--- /dev/null
+////////////////////////////////////////////////////////////////////////////////
+///
+/// Win32 version of the AMD 3DNow! optimized routines for AMD K6-2/Athlon
+/// processors. All 3DNow! optimized functions have been gathered into this
+/// single source code file, regardless to their class or original source code
+/// file, in order to ease porting the library to other compiler and processor
+/// platforms.
+///
+/// By the way; the performance gain depends heavily on the CPU generation: On
+/// K6-2 these routines provided speed-up of even 2.4 times, while on Athlon the
+/// difference to the original routines stayed at unremarkable 8%! Such a small
+/// improvement on Athlon is due to 3DNow can perform only two operations in
+/// parallel, and obviously also the Athlon FPU is doing a very good job with
+/// the standard C floating point routines! Here these routines are anyway,
+/// although it might not be worth the effort to convert these to GCC platform,
+/// for Athlon CPU at least. The situation is different regarding the SSE
+/// optimizations though, thanks to the four parallel operations of SSE that
+/// already make a difference.
+///
+/// This file is to be compiled in Windows platform with Microsoft Visual C++
+/// Compiler. Please see '3dnow_gcc.cpp' for the gcc compiler version for all
+/// GNU platforms (if file supplied).
+///
+/// NOTICE: If using Visual Studio 6.0, you'll need to install the "Visual C++
+/// 6.0 processor pack" update to support 3DNow! instruction set. The update is
+/// available for download at Microsoft Developers Network, see here:
+/// http://msdn.microsoft.com/vstudio/downloads/tools/ppack/default.aspx
+///
+/// If the above URL is expired or removed, go to "http://msdn.microsoft.com" and
+/// perform a search with keywords "processor pack".
+///
+/// Author : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai @ iki.fi
+/// SoundTouch WWW: http://www.iki.fi/oparviai/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// Last changed : $Date$
+// File revision : $Revision$
+//
+// $Id$
+//
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+// SoundTouch audio processing library
+// Copyright (c) Olli Parviainen
+//
+// This library is free software; you can redistribute it and/or
+// modify it under the terms of the GNU Lesser General Public
+// License as published by the Free Software Foundation; either
+// version 2.1 of the License, or (at your option) any later version.
+//
+// This library is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+// Lesser General Public License for more details.
+//
+// You should have received a copy of the GNU Lesser General Public
+// License along with this library; if not, write to the Free Software
+// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#include "cpu_detect.h"
+#include "STTypes.h"
+
+#ifndef WIN32
+#error "wrong platform - this source code file is exclusively for Win32 platform"
+#endif
+
+using namespace soundtouch;
+
+#ifdef ALLOW_3DNOW
+// 3DNow! routines available only with float sample type
+
+//////////////////////////////////////////////////////////////////////////////
+//
+// implementation of 3DNow! optimized functions of class 'TDStretch3DNow'
+//
+//////////////////////////////////////////////////////////////////////////////
+
+#include "TDStretch.h"
+#include <limits.h>
+
+// these are declared in 'TDStretch.cpp'
+extern int scanOffsets[4][24];
+
+
+// Calculates cross correlation of two buffers
+double TDStretch3DNow::calcCrossCorrStereo(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;
+}
+
+
+
+
+//////////////////////////////////////////////////////////////////////////////
+//
+// 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
+ delete[] filterCoeffsUnalign;
+ 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
--- /dev/null
+////////////////////////////////////////////////////////////////////////////////
+///
+/// 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
+/// SoundTouch WWW: http://www.iki.fi/oparviai/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// Last changed : $Date$
+// File revision : $Revision$
+//
+// $Id$
+//
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+// SoundTouch audio processing library
+// Copyright (c) Olli Parviainen
+//
+// This library is free software; you can redistribute it and/or
+// modify it under the terms of the GNU Lesser General Public
+// License as published by the Free Software Foundation; either
+// version 2.1 of the License, or (at your option) any later version.
+//
+// This library is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+// Lesser General Public License for more details.
+//
+// You should have received a copy of the GNU Lesser General Public
+// License along with this library; if not, write to the Free Software
+// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#include <memory.h>
+#include <assert.h>
+#include <math.h>
+#include <stdlib.h>
+#include "AAFilter.h"
+#include "FIRFilter.h"
+
+using namespace soundtouch;
+
+#define PI 3.141592655357989
+#define TWOPI (2 * PI)
+
+/*****************************************************************************
+ *
+ * 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();
+}
--- /dev/null
+////////////////////////////////////////////////////////////////////////////////
+///
+/// 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
+/// SoundTouch WWW: http://www.iki.fi/oparviai/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// Last changed : $Date$
+// File revision : $Revision$
+//
+// $Id$
+//
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+// SoundTouch audio processing library
+// Copyright (c) Olli Parviainen
+//
+// This library is free software; you can redistribute it and/or
+// modify it under the terms of the GNU Lesser General Public
+// License as published by the Free Software Foundation; either
+// version 2.1 of the License, or (at your option) any later version.
+//
+// This library is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+// Lesser General Public License for more details.
+//
+// You should have received a copy of the GNU Lesser General Public
+// License along with this library; if not, write to the Free Software
+// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#ifndef AAFilter_H
+#define AAFilter_H
+
+#include "STTypes.h"
+
+namespace soundtouch
+{
+
+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(SAMPLETYPE *dest,
+ const SAMPLETYPE *src,
+ uint numSamples,
+ uint numChannels) const;
+};
+
+}
+
+#endif
--- /dev/null
+////////////////////////////////////////////////////////////////////////////////
+///
+/// Beats-per-minute (BPM) detection routine.
+///
+/// The beat detection algorithm works as follows:
+/// - Use function 'inputSamples' to input a chunks of samples to the class for
+/// analysis. It's a good idea to enter a large sound file or stream in smallish
+/// chunks of around few kilosamples in order not to extinguish too much RAM memory.
+/// - Input sound data is decimated to approx 500 Hz to reduce calculation burden,
+/// which is basically ok as low (bass) frequencies mostly determine the beat rate.
+/// Simple averaging is used for anti-alias filtering because the resulting signal
+/// quality isn't of that high importance.
+/// - Decimated sound data is enveloped, i.e. the amplitude shape is detected by
+/// taking absolute value that's smoothed by sliding average. Signal levels that
+/// are below a couple of times the general RMS amplitude level are cut away to
+/// leave only notable peaks there.
+/// - Repeating sound patterns (e.g. beats) are detected by calculating short-term
+/// autocorrelation function of the enveloped signal.
+/// - After whole sound data file has been analyzed as above, the bpm level is
+/// detected by function 'getBpm' that finds the highest peak of the autocorrelation
+/// function, calculates it's precise location and converts this reading to bpm's.
+///
+/// Author : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai @ iki.fi
+/// SoundTouch WWW: http://www.iki.fi/oparviai/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// Last changed : $Date$
+// File revision : $Revision$
+//
+// $Id$
+//
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+// SoundTouch audio processing library
+// Copyright (c) Olli Parviainen
+//
+// This library is free software; you can redistribute it and/or
+// modify it under the terms of the GNU Lesser General Public
+// License as published by the Free Software Foundation; either
+// version 2.1 of the License, or (at your option) any later version.
+//
+// This library is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+// Lesser General Public License for more details.
+//
+// You should have received a copy of the GNU Lesser General Public
+// License along with this library; if not, write to the Free Software
+// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#ifndef _BPMDetect_H_
+#define _BPMDetect_H_
+
+#include "STTypes.h"
+#include "FIFOSampleBuffer.h"
+
+/// Minimum allowed BPM rate. Used to restrict accepted result above a reasonable limit.
+#define MIN_BPM 45
+
+/// Maximum allowed BPM rate. Used to restrict accepted result below a reasonable limit.
+#define MAX_BPM 230
+
+
+/// Class for calculating BPM rate for audio data.
+class BPMDetect
+{
+protected:
+ /// Auto-correlation accumulator bins.
+ float *xcorr;
+
+ /// Amplitude envelope sliding average approximation level accumulator
+ float envelopeAccu;
+
+ /// RMS volume sliding average approximation level accumulator
+ float RMSVolumeAccu;
+
+ /// Sample average counter.
+ int decimateCount;
+
+ /// Sample average accumulator for FIFO-like decimation.
+ soundtouch::LONG_SAMPLETYPE decimateSum;
+
+ /// Decimate sound by this coefficient to reach approx. 500 Hz.
+ int decimateBy;
+
+ /// Auto-correlation window length
+ int windowLen;
+
+ /// Number of channels (1 = mono, 2 = stereo)
+ int channels;
+
+ /// sample rate
+ int sampleRate;
+
+ /// Beginning of auto-correlation window: Autocorrelation isn't being updated for
+ /// the first these many correlation bins.
+ int windowStart;
+
+ /// FIFO-buffer for decimated processing samples.
+ soundtouch::FIFOSampleBuffer *buffer;
+
+ /// Initialize the class for processing.
+ void init(int numChannels, int sampleRate);
+
+ /// Updates auto-correlation function for given number of decimated samples that
+ /// are read from the internal 'buffer' pipe (samples aren't removed from the pipe
+ /// though).
+ void updateXCorr(int process_samples /// How many samples are processed.
+ );
+
+ /// Decimates samples to approx. 500 Hz.
+ ///
+ /// \return Number of output samples.
+ int decimate(soundtouch::SAMPLETYPE *dest, ///< Destination buffer
+ const soundtouch::SAMPLETYPE *src, ///< Source sample buffer
+ int numsamples ///< Number of source samples.
+ );
+
+ /// Calculates amplitude envelope for the buffer of samples.
+ /// Result is output to 'samples'.
+ void calcEnvelope(soundtouch::SAMPLETYPE *samples, ///< Pointer to input/output data buffer
+ int numsamples ///< Number of samples in buffer
+ );
+
+public:
+ /// Constructor.
+ BPMDetect(int numChannels, ///< Number of channels in sample data.
+ int sampleRate ///< Sample rate in Hz.
+ );
+
+ /// Destructor.
+ virtual ~BPMDetect();
+
+ /// Inputs a block of samples for analyzing: Envelopes the samples and then
+ /// updates the autocorrelation estimation. When whole song data has been input
+ /// in smaller blocks using this function, read the resulting bpm with 'getBpm'
+ /// function.
+ ///
+ /// Notice that data in 'samples' array can be disrupted in processing.
+ void inputSamples(soundtouch::SAMPLETYPE *samples, ///< Pointer to input/working data buffer
+ int numSamples ///< Number of samples in buffer
+ );
+
+
+ /// Analyzes the results and returns the BPM rate. Use this function to read result
+ /// after whole song data has been input to the class by consecutive calls of
+ /// 'inputSamples' function.
+ ///
+ /// \return Beats-per-minute rate, or zero if detection failed.
+ float getBpm();
+};
+
+#endif // _BPMDetect_H_
--- /dev/null
+ GNU GENERAL PUBLIC LICENSE
+ Version 2, June 1991
+
+ Copyright (C) 1989, 1991 Free Software Foundation, Inc.
+ 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ Everyone is permitted to copy and distribute verbatim copies
+ of this license document, but changing it is not allowed.
+
+ Preamble
+
+ The licenses for most software are designed to take away your
+freedom to share and change it. By contrast, the GNU General Public
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+\f
+ GNU GENERAL PUBLIC LICENSE
+ TERMS AND CONDITIONS FOR COPYING, DISTRIBUTION AND MODIFICATION
+
+ 0. This License applies to any program or other work which contains
+a notice placed by the copyright holder saying it may be distributed
+under the terms of this General Public License. The "Program", below,
+refers to any such program or work, and a "work based on the Program"
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+
+Activities other than copying, distribution and modification are not
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+Whether that is true depends on what the Program does.
+
+ 1. You may copy and distribute verbatim copies of the Program's
+source code as you receive it, in any medium, provided that you
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+
+You may charge a fee for the physical act of transferring a copy, and
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+
+ 2. You may modify your copy or copies of the Program or any portion
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+
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+
+ b) You must cause any work that you distribute or publish, that in
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+
+ c) If the modified program normally reads commands interactively
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+ announcement including an appropriate copyright notice and a
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+\f
+These requirements apply to the modified work as a whole. If
+identifiable sections of that work are not derived from the Program,
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+
+In addition, mere aggregation of another work not based on the Program
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+a storage or distribution medium does not bring the other work under
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+
+ 3. You may copy and distribute the Program (or a work based on it,
+under Section 2) in object code or executable form under the terms of
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+
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+\f
+ 4. You may not copy, modify, sublicense, or distribute the Program
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+This section is intended to make thoroughly clear what is believed to
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+\f
+ 8. If the distribution and/or use of the Program is restricted in
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+ 10. If you wish to incorporate parts of the Program into other free
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+ NO WARRANTY
+
+ 11. BECAUSE THE PROGRAM IS LICENSED FREE OF CHARGE, THERE IS NO WARRANTY
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+
+ END OF TERMS AND CONDITIONS
+\f
+ How to Apply These Terms to Your New Programs
+
+ If you develop a new program, and you want it to be of the greatest
+possible use to the public, the best way to achieve this is to make it
+free software which everyone can redistribute and change under these terms.
+
+ To do so, attach the following notices to the program. It is safest
+to attach them to the start of each source file to most effectively
+convey the exclusion of warranty; and each file should have at least
+the "copyright" line and a pointer to where the full notice is found.
+
+ <one line to give the program's name and a brief idea of what it does.>
+ Copyright (C) 19yy <name of author>
+
+ This program 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.
+
+ This program 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
+
+
+Also add information on how to contact you by electronic and paper mail.
+
+If the program is interactive, make it output a short notice like this
+when it starts in an interactive mode:
+
+ Gnomovision version 69, Copyright (C) 19yy name of author
+ Gnomovision comes with ABSOLUTELY NO WARRANTY; for details type `show w'.
+ This is free software, and you are welcome to redistribute it
+ under certain conditions; type `show c' for details.
+
+The hypothetical commands `show w' and `show c' should show the appropriate
+parts of the General Public License. Of course, the commands you use may
+be called something other than `show w' and `show c'; they could even be
+mouse-clicks or menu items--whatever suits your program.
+
+You should also get your employer (if you work as a programmer) or your
+school, if any, to sign a "copyright disclaimer" for the program, if
+necessary. Here is a sample; alter the names:
+
+ Yoyodyne, Inc., hereby disclaims all copyright interest in the program
+ `Gnomovision' (which makes passes at compilers) written by James Hacker.
+
+ <signature of Ty Coon>, 1 April 1989
+ Ty Coon, President of Vice
+
+This General Public License does not permit incorporating your program into
+proprietary programs. If your program is a subroutine library, you may
+consider it more useful to permit linking proprietary applications with the
+library. If this is what you want to do, use the GNU Library General
+Public License instead of this License.
--- /dev/null
+////////////////////////////////////////////////////////////////////////////////
+///
+/// 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
+/// SoundTouch WWW: http://www.iki.fi/oparviai/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// Last changed : $Date$
+// File revision : $Revision$
+//
+// $Id$
+//
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+// SoundTouch audio processing library
+// Copyright (c) Olli Parviainen
+//
+// This library is free software; you can redistribute it and/or
+// modify it under the terms of the GNU Lesser General Public
+// License as published by the Free Software Foundation; either
+// version 2.1 of the License, or (at your option) any later version.
+//
+// This library is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+// Lesser General Public License for more details.
+//
+// You should have received a copy of the GNU Lesser General Public
+// License along with this library; if not, write to the Free Software
+// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#include <stdlib.h>
+#include <memory.h>
+#include <string.h>
+#include <assert.h>
+#include <stdexcept>
+
+#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;
+}
--- /dev/null
+////////////////////////////////////////////////////////////////////////////////
+///
+/// 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
+/// output samples from the buffer as well as grows the storage size
+/// whenever necessary.
+///
+/// Author : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai @ iki.fi
+/// SoundTouch WWW: http://www.iki.fi/oparviai/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// Last changed : $Date$
+// File revision : $Revision$
+//
+// $Id$
+//
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+// SoundTouch audio processing library
+// Copyright (c) Olli Parviainen
+//
+// This library is free software; you can redistribute it and/or
+// modify it under the terms of the GNU Lesser General Public
+// License as published by the Free Software Foundation; either
+// version 2.1 of the License, or (at your option) any later version.
+//
+// This library is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+// Lesser General Public License for more details.
+//
+// You should have received a copy of the GNU Lesser General Public
+// License along with this library; if not, write to the Free Software
+// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#ifndef FIFOSampleBuffer_H
+#define FIFOSampleBuffer_H
+
+#include "FIFOSamplePipe.h"
+
+namespace soundtouch
+{
+
+/// Sample buffer working in FIFO (first-in-first-out) principle. The class takes
+/// care of storage size adjustment and data moving during input/output operations.
+///
+/// Notice that in case of stereo audio, one sample is considered to consist of
+/// both channel data.
+class FIFOSampleBuffer : public FIFOSamplePipe
+{
+private:
+ /// Sample buffer.
+ SAMPLETYPE *buffer;
+
+ // Raw unaligned buffer memory. 'buffer' is made aligned by pointing it to first
+ // 16-byte aligned location of this buffer
+ SAMPLETYPE *bufferUnaligned;
+
+ /// Sample buffer size in bytes
+ uint sizeInBytes;
+
+ /// How many samples are currently in buffer.
+ uint samplesInBuffer;
+
+ /// Channels, 1=mono, 2=stereo.
+ uint channels;
+
+ /// Current position pointer to the buffer. This pointer is increased when samples are
+ /// removed from the pipe so that it's necessary to actually rewind buffer (move data)
+ /// only new data when is put to the pipe.
+ uint bufferPos;
+
+ /// Rewind the buffer by moving data from position pointed by 'bufferPos' to real
+ /// beginning of the buffer.
+ void rewind();
+
+ /// Ensures that the buffer has capacity for at least this many samples.
+ void ensureCapacity(const uint capacityRequirement);
+
+ /// Returns current capacity.
+ uint getCapacity() const;
+
+public:
+
+ /// Constructor
+ FIFOSampleBuffer(uint numChannels = 2 ///< Number of channels, 1=mono, 2=stereo.
+ ///< Default is stereo.
+ );
+
+ /// destructor
+ virtual ~FIFOSampleBuffer();
+
+ /// Returns a pointer to the beginning of the output samples.
+ /// This function is provided for accessing the output samples directly.
+ /// Please be careful for not to corrupt the book-keeping!
+ ///
+ /// When using this function to output samples, also remember to 'remove' the
+ /// output samples from the buffer by calling the
+ /// 'receiveSamples(numSamples)' function
+ virtual SAMPLETYPE *ptrBegin() const;
+
+ /// 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
+ /// not corrupt the book-keeping!
+ ///
+ /// 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 *ptrEnd(
+ uint slackCapacity ///< How much free capacity (in samples) there _at least_
+ ///< should be so that the caller can succesfully insert the
+ ///< desired samples to the buffer. If necessary, the function
+ ///< grows the buffer size to comply with this requirement.
+ );
+
+ /// Adds 'numSamples' pcs of samples from the 'samples' memory position to
+ /// the sample buffer.
+ virtual void putSamples(const SAMPLETYPE *samples, ///< Pointer to samples.
+ uint numSamples ///< Number of samples to insert.
+ );
+
+ /// Adjusts the book-keeping to increase 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!
+ virtual void putSamples(uint numSamples ///< Number of samples been inserted.
+ );
+
+ /// Output samples from beginning of the sample buffer. Copies requested samples to
+ /// output buffer and removes them from the sample buffer. If there are less than
+ /// 'numsample' samples in the buffer, returns all that available.
+ ///
+ /// \return Number of samples returned.
+ virtual uint receiveSamples(SAMPLETYPE *output, ///< Buffer where to copy output samples.
+ uint maxSamples ///< How many samples to receive at max.
+ );
+
+ /// Adjusts book-keeping so that given number of samples are removed from beginning of the
+ /// sample buffer without copying them anywhere.
