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- Issue #1949 - Part 1: Update soundtouch library to 2.3.1. (e98c7b8f)
- Issue #1949 - Part 2: Disable exceptions in soundtouch's STTypes.h. (f170360d)
This commit is contained in:
roytam1
2022-07-06 10:03:43 +08:00
parent bd44f86100
commit 598aa0cea6
33 changed files with 812 additions and 589 deletions
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media/libsoundtouch/AUTHORS
+1 -1
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@@ -1,4 +1,4 @@
The SoundTouch Library
Copyright © Olli Parviainen 2001-2012
Copyright © Olli Parviainen 2001-2021
http://www.surina.net/soundtouch/
media/libsoundtouch/LICENSE
+3 -3
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@@ -2,7 +2,7 @@
Version 2.1, February 1999
Copyright (C) 1991, 1999 Free Software Foundation, Inc.
59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
Everyone is permitted to copy and distribute verbatim copies
of this license document, but changing it is not allowed.
@@ -117,7 +117,7 @@ be combined with the library in order to run.
0. This License Agreement applies to any software library or other
program which contains a notice placed by the copyright holder or
other authoried party saying it may be distributed under the terms of
other authorized party saying it may be distributed under the terms of
this Lesser General Public License (also called "this License").
Each licensee is addressed as "you".
@@ -455,4 +455,4 @@ FAILURE OF THE LIBRARY TO OPERATE WITH ANY OTHER SOFTWARE), EVEN IF
SUCH HOLDER OR OTHER PARTY HAS BEEN ADVISED OF THE POSSIBILITY OF SUCH
DAMAGES.
END OF TERMS AND CONDITIONS
END OF TERMS AND CONDITIONS
media/libsoundtouch/README_MOZILLA → media/libsoundtouch/README_MCP
+2 -2
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@@ -1,6 +1,6 @@
These files are from the SoundTouch library (http://www.surina.net/soundtouch/),
and are extracted from the revision r222 of the svn repository at
https://soundtouch.svn.sourceforge.net/svnroot/soundtouch/trunk.
and are extracted from tagged version 2.3.1 of the git repository at
https://codeberg.org/soundtouch/soundtouch/.
The whole library is not used, only the relevant files are imported in the tree,
using the script `update.sh`. Some changes have been made to the files, using
media/libsoundtouch/mcp-disableexceptions.patch
+13
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@@ -0,0 +1,13 @@
diff -u /src/STTypes.h /src/STTypes.h
--- /src/STTypes.h
+++ /src/STTypes.h
@@ -174,7 +174,7 @@
}
// define ST_NO_EXCEPTION_HANDLING switch to disable throwing std exceptions:
-// #define ST_NO_EXCEPTION_HANDLING 1
+#define ST_NO_EXCEPTION_HANDLING 1
#ifdef ST_NO_EXCEPTION_HANDLING
// Exceptions disabled. Throw asserts instead if enabled.
#include <assert.h>
media/libsoundtouch/moz-libsoundtouch.patch
+14 -19
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@@ -56,7 +56,7 @@ diff -u /src/cpu_detect_x86.cpp /src/cpu_detect_x86.cpp
diff -u /src/STTypes.h /src/STTypes.h
--- /src/STTypes.h
+++ /src/STTypes.h
@@ -54,12 +54,13 @@
@@ -54,12 +54,17 @@
#define SOUNDTOUCH_ALIGN_POINTER_16(x) ( ( (ulongptr)(x) + 15 ) & ~(ulongptr)15 )
@@ -68,9 +68,13 @@ diff -u /src/STTypes.h /src/STTypes.h
+#include "soundtouch_config.h"
+#if defined(WIN32)
+#define EXPORT __declspec(dllexport)
+#if defined(BUILDING_SOUNDTOUCH)
+#define SOUNDTOUCH_API __declspec(dllexport)
+#else
+#define EXPORT
+#define SOUNDTOUCH_API __declspec(dllimport)
+#endif
+#else
+#define SOUNDTOUCH_API
+#endif
namespace soundtouch
@@ -83,7 +87,7 @@ diff -u /src/SoundTouch.h /src/SoundTouch.h
#define SETTING_NOMINAL_OUTPUT_SEQUENCE 7
-class SoundTouch : public FIFOProcessor
+class EXPORT SoundTouch : public FIFOProcessor
+class SOUNDTOUCH_API SoundTouch : public FIFOProcessor
{
private:
/// Rate transposer class instance
@@ -116,26 +120,17 @@ diff -u /src/TDStretch.cpp /src/TDStretch.cpp
+#endif
// Check if MMX/SSE instruction set extensions supported by CPU
diff --git a/media/libsoundtouch/src/AAFilter.cpp b/media/libsoundtouch/src/AAFilter.cpp
--- a/media/libsoundtouch/src/AAFilter.cpp
+++ b/media/libsoundtouch/src/AAFilter.cpp
@@ -44,17 +44,17 @@
#include <assert.h>
#include <math.h>
#include <stdlib.h>
#include "AAFilter.h"
#include "FIRFilter.h"
diff -u /src/AAFilter.cpp /src/AAFilter.cpp
--- /src/AAFilter.cpp
+++ /src/AAFilter.cpp
@@ -42,7 +42,7 @@
using namespace soundtouch;
-#define PI 3.141592655357989
-#define PI 3.14159265358979323846
+#define PI M_PI
#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>
media/libsoundtouch/src/AAFilter.cpp
-14
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@@ -12,13 +12,6 @@
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2014-01-05 21:40:22 +0000 (Sun, 05 Jan 2014) $
// File revision : $Revision: 4 $
//
// $Id: AAFilter.cpp 177 2014-01-05 21:40:22Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
@@ -75,7 +68,6 @@ using namespace soundtouch;
#define _DEBUG_SAVE_AAFIR_COEFFS(x, y)
#endif
/*****************************************************************************
*
* Implementation of the class 'AAFilter'
@@ -90,14 +82,12 @@ AAFilter::AAFilter(uint len)
}
AAFilter::~AAFilter()
{
delete pFIR;
}
// Sets new anti-alias filter cut-off edge frequency, scaled to
// sampling frequency (nyquist frequency = 0.5).
// The filter will cut frequencies higher than the given frequency.
@@ -108,7 +98,6 @@ void AAFilter::setCutoffFreq(double newCutoffFreq)
}
// Sets number of FIR filter taps
void AAFilter::setLength(uint newLength)
{
@@ -117,7 +106,6 @@ void AAFilter::setLength(uint newLength)
}
// Calculates coefficients for a low-pass FIR filter using Hamming window
void AAFilter::calculateCoeffs()
{
@@ -177,12 +165,10 @@ void AAFilter::calculateCoeffs()
for (i = 0; i < length; i ++)
{
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;
}
media/libsoundtouch/src/AAFilter.h
-7
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@@ -13,13 +13,6 @@
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2014-01-07 19:41:23 +0000 (Tue, 07 Jan 2014) $
// File revision : $Revision: 4 $
//
// $Id: AAFilter.h 187 2014-01-07 19:41:23Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
media/libsoundtouch/src/FIFOSampleBuffer.cpp
+11 -10
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@@ -15,13 +15,6 @@
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2012-11-08 18:53:01 +0000 (Thu, 08 Nov 2012) $
// File revision : $Revision: 4 $
//
// $Id: FIFOSampleBuffer.cpp 160 2012-11-08 18:53:01Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
@@ -80,7 +73,8 @@ void FIFOSampleBuffer::setChannels(int numChannels)
{
uint usedBytes;
assert(numChannels > 0);
if (!verifyNumberOfChannels(numChannels)) return;
usedBytes = channels * samplesInBuffer;
channels = (uint)numChannels;
samplesInBuffer = usedBytes / channels;
@@ -131,7 +125,7 @@ void FIFOSampleBuffer::putSamples(uint nSamples)
//
// Parameter 'slackCapacity' tells the function how much free capacity (in
// terms of samples) there _at least_ should be, in order to the caller to
// succesfully insert all the required samples to the buffer. When necessary,
// successfully insert all the required samples to the buffer. When necessary,
// the function grows the buffer size to comply with this requirement.
//
// When using this function as means for inserting new samples, also remember
@@ -158,7 +152,7 @@ SAMPLETYPE *FIFOSampleBuffer::ptrBegin()
}
// Ensures that the buffer has enought capacity, i.e. space for _at least_
// Ensures that the buffer has enough capacity, i.e. space for _at least_
// 'capacityRequirement' number of samples. The buffer is grown in steps of
// 4 kilobytes to eliminate the need for frequently growing up the buffer,
// as well as to round the buffer size up to the virtual memory page size.
@@ -272,3 +266,10 @@ uint FIFOSampleBuffer::adjustAmountOfSamples(uint numSamples)
return samplesInBuffer;
}
/// Add silence to end of buffer
void FIFOSampleBuffer::addSilent(uint nSamples)
{
memset(ptrEnd(nSamples), 0, sizeof(SAMPLETYPE) * nSamples * channels);
samplesInBuffer += nSamples;
}
media/libsoundtouch/src/FIFOSampleBuffer.h
+4 -8
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@@ -15,13 +15,6 @@
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2014-01-05 21:40:22 +0000 (Sun, 05 Jan 2014) $
// File revision : $Revision: 4 $
//
// $Id: FIFOSampleBuffer.h 177 2014-01-05 21:40:22Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
@@ -119,7 +112,7 @@ public:
/// 'putSamples(numSamples)' function.
SAMPLETYPE *ptrEnd(
uint slackCapacity ///< How much free capacity (in samples) there _at least_
///< should be so that the caller can succesfully insert the
///< should be so that the caller can successfully insert the
///< desired samples to the buffer. If necessary, the function
///< grows the buffer size to comply with this requirement.
);
@@ -177,6 +170,9 @@ public:
/// allow trimming (downwards) amount of samples in pipeline.
/// Returns adjusted amount of samples
uint adjustAmountOfSamples(uint numSamples);
/// Add silence to end of buffer
void addSilent(uint nSamples);
};
}
media/libsoundtouch/src/FIFOSamplePipe.h
+12 -16
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@@ -17,13 +17,6 @@
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2012-06-13 19:29:53 +0000 (Wed, 13 Jun 2012) $
// File revision : $Revision: 4 $
//
// $Id: FIFOSamplePipe.h 143 2012-06-13 19:29:53Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
@@ -58,6 +51,18 @@ namespace soundtouch
/// Abstract base class for FIFO (first-in-first-out) sample processing classes.
class FIFOSamplePipe
{
protected:
bool verifyNumberOfChannels(int nChannels) const
{
if ((nChannels > 0) && (nChannels <= SOUNDTOUCH_MAX_CHANNELS))
{
return true;
}
ST_THROW_RT_ERROR("Error: Illegal number of channels");
return false;
}
public:
// virtual default destructor
virtual ~FIFOSamplePipe() {}
@@ -122,7 +127,6 @@ public:
};
/// Base-class for sound processing routines working in FIFO principle. With this base
/// class it's easy to implement sound processing stages that can be chained together,
/// so that samples that are fed into beginning of the pipe automatically go through
@@ -145,7 +149,6 @@ protected:
output = pOutput;
}
/// Constructor. Doesn't define output pipe; it has to be set be
/// 'setOutPipe' function.
FIFOProcessor()
@@ -153,7 +156,6 @@ protected:
output = NULL;
}
/// Constructor. Configures output pipe.
FIFOProcessor(FIFOSamplePipe *pOutput ///< Output pipe.
)
@@ -161,13 +163,11 @@ protected:
output = pOutput;
}
/// Destructor.
virtual ~FIFOProcessor()
{
}
/// Returns a pointer to the beginning of the output samples.
/// This function is provided for accessing the output samples directly.
/// Please be careful for not to corrupt the book-keeping!
@@ -194,7 +194,6 @@ public:
return output->receiveSamples(outBuffer, maxSamples);
}
/// Adjusts book-keeping so that given number of samples are removed from beginning of the
/// sample buffer without copying them anywhere.
///
@@ -206,14 +205,12 @@ public:
return output->receiveSamples(maxSamples);
}
/// Returns number of samples currently available.
virtual uint numSamples() const
{
return output->numSamples();
}
/// Returns nonzero if there aren't any samples available for outputting.
virtual int isEmpty() const
{
@@ -226,7 +223,6 @@ public:
{
return output->adjustAmountOfSamples(numSamples);
}
};
}
media/libsoundtouch/src/FIRFilter.cpp
+52 -51
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@@ -2,22 +2,21 @@
///
/// General FIR digital filter routines with MMX optimization.
