1.1 --- /dev/null Thu Jan 01 00:00:00 1970 +0000
1.2 +++ b/media/libsoundtouch/src/AAFilter.cpp Wed Dec 31 06:09:35 2014 +0100
1.3 @@ -0,0 +1,236 @@
1.4 +////////////////////////////////////////////////////////////////////////////////
1.5 +///
1.6 +/// FIR low-pass (anti-alias) filter with filter coefficient design routine and
1.7 +/// MMX optimization.
1.8 +///
1.9 +/// Anti-alias filter is used to prevent folding of high frequencies when
1.10 +/// transposing the sample rate with interpolation.
1.11 +///
1.12 +/// Author : Copyright (c) Olli Parviainen
1.13 +/// Author e-mail : oparviai 'at' iki.fi
1.14 +/// SoundTouch WWW: http://www.surina.net/soundtouch
1.15 +///
1.16 +////////////////////////////////////////////////////////////////////////////////
1.17 +//
1.18 +// Last changed : $Date: 2014-01-05 15:40:22 -0600 (Sun, 05 Jan 2014) $
1.19 +// File revision : $Revision: 4 $
1.20 +//
1.21 +// $Id: AAFilter.cpp 177 2014-01-05 21:40:22Z oparviai $
1.22 +//
1.23 +////////////////////////////////////////////////////////////////////////////////
1.24 +//
1.25 +// License :
1.26 +//
1.27 +// SoundTouch audio processing library
1.28 +// Copyright (c) Olli Parviainen
1.29 +//
1.30 +// This library is free software; you can redistribute it and/or
1.31 +// modify it under the terms of the GNU Lesser General Public
1.32 +// License as published by the Free Software Foundation; either
1.33 +// version 2.1 of the License, or (at your option) any later version.
1.34 +//
1.35 +// This library is distributed in the hope that it will be useful,
1.36 +// but WITHOUT ANY WARRANTY; without even the implied warranty of
1.37 +// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
1.38 +// Lesser General Public License for more details.
1.39 +//
1.40 +// You should have received a copy of the GNU Lesser General Public
1.41 +// License along with this library; if not, write to the Free Software
1.42 +// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
1.43 +//
1.44 +////////////////////////////////////////////////////////////////////////////////
1.45 +
1.46 +#include <memory.h>
1.47 +#include <assert.h>
1.48 +#include <math.h>
1.49 +#include <stdlib.h>
1.50 +#include "AAFilter.h"
1.51 +#include "FIRFilter.h"
1.52 +
1.53 +using namespace soundtouch;
1.54 +
1.55 +#define PI 3.141592655357989
1.56 +#define TWOPI (2 * PI)
1.57 +
1.58 +// define this to save AA filter coefficients to a file
1.59 +// #define _DEBUG_SAVE_AAFILTER_COEFFICIENTS 1
1.60 +
1.61 +#ifdef _DEBUG_SAVE_AAFILTER_COEFFICIENTS
1.62 + #include <stdio.h>
1.63 +
1.64 + static void _DEBUG_SAVE_AAFIR_COEFFS(SAMPLETYPE *coeffs, int len)
1.65 + {
1.66 + FILE *fptr = fopen("aa_filter_coeffs.txt", "wt");
1.67 + if (fptr == NULL) return;
1.68 +
1.69 + for (int i = 0; i < len; i ++)
1.70 + {
1.71 + double temp = coeffs[i];
1.72 + fprintf(fptr, "%lf\n", temp);
1.73 + }
1.74 + fclose(fptr);
1.75 + }
1.76 +
1.77 +#else
1.78 + #define _DEBUG_SAVE_AAFIR_COEFFS(x, y)
1.79 +#endif
1.80 +
1.81 +
1.82 +/*****************************************************************************
1.83 + *
1.84 + * Implementation of the class 'AAFilter'
1.85 + *
1.86 + *****************************************************************************/
1.87 +
1.88 +AAFilter::AAFilter(uint len)
1.89 +{
1.90 + pFIR = FIRFilter::newInstance();
1.91 + cutoffFreq = 0.5;
1.92 + setLength(len);
1.93 +}
1.94 +
1.95 +
1.96 +
1.97 +AAFilter::~AAFilter()
1.98 +{
1.99 + delete pFIR;
1.100 +}
1.101 +
1.102 +
1.103 +
1.104 +// Sets new anti-alias filter cut-off edge frequency, scaled to
1.105 +// sampling frequency (nyquist frequency = 0.5).
1.106 +// The filter will cut frequencies higher than the given frequency.
