Updated SoundTouch library to 1.8.1 [r198]

Externals/soundtouch/AAFilter.cpp

@@ -12,10 +12,10 @@
 ///
 ////////////////////////////////////////////////////////////////////////////////
 //
-// Last changed  : $Date: 2009-01-11 11:34:24 +0000 (Sun, 11 Jan 2009) $
+// Last changed  : $Date: 2014-01-06 08:40:22 +1100 (Mon, 06 Jan 2014) $
 // File revision : $Revision: 4 $
 //
-// $Id: AAFilter.cpp 45 2009-01-11 11:34:24Z oparviai $
+// $Id: AAFilter.cpp 177 2014-01-05 21:40:22Z oparviai $
 //
 ////////////////////////////////////////////////////////////////////////////////
 //
@@ -52,6 +52,30 @@ using namespace soundtouch;
 #define PI        3.141592655357989
 #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'
@@ -99,7 +123,7 @@ void AAFilter::calculateCoeffs()
 {
     uint i;
     double cntTemp, temp, tempCoeff,h, w;
-    double fc2, wc;
+    double wc;
     double scaleCoeff, sum;
     double *work;
     SAMPLETYPE *coeffs;
@@ -112,8 +136,7 @@ void AAFilter::calculateCoeffs()
     work = new double[length];
     coeffs = new SAMPLETYPE[length];
 
-    fc2 = 2.0 * cutoffFreq; 
+    wc = 2.0 * PI * cutoffFreq;
-    wc = PI * fc2;
     tempCoeff = TWOPI / (double)length;
 
     sum = 0;
@@ -124,7 +147,7 @@ void AAFilter::calculateCoeffs()
         temp = cntTemp * wc;
         if (temp != 0) 
         {
-            h = fc2 * sin(temp) / temp;                     // sinc function
+            h = sin(temp) / temp;                     // sinc function
         } 
         else 
         {
@@ -153,17 +176,21 @@ void AAFilter::calculateCoeffs()
 
     for (i = 0; i < length; i ++) 
     {
-        // scale & round to nearest integer
         temp = work[i] * scaleCoeff;
+//#if SOUNDTOUCH_INTEGER_SAMPLES
+        // scale & round to nearest integer
         temp += (temp >= 0) ? 0.5 : -0.5;
         // ensure no overfloods
         assert(temp >= -32768 && temp <= 32767);
+//#endif
         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;
 }
@@ -178,6 +205,31 @@ uint AAFilter::evaluate(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples
 }
 
 
+/// 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();

Externals/soundtouch/AAFilter.h

@@ -13,10 +13,10 @@
 ///
 ////////////////////////////////////////////////////////////////////////////////
 //
-// Last changed  : $Date: 2008-02-10 16:26:55 +0000 (Sun, 10 Feb 2008) $
+// Last changed  : $Date: 2014-01-08 06:41:23 +1100 (Wed, 08 Jan 2014) $
 // File revision : $Revision: 4 $
 //
-// $Id: AAFilter.h 11 2008-02-10 16:26:55Z oparviai $
+// $Id: AAFilter.h 187 2014-01-07 19:41:23Z oparviai $
 //
 ////////////////////////////////////////////////////////////////////////////////
 //
@@ -45,6 +45,7 @@
 #define AAFilter_H
 
 #include "STTypes.h"
+#include "FIFOSampleBuffer.h"
 
 namespace soundtouch
 {
@@ -84,6 +85,14 @@ public:
                   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;
+
 };
 
 }

Externals/soundtouch/BPMDetect.cpp

@@ -26,7 +26,7 @@
 ///
 ////////////////////////////////////////////////////////////////////////////////
 //
-// Last changed  : $Date: 2012-08-30 19:45:25 +0000 (Thu, 30 Aug 2012) $
+// Last changed  : $Date: 2012-08-31 05:45:25 +1000 (Fri, 31 Aug 2012) $
 // File revision : $Revision: 4 $
 //
 // $Id: BPMDetect.cpp 149 2012-08-30 19:45:25Z oparviai $

Externals/soundtouch/CMakeLists.txt

@@ -4,6 +4,9 @@ set(SRCS
 	cpu_detect_x86.cpp
 	FIFOSampleBuffer.cpp
 	FIRFilter.cpp
+	InterpolateCubic.cpp
+	InterpolateLinear.cpp
+	InterpolateShannon.cpp
 	mmx_optimized.cpp
 	PeakFinder.cpp
 	RateTransposer.cpp

Externals/soundtouch/FIFOSampleBuffer.cpp

@@ -15,7 +15,7 @@
 ///
 ////////////////////////////////////////////////////////////////////////////////
 //
-// Last changed  : $Date: 2012-11-08 18:53:01 +0000 (Thu, 08 Nov 2012) $
+// Last changed  : $Date: 2012-11-09 05:53:01 +1100 (Fri, 09 Nov 2012) $
 // File revision : $Revision: 4 $
 //
 // $Id: FIFOSampleBuffer.cpp 160 2012-11-08 18:53:01Z oparviai $
@@ -86,6 +86,10 @@ void FIFOSampleBuffer::setChannels(int numChannels)
     samplesInBuffer = usedBytes / channels;
 }
 
+int FIFOSampleBuffer::getChannels()
+{
+	return channels;
+}
 
 // if output location pointer 'bufferPos' isn't zero, 'rewinds' the buffer and
 // zeroes this pointer by copying samples from the 'bufferPos' pointer 

Externals/soundtouch/FIFOSampleBuffer.h

@@ -161,6 +161,7 @@ public:
 
     /// Sets number of channels, 1 = mono, 2 = stereo.
     void setChannels(int numChannels);
+	int getChannels();
 
     /// Returns nonzero if there aren't any samples available for outputting.
     virtual int isEmpty() const;

Externals/soundtouch/FIRFilter.cpp

@@ -11,7 +11,7 @@
 ///
 ////////////////////////////////////////////////////////////////////////////////
 //
-// Last changed  : $Date: 2013-06-12 15:24:44 +0000 (Wed, 12 Jun 2013) $
+// Last changed  : $Date: 2013-06-13 01:24:44 +1000 (Thu, 13 Jun 2013) $
 // File revision : $Revision: 4 $
 //
 // $Id: FIRFilter.cpp 171 2013-06-12 15:24:44Z oparviai $
@@ -217,7 +217,6 @@ uint FIRFilter::evaluateFilterMulti(SAMPLETYPE *dest, const SAMPLETYPE *src, uin
             sum[c] = 0;
         }
     }
-    free(sum);
     return numSamples - length;
 }
 

Externals/soundtouch/FIRFilter.h

@@ -11,7 +11,7 @@
 ///
 ////////////////////////////////////////////////////////////////////////////////
 //
-// Last changed  : $Date: 2013-06-12 15:24:44 +0000 (Wed, 12 Jun 2013) $
+// Last changed  : $Date: 2013-06-13 01:24:44 +1000 (Thu, 13 Jun 2013) $
 // File revision : $Revision: 4 $
 //
 // $Id: FIRFilter.h 171 2013-06-12 15:24:44Z oparviai $

Externals/soundtouch/InterpolateCubic.cpp

@@ -0,0 +1,200 @@
+////////////////////////////////////////////////////////////////////////////////
+/// 
+/// Cubic interpolation routine.
+///
+/// Author        : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// $Id: InterpolateCubic.cpp 179 2014-01-06 18:41:42Z oparviai $
+//
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+//  SoundTouch audio processing library
+//  Copyright (c) Olli Parviainen
+//
+//  This library is free software; you can redistribute it and/or
+//  modify it under the terms of the GNU Lesser General Public
+//  License as published by the Free Software Foundation; either
+//  version 2.1 of the License, or (at your option) any later version.
+//
+//  This library is distributed in the hope that it will be useful,
+//  but WITHOUT ANY WARRANTY; without even the implied warranty of
+//  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+//  Lesser General Public License for more details.
+//
+//  You should have received a copy of the GNU Lesser General Public
+//  License along with this library; if not, write to the Free Software
+//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#include <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;
+}

Externals/soundtouch/InterpolateCubic.h

@@ -0,0 +1,67 @@
+////////////////////////////////////////////////////////////////////////////////
+/// 
+/// Cubic interpolation routine.
+///
+/// Author        : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// $Id: InterpolateCubic.h 179 2014-01-06 18:41:42Z oparviai $
+//
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+//  SoundTouch audio processing library
+//  Copyright (c) Olli Parviainen
+//
+//  This library is free software; you can redistribute it and/or
+//  modify it under the terms of the GNU Lesser General Public
+//  License as published by the Free Software Foundation; either
+//  version 2.1 of the License, or (at your option) any later version.
+//
+//  This library is distributed in the hope that it will be useful,
+//  but WITHOUT ANY WARRANTY; without even the implied warranty of
+//  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+//  Lesser General Public License for more details.
+//
+//  You should have received a copy of the GNU Lesser General Public
+//  License along with this library; if not, write to the Free Software
+//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#ifndef _InterpolateCubic_H_
+#define _InterpolateCubic_H_
+
+#include "RateTransposer.h"
+#include "STTypes.h"
+
+namespace soundtouch
+{
+
+class InterpolateCubic : public TransposerBase
+{
+protected:
+    virtual void resetRegisters();
+    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);
+
+    float fract;
+
+public:
+    InterpolateCubic();
+};
+
+}
+
+#endif

