Use LocalMediaStreamAudioSource for screen-casting use cases.

content/renderer/media/webrtc/processed_local_audio_source.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/processed_local_audio_source.h"
 
 #include "base/logging.h"
 #include "base/metrics/histogram.h"
 #include "base/strings/stringprintf.h"
 #include "content/renderer/media/audio_device_factory.h"
@@ skipping 332 matching lines @@
 
 media::AudioParameters ProcessedLocalAudioSource::GetInputFormat() const {
   return audio_processor_ ? audio_processor_->InputFormat()
                           : media::AudioParameters();
 }
 
 int ProcessedLocalAudioSource::GetBufferSize(int sample_rate) const {
   DCHECK(thread_checker_.CalledOnValidThread());
 #if defined(OS_ANDROID)
   // TODO(henrika): Re-evaluate whether to use same logic as other platforms.
+  // http://crbug.com/638081
   return (2 * sample_rate / 100);
 #endif
 
   // If audio processing is turned on, require 10ms buffers.
   if (audio_processor_->has_audio_processing())
     return (sample_rate / 100);
 
   // If audio processing is off and the native hardware buffer size was
   // provided, use it. It can be harmful, in terms of CPU/power consumption, to
   // use smaller buffer sizes than the native size (http://crbug.com/362261).
   if (int hardware_buffer_size = device_info().device.input.frames_per_buffer)
     return hardware_buffer_size;
 
   // If the buffer size is missing from the StreamDeviceInfo, provide 10ms as a
   // fall-back.
   //
   // TODO(miu): Identify where/why the buffer size might be missing, fix the
-  // code, and then require it here.
+  // code, and then require it here. http://crbug.com/638081
   return (sample_rate / 100);
 }
 
 }  // namespace content