1.107 +void AAFilter::setCutoffFreq(double newCutoffFreq)
1.108 +{
1.109 + cutoffFreq = newCutoffFreq;
1.110 + calculateCoeffs();
1.111 +}
1.112 +
1.113 +
1.114 +
1.115 +// Sets number of FIR filter taps
1.116 +void AAFilter::setLength(uint newLength)
1.117 +{
1.118 + length = newLength;
1.119 + calculateCoeffs();
1.120 +}
1.121 +
1.122 +
1.123 +
1.124 +// Calculates coefficients for a low-pass FIR filter using Hamming window
1.125 +void AAFilter::calculateCoeffs()
1.126 +{
1.127 + uint i;
1.128 + double cntTemp, temp, tempCoeff,h, w;
1.129 + double wc;
1.130 + double scaleCoeff, sum;
1.131 + double *work;
1.132 + SAMPLETYPE *coeffs;
1.133 +
1.134 + assert(length >= 2);
1.135 + assert(length % 4 == 0);
1.136 + assert(cutoffFreq >= 0);
1.137 + assert(cutoffFreq <= 0.5);
1.138 +
1.139 + work = new double[length];
1.140 + coeffs = new SAMPLETYPE[length];
1.141 +
1.142 + wc = 2.0 * PI * cutoffFreq;
1.143 + tempCoeff = TWOPI / (double)length;
1.144 +
1.145 + sum = 0;
1.146 + for (i = 0; i < length; i ++)
1.147 + {
1.148 + cntTemp = (double)i - (double)(length / 2);
1.149 +
1.150 + temp = cntTemp * wc;
1.151 + if (temp != 0)
1.152 + {
1.153 + h = sin(temp) / temp; // sinc function
1.154 + }
1.155 + else
1.156 + {
1.157 + h = 1.0;
1.158 + }
1.159 + w = 0.54 + 0.46 * cos(tempCoeff * cntTemp); // hamming window
1.160 +
1.161 + temp = w * h;
1.162 + work[i] = temp;
1.163 +
1.164 + // calc net sum of coefficients
1.165 + sum += temp;
1.166 + }
1.167 +
1.168 + // ensure the sum of coefficients is larger than zero
1.169 + assert(sum > 0);
1.170 +
1.171 + // ensure we've really designed a lowpass filter...
1.172 + assert(work[length/2] > 0);
1.173 + assert(work[length/2 + 1] > -1e-6);
1.174 + assert(work[length/2 - 1] > -1e-6);
1.175 +
1.176 + // Calculate a scaling coefficient in such a way that the result can be
1.177 + // divided by 16384
1.178 + scaleCoeff = 16384.0f / sum;
1.179 +
1.180 + for (i = 0; i < length; i ++)
1.181 + {
1.182 + temp = work[i] * scaleCoeff;
1.183 +//#if SOUNDTOUCH_INTEGER_SAMPLES
1.184 + // scale & round to nearest integer
1.185 + temp += (temp >= 0) ? 0.5 : -0.5;
1.186 + // ensure no overfloods
1.187 + assert(temp >= -32768 && temp <= 32767);
1.188 +//#endif
1.189 + coeffs[i] = (SAMPLETYPE)temp;
1.190 + }
1.191 +
1.192 + // Set coefficients. Use divide factor 14 => divide result by 2^14 = 16384
1.193 + pFIR->setCoefficients(coeffs, length, 14);
1.194 +
1.195 + _DEBUG_SAVE_AAFIR_COEFFS(coeffs, length);
1.196 +
1.197 + delete[] work;
1.198 + delete[] coeffs;
1.199 +}
1.200 +
1.201 +
1.202 +// Applies the filter to the given sequence of samples.
1.203 +// Note : The amount of outputted samples is by value of 'filter length'
1.204 +// smaller than the amount of input samples.
1.205 +uint AAFilter::evaluate(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples, uint numChannels) const
1.206 +{
1.207 + return pFIR->evaluate(dest, src, numSamples, numChannels);
1.208 +}
1.209 +
1.210 +
1.211 +/// Applies the filter to the given src & dest pipes, so that processed amount of
1.212 +/// samples get removed from src, and produced amount added to dest
1.213 +/// Note : The amount of outputted samples is by value of 'filter length'
1.214 +/// smaller than the amount of input samples.
1.215 +uint AAFilter::evaluate(FIFOSampleBuffer &dest, FIFOSampleBuffer &src) const
1.216 +{
1.217 + SAMPLETYPE *pdest;
1.218 + const SAMPLETYPE *psrc;
1.219 + uint numSrcSamples;
1.220 + uint result;
1.221 + int numChannels = src.getChannels();
1.222 +
1.223 + assert(numChannels == dest.getChannels());
1.224 +
1.225 + numSrcSamples = src.numSamples();
1.226 + psrc = src.ptrBegin();
1.227 + pdest = dest.ptrEnd(numSrcSamples);
1.228 + result = pFIR->evaluate(pdest, psrc, numSrcSamples, numChannels);
1.229 + src.receiveSamples(result);
1.230 + dest.putSamples(result);
1.231 +
1.232 + return result;
1.233 +}
1.234 +
1.235 +
1.236 +uint AAFilter::getLength() const
1.237 +{
1.238 + return pFIR->getLength();
1.239 +}