///
/// Note : MMX optimized functions reside in a separate, platform-specific file,
/// Notes : MMX optimized functions reside in a separate, platform-specific file,
/// e.g. 'mmx_win.cpp' or 'mmx_gcc.cpp'
///
/// This source file contains OpenMP optimizations that allow speeding up the
/// corss-correlation algorithm by executing it in several threads / CPU cores
/// in parallel. See the following article link for more detailed discussion
/// about SoundTouch OpenMP optimizations:
/// http://www.softwarecoven.com/parallel-computing-in-embedded-mobile-devices
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2015-02-21 21:24:29 +0000 (Sat, 21 Feb 2015) $
// File revision : $Revision: 4 $
//
// $Id: FIRFilter.cpp 202 2015-02-21 21:24:29Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
@@ -61,14 +60,17 @@ FIRFilter::FIRFilter()
length = 0;
lengthDiv8 = 0;
filterCoeffs = NULL;
filterCoeffsStereo = NULL;
}
FIRFilter::~FIRFilter()
{
delete[] filterCoeffs;
delete[] filterCoeffsStereo;
}
// Usual C-version of the filter routine for stereo sound
uint FIRFilter::evaluateFilterStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples) const
{
@@ -78,35 +80,26 @@ uint FIRFilter::evaluateFilterStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, ui
// because division is much slower operation than multiplying.
double dScaler = 1.0 / (double)resultDivider;
#endif
// hint compiler autovectorization that loop length is divisible by 8
int ilength = length & -8;
assert(length != 0);
assert(src != NULL);
assert(dest != NULL);
assert(filterCoeffs != NULL);
assert((length != 0) && (length == ilength) && (src != NULL) && (dest != NULL) && (filterCoeffs != NULL));
end = 2 * (numSamples - length);
end = 2 * (numSamples - ilength);
#pragma omp parallel for
for (j = 0; j < end; j += 2)
{
const SAMPLETYPE *ptr;
LONG_SAMPLETYPE suml, sumr;
uint i;
suml = sumr = 0;
ptr = src + j;
for (i = 0; i < length; i += 4)
for (int i = 0; i < ilength; i ++)
{
// loop is unrolled by factor of 4 here for efficiency
suml += ptr[2 * i + 0] * filterCoeffs[i + 0] +
ptr[2 * i + 2] * filterCoeffs[i + 1] +
ptr[2 * i + 4] * filterCoeffs[i + 2] +
ptr[2 * i + 6] * filterCoeffs[i + 3];
sumr += ptr[2 * i + 1] * filterCoeffs[i + 0] +
ptr[2 * i + 3] * filterCoeffs[i + 1] +
ptr[2 * i + 5] * filterCoeffs[i + 2] +
ptr[2 * i + 7] * filterCoeffs[i + 3];
suml += ptr[2 * i] * filterCoeffsStereo[2 * i];
sumr += ptr[2 * i + 1] * filterCoeffsStereo[2 * i + 1];
}
#ifdef SOUNDTOUCH_INTEGER_SAMPLES
@@ -116,19 +109,14 @@ uint FIRFilter::evaluateFilterStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, ui
suml = (suml < -32768) ? -32768 : (suml > 32767) ? 32767 : suml;
// saturate to 16 bit integer limits
sumr = (sumr < -32768) ? -32768 : (sumr > 32767) ? 32767 : sumr;
#else
suml *= dScaler;
sumr *= dScaler;
#endif // SOUNDTOUCH_INTEGER_SAMPLES
dest[j] = (SAMPLETYPE)suml;
dest[j + 1] = (SAMPLETYPE)sumr;
}
return numSamples - length;
return numSamples - ilength;
}
// Usual C-version of the filter routine for mono sound
uint FIRFilter::evaluateFilterMono(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples) const
{
@@ -139,31 +127,28 @@ uint FIRFilter::evaluateFilterMono(SAMPLETYPE *dest, const SAMPLETYPE *src, uint
double dScaler = 1.0 / (double)resultDivider;
#endif
assert(length != 0);
// hint compiler autovectorization that loop length is divisible by 8
int ilength = length & -8;
end = numSamples - length;
assert(ilength != 0);
end = numSamples - ilength;
#pragma omp parallel for
for (j = 0; j < end; j ++)
for (j = 0; j < end; j ++)
{
const SAMPLETYPE *pSrc = src + j;
LONG_SAMPLETYPE sum;
uint i;
int i;
sum = 0;
for (i = 0; i < length; i += 4)
for (i = 0; i < ilength; i ++)
{
// loop is unrolled by factor of 4 here for efficiency
sum += pSrc[i + 0] * filterCoeffs[i + 0] +
pSrc[i + 1] * filterCoeffs[i + 1] +
pSrc[i + 2] * filterCoeffs[i + 2] +
pSrc[i + 3] * filterCoeffs[i + 3];
sum += pSrc[i] * filterCoeffs[i];
}
#ifdef SOUNDTOUCH_INTEGER_SAMPLES
sum >>= resultDivFactor;
// saturate to 16 bit integer limits
sum = (sum < -32768) ? -32768 : (sum > 32767) ? 32767 : sum;
#else
sum *= dScaler;
#endif // SOUNDTOUCH_INTEGER_SAMPLES
dest[j] = (SAMPLETYPE)sum;
}
@@ -187,14 +172,18 @@ uint FIRFilter::evaluateFilterMulti(SAMPLETYPE *dest, const SAMPLETYPE *src, uin
assert(filterCoeffs != NULL);
assert(numChannels < 16);
end = numChannels * (numSamples - length);
// hint compiler autovectorization that loop length is divisible by 8
int ilength = length & -8;
end = numChannels * (numSamples - ilength);
#pragma omp parallel for
for (j = 0; j < end; j += numChannels)
{
const SAMPLETYPE *ptr;
LONG_SAMPLETYPE sums[16];
uint c, i;
uint c;
int i;
for (c = 0; c < numChannels; c ++)
{
@@ -203,7 +192,7 @@ uint FIRFilter::evaluateFilterMulti(SAMPLETYPE *dest, const SAMPLETYPE *src, uin
ptr = src + j;
for (i = 0; i < length; i ++)
for (i = 0; i < ilength; i ++)
{
SAMPLETYPE coef=filterCoeffs[i];
for (c = 0; c < numChannels; c ++)
@@ -217,13 +206,11 @@ uint FIRFilter::evaluateFilterMulti(SAMPLETYPE *dest, const SAMPLETYPE *src, uin
{
#ifdef SOUNDTOUCH_INTEGER_SAMPLES
sums[c] >>= resultDivFactor;
#else
sums[c] *= dScaler;
#endif // SOUNDTOUCH_INTEGER_SAMPLES
dest[j+c] = (SAMPLETYPE)sums[c];
}
}
return numSamples - length;
return numSamples - ilength;
}
@@ -235,6 +222,13 @@ void FIRFilter::setCoefficients(const SAMPLETYPE *coeffs, uint newLength, uint u
assert(newLength > 0);
if (newLength % 8) ST_THROW_RT_ERROR("FIR filter length not divisible by 8");
#ifdef SOUNDTOUCH_FLOAT_SAMPLES
// scale coefficients already here if using floating samples
double scale = 1.0 / resultDivider;
#else
short scale = 1;
#endif
lengthDiv8 = newLength / 8;
length = lengthDiv8 * 8;
assert(length == newLength);
@@ -244,7 +238,16 @@ void FIRFilter::setCoefficients(const SAMPLETYPE *coeffs, uint newLength, uint u
delete[] filterCoeffs;
filterCoeffs = new SAMPLETYPE[length];
memcpy(filterCoeffs, coeffs, length * sizeof(SAMPLETYPE));
delete[] filterCoeffsStereo;
filterCoeffsStereo = new SAMPLETYPE[length*2];
for (uint i = 0; i < length; i ++)
{
filterCoeffs[i] = (SAMPLETYPE)(coeffs[i] * scale);
// create also stereo set of filter coefficients: this allows compiler
// to autovectorize filter evaluation much more efficiently
filterCoeffsStereo[2 * i] = (SAMPLETYPE)(coeffs[i] * scale);
filterCoeffsStereo[2 * i + 1] = (SAMPLETYPE)(coeffs[i] * scale);
}
}
@@ -254,7 +257,6 @@ uint FIRFilter::getLength() const
}
// Applies the filter to the given sequence of samples.
//
// Note : The amount of outputted samples is by value of 'filter_length'
@@ -284,7 +286,6 @@ uint FIRFilter::evaluate(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSample
}
// Operator 'new' is overloaded so that it automatically creates a suitable instance
// depending on if we've a MMX-capable CPU available or not.
void * FIRFilter::operator new(size_t s)
media/libsoundtouch/src/FIRFilter.h
+1 -7
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@@ -11,13 +11,6 @@
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2015-02-21 21:24:29 +0000 (Sat, 21 Feb 2015) $
// File revision : $Revision: 4 $
//
// $Id: FIRFilter.h 202 2015-02-21 21:24:29Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
@@ -64,6 +57,7 @@ protected:
// Memory for filter coefficients
SAMPLETYPE *filterCoeffs;
SAMPLETYPE *filterCoeffsStereo;
virtual uint evaluateFilterStereo(SAMPLETYPE *dest,
const SAMPLETYPE *src,
media/libsoundtouch/src/InterpolateCubic.cpp
-4
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@@ -8,10 +8,6 @@
///
////////////////////////////////////////////////////////////////////////////////
//
// $Id: InterpolateCubic.cpp 179 2014-01-06 18:41:42Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
media/libsoundtouch/src/InterpolateCubic.h
+8 -6
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@@ -8,10 +8,6 @@
///
////////////////////////////////////////////////////////////////////////////////
//
// $Id: InterpolateCubic.h 179 2014-01-06 18:41:42Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
@@ -45,7 +41,6 @@ namespace soundtouch
class InterpolateCubic : public TransposerBase
{
protected:
virtual void resetRegisters();
virtual int transposeMono(SAMPLETYPE *dest,
const SAMPLETYPE *src,
int &srcSamples);
@@ -56,10 +51,17 @@ protected:
const SAMPLETYPE *src,
int &srcSamples);
float fract;
double fract;
public:
InterpolateCubic();
virtual void resetRegisters();
int getLatency() const
{
return 1;
}
};
}
media/libsoundtouch/src/InterpolateLinear.cpp
+8 -11
View File
@@ -8,10 +8,6 @@
///
////////////////////////////////////////////////////////////////////////////////
//
// $Id: InterpolateLinear.cpp 180 2014-01-06 19:16:02Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
@@ -146,7 +142,7 @@ int InterpolateLinearInteger::transposeMulti(SAMPLETYPE *dest, const SAMPLETYPE
LONG_SAMPLETYPE temp, vol1;
assert(iFract < SCALE);
vol1 = (SCALE - iFract);
vol1 = (LONG_SAMPLETYPE)(SCALE - iFract);
for (int c = 0; c < numChannels; c ++)
{
temp = vol1 * src[c] + iFract * src[c + numChannels];
@@ -170,9 +166,9 @@ int InterpolateLinearInteger::transposeMulti(SAMPLETYPE *dest, const SAMPLETYPE
// Sets new target iRate. Normal iRate = 1.0, smaller values represent slower
// iRate, larger faster iRates.
