/*
  ==============================================================================

   This file is part of the JUCE library.
   Copyright (c) 2016 - ROLI Ltd.

   Permission is granted to use this software under the terms of the ISC license
   http://www.isc.org/downloads/software-support-policy/isc-license/

   Permission to use, copy, modify, and/or distribute this software for any
   purpose with or without fee is hereby granted, provided that the above
   copyright notice and this permission notice appear in all copies.

   THE SOFTWARE IS PROVIDED "AS IS" AND ISC DISCLAIMS ALL WARRANTIES WITH REGARD
   TO THIS SOFTWARE INCLUDING ALL IMPLIED WARRANTIES OF MERCHANTABILITY AND
   FITNESS. IN NO EVENT SHALL ISC BE LIABLE FOR ANY SPECIAL, DIRECT, INDIRECT,
   OR CONSEQUENTIAL DAMAGES OR ANY DAMAGES WHATSOEVER RESULTING FROM LOSS OF
   USE, DATA OR PROFITS, WHETHER IN AN ACTION OF CONTRACT, NEGLIGENCE OR OTHER
   TORTIOUS ACTION, ARISING OUT OF OR IN CONNECTION WITH THE USE OR PERFORMANCE
   OF THIS SOFTWARE.

   -----------------------------------------------------------------------------

   To release a closed-source product which uses other parts of JUCE not
   licensed under the ISC terms, commercial licenses are available: visit
   www.juce.com for more information.

  ==============================================================================
*/

ResamplingAudioSource::ResamplingAudioSource (AudioSource* const inputSource,
                                              const bool deleteInputWhenDeleted,
                                              const int channels)
    : input (inputSource, deleteInputWhenDeleted),
      ratio (1.0),
      lastRatio (1.0),
      bufferPos (0),
      sampsInBuffer (0),
      subSampleOffset (0),
      numChannels (channels)
{
    jassert (input != nullptr);
    zeromem (coefficients, sizeof (coefficients));
}

ResamplingAudioSource::~ResamplingAudioSource() {}

void ResamplingAudioSource::setResamplingRatio (const double samplesInPerOutputSample)
{
    jassert (samplesInPerOutputSample > 0);

    const SpinLock::ScopedLockType sl (ratioLock);
    ratio = jmax (0.0, samplesInPerOutputSample);
}

void ResamplingAudioSource::prepareToPlay (int samplesPerBlockExpected, double sampleRate)
{
    const SpinLock::ScopedLockType sl (ratioLock);

    const int scaledBlockSize = roundToInt (samplesPerBlockExpected * ratio);
    input->prepareToPlay (scaledBlockSize, sampleRate * ratio);

    buffer.setSize (numChannels, scaledBlockSize + 32);

    filterStates.calloc ((size_t) numChannels);
    srcBuffers.calloc ((size_t) numChannels);
    destBuffers.calloc ((size_t) numChannels);
    createLowPass (ratio);

    flushBuffers();
}

void ResamplingAudioSource::flushBuffers()
{
    buffer.clear();
    bufferPos = 0;
    sampsInBuffer = 0;
    subSampleOffset = 0.0;
    resetFilters();
}

void ResamplingAudioSource::releaseResources()
{
    input->releaseResources();
    buffer.setSize (numChannels, 0);
}

void ResamplingAudioSource::getNextAudioBlock (const AudioSourceChannelInfo& info)
{
    double localRatio;

    {
        const SpinLock::ScopedLockType sl (ratioLock);
        localRatio = ratio;
    }

    if (lastRatio != localRatio)
    {
        createLowPass (localRatio);
        lastRatio = localRatio;
    }

    const int sampsNeeded = roundToInt (info.numSamples * localRatio) + 3;

    int bufferSize = buffer.getNumSamples();

    if (bufferSize < sampsNeeded + 8)
    {
        bufferPos %= bufferSize;
        bufferSize = sampsNeeded + 32;
        buffer.setSize (buffer.getNumChannels(), bufferSize, true, true);
    }

    bufferPos %= bufferSize;

    int endOfBufferPos = bufferPos + sampsInBuffer;
    const int channelsToProcess = jmin (numChannels, info.buffer->getNumChannels());

    while (sampsNeeded > sampsInBuffer)
    {
        endOfBufferPos %= bufferSize;

        int numToDo = jmin (sampsNeeded - sampsInBuffer,
                            bufferSize - endOfBufferPos);

