root/Source/modules/webaudio/ScriptProcessorNode.h

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 * Copyright (C) 2010, Google Inc. All rights reserved.
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#ifndef ScriptProcessorNode_h
#define ScriptProcessorNode_h

#include "platform/audio/AudioBus.h"
#include "modules/webaudio/AudioNode.h"
#include "wtf/Forward.h"
#include "wtf/PassRefPtr.h"
#include "wtf/RefPtr.h"
#include "wtf/Vector.h"

namespace WebCore {

class AudioBuffer;
class AudioContext;
class AudioProcessingEvent;

// ScriptProcessorNode is an AudioNode which allows for arbitrary synthesis or processing directly using JavaScript.
// The API allows for a variable number of inputs and outputs, although it must have at least one input or output.
// This basic implementation supports no more than one input and output.
// The "onaudioprocess" attribute is an event listener which will get called periodically with an AudioProcessingEvent which has
// AudioBuffers for each input and output.

class ScriptProcessorNode FINAL : public AudioNode {
public:
    // bufferSize must be one of the following values: 256, 512, 1024, 2048, 4096, 8192, 16384.
    // This value controls how frequently the onaudioprocess event handler is called and how many sample-frames need to be processed each call.
    // Lower numbers for bufferSize will result in a lower (better) latency. Higher numbers will be necessary to avoid audio breakup and glitches.
    // The value chosen must carefully balance between latency and audio quality.
    static PassRefPtr<ScriptProcessorNode> create(AudioContext*, float sampleRate, size_t bufferSize, unsigned numberOfInputChannels, unsigned numberOfOutputChannels);

    virtual ~ScriptProcessorNode();

    // AudioNode
    virtual void process(size_t framesToProcess) OVERRIDE;
    virtual void initialize() OVERRIDE;
    virtual void uninitialize() OVERRIDE;

    size_t bufferSize() const { return m_bufferSize; }

    DEFINE_ATTRIBUTE_EVENT_LISTENER(audioprocess);


private:
    virtual double tailTime() const OVERRIDE;
    virtual double latencyTime() const OVERRIDE;

    ScriptProcessorNode(AudioContext*, float sampleRate, size_t bufferSize, unsigned numberOfInputChannels, unsigned numberOfOutputChannels);

    static void fireProcessEventDispatch(void* userData);
    void fireProcessEvent();

    // Double buffering
    unsigned doubleBufferIndex() const { return m_doubleBufferIndex; }
    void swapBuffers() { m_doubleBufferIndex = 1 - m_doubleBufferIndex; }
    unsigned m_doubleBufferIndex;
    unsigned m_doubleBufferIndexForEvent;
    Vector<RefPtr<AudioBuffer> > m_inputBuffers;
    Vector<RefPtr<AudioBuffer> > m_outputBuffers;

    size_t m_bufferSize;
    unsigned m_bufferReadWriteIndex;

    unsigned m_numberOfInputChannels;
    unsigned m_numberOfOutputChannels;

    RefPtr<AudioBus> m_internalInputBus;

    // Synchronize process() with fireProcessEvent().
    mutable Mutex m_processEventLock;
};

} // namespace WebCore

#endif // ScriptProcessorNode_h

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