void InterpolateLinearInteger::setRate(float newRate)
void InterpolateLinearInteger::setRate(double newRate)
{
iRate = (int)(newRate * SCALE + 0.5f);
iRate = (int)(newRate * SCALE + 0.5);
TransposerBase::setRate(newRate);
}
@@ -190,7 +186,7 @@ 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);
setRate(1.0);
}
@@ -275,12 +271,13 @@ int InterpolateLinearFloat::transposeMulti(SAMPLETYPE *dest, const SAMPLETYPE *s
i = 0;
while (srcCount < srcSampleEnd)
{
float temp, vol1;
float temp, vol1, fract_float;
vol1 = (1.0f- fract);
vol1 = (float)(1.0 - fract);
fract_float = (float)fract;
for (int c = 0; c < numChannels; c ++)
{
temp = vol1 * src[c] + fract * src[c + numChannels];
temp = vol1 * src[c] + fract_float * src[c + numChannels];
*dest = (SAMPLETYPE)temp;
dest ++;
}
media/libsoundtouch/src/InterpolateLinear.h
+18 -12
View File
@@ -8,10 +8,6 @@
///
////////////////////////////////////////////////////////////////////////////////
//
// $Id: InterpolateLinear.h 179 2014-01-06 18:41:42Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
@@ -42,15 +38,13 @@
namespace soundtouch
{
/// Linear transposer class that uses integer arithmetics
/// Linear transposer class that uses integer arithmetic
class InterpolateLinearInteger : public TransposerBase
{
protected:
int iFract;
int iRate;
virtual void resetRegisters();
virtual int transposeMono(SAMPLETYPE *dest,
const SAMPLETYPE *src,
int &srcSamples);
@@ -63,17 +57,22 @@ public:
/// Sets new target rate. Normal rate = 1.0, smaller values represent slower
/// rate, larger faster rates.
virtual void setRate(float newRate);
virtual void setRate(double newRate);
virtual void resetRegisters();
int getLatency() const
{
return 0;
}
};
/// Linear transposer class that uses floating point arithmetics
/// Linear transposer class that uses floating point arithmetic
class InterpolateLinearFloat : public TransposerBase
{
protected:
float fract;
virtual void resetRegisters();
double fract;
virtual int transposeMono(SAMPLETYPE *dest,
const SAMPLETYPE *src,
@@ -85,6 +84,13 @@ protected:
public:
InterpolateLinearFloat();
virtual void resetRegisters();
int getLatency() const
{
return 0;
}
};
}
media/libsoundtouch/src/InterpolateShannon.cpp
-4
View File
@@ -13,10 +13,6 @@
///
////////////////////////////////////////////////////////////////////////////////
//
// $Id: InterpolateShannon.cpp 195 2014-04-06 15:57:21Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
media/libsoundtouch/src/InterpolateShannon.h
+8 -6
View File
@@ -13,10 +13,6 @@
///
////////////////////////////////////////////////////////////////////////////////
//
// $Id: InterpolateShannon.h 179 2014-01-06 18:41:42Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
@@ -50,7 +46,6 @@ namespace soundtouch
class InterpolateShannon : public TransposerBase
{
protected:
void resetRegisters();
int transposeMono(SAMPLETYPE *dest,
const SAMPLETYPE *src,
int &srcSamples);
@@ -61,10 +56,17 @@ protected:
const SAMPLETYPE *src,
int &srcSamples);
float fract;
double fract;
public:
InterpolateShannon();
void resetRegisters();
int getLatency() const
{
return 3;
}
};
}
media/libsoundtouch/src/RateTransposer.cpp
+34 -21
View File
@@ -10,13 +10,6 @@
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2014-04-06 15:57:21 +0000 (Sun, 06 Apr 2014) $
// File revision : $Revision: 4 $
//
// $Id: RateTransposer.cpp 195 2014-04-06 15:57:21Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
@@ -57,15 +50,21 @@ TransposerBase::ALGORITHM TransposerBase::algorithm = TransposerBase::CUBIC;
// Constructor
RateTransposer::RateTransposer() : FIFOProcessor(&outputBuffer)
{
bUseAAFilter = true;
bUseAAFilter =
#ifndef SOUNDTOUCH_PREVENT_CLICK_AT_RATE_CROSSOVER
true;
#else
// Disable Anti-alias filter if desirable to avoid click at rate change zero value crossover
false;
#endif
// Instantiates the anti-alias filter
pAAFilter = new AAFilter(64);
pTransposer = TransposerBase::newInstance();
clear();
}
RateTransposer::~RateTransposer()
{
delete pAAFilter;
@@ -73,11 +72,14 @@ RateTransposer::~RateTransposer()
}
/// Enables/disables the anti-alias filter. Zero to disable, nonzero to enable
void RateTransposer::enableAAFilter(bool newMode)
{
#ifndef SOUNDTOUCH_PREVENT_CLICK_AT_RATE_CROSSOVER
// Disable Anti-alias filter if desirable to avoid click at rate change zero value crossover
bUseAAFilter = newMode;
clear();
#endif
}
@@ -94,23 +96,22 @@ AAFilter *RateTransposer::getAAFilter()
}
// Sets new target iRate. Normal iRate = 1.0, smaller values represent slower
// iRate, larger faster iRates.
void RateTransposer::setRate(float newRate)
void RateTransposer::setRate(double newRate)
{
double fCutoff;
pTransposer->setRate(newRate);
// design a new anti-alias filter
if (newRate > 1.0f)
if (newRate > 1.0)
{
fCutoff = 0.5f / newRate;
fCutoff = 0.5 / newRate;
}
else
{
fCutoff = 0.5f * newRate;
fCutoff = 0.5 * newRate;
}
pAAFilter->setCutoffFreq(fCutoff);
}
@@ -177,11 +178,10 @@ void RateTransposer::processSamples(const SAMPLETYPE *src, uint nSamples)
// Sets the number of channels, 1 = mono, 2 = stereo
void RateTransposer::setChannels(int nChannels)
{
assert(nChannels > 0);
if (!verifyNumberOfChannels(nChannels) ||
(pTransposer->numChannels == nChannels)) return;
if (pTransposer->numChannels == nChannels) return;
pTransposer->setChannels(nChannels);
inputBuffer.setChannels(nChannels);
midBuffer.setChannels(nChannels);
outputBuffer.setChannels(nChannels);
@@ -194,6 +194,11 @@ void RateTransposer::clear()
outputBuffer.clear();
midBuffer.clear();
inputBuffer.clear();
pTransposer->resetRegisters();
// prefill buffer to avoid losing first samples at beginning of stream
int prefill = getLatency();
inputBuffer.addSilent(prefill);
}
@@ -208,6 +213,14 @@ int RateTransposer::isEmpty() const
}
/// Return approximate initial input-output latency
int RateTransposer::getLatency() const
{
return pTransposer->getLatency() +
((bUseAAFilter) ? (pAAFilter->getLength() / 2) : 0);
}
//////////////////////////////////////////////////////////////////////////////
//
// TransposerBase - Base class for interpolation
@@ -225,7 +238,7 @@ void TransposerBase::setAlgorithm(TransposerBase::ALGORITHM a)
int TransposerBase::transpose(FIFOSampleBuffer &dest, FIFOSampleBuffer &src)
{
int numSrcSamples = src.numSamples();
int sizeDemand = (int)((float)numSrcSamples / rate) + 8;
int sizeDemand = (int)((double)numSrcSamples / rate) + 8;
int numOutput;
SAMPLETYPE *psrc = src.ptrBegin();
SAMPLETYPE *pdest = dest.ptrEnd(sizeDemand);
@@ -270,7 +283,7 @@ void TransposerBase::setChannels(int channels)
}
void TransposerBase::setRate(float newRate)
void TransposerBase::setRate(double newRate)
{
rate = newRate;
}
@@ -280,7 +293,7 @@ void TransposerBase::setRate(float newRate)
TransposerBase *TransposerBase::newInstance()
{
#ifdef SOUNDTOUCH_INTEGER_SAMPLES
// Notice: For integer arithmetics support only linear algorithm (due to simplest calculus)
// Notice: For integer arithmetic support only linear algorithm (due to simplest calculus)
return ::new InterpolateLinearInteger;
#else
switch (algorithm)
media/libsoundtouch/src/RateTransposer.h
+9 -24
View File
@@ -14,13 +14,6 @@
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2014-04-06 15:57:21 +0000 (Sun, 06 Apr 2014) $
// File revision : $Revision: 4 $
//
// $Id: RateTransposer.h 195 2014-04-06 15:57:21Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
@@ -66,8 +59,6 @@ public:
};
protected:
virtual void resetRegisters() = 0;
virtual int transposeMono(SAMPLETYPE *dest,
const SAMPLETYPE *src,
int &srcSamples) = 0;
@@ -81,15 +72,18 @@ protected:
static ALGORITHM algorithm;
public:
float rate;
double rate;
int numChannels;
TransposerBase();
virtual ~TransposerBase();
virtual int transpose(FIFOSampleBuffer &dest, FIFOSampleBuffer &src);
virtual void setRate(float newRate);
virtual void setRate(double newRate);
virtual void setChannels(int channels);
virtual int getLatency() const = 0;
virtual void resetRegisters() = 0;
// static factory function
static TransposerBase *newInstance();
@@ -132,21 +126,9 @@ public:
RateTransposer();
virtual ~RateTransposer();
/// Operator 'new' is overloaded so that it automatically creates a suitable instance
/// depending on if we're to use integer or floating point arithmetics.
// static void *operator new(size_t s);
/// Use this function instead of "new" operator to create a new instance of this class.
/// This function automatically chooses a correct implementation, depending on if
/// integer ot floating point arithmetics are to be used.
// static RateTransposer *newInstance();
/// Returns the output buffer object
FIFOSamplePipe *getOutput() { return &outputBuffer; };
/// Returns the store buffer object
// FIFOSamplePipe *getStore() { return &storeBuffer; };
/// Return anti-alias filter object
AAFilter *getAAFilter();
@@ -158,7 +140,7 @@ public:
/// Sets new target rate. Normal rate = 1.0, smaller values represent slower
/// rate, larger faster rates.
virtual void setRate(float newRate);
virtual void setRate(double newRate);
/// Sets the number of channels, 1 = mono, 2 = stereo
void setChannels(int channels);
@@ -172,6 +154,9 @@ public:
/// Returns nonzero if there aren't any samples available for outputting.
int isEmpty() const;
/// Return approximate initial input-output latency
int getLatency() const;
};
}
media/libsoundtouch/src/STTypes.h
+28 -19
View File
@@ -8,13 +8,6 @@
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2015-05-18 15:25:07 +0000 (Mon, 18 May 2015) $
// File revision : $Revision: 3 $
//
// $Id: STTypes.h 215 2015-05-18 15:25:07Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
@@ -57,13 +50,20 @@ typedef unsigned long ulong;
#include "soundtouch_config.h"
#if defined(WIN32)
#define EXPORT __declspec(dllexport)
#if defined(BUILDING_SOUNDTOUCH)
#define SOUNDTOUCH_API __declspec(dllexport)
#else
#define EXPORT
#define SOUNDTOUCH_API __declspec(dllimport)
#endif
#else
#define SOUNDTOUCH_API
#endif
namespace soundtouch
{
/// Max allowed number of channels
#define SOUNDTOUCH_MAX_CHANNELS 16
/// Activate these undef's to overrule the possible sampletype
/// setting inherited from some other header file:
//#undef SOUNDTOUCH_INTEGER_SAMPLES
@@ -126,10 +126,10 @@ namespace soundtouch
#endif
// If defined, allows the SIMD-optimized routines to take minor shortcuts
// for improved performance. Undefine to require faithfully similar SIMD
// calculations as in normal C implementation.
#define SOUNDTOUCH_ALLOW_NONEXACT_SIMD_OPTIMIZATION 1
// If defined, allows the SIMD-optimized routines to skip unevenly aligned
// memory offsets that can cause performance penalty in some SIMD implementations.
// Causes slight compromise in sound quality.
// #define SOUNDTOUCH_ALLOW_NONEXACT_SIMD_OPTIMIZATION 1
#ifdef SOUNDTOUCH_INTEGER_SAMPLES
@@ -144,16 +144,19 @@ namespace soundtouch
#endif // SOUNDTOUCH_FLOAT_SAMPLES
#ifdef SOUNDTOUCH_ALLOW_X86_OPTIMIZATIONS
// Allow MMX optimizations
#define SOUNDTOUCH_ALLOW_MMX 1
// Allow MMX optimizations (not available in X64 mode)
#if (!_M_X64)
#define SOUNDTOUCH_ALLOW_MMX 1
#endif
#endif
#else
// floating point samples
typedef float SAMPLETYPE;
// data type for sample accumulation: Use double to utilize full precision.
typedef double LONG_SAMPLETYPE;
// data type for sample accumulation: Use float also here to enable
// efficient autovectorization
typedef float LONG_SAMPLETYPE;
#ifdef SOUNDTOUCH_ALLOW_X86_OPTIMIZATIONS
// Allow SSE optimizations
@@ -162,10 +165,16 @@ namespace soundtouch
#endif // SOUNDTOUCH_INTEGER_SAMPLES
};
#if ((SOUNDTOUCH_ALLOW_SSE) || (__SSE__) || (SOUNDTOUCH_USE_NEON))
#if SOUNDTOUCH_ALLOW_NONEXACT_SIMD_OPTIMIZATION
#define ST_SIMD_AVOID_UNALIGNED
#endif
#endif
}
// define ST_NO_EXCEPTION_HANDLING switch to disable throwing std exceptions:
// #define ST_NO_EXCEPTION_HANDLING 1
#define ST_NO_EXCEPTION_HANDLING 1
#ifdef ST_NO_EXCEPTION_HANDLING
// Exceptions disabled. Throw asserts instead if enabled.
