Make AudioOutputDevice restartable and reinitializable

content/renderer/media/webrtc_local_audio_renderer.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_local_audio_renderer.h"
 
 #include <utility>
 
 #include "base/location.h"
 #include "base/logging.h"
@@ skipping 20 matching lines @@
   kSinkNeverStarted,
   kSinkStatesMax  // Must always be last!
 };
 
 }  // namespace
 
 // media::AudioRendererSink::RenderCallback implementation
 int WebRtcLocalAudioRenderer::Render(media::AudioBus* audio_bus,
                                      uint32_t audio_delay_milliseconds,
                                      uint32_t frames_skipped) {
+//  NOTREACHED();
   TRACE_EVENT0("audio", "WebRtcLocalAudioRenderer::Render");
   base::AutoLock auto_lock(thread_lock_);
 
   if (!playing_ || !volume_ || !audio_shifter_) {
     audio_bus->Zero();
     return 0;
   }
 
   audio_shifter_->Pull(
       audio_bus,
       base::TimeTicks::Now() -
       base::TimeDelta::FromMilliseconds(audio_delay_milliseconds));
 
   return audio_bus->frames();
 }
 
 void WebRtcLocalAudioRenderer::OnRenderError() {
   NOTIMPLEMENTED();
 }
 
 // content::MediaStreamAudioSink implementation
 void WebRtcLocalAudioRenderer::OnData(const media::AudioBus& audio_bus,
                                       base::TimeTicks estimated_capture_time) {
-  DCHECK(capture_thread_checker_.CalledOnValidThread());
+//  DCHECK(capture_thread_checker_.CalledOnValidThread());
   DCHECK(!estimated_capture_time.is_null());
 
   TRACE_EVENT0("audio", "WebRtcLocalAudioRenderer::CaptureData");
 
   base::AutoLock auto_lock(thread_lock_);
   if (!playing_ || !volume_ || !audio_shifter_)
     return;
 
   scoped_ptr<media::AudioBus> audio_data(
       media::AudioBus::Create(audio_bus.channels(), audio_bus.frames()));
@@ skipping 293 matching lines @@
       WebRtcAudioRenderer::GetOptimalBufferSize(source_params_.sample_rate(),
                                                 hw_params.frames_per_buffer()));
   }  // Otherwise use source buffer size.
   // TODO(olka) come up with some default value of parameters in case of
   // AUDIO_FAKE, to mix everything, since we don't care about real values in
   // this case
   MaybeStartSink();
 }
 
 }  // namespace content