diff options
| author |  waker <wakeroid@gmail.com> | 2011-03-23 21:26:26 +0100 |
|---|---|---|
| committer | waker <wakeroid@gmail.com> | 2011-03-23 21:26:26 +0100 |
| commit | fd302d7abc36942e7ff14b22fae1e72b4495bef1 (patch) | |
| tree | 2a36f8361c907a5bea91a9d905957a709f31ea64 /plugins/soundtouch/soundtouch/source | |
| parent | 11e63b53b8c91da89592c373bb32fc2b656c6024 (diff) | |
renamed soundtouch and uade2 plugin folders and output .so
Diffstat (limited to 'plugins/soundtouch/soundtouch/source')
19 files changed, 5702 insertions, 0 deletions
diff --git a/plugins/soundtouch/soundtouch/source/SoundTouch/3dnow_win.cpp b/plugins/soundtouch/soundtouch/source/SoundTouch/3dnow_win.cpp new file mode 100644 index 00000000..f0a9d7ec --- /dev/null +++ b/plugins/soundtouch/soundtouch/source/SoundTouch/3dnow_win.cpp @@ -0,0 +1,349 @@ +////////////////////////////////////////////////////////////////////////////////
+///
+/// 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/en-us/vstudio/aa718349.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: 2009-02-21 18:00:14 +0200 (Sat, 21 Feb 2009) $
+// File revision : $Revision: 4 $
+//
+// $Id: 3dnow_win.cpp 63 2009-02-21 16:00:14Z oparviai $
+//
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+// SoundTouch audio processing library
+// Copyright (c) Olli Parviainen
+//
+// This library is free software; you can redistribute it and/or
+// modify it under the terms of the GNU Lesser General Public
+// License as published by the Free Software Foundation; either
+// version 2.1 of the License, or (at your option) any later version.
+//
+// This library is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+// Lesser General Public License for more details.
+//
+// You should have received a copy of the GNU Lesser General Public
+// License along with this library; if not, write to the Free Software
+// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#include "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"
+
+
+// Calculates cross correlation of two buffers
+double TDStretch3DNow::calcCrossCorrStereo(const float *pV1, const float *pV2) const
+{
+ int overlapLengthLocal = overlapLength;
+ float corr = 0;
+
+ // 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;
+ filterCoeffsAlign = NULL;
+}
+
+
+FIRFilter3DNow::~FIRFilter3DNow()
+{
+ delete[] filterCoeffsUnalign;
+ filterCoeffsUnalign = NULL;
+ filterCoeffsAlign = NULL;
+}
+
+
+// (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) & (uint)-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, uint numSamples) const
+{
+ float *filterCoeffsLocal = filterCoeffsAlign;
+ uint count = (numSamples - length) & (uint)-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
diff --git a/plugins/soundtouch/soundtouch/source/SoundTouch/AAFilter.cpp b/plugins/soundtouch/soundtouch/source/SoundTouch/AAFilter.cpp new file mode 100644 index 00000000..96abda49 --- /dev/null +++ b/plugins/soundtouch/soundtouch/source/SoundTouch/AAFilter.cpp @@ -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: 2009-01-11 13:34:24 +0200 (Sun, 11 Jan 2009) $
+// File revision : $Revision: 4 $
+//
+// $Id: AAFilter.cpp 45 2009-01-11 11:34:24Z oparviai $
+//
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+// SoundTouch audio processing library
+// Copyright (c) Olli Parviainen
+//
+// This library is free software; you can redistribute it and/or
+// modify it under the terms of the GNU Lesser General Public
+// License as published by the Free Software Foundation; either
+// version 2.1 of the License, or (at your option) any later version.
+//
+// This library is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+// Lesser General Public License for more details.
+//
+// You should have received a copy of the GNU Lesser General Public
+// License along with this library; if not, write to the Free Software
+// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#include <memory.h>
+#include <assert.h>
+#include <math.h>
+#include <stdlib.h>
+#include "AAFilter.h"
+#include "FIRFilter.h"
+
+using namespace soundtouch;
+
+#define PI 3.141592655357989
+#define TWOPI (2 * PI)
+
+/*****************************************************************************
+ *
+ * Implementation of the class 'AAFilter'
+ *
+ *****************************************************************************/
+
+AAFilter::AAFilter(uint len)
+{
+ pFIR = FIRFilter::newInstance();
+ cutoffFreq = 0.5;
+ setLength(len);
+}
+
+
+
+AAFilter::~AAFilter()
+{
+ delete pFIR;
+}
+
+
+
+// Sets new anti-alias filter cut-off edge frequency, scaled to
+// sampling frequency (nyquist frequency = 0.5).
+// The filter will cut frequencies higher than the given frequency.
+void AAFilter::setCutoffFreq(double newCutoffFreq)
+{
+ cutoffFreq = newCutoffFreq;
+ calculateCoeffs();
+}
+
+
+
+// Sets number of FIR filter taps
+void AAFilter::setLength(uint newLength)
+{
+ length = newLength;
+ calculateCoeffs();
+}
+
+
+
+// Calculates coefficients for a low-pass FIR filter using Hamming window
+void AAFilter::calculateCoeffs()
+{
+ uint i;
+ double cntTemp, temp, tempCoeff,h, w;
+ double fc2, wc;
+ double scaleCoeff, sum;
+ double *work;
+ SAMPLETYPE *coeffs;
+
+ assert(length >= 2);
+ assert(length % 4 == 0);
+ assert(cutoffFreq >= 0);
+ assert(cutoffFreq <= 0.5);
+
+ work = new double[length];
+ coeffs = new SAMPLETYPE[length];
+
+ 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();
+}
diff --git a/plugins/soundtouch/soundtouch/source/SoundTouch/AAFilter.h b/plugins/soundtouch/soundtouch/source/SoundTouch/AAFilter.h new file mode 100644 index 00000000..d5c8ce4c --- /dev/null +++ b/plugins/soundtouch/soundtouch/source/SoundTouch/AAFilter.h @@ -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: 2008-02-10 18:26:55 +0200 (Sun, 10 Feb 2008) $
+// File revision : $Revision: 4 $
+//
+// $Id: AAFilter.h 11 2008-02-10 16:26:55Z oparviai $
+//
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+// SoundTouch audio processing library
+// Copyright (c) Olli Parviainen
+//
+// This library is free software; you can redistribute it and/or
+// modify it under the terms of the GNU Lesser General Public
+// License as published by the Free Software Foundation; either
+// version 2.1 of the License, or (at your option) any later version.
+//
+// This library is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+// Lesser General Public License for more details.
+//
+// You should have received a copy of the GNU Lesser General Public
+// License along with this library; if not, write to the Free Software
+// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#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
diff --git a/plugins/soundtouch/soundtouch/source/SoundTouch/BPMDetect.cpp b/plugins/soundtouch/soundtouch/source/SoundTouch/BPMDetect.cpp new file mode 100644 index 00000000..405f514b --- /dev/null +++ b/plugins/soundtouch/soundtouch/source/SoundTouch/BPMDetect.cpp @@ -0,0 +1,308 @@ +////////////////////////////////////////////////////////////////////////////////
+///
+/// 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.
+/// - Inputted 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: 2009-02-21 18:00:14 +0200 (Sat, 21 Feb 2009) $
+// File revision : $Revision: 4 $
+//
+// $Id: BPMDetect.cpp 63 2009-02-21 16:00:14Z oparviai $
+//
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+// SoundTouch audio processing library
+// Copyright (c) Olli Parviainen
+//
+// This library is free software; you can redistribute it and/or
+// modify it under the terms of the GNU Lesser General Public
+// License as published by the Free Software Foundation; either
+// version 2.1 of the License, or (at your option) any later version.
+//
+// This library is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+// Lesser General Public License for more details.
+//
+// You should have received a copy of the GNU Lesser General Public
+// License along with this library; if not, write to the Free Software
+// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#include <math.h>
+#include <assert.h>
+#include <string.h>
+#include "FIFOSampleBuffer.h"
+#include "PeakFinder.h"
+#include "BPMDetect.h"
+
+using namespace soundtouch;
+
+#define INPUT_BLOCK_SAMPLES 2048
+#define DECIMATED_BLOCK_SAMPLES 256
+
+/// decay constant for calculating RMS volume sliding average approximation
+/// (time constant is about 10 sec)
+const float avgdecay = 0.99986f;
+
+/// Normalization coefficient for calculating RMS sliding average approximation.
+const float avgnorm = (1 - avgdecay);
+
+
+
+BPMDetect::BPMDetect(int numChannels, int aSampleRate)
+{
+ this->sampleRate = aSampleRate;
+ this->channels = numChannels;
+
+ decimateSum = 0;
+ decimateCount = 0;
+
+ envelopeAccu = 0;
+
+ // Initialize RMS volume accumulator to RMS level of 3000 (out of 32768) that's
+ // a typical RMS signal level value for song data. This value is then adapted
+ // to the actual level during processing.
+#ifdef INTEGER_SAMPLES
+ // integer samples
+ RMSVolumeAccu = (3000 * 3000) / avgnorm;
+#else
+ // float samples, scaled to range [-1..+1[
+ RMSVolumeAccu = (0.092f * 0.092f) / avgnorm;
+#endif
+
+ // choose decimation factor so that result is approx. 500 Hz
+ decimateBy = sampleRate / 500;
+ assert(decimateBy > 0);
+ assert(INPUT_BLOCK_SAMPLES < decimateBy * DECIMATED_BLOCK_SAMPLES);
+
+ // Calculate window length & starting item according to desired min & max bpms
+ windowLen = (60 * sampleRate) / (decimateBy * MIN_BPM);
+ windowStart = (60 * sampleRate) / (decimateBy * MAX_BPM);
+
+ assert(windowLen > windowStart);
+
+ // allocate new working objects
+ xcorr = new float[windowLen];
+ memset(xcorr, 0, windowLen * sizeof(float));
+
+ // allocate processing buffer
+ buffer = new FIFOSampleBuffer();
+ // we do processing in mono mode
+ buffer->setChannels(1);
+ buffer->clear();
+}
+
+
+
+BPMDetect::~BPMDetect()
+{
+ delete[] xcorr;
+ delete buffer;
+}
+
+
+
+/// convert to mono, low-pass filter & decimate to about 500 Hz.
+/// return number of outputted samples.
+///
+/// Decimation is used to remove the unnecessary frequencies and thus to reduce
+/// the amount of data needed to be processed as calculating autocorrelation
+/// function is a very-very heavy operation.
+///
+/// Anti-alias filtering is done simply by averaging the samples. This is really a
+/// poor-man's anti-alias filtering, but it's not so critical in this kind of application
+/// (it'd also be difficult to design a high-quality filter with steep cut-off at very
+/// narrow band)
+int BPMDetect::decimate(SAMPLETYPE *dest, const SAMPLETYPE *src, int numsamples)
+{
+ int count, outcount;
+ LONG_SAMPLETYPE out;
+
+ assert(channels > 0);
+ assert(decimateBy > 0);
+ outcount = 0;
+ for (count = 0; count < numsamples; count ++)
+ {
+ int j;
+
+ // convert to mono and accumulate
+ for (j = 0; j < channels; j ++)
+ {
+ decimateSum += src[j];
+ }
+ src += j;
+
+ decimateCount ++;
+ if (decimateCount >= decimateBy)
+ {
+ // Store every Nth sample only
+ out = (LONG_SAMPLETYPE)(decimateSum / (decimateBy * channels));
+ decimateSum = 0;
+ decimateCount = 0;
+#ifdef INTEGER_SAMPLES
+ // check ranges for sure (shouldn't actually be necessary)
+ if (out > 32767)
+ {
+ out = 32767;
+ }
+ else if (out < -32768)
+ {
+ out = -32768;
+ }
+#endif // INTEGER_SAMPLES
+ dest[outcount] = (SAMPLETYPE)out;
+ outcount ++;
+ }
+ }
+ return outcount;
+}
+
+
+
+// Calculates autocorrelation function of the sample history buffer
+void BPMDetect::updateXCorr(int process_samples)
+{
+ int offs;
+ SAMPLETYPE *pBuffer;
+
+ assert(buffer->numSamples() >= (uint)(process_samples + windowLen));
+
+ pBuffer = buffer->ptrBegin();
+ for (offs = windowStart; offs < windowLen; offs ++)
+ {
+ LONG_SAMPLETYPE sum;
+ int i;
+
+ sum = 0;
+ for (i = 0; i < process_samples; i ++)
+ {
+ sum += pBuffer[i] * pBuffer[i + offs]; // scaling the sub-result shouldn't be necessary
+ }
+// xcorr[offs] *= xcorr_decay; // decay 'xcorr' here with suitable coefficients
+ // if it's desired that the system adapts automatically to
+ // various bpms, e.g. in processing continouos music stream.
