Initial community commit

Src/external_dependencies/openmpt-trunk/include/soundtouch/source/Makefile.am

@@ -0,0 +1,24 @@
+## Process this file with automake to create Makefile.in
+##
+## This file is part of SoundTouch, an audio processing library for pitch/time adjustments
+## 
+## SoundTouch is free software; you can redistribute it and/or modify it under the
+## terms of the GNU General Public License as published by the Free Software
+## Foundation; either version 2 of the License, or (at your option) any later
+## version.
+## 
+## SoundTouch is distributed in the hope that it will be useful, but WITHOUT ANY
+## WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
+## A PARTICULAR PURPOSE.  See the GNU General Public License for more details.
+## 
+## You should have received a copy of the GNU General Public License along with
+## this program; if not, write to the Free Software Foundation, Inc., 59 Temple
+## Place - Suite 330, Boston, MA  02111-1307, USA
+
+include $(top_srcdir)/config/am_include.mk
+
+SUBDIRS=SoundTouch SoundStretch
+
+# set to something if you want other stuff to be included in the distribution tarball
+#EXTRA_DIST=
+

Src/external_dependencies/openmpt-trunk/include/soundtouch/source/SoundTouch/AAFilter.cpp

@@ -0,0 +1,222 @@
+////////////////////////////////////////////////////////////////////////////////
+///
+/// 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
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// 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       3.14159265358979323846
+#define TWOPI    (2 * PI)
+
+// define this to save AA filter coefficients to a file
+// #define _DEBUG_SAVE_AAFILTER_COEFFICIENTS   1
+
+#ifdef _DEBUG_SAVE_AAFILTER_COEFFICIENTS
+    #include <stdio.h>
+
+    static void _DEBUG_SAVE_AAFIR_COEFFS(SAMPLETYPE *coeffs, int len)
+    {
+        FILE *fptr = fopen("aa_filter_coeffs.txt", "wt");
+        if (fptr == NULL) return;
+
+        for (int i = 0; i < len; i ++)
+        {
+            double temp = coeffs[i];
+            fprintf(fptr, "%lf\n", temp);
+        }
+        fclose(fptr);
+    }
+
+#else
+    #define _DEBUG_SAVE_AAFIR_COEFFS(x, y)
+#endif
+
+/*****************************************************************************
+ *
+ * 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 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];
+
+    wc = 2.0 * PI * cutoffFreq;
+    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 = 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 ++) 
+    {
+        temp = work[i] * scaleCoeff;
+        // scale & round to nearest integer
+        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);
+
+    _DEBUG_SAVE_AAFIR_COEFFS(coeffs, length);
+
+    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);
+}
+
+
+/// Applies the filter to the given src & dest pipes, so that processed amount of
+/// samples get removed from src, and produced amount added to dest 
+/// Note : The amount of outputted samples is by value of 'filter length' 
+/// smaller than the amount of input samples.
+uint AAFilter::evaluate(FIFOSampleBuffer &dest, FIFOSampleBuffer &src) const
+{
+    SAMPLETYPE *pdest;
+    const SAMPLETYPE *psrc;
+    uint numSrcSamples;
+    uint result;
+    int numChannels = src.getChannels();
+
+    assert(numChannels == dest.getChannels());
+
+    numSrcSamples = src.numSamples();
+    psrc = src.ptrBegin();
+    pdest = dest.ptrEnd(numSrcSamples);
+    result = pFIR->evaluate(pdest, psrc, numSrcSamples, numChannels);
+    src.receiveSamples(result);
+    dest.putSamples(result);
+
+    return result;
+}
+
+
+uint AAFilter::getLength() const
+{
+    return pFIR->getLength();
+}

Src/external_dependencies/openmpt-trunk/include/soundtouch/source/SoundTouch/AAFilter.h

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+////////////////////////////////////////////////////////////////////////////////
+///
+/// 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
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// 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"
+#include "FIFOSampleBuffer.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;
+
+    /// Applies the filter to the given src & dest pipes, so that processed amount of
+    /// samples get removed from src, and produced amount added to dest 
+    /// Note : The amount of outputted samples is by value of 'filter length' 
+    /// smaller than the amount of input samples.
+    uint evaluate(FIFOSampleBuffer &dest, 
+                  FIFOSampleBuffer &src) const;
+
+};
+
+}
+
+#endif

Src/external_dependencies/openmpt-trunk/include/soundtouch/source/SoundTouch/BPMDetect.cpp

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+////////////////////////////////////////////////////////////////////////////////
+///
+/// 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
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// 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
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#define _USE_MATH_DEFINES
+
+#include <math.h>
+#include <assert.h>
+#include <string.h>
+#include <stdio.h>
+#include <cfloat>
+#include "FIFOSampleBuffer.h"
+#include "PeakFinder.h"
+#include "BPMDetect.h"
+
+using namespace soundtouch;
+
+// algorithm input sample block size
+static const int INPUT_BLOCK_SIZE = 2048;
+
+// decimated sample block size
+static const int DECIMATED_BLOCK_SIZE = 256;
+
+/// Target sample rate after decimation
+static const int TARGET_SRATE = 1000;
+
+/// XCorr update sequence size, update in about 200msec chunks
+static const int XCORR_UPDATE_SEQUENCE = (int)(TARGET_SRATE / 5);
+
+/// Moving average N size
+static const int MOVING_AVERAGE_N = 15;
+
+/// XCorr decay time constant, decay to half in 30 seconds
+/// If it's desired to have the system adapt quicker to beat rate 
+/// changes within a continuing music stream, then the 
+/// 'xcorr_decay_time_constant' value can be reduced, yet that
+/// can increase possibility of glitches in bpm detection.
+static const double XCORR_DECAY_TIME_CONSTANT = 30.0;
+
+/// Data overlap factor for beat detection algorithm
+static const int OVERLAP_FACTOR = 4;
+
+static const double TWOPI = (2 * M_PI);
+
+////////////////////////////////////////////////////////////////////////////////
+
+// Enable following define to create bpm analysis file:
+
+//#define _CREATE_BPM_DEBUG_FILE
+
+#ifdef _CREATE_BPM_DEBUG_FILE
+
+    static void _SaveDebugData(const char *name, const float *data, int minpos, int maxpos, double coeff)
+    {
+        FILE *fptr = fopen(name, "wt");
+        int i;
+
+        if (fptr)
+        {
+            printf("\nWriting BPM debug data into file %s\n", name);
+            for (i = minpos; i < maxpos; i ++)
+            {
+                fprintf(fptr, "%d\t%.1lf\t%f\n", i, coeff / (double)i, data[i]);
+            }
+            fclose(fptr);
+        }
+    }
+
+    void _SaveDebugBeatPos(const char *name, const std::vector<BEAT> &beats)
+    {
+        printf("\nWriting beat detections data into file %s\n", name);
+
+        FILE *fptr = fopen(name, "wt");
+        if (fptr)
+        {
+            for (uint i = 0; i < beats.size(); i++)
+            {
+                BEAT b = beats[i];
+                fprintf(fptr, "%lf\t%lf\n", b.pos, b.strength);
+            }
+            fclose(fptr);
+        }
+    }
+#else
+    #define _SaveDebugData(name, a,b,c,d)
+    #define _SaveDebugBeatPos(name, b)
+#endif
+
+// Hamming window
+void hamming(float *w, int N)
+{
+    for (int i = 0; i < N; i++)
+    {
+        w[i] = (float)(0.54 - 0.46 * cos(TWOPI * i / (N - 1)));
+    }
+
+}
+
+////////////////////////////////////////////////////////////////////////////////
+//
+// IIR2_filter - 2nd order IIR filter
+
+IIR2_filter::IIR2_filter(const double *lpf_coeffs)
+{
+    memcpy(coeffs, lpf_coeffs, 5 * sizeof(double));
+    memset(prev, 0, sizeof(prev));
+}
+
+
+float IIR2_filter::update(float x)
+{
+    prev[0] = x;
+    double y = x * coeffs[0];
+
+    for (int i = 4; i >= 1; i--)
+    {
+        y += coeffs[i] * prev[i];
+        prev[i] = prev[i - 1];
+    }
+
+    prev[3] = y;
+    return (float)y;
+}
+
+
+// IIR low-pass filter coefficients, calculated with matlab/octave cheby2(2,40,0.05)
+const double _LPF_coeffs[5] = { 0.00996655391939, -0.01944529148401, 0.00996655391939, 1.96867605796247, -0.96916387431724 };
+
+////////////////////////////////////////////////////////////////////////////////
+
+BPMDetect::BPMDetect(int numChannels, int aSampleRate) :
+    beat_lpf(_LPF_coeffs)
+{
+    beats.reserve(250); // initial reservation to prevent frequent reallocation
+
+    this->sampleRate = aSampleRate;
+    this->channels = numChannels;
+
+    decimateSum = 0;
+    decimateCount = 0;
+
+    // choose decimation factor so that result is approx. 1000 Hz
+    decimateBy = sampleRate / TARGET_SRATE;
+    if ((decimateBy <= 0) || (decimateBy * DECIMATED_BLOCK_SIZE < INPUT_BLOCK_SIZE))
+    {
+        ST_THROW_RT_ERROR("Too small samplerate");
+    }
+
+    // Calculate window length & starting item according to desired min & max bpms
+    windowLen = (60 * sampleRate) / (decimateBy * MIN_BPM);
+    windowStart = (60 * sampleRate) / (decimateBy * MAX_BPM_RANGE);
+
+    assert(windowLen > windowStart);
+
+    // allocate new working objects
+    xcorr = new float[windowLen];
+    memset(xcorr, 0, windowLen * sizeof(float));
+
+    pos = 0;
+    peakPos = 0;
+    peakVal = 0;
+    init_scaler = 1;
+    beatcorr_ringbuffpos = 0;
+    beatcorr_ringbuff = new float[windowLen];
+    memset(beatcorr_ringbuff, 0, windowLen * sizeof(float));
+
+    // allocate processing buffer
+    buffer = new FIFOSampleBuffer();
+    // we do processing in mono mode
+    buffer->setChannels(1);
+    buffer->clear();
+
+    // calculate hamming windows
+    hamw = new float[XCORR_UPDATE_SEQUENCE];
+    hamming(hamw, XCORR_UPDATE_SEQUENCE);
+    hamw2 = new float[XCORR_UPDATE_SEQUENCE / 2];
+    hamming(hamw2, XCORR_UPDATE_SEQUENCE / 2);
+}
+
+
+BPMDetect::~BPMDetect()
+{
+    delete[] xcorr;
+    delete[] beatcorr_ringbuff;
+    delete[] hamw;
+    delete[] hamw2;
+    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 SOUNDTOUCH_INTEGER_SAMPLES
+            // check ranges for sure (shouldn't actually be necessary)
+            if (out > 32767) 
+            {
+                out = 32767;
+            } 
+            else if (out < -32768) 
+            {
+                out = -32768;
+            }
+#endif // SOUNDTOUCH_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));
+    assert(process_samples == XCORR_UPDATE_SEQUENCE);
+
+    pBuffer = buffer->ptrBegin();
+
+    // calculate decay factor for xcorr filtering
+    float xcorr_decay = (float)pow(0.5, 1.0 / (XCORR_DECAY_TIME_CONSTANT * TARGET_SRATE / process_samples));
+
+    // prescale pbuffer
+    float tmp[XCORR_UPDATE_SEQUENCE];
+    for (int i = 0; i < process_samples; i++)
+    {
+        tmp[i] = hamw[i] * hamw[i] * pBuffer[i];
+    }
+
+    #pragma omp parallel for
+    for (offs = windowStart; offs < windowLen; offs ++) 
+    {
+        float sum;
+        int i;
+
+        sum = 0;
+        for (i = 0; i < process_samples; i ++) 
+        {
+            sum += tmp[i] * pBuffer[i + offs];  // scaling the sub-result shouldn't be necessary
+        }
+        xcorr[offs] *= xcorr_decay;   // decay 'xcorr' here with suitable time constant.
+
+        xcorr[offs] += (float)fabs(sum);
+    }
+}
+
+
+// Detect individual beat positions
+void BPMDetect::updateBeatPos(int process_samples)
+{
+    SAMPLETYPE *pBuffer;
+
+    assert(buffer->numSamples() >= (uint)(process_samples + windowLen));
+
+    pBuffer = buffer->ptrBegin();
+    assert(process_samples == XCORR_UPDATE_SEQUENCE / 2);
+
+    //    static double thr = 0.0003;
+    double posScale = (double)this->decimateBy / (double)this->sampleRate;
+    int resetDur = (int)(0.12 / posScale + 0.5);
+
+    // prescale pbuffer
+    float tmp[XCORR_UPDATE_SEQUENCE / 2];
+    for (int i = 0; i < process_samples; i++)
+    {
+        tmp[i] = hamw2[i] * hamw2[i] * pBuffer[i];
+    }
+
+    #pragma omp parallel for
+    for (int offs = windowStart; offs < windowLen; offs++)
+    {
+        float sum = 0;
+        for (int i = 0; i < process_samples; i++)
+        {
+            sum += tmp[i] * pBuffer[offs + i];
+        }
+        beatcorr_ringbuff[(beatcorr_ringbuffpos + offs) % windowLen] += (float)((sum > 0) ? sum : 0); // accumulate only positive correlations
+    }
+
+    int skipstep = XCORR_UPDATE_SEQUENCE / OVERLAP_FACTOR;
+
+    // compensate empty buffer at beginning by scaling coefficient
+    float scale = (float)windowLen / (float)(skipstep * init_scaler);
+    if (scale > 1.0f)
+    {
+        init_scaler++;
+    }
+    else
+    {
+        scale = 1.0f;
+    }
+
+    // detect beats
+    for (int i = 0; i < skipstep; i++)
+    {
+        LONG_SAMPLETYPE max = 0;
+
+        float sum = beatcorr_ringbuff[beatcorr_ringbuffpos];
+        sum -= beat_lpf.update(sum);
+
+        if (sum > peakVal)
+        {
+            // found new local largest value
+            peakVal = sum;
+            peakPos = pos;
+        }
+        if (pos > peakPos + resetDur)
+        {
+            // largest value not updated for 200msec => accept as beat
+            peakPos += skipstep;
+            if (peakVal > 0)
+            {
+                // add detected beat to end of "beats" vector
+                BEAT temp = { (float)(peakPos * posScale), (float)(peakVal * scale) };
+                beats.push_back(temp);
+            }
+
+            peakVal = 0;
+            peakPos = pos;
+        }
+
+        beatcorr_ringbuff[beatcorr_ringbuffpos] = 0;
+        pos++;
+        beatcorr_ringbuffpos = (beatcorr_ringbuffpos + 1) % windowLen;
+    }
+}
+
+
+#define max(x,y) ((x) > (y) ? (x) : (y))
+
+void BPMDetect::inputSamples(const SAMPLETYPE *samples, int numSamples)
+{
+    SAMPLETYPE decimated[DECIMATED_BLOCK_SIZE];
+
+    // iterate so that max INPUT_BLOCK_SAMPLES processed per iteration
+    while (numSamples > 0)
+    {
+        int block;
+        int decSamples;
+
+        block = (numSamples > INPUT_BLOCK_SIZE) ? INPUT_BLOCK_SIZE : numSamples;
+
+        // decimate. note that converts to mono at the same time
+        decSamples = decimate(decimated, samples, block);
+        samples += block * channels;
+        numSamples -= block;
+
+        buffer->putSamples(decimated, decSamples);
+    }
+
+    // when the buffer has enough samples for processing...
+    int req = max(windowLen + XCORR_UPDATE_SEQUENCE, 2 * XCORR_UPDATE_SEQUENCE);
+    while ((int)buffer->numSamples() >= req) 
+    {
+        // ... update autocorrelations...
+        updateXCorr(XCORR_UPDATE_SEQUENCE);
+        // ...update beat position calculation...
+        updateBeatPos(XCORR_UPDATE_SEQUENCE / 2);
+        // ... and remove proceessed samples from the buffer
+        int n = XCORR_UPDATE_SEQUENCE / OVERLAP_FACTOR;
+        buffer->receiveSamples(n);
+    }
+}
+
+
+void BPMDetect::removeBias()
+{
+    int i;
+
+    // Remove linear bias: calculate linear regression coefficient
+    // 1. calc mean of 'xcorr' and 'i'
+    double mean_i = 0;
+    double mean_x = 0;
+    for (i = windowStart; i < windowLen; i++)
+    {
+        mean_x += xcorr[i];
+    }
+    mean_x /= (windowLen - windowStart);
+    mean_i = 0.5 * (windowLen - 1 + windowStart);
+
+    // 2. calculate linear regression coefficient
+    double b = 0;
+    double div = 0;
+    for (i = windowStart; i < windowLen; i++)
+    {
+        double xt = xcorr[i] - mean_x;
+        double xi = i - mean_i;
+        b += xt * xi;
+        div += xi * xi;
+    }
+    b /= div;
+
+    // subtract linear regression and resolve min. value bias
+    float minval = FLT_MAX;   // arbitrary large number
+    for (i = windowStart; i < windowLen; i ++)
+    {
+        xcorr[i] -= (float)(b * i);
+        if (xcorr[i] < minval)
+        {
+            minval = xcorr[i];
+        }
+    }
+
+    // subtract min.value
+    for (i = windowStart; i < windowLen; i ++)
+    {
+        xcorr[i] -= minval;
+    }
+}
+
+
+// Calculate N-point moving average for "source" values
+void MAFilter(float *dest, const float *source, int start, int end, int N)
+{
+    for (int i = start; i < end; i++)
+    {
+        int i1 = i - N / 2;
+        int i2 = i + N / 2 + 1;
+        if (i1 < start) i1 = start;
+        if (i2 > end)   i2 = end;
+
+        double sum = 0;
+        for (int j = i1; j < i2; j ++)
+        { 
+            sum += source[j];
+        }
+        dest[i] = (float)(sum / (i2 - i1));
+    }
+}
+
+
+float BPMDetect::getBpm()
+{
+    double peakPos;
+    double coeff;
+    PeakFinder peakFinder;
+
+    // remove bias from xcorr data
+    removeBias();
+
+    coeff = 60.0 * ((double)sampleRate / (double)decimateBy);
+
+    // save bpm debug data if debug data writing enabled
+    _SaveDebugData("soundtouch-bpm-xcorr.txt", xcorr, windowStart, windowLen, coeff);
+
+    // Smoothen by N-point moving-average
+    float *data = new float[windowLen];
+    memset(data, 0, sizeof(float) * windowLen);
+    MAFilter(data, xcorr, windowStart, windowLen, MOVING_AVERAGE_N);
+
+    // find peak position
+    peakPos = peakFinder.detectPeak(data, windowStart, windowLen);
+
+    // save bpm debug data if debug data writing enabled
+    _SaveDebugData("soundtouch-bpm-smoothed.txt", data, windowStart, windowLen, coeff);
+
+    delete[] data;
+
+    assert(decimateBy != 0);
+    if (peakPos < 1e-9) return 0.0; // detection failed.
+
+    _SaveDebugBeatPos("soundtouch-detected-beats.txt", beats);
+
+    // calculate BPM
+    float bpm = (float)(coeff / peakPos);
+    return (bpm >= MIN_BPM && bpm <= MAX_BPM_VALID) ? bpm : 0;
+}
+
+
+/// Get beat position arrays. Note: The array includes also really low beat detection values 
+/// in absence of clear strong beats. Consumer may wish to filter low values away.
+/// - "pos" receive array of beat positions
+/// - "values" receive array of beat detection strengths
+/// - max_num indicates max.size of "pos" and "values" array.  
+///
+/// You can query a suitable array sized by calling this with NULL in "pos" & "values".
+///
+/// \return number of beats in the arrays.
+int BPMDetect::getBeats(float *pos, float *values, int max_num)
+{
+    int num = (int)beats.size();
+    if ((!pos) || (!values)) return num;    // pos or values NULL, return just size
+
+    for (int i = 0; (i < num) && (i < max_num); i++)
+    {
+        pos[i] = beats[i].pos;
+        values[i] = beats[i].strength;
+    }
+    return num;
+}

