/ 1100308561 0 0 0 5444 `
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FIFOSampleBuffer.o/
RateTransposer.o/
cpu_detect_x86_gcc.o/
AAFilter.o/ 1100308513 1001 10 100644 5044 `
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d m 3 | > P x M \ FIRFilter.o/ 1100308521 1001 10 100644 7348 `
ELF 4 4 ( .shstrtab .gnu.linkonce.r._ZTV9FIRFilter .gnu.linkonce.r._ZTI9FIRFilter .gnu.linkonce.r._ZTS9FIRFilter .text .rodata .gcc_except_table .eh_frame .symtab .strtab .rela.gnu.linkonce.r._ZTV9FIRFilter .rela.gnu.linkonce.r._ZTI9FIRFilter .rela.text .rela.eh_frame .comment 9FIRFilter ` " " " " " ` " " " " " 㿐 `
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length != 0 FIRFilter.cpp numChannels == 1 || numChannels == 2 lengthDiv8 * 8 == length length > 0 Don't use 'new FIRFilter', use 'newInstance' member instead! newLength > 0 length == newLength FIR filter length not divisible by 8 A A A ( d \ 4 [ @ 4 \ t 8 , \ zPL |P
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T H l l \ /0 1100308528 1001 10 100644 6576 `
ELF h 4 ( .shstrtab .gnu.linkonce.r._ZTV16FIFOSampleBuffer .gnu.linkonce.r._ZTI16FIFOSampleBuffer .gnu.linkonce.r._ZTI14FIFOSamplePipe .gnu.linkonce.r._ZTS14FIFOSamplePipe .gnu.linkonce.r._ZTS16FIFOSampleBuffer .gnu.linkonce.r._ZTV14FIFOSamplePipe .text .rodata .gcc_except_table .eh_frame .symtab .strtab .rela.gnu.linkonce.r._ZTV16FIFOSampleBuffer .rela.gnu.linkonce.r._ZTI16FIFOSampleBuffer .rela.gnu.linkonce.r._ZTI14FIFOSamplePipe .rela.gnu.linkonce.r._ZTV14FIFOSamplePipe .rela.text .rela.eh_frame .comment 14FIFOSamplePipe 16FIFOSampleBuffer " ` " " " " " " " ` " " " " " " ` @ ` ` @ ` 㿐 @ @ & & 㿐 ` @ @ @ @ & 㿐 @ 㿐 @ @ 㿐 @8 @ @ @ ` @ @ @ " @ " " `?" " x @ * k @ @ ]& @ op & @ @ @ @ & & =& @ & & @ @ @ ? @2 ? # ! ` @ @ @ ? ?@ ? @ 0@ ? 4 @ 0 㿐 @ @ @ 㿐 @ @ @
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GCC: (GNU) 3.4.1 4 0 ( 2 X Y d ~ p $ R
! , ) l U | $ ` ` \ mmx_gcc.o/ 1100308533 1001 10 100644 804 `
ELF 4 4 ( .shstrtab .text .symtab .strtab .comment
6 K _ r mmx_gcc.cpp pthread_mutex_lock pthread_mutex_trylock pthread_mutex_unlock pthread_setspecific pthread_key_delete pthread_key_create pthread_once pthread_getspecific pthread_create as: Sun WorkShop 6 update 2 Compiler Common 6.2 Solaris_9_CBE 2001/04/02
GCC: (GNU) 3.4.1 4 * ^ ` ! \ /20 1100308538 1001 10 100644 13752 `
ELF 0@ 4 ( # .shstrtab .gnu.linkonce.r._ZTV14RateTransposer .gnu.linkonce.r._ZTI14RateTransposer .gnu.linkonce.r._ZTI13FIFOProcessor .gnu.linkonce.r._ZTI14FIFOSamplePipe .gnu.linkonce.r._ZTS14FIFOSamplePipe .gnu.linkonce.r._ZTS13FIFOProcessor .gnu.linkonce.r._ZTS14RateTransposer .gnu.linkonce.r._ZTV13FIFOProcessor .gnu.linkonce.r._ZTV14FIFOSamplePipe .gnu.linkonce.t._ZN13FIFOProcessorD1Ev .gnu.linkonce.t._ZN13FIFOProcessorD0Ev .gnu.linkonce.t._ZNK13FIFOProcessor8ptrBeginEv .gnu.linkonce.t._ZN13FIFOProcessor14receiveSamplesEPsj .gnu.linkonce.t._ZN13FIFOProcessor14receiveSamplesEj .gnu.linkonce.t._ZNK13FIFOProcessor10numSamplesEv .gnu.linkonce.t._ZNK13FIFOProcessor7isEmptyEv .text .gcc_except_table .rodata .eh_frame .symtab .strtab .rela.gnu.linkonce.r._ZTV14RateTransposer .rela.gnu.linkonce.r._ZTI14RateTransposer .rela.gnu.linkonce.r._ZTI13FIFOProcessor .rela.gnu.linkonce.r._ZTI14FIFOSamplePipe .rela.gnu.linkonce.r._ZTV13FIFOProcessor .rela.gnu.linkonce.r._ZTV14FIFOSamplePipe .rela.gnu.linkonce.t._ZN13FIFOProcessorD1Ev .rela.gnu.linkonce.t._ZN13FIFOProcessorD0Ev .rela.text .rela.eh_frame .comment 14FIFOSamplePipe 13FIFOProcessor 14RateTransposer ` " ` " ` 㿐 @ @ 㿐 ` @ 㿐 ` @ 㿐 `@ 㿐 `@ 2 " 2 㿐 & @ @ @ T@ 8@ @ 8@ $ @ @ T00 $ 㿐 & @ @ @ T@ 8@ @ 8@ $ @ @ T00 $ 㿐 & @ @ @ T@ 8@ $ @ @ 8@ $ @ @ T00 " p p 㿐 % @ $@ U V @ @: 2@ :`6@ ? ) @ ? @
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&x & 0 * & 5 . D / \ SoundTouch.o/ 1100308547 1001 10 100644 16760 `
ELF ; 4 ( % .shstrtab .gnu.linkonce.r._ZTV10SoundTouch .gnu.linkonce.r._ZTI10SoundTouch .gnu.linkonce.r._ZTI13FIFOProcessor .gnu.linkonce.r._ZTI14FIFOSamplePipe .gnu.linkonce.r._ZTS14FIFOSamplePipe .gnu.linkonce.r._ZTS13FIFOProcessor .gnu.linkonce.r._ZTS10SoundTouch .gnu.linkonce.r._ZTV13FIFOProcessor .gnu.linkonce.r._ZTV14FIFOSamplePipe .gnu.linkonce.t._ZN13FIFOProcessorD1Ev .gnu.linkonce.t._ZN13FIFOProcessorD0Ev .gnu.linkonce.t._ZNK13FIFOProcessor8ptrBeginEv .gnu.linkonce.t._ZN13FIFOProcessor14receiveSamplesEPsj .gnu.linkonce.t._ZN13FIFOProcessor14receiveSamplesEj .gnu.linkonce.t._ZNK13FIFOProcessor10numSamplesEv .gnu.linkonce.t._ZNK13FIFOProcessor7isEmptyEv .rodata .text .gcc_except_table .data .eh_frame .symtab .strtab .rela.gnu.linkonce.r._ZTV10SoundTouch .rela.gnu.linkonce.r._ZTI10SoundTouch .rela.gnu.linkonce.r._ZTI13FIFOProcessor .rela.gnu.linkonce.r._ZTI14FIFOSamplePipe .rela.gnu.linkonce.r._ZTV13FIFOProcessor .rela.gnu.linkonce.r._ZTV14FIFOSamplePipe .rela.gnu.linkonce.t._ZN13FIFOProcessorD1Ev .rela.gnu.linkonce.t._ZN13FIFOProcessorD0Ev .rela.text .rela.data .rela.eh_frame .comment 14FIFOSamplePipe 13FIFOProcessor 10SoundTouch ` " ` " ` 㿐 @ @ 㿐 ` @ 㿐 ` @ 㿐 `@ 㿐 `@ SoundTouch Version: %s
1.2.1 output == pTDStretch SoundTouch.cpp output == pRateTransposer ? ?1rA@ ?1rA@ ?1r<#
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// these are declared in 'TDStretch.cpp'
extern int scanOffsets[4][24];
// Calculates cross correlation of two buffers
_inline float TDStretch3DNow::calculateCrossCorrelation(const float *pV1, const float *pV2) const
{
uint overlapLengthLocal = overlapLength;
float corr;
// Calculates the cross-correlation value between 'pV1' and 'pV2' vectors
/*
c-pseudocode:
corr = 0;
for (i = 0; i < overlapLength / 4; i ++)
{
corr += pV1[0] * pV2[0];
pV1[1] * pV2[1];
pV1[2] * pV2[2];
pV1[3] * pV2[3];
pV1[4] * pV2[4];
pV1[5] * pV2[5];
pV1[6] * pV2[6];
pV1[7] * pV2[7];
pV1 += 8;
pV2 += 8;
}
*/
_asm
{
// give prefetch hints to CPU of what data are to be needed soonish.
