Integrate HTMLMediaElement with Web Audio API's MediaElementAudioSourceNode

content/renderer/media/audio_renderer_impl.cc

 // Copyright (c) 2011 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/audio_renderer_impl.h"
 
 #include <math.h>
 
 #include <algorithm>
 
@@ skipping 20 matching lines @@
   if (sample_rate <= 48000)
     return kNominalBufferSize;
   else if (sample_rate <= 96000)
     return kNominalBufferSize * 2;
   return kNominalBufferSize * 4;
 }
 
 AudioRendererImpl::AudioRendererImpl()
     : AudioRendererBase(),
       bytes_per_second_(0),
-      stopped_(false) {
+      stopped_(false),
+      is_initialized_(false) {
   // We create the AudioDevice here because it must be created in the
   // main thread.  But we don't yet know the audio format (sample-rate, etc.)
   // at this point.  Later, when OnInitialize() is called, we have
   // the audio format information and call the AudioDevice::Initialize()
   // method to fully initialize it.
   audio_device_ = new AudioDevice();
+
+  // Assign the AudioDevice as the default audio sink.
+  sink_ = audio_device_.get();
 }
 
 AudioRendererImpl::~AudioRendererImpl() {
 }
 
 base::TimeDelta AudioRendererImpl::ConvertToDuration(int bytes) {
   if (bytes_per_second_) {
     return base::TimeDelta::FromMicroseconds(
         base::Time::kMicrosecondsPerSecond * bytes / bytes_per_second_);
   }
@@ skipping 25 matching lines @@
   // Please see: http://code.google.com/p/chromium/issues/detail?id=103627
   // for more details.
   audio_parameters_ = AudioParameters(AudioParameters::AUDIO_PCM_LINEAR,
                                       channel_layout,
                                       sample_rate,
                                       bits_per_channel,
                                       0);
 
   bytes_per_second_ = audio_parameters_.GetBytesPerSecond();
 
-  DCHECK(audio_device_.get());
+  base::AutoLock auto_lock(sink_lock_);
 
-  if (!audio_device_->IsInitialized()) {
-    audio_device_->Initialize(
+  DCHECK(sink_);
+
+  if (!is_initialized_) {
+    sink_->Initialize(
         GetBufferSizeForSampleRate(sample_rate),
         audio_parameters_.channels,
         audio_parameters_.sample_rate,
         audio_parameters_.format,
         this);
 
-    audio_device_->Start();
+    sink_->Start();
+    is_initialized_ = true;
   }
 
   return true;
 }
 
 void AudioRendererImpl::OnStop() {
   if (stopped_)
     return;
 
-  DCHECK(audio_device_.get());
-  audio_device_->Stop();
+  base::AutoLock auto_lock(sink_lock_);
+  DCHECK(sink_);
+  sink_->Stop();
 
   stopped_ = true;
 }
 
 void AudioRendererImpl::SetPlaybackRate(float rate) {
   DCHECK_LE(0.0f, rate);
 
   // Handle the case where we stopped due to IO message loop dying.
   if (stopped_) {
     AudioRendererBase::SetPlaybackRate(rate);
@@ skipping 37 matching lines @@
   if (GetPlaybackRate() != 0.0f) {
     DoPlay();
   } else {
     DoPause();
   }
 }
 
 void AudioRendererImpl::SetVolume(float volume) {
   if (stopped_)
     return;
-  DCHECK(audio_device_.get());
-  audio_device_->SetVolume(volume);
+  base::AutoLock auto_lock(sink_lock_);
+  DCHECK(sink_);
+  sink_->SetVolume(volume);
 }
 
 void AudioRendererImpl::DoPlay() {
   earliest_end_time_ = base::Time::Now();
-  DCHECK(audio_device_.get());
-  audio_device_->Play();
+  base::AutoLock auto_lock(sink_lock_);
+  DCHECK(sink_);
+  sink_->Play();
 }
 
 void AudioRendererImpl::DoPause() {
-  DCHECK(audio_device_.get());
-  audio_device_->Pause(false);
+  base::AutoLock auto_lock(sink_lock_);
+  DCHECK(sink_);
+  sink_->Pause(false);
 }
 
 void AudioRendererImpl::DoSeek() {
   earliest_end_time_ = base::Time::Now();
 
   // Pause and flush the stream when we seek to a new location.
-  DCHECK(audio_device_.get());
-  audio_device_->Pause(true);
+  base::AutoLock auto_lock(sink_lock_);
+  DCHECK(sink_);
+  sink_->Pause(true);
 }
 
 void AudioRendererImpl::Render(const std::vector<float*>& audio_data,
                                size_t number_of_frames,
                                size_t audio_delay_milliseconds) {
   if (stopped_ || GetPlaybackRate() == 0.0f) {
     // Output silence if stopped.
     for (size_t i = 0; i < audio_data.size(); ++i)
       memset(audio_data[i], 0, sizeof(float) * number_of_frames);
     return;
@@ skipping 35 matching lines @@
                                     bytes_per_frame / channels,
                                     filled_frames);
 
     // If FillBuffer() didn't give us enough data then zero out the remainder.
     if (filled_frames < number_of_frames) {
       int frames_to_zero = number_of_frames - filled_frames;
       memset(audio_data[channel_index], 0, sizeof(float) * frames_to_zero);
     }
   }
 }
+
+void AudioRendererImpl::SetAudioRendererSink(media::AudioRendererSink* sink) {
+  // NULL should be passed in to use the default AudioDevice sink.
+  DCHECK_NE(sink, audio_device_.get());
+
+  // Synchronize with other uses of sink_ since we're messing with it here.
+  base::AutoLock auto_lock(sink_lock_);
+
+  if (sink && sink != sink_) {
+    // We're switching rendering to a different sink than the AudioDevice.
+
+    // Since we're no longer rendering to the AudioDevice, we don't want
+    // it to make callbacks to us any more.

[tommi (sloooow) - chröme, 2011/12/19 15:26:24]

nit: anymore

[Chris Rogers, 2011/12/19 21:40:35]

Done.

+    if (sink_ == audio_device_.get())
+      audio_device_->Pause(true);
+
+    sink_ = sink;
+    if (is_initialized_) {
+      // The sink needs to be notified of the audio format, if available.
+      // If the format is not yet available, it will later be called
+      // in OnInitialize().
+      sink_->Initialize(
+          GetBufferSizeForSampleRate(audio_parameters_.sample_rate),
+          audio_parameters_.channels,
+          audio_parameters_.sample_rate,
+          audio_parameters_.format,
+          this);
+    }
+  } else if (!sink && sink_ != audio_device_.get()) {
+    // Use default sink.
+    sink_ = audio_device_.get();
+
+    // !!!!! should only call Play() if we're in the playing state.

[tommi (sloooow) - chröme, 2011/12/19 15:26:24]

will you fix this or is it a todo?
do we need to check for is_initialized_ here as well?

[Chris Rogers, 2011/12/19 21:40:35]

Actually, for now I think it's better to do nothing special here.  I'll address
this by *not* calling Play(), and adding a TODO that we should call Play() under
the right circumstances.  The case where this comes up is very much an edge
case, when somebody has detached an <audio> tag from normal playback (using the
Web Audio API), then put it back in the *normal* state.  It's not going to hit
anybody who's currently using <audio>, and even with the Web Audio API
integration, this case is unlikely to be important.  Will add TODO

On 2011/12/19 15:26:24, tommi wrote:

[tommi (sloooow) - chröme, 2011/12/20 15:05:46]

sounds good.

+    audio_device_->Play();
+  }
+}