+ // The 'xcorr_decay' should be a value that's smaller than but
+ // close to one, and should also depend on 'process_samples' value.
+
+ xcorr[offs] += (float)sum;
+ }
+}
+
+
+
+// Calculates envelope of the sample data
+void BPMDetect::calcEnvelope(SAMPLETYPE *samples, int numsamples)
+{
+ const float decay = 0.7f; // decay constant for smoothing the envelope
+ const float norm = (1 - decay);
+
+ int i;
+ LONG_SAMPLETYPE out;
+ float val;
+
+ for (i = 0; i < numsamples; i ++)
+ {
+ // calc average RMS volume
+ RMSVolumeAccu *= avgdecay;
+ val = (float)fabs((float)samples[i]);
+ RMSVolumeAccu += val * val;
+
+ // cut amplitudes that are below 2 times average RMS volume
+ // (we're interested in peak values, not the silent moments)
+ val -= 2 * (float)sqrt(RMSVolumeAccu * avgnorm);
+ val = (val > 0) ? val : 0;
+
+ // smooth amplitude envelope
+ envelopeAccu *= decay;
+ envelopeAccu += val;
+ out = (LONG_SAMPLETYPE)(envelopeAccu * norm);
+
+#ifdef INTEGER_SAMPLES
+ // cut peaks (shouldn't be necessary though)
+ if (out > 32767) out = 32767;
+#endif // INTEGER_SAMPLES
+ samples[i] = (SAMPLETYPE)out;
+ }
+}
+
+
+
+void BPMDetect::inputSamples(const SAMPLETYPE *samples, int numSamples)
+{
+ SAMPLETYPE decimated[DECIMATED_BLOCK_SAMPLES];
+
+ // iterate so that max INPUT_BLOCK_SAMPLES processed per iteration
+ while (numSamples > 0)
+ {
+ int block;
+ int decSamples;
+
+ block = (numSamples > INPUT_BLOCK_SAMPLES) ? INPUT_BLOCK_SAMPLES : numSamples;
+
+ // decimate. note that converts to mono at the same time
+ decSamples = decimate(decimated, samples, block);
+ samples += block * channels;
+ numSamples -= block;
+
+ // envelope new samples and add them to buffer
+ calcEnvelope(decimated, decSamples);
+ buffer->putSamples(decimated, decSamples);
+ }
+
+ // when the buffer has enought samples for processing...
+ if ((int)buffer->numSamples() > windowLen)
+ {
+ int processLength;
+
+ // how many samples are processed
+ processLength = (int)buffer->numSamples() - windowLen;
+
+ // ... calculate autocorrelations for oldest samples...
+ updateXCorr(processLength);
+ // ... and remove them from the buffer
+ buffer->receiveSamples(processLength);
+ }
+}
+
+
+
+float BPMDetect::getBpm()
+{
+ double peakPos;
+ PeakFinder peakFinder;
+
+ // find peak position
+ peakPos = peakFinder.detectPeak(xcorr, windowStart, windowLen);
+
+ assert(decimateBy != 0);
+ if (peakPos < 1e-6) return 0.0; // detection failed.
+
+ // calculate BPM
+ return (float)(60.0 * (((double)sampleRate / (double)decimateBy) / peakPos));
+}
diff --git a/plugins/soundtouch/soundtouch/source/SoundTouch/FIFOSampleBuffer.cpp b/plugins/soundtouch/soundtouch/source/SoundTouch/FIFOSampleBuffer.cpp new file mode 100644 index 00000000..01f64b08 --- /dev/null +++ b/plugins/soundtouch/soundtouch/source/SoundTouch/FIFOSampleBuffer.cpp @@ -0,0 +1,262 @@ +////////////////////////////////////////////////////////////////////////////////
+///
+/// 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: 2009-02-27 19:24:42 +0200 (Fri, 27 Feb 2009) $
+// File revision : $Revision: 4 $
+//
+// $Id: FIFOSampleBuffer.cpp 68 2009-02-27 17:24:42Z oparviai $
+//
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+// SoundTouch audio processing library
+// Copyright (c) Olli Parviainen
+//
+// This library is free software; you can redistribute it and/or
+// modify it under the terms of the GNU Lesser General Public
+// License as published by the Free Software Foundation; either
+// version 2.1 of the License, or (at your option) any later version.
+//
+// This library is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+// Lesser General Public License for more details.
+//
+// You should have received a copy of the GNU Lesser General Public
+// License along with this library; if not, write to the Free Software
+// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#include <stdlib.h>
+#include <memory.h>
+#include <string.h>
+#include <assert.h>
+#include <stdexcept>
+
+#include "FIFOSampleBuffer.h"
+
+using namespace soundtouch;
+
+// Constructor
+FIFOSampleBuffer::FIFOSampleBuffer(int numChannels)
+{
+ assert(numChannels > 0);
+ sizeInBytes = 0; // reasonable initial value
+ buffer = NULL;
+ bufferUnaligned = NULL;
+ samplesInBuffer = 0;
+ bufferPos = 0;
+ channels = (uint)numChannels;
+ ensureCapacity(32); // allocate initial capacity
+}
+
+
+// destructor
+FIFOSampleBuffer::~FIFOSampleBuffer()
+{
+ delete[] bufferUnaligned;
+ bufferUnaligned = NULL;
+ buffer = NULL;
+}
+
+
+// Sets number of channels, 1 = mono, 2 = stereo
+void FIFOSampleBuffer::setChannels(int numChannels)
+{
+ uint usedBytes;
+
+ assert(numChannels > 0);
+ usedBytes = channels * samplesInBuffer;
+ channels = (uint)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 (buffer && 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 nSamples)
+{
+ memcpy(ptrEnd(nSamples), samples, sizeof(SAMPLETYPE) * nSamples * channels);
+ samplesInBuffer += nSamples;
+}
+
+
+// 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 nSamples)
+{
+ uint req;
+
+ req = samplesInBuffer + nSamples;
+ ensureCapacity(req);
+ samplesInBuffer += nSamples;
+}
+
+
+// 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()
+{
+ assert(buffer);
+ 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) & (uint)-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");
+ }
+ // Align the buffer to begin at 16byte cache line boundary for optimal performance
+ temp = (SAMPLETYPE *)(((ulong)tempUnaligned + 15) & (ulong)-16);
+ if (samplesInBuffer)
+ {
+ 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;
+}
diff --git a/plugins/soundtouch/soundtouch/source/SoundTouch/FIRFilter.cpp b/plugins/soundtouch/soundtouch/source/SoundTouch/FIRFilter.cpp new file mode 100644 index 00000000..231263ad --- /dev/null +++ b/plugins/soundtouch/soundtouch/source/SoundTouch/FIRFilter.cpp @@ -0,0 +1,269 @@ +////////////////////////////////////////////////////////////////////////////////
+///
+/// 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 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// Last changed : $Date: 2009-02-25 19:13:51 +0200 (Wed, 25 Feb 2009) $
+// File revision : $Revision: 4 $
+//
+// $Id: FIRFilter.cpp 67 2009-02-25 17:13:51Z oparviai $
+//
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+// SoundTouch audio processing library
+// Copyright (c) Olli Parviainen
+//
+// This library is free software; you can redistribute it and/or
+// modify it under the terms of the GNU Lesser General Public
+// License as published by the Free Software Foundation; either
+// version 2.1 of the License, or (at your option) any later version.
+//
+// This library is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+// Lesser General Public License for more details.
+//
+// You should have received a copy of the GNU Lesser General Public
+// License along with this library; if not, write to the Free Software
+// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#include <memory.h>
+#include <assert.h>
+#include <math.h>
+#include <stdlib.h>
+#include <stdexcept>
+#include "FIRFilter.h"
+#include "cpu_detect.h"
+
+using namespace soundtouch;
+
+/*****************************************************************************
+ *
+ * Implementation of the class 'FIRFilter'
+ *
+ *****************************************************************************/
+
+FIRFilter::FIRFilter()
+{
+ resultDivFactor = 0;
+ resultDivider = 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);
+ assert(src != NULL);
+ assert(dest != NULL);
+ assert(filterCoeffs != NULL);
+
+ 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 = (SAMPLETYPE)::pow(2.0, (int)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;
+ if (numChannels == 2)
+ {
+ return evaluateFilterStereo(dest, src, numSamples);
+ } else {
+ return evaluateFilterMono(dest, src, numSamples);
+ }
+}
+
+
+
+// Operator 'new' is overloaded so that it automatically creates a suitable instance
+// depending on if we've a MMX-capable CPU available or not.
+void * FIRFilter::operator new(size_t s)
+{
+ // Notice! don't use "new FIRFilter" directly, use "newInstance" to create a new instance instead!
+ throw std::runtime_error("Error in FIRFilter::new: Don't use 'new FIRFilter', use 'newInstance' member instead!");
+ return NULL;
+}
+
+
+FIRFilter * FIRFilter::newInstance()
+{
+ uint uExtensions;
+
+ uExtensions = detectCPUextensions();
+
+ // Check if MMX/SSE/3DNow! instruction set extensions supported by CPU
+
+#ifdef ALLOW_MMX
+ // MMX routines available only with integer sample types
+ if (uExtensions & SUPPORT_MMX)
+ {
+ return ::new FIRFilterMMX;
+ }
+ else
+#endif // ALLOW_MMX
+
+#ifdef ALLOW_SSE
+ if (uExtensions & SUPPORT_SSE)
+ {
+ // SSE support
+ return ::new FIRFilterSSE;
+ }
+ else
+#endif // ALLOW_SSE
+
+#ifdef ALLOW_3DNOW
+ if (uExtensions & SUPPORT_3DNOW)
+ {
+ // 3DNow! support
+ return ::new FIRFilter3DNow;
+ }
+ else
+#endif // ALLOW_3DNOW
+
+ {
+ // ISA optimizations not supported, use plain C version
+ return ::new FIRFilter;
+ }
+}
diff --git a/plugins/soundtouch/soundtouch/source/SoundTouch/FIRFilter.h b/plugins/soundtouch/soundtouch/source/SoundTouch/FIRFilter.h new file mode 100644 index 00000000..5713f7bb --- /dev/null +++ b/plugins/soundtouch/soundtouch/source/SoundTouch/FIRFilter.h @@ -0,0 +1,164 @@ +////////////////////////////////////////////////////////////////////////////////
+///
+/// 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 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// Last changed : $Date: 2009-02-21 18:00:14 +0200 (Sat, 21 Feb 2009) $
+// File revision : $Revision: 4 $
+//
+// $Id: FIRFilter.h 63 2009-02-21 16:00:14Z oparviai $
+//
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+// SoundTouch audio processing library
+// Copyright (c) Olli Parviainen
+//
+// This library is free software; you can redistribute it and/or
+// modify it under the terms of the GNU Lesser General Public
+// License as published by the Free Software Foundation; either
+// version 2.1 of the License, or (at your option) any later version.
+//
+// This library is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+// Lesser General Public License for more details.
+//
+// You should have received a copy of the GNU Lesser General Public
+// License along with this library; if not, write to the Free Software
+// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#ifndef FIRFilter_H
+#define FIRFilter_H
+
+#include <stddef.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;
+
+public:
+ FIRFilter();
+ virtual ~FIRFilter();
+
+ /// Operator 'new' is overloaded so that it automatically creates a suitable instance
+ /// depending on if we've a MMX-capable CPU available or not.
+ static void * operator new(size_t s);
+
+ static FIRFilter *newInstance();
+
+ /// Applies the filter to the given sequence of samples.
+ /// Note : The amount of outputted samples is by value of 'filter_length'
+ /// smaller than the amount of input samples.
+ ///
+ /// \return Number of samples copied to 'dest'.
+ uint evaluate(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
diff --git a/plugins/soundtouch/soundtouch/source/SoundTouch/PeakFinder.cpp b/plugins/soundtouch/soundtouch/source/SoundTouch/PeakFinder.cpp new file mode 100644 index 00000000..03f60bfa --- /dev/null +++ b/plugins/soundtouch/soundtouch/source/SoundTouch/PeakFinder.cpp @@ -0,0 +1,239 @@ +////////////////////////////////////////////////////////////////////////////////
+///
+/// Peak detection routine.
+///
+/// The routine detects highest value on an array of values and calculates the
+/// precise peak location as a mass-center of the 'hump' around the peak value.
+///
+/// Author : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// Last changed : $Date: 2009-02-21 18:00:14 +0200 (Sat, 21 Feb 2009) $
+// File revision : $Revision: 4 $
+//
+// $Id: PeakFinder.cpp 63 2009-02-21 16:00:14Z oparviai $
+//
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+// SoundTouch audio processing library
+// Copyright (c) Olli Parviainen
+//
+// This library is free software; you can redistribute it and/or
+// modify it under the terms of the GNU Lesser General Public
+// License as published by the Free Software Foundation; either
+// version 2.1 of the License, or (at your option) any later version.