// give more aggressive hints on pV1 as that changes more between different calls
// while pV2 stays the same.
prefetch [pV1]
prefetch [pV2]
prefetch [pV1 + 32]
mov eax, dword ptr pV2
mov ebx, dword ptr pV1
pxor mm0, mm0
mov ecx, overlapLengthLocal
shr ecx, 2 // div by four
loop1:
movq mm1, [eax]
prefetch [eax + 32] // give a prefetch hint to CPU what data are to be needed soonish
pfmul mm1, [ebx]
prefetch [ebx + 64] // give a prefetch hint to CPU what data are to be needed soonish
movq mm2, [eax + 8]
pfadd mm0, mm1
pfmul mm2, [ebx + 8]
movq mm3, [eax + 16]
pfadd mm0, mm2
pfmul mm3, [ebx + 16]
movq mm4, [eax + 24]
pfadd mm0, mm3
pfmul mm4, [ebx + 24]
add eax, 32
pfadd mm0, mm4
add ebx, 32
dec ecx
jnz loop1
// add halfs of mm0 together and return the result.
// note: mm1 is used as a dummy parameter only, we actually don't care about it's value
pfacc mm0, mm1
movd corr, mm0
femms
}
return corr;
}
// 3DNow!-optimized version of the function "seekBestOverlapPositionStereo"
uint TDStretch3DNow::seekBestOverlapPositionStereo(const float *other)
{
uint bestoffs;
float bestcorr, corr;
uint corrOffset, i;
// Slopes the amplitude of the "midBuffer" samples
slopeReferenceSamplesStereo();
bestcorr = INT_MIN;
bestoffs = 0;
corrOffset = 0;
// Scans for the best correlation value by testing each possible position
// over the permitted range.
for (i = 0; i < seekLength; i ++)
{
// Calculates the cross-correlation value for the mixing position
// corresponding to "i"
corr = calculateCrossCorrelation(other + 2 * i, pRefMidBuffer);
// Checks for the highest correlation value
if (corr > bestcorr)
{
bestoffs = i;
bestcorr = corr;
}
}
return bestoffs;
}
// 3DNow!-optimized version of the function "seekBestOverlapPositionStereoQuick"
uint TDStretch3DNow::seekBestOverlapPositionStereoQuick(const float *other)
{
float *local_refMidBuffer = pRefMidBuffer;
uint bestpos, scancount, i;
float bestcorr,corr;
uint corrPos, tempPos;
// Slopes the amplitude of the "midBuffer" samples
slopeReferenceSamplesStereo();
bestcorr = INT_MIN;
bestpos = scanOffsets[0][0];
corrPos = 0;
tempPos = 0;
// Scans for the best correlation value using four-pass hierarchical search.
//
// The look-up table "scans" has hierarchical position adjusting steps.
// In first pass the routine searhes for the highest correlation with
// relatively coarse steps, then rescans the neighbourhood of the highest
// correlation with better resolution and so on.
for (scancount = 0;scancount < 4; scancount ++)
{
i = 0;
while (scanOffsets[scancount][i])
{
tempPos = corrPos + scanOffsets[scancount][i];
if (tempPos >= seekLength) break;
// Calculates correlation value for the mixing position corresponding
// to "tempPos"
corr = calculateCrossCorrelation(other + 2 * tempPos, pRefMidBuffer);
// Checks for the highest correlation value
if (corr > bestcorr)
{
bestpos = tempPos;
bestcorr = corr;
}
i ++;
}
corrPos = bestpos;
}
return bestpos;
}
// 3DNow!-optimized version of the function overlapStereo
void TDStretch3DNow::overlapStereo(float *output, const float *input) const
{
int i;
uint cnt2;
float fTemp;
float fScale;
float fi;
fScale = 1.0f / (float)overlapLength;
for (i = 0; i < (int)overlapLength ; i ++)
{
fTemp = (float)(overlapLength - i) * fScale;
fi = (float)i * fScale;
cnt2 = 2 * i;
output[cnt2 + 0] = input[cnt2 + 0] * fi + pMidBuffer[cnt2 + 0] * fTemp;
output[cnt2 + 1] = input[cnt2 + 1] * fi + pMidBuffer[cnt2 + 1] * fTemp;
}
}
//////////////////////////////////////////////////////////////////////////////
//
// implementation of 3DNow! optimized functions of class 'FIRFilter'
//
//////////////////////////////////////////////////////////////////////////////
#include "FIRFilter.h"
FIRFilter3DNow::FIRFilter3DNow() : FIRFilter()
{
filterCoeffsUnalign = NULL;
}
FIRFilter3DNow::~FIRFilter3DNow()
{
delete[] filterCoeffsUnalign;
}
// (overloaded) Calculates filter coefficients for 3DNow! routine
void FIRFilter3DNow::setCoefficients(const float *coeffs, uint newLength, uint uResultDivFactor)
{
uint i;
float fDivider;
FIRFilter::setCoefficients(coeffs, newLength, uResultDivFactor);
// Scale the filter coefficients so that it won't be necessary to scale the filtering result
// also rearrange coefficients suitably for 3DNow!
// Ensure that filter coeffs array is aligned to 16-byte boundary
filterCoeffsUnalign = new float[2 * newLength + 4];
filterCoeffsAlign = (float *)(((uint)filterCoeffsUnalign + 15) & -16);
fDivider = (float)resultDivider;
// rearrange the filter coefficients for mmx routines
for (i = 0; i < newLength; i ++)
{
filterCoeffsAlign[2 * i + 0] =
filterCoeffsAlign[2 * i + 1] = coeffs[i + 0] / fDivider;
}
}
// 3DNow!-optimized version of the filter routine for stereo sound
uint FIRFilter3DNow::evaluateFilterStereo(float *dest, const float *src, const uint numSamples) const
{
float *filterCoeffsLocal = filterCoeffsAlign;
uint count = (numSamples - length) & -2;
uint lengthLocal = length / 4;
assert(length != 0);
assert(count % 2 == 0);
/* original code:
double suml1, suml2;
double sumr1, sumr2;
uint i, j;
for (j = 0; j < count; j += 2)
{
const float *ptr;
suml1 = sumr1 = 0.0;
suml2 = sumr2 = 0.0;
ptr = src;
filterCoeffsLocal = filterCoeffs;
for (i = 0; i < lengthLocal; i ++)
{
// unroll loop for efficiency.
suml1 += ptr[0] * filterCoeffsLocal[0] +
ptr[2] * filterCoeffsLocal[2] +
ptr[4] * filterCoeffsLocal[4] +
ptr[6] * filterCoeffsLocal[6];
sumr1 += ptr[1] * filterCoeffsLocal[1] +
ptr[3] * filterCoeffsLocal[3] +
ptr[5] * filterCoeffsLocal[5] +
ptr[7] * filterCoeffsLocal[7];
suml2 += ptr[8] * filterCoeffsLocal[0] +
ptr[10] * filterCoeffsLocal[2] +
ptr[12] * filterCoeffsLocal[4] +
ptr[14] * filterCoeffsLocal[6];
sumr2 += ptr[9] * filterCoeffsLocal[1] +
ptr[11] * filterCoeffsLocal[3] +
ptr[13] * filterCoeffsLocal[5] +
ptr[15] * filterCoeffsLocal[7];
ptr += 16;
filterCoeffsLocal += 8;
}
dest[0] = (float)suml1;
dest[1] = (float)sumr1;
dest[2] = (float)suml2;
dest[3] = (float)sumr2;
src += 4;
dest += 4;
}
*/
_asm
{
mov eax, dword ptr dest
mov ebx, dword ptr src
mov edx, count
shr edx, 1
loop1:
// "outer loop" : during each round 2*2 output samples are calculated
prefetch [ebx] // give a prefetch hint to CPU what data are to be needed soonish
prefetch [filterCoeffsLocal] // give a prefetch hint to CPU what data are to be needed soonish
mov esi, ebx
mov edi, filterCoeffsLocal
pxor mm0, mm0
pxor mm1, mm1
mov ecx, lengthLocal
loop2:
// "inner loop" : during each round four FIR filter taps are evaluated for 2*2 output samples
movq mm2, [edi]
movq mm3, mm2
prefetch [edi + 32] // give a prefetch hint to CPU what data are to be needed soonish
pfmul mm2, [esi]
prefetch [esi + 32] // give a prefetch hint to CPU what data are to be needed soonish
pfmul mm3, [esi + 8]
movq mm4, [edi + 8]
movq mm5, mm4
pfadd mm0, mm2
pfmul mm4, [esi + 8]
pfadd mm1, mm3
pfmul mm5, [esi + 16]
movq mm2, [edi + 16]
movq mm6, mm2
pfadd mm0, mm4
pfmul mm2, [esi + 16]
pfadd mm1, mm5
pfmul mm6, [esi + 24]
movq mm3, [edi + 24]
movq mm7, mm3
pfadd mm0, mm2
pfmul mm3, [esi + 24]
pfadd mm1, mm6
pfmul mm7, [esi + 32]
add esi, 32
pfadd mm0, mm3
add edi, 32
pfadd mm1, mm7
dec ecx
jnz loop2
movq [eax], mm0
add ebx, 16
movq [eax + 8], mm1
add eax, 16
dec edx
jnz loop1
femms
}
return count;
}
#endif // ALLOW_3DNOW
0707010001e15b000081b60000000200000002000000013fed755600001395000000200000000000000000000000000000002500000000reloc/source/SoundTouch/AAFilter.cpp /*****************************************************************************
*
* 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 @ iki.fi
* File created : 13-Jan-2002
*
* Last changed : $Date: 2003/12/27 10:00:51 $
* File revision : $Revision: 1.7 $
*
* $Id: AAFilter.cpp,v 1.7 2003/12/27 10:00:51 Olli Exp $
*
* License :
*
* SoundTouch sound 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
#include
#include
#include
#include "AAFilter.h"
#include "FIRFilter.h"
using namespace soundtouch;
#define PI 3.141592655357989
#define TWOPI (2 * PI)
/*****************************************************************************
*
* Implementation of the class 'AAFilter'
*
*****************************************************************************/
AAFilter::AAFilter(const uint length)
{
pFIR = FIRFilter::newInstance();
cutoffFreq = 0.5;
setLength(length);
}
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(const double newCutoffFreq)
{
cutoffFreq = newCutoffFreq;
calculateCoeffs();
}
// Sets number of FIR filter taps
void AAFilter::setLength(const 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 fc2, wc;
double scaleCoeff, sum;
double *work;
SAMPLETYPE *coeffs;
assert(length > 0);
assert(length % 4 == 0);
assert(cutoffFreq >= 0);
assert(cutoffFreq <= 0.5);
work = new double[length];
coeffs = new SAMPLETYPE[length];
fc2 = 2.0 * cutoffFreq;
wc = PI * fc2;
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 = fc2 * 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 ++)
{
// scale & round to nearest integer
temp = work[i] * scaleCoeff;
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);
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);
}
uint AAFilter::getLength() const
{
return pFIR->getLength();
}
0707010001e15c000081b60000000200000002000000013fed755600000b20000000200000000000000000000000000000002300000000reloc/source/SoundTouch/AAFilter.h /*****************************************************************************
*
* Sampled sound tempo changer/time stretch algorithm. Changes the sound tempo
* while maintaining the original pitch by using a time domain WSOLA-like method
* with several performance-increasing tweaks.