Src/external_dependencies/openmpt-trunk/include/soundtouch/source/SoundTouch/FIFOSampleBuffer.cpp

@@ -0,0 +1,275 @@
+////////////////////////////////////////////////////////////////////////////////
+///
+/// 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
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// 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 "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;
+
+    if (!verifyNumberOfChannels(numChannels)) return;
+
+    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
+// successfully 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 enough 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)
+        {
+            ST_THROW_RT_ERROR("Couldn't allocate memory!\n");
+        }
+        // Align the buffer to begin at 16byte cache line boundary for optimal performance
+        temp = (SAMPLETYPE *)SOUNDTOUCH_ALIGN_POINTER_16(tempUnaligned);
+        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;
+}
+
+
+/// allow trimming (downwards) amount of samples in pipeline.
+/// Returns adjusted amount of samples
+uint FIFOSampleBuffer::adjustAmountOfSamples(uint numSamples)
+{
+    if (numSamples < samplesInBuffer)
+    {
+        samplesInBuffer = numSamples;
+    }
+    return samplesInBuffer;
+}
+
+
+/// Add silence to end of buffer
+void FIFOSampleBuffer::addSilent(uint nSamples)
+{
+    memset(ptrEnd(nSamples), 0, sizeof(SAMPLETYPE) * nSamples * channels);
+    samplesInBuffer += nSamples;
+}

Src/external_dependencies/openmpt-trunk/include/soundtouch/source/SoundTouch/FIRFilter.cpp

@@ -0,0 +1,329 @@
+////////////////////////////////////////////////////////////////////////////////
+///
+/// General FIR digital filter routines with MMX optimization. 
+///
+/// Notes : MMX optimized functions reside in a separate, platform-specific file, 
+/// e.g. 'mmx_win.cpp' or 'mmx_gcc.cpp'
+///
+/// This source file contains OpenMP optimizations that allow speeding up the
+/// corss-correlation algorithm by executing it in several threads / CPU cores 
+/// in parallel. See the following article link for more detailed discussion 
+/// about SoundTouch OpenMP optimizations:
+/// http://www.softwarecoven.com/parallel-computing-in-embedded-mobile-devices
+///
+/// Author        : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// 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 "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;
+    filterCoeffsStereo = NULL;
+}
+
+
+FIRFilter::~FIRFilter()
+{
+    delete[] filterCoeffs;
+    delete[] filterCoeffsStereo;
+}
+
+
+// Usual C-version of the filter routine for stereo sound
+uint FIRFilter::evaluateFilterStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples) const
+{
+    int j, end;
+#ifdef SOUNDTOUCH_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
+    // hint compiler autovectorization that loop length is divisible by 8
+    int ilength = length & -8;
+
+    assert((length != 0) && (length == ilength) && (src != NULL) && (dest != NULL) && (filterCoeffs != NULL));
+
+    end = 2 * (numSamples - ilength);
+
+    #pragma omp parallel for
+    for (j = 0; j < end; j += 2) 
+    {
+        const SAMPLETYPE *ptr;
+        LONG_SAMPLETYPE suml, sumr;
+
+        suml = sumr = 0;
+        ptr = src + j;
+
+        for (int i = 0; i < ilength; i ++)
+        {
+            suml += ptr[2 * i] * filterCoeffsStereo[2 * i];
+            sumr += ptr[2 * i + 1] * filterCoeffsStereo[2 * i + 1];
+        }
+
+#ifdef SOUNDTOUCH_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;
+#endif // SOUNDTOUCH_INTEGER_SAMPLES
+        dest[j] = (SAMPLETYPE)suml;
+        dest[j + 1] = (SAMPLETYPE)sumr;
+    }
+    return numSamples - ilength;
+}
+
+
+// Usual C-version of the filter routine for mono sound
+uint FIRFilter::evaluateFilterMono(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples) const
+{
+    int j, end;
+#ifdef SOUNDTOUCH_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
+
+    // hint compiler autovectorization that loop length is divisible by 8
+    int ilength = length & -8;
+
+    assert(ilength != 0);
+
+    end = numSamples - ilength;
+    #pragma omp parallel for
+    for (j = 0; j < end; j ++)
+    {
+        const SAMPLETYPE *pSrc = src + j;
+        LONG_SAMPLETYPE sum;
+        int i;
+
+        sum = 0;
+        for (i = 0; i < ilength; i ++)
+        {
+            sum += pSrc[i] * filterCoeffs[i];
+        }
+#ifdef SOUNDTOUCH_INTEGER_SAMPLES
+        sum >>= resultDivFactor;
+        // saturate to 16 bit integer limits
+        sum = (sum < -32768) ? -32768 : (sum > 32767) ? 32767 : sum;
+#endif // SOUNDTOUCH_INTEGER_SAMPLES
+        dest[j] = (SAMPLETYPE)sum;
+    }
+    return end;
+}
+
+
+uint FIRFilter::evaluateFilterMulti(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples, uint numChannels)
+{
+    int j, end;
+
+#ifdef SOUNDTOUCH_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);
+    assert(numChannels < 16);
+
+    // hint compiler autovectorization that loop length is divisible by 8
+    int ilength = length & -8;
+
+    end = numChannels * (numSamples - ilength);
+
+    #pragma omp parallel for
+    for (j = 0; j < end; j += numChannels)
+    {
+        const SAMPLETYPE *ptr;
+        LONG_SAMPLETYPE sums[16];
+        uint c;
+        int i;
+
+        for (c = 0; c < numChannels; c ++)
+        {
+            sums[c] = 0;
+        }
+
+        ptr = src + j;
+
+        for (i = 0; i < ilength; i ++)
+        {
+            SAMPLETYPE coef=filterCoeffs[i];
+            for (c = 0; c < numChannels; c ++)
+            {
+                sums[c] += ptr[0] * coef;
+                ptr ++;
+            }
+        }
+        
+        for (c = 0; c < numChannels; c ++)
+        {
+#ifdef SOUNDTOUCH_INTEGER_SAMPLES
+            sums[c] >>= resultDivFactor;
+#endif // SOUNDTOUCH_INTEGER_SAMPLES
+            dest[j+c] = (SAMPLETYPE)sums[c];
+        }
+    }
+    return numSamples - ilength;
+}
+
+
+// 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) ST_THROW_RT_ERROR("FIR filter length not divisible by 8");
+
+    #ifdef SOUNDTOUCH_FLOAT_SAMPLES
+        // scale coefficients already here if using floating samples
+        double scale = 1.0 / resultDivider;
+    #else
+        short scale = 1;
+    #endif
+
+    lengthDiv8 = newLength / 8;
+    length = lengthDiv8 * 8;
+    assert(length == newLength);
+
+    resultDivFactor = uResultDivFactor;
+    resultDivider = (SAMPLETYPE)::pow(2.0, (int)resultDivFactor);
+
+    delete[] filterCoeffs;
+    filterCoeffs = new SAMPLETYPE[length];
+    delete[] filterCoeffsStereo;
+    filterCoeffsStereo = new SAMPLETYPE[length*2];
+    for (uint i = 0; i < length; i ++)
+    {
+        filterCoeffs[i] = (SAMPLETYPE)(coeffs[i] * scale);
+        // create also stereo set of filter coefficients: this allows compiler
+        // to autovectorize filter evaluation much more efficiently
+        filterCoeffsStereo[2 * i] = (SAMPLETYPE)(coeffs[i] * scale);
+        filterCoeffsStereo[2 * i + 1] = (SAMPLETYPE)(coeffs[i] * scale);
+    }
+}
+
+
+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) 
+{
+    assert(length > 0);
+    assert(lengthDiv8 * 8 == length);
+
+    if (numSamples < length) return 0;
+
+#ifndef USE_MULTICH_ALWAYS
+    if (numChannels == 1)
+    {
+        return evaluateFilterMono(dest, src, numSamples);
+    } 
+    else if (numChannels == 2)
+    {
+        return evaluateFilterStereo(dest, src, numSamples);
+    }
+    else
+#endif // USE_MULTICH_ALWAYS
+    {
+        assert(numChannels > 0);
+        return evaluateFilterMulti(dest, src, numSamples, numChannels);
+    }
+}
+
+
+// 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!
+    ST_THROW_RT_ERROR("Error in FIRFilter::new: Don't use 'new FIRFilter', use 'newInstance' member instead!");
+    return newInstance();
+}
+
+
+FIRFilter * FIRFilter::newInstance()
+{
+    uint uExtensions;
+
+    uExtensions = detectCPUextensions();
+
+    // Check if MMX/SSE instruction set extensions supported by CPU
+
+#ifdef SOUNDTOUCH_ALLOW_MMX
+    // MMX routines available only with integer sample types
+    if (uExtensions & SUPPORT_MMX)
+    {
+        return ::new FIRFilterMMX;
+    }
+    else
+#endif // SOUNDTOUCH_ALLOW_MMX
+
+#ifdef SOUNDTOUCH_ALLOW_SSE
+    if (uExtensions & SUPPORT_SSE)
+    {
+        // SSE support
+        return ::new FIRFilterSSE;
+    }
+    else
+#endif // SOUNDTOUCH_ALLOW_SSE
+
+    {
+        // ISA optimizations not supported, use plain C version
+        return ::new FIRFilter;
+    }
+}

Src/external_dependencies/openmpt-trunk/include/soundtouch/source/SoundTouch/FIRFilter.h

@@ -0,0 +1,140 @@
+////////////////////////////////////////////////////////////////////////////////
+///
+/// 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
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// 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;
+    SAMPLETYPE *filterCoeffsStereo;
+
+    virtual uint evaluateFilterStereo(SAMPLETYPE *dest, 
+                                      const SAMPLETYPE *src, 
+                                      uint numSamples) const;
+    virtual uint evaluateFilterMono(SAMPLETYPE *dest, 
+                                    const SAMPLETYPE *src, 
+                                    uint numSamples) const;
+    virtual uint evaluateFilterMulti(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples, uint numChannels);
+
+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);
+
+    uint getLength() const;
+
+    virtual void setCoefficients(const SAMPLETYPE *coeffs, 
+                                 uint newLength, 
+                                 uint uResultDivFactor);
+};
+
+
+// Optional subclasses that implement CPU-specific optimizations:
+
+#ifdef SOUNDTOUCH_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 // SOUNDTOUCH_ALLOW_MMX
+
+
+#ifdef SOUNDTOUCH_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 // SOUNDTOUCH_ALLOW_SSE
+
+}
+
+#endif  // FIRFilter_H

Src/external_dependencies/openmpt-trunk/include/soundtouch/source/SoundTouch/InterpolateCubic.cpp