#include <assert.h>
media/libsoundtouch/src/SoundTouch.cpp
+121 -87
View File
@@ -41,13 +41,6 @@
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2014-10-08 15:26:57 +0000 (Wed, 08 Oct 2014) $
// File revision : $Revision: 4 $
//
// $Id: SoundTouch.cpp 201 2014-10-08 15:26:57Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
@@ -110,12 +103,14 @@ SoundTouch::SoundTouch()
calcEffectiveRateAndTempo();
samplesExpectedOut = 0;
samplesOutput = 0;
channels = 0;
bSrateSet = false;
}
SoundTouch::~SoundTouch()
{
delete pRateTransposer;
@@ -123,7 +118,6 @@ SoundTouch::~SoundTouch()
}
/// Get SoundTouch library version string
const char *SoundTouch::getVersionString()
{
@@ -143,90 +137,79 @@ uint SoundTouch::getVersionId()
// Sets the number of channels, 1 = mono, 2 = stereo
void SoundTouch::setChannels(uint numChannels)
{
/*if (numChannels != 1 && numChannels != 2)
{
//ST_THROW_RT_ERROR("Illegal number of channels");
return;
}*/
if (!verifyNumberOfChannels(numChannels)) return;
channels = numChannels;
pRateTransposer->setChannels((int)numChannels);
pTDStretch->setChannels((int)numChannels);
}
// Sets new rate control value. Normal rate = 1.0, smaller values
// represent slower rate, larger faster rates.
void SoundTouch::setRate(float newRate)
void SoundTouch::setRate(double newRate)
{
virtualRate = newRate;
calcEffectiveRateAndTempo();
}
// Sets new rate control value as a difference in percents compared
// to the original rate (-50 .. +100 %)
void SoundTouch::setRateChange(float newRate)
void SoundTouch::setRateChange(double newRate)
{
virtualRate = 1.0f + 0.01f * newRate;
virtualRate = 1.0 + 0.01 * newRate;
calcEffectiveRateAndTempo();
}
// Sets new tempo control value. Normal tempo = 1.0, smaller values
// represent slower tempo, larger faster tempo.
void SoundTouch::setTempo(float newTempo)
void SoundTouch::setTempo(double newTempo)
{
virtualTempo = newTempo;
calcEffectiveRateAndTempo();
}
// Sets new tempo control value as a difference in percents compared
// to the original tempo (-50 .. +100 %)
void SoundTouch::setTempoChange(float newTempo)
void SoundTouch::setTempoChange(double newTempo)
{
virtualTempo = 1.0f + 0.01f * newTempo;
virtualTempo = 1.0 + 0.01 * newTempo;
calcEffectiveRateAndTempo();
}
// Sets new pitch control value. Original pitch = 1.0, smaller values
// represent lower pitches, larger values higher pitch.
void SoundTouch::setPitch(float newPitch)
void SoundTouch::setPitch(double newPitch)
{
virtualPitch = newPitch;
calcEffectiveRateAndTempo();
}
// Sets pitch change in octaves compared to the original pitch
// (-1.00 .. +1.00)
void SoundTouch::setPitchOctaves(float newPitch)
void SoundTouch::setPitchOctaves(double newPitch)
{
virtualPitch = (float)exp(0.69314718056f * newPitch);
virtualPitch = exp(0.69314718056 * newPitch);
calcEffectiveRateAndTempo();
}
// Sets pitch change in semi-tones compared to the original pitch
// (-12 .. +12)
void SoundTouch::setPitchSemiTones(int newPitch)
{
setPitchOctaves((float)newPitch / 12.0f);
setPitchOctaves((double)newPitch / 12.0);
}
void SoundTouch::setPitchSemiTones(float newPitch)
void SoundTouch::setPitchSemiTones(double newPitch)
{
setPitchOctaves(newPitch / 12.0f);
setPitchOctaves(newPitch / 12.0);
}
@@ -234,8 +217,8 @@ void SoundTouch::setPitchSemiTones(float newPitch)
// nominal control values.
void SoundTouch::calcEffectiveRateAndTempo()
{
float oldTempo = tempo;
float oldRate = rate;
double oldTempo = tempo;
double oldRate = rate;
tempo = virtualTempo / virtualPitch;
rate = virtualPitch * virtualRate;
@@ -302,23 +285,12 @@ void SoundTouch::putSamples(const SAMPLETYPE *samples, uint nSamples)
ST_THROW_RT_ERROR("SoundTouch : Number of channels not defined");
}
// Transpose the rate of the new samples if necessary
/* Bypass the nominal setting - can introduce a click in sound when tempo/pitch control crosses the nominal value...
if (rate == 1.0f)
{
// The rate value is same as the original, simply evaluate the tempo changer.
assert(output == pTDStretch);
if (pRateTransposer->isEmpty() == 0)
{
// yet flush the last samples in the pitch transposer buffer
// (may happen if 'rate' changes from a non-zero value to zero)
pTDStretch->moveSamples(*pRateTransposer);
}
pTDStretch->putSamples(samples, nSamples);
}
*/
// accumulate how many samples are expected out from processing, given the current
// processing setting
samplesExpectedOut += (double)nSamples / ((double)rate * (double)tempo);
#ifndef SOUNDTOUCH_PREVENT_CLICK_AT_RATE_CROSSOVER
else if (rate <= 1.0f)
if (rate <= 1.0f)
{
// transpose the rate down, output the transposed sound to tempo changer buffer
assert(output == pTDStretch);
@@ -346,44 +318,30 @@ void SoundTouch::putSamples(const SAMPLETYPE *samples, uint nSamples)
void SoundTouch::flush()
{
int i;
int nUnprocessed;
int nOut;
SAMPLETYPE *buff = new SAMPLETYPE[64 * channels];
// check how many samples still await processing, and scale
// that by tempo & rate to get expected output sample count
nUnprocessed = numUnprocessedSamples();
nUnprocessed = (int)((double)nUnprocessed / (tempo * rate) + 0.5);
int numStillExpected;
SAMPLETYPE *buff = new SAMPLETYPE[128 * channels];
nOut = numSamples(); // ready samples currently in buffer ...
nOut += nUnprocessed; // ... and how many we expect there to be in the end
memset(buff, 0, 64 * channels * sizeof(SAMPLETYPE));
// how many samples are still expected to output
numStillExpected = (int)((long)(samplesExpectedOut + 0.5) - samplesOutput);
if (numStillExpected < 0) numStillExpected = 0;
memset(buff, 0, 128 * channels * sizeof(SAMPLETYPE));
// "Push" the last active samples out from the processing pipeline by
// feeding blank samples into the processing pipeline until new,
// processed samples appear in the output (not however, more than
// 8ksamples in any case)
for (i = 0; i < 128; i ++)
// 24ksamples in any case)
for (i = 0; (numStillExpected > (int)numSamples()) && (i < 200); i ++)
{
putSamples(buff, 64);
if ((int)numSamples() >= nOut)
{
// Enough new samples have appeared into the output!
// As samples come from processing with bigger chunks, now truncate it
// back to maximum "nOut" samples to improve duration accuracy
adjustAmountOfSamples(nOut);
// finish
break;
}
putSamples(buff, 128);
}
adjustAmountOfSamples(numStillExpected);
delete[] buff;
// Clear working buffers
pRateTransposer->clear();
// Clear input buffers
pTDStretch->clearInput();
// yet leave the 'tempoChanger' output intouched as that's where the
// yet leave the output intouched as that's where the
// flushed samples are!
}
@@ -452,7 +410,7 @@ int SoundTouch::getSetting(int settingId) const
return pRateTransposer->getAAFilter()->getLength();
case SETTING_USE_QUICKSEEK :
return (uint) pTDStretch->isQuickSeekEnabled();
return (uint)pTDStretch->isQuickSeekEnabled();
case SETTING_SEQUENCE_MS:
pTDStretch->getParameters(NULL, &temp, NULL, NULL);
@@ -466,13 +424,53 @@ int SoundTouch::getSetting(int settingId) const
pTDStretch->getParameters(NULL, NULL, NULL, &temp);
return temp;
case SETTING_NOMINAL_INPUT_SEQUENCE :
return pTDStretch->getInputSampleReq();
case SETTING_NOMINAL_INPUT_SEQUENCE :
{
int size = pTDStretch->getInputSampleReq();
case SETTING_NOMINAL_OUTPUT_SEQUENCE :
return pTDStretch->getOutputBatchSize();
#ifndef SOUNDTOUCH_PREVENT_CLICK_AT_RATE_CROSSOVER
if (rate <= 1.0)
{
// transposing done before timestretch, which impacts latency
return (int)(size * rate + 0.5);
}
#endif
return size;
}
default :
case SETTING_NOMINAL_OUTPUT_SEQUENCE :
{
int size = pTDStretch->getOutputBatchSize();
if (rate > 1.0)
{
// transposing done after timestretch, which impacts latency
return (int)(size / rate + 0.5);
}
return size;
}
case SETTING_INITIAL_LATENCY:
{
double latency = pTDStretch->getLatency();
int latency_tr = pRateTransposer->getLatency();
#ifndef SOUNDTOUCH_PREVENT_CLICK_AT_RATE_CROSSOVER
if (rate <= 1.0)
{
// transposing done before timestretch, which impacts latency
latency = (latency + latency_tr) * rate;
}
else
#endif
{
latency += (double)latency_tr / rate;
}
return (int)(latency + 0.5);
}
default :
return 0;
}
}
@@ -482,12 +480,13 @@ int SoundTouch::getSetting(int settingId) const
// buffers.
void SoundTouch::clear()
{
samplesExpectedOut = 0;
samplesOutput = 0;
pRateTransposer->clear();
pTDStretch->clear();
}
/// Returns number of samples currently unprocessed.
uint SoundTouch::numUnprocessedSamples() const
{
@@ -502,3 +501,38 @@ uint SoundTouch::numUnprocessedSamples() const
}
return 0;
}
/// Output samples from beginning of the sample buffer. Copies requested samples to
/// output buffer and removes them from the sample buffer. If there are less than
/// 'numsample' samples in the buffer, returns all that available.
///
/// \return Number of samples returned.
uint SoundTouch::receiveSamples(SAMPLETYPE *output, uint maxSamples)
{
uint ret = FIFOProcessor::receiveSamples(output, maxSamples);
samplesOutput += (long)ret;
return ret;
}
/// Adjusts book-keeping so that given number of samples are removed from beginning of the
/// sample buffer without copying them anywhere.
///
/// Used to reduce the number of samples in the buffer when accessing the sample buffer directly
/// with 'ptrBegin' function.
uint SoundTouch::receiveSamples(uint maxSamples)
{
uint ret = FIFOProcessor::receiveSamples(maxSamples);
samplesOutput += (long)ret;
return ret;
}
/// Get ratio between input and output audio durations, useful for calculating
/// processed output duration: if you'll process a stream of N samples, then
/// you can expect to get out N * getInputOutputSampleRatio() samples.
double SoundTouch::getInputOutputSampleRatio()
{
return 1.0 / (tempo * rate);
}
media/libsoundtouch/src/SoundTouch.h
+106 -35
View File
@@ -41,13 +41,6 @@
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2015-05-18 15:28:41 +0000 (Mon, 18 May 2015) $
// File revision : $Revision: 4 $
//
// $Id: SoundTouch.h 216 2015-05-18 15:28:41Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
@@ -79,10 +72,10 @@ namespace soundtouch
{
/// Soundtouch library version string
#define SOUNDTOUCH_VERSION "1.9.0"
#define SOUNDTOUCH_VERSION "2.3.1"
/// SoundTouch library version id
#define SOUNDTOUCH_VERSION_ID (10900)
#define SOUNDTOUCH_VERSION_ID (20301)
//
// Available setting IDs for the 'setSetting' & 'get_setting' functions:
@@ -116,32 +109,63 @@ namespace soundtouch
#define SETTING_OVERLAP_MS 5
/// Call "getSetting" with this ID to query nominal average processing sequence
/// size in samples. This value tells approcimate value how many input samples
/// SoundTouch needs to gather before it does DSP processing run for the sample batch.