*
* 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 @ iki.fi
* File created : 13-Jan-2002
*
* Last changed : $Date: 2003/12/27 10:00:51 $
* File revision : $Revision: 1.7 $
*
* $Id: AAFilter.h,v 1.7 2003/12/27 10:00:51 Olli Exp $
*
* License :
*
* SoundTouch sound 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
*
*****************************************************************************/
#ifndef AAFilter_H
#define AAFilter_H
#include "STTypes.h"
class AAFilter
{
protected:
class FIRFilter *pFIR;
/// Low-pass filter cut-off frequency, negative = invalid
double cutoffFreq;
/// num of filter taps
uint length;
/// Calculate the FIR coefficients realizing the given cutoff-frequency
void calculateCoeffs();
public:
AAFilter(uint length);
~AAFilter();
/// Sets new anti-alias filter cut-off edge frequency, scaled to sampling
/// frequency (nyquist frequency = 0.5). The filter will cut off the
/// frequencies than that.
void setCutoffFreq(double newCutoffFreq);
/// Sets number of FIR filter taps, i.e. ~filter complexity
void setLength(uint newLength);
uint getLength() const;
/// 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 evaluate(soundtouch::SAMPLETYPE *dest,
const soundtouch::SAMPLETYPE *src,
uint numSamples,
uint numChannels) const;
};
#endif
0707010001e15d000081a4000000020000000200000001419560210000000a000000200000000000000000000000000000002400000000reloc/source/SoundTouch/AAFilter.lo timestamp
0707010001e15e000081a400000002000000020000000141956021000013b4000000200000000000000000000000000000002300000000reloc/source/SoundTouch/AAFilter.o ELF $ 4 (
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d m 3 | > P x M \ 0707010001e15f000081b60000000200000002000000013fed755600001f69000000200000000000000000000000000000002d00000000reloc/source/SoundTouch/FIFOSampleBuffer.cpp /*****************************************************************************
*
* A buffer class for temporarily storaging sound samples, operates as a
* first-in-first-out pipe.
*
* Samples are added to the end of the sample buffer with the 'putSamples'
* function, and are received from the beginning of the buffer by calling
* the 'receiveSamples' function. The class automatically removes the
* outputted samples from the buffer, as well as grows the buffer size
* whenever necessary.
*
* Author : Copyright (c) Olli Parviainen
* Author e-mail : oparviai @ iki.fi
* File created : 13-Jan-2002
*
* Last changed : $Date: 2003/12/27 10:00:51 $
* File revision : $Revision: 1.9 $
*
* $Id: FIFOSampleBuffer.cpp,v 1.9 2003/12/27 10:00:51 Olli Exp $
*
* License :
*
* SoundTouch sound 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
#include
#include
#include
#include
#include "FIFOSampleBuffer.h"
using namespace soundtouch;
// Constructor
FIFOSampleBuffer::FIFOSampleBuffer(uint numChannels)
{
sizeInBytes = 0; // reasonable initial value
buffer = NULL; //new SAMPLETYPE[sizeInBytes / sizeof(SAMPLETYPE)];
bufferUnaligned = NULL;
samplesInBuffer = 0;
bufferPos = 0;
channels = numChannels;
}
// destructor
FIFOSampleBuffer::~FIFOSampleBuffer()
{
delete[] bufferUnaligned;
}
// Sets number of channels, 1 = mono, 2 = stereo
void FIFOSampleBuffer::setChannels(const uint numChannels)
{
uint usedBytes;
usedBytes = channels * samplesInBuffer;
channels = numChannels;
samplesInBuffer = usedBytes / channels;
}
// if output location pointer 'bufferPos' isn't zero, 'rewinds' the buffer and
// zeroes this pointer by copying samples from the 'bufferPos' pointer
// location on to the beginning of the buffer.
void FIFOSampleBuffer::rewind()
{
if (bufferPos)
{
memmove(buffer, ptrBegin(), sizeof(SAMPLETYPE) * channels * samplesInBuffer);
bufferPos = 0;
}
}
// Adds 'numSamples' pcs of samples from the 'samples' memory position to
// the sample buffer.
void FIFOSampleBuffer::putSamples(const SAMPLETYPE *samples, uint numSamples)
{
memcpy(ptrEnd(numSamples), samples, sizeof(SAMPLETYPE) * numSamples * channels);
samplesInBuffer += numSamples;
}
// Increases the number of samples in the buffer without copying any actual
// samples.
//
// This function is used to update the number of samples in the sample buffer
// when accessing the buffer directly with 'ptrEnd' function. Please be
// careful though!
void FIFOSampleBuffer::putSamples(uint numSamples)
{
uint req;
req = samplesInBuffer + numSamples;
ensureCapacity(req);
samplesInBuffer += numSamples;
}
// Returns a pointer to the end of the used part of the sample buffer (i.e.
// where the new samples are to be inserted). This function may be used for
// inserting new samples into the sample buffer directly. Please be careful!
//
// Parameter 'slackCapacity' tells the function how much free capacity (in
// terms of samples) there _at least_ should be, in order to the caller to
// succesfully insert all the required samples to the buffer. When necessary,
// the function grows the buffer size to comply with this requirement.
//
// When using this function as means for inserting new samples, also remember
// to increase the sample count afterwards, by calling the
// 'putSamples(numSamples)' function.
SAMPLETYPE *FIFOSampleBuffer::ptrEnd(uint slackCapacity)
{
ensureCapacity(samplesInBuffer + slackCapacity);
return buffer + samplesInBuffer * channels;
}
// Returns a pointer to the beginning of the currently non-outputted samples.
// This function is provided for accessing the output samples directly.
// Please be careful!
//
// When using this function to output samples, also remember to 'remove' the
// outputted samples from the buffer by calling the
// 'receiveSamples(numSamples)' function
SAMPLETYPE *FIFOSampleBuffer::ptrBegin() const
{
return buffer + bufferPos * channels;
}
// Ensures that the buffer has enought capacity, i.e. space for _at least_
// 'capacityRequirement' number of samples. The buffer is grown in steps of
// 4 kilobytes to eliminate the need for frequently growing up the buffer,
// as well as to round the buffer size up to the virtual memory page size.
void FIFOSampleBuffer::ensureCapacity(uint capacityRequirement)
{
SAMPLETYPE *tempUnaligned, *temp;
if (capacityRequirement > getCapacity())
{
// enlarge the buffer in 4kbyte steps (round up to next 4k boundary)
sizeInBytes = (capacityRequirement * channels * sizeof(SAMPLETYPE) + 4095) & -4096;
assert(sizeInBytes % 2 == 0);
tempUnaligned = new SAMPLETYPE[sizeInBytes / sizeof(SAMPLETYPE) + 16 / sizeof(SAMPLETYPE)];
if (tempUnaligned == NULL)
{
throw std::runtime_error("Couldn't allocate memory!\n");
}
temp = (SAMPLETYPE *)(((ulong)tempUnaligned + 15) & -16);
memcpy(temp, ptrBegin(), samplesInBuffer * channels * sizeof(SAMPLETYPE));
delete[] bufferUnaligned;
buffer = temp;
bufferUnaligned = tempUnaligned;
bufferPos = 0;
}
else
{
// simply rewind the buffer (if necessary)
rewind();
}
}
// Returns the current buffer capacity in terms of samples
uint FIFOSampleBuffer::getCapacity() const
{
return sizeInBytes / (channels * sizeof(SAMPLETYPE));
}
// Returns the number of samples currently in the buffer
uint FIFOSampleBuffer::numSamples() const
{
return samplesInBuffer;
}
// Output samples from beginning of the sample buffer. Copies demanded number
// of samples to output and removes them from the sample buffer. If there
// are less than 'numsample' samples in the buffer, returns all available.