@@ -0,0 +1,196 @@
+////////////////////////////////////////////////////////////////////////////////
+/// 
+/// Cubic interpolation routine.
+///
+/// Author        : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// 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 <stddef.h>
+#include <math.h>
+#include "InterpolateCubic.h"
+#include "STTypes.h"
+
+using namespace soundtouch;
+
+// cubic interpolation coefficients
+static const float _coeffs[]= 
+{ -0.5f,  1.0f, -0.5f, 0.0f,
+   1.5f, -2.5f,  0.0f, 1.0f,
+  -1.5f,  2.0f,  0.5f, 0.0f,
+   0.5f, -0.5f,  0.0f, 0.0f};
+
+
+InterpolateCubic::InterpolateCubic()
+{
+    fract = 0;
+}
+
+
+void InterpolateCubic::resetRegisters()
+{
+    fract = 0;
+}
+
+
+/// Transpose mono audio. Returns number of produced output samples, and 
+/// updates "srcSamples" to amount of consumed source samples
+int InterpolateCubic::transposeMono(SAMPLETYPE *pdest, 
+                    const SAMPLETYPE *psrc, 
+                    int &srcSamples)
+{
+    int i;
+    int srcSampleEnd = srcSamples - 4;
+    int srcCount = 0;
+
+    i = 0;
+    while (srcCount < srcSampleEnd)
+    {
+        float out;
+        const float x3 = 1.0f;
+        const float x2 = (float)fract;    // x
+        const float x1 = x2*x2;           // x^2
+        const float x0 = x1*x2;           // x^3
+        float y0, y1, y2, y3;
+
+        assert(fract < 1.0);
+
+        y0 =  _coeffs[0] * x0 +  _coeffs[1] * x1 +  _coeffs[2] * x2 +  _coeffs[3] * x3;
+        y1 =  _coeffs[4] * x0 +  _coeffs[5] * x1 +  _coeffs[6] * x2 +  _coeffs[7] * x3;
+        y2 =  _coeffs[8] * x0 +  _coeffs[9] * x1 + _coeffs[10] * x2 + _coeffs[11] * x3;
+        y3 = _coeffs[12] * x0 + _coeffs[13] * x1 + _coeffs[14] * x2 + _coeffs[15] * x3;
+
+        out = y0 * psrc[0] + y1 * psrc[1] + y2 * psrc[2] + y3 * psrc[3];
+
+        pdest[i] = (SAMPLETYPE)out;
+        i ++;
+
+        // update position fraction
+        fract += rate;
+        // update whole positions
+        int whole = (int)fract;
+        fract -= whole;
+        psrc += whole;
+        srcCount += whole;
+    }
+    srcSamples = srcCount;
+    return i;
+}
+
+
+/// Transpose stereo audio. Returns number of produced output samples, and 
+/// updates "srcSamples" to amount of consumed source samples
+int InterpolateCubic::transposeStereo(SAMPLETYPE *pdest, 
+                    const SAMPLETYPE *psrc, 
+                    int &srcSamples)
+{
+    int i;
+    int srcSampleEnd = srcSamples - 4;
+    int srcCount = 0;
+
+    i = 0;
+    while (srcCount < srcSampleEnd)
+    {
+        const float x3 = 1.0f;
+        const float x2 = (float)fract;    // x
+        const float x1 = x2*x2;           // x^2
+        const float x0 = x1*x2;           // x^3
+        float y0, y1, y2, y3;
+        float out0, out1;
+
+        assert(fract < 1.0);
+
+        y0 =  _coeffs[0] * x0 +  _coeffs[1] * x1 +  _coeffs[2] * x2 +  _coeffs[3] * x3;
+        y1 =  _coeffs[4] * x0 +  _coeffs[5] * x1 +  _coeffs[6] * x2 +  _coeffs[7] * x3;
+        y2 =  _coeffs[8] * x0 +  _coeffs[9] * x1 + _coeffs[10] * x2 + _coeffs[11] * x3;
+        y3 = _coeffs[12] * x0 + _coeffs[13] * x1 + _coeffs[14] * x2 + _coeffs[15] * x3;
+
+        out0 = y0 * psrc[0] + y1 * psrc[2] + y2 * psrc[4] + y3 * psrc[6];
+        out1 = y0 * psrc[1] + y1 * psrc[3] + y2 * psrc[5] + y3 * psrc[7];
+
+        pdest[2*i]   = (SAMPLETYPE)out0;
+        pdest[2*i+1] = (SAMPLETYPE)out1;
+        i ++;
+
+        // update position fraction
+        fract += rate;
+        // update whole positions
+        int whole = (int)fract;
+        fract -= whole;
+        psrc += 2*whole;
+        srcCount += whole;
+    }
+    srcSamples = srcCount;
+    return i;
+}
+
+
+/// Transpose multi-channel audio. Returns number of produced output samples, and 
+/// updates "srcSamples" to amount of consumed source samples
+int InterpolateCubic::transposeMulti(SAMPLETYPE *pdest, 
+                    const SAMPLETYPE *psrc, 
+                    int &srcSamples)
+{
+    int i;
+    int srcSampleEnd = srcSamples - 4;
+    int srcCount = 0;
+
+    i = 0;
+    while (srcCount < srcSampleEnd)
+    {
+        const float x3 = 1.0f;
+        const float x2 = (float)fract;    // x
+        const float x1 = x2*x2;           // x^2
+        const float x0 = x1*x2;           // x^3
+        float y0, y1, y2, y3;
+
+        assert(fract < 1.0);
+
+        y0 =  _coeffs[0] * x0 +  _coeffs[1] * x1 +  _coeffs[2] * x2 +  _coeffs[3] * x3;
+        y1 =  _coeffs[4] * x0 +  _coeffs[5] * x1 +  _coeffs[6] * x2 +  _coeffs[7] * x3;
+        y2 =  _coeffs[8] * x0 +  _coeffs[9] * x1 + _coeffs[10] * x2 + _coeffs[11] * x3;
+        y3 = _coeffs[12] * x0 + _coeffs[13] * x1 + _coeffs[14] * x2 + _coeffs[15] * x3;
+
+        for (int c = 0; c < numChannels; c ++)
+        {
+            float out;
+            out = y0 * psrc[c] + y1 * psrc[c + numChannels] + y2 * psrc[c + 2 * numChannels] + y3 * psrc[c + 3 * numChannels];
+            pdest[0] = (SAMPLETYPE)out;
+            pdest ++;
+        }
+        i ++;
+
+        // update position fraction
+        fract += rate;
+        // update whole positions
+        int whole = (int)fract;
+        fract -= whole;
+        psrc += numChannels*whole;
+        srcCount += whole;
+    }
+    srcSamples = srcCount;
+    return i;
+}

Src/external_dependencies/openmpt-trunk/include/soundtouch/source/SoundTouch/InterpolateCubic.h

@@ -0,0 +1,69 @@
+////////////////////////////////////////////////////////////////////////////////
+/// 
+/// Cubic interpolation routine.
+///
+/// Author        : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// 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 _InterpolateCubic_H_
+#define _InterpolateCubic_H_
+
+#include "RateTransposer.h"
+#include "STTypes.h"
+
+namespace soundtouch
+{
+
+class InterpolateCubic : public TransposerBase
+{
+protected:
+    virtual int transposeMono(SAMPLETYPE *dest, 
+                        const SAMPLETYPE *src, 
+                        int &srcSamples);
+    virtual int transposeStereo(SAMPLETYPE *dest, 
+                        const SAMPLETYPE *src, 
+                        int &srcSamples);
+    virtual int transposeMulti(SAMPLETYPE *dest, 
+                        const SAMPLETYPE *src, 
+                        int &srcSamples);
+
+    double fract;
+
+public:
+    InterpolateCubic();
+
+    virtual void resetRegisters();
+
+    int getLatency() const
+    {
+        return 1;
+    }
+};
+
+}
+
+#endif

Src/external_dependencies/openmpt-trunk/include/soundtouch/source/SoundTouch/InterpolateLinear.cpp

@@ -0,0 +1,296 @@
+////////////////////////////////////////////////////////////////////////////////
+/// 
+/// Linear interpolation algorithm.
+///
+/// Author        : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// 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 "InterpolateLinear.h"
+
+using namespace soundtouch;
+
+//////////////////////////////////////////////////////////////////////////////
+//
+// InterpolateLinearInteger - integer arithmetic implementation
+// 
+
+/// fixed-point interpolation routine precision
+#define SCALE    65536
+
+
+// Constructor
+InterpolateLinearInteger::InterpolateLinearInteger() : TransposerBase()
+{
+    // Notice: use local function calling syntax for sake of clarity, 
+    // to indicate the fact that C++ constructor can't call virtual functions.
+    resetRegisters();
+    setRate(1.0f);
+}
+
+
+void InterpolateLinearInteger::resetRegisters()
+{
+    iFract = 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
+int InterpolateLinearInteger::transposeMono(SAMPLETYPE *dest, const SAMPLETYPE *src, int &srcSamples)
+{
+    int i;
+    int srcSampleEnd = srcSamples - 1;
+    int srcCount = 0;
+
+    i = 0;
+    while (srcCount < srcSampleEnd)
+    {
+        LONG_SAMPLETYPE temp;
+    
+        assert(iFract < SCALE);
+
+        temp = (SCALE - iFract) * src[0] + iFract * src[1];
+        dest[i] = (SAMPLETYPE)(temp / SCALE);
+        i++;
+
+        iFract += iRate;
+
+        int iWhole = iFract / SCALE;
+        iFract -= iWhole * SCALE;
+        srcCount += iWhole;
+        src += iWhole;
+    }
+    srcSamples = srcCount;
+
+    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
+int InterpolateLinearInteger::transposeStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, int &srcSamples)
+{
+    int i;
+    int srcSampleEnd = srcSamples - 1;
+    int srcCount = 0;
+
+    i = 0;
+    while (srcCount < srcSampleEnd)
+    {
+        LONG_SAMPLETYPE temp0;
+        LONG_SAMPLETYPE temp1;
+    
+        assert(iFract < SCALE);
+
+        temp0 = (SCALE - iFract) * src[0] + iFract * src[2];
+        temp1 = (SCALE - iFract) * src[1] + iFract * src[3];
+        dest[0] = (SAMPLETYPE)(temp0 / SCALE);
+        dest[1] = (SAMPLETYPE)(temp1 / SCALE);
+        dest += 2;
+        i++;
+
+        iFract += iRate;
+
+        int iWhole = iFract / SCALE;
+        iFract -= iWhole * SCALE;
+        srcCount += iWhole;
+        src += 2*iWhole;
+    }
+    srcSamples = srcCount;
+
+    return i;
+}
+
+
+int InterpolateLinearInteger::transposeMulti(SAMPLETYPE *dest, const SAMPLETYPE *src, int &srcSamples)
+{
+    int i;
+    int srcSampleEnd = srcSamples - 1;
+    int srcCount = 0;
+
+    i = 0;
+    while (srcCount < srcSampleEnd)
+    {
+        LONG_SAMPLETYPE temp, vol1;
+    
+        assert(iFract < SCALE);
+        vol1 = (LONG_SAMPLETYPE)(SCALE - iFract);
+        for (int c = 0; c < numChannels; c ++)
+        {
+            temp = vol1 * src[c] + iFract * src[c + numChannels];
+            dest[0] = (SAMPLETYPE)(temp / SCALE);
+            dest ++;
+        }
+        i++;
+
+        iFract += iRate;
+
+        int iWhole = iFract / SCALE;
+        iFract -= iWhole * SCALE;
+        srcCount += iWhole;
+        src += iWhole * numChannels;
+    }
+    srcSamples = srcCount;
+
+    return i;
+}
+
+
+// Sets new target iRate. Normal iRate = 1.0, smaller values represent slower 
+// iRate, larger faster iRates.
+void InterpolateLinearInteger::setRate(double newRate)
+{
+    iRate = (int)(newRate * SCALE + 0.5);
+    TransposerBase::setRate(newRate);
+}
+
+
+//////////////////////////////////////////////////////////////////////////////
+//
+// InterpolateLinearFloat - floating point arithmetic implementation
+// 
+//////////////////////////////////////////////////////////////////////////////
+
+
+// Constructor
+InterpolateLinearFloat::InterpolateLinearFloat() : TransposerBase()
+{
+    // Notice: use local function calling syntax for sake of clarity, 
+    // to indicate the fact that C++ constructor can't call virtual functions.
+    resetRegisters();
+    setRate(1.0);
+}
+
+
+void InterpolateLinearFloat::resetRegisters()
+{
+    fract = 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
+int InterpolateLinearFloat::transposeMono(SAMPLETYPE *dest, const SAMPLETYPE *src, int &srcSamples)
+{
+    int i;
+    int srcSampleEnd = srcSamples - 1;
+    int srcCount = 0;
+
+    i = 0;
+    while (srcCount < srcSampleEnd)
+    {
+        double out;
+        assert(fract < 1.0);
+
+        out = (1.0 - fract) * src[0] + fract * src[1];
+        dest[i] = (SAMPLETYPE)out;
+        i ++;
+
+        // update position fraction
+        fract += rate;
+        // update whole positions
+        int whole = (int)fract;
+        fract -= whole;
+        src += whole;
+        srcCount += whole;
+    }
+    srcSamples = srcCount;
+    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
+int InterpolateLinearFloat::transposeStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, int &srcSamples)
+{
+    int i;
+    int srcSampleEnd = srcSamples - 1;
+    int srcCount = 0;
+
+    i = 0;
+    while (srcCount < srcSampleEnd)
+    {
+        double out0, out1;
+        assert(fract < 1.0);
+
+        out0 = (1.0 - fract) * src[0] + fract * src[2];
+        out1 = (1.0 - fract) * src[1] + fract * src[3];
+        dest[2*i]   = (SAMPLETYPE)out0;
+        dest[2*i+1] = (SAMPLETYPE)out1;
+        i ++;
+
+        // update position fraction
+        fract += rate;
+        // update whole positions
+        int whole = (int)fract;
+        fract -= whole;
+        src += 2*whole;
+        srcCount += whole;
+    }
+    srcSamples = srcCount;
+    return i;
+}
+
+
+int InterpolateLinearFloat::transposeMulti(SAMPLETYPE *dest, const SAMPLETYPE *src, int &srcSamples)
+{
+    int i;
+    int srcSampleEnd = srcSamples - 1;
+    int srcCount = 0;
+
+    i = 0;
+    while (srcCount < srcSampleEnd)
+    {
+        float temp, vol1, fract_float;
+    
+        vol1 = (float)(1.0 - fract);
+		fract_float = (float)fract;
+        for (int c = 0; c < numChannels; c ++)
+        {
+			temp = vol1 * src[c] + fract_float * src[c + numChannels];
+            *dest = (SAMPLETYPE)temp;
+            dest ++;
+        }
+        i++;
+
+        fract += rate;
+
+        int iWhole = (int)fract;
+        fract -= iWhole;
+        srcCount += iWhole;
+        src += iWhole * numChannels;
+    }
+    srcSamples = srcCount;
+
+    return i;
+}

Src/external_dependencies/openmpt-trunk/include/soundtouch/source/SoundTouch/InterpolateLinear.h

@@ -0,0 +1,98 @@
+////////////////////////////////////////////////////////////////////////////////
+/// 
+/// Linear interpolation routine.
+///
+/// Author        : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// 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 _InterpolateLinear_H_
+#define _InterpolateLinear_H_
+
+#include "RateTransposer.h"
+#include "STTypes.h"
+
+namespace soundtouch
+{
+
+/// Linear transposer class that uses integer arithmetic
+class InterpolateLinearInteger : public TransposerBase
+{
+protected:
+    int iFract;
+    int iRate;
+
+    virtual int transposeMono(SAMPLETYPE *dest, 
+                       const SAMPLETYPE *src, 
+                       int &srcSamples);
+    virtual int transposeStereo(SAMPLETYPE *dest, 
+                         const SAMPLETYPE *src, 
+                         int &srcSamples);
+    virtual int transposeMulti(SAMPLETYPE *dest, const SAMPLETYPE *src, int &srcSamples);
+public:
+    InterpolateLinearInteger();
+
+    /// Sets new target rate. Normal rate = 1.0, smaller values represent slower 
+    /// rate, larger faster rates.
+    virtual void setRate(double newRate);
+
+    virtual void resetRegisters();
+
+    int getLatency() const
+    {
+        return 0;
+    }
+};
+
+
+/// Linear transposer class that uses floating point arithmetic
+class InterpolateLinearFloat : public TransposerBase
+{
+protected:
+    double fract;
+
+    virtual int transposeMono(SAMPLETYPE *dest, 
+                       const SAMPLETYPE *src, 
+                       int &srcSamples);
+    virtual int transposeStereo(SAMPLETYPE *dest, 
+                         const SAMPLETYPE *src, 
+                         int &srcSamples);
+    virtual int transposeMulti(SAMPLETYPE *dest, const SAMPLETYPE *src, int &srcSamples);
+
+public:
+    InterpolateLinearFloat();
+
+    virtual void resetRegisters();
+
+    int getLatency() const
+    {
+        return 0;
+    }
+};
+
+}
+
+#endif