+//
+// This library is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+// Lesser General Public License for more details.
+//
+// You should have received a copy of the GNU Lesser General Public
+// License along with this library; if not, write to the Free Software
+// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#include <math.h>
+#include <assert.h>
+
+#include "PeakFinder.h"
+
+using namespace soundtouch;
+
+#define max(x, y) (((x) > (y)) ? (x) : (y))
+
+
+PeakFinder::PeakFinder()
+{
+ minPos = maxPos = 0;
+}
+
+
+// Finds 'ground level' of a peak hump by starting from 'peakpos' and proceeding
+// to direction defined by 'direction' until next 'hump' after minimum value will
+// begin
+int PeakFinder::findGround(const float *data, int peakpos, int direction) const
+{
+ float refvalue;
+ int lowpos;
+ int pos;
+ int climb_count;
+ float delta;
+
+ climb_count = 0;
+ refvalue = data[peakpos];
+ lowpos = peakpos;
+
+ pos = peakpos;
+
+ while ((pos > minPos) && (pos < maxPos))
+ {
+ int prevpos;
+
+ prevpos = pos;
+ pos += direction;
+
+ // calculate derivate
+ delta = data[pos] - data[prevpos];
+ if (delta <= 0)
+ {
+ // going downhill, ok
+ if (climb_count)
+ {
+ climb_count --; // decrease climb count
+ }
+
+ // check if new minimum found
+ if (data[pos] < refvalue)
+ {
+ // new minimum found
+ lowpos = pos;
+ refvalue = data[pos];
+ }
+ }
+ else
+ {
+ // going uphill, increase climbing counter
+ climb_count ++;
+ if (climb_count > 5) break; // we've been climbing too long => it's next uphill => quit
+ }
+ }
+ return lowpos;
+}
+
+
+// Find offset where the value crosses the given level, when starting from 'peakpos' and
+// proceeds to direction defined in 'direction'
+int PeakFinder::findCrossingLevel(const float *data, float level, int peakpos, int direction) const
+{
+ float peaklevel;
+ int pos;
+
+ peaklevel = data[peakpos];
+ assert(peaklevel >= level);
+ pos = peakpos;
+ while ((pos >= minPos) && (pos < maxPos))
+ {
+ if (data[pos + direction] < level) return pos; // crossing found
+ pos += direction;
+ }
+ return -1; // not found
+}
+
+
+// Calculates the center of mass location of 'data' array items between 'firstPos' and 'lastPos'
+double PeakFinder::calcMassCenter(const float *data, int firstPos, int lastPos) const
+{
+ int i;
+ float sum;
+ float wsum;
+
+ sum = 0;
+ wsum = 0;
+ for (i = firstPos; i <= lastPos; i ++)
+ {
+ sum += (float)i * data[i];
+ wsum += data[i];
+ }
+
+ if (wsum < 1e-6) return 0;
+ return sum / wsum;
+}
+
+
+
+/// get exact center of peak near given position by calculating local mass of center
+double PeakFinder::getPeakCenter(const float *data, int peakpos) const
+{
+ float peakLevel; // peak level
+ int crosspos1, crosspos2; // position where the peak 'hump' crosses cutting level
+ float cutLevel; // cutting value
+ float groundLevel; // ground level of the peak
+ int gp1, gp2; // bottom positions of the peak 'hump'
+
+ // find ground positions.
+ gp1 = findGround(data, peakpos, -1);
+ gp2 = findGround(data, peakpos, 1);
+
+ groundLevel = max(data[gp1], data[gp2]);
+ peakLevel = data[peakpos];
+
+ if (groundLevel < 1e-6) return 0; // ground level too small => detection failed
+ if ((peakLevel / groundLevel) < 1.3) return 0; // peak less than 30% of the ground level => no good peak detected
+
+ // calculate 70%-level of the peak
+ cutLevel = 0.70f * peakLevel + 0.30f * groundLevel;
+ // find mid-level crossings
+ crosspos1 = findCrossingLevel(data, cutLevel, peakpos, -1);
+ crosspos2 = findCrossingLevel(data, cutLevel, peakpos, 1);
+
+ if ((crosspos1 < 0) || (crosspos2 < 0)) return 0; // no crossing, no peak..
+
+ // calculate mass center of the peak surroundings
+ return calcMassCenter(data, crosspos1, crosspos2);
+}
+
+
+
+double PeakFinder::detectPeak(const float *data, int aminPos, int amaxPos)
+{
+
+ int i;
+ int peakpos; // position of peak level
+ double highPeak, peak;
+
+ this->minPos = aminPos;
+ this->maxPos = amaxPos;
+
+ // find absolute peak
+ peakpos = minPos;
+ peak = data[minPos];
+ for (i = minPos + 1; i < maxPos; i ++)
+ {
+ if (data[i] > peak)
+ {
+ peak = data[i];
+ peakpos = i;
+ }
+ }
+
+ // Calculate exact location of the highest peak mass center
+ highPeak = getPeakCenter(data, peakpos);
+ peak = highPeak;
+
+ // Now check if the highest peak were in fact harmonic of the true base beat peak
+ // - sometimes the highest peak can be Nth harmonic of the true base peak yet
+ // just a slightly higher than the true base
+ for (i = 2; i < 10; i ++)
+ {
+ double peaktmp, tmp;
+ int i1,i2;
+
+ peakpos = (int)(highPeak / (double)i + 0.5f);
+ if (peakpos < minPos) break;
+
+ // calculate mass-center of possible base peak
+ peaktmp = getPeakCenter(data, peakpos);
+
+ // now compare to highest detected peak
+ i1 = (int)(highPeak + 0.5);
+ i2 = (int)(peaktmp + 0.5);
+ tmp = 2 * (data[i2] - data[i1]) / (data[i2] + data[i1]);
+ if (fabs(tmp) < 0.1)
+ {
+ // The highest peak is harmonic of almost as high base peak,
+ // thus use the base peak instead
+ peak = peaktmp;
+ }
+ }
+
+ return peak;
+}
+
+
diff --git a/plugins/soundtouch/soundtouch/source/SoundTouch/PeakFinder.h b/plugins/soundtouch/soundtouch/source/SoundTouch/PeakFinder.h new file mode 100644 index 00000000..e3640cc6 --- /dev/null +++ b/plugins/soundtouch/soundtouch/source/SoundTouch/PeakFinder.h @@ -0,0 +1,93 @@ +////////////////////////////////////////////////////////////////////////////////
+///
+/// The routine detects highest value on an array of values and calculates the
+/// precise peak location as a mass-center of the 'hump' around the peak value.
+///
+/// Author : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// Last changed : $Date: 2009-02-21 18:00:14 +0200 (Sat, 21 Feb 2009) $
+// File revision : $Revision: 4 $
+//
+// $Id: PeakFinder.h 63 2009-02-21 16:00:14Z oparviai $
+//
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+// SoundTouch audio processing library
+// Copyright (c) Olli Parviainen
+//
+// This library is free software; you can redistribute it and/or
+// modify it under the terms of the GNU Lesser General Public
+// License as published by the Free Software Foundation; either
+// version 2.1 of the License, or (at your option) any later version.
+//
+// This library is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+// Lesser General Public License for more details.
+//
+// You should have received a copy of the GNU Lesser General Public
+// License along with this library; if not, write to the Free Software
+// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#ifndef _PeakFinder_H_
+#define _PeakFinder_H_
+
+namespace soundtouch
+{
+
+class PeakFinder
+{
+protected:
+ /// Min, max allowed peak positions within the data vector
+ int minPos, maxPos;
+
+ /// Calculates the mass center between given vector items.
+ double calcMassCenter(const float *data, ///< Data vector.
+ int firstPos, ///< Index of first vector item beloging to the peak.
+ int lastPos ///< Index of last vector item beloging to the peak.
+ ) const;
+
+ /// Finds the data vector index where the monotoniously decreasing signal crosses the
+ /// given level.
+ int findCrossingLevel(const float *data, ///< Data vector.
+ float level, ///< Goal crossing level.
+ int peakpos, ///< Peak position index within the data vector.
+ int direction /// Direction where to proceed from the peak: 1 = right, -1 = left.
+ ) const;
+
+ /// Finds the 'ground' level, i.e. smallest level between two neighbouring peaks, to right-
+ /// or left-hand side of the given peak position.
+ int findGround(const float *data, /// Data vector.
+ int peakpos, /// Peak position index within the data vector.
+ int direction /// Direction where to proceed from the peak: 1 = right, -1 = left.
+ ) const;
+
+ /// get exact center of peak near given position by calculating local mass of center
+ double getPeakCenter(const float *data, int peakpos) const;
+
+public:
+ /// Constructor.
+ PeakFinder();
+
+ /// Detect exact peak position of the data vector by finding the largest peak 'hump'
+ /// and calculating the mass-center location of the peak hump.
+ ///
+ /// \return The location of the largest base harmonic peak hump.
+ double detectPeak(const float *data, /// Data vector to be analyzed. The data vector has
+ /// to be at least 'maxPos' items long.
+ int minPos, ///< Min allowed peak location within the vector data.
+ int maxPos ///< Max allowed peak location within the vector data.
+ );
+};
+
+}
+
+#endif // _PeakFinder_H_
diff --git a/plugins/soundtouch/soundtouch/source/SoundTouch/RateTransposer.cpp b/plugins/soundtouch/soundtouch/source/SoundTouch/RateTransposer.cpp new file mode 100644 index 00000000..7e0b277d --- /dev/null +++ b/plugins/soundtouch/soundtouch/source/SoundTouch/RateTransposer.cpp @@ -0,0 +1,628 @@ +////////////////////////////////////////////////////////////////////////////////
+///
+/// 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 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// Last changed : $Date: 2009-10-31 16:37:24 +0200 (Sat, 31 Oct 2009) $
+// File revision : $Revision: 4 $
+//
+// $Id: RateTransposer.cpp 74 2009-10-31 14:37:24Z oparviai $
+//
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+// SoundTouch audio processing library
+// Copyright (c) Olli Parviainen
+//
+// This library is free software; you can redistribute it and/or
+// modify it under the terms of the GNU Lesser General Public
+// License as published by the Free Software Foundation; either
+// version 2.1 of the License, or (at your option) any later version.
+//
+// This library is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+// Lesser General Public License for more details.
+//
+// You should have received a copy of the GNU Lesser General Public
+// License along with this library; if not, write to the Free Software
+// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#include <memory.h>
+#include <assert.h>
+#include <stdlib.h>
+#include <stdio.h>
+#include <stdexcept>
+#include "RateTransposer.h"
+#include "AAFilter.h"
+
+using namespace std;
+using namespace soundtouch;
+
+
+/// A linear samplerate transposer class that uses integer arithmetics.
+/// for the transposing.
+class RateTransposerInteger : public RateTransposer
+{
+protected:
+ int iSlopeCount;
+ int iRate;
+ 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;
+ 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();
+};
+
+
+
+
+// Operator 'new' is overloaded so that it automatically creates a suitable instance
+// depending on if we've a MMX/SSE/etc-capable CPU available or not.
+void * RateTransposer::operator new(size_t s)
+{
+ throw runtime_error("Error in RateTransoser::new: don't use \"new TDStretch\" directly, use \"newInstance\" to create a new instance instead!");
+ return NULL;
+}
+
+
+RateTransposer *RateTransposer::newInstance()
+{
+#ifdef INTEGER_SAMPLES
+ return ::new RateTransposerInteger;
+#else
+ return ::new RateTransposerFloat;
+#endif
+}
+
+
+// Constructor
+RateTransposer::RateTransposer() : FIFOProcessor(&outputBuffer)
+{
+ numChannels = 2;
+ bUseAAFilter = TRUE;
+ fRate = 0;
+
+ // 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(BOOL newMode)
+{
+ bUseAAFilter = newMode;
+}
+
+
+/// Returns nonzero if anti-alias filter is enabled.
+BOOL RateTransposer::isAAFilterEnabled() const
+{
+ return bUseAAFilter;
+}
+
+
+AAFilter *RateTransposer::getAAFilter()
+{
+ return pAAFilter;
+}
+
+
+
+// Sets new target iRate. Normal iRate = 1.0, smaller values represent slower
+// iRate, larger faster iRates.
+void RateTransposer::setRate(float newRate)
+{
+ double 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 'nSamples' pcs of samples from the 'samples' memory position into
+// the input of the object.