/// Call "getSetting" with this ID to query processing sequence size in samples.
/// This value gives approximate value of how many input samples you'll need to
/// feed into SoundTouch after initial buffering to get out a new batch of
/// output samples.
///
/// This value does not include initial buffering at beginning of a new processing
/// stream, use SETTING_INITIAL_LATENCY to get the initial buffering size.
///
/// Notices:
/// - This is read-only parameter, i.e. setSetting ignores this parameter
/// - Returned value is approximate average value, exact processing batch
/// size may wary from time to time
/// - This parameter value is not constant but may change depending on
/// - This parameter value is not constant but change depending on
/// tempo/pitch/rate/samplerate settings.
#define SETTING_NOMINAL_INPUT_SEQUENCE 6
#define SETTING_NOMINAL_INPUT_SEQUENCE 6
/// Call "getSetting" with this ID to query nominal average processing output
/// size in samples. This value tells approcimate value how many output samples
/// SoundTouch outputs once it does DSP processing run for a batch of input samples.
///
///
/// Notices:
/// - This is read-only parameter, i.e. setSetting ignores this parameter
/// - Returned value is approximate average value, exact processing batch
/// size may wary from time to time
/// - This parameter value is not constant but may change depending on
/// - This parameter value is not constant but change depending on
/// tempo/pitch/rate/samplerate settings.
#define SETTING_NOMINAL_OUTPUT_SEQUENCE 7
#define SETTING_NOMINAL_OUTPUT_SEQUENCE 7
class EXPORT SoundTouch : public FIFOProcessor
/// Call "getSetting" with this ID to query initial processing latency, i.e.
/// approx. how many samples you'll need to enter to SoundTouch pipeline before
/// you can expect to get first batch of ready output samples out.
///
/// After the first output batch, you can then expect to get approx.
/// SETTING_NOMINAL_OUTPUT_SEQUENCE ready samples out for every
/// SETTING_NOMINAL_INPUT_SEQUENCE samples that you enter into SoundTouch.
///
/// Example:
/// processing with parameter -tempo=5
/// => initial latency = 5509 samples
/// input sequence = 4167 samples
/// output sequence = 3969 samples
///
/// Accordingly, you can expect to feed in approx. 5509 samples at beginning of
/// the stream, and then you'll get out the first 3969 samples. After that, for
/// every approx. 4167 samples that you'll put in, you'll receive again approx.
/// 3969 samples out.
///
/// This also means that average latency during stream processing is
/// INITIAL_LATENCY-OUTPUT_SEQUENCE/2, in the above example case 5509-3969/2
/// = 3524 samples
///
/// Notices:
/// - This is read-only parameter, i.e. setSetting ignores this parameter
/// - This parameter value is not constant but change depending on
/// tempo/pitch/rate/samplerate settings.
#define SETTING_INITIAL_LATENCY 8
class SOUNDTOUCH_API SoundTouch : public FIFOProcessor
{
private:
/// Rate transposer class instance
@@ -151,17 +175,24 @@ private:
class TDStretch *pTDStretch;
/// Virtual pitch parameter. Effective rate & tempo are calculated from these parameters.
float virtualRate;
double virtualRate;
/// Virtual pitch parameter. Effective rate & tempo are calculated from these parameters.
float virtualTempo;
double virtualTempo;
/// Virtual pitch parameter. Effective rate & tempo are calculated from these parameters.
float virtualPitch;
double virtualPitch;
/// Flag: Has sample rate been set?
bool bSrateSet;
/// Accumulator for how many samples in total will be expected as output vs. samples put in,
/// considering current processing settings.
double samplesExpectedOut;
/// Accumulator for how many samples in total have been read out from the processing so far
long samplesOutput;
/// Calculates effective rate & tempo valuescfrom 'virtualRate', 'virtualTempo' and
/// 'virtualPitch' parameters.
void calcEffectiveRateAndTempo();
@@ -171,10 +202,10 @@ protected :
uint channels;
/// Effective 'rate' value calculated from 'virtualRate', 'virtualTempo' and 'virtualPitch'
float rate;
double rate;
/// Effective 'tempo' value calculated from 'virtualRate', 'virtualTempo' and 'virtualPitch'
float tempo;
double tempo;
public:
SoundTouch();
@@ -188,32 +219,32 @@ public:
/// Sets new rate control value. Normal rate = 1.0, smaller values
/// represent slower rate, larger faster rates.
void setRate(float newRate);
void setRate(double newRate);
/// Sets new tempo control value. Normal tempo = 1.0, smaller values
/// represent slower tempo, larger faster tempo.
void setTempo(float newTempo);
void setTempo(double newTempo);
/// Sets new rate control value as a difference in percents compared
/// to the original rate (-50 .. +100 %)
void setRateChange(float newRate);
void setRateChange(double newRate);
/// Sets new tempo control value as a difference in percents compared
/// to the original tempo (-50 .. +100 %)
void setTempoChange(float newTempo);
void setTempoChange(double newTempo);
/// Sets new pitch control value. Original pitch = 1.0, smaller values
/// represent lower pitches, larger values higher pitch.
void setPitch(float newPitch);
void setPitch(double newPitch);
/// Sets pitch change in octaves compared to the original pitch
/// (-1.00 .. +1.00)
void setPitchOctaves(float newPitch);
void setPitchOctaves(double newPitch);
/// Sets pitch change in semi-tones compared to the original pitch
/// (-12 .. +12)
void setPitchSemiTones(int newPitch);
void setPitchSemiTones(float newPitch);
void setPitchSemiTones(double newPitch);
/// Sets the number of channels, 1 = mono, 2 = stereo
void setChannels(uint numChannels);
@@ -221,6 +252,24 @@ public:
/// Sets sample rate.
void setSampleRate(uint srate);
/// Get ratio between input and output audio durations, useful for calculating
/// processed output duration: if you'll process a stream of N samples, then
/// you can expect to get out N * getInputOutputSampleRatio() samples.
///
/// This ratio will give accurate target duration ratio for a full audio track,
/// given that the the whole track is processed with same processing parameters.
///
/// If this ratio is applied to calculate intermediate offsets inside a processing
/// stream, then this ratio is approximate and can deviate +- some tens of milliseconds
/// from ideal offset, yet by end of the audio stream the duration ratio will become
/// exact.
///
/// Example: if processing with parameters "-tempo=15 -pitch=-3", the function
/// will return value 0.8695652... Now, if processing an audio stream whose duration
/// is exactly one million audio samples, then you can expect the processed
/// output duration be 0.869565 * 1000000 = 869565 samples.
double getInputOutputSampleRatio();
/// Flushes the last samples from the processing pipeline to the output.
/// Clears also the internal processing buffers.
//
@@ -240,6 +289,23 @@ public:
///< contains data for both channels.
);
/// Output samples from beginning of the sample buffer. Copies requested samples to
/// output buffer and removes them from the sample buffer. If there are less than
/// 'numsample' samples in the buffer, returns all that available.
///
/// \return Number of samples returned.
virtual uint receiveSamples(SAMPLETYPE *output, ///< Buffer where to copy output samples.
uint maxSamples ///< How many samples to receive at max.
);
/// Adjusts book-keeping so that given number of samples are removed from beginning of the
/// sample buffer without copying them anywhere.
///
/// Used to reduce the number of samples in the buffer when accessing the sample buffer directly
/// with 'ptrBegin' function.
virtual uint receiveSamples(uint maxSamples ///< Remove this many samples from the beginning of pipe.
);
/// Clears all the samples in the object's output and internal processing
/// buffers.
virtual void clear();
@@ -247,7 +313,7 @@ public:
/// Changes a setting controlling the processing system behaviour. See the
/// 'SETTING_...' defines for available setting ID's.
///
/// \return 'true' if the setting was succesfully changed
/// \return 'true' if the setting was successfully changed
bool setSetting(int settingId, ///< Setting ID number. see SETTING_... defines.
int value ///< New setting value.
);
@@ -262,6 +328,11 @@ public:
/// Returns number of samples currently unprocessed.
virtual uint numUnprocessedSamples() const;
/// Return number of channels
uint numChannels() const
{
return channels;
}
/// Other handy functions that are implemented in the ancestor classes (see
/// classes 'FIFOProcessor' and 'FIFOSamplePipe')
media/libsoundtouch/src/SoundTouchFactory.cpp
+2 -2
View File
@@ -9,14 +9,14 @@
namespace soundtouch
{
EXPORT
SOUNDTOUCH_API
soundtouch::SoundTouch*
createSoundTouchObj()
{
return new soundtouch::SoundTouch();
}
EXPORT
SOUNDTOUCH_API
void
destroySoundTouchObj(soundtouch::SoundTouch* aObj)
{
media/libsoundtouch/src/SoundTouchFactory.h
+2 -2
View File
@@ -12,11 +12,11 @@
namespace soundtouch
{
EXPORT
SOUNDTOUCH_API
soundtouch::SoundTouch*
createSoundTouchObj();
EXPORT
SOUNDTOUCH_API
void
destroySoundTouchObj(soundtouch::SoundTouch* aObj);
}
media/libsoundtouch/src/TDStretch.cpp
+288 -151
View File
@@ -1,11 +1,17 @@
////////////////////////////////////////////////////////////////////////////////
///////////////////////////////////////////////////////////////////////////////
///
/// Sampled sound tempo changer/time stretch algorithm. Changes the sound tempo
/// while maintaining the original pitch by using a time domain WSOLA-like
/// method with several performance-increasing tweaks.
///
/// Note : MMX optimized functions reside in a separate, platform-specific
/// file, e.g. 'mmx_win.cpp' or 'mmx_gcc.cpp'
/// Notes : MMX optimized functions reside in a separate, platform-specific
/// file, e.g. 'mmx_win.cpp' or 'mmx_gcc.cpp'.
///
/// This source file contains OpenMP optimizations that allow speeding up the
/// corss-correlation algorithm by executing it in several threads / CPU cores
/// in parallel. See the following article link for more detailed discussion
/// about SoundTouch OpenMP optimizations:
/// http://www.softwarecoven.com/parallel-computing-in-embedded-mobile-devices
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
@@ -13,13 +19,6 @@
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2015-02-22 15:07:12 +0000 (Sun, 22 Feb 2015) $
// File revision : $Revision: 1.12 $
//
// $Id: TDStretch.cpp 205 2015-02-22 15:07:12Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
@@ -55,7 +54,6 @@ using namespace soundtouch;
#define max(x, y) (((x) > (y)) ? (x) : (y))
/*****************************************************************************
*
* Constant definitions
@@ -63,7 +61,7 @@ using namespace soundtouch;
*****************************************************************************/
// Table for the hierarchical mixing position seeking algorithm
static const short _scanOffsets[5][24]={
const short _scanOffsets[5][24]={
{ 124, 186, 248, 310, 372, 434, 496, 558, 620, 682, 744, 806,
868, 930, 992, 1054, 1116, 1178, 1240, 1302, 1364, 1426, 1488, 0},
{-100, -75, -50, -25, 25, 50, 75, 100, 0, 0, 0, 0,
@@ -94,9 +92,6 @@ TDStretch::TDStretch() : FIFOProcessor(&outputBuffer)
bAutoSeqSetting = true;
bAutoSeekSetting = true;
// outDebt = 0;
skipFract = 0;
tempo = 1.0f;
setParameters(44100, DEFAULT_SEQUENCE_MS, DEFAULT_SEEKWINDOW_MS, DEFAULT_OVERLAP_MS);
setTempo(1.0f);
@@ -202,7 +197,7 @@ void TDStretch::overlapMono(SAMPLETYPE *pOutput, const SAMPLETYPE *pInput) const
m1 = (SAMPLETYPE)0;
m2 = (SAMPLETYPE)overlapLength;
for (i = 0; i < overlapLength ; i ++)
for (i = 0; i < overlapLength ; i ++)
{
pOutput[i] = (pInput[i] * m1 + pMidBuffer[i] * m2 ) / overlapLength;
m1 += 1;
@@ -222,6 +217,10 @@ void TDStretch::clearInput()
{
inputBuffer.clear();
clearMidBuffer();
isBeginning = true;
maxnorm = 0;
maxnormf = 1e8;
skipFract = 0;
}
@@ -255,7 +254,7 @@ int TDStretch::seekBestOverlapPosition(const SAMPLETYPE *refPos)
if (bQuickSeek)
{
return seekBestOverlapPositionQuick(refPos);
}
}
else
{
return seekBestOverlapPositionFull(refPos);
@@ -287,7 +286,6 @@ inline void TDStretch::overlap(SAMPLETYPE *pOutput, const SAMPLETYPE *pInput, ui
}
// Seeks for the optimal overlap-mixing position. The 'stereo' version of the
// routine
//
@@ -301,21 +299,23 @@ int TDStretch::seekBestOverlapPositionFull(const SAMPLETYPE *refPos)
int i;
double norm;
bestCorr = FLT_MIN;
bestCorr = -FLT_MAX;
bestOffs = 0;
// Scans for the best correlation value by testing each possible position
// over the permitted range.
