////////////////////////////////////////////////////////////////////////////////
///
/// FIR low-pass (anti-alias) filter with filter coefficient design routine and
/// MMX optimization.
///
/// Anti-alias filter is used to prevent folding of high frequencies when
/// transposing the sample rate with interpolation.
///
/// Author        : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
//  SoundTouch audio processing library
//  Copyright (c) Olli Parviainen
//
//  This library is free software; you can redistribute it and/or
//  modify it under the terms of the GNU Lesser General Public
//  License as published by the Free Software Foundation; either
//  version 2.1 of the License, or (at your option) any later version.
//
//  This library is distributed in the hope that it will be useful,
//  but WITHOUT ANY WARRANTY; without even the implied warranty of
//  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
//  Lesser General Public License for more details.
//
//  You should have received a copy of the GNU Lesser General Public
//  License along with this library; if not, write to the Free Software
//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
//
////////////////////////////////////////////////////////////////////////////////

#include "AAFilter.h"
#include "ADM_default.h"
#include "FIRFilter.h"
#include <math.h>
#include <memory.h>
#include <stdlib.h>

using namespace soundtouch;

#define PI 3.14159265358979323846
#define TWOPI (2 * PI)

// define this to save AA filter coefficients to a file
// #define _DEBUG_SAVE_AAFILTER_COEFFICIENTS   1

#ifdef _DEBUG_SAVE_AAFILTER_COEFFICIENTS
#include <stdio.h>

static void _DEBUG_SAVE_AAFIR_COEFFS(SAMPLETYPE *coeffs, int len)
{
    FILE *fptr = fopen("aa_filter_coeffs.txt", "wt");
    if (fptr == NULL)
        return;

    for (int i = 0; i < len; i++)
    {
        double temp = coeffs[i];
        fprintf(fptr, "%lf\n", temp);
    }
    fclose(fptr);
}

#else
#define _DEBUG_SAVE_AAFIR_COEFFS(x, y)
#endif

/*****************************************************************************
 *
 * Implementation of the class 'AAFilter'
 *
 *****************************************************************************/

AAFilter::AAFilter(uint len)
{
    pFIR = FIRFilter::newInstance();
    cutoffFreq = 0.5;
    setLength(len);
}

AAFilter::~AAFilter()
{
    delete pFIR;
}

// Sets new anti-alias filter cut-off edge frequency, scaled to
// sampling frequency (nyquist frequency = 0.5).
// The filter will cut frequencies higher than the given frequency.
void AAFilter::setCutoffFreq(double newCutoffFreq)
{
    cutoffFreq = newCutoffFreq;
    calculateCoeffs();
}

// Sets number of FIR filter taps
void AAFilter::setLength(uint newLength)
{
    length = newLength;
    calculateCoeffs();
}

// Calculates coefficients for a low-pass FIR filter using Hamming window
void AAFilter::calculateCoeffs()
{
    uint i;
    double cntTemp, temp, tempCoeff, h, w;
    double wc;
    double scaleCoeff, sum;
    double *work;
    SAMPLETYPE *coeffs;

    ADM_assert(length >= 2);
    ADM_assert(length % 4 == 0);
    ADM_assert(cutoffFreq >= 0);
    ADM_assert(cutoffFreq <= 0.5);

    work = new double[length];
    coeffs = new SAMPLETYPE[length];

    wc = 2.0 * PI * cutoffFreq;
    tempCoeff = TWOPI / (double)length;

    sum = 0;
    for (i = 0; i < length; i++)
    {
        cntTemp = (double)i - (double)(length / 2);

        temp = cntTemp * wc;
        if (temp != 0)
        {
            h = sin(temp) / temp; // sinc function
        }
        else
        {
            h = 1.0;
        }
        w = 0.54 + 0.46 * cos(tempCoeff * cntTemp); // hamming window

        temp = w * h;
        work[i] = temp;

        // calc net sum of coefficients
        sum += temp;
    }

    // ensure the sum of coefficients is larger than zero
    ADM_assert(sum > 0);

    // ensure we've really designed a lowpass filter...
    ADM_assert(work[length / 2] > 0);
    ADM_assert(work[length / 2 + 1] > -1e-6);
    ADM_assert(work[length / 2 - 1] > -1e-6);

    // Calculate a scaling coefficient in such a way that the result can be
    // divided by 16384
    scaleCoeff = 16384.0f / sum;

    for (i = 0; i < length; i++)
    {
        temp = work[i] * scaleCoeff;
        // scale & round to nearest integer
        temp += (temp >= 0) ? 0.5 : -0.5;
        // ensure no overfloods
        ADM_assert(temp >= -32768 && temp <= 32767);
        coeffs[i] = (SAMPLETYPE)temp;
    }

    // Set coefficients. Use divide factor 14 => divide result by 2^14 = 16384
    pFIR->setCoefficients(coeffs, length, 14);

    _DEBUG_SAVE_AAFIR_COEFFS(coeffs, length);

    delete[] work;
    delete[] coeffs;
}

// Applies the filter to the given sequence of samples.
// Note : The amount of outputted samples is by value of 'filter length'
// smaller than the amount of input samples.
uint AAFilter::evaluate(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples, uint numChannels) const
{
    return pFIR->evaluate(dest, src, numSamples, numChannels);
}

/// Applies the filter to the given src & dest pipes, so that processed amount of
/// samples get removed from src, and produced amount added to dest
/// Note : The amount of outputted samples is by value of 'filter length'
/// smaller than the amount of input samples.
uint AAFilter::evaluate(FIFOSampleBuffer &dest, FIFOSampleBuffer &src) const
{
    SAMPLETYPE *pdest;
    const SAMPLETYPE *psrc;
    uint numSrcSamples;
    uint result;
    int numChannels = src.getChannels();

    ADM_assert(numChannels == dest.getChannels());

    numSrcSamples = src.numSamples();
    psrc = src.ptrBegin();
    pdest = dest.ptrEnd(numSrcSamples);
    result = pFIR->evaluate(pdest, psrc, numSrcSamples, numChannels);
    src.receiveSamples(result);
    dest.putSamples(result);

    return result;
}

uint AAFilter::getLength() const
{
    return pFIR->getLength();
}