        AudioSourceChannelInfo readInfo (&buffer, endOfBufferPos, numToDo);
        input->getNextAudioBlock (readInfo);

        if (localRatio > 1.0001)
        {
            // for down-sampling, pre-apply the filter..

            for (int i = channelsToProcess; --i >= 0;)
                applyFilter (buffer.getWritePointer (i, endOfBufferPos), numToDo, filterStates[i]);
        }

        sampsInBuffer += numToDo;
        endOfBufferPos += numToDo;
    }

    for (int channel = 0; channel < channelsToProcess; ++channel)
    {
        destBuffers[channel] = info.buffer->getWritePointer (channel, info.startSample);
        srcBuffers[channel] = buffer.getReadPointer (channel);
    }

    int nextPos = (bufferPos + 1) % bufferSize;

    for (int m = info.numSamples; --m >= 0;)
    {
        jassert (sampsInBuffer > 0 && nextPos != endOfBufferPos);

        const float alpha = (float) subSampleOffset;

        for (int channel = 0; channel < channelsToProcess; ++channel)
            *destBuffers[channel]++ = srcBuffers[channel][bufferPos]
                                        + alpha * (srcBuffers[channel][nextPos] - srcBuffers[channel][bufferPos]);

        subSampleOffset += localRatio;

        while (subSampleOffset >= 1.0)
        {
            if (++bufferPos >= bufferSize)
                bufferPos = 0;

            --sampsInBuffer;

            nextPos = (bufferPos + 1) % bufferSize;
            subSampleOffset -= 1.0;
        }
    }

    if (localRatio < 0.9999)
    {
        // for up-sampling, apply the filter after transposing..
        for (int i = channelsToProcess; --i >= 0;)
            applyFilter (info.buffer->getWritePointer (i, info.startSample), info.numSamples, filterStates[i]);
    }
    else if (localRatio <= 1.0001 && info.numSamples > 0)
    {
        // if the filter's not currently being applied, keep it stoked with the last couple of samples to avoid discontinuities
        for (int i = channelsToProcess; --i >= 0;)
        {
            const float* const endOfBuffer = info.buffer->getReadPointer (i, info.startSample + info.numSamples - 1);
            FilterState& fs = filterStates[i];

            if (info.numSamples > 1)
            {
                fs.y2 = fs.x2 = *(endOfBuffer - 1);
            }
            else
            {
                fs.y2 = fs.y1;
                fs.x2 = fs.x1;
            }

            fs.y1 = fs.x1 = *endOfBuffer;
        }
    }

    jassert (sampsInBuffer >= 0);
}

void ResamplingAudioSource::createLowPass (const double frequencyRatio)
{
    const double proportionalRate = (frequencyRatio > 1.0) ? 0.5 / frequencyRatio
                                                           : 0.5 * frequencyRatio;

    const double n = 1.0 / std::tan (double_Pi * jmax (0.001, proportionalRate));
    const double nSquared = n * n;
    const double c1 = 1.0 / (1.0 + std::sqrt (2.0) * n + nSquared);

    setFilterCoefficients (c1,
                           c1 * 2.0f,
                           c1,
                           1.0,
                           c1 * 2.0 * (1.0 - nSquared),
                           c1 * (1.0 - std::sqrt (2.0) * n + nSquared));
}

void ResamplingAudioSource::setFilterCoefficients (double c1, double c2, double c3, double c4, double c5, double c6)
{
    const double a = 1.0 / c4;

    c1 *= a;
    c2 *= a;
    c3 *= a;
    c5 *= a;
    c6 *= a;

    coefficients[0] = c1;
    coefficients[1] = c2;
    coefficients[2] = c3;
    coefficients[3] = c4;
    coefficients[4] = c5;
    coefficients[5] = c6;
}

void ResamplingAudioSource::resetFilters()
{
    if (filterStates != nullptr)
        filterStates.clear ((size_t) numChannels);
}

void ResamplingAudioSource::applyFilter (float* samples, int num, FilterState& fs)
{
    while (--num >= 0)
    {
        const double in = *samples;

        double out = coefficients[0] * in
                     + coefficients[1] * fs.x1
                     + coefficients[2] * fs.x2
                     - coefficients[4] * fs.y1
                     - coefficients[5] * fs.y2;

       #if JUCE_INTEL
        if (! (out < -1.0e-8 || out > 1.0e-8))
            out = 0;
       #endif

        fs.x2 = fs.x1;
        fs.x1 = in;
        fs.y2 = fs.y1;
        fs.y1 = out;

        *samples++ = (float) out;
    }
}
