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AAFilter.cpp (6143B)

---

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

using namespace soundtouch;

#define PI       M_PI
#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;

   assert(length >= 2);
   assert(length % 4 == 0);
   assert(cutoffFreq >= 0);
   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
   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 ++) 
   {
       temp = work[i] * scaleCoeff;
       // scale & round to nearest integer
       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);

   _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();

   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();
}