+void RateTransposer::putSamples(const SAMPLETYPE *samples, uint nSamples)
+{
+ processSamples(samples, nSamples);
+}
+
+
+
+// Transposes up the sample rate, causing the observed playback 'rate' of the
+// sound to decrease
+void RateTransposer::upsample(const SAMPLETYPE *src, uint nSamples)
+{
+ uint 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 = (uint)((float)nSamples / fRate + 16.0f);
+
+ // Transpose the samples, store the result into the end of "storeBuffer"
+ count = transpose(storeBuffer.ptrEnd(sizeTemp), src, nSamples);
+ 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, (uint)numChannels);
+ 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 nSamples)
+{
+ uint 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, nSamples);
+
+ // 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, (uint)numChannels);
+
+ if (count == 0) return;
+
+ // Remove the filtered samples from 'storeBuffer'
+ storeBuffer.receiveSamples(count);
+
+ // Transpose the samples (+16 is to reserve some slack in the destination buffer)
+ sizeTemp = (uint)((float)nSamples / 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 nSamples)
+{
+ uint count;
+ uint sizeReq;
+
+ if (nSamples == 0) return;
+ assert(pAAFilter);
+
+ // If anti-alias filter is turned off, simply transpose without applying
+ // the filter
+ if (bUseAAFilter == FALSE)
+ {
+ sizeReq = (uint)((float)nSamples / fRate + 1.0f);
+ count = transpose(outputBuffer.ptrEnd(sizeReq), src, nSamples);
+ outputBuffer.putSamples(count);
+ return;
+ }
+
+ // Transpose with anti-alias filter
+ if (fRate < 1.0f)
+ {
+ upsample(src, nSamples);
+ }
+ else
+ {
+ downsample(src, nSamples);
+ }
+}
+
+
+// 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 nSamples)
+{
+ if (numChannels == 2)
+ {
+ return transposeStereo(dest, src, nSamples);
+ }
+ else
+ {
+ return transposeMono(dest, src, nSamples);
+ }
+}
+
+
+// Sets the number of channels, 1 = mono, 2 = stereo
+void RateTransposer::setChannels(int nChannels)
+{
+ assert(nChannels > 0);
+ if (numChannels == nChannels) return;
+
+ assert(nChannels == 1 || nChannels == 2);
+ numChannels = nChannels;
+
+ storeBuffer.setChannels(numChannels);
+ tempBuffer.setChannels(numChannels);
+ outputBuffer.setChannels(numChannels);
+
+ // 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.
+int RateTransposer::isEmpty() const
+{
+ 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()
+{
+ // Notice: use local function calling syntax for sake of clarity,
+ // to indicate the fact that C++ constructor can't call virtual functions.
+ RateTransposerInteger::resetRegisters();
+ RateTransposerInteger::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 nSamples)
+{
+ unsigned int i, used;
+ LONG_SAMPLETYPE temp, vol1;
+
+ if (nSamples == 0) return 0; // no samples, no work
+
+ 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 += iRate;
+ }
+ // now always (iSlopeCount > SCALE)
+ iSlopeCount -= SCALE;
+
+ while (1)
+ {
+ while (iSlopeCount > SCALE)
+ {
+ iSlopeCount -= SCALE;
+ used ++;
+ if (used >= nSamples - 1) goto end;
+ }
+ vol1 = (LONG_SAMPLETYPE)(SCALE - iSlopeCount);
+ temp = src[used] * vol1 + iSlopeCount * src[used + 1];
+ dest[i] = (SAMPLETYPE)(temp / SCALE);
+
+ i++;
+ iSlopeCount += iRate;
+ }
+end:
+ // Store the last sample for the next round
+ sPrevSampleL = src[nSamples - 1];
+
+ return i;
+}
+
+
+// Transposes the sample rate of the given samples using linear interpolation.
+// 'Stereo' version of the routine. Returns the number of samples returned in
+// the "dest" buffer
+uint RateTransposerInteger::transposeStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, uint nSamples)
+{
+ unsigned int srcPos, i, used;
+ LONG_SAMPLETYPE temp, vol1;
+
+ if (nSamples == 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 += iRate;
+ }
+ // now always (iSlopeCount > SCALE)
+ iSlopeCount -= SCALE;
+
+ while (1)
+ {
+ while (iSlopeCount > SCALE)
+ {
+ iSlopeCount -= SCALE;
+ used ++;
+ if (used >= nSamples - 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 += iRate;
+ }
+end:
+ // Store the last sample for the next round
+ sPrevSampleL = src[2 * nSamples - 2];
+ sPrevSampleR = src[2 * nSamples - 1];
+
+ return i;
+}
+
+
+// Sets new target iRate. Normal iRate = 1.0, smaller values represent slower
+// iRate, larger faster iRates.
+void RateTransposerInteger::setRate(float newRate)
+{
+ iRate = (int)(newRate * SCALE + 0.5f);
+ RateTransposer::setRate(newRate);
+}
+
+
+//////////////////////////////////////////////////////////////////////////////
+//
+// RateTransposerFloat - floating point arithmetic implementation
+//
+//////////////////////////////////////////////////////////////////////////////
+
+// Constructor
+RateTransposerFloat::RateTransposerFloat() : RateTransposer()
+{
+ // Notice: use local function calling syntax for sake of clarity,
+ // to indicate the fact that C++ constructor can't call virtual functions.
+ RateTransposerFloat::resetRegisters();
+ RateTransposerFloat::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 nSamples)
+{
+ 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;
+
+ if (nSamples > 1)
+ {
+ while (1)
+ {
+ while (fSlopeCount > 1.0f)
+ {
+ fSlopeCount -= 1.0f;
+ used ++;
+ if (used >= nSamples - 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[nSamples - 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 nSamples)
+{
+ unsigned int srcPos, i, used;
+
+ if (nSamples == 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;
+
+ if (nSamples > 1)
+ {
+ while (1)
+ {
+ while (fSlopeCount > 1.0f)
+ {
+ fSlopeCount -= 1.0f;
+ used ++;
+ if (used >= nSamples - 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 * nSamples - 2];
+ sPrevSampleR = src[2 * nSamples - 1];
+
+ return i;
+}
diff --git a/plugins/soundtouch/soundtouch/source/SoundTouch/RateTransposer.h b/plugins/soundtouch/soundtouch/source/SoundTouch/RateTransposer.h new file mode 100644 index 00000000..f035af2c --- /dev/null +++ b/plugins/soundtouch/soundtouch/source/SoundTouch/RateTransposer.h @@ -0,0 +1,159 @@ +////////////////////////////////////////////////////////////////////////////////
+///
+/// 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 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// Last changed : $Date: 2009-02-21 18:00:14 +0200 (Sat, 21 Feb 2009) $
+// File revision : $Revision: 4 $
+//
+// $Id: RateTransposer.h 63 2009-02-21 16:00:14Z oparviai $
+//
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+// SoundTouch audio processing library
+// Copyright (c) Olli Parviainen
+//
+// This library is free software; you can redistribute it and/or
+// modify it under the terms of the GNU Lesser General Public
+// License as published by the Free Software Foundation; either
+// version 2.1 of the License, or (at your option) any later version.
+//
+// This library is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+// Lesser General Public License for more details.
+//
+// You should have received a copy of the GNU Lesser General Public
+// License along with this library; if not, write to the Free Software
+// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#ifndef RateTransposer_H
+#define RateTransposer_H
+
+#include <stddef.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;
+
+ int numChannels;
+
+ /// 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;
+
+ 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;
+ inline uint transpose(SAMPLETYPE *dest,
+ const SAMPLETYPE *src,
+ uint numSamples);
+
+ 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);
+
+
+public:
+ RateTransposer();
+ virtual ~RateTransposer();
+
+ /// Operator 'new' is overloaded so that it automatically creates a suitable instance
+ /// depending on if we're to use integer or floating point arithmetics.
+ static void *operator new(size_t s);
+
+ /// 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();
+
+ /// 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(int 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.
+ int isEmpty() const;
+};
+
+}
+
+#endif
diff --git a/plugins/soundtouch/soundtouch/source/SoundTouch/SoundTouch.cpp b/plugins/soundtouch/soundtouch/source/SoundTouch/SoundTouch.cpp new file mode 100644 index 00000000..aa7ac028 --- /dev/null +++ b/plugins/soundtouch/soundtouch/source/SoundTouch/SoundTouch.cpp @@ -0,0 +1,480 @@ +//////////////////////////////////////////////////////////////////////////////
+///
+/// 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 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// Last changed : $Date: 2009-05-19 07:57:30 +0300 (Tue, 19 May 2009) $
+// File revision : $Revision: 4 $
+//
+// $Id: SoundTouch.cpp 73 2009-05-19 04:57:30Z oparviai $
+//
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+// SoundTouch audio processing library
+// Copyright (c) Olli Parviainen
+//
+// This library is free software; you can redistribute it and/or
+// modify it under the terms of the GNU Lesser General Public
+// License as published by the Free Software Foundation; either
+// version 2.1 of the License, or (at your option) any later version.
+//
+// This library is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+// Lesser General Public License for more details.
+//
+// You should have received a copy of the GNU Lesser General Public
+// License along with this library; if not, write to the Free Software
+// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#include <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;
+
+/// test if two floating point numbers are equal
+#define TEST_FLOAT_EQUAL(a, b) (fabs(a - b) < 1e-10)
+
+
+/// Print library version string for autoconf
+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((int)numChannels);
+ pTDStretch->setChannels((int)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 (!TEST_FLOAT_EQUAL(rate,oldRate)) pRateTransposer->setRate(rate);
+ if (!TEST_FLOAT_EQUAL(tempo, oldTempo)) pTDStretch->setTempo(tempo);
+
+#ifndef PREVENT_CLICK_AT_RATE_CROSSOVER
+ if (rate <= 1.0f)
+ {
+ 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;
+ }
+ }
+ else
+#endif
+ {
+ 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;
+ }
+ }
+}
+
+
+// Sets sample rate.
+void SoundTouch::setSampleRate(uint srate)
+{
+ bSrateSet = TRUE;
+ // set sample rate, leave other tempo changer parameters as they are.
+ pTDStretch->setParameters((int)srate);
+}
+
+
+// Adds 'numSamples' pcs of samples from the 'samples' memory position into
+// the input of the object.
+void SoundTouch::putSamples(const SAMPLETYPE *samples, uint nSamples)
+{
+ 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
+ /* Bypass the nominal setting - can introduce a click in sound when tempo/pitch control crosses the nominal value...
+ 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, nSamples);
+ }
+ */
+#ifndef PREVENT_CLICK_AT_RATE_CROSSOVER
+ else if (rate <= 1.0f)
+ {
+ // transpose the rate down, output the transposed sound to tempo changer buffer
+ assert(output == pTDStretch);
+ pRateTransposer->putSamples(samples, nSamples);
+ pTDStretch->moveSamples(*pRateTransposer);
+ }
+ else
+#endif
+ {
+ // evaluate the tempo changer, then transpose the rate up,
+ assert(output == pRateTransposer);
+ pTDStretch->putSamples(samples, nSamples);
+ 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(int settingId, int value)
+{
+ int 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.
+int SoundTouch::getSetting(int settingId) const
+{
+ int temp;
+
+ switch (settingId)
+ {
+ case SETTING_USE_AA_FILTER :
+ return (uint)pRateTransposer->isAAFilterEnabled();
+
+ case SETTING_AA_FILTER_LENGTH :
+ return pRateTransposer->getAAFilter()->getLength();
+
+ case SETTING_USE_QUICKSEEK :
+ return (uint) 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;
+}
diff --git a/plugins/soundtouch/soundtouch/source/SoundTouch/TDStretch.cpp b/plugins/soundtouch/soundtouch/source/SoundTouch/TDStretch.cpp new file mode 100644 index 00000000..232133b5 --- /dev/null +++ b/plugins/soundtouch/soundtouch/source/SoundTouch/TDStretch.cpp @@ -0,0 +1,1045 @@ +////////////////////////////////////////////////////////////////////////////////
+///
+/// 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 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// Last changed : $Date: 2009-12-28 21:27:04 +0200 (Mon, 28 Dec 2009) $
+// File revision : $Revision: 1.12 $
+//
+// $Id: TDStretch.cpp 77 2009-12-28 19:27:04Z oparviai $
+//
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+// SoundTouch audio processing library
+// Copyright (c) Olli Parviainen
+//
+// This library is free software; you can redistribute it and/or
+// modify it under the terms of the GNU Lesser General Public
+// License as published by the Free Software Foundation; either
+// version 2.1 of the License, or (at your option) any later version.
+//
+// This library is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+// Lesser General Public License for more details.