Externals/soundtouch/InterpolateLinear.cpp

@@ -0,0 +1,299 @@
+////////////////////////////////////////////////////////////////////////////////
+/// 
+/// Linear interpolation algorithm.
+///
+/// Author        : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// $Id: InterpolateLinear.cpp 180 2014-01-06 19:16:02Z oparviai $
+//
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+//  SoundTouch audio processing library
+//  Copyright (c) Olli Parviainen
+//
+//  This library is free software; you can redistribute it and/or
+//  modify it under the terms of the GNU Lesser General Public
+//  License as published by the Free Software Foundation; either
+//  version 2.1 of the License, or (at your option) any later version.
+//
+//  This library is distributed in the hope that it will be useful,
+//  but WITHOUT ANY WARRANTY; without even the implied warranty of
+//  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+//  Lesser General Public License for more details.
+//
+//  You should have received a copy of the GNU Lesser General Public
+//  License along with this library; if not, write to the Free Software
+//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#include <assert.h>
+#include <stdlib.h>
+#include "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 = (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(float newRate)
+{
+    iRate = (int)(newRate * SCALE + 0.5f);
+    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(1.0f);
+}
+
+
+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;
+    
+        vol1 = (1.0f- fract);
+        for (int c = 0; c < numChannels; c ++)
+        {
+            temp = vol1 * src[c] + fract * 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;
+}

Externals/soundtouch/InterpolateLinear.h

@@ -0,0 +1,92 @@
+////////////////////////////////////////////////////////////////////////////////
+/// 
+/// Linear interpolation routine.
+///
+/// Author        : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// $Id: InterpolateLinear.h 179 2014-01-06 18:41:42Z oparviai $
+//
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+//  SoundTouch audio processing library
+//  Copyright (c) Olli Parviainen
+//
+//  This library is free software; you can redistribute it and/or
+//  modify it under the terms of the GNU Lesser General Public
+//  License as published by the Free Software Foundation; either
+//  version 2.1 of the License, or (at your option) any later version.
+//
+//  This library is distributed in the hope that it will be useful,
+//  but WITHOUT ANY WARRANTY; without even the implied warranty of
+//  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+//  Lesser General Public License for more details.
+//
+//  You should have received a copy of the GNU Lesser General Public
+//  License along with this library; if not, write to the Free Software
+//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#ifndef _InterpolateLinear_H_
+#define _InterpolateLinear_H_
+
+#include "RateTransposer.h"
+#include "STTypes.h"
+
+namespace soundtouch
+{
+
+/// Linear transposer class that uses integer arithmetics
+class InterpolateLinearInteger : public TransposerBase
+{
+protected:
+    int iFract;
+    int iRate;
+
+    virtual void resetRegisters();
+
+    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(float newRate);
+};
+
+
+/// Linear transposer class that uses floating point arithmetics
+class InterpolateLinearFloat : public TransposerBase
+{
+protected:
+    float fract;
+
+    virtual void resetRegisters();
+
+    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();
+};
+
+}
+
+#endif

Externals/soundtouch/InterpolateShannon.cpp

@@ -0,0 +1,185 @@
+////////////////////////////////////////////////////////////////////////////////
+/// 
+/// 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
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// $Id: InterpolateShannon.cpp 195 2014-04-06 15:57:21Z oparviai $
+//
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+//  SoundTouch audio processing library
+//  Copyright (c) Olli Parviainen
+//
+//  This library is free software; you can redistribute it and/or
+//  modify it under the terms of the GNU Lesser General Public
+//  License as published by the Free Software Foundation; either
+//  version 2.1 of the License, or (at your option) any later version.
+//
+//  This library is distributed in the hope that it will be useful,
+//  but WITHOUT ANY WARRANTY; without even the implied warranty of
+//  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+//  Lesser General Public License for more details.
+//
+//  You should have received a copy of the GNU Lesser General Public
+//  License along with this library; if not, write to the Free Software
+//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#include <math.h>
+#include "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;
+}

Externals/soundtouch/InterpolateShannon.h

@@ -0,0 +1,72 @@
+////////////////////////////////////////////////////////////////////////////////
+/// 
+/// 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
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// $Id: InterpolateShannon.h 179 2014-01-06 18:41:42Z oparviai $
+//
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+//  SoundTouch audio processing library
+//  Copyright (c) Olli Parviainen
+//
+//  This library is free software; you can redistribute it and/or
+//  modify it under the terms of the GNU Lesser General Public
+//  License as published by the Free Software Foundation; either
+//  version 2.1 of the License, or (at your option) any later version.
+//
+//  This library is distributed in the hope that it will be useful,
+//  but WITHOUT ANY WARRANTY; without even the implied warranty of
+//  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+//  Lesser General Public License for more details.
+//
+//  You should have received a copy of the GNU Lesser General Public
+//  License along with this library; if not, write to the Free Software
+//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#ifndef _InterpolateShannon_H_
+#define _InterpolateShannon_H_
+
+#include "RateTransposer.h"
+#include "STTypes.h"
+
+namespace soundtouch
+{
+
+class InterpolateShannon : public TransposerBase
+{
+protected:
+    void resetRegisters();
+    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);
+
+    float fract;
+
+public:
+    InterpolateShannon();
+};
+
+}
+
+#endif

Externals/soundtouch/PeakFinder.cpp

@@ -11,7 +11,7 @@
 ///
 ////////////////////////////////////////////////////////////////////////////////
 //
-// Last changed  : $Date: 2012-12-28 19:52:47 +0000 (Fri, 28 Dec 2012) $
+// Last changed  : $Date: 2012-12-29 06:52:47 +1100 (Sat, 29 Dec 2012) $
 // File revision : $Revision: 4 $
 //
 // $Id: PeakFinder.cpp 164 2012-12-28 19:52:47Z oparviai $

Externals/soundtouch/PeakFinder.h

@@ -9,7 +9,7 @@
 ///
 ////////////////////////////////////////////////////////////////////////////////
 //
-// Last changed  : $Date: 2011-12-30 20:33:46 +0000 (Fri, 30 Dec 2011) $
+// Last changed  : $Date: 2011-12-31 07:33:46 +1100 (Sat, 31 Dec 2011) $
 // File revision : $Revision: 4 $
 //
 // $Id: PeakFinder.h 132 2011-12-30 20:33:46Z oparviai $

Externals/soundtouch/RateTransposer.cpp

@@ -10,10 +10,10 @@
 ///
 ////////////////////////////////////////////////////////////////////////////////
 //
-// Last changed  : $Date: 2013-06-14 17:34:33 +0000 (Fri, 14 Jun 2013) $
+// Last changed  : $Date: 2014-04-07 01:57:21 +1000 (Mon, 07 Apr 2014) $
 // File revision : $Revision: 4 $
 //
-// $Id: RateTransposer.cpp 172 2013-06-14 17:34:33Z oparviai $
+// $Id: RateTransposer.cpp 195 2014-04-06 15:57:21Z oparviai $
 //
 ////////////////////////////////////////////////////////////////////////////////
 //
@@ -43,95 +43,25 @@
 #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
-/// A linear samplerate transposer class that uses integer arithmetics.
+TransposerBase::ALGORITHM TransposerBase::algorithm = TransposerBase::CUBIC;
-/// for the transposing.
-class RateTransposerInteger : public RateTransposer
-{
-protected:
-    int iSlopeCount;
-    int iRate;
-    SAMPLETYPE *sPrevSample;
-
-    virtual void resetRegisters();
-
-    virtual int transposeStereo(SAMPLETYPE *dest, 
-                         const SAMPLETYPE *src, 
-                         uint numSamples);
-    virtual int transposeMono(SAMPLETYPE *dest, 
-                       const SAMPLETYPE *src, 
-                       uint numSamples);
-    virtual int transposeMulti(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples);
-public:
-    RateTransposerInteger();
-    virtual ~RateTransposerInteger();
-
-    /// Sets new target rate. Normal rate = 1.0, smaller values represent slower 
-    /// rate, larger faster rates.
-    virtual void setRate(float newRate);
-
-};
-
-
-/// A linear samplerate transposer class that uses floating point arithmetics
-/// for the transposing.
-class RateTransposerFloat : public RateTransposer
-{
-protected:
-    float fSlopeCount;
-    SAMPLETYPE *sPrevSample;
-
-    virtual void resetRegisters();
-
-    virtual int transposeStereo(SAMPLETYPE *dest, 
-                         const SAMPLETYPE *src, 
-                         uint numSamples);
-    virtual int transposeMono(SAMPLETYPE *dest, 
-                       const SAMPLETYPE *src, 
-                       uint numSamples);
-    virtual int transposeMulti(SAMPLETYPE *dest, const SAMPLETYPE *src, uint nSamples);
-
-public:
-    RateTransposerFloat();
-    virtual ~RateTransposerFloat();
-};
-
-
-
-
-// Operator 'new' is overloaded so that it automatically creates a suitable instance 
-// depending on if we've a MMX/SSE/etc-capable CPU available or not.
-void * RateTransposer::operator new(size_t s)
-{
-    ST_THROW_RT_ERROR("Error in RateTransoser::new: don't use \"new TDStretch\" directly, use \"newInstance\" to create a new instance instead!");
-    return newInstance();
-}
-
-
-RateTransposer *RateTransposer::newInstance()
-{
-#ifdef SOUNDTOUCH_INTEGER_SAMPLES
-    return ::new RateTransposerInteger;
-#else
-    return ::new RateTransposerFloat;
-#endif
-}
 
 
 // Constructor
 RateTransposer::RateTransposer() : FIFOProcessor(&outputBuffer)
 {
-    numChannels = 2;
+    bUseAAFilter = true;
-    bUseAAFilter = TRUE;
-    fRate = 0;
 
