MediaStream audio: Refactor 3 separate "glue" implementations into one.

content/renderer/media/webaudio_media_stream_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/webaudio_capturer_source.h"
+#include "content/renderer/media/webaudio_media_stream_source.h"
 
 #include "base/bind.h"
 #include "base/bind_helpers.h"
 #include "base/logging.h"
-#include "content/renderer/media/webrtc_local_audio_track.h"
-
-using media::AudioBus;
-using media::AudioParameters;
-using media::ChannelLayout;
-using media::CHANNEL_LAYOUT_MONO;
-using media::CHANNEL_LAYOUT_STEREO;
 
 namespace content {
 
-WebAudioCapturerSource::WebAudioCapturerSource(
+WebAudioMediaStreamSource::WebAudioMediaStreamSource(
     blink::WebMediaStreamSource* blink_source)
-    : track_(NULL),
-      audio_format_changed_(false),
-      fifo_(base::Bind(&WebAudioCapturerSource::DeliverRebufferedAudio,
+    : MediaStreamAudioSource(false /* is_remote */),
+      is_registered_consumer_(false),
+      fifo_(base::Bind(&WebAudioMediaStreamSource::DeliverRebufferedAudio,
                        base::Unretained(this))),
       blink_source_(*blink_source) {
-  DCHECK(blink_source);
-  DCHECK(!blink_source_.isNull());
-  DVLOG(1) << "WebAudioCapturerSource::WebAudioCapturerSource()";
-  blink_source_.addAudioConsumer(this);
+  DVLOG(1) << "WebAudioMediaStreamSource::WebAudioMediaStreamSource()";
 }
 
-WebAudioCapturerSource::~WebAudioCapturerSource() {
-  DCHECK(thread_checker_.CalledOnValidThread());
-  DVLOG(1) << "WebAudioCapturerSource::~WebAudioCapturerSource()";
-  DeregisterFromBlinkSource();
+WebAudioMediaStreamSource::~WebAudioMediaStreamSource() {
+  DVLOG(1) << "WebAudioMediaStreamSource::~WebAudioMediaStreamSource()";
+  EnsureSourceIsStopped();
 }
 
-void WebAudioCapturerSource::setFormat(
-    size_t number_of_channels, float sample_rate) {
+void WebAudioMediaStreamSource::setFormat(size_t number_of_channels,
+                                          float sample_rate) {
   DCHECK(thread_checker_.CalledOnValidThread());
-  DVLOG(1) << "WebAudioCapturerSource::setFormat(sample_rate="
-           << sample_rate << ")";
+  VLOG(1) << "WebAudio media stream source changed format to: channels="
+          << number_of_channels << ", sample_rate=" << sample_rate;
 
   // If the channel count is greater than 8, use discrete layout. However,
-  // anything beyond 8 is ignored by the subsequent (WebRTC) audio pipeline.
-  ChannelLayout channel_layout =
+  // anything beyond 8 is ignored by some audio tracks/sinks.
+  media::ChannelLayout channel_layout =
       number_of_channels > 8 ? media::CHANNEL_LAYOUT_DISCRETE
                              : media::GuessChannelLayout(number_of_channels);
 
-  base::AutoLock auto_lock(lock_);
+  // Set the format used by this WebAudioMediaStreamSource. We are using 10ms
+  // data as a buffer size since that is the native buffer size of WebRtc packet
+  // running on.
+  //
+  // TODO(miu): Re-evaluate whether this is needed.  For now (this refactoring),
+  // I did not want to change behavior.  http://crbug.com/577874
+  fifo_.Reset(sample_rate / 100);
+  media::AudioParameters params(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
+                                channel_layout, sample_rate, 16,
+                                fifo_.frames_per_buffer());
+  // Take care of the discrete channel layout case.
+  params.set_channels_for_discrete(number_of_channels);
+  MediaStreamAudioSource::SetFormat(params);
 
-  // Set the format used by this WebAudioCapturerSource. We are using 10ms data
-  // as buffer size since that is the native buffer size of WebRtc packet
-  // running on.
-  fifo_.Reset(sample_rate / 100);
-  params_.Reset(media::AudioParameters::AUDIO_PCM_LOW_LATENCY, channel_layout,
-                sample_rate, 16, fifo_.frames_per_buffer());
+  if (!wrapper_bus_ || wrapper_bus_->channels() != params.channels())
+    wrapper_bus_ = media::AudioBus::CreateWrapper(params.channels());
+}
 