+//
+// You should have received a copy of the GNU Lesser General Public
+// License along with this library; if not, write to the Free Software
+// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#include <string.h>
+#include <limits.h>
+#include <assert.h>
+#include <math.h>
+#include <float.h>
+#include <stdexcept>
+
+#include "STTypes.h"
+#include "cpu_detect.h"
+#include "TDStretch.h"
+
+#include <stdio.h>
+
+using namespace soundtouch;
+
+#define max(x, y) (((x) > (y)) ? (x) : (y))
+
+
+/*****************************************************************************
+ *
+ * Constant definitions
+ *
+ *****************************************************************************/
+
+// Table for the hierarchical mixing position seeking algorithm
+static const short _scanOffsets[5][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},
+ { 121, 114, 97, 114, 98, 105, 108, 32, 104, 99, 117, 111,
+ 116, 100, 110, 117, 111, 115, 0, 0, 0, 0, 0, 0}};
+
+/*****************************************************************************
+ *
+ * Implementation of the class 'TDStretch'
+ *
+ *****************************************************************************/
+
+
+TDStretch::TDStretch() : FIFOProcessor(&outputBuffer)
+{
+ bQuickSeek = FALSE;
+ channels = 2;
+
+ pMidBuffer = NULL;
+ pRefMidBufferUnaligned = NULL;
+ overlapLength = 0;
+
+ bAutoSeqSetting = TRUE;
+ bAutoSeekSetting = TRUE;
+
+// outDebt = 0;
+ skipFract = 0;
+
+ tempo = 1.0f;
+ setParameters(44100, DEFAULT_SEQUENCE_MS, DEFAULT_SEEKWINDOW_MS, DEFAULT_OVERLAP_MS);
+ setTempo(1.0f);
+
+ clear();
+}
+
+
+
+TDStretch::~TDStretch()
+{
+ delete[] pMidBuffer;
+ delete[] pRefMidBufferUnaligned;
+}
+
+
+
+// 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(int aSampleRate, int aSequenceMS,
+ int aSeekWindowMS, int aOverlapMS)
+{
+ // accept only positive parameter values - if zero or negative, use old values instead
+ if (aSampleRate > 0) this->sampleRate = aSampleRate;
+ if (aOverlapMS > 0) this->overlapMs = aOverlapMS;
+
+ if (aSequenceMS > 0)
+ {
+ this->sequenceMs = aSequenceMS;
+ bAutoSeqSetting = FALSE;
+ }
+ else if (aSequenceMS == 0)
+ {
+ // if zero, use automatic setting
+ bAutoSeqSetting = TRUE;
+ }
+
+ if (aSeekWindowMS > 0)
+ {
+ this->seekWindowMs = aSeekWindowMS;
+ bAutoSeekSetting = FALSE;
+ }
+ else if (aSeekWindowMS == 0)
+ {
+ // if zero, use automatic setting
+ bAutoSeekSetting = TRUE;
+ }
+
+ calcSeqParameters();
+
+ 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(int *pSampleRate, int *pSequenceMs, int *pSeekWindowMs, int *pOverlapMs) const
+{
+ if (pSampleRate)
+ {
+ *pSampleRate = sampleRate;
+ }
+
+ if (pSequenceMs)
+ {
+ *pSequenceMs = (bAutoSeqSetting) ? (USE_AUTO_SEQUENCE_LEN) : sequenceMs;
+ }
+
+ if (pSeekWindowMs)
+ {
+ *pSeekWindowMs = (bAutoSeekSetting) ? (USE_AUTO_SEEKWINDOW_LEN) : seekWindowMs;
+ }
+
+ if (pOverlapMs)
+ {
+ *pOverlapMs = overlapMs;
+ }
+}
+
+
+// Overlaps samples in 'midBuffer' with the samples in 'pInput'
+void TDStretch::overlapMono(SAMPLETYPE *pOutput, const SAMPLETYPE *pInput) const
+{
+ int i, itemp;
+
+ for (i = 0; i < overlapLength ; i ++)
+ {
+ itemp = overlapLength - i;
+ pOutput[i] = (pInput[i] * i + pMidBuffer[i] * itemp ) / overlapLength; // >> overlapDividerBits;
+ }
+}
+
+
+
+void TDStretch::clearMidBuffer()
+{
+ memset(pMidBuffer, 0, 2 * sizeof(SAMPLETYPE) * overlapLength);
+}
+
+
+void TDStretch::clearInput()
+{
+ inputBuffer.clear();
+ clearMidBuffer();
+}
+
+
+// Clears the sample buffers
+void TDStretch::clear()
+{
+ outputBuffer.clear();
+ clearInput();
+}
+
+
+
+// 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.
+int 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 'pInputBuffer' at position
+// of 'ovlPos'.
+inline void TDStretch::overlap(SAMPLETYPE *pOutput, const SAMPLETYPE *pInput, uint ovlPos) const
+{
+ if (channels == 2)
+ {
+ // stereo sound
+ overlapStereo(pOutput, pInput + 2 * ovlPos);
+ } else {
+ // mono sound.
+ overlapMono(pOutput, pInput + 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
+int TDStretch::seekBestOverlapPositionStereo(const SAMPLETYPE *refPos)
+{
+ int bestOffs;
+ double bestCorr, corr;
+ int i;
+
+ // Slopes the amplitudes of the 'midBuffer' samples
+ precalcCorrReferenceStereo();
+
+ bestCorr = FLT_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 = (double)calcCrossCorrStereo(refPos + 2 * i, pRefMidBuffer);
+ // heuristic rule to slightly favour values close to mid of the range
+ double tmp = (double)(2 * i - seekLength) / (double)seekLength;
+ corr = ((corr + 0.1) * (1.0 - 0.25 * tmp * tmp));
+
+ // 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
+int TDStretch::seekBestOverlapPositionStereoQuick(const SAMPLETYPE *refPos)
+{
+ int j;
+ int bestOffs;
+ double bestCorr, corr;
+ int scanCount, corrOffset, tempOffset;
+
+ // Slopes the amplitude of the 'midBuffer' samples
+ precalcCorrReferenceStereo();
+
+ bestCorr = FLT_MIN;
+ bestOffs = _scanOffsets[0][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 = (double)calcCrossCorrStereo(refPos + 2 * tempOffset, pRefMidBuffer);
+ // heuristic rule to slightly favour values close to mid of the range
+ double tmp = (double)(2 * tempOffset - seekLength) / seekLength;
+ corr = ((corr + 0.1) * (1.0 - 0.25 * tmp * tmp));
+
+ // 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
+int TDStretch::seekBestOverlapPositionMono(const SAMPLETYPE *refPos)
+{
+ int bestOffs;
+ double bestCorr, corr;
+ int tempOffset;
+ const SAMPLETYPE *compare;
+
+ // Slopes the amplitude of the 'midBuffer' samples
+ precalcCorrReferenceMono();
+
+ bestCorr = FLT_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 = (double)calcCrossCorrMono(pRefMidBuffer, compare);
+ // heuristic rule to slightly favour values close to mid of the range
+ double tmp = (double)(2 * tempOffset - seekLength) / seekLength;
+ corr = ((corr + 0.1) * (1.0 - 0.25 * tmp * tmp));
+
+ // 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
+int TDStretch::seekBestOverlapPositionMonoQuick(const SAMPLETYPE *refPos)
+{
+ int j;
+ int bestOffs;
+ double bestCorr, corr;
+ int scanCount, corrOffset, tempOffset;
+
+ // Slopes the amplitude of the 'midBuffer' samples
+ precalcCorrReferenceMono();
+
+ bestCorr = FLT_MIN;
+ bestOffs = _scanOffsets[0][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 = (double)calcCrossCorrMono(refPos + tempOffset, pRefMidBuffer);
+ // heuristic rule to slightly favour values close to mid of the range
+ double tmp = (double)(2 * tempOffset - seekLength) / seekLength;
+ corr = ((corr + 0.1) * (1.0 - 0.25 * tmp * tmp));
+
+ // 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.
+}
+
+
+/// Calculates processing sequence length according to tempo setting
+void TDStretch::calcSeqParameters()
+{
+ // Adjust tempo param according to tempo, so that variating processing sequence length is used
+ // at varius tempo settings, between the given low...top limits
+ #define AUTOSEQ_TEMPO_LOW 0.5 // auto setting low tempo range (-50%)
+ #define AUTOSEQ_TEMPO_TOP 2.0 // auto setting top tempo range (+100%)
+
+ // sequence-ms setting values at above low & top tempo
+ #define AUTOSEQ_AT_MIN 125.0
+ #define AUTOSEQ_AT_MAX 50.0
+ #define AUTOSEQ_K ((AUTOSEQ_AT_MAX - AUTOSEQ_AT_MIN) / (AUTOSEQ_TEMPO_TOP - AUTOSEQ_TEMPO_LOW))
+ #define AUTOSEQ_C (AUTOSEQ_AT_MIN - (AUTOSEQ_K) * (AUTOSEQ_TEMPO_LOW))
+
+ // seek-window-ms setting values at above low & top tempo
+ #define AUTOSEEK_AT_MIN 25.0
+ #define AUTOSEEK_AT_MAX 15.0
+ #define AUTOSEEK_K ((AUTOSEEK_AT_MAX - AUTOSEEK_AT_MIN) / (AUTOSEQ_TEMPO_TOP - AUTOSEQ_TEMPO_LOW))
+ #define AUTOSEEK_C (AUTOSEEK_AT_MIN - (AUTOSEEK_K) * (AUTOSEQ_TEMPO_LOW))
+
+ #define CHECK_LIMITS(x, mi, ma) (((x) < (mi)) ? (mi) : (((x) > (ma)) ? (ma) : (x)))
+
+ double seq, seek;
+
+ if (bAutoSeqSetting)
+ {
+ seq = AUTOSEQ_C + AUTOSEQ_K * tempo;
+ seq = CHECK_LIMITS(seq, AUTOSEQ_AT_MAX, AUTOSEQ_AT_MIN);
+ sequenceMs = (int)(seq + 0.5);
+ }
+
+ if (bAutoSeekSetting)
+ {
+ seek = AUTOSEEK_C + AUTOSEEK_K * tempo;
+ seek = CHECK_LIMITS(seek, AUTOSEEK_AT_MAX, AUTOSEEK_AT_MIN);
+ seekWindowMs = (int)(seek + 0.5);
+ }
+
+ // Update seek window lengths
+ seekWindowLength = (sampleRate * sequenceMs) / 1000;
+ if (seekWindowLength < 2 * overlapLength)
+ {
+ seekWindowLength = 2 * overlapLength;
+ }
+ seekLength = (sampleRate * seekWindowMs) / 1000;
+}
+
+
+
+// Sets new target tempo. Normal tempo = 'SCALE', smaller values represent slower
+// tempo, larger faster tempo.
+void TDStretch::setTempo(float newTempo)
+{
+ int intskip;
+
+ tempo = newTempo;
+
+ // Calculate new sequence duration
+ calcSeqParameters();
+
+ // Calculate ideal skip length (according to tempo value)
+ nominalSkip = tempo * (seekWindowLength - overlapLength);
+ 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) + seekLength / 2;
+ sampleReq = max(intskip + overlapLength, seekWindowLength) + seekLength;
+}
+
+
+
+// Sets the number of channels, 1 = mono, 2 = stereo
+void TDStretch::setChannels(int numChannels)
+{
+ assert(numChannels > 0);
+ 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);
+}
+*/
+
+#include <stdio.h>
+
+// Processes as many processing frames of the samples 'inputBuffer', store
+// the result into 'outputBuffer'
+void TDStretch::processSamples()
+{
+ int ovlSkip, offset;
+ int temp;
+
+ /* Removed this small optimization - can introduce a click to sound when tempo setting
+ crosses the nominal value
+ if (tempo == 1.0f)
+ {
+ // tempo not changed from the original, so bypass the processing
+ processNominalTempo();
+ return;
+ }
+ */
+
+ // Process samples as long as there are enough samples in 'inputBuffer'
+ // to form a processing frame.
+// while ((int)inputBuffer.numSamples() >= sampleReq - (outDebt / 4))
+ while ((int)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((uint)overlapLength), inputBuffer.ptrBegin(), (uint)offset);
+ outputBuffer.putSamples((uint)overlapLength);
+
+ // ... then copy sequence samples from 'inputBuffer' to output:
+ temp = (seekLength / 2 - offset);
+
+ // compensate cumulated output length diff vs. ideal output
+// temp -= outDebt / 4;
+
+ // update ideal vs. true output difference
+// outDebt += temp;
+
+ // length of sequence
+// temp += (seekWindowLength - 2 * overlapLength);
+ temp = (seekWindowLength - 2 * overlapLength);
+
+ // crosscheck that we don't have buffer overflow...
+ if ((int)inputBuffer.numSamples() < (offset + temp + overlapLength * 2))
+ {
+ continue; // just in case, shouldn't really happen
+ }
+
+ outputBuffer.putSamples(inputBuffer.ptrBegin() + channels * (offset + overlapLength), (uint)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 + temp + overlapLength * 2) <= (int)inputBuffer.numSamples());
+ memcpy(pMidBuffer, inputBuffer.ptrBegin() + channels * (offset + temp + overlapLength),
+ channels * sizeof(SAMPLETYPE) * overlapLength);
+
+ // 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((uint)ovlSkip);
+ }
+}
+
+
+// Adds 'numsamples' pcs of samples from the 'samples' memory position into
+// the input of the object.
