Support for AudioContextOptions latencyHint.

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 27 matching lines @@
 #include "wtf/Noncopyable.h"
 #include "wtf/PassRefPtr.h"
 #include "wtf/text/WTFString.h"
 #include <memory>
 
 namespace blink {
 
 class AudioFIFO;
 class AudioPullFIFO;
 class SecurityOrigin;
+class WebAudioLatencyHint;
 
 // An AudioDestination using Chromium's audio system
 
 class PLATFORM_EXPORT AudioDestination : public WebAudioDevice::RenderCallback,
                                          public AudioSourceProvider {
   USING_FAST_MALLOC(AudioDestination);
   WTF_MAKE_NONCOPYABLE(AudioDestination);
 
  public:
   AudioDestination(AudioIOCallback&,
                    const String& inputDeviceId,
                    unsigned numberOfInputChannels,
                    unsigned numberOfOutputChannels,
-                   float sampleRate,
+                   const WebAudioLatencyHint&,
                    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,
+      const WebAudioLatencyHint&,
       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*>& sourceData,
               const WebVector<float*>& audioData,
               size_t numberOfFrames) override;
 
   // AudioSourceProvider
   void provideInput(AudioBus*, size_t framesToProcess) override;
 
   static float hardwareSampleRate();
 
   // 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.
   static unsigned long maxChannelCount();
 
  private:
   AudioIOCallback& m_callback;
   unsigned m_numberOfOutputChannels;
   RefPtr<AudioBus> m_inputBus;
   RefPtr<AudioBus> m_renderBus;
-  float m_sampleRate;

[hongchan, 2016/11/15 23:46:01]

How do I query the sample rate of the destination then? This is the WebAudio
abstraction of the underlying device/thread, we don't want to lose the critical
info like the sample rate.

I am not sure if WebAudioLatencyHint can completely replace the m_sampleRate
variable.

[Andrew MacPherson, 2016/11/16 10:58:30]

My understanding is that this should be queried using
m_audioDevice->sampleRate().

   bool m_isPlaying;
   std::unique_ptr<WebAudioDevice> m_audioDevice;
   size_t m_callbackBufferSize;
 
   std::unique_ptr<AudioFIFO> m_inputFifo;
   std::unique_ptr<AudioPullFIFO> m_fifo;
 };
 
 }  // namespace blink
 
 #endif  // AudioDestination_h