-    // Instantiates the anti-alias filter with default tap length
+    // Instantiates the anti-alias filter
-    // of 32
+    pAAFilter = new AAFilter(64);
-    pAAFilter = new AAFilter(32);
+    pTransposer = TransposerBase::newInstance();
 }
 
 
@@ -139,19 +69,20 @@ RateTransposer::RateTransposer() : FIFOProcessor(&outputBuffer)
 RateTransposer::~RateTransposer()
 {
     delete pAAFilter;
+    delete pTransposer;
 }
 
 
 
 /// Enables/disables the anti-alias filter. Zero to disable, nonzero to enable
-void RateTransposer::enableAAFilter(BOOL newMode)
+void RateTransposer::enableAAFilter(bool newMode)
 {
     bUseAAFilter = newMode;
 }
 
 
 /// Returns nonzero if anti-alias filter is enabled.
-BOOL RateTransposer::isAAFilterEnabled() const
+bool RateTransposer::isAAFilterEnabled() const
 {
     return bUseAAFilter;
 }
@@ -170,7 +101,7 @@ void RateTransposer::setRate(float newRate)
 {
     double fCutoff;
 
-    fRate = newRate;
+    pTransposer->setRate(newRate);
 
     // design a new anti-alias filter
     if (newRate > 1.0f) 
@@ -185,22 +116,6 @@ void RateTransposer::setRate(float newRate)
 }
 
 
-// Outputs as many samples of the 'outputBuffer' as possible, and if there's
-// any room left, outputs also as many of the incoming samples as possible.
-// The goal is to drive the outputBuffer empty.
-//
-// It's allowed for 'output' and 'input' parameters to point to the same
-// memory position.
-/*
-void RateTransposer::flushStoreBuffer()
-{
-    if (storeBuffer.isEmpty()) return;
-
-    outputBuffer.moveSamples(storeBuffer);
-}
-*/
-
-
 // Adds 'nSamples' pcs of samples from the 'samples' memory position into
 // the input of the object.
 void RateTransposer::putSamples(const SAMPLETYPE *samples, uint nSamples)
@@ -209,70 +124,6 @@ void RateTransposer::putSamples(const SAMPLETYPE *samples, uint nSamples)
 }
 
 
-
-// Transposes up the sample rate, causing the observed playback 'rate' of the
-// sound to decrease
-void RateTransposer::upsample(const SAMPLETYPE *src, uint nSamples)
-{
-    uint count, sizeTemp, num;
-
-    // If the parameter 'uRate' value is smaller than 'SCALE', first transpose
-    // the samples and then apply the anti-alias filter to remove aliasing.
-
-    // First check that there's enough room in 'storeBuffer' 
-    // (+16 is to reserve some slack in the destination buffer)
-    sizeTemp = (uint)((float)nSamples / fRate + 16.0f);
-
-    // Transpose the samples, store the result into the end of "storeBuffer"
-    count = transpose(storeBuffer.ptrEnd(sizeTemp), src, nSamples);
-    storeBuffer.putSamples(count);
-
-    // Apply the anti-alias filter to samples in "store output", output the
-    // result to "dest"
-    num = storeBuffer.numSamples();
-    count = pAAFilter->evaluate(outputBuffer.ptrEnd(num), 
-        storeBuffer.ptrBegin(), num, (uint)numChannels);
-    outputBuffer.putSamples(count);
-
-    // Remove the processed samples from "storeBuffer"
-    storeBuffer.receiveSamples(count);
-}
-
-
-// Transposes down the sample rate, causing the observed playback 'rate' of the
-// sound to increase
-void RateTransposer::downsample(const SAMPLETYPE *src, uint nSamples)
-{
-    uint count, sizeTemp;
-
-    // If the parameter 'uRate' value is larger than 'SCALE', first apply the
-    // anti-alias filter to remove high frequencies (prevent them from folding
-    // over the lover frequencies), then transpose.
-
-    // Add the new samples to the end of the storeBuffer
-    storeBuffer.putSamples(src, nSamples);
-
-    // Anti-alias filter the samples to prevent folding and output the filtered 
-    // data to tempBuffer. Note : because of the FIR filter length, the
-    // filtering routine takes in 'filter_length' more samples than it outputs.
-    assert(tempBuffer.isEmpty());
-    sizeTemp = storeBuffer.numSamples();
-
-    count = pAAFilter->evaluate(tempBuffer.ptrEnd(sizeTemp), 
-        storeBuffer.ptrBegin(), sizeTemp, (uint)numChannels);
-
-    if (count == 0) return;
-
-    // Remove the filtered samples from 'storeBuffer'
-    storeBuffer.receiveSamples(count);
-
-    // Transpose the samples (+16 is to reserve some slack in the destination buffer)
-    sizeTemp = (uint)((float)nSamples / fRate + 16.0f);
-    count = transpose(outputBuffer.ptrEnd(sizeTemp), tempBuffer.ptrBegin(), count);
-    outputBuffer.putSamples(count);
-}
-
-
 // Transposes sample rate by applying anti-alias filter to prevent folding. 
 // Returns amount of samples returned in the "dest" buffer.
 // The maximum amount of samples that can be returned at a time is set by
@@ -280,51 +131,45 @@ void RateTransposer::downsample(const SAMPLETYPE *src, uint nSamples)
 void RateTransposer::processSamples(const SAMPLETYPE *src, uint nSamples)
 {
     uint count;
-    uint sizeReq;
 
     if (nSamples == 0) return;
-    assert(pAAFilter);
+
+    // 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) 
+    if (bUseAAFilter == false) 
     {
-        sizeReq = (uint)((float)nSamples / fRate + 1.0f);
+        count = pTransposer->transpose(outputBuffer, inputBuffer);
-        count = transpose(outputBuffer.ptrEnd(sizeReq), src, nSamples);
-        outputBuffer.putSamples(count);
         return;
     }
 
+    assert(pAAFilter);
+
     // Transpose with anti-alias filter
-    if (fRate < 1.0f) 
+    if (pTransposer->rate < 1.0f) 
     {
-        upsample(src, nSamples);
+        // 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  
     {
-        downsample(src, nSamples);
+        // 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);
 
-// Transposes the sample rate of the given samples using linear interpolation. 
+        // Transpose the AA-filtered samples in "midBuffer"
-// Returns the number of samples returned in the "dest" buffer
+        pTransposer->transpose(outputBuffer, midBuffer);
-inline int RateTransposer::transpose(SAMPLETYPE *dest, const SAMPLETYPE *src, uint nSamples)
-{
-#ifndef USE_MULTICH_ALWAYS
-    if (numChannels == 1)
-    {
-        return transposeMono(dest, src, nSamples);
-    }
-    else if (numChannels == 2) 
-    {
-        return transposeStereo(dest, src, nSamples);
-    } 
-    else 
-#endif // USE_MULTICH_ALWAYS
-    {
-        assert(numChannels > 0);
-        return transposeMulti(dest, src, nSamples);
     }
 }
 
@@ -333,17 +178,13 @@ inline int RateTransposer::transpose(SAMPLETYPE *dest, const SAMPLETYPE *src, ui
 void RateTransposer::setChannels(int nChannels)
 {
     assert(nChannels > 0);
-    if (numChannels == nChannels) return;
 
-//    assert(nChannels == 1 || nChannels == 2);
+    if (pTransposer->numChannels == nChannels) return;
-    numChannels = nChannels;
+    pTransposer->setChannels(nChannels);
 
-    storeBuffer.setChannels(numChannels);
+    inputBuffer.setChannels(nChannels);
-    tempBuffer.setChannels(numChannels);
+    midBuffer.setChannels(nChannels);
-    outputBuffer.setChannels(numChannels);
+    outputBuffer.setChannels(nChannels);
-
-    // Inits the linear interpolation registers
-    resetRegisters();
 }
 
 
@@ -351,7 +192,8 @@ void RateTransposer::setChannels(int nChannels)
 void RateTransposer::clear()
 {
     outputBuffer.clear();
-    storeBuffer.clear();
+    midBuffer.clear();
+    inputBuffer.clear();
 }
 
 
@@ -362,387 +204,99 @@ int RateTransposer::isEmpty() const
 
     res = FIFOProcessor::isEmpty();
     if (res == 0) return 0;
-    return storeBuffer.isEmpty();
+    return inputBuffer.isEmpty();
 }
 
 
 //////////////////////////////////////////////////////////////////////////////
 //
-// RateTransposerInteger - integer arithmetic implementation
+// TransposerBase - Base class for interpolation
 // 
 