+ ///
+ /// Used to reduce the number of samples in the buffer when accessing the sample buffer directly
+ /// with 'ptrBegin' function.
+ virtual uint receiveSamples(uint maxSamples ///< Remove this many samples from the beginning of pipe.
+ );
+
+ /// Returns number of samples currently available.
+ virtual uint numSamples() const;
+
+ /// Sets number of channels, 1 = mono, 2 = stereo.
+ void setChannels(uint numChannels);
+
+ /// Returns nonzero if there aren't any samples available for outputting.
+ virtual int isEmpty() const;
+
+ /// Clears all the samples.
+ virtual void clear();
+};
+
+}
+
+#endif
--- /dev/null
+////////////////////////////////////////////////////////////////////////////////
+///
+/// 'FIFOSamplePipe' : An abstract base class for classes that manipulate sound
+/// samples by operating like a first-in-first-out pipe: New samples are fed
+/// into one end of the pipe with the 'putSamples' function, and the processed
+/// samples are received from the other end with the 'receiveSamples' function.
+///
+/// 'FIFOProcessor' : A base class for classes the do signal processing with
+/// the samples while operating like a first-in-first-out pipe. When samples
+/// are input with the 'putSamples' function, the class processes them
+/// and moves the processed samples to the given 'output' pipe object, which
+/// may be either another processing stage, or a fifo sample buffer object.
+///
+/// Author : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai @ iki.fi
+/// SoundTouch WWW: http://www.iki.fi/oparviai/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// Last changed : $Date$
+// File revision : $Revision$
+//
+// $Id$
+//
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+// SoundTouch audio processing library
+// Copyright (c) Olli Parviainen
+//
+// This library is free software; you can redistribute it and/or
+// modify it under the terms of the GNU Lesser General Public
+// License as published by the Free Software Foundation; either
+// version 2.1 of the License, or (at your option) any later version.
+//
+// This library is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+// Lesser General Public License for more details.
+//
+// You should have received a copy of the GNU Lesser General Public
+// License along with this library; if not, write to the Free Software
+// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#ifndef FIFOSamplePipe_H
+#define FIFOSamplePipe_H
+
+#include <assert.h>
+#include <stdlib.h>
+#include "STTypes.h"
+
+namespace soundtouch
+{
+
+/// Abstract base class for FIFO (first-in-first-out) sample processing classes.
+class FIFOSamplePipe
+{
+public:
+ /// Returns a pointer to the beginning of the output samples.
+ /// This function is provided for accessing the output samples directly.
+ /// Please be careful for not to corrupt the book-keeping!
+ ///
+ /// When using this function to output samples, also remember to 'remove' the
+ /// output samples from the buffer by calling the
+ /// 'receiveSamples(numSamples)' function
+ virtual SAMPLETYPE *ptrBegin() const = 0;
+
+ /// Adds 'numSamples' pcs of samples from the 'samples' memory position to
+ /// the sample buffer.
+ virtual void putSamples(const SAMPLETYPE *samples, ///< Pointer to samples.
+ uint numSamples ///< Number of samples to insert.
+ ) = 0;
+
+
+ // Moves samples from the 'other' pipe instance to this instance.
+ void moveSamples(FIFOSamplePipe &other ///< Other pipe instance where from the receive the data.
+ )
+ {
+ int oNumSamples = other.numSamples();
+
+ putSamples(other.ptrBegin(), oNumSamples);
+ other.receiveSamples(oNumSamples);
+ };
+
+ /// Output samples from beginning of the sample buffer. Copies requested samples to
+ /// output buffer and removes them from the sample buffer. If there are less than
+ /// 'numsample' samples in the buffer, returns all that available.
+ ///
+ /// \return Number of samples returned.
+ virtual uint receiveSamples(SAMPLETYPE *output, ///< Buffer where to copy output samples.
+ uint maxSamples ///< How many samples to receive at max.
+ ) = 0;
+
+ /// Adjusts book-keeping so that given number of samples are removed from beginning of the
+ /// sample buffer without copying them anywhere.
+ ///
+ /// Used to reduce the number of samples in the buffer when accessing the sample buffer directly
+ /// with 'ptrBegin' function.
+ virtual uint receiveSamples(uint maxSamples ///< Remove this many samples from the beginning of pipe.
+ ) = 0;
+
+ /// Returns number of samples currently available.
+ virtual uint numSamples() const = 0;
+
+ // Returns nonzero if there aren't any samples available for outputting.
+ virtual int isEmpty() const = 0;
+
+ /// Clears all the samples.
+ virtual void clear() = 0;
+};
+
+
+
+/// Base-class for sound processing routines working in FIFO principle. With this base
+/// class it's easy to implement sound processing stages that can be chained together,
+/// so that samples that are fed into beginning of the pipe automatically go through
+/// all the processing stages.
+///
+/// When samples are input to this class, they're first processed and then put to
+/// the FIFO pipe that's defined as output of this class. This output pipe can be
+/// either other processing stage or a FIFO sample buffer.
+class FIFOProcessor :public FIFOSamplePipe
+{
+protected:
+ /// Internal pipe where processed samples are put.
+ FIFOSamplePipe *output;
+
+ /// Sets output pipe.
+ void setOutPipe(FIFOSamplePipe *pOutput)
+ {
+ assert(output == NULL);
+ assert(pOutput != NULL);
+ output = pOutput;
+ }
+
+
+ /// Constructor. Doesn't define output pipe; it has to be set be
+ /// 'setOutPipe' function.
+ FIFOProcessor()
+ {
+ output = NULL;
+ }
+
+
+ /// Constructor. Configures output pipe.
+ FIFOProcessor(FIFOSamplePipe *pOutput ///< Output pipe.
+ )
+ {
+ output = pOutput;
+ }
+
+
+ /// Destructor.
+ virtual ~FIFOProcessor()
+ {
+ }
+
+
+ /// Returns a pointer to the beginning of the output samples.
+ /// This function is provided for accessing the output samples directly.
+ /// Please be careful for not to corrupt the book-keeping!
+ ///
+ /// When using this function to output samples, also remember to 'remove' the
+ /// output samples from the buffer by calling the
+ /// 'receiveSamples(numSamples)' function
+ virtual SAMPLETYPE *ptrBegin() const
+ {
+ return output->ptrBegin();
+ }
+
+public:
+
+ /// Output samples from beginning of the sample buffer. Copies requested samples to
+ /// output buffer and removes them from the sample buffer. If there are less than
+ /// 'numsample' samples in the buffer, returns all that available.
+ ///
+ /// \return Number of samples returned.
+ virtual uint receiveSamples(SAMPLETYPE *outBuffer, ///< Buffer where to copy output samples.
+ uint maxSamples ///< How many samples to receive at max.
+ )
+ {
+ return output->receiveSamples(outBuffer, maxSamples);
+ }
+
+
+ /// Adjusts book-keeping so that given number of samples are removed from beginning of the
+ /// sample buffer without copying them anywhere.
+ ///
+ /// Used to reduce the number of samples in the buffer when accessing the sample buffer directly
+ /// with 'ptrBegin' function.
+ virtual uint receiveSamples(uint maxSamples ///< Remove this many samples from the beginning of pipe.
+ )
+ {
+ return output->receiveSamples(maxSamples);
+ }
+
+
+ /// Returns number of samples currently available.
+ virtual uint numSamples() const
+ {
+ return output->numSamples();
+ }
+
+
+ /// Returns nonzero if there aren't any samples available for outputting.
+ virtual int isEmpty() const
+ {
+ return output->isEmpty();
+ }
+};
+
+}
+
+#endif
--- /dev/null
+////////////////////////////////////////////////////////////////////////////////
+///
+/// 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
+/// SoundTouch WWW: http://www.iki.fi/oparviai/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// Last changed : $Date$
+// File revision : $Revision$
+//
+// $Id$
+//
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+// SoundTouch audio processing library
+// Copyright (c) Olli Parviainen
+//
+// This library is free software; you can redistribute it and/or
+// modify it under the terms of the GNU Lesser General Public
+// License as published by the Free Software Foundation; either
+// version 2.1 of the License, or (at your option) any later version.
+//
+// This library is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+// Lesser General Public License for more details.
+//
+// You should have received a copy of the GNU Lesser General Public
+// License along with this library; if not, write to the Free Software
+// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#include <memory.h>
+#include <assert.h>
+#include <math.h>
+#include <stdlib.h>
+#include <stdexcept>
+#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);
+ }
+}
+
+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;
+ }
+}
--- /dev/null
+////////////////////////////////////////////////////////////////////////////////
+///
+/// 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
+/// SoundTouch WWW: http://www.iki.fi/oparviai/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// Last changed : $Date$
+// File revision : $Revision$
+//
+// $Id$
+//
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+// SoundTouch audio processing library
+// Copyright (c) Olli Parviainen
+//
+// This library is free software; you can redistribute it and/or
+// modify it under the terms of the GNU Lesser General Public
+// License as published by the Free Software Foundation; either
+// version 2.1 of the License, or (at your option) any later version.
+//
+// This library is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+// Lesser General Public License for more details.
+//
+// You should have received a copy of the GNU Lesser General Public
+// License along with this library; if not, write to the Free Software
+// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#ifndef FIRFilter_H
+#define FIRFilter_H
+
+#include "STTypes.h"
+
+namespace soundtouch
+{
+
+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.
+ SAMPLETYPE resultDivider;
+
+ // Memory for filter coefficients
+ SAMPLETYPE *filterCoeffs;
+
+ virtual uint evaluateFilterStereo(SAMPLETYPE *dest,
+ const SAMPLETYPE *src,
+ uint numSamples) const;
+ virtual uint evaluateFilterMono(SAMPLETYPE *dest,
+ const SAMPLETYPE *src,
+ uint numSamples) const;
+
+ FIRFilter();
+
+public:
+ virtual ~FIRFilter();
+
+ 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(SAMPLETYPE *dest,
+ const SAMPLETYPE *src,
+ uint numSamples,
+ uint numChannels) const;
+
+ uint getLength() const;
+
+ virtual void setCoefficients(const 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
--- /dev/null
+SoundTouch sound processing library v1.01
+=========================================
+Copyright (c) Olli Parviainen 2002
+
+A library for changing tempo, pitch and playback rate of digital sound.
+
+
+SoundStretch sound processing application v1.1
+==============================================
+Copyright (c) Olli Parviainen 2002-2003
+
+A command-line application for changing tempo, pitch and playback rates
+of WAV sound files. This program also demonstrates how the "SoundTouch"
+library can be used to process sound in own programs.
+
+
+SoundStretch Usage Instructions
+===============================
+
+SoundStretch Usage syntax:
+ soundstretch infile.wav outfile.wav [switches]
+
+Where:
+
+ "infile.wav" is the name of the input sound data file (in .WAV audio
+ file format).
+
+ "outfile.wav" is the name of the output sound file where the resulting
+ sound is saved (in .WAV audio file format).
+
+ [switches] are one or more control switches.
+
+Available control switches are:
+
+ -tempo=n : Change sound tempo by n percents (n = -95.0 .. +5000.0 %)
+
+ -pitch=n : Change sound pitch by n semitones (n = -60.0 .. + 60.0 semitones)
+
+ -rate=n : Change sound playback rate by n percents (n = -95.0 .. +5000.0 %)
+
+ -bpm=n : Detect the Beats-Per-Minute (BPM) rate of the sound and adjust the
+ tempo to meet 'n' BPMs. If this switch is defined, the "-tempo=n"
+ switch value is ignored.
+
+ If "=n" is omitted, i.e. switch "-bpm" is used alone, the
+ program just calculates and displays the BPM rate but doesn't
+ adjust tempo according to the BPM value.
+
+ -quick : Use quicker tempo change algorithm. Gains speed but loses sound
+ quality.
+
+ -naa : Don't use anti-alias filtering in samplerate transposing. Gains
+ speed but loses sound quality.
+
+ -license : Displays the program license text (GPL)
+
+Notes:
+ * The numerical switch values can be entered using either integer (e.g.
+ "-tempo=123") or decimal (e.g. "-tempo=123.45") numbers.
+
+ * The "-naa" and/or "-quick" switches can be used to reduce CPU usage
+ while compromising some sound quality
+
+ * The BPM detection algorithm works by detecting repeating low-frequency
+ (<250Hz) sound patterns and thus works mostly with most rock/pop music
+ with bass or drum beat. The BPM detection doesn't work on pieces such
+ as classical music without distinct, repeating bass frequency patterns.
+ Also pieces with varying tempo, varying bass patterns or very complex
+ bass patterns (jazz, hiphop) may produce odd BPM readings.
+
+ In cases when the bass pattern drifts a bit around a nominal beat rate
+ (e.g. drummer is again drunken :), the BPM algorithm may report incorrect
+ harmonic one-halft of one-thirdth of the correct BPM value; in such case
+ the system could for example report BPM value of 50 or 100 instead of
+ correct BPM value of 150.
+
+
+Usage examples:
+===============
+
+ Example 1
+ =========
+
+ The following command increases tempo of the sound file "originalfile.wav"
+ by 12.5% and saves result to file "destinationfile.wav":
+
+ soundstretch originalfile.wav destinationfile.wav -tempo=12.5
+
+
+ Example 2
+ =========
+
+ The following command decreases the sound pitch (key) of the sound file
+ "orig.wav" by two semitones and saves the result to file "dest.wav":
+
+ soundstretch orig.wav dest.wav -pitch=-2
+
+
+ Example 3
+ =========
+
+ The following command processes the file "orig.wav" by decreasing the
+ sound tempo by 25.3% and increasing the sound pitch (key) by 1.5 semitones.
+ Result is saved to file "dest.wav":
+
+ soundstretch orig.wav dest.wav -tempo=-25.3 -pitch=1.5
+
+
+ Example 4
+ =========
+
+ The following command detects the BPM rate of the file "orig.wav" and
+ adjusts the tempo to match 100 beats per minute. Result is saved to
+ file "dest.wav":
+
+ soundstretch orig.wav dest.wav -bpm=100
+
+
+
+Building Instructions
+=====================
+
+The package contains executable binaries for Win32 platform in the "bin"
+directory.
+
+To build the library and application executable for other platforms or to
+re-build the delivered binaries, run either of the scripts in the package
+root directory:
+
+"make-win.bat" for Microsoft Windows environment, or
+"make-gcc" for GNU/Linux or Unix environment with a gcc compiler.
+
+
+
+Change History
+==============
+
+
+ SoundTouch library Change History
+ =================================
+
+ v1.01:
+ - "mmx_gcc.cpp": Added "using namespace std" and removed "return 0" from a
+ function with void return value to fix compiler errors when compiling
+ the library in Solaris environment.
+
+ - Moved file "FIFOSampleBuffer.h" to "include" directory to allow accessing
+ the FIFOSampleBuffer class from external files.
+
+ v1.0: Initial release
+
+
+ SoundStretch application Change History
+ =======================================
+
+ v1.1:
+ - Fixed "Release" settings in Microsoft Visual C++ project file (.dsp)
+
+ - Added beats-per-minute (BPM) detection routine and command-line switch
+ "-bpm"
+
+ v1.01: Initial release
+
+
+Acknowledgements
+================
+
+Many thanks to Stuart Lamble for translating the MMX optimizations from
+MS Visual C++ syntax into gcc syntax for joy of all Linux users.
+
+Thanks also to Manish Bajpai, whose WAV file reading routines I've used
+as base of the WavInFile & WavOutFile classes, that are being used in
+the soundstrecth program for accessing WAV audio files.
+
+
+LICENSE:
+========
+
+This program 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.
+
+This program 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.\n"
+
+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
--- /dev/null
+////////////////////////////////////////////////////////////////////////////////
+///
+/// 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
+/// SoundTouch WWW: http://www.iki.fi/oparviai/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// Last changed : $Date$
+// File revision : $Revision$
+//
+// $Id$
+//
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+// SoundTouch audio processing library
+// Copyright (c) Olli Parviainen
+//
+// This library is free software; you can redistribute it and/or
+// modify it under the terms of the GNU Lesser General Public
+// License as published by the Free Software Foundation; either
+// version 2.1 of the License, or (at your option) any later version.
+//
+// This library is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+// Lesser General Public License for more details.
+//
+// You should have received a copy of the GNU Lesser General Public
+// License along with this library; if not, write to the Free Software
+// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#include <memory.h>
+#include <assert.h>
+#include <stdlib.h>
+#include <stdio.h>
+#include <limits.h>
+#include "RateTransposer.h"
+#include "AAFilter.h"
+
+using namespace soundtouch;
+
+
+/// A linear samplerate transposer class that uses integer arithmetics.
+/// for the transposing.
+class RateTransposerInteger : public RateTransposer
+{
+protected:
+ int iSlopeCount;
+ uint uRate;
+ SAMPLETYPE sPrevSampleL, sPrevSampleR;
+
+ virtual void resetRegisters();
+
+ virtual uint transposeStereo(SAMPLETYPE *dest,
+ const SAMPLETYPE *src,
+ uint numSamples);
+ virtual uint transposeMono(SAMPLETYPE *dest,
+ const SAMPLETYPE *src,
+ uint numSamples);
+
+public:
+ RateTransposerInteger();
+ virtual ~RateTransposerInteger();
+
+ /// Sets new target rate. Normal rate = 1.0, smaller values represent slower
+ /// rate, larger faster rates.
+ virtual void setRate(float newRate);
+
+};
+
+
+/// A linear samplerate transposer class that uses floating point arithmetics
+/// for the transposing.
+class RateTransposerFloat : public RateTransposer
+{
+protected:
+ float fSlopeCount;
+ float fRateStep;
+ SAMPLETYPE sPrevSampleL, sPrevSampleR;
+
+ virtual void resetRegisters();
+
+ virtual uint transposeStereo(SAMPLETYPE *dest,
+ const SAMPLETYPE *src,
+ uint numSamples);
+ virtual uint transposeMono(SAMPLETYPE *dest,
+ const SAMPLETYPE *src,
+ uint numSamples);
+
+public:
+ RateTransposerFloat();
+ virtual ~RateTransposerFloat();
+};
+
+
+
+#ifndef min
+#define min(a,b) ((a > b) ? b : a)
+#define max(a,b) ((a < b) ? b : a)
+#endif
+
+RateTransposer *RateTransposer::newInstance()
+{
+#ifdef INTEGER_SAMPLES
+ return ::new RateTransposerInteger;
+#else
+ return ::new RateTransposerFloat;
+#endif
+}
+
+
+// Constructor
+RateTransposer::RateTransposer() : FIFOProcessor(&outputBuffer)
+{
+ uChannels = 2;
+ bUseAAFilter = TRUE;
+
+ // Instantiates the anti-alias filter with default tap length
+ // of 32
+ pAAFilter = new AAFilter(32);
+}
+
+
+
+RateTransposer::~RateTransposer()
+{
+ delete pAAFilter;
+}
+
+
+
+/// 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;
+}
+
+
+
+// Sets new target uRate. Normal uRate = 1.0, smaller values represent slower
+// uRate, larger faster uRates.
+void RateTransposer::setRate(float newRate)
+{
+ float fCutoff;
+
+ fRate = newRate;
+
+ // 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 = (int)((float)numSamples / fRate + 16.0f);
+
+ // 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 = (int)((float)numSamples / fRate + 16.0f);
+ 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);
+
+ // If anti-alias filter is turned off, simply transpose without applying
+ // the filter
+ if (bUseAAFilter == FALSE)
+ {
+ sizeReq = (int)((float)numSamples / fRate + 1.0f);
+ count = transpose(outputBuffer.ptrEnd(sizeReq), src, numSamples);
+ outputBuffer.putSamples(count);
+ return;
+ }
+
+ // Transpose with anti-alias filter
+ if (fRate < 1.0f)
+ {
+ 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
+ resetRegisters();
+}
+
+
+// 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();
+}
+
+
+//////////////////////////////////////////////////////////////////////////////
+//
+// RateTransposerInteger - integer arithmetic implementation
+//
+
+/// fixed-point interpolation routine precision
+#define SCALE 65536
+
+// Constructor
+RateTransposerInteger::RateTransposerInteger() : RateTransposer()
+{
+ // call these here as these are virtual functions; calling these
+ // from the base class constructor wouldn't execute the overloaded
+ // versions (<master yoda>peculiar C++ can be</my>).
