Add WebThread in AudioDestination to support AudioWorkletThread

third_party/WebKit/Source/platform/audio/AudioDestination.h

 /*
  * Copyright (C) 2010 Google Inc. All rights reserved.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions
  * are met:
  *
  * 1.  Redistributions of source code must retain the above copyright
  *     notice, this list of conditions and the following disclaimer.
  * 2.  Redistributions in binary form must reproduce the above copyright
@@ skipping 25 matching lines @@
 #include "platform/wtf/Noncopyable.h"
 #include "platform/wtf/text/WTFString.h"
 #include "public/platform/WebAudioDevice.h"
 #include "public/platform/WebVector.h"
 
 namespace blink {
 
 class PushPullFIFO;
 class SecurityOrigin;
 class WebAudioLatencyHint;
+class WebThread;
 
 // The AudioDestination class is an audio sink interface between the media
 // renderer and the Blink's WebAudio module. It has a FIFO to adapt the
 // different processing block sizes of WebAudio renderer and actual hardware
 // audio callback.
 class PLATFORM_EXPORT AudioDestination : public WebAudioDevice::RenderCallback {
   USING_FAST_MALLOC(AudioDestination);
   WTF_MAKE_NONCOPYABLE(AudioDestination);
 
  public:
@@ skipping 10 matching lines @@
       PassRefPtr<SecurityOrigin>);
 
   // The actual render function (WebAudioDevice::RenderCallback) isochronously
   // invoked by the media renderer.
   void Render(const WebVector<float*>& destination_data,
               size_t number_of_frames,
               double delay,
               double delay_timestamp,
               size_t prior_frames_skipped) override;
 
+  // FIFO calls back this function when more data is needed to fulfill the
+  // request from AudioDeviceThread.
+  void RequestRender(double delay,
+                     double delay_timestamp,
+                     size_t prior_frames_skipped,
+                     size_t frames_requested,
+                     size_t frames_to_render);
+
+  // The actual render request to the WebAudio destination node. This triggers
+  // the WebAudio rendering pipe line on the web thread.
+  void RequestRenderOnWebThread(size_t frames_requested,
+                                size_t frames_to_render,
+                                double delay,
+                                double delay_timestamp,
+                                size_t prior_frames_skipped);
+
   virtual void Start();
   virtual void Stop();
 
   size_t CallbackBufferSize() const { return callback_buffer_size_; }
   bool IsPlaying() { return is_playing_; }
 
   double SampleRate() const { return web_audio_device_->SampleRate(); }
 
   // Returns the audio buffer size in frames used by the underlying audio
   // hardware.
@@ skipping 15 matching lines @@
   // To pass the data from FIFO to the audio device callback.
   RefPtr<AudioBus> output_bus_;
 
   // To push the rendered result from WebAudio graph into the FIFO.
   RefPtr<AudioBus> render_bus_;
 
   // Resolves the buffer size mismatch between the WebAudio engine and
   // the callback function from the actual audio device.
   std::unique_ptr<PushPullFIFO> fifo_;
 
+  // WebThread for WebAudio rendering.
+  std::unique_ptr<WebThread> rendering_thread_;
+
   size_t frames_elapsed_;
   AudioIOPosition output_position_;
   base::TimeTicks output_position_received_timestamp_;
 
   // Check if the buffer size chosen by the WebAudioDevice is too large.
   bool CheckBufferSize();
 
   size_t HardwareBufferSize();
+
+  bool IsRenderingThread();
 };
 
 }  // namespace blink
 
 #endif  // AudioDestination_h