bestCorr = calcCrossCorr(refPos, pMidBuffer, norm);
bestCorr = (bestCorr + 0.1) * 0.75;
#pragma omp parallel for
for (i = 1; i < seekLength; i ++)
for (i = 1; i < seekLength; i ++)
{
double corr;
// Calculates correlation value for the mixing position corresponding to 'i'
#ifdef _OPENMP
#if defined(_OPENMP) || defined(ST_SIMD_AVOID_UNALIGNED)
// in parallel OpenMP mode, can't use norm accumulator version as parallel executor won't
// iterate the loop in sequential order
// in SIMD mode, avoid accumulator version to allow avoiding unaligned positions
corr = calcCrossCorr(refPos + channels * i, pMidBuffer, norm);
#else
// In non-parallel version call "calcCrossCorrAccumulate" that is otherwise same
@@ -341,6 +341,11 @@ int TDStretch::seekBestOverlapPositionFull(const SAMPLETYPE *refPos)
}
}
}
#ifdef SOUNDTOUCH_INTEGER_SAMPLES
adaptNormalizer();
#endif
// clear cross correlation routine state if necessary (is so e.g. in MMX routines).
clearCrossCorrState();
@@ -348,64 +353,157 @@ int TDStretch::seekBestOverlapPositionFull(const SAMPLETYPE *refPos)
}
// Seeks for the optimal overlap-mixing position. The 'stereo' version of the
// routine
// Quick seek algorithm for improved runtime-performance: First roughly scans through the
// correlation area, and then scan surroundings of two best preliminary correlation candidates
// with improved precision
//
// The best position is determined as the position where the two overlapped
// sample sequences are 'most alike', in terms of the highest cross-correlation
// value over the overlapping period
int TDStretch::seekBestOverlapPositionQuick(const SAMPLETYPE *refPos)
// Based on testing:
// - This algorithm gives on average 99% as good match as the full algorithm
// - this quick seek algorithm finds the best match on ~90% of cases
// - on those 10% of cases when this algorithm doesn't find best match,
// it still finds on average ~90% match vs. the best possible match
int TDStretch::seekBestOverlapPositionQuick(const SAMPLETYPE *refPos)
{
int j;
#define _MIN(a, b) (((a) < (b)) ? (a) : (b))
#define SCANSTEP 16
#define SCANWIND 8
int bestOffs;
double bestCorr, corr;
int scanCount, corrOffset, tempOffset;
int i;
int bestOffs2;
float bestCorr, corr;
float bestCorr2;
double norm;
bestCorr = FLT_MIN;
bestOffs = _scanOffsets[0][0];
corrOffset = 0;
tempOffset = 0;
// note: 'float' types used in this function in case that the platform would need to use software-fp
// Scans for the best correlation value using four-pass hierarchical search.
bestCorr =
bestCorr2 = -FLT_MAX;
bestOffs =
bestOffs2 = SCANWIND;
// Scans for the best correlation value by testing each possible position
// over the permitted range. Look for two best matches on the first pass to
// increase possibility of ideal match.
//
// The look-up table 'scans' has hierarchical position adjusting steps.
// In first pass the routine searhes for the highest correlation with
// relatively coarse steps, then rescans the neighbourhood of the highest
// correlation with better resolution and so on.
for (scanCount = 0;scanCount < 4; scanCount ++)
// Begin from "SCANSTEP" instead of SCANWIND to make the calculation
// catch the 'middlepoint' of seekLength vector as that's the a-priori
// expected best match position
//
// Roughly:
// - 15% of cases find best result directly on the first round,
// - 75% cases find better match on 2nd round around the best match from 1st round
// - 10% cases find better match on 2nd round around the 2nd-best-match from 1st round
for (i = SCANSTEP; i < seekLength - SCANWIND - 1; i += SCANSTEP)
{
j = 0;
while (_scanOffsets[scanCount][j])
// Calculates correlation value for the mixing position corresponding
// to 'i'
corr = (float)calcCrossCorr(refPos + channels*i, pMidBuffer, norm);
// heuristic rule to slightly favour values close to mid of the seek range
float tmp = (float)(2 * i - seekLength - 1) / (float)seekLength;
corr = ((corr + 0.1f) * (1.0f - 0.25f * tmp * tmp));
// Checks for the highest correlation value
if (corr > bestCorr)
{
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, 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));
// Checks for the highest correlation value
if (corr > bestCorr)
{
bestCorr = corr;
bestOffs = tempOffset;
}
j ++;
// found new best match. keep the previous best as 2nd best match
bestCorr2 = bestCorr;
bestOffs2 = bestOffs;
bestCorr = corr;
bestOffs = i;
}
else if (corr > bestCorr2)
{
// not new best, but still new 2nd best match
bestCorr2 = corr;
bestOffs2 = i;
}
corrOffset = bestOffs;
}
// Scans surroundings of the found best match with small stepping
int end = _MIN(bestOffs + SCANWIND + 1, seekLength);
for (i = bestOffs - SCANWIND; i < end; i++)
{
if (i == bestOffs) continue; // this offset already calculated, thus skip
// Calculates correlation value for the mixing position corresponding
// to 'i'
corr = (float)calcCrossCorr(refPos + channels*i, pMidBuffer, norm);
// heuristic rule to slightly favour values close to mid of the range
float tmp = (float)(2 * i - seekLength - 1) / (float)seekLength;
corr = ((corr + 0.1f) * (1.0f - 0.25f * tmp * tmp));
// Checks for the highest correlation value
if (corr > bestCorr)
{
bestCorr = corr;
bestOffs = i;
}
}
// Scans surroundings of the 2nd best match with small stepping
end = _MIN(bestOffs2 + SCANWIND + 1, seekLength);
for (i = bestOffs2 - SCANWIND; i < end; i++)
{
if (i == bestOffs2) continue; // this offset already calculated, thus skip
// Calculates correlation value for the mixing position corresponding
// to 'i'
corr = (float)calcCrossCorr(refPos + channels*i, pMidBuffer, norm);
// heuristic rule to slightly favour values close to mid of the range
float tmp = (float)(2 * i - seekLength - 1) / (float)seekLength;
corr = ((corr + 0.1f) * (1.0f - 0.25f * tmp * tmp));
// Checks for the highest correlation value
if (corr > bestCorr)
{
bestCorr = corr;
bestOffs = i;
}
}
// clear cross correlation routine state if necessary (is so e.g. in MMX routines).
clearCrossCorrState();
#ifdef SOUNDTOUCH_INTEGER_SAMPLES
adaptNormalizer();
#endif
return bestOffs;
}
/// For integer algorithm: adapt normalization factor divider with music so that
/// it'll not be pessimistically restrictive that can degrade quality on quieter sections
/// yet won't cause integer overflows either
void TDStretch::adaptNormalizer()
{
// Do not adapt normalizer over too silent sequences to avoid averaging filter depleting to
// too low values during pauses in music
if ((maxnorm > 1000) || (maxnormf > 40000000))
{
//norm averaging filter
maxnormf = 0.9f * maxnormf + 0.1f * (float)maxnorm;
if ((maxnorm > 800000000) && (overlapDividerBitsNorm < 16))
{
// large values, so increase divider
overlapDividerBitsNorm++;
if (maxnorm > 1600000000) overlapDividerBitsNorm++; // extra large value => extra increase
}
else if ((maxnormf < 1000000) && (overlapDividerBitsNorm > 0))
{
// extra small values, decrease divider
overlapDividerBitsNorm--;
}
}
maxnorm = 0;
}
/// clear cross correlation routine state if necessary
void TDStretch::clearCrossCorrState()
{
@@ -417,18 +515,18 @@ void TDStretch::clearCrossCorrState()
void TDStretch::calcSeqParameters()
{
// Adjust tempo param according to tempo, so that variating processing sequence length is used
// at varius tempo settings, between the given low...top limits
// at various tempo settings, between the given low...top limits
#define AUTOSEQ_TEMPO_LOW 0.5 // auto setting low tempo range (-50%)
#define AUTOSEQ_TEMPO_TOP 2.0 // auto setting top tempo range (+100%)
// sequence-ms setting values at above low & top tempo
#define AUTOSEQ_AT_MIN 125.0
#define AUTOSEQ_AT_MAX 50.0
#define AUTOSEQ_AT_MIN 90.0
#define AUTOSEQ_AT_MAX 40.0
#define AUTOSEQ_K ((AUTOSEQ_AT_MAX - AUTOSEQ_AT_MIN) / (AUTOSEQ_TEMPO_TOP - AUTOSEQ_TEMPO_LOW))
#define AUTOSEQ_C (AUTOSEQ_AT_MIN - (AUTOSEQ_K) * (AUTOSEQ_TEMPO_LOW))
// seek-window-ms setting values at above low & top tempo
#define AUTOSEEK_AT_MIN 25.0
// seek-window-ms setting values at above low & top tempoq
#define AUTOSEEK_AT_MIN 20.0
#define AUTOSEEK_AT_MAX 15.0
#define AUTOSEEK_K ((AUTOSEEK_AT_MAX - AUTOSEEK_AT_MIN) / (AUTOSEQ_TEMPO_TOP - AUTOSEQ_TEMPO_LOW))
#define AUTOSEEK_C (AUTOSEEK_AT_MIN - (AUTOSEEK_K) * (AUTOSEQ_TEMPO_LOW))
@@ -464,7 +562,7 @@ void TDStretch::calcSeqParameters()
// Sets new target tempo. Normal tempo = 'SCALE', smaller values represent slower
// tempo, larger faster tempo.
void TDStretch::setTempo(float newTempo)
void TDStretch::setTempo(double newTempo)
{
int intskip;
@@ -475,7 +573,7 @@ void TDStretch::setTempo(float newTempo)
// Calculate ideal skip length (according to tempo value)
nominalSkip = tempo * (seekWindowLength - overlapLength);
intskip = (int)(nominalSkip + 0.5f);
intskip = (int)(nominalSkip + 0.5);
// Calculate how many samples are needed in the 'inputBuffer' to
// process another batch of samples
@@ -488,9 +586,8 @@ void TDStretch::setTempo(float newTempo)
// Sets the number of channels, 1 = mono, 2 = stereo
void TDStretch::setChannels(int numChannels)
{
assert(numChannels > 0);
if (channels == numChannels) return;
// assert(numChannels == 1 || numChannels == 2);
if (!verifyNumberOfChannels(numChannels) ||
(channels == numChannels)) return;
channels = numChannels;
inputBuffer.setChannels(channels);
@@ -539,7 +636,8 @@ void TDStretch::processNominalTempo()
// the result into 'outputBuffer'
void TDStretch::processSamples()
{
int ovlSkip, offset;
int ovlSkip;
int offset = 0;
int temp;
/* Removed this small optimization - can introduce a click to sound when tempo setting
@@ -556,35 +654,62 @@ void TDStretch::processSamples()
// to form a processing frame.