//
// Returns number of samples copied.
uint FIFOSampleBuffer::receiveSamples(SAMPLETYPE *output, uint maxSamples)
{
uint num;
num = (maxSamples > samplesInBuffer) ? samplesInBuffer : maxSamples;
memcpy(output, ptrBegin(), channels * sizeof(SAMPLETYPE) * num);
return receiveSamples(num);
}
// Removes samples from the beginning of the sample buffer without copying them
// anywhere. Used to reduce the number of samples in the buffer, when accessing
// the sample buffer with the 'ptrBegin' function.
uint FIFOSampleBuffer::receiveSamples(uint maxSamples)
{
if (maxSamples >= samplesInBuffer)
{
uint temp;
temp = samplesInBuffer;
samplesInBuffer = 0;
return temp;
}
samplesInBuffer -= maxSamples;
bufferPos += maxSamples;
return maxSamples;
}
// Returns nonzero if the sample buffer is empty
int FIFOSampleBuffer::isEmpty() const
{
return (samplesInBuffer == 0) ? 1 : 0;
}
// Clears the sample buffer
void FIFOSampleBuffer::clear()
{
samplesInBuffer = 0;
bufferPos = 0;
}
0707010001e160000081a4000000020000000200000001419560300000000a000000200000000000000000000000000000002c00000000reloc/source/SoundTouch/FIFOSampleBuffer.lo timestamp
0707010001e161000081a400000002000000020000000141956030000019b0000000200000000000000000000000000000002b00000000reloc/source/SoundTouch/FIFOSampleBuffer.o ELF h 4 ( .shstrtab .gnu.linkonce.r._ZTV16FIFOSampleBuffer .gnu.linkonce.r._ZTI16FIFOSampleBuffer .gnu.linkonce.r._ZTI14FIFOSamplePipe .gnu.linkonce.r._ZTS14FIFOSamplePipe .gnu.linkonce.r._ZTS16FIFOSampleBuffer .gnu.linkonce.r._ZTV14FIFOSamplePipe .text .rodata .gcc_except_table .eh_frame .symtab .strtab .rela.gnu.linkonce.r._ZTV16FIFOSampleBuffer .rela.gnu.linkonce.r._ZTI16FIFOSampleBuffer .rela.gnu.linkonce.r._ZTI14FIFOSamplePipe .rela.gnu.linkonce.r._ZTV14FIFOSamplePipe .rela.text .rela.eh_frame .comment 14FIFOSamplePipe 16FIFOSampleBuffer " ` " " " " " " " ` " " " " " " ` @ ` ` @ ` 㿐 @ @ & & 㿐 ` @ @ @ @ & 㿐 @ 㿐 @ @ 㿐 @8 @ @ @ ` @ @ @ " @ " " `?" " x @ * k @ @ ]& @ op & @ @ @ @ & & =& @ & & @ @ @ ? @2 ? # ! ` @ @ @ ? ?@ ? @ 0@ ? 4 @ 0 㿐 @ @ @ 㿐 @ @ @
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! , ) l U | $ ` ` \ 0707010001e162000081b60000000200000002000000013fed755600001e60000000200000000000000000000000000000002600000000reloc/source/SoundTouch/FIRFilter.cpp /*****************************************************************************
*
* General FIR digital filter routines with MMX optimization.
*
* Note : MMX optimized functions reside in a separate, platform-specific file,
* e.g. 'mmx_win.cpp' or 'mmx_gcc.cpp'
*
* Author : Copyright (c) Olli Parviainen
* Author e-mail : oparviai @ iki.fi
* File created : 13-Jan-2002
*
* Last changed : $Date: 2003/12/27 10:00:52 $
* File revision : $Revision: 1.13 $
*
* $Id: FIRFilter.cpp,v 1.13 2003/12/27 10:00:52 Olli Exp $
*
* License :
*
* SoundTouch sound 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
#include
#include
#include
#include
#include "FIRFilter.h"
#include "cpu_detect.h"
using namespace soundtouch;
/*****************************************************************************
*
* Implementation of the class 'FIRFilter'
*
*****************************************************************************/
FIRFilter::FIRFilter()
{
resultDivFactor = 0;
length = 0;
lengthDiv8 = 0;
filterCoeffs = NULL;
}
FIRFilter::~FIRFilter()
{
delete[] filterCoeffs;
}
// Usual C-version of the filter routine for stereo sound
uint FIRFilter::evaluateFilterStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples) const
{
uint i, j, end;
LONG_SAMPLETYPE suml, sumr;
#ifdef FLOAT_SAMPLES
// when using floating point samples, use a scaler instead of a divider
// because division is much slower operation than multiplying.
double dScaler = 1.0 / (double)resultDivider;
#endif
assert(length != 0);
end = 2 * (numSamples - length);
for (j = 0; j < end; j += 2)
{
const SAMPLETYPE *ptr;
suml = sumr = 0;
ptr = src + j;
for (i = 0; i < length; i += 4)
{
// loop is unrolled by factor of 4 here for efficiency
suml += ptr[2 * i + 0] * filterCoeffs[i + 0] +
ptr[2 * i + 2] * filterCoeffs[i + 1] +
ptr[2 * i + 4] * filterCoeffs[i + 2] +
ptr[2 * i + 6] * filterCoeffs[i + 3];
sumr += ptr[2 * i + 1] * filterCoeffs[i + 0] +
ptr[2 * i + 3] * filterCoeffs[i + 1] +
ptr[2 * i + 5] * filterCoeffs[i + 2] +
ptr[2 * i + 7] * filterCoeffs[i + 3];
}
#ifdef INTEGER_SAMPLES
suml >>= resultDivFactor;
sumr >>= resultDivFactor;
// saturate to 16 bit integer limits
suml = (suml < -32768) ? -32768 : (suml > 32767) ? 32767 : suml;
// saturate to 16 bit integer limits
sumr = (sumr < -32768) ? -32768 : (sumr > 32767) ? 32767 : sumr;
#else
suml *= dScaler;
sumr *= dScaler;
#endif // INTEGER_SAMPLES
dest[j] = (SAMPLETYPE)suml;
dest[j + 1] = (SAMPLETYPE)sumr;
}
return numSamples - length;
}
// Usual C-version of the filter routine for mono sound
uint FIRFilter::evaluateFilterMono(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples) const
{
uint i, j, end;
LONG_SAMPLETYPE sum;
#ifdef FLOAT_SAMPLES
// when using floating point samples, use a scaler instead of a divider
// because division is much slower operation than multiplying.
double dScaler = 1.0 / (double)resultDivider;
#endif
assert(length != 0);
end = numSamples - length;
for (j = 0; j < end; j ++)
{
sum = 0;
for (i = 0; i < length; i += 4)
{
// loop is unrolled by factor of 4 here for efficiency
sum += src[i + 0] * filterCoeffs[i + 0] +
src[i + 1] * filterCoeffs[i + 1] +
src[i + 2] * filterCoeffs[i + 2] +
src[i + 3] * filterCoeffs[i + 3];
}
#ifdef INTEGER_SAMPLES
sum >>= resultDivFactor;
// saturate to 16 bit integer limits
sum = (sum < -32768) ? -32768 : (sum > 32767) ? 32767 : sum;
#else
sum *= dScaler;
#endif // INTEGER_SAMPLES
dest[j] = (SAMPLETYPE)sum;
src ++;
}
return end;
}
// Set filter coeffiecients and length.
//
// Throws an exception if filter length isn't divisible by 8
void FIRFilter::setCoefficients(const SAMPLETYPE *coeffs, uint newLength, uint uResultDivFactor)
{
assert(newLength > 0);
if (newLength % 8) throw std::runtime_error("FIR filter length not divisible by 8");
lengthDiv8 = newLength / 8;
length = lengthDiv8 * 8;
assert(length == newLength);
resultDivFactor = uResultDivFactor;
resultDivider = (uint)pow(2, resultDivFactor);
delete[] filterCoeffs;
filterCoeffs = new SAMPLETYPE[length];
memcpy(filterCoeffs, coeffs, length * sizeof(SAMPLETYPE));
}
uint FIRFilter::getLength() const
{
return length;
}
// 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 FIRFilter::evaluate(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples, uint numChannels) const
{
assert(numChannels == 1 || numChannels == 2);
assert(length > 0);
assert(lengthDiv8 * 8 == length);
if (numSamples < length) return 0;
assert(resultDivFactor >= 0);
if (numChannels == 2)
{
return evaluateFilterStereo(dest, src, numSamples);
} else {
return evaluateFilterMono(dest, src, numSamples);
}
}
// Operator 'new' is overloaded so that it automatically creates a suitable instance
// depending on if we've a MMX-capable CPU available or not.
void * FIRFilter::operator new(size_t s)
{
// Notice! don't use "new FIRFilter" directly, use "newInstance" to create a new instance instead!
throw std::runtime_error("Don't use 'new FIRFilter', use 'newInstance' member instead!");
return NULL;
}
FIRFilter * FIRFilter::newInstance()
{
uint uExtensions;
uExtensions = detectCPUextensions();
// Check if MMX/SSE/3DNow! instruction set extensions supported by CPU
#ifdef ALLOW_MMX
// MMX routines available only with integer sample types
if (uExtensions & SUPPORT_MMX)
{
return ::new FIRFilterMMX;
}
else
#endif // ALLOW_MMX
#ifdef ALLOW_SSE
if (uExtensions & SUPPORT_SSE)
{
// SSE support
return ::new FIRFilterSSE;
}
else
#endif // ALLOW_SSE
#ifdef ALLOW_3DNOW
if (uExtensions & SUPPORT_3DNOW)
{
// 3DNow! support
return ::new FIRFilter3DNow;
}
else
#endif // ALLOW_3DNOW
{
// ISA optimizations not supported, use plain C version
return ::new FIRFilter;
}
}
0707010001e163000081b60000000200000002000000013fed7556000013e7000000200000000000000000000000000000002400000000reloc/source/SoundTouch/FIRFilter.h /*****************************************************************************
*
* General FIR digital filter routines with MMX optimization.
