImpl::current().
+CONTENT_EXPORT uint32 GetInputChannelCount();
+
 // Forces the next call to any of the Get functions to query the hardware
 // and repopulate the cache.
 CONTENT_EXPORT void ResetCache();
 }  // namespace audio_hardware
 
 #endif  // CONTENT_RENDERER_MEDIA_AUDIO_HARDWARE_H_