while ((int)inputBuffer.numSamples() >= sampleReq)
{
// If tempo differs from the normal ('SCALE'), scan for the best overlapping
// position
offset = seekBestOverlapPosition(inputBuffer.ptrBegin());
if (isBeginning == false)
{
// apart from the very beginning of the track,
// scan for the best overlapping position & do overlap-add
offset = seekBestOverlapPosition(inputBuffer.ptrBegin());
// Mix the samples in the 'inputBuffer' at position of 'offset' with the
// samples in 'midBuffer' using sliding overlapping
// ... first partially overlap with the end of the previous sequence
// (that's in 'midBuffer')
overlap(outputBuffer.ptrEnd((uint)overlapLength), inputBuffer.ptrBegin(), (uint)offset);
outputBuffer.putSamples((uint)overlapLength);
// Mix the samples in the 'inputBuffer' at position of 'offset' with the
// samples in 'midBuffer' using sliding overlapping
// ... first partially overlap with the end of the previous sequence
// (that's in 'midBuffer')
overlap(outputBuffer.ptrEnd((uint)overlapLength), inputBuffer.ptrBegin(), (uint)offset);
outputBuffer.putSamples((uint)overlapLength);
offset += overlapLength;
}
else
{
// Adjust processing offset at beginning of track by not perform initial overlapping
// and compensating that in the 'input buffer skip' calculation
isBeginning = false;
int skip = (int)(tempo * overlapLength + 0.5 * seekLength + 0.5);
#ifdef ST_SIMD_AVOID_UNALIGNED
// in SIMD mode, round the skip amount to value corresponding to aligned memory address
if (channels == 1)
{
skip &= -4;
}
else if (channels == 2)
{
skip &= -2;
}
#endif
skipFract -= skip;
if (skipFract <= -nominalSkip)
{
skipFract = -nominalSkip;
}
}
// ... then copy sequence samples from 'inputBuffer' to output:
// length of sequence
temp = (seekWindowLength - 2 * overlapLength);
// crosscheck that we don't have buffer overflow...
if ((int)inputBuffer.numSamples() < (offset + temp + overlapLength * 2))
if ((int)inputBuffer.numSamples() < (offset + seekWindowLength - overlapLength))
{
continue; // just in case, shouldn't really happen
}
outputBuffer.putSamples(inputBuffer.ptrBegin() + channels * (offset + overlapLength), (uint)temp);
// length of sequence
temp = (seekWindowLength - 2 * overlapLength);
outputBuffer.putSamples(inputBuffer.ptrBegin() + channels * offset, (uint)temp);
// Copies the end of the current sequence from 'inputBuffer' to
// 'midBuffer' for being mixed with the beginning of the next
// processing sequence and so on
assert((offset + temp + overlapLength * 2) <= (int)inputBuffer.numSamples());
memcpy(pMidBuffer, inputBuffer.ptrBegin() + channels * (offset + temp + overlapLength),
assert((offset + temp + overlapLength) <= (int)inputBuffer.numSamples());
memcpy(pMidBuffer, inputBuffer.ptrBegin() + channels * (offset + temp),
channels * sizeof(SAMPLETYPE) * overlapLength);
// Remove the processed samples from the input buffer. Update
@@ -680,7 +805,7 @@ TDStretch * TDStretch::newInstance()
//////////////////////////////////////////////////////////////////////////////
//
// Integer arithmetics specific algorithm implementations.
// Integer arithmetic specific algorithm implementations.
//
//////////////////////////////////////////////////////////////////////////////
@@ -694,7 +819,7 @@ void TDStretch::overlapStereo(short *poutput, const short *input) const
short temp;
int cnt2;
for (i = 0; i < overlapLength ; i ++)
for (i = 0; i < overlapLength ; i ++)
{
temp = (short)(overlapLength - i);
cnt2 = 2 * i;
@@ -706,21 +831,19 @@ void TDStretch::overlapStereo(short *poutput, const short *input) const
// Overlaps samples in 'midBuffer' with the samples in 'input'. The 'Multi'
// version of the routine.
void TDStretch::overlapMulti(SAMPLETYPE *poutput, const SAMPLETYPE *input) const
void TDStretch::overlapMulti(short *poutput, const short *input) const
{
SAMPLETYPE m1=(SAMPLETYPE)0;
SAMPLETYPE m2;
int i=0;
short m1;
int i = 0;
for (m2 = (SAMPLETYPE)overlapLength; m2; m2 --)
for (m1 = 0; m1 < overlapLength; m1 ++)
{
short m2 = (short)(overlapLength - m1);
for (int c = 0; c < channels; c ++)
{
poutput[i] = (input[i] * m1 + pMidBuffer[i] * m2) / overlapLength;
i++;
}
m1++;
}
}
@@ -743,13 +866,15 @@ void TDStretch::calculateOverlapLength(int aoverlapMs)
// calculate overlap length so that it's power of 2 - thus it's easy to do
// integer division by right-shifting. Term "-1" at end is to account for
// the extra most significatnt bit left unused in result by signed multiplication
overlapDividerBits = _getClosest2Power((sampleRate * aoverlapMs) / 1000.0) - 1;
if (overlapDividerBits > 9) overlapDividerBits = 9;
if (overlapDividerBits < 3) overlapDividerBits = 3;
newOvl = (int)pow(2.0, (int)overlapDividerBits + 1); // +1 => account for -1 above
overlapDividerBitsPure = _getClosest2Power((sampleRate * aoverlapMs) / 1000.0) - 1;
if (overlapDividerBitsPure > 9) overlapDividerBitsPure = 9;
if (overlapDividerBitsPure < 3) overlapDividerBitsPure = 3;
newOvl = (int)pow(2.0, (int)overlapDividerBitsPure + 1); // +1 => account for -1 above
acceptNewOverlapLength(newOvl);
overlapDividerBitsNorm = overlapDividerBitsPure;
// calculate sloping divider so that crosscorrelation operation won't
// overflow 32-bit register. Max. sum of the crosscorrelation sum without
// divider would be 2^30*(N^3-N)/3, where N = overlap length
@@ -757,28 +882,40 @@ void TDStretch::calculateOverlapLength(int aoverlapMs)
}
double TDStretch::calcCrossCorr(const short *mixingPos, const short *compare, double &norm) const
double TDStretch::calcCrossCorr(const short *mixingPos, const short *compare, double &norm)
{
long corr;
long lnorm;
unsigned long lnorm;
int i;
#ifdef ST_SIMD_AVOID_UNALIGNED
// in SIMD mode skip 'mixingPos' positions that aren't aligned to 16-byte boundary
if (((ulongptr)mixingPos) & 15) return -1e50;
#endif
// hint compiler autovectorization that loop length is divisible by 8
int ilength = (channels * overlapLength) & -8;
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)
// Same routine for stereo and mono
for (i = 0; i < ilength; i += 2)
{
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;
mixingPos[i + 1] * compare[i + 1]) >> overlapDividerBitsNorm;
lnorm += (mixingPos[i] * mixingPos[i] +
mixingPos[i + 1] * mixingPos[i + 1]) >> overlapDividerBits; // notice: do intermediate division here to avoid integer overflow
lnorm += (mixingPos[i + 2] * mixingPos[i + 2] +
mixingPos[i + 3] * mixingPos[i + 3]) >> overlapDividerBits;
mixingPos[i + 1] * mixingPos[i + 1]) >> overlapDividerBitsNorm;
// do intermediate scalings to avoid integer overflow
}
if (lnorm > maxnorm)
{
// modify 'maxnorm' inside critical section to avoid multi-access conflict if in OpenMP mode
#pragma omp critical
if (lnorm > maxnorm)
{
maxnorm = lnorm;
}
}
// Normalize result by dividing by sqrt(norm) - this step is easiest
// done using floating point operation
norm = (double)lnorm;
@@ -787,38 +924,42 @@ double TDStretch::calcCrossCorr(const short *mixingPos, const short *compare, do
/// Update cross-correlation by accumulating "norm" coefficient by previously calculated value
double TDStretch::calcCrossCorrAccumulate(const short *mixingPos, const short *compare, double &norm) const
double TDStretch::calcCrossCorrAccumulate(const short *mixingPos, const short *compare, double &norm)
{
long corr;
long lnorm;
int i;
// hint compiler autovectorization that loop length is divisible by 8
int ilength = (channels * overlapLength) & -8;
// cancel first normalizer tap from previous round
lnorm = 0;
for (i = 1; i <= channels; i ++)
{
lnorm -= (mixingPos[-i] * mixingPos[-i]) >> overlapDividerBits;
lnorm -= (mixingPos[-i] * mixingPos[-i]) >> overlapDividerBitsNorm;
}
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)
// Same routine for stereo and mono.
for (i = 0; i < ilength; i += 2)
{
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;
mixingPos[i + 1] * compare[i + 1]) >> overlapDividerBitsNorm;
}
// update normalizer with last samples of this round
for (int j = 0; j < channels; j ++)
{
i --;
lnorm += (mixingPos[i] * mixingPos[i]) >> overlapDividerBits;
lnorm += (mixingPos[i] * mixingPos[i]) >> overlapDividerBitsNorm;
}
norm += (double)lnorm;
if (norm > maxnorm)
{
maxnorm = (unsigned long)norm;
}
// Normalize result by dividing by sqrt(norm) - this step is easiest
// done using floating point operation
@@ -829,7 +970,7 @@ double TDStretch::calcCrossCorrAccumulate(const short *mixingPos, const short *c
//////////////////////////////////////////////////////////////////////////////
//
// Floating point arithmetics specific algorithm implementations.
// Floating point arithmetic specific algorithm implementations.
//
#ifdef SOUNDTOUCH_FLOAT_SAMPLES
@@ -903,29 +1044,26 @@ void TDStretch::calculateOverlapLength(int overlapInMsec)
/// Calculate cross-correlation
double TDStretch::calcCrossCorr(const float *mixingPos, const float *compare, double &anorm) const
double TDStretch::calcCrossCorr(const float *mixingPos, const float *compare, double &anorm)
{
double corr;
double norm;
float corr;
float norm;
int i;
#ifdef ST_SIMD_AVOID_UNALIGNED
// in SIMD mode skip 'mixingPos' positions that aren't aligned to 16-byte boundary
if (((ulongptr)mixingPos) & 15) return -1e50;
#endif
// hint compiler autovectorization that loop length is divisible by 8
int ilength = (channels * overlapLength) & -8;
corr = norm = 0;
// 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)
// Same routine for stereo and mono
for (i = 0; i < ilength; i ++)
{
corr += mixingPos[i] * compare[i] +
mixingPos[i + 1] * compare[i + 1];
norm += mixingPos[i] * mixingPos[i] +
mixingPos[i + 1] * mixingPos[i + 1];
// unroll the loop for better CPU efficiency:
corr += mixingPos[i + 2] * compare[i + 2] +
mixingPos[i + 3] * compare[i + 3];
norm += mixingPos[i + 2] * mixingPos[i + 2] +
mixingPos[i + 3] * mixingPos[i + 3];
corr += mixingPos[i] * compare[i];
norm += mixingPos[i] * mixingPos[i];
}
anorm = norm;
@@ -934,9 +1072,9 @@ double TDStretch::calcCrossCorr(const float *mixingPos, const float *compare, do
/// Update cross-correlation by accumulating "norm" coefficient by previously calculated value
double TDStretch::calcCrossCorrAccumulate(const float *mixingPos, const float *compare, double &norm) const
double TDStretch::calcCrossCorrAccumulate(const float *mixingPos, const float *compare, double &norm)
{
double corr;
float corr;
int i;
corr = 0;
@@ -947,14 +1085,13 @@ double TDStretch::calcCrossCorrAccumulate(const float *mixingPos, const float *c
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)
// hint compiler autovectorization that loop length is divisible by 8
int ilength = (channels * overlapLength) & -8;
// Same routine for stereo and mono
for (i = 0; i < ilength; i ++)
{
corr += mixingPos[i] * compare[i] +
mixingPos[i + 1] * compare[i + 1] +
mixingPos[i + 2] * compare[i + 2] +
mixingPos[i + 3] * compare[i + 3];
corr += mixingPos[i] * compare[i];
}
// update normalizer with last samples of this round
media/libsoundtouch/src/TDStretch.h
+33 -26
View File
@@ -13,13 +13,6 @@
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2014-04-06 15:57:21 +0000 (Sun, 06 Apr 2014) $
// File revision : $Revision: 4 $
//
// $Id: TDStretch.h 195 2014-04-06 15:57:21Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
@@ -112,39 +105,47 @@ class TDStretch : public FIFOProcessor
protected:
int channels;
int sampleReq;
float tempo;
SAMPLETYPE *pMidBuffer;
SAMPLETYPE *pMidBufferUnaligned;
int overlapLength;
int seekLength;
int seekWindowLength;
int overlapDividerBits;
int overlapDividerBitsNorm;
int overlapDividerBitsPure;
int slopingDivider;
float nominalSkip;
float skipFract;
FIFOSampleBuffer outputBuffer;
FIFOSampleBuffer inputBuffer;
bool bQuickSeek;
int sampleRate;
int sequenceMs;
int seekWindowMs;
int overlapMs;
unsigned long maxnorm;
float maxnormf;
double tempo;
double nominalSkip;
double skipFract;
bool bQuickSeek;
bool bAutoSeqSetting;
bool bAutoSeekSetting;
bool isBeginning;
SAMPLETYPE *pMidBuffer;
SAMPLETYPE *pMidBufferUnaligned;
FIFOSampleBuffer outputBuffer;
FIFOSampleBuffer inputBuffer;
void acceptNewOverlapLength(int newOverlapLength);
virtual void clearCrossCorrState();
void calculateOverlapLength(int overlapMs);
virtual double calcCrossCorr(const SAMPLETYPE *mixingPos, const SAMPLETYPE *compare, double &norm) const;
virtual double calcCrossCorrAccumulate(const SAMPLETYPE *mixingPos, const SAMPLETYPE *compare, double &norm) const;
virtual double calcCrossCorr(const SAMPLETYPE *mixingPos, const SAMPLETYPE *compare, double &norm);
virtual double calcCrossCorrAccumulate(const SAMPLETYPE *mixingPos, const SAMPLETYPE *compare, double &norm);
virtual int seekBestOverlapPositionFull(const SAMPLETYPE *refPos);
virtual int seekBestOverlapPositionQuick(const SAMPLETYPE *refPos);
int seekBestOverlapPosition(const SAMPLETYPE *refPos);
virtual int seekBestOverlapPosition(const SAMPLETYPE *refPos);
virtual void overlapStereo(SAMPLETYPE *output, const SAMPLETYPE *input) const;
virtual void overlapMono(SAMPLETYPE *output, const SAMPLETYPE *input) const;
@@ -154,6 +155,7 @@ protected:
void overlap(SAMPLETYPE *output, const SAMPLETYPE *input, uint ovlPos) const;
void calcSeqParameters();
void adaptNormalizer();
/// Changes the tempo of the given sound samples.
