Source/WebCore/webaudio/AudioParam.cpp

/*
 * Copyright (C) 2010 Google Inc. All rights reserved.
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#include "config.h"

#if ENABLE(WEB_AUDIO)

#include "AudioParam.h"

#include "AudioNode.h"
#include "AudioUtilities.h"
#include <wtf/MathExtras.h>

namespace WebCore {

const double AudioParam::DefaultSmoothingConstant = 0.05;
const double AudioParam::SnapThreshold = 0.001;

float AudioParam::value()
{
    // Update value for timeline.
    if (context() && context()->isAudioThread()) {
        bool hasValue;
        float timelineValue = m_timeline.valueForContextTime(context(), m_value, hasValue);

        if (hasValue)
            m_value = timelineValue;
    }

    return static_cast<float>(m_value);
}

void AudioParam::setValue(float value)
{
    // Check against JavaScript giving us bogus floating-point values.
    // Don't ASSERT, since this can happen if somebody writes bad JS.
    if (!isnan(value) && !isinf(value))
        m_value = value;
}

float AudioParam::smoothedValue()
{
    return static_cast<float>(m_smoothedValue);
}

bool AudioParam::smooth()
{
    // If values have been explicitly scheduled on the timeline, then use the exact value.
    // Smoothing effectively is performed by the timeline.
    bool useTimelineValue = false;
    if (context())
        m_value = m_timeline.valueForContextTime(context(), m_value, useTimelineValue);
    
    if (m_smoothedValue == m_value) {
        // Smoothed value has already approached and snapped to value.
        return true;
    }
    
    if (useTimelineValue)
        m_smoothedValue = m_value;
    else {
        // Dezipper - exponential approach.
        m_smoothedValue += (m_value - m_smoothedValue) * m_smoothingConstant;

        // If we get close enough then snap to actual value.
        if (fabs(m_smoothedValue - m_value) < SnapThreshold) // FIXME: the threshold needs to be adjustable depending on range - but this is OK general purpose value.
            m_smoothedValue = m_value;
    }

    return false;
}

void AudioParam::calculateSampleAccurateValues(float* values, unsigned numberOfValues)
{
    bool isSafe = context() && context()->isAudioThread() && values;
    ASSERT(isSafe);
    if (!isSafe)
        return;

    // Calculate values for this render quantum.
    // Normally numberOfValues will equal AudioNode::ProcessingSizeInFrames (the render quantum size).
    float sampleRate = context()->sampleRate();
    float startTime = context()->currentTime();
    float endTime = startTime + numberOfValues / sampleRate;

    // Note we're running control rate at the sample-rate.
    // Pass in the current value as default value.
    m_value = m_timeline.valuesForTimeRange(startTime, endTime, m_value, values, numberOfValues, sampleRate, sampleRate);
}

} // namespace WebCore

#endif // ENABLE(WEB_AUDIO)