Src/external_dependencies/openmpt-trunk/include/soundtouch/source/SoundTouch/InterpolateShannon.cpp

@@ -0,0 +1,181 @@
+////////////////////////////////////////////////////////////////////////////////
+/// 
+/// Sample interpolation routine using 8-tap band-limited Shannon interpolation 
+/// with kaiser window.
+///
+/// Notice. This algorithm is remarkably much heavier than linear or cubic
+/// interpolation, and not remarkably better than cubic algorithm. Thus mostly
+/// for experimental purposes
+///
+/// Author        : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// 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 "InterpolateShannon.h"
+#include "STTypes.h"
+
+using namespace soundtouch;
+
+
+/// Kaiser window with beta = 2.0
+/// Values scaled down by 5% to avoid overflows
+static const double _kaiser8[8] = 
+{
+   0.41778693317814,
+   0.64888025049173,
+   0.83508562409944,
+   0.93887857733412,
+   0.93887857733412,
+   0.83508562409944,
+   0.64888025049173,
+   0.41778693317814
+};
+
+
+InterpolateShannon::InterpolateShannon()
+{
+    fract = 0;
+}
+
+
+void InterpolateShannon::resetRegisters()
+{
+    fract = 0;
+}
+
+
+#define PI 3.1415926536
+#define sinc(x) (sin(PI * (x)) / (PI * (x)))
+
+/// Transpose mono audio. Returns number of produced output samples, and 
+/// updates "srcSamples" to amount of consumed source samples
+int InterpolateShannon::transposeMono(SAMPLETYPE *pdest, 
+                    const SAMPLETYPE *psrc, 
+                    int &srcSamples)
+{
+    int i;
+    int srcSampleEnd = srcSamples - 8;
+    int srcCount = 0;
+
+    i = 0;
+    while (srcCount < srcSampleEnd)
+    {
+        double out;
+        assert(fract < 1.0);
+
+        out  = psrc[0] * sinc(-3.0 - fract) * _kaiser8[0];
+        out += psrc[1] * sinc(-2.0 - fract) * _kaiser8[1];
+        out += psrc[2] * sinc(-1.0 - fract) * _kaiser8[2];
+        if (fract < 1e-6)
+        {
+            out += psrc[3] * _kaiser8[3];     // sinc(0) = 1
+        }
+        else
+        {
+            out += psrc[3] * sinc(- fract) * _kaiser8[3];
+        }
+        out += psrc[4] * sinc( 1.0 - fract) * _kaiser8[4];
+        out += psrc[5] * sinc( 2.0 - fract) * _kaiser8[5];
+        out += psrc[6] * sinc( 3.0 - fract) * _kaiser8[6];
+        out += psrc[7] * sinc( 4.0 - fract) * _kaiser8[7];
+
+        pdest[i] = (SAMPLETYPE)out;
+        i ++;
+
+        // update position fraction
+        fract += rate;
+        // update whole positions
+        int whole = (int)fract;
+        fract -= whole;
+        psrc += whole;
+        srcCount += whole;
+    }
+    srcSamples = srcCount;
+    return i;
+}
+
+
+/// Transpose stereo audio. Returns number of produced output samples, and 
+/// updates "srcSamples" to amount of consumed source samples
+int InterpolateShannon::transposeStereo(SAMPLETYPE *pdest, 
+                    const SAMPLETYPE *psrc, 
+                    int &srcSamples)
+{
+    int i;
+    int srcSampleEnd = srcSamples - 8;
+    int srcCount = 0;
+
+    i = 0;
+    while (srcCount < srcSampleEnd)
+    {
+        double out0, out1, w;
+        assert(fract < 1.0);
+
+        w = sinc(-3.0 - fract) * _kaiser8[0];
+        out0 = psrc[0] * w; out1 = psrc[1] * w;
+        w = sinc(-2.0 - fract) * _kaiser8[1];
+        out0 += psrc[2] * w; out1 += psrc[3] * w;
+        w = sinc(-1.0 - fract) * _kaiser8[2];
+        out0 += psrc[4] * w; out1 += psrc[5] * w;
+        w = _kaiser8[3] * ((fract < 1e-5) ? 1.0 : sinc(- fract));   // sinc(0) = 1
+        out0 += psrc[6] * w; out1 += psrc[7] * w;
+        w = sinc( 1.0 - fract) * _kaiser8[4];
+        out0 += psrc[8] * w; out1 += psrc[9] * w;
+        w = sinc( 2.0 - fract) * _kaiser8[5];
+        out0 += psrc[10] * w; out1 += psrc[11] * w;
+        w = sinc( 3.0 - fract) * _kaiser8[6];
+        out0 += psrc[12] * w; out1 += psrc[13] * w;
+        w = sinc( 4.0 - fract) * _kaiser8[7];
+        out0 += psrc[14] * w; out1 += psrc[15] * w;
+
+        pdest[2*i]   = (SAMPLETYPE)out0;
+        pdest[2*i+1] = (SAMPLETYPE)out1;
+        i ++;
+
+        // update position fraction
+        fract += rate;
+        // update whole positions
+        int whole = (int)fract;
+        fract -= whole;
+        psrc += 2*whole;
+        srcCount += whole;
+    }
+    srcSamples = srcCount;
+    return i;
+}
+
+
+/// Transpose stereo audio. Returns number of produced output samples, and 
+/// updates "srcSamples" to amount of consumed source samples
+int InterpolateShannon::transposeMulti(SAMPLETYPE *pdest, 
+                    const SAMPLETYPE *psrc, 
+                    int &srcSamples)
+{
+    // not implemented
+    assert(false);
+    return 0;
+}

Src/external_dependencies/openmpt-trunk/include/soundtouch/source/SoundTouch/InterpolateShannon.h

@@ -0,0 +1,74 @@
+////////////////////////////////////////////////////////////////////////////////
+/// 
+/// Sample interpolation routine using 8-tap band-limited Shannon interpolation 
+/// with kaiser window.
+///
+/// Notice. This algorithm is remarkably much heavier than linear or cubic
+/// interpolation, and not remarkably better than cubic algorithm. Thus mostly
+/// for experimental purposes
+///
+/// Author        : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// 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 _InterpolateShannon_H_
+#define _InterpolateShannon_H_
+
+#include "RateTransposer.h"
+#include "STTypes.h"
+
+namespace soundtouch
+{
+
+class InterpolateShannon : public TransposerBase
+{
+protected:
+    int transposeMono(SAMPLETYPE *dest, 
+                        const SAMPLETYPE *src, 
+                        int &srcSamples);
+    int transposeStereo(SAMPLETYPE *dest, 
+                        const SAMPLETYPE *src, 
+                        int &srcSamples);
+    int transposeMulti(SAMPLETYPE *dest, 
+                        const SAMPLETYPE *src, 
+                        int &srcSamples);
+
+    double fract;
+
+public:
+    InterpolateShannon();
+
+    void resetRegisters();
+
+    int getLatency() const
+    {
+        return 3;
+    }
+};
+
+}
+
+#endif

Src/external_dependencies/openmpt-trunk/include/soundtouch/source/SoundTouch/Makefile.am

@@ -0,0 +1,74 @@
+## Process this file with automake to create Makefile.in
+##
+## This file is part of SoundTouch, an audio processing library for pitch/time adjustments
+## 
+## SoundTouch is free software; you can redistribute it and/or modify it under the
+## terms of the GNU General Public License as published by the Free Software
+## Foundation; either version 2 of the License, or (at your option) any later
+## version.
+## 
+## SoundTouch is distributed in the hope that it will be useful, but WITHOUT ANY
+## WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
+## A PARTICULAR PURPOSE.  See the GNU General Public License for more details.
+## 
+## You should have received a copy of the GNU General Public License along with
+## this program; if not, write to the Free Software Foundation, Inc., 59 Temple
+## Place - Suite 330, Boston, MA  02111-1307, USA
+
+
+include $(top_srcdir)/config/am_include.mk
+
+
+# set to something if you want other stuff to be included in the distribution tarball
+EXTRA_DIST=SoundTouch.sln SoundTouch.vcxproj
+
+noinst_HEADERS=AAFilter.h cpu_detect.h cpu_detect_x86.cpp FIRFilter.h RateTransposer.h TDStretch.h PeakFinder.h \
+    InterpolateCubic.h InterpolateLinear.h InterpolateShannon.h
+
+lib_LTLIBRARIES=libSoundTouch.la
+#
+libSoundTouch_la_SOURCES=AAFilter.cpp FIRFilter.cpp FIFOSampleBuffer.cpp    \
+    RateTransposer.cpp SoundTouch.cpp TDStretch.cpp cpu_detect_x86.cpp      \
+    BPMDetect.cpp PeakFinder.cpp InterpolateLinear.cpp InterpolateCubic.cpp \
+    InterpolateShannon.cpp
+
+# Compiler flags
+#AM_CXXFLAGS+=
+
+# Compile the files that need MMX and SSE individually.
+libSoundTouch_la_LIBADD=libSoundTouchMMX.la libSoundTouchSSE.la
+noinst_LTLIBRARIES=libSoundTouchMMX.la libSoundTouchSSE.la
+libSoundTouchMMX_la_SOURCES=mmx_optimized.cpp
+libSoundTouchSSE_la_SOURCES=sse_optimized.cpp
+
+# We enable optimizations by default.
+# If MMX is supported compile with -mmmx.
+# Do not assume -msse is also supported.
+if HAVE_MMX
+libSoundTouchMMX_la_CXXFLAGS = -mmmx $(AM_CXXFLAGS)
+else
+libSoundTouchMMX_la_CXXFLAGS = $(AM_CXXFLAGS)
+endif
+
+# We enable optimizations by default.
+# If SSE is supported compile with -msse.
+if HAVE_SSE
+libSoundTouchSSE_la_CXXFLAGS = -msse $(AM_CXXFLAGS)
+else
+libSoundTouchSSE_la_CXXFLAGS = $(AM_CXXFLAGS)
+endif
+
+# Let the user disable optimizations if he wishes to.
+if !X86_OPTIMIZATIONS
+libSoundTouchMMX_la_CXXFLAGS = $(AM_CXXFLAGS)
+libSoundTouchSSE_la_CXXFLAGS = $(AM_CXXFLAGS)
+endif
+
+# Modify the default 0.0.0 to LIB_SONAME.0.0
+libSoundTouch_la_LDFLAGS=-version-info @LIB_SONAME@
+
+# other linking flags to add
+# noinst_LTLIBRARIES = libSoundTouchOpt.la
+# libSoundTouch_la_LIBADD = libSoundTouchOpt.la
+# libSoundTouchOpt_la_SOURCES = mmx_optimized.cpp sse_optimized.cpp 
+# libSoundTouchOpt_la_CXXFLAGS = -O3 -msse -fcheck-new -I../../include

Src/external_dependencies/openmpt-trunk/include/soundtouch/source/SoundTouch/PeakFinder.cpp

@@ -0,0 +1,277 @@
+////////////////////////////////////////////////////////////////////////////////
+///
+/// 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
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// 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 real 'top' of a peak hump from neighnourhood of the given 'peakpos'.
+int PeakFinder::findTop(const float *data, int peakpos) const
+{
+    int i;
+    int start, end;
+    float refvalue;
+
+    refvalue = data[peakpos];
+
+    // seek within ±10 points
+    start = peakpos - 10;
+    if (start < minPos) start = minPos;
+    end = peakpos + 10;
+    if (end > maxPos) end = maxPos;
+
+    for (i = start; i <= end; i ++)
+    {
+        if (data[i] > refvalue)
+        {
+            peakpos = i;
+            refvalue = data[i];
+        }
+    }
+
+    // failure if max value is at edges of seek range => it's not peak, it's at slope.
+    if ((peakpos == start) || (peakpos == end)) return 0;
+
+    return peakpos;
+}
+
+
+// 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
+{
+    int lowpos;
+    int pos;
+    int climb_count;
+    float refvalue;
+    float delta;
+
+    climb_count = 0;
+    refvalue = data[peakpos];
+    lowpos = peakpos;
+
+    pos = peakpos;
+
+    while ((pos > minPos+1) && (pos < maxPos-1))
+    {
+        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 + direction < 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);
+
+    peakLevel = data[peakpos];
+
+    if (gp1 == gp2) 
+    {
+        // avoid rounding errors when all are equal
+        assert(gp1 == peakpos);
+        cutLevel = groundLevel = peakLevel;
+    } else {
+        // get average of the ground levels
+        groundLevel = 0.5f * (data[gp1] + data[gp2]);
+
+        // 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 = 1; i < 3; i ++)
+    {
+        double peaktmp, harmonic;
+        int i1,i2;
+
+        harmonic = (double)pow(2.0, i);
+        peakpos = (int)(highPeak / harmonic + 0.5f);
+        if (peakpos < minPos) break;
+        peakpos = findTop(data, peakpos);   // seek true local maximum index
+        if (peakpos == 0) continue;         // no local max here
+
+        // calculate mass-center of possible harmonic peak
+        peaktmp = getPeakCenter(data, peakpos);
+
+        // accept harmonic peak if 
+        // (a) it is found
+        // (b) is within ±4% of the expected harmonic interval
+        // (c) has at least half x-corr value of the max. peak
+
+        double diff = harmonic * peaktmp / highPeak;
+        if ((diff < 0.96) || (diff > 1.04)) continue;   // peak too afar from expected
+
+        // now compare to highest detected peak
+        i1 = (int)(highPeak + 0.5);
+        i2 = (int)(peaktmp + 0.5);
+        if (data[i2] >= 0.4*data[i1])
+        {
+            // The harmonic is at least half as high primary peak,
+            // thus use the harmonic peak instead
+            peak = peaktmp;
+        }
+    }
+
+    return peak;
+}

Src/external_dependencies/openmpt-trunk/include/soundtouch/source/SoundTouch/PeakFinder.h

@@ -0,0 +1,90 @@
+////////////////////////////////////////////////////////////////////////////////
+///
+/// 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
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// 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 belonging to the peak.
+                         int lastPos        ///< Index of last vector item belonging 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 real 'top' of a peak hump from neighnourhood of the given 'peakpos'.
+    int findTop(const float *data, int peakpos) 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_

Src/external_dependencies/openmpt-trunk/include/soundtouch/source/SoundTouch/RateTransposer.cpp