+void TDStretch::putSamples(const SAMPLETYPE *samples, uint nSamples)
+{
+ // Add the samples into the input buffer
+ inputBuffer.putSamples(samples, nSamples);
+ // Process the samples in input buffer
+ processSamples();
+}
+
+
+
+/// Set new overlap length parameter & reallocate RefMidBuffer if necessary.
+void TDStretch::acceptNewOverlapLength(int newOverlapLength)
+{
+ int prevOvl;
+
+ assert(newOverlapLength >= 0);
+ prevOvl = overlapLength;
+ overlapLength = newOverlapLength;
+
+ if (overlapLength > prevOvl)
+ {
+ delete[] pMidBuffer;
+ delete[] pRefMidBufferUnaligned;
+
+ pMidBuffer = new SAMPLETYPE[overlapLength * 2];
+ 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) & (ulong)-16);
+ }
+}
+
+
+// Operator 'new' is overloaded so that it automatically creates a suitable instance
+// depending on if we've a MMX/SSE/etc-capable CPU available or not.
+void * TDStretch::operator new(size_t s)
+{
+ // Notice! don't use "new TDStretch" directly, use "newInstance" to create a new instance instead!
+ throw std::runtime_error("Error in TDStretch::new: Don't use 'new TDStretch' directly, use 'newInstance' member instead!");
+ return NULL;
+}
+
+
+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 *poutput, const short *input) const
+{
+ int i;
+ short temp;
+ int cnt2;
+
+ for (i = 0; i < overlapLength ; i ++)
+ {
+ temp = (short)(overlapLength - i);
+ cnt2 = 2 * i;
+ poutput[cnt2] = (input[cnt2] * i + pMidBuffer[cnt2] * temp ) / overlapLength;
+ poutput[cnt2 + 1] = (input[cnt2 + 1] * i + pMidBuffer[cnt2 + 1] * temp ) / overlapLength;
+ }
+}
+
+// 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);
+}
+
+
+/// Calculates overlap period length in samples.
+/// Integer version rounds overlap length to closest power of 2
+/// for a divide scaling operation.
+void TDStretch::calculateOverlapLength(int aoverlapMs)
+{
+ int newOvl;
+
+ assert(aoverlapMs >= 0);
+
+ // calculate overlap length so that it's power of 2 - thus it's easy to do
+ // integer division by right-shifting. Term "-1" at end is to account for
+ // the extra most significatnt bit left unused in result by signed multiplication
+ overlapDividerBits = _getClosest2Power((sampleRate * aoverlapMs) / 1000.0) - 1;
+ if (overlapDividerBits > 9) overlapDividerBits = 9;
+ if (overlapDividerBits < 3) overlapDividerBits = 3;
+ newOvl = (int)pow(2.0, (int)overlapDividerBits + 1); // +1 => account for -1 above
+
+ 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;
+ long norm;
+ int i;
+
+ corr = norm = 0;
+ for (i = 1; i < overlapLength; i ++)
+ {
+ corr += (mixingPos[i] * compare[i]) >> overlapDividerBits;
+ norm += (mixingPos[i] * mixingPos[i]) >> overlapDividerBits;
+ }
+
+ // Normalize result by dividing by sqrt(norm) - this step is easiest
+ // done using floating point operation
+ if (norm == 0) norm = 1; // to avoid div by zero
+ return (long)((double)corr * SHRT_MAX / sqrt((double)norm));
+}
+
+
+long TDStretch::calcCrossCorrStereo(const short *mixingPos, const short *compare) const
+{
+ long corr;
+ long norm;
+ int i;
+
+ corr = norm = 0;
+ for (i = 2; i < 2 * overlapLength; i += 2)
+ {
+ corr += (mixingPos[i] * compare[i] +
+ mixingPos[i + 1] * compare[i + 1]) >> overlapDividerBits;
+ norm += (mixingPos[i] * mixingPos[i] + mixingPos[i + 1] * mixingPos[i + 1]) >> overlapDividerBits;
+ }
+
+ // Normalize result by dividing by sqrt(norm) - this step is easiest
+ // done using floating point operation
+ if (norm == 0) norm = 1; // to avoid div by zero
+ return (long)((double)corr * SHRT_MAX / sqrt((double)norm));
+}
+
+#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);
+ }
+}
+
+
+// Overlaps samples in 'midBuffer' with the samples in 'pInput'
+void TDStretch::overlapStereo(float *pOutput, const float *pInput) const
+{
+ int i;
+ int 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;
+ pOutput[cnt2 + 0] = pInput[cnt2 + 0] * fi + pMidBuffer[cnt2 + 0] * fTemp;
+ pOutput[cnt2 + 1] = pInput[cnt2 + 1] * fi + pMidBuffer[cnt2 + 1] * fTemp;
+ }
+}
+
+
+/// Calculates overlapInMsec period length in samples.
+void TDStretch::calculateOverlapLength(int overlapInMsec)
+{
+ int newOvl;
+
+ assert(overlapInMsec >= 0);
+ newOvl = (sampleRate * overlapInMsec) / 1000;
+ if (newOvl < 16) newOvl = 16;
+
+ // must be divisible by 8
+ newOvl -= newOvl % 8;
+
+ acceptNewOverlapLength(newOvl);
+}
+
+
+
+double TDStretch::calcCrossCorrMono(const float *mixingPos, const float *compare) const
+{
+ double corr;
+ double norm;
+ int i;
+
+ corr = norm = 0;
+ for (i = 1; i < overlapLength; i ++)
+ {
+ corr += mixingPos[i] * compare[i];
+ norm += mixingPos[i] * mixingPos[i];
+ }
+
+ if (norm < 1e-9) norm = 1.0; // to avoid div by zero
+ return corr / sqrt(norm);
+}
+
+
+double TDStretch::calcCrossCorrStereo(const float *mixingPos, const float *compare) const
+{
+ double corr;
+ double norm;
+ int i;
+
+ corr = norm = 0;
+ for (i = 2; i < 2 * overlapLength; i += 2)
+ {
+ corr += mixingPos[i] * compare[i] +
+ mixingPos[i + 1] * compare[i + 1];
+ norm += mixingPos[i] * mixingPos[i] +
+ mixingPos[i + 1] * mixingPos[i + 1];
+ }
+
+ if (norm < 1e-9) norm = 1.0; // to avoid div by zero
+ return corr / sqrt(norm);
+}
+
+#endif // FLOAT_SAMPLES
diff --git a/plugins/soundtouch/soundtouch/source/SoundTouch/TDStretch.h b/plugins/soundtouch/soundtouch/source/SoundTouch/TDStretch.h new file mode 100644 index 00000000..00d1f3e3 --- /dev/null +++ b/plugins/soundtouch/soundtouch/source/SoundTouch/TDStretch.h @@ -0,0 +1,275 @@ +////////////////////////////////////////////////////////////////////////////////
+///
+/// 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/SSE optimized functions reside in separate, platform-specific files
+/// 'mmx_optimized.cpp' and 'sse_optimized.cpp'
+///
+/// Author : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// Last changed : $Date: 2009-05-17 14:35:13 +0300 (Sun, 17 May 2009) $
+// File revision : $Revision: 4 $
+//
+// $Id: TDStretch.h 71 2009-05-17 11:35:13Z oparviai $
+//
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+// SoundTouch audio processing library
+// Copyright (c) Olli Parviainen
+//
+// This library is free software; you can redistribute it and/or
+// modify it under the terms of the GNU Lesser General Public
+// License as published by the Free Software Foundation; either
+// version 2.1 of the License, or (at your option) any later version.
+//
+// This library is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+// Lesser General Public License for more details.
+//
+// You should have received a copy of the GNU Lesser General Public
+// License along with this library; if not, write to the Free Software
+// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#ifndef TDStretch_H
+#define TDStretch_H
+
+#include <stddef.h>
+#include "STTypes.h"
+#include "RateTransposer.h"
+#include "FIFOSamplePipe.h"
+
+namespace soundtouch
+{
+
+/// Default values for sound processing parameters:
+/// Notice that the default parameters are tuned for contemporary popular music
+/// processing. For speech processing applications these parameters suit better:
+/// #define DEFAULT_SEQUENCE_MS 40
+/// #define DEFAULT_SEEKWINDOW_MS 15
+/// #define DEFAULT_OVERLAP_MS 8
+///
+
+/// 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 40
+#define DEFAULT_SEQUENCE_MS USE_AUTO_SEQUENCE_LEN
+
+/// Giving this value for the sequence length sets automatic parameter value
+/// according to tempo setting (recommended)
+#define USE_AUTO_SEQUENCE_LEN 0
+
+/// 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 15
+#define DEFAULT_SEEKWINDOW_MS USE_AUTO_SEEKWINDOW_LEN
+
+/// Giving this value for the seek window length sets automatic parameter value
+/// according to tempo setting (recommended)
+#define USE_AUTO_SEEKWINDOW_LEN 0
+
+/// 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 8
+
+
+/// Class that does the time-stretch (tempo change) effect for the processed
+/// sound.
+class TDStretch : public FIFOProcessor
+{
+protected:
+ int channels;
+ int sampleReq;
+ float tempo;
+
+ SAMPLETYPE *pMidBuffer;
+ SAMPLETYPE *pRefMidBuffer;
+ SAMPLETYPE *pRefMidBufferUnaligned;
+ int overlapLength;
+ int seekLength;
+ int seekWindowLength;
+ int overlapDividerBits;
+ int slopingDivider;
+ float nominalSkip;
+ float skipFract;
+ FIFOSampleBuffer outputBuffer;
+ FIFOSampleBuffer inputBuffer;
+ BOOL bQuickSeek;
+// int outDebt;
+// BOOL bMidBufferDirty;
+
+ int sampleRate;
+ int sequenceMs;
+ int seekWindowMs;
+ int overlapMs;
+ BOOL bAutoSeqSetting;
+ BOOL bAutoSeekSetting;
+
+ void acceptNewOverlapLength(int newOverlapLength);
+
+ virtual void clearCrossCorrState();
+ void calculateOverlapLength(int overlapMs);
+
+ virtual LONG_SAMPLETYPE calcCrossCorrStereo(const SAMPLETYPE *mixingPos, const SAMPLETYPE *compare) const;
+ virtual LONG_SAMPLETYPE calcCrossCorrMono(const SAMPLETYPE *mixingPos, const SAMPLETYPE *compare) const;
+
+ virtual int seekBestOverlapPositionStereo(const SAMPLETYPE *refPos);
+ virtual int seekBestOverlapPositionStereoQuick(const SAMPLETYPE *refPos);
+ virtual int seekBestOverlapPositionMono(const SAMPLETYPE *refPos);
+ virtual int seekBestOverlapPositionMonoQuick(const SAMPLETYPE *refPos);
+ int 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 calcSeqParameters();
+
+ /// 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();
+
+public:
+ TDStretch();
+ virtual ~TDStretch();
+
+ /// Operator 'new' is overloaded so that it automatically creates a suitable instance
+ /// depending on if we've a MMX/SSE/etc-capable CPU available or not.
+ static void *operator new(size_t s);
+
+ /// 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(int 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(int sampleRate, ///< Samplerate of sound being processed (Hz)
+ int sequenceMS = -1, ///< Single processing sequence length (ms)
+ int seekwindowMS = -1, ///< Offset seeking window length (ms)
+ int overlapMS = -1 ///< 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(int *pSampleRate, int *pSequenceMs, int *pSeekWindowMs, int *pOverlapMs) const;
+
+ /// 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
diff --git a/plugins/soundtouch/soundtouch/source/SoundTouch/cpu_detect.h b/plugins/soundtouch/soundtouch/source/SoundTouch/cpu_detect.h new file mode 100644 index 00000000..025781da --- /dev/null +++ b/plugins/soundtouch/soundtouch/source/SoundTouch/cpu_detect.h @@ -0,0 +1,62 @@ +////////////////////////////////////////////////////////////////////////////////
+///
+/// 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 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// Last changed : $Date: 2008-02-10 18:26:55 +0200 (Sun, 10 Feb 2008) $
+// File revision : $Revision: 4 $
+//
+// $Id: cpu_detect.h 11 2008-02-10 16:26:55Z oparviai $
+//
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+// SoundTouch audio processing library
+// Copyright (c) Olli Parviainen
+//
+// This library is free software; you can redistribute it and/or
+// modify it under the terms of the GNU Lesser General Public
+// License as published by the Free Software Foundation; either
+// version 2.1 of the License, or (at your option) any later version.
+//
+// This library is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+// Lesser General Public License for more details.
