Introduce AudioHardwareConfig for renderer side audio device info.

content/renderer/media/webrtc_audio_capturer.cc

 // 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.
 
 #include "content/renderer/media/webrtc_audio_capturer.h"
 
 #include "base/bind.h"
 #include "base/logging.h"
 #include "base/metrics/histogram.h"
 #include "base/string_util.h"
 #include "content/common/child_process.h"
 #include "content/renderer/media/audio_device_factory.h"
-#include "content/renderer/media/audio_hardware.h"
 #include "content/renderer/media/webrtc_audio_device_impl.h"
 #include "content/renderer/media/webrtc_local_audio_renderer.h"
 #include "media/audio/audio_util.h"
 #include "media/audio/sample_rates.h"
 
 namespace content {
 
 // Supported hardware sample rates for input and output sides.
 #if defined(OS_WIN) || defined(OS_MACOSX)
 // media::GetAudioInputHardwareSampleRate() asks the audio layer
@@ skipping 394 matching lines @@
   // Inform the local renderer about the stopped device.
   // The renderer can then save resources by not asking for more data from
   // the stopped source. We are on the IO thread but the callback task will
   // be posted on the message loop of the main render thread thanks to
   // usage of BindToLoop() when the callback was initialized.
   if (!on_device_stopped_cb_.is_null())
     on_device_stopped_cb_.Run();
 }
 
 }  // namespace content