+ resetRegisters();
+ setRate(1.0f);
+}
+
+
+RateTransposerInteger::~RateTransposerInteger()
+{
+}
+
+
+void RateTransposerInteger::resetRegisters()
+{
+ iSlopeCount = 0;
+ sPrevSampleL =
+ sPrevSampleR = 0;
+}
+
+
+
+// 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 RateTransposerInteger::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 RateTransposerInteger::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 RateTransposerInteger::setRate(float newRate)
+{
+ uRate = (int)(newRate * SCALE + 0.5f);
+ RateTransposer::setRate(newRate);
+}
+
+
+//////////////////////////////////////////////////////////////////////////////
+//
+// RateTransposerFloat - floating point arithmetic implementation
+//
+//////////////////////////////////////////////////////////////////////////////
+
+// Constructor
+RateTransposerFloat::RateTransposerFloat() : RateTransposer()
+{
+ // call these here as these are virtual functions; calling these
+ // from the base class constructor wouldn't execute the overloaded
+ // versions (<master yoda>peculiar C++ can be</my>).
+ resetRegisters();
+ setRate(1.0f);
+}
+
+
+RateTransposerFloat::~RateTransposerFloat()
+{
+}
+
+
+void RateTransposerFloat::resetRegisters()
+{
+ fSlopeCount = 0;
+ sPrevSampleL =
+ sPrevSampleR = 0;
+}
+
+
+
+// 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 RateTransposerFloat::transposeMono(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples)
+{
+ unsigned int i, used;
+
+ used = 0;
+ i = 0;
+
+ // Process the last sample saved from the previous call first...
+ while (fSlopeCount <= 1.0f)
+ {
+ dest[i] = (SAMPLETYPE)((1.0f - fSlopeCount) * sPrevSampleL + fSlopeCount * src[0]);
+ i++;
+ fSlopeCount += fRate;
+ }
+ fSlopeCount -= 1.0f;
+
+ while (1)
+ {
+ while (fSlopeCount > 1.0f)
+ {
+ fSlopeCount -= 1.0f;
+ used ++;
+ if (used >= numSamples - 1) goto end;
+ }
+ dest[i] = (SAMPLETYPE)((1.0f - fSlopeCount) * src[used] + fSlopeCount * src[used + 1]);
+ i++;
+ fSlopeCount += fRate;
+ }
+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 RateTransposerFloat::transposeStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples)
+{
+ unsigned int srcPos, i, used;
+
+ if (numSamples == 0) return 0; // no samples, no work
+
+ used = 0;
+ i = 0;
+
+ // Process the last sample saved from the sPrevSampleLious call first...
+ while (fSlopeCount <= 1.0f)
+ {
+ dest[2 * i] = (SAMPLETYPE)((1.0f - fSlopeCount) * sPrevSampleL + fSlopeCount * src[0]);
+ dest[2 * i + 1] = (SAMPLETYPE)((1.0f - fSlopeCount) * sPrevSampleR + fSlopeCount * src[1]);
+ i++;
+ fSlopeCount += fRate;
+ }
+ // now always (iSlopeCount > 1.0f)
+ fSlopeCount -= 1.0f;
+
+ while (1)
+ {
+ while (fSlopeCount > 1.0f)
+ {
+ fSlopeCount -= 1.0f;
+ used ++;
+ if (used >= numSamples - 1) goto end;
+ }
+ srcPos = 2 * used;
+
+ dest[2 * i] = (SAMPLETYPE)((1.0f - fSlopeCount) * src[srcPos]
+ + fSlopeCount * src[srcPos + 2]);
+ dest[2 * i + 1] = (SAMPLETYPE)((1.0f - fSlopeCount) * src[srcPos + 1]
+ + fSlopeCount * src[srcPos + 3]);
+
+ i++;
+ fSlopeCount += fRate;
+ }
+end:
+ // Store the last sample for the next round
+ sPrevSampleL = src[2 * numSamples - 2];
+ sPrevSampleR = src[2 * numSamples - 1];
+
+ return i;
+}
--- /dev/null
+////////////////////////////////////////////////////////////////////////////////
+///
+/// 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).
+///
+/// Use either of the derived classes of 'RateTransposerInteger' or
+/// 'RateTransposerFloat' for corresponding integer/floating point tranposing
+/// algorithm implementation.
+///
+/// Author : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai @ iki.fi
+/// SoundTouch WWW: http://www.iki.fi/oparviai/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// Last changed : $Date$
+// File revision : $Revision$
+//
+// $Id$
+//
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+// SoundTouch audio processing library
+// Copyright (c) Olli Parviainen
+//
+// This library is free software; you can redistribute it and/or
+// modify it under the terms of the GNU Lesser General Public
+// License as published by the Free Software Foundation; either
+// version 2.1 of the License, or (at your option) any later version.
+//
+// This library is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+// Lesser General Public License for more details.
+//
+// You should have received a copy of the GNU Lesser General Public
+// License along with this library; if not, write to the Free Software
+// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#ifndef RateTransposer_H
+#define RateTransposer_H
+
+#include "AAFilter.h"
+#include "FIFOSamplePipe.h"
+#include "FIFOSampleBuffer.h"
+
+#include "STTypes.h"
+
+namespace soundtouch
+{
+
+/// A common linear samplerate transposer class.
+///
+/// Note: Use function "RateTransposer::newInstance()" to create a new class
+/// instance instead of the "new" operator; that function automatically
+/// chooses a correct implementation depending on if integer or floating
+/// arithmetics are to be used.
+class RateTransposer : public FIFOProcessor
+{
+protected:
+ /// Anti-alias filter object
+ AAFilter *pAAFilter;
+
+ float fRate;
+
+ uint uChannels;
+
+ /// 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 init();
+
+ virtual void resetRegisters() = 0;
+
+ virtual uint transposeStereo(SAMPLETYPE *dest,
+ const SAMPLETYPE *src,
+ uint numSamples) = 0;
+ virtual uint transposeMono(SAMPLETYPE *dest,
+ const SAMPLETYPE *src,
+ uint numSamples) = 0;
+ uint transpose(SAMPLETYPE *dest,
+ const SAMPLETYPE *src,
+ uint numSamples);
+
+ void flushStoreBuffer();
+
+ void downsample(const SAMPLETYPE *src,
+ uint numSamples);
+ void upsample(const 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 SAMPLETYPE *src,
+ uint numSamples);
+
+ RateTransposer();
+
+public:
+ virtual ~RateTransposer();
+
+ /// Use this function instead of "new" operator to create a new instance of this class.
+ /// This function automatically chooses a correct implementation, depending on if
+ /// integer ot floating point arithmetics are to be used.
+ static RateTransposer *newInstance();
+
+ /// 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.
+ virtual 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.
+ void putSamples(const SAMPLETYPE *samples, uint numSamples);
+
+ /// Clears all the samples in the object
+ void clear();
+
+ /// Returns nonzero if there aren't any samples available for outputting.
+ uint isEmpty();
+};
+
+}
+
+#endif
--- /dev/null
+# -*- python -*-
+
+import glob
+
+soundtouch_files = Split("""
+AAFilter.cpp
+FIFOSampleBuffer.cpp
+FIRFilter.cpp
+RateTransposer.cpp
+SoundTouch.cpp
+TDStretch.cpp
+mmx_gcc.cpp
+cpu_detect_x86_gcc.cpp
+""")
+
+Import('env')
+st = env.Copy()
+st.Append(CCFLAGS="-DHAVE_CONFIG_H -D_REENTRANT -D_LARGEFILE_SOURCE -D_LARGEFILE64_SOURCE")
+libst = st.StaticLibrary('soundtouch', soundtouch_files)
+Default(libst)
+
+env.Alias('tarball', env.Distribute (env['DISTTREE'],
+ [ 'SConscript'] + soundtouch_files + glob.glob('*.h')))
--- /dev/null
+////////////////////////////////////////////////////////////////////////////////
+///
+/// Common type definitions for SoundTouch audio processing library.
+///
+/// Author : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai @ iki.fi
+/// SoundTouch WWW: http://www.iki.fi/oparviai/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// Last changed : $Date$
+// File revision : $Revision$
+//
+// $Id$
+//
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+// SoundTouch audio processing library
+// Copyright (c) Olli Parviainen
+//
+// This library is free software; you can redistribute it and/or
+// modify it under the terms of the GNU Lesser General Public
+// License as published by the Free Software Foundation; either
+// version 2.1 of the License, or (at your option) any later version.
+//
+// This library is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+// Lesser General Public License for more details.
+//
+// You should have received a copy of the GNU Lesser General Public
+// License along with this library; if not, write to the Free Software
+// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#ifndef STTypes_H
+#define STTypes_H
+
+typedef unsigned int uint;
+typedef unsigned long ulong;
+
+#ifndef _WINDEF_
+ // if these aren't defined already by Windows headers, define now
+
+ typedef unsigned int BOOL;
+
+ #define FALSE 0
+ #define TRUE 1
+
+#endif // _WINDEF_
+
+
+namespace soundtouch
+{
+ /// Enable one of the following defines to choose either 16bit integer or
+ /// 32bit float sample type. If you don't have opinion, using integer samples
+ /// is generally faster.
+ /// #define INTEGER_SAMPLES //< 16bit integer samples
+ #define FLOAT_SAMPLES //< 32bit float samples
+
+
+ /// Define this to allow CPU-specific assembler optimizations. Notice that
+ /// having this enabled on non-x86 platforms doesn't matter; the compiler can
+ /// drop unsupported extensions on different platforms automatically.
+ /// However, if you're having difficulties getting the optimized routines
+ /// compiled with your compler (e.g. some gcc compiler versions may be picky),
+ /// you may wish to disable the optimizations to make the library compile.
+ #define ALLOW_OPTIMIZATIONS 1
+
+
+ #ifdef INTEGER_SAMPLES
+ // 16bit integer sample type
+ typedef short SAMPLETYPE;
+ // data type for sample accumulation: Use 32bit integer to prevent overflows
+ typedef long LONG_SAMPLETYPE;
+
+ #ifdef FLOAT_SAMPLES
+ // check that only one sample type is defined
+ #error "conflicting sample types defined"
+ #endif // FLOAT_SAMPLES
+
+ #ifdef ALLOW_OPTIMIZATIONS
+ #if WIN32 || __i386__
+ // Allow MMX optimizations
+ #define ALLOW_MMX 1
+ #endif
+ #endif
+
+ #else
+
+ // floating point samples
+ typedef float SAMPLETYPE;
+ // data type for sample accumulation: Use double to utilize full precision.
+ typedef double LONG_SAMPLETYPE;
+
+ #ifdef ALLOW_OPTIMIZATIONS
+ #ifdef WIN32
+ // Allow 3DNow! and SSE optimizations
+ #define ALLOW_3DNOW 1
+ #define ALLOW_SSE 1
+ #endif // WIN32
+ #endif
+
+ #endif // INTEGER_SAMPLES
+};
+
+#endif
--- /dev/null
+//////////////////////////////////////////////////////////////////////////////
+///
+/// SoundTouch - main class for tempo/pitch/rate adjusting routines.
+///
+/// Notes:
+/// - Initialize the SoundTouch object instance by setting up the sound stream
+/// parameters with functions 'setSampleRate' and 'setChannels', then set
+/// desired tempo/pitch/rate settings with the corresponding functions.
+///
+/// - The SoundTouch class behaves like a first-in-first-out pipeline: The
+/// samples that are to be processed are fed into one of the pipe by calling
+/// function 'putSamples', while the ready processed samples can be read
+/// from the other end of the pipeline with function 'receiveSamples'.
+///
+/// - The SoundTouch processing classes require certain sized 'batches' of
+/// samples in order to process the sound. For this reason the classes buffer
+/// incoming samples until there are enough of samples available for
+/// processing, then they carry out the processing step and consequently
+/// make the processed samples available for outputting.
+///
+/// - For the above reason, the processing routines introduce a certain
+/// 'latency' between the input and output, so that the samples input to
+/// SoundTouch may not be immediately available in the output, and neither
+/// the amount of outputtable samples may not immediately be in direct
+/// relationship with the amount of previously input samples.
+///
+/// - The tempo/pitch/rate control parameters can be altered during processing.
+/// Please notice though that they aren't currently protected by semaphores,
+/// so in multi-thread application external semaphore protection may be
+/// required.
+///
+/// - This class utilizes classes 'TDStretch' for tempo change (without modifying
+/// pitch) and 'RateTransposer' for changing the playback rate (that is, both
+/// tempo and pitch in the same ratio) of the sound. The third available control
+/// 'pitch' (change pitch but maintain tempo) is produced by a combination of
+/// combining the two other controls.
+///
+/// Author : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai @ iki.fi
+/// SoundTouch WWW: http://www.iki.fi/oparviai/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// Last changed : $Date$
+// File revision : $Revision$
+//
+// $Id$
+//
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+// SoundTouch audio processing library
+// Copyright (c) Olli Parviainen
+//
+// This library is free software; you can redistribute it and/or
+// modify it under the terms of the GNU Lesser General Public
+// License as published by the Free Software Foundation; either
+// version 2.1 of the License, or (at your option) any later version.
+//
+// This library is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+// Lesser General Public License for more details.
+//
+// You should have received a copy of the GNU Lesser General Public
+// License along with this library; if not, write to the Free Software
+// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#include <assert.h>
+#include <stdlib.h>
+#include <memory.h>
+#include <math.h>
+#include <stdexcept>
+#include <stdio.h>
+
+#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 = RateTransposer::newInstance();
+ 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();
+}
+
+
+
+/// Returns number of samples currently unprocessed.
+uint SoundTouch::numUnprocessedSamples() const
+{
+ FIFOSamplePipe * psp;
+ if (pTDStretch)
+ {
+ psp = pTDStretch->getInput();
+ if (psp)
+ {
+ return psp->numSamples();
+ }
+ }
+ return 0;
+}
--- /dev/null
+//////////////////////////////////////////////////////////////////////////////
+///
+/// SoundTouch - main class for tempo/pitch/rate adjusting routines.
+///
+/// Notes:
+/// - Initialize the SoundTouch object instance by setting up the sound stream
+/// parameters with functions 'setSampleRate' and 'setChannels', then set
+/// desired tempo/pitch/rate settings with the corresponding functions.
+///
+/// - The SoundTouch class behaves like a first-in-first-out pipeline: The
+/// samples that are to be processed are fed into one of the pipe by calling
+/// function 'putSamples', while the ready processed samples can be read
+/// from the other end of the pipeline with function 'receiveSamples'.
+///
+/// - The SoundTouch processing classes require certain sized 'batches' of
+/// samples in order to process the sound. For this reason the classes buffer
+/// incoming samples until there are enough of samples available for
+/// processing, then they carry out the processing step and consequently
+/// make the processed samples available for outputting.
+///
+/// - For the above reason, the processing routines introduce a certain
+/// 'latency' between the input and output, so that the samples input to
+/// SoundTouch may not be immediately available in the output, and neither
+/// the amount of outputtable samples may not immediately be in direct
+/// relationship with the amount of previously input samples.
+///
+/// - The tempo/pitch/rate control parameters can be altered during processing.
+/// Please notice though that they aren't currently protected by semaphores,
+/// so in multi-thread application external semaphore protection may be
+/// required.
+///
+/// - This class utilizes classes 'TDStretch' for tempo change (without modifying
+/// pitch) and 'RateTransposer' for changing the playback rate (that is, both
+/// tempo and pitch in the same ratio) of the sound. The third available control
+/// 'pitch' (change pitch but maintain tempo) is produced by a combination of
+/// combining the two other controls.
+///
+/// Author : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai @ iki.fi
+/// SoundTouch WWW: http://www.iki.fi/oparviai/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// Last changed : $Date$
+// File revision : $Revision$
+//
+// $Id$
+//
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+// SoundTouch audio processing library
+// Copyright (c) Olli Parviainen
+//
+// This library is free software; you can redistribute it and/or
+// modify it under the terms of the GNU Lesser General Public
+// License as published by the Free Software Foundation; either
+// version 2.1 of the License, or (at your option) any later version.
+//
+// This library is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+// Lesser General Public License for more details.
+//
+// You should have received a copy of the GNU Lesser General Public
+// License along with this library; if not, write to the Free Software
+// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#ifndef SoundTouch_H
+#define SoundTouch_H
+
+#include "FIFOSamplePipe.h"
+#include "STTypes.h"
+
+namespace soundtouch
+{
+
+/// Soundtouch library version string
+#define SOUNDTOUCH_VERSION "1.3.0"
+
+/// SoundTouch library version id
+#define SOUNDTOUCH_VERSION_ID 010300
+
+//
+// Available setting IDs for the 'setSetting' & 'get_setting' functions:
+
+/// Enable/disable anti-alias filter in pitch transposer (0 = disable)
+#define SETTING_USE_AA_FILTER 0
+
+/// Pitch transposer anti-alias filter length (8 .. 128 taps, default = 32)
+#define SETTING_AA_FILTER_LENGTH 1
+
+/// Enable/disable quick seeking algorithm in tempo changer routine
+/// (enabling quick seeking lowers CPU utilization but causes a minor sound
+/// quality compromising)
+#define SETTING_USE_QUICKSEEK 2
+
+/// Time-stretch algorithm single processing sequence length in milliseconds. This determines
+/// to how long sequences the original sound is chopped in the time-stretch algorithm.
+/// See "STTypes.h" or README for more information.
+#define SETTING_SEQUENCE_MS 3
+
+/// Time-stretch algorithm seeking window length in milliseconds for algorithm that finds the
+/// best possible overlapping location. This determines from how wide window the algorithm
+/// may look for an optimal joining location when mixing the sound sequences back together.
+/// See "STTypes.h" or README for more information.
+#define SETTING_SEEKWINDOW_MS 4
+
+/// Time-stretch algorithm overlap length in milliseconds. When the chopped sound sequences
+/// are mixed back together, to form a continuous sound stream, this parameter defines over
+/// how long period the two consecutive sequences are let to overlap each other.
+/// See "STTypes.h" or README for more information.
+#define SETTING_OVERLAP_MS 5
+
+
+class SoundTouch : public FIFOProcessor
+{
+private:
+ /// Rate transposer class instance
+ class RateTransposer *pRateTransposer;
+
+ /// Time-stretch class instance
+ class TDStretch *pTDStretch;
+
+ /// Virtual pitch parameter. Effective rate & tempo are calculated from these parameters.
+ float virtualRate;
+
+ /// Virtual pitch parameter. Effective rate & tempo are calculated from these parameters.
+ float virtualTempo;
+
+ /// Virtual pitch parameter. Effective rate & tempo are calculated from these parameters.
+ float virtualPitch;
+
+ /// Flag: Has sample rate been set?
+ BOOL bSrateSet;
+
+ /// Calculates effective rate & tempo valuescfrom 'virtualRate', 'virtualTempo' and
+ /// 'virtualPitch' parameters.
+ void calcEffectiveRateAndTempo();
+
+protected :
+ /// Number of channels
+ uint channels;
+
+ /// Effective 'rate' value calculated from 'virtualRate', 'virtualTempo' and 'virtualPitch'
+ float rate;
+
+ /// Effective 'tempo' value calculated from 'virtualRate', 'virtualTempo' and 'virtualPitch'
+ float tempo;
+
+public:
+ SoundTouch();
+ virtual ~SoundTouch();
+
+ /// Get SoundTouch library version string
+ static const char *getVersionString();
+
+ /// Get SoundTouch library version Id
+ static uint SoundTouch::getVersionId();
+
+ /// Sets new rate control value. Normal rate = 1.0, smaller values
+ /// represent slower rate, larger faster rates.