-/// fixed-point interpolation routine precision
+// static function to set interpolation algorithm
-#define SCALE    65536
+void TransposerBase::setAlgorithm(TransposerBase::ALGORITHM a)
-
-// Constructor
-RateTransposerInteger::RateTransposerInteger() : RateTransposer()
 {
-    // Notice: use local function calling syntax for sake of clarity, 
+    TransposerBase::algorithm = a;
-    // to indicate the fact that C++ constructor can't call virtual functions.
-    sPrevSample=0;
-    RateTransposerInteger::resetRegisters();
-    RateTransposerInteger::setRate(1.0f);
-}
-
-
-RateTransposerInteger::~RateTransposerInteger()
-{
-    if (sPrevSample) delete[] sPrevSample;
-}
-
-
-void RateTransposerInteger::resetRegisters()
-{
-    iSlopeCount = 0;
-    delete[] sPrevSample;
-    sPrevSample = new SAMPLETYPE[numChannels];
-    memset(sPrevSample, 0, numChannels * sizeof(SAMPLETYPE));
-}
-
-
-
-// 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 RateTransposerInteger::transposeMono(SAMPLETYPE *dest, const SAMPLETYPE *src, uint nSamples)
-{
-    int i, remain;
-    LONG_SAMPLETYPE temp, vol1;
-
-    if (nSamples == 0) return 0;  // no samples, no work
-
-    remain = nSamples - 1;
-    i = 0;
-
-    // Process the last sample saved from the previous call first...
-    while (iSlopeCount <= SCALE) 
-    {
-        vol1 = (LONG_SAMPLETYPE)(SCALE - iSlopeCount);
-        temp = vol1 * sPrevSample[0] + iSlopeCount * src[0];
-        dest[i] = (SAMPLETYPE)(temp / SCALE);
-        i++;
-        iSlopeCount += iRate;
-    }
-    // now always (iSlopeCount > SCALE)
-    iSlopeCount -= SCALE;
-
-    while (1)
-    {
-        while (iSlopeCount > SCALE) 
-        {
-            iSlopeCount -= SCALE;
-            src ++;
-            remain --;
-            if (remain == 0) goto end;
-        }
-        vol1 = (LONG_SAMPLETYPE)(SCALE - iSlopeCount);
-        temp = src[0] * vol1 + iSlopeCount * src[1];
-        dest[i] = (SAMPLETYPE)(temp / SCALE);
-
-        i++;
-        iSlopeCount += iRate;
-    }
-end:
-    // Store the last sample for the next round
-    sPrevSample[0] = src[0];
-
-    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 
+// Returns the number of samples returned in the "dest" buffer
-// the "dest" buffer
+int TransposerBase::transpose(FIFOSampleBuffer &dest, FIFOSampleBuffer &src)
-int RateTransposerInteger::transposeStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, uint nSamples)
 {
-    int i, remain;
+    int numSrcSamples = src.numSamples();
-    LONG_SAMPLETYPE temp, vol1;
+    int sizeDemand = (int)((float)numSrcSamples / rate) + 8;
+    int numOutput;
+    SAMPLETYPE *psrc = src.ptrBegin();
+    SAMPLETYPE *pdest = dest.ptrEnd(sizeDemand);
 
-    if (nSamples == 0) return 0;  // no samples, no work
+#ifndef USE_MULTICH_ALWAYS
-
+    if (numChannels == 1)
-    remain = nSamples - 1;
-    i = 0;
-
-    // Process the last sample saved from the sPrevSampleLious call first...
-    while (iSlopeCount <= SCALE) 
     {
-        vol1 = (LONG_SAMPLETYPE)(SCALE - iSlopeCount);
+        numOutput = transposeMono(pdest, psrc, numSrcSamples);
-        temp = vol1 * sPrevSample[0] + iSlopeCount * src[0];
-        dest[2 * i] = (SAMPLETYPE)(temp / SCALE);
-        temp = vol1 * sPrevSample[1] + iSlopeCount * src[1];
-        dest[2 * i + 1] = (SAMPLETYPE)(temp / SCALE);
-        i++;
-        iSlopeCount += iRate;
     }
-    // now always (iSlopeCount > SCALE)
+    else if (numChannels == 2) 
-    iSlopeCount -= SCALE;
-
-    while (1)
     {
-        while (iSlopeCount > SCALE) 
+        numOutput = transposeStereo(pdest, psrc, numSrcSamples);
-        {
+    } 
-            iSlopeCount -= SCALE;
+    else 
-            remain --;
+#endif // USE_MULTICH_ALWAYS
-            src += 2;
+    {
-            if (remain == 0) goto end;
+        assert(numChannels > 0);
-        }
+        numOutput = transposeMulti(pdest, psrc, numSrcSamples);
-        vol1 = (LONG_SAMPLETYPE)(SCALE - iSlopeCount);
-        temp = src[0] * vol1 + iSlopeCount * src[2];
-        dest[2 * i] = (SAMPLETYPE)(temp / SCALE);
-        temp = src[1] * vol1 + iSlopeCount * src[3];
-        dest[2 * i + 1] = (SAMPLETYPE)(temp / SCALE);
-
-        i++;
-        iSlopeCount += iRate;
     }
-end:
+    dest.putSamples(numOutput);
-    // Store the last sample for the next round
+    src.receiveSamples(numSrcSamples);
-    sPrevSample[0] = src[0];
+    return numOutput;
-    sPrevSample[1] = src[1];
-
-    return i;
 }
 
 
-int RateTransposerInteger::transposeMulti(SAMPLETYPE *dest, const SAMPLETYPE *src, uint nSamples)
+TransposerBase::TransposerBase()
 {
-    int i, remaining;
+    numChannels = 0;
-    LONG_SAMPLETYPE temp, vol1;
+    rate = 1.0f;
-
-    if (nSamples == 0) return 0;  // no samples, no work
-
-    remaining = nSamples - 1;
-    i = 0;
-
-    // Process the last sample saved from the sPrevSampleLious call first...
-    while (iSlopeCount <= SCALE)
-    {
-        for (int c = 0; c < numChannels; c ++)
-        {
-            vol1 = (SCALE - iSlopeCount);
-            temp = vol1 * sPrevSample[c] + iSlopeCount * src[c];
-            *dest = (SAMPLETYPE)(temp / SCALE);
-            dest ++;
-        }
-        i++;
-
-        iSlopeCount += iRate;
-    }
-    // now always (iSlopeCount > SCALE)
-    iSlopeCount -= SCALE;
-
-    while (1)
-    {
-        while (iSlopeCount > SCALE) 
-        {
-            iSlopeCount -= SCALE;
-            src += numChannels;
-            remaining --;
-            if (remaining == 0) goto end;
-        }
-
-        for (int c = 0; c < numChannels; c ++)
-        {
-            vol1 = (SCALE - iSlopeCount);
-            temp = src[c] * vol1 + iSlopeCount * src[c + numChannels];
-            *dest = (SAMPLETYPE)(temp / SCALE);
-            dest++;
-        }
-
-        i++;
-        iSlopeCount += iRate;
-    }
-end:
-    // Store the last sample for the next round
-    memcpy(sPrevSample, src, numChannels * sizeof(SAMPLETYPE));
-
-    return i;
-}
-
-// Sets new target iRate. Normal iRate = 1.0, smaller values represent slower 
-// iRate, larger faster iRates.
-void RateTransposerInteger::setRate(float newRate)
-{
-    iRate = (int)(newRate * SCALE + 0.5f);
-    RateTransposer::setRate(newRate);
 }
 
 
-//////////////////////////////////////////////////////////////////////////////
+TransposerBase::~TransposerBase()
-//
-// RateTransposerFloat - floating point arithmetic implementation
-// 
-//////////////////////////////////////////////////////////////////////////////
-
-// Constructor
-RateTransposerFloat::RateTransposerFloat() : RateTransposer()
 {
-    // Notice: use local function calling syntax for sake of clarity, 
-    // to indicate the fact that C++ constructor can't call virtual functions.
-    sPrevSample = NULL;
-    RateTransposerFloat::resetRegisters();
-    RateTransposerFloat::setRate(1.0f);
 }
 
 
-RateTransposerFloat::~RateTransposerFloat()
+void TransposerBase::setChannels(int channels)
 {
-    delete[] sPrevSample;
+    numChannels = channels;
+    resetRegisters();
 }
 
 
-void RateTransposerFloat::resetRegisters()
+void TransposerBase::setRate(float newRate)
 {
-    fSlopeCount = 0;
+    rate = newRate;
-    delete[] sPrevSample;
-    sPrevSample = new SAMPLETYPE[numChannels];
-    memset(sPrevSample, 0, numChannels * sizeof(SAMPLETYPE));
 }
 
 
-
+// static factory function
-// Transposes the sample rate of the given samples using linear interpolation. 
+TransposerBase *TransposerBase::newInstance()
-// 'Mono' version of the routine. Returns the number of samples returned in 
-// the "dest" buffer
-int RateTransposerFloat::transposeMono(SAMPLETYPE *dest, const SAMPLETYPE *src, uint nSamples)
 {
-    int i, remain;
+#ifdef SOUNDTOUCH_INTEGER_SAMPLES
-
+    // Notice: For integer arithmetics support only linear algorithm (due to simplest calculus)
-    remain = 0;
+    return ::new InterpolateLinearInteger;
-    i = 0;
+#else
-
+    switch (algorithm)
-    // Process the last sample saved from the previous call first...
-    while (fSlopeCount <= 1.0f) 
     {
-        dest[i] = (SAMPLETYPE)((1.0f - fSlopeCount) * sPrevSample[0] + fSlopeCount * src[0]);
+        case LINEAR:
-        i++;
+            return new InterpolateLinearFloat;
-        fSlopeCount += fRate;
-    }
-    fSlopeCount -= 1.0f;
 
-    if (nSamples > 1)
+        case CUBIC:
-    {
+            return new InterpolateCubic;
-        while (1)
-        {
-            while (fSlopeCount > 1.0f) 
-            {
-                fSlopeCount -= 1.0f;
-                src ++;
-                remain --;
-                if (remain == 0) goto end;
-            }
-            dest[i] = (SAMPLETYPE)((1.0f - fSlopeCount) * src[0] + fSlopeCount * src[1]);
-            i++;
-            fSlopeCount += fRate;
-        }
-    }
-end:
-    // Store the last sample for the next round
-    sPrevSample[0] = src[0];
 