@@ -0,0 +1,315 @@
+////////////////////////////////////////////////////////////////////////////////
+/// 
+/// 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
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// 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 "RateTransposer.h"
+#include "InterpolateLinear.h"
+#include "InterpolateCubic.h"
+#include "InterpolateShannon.h"
+#include "AAFilter.h"
+
+using namespace soundtouch;
+
+// Define default interpolation algorithm here
+TransposerBase::ALGORITHM TransposerBase::algorithm = TransposerBase::CUBIC;
+
+
+// Constructor
+RateTransposer::RateTransposer() : FIFOProcessor(&outputBuffer)
+{
+    bUseAAFilter = 
+#ifndef SOUNDTOUCH_PREVENT_CLICK_AT_RATE_CROSSOVER
+        true;
+#else
+        // Disable Anti-alias filter if desirable to avoid click at rate change zero value crossover
+        false;
+#endif
+
+    // Instantiates the anti-alias filter
+    pAAFilter = new AAFilter(64);
+    pTransposer = TransposerBase::newInstance();
+    clear();
+}
+
+
+RateTransposer::~RateTransposer()
+{
+    delete pAAFilter;
+    delete pTransposer;
+}
+
+
+/// Enables/disables the anti-alias filter. Zero to disable, nonzero to enable
+void RateTransposer::enableAAFilter(bool newMode)
+{
+#ifndef SOUNDTOUCH_PREVENT_CLICK_AT_RATE_CROSSOVER
+    // Disable Anti-alias filter if desirable to avoid click at rate change zero value crossover
+    bUseAAFilter = newMode;
+    clear();
+#endif
+}
+
+
+/// 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(double newRate)
+{
+    double fCutoff;
+
+    pTransposer->setRate(newRate);
+
+    // design a new anti-alias filter
+    if (newRate > 1.0) 
+    {
+        fCutoff = 0.5 / newRate;
+    } 
+    else 
+    {
+        fCutoff = 0.5 * newRate;
+    }
+    pAAFilter->setCutoffFreq(fCutoff);
+}
+
+
+// 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 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;
+
+    if (nSamples == 0) return;
+
+    // Store samples to input buffer
+    inputBuffer.putSamples(src, nSamples);
+
+    // If anti-alias filter is turned off, simply transpose without applying
+    // the filter
+    if (bUseAAFilter == false) 
+    {
+        count = pTransposer->transpose(outputBuffer, inputBuffer);
+        return;
+    }
+
+    assert(pAAFilter);
+
+    // Transpose with anti-alias filter
+    if (pTransposer->rate < 1.0f) 
+    {
+        // If the parameter 'Rate' value is smaller than 1, first transpose
+        // the samples and then apply the anti-alias filter to remove aliasing.
+
+        // Transpose the samples, store the result to end of "midBuffer"
+        pTransposer->transpose(midBuffer, inputBuffer);
+
+        // Apply the anti-alias filter for transposed samples in midBuffer
+        pAAFilter->evaluate(outputBuffer, midBuffer);
+    } 
+    else  
+    {
+        // If the parameter 'Rate' value is larger than 1, first apply the
+        // anti-alias filter to remove high frequencies (prevent them from folding
+        // over the lover frequencies), then transpose.
+
+        // Apply the anti-alias filter for samples in inputBuffer
+        pAAFilter->evaluate(midBuffer, inputBuffer);
+
+        // Transpose the AA-filtered samples in "midBuffer"
+        pTransposer->transpose(outputBuffer, midBuffer);
+    }
+}
+
+
+// Sets the number of channels, 1 = mono, 2 = stereo
+void RateTransposer::setChannels(int nChannels)
+{
+    if (!verifyNumberOfChannels(nChannels) ||
+        (pTransposer->numChannels == nChannels)) return;
+
+    pTransposer->setChannels(nChannels);
+    inputBuffer.setChannels(nChannels);
+    midBuffer.setChannels(nChannels);
+    outputBuffer.setChannels(nChannels);
+}
+
+
+// Clears all the samples in the object
+void RateTransposer::clear()
+{
+    outputBuffer.clear();
+    midBuffer.clear();
+    inputBuffer.clear();
+    pTransposer->resetRegisters();
+
+    // prefill buffer to avoid losing first samples at beginning of stream
+    int prefill = getLatency();
+    inputBuffer.addSilent(prefill);
+}
+
+
+// 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 inputBuffer.isEmpty();
+}
+
+
+/// Return approximate initial input-output latency
+int RateTransposer::getLatency() const
+{
+    return pTransposer->getLatency() +
+        ((bUseAAFilter) ? (pAAFilter->getLength() / 2) : 0);
+}
+
+
+//////////////////////////////////////////////////////////////////////////////
+//
+// TransposerBase - Base class for interpolation
+// 
+
+// static function to set interpolation algorithm
+void TransposerBase::setAlgorithm(TransposerBase::ALGORITHM a)
+{
+    TransposerBase::algorithm = a;
+}
+
+
+// Transposes the sample rate of the given samples using linear interpolation. 
+// Returns the number of samples returned in the "dest" buffer
+int TransposerBase::transpose(FIFOSampleBuffer &dest, FIFOSampleBuffer &src)
+{
+    int numSrcSamples = src.numSamples();
+    int sizeDemand = (int)((double)numSrcSamples / rate) + 8;
+    int numOutput;
+    SAMPLETYPE *psrc = src.ptrBegin();
+    SAMPLETYPE *pdest = dest.ptrEnd(sizeDemand);
+
+#ifndef USE_MULTICH_ALWAYS
+    if (numChannels == 1)
+    {
+        numOutput = transposeMono(pdest, psrc, numSrcSamples);
+    }
+    else if (numChannels == 2) 
+    {
+        numOutput = transposeStereo(pdest, psrc, numSrcSamples);
+    } 
+    else 
+#endif // USE_MULTICH_ALWAYS
+    {
+        assert(numChannels > 0);
+        numOutput = transposeMulti(pdest, psrc, numSrcSamples);
+    }
+    dest.putSamples(numOutput);
+    src.receiveSamples(numSrcSamples);
+    return numOutput;
+}
+
+
+TransposerBase::TransposerBase()
+{
+    numChannels = 0;
+    rate = 1.0f;
+}
+
+
+TransposerBase::~TransposerBase()
+{
+}
+
+
+void TransposerBase::setChannels(int channels)
+{
+    numChannels = channels;
+    resetRegisters();
+}
+
+
+void TransposerBase::setRate(double newRate)
+{
+    rate = newRate;
+}
+
+
+// static factory function
+TransposerBase *TransposerBase::newInstance()
+{
+#ifdef SOUNDTOUCH_INTEGER_SAMPLES
+    // Notice: For integer arithmetic support only linear algorithm (due to simplest calculus)
+    return ::new InterpolateLinearInteger;
+#else
+    switch (algorithm)
+    {
+        case LINEAR:
+            return new InterpolateLinearFloat;
+
+        case CUBIC:
+            return new InterpolateCubic;
+
+        case SHANNON:
+            return new InterpolateShannon;
+
+        default:
+            assert(false);
+            return NULL;
+    }
+#endif
+}

Src/external_dependencies/openmpt-trunk/include/soundtouch/source/SoundTouch/RateTransposer.h

@@ -0,0 +1,164 @@
+////////////////////////////////////////////////////////////////////////////////
+/// 
+/// 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
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// 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
+{
+
+/// Abstract base class for transposer implementations (linear, advanced vs integer, float etc)
+class TransposerBase
+{
+public:
+        enum ALGORITHM {
+        LINEAR = 0,
+        CUBIC,
+        SHANNON
+    };
+
+protected:
+    virtual int transposeMono(SAMPLETYPE *dest, 
+                        const SAMPLETYPE *src, 
+                        int &srcSamples)  = 0;
+    virtual int transposeStereo(SAMPLETYPE *dest, 
+                        const SAMPLETYPE *src, 
+                        int &srcSamples) = 0;
+    virtual int transposeMulti(SAMPLETYPE *dest, 
+                        const SAMPLETYPE *src, 
+                        int &srcSamples) = 0;
+
+    static ALGORITHM algorithm;
+
+public:
+    double rate;
+    int numChannels;
+
+    TransposerBase();
+    virtual ~TransposerBase();
+
+    virtual int transpose(FIFOSampleBuffer &dest, FIFOSampleBuffer &src);
+    virtual void setRate(double newRate);
+    virtual void setChannels(int channels);
+    virtual int getLatency() const = 0;
+
+    virtual void resetRegisters() = 0;
+
+    // static factory function
+    static TransposerBase *newInstance();
+
+    // static function to set interpolation algorithm
+    static void setAlgorithm(ALGORITHM a);
+};
+
+
+/// A common linear samplerate transposer class.
+///
+class RateTransposer : public FIFOProcessor
+{
+protected:
+    /// Anti-alias filter object
+    AAFilter *pAAFilter;
+    TransposerBase *pTransposer;
+
+    /// Buffer for collecting samples to feed the anti-alias filter between
+    /// two batches
+    FIFOSampleBuffer inputBuffer;
+
+    /// Buffer for keeping samples between transposing & anti-alias filter
+    FIFOSampleBuffer midBuffer;
+
+    /// Output sample buffer
+    FIFOSampleBuffer outputBuffer;
+
+    bool bUseAAFilter;
+
+
+    /// 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();
+
+    /// Returns the output buffer object
+    FIFOSamplePipe *getOutput() { return &outputBuffer; };
+
+    /// 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(double 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;
+
+    /// Return approximate initial input-output latency
+    int getLatency() const;
+};
+
+}
+
+#endif

Src/external_dependencies/openmpt-trunk/include/soundtouch/source/SoundTouch/SoundTouch.cpp

@@ -0,0 +1,538 @@
+//////////////////////////////////////////////////////////////////////////////
+///
+/// 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
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// 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 <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 = new RateTransposer();
+    pTDStretch = TDStretch::newInstance();
+
+    setOutPipe(pTDStretch);
+
+    rate = tempo = 0;
+
+    virtualPitch = 
+    virtualRate = 
+    virtualTempo = 1.0;
+
+    calcEffectiveRateAndTempo();
+
+    samplesExpectedOut = 0;
+    samplesOutput = 0;
+
+    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 (!verifyNumberOfChannels(numChannels)) return;
+
+    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(double newRate)
+{
+    virtualRate = newRate;
+    calcEffectiveRateAndTempo();
+}
+
+
+// Sets new rate control value as a difference in percents compared
+// to the original rate (-50 .. +100 %)
+void SoundTouch::setRateChange(double newRate)
+{
+    virtualRate = 1.0 + 0.01 * newRate;
+    calcEffectiveRateAndTempo();
+}
+
+
+// Sets new tempo control value. Normal tempo = 1.0, smaller values
+// represent slower tempo, larger faster tempo.
+void SoundTouch::setTempo(double newTempo)
+{
+    virtualTempo = newTempo;
+    calcEffectiveRateAndTempo();
+}
+
+
+// Sets new tempo control value as a difference in percents compared
+// to the original tempo (-50 .. +100 %)
+void SoundTouch::setTempoChange(double newTempo)
+{
+    virtualTempo = 1.0 + 0.01 * newTempo;
+    calcEffectiveRateAndTempo();
+}
+
+
+// Sets new pitch control value. Original pitch = 1.0, smaller values
+// represent lower pitches, larger values higher pitch.
+void SoundTouch::setPitch(double newPitch)
+{
+    virtualPitch = newPitch;
+    calcEffectiveRateAndTempo();
+}
+
+
+// Sets pitch change in octaves compared to the original pitch
+// (-1.00 .. +1.00)
+void SoundTouch::setPitchOctaves(double newPitch)
+{
+    virtualPitch = exp(0.69314718056 * newPitch);
+    calcEffectiveRateAndTempo();
+}
+
+
+// Sets pitch change in semi-tones compared to the original pitch
+// (-12 .. +12)
+void SoundTouch::setPitchSemiTones(int newPitch)
+{
+    setPitchOctaves((double)newPitch / 12.0);
+}
+
+
+void SoundTouch::setPitchSemiTones(double newPitch)
+{
+    setPitchOctaves(newPitch / 12.0);
+}
+
+
+// Calculates 'effective' rate and tempo values from the
+// nominal control values.
+void SoundTouch::calcEffectiveRateAndTempo()
+{
+    double oldTempo = tempo;
+    double 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 SOUNDTOUCH_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
+            // deprecated : 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)
+{
+    // set sample rate, leave other tempo changer parameters as they are.
+    pTDStretch->setParameters((int)srate);
+    bSrateSet = true;
+}
+
+
+// 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) 
+    {
+        ST_THROW_RT_ERROR("SoundTouch : Sample rate not defined");
+    } 
+    else if (channels == 0) 
+    {
+        ST_THROW_RT_ERROR("SoundTouch : Number of channels not defined");
+    }
+
+    // accumulate how many samples are expected out from processing, given the current 
+    // processing setting
+    samplesExpectedOut += (double)nSamples / ((double)rate * (double)tempo);
+
+#ifndef SOUNDTOUCH_PREVENT_CLICK_AT_RATE_CROSSOVER
+    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;
+    int numStillExpected;
+    SAMPLETYPE *buff = new SAMPLETYPE[128 * channels];
+
+    // how many samples are still expected to output
+    numStillExpected = (int)((long)(samplesExpectedOut + 0.5) - samplesOutput);
+    if (numStillExpected < 0) numStillExpected = 0;
+
+    memset(buff, 0, 128 * channels * 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 
+    // 24ksamples in any case)
+    for (i = 0; (numStillExpected > (int)numSamples()) && (i < 200); i ++)
+    {
+        putSamples(buff, 128);
+    }
+
+    adjustAmountOfSamples(numStillExpected);
+
+    delete[] buff;
+
+    // Clear input buffers
+    pTDStretch->clearInput();
+    // yet leave the 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;
+
+        case SETTING_NOMINAL_INPUT_SEQUENCE :
+        {
+            int size = pTDStretch->getInputSampleReq();
+
+#ifndef SOUNDTOUCH_PREVENT_CLICK_AT_RATE_CROSSOVER
+            if (rate <= 1.0)
+            {
+                // transposing done before timestretch, which impacts latency
+                return (int)(size * rate + 0.5);
+            }
+#endif
+            return size;
+        }
+
+        case SETTING_NOMINAL_OUTPUT_SEQUENCE :
+        {
+            int size = pTDStretch->getOutputBatchSize();
+
+            if (rate > 1.0)
+            {
+                // transposing done after timestretch, which impacts latency
+                return (int)(size / rate + 0.5);
+            }
+            return size;
+        }
+
+        case SETTING_INITIAL_LATENCY:
+        {
+            double latency = pTDStretch->getLatency();
+            int latency_tr = pRateTransposer->getLatency();
+
+#ifndef SOUNDTOUCH_PREVENT_CLICK_AT_RATE_CROSSOVER
+            if (rate <= 1.0)
+            {
+                // transposing done before timestretch, which impacts latency
+                latency = (latency + latency_tr) * rate;
+            }
+            else
+#endif
+            {
+                latency += (double)latency_tr / rate;
+            }
+
+            return (int)(latency + 0.5);
+        }
+
+        default :
+            return 0;
+    }
+}
+
+
+// Clears all the samples in the object's output and internal processing
+// buffers.
+void SoundTouch::clear()
+{
+    samplesExpectedOut = 0;
+    samplesOutput = 0;
+    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;
+}
+
+
+/// Output samples from beginning of the sample buffer. Copies requested samples to 
+/// output buffer and removes them from the sample buffer. If there are less than 
+/// 'numsample' samples in the buffer, returns all that available.
+///
+/// \return Number of samples returned.
+uint SoundTouch::receiveSamples(SAMPLETYPE *output, uint maxSamples)
+{
+    uint ret = FIFOProcessor::receiveSamples(output, maxSamples);
+    samplesOutput += (long)ret;
+    return ret;
+}
+
+
+/// Adjusts book-keeping so that given number of samples are removed from beginning of the 
+/// sample buffer without copying them anywhere. 
+///
+/// Used to reduce the number of samples in the buffer when accessing the sample buffer directly
+/// with 'ptrBegin' function.
+uint SoundTouch::receiveSamples(uint maxSamples)
+{
+    uint ret = FIFOProcessor::receiveSamples(maxSamples);
+    samplesOutput += (long)ret;
+    return ret;
+}
+
+
+/// Get ratio between input and output audio durations, useful for calculating
+/// processed output duration: if you'll process a stream of N samples, then 
+/// you can expect to get out N * getInputOutputSampleRatio() samples.
+double SoundTouch::getInputOutputSampleRatio()
+{
+    return 1.0 / (tempo * rate);
+}

Src/external_dependencies/openmpt-trunk/include/soundtouch/source/SoundTouch/TDStretch.h

