Add ability to pause and resume the audio sink from blink.

Add state to support pausing a web audio device such that it can be
restarted when the frame is returned to be visible.

BUG=907125

Change-Id: I01aef9c19c81e35e68ee1e299e16cd6c9946ebde
Reviewed-on: https://chromium-review.googlesource.com/c/1344265Reviewed-by: Dale Curtis <dalecurtis@chromium.org>
Reviewed-by: Rick Byers <rbyers@chromium.org>
Reviewed-by: Hongchan Choi <hongchan@chromium.org>
Commit-Queue: Dave Tapuska <dtapuska@chromium.org>
Cr-Commit-Position: refs/heads/master@{#611881}

content/renderer/media/renderer_webaudiodevice_impl.cc

@@ -188,6 +188,18 @@ void RendererWebAudioDeviceImpl::Start() {
   sink_->Play();
 }
 
+void RendererWebAudioDeviceImpl::Pause() {
+  DCHECK(thread_checker_.CalledOnValidThread());
+  if (sink_)
+    sink_->Pause();
+}
+
+void RendererWebAudioDeviceImpl::Resume() {
+  DCHECK(thread_checker_.CalledOnValidThread());
+  if (sink_)
+    sink_->Play();
+}
+
 void RendererWebAudioDeviceImpl::Stop() {
   DCHECK(thread_checker_.CalledOnValidThread());
   if (sink_) {

content/renderer/media/renderer_webaudiodevice_impl.h

@@ -44,6 +44,8 @@ class CONTENT_EXPORT RendererWebAudioDeviceImpl
   // blink::WebAudioDevice implementation.
   void Start() override;
   void Stop() override;
+  void Pause() override;
+  void Resume() override;
   double SampleRate() override;
   int FramesPerBuffer() override;
 

third_party/blink/public/platform/web_audio_device.h

@@ -58,6 +58,16 @@ class WebAudioDevice {
 
   virtual void Start() = 0;
   virtual void Stop() = 0;
+
+  // Pause an audio device so that the device doesn't produce
+  // requests for audio frames. This is used when a web frame is
+  // frozen. This shouldn't be observable to the web application.
+  virtual void Pause() = 0;
+
+  // Resume an audio device that was previously paused. Used
+  // for frozen frames.
+  virtual void Resume() = 0;
+
   virtual double SampleRate() = 0;
   virtual int FramesPerBuffer() = 0;
 };

third_party/blink/renderer/modules/webaudio/audio_context_autoplay_test.cc

@@ -64,6 +64,8 @@ class MockWebAudioDeviceForAutoplayTest : public WebAudioDevice {
 
   void Start() override {}
   void Stop() override {}
+  void Pause() override {}
+  void Resume() override {}
   double SampleRate() override { return sample_rate_; }
   int FramesPerBuffer() override { return frames_per_buffer_; }
 

third_party/blink/renderer/modules/webaudio/audio_context_test.cc

@@ -27,6 +27,8 @@ class MockWebAudioDeviceForAudioContext : public WebAudioDevice {
 
   void Start() override {}
   void Stop() override {}
+  void Pause() override {}
+  void Resume() override {}
   double SampleRate() override { return sample_rate_; }
   int FramesPerBuffer() override { return frames_per_buffer_; }
 

third_party/blink/renderer/platform/audio/audio_destination.cc

@@ -64,7 +64,7 @@ AudioDestination::AudioDestination(AudioIOCallback& callback,
                                    unsigned number_of_output_channels,
                                    const WebAudioLatencyHint& latency_hint)
     : number_of_output_channels_(number_of_output_channels),
-      is_playing_(false),
+      play_state_(PlayState::kStopped),
       fifo_(
           std::make_unique<PushPullFIFO>(number_of_output_channels, kFIFOSize)),
       output_bus_(AudioBus::Create(number_of_output_channels,
@@ -198,10 +198,10 @@ void AudioDestination::Start() {
   DCHECK(IsMainThread());
 
   // Start the "audio device" after the rendering thread is ready.
-  if (web_audio_device_ && !is_playing_) {
+  if (web_audio_device_ && play_state_ == PlayState::kStopped) {
     TRACE_EVENT0("webaudio", "AudioDestination::Start");
     web_audio_device_->Start();
-    is_playing_ = true;
+    play_state_ = PlayState::kPlaying;
   }
 }
 
@@ -209,11 +209,11 @@ void AudioDestination::StartWithWorkletTaskRunner(
     scoped_refptr<base::SingleThreadTaskRunner> worklet_task_runner) {
   DCHECK(IsMainThread());
 
-  if (web_audio_device_ && !is_playing_) {
+  if (web_audio_device_ && play_state_ == PlayState::kStopped) {
     TRACE_EVENT0("webaudio", "AudioDestination::Start");
     worklet_task_runner_ = std::move(worklet_task_runner);
     web_audio_device_->Start();
-    is_playing_ = true;
+    play_state_ = PlayState::kPlaying;
   }
 }
 
@@ -222,11 +222,27 @@ void AudioDestination::Stop() {
 
   // This assumes stopping the "audio device" is synchronous and dumping the
   // rendering thread is safe after that.
-  if (web_audio_device_ && is_playing_) {
+  if (web_audio_device_ && play_state_ != PlayState::kStopped) {
     TRACE_EVENT0("webaudio", "AudioDestination::Stop");
     web_audio_device_->Stop();
     worklet_task_runner_ = nullptr;
-    is_playing_ = false;
+    play_state_ = PlayState::kStopped;
+  }
+}
+
+void AudioDestination::Pause() {
+  DCHECK(IsMainThread());
+  if (web_audio_device_ && play_state_ == PlayState::kPlaying) {
+    web_audio_device_->Pause();
+    play_state_ = PlayState::kPaused;
+  }
+}
+
+void AudioDestination::Resume() {
+  DCHECK(IsMainThread());
+  if (web_audio_device_ && play_state_ == PlayState::kPaused) {
+    web_audio_device_->Resume();
+    play_state_ = PlayState::kPlaying;
   }
 }
 
@@ -237,7 +253,7 @@ size_t AudioDestination::CallbackBufferSize() const {
 
 bool AudioDestination::IsPlaying() {
   DCHECK(IsMainThread());
-  return is_playing_;
+  return play_state_ == PlayState::kPlaying;
 }
 
 int AudioDestination::FramesPerBuffer() const {

third_party/blink/renderer/platform/audio/audio_destination.h

@@ -91,6 +91,8 @@ class PLATFORM_EXPORT AudioDestination
 
   virtual void Start();
   virtual void Stop();
+  virtual void Pause();
+  virtual void Resume();
 
   // Starts the destination with the AudioWorklet support.
   void StartWithWorkletTaskRunner(
@@ -112,6 +114,8 @@ class PLATFORM_EXPORT AudioDestination
   static uint32_t MaxChannelCount();
 
  private:
+  enum class PlayState { kStopped, kPlaying, kPaused };
+
   // Check if the buffer size chosen by the WebAudioDevice is too large.
   bool CheckBufferSize();
 
@@ -121,7 +125,7 @@ class PLATFORM_EXPORT AudioDestination
   std::unique_ptr<WebAudioDevice> web_audio_device_;
   const unsigned number_of_output_channels_;
   size_t callback_buffer_size_;
-  bool is_playing_;
+  PlayState play_state_;
 
   // The task runner for AudioWorklet operation. This is only valid when
   // the AudioWorklet is activated.