sonyoss-vita-webkit/Source/WebCore/Modules/webaudio/DelayDSPKernel.cpp

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

#if ENABLE(WEB_AUDIO)

#include "DelayDSPKernel.h"

#include "AudioUtilities.h"
#include <algorithm>

using namespace std;

namespace WebCore {

const float SmoothingTimeConstant = 0.020f; // 20ms

DelayDSPKernel::DelayDSPKernel(DelayProcessor* processor)
    : AudioDSPKernel(processor)
    , m_writeIndex(0)
    , m_firstTime(true)
    , m_delayTimes(AudioNode::ProcessingSizeInFrames)
{
    ASSERT(processor && processor->sampleRate() > 0);
    if (!(processor && processor->sampleRate() > 0))
        return;

    m_maxDelayTime = processor->maxDelayTime();
    ASSERT(m_maxDelayTime >= 0);
    if (m_maxDelayTime < 0)
        return;

    m_buffer.allocate(bufferLengthForDelay(m_maxDelayTime, processor->sampleRate()));
    m_buffer.zero();

    m_smoothingRate = AudioUtilities::discreteTimeConstantForSampleRate(SmoothingTimeConstant, processor->sampleRate());
}

DelayDSPKernel::DelayDSPKernel(double maxDelayTime, float sampleRate)
    : AudioDSPKernel(sampleRate)
    , m_maxDelayTime(maxDelayTime)
    , m_writeIndex(0)
    , m_firstTime(true)
{
    ASSERT(maxDelayTime > 0.0);
    if (maxDelayTime <= 0.0)
        return;

    size_t bufferLength = bufferLengthForDelay(maxDelayTime, sampleRate);
    ASSERT(bufferLength);
    if (!bufferLength)
        return;

    m_buffer.allocate(bufferLength);
    m_buffer.zero();

    m_smoothingRate = AudioUtilities::discreteTimeConstantForSampleRate(SmoothingTimeConstant, sampleRate);
}

size_t DelayDSPKernel::bufferLengthForDelay(double maxDelayTime, double sampleRate) const
{
    // Compute the length of the buffer needed to handle a max delay of |maxDelayTime|. One is
    // added to handle the case where the actual delay equals the maximum delay.
    return 1 + AudioUtilities::timeToSampleFrame(maxDelayTime, sampleRate);
}

void DelayDSPKernel::process(const float* source, float* destination, size_t framesToProcess)
{
    size_t bufferLength = m_buffer.size();
    float* buffer = m_buffer.data();

    ASSERT(bufferLength);
    if (!bufferLength)
        return;

    ASSERT(source && destination);
    if (!source || !destination)
        return;

    float sampleRate = this->sampleRate();
    double delayTime = 0;
    float* delayTimes = m_delayTimes.data();
    double maxTime = maxDelayTime();

    bool sampleAccurate = delayProcessor() && delayProcessor()->delayTime()->hasSampleAccurateValues();

    if (sampleAccurate)
        delayProcessor()->delayTime()->calculateSampleAccurateValues(delayTimes, framesToProcess);
    else {
        delayTime = delayProcessor() ? delayProcessor()->delayTime()->finalValue() : m_desiredDelayFrames / sampleRate;

        // Make sure the delay time is in a valid range.
        delayTime = min(maxTime, delayTime);
        delayTime = max(0.0, delayTime);

        if (m_firstTime) {
            m_currentDelayTime = delayTime;
            m_firstTime = false;
        }
    }

    for (unsigned i = 0; i < framesToProcess; ++i) {
        if (sampleAccurate) {
            delayTime = delayTimes[i];
            delayTime = std::min(maxTime, delayTime);
            delayTime = std::max(0.0, delayTime);
            m_currentDelayTime = delayTime;
        } else {
            // Approach desired delay time.
            m_currentDelayTime += (delayTime - m_currentDelayTime) * m_smoothingRate;
        }

        double desiredDelayFrames = m_currentDelayTime * sampleRate;

        double readPosition = m_writeIndex + bufferLength - desiredDelayFrames;
        if (readPosition >= bufferLength)
            readPosition -= bufferLength;

        // Linearly interpolate in-between delay times.
        int readIndex1 = static_cast<int>(readPosition);
        int readIndex2 = (readIndex1 + 1) % bufferLength;
        double interpolationFactor = readPosition - readIndex1;

        double input = static_cast<float>(*source++);
        buffer[m_writeIndex] = static_cast<float>(input);
        m_writeIndex = (m_writeIndex + 1) % bufferLength;

        double sample1 = buffer[readIndex1];
        double sample2 = buffer[readIndex2];

        double output = (1.0 - interpolationFactor) * sample1 + interpolationFactor * sample2;

        *destination++ = static_cast<float>(output);
    }
}

void DelayDSPKernel::reset()
{
    m_firstTime = true;
    m_buffer.zero();
}

double DelayDSPKernel::tailTime() const
{
    return m_maxDelayTime;
}

double DelayDSPKernel::latencyTime() const
{
    return 0;
}

} // namespace WebCore

#endif // ENABLE(WEB_AUDIO)