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[OMPT-svn-commit] SF.net SVN: modplug:[227] trunk/OpenMPT
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From: <rel...@us...> - 2008-10-13 14:26:27
|
Revision: 227
http://modplug.svn.sourceforge.net/modplug/?rev=227&view=rev
Author: relabsoluness
Date: 2008-10-13 14:25:58 +0000 (Mon, 13 Oct 2008)
Log Message:
-----------
+ Added SoundTouch library to SVN(LGPL); to be used for time stretching.
Added Paths:
-----------
trunk/OpenMPT/soundtouch/
trunk/OpenMPT/soundtouch/3dnow_win.cpp
trunk/OpenMPT/soundtouch/AAFilter.cpp
trunk/OpenMPT/soundtouch/AAFilter.h
trunk/OpenMPT/soundtouch/BPMDetect.h
trunk/OpenMPT/soundtouch/COPYING.TXT
trunk/OpenMPT/soundtouch/FIFOSampleBuffer.cpp
trunk/OpenMPT/soundtouch/FIFOSampleBuffer.h
trunk/OpenMPT/soundtouch/FIFOSamplePipe.h
trunk/OpenMPT/soundtouch/FIRFilter.cpp
trunk/OpenMPT/soundtouch/FIRFilter.h
trunk/OpenMPT/soundtouch/README.html
trunk/OpenMPT/soundtouch/RateTransposer.cpp
trunk/OpenMPT/soundtouch/RateTransposer.h
trunk/OpenMPT/soundtouch/STTypes.h
trunk/OpenMPT/soundtouch/SoundTouch.cpp
trunk/OpenMPT/soundtouch/SoundTouch.h
trunk/OpenMPT/soundtouch/TDStretch.cpp
trunk/OpenMPT/soundtouch/TDStretch.h
trunk/OpenMPT/soundtouch/cpu_detect.h
trunk/OpenMPT/soundtouch/cpu_detect_x86_win.cpp
trunk/OpenMPT/soundtouch/mmx_optimized.cpp
trunk/OpenMPT/soundtouch/soundtouch-1.3.1.zip
trunk/OpenMPT/soundtouch/soundtouch.vcproj
trunk/OpenMPT/soundtouch/sse_optimized.cpp
Added: trunk/OpenMPT/soundtouch/3dnow_win.cpp
===================================================================
--- trunk/OpenMPT/soundtouch/3dnow_win.cpp (rev 0)
+++ trunk/OpenMPT/soundtouch/3dnow_win.cpp 2008-10-13 14:25:58 UTC (rev 227)
@@ -0,0 +1,350 @@
+////////////////////////////////////////////////////////////////////////////////
+///
+/// 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 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// Last changed : $Date: 2006/02/05 16:44:06 $
+// File revision : $Revision: 1.10 $
+//
+// $Id: 3dnow_win.cpp,v 1.10 2006/02/05 16:44:06 Olli Exp $
+//
+////////////////////////////////////////////////////////////////////////////////
+//
+// 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
Added: trunk/OpenMPT/soundtouch/AAFilter.cpp
===================================================================
--- trunk/OpenMPT/soundtouch/AAFilter.cpp (rev 0)
+++ trunk/OpenMPT/soundtouch/AAFilter.cpp 2008-10-13 14:25:58 UTC (rev 227)
@@ -0,0 +1,184 @@
+////////////////////////////////////////////////////////////////////////////////
+///
+/// FIR low-pass (anti-alias) filter with filter coefficient design routine and
+/// MMX optimization.
+///
+/// Anti-alias filter is used to prevent folding of high frequencies when
+/// transposing the sample rate with interpolation.
+///
+/// Author : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// Last changed : $Date: 2006/02/05 16:44:06 $
+// File revision : $Revision: 1.9 $
+//
+// $Id: AAFilter.cpp,v 1.9 2006/02/05 16:44:06 Olli Exp $
+//
+////////////////////////////////////////////////////////////////////////////////
+//
+// 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();
+}
Added: trunk/OpenMPT/soundtouch/AAFilter.h
===================================================================
--- trunk/OpenMPT/soundtouch/AAFilter.h (rev 0)
+++ trunk/OpenMPT/soundtouch/AAFilter.h 2008-10-13 14:25:58 UTC (rev 227)
@@ -0,0 +1,91 @@
+////////////////////////////////////////////////////////////////////////////////
+///
+/// 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 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// Last changed : $Date: 2006/02/05 16:44:06 $
+// File revision : $Revision: 1.10 $
+//
+// $Id: AAFilter.h,v 1.10 2006/02/05 16:44:06 Olli Exp $
+//
+////////////////////////////////////////////////////////////////////////////////
+//
+// 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
Added: trunk/OpenMPT/soundtouch/BPMDetect.h
===================================================================
--- trunk/OpenMPT/soundtouch/BPMDetect.h (rev 0)
+++ trunk/OpenMPT/soundtouch/BPMDetect.h 2008-10-13 14:25:58 UTC (rev 227)
@@ -0,0 +1,159 @@
+////////////////////////////////////////////////////////////////////////////////
+///
+/// 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 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// Last changed : $Date: 2006/02/05 16:44:06 $
+// File revision : $Revision: 1.5 $
+//
+// $Id: BPMDetect.h,v 1.5 2006/02/05 16:44:06 Olli Exp $
+//
+////////////////////////////////////////////////////////////////////////////////
+//
+// 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_
Added: trunk/OpenMPT/soundtouch/COPYING.TXT
===================================================================
--- trunk/OpenMPT/soundtouch/COPYING.TXT (rev 0)
+++ trunk/OpenMPT/soundtouch/COPYING.TXT 2008-10-13 14:25:58 UTC (rev 227)
@@ -0,0 +1,458 @@
+ GNU LESSER GENERAL PUBLIC LICENSE
+ Version 2.1, February 1999
+
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+ 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ Everyone is permitted to copy and distribute verbatim copies
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Added: trunk/OpenMPT/soundtouch/FIFOSampleBuffer.cpp
===================================================================
--- trunk/OpenMPT/soundtouch/FIFOSampleBuffer.cpp (rev 0)
+++ trunk/OpenMPT/soundtouch/FIFOSampleBuffer.cpp 2008-10-13 14:25:58 UTC (rev 227)
@@ -0,0 +1,252 @@
+////////////////////////////////////////////////////////////////////////////////
+///
+/// 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 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// Last changed : $Date: 2006/02/05 16:44:06 $
+// File revision : $Revision: 1.11 $
+//
+// $Id: FIFOSampleBuffer.cpp,v 1.11 2006/02/05 16:44:06 Olli Exp $
+//
+////////////////////////////////////////////////////////////////////////////////
+//
+// 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
+{
+...
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