*
* Note : MMX optimized functions reside in a separate, platform-specific file,
* e.g. 'mmx_win.cpp' or 'mmx_gcc.cpp'
*
* Author : Copyright (c) Olli Parviainen
* Author e-mail : oparviai @ iki.fi
* File created : 13-Jan-2002
*
* Last changed : $Date: 2003/12/27 10:00:52 $
* File revision : $Revision: 1.13 $
*
* $Id: FIRFilter.h,v 1.13 2003/12/27 10:00:52 Olli Exp $
*
* License :
*
* SoundTouch sound 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
*
*****************************************************************************/
#ifndef FIRFilter_H
#define FIRFilter_H
#include "STTypes.h"
class FIRFilter
{
protected:
// Number of FIR filter taps
uint length;
// Number of FIR filter taps divided by 8
uint lengthDiv8;
// Result divider factor in 2^k format
uint resultDivFactor;
// Result divider value.
soundtouch::SAMPLETYPE resultDivider;
// Memory for filter coefficients
soundtouch::SAMPLETYPE *filterCoeffs;
virtual uint evaluateFilterStereo(soundtouch::SAMPLETYPE *dest,
const soundtouch::SAMPLETYPE *src,
uint numSamples) const;
virtual uint evaluateFilterMono(soundtouch::SAMPLETYPE *dest,
const soundtouch::SAMPLETYPE *src,
uint numSamples) const;
public:
FIRFilter();
virtual ~FIRFilter();
/// Operator 'new' is overloaded so that it automatically creates a suitable instance
/// depending on if we've a MMX-capable CPU available or not.
void * operator new(size_t s);
static FIRFilter *newInstance();
/// 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.
///
/// \return Number of samples copied to 'dest'.
uint evaluate(soundtouch::SAMPLETYPE *dest,
const soundtouch::SAMPLETYPE *src,
uint numSamples,
uint numChannels) const;
uint getLength() const;
virtual void setCoefficients(const soundtouch::SAMPLETYPE *coeffs,
uint newLength,
uint uResultDivFactor);
};
// Optional subclasses that implement CPU-specific optimizations:
#ifdef ALLOW_MMX
/// Class that implements MMX optimized functions exclusive for 16bit integer samples type.
class FIRFilterMMX : public FIRFilter
{
protected:
short *filterCoeffsUnalign;
short *filterCoeffsAlign;
virtual uint evaluateFilterStereo(short *dest, const short *src, uint numSamples) const;
public:
FIRFilterMMX();
~FIRFilterMMX();
virtual void setCoefficients(const short *coeffs, uint newLength, uint uResultDivFactor);
};
#endif // ALLOW_MMX
#ifdef ALLOW_3DNOW
/// Class that implements 3DNow! optimized functions exclusive for floating point samples type.
class FIRFilter3DNow : public FIRFilter
{
protected:
float *filterCoeffsUnalign;
float *filterCoeffsAlign;
virtual uint evaluateFilterStereo(float *dest, const float *src, uint numSamples) const;
public:
FIRFilter3DNow();
~FIRFilter3DNow();
virtual void setCoefficients(const float *coeffs, uint newLength, uint uResultDivFactor);
};
#endif // ALLOW_3DNOW
#ifdef ALLOW_SSE
/// Class that implements SSE optimized functions exclusive for floating point samples type.
class FIRFilterSSE : public FIRFilter
{
protected:
float *filterCoeffsUnalign;
float *filterCoeffsAlign;
virtual uint evaluateFilterStereo(float *dest, const float *src, uint numSamples) const;
public:
FIRFilterSSE();
~FIRFilterSSE();
virtual void setCoefficients(const float *coeffs, uint newLength, uint uResultDivFactor);
};
#endif // ALLOW_SSE
#endif // FIRFilter_H
0707010001e164000081a4000000020000000200000001419560290000000a000000200000000000000000000000000000002500000000reloc/source/SoundTouch/FIRFilter.lo timestamp
0707010001e165000081a40000000200000002000000014195602900001cb4000000200000000000000000000000000000002400000000reloc/source/SoundTouch/FIRFilter.o ELF 4 4 ( .shstrtab .gnu.linkonce.r._ZTV9FIRFilter .gnu.linkonce.r._ZTI9FIRFilter .gnu.linkonce.r._ZTS9FIRFilter .text .rodata .gcc_except_table .eh_frame .symtab .strtab .rela.gnu.linkonce.r._ZTV9FIRFilter .rela.gnu.linkonce.r._ZTI9FIRFilter .rela.text .rela.eh_frame .comment 9FIRFilter ` " " " " " ` " " " " " 㿐 `
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length != 0 FIRFilter.cpp numChannels == 1 || numChannels == 2 lengthDiv8 * 8 == length length > 0 Don't use 'new FIRFilter', use 'newInstance' member instead! newLength > 0 length == newLength FIR filter length not divisible by 8 A A A ( d \ 4 [ @ 4 \ t 8 , \ zPL |P
- @
- `
- D
-
-
- 4 G Y k @ 4 t 4 ! 8 2 ! A H a } 1 D M ` y ! * C I D f z FIRFilter.cpp _ZN9FIRFilter15setCoefficientsEPKsjj pthread_mutex_lock _ZN9FIRFilterC2Ev _ZN9FIRFilterC1Ev _ZNK9FIRFilter8evaluateEPsPKsjj pow __cxa_free_exception _Znaj _Znwj pthread_mutex_trylock _ZN9FIRFilterD2Ev _ZN9FIRFilterD1Ev pthread_mutex_unlock _ZTI9FIRFilter _ZN9FIRFilterD0Ev _ZN9FIRFilternwEj _ZTV9FIRFilter memcpy _Z19detectCPUextensionsv _ZNSt13runtime_errorC1ERKSs _ZNK9FIRFilter20evaluateFilterStereoEPsPKsj _ZTISt13runtime_error _ZNK9FIRFilter9getLengthEv _ZdlPv __gxx_personality_v0 _ZN9__gnu_cxx18__exchange_and_addEPVii pthread_setspecific pthread_key_delete __assert pthread_key_create _ZNSt13runtime_errorD1Ev _ZNSs4_Rep10_M_destroyERKSaIcE _ZTS9FIRFilter _ZTVN10__cxxabiv117__class_type_infoE _Unwind_Resume pthread_once _ZNSsC1EPKcRKSaIcE _ZNSs4_Rep20_S_empty_rep_storageE __cxa_throw __cxa_allocate_exception .umul _ZN9FIRFilter11newInstanceEv pthread_getspecific _ZNK9FIRFilter18evaluateFilterMonoEPsPKsj pthread_create _ZdaPv 4 * ) $ D L ` 6 x 6 6 t 1 1 1 1 1 1 1 1 D H p t % 0 1 @ 1 P 1 ` 1 $ $ % ( 4 P 8 < @ P D % p p % ( % ( - 0 0 < H . P . ` d ' l p / t ' x . . + " ( " ( $ ( , 0 4 % ` h l 6 - 0 . . $ ' ( ' , / 0 4 8 < @ % P + X \ h t . | . " ( " ( ! $ D d E as: Sun WorkShop 6 update 2 Compiler Common 6.2 Solaris_9_CBE 2001/04/02
GCC: (GNU) 3.4.1 4
H * d I p h | n v
L ( p
T H l l \ 0707010001e166000081a400000002000000020000000141955fea00003873000000200000000000000000000000000000002100000000reloc/source/SoundTouch/Makefile # Makefile.in generated by automake 1.7.7 from Makefile.am.