-    return i;
+        case SHANNON:
-}
+            return new InterpolateShannon;
-
+
-
+        default:
-// Transposes the sample rate of the given samples using linear interpolation. 
+            assert(false);
-// 'Mono' version of the routine. Returns the number of samples returned in 
+            return NULL;
-// the "dest" buffer
+    }
-int RateTransposerFloat::transposeStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, uint nSamples)
+#endif
-{
-    int i, remain;
-
-    if (nSamples == 0) return 0;  // no samples, no work
-
-    remain = nSamples - 1;
-    i = 0;
-
-    // Process the last sample saved from the sPrevSampleLious call first...
-    while (fSlopeCount <= 1.0f) 
-    {
-        dest[2 * i] = (SAMPLETYPE)((1.0f - fSlopeCount) * sPrevSample[0] + fSlopeCount * src[0]);
-        dest[2 * i + 1] = (SAMPLETYPE)((1.0f - fSlopeCount) * sPrevSample[1] + fSlopeCount * src[1]);
-        i++;
-        fSlopeCount += fRate;
-    }
-    // now always (iSlopeCount > 1.0f)
-    fSlopeCount -= 1.0f;
-
-    if (nSamples > 1)
-    {
-        while (1)
-        {
-            while (fSlopeCount > 1.0f) 
-            {
-                fSlopeCount -= 1.0f;
-                remain --;
-                src += 2;
-                if (remain == 0) goto end;
-            }
-
-            dest[2 * i] = (SAMPLETYPE)((1.0f - fSlopeCount) * src[0] 
-                + fSlopeCount * src[2]);
-            dest[2 * i + 1] = (SAMPLETYPE)((1.0f - fSlopeCount) * src[1] 
-                + fSlopeCount * src[3]);
-
-            i++;
-            fSlopeCount += fRate;
-        }
-    }
-end:
-    // Store the last sample for the next round
-    sPrevSample[0] = src[0];
-    sPrevSample[1] = src[1];
-
-    return i;
-}
-
-int RateTransposerFloat::transposeMulti(SAMPLETYPE *dest, const SAMPLETYPE *src, uint nSamples)
-{
-    int i, remaining;
-
-    if (nSamples == 0) return 0;  // no samples, no work
-
-    remaining = nSamples - 1;
-    i = 0;
-
-    // Process the last sample saved from the sPrevSampleLious call first...
-    while (fSlopeCount <= 1.0f) 
-    {
-        for (int c = 0; c < numChannels; c ++)
-        {
-            *dest = (SAMPLETYPE)((1.0f - fSlopeCount) * sPrevSample[c] + fSlopeCount * src[c]);
-            dest ++;
-        }
-        i++;
-        fSlopeCount += fRate;
-    }
-    // now always (iSlopeCount > 1.0f)
-    fSlopeCount -= 1.0f;
-
-    while (remaining > 0)
-    {
-        while (fSlopeCount > 1.0f) 
-        {
-            fSlopeCount -= 1.0f;
-            src += numChannels;
-            remaining --;
-            if (remaining == 0) goto end;
-        }
-
-        for (int c = 0; c < numChannels; c ++)
-        {
-            *dest = (SAMPLETYPE)((1.0f - fSlopeCount) * src[c] 
-                + fSlopeCount * src[c + numChannels]);
-            dest++;
-        }
-
-        i++;
-        fSlopeCount += fRate;
-    }
-
-end:
-    // Store the last sample for the next round
-    memcpy(sPrevSample, src, numChannels * sizeof(SAMPLETYPE));
-
-    return i;
 }

Externals/soundtouch/RateTransposer.h

@@ -14,10 +14,10 @@
 ///
 ////////////////////////////////////////////////////////////////////////////////
 //
-// Last changed  : $Date: 2013-06-12 15:24:44 +0000 (Wed, 12 Jun 2013) $
+// Last changed  : $Date: 2014-04-07 01:57:21 +1000 (Mon, 07 Apr 2014) $
 // File revision : $Revision: 4 $
 //
-// $Id: RateTransposer.h 171 2013-06-12 15:24:44Z oparviai $
+// $Id: RateTransposer.h 195 2014-04-06 15:57:21Z oparviai $
 //
 ////////////////////////////////////////////////////////////////////////////////
 //
@@ -55,51 +55,71 @@
 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 void resetRegisters() = 0;
+
+    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:
+    float rate;
+    int numChannels;
+
+    TransposerBase();
+    virtual ~TransposerBase();
+
+    virtual int transpose(FIFOSampleBuffer &dest, FIFOSampleBuffer &src);
+    virtual void setRate(float newRate);
+    virtual void setChannels(int channels);
+
+    // static factory function
+    static TransposerBase *newInstance();
+
+    // static function to set interpolation algorithm
+    static void setAlgorithm(ALGORITHM a);
+};
+
+
 /// A common linear samplerate transposer class.
 ///
-/// Note: Use function "RateTransposer::newInstance()" to create a new class 
-/// instance instead of the "new" operator; that function automatically 
-/// chooses a correct implementation depending on if integer or floating 
-/// arithmetics are to be used.
 class RateTransposer : public FIFOProcessor
 {
 protected:
     /// Anti-alias filter object
     AAFilter *pAAFilter;
-
+    TransposerBase *pTransposer;
-    float fRate;
-
-    int numChannels;
 
     /// Buffer for collecting samples to feed the anti-alias filter between
     /// two batches
-    FIFOSampleBuffer storeBuffer;
+    FIFOSampleBuffer inputBuffer;
 
     /// Buffer for keeping samples between transposing & anti-alias filter
-    FIFOSampleBuffer tempBuffer;
+    FIFOSampleBuffer midBuffer;
 
     /// Output sample buffer
     FIFOSampleBuffer outputBuffer;
 
-    BOOL bUseAAFilter;
+    bool bUseAAFilter;
 
-    virtual void resetRegisters() = 0;
-
-    virtual int transposeStereo(SAMPLETYPE *dest, 
-                         const SAMPLETYPE *src, 
-                         uint numSamples) = 0;
-    virtual int transposeMono(SAMPLETYPE *dest, 
-                       const SAMPLETYPE *src, 
-                       uint numSamples) = 0;
-    virtual int transposeMulti(SAMPLETYPE *dest, const SAMPLETYPE *src, uint nSamples) = 0;
-    inline int transpose(SAMPLETYPE *dest, 
-                   const SAMPLETYPE *src, 
-                   uint numSamples);
-
-    void downsample(const SAMPLETYPE *src, 
-                    uint numSamples);
-    void upsample(const SAMPLETYPE *src, 
-                 uint numSamples);
 
     /// Transposes sample rate by applying anti-alias filter to prevent folding. 
     /// Returns amount of samples returned in the "dest" buffer.
@@ -108,34 +128,33 @@ protected:
     void processSamples(const SAMPLETYPE *src, 
                         uint numSamples);
 
-
 public:
     RateTransposer();
     virtual ~RateTransposer();
 
     /// Operator 'new' is overloaded so that it automatically creates a suitable instance 
     /// depending on if we're to use integer or floating point arithmetics.
-    static void *operator new(size_t s);
+//    static void *operator new(size_t s);
 
     /// Use this function instead of "new" operator to create a new instance of this class. 
     /// This function automatically chooses a correct implementation, depending on if 
     /// integer ot floating point arithmetics are to be used.
-    static RateTransposer *newInstance();
+//    static RateTransposer *newInstance();
 
     /// Returns the output buffer object
     FIFOSamplePipe *getOutput() { return &outputBuffer; };
 
     /// Returns the store buffer object
-    FIFOSamplePipe *getStore() { return &storeBuffer; };
+//    FIFOSamplePipe *getStore() { return &storeBuffer; };
 
     /// Return anti-alias filter object
     AAFilter *getAAFilter();
 
     /// Enables/disables the anti-alias filter. Zero to disable, nonzero to enable
-    void enableAAFilter(BOOL newMode);
+    void enableAAFilter(bool newMode);
 
     /// Returns nonzero if anti-alias filter is enabled.
-    BOOL isAAFilterEnabled() const;
+    bool isAAFilterEnabled() const;
 
     /// Sets new target rate. Normal rate = 1.0, smaller values represent slower 
     /// rate, larger faster rates.

Externals/soundtouch/SoundTouch.cpp

@@ -41,10 +41,10 @@
 ///
 ////////////////////////////////////////////////////////////////////////////////
 //
-// Last changed  : $Date: 2013-06-12 15:24:44 +0000 (Wed, 12 Jun 2013) $
+// Last changed  : $Date: 2014-04-07 01:57:21 +1000 (Mon, 07 Apr 2014) $
 // File revision : $Revision: 4 $
 //
-// $Id: SoundTouch.cpp 171 2013-06-12 15:24:44Z oparviai $
+// $Id: SoundTouch.cpp 195 2014-04-06 15:57:21Z oparviai $
 //
 ////////////////////////////////////////////////////////////////////////////////
 //
@@ -97,7 +97,7 @@ SoundTouch::SoundTouch()
 {
     // Initialize rate transposer and tempo changer instances
 
