Clean up and refactor platform/AudioDestination

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

Index: third_party/WebKit/Source/platform/audio/AudioDestination.h
diff --git a/third_party/WebKit/Source/platform/audio/AudioDestination.h b/third_party/WebKit/Source/platform/audio/AudioDestination.h
index f1d13c3d6ce715714496458f324d1c2cd7459813..695eb3c05171ba86914e1056a6578229c691f7c4 100644
--- a/third_party/WebKit/Source/platform/audio/AudioDestination.h
+++ b/third_party/WebKit/Source/platform/audio/AudioDestination.h
@@ -36,7 +36,6 @@
 #include "public/platform/WebVector.h"
 #include "wtf/Allocator.h"
 #include "wtf/Noncopyable.h"
-#include "wtf/PassRefPtr.h"
 #include "wtf/text/WTFString.h"
 #include <memory>
 
@@ -45,8 +44,10 @@ namespace blink {
 class AudioPullFIFO;
 class SecurityOrigin;
 
-// An AudioDestination using Chromium's audio system
-
+// 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,
                                          public AudioSourceProvider {
   USING_FAST_MALLOC(AudioDestination);
@@ -54,66 +55,60 @@ class PLATFORM_EXPORT AudioDestination : public WebAudioDevice::RenderCallback,
 
  public:
   AudioDestination(AudioIOCallback&,
-                   const String& inputDeviceId,
-                   unsigned numberOfInputChannels,
                    unsigned numberOfOutputChannels,
                    float sampleRate,
                    PassRefPtr<SecurityOrigin>);
   ~AudioDestination() override;
 
-  // Pass in (numberOfInputChannels > 0) if live/local audio input is desired.
-  // Port-specific device identification information for live/local input
-  // streams can be passed in the inputDeviceId.
   static std::unique_ptr<AudioDestination> create(
       AudioIOCallback&,
-      const String& inputDeviceId,
-      unsigned numberOfInputChannels,
       unsigned numberOfOutputChannels,
       float sampleRate,
       PassRefPtr<SecurityOrigin>);
 
-  virtual void start();
-  virtual void stop();
-  bool isPlaying() { return m_isPlaying; }
-
-  float sampleRate() const { return m_sampleRate; }
-
-  // WebAudioDevice::RenderCallback
-  void render(const WebVector<float*>& audioData,
+  // The actual render function (WebAudioDevice::RenderCallback) isochronously
+  // invoked by the media renderer.
+  void render(const WebVector<float*>& destinationData,
               size_t numberOfFrames,
               double delay,
               double delayTimestamp,
               size_t priorFramesSkipped) override;
 
-  // AudioSourceProvider
-  void provideInput(AudioBus*, size_t framesToProcess) override;
+  // AudioSourceProvider (FIFO)
+  void provideInput(AudioBus* outputBus, size_t framesToProcess) override;
 
-  static float hardwareSampleRate();
+  virtual void start();
+  virtual void stop();
 
   size_t callbackBufferSize() const { return m_callbackBufferSize; }
+  float sampleRate() const { return m_sampleRate; }
+  bool isPlaying() { return m_isPlaying; }
 
-  // maxChannelCount() returns the total number of output channels of the audio
-  // hardware.  A value of 0 indicates that the number of channels cannot be
-  // configured and that only stereo (2-channel) destinations can be created.
-  // The numberOfOutputChannels parameter of AudioDestination::create() is
-  // allowed to be a value: 1 <= numberOfOutputChannels <= maxChannelCount(),
-  // or if maxChannelCount() equals 0, then numberOfOutputChannels must be 2.
+  // The information from the actual audio hardware. (via Platform::current)
+  static size_t hardwareBufferSize();
+  static float hardwareSampleRate();
   static unsigned long maxChannelCount();
 
  private:
-  AudioIOCallback& m_callback;
+  std::unique_ptr<WebAudioDevice> m_webAudioDevice;
   unsigned m_numberOfOutputChannels;
-  RefPtr<AudioBus> m_renderBus;
+  size_t m_callbackBufferSize;
   float m_sampleRate;
   bool m_isPlaying;
-  std::unique_ptr<WebAudioDevice> m_audioDevice;
-  size_t m_callbackBufferSize;
 
+  // The render callback function of WebAudio engine. (i.e. DestinationNode)
+  AudioIOCallback& m_callback;
+
+  RefPtr<AudioBus> m_outputBus;
   std::unique_ptr<AudioPullFIFO> m_fifo;
 
   size_t m_framesElapsed;
   AudioIOPosition m_outputPosition;
   base::TimeTicks m_outputPositionReceivedTimestamp;
+
+  // Calculate the optimum buffer size for a given platform. Return false if the
+  // buffer size calculation fails.
+  bool calculateBufferSize();
 };
 
 }  // namespace blink