| Index: content/renderer/media/webrtc_local_audio_track.cc
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| diff --git a/content/renderer/media/webrtc_local_audio_track.cc b/content/renderer/media/webrtc_local_audio_track.cc
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| deleted file mode 100644
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| index 53c55b2d0d4c45a7c50c331b547c616dd4f717e2..0000000000000000000000000000000000000000
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| --- a/content/renderer/media/webrtc_local_audio_track.cc
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| +++ /dev/null
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| @@ -1,169 +0,0 @@
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| -// Copyright (c) 2012 The Chromium Authors. All rights reserved.
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| -// Use of this source code is governed by a BSD-style license that can be
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| -// found in the LICENSE file.
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| -
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| -#include "content/renderer/media/webrtc_local_audio_track.h"
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| -
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| -#include <stdint.h>
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| -
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| -#include <limits>
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| -
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| -#include "content/public/renderer/media_stream_audio_sink.h"
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| -#include "content/renderer/media/media_stream_audio_level_calculator.h"
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| -#include "content/renderer/media/media_stream_audio_processor.h"
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| -#include "content/renderer/media/media_stream_audio_sink_owner.h"
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| -#include "content/renderer/media/media_stream_audio_track_sink.h"
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| -#include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h"
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| -
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| -namespace content {
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| -
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| -WebRtcLocalAudioTrack::WebRtcLocalAudioTrack(
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| -    scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter)
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| -    : MediaStreamAudioTrack(true), adapter_(std::move(adapter)) {
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| -  signal_thread_checker_.DetachFromThread();
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| -  DVLOG(1) << "WebRtcLocalAudioTrack::WebRtcLocalAudioTrack()";
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| -
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| -  adapter_->Initialize(this);
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| -}
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| -
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| -WebRtcLocalAudioTrack::~WebRtcLocalAudioTrack() {
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| -  DCHECK(main_render_thread_checker_.CalledOnValidThread());
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| -  DVLOG(1) << "WebRtcLocalAudioTrack::~WebRtcLocalAudioTrack()";
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| -  // Ensure the track is stopped.
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| -  MediaStreamAudioTrack::Stop();
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| -}
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| -
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| -media::AudioParameters WebRtcLocalAudioTrack::GetOutputFormat() const {
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| -  DCHECK(main_render_thread_checker_.CalledOnValidThread());
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| -  base::AutoLock auto_lock(lock_);
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| -  return audio_parameters_;
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| -}
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| -
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| -void WebRtcLocalAudioTrack::Capture(const media::AudioBus& audio_bus,
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| -                                    base::TimeTicks estimated_capture_time) {
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| -  DCHECK(capture_thread_checker_.CalledOnValidThread());
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| -  DCHECK(!estimated_capture_time.is_null());
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| -
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| -  SinkList::ItemList sinks;
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| -  SinkList::ItemList sinks_to_notify_format;
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| -  {
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| -    base::AutoLock auto_lock(lock_);
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| -    sinks = sinks_.Items();
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| -    sinks_.RetrieveAndClearTags(&sinks_to_notify_format);
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| -  }
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| -
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| -  // Notify the tracks on when the format changes. This will do nothing if
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| -  // |sinks_to_notify_format| is empty. Note that accessing |audio_parameters_|
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| -  // without holding the |lock_| is valid since |audio_parameters_| is only
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| -  // changed on the current thread.
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| -  for (const auto& sink : sinks_to_notify_format)
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| -    sink->OnSetFormat(audio_parameters_);
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| -
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| -  // Feed the data to the sinks.
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| -  // TODO(jiayl): we should not pass the real audio data down if the track is
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| -  // disabled. This is currently done so to feed input to WebRTC typing
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| -  // detection and should be changed when audio processing is moved from
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| -  // WebRTC to the track.
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| -  for (const auto& sink : sinks)
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| -    sink->OnData(audio_bus, estimated_capture_time);
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| -}
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| -
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| -void WebRtcLocalAudioTrack::OnSetFormat(
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| -    const media::AudioParameters& params) {
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| -  DVLOG(1) << "WebRtcLocalAudioTrack::OnSetFormat()";
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| -  // If the source is restarted, we might have changed to another capture
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| -  // thread.
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| -  capture_thread_checker_.DetachFromThread();
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| -  DCHECK(capture_thread_checker_.CalledOnValidThread());
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| -
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| -  base::AutoLock auto_lock(lock_);
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| -  audio_parameters_ = params;
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| -  // Remember to notify all sinks of the new format.
