The Tor Browser: comparison media/libsoundtouch/src/AAFilter.cpp

--1:000000000000
+:f8c813a09aa9
+////////////////////////////////////////////////////////////////////////////////
+///
+/// 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: 2014-01-05 15:40:22 -0600 (Sun, 05 Jan 2014) $
+// File revision : $Revision: 4 $
+//
+// $Id: AAFilter.cpp 177 2014-01-05 21:40:22Z 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)
+// 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;
+//#if SOUNDTOUCH_INTEGER_SAMPLES
+// scale & round to nearest integer
+temp += (temp >= 0) ? 0.5 : -0.5;
+// ensure no overfloods
+assert(temp >= -32768 && temp <= 32767);
+//#endif
+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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comparison: media/libsoundtouch/src/AAFilter.cpp

media/libsoundtouch/src/AAFilter.cpp