/*
 * 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.
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 *    notice, this list of conditions and the following disclaimer in the
 *    documentation and/or other materials provided with the distribution.
 *
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 * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
 * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
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#ifndef AudioBufferSourceNode_h
#define AudioBufferSourceNode_h

#include "AudioBuffer.h"
#include "AudioBus.h"
#include "AudioGain.h"
#include "AudioPannerNode.h"
#include "AudioSourceNode.h"
#include <wtf/PassRefPtr.h>
#include <wtf/RefPtr.h>
#include <wtf/Threading.h>

namespace WebCore {

class AudioContext;

// AudioBufferSourceNode is an AudioNode representing an audio source from an in-memory audio asset represented by an AudioBuffer.
// It generally will be used for short sounds which require a high degree of scheduling flexibility (can playback in rhythmically perfect ways).

class AudioBufferSourceNode : public AudioSourceNode {
public:
    static PassRefPtr<AudioBufferSourceNode> create(AudioContext*, double sampleRate);

    virtual ~AudioBufferSourceNode();
    
    // AudioNode
    virtual void process(size_t framesToProcess);
    virtual void reset();
    
    // setBuffer() is called on the main thread.  This is the buffer we use for playback.
    void setBuffer(AudioBuffer*);
    AudioBuffer* buffer() { return m_buffer.get(); }
                    
    // numberOfChannels() returns the number of output channels.  This value equals the number of channels from the buffer.
    // If a new buffer is set with a different number of channels, then this value will dynamically change.
    unsigned numberOfChannels();
                    
    // Play-state
    // noteOn(), noteGrainOn(), and noteOff() must all be called from the main thread.
    void noteOn(double when);
    void noteGrainOn(double when, double grainOffset, double grainDuration);
    void noteOff(double when);

    bool looping() const { return m_isLooping; }
    void setLooping(bool looping) { m_isLooping = looping; }
    
    AudioGain* gain() { return m_gain.get(); }                                        
    AudioParam* playbackRate() { return m_playbackRate.get(); }

    // If a panner node is set, then we can incorporate doppler shift into the playback pitch rate.
    void setPannerNode(PassRefPtr<AudioPannerNode> pannerNode) { m_pannerNode = pannerNode; }

private:
    AudioBufferSourceNode(AudioContext*, double sampleRate);

    void renderFromBuffer(AudioBus*, unsigned destinationFrameOffset, size_t numberOfFrames);

    // m_buffer holds the sample data which this node outputs.
    RefPtr<AudioBuffer> m_buffer;

    // Used for the "gain" and "playbackRate" attributes.
    RefPtr<AudioGain> m_gain;
    RefPtr<AudioParam> m_playbackRate;

    // m_isPlaying is set to true when noteOn() or noteGrainOn() is called.
    bool m_isPlaying;

    // If m_isLooping is false, then this node will be done playing and become inactive after it reaches the end of the sample data in the buffer.
    // If true, it will wrap around to the start of the buffer each time it reaches the end.
    bool m_isLooping;

    // This node is considered finished when it reaches the end of the buffer's sample data after noteOn() has been called.
    // This will only be set to true if m_isLooping == false.
    bool m_hasFinished;

    // m_startTime is the time to start playing based on the context's timeline (0.0 or a time less than the context's current time means "now").
    double m_startTime; // in seconds

    // m_endTime is the time to stop playing based on the context's timeline (0.0 or a time less than the context's current time means "now").
    // If it hasn't been set explicitly, then the sound will not stop playing (if looping) or will stop when the end of the AudioBuffer
    // has been reached.
    double m_endTime; // in seconds
    
    // m_virtualReadIndex is a sample-frame index into our buffer representing the current playback position.
    // Since it's floating-point, it has sub-sample accuracy.
    double m_virtualReadIndex;

    // Granular playback
    bool m_isGrain;
    double m_grainOffset; // in seconds
    double m_grainDuration; // in seconds

    // totalPitchRate() returns the instantaneous pitch rate (non-time preserving).
    // It incorporates the base pitch rate, any sample-rate conversion factor from the buffer, and any doppler shift from an associated panner node.
    double totalPitchRate();

    // m_lastGain provides continuity when we dynamically adjust the gain.
    double m_lastGain;
    
    // We optionally keep track of a panner node which has a doppler shift that is incorporated into the pitch rate.
    RefPtr<AudioPannerNode> m_pannerNode;

    // This synchronizes process() with setBuffer() which can cause dynamic channel count changes.
    mutable Mutex m_processLock;

    // Handles the time when we reach the end of sample data (non-looping) or the noteOff() time has been reached.
    void finish();
};

} // namespace WebCore

#endif // AudioBufferSourceNode_h