+ void setRate(float newRate);
+
+ /// Sets new tempo control value. Normal tempo = 1.0, smaller values
+ /// represent slower tempo, larger faster tempo.
+ void setTempo(float newTempo);
+
+ /// Sets new rate control value as a difference in percents compared
+ /// to the original rate (-50 .. +100 %)
+ void setRateChange(float newRate);
+
+ /// Sets new tempo control value as a difference in percents compared
+ /// to the original tempo (-50 .. +100 %)
+ void setTempoChange(float newTempo);
+
+ /// Sets new pitch control value. Original pitch = 1.0, smaller values
+ /// represent lower pitches, larger values higher pitch.
+ void setPitch(float newPitch);
+
+ /// Sets pitch change in octaves compared to the original pitch
+ /// (-1.00 .. +1.00)
+ void setPitchOctaves(float newPitch);
+
+ /// Sets pitch change in semi-tones compared to the original pitch
+ /// (-12 .. +12)
+ void setPitchSemiTones(int newPitch);
+ void setPitchSemiTones(float newPitch);
+
+ /// Sets the number of channels, 1 = mono, 2 = stereo
+ void setChannels(uint numChannels);
+
+ /// Sets sample rate.
+ void setSampleRate(uint srate);
+
+ /// 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 flush();
+
+ /// Adds 'numSamples' pcs of samples from the 'samples' memory position into
+ /// the input of the object. Notice that sample rate _has_to_ be set before
+ /// calling this function, otherwise throws a runtime_error exception.
+ virtual void putSamples(
+ const SAMPLETYPE *samples, ///< Pointer to sample buffer.
+ uint numSamples ///< Number of samples in buffer. Notice
+ ///< that in case of stereo-sound a single sample
+ ///< contains data for both channels.
+ );
+
+ /// Clears all the samples in the object's output and internal processing
+ /// buffers.
+ virtual void clear();
+
+ /// Changes a setting controlling the processing system behaviour. See the
+ /// 'SETTING_...' defines for available setting ID's.
+ ///
+ /// \return 'TRUE' if the setting was succesfully changed
+ BOOL setSetting(uint settingId, ///< Setting ID number. see SETTING_... defines.
+ uint value ///< New setting value.
+ );
+
+ /// Reads a setting controlling the processing system behaviour. See the
+ /// 'SETTING_...' defines for available setting ID's.
+ ///
+ /// \return the setting value.
+ uint getSetting(uint settingId ///< Setting ID number, see SETTING_... defines.
+ ) const;
+
+ /// Returns number of samples currently unprocessed.
+ virtual uint numUnprocessedSamples() const;
+
+
+ /// Other handy functions that are implemented in the ancestor classes (see
+ /// classes 'FIFOProcessor' and 'FIFOSamplePipe')
+ ///
+ /// - receiveSamples() : Use this function to receive 'ready' processed samples from SoundTouch.
+ /// - numSamples() : Get number of 'ready' samples that can be received with
+ /// function 'receiveSamples()'
+ /// - isEmpty() : Returns nonzero if there aren't any 'ready' samples.
+ /// - clear() : Clears all samples from ready/processing buffers.
+};
+
+}
+#endif
--- /dev/null
+////////////////////////////////////////////////////////////////////////////////
+///
+/// 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
+/// SoundTouch WWW: http://www.iki.fi/oparviai/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// Last changed : $Date$
+// File revision : $Revision$
+//
+// $Id$
+//
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+// SoundTouch audio processing library
+// Copyright (c) Olli Parviainen
+//
+// This library is free software; you can redistribute it and/or
+// modify it under the terms of the GNU Lesser General Public
+// License as published by the Free Software Foundation; either
+// version 2.1 of the License, or (at your option) any later version.
+//
+// This library is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+// Lesser General Public License for more details.
+//
+// You should have received a copy of the GNU Lesser General Public
+// License along with this library; if not, write to the Free Software
+// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#include <string.h>
+#include <stdlib.h>
+#include <memory.h>
+#include <limits.h>
+#include <math.h>
+#include <assert.h>
+
+#include "STTypes.h"
+#include "cpu_detect.h"
+#include "TDStretch.h"
+
+using namespace soundtouch;
+
+#ifndef min
+#define min(a,b) ((a > b) ? b : a)
+#define max(a,b) ((a < b) ? b : a)
+#endif
+
+
+
+/*****************************************************************************
+ *
+ * Constant definitions
+ *
+ *****************************************************************************/
+
+
+#define MAX_SCAN_DELTA 124
+
+// Table for the hierarchical mixing position seeking algorithm
+int scanOffsets[4][24]={
+ { 124, 186, 248, 310, 372, 434, 496, 558, 620, 682, 744, 806,
+ 868, 930, 992, 1054, 1116, 1178, 1240, 1302, 1364, 1426, 1488, 0},
+ {-100, -75, -50, -25, 25, 50, 75, 100, 0, 0, 0, 0,
+ 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0},
+ { -20, -15, -10, -5, 5, 10, 15, 20, 0, 0, 0, 0,
+ 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0},
+ { -4, -3, -2, -1, 1, 2, 3, 4, 0, 0, 0, 0,
+ 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0}};
+
+/*****************************************************************************
+ *
+ * Implementation of the class 'TDStretch'
+ *
+ *****************************************************************************/
+
+
+TDStretch::TDStretch() : FIFOProcessor(&outputBuffer)
+{
+ bQuickseek = FALSE;
+ channels = 2;
+ bMidBufferDirty = FALSE;
+
+ pMidBuffer = NULL;
+ pRefMidBufferUnaligned = NULL;
+ overlapLength = 0;
+
+ setParameters(44100, DEFAULT_SEQUENCE_MS, DEFAULT_SEEKWINDOW_MS, DEFAULT_OVERLAP_MS);
+
+ setTempo(1.0f);
+}
+
+
+
+
+TDStretch::~TDStretch()
+{
+ delete[] pMidBuffer;
+ delete[] pRefMidBufferUnaligned;
+}
+
+
+
+// Calculates the x having the closest 2^x value for the given value
+static int _getClosest2Power(double value)
+{
+ return (int)(log(value) / log(2.0) + 0.5);
+}
+
+
+
+// Sets routine control parameters. These control are certain time constants
+// defining how the sound is stretched to the desired duration.
+//
+// 'sampleRate' = sample rate of the sound
+// 'sequenceMS' = one processing sequence length in milliseconds (default = 82 ms)
+// 'seekwindowMS' = seeking window length for scanning the best overlapping
+// position (default = 28 ms)
+// 'overlapMS' = overlapping length (default = 12 ms)
+
+void TDStretch::setParameters(uint aSampleRate, uint aSequenceMS,
+ uint aSeekWindowMS, uint aOverlapMS)
+{
+ this->sampleRate = aSampleRate;
+ this->sequenceMs = aSequenceMS;
+ this->seekWindowMs = aSeekWindowMS;
+ this->overlapMs = aOverlapMS;
+
+ seekLength = (sampleRate * seekWindowMs) / 1000;
+ seekWindowLength = (sampleRate * sequenceMs) / 1000;
+
+ maxOffset = seekLength;
+
+ calculateOverlapLength(overlapMs);
+
+ // set tempo to recalculate 'sampleReq'
+ setTempo(tempo);
+
+}
+
+
+
+/// Get routine control parameters, see setParameters() function.
+/// Any of the parameters to this function can be NULL, in such case corresponding parameter
+/// value isn't returned.
+void TDStretch::getParameters(uint *pSampleRate, uint *pSequenceMs, uint *pSeekWindowMs, uint *pOverlapMs)
+{
+ if (pSampleRate)
+ {
+ *pSampleRate = sampleRate;
+ }
+
+ if (pSequenceMs)
+ {
+ *pSequenceMs = sequenceMs;
+ }
+
+ if (pSeekWindowMs)
+ {
+ *pSeekWindowMs = seekWindowMs;
+ }
+
+ if (pOverlapMs)
+ {
+ *pOverlapMs = overlapMs;
+ }
+}
+
+
+// Overlaps samples in 'midBuffer' with the samples in 'input'
+void TDStretch::overlapMono(SAMPLETYPE *output, const SAMPLETYPE *input) const
+{
+ int i, itemp;
+
+ for (i = 0; i < (int)overlapLength ; i ++)
+ {
+ itemp = overlapLength - i;
+ output[i] = (input[i] * i + pMidBuffer[i] * itemp ) / overlapLength; // >> overlapDividerBits;
+ }
+}
+
+
+
+void TDStretch::clearMidBuffer()
+{
+ if (bMidBufferDirty)
+ {
+ memset(pMidBuffer, 0, 2 * sizeof(SAMPLETYPE) * overlapLength);
+ bMidBufferDirty = FALSE;
+ }
+}
+
+
+void TDStretch::clearInput()
+{
+ inputBuffer.clear();
+ clearMidBuffer();
+}
+
+
+// Clears the sample buffers
+void TDStretch::clear()
+{
+ outputBuffer.clear();
+ inputBuffer.clear();
+ clearMidBuffer();
+}
+
+
+
+// Enables/disables the quick position seeking algorithm. Zero to disable, nonzero
+// to enable
+void TDStretch::enableQuickSeek(BOOL enable)
+{
+ bQuickseek = enable;
+}
+
+
+// Returns nonzero if the quick seeking algorithm is enabled.
+BOOL TDStretch::isQuickSeekEnabled() const
+{
+ return bQuickseek;
+}
+
+
+// Seeks for the optimal overlap-mixing position.
+uint TDStretch::seekBestOverlapPosition(const SAMPLETYPE *refPos)
+{
+ if (channels == 2)
+ {
+ // stereo sound
+ if (bQuickseek)
+ {
+ return seekBestOverlapPositionStereoQuick(refPos);
+ }
+ else
+ {
+ return seekBestOverlapPositionStereo(refPos);
+ }
+ }
+ else
+ {
+ // mono sound
+ if (bQuickseek)
+ {
+ return seekBestOverlapPositionMonoQuick(refPos);
+ }
+ else
+ {
+ return seekBestOverlapPositionMono(refPos);
+ }
+ }
+}
+
+
+
+
+// Overlaps samples in 'midBuffer' with the samples in 'inputBuffer' at position
+// of 'ovlPos'.
+inline void TDStretch::overlap(SAMPLETYPE *output, const SAMPLETYPE *input, uint ovlPos) const
+{
+ if (channels == 2)
+ {
+ // stereo sound
+ overlapStereo(output, input + 2 * ovlPos);
+ } else {
+ // mono sound.
+ overlapMono(output, input + ovlPos);
+ }
+}
+
+
+
+
+// Seeks for the optimal overlap-mixing position. The 'stereo' version of the
+// routine
+//
+// The best position is determined as the position where the two overlapped
+// sample sequences are 'most alike', in terms of the highest cross-correlation
+// value over the overlapping period
+uint TDStretch::seekBestOverlapPositionStereo(const SAMPLETYPE *refPos)
+{
+ uint bestOffs;
+ LONG_SAMPLETYPE bestCorr, corr;
+ uint i;
+
+ // Slopes the amplitudes of the 'midBuffer' samples
+ precalcCorrReferenceStereo();
+
+ bestCorr = INT_MIN;
+ bestOffs = 0;
+
+ // Scans for the best correlation value by testing each possible position
+ // over the permitted range.
+ for (i = 0; i < seekLength; i ++)
+ {
+ // Calculates correlation value for the mixing position corresponding
+ // to 'i'
+ corr = calcCrossCorrStereo(refPos + 2 * i, pRefMidBuffer);
+
+ // Checks for the highest correlation value
+ if (corr > bestCorr)
+ {
+ bestCorr = corr;
+ bestOffs = i;
+ }
+ }
+ // clear cross correlation routine state if necessary (is so e.g. in MMX routines).
+ clearCrossCorrState();
+
+ return bestOffs;
+}
+
+
+// Seeks for the optimal overlap-mixing position. The 'stereo' version of the
+// routine
+//
+// The best position is determined as the position where the two overlapped
+// sample sequences are 'most alike', in terms of the highest cross-correlation
+// value over the overlapping period
+uint TDStretch::seekBestOverlapPositionStereoQuick(const SAMPLETYPE *refPos)
+{
+ uint j;
+ uint bestOffs;
+ LONG_SAMPLETYPE bestCorr, corr;
+ uint scanCount, corrOffset, tempOffset;
+
+ // Slopes the amplitude of the 'midBuffer' samples
+ precalcCorrReferenceStereo();
+
+ bestCorr = INT_MIN;
+ bestOffs = 0;
+ corrOffset = 0;
+ tempOffset = 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 ++)
+ {
+ j = 0;
+ while (scanOffsets[scanCount][j])
+ {
+ tempOffset = corrOffset + scanOffsets[scanCount][j];
+ if (tempOffset >= seekLength) break;
+
+ // Calculates correlation value for the mixing position corresponding
+ // to 'tempOffset'
+ corr = calcCrossCorrStereo(refPos + 2 * tempOffset, pRefMidBuffer);
+
+ // Checks for the highest correlation value
+ if (corr > bestCorr)
+ {
+ bestCorr = corr;
+ bestOffs = tempOffset;
+ }
+ j ++;
+ }
+ corrOffset = bestOffs;
+ }
+ // clear cross correlation routine state if necessary (is so e.g. in MMX routines).
+ clearCrossCorrState();
+
+ return bestOffs;
+}
+
+
+
+// Seeks for the optimal overlap-mixing position. The 'mono' version of the
+// routine
+//
+// The best position is determined as the position where the two overlapped
+// sample sequences are 'most alike', in terms of the highest cross-correlation
+// value over the overlapping period
+uint TDStretch::seekBestOverlapPositionMono(const SAMPLETYPE *refPos)
+{
+ uint bestOffs;
+ LONG_SAMPLETYPE bestCorr, corr;
+ uint tempOffset;
+ const SAMPLETYPE *compare;
+
+ // Slopes the amplitude of the 'midBuffer' samples
+ precalcCorrReferenceMono();
+
+ bestCorr = INT_MIN;
+ bestOffs = 0;
+
+ // Scans for the best correlation value by testing each possible position
+ // over the permitted range.
+ for (tempOffset = 0; tempOffset < seekLength; tempOffset ++)
+ {
+ compare = refPos + tempOffset;
+
+ // Calculates correlation value for the mixing position corresponding
+ // to 'tempOffset'
+ corr = calcCrossCorrMono(pRefMidBuffer, compare);
+
+ // Checks for the highest correlation value
+ if (corr > bestCorr)
+ {
+ bestCorr = corr;
+ bestOffs = tempOffset;
+ }
+ }
+ // clear cross correlation routine state if necessary (is so e.g. in MMX routines).
+ clearCrossCorrState();
+
+ return bestOffs;
+}
+
+
+// Seeks for the optimal overlap-mixing position. The 'mono' version of the
+// routine
+//
+// The best position is determined as the position where the two overlapped
+// sample sequences are 'most alike', in terms of the highest cross-correlation
+// value over the overlapping period
+uint TDStretch::seekBestOverlapPositionMonoQuick(const SAMPLETYPE *refPos)
+{
+ uint j;
+ uint bestOffs;
+ LONG_SAMPLETYPE bestCorr, corr;
+ uint scanCount, corrOffset, tempOffset;
+
+ // Slopes the amplitude of the 'midBuffer' samples
+ precalcCorrReferenceMono();
+
+ bestCorr = INT_MIN;
+ bestOffs = 0;
+ corrOffset = 0;
+ tempOffset = 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 ++)
+ {
+ j = 0;
+ while (scanOffsets[scanCount][j])
+ {
+ tempOffset = corrOffset + scanOffsets[scanCount][j];
+ if (tempOffset >= seekLength) break;
+
+ // Calculates correlation value for the mixing position corresponding
+ // to 'tempOffset'
+ corr = calcCrossCorrMono(refPos + tempOffset, pRefMidBuffer);
+
+ // Checks for the highest correlation value
+ if (corr > bestCorr)
+ {
+ bestCorr = corr;
+ bestOffs = tempOffset;
+ }
+ j ++;
+ }
+ corrOffset = bestOffs;
+ }
+ // clear cross correlation routine state if necessary (is so e.g. in MMX routines).
+ clearCrossCorrState();
+
+ return bestOffs;
+}
+
+
+/// clear cross correlation routine state if necessary
+void TDStretch::clearCrossCorrState()
+{
+ // default implementation is empty.
+}
+
+
+// Sets new target tempo. Normal tempo = 'SCALE', smaller values represent slower
+// tempo, larger faster tempo.
+void TDStretch::setTempo(float newTempo)
+{
+ uint intskip;
+
+ tempo = newTempo;
+
+ // Calculate ideal skip length (according to tempo value)
+ nominalSkip = tempo * (seekWindowLength - overlapLength);
+ skipFract = 0;
+ intskip = (int)(nominalSkip + 0.5f);
+
+ // Calculate how many samples are needed in the 'inputBuffer' to
+ // process another batch of samples
+ sampleReq = max(intskip + overlapLength, seekWindowLength) + maxOffset;
+}
+
+
+
+// Sets the number of channels, 1 = mono, 2 = stereo
+void TDStretch::setChannels(uint numChannels)
+{
+ if (channels == numChannels) return;
+ assert(numChannels == 1 || numChannels == 2);
+
+ channels = numChannels;
+ inputBuffer.setChannels(channels);
+ outputBuffer.setChannels(channels);
+}
+
+
+// nominal tempo, no need for processing, just pass the samples through
+// to outputBuffer
+void TDStretch::processNominalTempo()
+{
+ assert(tempo == 1.0f);
+
+ if (bMidBufferDirty)
+ {
+ // If there are samples in pMidBuffer waiting for overlapping,
+ // do a single sliding overlapping with them in order to prevent a
+ // clicking distortion in the output sound
+ if (inputBuffer.numSamples() < overlapLength)
+ {
+ // wait until we've got overlapLength input samples
+ return;
+ }
+ // Mix the samples in the beginning of 'inputBuffer' with the
+ // samples in 'midBuffer' using sliding overlapping
+ overlap(outputBuffer.ptrEnd(overlapLength), inputBuffer.ptrBegin(), 0);
+ outputBuffer.putSamples(overlapLength);
+ inputBuffer.receiveSamples(overlapLength);
+ clearMidBuffer();
+ // now we've caught the nominal sample flow and may switch to
+ // bypass mode
+ }
+
+ // Simply bypass samples from input to output
+ outputBuffer.moveSamples(inputBuffer);
+}
+
+
+// Processes as many processing frames of the samples 'inputBuffer', store
+// the result into 'outputBuffer'
+void TDStretch::processSamples()
+{
+ uint ovlSkip, offset;
+ int temp;
+
+ if (tempo == 1.0f)
+ {
+ // tempo not changed from the original, so bypass the processing
+ processNominalTempo();
+ return;
+ }
+
+ if (bMidBufferDirty == FALSE)
+ {
+ // if midBuffer is empty, move the first samples of the input stream
+ // into it
+ if (inputBuffer.numSamples() < overlapLength)
+ {
+ // wait until we've got overlapLength samples
+ return;
+ }
+ memcpy(pMidBuffer, inputBuffer.ptrBegin(), channels * overlapLength * sizeof(SAMPLETYPE));
+ inputBuffer.receiveSamples(overlapLength);
+ bMidBufferDirty = TRUE;
+ }
+
+ // Process samples as long as there are enough samples in 'inputBuffer'
+ // to form a processing frame.