-    pRateTransposer = RateTransposer::newInstance();
+    pRateTransposer = new RateTransposer();
     pTDStretch = TDStretch::newInstance();
 
     setOutPipe(pTDStretch);
@@ -111,7 +111,7 @@ SoundTouch::SoundTouch()
     calcEffectiveRateAndTempo();
 
     channels = 0;
-    bSrateSet = FALSE;
+    bSrateSet = false;
 }
 
 
@@ -255,7 +255,7 @@ void SoundTouch::calcEffectiveRateAndTempo()
             tempoOut = pTDStretch->getOutput();
             tempoOut->moveSamples(*output);
             // move samples in pitch transposer's store buffer to tempo changer's input
-            pTDStretch->moveSamples(*pRateTransposer->getStore());
+            // deprecated : pTDStretch->moveSamples(*pRateTransposer->getStore());
 
             output = pTDStretch;
         }
@@ -283,7 +283,7 @@ void SoundTouch::calcEffectiveRateAndTempo()
 // Sets sample rate.
 void SoundTouch::setSampleRate(uint srate)
 {
-    bSrateSet = TRUE;
+    bSrateSet = true;
     // set sample rate, leave other tempo changer parameters as they are.
     pTDStretch->setParameters((int)srate);
 }
@@ -293,7 +293,7 @@ void SoundTouch::setSampleRate(uint srate)
 // the input of the object.
 void SoundTouch::putSamples(const SAMPLETYPE *samples, uint nSamples)
 {
-    if (bSrateSet == FALSE) 
+    if (bSrateSet == false) 
     {
         ST_THROW_RT_ERROR("SoundTouch : Sample rate not defined");
     } 
@@ -383,13 +383,12 @@ void SoundTouch::flush()
     pTDStretch->clearInput();
     // yet leave the 'tempoChanger' output intouched as that's where the
     // flushed samples are!
-    free(buff);
 }
 
 
 // Changes a setting controlling the processing system behaviour. See the
 // 'SETTING_...' defines for available setting ID's.
-BOOL SoundTouch::setSetting(int settingId, int value)
+bool SoundTouch::setSetting(int settingId, int value)
 {
     int sampleRate, sequenceMs, seekWindowMs, overlapMs;
 
@@ -400,36 +399,36 @@ BOOL SoundTouch::setSetting(int settingId, int value)
     {
         case SETTING_USE_AA_FILTER :
             // enables / disabless anti-alias filter
-            pRateTransposer->enableAAFilter((value != 0) ? TRUE : FALSE);
+            pRateTransposer->enableAAFilter((value != 0) ? true : false);
-            return TRUE;
+            return true;
 
         case SETTING_AA_FILTER_LENGTH :
             // sets anti-alias filter length
             pRateTransposer->getAAFilter()->setLength(value);
-            return TRUE;
+            return true;
 
         case SETTING_USE_QUICKSEEK :
             // enables / disables tempo routine quick seeking algorithm
-            pTDStretch->enableQuickSeek((value != 0) ? TRUE : FALSE);
+            pTDStretch->enableQuickSeek((value != 0) ? true : false);
-            return TRUE;
+            return true;
 
         case SETTING_SEQUENCE_MS:
             // change time-stretch sequence duration parameter
             pTDStretch->setParameters(sampleRate, value, seekWindowMs, overlapMs);
-            return TRUE;
+            return true;
 
         case SETTING_SEEKWINDOW_MS:
             // change time-stretch seek window length parameter
             pTDStretch->setParameters(sampleRate, sequenceMs, value, overlapMs);
-            return TRUE;
+            return true;
 
         case SETTING_OVERLAP_MS:
             // change time-stretch overlap length parameter
             pTDStretch->setParameters(sampleRate, sequenceMs, seekWindowMs, value);
-            return TRUE;
+            return true;
 
         default :
-            return FALSE;
+            return false;
     }
 }
 

Externals/soundtouch/SoundTouch.h

@@ -79,10 +79,10 @@ namespace soundtouch
 {
 
 /// Soundtouch library version string
-#define SOUNDTOUCH_VERSION          "1.7.2 (dev)"
+#define SOUNDTOUCH_VERSION          "1.8.1 (r198)"
 
 /// SoundTouch library version id
-#define SOUNDTOUCH_VERSION_ID       (10702)
+#define SOUNDTOUCH_VERSION_ID       (10801)
 
 //
 // Available setting IDs for the 'setSetting' & 'get_setting' functions:
@@ -248,7 +248,7 @@ public:
     /// 'SETTING_...' defines for available setting ID's.
     /// 
     /// \return 'TRUE' if the setting was succesfully changed
-    BOOL setSetting(int settingId,   ///< Setting ID number. see SETTING_... defines.
+    bool setSetting(int settingId,   ///< Setting ID number. see SETTING_... defines.
                     int value        ///< New setting value.
                     );
 

Externals/soundtouch/SoundTouch.vcxproj

@@ -48,6 +48,9 @@
     <ClCompile Include="cpu_detect_x86.cpp" />
     <ClCompile Include="FIFOSampleBuffer.cpp" />
     <ClCompile Include="FIRFilter.cpp" />
+    <ClCompile Include="InterpolateCubic.cpp" />
+    <ClCompile Include="InterpolateLinear.cpp" />
+    <ClCompile Include="InterpolateShannon.cpp" />            
     <ClCompile Include="mmx_optimized.cpp" />
     <ClCompile Include="PeakFinder.cpp" />
     <ClCompile Include="RateTransposer.cpp" />
@@ -62,6 +65,9 @@
     <ClInclude Include="FIFOSampleBuffer.h" />
     <ClInclude Include="FIFOSamplePipe.h" />
     <ClInclude Include="FIRFilter.h" />
+    <ClInclude Include="InterpolateCubic.h" />
+    <ClInclude Include="InterpolateLinear.h" />
+    <ClInclude Include="InterpolateShannon.h" />            
     <ClInclude Include="PeakFinder.h" />
     <ClInclude Include="RateTransposer.h" />
     <ClInclude Include="SoundTouch.h" />
@@ -74,4 +80,4 @@
   <Import Project="$(VCTargetsPath)\Microsoft.Cpp.targets" />
   <ImportGroup Label="ExtensionTargets">
   </ImportGroup>
-</Project>
+</Project>

Externals/soundtouch/TDStretch.cpp

@@ -13,10 +13,10 @@
 ///
 ////////////////////////////////////////////////////////////////////////////////
 //
-// Last changed  : $Date: 2013-06-14 17:34:33 +0000 (Fri, 14 Jun 2013) $
+// Last changed  : $Date: 2014-04-07 01:57:21 +1000 (Mon, 07 Apr 2014) $
 // File revision : $Revision: 1.12 $
 //
-// $Id: TDStretch.cpp 172 2013-06-14 17:34:33Z oparviai $
+// $Id: TDStretch.cpp 195 2014-04-06 15:57:21Z oparviai $
 //
 ////////////////////////////////////////////////////////////////////////////////
 //
@@ -84,15 +84,15 @@ static const short _scanOffsets[5][24]={
 
 TDStretch::TDStretch() : FIFOProcessor(&outputBuffer)
 {
-    bQuickSeek = FALSE;
+    bQuickSeek = false;
     channels = 2;
 
     pMidBuffer = NULL;
     pMidBufferUnaligned = NULL;
     overlapLength = 0;
 
-    bAutoSeqSetting = TRUE;
+    bAutoSeqSetting = true;
-    bAutoSeekSetting = TRUE;
+    bAutoSeekSetting = true;
 
 //    outDebt = 0;
     skipFract = 0;
@@ -132,23 +132,23 @@ void TDStretch::setParameters(int aSampleRate, int aSequenceMS,
     if (aSequenceMS > 0)
     {
         this->sequenceMs = aSequenceMS;
-        bAutoSeqSetting = FALSE;
+        bAutoSeqSetting = false;
     } 
     else if (aSequenceMS == 0)
     {
         // if zero, use automatic setting
-        bAutoSeqSetting = TRUE;
+        bAutoSeqSetting = true;
     }
 
     if (aSeekWindowMS > 0) 
     {
         this->seekWindowMs = aSeekWindowMS;
-        bAutoSeekSetting = FALSE;
+        bAutoSeekSetting = false;
     } 
     else if (aSeekWindowMS == 0) 
     {
         // if zero, use automatic setting
-        bAutoSeekSetting = TRUE;
+        bAutoSeekSetting = true;
     }
 
     calcSeqParameters();
@@ -231,14 +231,14 @@ void TDStretch::clear()
 
 // Enables/disables the quick position seeking algorithm. Zero to disable, nonzero
 // to enable
-void TDStretch::enableQuickSeek(BOOL enable)
+void TDStretch::enableQuickSeek(bool enable)
 {
     bQuickSeek = enable;
 }
 
 
 // Returns nonzero if the quick seeking algorithm is enabled.
-BOOL TDStretch::isQuickSeekEnabled() const
+bool TDStretch::isQuickSeekEnabled() const
 {
     return bQuickSeek;
 }
@@ -293,6 +293,7 @@ int TDStretch::seekBestOverlapPositionFull(const SAMPLETYPE *refPos)
 {
     int bestOffs;
     double bestCorr, corr;
+    double norm;
     int i;
 
     bestCorr = FLT_MIN;
@@ -300,11 +301,15 @@ int TDStretch::seekBestOverlapPositionFull(const SAMPLETYPE *refPos)
 
     // Scans for the best correlation value by testing each possible position
     // over the permitted range.
-    for (i = 0; i < seekLength; i ++) 
+    bestCorr = calcCrossCorr(refPos, pMidBuffer, norm);
+    for (i = 1; i < seekLength; i ++) 
     {
         // Calculates correlation value for the mixing position corresponding
-        // to 'i'
+        // to 'i'. Now call "calcCrossCorrAccumulate" that is otherwise same as
-        corr = calcCrossCorr(refPos + channels * i, pMidBuffer);
+        // "calcCrossCorr", but saves time by reusing & updating previously stored 
+        // "norm" value
+        corr = calcCrossCorrAccumulate(refPos + channels * i, pMidBuffer, norm);
+
         // heuristic rule to slightly favour values close to mid of the range
         double tmp = (double)(2 * i - seekLength) / (double)seekLength;
         corr = ((corr + 0.1) * (1.0 - 0.25 * tmp * tmp));
@@ -352,12 +357,13 @@ int TDStretch::seekBestOverlapPositionQuick(const SAMPLETYPE *refPos)
         j = 0;
         while (_scanOffsets[scanCount][j]) 
         {
+            double norm;
             tempOffset = corrOffset + _scanOffsets[scanCount][j];
             if (tempOffset >= seekLength) break;
 