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# other linking flags to add
#libSoundTouch_la_LIBADD=
# Tell versions [3.59,3.63) of GNU make to not export all variables.
# Otherwise a system limit (for SysV at least) may be exceeded.
.NOEXPORT:
0707010001e167000081b60000000200000002000000013fed755600000732000000200000000000000000000000000000002400000000reloc/source/SoundTouch/Makefile.am ## Process this file with automake to create Makefile.in
##
## $Id: Makefile.am,v 1.1 2003/12/25 11:36:08 Olli Exp $
##
## Copyright (C) 2003 - David W. Durham
##
## This file is part of SoundTouch, an audio processing library for pitch/time adjustments
##
## SoundTouch is free software; you can redistribute it and/or modify it under the
## terms of the GNU General Public License as published by the Free Software
## Foundation; either version 2 of the License, or (at your option) any later
## version.
##
## SoundTouch 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 General Public License for more details.
##
## You should have received a copy of the GNU General Public License along with
## this program; if not, write to the Free Software Foundation, Inc., 59 Temple
## Place - Suite 330, Boston, MA 02111-1307, USA
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#libSoundTouch_la_LIBADD=
# Tell versions [3.59,3.63) of GNU make to not export all variables.
# Otherwise a system limit (for SysV at least) may be exceeded.
.NOEXPORT:
0707010001e169000081b60000000200000002000000013fed755700002f75000000200000000000000000000000000000002b00000000reloc/source/SoundTouch/RateTransposer.cpp /*****************************************************************************
*
* Sample rate transposer. Changes sample rate by using linear interpolation
* together with anti-alias filtering (first order interpolation with anti-
* alias filtering should be quite adequate for this application)
*
* Author : Copyright (c) Olli Parviainen
* Author e-mail : oparviai @ iki.fi
* File created : 13-Jan-2002
*
* Last changed : $Date: 2003/12/27 10:00:52 $
* File revision : $Revision: 1.7 $
*
* $Id: RateTransposer.cpp,v 1.7 2003/12/27 10:00:52 Olli Exp $
*
* License :
*
* SoundTouch sound 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
#include
#include
#include
#include
#include "RateTransposer.h"
#include "AAFilter.h"
using namespace soundtouch;
#ifndef min
#define min(a,b) ((a > b) ? b : a)
#define max(a,b) ((a < b) ? b : a)
#endif
/// fixed-point interpolation routine precision
#define SCALE 65536
// Constructor
RateTransposer::RateTransposer() : FIFOProcessor(&outputBuffer)
{
uChannels = 2;
bUseAAFilter = TRUE;
resetRegisters();
// Instantiates the anti-alias filter with default tap length
// of 32
pAAFilter = new AAFilter(32);
setRate(1.0f);
}
RateTransposer::~RateTransposer()
{
delete pAAFilter;
}
void RateTransposer::resetRegisters()
{
iSlopeCount = 0;
sPrevSampleL =
sPrevSampleR = 0;
}
// Enables/disables the anti-alias filter. Zero to disable, nonzero to enable
void RateTransposer::enableAAFilter(const BOOL newMode)
{
bUseAAFilter = newMode;
}
// Returns nonzero if anti-alias filter is enabled.
BOOL RateTransposer::isAAFilterEnabled() const
{
return bUseAAFilter;
}
AAFilter *RateTransposer::getAAFilter() const
{
return pAAFilter;
}
// Transposes the sample rate of the given samples using linear interpolation.
// 'Mono' version of the routine. Returns the number of samples returned in
// the "dest" buffer
uint RateTransposer::transposeMono(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples)
{
unsigned int i, used;
LONG_SAMPLETYPE temp, vol1;
used = 0;
i = 0;
// Process the last sample saved from the previous call first...
while (iSlopeCount <= SCALE)
{
vol1 = (LONG_SAMPLETYPE)(SCALE - iSlopeCount);
temp = vol1 * sPrevSampleL + iSlopeCount * src[0];
dest[i] = (SAMPLETYPE)(temp / SCALE);
i++;
iSlopeCount += uRate;
}
// now always (iSlopeCount > SCALE)
iSlopeCount -= SCALE;
while (1)
{
while (iSlopeCount > SCALE)
{
iSlopeCount -= SCALE;
used ++;
if (used >= numSamples - 1) goto end;
}
vol1 = (LONG_SAMPLETYPE)(SCALE - iSlopeCount);
temp = src[used] * vol1 + iSlopeCount * src[used + 1];
dest[i] = (SAMPLETYPE)(temp / SCALE);
i++;
iSlopeCount += uRate;
}
end:
// Store the last sample for the next round
sPrevSampleL = src[numSamples - 1];
return i;
}
// Transposes the sample rate of the given samples using linear interpolation.
// 'Mono' version of the routine. Returns the number of samples returned in
// the "dest" buffer
uint RateTransposer::transposeStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples)
{
unsigned int srcPos, i, used;
LONG_SAMPLETYPE temp, vol1;
if (numSamples == 0) return 0; // no samples, no work
used = 0;
i = 0;
// Process the last sample saved from the sPrevSampleLious call first...
while (iSlopeCount <= SCALE)
{
vol1 = (LONG_SAMPLETYPE)(SCALE - iSlopeCount);
temp = vol1 * sPrevSampleL + iSlopeCount * src[0];
dest[2 * i] = (SAMPLETYPE)(temp / SCALE);
temp = vol1 * sPrevSampleR + iSlopeCount * src[1];
dest[2 * i + 1] = (SAMPLETYPE)(temp / SCALE);
i++;
iSlopeCount += uRate;
}
// now always (iSlopeCount > SCALE)
iSlopeCount -= SCALE;
while (1)
{
while (iSlopeCount > SCALE)
{
iSlopeCount -= SCALE;
used ++;
if (used >= numSamples - 1) goto end;
}
srcPos = 2 * used;
vol1 = (LONG_SAMPLETYPE)(SCALE - iSlopeCount);
temp = src[srcPos] * vol1 + iSlopeCount * src[srcPos + 2];
dest[2 * i] = (SAMPLETYPE)(temp / SCALE);
temp = src[srcPos + 1] * vol1 + iSlopeCount * src[srcPos + 3];
dest[2 * i + 1] = (SAMPLETYPE)(temp / SCALE);
i++;
iSlopeCount += uRate;
}
end:
// Store the last sample for the next round
sPrevSampleL = src[2 * numSamples - 2];
sPrevSampleR = src[2 * numSamples - 1];
return i;
}
// Sets new target uRate. Normal uRate = 1.0, smaller values represent slower
// uRate, larger faster uRates.
void RateTransposer::setRate(float newRate)
{
float fCutoff;
uRate = (int)(newRate * SCALE + 0.5f);
// design a new anti-alias filter
if (newRate > 1.0f)
{
fCutoff = 0.5f / newRate;
}
else
{
fCutoff = 0.5f * newRate;
}
pAAFilter->setCutoffFreq(fCutoff);
}
// Outputs as many samples of the 'outputBuffer' as possible, and if there's
// any room left, outputs also as many of the incoming samples as possible.
// The goal is to drive the outputBuffer empty.
//
// It's allowed for 'output' and 'input' parameters to point to the same
// memory position.
void RateTransposer::flushStoreBuffer()
{
if (storeBuffer.isEmpty()) return;
outputBuffer.moveSamples(storeBuffer);
}
// Adds 'numSamples' pcs of samples from the 'samples' memory position into
// the input of the object.
void RateTransposer::putSamples(const SAMPLETYPE *samples, uint numSamples)
{
processSamples(samples, numSamples);
}
// Transposes up the sample rate, causing the observed playback 'rate' of the
// sound to decrease
void RateTransposer::upsample(const SAMPLETYPE *src, uint numSamples)
{
int count, sizeTemp, num;
// If the parameter 'uRate' value is smaller than 'SCALE', first transpose
// the samples and then apply the anti-alias filter to remove aliasing.
// First check that there's enough room in 'storeBuffer'
// (+16 is to reserve some slack in the destination buffer)
sizeTemp = SCALE * numSamples / uRate + 16;
// Transpose the samples, store the result into the end of "storeBuffer"
count = transpose(storeBuffer.ptrEnd(sizeTemp), src, numSamples);
storeBuffer.putSamples(count);
// Apply the anti-alias filter to samples in "store output", output the
// result to "dest"
num = storeBuffer.numSamples();
count = pAAFilter->evaluate(outputBuffer.ptrEnd(num),
storeBuffer.ptrBegin(), num, uChannels);
outputBuffer.putSamples(count);
// Remove the processed samples from "storeBuffer"
storeBuffer.receiveSamples(count);
}
// Transposes down the sample rate, causing the observed playback 'rate' of the
// sound to increase
void RateTransposer::downsample(const SAMPLETYPE *src, uint numSamples)
{
int count, sizeTemp;
// If the parameter 'uRate' value is larger than 'SCALE', first apply the
// anti-alias filter to remove high frequencies (prevent them from folding
// over the lover frequencies), then transpose. */
// Add the new samples to the end of the storeBuffer */
storeBuffer.putSamples(src, numSamples);
// Anti-alias filter the samples to prevent folding and output the filtered
// data to tempBuffer. Note : because of the FIR filter length, the
// filtering routine takes in 'filter_length' more samples than it outputs.
assert(tempBuffer.isEmpty());
sizeTemp = storeBuffer.numSamples();
count = pAAFilter->evaluate(tempBuffer.ptrEnd(sizeTemp),
storeBuffer.ptrBegin(), sizeTemp, uChannels);
// Remove the filtered samples from 'storeBuffer'
storeBuffer.receiveSamples(count);
// Transpose the samples (+16 is to reserve some slack in the destination buffer)
sizeTemp = SCALE * numSamples / uRate + 16;
count = transpose(outputBuffer.ptrEnd(sizeTemp), tempBuffer.ptrBegin(), count);
outputBuffer.putSamples(count);
}
// Transposes sample rate by applying anti-alias filter to prevent folding.
// Returns amount of samples returned in the "dest" buffer.