+ while (inputBuffer.numSamples() >= sampleReq)
+ {
+ // If tempo differs from the normal ('SCALE'), scan for the best overlapping
+ // position
+ offset = seekBestOverlapPosition(inputBuffer.ptrBegin());
+
+ // Mix the samples in the 'inputBuffer' at position of 'offset' with the
+ // samples in 'midBuffer' using sliding overlapping
+ // ... first partially overlap with the end of the previous sequence
+ // (that's in 'midBuffer')
+ overlap(outputBuffer.ptrEnd(overlapLength), inputBuffer.ptrBegin(), offset);
+ outputBuffer.putSamples(overlapLength);
+
+ // ... then copy sequence samples from 'inputBuffer' to output
+ temp = (seekWindowLength - 2 * overlapLength);// & 0xfffffffe;
+ if (temp > 0)
+ {
+ outputBuffer.putSamples(inputBuffer.ptrBegin() + channels * (offset + overlapLength), temp);
+ }
+
+ // Copies the end of the current sequence from 'inputBuffer' to
+ // 'midBuffer' for being mixed with the beginning of the next
+ // processing sequence and so on
+ assert(offset + seekWindowLength <= inputBuffer.numSamples());
+ memcpy(pMidBuffer, inputBuffer.ptrBegin() + channels * (offset + seekWindowLength - overlapLength),
+ channels * sizeof(SAMPLETYPE) * overlapLength);
+ bMidBufferDirty = TRUE;
+
+ // Remove the processed samples from the input buffer. Update
+ // the difference between integer & nominal skip step to 'skipFract'
+ // in order to prevent the error from accumulating over time.
+ skipFract += nominalSkip; // real skip size
+ ovlSkip = (int)skipFract; // rounded to integer skip
+ skipFract -= ovlSkip; // maintain the fraction part, i.e. real vs. integer skip
+ inputBuffer.receiveSamples(ovlSkip);
+ }
+}
+
+
+// Adds 'numsamples' pcs of samples from the 'samples' memory position into
+// the input of the object.
+void TDStretch::putSamples(const SAMPLETYPE *samples, uint numSamples)
+{
+ // Add the samples into the input buffer
+ inputBuffer.putSamples(samples, numSamples);
+ // Process the samples in input buffer
+ processSamples();
+}
+
+
+
+/// Set new overlap length parameter & reallocate RefMidBuffer if necessary.
+void TDStretch::acceptNewOverlapLength(uint newOverlapLength)
+{
+ uint prevOvl;
+
+ prevOvl = overlapLength;
+ overlapLength = newOverlapLength;
+
+ if (overlapLength > prevOvl)
+ {
+ delete[] pMidBuffer;
+ delete[] pRefMidBufferUnaligned;
+
+ pMidBuffer = new SAMPLETYPE[overlapLength * 2];
+ bMidBufferDirty = TRUE;
+ clearMidBuffer();
+
+ pRefMidBufferUnaligned = new SAMPLETYPE[2 * overlapLength + 16 / sizeof(SAMPLETYPE)];
+ // ensure that 'pRefMidBuffer' is aligned to 16 byte boundary for efficiency
+ pRefMidBuffer = (SAMPLETYPE *)((((ulong)pRefMidBufferUnaligned) + 15) & -16);
+ }
+}
+
+TDStretch * TDStretch::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 TDStretchMMX;
+ }
+ else
+#endif // ALLOW_MMX
+
+
+#ifdef ALLOW_SSE
+ if (uExtensions & SUPPORT_SSE)
+ {
+ // SSE support
+ return ::new TDStretchSSE;
+ }
+ else
+#endif // ALLOW_SSE
+
+
+#ifdef ALLOW_3DNOW
+ if (uExtensions & SUPPORT_3DNOW)
+ {
+ // 3DNow! support
+ return ::new TDStretch3DNow;
+ }
+ else
+#endif // ALLOW_3DNOW
+
+ {
+ // ISA optimizations not supported, use plain C version
+ return ::new TDStretch;
+ }
+}
+
+
+//////////////////////////////////////////////////////////////////////////////
+//
+// Integer arithmetics specific algorithm implementations.
+//
+//////////////////////////////////////////////////////////////////////////////
+
+#ifdef INTEGER_SAMPLES
+
+// Slopes the amplitude of the 'midBuffer' samples so that cross correlation
+// is faster to calculate
+void TDStretch::precalcCorrReferenceStereo()
+{
+ int i, cnt2;
+ int temp, temp2;
+
+ for (i=0 ; i < (int)overlapLength ;i ++)
+ {
+ temp = i * (overlapLength - i);
+ cnt2 = i * 2;
+
+ temp2 = (pMidBuffer[cnt2] * temp) / slopingDivider;
+ pRefMidBuffer[cnt2] = (short)(temp2);
+ temp2 = (pMidBuffer[cnt2 + 1] * temp) / slopingDivider;
+ pRefMidBuffer[cnt2 + 1] = (short)(temp2);
+ }
+}
+
+
+// Slopes the amplitude of the 'midBuffer' samples so that cross correlation
+// is faster to calculate
+void TDStretch::precalcCorrReferenceMono()
+{
+ int i;
+ long temp;
+ long temp2;
+
+ for (i=0 ; i < (int)overlapLength ;i ++)
+ {
+ temp = i * (overlapLength - i);
+ temp2 = (pMidBuffer[i] * temp) / slopingDivider;
+ pRefMidBuffer[i] = (short)temp2;
+ }
+}
+
+
+// Overlaps samples in 'midBuffer' with the samples in 'input'. The 'Stereo'
+// version of the routine.
+void TDStretch::overlapStereo(short *output, const short *input) const
+{
+ int i;
+ short temp;
+ uint cnt2;
+
+ for (i = 0; i < (int)overlapLength ; i ++)
+ {
+ temp = (short)(overlapLength - i);
+ cnt2 = 2 * i;
+ output[cnt2] = (input[cnt2] * i + pMidBuffer[cnt2] * temp ) / overlapLength;
+ output[cnt2 + 1] = (input[cnt2 + 1] * i + pMidBuffer[cnt2 + 1] * temp ) / overlapLength;
+ }
+}
+
+
+/// Calculates overlap period length in samples.
+/// Integer version rounds overlap length to closest power of 2
+/// for a divide scaling operation.
+void TDStretch::calculateOverlapLength(uint overlapMs)
+{
+ uint newOvl;
+
+ overlapDividerBits = _getClosest2Power((sampleRate * overlapMs) / 1000.0);
+ if (overlapDividerBits > 9) overlapDividerBits = 9;
+ if (overlapDividerBits < 4) overlapDividerBits = 4;
+ newOvl = (uint)pow(2, overlapDividerBits);
+
+ acceptNewOverlapLength(newOvl);
+
+ // calculate sloping divider so that crosscorrelation operation won't
+ // overflow 32-bit register. Max. sum of the crosscorrelation sum without
+ // divider would be 2^30*(N^3-N)/3, where N = overlap length
+ slopingDivider = (newOvl * newOvl - 1) / 3;
+}
+
+
+long TDStretch::calcCrossCorrMono(const short *mixingPos, const short *compare) const
+{
+ long corr;
+ uint i;
+
+ corr = 0;
+ for (i = 1; i < overlapLength; i ++)
+ {
+ corr += (mixingPos[i] * compare[i]) >> overlapDividerBits;
+ }
+
+ return corr;
+}
+
+
+long TDStretch::calcCrossCorrStereo(const short *mixingPos, const short *compare) const
+{
+ long corr;
+ uint i;
+
+ corr = 0;
+ for (i = 2; i < 2 * overlapLength; i += 2)
+ {
+ corr += (mixingPos[i] * compare[i] +
+ mixingPos[i + 1] * compare[i + 1]) >> overlapDividerBits;
+ }
+
+ return corr;
+}
+
+#endif // INTEGER_SAMPLES
+
+//////////////////////////////////////////////////////////////////////////////
+//
+// Floating point arithmetics specific algorithm implementations.
+//
+
+#ifdef FLOAT_SAMPLES
+
+
+// Slopes the amplitude of the 'midBuffer' samples so that cross correlation
+// is faster to calculate
+void TDStretch::precalcCorrReferenceStereo()
+{
+ int i, cnt2;
+ float temp;
+
+ for (i=0 ; i < (int)overlapLength ;i ++)
+ {
+ temp = (float)i * (float)(overlapLength - i);
+ cnt2 = i * 2;
+ pRefMidBuffer[cnt2] = (float)(pMidBuffer[cnt2] * temp);
+ pRefMidBuffer[cnt2 + 1] = (float)(pMidBuffer[cnt2 + 1] * temp);
+ }
+}
+
+
+// Slopes the amplitude of the 'midBuffer' samples so that cross correlation
+// is faster to calculate
+void TDStretch::precalcCorrReferenceMono()
+{
+ int i;
+ float temp;
+
+ for (i=0 ; i < (int)overlapLength ;i ++)
+ {
+ temp = (float)i * (float)(overlapLength - i);
+ pRefMidBuffer[i] = (float)(pMidBuffer[i] * temp);
+ }
+}
+
+
+// SSE-optimized version of the function overlapStereo
+void TDStretch::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;
+ }
+}
+
+
+/// Calculates overlap period length in samples.
+void TDStretch::calculateOverlapLength(uint overlapMs)
+{
+ uint newOvl;
+
+ newOvl = (sampleRate * overlapMs) / 1000;
+ if (newOvl < 16) newOvl = 16;
+
+ acceptNewOverlapLength(newOvl);
+}
+
+
+
+double TDStretch::calcCrossCorrMono(const float *mixingPos, const float *compare) const
+{
+ double corr;
+ uint i;
+
+ corr = 0;
+ for (i = 1; i < overlapLength; i ++)
+ {
+ corr += mixingPos[i] * compare[i];
+ }
+
+ return corr;
+}
+
+
+double TDStretch::calcCrossCorrStereo(const float *mixingPos, const float *compare) const
+{
+ double corr;
+ uint i;
+
+ corr = 0;
+ for (i = 2; i < 2 * overlapLength; i += 2)
+ {
+ corr += mixingPos[i] * compare[i] +
+ mixingPos[i + 1] * compare[i + 1];
+ }
+
+ return corr;
+}
+
+#endif // FLOAT_SAMPLES
--- /dev/null
+////////////////////////////////////////////////////////////////////////////////
+///
+/// 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
+/// SoundTouch WWW: http://www.iki.fi/oparviai/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// Last changed : $Date$
+// File revision : $Revision$
+//
+// $Id$
+//
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+// SoundTouch audio processing library
+// Copyright (c) Olli Parviainen
+//
+// This library is free software; you can redistribute it and/or
+// modify it under the terms of the GNU Lesser General Public
+// License as published by the Free Software Foundation; either
+// version 2.1 of the License, or (at your option) any later version.
+//
+// This library is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+// Lesser General Public License for more details.
+//
+// You should have received a copy of the GNU Lesser General Public
+// License along with this library; if not, write to the Free Software
+// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#ifndef TDStretch_H
+#define TDStretch_H
+
+#include "STTypes.h"
+#include "RateTransposer.h"
+#include "FIFOSamplePipe.h"
+
+namespace soundtouch
+{
+
+// Default values for sound processing parameters:
+
+/// Default length of a single processing sequence, in milliseconds. This determines to how
+/// long sequences the original sound is chopped in the time-stretch algorithm.
+///
+/// The larger this value is, the lesser sequences are used in processing. In principle
+/// a bigger value sounds better when slowing down tempo, but worse when increasing tempo
+/// and vice versa.
+///
+/// Increasing this value reduces computational burden & vice versa.
+#define DEFAULT_SEQUENCE_MS 82
+
+/// Seeking window default length in milliseconds for algorithm that finds the best possible
+/// overlapping location. This determines from how wide window the algorithm may look for an
+/// optimal joining location when mixing the sound sequences back together.
+///
+/// The bigger this window setting is, the higher the possibility to find a better mixing
+/// position will become, but at the same time large values may cause a "drifting" artifact
+/// because consequent sequences will be taken at more uneven intervals.
+///
+/// If there's a disturbing artifact that sounds as if a constant frequency was drifting
+/// around, try reducing this setting.
+///
+/// Increasing this value increases computational burden & vice versa.
+#define DEFAULT_SEEKWINDOW_MS 14
+
+/// Overlap length in milliseconds. When the chopped sound sequences are mixed back together,
+/// to form a continuous sound stream, this parameter defines over how long period the two
+/// consecutive sequences are let to overlap each other.
+///
+/// This shouldn't be that critical parameter. If you reduce the DEFAULT_SEQUENCE_MS setting
+/// by a large amount, you might wish to try a smaller value on this.
+///
+/// Increasing this value increases computational burden & vice versa.
+#define DEFAULT_OVERLAP_MS 12
+
+
+/// Class that does the time-stretch (tempo change) effect for the processed
+/// sound.
+class TDStretch : public FIFOProcessor
+{
+protected:
+ uint channels;
+ uint sampleReq;
+ float tempo;
+
+ SAMPLETYPE *pMidBuffer;
+ SAMPLETYPE *pRefMidBuffer;
+ SAMPLETYPE *pRefMidBufferUnaligned;
+ uint overlapLength;
+ uint overlapDividerBits;
+ uint slopingDivider;
+ uint seekLength;
+ uint seekWindowLength;
+ uint maxOffset;
+ float nominalSkip;
+ float skipFract;
+ FIFOSampleBuffer outputBuffer;
+ FIFOSampleBuffer inputBuffer;
+ BOOL bQuickseek;
+ BOOL bMidBufferDirty;
+
+ uint sampleRate;
+ uint sequenceMs;
+ uint seekWindowMs;
+ uint overlapMs;
+
+ void acceptNewOverlapLength(uint newOverlapLength);
+
+ virtual void clearCrossCorrState();
+ void calculateOverlapLength(uint overlapMs);
+
+ virtual LONG_SAMPLETYPE calcCrossCorrStereo(const SAMPLETYPE *mixingPos, const SAMPLETYPE *compare) const;
+ virtual LONG_SAMPLETYPE calcCrossCorrMono(const SAMPLETYPE *mixingPos, const SAMPLETYPE *compare) const;
+
+ virtual uint seekBestOverlapPositionStereo(const SAMPLETYPE *refPos);
+ virtual uint seekBestOverlapPositionStereoQuick(const SAMPLETYPE *refPos);
+ virtual uint seekBestOverlapPositionMono(const SAMPLETYPE *refPos);
+ virtual uint seekBestOverlapPositionMonoQuick(const SAMPLETYPE *refPos);
+ uint seekBestOverlapPosition(const SAMPLETYPE *refPos);
+
+ virtual void overlapStereo(SAMPLETYPE *output, const SAMPLETYPE *input) const;
+ virtual void overlapMono(SAMPLETYPE *output, const SAMPLETYPE *input) const;
+
+ void clearMidBuffer();
+ void overlap(SAMPLETYPE *output, const SAMPLETYPE *input, uint ovlPos) const;
+
+ void precalcCorrReferenceMono();
+ void precalcCorrReferenceStereo();
+
+ void processNominalTempo();
+
+ /// Changes the tempo of the given sound samples.
+ /// Returns amount of samples returned in the "output" buffer.
+ /// The maximum amount of samples that can be returned at a time is set by
+ /// the 'set_returnBuffer_size' function.
+ void processSamples();
+
+ TDStretch();
+
+public:
+ virtual ~TDStretch();
+
+ /// Use this function instead of "new" operator to create a new instance of this class.
+ /// This function automatically chooses a correct feature set depending on if the CPU
+ /// supports MMX/SSE/etc extensions.
+ static TDStretch *newInstance();
+
+ /// Returns the output buffer object
+ FIFOSamplePipe *getOutput() { return &outputBuffer; };
+
+ /// Returns the input buffer object
+ FIFOSamplePipe *getInput() { return &inputBuffer; };
+
+ /// Sets new target tempo. Normal tempo = 'SCALE', smaller values represent slower
+ /// tempo, larger faster tempo.
+ void setTempo(float newTempo);
+
+ /// Returns nonzero if there aren't any samples available for outputting.
+ virtual void clear();
+
+ /// Clears the input buffer
+ void clearInput();
+
+ /// Sets the number of channels, 1 = mono, 2 = stereo
+ void setChannels(uint numChannels);
+
+ /// Enables/disables the quick position seeking algorithm. Zero to disable,
+ /// nonzero to enable
+ void enableQuickSeek(BOOL enable);
+
+ /// Returns nonzero if the quick seeking algorithm is enabled.
+ BOOL isQuickSeekEnabled() const;
+
+ /// Sets routine control parameters. These control are certain time constants
+ /// defining how the sound is stretched to the desired duration.
+ //
+ /// 'sampleRate' = sample rate of the sound
+ /// 'sequenceMS' = one processing sequence length in milliseconds
+ /// 'seekwindowMS' = seeking window length for scanning the best overlapping
+ /// position
+ /// 'overlapMS' = overlapping length
+ void setParameters(uint sampleRate, ///< Samplerate of sound being processed (Hz)
+ uint sequenceMS = DEFAULT_SEQUENCE_MS, ///< Single processing sequence length (ms)
+ uint seekwindowMS = DEFAULT_SEEKWINDOW_MS, ///< Offset seeking window length (ms)
+ uint overlapMS = DEFAULT_OVERLAP_MS ///< Sequence overlapping length (ms)
+ );
+
+ /// Get routine control parameters, see setParameters() function.
+ /// Any of the parameters to this function can be NULL, in such case corresponding parameter
+ /// value isn't returned.
+ void getParameters(uint *pSampleRate, uint *pSequenceMs, uint *pSeekWindowMs, uint *pOverlapMs);
+
+ /// Adds 'numsamples' pcs of samples from the 'samples' memory position into
+ /// the input of the object.
+ virtual void putSamples(
+ const SAMPLETYPE *samples, ///< Input sample data
+ uint numSamples ///< Number of samples in 'samples' so that one sample
+ ///< contains both channels if stereo
+ );
+};
+
+
+
+// Implementation-specific class declarations:
+
+#ifdef ALLOW_MMX
+ /// Class that implements MMX optimized routines for 16bit integer samples type.
+ class TDStretchMMX : public TDStretch
+ {
+ protected:
+ long calcCrossCorrStereo(const short *mixingPos, const short *compare) const;
+ virtual void overlapStereo(short *output, const short *input) const;
+ virtual void clearCrossCorrState();
+ };
+#endif /// ALLOW_MMX
+
+
+#ifdef ALLOW_3DNOW
+ /// Class that implements 3DNow! optimized routines for floating point samples type.
+ class TDStretch3DNow : public TDStretch
+ {
+ protected:
+ double calcCrossCorrStereo(const float *mixingPos, const float *compare) const;
+ };
+#endif /// ALLOW_3DNOW
+
+
+#ifdef ALLOW_SSE
+ /// Class that implements SSE optimized routines for floating point samples type.
+ class TDStretchSSE : public TDStretch
+ {
+ protected:
+ double calcCrossCorrStereo(const float *mixingPos, const float *compare) const;
+ };
+
+#endif /// ALLOW_SSE
+
+}
+#endif /// TDStretch_H
--- /dev/null
+////////////////////////////////////////////////////////////////////////////////
+///
+/// A header file for detecting the Intel MMX instructions set extension.
+///
+/// Please see 'mmx_win.cpp', 'mmx_cpp.cpp' and 'mmx_non_x86.cpp' for the
+/// routine implementations for x86 Windows, x86 gnu version and non-x86
+/// platforms, respectively.
+///
+/// Author : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai @ iki.fi
+/// SoundTouch WWW: http://www.iki.fi/oparviai/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// Last changed : $Date$
+// File revision : $Revision$
+//
+// $Id$
+//
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+// SoundTouch audio processing library
+// Copyright (c) Olli Parviainen
+//
+// This library is free software; you can redistribute it and/or
+// modify it under the terms of the GNU Lesser General Public
+// License as published by the Free Software Foundation; either
+// version 2.1 of the License, or (at your option) any later version.
+//
+// This library is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+// Lesser General Public License for more details.
+//
+// You should have received a copy of the GNU Lesser General Public
+// License along with this library; if not, write to the Free Software
+// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#ifndef _CPU_DETECT_H_
+#define _CPU_DETECT_H_
+
+#include "STTypes.h"
+
+#define SUPPORT_MMX 0x0001
+#define SUPPORT_3DNOW 0x0002
+#define SUPPORT_ALTIVEC 0x0004
+#define SUPPORT_SSE 0x0008
+#define SUPPORT_SSE2 0x0010
+
+/// Checks which instruction set extensions are supported by the CPU.