@@ -0,0 +1,279 @@
+////////////////////////////////////////////////////////////////////////////////
+/// 
+/// 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
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// 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;
+
+    int overlapLength;
+    int seekLength;
+    int seekWindowLength;
+    int overlapDividerBitsNorm;
+    int overlapDividerBitsPure;
+    int slopingDivider;
+    int sampleRate;
+    int sequenceMs;
+    int seekWindowMs;
+    int overlapMs;
+
+    unsigned long maxnorm;
+    float maxnormf;
+
+    double tempo;
+    double nominalSkip;
+    double skipFract;
+
+    bool bQuickSeek;
+    bool bAutoSeqSetting;
+    bool bAutoSeekSetting;
+    bool isBeginning;
+
+    SAMPLETYPE *pMidBuffer;
+    SAMPLETYPE *pMidBufferUnaligned;
+
+    FIFOSampleBuffer outputBuffer;
+    FIFOSampleBuffer inputBuffer;
+
+    void acceptNewOverlapLength(int newOverlapLength);
+
+    virtual void clearCrossCorrState();
+    void calculateOverlapLength(int overlapMs);
+
+    virtual double calcCrossCorr(const SAMPLETYPE *mixingPos, const SAMPLETYPE *compare, double &norm);
+    virtual double calcCrossCorrAccumulate(const SAMPLETYPE *mixingPos, const SAMPLETYPE *compare, double &norm);
+
+    virtual int seekBestOverlapPositionFull(const SAMPLETYPE *refPos);
+    virtual int seekBestOverlapPositionQuick(const SAMPLETYPE *refPos);
+    virtual int seekBestOverlapPosition(const SAMPLETYPE *refPos);
+
+    virtual void overlapStereo(SAMPLETYPE *output, const SAMPLETYPE *input) const;
+    virtual void overlapMono(SAMPLETYPE *output, const SAMPLETYPE *input) const;
+    virtual void overlapMulti(SAMPLETYPE *output, const SAMPLETYPE *input) const;
+
+    void clearMidBuffer();
+    void overlap(SAMPLETYPE *output, const SAMPLETYPE *input, uint ovlPos) const;
+
+    void calcSeqParameters();
+    void adaptNormalizer();
+
+    /// 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(double 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
+            );
+
+    /// return nominal input sample requirement for triggering a processing batch
+    int getInputSampleReq() const
+    {
+        return (int)(nominalSkip + 0.5);
+    }
+
+    /// return nominal output sample amount when running a processing batch
+    int getOutputBatchSize() const
+    {
+        return seekWindowLength - overlapLength;
+    }
+
+	/// return approximate initial input-output latency
+	int getLatency() const
+	{
+		return sampleReq;
+	}
+};
+
+
+// Implementation-specific class declarations:
+
+#ifdef SOUNDTOUCH_ALLOW_MMX
+    /// Class that implements MMX optimized routines for 16bit integer samples type.
+    class TDStretchMMX : public TDStretch
+    {
+    protected:
+        double calcCrossCorr(const short *mixingPos, const short *compare, double &norm);
+        double calcCrossCorrAccumulate(const short *mixingPos, const short *compare, double &norm);
+        virtual void overlapStereo(short *output, const short *input) const;
+        virtual void clearCrossCorrState();
+    };
+#endif /// SOUNDTOUCH_ALLOW_MMX
+
+
+#ifdef SOUNDTOUCH_ALLOW_SSE
+    /// Class that implements SSE optimized routines for floating point samples type.
+    class TDStretchSSE : public TDStretch
+    {
+    protected:
+        double calcCrossCorr(const float *mixingPos, const float *compare, double &norm);
+        double calcCrossCorrAccumulate(const float *mixingPos, const float *compare, double &norm);
+    };
+
+#endif /// SOUNDTOUCH_ALLOW_SSE
+
+}
+#endif  /// TDStretch_H

Src/external_dependencies/openmpt-trunk/include/soundtouch/source/SoundTouch/cpu_detect.h

@@ -0,0 +1,55 @@
+////////////////////////////////////////////////////////////////////////////////
+///
+/// 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
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// 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_

Src/external_dependencies/openmpt-trunk/include/soundtouch/source/SoundTouch/cpu_detect_x86.cpp

@@ -0,0 +1,130 @@
+////////////////////////////////////////////////////////////////////////////////
+///
+/// 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
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// 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"
+
+
+#if defined(SOUNDTOUCH_ALLOW_X86_OPTIMIZATIONS)
+
+   #if defined(__GNUC__) && defined(__i386__)
+       // gcc
+       #include "cpuid.h"
+   #elif defined(_M_IX86)
+       // windows non-gcc
+       #include <intrin.h>
+   #endif
+
+   #define bit_MMX     (1 << 23)
+   #define bit_SSE     (1 << 25)
+   #define bit_SSE2    (1 << 26)
+#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)
+{
+/// If building for a 64bit system (no Itanium) and the user wants optimizations.
+/// Return the OR of SUPPORT_{MMX,SSE,SSE2}. 11001 or 0x19.
+/// Keep the _dwDisabledISA test (2 more operations, could be eliminated).
+#if ((defined(__GNUC__) && defined(__x86_64__)) \
+    || defined(_M_X64))  \
+    && defined(SOUNDTOUCH_ALLOW_X86_OPTIMIZATIONS)
+    return 0x19 & ~_dwDisabledISA;
+
+/// If building for a 32bit system and the user wants optimizations.
+/// Keep the _dwDisabledISA test (2 more operations, could be eliminated).
+#elif ((defined(__GNUC__) && defined(__i386__)) \
+    || defined(_M_IX86))  \
+    && defined(SOUNDTOUCH_ALLOW_X86_OPTIMIZATIONS)
+
+    if (_dwDisabledISA == 0xffffffff) return 0;
+ 
+    uint res = 0;
+ 
+#if defined(__GNUC__)
+    // GCC version of cpuid. Requires GCC 4.3.0 or later for __cpuid intrinsic support.
+    uint eax, ebx, ecx, edx;  // unsigned int is the standard type. uint is defined by the compiler and not guaranteed to be portable.
+
+    // Check if no cpuid support.
+    if (!__get_cpuid (1, &eax, &ebx, &ecx, &edx)) return 0; // always disable extensions.
+
+    if (edx & bit_MMX)  res = res | SUPPORT_MMX;
+    if (edx & bit_SSE)  res = res | SUPPORT_SSE;
+    if (edx & bit_SSE2) res = res | SUPPORT_SSE2;
+
+#else
+    // Window / VS version of cpuid. Notice that Visual Studio 2005 or later required 
+    // for __cpuid intrinsic support.
+    int reg[4] = {-1};
+
+    // Check if no cpuid support.
+    __cpuid(reg,0);
+    if ((unsigned int)reg[0] == 0) return 0; // always disable extensions.
+
+    __cpuid(reg,1);
+    if ((unsigned int)reg[3] & bit_MMX)  res = res | SUPPORT_MMX;
+    if ((unsigned int)reg[3] & bit_SSE)  res = res | SUPPORT_SSE;
+    if ((unsigned int)reg[3] & bit_SSE2) res = res | SUPPORT_SSE2;
+
+#endif
+
+    return res & ~_dwDisabledISA;
+
+#else
+
+/// One of these is true:
+/// 1) We don't want optimizations.
+/// 2) Using an unsupported compiler.
+/// 3) Running on a non-x86 platform.
+    return 0;
+
+#endif
+}

Src/external_dependencies/openmpt-trunk/include/soundtouch/source/SoundTouch/mmx_optimized.cpp

@@ -0,0 +1,396 @@
+////////////////////////////////////////////////////////////////////////////////
+///
+/// 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
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// 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 SOUNDTOUCH_ALLOW_MMX
+// MMX routines available only with integer sample type
+
+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
+double TDStretchMMX::calcCrossCorr(const short *pV1, const short *pV2, double &dnorm)
+{
+    const __m64 *pVec1, *pVec2;
+    __m64 shifter;
+    __m64 accu, normaccu;
+    long corr, norm;
+    int i;
+   
+    pVec1 = (__m64*)pV1;
+    pVec2 = (__m64*)pV2;
+
+    shifter = _m_from_int(overlapDividerBitsNorm);
+    normaccu = accu = _mm_setzero_si64();
+
+    // Process 4 parallel sets of 2 * stereo samples or 4 * mono samples 
+    // during each round for improved CPU-level parallellization.
+    for (i = 0; i < channels * overlapLength / 16; 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_sra_pi32(_mm_madd_pi16(pVec1[0], pVec2[0]), shifter),
+                            _mm_sra_pi32(_mm_madd_pi16(pVec1[1], pVec2[1]), shifter));
+        temp2 = _mm_add_pi32(_mm_sra_pi32(_mm_madd_pi16(pVec1[0], pVec1[0]), shifter),
+                            _mm_sra_pi32(_mm_madd_pi16(pVec1[1], pVec1[1]), shifter));
+        accu = _mm_add_pi32(accu, temp);
+        normaccu = _mm_add_pi32(normaccu, temp2);
+
+        temp = _mm_add_pi32(_mm_sra_pi32(_mm_madd_pi16(pVec1[2], pVec2[2]), shifter),
+                            _mm_sra_pi32(_mm_madd_pi16(pVec1[3], pVec2[3]), shifter));
+        temp2 = _mm_add_pi32(_mm_sra_pi32(_mm_madd_pi16(pVec1[2], pVec1[2]), shifter),
+                            _mm_sra_pi32(_mm_madd_pi16(pVec1[3], pVec1[3]), shifter));
+        accu = _mm_add_pi32(accu, temp);
+        normaccu = _mm_add_pi32(normaccu, temp2);
+
+        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();
+
+    if (norm > (long)maxnorm)
+    {
+        // modify 'maxnorm' inside critical section to avoid multi-access conflict if in OpenMP mode
+        #pragma omp critical
+        if (norm > (long)maxnorm)
+        {
+            maxnorm = norm;
+        }
+    }
+
+    // Normalize result by dividing by sqrt(norm) - this step is easiest 
+    // done using floating point operation
+    dnorm = (double)norm;
+
+    return (double)corr / sqrt(dnorm < 1e-9 ? 1.0 : dnorm);
+    // Note: Warning about the missing EMMS instruction is harmless
+    // as it'll be called elsewhere.
+}
+
+
+/// Update cross-correlation by accumulating "norm" coefficient by previously calculated value
+double TDStretchMMX::calcCrossCorrAccumulate(const short *pV1, const short *pV2, double &dnorm)
+{
+    const __m64 *pVec1, *pVec2;
+    __m64 shifter;
+    __m64 accu;
+    long corr, lnorm;
+    int i;
+   
+    // cancel first normalizer tap from previous round
+    lnorm = 0;
+    for (i = 1; i <= channels; i ++)
+    {
+        lnorm -= (pV1[-i] * pV1[-i]) >> overlapDividerBitsNorm;
+    }
+
+    pVec1 = (__m64*)pV1;
+    pVec2 = (__m64*)pV2;
+
+    shifter = _m_from_int(overlapDividerBitsNorm);
+    accu = _mm_setzero_si64();
+
+    // Process 4 parallel sets of 2 * stereo samples or 4 * mono samples 
+    // during each round for improved CPU-level parallellization.
+    for (i = 0; i < channels * overlapLength / 16; i ++)
+    {
+        __m64 temp;
+
+        // 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_sra_pi32(_mm_madd_pi16(pVec1[0], pVec2[0]), shifter),
+                            _mm_sra_pi32(_mm_madd_pi16(pVec1[1], pVec2[1]), shifter));
+        accu = _mm_add_pi32(accu, temp);
+
+        temp = _mm_add_pi32(_mm_sra_pi32(_mm_madd_pi16(pVec1[2], pVec2[2]), shifter),
+                            _mm_sra_pi32(_mm_madd_pi16(pVec1[3], pVec2[3]), shifter));
+        accu = _mm_add_pi32(accu, temp);
+
+        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);
+
+    // Clear MMS state
+    _m_empty();
+
+    // update normalizer with last samples of this round
+    pV1 = (short *)pVec1;
+    for (int j = 1; j <= channels; j ++)
+    {
+        lnorm += (pV1[-j] * pV1[-j]) >> overlapDividerBitsNorm;
+    }
+    dnorm += (double)lnorm;
+
+    if (lnorm > (long)maxnorm)
+    {
+        maxnorm = lnorm;
+    }
+
+    // Normalize result by dividing by sqrt(norm) - this step is easiest 
+    // done using floating point operation
+    return (double)corr / sqrt((dnorm < 1e-9) ? 1.0 : dnorm);
+}
+
+
+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(overlapDividerBitsPure + 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()
+{
+    filterCoeffsAlign = NULL;
+    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 *)SOUNDTOUCH_ALIGN_POINTER_16(filterCoeffsUnalign);
+
+    // 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;
+}
+
+#else
+
+// workaround to not complain about empty module
+bool _dontcomplain_mmx_empty;
+
+#endif  // SOUNDTOUCH_ALLOW_MMX

Src/external_dependencies/openmpt-trunk/include/soundtouch/source/SoundTouch/sse_optimized.cpp

@@ -0,0 +1,365 @@
+////////////////////////////////////////////////////////////////////////////////
+///
+/// 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
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// 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 SOUNDTOUCH_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::calcCrossCorr(const float *pV1, const float *pV2, double &anorm)
+{
+    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 SOUNDTOUCH_ALLOW_NONEXACT_SIMD_OPTIMIZATION is provided
+    // for choosing if this little cheating is allowed.
+
+#ifdef ST_SIMD_AVOID_UNALIGNED
+    // Little cheating allowed, return valid correlation only for 
+    // aligned locations, meaning every second round for stereo sound.
+
+    #define _MM_LOAD    _mm_load_ps
+
+    if (((ulongptr)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. Use same routine for
+    // stereo & mono, for mono it just means twice the amount of unrolling.
+    for (i = 0; i < channels * overlapLength / 16; 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;
+    float norm = (pvNorm[0] + pvNorm[1] + pvNorm[2] + pvNorm[3]);
+    anorm = norm;
+
+    float *pvSum = (float*)&vSum;
+    return (double)(pvSum[0] + pvSum[1] + pvSum[2] + pvSum[3]) / sqrt(norm < 1e-9 ? 1.0 : 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 < channels * overlapLength / 16; 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);
+    */
+}
+
+
+
+double TDStretchSSE::calcCrossCorrAccumulate(const float *pV1, const float *pV2, double &norm)
+{
+    // call usual calcCrossCorr function because SSE does not show big benefit of 
+    // accumulating "norm" value, and also the "norm" rolling algorithm would get 
+    // complicated due to SSE-specific alignment-vs-nonexact correlation rules.
+    return calcCrossCorr(pV1, pV2, norm);
+}
+
+
+//////////////////////////////////////////////////////////////////////////////
+//
+// 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 SSE
+    // Ensure that filter coeffs array is aligned to 16-byte boundary
+    delete[] filterCoeffsUnalign;
+    filterCoeffsUnalign = new float[2 * newLength + 4];
+    filterCoeffsAlign = (float *)SOUNDTOUCH_ALIGN_POINTER_16(filterCoeffsUnalign);
+
+    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(((ulongptr)filterCoeffsAlign) % 16 == 0);
+
+    // filter is evaluated for two stereo samples with each iteration, thus use of 'j += 2'
+    #pragma omp parallel for
+    for (j = 0; j < count; j += 2)
+    {
+        const float *pSrc;
+        float *pDest;
+        const __m128 *pFil;
+        __m128 sum1, sum2;
+        uint i;
+
+        pSrc = (const float*)source + j * 2;      // source audio data
+        pDest = dest + j * 2;                     // destination 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(pDest, _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
+                    ));
+    }
+
+    // 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;
+    }
+    */
+}
+
+#endif  // SOUNDTOUCH_ALLOW_SSE

Src/external_dependencies/openmpt-trunk/include/soundtouch/source/SoundTouchDLL/SoundTouchDLL.cpp

