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diff --git a/lib/soundtouch/AAFilter.cpp b/lib/soundtouch/AAFilter.cpp
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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 == 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();
-}