/*
 * Copyright (C) 2010 Google Inc. All rights reserved.
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 * 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
 *     notice, this list of conditions and the following disclaimer in the
 *     documentation and/or other materials provided with the distribution.
 * 3.  Neither the name of Apple Computer, Inc. ("Apple") nor the names of
 *     its contributors may be used to endorse or promote products derived
 *     from this software without specific prior written permission.
 *
 * THIS SOFTWARE IS PROVIDED BY APPLE AND ITS CONTRIBUTORS "AS IS" AND ANY
 * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
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 */

#ifndef AudioParam_h
#define AudioParam_h

#include "AudioContext.h"
#include "AudioParamTimeline.h"
#include "AudioSummingJunction.h"
#include <runtime/Float32Array.h>
#include <sys/types.h>
#include <wtf/PassRefPtr.h>
#include <wtf/RefCounted.h>
#include <wtf/text/WTFString.h>

namespace WebCore {

class AudioNodeOutput;

class AudioParam : public AudioSummingJunction, public RefCounted<AudioParam> {
public:
    static const double DefaultSmoothingConstant;
    static const double SnapThreshold;

    static PassRefPtr<AudioParam> create(AudioContext* context, const String& name, double defaultValue, double minValue, double maxValue, unsigned units = 0)
    {
        return adoptRef(new AudioParam(context, name, defaultValue, minValue, maxValue, units));
    }

    // AudioSummingJunction
    virtual bool canUpdateState() OVERRIDE { return true; }
    virtual void didUpdate() OVERRIDE { }

    // Intrinsic value.
    float value();
    void setValue(float);

    // Final value for k-rate parameters, otherwise use calculateSampleAccurateValues() for a-rate.
    // Must be called in the audio thread.
    float finalValue();

    String name() const { return m_name; }

    float minValue() const { return static_cast<float>(m_minValue); }
    float maxValue() const { return static_cast<float>(m_maxValue); }
    float defaultValue() const { return static_cast<float>(m_defaultValue); }
    unsigned units() const { return m_units; }

    // Value smoothing:

    // When a new value is set with setValue(), in our internal use of the parameter we don't immediately jump to it.
    // Instead we smoothly approach this value to avoid glitching.
    float smoothedValue();

    // Smoothly exponentially approaches to (de-zippers) the desired value.
    // Returns true if smoothed value has already snapped exactly to value.
    bool smooth();

    void resetSmoothedValue() { m_smoothedValue = m_value; }
    void setSmoothingConstant(double k) { m_smoothingConstant = k; }

    // Parameter automation.    
    void setValueAtTime(float value, float time) { m_timeline.setValueAtTime(value, time); }
    void linearRampToValueAtTime(float value, float time) { m_timeline.linearRampToValueAtTime(value, time); }
    void exponentialRampToValueAtTime(float value, float time) { m_timeline.exponentialRampToValueAtTime(value, time); }
    void setTargetAtTime(float target, float time, float timeConstant) { m_timeline.setTargetAtTime(target, time, timeConstant); }
    void setValueCurveAtTime(Float32Array* curve, float time, float duration) { m_timeline.setValueCurveAtTime(curve, time, duration); }
    void cancelScheduledValues(float startTime) { m_timeline.cancelScheduledValues(startTime); }

    bool hasSampleAccurateValues() { return m_timeline.hasValues() || numberOfRenderingConnections(); }
    
    // Calculates numberOfValues parameter values starting at the context's current time.
    // Must be called in the context's render thread.
    void calculateSampleAccurateValues(float* values, unsigned numberOfValues);

    // Connect an audio-rate signal to control this parameter.
    void connect(AudioNodeOutput*);
    void disconnect(AudioNodeOutput*);

protected:
    AudioParam(AudioContext* context, const String& name, double defaultValue, double minValue, double maxValue, unsigned units = 0)
        : AudioSummingJunction(context)
        , m_name(name)
        , m_value(defaultValue)
        , m_defaultValue(defaultValue)
        , m_minValue(minValue)
        , m_maxValue(maxValue)
        , m_units(units)
        , m_smoothedValue(defaultValue)
        , m_smoothingConstant(DefaultSmoothingConstant)
    {
    }

private:
    // sampleAccurate corresponds to a-rate (audio rate) vs. k-rate in the Web Audio specification.
    void calculateFinalValues(float* values, unsigned numberOfValues, bool sampleAccurate);
    void calculateTimelineValues(float* values, unsigned numberOfValues);

    String m_name;
    double m_value;
    double m_defaultValue;
    double m_minValue;
    double m_maxValue;
    unsigned m_units;

    // Smoothing (de-zippering)
    double m_smoothedValue;
    double m_smoothingConstant;
    
    AudioParamTimeline m_timeline;
};

} // namespace WebCore

#endif // AudioParam_h
