The Tor Browser: media/libsoundtouch/src/AAFilter.cpp@b8a032363ba2 (annotated)

media/libsoundtouch/src/AAFilter.cpp@b8a032363ba2 (annotated)

media/libsoundtouch/src/AAFilter.cpp

Thu, 22 Jan 2015 13:21:57 +0100

author: Michael Schloh von Bennewitz <michael@schloh.com>
date: Thu, 22 Jan 2015 13:21:57 +0100
branch: TOR_BUG_9701
changeset 15: b8a032363ba2
permissions: -rw-r--r--

Incorporate requested changes from Mozilla in review:
https://bugzilla.mozilla.org/show_bug.cgi?id=1123480#c6

 ////////////////////////////////////////////////////////////////////////////////
 ///
 /// 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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media/libsoundtouch/src/AAFilter.cpp@b8a032363ba2 (annotated)

media/libsoundtouch/src/AAFilter.cpp