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| -  sinks_.TagAll();
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| -}
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| -
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| -void WebRtcLocalAudioTrack::SetLevel(
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| -    scoped_refptr<MediaStreamAudioLevelCalculator::Level> level) {
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| -  adapter_->SetLevel(std::move(level));
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| -}
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| -
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| -void WebRtcLocalAudioTrack::SetAudioProcessor(
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| -    scoped_refptr<MediaStreamAudioProcessor> processor) {
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| -  adapter_->SetAudioProcessor(std::move(processor));
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| -}
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| -
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| -void WebRtcLocalAudioTrack::AddSink(MediaStreamAudioSink* sink) {
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| -  // This method is called from webrtc, on the signaling thread, when the local
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| -  // description is set and from the main thread from WebMediaPlayerMS::load
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| -  // (via WebRtcLocalAudioRenderer::Start).
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| -  DCHECK(main_render_thread_checker_.CalledOnValidThread() ||
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| -         signal_thread_checker_.CalledOnValidThread());
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| -  DVLOG(1) << "WebRtcLocalAudioTrack::AddSink()";
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| -  base::AutoLock auto_lock(lock_);
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| -
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| -  // Verify that |sink| is not already added to the list.
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| -  DCHECK(!sinks_.Contains(
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| -      MediaStreamAudioTrackSink::WrapsMediaStreamSink(sink)));
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| -
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| -  // Create (and add to the list) a new MediaStreamAudioTrackSink
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| -  // which owns the |sink| and delagates all calls to the
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| -  // MediaStreamAudioSink interface. It will be tagged in the list, so
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| -  // we remember to call OnSetFormat() on the new sink.
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| -  scoped_refptr<MediaStreamAudioTrackSink> sink_owner(
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| -      new MediaStreamAudioSinkOwner(sink));
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| -  sinks_.AddAndTag(sink_owner.get());
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| -}
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| -
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| -void WebRtcLocalAudioTrack::RemoveSink(MediaStreamAudioSink* sink) {
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| -  // See AddSink for additional context.  When local audio is stopped from
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| -  // webrtc, we'll be called here on the signaling thread.
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| -  DCHECK(main_render_thread_checker_.CalledOnValidThread() ||
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| -         signal_thread_checker_.CalledOnValidThread());
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| -  DVLOG(1) << "WebRtcLocalAudioTrack::RemoveSink()";
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| -
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| -  scoped_refptr<MediaStreamAudioTrackSink> removed_item;
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| -  {
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| -    base::AutoLock auto_lock(lock_);
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| -    removed_item = sinks_.Remove(
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| -        MediaStreamAudioTrackSink::WrapsMediaStreamSink(sink));
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| -  }
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| -
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| -  // Clear the delegate to ensure that no more capture callbacks will
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| -  // be sent to this sink. Also avoids a possible crash which can happen
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| -  // if this method is called while capturing is active.
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| -  if (removed_item.get())
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| -    removed_item->Reset();
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| -}
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| -
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| -void WebRtcLocalAudioTrack::SetEnabled(bool enabled) {
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| -  DCHECK(main_render_thread_checker_.CalledOnValidThread());
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| -  if (adapter_.get())
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| -    adapter_->set_enabled(enabled);
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| -}
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| -
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| -void WebRtcLocalAudioTrack::OnStop() {
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| -  DCHECK(main_render_thread_checker_.CalledOnValidThread());
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| -  DVLOG(1) << "WebRtcLocalAudioTrack::OnStop()";
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| -
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| -  // Protect the pointers using the lock when accessing |sinks_|.
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| -  SinkList::ItemList sinks;
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| -  {
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| -    base::AutoLock auto_lock(lock_);
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| -    sinks = sinks_.Items();
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| -    sinks_.Clear();
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| -  }
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| -
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| -  for (SinkList::ItemList::const_iterator it = sinks.begin();
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| -       it != sinks.end();
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| -       ++it){
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| -    (*it)->OnReadyStateChanged(blink::WebMediaStreamSource::ReadyStateEnded);
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| -    (*it)->Reset();
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| -  }
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| -}
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| -
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| -webrtc::AudioTrackInterface* WebRtcLocalAudioTrack::GetAudioAdapter() {
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| -  DCHECK(main_render_thread_checker_.CalledOnValidThread());
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| -  return adapter_.get();
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| -}
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| -
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| -}  // namespace content
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| 
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