             // Calculates correlation value for the mixing position corresponding
             // to 'tempOffset'
-            corr = (double)calcCrossCorr(refPos + channels * tempOffset, pMidBuffer);
+            corr = (double)calcCrossCorr(refPos + channels * tempOffset, pMidBuffer, norm);
             // heuristic rule to slightly favour values close to mid of the range
             double tmp = (double)(2 * tempOffset - seekLength) / seekLength;
             corr = ((corr + 0.1) * (1.0 - 0.25 * tmp * tmp));
@@ -729,32 +735,72 @@ void TDStretch::calculateOverlapLength(int aoverlapMs)
 }
 
 
-double TDStretch::calcCrossCorr(const short *mixingPos, const short *compare) const
+double TDStretch::calcCrossCorr(const short *mixingPos, const short *compare, double &norm) const
 {
     long corr;
-    long norm;
+    long lnorm;
     int i;
 
-    corr = norm = 0;
+    corr = lnorm = 0;
     // Same routine for stereo and mono. For stereo, unroll loop for better
     // efficiency and gives slightly better resolution against rounding. 
     // For mono it same routine, just  unrolls loop by factor of 4
     for (i = 0; i < channels * overlapLength; i += 4) 
     {
         corr += (mixingPos[i] * compare[i] + 
-                 mixingPos[i + 1] * compare[i + 1] +
+                 mixingPos[i + 1] * compare[i + 1]) >> overlapDividerBits;  // notice: do intermediate division here to avoid integer overflow
-                 mixingPos[i + 2] * compare[i + 2] + 
+        corr += (mixingPos[i + 2] * compare[i + 2] + 
                  mixingPos[i + 3] * compare[i + 3]) >> overlapDividerBits;
-        norm += (mixingPos[i] * mixingPos[i] + 
+        lnorm += (mixingPos[i] * mixingPos[i] + 
-                 mixingPos[i + 1] * mixingPos[i + 1] +
+                  mixingPos[i + 1] * mixingPos[i + 1]) >> overlapDividerBits; // notice: do intermediate division here to avoid integer overflow
-                 mixingPos[i + 2] * mixingPos[i + 2] + 
+        lnorm += (mixingPos[i + 2] * mixingPos[i + 2] + 
-                 mixingPos[i + 3] * mixingPos[i + 3]) >> overlapDividerBits;
+                  mixingPos[i + 3] * mixingPos[i + 3]) >> overlapDividerBits;
     }
 
     // Normalize result by dividing by sqrt(norm) - this step is easiest 
     // done using floating point operation
-    if (norm == 0) norm = 1;    // to avoid div by zero
+    norm = (double)lnorm;
-    return (double)corr / sqrt((double)norm);
+    return (double)corr / sqrt((norm < 1e-9) ? 1.0 : norm);
+}
+
+
+/// Update cross-correlation by accumulating "norm" coefficient by previously calculated value
+double TDStretch::calcCrossCorrAccumulate(const short *mixingPos, const short *compare, double &norm) const
+{
+    long corr;
+    long lnorm;
+    int i;
+
+    // cancel first normalizer tap from previous round
+    lnorm = 0;
+    for (i = 1; i <= channels; i ++)
+    {
+        lnorm -= (mixingPos[-i] * mixingPos[-i]) >> overlapDividerBits;
+    }
+
+    corr = 0;
+    // Same routine for stereo and mono. For stereo, unroll loop for better
+    // efficiency and gives slightly better resolution against rounding. 
+    // For mono it same routine, just  unrolls loop by factor of 4
+    for (i = 0; i < channels * overlapLength; i += 4) 
+    {
+        corr += (mixingPos[i] * compare[i] + 
+                 mixingPos[i + 1] * compare[i + 1]) >> overlapDividerBits;  // notice: do intermediate division here to avoid integer overflow
+        corr += (mixingPos[i + 2] * compare[i + 2] + 
+                 mixingPos[i + 3] * compare[i + 3]) >> overlapDividerBits;
+    }
+
+    // update normalizer with last samples of this round
+    for (int j = 0; j < channels; j ++)
+    {
+        i --;
+        lnorm += (mixingPos[i] * mixingPos[i]) >> overlapDividerBits;
+    }
+    norm += (double)lnorm;
+
+    // Normalize result by dividing by sqrt(norm) - this step is easiest 
+    // done using floating point operation
+    return (double)corr / sqrt((norm < 1e-9) ? 1.0 : norm);
 }
 
 #endif // SOUNDTOUCH_INTEGER_SAMPLES
@@ -834,10 +880,10 @@ void TDStretch::calculateOverlapLength(int overlapInMsec)
 }
 
 
-double TDStretch::calcCrossCorr(const float *mixingPos, const float *compare) const
+/// Calculate cross-correlation
+double TDStretch::calcCrossCorr(const float *mixingPos, const float *compare, double &norm) const
 {
     double corr;
-    double norm;
     int i;
 
     corr = norm = 0;
@@ -859,8 +905,43 @@ double TDStretch::calcCrossCorr(const float *mixingPos, const float *compare) co
                 mixingPos[i + 3] * mixingPos[i + 3];
     }
 
-    if (norm < 1e-9) norm = 1.0;    // to avoid div by zero
+    return corr / sqrt((norm < 1e-9 ? 1.0 : norm));
-    return corr / sqrt(norm);
 }
 
+
+/// Update cross-correlation by accumulating "norm" coefficient by previously calculated value
+double TDStretch::calcCrossCorrAccumulate(const float *mixingPos, const float *compare, double &norm) const
+{
+    double corr;
+    int i;
+
+    corr = 0;
+
+    // cancel first normalizer tap from previous round
+    for (i = 1; i <= channels; i ++)
+    {
+        norm -= mixingPos[-i] * mixingPos[-i];
+    }
+
+    // Same routine for stereo and mono. For Stereo, unroll by factor of 2.
+    // For mono it's same routine yet unrollsd by factor of 4.
+    for (i = 0; i < channels * overlapLength; i += 4) 
+    {
+        corr += mixingPos[i] * compare[i] +
+                mixingPos[i + 1] * compare[i + 1] +
+                mixingPos[i + 2] * compare[i + 2] +
+                mixingPos[i + 3] * compare[i + 3];
+    }
+
+    // update normalizer with last samples of this round
+    for (int j = 0; j < channels; j ++)
+    {
+        i --;
+        norm += mixingPos[i] * mixingPos[i];
+    }
+
+    return corr / sqrt((norm < 1e-9 ? 1.0 : norm));
+}
+
+
 #endif // SOUNDTOUCH_FLOAT_SAMPLES

Externals/soundtouch/TDStretch.h

@@ -13,10 +13,10 @@
 ///
 ////////////////////////////////////////////////////////////////////////////////
 //
-// Last changed  : $Date: 2013-06-12 15:24:44 +0000 (Wed, 12 Jun 2013) $
+// Last changed  : $Date: 2014-04-07 01:57:21 +1000 (Mon, 07 Apr 2014) $
 // File revision : $Revision: 4 $
 //
-// $Id: TDStretch.h 171 2013-06-12 15:24:44Z oparviai $
+// $Id: TDStretch.h 195 2014-04-06 15:57:21Z oparviai $
 //
 ////////////////////////////////////////////////////////////////////////////////
 //
@@ -125,21 +125,22 @@ protected:
     float skipFract;
     FIFOSampleBuffer outputBuffer;
     FIFOSampleBuffer inputBuffer;
-    BOOL bQuickSeek;
+    bool bQuickSeek;
 
     int sampleRate;
     int sequenceMs;
     int seekWindowMs;
     int overlapMs;
-    BOOL bAutoSeqSetting;
+    bool bAutoSeqSetting;
-    BOOL bAutoSeekSetting;
+    bool bAutoSeekSetting;
 
     void acceptNewOverlapLength(int newOverlapLength);
 
     virtual void clearCrossCorrState();
     void calculateOverlapLength(int overlapMs);
 
-    virtual double calcCrossCorr(const SAMPLETYPE *mixingPos, const SAMPLETYPE *compare) const;
+    virtual double calcCrossCorr(const SAMPLETYPE *mixingPos, const SAMPLETYPE *compare, double &norm) const;
+    virtual double calcCrossCorrAccumulate(const SAMPLETYPE *mixingPos, const SAMPLETYPE *compare, double &norm) const;
 
     virtual int seekBestOverlapPositionFull(const SAMPLETYPE *refPos);
     virtual int seekBestOverlapPositionQuick(const SAMPLETYPE *refPos);
@@ -194,10 +195,10 @@ public:
 
     /// Enables/disables the quick position seeking algorithm. Zero to disable, 
     /// nonzero to enable
-    void enableQuickSeek(BOOL enable);
+    void enableQuickSeek(bool enable);
 
     /// Returns nonzero if the quick seeking algorithm is enabled.
-    BOOL isQuickSeekEnabled() const;
+    bool isQuickSeekEnabled() const;
 