@@ -0,0 +1,527 @@
+//////////////////////////////////////////////////////////////////////////////
+///
+/// SoundTouch DLL wrapper - wraps SoundTouch routines into a Dynamic Load 
+/// Library interface.
+///
+/// Author        : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// 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
+//
+////////////////////////////////////////////////////////////////////////////////
+
+
+#if defined(_WIN32) || defined(WIN32)
+    #include <windows.h>
+
+    // DLL main in Windows compilation
+    BOOL APIENTRY DllMain( HANDLE hModule,
+                           DWORD  ul_reason_for_call,
+                           LPVOID lpReserved
+                         )
+    {
+        switch (ul_reason_for_call)
+        {
+        case DLL_PROCESS_ATTACH:
+        case DLL_THREAD_ATTACH:
+        case DLL_THREAD_DETACH:
+        case DLL_PROCESS_DETACH:
+            break;
+        }
+        return TRUE;
+    }
+#endif
+
+#include <limits.h>
+#include <string.h>
+#include "SoundTouchDLL.h"
+#include "SoundTouch.h"
+#include "BPMDetect.h"
+
+using namespace soundtouch;
+
+#ifdef SOUNDTOUCH_INTEGER_SAMPLES
+    #error "error - compile the dll version with float samples"
+#endif // SOUNDTOUCH_INTEGER_SAMPLES
+
+//////////////
+
+typedef struct
+{
+    DWORD dwMagic;
+    SoundTouch *pst;
+} STHANDLE;
+
+typedef struct
+{
+    DWORD dwMagic;
+    BPMDetect *pbpm;
+    uint numChannels;
+} BPMHANDLE;
+
+#define STMAGIC  0x1770C001
+#define BPMMAGIC 0x1771C10a
+
+SOUNDTOUCHDLL_API HANDLE __cdecl soundtouch_createInstance()
+{
+    STHANDLE *tmp = new STHANDLE;
+
+    if (tmp)
+    {
+        tmp->dwMagic = STMAGIC;
+        tmp->pst = new SoundTouch();
+        if (tmp->pst == NULL)
+        {
+            delete tmp;
+            tmp = NULL;
+        }
+    }
+    return (HANDLE)tmp;
+}
+
+
+SOUNDTOUCHDLL_API void __cdecl soundtouch_destroyInstance(HANDLE h)
+{
+    STHANDLE *sth = (STHANDLE*)h;
+    if (sth->dwMagic != STMAGIC) return;
+
+    sth->dwMagic = 0;
+    if (sth->pst) delete sth->pst;
+    sth->pst = NULL;
+    delete sth;
+}
+
+
+/// Get SoundTouch library version string
+SOUNDTOUCHDLL_API const char *__cdecl soundtouch_getVersionString()
+{
+    return SoundTouch::getVersionString();
+}
+
+
+/// Get SoundTouch library version string - alternative function for 
+/// environments that can't properly handle character string as return value
+SOUNDTOUCHDLL_API void __cdecl soundtouch_getVersionString2(char* versionString, int bufferSize)
+{
+    strncpy(versionString, SoundTouch::getVersionString(), bufferSize - 1);
+    versionString[bufferSize - 1] = 0;
+}
+
+
+/// Get SoundTouch library version Id
+SOUNDTOUCHDLL_API uint __cdecl soundtouch_getVersionId()
+{
+    return SoundTouch::getVersionId();
+}
+
+/// Sets new rate control value. Normal rate = 1.0, smaller values
+/// represent slower rate, larger faster rates.
+SOUNDTOUCHDLL_API void __cdecl soundtouch_setRate(HANDLE h, float newRate)
+{
+    STHANDLE *sth = (STHANDLE*)h;
+    if (sth->dwMagic != STMAGIC) return;
+
+    sth->pst->setRate(newRate);
+}
+
+
+/// Sets new tempo control value. Normal tempo = 1.0, smaller values
+/// represent slower tempo, larger faster tempo.
+SOUNDTOUCHDLL_API void __cdecl soundtouch_setTempo(HANDLE h, float newTempo)
+{
+    STHANDLE *sth = (STHANDLE*)h;
+    if (sth->dwMagic != STMAGIC) return;
+
+    sth->pst->setTempo(newTempo);
+}
+
+/// Sets new rate control value as a difference in percents compared
+/// to the original rate (-50 .. +100 %)
+SOUNDTOUCHDLL_API void __cdecl soundtouch_setRateChange(HANDLE h, float newRate)
+{
+    STHANDLE *sth = (STHANDLE*)h;
+    if (sth->dwMagic != STMAGIC) return;
+
+    sth->pst->setRateChange(newRate);
+}
+
+/// Sets new tempo control value as a difference in percents compared
+/// to the original tempo (-50 .. +100 %)
+SOUNDTOUCHDLL_API void __cdecl soundtouch_setTempoChange(HANDLE h, float newTempo)
+{
+    STHANDLE *sth = (STHANDLE*)h;
+    if (sth->dwMagic != STMAGIC) return;
+
+    sth->pst->setTempoChange(newTempo);
+}
+
+/// Sets new pitch control value. Original pitch = 1.0, smaller values
+/// represent lower pitches, larger values higher pitch.
+SOUNDTOUCHDLL_API void __cdecl soundtouch_setPitch(HANDLE h, float newPitch)
+{
+    STHANDLE *sth = (STHANDLE*)h;
+    if (sth->dwMagic != STMAGIC) return;
+
+    sth->pst->setPitch(newPitch);
+}
+
+/// Sets pitch change in octaves compared to the original pitch  
+/// (-1.00 .. +1.00)
+SOUNDTOUCHDLL_API void __cdecl soundtouch_setPitchOctaves(HANDLE h, float newPitch)
+{
+    STHANDLE *sth = (STHANDLE*)h;
+    if (sth->dwMagic != STMAGIC) return;
+
+    sth->pst->setPitchOctaves(newPitch);
+}
+
+/// Sets pitch change in semi-tones compared to the original pitch
+/// (-12 .. +12)
+SOUNDTOUCHDLL_API void __cdecl soundtouch_setPitchSemiTones(HANDLE h, float newPitch)
+{
+    STHANDLE *sth = (STHANDLE*)h;
+    if (sth->dwMagic != STMAGIC) return;
+
+    sth->pst->setPitchSemiTones(newPitch);
+}
+
+
+/// Sets the number of channels, 1 = mono, 2 = stereo
+SOUNDTOUCHDLL_API void __cdecl soundtouch_setChannels(HANDLE h, uint numChannels)
+{
+    STHANDLE *sth = (STHANDLE*)h;
+    if (sth->dwMagic != STMAGIC) return;
+
+    sth->pst->setChannels(numChannels);
+}
+
+/// Sets sample rate.
+SOUNDTOUCHDLL_API void __cdecl soundtouch_setSampleRate(HANDLE h, uint srate)
+{
+    STHANDLE *sth = (STHANDLE*)h;
+    if (sth->dwMagic != STMAGIC) return;
+
+    sth->pst->setSampleRate(srate);
+}
+
+/// Flushes the last samples from the processing pipeline to the output.
+/// Clears also the internal processing buffers.
+//
+/// Note: This function is meant for extracting the last samples of a sound
+/// stream. This function may introduce additional blank samples in the end
+/// of the sound stream, and thus it's not recommended to call this function
+/// in the middle of a sound stream.
+SOUNDTOUCHDLL_API void __cdecl soundtouch_flush(HANDLE h)
+{
+    STHANDLE *sth = (STHANDLE*)h;
+    if (sth->dwMagic != STMAGIC) return;
+
+    sth->pst->flush();
+}
+
+/// Adds 'numSamples' pcs of samples from the 'samples' memory position into
+/// the input of the object. Notice that sample rate _has_to_ be set before
+/// calling this function, otherwise throws a runtime_error exception.
+SOUNDTOUCHDLL_API void __cdecl soundtouch_putSamples(HANDLE h, 
+        const SAMPLETYPE *samples,      ///< Pointer to sample buffer.
+        unsigned int numSamples         ///< Number of samples in buffer. Notice
+                                        ///< that in case of stereo-sound a single sample
+                                        ///< contains data for both channels.
+        )
+{
+    STHANDLE *sth = (STHANDLE*)h;
+    if (sth->dwMagic != STMAGIC) return;
+
+    sth->pst->putSamples(samples, numSamples);
+}
+
+/// int16 version of soundtouch_putSamples(): This accept int16 (short) sample data
+/// and internally converts it to float format before processing
+SOUNDTOUCHDLL_API void __cdecl soundtouch_putSamples_i16(HANDLE h,
+        const short *samples,       ///< Pointer to sample buffer.
+        unsigned int numSamples     ///< Number of sample frames in buffer. Notice
+                                    ///< that in case of multi-channel sound a single sample
+                                    ///< contains data for all channels.
+        )
+{
+    STHANDLE *sth = (STHANDLE*)h;
+    if (sth->dwMagic != STMAGIC) return;
+
+    uint numChannels = sth->pst->numChannels();
+
+    // iterate until all samples converted & put to SoundTouch object
+    while (numSamples > 0)
+    {
+        float convert[8192];    // allocate temporary conversion buffer from stack
+
+        // how many multichannel samples fit into 'convert' buffer:
+        uint convSamples = 8192 / numChannels;
+
+        // convert max 'nround' values at a time to guarantee that these fit in the 'convert' buffer
+        uint n = (numSamples > convSamples) ? convSamples : numSamples;
+        for (uint i = 0; i < n * numChannels; i++)
+        {
+            convert[i] = samples[i];
+        }
+        // put the converted samples into SoundTouch
+        sth->pst->putSamples(convert, n);
+
+        numSamples -= n;
+        samples += n * numChannels;
+    }
+}
+
+/// Clears all the samples in the object's output and internal processing
+/// buffers.
+SOUNDTOUCHDLL_API void __cdecl soundtouch_clear(HANDLE h)
+{
+    STHANDLE *sth = (STHANDLE*)h;
+    if (sth->dwMagic != STMAGIC) return;
+
+    sth->pst->clear();
+}
+
+/// Changes a setting controlling the processing system behaviour. See the
+/// 'SETTING_...' defines for available setting ID's.
+/// 
+/// \return 'nonzero' if the setting was successfully changed
+SOUNDTOUCHDLL_API int __cdecl soundtouch_setSetting(HANDLE h, 
+        int settingId,   ///< Setting ID number. see SETTING_... defines.
+        int value        ///< New setting value.
+        )
+{
+    STHANDLE *sth = (STHANDLE*)h;
+    if (sth->dwMagic != STMAGIC) return FALSE;
+
+    return sth->pst->setSetting(settingId, value);
+}
+
+/// Reads a setting controlling the processing system behaviour. See the
+/// 'SETTING_...' defines for available setting ID's.
+///
+/// \return the setting value.
+SOUNDTOUCHDLL_API int __cdecl soundtouch_getSetting(HANDLE h, 
+        int settingId    ///< Setting ID number, see SETTING_... defines.
+        )
+{
+    STHANDLE *sth = (STHANDLE*)h;
+    if (sth->dwMagic != STMAGIC) return -1;
+
+    return sth->pst->getSetting(settingId);
+}
+
+
+/// Returns number of samples currently unprocessed.
+SOUNDTOUCHDLL_API uint __cdecl soundtouch_numUnprocessedSamples(HANDLE h)
+{
+    STHANDLE *sth = (STHANDLE*)h;
+    if (sth->dwMagic != STMAGIC) return 0;
+
+    return sth->pst->numUnprocessedSamples();
+}
+
+
+/// Adjusts book-keeping so that given number of samples are removed from beginning of the 
+/// sample buffer without copying them anywhere. 
+///
+/// Used to reduce the number of samples in the buffer when accessing the sample buffer directly
+/// with 'ptrBegin' function.
+SOUNDTOUCHDLL_API uint __cdecl soundtouch_receiveSamples(HANDLE h, 
+        SAMPLETYPE *outBuffer,      ///< Buffer where to copy output samples.
+        unsigned int maxSamples     ///< How many samples to receive at max.
+        )
+{
+    STHANDLE *sth = (STHANDLE*)h;
+    if (sth->dwMagic != STMAGIC) return 0;
+
+    if (outBuffer)
+    {
+        return sth->pst->receiveSamples(outBuffer, maxSamples);
+    }
+    else
+    {
+        return sth->pst->receiveSamples(maxSamples);
+    }
+}
+
+
+/// int16 version of soundtouch_receiveSamples(): This converts internal float samples
+/// into int16 (short) return data type
+SOUNDTOUCHDLL_API uint __cdecl soundtouch_receiveSamples_i16(HANDLE h,
+        short *outBuffer,           ///< Buffer where to copy output samples.
+        unsigned int maxSamples     ///< How many samples to receive at max.
+        )
+{
+    STHANDLE *sth = (STHANDLE*)h;
+    if (sth->dwMagic != STMAGIC) return 0;
+    uint outTotal = 0;
+
+    if (outBuffer == NULL)
+    {
+        // only reduce sample count, not receive samples
+        return sth->pst->receiveSamples(maxSamples);
+    }
+
+    uint numChannels = sth->pst->numChannels();
+
+    // iterate until all samples converted & put to SoundTouch object
+    while (maxSamples > 0)
+    {
+        float convert[8192];    // allocate temporary conversion buffer from stack
+
+        // how many multichannel samples fit into 'convert' buffer:
+        uint convSamples = 8192 / numChannels;
+
+        // request max 'nround' values at a time to guarantee that these fit in the 'convert' buffer
+        uint n = (maxSamples > convSamples) ? convSamples : maxSamples;
+
+        uint out = sth->pst->receiveSamples(convert, n);
+
+        // convert & saturate received samples to int16
+        for (uint i = 0; i < out * numChannels; i++)
+        {
+            // first convert value to int32, then saturate to int16 min/max limits
+            int value = (int)convert[i];
+            value = (value < SHRT_MIN) ? SHRT_MIN : (value > SHRT_MAX) ? SHRT_MAX : value;
+            outBuffer[i] = (short)value;
+        }
+        outTotal += out;
+        if (out < n) break;  // didn't get as many as asked => no more samples available => break here
+
+        maxSamples -= n;
+        outBuffer += out * numChannels;
+    }
+
+    // return number of processed samples
+    return outTotal;
+}
+
+
+/// Returns number of samples currently available.
+SOUNDTOUCHDLL_API uint __cdecl soundtouch_numSamples(HANDLE h)
+{
+    STHANDLE *sth = (STHANDLE*)h;
+    if (sth->dwMagic != STMAGIC) return 0;
+
+    return sth->pst->numSamples();
+}
+
+
+/// Returns nonzero if there aren't any samples available for outputting.
+SOUNDTOUCHDLL_API int __cdecl soundtouch_isEmpty(HANDLE h)
+{
+    STHANDLE *sth = (STHANDLE*)h;
+    if (sth->dwMagic != STMAGIC) return -1;
+
+    return sth->pst->isEmpty();
+}
+
+
+SOUNDTOUCHDLL_API HANDLE __cdecl bpm_createInstance(int numChannels, int sampleRate)
+{
+    BPMHANDLE *tmp = new BPMHANDLE;
+
+    if (tmp)
+    {
+        tmp->dwMagic = BPMMAGIC;
+        tmp->pbpm = new BPMDetect(numChannels, sampleRate);
+        if (tmp->pbpm == NULL)
+        {
+            delete tmp;
+            tmp = NULL;
+        }
+    }
+    return (HANDLE)tmp;
+}
+
+
+SOUNDTOUCHDLL_API void __cdecl bpm_destroyInstance(HANDLE h)
+{
+    BPMHANDLE *sth = (BPMHANDLE*)h;
+    if (sth->dwMagic != BPMMAGIC) return;
+
+    sth->dwMagic = 0;
+    if (sth->pbpm) delete sth->pbpm;
+    sth->pbpm = NULL;
+    delete sth;
+}
+
+
+/// Feed 'numSamples' sample frames from 'samples' into the BPM detection handler
+SOUNDTOUCHDLL_API void __cdecl bpm_putSamples(HANDLE h, 
+        const float *samples,
+        unsigned int numSamples)
+{
+    BPMHANDLE *bpmh = (BPMHANDLE*)h;
+    if (bpmh->dwMagic != BPMMAGIC) return;
+
+    bpmh->pbpm->inputSamples(samples, numSamples);
+}
+
+
+/// Feed 'numSamples' sample frames from 'samples' into the BPM detection handler.
+/// 16bit int sample format version.
+SOUNDTOUCHDLL_API void __cdecl bpm_putSamples_i16(HANDLE h, 
+        const short *samples,
+        unsigned int numSamples)
+{
+    BPMHANDLE *bpmh = (BPMHANDLE*)h;
+    if (bpmh->dwMagic != BPMMAGIC) return;
+
+    uint numChannels = bpmh->numChannels;
+
+    // iterate until all samples converted & put to SoundTouch object
+    while (numSamples > 0)
+    {
+        float convert[8192];    // allocate temporary conversion buffer from stack
+
+        // how many multichannel samples fit into 'convert' buffer:
+        uint convSamples = 8192 / numChannels;
+
+        // convert max 'nround' values at a time to guarantee that these fit in the 'convert' buffer
+        uint n = (numSamples > convSamples) ? convSamples : numSamples;
+        for (uint i = 0; i < n * numChannels; i++)
+        {
+            convert[i] = samples[i];
+        }
+        // put the converted samples into SoundTouch
+        bpmh->pbpm->inputSamples(convert, n);
+
+        numSamples -= n;
+        samples += n * numChannels;
+    }
+}
+
+
+/// Analyzes the results and returns the BPM rate. Use this function to read result
+/// after whole song data has been input to the class by consecutive calls of
+/// 'inputSamples' function.
+///
+/// \return Beats-per-minute rate, or zero if detection failed.
+SOUNDTOUCHDLL_API float __cdecl bpm_getBpm(HANDLE h)
+{
+    BPMHANDLE *bpmh = (BPMHANDLE*)h;
+    if (bpmh->dwMagic != BPMMAGIC) return 0;
+
+    return bpmh->pbpm->getBpm();
+}