+///
+/// \return A bitmask of supported extensions, see SUPPORT_... defines.
+uint detectCPUextensions(void);
+
+/// Disables given set of instruction extensions. See SUPPORT_... defines.
+void disableExtensions(uint wDisableMask);
+
+#endif // _CPU_DETECT_H_
--- /dev/null
+////////////////////////////////////////////////////////////////////////////////
+///
+/// gcc version of the x86 CPU detect routine.
+///
+/// This file is to be compiled on any platform with the GNU C compiler.
+/// Compiler. Please see 'cpu_detect_x86_win.cpp' for the x86 Windows version
+/// of this file.
+///
+/// Author : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai @ iki.fi
+/// SoundTouch WWW: http://www.iki.fi/oparviai/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// Last changed : $Date$
+// File revision : $Revision$
+//
+// $Id$
+//
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+// SoundTouch audio processing library
+// Copyright (c) Olli Parviainen
+//
+// This library is free software; you can redistribute it and/or
+// modify it under the terms of the GNU Lesser General Public
+// License as published by the Free Software Foundation; either
+// version 2.1 of the License, or (at your option) any later version.
+//
+// This library is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+// Lesser General Public License for more details.
+//
+// You should have received a copy of the GNU Lesser General Public
+// License along with this library; if not, write to the Free Software
+// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#include <stdexcept>
+#include <string>
+#include "cpu_detect.h"
+
+#ifndef __GNUC__
+#error wrong platform - this source code file is for the GNU C compiler.
+#endif
+
+using namespace std;
+
+#include <stdio.h>
+//////////////////////////////////////////////////////////////////////////////
+//
+// processor instructions extension detection routines
+//
+//////////////////////////////////////////////////////////////////////////////
+
+
+// Flag variable indicating whick ISA extensions are disabled (for debugging)
+static uint _dwDisabledISA = 0x00; // 0xffffffff; //<- use this to disable all extensions
+
+// Disables given set of instruction extensions. See SUPPORT_... defines.
+void disableExtensions(uint dwDisableMask)
+{
+ _dwDisabledISA = dwDisableMask;
+}
+
+
+
+/// Checks which instruction set extensions are supported by the CPU.
+uint detectCPUextensions(void)
+{
+#ifndef __i386__
+ return 0; // always disable extensions on non-x86 platforms.
+#else
+ uint res = 0;
+
+ if (_dwDisabledISA == 0xffffffff) return 0;
+
+ asm volatile(
+ "\n\txor %%esi, %%esi" // clear %%esi = result register
+ // check if 'cpuid' instructions is available by toggling eflags bit 21
+
+ "\n\tpushf" // save eflags to stack
+ "\n\tpop %%eax" // load eax from stack (with eflags)
+ "\n\tmovl %%eax, %%ecx" // save the original eflags values to ecx
+ "\n\txor $0x00200000, %%eax" // toggle bit 21
+ "\n\tpush %%eax" // store toggled eflags to stack
+ "\n\tpopf" // load eflags from stack
+ "\n\tpushf" // save updated eflags to stack
+ "\n\tpop %%eax" // load from stack
+ "\n\txor %%edx, %%edx" // clear edx for defaulting no mmx
+ "\n\tcmp %%ecx, %%eax" // compare to original eflags values
+ "\n\tjz end" // jumps to 'end' if cpuid not present
+
+ // cpuid instruction available, test for presence of mmx instructions
+
+ "\n\tmovl $1, %%eax"
+ "\n\tcpuid"
+// movl $0x00800000, %edx // force enable MMX
+ "\n\ttest $0x00800000, %%edx"
+ "\n\tjz end" // branch if MMX not available
+
+ "\n\tor $0x01, %%esi" // otherwise add MMX support bit
+
+ "\n\ttest $0x02000000, %%edx"
+ "\n\tjz test3DNow" // branch if SSE not available
+
+ "\n\tor $0x08, %%esi" // otherwise add SSE support bit
+
+ "\n\ttest3DNow:"
+ // test for precense of AMD extensions
+ "\n\tmov $0x80000000, %%eax"
+ "\n\tcpuid"
+ "\n\tcmp $0x80000000, %%eax"
+ "\n\tjbe end" // branch if no AMD extensions detected
+
+ // test for precense of 3DNow! extension
+ "\n\tmov $0x80000001, %%eax"
+ "\n\tcpuid"
+ "\n\ttest $0x80000000, %%edx"
+ "\n\tjz end" // branch if 3DNow! not detected
+
+ "\n\tor $0x02, %%esi" // otherwise add 3DNow support bit
+
+ "\n\tend:"
+
+ "\n\tmov %%esi, %0"
+
+ : "=r" (res)
+ : /* no inputs */
+ : "%edx", "%eax", "%ecx", "%esi" );
+
+ return res & ~_dwDisabledISA;
+#endif
+}
--- /dev/null
+////////////////////////////////////////////////////////////////////////////////
+///
+/// Win32 version of the x86 CPU detect routine.
+///
+/// This file is to be compiled in Windows platform with Microsoft Visual C++
+/// Compiler. Please see 'cpu_detect_x86_gcc.cpp' for the gcc compiler version
+/// for all GNU platforms.
+///
+/// Author : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai @ iki.fi
+/// SoundTouch WWW: http://www.iki.fi/oparviai/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// Last changed : $Date$
+// File revision : $Revision$
+//
+// $Id$
+//
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+// SoundTouch audio processing library
+// Copyright (c) Olli Parviainen
+//
+// This library is free software; you can redistribute it and/or
+// modify it under the terms of the GNU Lesser General Public
+// License as published by the Free Software Foundation; either
+// version 2.1 of the License, or (at your option) any later version.
+//
+// This library is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+// Lesser General Public License for more details.
+//
+// You should have received a copy of the GNU Lesser General Public
+// License along with this library; if not, write to the Free Software
+// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#include "cpu_detect.h"
+
+#ifndef WIN32
+#error wrong platform - this source code file is exclusively for Win32 platform
+#endif
+
+//////////////////////////////////////////////////////////////////////////////
+//
+// processor instructions extension detection routines
+//
+//////////////////////////////////////////////////////////////////////////////
+
+// Flag variable indicating whick ISA extensions are disabled (for debugging)
+static uint _dwDisabledISA = 0x00; // 0xffffffff; //<- use this to disable all extensions
+
+
+// Disables given set of instruction extensions. See SUPPORT_... defines.
+void disableExtensions(uint dwDisableMask)
+{
+ _dwDisabledISA = dwDisableMask;
+}
+
+
+
+/// Checks which instruction set extensions are supported by the CPU.
+uint detectCPUextensions(void)
+{
+ uint res = 0;
+
+ if (_dwDisabledISA == 0xffffffff) return 0;
+
+ _asm
+ {
+ ; check if 'cpuid' instructions is available by toggling eflags bit 21
+ ;
+ xor esi, esi ; clear esi = result register
+
+ pushfd ; save eflags to stack
+ pop eax ; load eax from stack (with eflags)
+ mov ecx, eax ; save the original eflags values to ecx
+ xor eax, 0x00200000 ; toggle bit 21
+ push eax ; store toggled eflags to stack
+ popfd ; load eflags from stack
+ pushfd ; save updated eflags to stack
+ pop eax ; load from stack
+ xor edx, edx ; clear edx for defaulting no mmx
+ cmp eax, ecx ; compare to original eflags values
+ jz end ; jumps to 'end' if cpuid not present
+
+ ; cpuid instruction available, test for presence of mmx instructions
+ mov eax, 1
+ cpuid
+ test edx, 0x00800000
+ jz end ; branch if MMX not available
+
+ or esi, SUPPORT_MMX ; otherwise add MMX support bit
+
+ test edx, 0x02000000
+ jz test3DNow ; branch if SSE not available
+
+ or esi, SUPPORT_SSE ; otherwise add SSE support bit
+
+ test3DNow:
+ ; test for precense of AMD extensions
+ mov eax, 0x80000000
+ cpuid
+ cmp eax, 0x80000000
+ jbe end ; branch if no AMD extensions detected
+
+ ; test for precense of 3DNow! extension
+ mov eax, 0x80000001
+ cpuid
+ test edx, 0x80000000
+ jz end ; branch if 3DNow! not detected
+
+ or esi, SUPPORT_3DNOW ; otherwise add 3DNow support bit
+
+ end:
+
+ mov res, esi
+ }
+
+ return res & ~_dwDisabledISA;
+}
--- /dev/null
+////////////////////////////////////////////////////////////////////////////////
+///
+/// gcc version of the MMX optimized routines. All MMX optimized functions
+/// have been gathered into this single source code file, regardless to their
+/// class or original source code file, in order to ease porting the library
+/// to other compiler and processor platforms.
+///
+/// This file is to be compiled on any platform with the GNU C compiler.
+/// Compiler. Please see 'mmx_win.cpp' for the x86 Windows version of this
+/// file.
+///
+/// Author : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai @ iki.fi
+/// SoundTouch WWW: http://www.iki.fi/oparviai/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// Last changed : $Date$
+// File revision : $Revision$
+//
+// $Id$
+//
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+// SoundTouch audio processing library
+// Copyright (c) Olli Parviainen
+//
+// This library is free software; you can redistribute it and/or
+// modify it under the terms of the GNU Lesser General Public
+// License as published by the Free Software Foundation; either
+// version 2.1 of the License, or (at your option) any later version.
+//
+// This library is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+// Lesser General Public License for more details.
+//
+// You should have received a copy of the GNU Lesser General Public
+// License along with this library; if not, write to the Free Software
+// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#include <stdexcept>
+#include <string>
+#include "cpu_detect.h"
+
+#ifndef __GNUC__
+#error "wrong platform - this source code file is for the GNU C compiler."
+#endif
+
+using namespace std;
+using namespace soundtouch;
+
+
+#ifdef ALLOW_MMX
+// MMX routines available only with integer sample type
+
+//////////////////////////////////////////////////////////////////////////////
+//
+// implementation of MMX optimized functions of class 'TDStretch'
+//
+// NOTE: ebx in gcc 3.x is not preserved if -fPIC and -DPIC
+// gcc-3.4 correctly flags this error and wont let you continue.
+// gcc-2.95 preserves esi correctly
+//
+//////////////////////////////////////////////////////////////////////////////
+
+#include "TDStretch.h"
+#include <limits.h>
+
+// these are declared in 'TDStretch.cpp'
+extern int scanOffsets[4][24];
+
+// Calculates cross correlation of two buffers
+long TDStretchMMX::calcCrossCorrStereo(const short *pV1, const short *pV2) const
+{
+#ifdef __i386__
+ int corr;
+ uint local_overlapLength = overlapLength;
+ uint local_overlapDividerBits = overlapDividerBits;
+
+ asm volatile(
+ // Calculate cross-correlation between the tempOffset and tmpbid_buffer.
+
+ // Process 4 parallel batches of 2 * stereo samples each during one
+ // round to improve CPU-level parallellization.
+
+ // load address of sloped pV2 buffer to eax
+ // load address of mixing point of the sample data buffer to edi
+ // load counter to ecx = overlapLength / 8 - 1
+ // empty the mm0
+
+ // prepare to the first round by loading
+ // load mm1 = eax[0]
+ // load mm2 = eax[1];
+
+ "\n\tmovl %1, %%eax"
+ "\n\tmovl %2, %%edi"
+
+ "\n\tmovq (%%eax), %%mm1"
+ "\n\tmovl %3, %%ecx"
+
+ "\n\tmovq 8(%%eax), %%mm2"
+ "\n\tshr $3, %%ecx"
+
+ "\n\tpxor %%mm0, %%mm0"
+ "\n\tsub $1, %%ecx"
+
+ "\n\tmovd %4, %%mm5"
+
+ "\n1:"
+ // multiply-add mm1 = mm1 * edi[0]
+ // multiply-add mm2 = mm2 * edi[1]
+ //
+ // add mm2 += mm1
+ // mm2 >>= mm5 (=overlapDividerBits)
+ // add mm0 += mm2
+ //
+ // load mm3 = eax[2]
+ // multiply-add mm3 = mm3 * edi[2]
+ //
+ // load mm4 = eax[3]
+ // multiply-add mm4 = mm4 * edi[3]
+ //
+ // add mm3 += mm4
+ // mm3 >>= mm5 (=overlapDividerBits)
+ // add mm0 += mm3
+ //
+ // add eax += 4
+ // add edi += 4
+ // load mm1 = eax[0] (~eax[4])
+ // load mm2 = eax[1] (~eax[5])
+ //
+ // loop
+
+ "\n\tpmaddwd (%%edi), %%mm1" // qword ptr [edi]
+ "\n\tmovq 16(%%eax), %%mm3" // qword ptr [eax+16]
+
+ "\n\tpmaddwd 8(%%edi), %%mm2" // qword ptr [edi+8]
+ "\n\tmovq 24(%%eax), %%mm4" // qword ptr [eax+24]
+
+ "\n\tpmaddwd 16(%%edi), %%mm3" // qword ptr [edi+16]
+ "\n\tpaddd %%mm1, %%mm2"
+
+ "\n\tpmaddwd 24(%%edi), %%mm4" // qword ptr [edi+24]
+ "\n\tmovq 32(%%eax), %%mm1" // qword ptr [eax+32]
+
+ "\n\tpsrad %%mm5, %%mm2"
+ "\n\tadd $32, %%eax"
+
+ "\n\tpaddd %%mm4, %%mm3"
+ "\n\tpaddd %%mm2, %%mm0"
+
+ "\n\tmovq 8(%%eax), %%mm2" // qword ptr [eax+8]
+ "\n\tpsrad %%mm5, %%mm3"
+
+ "\n\tadd $32, %%edi"
+ "\n\tpaddd %%mm3, %%mm0"
+
+ "\n\tdec %%ecx"
+ "\n\tjnz 1b"
+
+ // Finalize the last partial loop:
+
+ "\n\tmovq 16(%%eax), %%mm3" // qword ptr [eax+16]
+ "\n\tpmaddwd (%%edi), %%mm1" // qword ptr [edi]
+
+ "\n\tmovq 24(%%eax), %%mm4" // qword ptr [eax+24]
+ "\n\tpmaddwd 8(%%edi), %%mm2" // qword ptr [edi+8]
+
+ "\n\tpmaddwd 16(%%edi), %%mm3" // qword ptr [edi+16]
+ "\n\tpaddd %%mm1, %%mm2"
+
+ "\n\tpmaddwd 24(%%edi), %%mm4" // qword ptr [edi+24]
+ "\n\tpsrad %%mm5, %%mm2"
+
+ "\n\tpaddd %%mm4, %%mm3"
+ "\n\tpaddd %%mm2, %%mm0"
+
+ "\n\tpsrad %%mm5, %%mm3"
+ "\n\tpaddd %%mm3, %%mm0"
+
+ // copy hi-dword of mm0 to lo-dword of mm1, then sum mmo+mm1
+ // and finally store the result into the variable "corr"
+
+ "\n\tmovq %%mm0, %%mm1"
+ "\n\tpsrlq $32, %%mm1"
+ "\n\tpaddd %%mm1, %%mm0"
+ "\n\tmovd %%mm0, %0"
+ : "=rm" (corr)
+ : "rim" (pV1), "rim" (pV2), "rim" (local_overlapLength),
+ "rim" (local_overlapDividerBits)
+ : "%ecx", "%eax", "%edi"
+ );
+ return corr;
+
+ // Note: Warning about the missing EMMS instruction is harmless
+ // as it'll be called elsewhere.
+#else
+ throw runtime_error("MMX not supported");
+#endif
+}
+
+void TDStretchMMX::clearCrossCorrState()
+{
+#ifdef __i386__
+ asm volatile("EMMS");
+#endif
+}
+
+// MMX-optimized version of the function overlapStereo
+void TDStretchMMX::overlapStereo(short *output, const short *input) const
+{
+#ifdef __i386__
+ short *local_midBuffer = pMidBuffer;
+ uint local_overlapLength = overlapLength;
+ uint local_overlapDividerBits = overlapDividerBits;
+
+ asm volatile(
+ "\n\t"
+ // load sliding mixing value counter to mm6 and mm7
+ // load counter value to ecx = overlapLength / 4
+ // load divider-shifter value to esi
+ // load mixing value adder to mm5
+ // load address of midBuffer to eax
+ // load address of inputBuffer added with ovlOffset to edi
+ // load address of end of the outputBuffer to edx
+ //
+ // We need to preserve esi, since gcc uses it for the
+ // stack frame.
+
+ "movl %0, %%eax\n\t" // ecx = 0x0000 OVL_
+ "movl $0x0002fffe, %%edi\n\t" // ecx = 0x0002 fffe
+
+ "movl %1, %%esi\n\t"
+ "movd %%eax, %%mm6\n\t" // mm6 = 0x0000 0000 0000 OVL_
+
+ "movl %%eax, %%ecx\n\t"
+ "sub $1, %%eax\n\t"
+
+ "punpckldq %%mm6, %%mm6\n\t" // mm6 = 0x0000 OVL_ 0000 OVL_
+
+ "or $0x00010000, %%eax\n\t" // eax = 0x0001 overlapLength-1
+
+ "movd %%edi, %%mm5\n\t" // mm5 = 0x0000 0000 0002 fffe
+ "movd %%eax, %%mm7\n\t" // mm7 = 0x0000 0000 0001 01ff
+
+ "movl %3, %%edi\n\t"
+
+ "movl %4, %%eax\n\t" // dword ptr local_midBuffer
+ "punpckldq %%mm5, %%mm5\n\t" // mm5 = 0x0002 fffe 0002 fffe
+
+ "shr $2, %%ecx\n\t" // ecx = overlapLength / 2
+ "punpckldq %%mm7, %%mm7\n\t" // mm7 = 0x0001 01ff 0001 01ff
+
+ "movl %2, %%edx\n"
+
+ "2:\n\t"
+ // Process two parallel batches of 2+2 stereo samples during each round
+ // to improve CPU-level parallellization.