// The maximum amount of samples that can be returned at a time is set by
// the 'set_returnBuffer_size' function.
void RateTransposer::processSamples(const SAMPLETYPE *src, uint numSamples)
{
uint count;
uint sizeReq;
if (numSamples == 0) return;
assert(pAAFilter);
/* Enable this code snippet to disable transposing algorithm when no rate change
is required; this will reduce cpu burden by a small amount, but also cause a
'click' in the sound if rate changed goes from a non-zero setting to zero during
the sound stream.
if (uRate == SCALE)
{
// Rate not changed from the original; simply pass through the samples
flushStoreBuffer();
outputBuffer.putSamples(src, numSamples);
resetRegisters();
return;
}
*/
// If anti-alias filter is turned off, simply transpose without applying
// the filter
if (bUseAAFilter == FALSE)
{
sizeReq = SCALE * numSamples / uRate + 1;
count = transpose(outputBuffer.ptrEnd(sizeReq), src, numSamples);
outputBuffer.putSamples(count);
return;
}
// Transpose with anti-alias filter
if (uRate < SCALE)
{
upsample(src, numSamples);
}
else
{
downsample(src, numSamples);
}
}
// Transposes the sample rate of the given samples using linear interpolation.
// Returns the number of samples returned in the "dest" buffer
inline uint RateTransposer::transpose(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples)
{
if (uChannels == 2)
{
return transposeStereo(dest, src, numSamples);
}
else
{
return transposeMono(dest, src, numSamples);
}
}
// Sets the number of channels, 1 = mono, 2 = stereo
void RateTransposer::setChannels(const uint numchannels)
{
if (uChannels == numchannels) return;
assert(numchannels == 1 || numchannels == 2);
uChannels = numchannels;
storeBuffer.setChannels(uChannels);
tempBuffer.setChannels(uChannels);
outputBuffer.setChannels(uChannels);
// Inits the linear interpolation registers
sPrevSampleL =
sPrevSampleR = 0;
iSlopeCount = 0;
}
// Clears all the samples in the object
void RateTransposer::clear()
{
outputBuffer.clear();
storeBuffer.clear();
}
// Returns nonzero if there aren't any samples available for outputting.
uint RateTransposer::isEmpty()
{
int res;
res = FIFOProcessor::isEmpty();
if (res == 0) return 0;
return storeBuffer.isEmpty();
}
0707010001e16a000081b60000000200000002000000013fed755700001201000000200000000000000000000000000000002900000000reloc/source/SoundTouch/RateTransposer.h /*****************************************************************************
*
* Sample rate transposer. Changes sample rate by using linear interpolation
* together with anti-alias filtering (first order interpolation with anti-
* alias filtering should be quite adequate for this application)
*
* Author : Copyright (c) Olli Parviainen
* Author e-mail : oparviai @ iki.fi
* File created : 13-Jan-2002
*
* Last changed : $Date: 2003/12/27 10:00:52 $
* File revision : $Revision: 1.5 $
*
* $Id: RateTransposer.h,v 1.5 2003/12/27 10:00:52 Olli Exp $
*
* License :
*
* SoundTouch sound 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
*
*****************************************************************************/
#ifndef RateTransposer_H
#define RateTransposer_H
#include "AAFilter.h"
#include "FIFOSamplePipe.h"
#include "FIFOSampleBuffer.h"
#include "STTypes.h"
class RateTransposer : public FIFOProcessor
{
private:
/// Anti-alias filter object
AAFilter *pAAFilter;
int iSlopeCount;
uint uRate;
uint uChannels;
soundtouch::SAMPLETYPE sPrevSampleL, sPrevSampleR;
/// Buffer for collecting samples to feed the anti-alias filter between
/// two batches
FIFOSampleBuffer storeBuffer;
/// Buffer for keeping samples between transposing & anti-alias filter
FIFOSampleBuffer tempBuffer;
/// Output sample buffer
FIFOSampleBuffer outputBuffer;
BOOL bUseAAFilter;
void resetRegisters();
uint transposeStereo(soundtouch::SAMPLETYPE *dest,
const soundtouch::SAMPLETYPE *src,
uint numSamples);
uint transposeMono(soundtouch::SAMPLETYPE *dest,
const soundtouch::SAMPLETYPE *src,
uint numSamples);
uint transpose(soundtouch::SAMPLETYPE *dest,
const soundtouch::SAMPLETYPE *src,
uint numSamples);
void flushStoreBuffer();
void downsample(const soundtouch::SAMPLETYPE *src,
uint numSamples);
void upsample(const soundtouch::SAMPLETYPE *src,
uint numSamples);
/// Transposes sample rate by applying anti-alias filter to prevent folding.
/// Returns amount of samples returned in the "dest" buffer.
/// The maximum amount of samples that can be returned at a time is set by
/// the 'set_returnBuffer_size' function.
void processSamples(const soundtouch::SAMPLETYPE *src,
uint numSamples);
public:
RateTransposer();
virtual ~RateTransposer();
/// Returns the output buffer object
FIFOSamplePipe *getOutput() { return &outputBuffer; };
/// Returns the store buffer object
FIFOSamplePipe *getStore() { return &storeBuffer; };
/// Return anti-alias filter object
AAFilter *getAAFilter() const;
/// Enables/disables the anti-alias filter. Zero to disable, nonzero to enable
void enableAAFilter(BOOL newMode);
/// Returns nonzero if anti-alias filter is enabled.
BOOL isAAFilterEnabled() const;
/// Sets new target rate. Normal rate = 1.0, smaller values represent slower
/// rate, larger faster rates.
void setRate(float newRate);
/// Sets the number of channels, 1 = mono, 2 = stereo
void setChannels(uint channels);
/// Adds 'numSamples' pcs of samples from the 'samples' memory position into
/// the input of the object.
virtual void putSamples(const soundtouch::SAMPLETYPE *samples, uint numSamples);
/// Clears all the samples in the object
virtual void clear();
/// Returns nonzero if there aren't any samples available for outputting.
virtual uint isEmpty();
};
#endif
0707010001e16b000081a40000000200000002000000014195603b0000000a000000200000000000000000000000000000002a00000000reloc/source/SoundTouch/RateTransposer.lo timestamp
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&x & 0 * & 5 . D / \ 0707010001e16d000081b60000000200000002000000013fed7557000033d0000000200000000000000000000000000000002700000000reloc/source/SoundTouch/SoundTouch.cpp /*****************************************************************************
*
* A main class for tempo/pitch/rate adjusting routines.
*
* Initialize the object by setting up the sound stream parameters with the
* 'setSampleRate' and 'setChannels' functions, and set the desired
* tempo/pitch/rate settings with the corresponding functions.
*
* Notes:
* * This class behaves like an first-in-first-out pipe: The samples to be
* processed are fed into one of the pipe with the 'putSamples' function,
* and the ready, processed samples are read from the other end with the
* 'receiveSamples' function.
*
* * The tempo/pitch/rate control parameters may freely be altered during
* processing.
*
* * The processing routines introduce a certain 'latency' between the
* input and output, so that the inputted samples aren't immediately
* transferred to the output, and neither the number of output samples
* immediately available after inputting some samples isn't in direct
* relationship to the number of previously inputted samples.
*
* * This class utilizes classes 'tempochanger' to change tempo of the
* sound (without changing pitch) and 'transposer' to change rate
* (that is, both tempo and pitch) of the sound. The third available control
* 'pitch' (change pitch but maintain tempo) is produced by suitably
* combining the two preceding controls.
*
* Author : Copyright (c) Olli Parviainen
* Author e-mail : oparviai @ iki.fi
* File created : 13-Jan-2002
*
* Last changed : $Date: 2003/12/27 10:00:30 $
* File revision : $Revision: 1.8 $
*
* $Id: SoundTouch.cpp,v 1.8 2003/12/27 10:00:30 Olli Exp $
*
* License :
*
* SoundTouch sound 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
#include
#include
#include
#include
#include
#include "SoundTouch.h"
#include "TDStretch.h"
#include "RateTransposer.h"
#include "cpu_detect.h"
using namespace soundtouch;
/// Print library version string
extern "C" void soundtouch_ac_test()
{
printf("SoundTouch Version: %s\n",SOUNDTOUCH_VERSION);
}
SoundTouch::SoundTouch()
{
// Initialize rate transposer and tempo changer instances
pRateTransposer = new RateTransposer;
pTDStretch = TDStretch::newInstance();
setOutPipe(pTDStretch);
rate = tempo = 0;
virtualPitch =
virtualRate =
virtualTempo = 1.0;
calcEffectiveRateAndTempo();
channels = 0;
bSrateSet = FALSE;
}
SoundTouch::~SoundTouch()
{
delete pRateTransposer;
delete pTDStretch;
}
/// Get SoundTouch library version string
const char *SoundTouch::getVersionString()
{
static const char *_version = SOUNDTOUCH_VERSION;
return _version;
}
/// Get SoundTouch library version Id
uint SoundTouch::getVersionId()
{
return SOUNDTOUCH_VERSION_ID;
}
// Sets the number of channels, 1 = mono, 2 = stereo
void SoundTouch::setChannels(uint numChannels)
{
if (numChannels != 1 && numChannels != 2)
{
throw std::runtime_error("Illegal number of channels");
}
channels = numChannels;
pRateTransposer->setChannels(numChannels);
pTDStretch->setChannels(numChannels);
}
// Sets new rate control value. Normal rate = 1.0, smaller values
// represent slower rate, larger faster rates.