Src/external_dependencies/openmpt-trunk/include/soundtouch/source/SoundTouchDLL/SoundTouchDLL.h

@@ -0,0 +1,229 @@
+//////////////////////////////////////////////////////////////////////////////
+///
+/// SoundTouch DLL wrapper - wraps SoundTouch routines into a Dynamic Load 
+/// Library interface.
+///
+/// Author        : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// 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 _SoundTouchDLL_h_
+#define _SoundTouchDLL_h_
+
+#if defined(_WIN32) || defined(WIN32)
+    // Windows
+    #ifndef __cplusplus
+        #error "Expected g++"
+    #endif
+
+    #ifdef DLL_EXPORTS
+        #define SOUNDTOUCHDLL_API extern "C" __declspec(dllexport)
+    #else
+        #define SOUNDTOUCHDLL_API extern "C" __declspec(dllimport)
+    #endif
+
+#else
+    // GNU version
+
+    #if defined(DLL_EXPORTS) || defined(SoundTouchDLL_EXPORTS)
+        // GCC declaration for exporting functions
+        #define SOUNDTOUCHDLL_API extern "C" __attribute__((__visibility__("default")))
+    #else
+        // GCC doesn't require DLL imports
+        #define SOUNDTOUCHDLL_API
+    #endif
+
+    // Linux-replacements for Windows declarations:
+    #define __cdecl
+    typedef unsigned int DWORD;
+    #define FALSE    0
+    #define TRUE    1
+
+#endif
+
+typedef void * HANDLE;
+
+/// Create a new instance of SoundTouch processor.
+SOUNDTOUCHDLL_API HANDLE __cdecl soundtouch_createInstance();
+
+/// Destroys a SoundTouch processor instance.
+SOUNDTOUCHDLL_API void __cdecl soundtouch_destroyInstance(HANDLE h);
+
+/// Get SoundTouch library version string
+SOUNDTOUCHDLL_API const char *__cdecl soundtouch_getVersionString();
+
+/// Get SoundTouch library version string - alternative function for 
+/// environments that can't properly handle character string as return value
+SOUNDTOUCHDLL_API void __cdecl soundtouch_getVersionString2(char* versionString, int bufferSize);
+
+/// Get SoundTouch library version Id
+SOUNDTOUCHDLL_API unsigned int __cdecl soundtouch_getVersionId();
+
+/// Sets new rate control value. Normal rate = 1.0, smaller values
+/// represent slower rate, larger faster rates.
+SOUNDTOUCHDLL_API void __cdecl soundtouch_setRate(HANDLE h, float newRate);
+
+/// Sets new tempo control value. Normal tempo = 1.0, smaller values
+/// represent slower tempo, larger faster tempo.
+SOUNDTOUCHDLL_API void __cdecl soundtouch_setTempo(HANDLE h, float newTempo);
+
+/// Sets new rate control value as a difference in percents compared
+/// to the original rate (-50 .. +100 %);
+SOUNDTOUCHDLL_API void __cdecl soundtouch_setRateChange(HANDLE h, float newRate);
+
+/// Sets new tempo control value as a difference in percents compared
+/// to the original tempo (-50 .. +100 %);
+SOUNDTOUCHDLL_API void __cdecl soundtouch_setTempoChange(HANDLE h, float newTempo);
+
+/// Sets new pitch control value. Original pitch = 1.0, smaller values
+/// represent lower pitches, larger values higher pitch.
+SOUNDTOUCHDLL_API void __cdecl soundtouch_setPitch(HANDLE h, float newPitch);
+
+/// Sets pitch change in octaves compared to the original pitch  
+/// (-1.00 .. +1.00);
+SOUNDTOUCHDLL_API void __cdecl soundtouch_setPitchOctaves(HANDLE h, float newPitch);
+
+/// Sets pitch change in semi-tones compared to the original pitch
+/// (-12 .. +12);
+SOUNDTOUCHDLL_API void __cdecl soundtouch_setPitchSemiTones(HANDLE h, float newPitch);
+
+
+/// Sets the number of channels, 1 = mono, 2 = stereo, n = multichannel
+SOUNDTOUCHDLL_API void __cdecl soundtouch_setChannels(HANDLE h, unsigned int numChannels);
+
+/// Sets sample rate.
+SOUNDTOUCHDLL_API void __cdecl soundtouch_setSampleRate(HANDLE h, unsigned int srate);
+
+/// Flushes the last samples from the processing pipeline to the output.
+/// Clears also the internal processing buffers.
+//
+/// Note: This function is meant for extracting the last samples of a sound
+/// stream. This function may introduce additional blank samples in the end
+/// of the sound stream, and thus it's not recommended to call this function
+/// in the middle of a sound stream.
+SOUNDTOUCHDLL_API void __cdecl soundtouch_flush(HANDLE h);
+
+/// Adds 'numSamples' pcs of samples from the 'samples' memory position into
+/// the input of the object. Notice that sample rate _has_to_ be set before
+/// calling this function, otherwise throws a runtime_error exception.
+SOUNDTOUCHDLL_API void __cdecl soundtouch_putSamples(HANDLE h, 
+        const float *samples,       ///< Pointer to sample buffer.
+        unsigned int numSamples     ///< Number of sample frames in buffer. Notice
+                                    ///< that in case of multi-channel sound a single 
+                                    ///< sample frame contains data for all channels.
+);
+
+/// int16 version of soundtouch_putSamples(): This accept int16 (short) sample data
+/// and internally converts it to float format before processing
+SOUNDTOUCHDLL_API void __cdecl soundtouch_putSamples_i16(HANDLE h,
+        const short *samples,       ///< Pointer to sample buffer.
+        unsigned int numSamples     ///< Number of sample frames in buffer. Notice
+                                    ///< that in case of multi-channel sound a single 
+                                    ///< sample frame contains data for all channels.
+);
+
+
+/// Clears all the samples in the object's output and internal processing
+/// buffers.
+SOUNDTOUCHDLL_API void __cdecl soundtouch_clear(HANDLE h);
+
+/// Changes a setting controlling the processing system behaviour. See the
+/// 'SETTING_...' defines for available setting ID's.
+/// 
+/// \return 'nonzero' if the setting was successfully changed, otherwise zero
+SOUNDTOUCHDLL_API int __cdecl soundtouch_setSetting(HANDLE h, 
+        int settingId,   ///< Setting ID number. see SETTING_... defines.
+        int value        ///< New setting value.
+);
+
+/// Reads a setting controlling the processing system behaviour. See the
+/// 'SETTING_...' defines for available setting ID's.
+///
+/// \return the setting value.
+SOUNDTOUCHDLL_API int __cdecl soundtouch_getSetting(HANDLE h, 
+        int settingId    ///< Setting ID number, see SETTING_... defines.
+);
+
+
+/// Returns number of samples currently unprocessed.
+SOUNDTOUCHDLL_API unsigned int __cdecl soundtouch_numUnprocessedSamples(HANDLE h);
+
+/// Adjusts book-keeping so that given number of samples are removed from beginning of the 
+/// sample buffer without copying them anywhere. 
+///
+/// Used to reduce the number of samples in the buffer when accessing the sample buffer directly
+/// with 'ptrBegin' function.
+SOUNDTOUCHDLL_API unsigned int __cdecl soundtouch_receiveSamples(HANDLE h, 
+        float *outBuffer,           ///< Buffer where to copy output samples.
+        unsigned int maxSamples     ///< How many samples to receive at max.
+);
+
+
+/// int16 version of soundtouch_receiveSamples(): This converts internal float samples
+/// into int16 (short) return data type
+SOUNDTOUCHDLL_API unsigned int __cdecl soundtouch_receiveSamples_i16(HANDLE h,
+        short *outBuffer,           ///< Buffer where to copy output samples.
+        unsigned int maxSamples     ///< How many samples to receive at max.
+);
+
+/// Returns number of samples currently available.
+SOUNDTOUCHDLL_API unsigned int __cdecl soundtouch_numSamples(HANDLE h);
+
+/// Returns nonzero if there aren't any samples available for outputting.
+SOUNDTOUCHDLL_API int __cdecl soundtouch_isEmpty(HANDLE h);
+
+/// Create a new instance of BPM detector
+SOUNDTOUCHDLL_API HANDLE __cdecl bpm_createInstance(int numChannels, int sampleRate);
+
+/// Destroys a BPM detector instance.
+SOUNDTOUCHDLL_API void __cdecl bpm_destroyInstance(HANDLE h);
+
+/// Feed 'numSamples' sample frames from 'samples' into the BPM detector.
+SOUNDTOUCHDLL_API void __cdecl bpm_putSamples(HANDLE h, 
+        const float *samples,           ///< Pointer to sample buffer.
+        unsigned int numSamples         ///< Number of samples in buffer. Notice
+                                        ///< that in case of stereo-sound a single sample
+                                        ///< contains data for both channels.
+        );
+
+/// Feed 'numSamples' sample frames from 'samples' into the BPM detector.
+/// 16bit int sample format version.
+SOUNDTOUCHDLL_API void __cdecl bpm_putSamples_i16(HANDLE h, 
+        const short *samples,           ///< Pointer to sample buffer.
+        unsigned int numSamples         ///< Number of samples in buffer. Notice
+                                        ///< that in case of stereo-sound a single sample
+                                        ///< contains data for both channels.
+        );
+
+/// Analyzes the results and returns the BPM rate. Use this function to read result
+/// after whole song data has been input to the class by consecutive calls of
+/// 'inputSamples' function.
+///
+/// \return Beats-per-minute rate, or zero if detection failed.
+SOUNDTOUCHDLL_API float __cdecl bpm_getBpm(HANDLE h);
+
+#endif  // _SoundTouchDLL_h_
+

Src/external_dependencies/openmpt-trunk/include/soundtouch/source/SoundTouchDLL/SoundTouchDLL.rc

@@ -0,0 +1,100 @@
+// Microsoft Visual C++ generated resource script.
+//
+#include "resource.h"
+
+#define APSTUDIO_READONLY_SYMBOLS
+/////////////////////////////////////////////////////////////////////////////
+//
+// Generated from the TEXTINCLUDE 2 resource.
+//
+#include "afxres.h"
+
+/////////////////////////////////////////////////////////////////////////////
+#undef APSTUDIO_READONLY_SYMBOLS
+
+/////////////////////////////////////////////////////////////////////////////
+// English (United States) resources
+
+#if !defined(AFX_RESOURCE_DLL) || defined(AFX_TARG_ENU)
+LANGUAGE LANG_ENGLISH, SUBLANG_ENGLISH_US
+#pragma code_page(1252)
+
+#ifdef APSTUDIO_INVOKED
+/////////////////////////////////////////////////////////////////////////////
+//
+// TEXTINCLUDE
+//
+
+1 TEXTINCLUDE 
+BEGIN
+    "resource.h\0"
+END
+
+2 TEXTINCLUDE 
+BEGIN
+    "#include ""afxres.h""\r\n"
+    "\0"
+END
+
+3 TEXTINCLUDE 
+BEGIN
+    "\r\n"
+    "\0"
+END
+
+#endif    // APSTUDIO_INVOKED
+
+
+/////////////////////////////////////////////////////////////////////////////
+//
+// Version
+//
+
+VS_VERSION_INFO VERSIONINFO
+ FILEVERSION 2,2,0,0
+ PRODUCTVERSION 2,2,0,0
+ FILEFLAGSMASK 0x17L
+#ifdef _DEBUG
+ FILEFLAGS 0x1L
+#else
+ FILEFLAGS 0x0L
+#endif
+ FILEOS 0x4L
+ FILETYPE 0x2L
+ FILESUBTYPE 0x0L
+BEGIN
+    BLOCK "StringFileInfo"
+    BEGIN
+        BLOCK "000004b0"
+        BEGIN
+            VALUE "Comments", "SoundTouch Library licensed for 3rd party applications subject to LGPL license v2.1. Visit http://www.surina.net/soundtouch for more information about the SoundTouch library."
+            VALUE "FileDescription", "SoundTouch Dynamic Link Library"
+            VALUE "FileVersion", "2.3.1.0"
+            VALUE "InternalName", "SoundTouch"
+            VALUE "LegalCopyright", "Copyright (C) Olli Parviainen 2021"
+            VALUE "OriginalFilename", "SoundTouch.dll"
+            VALUE "ProductName", " SoundTouch Dynamic Link Library"
+            VALUE "ProductVersion", "2.3.1.0"
+        END
+    END
+    BLOCK "VarFileInfo"
+    BEGIN
+        VALUE "Translation", 0x0, 1200
+    END
+END
+
+#endif    // English (United States) resources
+/////////////////////////////////////////////////////////////////////////////
+
+
+
+#ifndef APSTUDIO_INVOKED
+/////////////////////////////////////////////////////////////////////////////
+//
+// Generated from the TEXTINCLUDE 3 resource.
+//
+
+
+/////////////////////////////////////////////////////////////////////////////
+#endif    // not APSTUDIO_INVOKED
+

Src/external_dependencies/openmpt-trunk/include/soundtouch/source/SoundTouchDLL/resource.h

@@ -0,0 +1,15 @@
+//{{NO_DEPENDENCIES}}
+// Microsoft Visual C++ generated include file.
+// Used by SoundTouchDLL.rc
+//
+
+// Next default values for new objects
+// 
+#ifdef APSTUDIO_INVOKED
+#ifndef APSTUDIO_READONLY_SYMBOLS
+#define _APS_NEXT_RESOURCE_VALUE        101
+#define _APS_NEXT_COMMAND_VALUE         40001
+#define _APS_NEXT_CONTROL_VALUE         1000
+#define _APS_NEXT_SYMED_VALUE           101
+#endif
+#endif