-  // Take care of the discrete channel layout case.
-  params_.set_channels_for_discrete(number_of_channels);
+bool WebAudioMediaStreamSource::EnsureSourceIsStarted() {
+  DCHECK(thread_checker_.CalledOnValidThread());
+  if (is_registered_consumer_)
+    return true;
+  if (blink_source_.isNull() || !blink_source_.requiresAudioConsumer())
+    return false;
+  VLOG(1) << "Starting WebAudio media stream source.";
+  blink_source_.addAudioConsumer(this);
+  is_registered_consumer_ = true;
+  return true;
+}
 
-  audio_format_changed_ = true;
-
-  if (!wrapper_bus_ ||
-      wrapper_bus_->channels() != static_cast<int>(number_of_channels)) {
-    wrapper_bus_ = AudioBus::CreateWrapper(params_.channels());
+void WebAudioMediaStreamSource::EnsureSourceIsStopped() {
+  DCHECK(thread_checker_.CalledOnValidThread());
+  if (is_registered_consumer_) {
+    if (!blink_source_.isNull()) {
+      blink_source_.removeAudioConsumer(this);
+      blink_source_.reset();
+      is_registered_consumer_ = false;
+      VLOG(1) << "Stopped WebAudio media stream source.  Final audio "
+                 "parameters={"
+              << GetAudioParameters().AsHumanReadableString() << "}.";
+    }
   }
 }
 
-void WebAudioCapturerSource::Start(WebRtcLocalAudioTrack* track) {
-  DCHECK(thread_checker_.CalledOnValidThread());
-  DCHECK(track);
-  base::AutoLock auto_lock(lock_);
-  track_ = track;
-}
-
-void WebAudioCapturerSource::Stop() {
-  DCHECK(thread_checker_.CalledOnValidThread());
-  {
-    base::AutoLock auto_lock(lock_);
-    track_ = NULL;
-  }
-  // DeregisterFromBlinkSource() should not be called while |lock_| is acquired,
-  // as it could result in a deadlock.
-  DeregisterFromBlinkSource();
-}
-
-void WebAudioCapturerSource::consumeAudio(
+void WebAudioMediaStreamSource::consumeAudio(
     const blink::WebVector<const float*>& audio_data,
     size_t number_of_frames) {
   // TODO(miu): Plumbing is needed to determine the actual capture timestamp
   // of the audio, instead of just snapshotting TimeTicks::Now(), for proper
   // audio/video sync.  http://crbug.com/335335
   current_reference_time_ = base::TimeTicks::Now();
 
-  base::AutoLock auto_lock(lock_);
-  if (!track_)
-    return;
-
-  // Update the downstream client if the audio format has been changed.
-  if (audio_format_changed_) {
-    track_->OnSetFormat(params_);
-    audio_format_changed_ = false;
-  }
-
   wrapper_bus_->set_frames(number_of_frames);

[o1ka, 2016/04/01 15:11:41]

Don't you need to protect |wrapper_bus_| from the concurrent access here and in
setFormat()? If not, than it would be nice to have a comment/DCHECK explaining
why.

[miu, 2016/04/08 18:36:47]

The header file includes a comment for these methods: It explains that blink
will not call into them concurrently.

-  DCHECK_EQ(params_.channels(), static_cast<int>(audio_data.size()));
+  DCHECK_EQ(wrapper_bus_->channels(), static_cast<int>(audio_data.size()));
   for (size_t i = 0; i < audio_data.size(); ++i)
     wrapper_bus_->SetChannelData(i, const_cast<float*>(audio_data[i]));
 
   // The following will result in zero, one, or multiple synchronous calls to
   // DeliverRebufferedAudio().
   fifo_.Push(*wrapper_bus_);
 }
 
-void WebAudioCapturerSource::DeliverRebufferedAudio(
+void WebAudioMediaStreamSource::DeliverRebufferedAudio(
     const media::AudioBus& audio_bus,
     int frame_delay) {
-  lock_.AssertAcquired();
   const base::TimeTicks reference_time =
       current_reference_time_ +
-      base::TimeDelta::FromMicroseconds(frame_delay *
-                                        base::Time::kMicrosecondsPerSecond /
-                                        params_.sample_rate());
-  track_->Capture(audio_bus, reference_time);
-}
-
-void WebAudioCapturerSource::DeregisterFromBlinkSource() {
-  if (!blink_source_.isNull()) {
-    blink_source_.removeAudioConsumer(this);
-    blink_source_.reset();
-  }
+      base::TimeDelta::FromMicroseconds(
+          frame_delay * base::Time::kMicrosecondsPerSecond /
+          MediaStreamAudioSource::GetAudioParameters().sample_rate());
+  MediaStreamAudioSource::DeliverDataToTracks(audio_bus, reference_time);
 }
 
 }  // namespace content