/// Returns amount of samples returned in the "output" buffer.
@@ -182,7 +184,7 @@ public:
/// Sets new target tempo. Normal tempo = 'SCALE', smaller values represent slower
/// tempo, larger faster tempo.
void setTempo(float newTempo);
void setTempo(double newTempo);
/// Returns nonzero if there aren't any samples available for outputting.
virtual void clear();
@@ -238,8 +240,13 @@ public:
{
return seekWindowLength - overlapLength;
}
};
/// return approximate initial input-output latency
int getLatency() const
{
return sampleReq;
}
};
// Implementation-specific class declarations:
@@ -249,8 +256,8 @@ public:
class TDStretchMMX : public TDStretch
{
protected:
double calcCrossCorr(const short *mixingPos, const short *compare, double &norm) const;
double calcCrossCorrAccumulate(const short *mixingPos, const short *compare, double &norm) const;
double calcCrossCorr(const short *mixingPos, const short *compare, double &norm);
double calcCrossCorrAccumulate(const short *mixingPos, const short *compare, double &norm);
virtual void overlapStereo(short *output, const short *input) const;
virtual void clearCrossCorrState();
};
@@ -262,8 +269,8 @@ public:
class TDStretchSSE : public TDStretch
{
protected:
double calcCrossCorr(const float *mixingPos, const float *compare, double &norm) const;
double calcCrossCorrAccumulate(const float *mixingPos, const float *compare, double &norm) const;
double calcCrossCorr(const float *mixingPos, const float *compare, double &norm);
double calcCrossCorrAccumulate(const float *mixingPos, const float *compare, double &norm);
};
#endif /// SOUNDTOUCH_ALLOW_SSE
media/libsoundtouch/src/cpu_detect.h
-7
View File
@@ -12,13 +12,6 @@
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2008-02-10 16:26:55 +0000 (Sun, 10 Feb 2008) $
// File revision : $Revision: 4 $
//
// $Id: cpu_detect.h 11 2008-02-10 16:26:55Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
media/libsoundtouch/src/cpu_detect_x86.cpp
-8
View File
@@ -11,13 +11,6 @@
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2014-01-07 18:24:28 +0000 (Tue, 07 Jan 2014) $
// File revision : $Revision: 4 $
//
// $Id: cpu_detect_x86.cpp 183 2014-01-07 18:24:28Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
@@ -74,7 +67,6 @@ void disableExtensions(uint dwDisableMask)
}
/// Checks which instruction set extensions are supported by the CPU.
uint detectCPUextensions(void)
{
media/libsoundtouch/src/mmx_optimized.cpp
+27 -16
View File
@@ -20,13 +20,6 @@
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2015-02-22 15:10:38 +0000 (Sun, 22 Feb 2015) $
// File revision : $Revision: 4 $
//
// $Id: mmx_optimized.cpp 206 2015-02-22 15:10:38Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
@@ -68,7 +61,7 @@ using namespace soundtouch;
// Calculates cross correlation of two buffers
double TDStretchMMX::calcCrossCorr(const short *pV1, const short *pV2, double &dnorm) const
double TDStretchMMX::calcCrossCorr(const short *pV1, const short *pV2, double &dnorm)
{
const __m64 *pVec1, *pVec2;
__m64 shifter;
@@ -79,7 +72,7 @@ double TDStretchMMX::calcCrossCorr(const short *pV1, const short *pV2, double &d
pVec1 = (__m64*)pV1;
pVec2 = (__m64*)pV2;
shifter = _m_from_int(overlapDividerBits);
shifter = _m_from_int(overlapDividerBitsNorm);
normaccu = accu = _mm_setzero_si64();
// Process 4 parallel sets of 2 * stereo samples or 4 * mono samples
@@ -123,6 +116,16 @@ double TDStretchMMX::calcCrossCorr(const short *pV1, const short *pV2, double &d
// Clear MMS state
_m_empty();
if (norm > (long)maxnorm)
{
// modify 'maxnorm' inside critical section to avoid multi-access conflict if in OpenMP mode
#pragma omp critical
if (norm > (long)maxnorm)
{
maxnorm = norm;
}
}
// Normalize result by dividing by sqrt(norm) - this step is easiest
// done using floating point operation
dnorm = (double)norm;
@@ -134,7 +137,7 @@ double TDStretchMMX::calcCrossCorr(const short *pV1, const short *pV2, double &d
/// Update cross-correlation by accumulating "norm" coefficient by previously calculated value
double TDStretchMMX::calcCrossCorrAccumulate(const short *pV1, const short *pV2, double &dnorm) const
double TDStretchMMX::calcCrossCorrAccumulate(const short *pV1, const short *pV2, double &dnorm)
{
const __m64 *pVec1, *pVec2;
__m64 shifter;
@@ -146,13 +149,13 @@ double TDStretchMMX::calcCrossCorrAccumulate(const short *pV1, const short *pV2,
lnorm = 0;
for (i = 1; i <= channels; i ++)
{
lnorm -= (pV1[-i] * pV1[-i]) >> overlapDividerBits;
lnorm -= (pV1[-i] * pV1[-i]) >> overlapDividerBitsNorm;
}
pVec1 = (__m64*)pV1;
pVec2 = (__m64*)pV2;
shifter = _m_from_int(overlapDividerBits);
shifter = _m_from_int(overlapDividerBitsNorm);
accu = _mm_setzero_si64();
// Process 4 parallel sets of 2 * stereo samples or 4 * mono samples
@@ -191,10 +194,15 @@ double TDStretchMMX::calcCrossCorrAccumulate(const short *pV1, const short *pV2,
pV1 = (short *)pVec1;
for (int j = 1; j <= channels; j ++)
{
lnorm += (pV1[-j] * pV1[-j]) >> overlapDividerBits;
lnorm += (pV1[-j] * pV1[-j]) >> overlapDividerBitsNorm;
}
dnorm += (double)lnorm;
if (lnorm > (long)maxnorm)
{
maxnorm = lnorm;
}
// Normalize result by dividing by sqrt(norm) - this step is easiest
// done using floating point operation
return (double)corr / sqrt((dnorm < 1e-9) ? 1.0 : dnorm);
@@ -209,7 +217,6 @@ void TDStretchMMX::clearCrossCorrState()
}
// MMX-optimized version of the function overlapStereo
void TDStretchMMX::overlapStereo(short *output, const short *input) const
{
@@ -233,7 +240,7 @@ void TDStretchMMX::overlapStereo(short *output, const short *input) const
// Overlaplength-division by shifter. "+1" is to account for "-1" deduced in
// overlapDividerBits calculation earlier.
shifter = _m_from_int(overlapDividerBits + 1);
shifter = _m_from_int(overlapDividerBitsPure + 1);
for (i = 0; i < overlapLength / 4; i ++)
{
@@ -325,7 +332,6 @@ void FIRFilterMMX::setCoefficients(const short *coeffs, uint newLength, uint uRe
}
// mmx-optimized version of the filter routine for stereo sound
uint FIRFilterMMX::evaluateFilterStereo(short *dest, const short *src, uint numSamples) const
{
@@ -382,4 +388,9 @@ uint FIRFilterMMX::evaluateFilterStereo(short *dest, const short *src, uint numS
return (numSamples & 0xfffffffe) - length;
}
#else
// workaround to not complain about empty module
bool _dontcomplain_mmx_empty;
#endif // SOUNDTOUCH_ALLOW_MMX
media/libsoundtouch/src/moz.build
+2
View File
@@ -42,6 +42,8 @@ else:
# Windows need alloca renamed to _alloca
DEFINES['alloca'] = '_alloca'
DEFINES['BUILDING_SOUNDTOUCH'] = 1
# We allow warnings for third-party code that can be updated from upstream.
ALLOW_COMPILER_WARNINGS = True
media/libsoundtouch/src/sse_optimized.cpp
+3 -10
View File
@@ -23,13 +23,6 @@
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2015-02-21 21:24:29 +0000 (Sat, 21 Feb 2015) $
// File revision : $Revision: 4 $
//
// $Id: sse_optimized.cpp 202 2015-02-21 21:24:29Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
@@ -71,7 +64,7 @@ using namespace soundtouch;
#include <math.h>
// Calculates cross correlation of two buffers
double TDStretchSSE::calcCrossCorr(const float *pV1, const float *pV2, double &anorm) const
double TDStretchSSE::calcCrossCorr(const float *pV1, const float *pV2, double &anorm)
{
int i;
const float *pVec1;
@@ -87,7 +80,7 @@ double TDStretchSSE::calcCrossCorr(const float *pV1, const float *pV2, double &a
// Compile-time define SOUNDTOUCH_ALLOW_NONEXACT_SIMD_OPTIMIZATION is provided
// for choosing if this little cheating is allowed.
#ifdef SOUNDTOUCH_ALLOW_NONEXACT_SIMD_OPTIMIZATION
#ifdef ST_SIMD_AVOID_UNALIGNED
// Little cheating allowed, return valid correlation only for
// aligned locations, meaning every second round for stereo sound.
@@ -183,7 +176,7 @@ double TDStretchSSE::calcCrossCorr(const float *pV1, const float *pV2, double &a
double TDStretchSSE::calcCrossCorrAccumulate(const float *pV1, const float *pV2, double &norm) const
double TDStretchSSE::calcCrossCorrAccumulate(const float *pV1, const float *pV2, double &norm)
{
// call usual calcCrossCorr function because SSE does not show big benefit of
// accumulating "norm" value, and also the "norm" rolling algorithm would get
media/libsoundtouch/update.sh
+2
View File
@@ -43,4 +43,6 @@ done
# Patch the imported files.
patch -p1 < moz-libsoundtouch.patch
patch -p1 < mcp-disableexceptions.patch
echo "Remember to update README_MCP with the version details."