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update soundtouch to svn revision 173
This commit is contained in:
368
Externals/soundtouch/AAFilter.cpp
vendored
368
Externals/soundtouch/AAFilter.cpp
vendored
@ -1,184 +1,184 @@
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////////////////////////////////////////////////////////////////////////////////
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///
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/// FIR low-pass (anti-alias) filter with filter coefficient design routine and
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/// MMX optimization.
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///
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/// Anti-alias filter is used to prevent folding of high frequencies when
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/// transposing the sample rate with interpolation.
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///
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/// Author : Copyright (c) Olli Parviainen
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/// Author e-mail : oparviai 'at' iki.fi
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/// SoundTouch WWW: http://www.surina.net/soundtouch
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///
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////////////////////////////////////////////////////////////////////////////////
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//
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// Last changed : $Date: 2009-01-11 13:34:24 +0200 (Sun, 11 Jan 2009) $
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// File revision : $Revision: 4 $
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//
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// $Id: AAFilter.cpp 45 2009-01-11 11:34:24Z oparviai $
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//
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////////////////////////////////////////////////////////////////////////////////
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//
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// License :
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//
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// SoundTouch audio processing library
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// Copyright (c) Olli Parviainen
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//
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// This library is free software; you can redistribute it and/or
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// modify it under the terms of the GNU Lesser General Public
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// License as published by the Free Software Foundation; either
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// version 2.1 of the License, or (at your option) any later version.
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//
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// This library is distributed in the hope that it will be useful,
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// but WITHOUT ANY WARRANTY; without even the implied warranty of
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// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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// Lesser General Public License for more details.
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//
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// You should have received a copy of the GNU Lesser General Public
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// License along with this library; if not, write to the Free Software
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// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
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//
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////////////////////////////////////////////////////////////////////////////////
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#include <memory.h>
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#include <assert.h>
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#include <math.h>
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#include <stdlib.h>
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#include "AAFilter.h"
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#include "FIRFilter.h"
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using namespace soundtouch;
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#define PI 3.141592655357989
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#define TWOPI (2 * PI)
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/*****************************************************************************
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*
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* Implementation of the class 'AAFilter'
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*
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*****************************************************************************/
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AAFilter::AAFilter(uint len)
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{
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pFIR = FIRFilter::newInstance();
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cutoffFreq = 0.5;
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setLength(len);
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}
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AAFilter::~AAFilter()
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{
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delete pFIR;
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}
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// Sets new anti-alias filter cut-off edge frequency, scaled to
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// sampling frequency (nyquist frequency = 0.5).
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// The filter will cut frequencies higher than the given frequency.
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void AAFilter::setCutoffFreq(double newCutoffFreq)
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{
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cutoffFreq = newCutoffFreq;
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calculateCoeffs();
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}
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// Sets number of FIR filter taps
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void AAFilter::setLength(uint newLength)
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{
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length = newLength;
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calculateCoeffs();
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}
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// Calculates coefficients for a low-pass FIR filter using Hamming window
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void AAFilter::calculateCoeffs()
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{
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uint i;
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double cntTemp, temp, tempCoeff,h, w;
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double fc2, wc;
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double scaleCoeff, sum;
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double *work;
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SAMPLETYPE *coeffs;
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assert(length >= 2);
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assert(length % 4 == 0);
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assert(cutoffFreq >= 0);
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assert(cutoffFreq <= 0.5);
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work = new double[length];
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coeffs = new SAMPLETYPE[length];
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fc2 = 2.0 * cutoffFreq;
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wc = PI * fc2;
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tempCoeff = TWOPI / (double)length;
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sum = 0;
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for (i = 0; i < length; i ++)
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{
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cntTemp = (double)i - (double)(length / 2);
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temp = cntTemp * wc;
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if (temp != 0)
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{
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h = fc2 * sin(temp) / temp; // sinc function
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}
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else
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{
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h = 1.0;
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}
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w = 0.54 + 0.46 * cos(tempCoeff * cntTemp); // hamming window
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temp = w * h;
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work[i] = temp;
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// calc net sum of coefficients
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sum += temp;
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}
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// ensure the sum of coefficients is larger than zero
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assert(sum > 0);
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// ensure we've really designed a lowpass filter...
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assert(work[length/2] > 0);
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assert(work[length/2 + 1] > -1e-6);
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assert(work[length/2 - 1] > -1e-6);
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// Calculate a scaling coefficient in such a way that the result can be
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// divided by 16384
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scaleCoeff = 16384.0f / sum;
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for (i = 0; i < length; i ++)
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{
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// scale & round to nearest integer
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temp = work[i] * scaleCoeff;
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temp += (temp >= 0) ? 0.5 : -0.5;
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// ensure no overfloods
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assert(temp >= -32768 && temp <= 32767);
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coeffs[i] = (SAMPLETYPE)temp;
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}
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// Set coefficients. Use divide factor 14 => divide result by 2^14 = 16384
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pFIR->setCoefficients(coeffs, length, 14);
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delete[] work;
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delete[] coeffs;
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}
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// Applies the filter to the given sequence of samples.
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// Note : The amount of outputted samples is by value of 'filter length'
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// smaller than the amount of input samples.