     /// Sets routine control parameters. These control are certain time constants
     /// defining how the sound is stretched to the desired duration.
@@ -248,7 +249,8 @@ public:
     class TDStretchMMX : public TDStretch
     {
     protected:
-        double calcCrossCorr(const short *mixingPos, const short *compare) const;
+        double calcCrossCorr(const short *mixingPos, const short *compare, double &norm) const;
+        double calcCrossCorrAccumulate(const short *mixingPos, const short *compare, double &norm) const;
         virtual void overlapStereo(short *output, const short *input) const;
         virtual void clearCrossCorrState();
     };
@@ -260,7 +262,8 @@ public:
     class TDStretchSSE : public TDStretch
     {
     protected:
-        double calcCrossCorr(const float *mixingPos, const float *compare) const;
+        double calcCrossCorr(const float *mixingPos, const float *compare, double &norm) const;
+        double calcCrossCorrAccumulate(const float *mixingPos, const float *compare, double &norm) const;
     };
 
 #endif /// SOUNDTOUCH_ALLOW_SSE

Externals/soundtouch/cpu_detect.h

@@ -12,7 +12,7 @@
 ///
 ////////////////////////////////////////////////////////////////////////////////
 //
-// Last changed  : $Date: 2008-02-10 16:26:55 +0000 (Sun, 10 Feb 2008) $
+// Last changed  : $Date: 2008-02-11 03:26:55 +1100 (Mon, 11 Feb 2008) $
 // File revision : $Revision: 4 $
 //
 // $Id: cpu_detect.h 11 2008-02-10 16:26:55Z oparviai $

Externals/soundtouch/cpu_detect_x86.cpp

@@ -11,10 +11,10 @@
 ///
 ////////////////////////////////////////////////////////////////////////////////
 //
-// Last changed  : $Date: 2012-11-08 18:44:37 +0000 (Thu, 08 Nov 2012) $
+// Last changed  : $Date: 2014-01-08 05:24:28 +1100 (Wed, 08 Jan 2014) $
 // File revision : $Revision: 4 $
 //
-// $Id: cpu_detect_x86.cpp 159 2012-11-08 18:44:37Z oparviai $
+// $Id: cpu_detect_x86.cpp 183 2014-01-07 18:24:28Z oparviai $
 //
 ////////////////////////////////////////////////////////////////////////////////
 //
@@ -42,22 +42,20 @@
 #include "cpu_detect.h"
 #include "STTypes.h"
 
+
 #if defined(SOUNDTOUCH_ALLOW_X86_OPTIMIZATIONS)
 
-    #if defined(__GNUC__) && defined(__i386__)
+   #if defined(__GNUC__) && defined(__i386__)
-        // gcc
+       // gcc
-        #include "cpuid.h"
+       #include "cpuid.h"
-    #elif defined(_M_IX86)
+   #elif defined(_M_IX86)
-        // windows non-gcc
+       // windows non-gcc
-        #include <intrin.h>
+       #include <intrin.h>
-    #endif
+   #endif
-    #ifndef bit_MMX
-        #define bit_MMX     (1 << 23)
-        #define bit_MMX     (1 << 23)
-        #define bit_SSE     (1 << 25)
-        #define bit_SSE2    (1 << 26)
-    #endif
 
+   #define bit_MMX     (1 << 23)
+   #define bit_SSE     (1 << 25)
+   #define bit_SSE2    (1 << 26)
 #endif
 
 

Externals/soundtouch/mmx_optimized.cpp

@@ -20,10 +20,10 @@
 ///
 ////////////////////////////////////////////////////////////////////////////////
 //
-// Last changed  : $Date: 2012-11-08 18:53:01 +0000 (Thu, 08 Nov 2012) $
+// Last changed  : $Date: 2014-01-08 05:25:40 +1100 (Wed, 08 Jan 2014) $
 // File revision : $Revision: 4 $
 //
-// $Id: mmx_optimized.cpp 160 2012-11-08 18:53:01Z oparviai $
+// $Id: mmx_optimized.cpp 184 2014-01-07 18:25:40Z oparviai $
 //
 ////////////////////////////////////////////////////////////////////////////////
 //
@@ -68,7 +68,7 @@ using namespace soundtouch;
 
 
 // Calculates cross correlation of two buffers
-double TDStretchMMX::calcCrossCorr(const short *pV1, const short *pV2) const
+double TDStretchMMX::calcCrossCorr(const short *pV1, const short *pV2, double &dnorm) const
 {
     const __m64 *pVec1, *pVec2;
     __m64 shifter;
@@ -93,19 +93,19 @@ double TDStretchMMX::calcCrossCorr(const short *pV1, const short *pV2) const
         // _mm_add_pi32 : 2*32bit add
         // _m_psrad     : 32bit right-shift
 
-        temp = _mm_add_pi32(_mm_madd_pi16(pVec1[0], pVec2[0]),
+        temp = _mm_add_pi32(_mm_sra_pi32(_mm_madd_pi16(pVec1[0], pVec2[0]), shifter),
-                            _mm_madd_pi16(pVec1[1], pVec2[1]));
+                            _mm_sra_pi32(_mm_madd_pi16(pVec1[1], pVec2[1]), shifter));
-        temp2 = _mm_add_pi32(_mm_madd_pi16(pVec1[0], pVec1[0]),
+        temp2 = _mm_add_pi32(_mm_sra_pi32(_mm_madd_pi16(pVec1[0], pVec1[0]), shifter),
-                             _mm_madd_pi16(pVec1[1], pVec1[1]));
+                            _mm_sra_pi32(_mm_madd_pi16(pVec1[1], pVec1[1]), shifter));
-        accu = _mm_add_pi32(accu, _mm_sra_pi32(temp, shifter));
+        accu = _mm_add_pi32(accu, temp);
-        normaccu = _mm_add_pi32(normaccu, _mm_sra_pi32(temp2, shifter));
+        normaccu = _mm_add_pi32(normaccu, temp2);
 
-        temp = _mm_add_pi32(_mm_madd_pi16(pVec1[2], pVec2[2]),
+        temp = _mm_add_pi32(_mm_sra_pi32(_mm_madd_pi16(pVec1[2], pVec2[2]), shifter),
-                            _mm_madd_pi16(pVec1[3], pVec2[3]));
+                            _mm_sra_pi32(_mm_madd_pi16(pVec1[3], pVec2[3]), shifter));
-        temp2 = _mm_add_pi32(_mm_madd_pi16(pVec1[2], pVec1[2]),
+        temp2 = _mm_add_pi32(_mm_sra_pi32(_mm_madd_pi16(pVec1[2], pVec1[2]), shifter),
-                             _mm_madd_pi16(pVec1[3], pVec1[3]));
+                            _mm_sra_pi32(_mm_madd_pi16(pVec1[3], pVec1[3]), shifter));
-        accu = _mm_add_pi32(accu, _mm_sra_pi32(temp, shifter));
+        accu = _mm_add_pi32(accu, temp);
-        normaccu = _mm_add_pi32(normaccu, _mm_sra_pi32(temp2, shifter));
+        normaccu = _mm_add_pi32(normaccu, temp2);
 
         pVec1 += 4;
         pVec2 += 4;
@@ -125,14 +125,81 @@ double TDStretchMMX::calcCrossCorr(const short *pV1, const short *pV2) const
 
     // Normalize result by dividing by sqrt(norm) - this step is easiest 
     // done using floating point operation
-    if (norm == 0) norm = 1;    // to avoid div by zero
+    dnorm = (double)norm;
 
-    return (double)corr / sqrt((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
+{
+    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]) >> overlapDividerBits;
+    }
+
+    pVec1 = (__m64*)pV1;
+    pVec2 = (__m64*)pV2;
+
+    shifter = _m_from_int(overlapDividerBits);
+    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]) >> overlapDividerBits;
+    }
+    dnorm += (double)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()
 {

Externals/soundtouch/sse_optimized.cpp

@@ -23,10 +23,10 @@
 ///
 ////////////////////////////////////////////////////////////////////////////////
 //
-// Last changed  : $Date: 2012-11-08 18:53:01 +0000 (Thu, 08 Nov 2012) $
+// Last changed  : $Date: 2014-01-08 05:25:40 +1100 (Wed, 08 Jan 2014) $
 // File revision : $Revision: 4 $
 //
-// $Id: sse_optimized.cpp 160 2012-11-08 18:53:01Z oparviai $
+// $Id: sse_optimized.cpp 184 2014-01-07 18:25:40Z oparviai $
 //
 ////////////////////////////////////////////////////////////////////////////////
 //
@@ -71,7 +71,7 @@ using namespace soundtouch;
 #include <math.h>
 
 // Calculates cross correlation of two buffers
-double TDStretchSSE::calcCrossCorr(const float *pV1, const float *pV2) const
+double TDStretchSSE::calcCrossCorr(const float *pV1, const float *pV2, double &norm) const
 {
     int i;
     const float *pVec1;
@@ -141,11 +141,10 @@ double TDStretchSSE::calcCrossCorr(const float *pV1, const float *pV2) const
 
     // return value = vSum[0] + vSum[1] + vSum[2] + vSum[3]
     float *pvNorm = (float*)&vNorm;
-    double norm = sqrt(pvNorm[0] + pvNorm[1] + pvNorm[2] + pvNorm[3]);
+    norm = (pvNorm[0] + pvNorm[1] + pvNorm[2] + pvNorm[3]);
-    if (norm < 1e-9) norm = 1.0;    // to avoid div by zero
 
     float *pvSum = (float*)&vSum;
-    return (double)(pvSum[0] + pvSum[1] + pvSum[2] + pvSum[3]) / norm;
+    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;
@@ -182,6 +181,16 @@ double TDStretchSSE::calcCrossCorr(const float *pV1, const float *pV2) const
 }
 
 
+
+double TDStretchSSE::calcCrossCorrAccumulate(const float *pV1, const float *pV2, double &norm) const
+{
+    // 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'