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- ////////////////////////////////////////////////////////////////////////////////
- ///
- /// 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 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 == nullptr) 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();
- }
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