void SoundTouch::setRate(float newRate)
{
virtualRate = newRate;
calcEffectiveRateAndTempo();
}
// Sets new rate control value as a difference in percents compared
// to the original rate (-50 .. +100 %)
void SoundTouch::setRateChange(float newRate)
{
virtualRate = 1.0f + 0.01f * newRate;
calcEffectiveRateAndTempo();
}
// Sets new tempo control value. Normal tempo = 1.0, smaller values
// represent slower tempo, larger faster tempo.
void SoundTouch::setTempo(float newTempo)
{
virtualTempo = newTempo;
calcEffectiveRateAndTempo();
}
// Sets new tempo control value as a difference in percents compared
// to the original tempo (-50 .. +100 %)
void SoundTouch::setTempoChange(float newTempo)
{
virtualTempo = 1.0f + 0.01f * newTempo;
calcEffectiveRateAndTempo();
}
// Sets new pitch control value. Original pitch = 1.0, smaller values
// represent lower pitches, larger values higher pitch.
void SoundTouch::setPitch(float newPitch)
{
virtualPitch = newPitch;
calcEffectiveRateAndTempo();
}
// Sets pitch change in octaves compared to the original pitch
// (-1.00 .. +1.00)
void SoundTouch::setPitchOctaves(float newPitch)
{
virtualPitch = (float)exp(0.69314718056f * newPitch);
calcEffectiveRateAndTempo();
}
// Sets pitch change in semi-tones compared to the original pitch
// (-12 .. +12)
void SoundTouch::setPitchSemiTones(int newPitch)
{
setPitchOctaves((float)newPitch / 12.0f);
}
void SoundTouch::setPitchSemiTones(float newPitch)
{
setPitchOctaves(newPitch / 12.0f);
}
// Calculates 'effective' rate and tempo values from the
// nominal control values.
void SoundTouch::calcEffectiveRateAndTempo()
{
float oldTempo = tempo;
float oldRate = rate;
tempo = virtualTempo / virtualPitch;
rate = virtualPitch * virtualRate;
if (rate != oldRate) pRateTransposer->setRate(rate);
if (tempo != oldTempo) pTDStretch->setTempo(tempo);
if (rate > 1.0f)
{
if (output != pRateTransposer)
{
FIFOSamplePipe *transOut;
assert(output == pTDStretch);
// move samples in the current output buffer to the output of pRateTransposer
transOut = pRateTransposer->getOutput();
transOut->moveSamples(*output);
// move samples in tempo changer's input to pitch transposer's input
pRateTransposer->moveSamples(*pTDStretch->getInput());
output = pRateTransposer;
}
}
else
{
if (output != pTDStretch)
{
FIFOSamplePipe *tempoOut;
assert(output == pRateTransposer);
// move samples in the current output buffer to the output of pTDStretch
tempoOut = pTDStretch->getOutput();
tempoOut->moveSamples(*output);
// move samples in pitch transposer's store buffer to tempo changer's input
pTDStretch->moveSamples(*pRateTransposer->getStore());
output = pTDStretch;
}
}
}
// Sets sample rate.
void SoundTouch::setSampleRate(uint srate)
{
bSrateSet = TRUE;
// set sample rate, leave other tempo changer parameters as they are.
pTDStretch->setParameters(srate);
}
// Adds 'numSamples' pcs of samples from the 'samples' memory position into
// the input of the object.
void SoundTouch::putSamples(const SAMPLETYPE *samples, uint numSamples)
{
if (bSrateSet == FALSE)
{
throw std::runtime_error("SoundTouch : Sample rate not defined");
}
else if (channels == 0)
{
throw std::runtime_error("SoundTouch : Number of channels not defined");
}
// Transpose the rate of the new samples if necessary
if (rate == 1.0f)
{
// The rate value is same as the original, simply evaluate the tempo changer.
assert(output == pTDStretch);
if (pRateTransposer->isEmpty() == 0)
{
// yet flush the last samples in the pitch transposer buffer
// (may happen if 'rate' changes from a non-zero value to zero)
pTDStretch->moveSamples(*pRateTransposer);
}
pTDStretch->putSamples(samples, numSamples);
}
else if (rate < 1.0f)
{
// transpose the rate down, output the transposed sound to tempo changer buffer
assert(output == pTDStretch);
pRateTransposer->putSamples(samples, numSamples);
pTDStretch->moveSamples(*pRateTransposer);
}
else
{
assert(rate > 1.0f);
// evaluate the tempo changer, then transpose the rate up,
assert(output == pRateTransposer);
pTDStretch->putSamples(samples, numSamples);
pRateTransposer->moveSamples(*pTDStretch);
}
}
// Flushes the last samples from the processing pipeline to the output.
// Clears also the internal processing buffers.
//
// Note: This function is meant for extracting the last samples of a sound
// stream. This function may introduce additional blank samples in the end
// of the sound stream, and thus it's not recommended to call this function
// in the middle of a sound stream.
void SoundTouch::flush()
{
int i;
uint nOut;
SAMPLETYPE buff[128];
nOut = numSamples();
memset(buff, 0, 128 * sizeof(SAMPLETYPE));
// "Push" the last active samples out from the processing pipeline by
// feeding blank samples into the processing pipeline until new,
// processed samples appear in the output (not however, more than
// 8ksamples in any case)
for (i = 0; i < 128; i ++)
{
putSamples(buff, 64);
if (numSamples() != nOut) break; // new samples have appeared in the output!
}
// Clear working buffers
pRateTransposer->clear();
pTDStretch->clearInput();
// yet leave the 'tempoChanger' output intouched as that's where the
// flushed samples are!
}
// Changes a setting controlling the processing system behaviour. See the
// 'SETTING_...' defines for available setting ID's.
BOOL SoundTouch::setSetting(uint settingId, uint value)
{
uint sampleRate, sequenceMs, seekWindowMs, overlapMs;
// read current tdstretch routine parameters
pTDStretch->getParameters(&sampleRate, &sequenceMs, &seekWindowMs, &overlapMs);
switch (settingId)
{
case SETTING_USE_AA_FILTER :
// enables / disabless anti-alias filter
pRateTransposer->enableAAFilter((value != 0) ? TRUE : FALSE);
return TRUE;
case SETTING_AA_FILTER_LENGTH :
// sets anti-alias filter length
pRateTransposer->getAAFilter()->setLength(value);
return TRUE;
case SETTING_USE_QUICKSEEK :
// enables / disables tempo routine quick seeking algorithm
pTDStretch->enableQuickSeek((value != 0) ? TRUE : FALSE);
return TRUE;
case SETTING_SEQUENCE_MS:
// change time-stretch sequence duration parameter
pTDStretch->setParameters(sampleRate, value, seekWindowMs, overlapMs);
return TRUE;
case SETTING_SEEKWINDOW_MS:
// change time-stretch seek window length parameter
pTDStretch->setParameters(sampleRate, sequenceMs, value, overlapMs);
return TRUE;
case SETTING_OVERLAP_MS:
// change time-stretch overlap length parameter
pTDStretch->setParameters(sampleRate, sequenceMs, seekWindowMs, value);
return TRUE;
default :
return FALSE;
}
}
// Reads a setting controlling the processing system behaviour. See the
// 'SETTING_...' defines for available setting ID's.
//
// Returns the setting value.
uint SoundTouch::getSetting(uint settingId) const
{
uint temp;
switch (settingId)
{
case SETTING_USE_AA_FILTER :
return pRateTransposer->isAAFilterEnabled();
case SETTING_AA_FILTER_LENGTH :
return pRateTransposer->getAAFilter()->getLength();
case SETTING_USE_QUICKSEEK :
return pTDStretch->isQuickSeekEnabled();
case SETTING_SEQUENCE_MS:
pTDStretch->getParameters(NULL, &temp, NULL, NULL);
return temp;
case SETTING_SEEKWINDOW_MS:
pTDStretch->getParameters(NULL, NULL, &temp, NULL);
return temp;
case SETTING_OVERLAP_MS:
pTDStretch->getParameters(NULL, NULL, NULL, &temp);
return temp;
default :
return 0;
}
}
// Clears all the samples in the object's output and internal processing
// buffers.
void SoundTouch::clear()
{
pRateTransposer->clear();
pTDStretch->clear();
}
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- - 0 - ' 9 2 9 A ;T \ 0707010001e172000081b60000000200000002000000013fed755700005ef0000000200000000000000000000000000000002600000000reloc/source/SoundTouch/TDStretch.cpp /*****************************************************************************
*
* Sampled sound tempo changer/time stretch algorithm. Changes the sound tempo
* while maintaining the original pitch by using a time domain WSOLA-like
* method with several performance-increasing tweaks.
*
* Note : MMX optimized functions reside in a separate, platform-specific
* file, e.g. 'mmx_win.cpp' or 'mmx_gcc.cpp'
*
* Author : Copyright (c) Olli Parviainen
* Author e-mail : oparviai @ iki.fi
* File created : 13-Jan-2002
*
* Last changed : $Date: 2003/12/27 10:00:52 $
* File revision : $Revision: 1.14 $
*
* $Id: TDStretch.cpp,v 1.14 2003/12/27 10:00:52 Olli Exp $
*
* License :
*
* SoundTouch sound 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
#include
#include
#include
#include