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uint AAFilter::evaluate(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples, uint numChannels) const
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{
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return pFIR->evaluate(dest, src, numSamples, numChannels);
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}
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uint AAFilter::getLength() const
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{
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return pFIR->getLength();
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}
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////////////////////////////////////////////////////////////////////////////////
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///
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/// FIR low-pass (anti-alias) filter with filter coefficient design routine and
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/// MMX optimization.
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///
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/// Anti-alias filter is used to prevent folding of high frequencies when
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/// transposing the sample rate with interpolation.
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///
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/// Author : Copyright (c) Olli Parviainen
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/// Author e-mail : oparviai 'at' iki.fi
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/// SoundTouch WWW: http://www.surina.net/soundtouch
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///
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////////////////////////////////////////////////////////////////////////////////
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//
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// Last changed : $Date: 2009-01-11 11:34:24 +0000 (Sun, 11 Jan 2009) $
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// File revision : $Revision: 4 $
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//
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// $Id: AAFilter.cpp 45 2009-01-11 11:34:24Z oparviai $
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//
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////////////////////////////////////////////////////////////////////////////////
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//
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// License :
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//
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// SoundTouch audio processing library
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// Copyright (c) Olli Parviainen
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//
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// This library is free software; you can redistribute it and/or
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// modify it under the terms of the GNU Lesser General Public
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// License as published by the Free Software Foundation; either
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// version 2.1 of the License, or (at your option) any later version.
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//
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// This library is distributed in the hope that it will be useful,
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// but WITHOUT ANY WARRANTY; without even the implied warranty of
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// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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// Lesser General Public License for more details.
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//
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// You should have received a copy of the GNU Lesser General Public
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// License along with this library; if not, write to the Free Software
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// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
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//
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////////////////////////////////////////////////////////////////////////////////
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#include <memory.h>
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#include <assert.h>
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#include <math.h>
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#include <stdlib.h>
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#include "AAFilter.h"
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#include "FIRFilter.h"
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using namespace soundtouch;
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#define PI 3.141592655357989
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#define TWOPI (2 * PI)
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/*****************************************************************************
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*
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* Implementation of the class 'AAFilter'
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*
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*****************************************************************************/
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AAFilter::AAFilter(uint len)
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{
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pFIR = FIRFilter::newInstance();
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cutoffFreq = 0.5;
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setLength(len);
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}
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AAFilter::~AAFilter()
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{
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delete pFIR;
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}
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// Sets new anti-alias filter cut-off edge frequency, scaled to
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// sampling frequency (nyquist frequency = 0.5).
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// The filter will cut frequencies higher than the given frequency.
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void AAFilter::setCutoffFreq(double newCutoffFreq)
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{
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cutoffFreq = newCutoffFreq;
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calculateCoeffs();
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}
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// Sets number of FIR filter taps
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void AAFilter::setLength(uint newLength)
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{
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length = newLength;
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calculateCoeffs();
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}
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// Calculates coefficients for a low-pass FIR filter using Hamming window
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void AAFilter::calculateCoeffs()
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{
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uint i;
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double cntTemp, temp, tempCoeff,h, w;
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double fc2, wc;
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double scaleCoeff, sum;
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double *work;
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SAMPLETYPE *coeffs;
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assert(length >= 2);
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assert(length % 4 == 0);
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assert(cutoffFreq >= 0);
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assert(cutoffFreq <= 0.5);
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work = new double[length];
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coeffs = new SAMPLETYPE[length];
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fc2 = 2.0 * cutoffFreq;
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wc = PI * fc2;
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tempCoeff = TWOPI / (double)length;
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sum = 0;
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for (i = 0; i < length; i ++)
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{
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cntTemp = (double)i - (double)(length / 2);
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temp = cntTemp * wc;
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if (temp != 0)
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{
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h = fc2 * sin(temp) / temp; // sinc function
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}
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else
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{
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h = 1.0;
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}
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w = 0.54 + 0.46 * cos(tempCoeff * cntTemp); // hamming window
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temp = w * h;
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work[i] = temp;
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// calc net sum of coefficients
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sum += temp;
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}
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// ensure the sum of coefficients is larger than zero
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assert(sum > 0);
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// ensure we've really designed a lowpass filter...
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assert(work[length/2] > 0);
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assert(work[length/2 + 1] > -1e-6);
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assert(work[length/2 - 1] > -1e-6);
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// Calculate a scaling coefficient in such a way that the result can be
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// divided by 16384
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scaleCoeff = 16384.0f / sum;
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for (i = 0; i < length; i ++)
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{
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// scale & round to nearest integer
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temp = work[i] * scaleCoeff;
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temp += (temp >= 0) ? 0.5 : -0.5;
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// ensure no overfloods
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assert(temp >= -32768 && temp <= 32767);
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coeffs[i] = (SAMPLETYPE)temp;
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}
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// Set coefficients. Use divide factor 14 => divide result by 2^14 = 16384
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pFIR->setCoefficients(coeffs, length, 14);
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delete[] work;
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delete[] coeffs;
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}
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// Applies the filter to the given sequence of samples.
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// Note : The amount of outputted samples is by value of 'filter length'
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// smaller than the amount of input samples.
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uint AAFilter::evaluate(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples, uint numChannels) const
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{
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return pFIR->evaluate(dest, src, numSamples, numChannels);
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}
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uint AAFilter::getLength() const
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{
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return pFIR->getLength();
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}
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