+ //
+ // Load [eax] into mm0 and mm1
+ // Load [edi] into mm3
+ // unpack words of mm0, mm1 and mm3 into mm0 and mm1
+ // multiply-add mm0*mm6 and mm1*mm7, store results into mm0 and mm1
+ // divide mm0 and mm1 by 512 (=right-shift by overlapDividerBits)
+ // pack the result into mm0 and store into [edx]
+ //
+ // Load [eax+8] into mm2 and mm3
+ // Load [edi+8] into mm4
+ // unpack words of mm2, mm3 and mm4 into mm2 and mm3
+ // multiply-add mm2*mm6 and mm3*mm7, store results into mm2 and mm3
+ // divide mm2 and mm3 by 512 (=right-shift by overlapDividerBits)
+ // pack the result into mm2 and store into [edx+8]
+
+
+ "movq (%%eax), %%mm0\n\t" // mm0 = m1l m1r m0l m0r
+ "add $16, %%edx\n\t"
+
+ "movq (%%edi), %%mm3\n\t" // mm3 = i1l i1r i0l i0r
+ "movq %%mm0, %%mm1\n\t" // mm1 = m1l m1r m0l m0r
+
+ "movq 8(%%eax), %%mm2\n\t" // mm2 = m3l m3r m2l m2r
+ "punpcklwd %%mm3, %%mm0\n\t" // mm0 = i0l m0l i0r m0r
+
+ "movq 8(%%edi), %%mm4\n\t" // mm4 = i3l i3r i2l i2r
+ "punpckhwd %%mm3, %%mm1\n\t" // mm1 = i1l m1l i1r m1r
+
+ "movq %%mm2, %%mm3\n\t" // mm3 = m3l m3r m2l m2r
+ "punpcklwd %%mm4, %%mm2\n\t" // mm2 = i2l m2l i2r m2r
+
+ "pmaddwd %%mm6, %%mm0\n\t" // mm0 = i0l*m63+m0l*m62 i0r*m61+m0r*m60
+ "punpckhwd %%mm4, %%mm3\n\t" // mm3 = i3l m3l i3r m3r
+
+ "movd %%esi, %%mm4\n\t" // mm4 = overlapDividerBits
+
+ "pmaddwd %%mm7, %%mm1\n\t" // mm1 = i1l*m73+m1l*m72 i1r*m71+m1r*m70
+ "paddw %%mm5, %%mm6\n\t"
+
+ "paddw %%mm5, %%mm7\n\t"
+ "psrad %%mm4, %%mm0\n\t" // mmo >>= overlapDividerBits
+
+ "pmaddwd %%mm6, %%mm2\n\t" // mm2 = i2l*m63+m2l*m62 i2r*m61+m2r*m60
+ "psrad %%mm4, %%mm1\n\t" // mm1 >>= overlapDividerBits
+
+ "pmaddwd %%mm7, %%mm3\n\t" // mm3 = i3l*m73+m3l*m72 i3r*m71+m3r*m70
+ "psrad %%mm4, %%mm2\n\t" // mm2 >>= overlapDividerBits
+
+ "packssdw %%mm1, %%mm0\n\t" // mm0 = mm1h mm1l mm0h mm0l
+ "psrad %%mm4, %%mm3\n\t" // mm3 >>= overlapDividerBits
+
+ "add $16, %%eax\n\t"
+ "paddw %%mm5, %%mm6\n\t"
+
+ "packssdw %%mm3, %%mm2\n\t" // mm2 = mm2h mm2l mm3h mm3l
+ "paddw %%mm5, %%mm7\n\t"
+
+ "movq %%mm0, -16(%%edx)\n\t"
+ "add $16, %%edi\n\t"
+
+ "movq %%mm2, -8(%%edx)\n\t"
+ "dec %%ecx\n\t"
+
+ "jnz 2b\n\t"
+
+ "emms\n\t"
+
+ :
+ : "rim" (local_overlapLength),
+ "rim" (local_overlapDividerBits),
+ "rim" (output),
+ "rim" (input),
+ "rim" (local_midBuffer)
+ /* input */
+ : "%edi", "%ecx", "%edx", "%eax", "%esi" /* regs */
+ );
+#else
+ throw runtime_error("MMX not supported");
+#endif
+}
+
+
+//////////////////////////////////////////////////////////////////////////////
+//
+// implementation of MMX optimized functions of class 'FIRFilter'
+//
+//////////////////////////////////////////////////////////////////////////////
+
+#include "FIRFilter.h"
+
+FIRFilterMMX::FIRFilterMMX() : FIRFilter()
+{
+ filterCoeffsUnalign = NULL;
+}
+
+
+FIRFilterMMX::~FIRFilterMMX()
+{
+ delete[] filterCoeffsUnalign;
+}
+
+
+#if 1
+// (overloaded) Calculates filter coefficients for MMX routine
+void FIRFilterMMX::setCoefficients(const short *coeffs, uint newLength, uint uResultDivFactor)
+{
+#ifdef __i386__
+ uint i;
+ FIRFilter::setCoefficients(coeffs, newLength, uResultDivFactor);
+
+ // Ensure that filter coeffs array is aligned to 16-byte boundary
+ delete[] filterCoeffsUnalign;
+ filterCoeffsUnalign = new short[2 * newLength + 8];
+ filterCoeffsAlign = (short *)(((uint)filterCoeffsUnalign + 15) & -16);
+
+ // rearrange the filter coefficients for mmx routines
+ for (i = 0;i < length; i += 4)
+ {
+ filterCoeffsAlign[2 * i + 0] = coeffs[i + 0];
+ filterCoeffsAlign[2 * i + 1] = coeffs[i + 2];
+ filterCoeffsAlign[2 * i + 2] = coeffs[i + 0];
+ filterCoeffsAlign[2 * i + 3] = coeffs[i + 2];
+
+ filterCoeffsAlign[2 * i + 4] = coeffs[i + 1];
+ filterCoeffsAlign[2 * i + 5] = coeffs[i + 3];
+ filterCoeffsAlign[2 * i + 6] = coeffs[i + 1];
+ filterCoeffsAlign[2 * i + 7] = coeffs[i + 3];
+ }
+#else
+ throw runtime_error("MMX not supported");
+#endif
+}
+
+
+
+// mmx-optimized version of the filter routine for stereo sound
+uint FIRFilterMMX::evaluateFilterStereo(short *dest, const short *src, const uint numSamples) const
+{
+#ifdef __i386__
+ // Create stack copies of the needed member variables for asm routines :
+ uint local_length = length;
+ uint local_lengthDiv8 = lengthDiv8;
+ uint local_resultDivider = resultDivFactor;
+ short *local_filterCoeffs = (short*)filterCoeffsAlign;
+ short *local_src = (short *)src;
+
+ asm volatile(
+ "\n\t"
+ // Load (num_samples-aa_filter_length)/2 to edi as a i
+ // Load a pointer to samples to esi
+ // Load a pointer to destination to edx
+
+ "movl %0, %%edi\n\t"
+ "subl %2, %%edi\n\t"
+ "movl %3, %%edx\n\t"
+ "sar $1, %%edi\n"
+
+ // Load filter length/8 to ecx
+ // Load pointer to samples from esi to ebx
+ // Load counter from edi to ecx
+ // Load [ebx] to mm3
+ // Load pointer to filter coefficients to eax
+ "3:\n\t"
+ "movl %1, %%esi\n\t"
+ "pxor %%mm0, %%mm0\n\t"
+
+ "movl %4, %%ecx\n\t"
+ "pxor %%mm7, %%mm7\n\t"
+
+ "movq (%%esi), %%mm1\n\t" // mm1 = l1 r1 l0 r0
+ "movl %5, %%eax\n"
+ "4:\n\t"
+
+ "movq 8(%%esi), %%mm2\n\t" // mm2 = l3 r3 l2 r2
+ "movq %%mm1, %%mm4\n\t" // mm4 = l1 r1 l0 r0
+
+ "movq 16(%%esi), %%mm3\n\t" // mm3 = l5 r5 l4 r4
+ "punpckhwd %%mm2, %%mm1\n\t" // mm1 = l3 l1 r3 r1
+
+ "movq %%mm2, %%mm6\n\t" // mm6 = l3 r3 l2 r2
+ "punpcklwd %%mm2, %%mm4\n\t" // mm4 = l2 l0 r2 r0
+
+ "movq (%%eax), %%mm2\n\t" // mm2 = f2 f0 f2 f0
+ "movq %%mm1, %%mm5\n\t" // mm5 = l3 l1 r3 r1
+
+ "punpcklwd %%mm3, %%mm6\n\t" // mm6 = l4 l2 r4 r2
+ "pmaddwd %%mm2, %%mm4\n\t" // mm4 = l2*f2+l0*f0 r2*f2+r0*f0
+
+ "pmaddwd %%mm2, %%mm5\n\t" // mm5 = l3*f2+l1*f0 r3*f2+l1*f0
+ "movq 8(%%eax), %%mm2\n\t" // mm2 = f3 f1 f3 f1
+
+ "paddd %%mm4, %%mm0\n\t" // mm0 += s02*f02
+ "movq %%mm3, %%mm4\n\t" // mm4 = l1 r1 l0 r0
+
+ "pmaddwd %%mm2, %%mm1\n\t" // mm1 = l3*f3+l1*f1 r3*f3+l1*f1
+ "paddd %%mm5, %%mm7\n\t" // mm7 += s13*f02
+
+ "pmaddwd %%mm2, %%mm6\n\t" // mm6 = l4*f3+l2*f1 r4*f3+f4*f1
+ "movq 24(%%esi), %%mm2\n\t" // mm2 = l3 r3 l2 r2
+
+ "paddd %%mm1, %%mm0\n\t" // mm0 += s31*f31
+ "movq 32(%%esi), %%mm1\n\t" // mm1 = l5 r5 l4 r4
+
+ "paddd %%mm6, %%mm7\n\t" // mm7 += s42*f31
+ "punpckhwd %%mm2, %%mm3\n\t" // mm3 = l3 l1 r3 r1
+
+ "movq %%mm2, %%mm6\n\t" // mm6 = l3 r3 l2 r2
+ "punpcklwd %%mm2, %%mm4\n\t" // mm4 = l2 l0 r2 r0
+
+ "movq 16(%%eax), %%mm2\n\t" // mm2 = f2 f0 f2 f0
+ "movq %%mm3, %%mm5\n\t" // mm5 = l3 l1 r3 r1
+
+ "punpcklwd %%mm1, %%mm6\n\t" // mm6 = l4 l2 r4 r2
+ "add $32, %%eax\n\t"
+
+ "pmaddwd %%mm2, %%mm4\n\t" // mm4 = l2*f2+l0*f0 r2*f2+r0*f0
+ "add $32, %%esi\n\t"
+
+ "pmaddwd %%mm2, %%mm5\n\t" // mm5 = l3*f2+l1*f0 r3*f2+l1*f0
+ "movq -8(%%eax), %%mm2\n\t" // mm2 = f3 f1 f3 f1
+
+ "paddd %%mm4, %%mm0\n\t" // mm0 += s02*f02
+ "pmaddwd %%mm2, %%mm3\n\t" // mm3 = l3*f3+l1*f1 r3*f3+l1*f1
+
+ "paddd %%mm5, %%mm7\n\t" // mm7 += s13*f02
+ "pmaddwd %%mm2, %%mm6\n\t" // mm6 = l4*f3+l2*f1 r4*f3+f4*f1
+
+ "paddd %%mm3, %%mm0\n\t" // mm0 += s31*f31
+ "paddd %%mm6, %%mm7\n\t" // mm7 += s42*f31
+
+ "dec %%ecx\n\t"
+ "jnz 4b\n\t"
+
+ // Divide mm0 and mm7 by 8192 (= right-shift by 13),
+ // pack and store to [edx]
+ "movd %6, %%mm4\n\t"
+
+ "psrad %%mm4, %%mm0\n\t" // divide the result
+
+ "add $8, %%edx\n\t"
+ "psrad %%mm4, %%mm7\n\t" // divide the result
+
+ "add $8, %1\n\t"
+ "packssdw %%mm7, %%mm0\n\t"
+
+ "movq %%mm0, -8(%%edx)\n\t"
+ "dec %%edi\n\t"
+
+ "jnz 3b\n\t"
+
+ "emms\n\t"
+
+ :
+ : "rim" (numSamples),
+ "rim" (local_src),
+ "rim" (local_length),
+ "rim" (dest),
+ "rim" (local_lengthDiv8),
+ "rim" (local_filterCoeffs),
+ "rim" (local_resultDivider) /* input */
+ : "%eax", "%ecx", "%edx", "%edi", "%esi" /* regs */
+ );
+ return (numSamples & 0xfffffffe) - local_length;
+#else
+ throw runtime_error("MMX not supported");
+ return 0;
+#endif
+}
+#endif
+
+#endif // ALLOW_MMX
--- /dev/null
+////////////////////////////////////////////////////////////////////////////////
+///
+/// Win32 version of the MMX optimized routines. All MMX optimized functions
+/// have been gathered into this single source code file, regardless to their
+/// class or original source code file, in order to ease porting the library
+/// to other compiler and processor platforms.
+///
+/// This file is to be compiled in Windows platform with Microsoft Visual C++
+/// Compiler. Please see 'mmx_gcc.cpp' for the gcc compiler version for all
+/// GNU platforms.
+///
+/// Author : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai @ iki.fi
+/// SoundTouch WWW: http://www.iki.fi/oparviai/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// Last changed : $Date$
+// File revision : $Revision$
+//
+// $Id$
+//
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+// SoundTouch audio processing library
+// Copyright (c) Olli Parviainen
+//
+// This library is free software; you can redistribute it and/or
+// modify it under the terms of the GNU Lesser General Public
+// License as published by the Free Software Foundation; either
+// version 2.1 of the License, or (at your option) any later version.
+//
+// This library is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+// Lesser General Public License for more details.
+//
+// You should have received a copy of the GNU Lesser General Public
+// License along with this library; if not, write to the Free Software
+// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#include "STTypes.h"
+
+#ifndef WIN32
+#error "wrong platform - this source code file is exclusively for Win32 platform"
+#endif
+
+using namespace soundtouch;
+
+#ifdef ALLOW_MMX
+// MMX routines available only with integer sample type
+
+//////////////////////////////////////////////////////////////////////////////
+//
+// implementation of MMX optimized functions of class 'TDStretchMMX'
+//
+//////////////////////////////////////////////////////////////////////////////
+
+#include "TDStretch.h"
+#include <limits.h>
+
+// these are declared in 'TDStretch.cpp'
+extern int scanOffsets[4][24];
+
+// Calculates cross correlation of two buffers
+long TDStretchMMX::calcCrossCorrStereo(const short *pV1, const short *pV2) const
+{
+ long corr;
+ uint local_overlapLength = overlapLength;
+ uint local_overlapDividerBits = overlapDividerBits;
+
+ _asm
+ {
+ ; Calculate cross-correlation between the tempOffset and tmpbid_buffer.
+ ;
+ ; Process 4 parallel batches of 2 * stereo samples each during one
+ ; round to improve CPU-level parallellization.
+ ;
+ ; load address of sloped pV2 buffer to eax
+ ; load address of mixing point of the sample data buffer to ebx
+ ; load counter to ecx = overlapLength / 8 - 1
+ ; empty the mm0
+ ;
+ ; prepare to the first round by loading
+ ; load mm1 = eax[0]
+ ; load mm2 = eax[1];
+
+ mov eax, dword ptr pV1
+ mov ebx, dword ptr pV2
+
+ movq mm1, qword ptr [eax]
+ mov ecx, local_overlapLength
+
+ movq mm2, qword ptr [eax+8]
+ shr ecx, 3
+
+ pxor mm0, mm0
+ sub ecx, 1
+
+ movd mm5, local_overlapDividerBits
+
+ loop1:
+ ; multiply-add mm1 = mm1 * ebx[0]
+ ; multiply-add mm2 = mm2 * ebx[1]
+ ;
+ ; add mm2 += mm1
+ ; mm2 >>= mm5 (=overlapDividerBits)
+ ; add mm0 += mm2
+ ;
+ ; load mm3 = eax[2]
+ ; multiply-add mm3 = mm3 * ebx[2]
+ ;
+ ; load mm4 = eax[3]
+ ; multiply-add mm4 = mm4 * ebx[3]
+ ;
+ ; add mm3 += mm4
+ ; mm3 >>= mm5 (=overlapDividerBits)
+ ; add mm0 += mm3
+ ;
+ ; add eax += 4;
+ ; add ebx += 4
+ ; load mm1 = eax[0] (~eax[4])
+ ; load mm2 = eax[1] (~eax[5])
+ ;
+ ; loop
+
+ pmaddwd mm1, qword ptr [ebx]
+ movq mm3, qword ptr [eax+16]
+
+ pmaddwd mm2, qword ptr [ebx+8]
+ movq mm4, qword ptr [eax+24]
+
+ pmaddwd mm3, qword ptr [ebx+16]
+ paddd mm2, mm1
+
+ pmaddwd mm4, qword ptr [ebx+24]
+ movq mm1, qword ptr [eax+32]
+
+ psrad mm2, mm5
+ add eax, 32
+
+ paddd mm3, mm4
+ paddd mm0, mm2
+
+ movq mm2, qword ptr [eax+8]
+ psrad mm3, mm5
+
+ add ebx, 32
+ paddd mm0, mm3
+
+ dec ecx
+ jnz loop1
+
+ ; Finalize the last partial loop:
+
+ movq mm3, qword ptr [eax+16]
+ pmaddwd mm1, qword ptr [ebx]
+
+ movq mm4, qword ptr [eax+24]
+ pmaddwd mm2, qword ptr [ebx+8]
+
+ pmaddwd mm3, qword ptr [ebx+16]
+ paddd mm2, mm1
+
+ pmaddwd mm4, qword ptr [ebx+24]
+ psrad mm2, mm5
+
+ paddd mm3, mm4
+ paddd mm0, mm2
+
+ psrad mm3, mm5
+ paddd mm0, mm3
+
+ ; copy hi-dword of mm0 to lo-dword of mm1, then sum mmo+mm1
+ ; and finally store the result into the variable "corr"
+
+ movq mm1, mm0
+ psrlq mm1, 32
+ paddd mm0, mm1
+ movd corr, mm0
+ }
+ return corr;
+
+ // Note: Warning about the missing EMMS instruction is harmless
+ // as it'll be called elsewhere.
+}
+
+
+
+void TDStretchMMX::clearCrossCorrState()
+{
+ _asm EMMS;
+}
+
+
+
+
+
+// MMX-optimized version of the function overlapStereo
+void TDStretchMMX::overlapStereo(short *output, const short *input) const
+{
+ short *local_midBuffer = pMidBuffer;
+ uint local_overlapLength = overlapLength;
+ uint local_overlapDividerBits = overlapDividerBits;
+
+ _asm
+ {
+ ; load sliding mixing value counter to mm6 and mm7
+ ; load counter value to ecx = overlapLength / 4
+ ; load divider-shifter value to esi
+ ; load mixing value adder to mm5
+ ; load address of midBuffer to eax
+ ; load address of inputBuffer added with ovlOffset to ebx
+ ; load address of end of the outputBuffer to edx
+
+ mov eax, local_overlapLength ; ecx = 0x0000 OVL_
+ mov edi, 0x0002fffe ; ecx = 0x0002 fffe
+
+ mov esi, local_overlapDividerBits
+ movd mm6, eax ; mm6 = 0x0000 0000 0000 OVL_
+
+ mov ecx, eax;
+ sub eax, 1
+
+ punpckldq mm6, mm6 ; mm6 = 0x0000 OVL_ 0000 OVL_
+ mov edx, output
+
+ or eax, 0x00010000 ; eax = 0x0001 overlapLength-1
+ mov ebx, dword ptr input
+
+ movd mm5, edi ; mm5 = 0x0000 0000 0002 fffe
+ movd mm7, eax ; mm7 = 0x0000 0000 0001 01ff
+
+ mov eax, dword ptr local_midBuffer
+ punpckldq mm5, mm5 ; mm5 = 0x0002 fffe 0002 fffe
+
+ shr ecx, 2 ; ecx = overlapLength / 2
+ punpckldq mm7, mm7 ; mm7 = 0x0001 01ff 0001 01ff
+
+ loop1:
+ ; Process two parallel batches of 2+2 stereo samples during each round
+ ; to improve CPU-level parallellization.