+//
+// You should have received a copy of the GNU Lesser General Public
+// License along with this library; if not, write to the Free Software
+// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#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_
diff --git a/plugins/soundtouch/soundtouch/source/SoundTouch/cpu_detect_x86_gcc.cpp b/plugins/soundtouch/soundtouch/source/SoundTouch/cpu_detect_x86_gcc.cpp new file mode 100644 index 00000000..b0d0a693 --- /dev/null +++ b/plugins/soundtouch/soundtouch/source/SoundTouch/cpu_detect_x86_gcc.cpp @@ -0,0 +1,135 @@ +////////////////////////////////////////////////////////////////////////////////
+///
+/// Generic version of the x86 CPU extension detection routine.
+///
+/// This file is for GNU & other non-Windows compilers, see 'cpu_detect_x86_win.cpp'
+/// for the Microsoft compiler version.
+///
+/// Author : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// Last changed : $Date: 2009-02-25 19:13:51 +0200 (Wed, 25 Feb 2009) $
+// File revision : $Revision: 4 $
+//
+// $Id: cpu_detect_x86_gcc.cpp 67 2009-02-25 17:13:51Z oparviai $
+//
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+// SoundTouch audio processing library
+// Copyright (c) Olli Parviainen
+//
+// This library is free software; you can redistribute it and/or
+// modify it under the terms of the GNU Lesser General Public
+// License as published by the Free Software Foundation; either
+// version 2.1 of the License, or (at your option) any later version.
+//
+// This library is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+// Lesser General Public License for more details.
+//
+// You should have received a copy of the GNU Lesser General Public
+// License along with this library; if not, write to the Free Software
+// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#include <stdexcept>
+#include <string>
+#include "cpu_detect.h"
+#include "STTypes.h"
+
+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)
+{
+#if (!(ALLOW_X86_OPTIMIZATIONS) || !(__GNUC__))
+
+ 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\tmovl (%%esp), %%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\tmovl %%eax, (%%esp)" // store toggled eflags to stack
+ "\n\tpopf" // load eflags from stack
+ "\n\tpushf" // save updated eflags to stack
+ "\n\tmovl (%%esp), %%eax" // load eax from stack
+ "\n\tpopf" // pop stack to restore esp
+ "\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"
+ "\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
+}
diff --git a/plugins/soundtouch/soundtouch/source/SoundTouch/cpu_detect_x86_win.cpp b/plugins/soundtouch/soundtouch/source/SoundTouch/cpu_detect_x86_win.cpp new file mode 100644 index 00000000..c6c54246 --- /dev/null +++ b/plugins/soundtouch/soundtouch/source/SoundTouch/cpu_detect_x86_win.cpp @@ -0,0 +1,129 @@ +////////////////////////////////////////////////////////////////////////////////
+///
+/// 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 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// Last changed : $Date: 2009-02-13 18:22:48 +0200 (Fri, 13 Feb 2009) $
+// File revision : $Revision: 4 $
+//
+// $Id: cpu_detect_x86_win.cpp 62 2009-02-13 16:22:48Z oparviai $
+//
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+// SoundTouch audio processing library
+// Copyright (c) Olli Parviainen
+//
+// This library is free software; you can redistribute it and/or
+// modify it under the terms of the GNU Lesser General Public
+// License as published by the Free Software Foundation; either
+// version 2.1 of the License, or (at your option) any later version.
+//
+// This library is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+// Lesser General Public License for more details.
+//
+// You should have received a copy of the GNU Lesser General Public
+// License along with this library; if not, write to the Free Software
+// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#include "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
+ mov eax,dword ptr [esp] ; load eax from stack (with eflags)
+ mov ecx, eax ; save the original eflags values to ecx
+ xor eax, 0x00200000 ; toggle bit 21
+ mov dword ptr [esp],eax ; store toggled eflags to stack
+ popfd ; load eflags from stack
+
+ pushfd ; save updated eflags to stack
+ mov eax,dword ptr [esp] ; load eax from stack
+ popfd ; pop stack to restore stack pointer
+
+ 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;
+}
diff --git a/plugins/soundtouch/soundtouch/source/SoundTouch/mmx_optimized.cpp b/plugins/soundtouch/soundtouch/source/SoundTouch/mmx_optimized.cpp new file mode 100644 index 00000000..539ee57c --- /dev/null +++ b/plugins/soundtouch/soundtouch/source/SoundTouch/mmx_optimized.cpp @@ -0,0 +1,320 @@ +////////////////////////////////////////////////////////////////////////////////
+///
+/// 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.
+///
+/// The MMX-optimizations are programmed using MMX compiler intrinsics that
+/// are supported both by Microsoft Visual C++ and GCC compilers, so this file
+/// should compile with both toolsets.
+///
+/// NOTICE: If using Visual Studio 6.0, you'll need to install the "Visual C++
+/// 6.0 processor pack" update to support compiler intrinsic syntax. The update
+/// is available for download at Microsoft Developers Network, see here:
+/// http://msdn.microsoft.com/en-us/vstudio/aa718349.aspx
+///
+/// Author : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// Last changed : $Date: 2009-10-31 16:53:23 +0200 (Sat, 31 Oct 2009) $
+// File revision : $Revision: 4 $
+//
+// $Id: mmx_optimized.cpp 75 2009-10-31 14:53:23Z oparviai $
+//
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+// SoundTouch audio processing library
+// Copyright (c) Olli Parviainen
+//
+// This library is free software; you can redistribute it and/or
+// modify it under the terms of the GNU Lesser General Public
+// License as published by the Free Software Foundation; either
+// version 2.1 of the License, or (at your option) any later version.
+//
+// This library is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+// Lesser General Public License for more details.
+//
+// You should have received a copy of the GNU Lesser General Public
+// License along with this library; if not, write to the Free Software
+// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#include "STTypes.h"
+
+#ifdef ALLOW_MMX
+// MMX routines available only with integer sample type
+
+#if !(WIN32 || __i386__ || __x86_64__)
+#error "wrong platform - this source code file is exclusively for x86 platforms"
+#endif
+
+using namespace soundtouch;
+
+//////////////////////////////////////////////////////////////////////////////
+//
+// implementation of MMX optimized functions of class 'TDStretchMMX'
+//
+//////////////////////////////////////////////////////////////////////////////
+
+#include "TDStretch.h"
+#include <mmintrin.h>
+#include <limits.h>
+#include <math.h>
+
+
+// Calculates cross correlation of two buffers
+long TDStretchMMX::calcCrossCorrStereo(const short *pV1, const short *pV2) const
+{
+ const __m64 *pVec1, *pVec2;
+ __m64 shifter;
+ __m64 accu, normaccu;
+ long corr, norm;
+ int i;
+
+ pVec1 = (__m64*)pV1;
+ pVec2 = (__m64*)pV2;
+
+ shifter = _m_from_int(overlapDividerBits);
+ normaccu = accu = _mm_setzero_si64();
+
+ // Process 4 parallel sets of 2 * stereo samples each during each
+ // round to improve CPU-level parallellization.
+ for (i = 0; i < overlapLength / 8; i ++)
+ {
+ __m64 temp, temp2;
+
+ // dictionary of instructions:
+ // _m_pmaddwd : 4*16bit multiply-add, resulting two 32bits = [a0*b0+a1*b1 ; a2*b2+a3*b3]
+ // _mm_add_pi32 : 2*32bit add
+ // _m_psrad : 32bit right-shift
+
+ temp = _mm_add_pi32(_mm_madd_pi16(pVec1[0], pVec2[0]),
+ _mm_madd_pi16(pVec1[1], pVec2[1]));
+ temp2 = _mm_add_pi32(_mm_madd_pi16(pVec1[0], pVec1[0]),
+ _mm_madd_pi16(pVec1[1], pVec1[1]));
+ accu = _mm_add_pi32(accu, _mm_sra_pi32(temp, shifter));
+ normaccu = _mm_add_pi32(normaccu, _mm_sra_pi32(temp2, shifter));
+
+ temp = _mm_add_pi32(_mm_madd_pi16(pVec1[2], pVec2[2]),
+ _mm_madd_pi16(pVec1[3], pVec2[3]));
+ temp2 = _mm_add_pi32(_mm_madd_pi16(pVec1[2], pVec1[2]),
+ _mm_madd_pi16(pVec1[3], pVec1[3]));
+ accu = _mm_add_pi32(accu, _mm_sra_pi32(temp, shifter));
+ normaccu = _mm_add_pi32(normaccu, _mm_sra_pi32(temp2, shifter));
+
+ pVec1 += 4;
+ pVec2 += 4;
+ }
+
+ // copy hi-dword of mm0 to lo-dword of mm1, then sum mmo+mm1
+ // and finally store the result into the variable "corr"
+
+ accu = _mm_add_pi32(accu, _mm_srli_si64(accu, 32));
+ corr = _m_to_int(accu);
+
+ normaccu = _mm_add_pi32(normaccu, _mm_srli_si64(normaccu, 32));
+ norm = _m_to_int(normaccu);
+
+ // Clear MMS state
+ _m_empty();
+
+ // Normalize result by dividing by sqrt(norm) - this step is easiest
+ // done using floating point operation
+ if (norm == 0) norm = 1; // to avoid div by zero
+ return (long)((double)corr * USHRT_MAX / sqrt((double)norm));
+ // Note: Warning about the missing EMMS instruction is harmless
+ // as it'll be called elsewhere.
+}
+
+
+
+void TDStretchMMX::clearCrossCorrState()
+{
+ // Clear MMS state
+ _m_empty();
+ //_asm EMMS;
+}
+
+
+
+// MMX-optimized version of the function overlapStereo
+void TDStretchMMX::overlapStereo(short *output, const short *input) const
+{
+ const __m64 *pVinput, *pVMidBuf;
+ __m64 *pVdest;
+ __m64 mix1, mix2, adder, shifter;
+ int i;
+
+ pVinput = (const __m64*)input;
+ pVMidBuf = (const __m64*)pMidBuffer;
+ pVdest = (__m64*)output;
+
+ // mix1 = mixer values for 1st stereo sample
+ // mix1 = mixer values for 2nd stereo sample
+ // adder = adder for updating mixer values after each round
+
+ mix1 = _mm_set_pi16(0, overlapLength, 0, overlapLength);
+ adder = _mm_set_pi16(1, -1, 1, -1);
+ mix2 = _mm_add_pi16(mix1, adder);
+ adder = _mm_add_pi16(adder, adder);
+
+ // Overlaplength-division by shifter. "+1" is to account for "-1" deduced in
+ // overlapDividerBits calculation earlier.
