AAFilter.cpp 5.1 KB
Newer Older
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 31 32 33 34 35 36 37 38 39 40 41 42 43 44 45 46 47 48 49 50 51 52 53 54 55 56 57 58 59 60 61 62 63 64 65 66 67 68 69 70 71 72 73 74 75 76 77 78 79 80 81 82 83 84 85 86 87 88 89 90 91 92 93 94 95 96 97 98 99 100 101 102 103 104 105 106 107 108 109 110 111 112 113 114 115 116 117 118 119 120 121 122 123 124 125 126 127 128 129 130 131 132 133 134 135 136 137 138 139 140 141 142 143 144 145 146 147 148 149 150 151 152 153 154 155 156 157 158 159 160 161 162 163 164 165 166 167 168 169 170 171 172 173 174 175 176 177 178 179 180 181 182 183 184
////////////////////////////////////////////////////////////////////////////////
///
/// 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
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed  : $Date: 2009-01-11 13:34:24 +0200 (Sun, 11 Jan 2009) $
// File revision : $Revision: 4 $
//
// $Id: AAFilter.cpp 45 2009-01-11 11:34:24Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// 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 <memory.h>
#include <assert.h>
#include <math.h>
#include <stdlib.h>
#include "AAFilter.h"
#include "FIRFilter.h"

using namespace soundtouch;

#define PI        3.141592655357989
#define TWOPI    (2 * PI)

/*****************************************************************************
 *
 * 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 fc2, wc;
    double scaleCoeff, sum;
    double *work;
    SAMPLETYPE *coeffs;

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

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

    fc2 = 2.0 * cutoffFreq; 
    wc = PI * fc2;
    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 = fc2 * 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
    assert(sum > 0);

    // ensure we've really designed a lowpass filter...
    assert(work[length/2] > 0);
    assert(work[length/2 + 1] > -1e-6);
    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 ++) 
    {
        // scale & round to nearest integer
        temp = work[i] * scaleCoeff;
        temp += (temp >= 0) ? 0.5 : -0.5;
        // ensure no overfloods
        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);

    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);
}


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