+ ;
+ ; Load [eax] into mm0 and mm1
+ ; Load [ebx] into mm3
+ ; unpack words of mm0, mm1 and mm3 into mm0 and mm1
+ ; multiply-add mm0*mm6 and mm1*mm7, store results into mm0 and mm1
+ ; divide mm0 and mm1 by 512 (=right-shift by overlapDividerBits)
+ ; pack the result into mm0 and store into [edx]
+ ;
+ ; Load [eax+8] into mm2 and mm3
+ ; Load [ebx+8] into mm4
+ ; unpack words of mm2, mm3 and mm4 into mm2 and mm3
+ ; multiply-add mm2*mm6 and mm3*mm7, store results into mm2 and mm3
+ ; divide mm2 and mm3 by 512 (=right-shift by overlapDividerBits)
+ ; pack the result into mm2 and store into [edx+8]
+
+
+ movq mm0, qword ptr [eax] ; mm0 = m1l m1r m0l m0r
+ add edx, 16
+
+ movq mm3, qword ptr [ebx] ; mm3 = i1l i1r i0l i0r
+ movq mm1, mm0 ; mm1 = m1l m1r m0l m0r
+
+ movq mm2, qword ptr [eax+8] ; mm2 = m3l m3r m2l m2r
+ punpcklwd mm0, mm3 ; mm0 = i0l m0l i0r m0r
+
+ movq mm4, qword ptr [ebx+8] ; mm4 = i3l i3r i2l i2r
+ punpckhwd mm1, mm3 ; mm1 = i1l m1l i1r m1r
+
+ movq mm3, mm2 ; mm3 = m3l m3r m2l m2r
+ punpcklwd mm2, mm4 ; mm2 = i2l m2l i2r m2r
+
+ pmaddwd mm0, mm6 ; mm0 = i0l*m63+m0l*m62 i0r*m61+m0r*m60
+ punpckhwd mm3, mm4 ; mm3 = i3l m3l i3r m3r
+
+ movd mm4, esi ; mm4 = overlapDividerBits
+
+ pmaddwd mm1, mm7 ; mm1 = i1l*m73+m1l*m72 i1r*m71+m1r*m70
+ paddw mm6, mm5
+
+ paddw mm7, mm5
+ psrad mm0, mm4 ; mmo >>= overlapDividerBits
+
+ pmaddwd mm2, mm6 ; mm2 = i2l*m63+m2l*m62 i2r*m61+m2r*m60
+ psrad mm1, mm4 ; mm1 >>= overlapDividerBits
+
+ pmaddwd mm3, mm7 ; mm3 = i3l*m73+m3l*m72 i3r*m71+m3r*m70
+ psrad mm2, mm4 ; mm2 >>= overlapDividerBits
+
+ packssdw mm0, mm1 ; mm0 = mm1h mm1l mm0h mm0l
+ psrad mm3, mm4 ; mm3 >>= overlapDividerBits
+
+ add eax, 16
+ paddw mm6, mm5
+
+ packssdw mm2, mm3 ; mm2 = mm2h mm2l mm3h mm3l
+ paddw mm7, mm5
+
+ movq qword ptr [edx-16], mm0
+ add ebx, 16
+
+ movq qword ptr [edx-8], mm2
+ dec ecx
+
+ jnz loop1
+
+ emms
+ }
+}
+
+
+//////////////////////////////////////////////////////////////////////////////
+//
+// implementation of MMX optimized functions of class 'FIRFilter'
+//
+//////////////////////////////////////////////////////////////////////////////
+
+#include "FIRFilter.h"
+
+
+FIRFilterMMX::FIRFilterMMX() : FIRFilter()
+{
+ filterCoeffsUnalign = NULL;
+}
+
+
+FIRFilterMMX::~FIRFilterMMX()
+{
+ delete[] filterCoeffsUnalign;
+}
+
+
+// (overloaded) Calculates filter coefficients for MMX routine
+void FIRFilterMMX::setCoefficients(const short *coeffs, uint newLength, uint uResultDivFactor)
+{
+ uint i;
+ FIRFilter::setCoefficients(coeffs, newLength, uResultDivFactor);
+
+ // Ensure that filter coeffs array is aligned to 16-byte boundary
+ delete[] filterCoeffsUnalign;
+ filterCoeffsUnalign = new short[2 * newLength + 8];
+ filterCoeffsAlign = (short *)(((uint)filterCoeffsUnalign + 15) & -16);
+
+ // rearrange the filter coefficients for mmx routines
+ for (i = 0;i < length; i += 4)
+ {
+ filterCoeffsAlign[2 * i + 0] = coeffs[i + 0];
+ filterCoeffsAlign[2 * i + 1] = coeffs[i + 2];
+ filterCoeffsAlign[2 * i + 2] = coeffs[i + 0];
+ filterCoeffsAlign[2 * i + 3] = coeffs[i + 2];
+
+ filterCoeffsAlign[2 * i + 4] = coeffs[i + 1];
+ filterCoeffsAlign[2 * i + 5] = coeffs[i + 3];
+ filterCoeffsAlign[2 * i + 6] = coeffs[i + 1];
+ filterCoeffsAlign[2 * i + 7] = coeffs[i + 3];
+ }
+}
+
+
+
+// mmx-optimized version of the filter routine for stereo sound
+uint FIRFilterMMX::evaluateFilterStereo(short *dest, const short *src, const uint numSamples) const
+{
+ // Create stack copies of the needed member variables for asm routines :
+ uint local_length = length;
+ uint local_lengthDiv8 = lengthDiv8;
+ uint local_resultDivider = resultDivFactor;
+ short *local_filterCoeffs = (short*)filterCoeffsAlign;
+
+ if (local_length < 2) return 0;
+
+ _asm
+ {
+ ; Load (num_samples-aa_filter_length)/2 to edi as a i
+ ; Load a pointer to samples to esi
+ ; Load a pointer to destination to edx
+
+ mov edi, numSamples
+ mov esi, dword ptr src
+ sub edi, local_length
+ mov edx, dword ptr dest
+ sar edi, 1
+
+ ; Load filter length/8 to ecx
+ ; Load pointer to samples from esi to ebx
+ ; Load counter from edi to ecx
+ ; Load [ebx] to mm3
+ ; Load pointer to filter coefficients to eax
+loop1:
+ mov ebx, esi
+ pxor mm0, mm0
+
+ mov ecx, local_lengthDiv8
+ pxor mm7, mm7
+
+ movq mm1, [ebx] ; mm1 = l1 r1 l0 r0
+ mov eax, local_filterCoeffs
+loop2:
+
+ movq mm2, [ebx+8] ; mm2 = l3 r3 l2 r2
+ movq mm4, mm1 ; mm4 = l1 r1 l0 r0
+
+ movq mm3, [ebx+16] ; mm3 = l5 r5 l4 r4
+ punpckhwd mm1, mm2 ; mm1 = l3 l1 r3 r1
+
+ movq mm6, mm2 ; mm6 = l3 r3 l2 r2
+ punpcklwd mm4, mm2 ; mm4 = l2 l0 r2 r0
+
+ movq mm2, qword ptr [eax] ; mm2 = f2 f0 f2 f0
+ movq mm5, mm1 ; mm5 = l3 l1 r3 r1
+
+ punpcklwd mm6, mm3 ; mm6 = l4 l2 r4 r2
+ pmaddwd mm4, mm2 ; mm4 = l2*f2+l0*f0 r2*f2+r0*f0
+
+ pmaddwd mm5, mm2 ; mm5 = l3*f2+l1*f0 r3*f2+l1*f0
+ movq mm2, qword ptr [eax+8] ; mm2 = f3 f1 f3 f1
+
+ paddd mm0, mm4 ; mm0 += s02*f02
+ movq mm4, mm3 ; mm4 = l1 r1 l0 r0
+
+ pmaddwd mm1, mm2 ; mm1 = l3*f3+l1*f1 r3*f3+l1*f1
+ paddd mm7, mm5 ; mm7 += s13*f02
+
+ pmaddwd mm6, mm2 ; mm6 = l4*f3+l2*f1 r4*f3+f4*f1
+ movq mm2, [ebx+24] ; mm2 = l3 r3 l2 r2
+
+ paddd mm0, mm1 ; mm0 += s31*f31
+ movq mm1, [ebx+32] ; mm1 = l5 r5 l4 r4
+
+ paddd mm7, mm6 ; mm7 += s42*f31
+ punpckhwd mm3, mm2 ; mm3 = l3 l1 r3 r1
+
+ movq mm6, mm2 ; mm6 = l3 r3 l2 r2
+ punpcklwd mm4, mm2 ; mm4 = l2 l0 r2 r0
+
+ movq mm2, qword ptr [eax+16] ; mm2 = f2 f0 f2 f0
+ movq mm5, mm3 ; mm5 = l3 l1 r3 r1
+
+ punpcklwd mm6, mm1 ; mm6 = l4 l2 r4 r2
+ add eax, 32
+
+ pmaddwd mm4, mm2 ; mm4 = l2*f2+l0*f0 r2*f2+r0*f0
+ add ebx, 32
+
+ pmaddwd mm5, mm2 ; mm5 = l3*f2+l1*f0 r3*f2+l1*f0
+ movq mm2, qword ptr [eax-8] ; mm2 = f3 f1 f3 f1
+
+ paddd mm0, mm4 ; mm0 += s02*f02
+ pmaddwd mm3, mm2 ; mm3 = l3*f3+l1*f1 r3*f3+l1*f1
+
+ paddd mm7, mm5 ; mm7 += s13*f02
+ pmaddwd mm6, mm2 ; mm6 = l4*f3+l2*f1 r4*f3+f4*f1
+
+ paddd mm0, mm3 ; mm0 += s31*f31
+ paddd mm7, mm6 ; mm7 += s42*f31
+
+ dec ecx
+ jnz loop2
+
+ ; Divide mm0 and mm7 by 8192 (= right-shift by 13),
+ ; pack and store to [edx]
+ movd mm4, local_resultDivider;
+
+ psrad mm0, mm4 ; divider the result
+
+ add edx, 8
+ psrad mm7, mm4 ; divider the result
+
+ add esi, 8
+ packssdw mm0, mm7
+
+ movq qword ptr [edx-8], mm0
+ dec edi
+
+ jnz loop1
+
+ emms
+ }
+ return (numSamples & 0xfffffffe) - local_length;
+}
+
+#endif // ALLOW_MMX
--- /dev/null
+////////////////////////////////////////////////////////////////////////////////
+///
+/// Win32 version of the SSE optimized routines for Pentium-III, Athlon-XP and
+/// later. All SSE optimized functions have been gathered into this single source
+/// code file, regardless to their class or original source code file, in order
+/// to ease porting the library to other compiler and processor platforms.
+///
+/// NOTICE: If using Visual Studio 6.0, you'll need to install the "Visual C++
+/// 6.0 processor pack" update to support SSE instruction set. The update is
+/// available for download at Microsoft Developers Network, see here:
+/// http://msdn.microsoft.com/vstudio/downloads/tools/ppack/default.aspx
+///
+/// If the above URL is expired or removed, go to "http://msdn.microsoft.com" and
+/// perform a search with keywords "processor pack".
+///
+/// This file is to be compiled in Windows platform with Microsoft Visual C++
+/// Compiler. Please see 'sse_gcc.cpp' for the gcc compiler version for all
+/// GNU platforms (if file supplied).
+///
+/// Author : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai @ iki.fi
+/// SoundTouch WWW: http://www.iki.fi/oparviai/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// Last changed : $Date$
+// File revision : $Revision$
+//
+// $Id$
+//
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+// SoundTouch audio processing library
+// Copyright (c) Olli Parviainen
+//
+// This library is free software; you can redistribute it and/or
+// modify it under the terms of the GNU Lesser General Public
+// License as published by the Free Software Foundation; either
+// version 2.1 of the License, or (at your option) any later version.
+//
+// This library is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+// Lesser General Public License for more details.
+//
+// You should have received a copy of the GNU Lesser General Public
+// License along with this library; if not, write to the Free Software
+// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#include "cpu_detect.h"
+#include "STTypes.h"
+
+#ifndef WIN32
+#error "wrong platform - this source code file is exclusively for Win32 platform"
+#endif
+
+using namespace soundtouch;
+
+#ifdef ALLOW_SSE
+// SSE routines available only with float sample type
+
+//////////////////////////////////////////////////////////////////////////////
+//
+// implementation of SSE optimized functions of class 'TDStretchSSE'
+//
+//////////////////////////////////////////////////////////////////////////////
+
+#include "TDStretch.h"
+#include <limits.h>
+
+// these are declared in 'TDStretch.cpp'
+extern int scanOffsets[4][24];
+
+// Calculates cross correlation of two buffers
+double TDStretchSSE::calcCrossCorrStereo(const float *pV1, const float *pV2) const
+{
+ uint overlapLengthLocal = overlapLength;
+ float corr;
+
+ /*
+ double corr;
+ uint i;
+
+ // Calculates the cross-correlation value between 'pV1' and 'pV2' vectors
+ corr = 0.0;
+ for (i = 0; i < overlapLength / 8; 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] +
+ pV1[9] * pV2[9] +
+ pV1[10] * pV2[10] +
+ pV1[11] * pV2[11] +
+ pV1[12] * pV2[12] +
+ pV1[13] * pV2[13] +
+ pV1[14] * pV2[14] +
+ pV1[15] * pV2[15];
+
+ pV1 += 16;
+ pV2 += 16;
+ }
+ */
+
+ _asm
+ {
+ // Very important note: data in 'pV2' _must_ be aligned to
+ // 16-byte boundary!
+
+ // give prefetch hints to CPU of what data are to be needed soonish
+ // give more aggressive hints on pV1 as that changes while pV2 stays
+ // same between runs
+ prefetcht0 [pV1]
+ prefetcht0 [pV2]
+ prefetcht0 [pV1 + 32]
+
+ mov eax, dword ptr pV1
+ mov ebx, dword ptr pV2
+
+ xorps xmm0, xmm0
+
+ mov ecx, overlapLengthLocal
+ shr ecx, 3 // div by eight
+
+ loop1:
+ prefetcht0 [eax + 64] // give a prefetch hint to CPU what data are to be needed soonish
+ prefetcht0 [ebx + 32] // give a prefetch hint to CPU what data are to be needed soonish
+ movups xmm1, [eax]
+ mulps xmm1, [ebx]
+ addps xmm0, xmm1
+
+ movups xmm2, [eax + 16]
+ mulps xmm2, [ebx + 16]
+ addps xmm0, xmm2
+
+ prefetcht0 [eax + 96] // give a prefetch hint to CPU what data are to be needed soonish
+ prefetcht0 [ebx + 64] // give a prefetch hint to CPU what data are to be needed soonish
+
+ movups xmm3, [eax + 32]
+ mulps xmm3, [ebx + 32]
+ addps xmm0, xmm3
+
+ movups xmm4, [eax + 48]
+ mulps xmm4, [ebx + 48]
+ addps xmm0, xmm4
+
+ add eax, 64
+ add ebx, 64
+
+ dec ecx
+ jnz loop1
+
+ // add the four floats of xmm0 together and return the result.
+
+ movhlps xmm1, xmm0 // move 3 & 4 of xmm0 to 1 & 2 of xmm1
+ addps xmm1, xmm0
+ movaps xmm2, xmm1
+ shufps xmm2, xmm2, 0x01 // move 2 of xmm2 as 1 of xmm2
+ addss xmm2, xmm1
+ movss corr, xmm2
+ }
+
+ return (double)corr;
+}
+
+
+//////////////////////////////////////////////////////////////////////////////
+//
+// implementation of SSE optimized functions of class 'FIRFilter'
+//
+//////////////////////////////////////////////////////////////////////////////
+
+#include "FIRFilter.h"
+
+FIRFilterSSE::FIRFilterSSE() : FIRFilter()
+{
+ filterCoeffsUnalign = NULL;
+}
+
+
+FIRFilterSSE::~FIRFilterSSE()
+{
+ delete[] filterCoeffsUnalign;
+}
+
+
+// (overloaded) Calculates filter coefficients for SSE routine
+void FIRFilterSSE::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
+ delete[] filterCoeffsUnalign;
+ 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;
+ }
+}
+
+
+
+// SSE-optimized version of the filter routine for stereo sound
+uint FIRFilterSSE::evaluateFilterStereo(float *dest, const float *src, const uint numSamples) const
+{
+ int count = (numSamples - length) & -2;
+ uint lengthLocal = length / 8;
+ float *filterCoeffsLocal = filterCoeffsAlign;
+
+ assert(count % 2 == 0);
+
+ if (count < 2) return 0;
+
+ /*
+ double suml1, suml2;
+ double sumr1, sumr2;
+ uint i, j;
+
+ for (j = 0; j < count; j += 2)
+ {
+ const float *ptr;
+ const float *pFil;
+
+ suml1 = sumr1 = 0.0;
+ suml2 = sumr2 = 0.0;
+ ptr = src;
+ pFil = filterCoeffs;
+ for (i = 0; i < lengthLocal; i ++)
+ {
+ // unroll loop for efficiency.
+
+ suml1 += ptr[0] * pFil[0] +
+ ptr[2] * pFil[2] +
+ ptr[4] * pFil[4] +
+ ptr[6] * pFil[6];
+
+ sumr1 += ptr[1] * pFil[1] +
+ ptr[3] * pFil[3] +
+ ptr[5] * pFil[5] +
+ ptr[7] * pFil[7];
+
+ suml2 += ptr[8] * pFil[0] +
+ ptr[10] * pFil[2] +
+ ptr[12] * pFil[4] +
+ ptr[14] * pFil[6];
+
+ sumr2 += ptr[9] * pFil[1] +
+ ptr[11] * pFil[3] +
+ ptr[13] * pFil[5] +
+ ptr[15] * pFil[7];
+
+ ptr += 16;
+ pFil += 8;
+ }
+ dest[0] = (float)suml1;
+ dest[1] = (float)sumr1;
+ dest[2] = (float)suml2;
+ dest[3] = (float)sumr2;
+
+ src += 4;
+ dest += 4;
+ }
+ */
+
+ _asm
+ {
+ // Very important note: data in 'src' _must_ be aligned to
+ // 16-byte boundary!
+ mov edx, count
+ mov ebx, dword ptr src
+ mov eax, dword ptr dest
+ shr edx, 1
+
+ loop1:
+ // "outer loop" : during each round 2*2 output samples are calculated
+
+ // give prefetch hints to CPU of what data are to be needed soonish
+ prefetcht0 [ebx]
+ prefetcht0 [filterCoeffsLocal]
+
+ mov esi, ebx
+ mov edi, filterCoeffsLocal
+ xorps xmm0, xmm0
+ xorps xmm1, xmm1
+ mov ecx, lengthLocal
+
+ loop2:
+ // "inner loop" : during each round eight FIR filter taps are evaluated for 2*2 samples
+ prefetcht0 [esi + 32] // give a prefetch hint to CPU what data are to be needed soonish
+ prefetcht0 [edi + 32] // give a prefetch hint to CPU what data are to be needed soonish
+
+ movups xmm2, [esi] // possibly unaligned load
+ movups xmm3, [esi + 8] // possibly unaligned load
+ mulps xmm2, [edi]
+ mulps xmm3, [edi]
+ addps xmm0, xmm2
+ addps xmm1, xmm3
+
+ movups xmm4, [esi + 16] // possibly unaligned load
+ movups xmm5, [esi + 24] // possibly unaligned load
+ mulps xmm4, [edi + 16]
+ mulps xmm5, [edi + 16]
+ addps xmm0, xmm4
+ addps xmm1, xmm5
+
+ prefetcht0 [esi + 64] // give a prefetch hint to CPU what data are to be needed soonish
+ prefetcht0 [edi + 64] // give a prefetch hint to CPU what data are to be needed soonish
+
+ movups xmm6, [esi + 32] // possibly unaligned load
+ movups xmm7, [esi + 40] // possibly unaligned load
+ mulps xmm6, [edi + 32]
+ mulps xmm7, [edi + 32]
+ addps xmm0, xmm6
+ addps xmm1, xmm7
+
+ movups xmm4, [esi + 48] // possibly unaligned load
+ movups xmm5, [esi + 56] // possibly unaligned load
+ mulps xmm4, [edi + 48]
+ mulps xmm5, [edi + 48]
+ addps xmm0, xmm4
+ addps xmm1, xmm5
+
+ add esi, 64
+ add edi, 64
+ dec ecx
+ jnz loop2
+
+ // Now xmm0 and xmm1 both have a filtered 2-channel sample each, but we still need
+ // to sum the two hi- and lo-floats of these registers together.
+
+ movhlps xmm2, xmm0 // xmm2 = xmm2_3 xmm2_2 xmm0_3 xmm0_2
+ movlhps xmm2, xmm1 // xmm2 = xmm1_1 xmm1_0 xmm0_3 xmm0_2
+ shufps xmm0, xmm1, 0xe4 // xmm0 = xmm1_3 xmm1_2 xmm0_1 xmm0_0
+ addps xmm0, xmm2
+
+ movaps [eax], xmm0
+ add ebx, 16
+ add eax, 16
+
+ dec edx
+ jnz loop1
+ }
+
+ return (uint)count;
+}
+
+#endif // ALLOW_SSE
projects / ardour.git / commitdiff