+ shifter = _m_from_int(overlapDividerBits + 1);
+
+ for (i = 0; i < overlapLength / 4; i ++)
+ {
+ __m64 temp1, temp2;
+
+ // load & shuffle data so that input & mixbuffer data samples are paired
+ temp1 = _mm_unpacklo_pi16(pVMidBuf[0], pVinput[0]); // = i0l m0l i0r m0r
+ temp2 = _mm_unpackhi_pi16(pVMidBuf[0], pVinput[0]); // = i1l m1l i1r m1r
+
+ // temp = (temp .* mix) >> shifter
+ temp1 = _mm_sra_pi32(_mm_madd_pi16(temp1, mix1), shifter);
+ temp2 = _mm_sra_pi32(_mm_madd_pi16(temp2, mix2), shifter);
+ pVdest[0] = _mm_packs_pi32(temp1, temp2); // pack 2*2*32bit => 4*16bit
+
+ // update mix += adder
+ mix1 = _mm_add_pi16(mix1, adder);
+ mix2 = _mm_add_pi16(mix2, adder);
+
+ // --- second round begins here ---
+
+ // load & shuffle data so that input & mixbuffer data samples are paired
+ temp1 = _mm_unpacklo_pi16(pVMidBuf[1], pVinput[1]); // = i2l m2l i2r m2r
+ temp2 = _mm_unpackhi_pi16(pVMidBuf[1], pVinput[1]); // = i3l m3l i3r m3r
+
+ // temp = (temp .* mix) >> shifter
+ temp1 = _mm_sra_pi32(_mm_madd_pi16(temp1, mix1), shifter);
+ temp2 = _mm_sra_pi32(_mm_madd_pi16(temp2, mix2), shifter);
+ pVdest[1] = _mm_packs_pi32(temp1, temp2); // pack 2*2*32bit => 4*16bit
+
+ // update mix += adder
+ mix1 = _mm_add_pi16(mix1, adder);
+ mix2 = _mm_add_pi16(mix2, adder);
+
+ pVinput += 2;
+ pVMidBuf += 2;
+ pVdest += 2;
+ }
+
+ _m_empty(); // clear MMS state
+}
+
+
+//////////////////////////////////////////////////////////////////////////////
+//
+// 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 *)(((ulong)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, uint numSamples) const
+{
+ // Create stack copies of the needed member variables for asm routines :
+ uint i, j;
+ __m64 *pVdest = (__m64*)dest;
+
+ if (length < 2) return 0;
+
+ for (i = 0; i < (numSamples - length) / 2; i ++)
+ {
+ __m64 accu1;
+ __m64 accu2;
+ const __m64 *pVsrc = (const __m64*)src;
+ const __m64 *pVfilter = (const __m64*)filterCoeffsAlign;
+
+ accu1 = accu2 = _mm_setzero_si64();
+ for (j = 0; j < lengthDiv8 * 2; j ++)
+ {
+ __m64 temp1, temp2;
+
+ temp1 = _mm_unpacklo_pi16(pVsrc[0], pVsrc[1]); // = l2 l0 r2 r0
+ temp2 = _mm_unpackhi_pi16(pVsrc[0], pVsrc[1]); // = l3 l1 r3 r1
+
+ accu1 = _mm_add_pi32(accu1, _mm_madd_pi16(temp1, pVfilter[0])); // += l2*f2+l0*f0 r2*f2+r0*f0
+ accu1 = _mm_add_pi32(accu1, _mm_madd_pi16(temp2, pVfilter[1])); // += l3*f3+l1*f1 r3*f3+r1*f1
+
+ temp1 = _mm_unpacklo_pi16(pVsrc[1], pVsrc[2]); // = l4 l2 r4 r2
+
+ accu2 = _mm_add_pi32(accu2, _mm_madd_pi16(temp2, pVfilter[0])); // += l3*f2+l1*f0 r3*f2+r1*f0
+ accu2 = _mm_add_pi32(accu2, _mm_madd_pi16(temp1, pVfilter[1])); // += l4*f3+l2*f1 r4*f3+r2*f1
+
+ // accu1 += l2*f2+l0*f0 r2*f2+r0*f0
+ // += l3*f3+l1*f1 r3*f3+r1*f1
+
+ // accu2 += l3*f2+l1*f0 r3*f2+r1*f0
+ // l4*f3+l2*f1 r4*f3+r2*f1
+
+ pVfilter += 2;
+ pVsrc += 2;
+ }
+ // accu >>= resultDivFactor
+ accu1 = _mm_srai_pi32(accu1, resultDivFactor);
+ accu2 = _mm_srai_pi32(accu2, resultDivFactor);
+
+ // pack 2*2*32bits => 4*16 bits
+ pVdest[0] = _mm_packs_pi32(accu1, accu2);
+ src += 4;
+ pVdest ++;
+ }
+
+ _m_empty(); // clear emms state
+
+ return (numSamples & 0xfffffffe) - length;
+}
+
+#endif // ALLOW_MMX
diff --git a/plugins/soundtouch/soundtouch/source/SoundTouch/sse_optimized.cpp b/plugins/soundtouch/soundtouch/source/SoundTouch/sse_optimized.cpp new file mode 100644 index 00000000..7659be68 --- /dev/null +++ b/plugins/soundtouch/soundtouch/source/SoundTouch/sse_optimized.cpp @@ -0,0 +1,510 @@ +////////////////////////////////////////////////////////////////////////////////
+///
+/// SSE optimized routines for Pentium-III, Athlon-XP and later CPUs. 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.
+///
+/// The SSE-optimizations are programmed using SSE compiler intrinsics that
+/// are supported both by Microsoft Visual C++ and GCC compilers, so this file
+/// should compile with both toolsets.
+///
+/// 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/en-us/vstudio/aa718349.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: 2009-12-28 22:32:57 +0200 (Mon, 28 Dec 2009) $
+// File revision : $Revision: 4 $
+//
+// $Id: sse_optimized.cpp 80 2009-12-28 20:32:57Z oparviai $
+//
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+// SoundTouch audio processing library
+// Copyright (c) Olli Parviainen
+//
+// This library is free software; you can redistribute it and/or
+// modify it under the terms of the GNU Lesser General Public
+// License as published by the Free Software Foundation; either
+// version 2.1 of the License, or (at your option) any later version.
+//
+// This library is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+// Lesser General Public License for more details.
+//
+// You should have received a copy of the GNU Lesser General Public
+// License along with this library; if not, write to the Free Software
+// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#include "cpu_detect.h"
+#include "STTypes.h"
+
+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 <xmmintrin.h>
+#include <math.h>
+
+// Calculates cross correlation of two buffers
+double TDStretchSSE::calcCrossCorrStereo(const float *pV1, const float *pV2) const
+{
+ int i;
+ const float *pVec1;
+ const __m128 *pVec2;
+ __m128 vSum, vNorm;
+
+ // Note. It means a major slow-down if the routine needs to tolerate
+ // unaligned __m128 memory accesses. It's way faster if we can skip
+ // unaligned slots and use _mm_load_ps instruction instead of _mm_loadu_ps.
+ // This can mean up to ~ 10-fold difference (incl. part of which is
+ // due to skipping every second round for stereo sound though).
+ //
+ // Compile-time define ALLOW_NONEXACT_SIMD_OPTIMIZATION is provided
+ // for choosing if this little cheating is allowed.
+
+#ifdef ALLOW_NONEXACT_SIMD_OPTIMIZATION
+ // Little cheating allowed, return valid correlation only for
+ // aligned locations, meaning every second round for stereo sound.
+
+ #define _MM_LOAD _mm_load_ps
+
+ if (((ulong)pV1) & 15) return -1e50; // skip unaligned locations
+
+#else
+ // No cheating allowed, use unaligned load & take the resulting
+ // performance hit.
+ #define _MM_LOAD _mm_loadu_ps
+#endif
+
+ // ensure overlapLength is divisible by 8
+ assert((overlapLength % 8) == 0);
+
+ // Calculates the cross-correlation value between 'pV1' and 'pV2' vectors
+ // Note: pV2 _must_ be aligned to 16-bit boundary, pV1 need not.
+ pVec1 = (const float*)pV1;
+ pVec2 = (const __m128*)pV2;
+ vSum = vNorm = _mm_setzero_ps();
+
+ // Unroll the loop by factor of 4 * 4 operations
+ for (i = 0; i < overlapLength / 8; i ++)
+ {
+ __m128 vTemp;
+ // vSum += pV1[0..3] * pV2[0..3]
+ vTemp = _MM_LOAD(pVec1);
+ vSum = _mm_add_ps(vSum, _mm_mul_ps(vTemp ,pVec2[0]));
+ vNorm = _mm_add_ps(vNorm, _mm_mul_ps(vTemp ,vTemp));
+
+ // vSum += pV1[4..7] * pV2[4..7]
+ vTemp = _MM_LOAD(pVec1 + 4);
+ vSum = _mm_add_ps(vSum, _mm_mul_ps(vTemp, pVec2[1]));
+ vNorm = _mm_add_ps(vNorm, _mm_mul_ps(vTemp ,vTemp));
+
+ // vSum += pV1[8..11] * pV2[8..11]
+ vTemp = _MM_LOAD(pVec1 + 8);
+ vSum = _mm_add_ps(vSum, _mm_mul_ps(vTemp, pVec2[2]));
+ vNorm = _mm_add_ps(vNorm, _mm_mul_ps(vTemp ,vTemp));
+
+ // vSum += pV1[12..15] * pV2[12..15]
+ vTemp = _MM_LOAD(pVec1 + 12);
+ vSum = _mm_add_ps(vSum, _mm_mul_ps(vTemp, pVec2[3]));
+ vNorm = _mm_add_ps(vNorm, _mm_mul_ps(vTemp ,vTemp));
+
+ pVec1 += 16;
+ pVec2 += 4;
+ }
+
+ // return value = vSum[0] + vSum[1] + vSum[2] + vSum[3]
+ float *pvNorm = (float*)&vNorm;
+ double norm = sqrt(pvNorm[0] + pvNorm[1] + pvNorm[2] + pvNorm[3]);
+ if (norm < 1e-9) norm = 1.0; // to avoid div by zero
+
+ float *pvSum = (float*)&vSum;
+ return (double)(pvSum[0] + pvSum[1] + pvSum[2] + pvSum[3]) / norm;
+
+ /* This is approximately corresponding routine in C-language yet without normalization:
+ double corr, norm;
+ uint i;
+
+ // Calculates the cross-correlation value between 'pV1' and 'pV2' vectors
+ corr = norm = 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];
+
+ for (j = 0; j < 15; j ++) norm += pV1[j] * pV1[j];
+
+ pV1 += 16;
+ pV2 += 16;
+ }
+ return corr / sqrt(norm);
+ */
+
+ /* This is a bit outdated, corresponding routine in assembler. This may be teeny-weeny bit
+ faster than intrinsic version, but more difficult to maintain & get compiled on multiple
+ platforms.
+
+ uint overlapLengthLocal = overlapLength;
+ float corr;
+
+ _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()
+{
+ filterCoeffsAlign = NULL;
+ filterCoeffsUnalign = NULL;
+}
+
+
+FIRFilterSSE::~FIRFilterSSE()
+{
+ delete[] filterCoeffsUnalign;
+ filterCoeffsAlign = NULL;
+ filterCoeffsUnalign = NULL;
+}
+
+
+// (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 *)(((unsigned long)filterCoeffsUnalign + 15) & (ulong)-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 *source, uint numSamples) const
+{
+ int count = (int)((numSamples - length) & (uint)-2);
+ int j;
+
+ assert(count % 2 == 0);
+
+ if (count < 2) return 0;
+
+ assert(source != NULL);
+ assert(dest != NULL);
+ assert((length % 8) == 0);
+ assert(filterCoeffsAlign != NULL);
+ assert(((ulong)filterCoeffsAlign) % 16 == 0);
+
+ // filter is evaluated for two stereo samples with each iteration, thus use of 'j += 2'
+ for (j = 0; j < count; j += 2)
+ {
+ const float *pSrc;
+ const __m128 *pFil;
+ __m128 sum1, sum2;
+ uint i;
+
+ pSrc = (const float*)source; // source audio data
+ pFil = (const __m128*)filterCoeffsAlign; // filter coefficients. NOTE: Assumes coefficients
+ // are aligned to 16-byte boundary
+ sum1 = sum2 = _mm_setzero_ps();
+
+ for (i = 0; i < length / 8; i ++)
+ {
+ // Unroll loop for efficiency & calculate filter for 2*2 stereo samples
+ // at each pass
+
+ // sum1 is accu for 2*2 filtered stereo sound data at the primary sound data offset
+ // sum2 is accu for 2*2 filtered stereo sound data for the next sound sample offset.
+
+ sum1 = _mm_add_ps(sum1, _mm_mul_ps(_mm_loadu_ps(pSrc) , pFil[0]));
+ sum2 = _mm_add_ps(sum2, _mm_mul_ps(_mm_loadu_ps(pSrc + 2), pFil[0]));
+
+ sum1 = _mm_add_ps(sum1, _mm_mul_ps(_mm_loadu_ps(pSrc + 4), pFil[1]));
+ sum2 = _mm_add_ps(sum2, _mm_mul_ps(_mm_loadu_ps(pSrc + 6), pFil[1]));
+
+ sum1 = _mm_add_ps(sum1, _mm_mul_ps(_mm_loadu_ps(pSrc + 8) , pFil[2]));
+ sum2 = _mm_add_ps(sum2, _mm_mul_ps(_mm_loadu_ps(pSrc + 10), pFil[2]));
+
+ sum1 = _mm_add_ps(sum1, _mm_mul_ps(_mm_loadu_ps(pSrc + 12), pFil[3]));
+ sum2 = _mm_add_ps(sum2, _mm_mul_ps(_mm_loadu_ps(pSrc + 14), pFil[3]));
+
+ pSrc += 16;
+ pFil += 4;
+ }
+
+ // Now sum1 and sum2 both have a filtered 2-channel sample each, but we still need
+ // to sum the two hi- and lo-floats of these registers together.
+
+ // post-shuffle & add the filtered values and store to dest.
+ _mm_storeu_ps(dest, _mm_add_ps(
+ _mm_shuffle_ps(sum1, sum2, _MM_SHUFFLE(1,0,3,2)), // s2_1 s2_0 s1_3 s1_2
+ _mm_shuffle_ps(sum1, sum2, _MM_SHUFFLE(3,2,1,0)) // s2_3 s2_2 s1_1 s1_0
+ ));
+ source += 4;
+ dest += 4;
+ }
+
+ // Ideas for further improvement:
+ // 1. If it could be guaranteed that 'source' were always aligned to 16-byte
+ // boundary, a faster aligned '_mm_load_ps' instruction could be used.
+ // 2. If it could be guaranteed that 'dest' were always aligned to 16-byte
+ // boundary, a faster '_mm_store_ps' instruction could be used.
+
+ return (uint)count;
+
+ /* original routine in C-language. please notice the C-version has differently
+ organized coefficients though.
+ 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;
+ }
+ */
+
+
+ /* Similar routine in assembly, again obsoleted due to maintainability
+ _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
+ }
+ */
+}
+
+#endif // ALLOW_SSE
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