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

  This file is part of the dRowAudio JUCE module
  Copyright 2004-13 by dRowAudio.

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

  dRowAudio is provided under the terms of The MIT License (MIT):

  Permission is hereby granted, free of charge, to any person obtaining a copy
  of this software and associated documentation files (the "Software"), to deal
  in the Software without restriction, including without limitation the rights
  to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
  copies of the Software, and to permit persons to whom the Software is
  furnished to do so, subject to the following conditions:

  The above copyright notice and this permission notice shall be included in all
  copies or substantial portions of the Software.

  THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
  IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, 
  FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE 
  AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER 
  LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
  OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE 
  SOFTWARE.

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



//==============================================================================
SampleRateConverter::SampleRateConverter (const int numChannels_)
    : ratio (1.0),
      numChannels (numChannels_)
{
    filterStates.calloc (numChannels);

    createLowPass (ratio);
    resetFilters();    
}

SampleRateConverter::~SampleRateConverter()
{
}

//==============================================================================
void SampleRateConverter::process (float** inputChannelData, int numInputChannels, int numInputSamples,
                                   float** outputChannelData, int numOutputChannels, int numOutputSamples)
{
    // take a copy of the outpur channel pointers so they can be re-used afterwards
    float* outputChannels[256];
    for (int i = 0; i < numOutputChannels; ++i)
        outputChannels[i] = outputChannelData[i];
    
    outputChannels[numOutputChannels] = nullptr;
    
    const int channelsToProcess = jmin (numInputChannels, numOutputChannels);
    const double localRatio = numInputSamples / (double) numOutputSamples;
        
    if (localRatio != ratio)
    {
        createLowPass (localRatio);
        ratio = localRatio;
    }
    
    if (localRatio > 1.0001)
    {
        // for down-sampling, pre-apply the filter..
        for (int i = channelsToProcess; --i >= 0;)
            applyFilter (inputChannelData[i], numInputSamples, filterStates[i]);
    }
    
    float nextSample = 0.0f;
    for (int s = 0; s < numOutputSamples; ++s)
    {
        for (int channel = 0; channel < channelsToProcess; ++channel)
            *outputChannels[channel]++ = linearInterpolate (inputChannelData[channel], numInputSamples, nextSample);
        
        nextSample += (float) localRatio;
    }
    
    if (localRatio < 0.9999)
    {
        // for up-sampling, apply the filter after transposing..
        for (int i = channelsToProcess; --i >= 0;)
            applyFilter (inputChannelData[i], numInputSamples, filterStates[i]);
    }
    else if (localRatio <= 1.0001 && numInputSamples > 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 = &inputChannelData[i][numInputSamples - 1];
            FilterState& fs = filterStates[i];
            
            if (numInputSamples > 1)
            {
                fs.y2 = fs.x2 = *(endOfBuffer - 1);
            }
            else
            {
                fs.y2 = fs.y1;
                fs.x2 = fs.x1;
            }
            
            fs.y1 = fs.x1 = *endOfBuffer;
        }
    }
}

//==============================================================================
void SampleRateConverter::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 SampleRateConverter::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 SampleRateConverter::resetFilters()
{
    filterStates.clear (numChannels);
}

void SampleRateConverter::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;
    }
}