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Initial commit as per SoundTouch v1.3.1

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GNU LESSER GENERAL PUBLIC LICENSE
Version 2.1, February 1999
Copyright (C) 1991, 1999 Free Software Foundation, Inc.
59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
Everyone is permitted to copy and distribute verbatim copies
of this license document, but changing it is not allowed.
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as the successor of the GNU Library Public License, version 2, hence
the version number 2.1.]
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## Process this file with automake to create Makefile.in
##
## $Id: Makefile.am 4 2008-02-10 16:07:12Z oparviai $
##
## Copyright (C) 2003 - David W. Durham
##
## This file is part of SoundTouch, an audio processing library for pitch/time adjustments
##
## SoundTouch is free software; you can redistribute it and/or modify it under the
## terms of the GNU General Public License as published by the Free Software
## Foundation; either version 2 of the License, or (at your option) any later
## version.
##
## SoundTouch is distributed in the hope that it will be useful, but WITHOUT ANY
## WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
## A PARTICULAR PURPOSE. See the GNU General Public License for more details.
##
## You should have received a copy of the GNU General Public License along with
## this program; if not, write to the Free Software Foundation, Inc., 59 Temple
## Place - Suite 330, Boston, MA 02111-1307, USA
## I used config/am_include.mk for common definitions
include $(top_srcdir)/config/am_include.mk
## Descend into SUBDIRS and run make. Look at the Makefile.am files in the
## subdirectories Start at frontend_fox/Makefile.am to see how everything works.
SUBDIRS=include source
# list files that are documentation to be packaged in a release tarball and installed
pkgdoc_DATA=COPYING.TXT README.html
# extra data files that are to be pacakged in a release tarball and installed into the data directory
#pkgdata_DATA=
# sets up for soundtouch.m4 to be installed
m4datadir=$(datadir)/aclocal
m4data_DATA=soundtouch.m4
## These extra files and directories will be included in the distribution. by
## using make-dist by default many common filenames are automatically included
## such as AUTHORS, COPYING, etc the bootstrap script really shouldn't be a part
## of a final package, but it is useful for developers who might want to make
## changes to the configure scripts or makefiles.
# NOTE: wouldn't have to list the .TXT file if it were named without the .TXT
EXTRA_DIST= \
soundtouch.m4 \
config/m4 \
bootstrap \
make-win.bat \
COPYING.TXT \
README.html
pkgconfigdir = $(libdir)/pkgconfig
pkgconfig_DATA = soundtouch-1.0.pc
## This removes stuff from the distribution which may be present
## from a cvs checkout or other build reasons
dist-hook:
rm -rf `find $(distdir) -type d -name CVS` # remove all CVS directories
echo rm -rf `find $(distdir) -type f -name \.\#\*` # CVS revision files left around for some reason
## This happens at 'make distclean'
#distcleancheck:
# rm -rf files-that-may-also-need-to-be-deleted-on-'make distclean'
# flag to aclocal where to find m4 macros for tests
ACLOCAL_AMFLAGS = -I config/m4

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# $Id: bootstrap 4 2008-02-10 16:07:12Z oparviai $
#!/bin/sh
if [ "$1" = "--clean" ]
then
if [ -a Makefile ]
then
make maintainer-clean
elif [ -a configure ]
then
configure && $0 --clean
else
bootstrap && configure && $0 --clean
fi
rm -rf configure libtool aclocal.m4 `find . -name Makefile.in` autom4te*.cache config/config.guess config/config.h.in config/config.sub config/depcomp config/install-sh config/ltmain.sh config/missing config/mkinstalldirs config/stamp-h config/stamp-h.in
#gettextize files
#rm -f ABOUT-NLS config/config.rpath config/m4/codeset.m4 config/m4/gettext.m4 config/m4/glibc21.m4 config/m4/iconv.m4 config/m4/intdiv0.m4 config/m4/inttypes-pri.m4 config/m4/inttypes.m4 config/m4/inttypes_h.m4 config/m4/isc-posix.m4 config/m4/lcmessage.m4 config/m4/lib-ld.m4 config/m4/lib-link.m4 config/m4/lib-prefix.m4 config/m4/progtest.m4 config/m4/stdint_h.m4 config/m4/uintmax_t.m4 config/m4/ulonglong.m4 po/Makefile.in.in po/Rules-quot po/boldquot.sed po/en@boldquot.header po/en@quot.header po/insert-header.sin po/quot.sed po/remove-potcdate.sin
else
export AUTOMAKE="automake --add-missing --foreign"
autoreconf -fisv && rm -f `find . -name "*~"` && rm -f ChangeLog
exit $?
fi

3
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Files in this directory are used by GNU autoconf/automake system.
These files aren't used/needed in the Windows environment.

51
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## vim:tw=78
## Process this file with automake to create Makefile.in
##
## $Id: am_include.mk 4 2008-02-10 16:07:12Z oparviai $
##
## Copyright (C) 2003 - David W. Durham
##
## This file is part of SoundTouch, an audio processing library for pitch/time adjustments
##
## SoundTouch is free software; you can redistribute it and/or modify it under the
## terms of the GNU General Public License as published by the Free Software
## Foundation; either version 2 of the License, or (at your option) any later
## version.
##
## SoundTouch is distributed in the hope that it will be useful, but WITHOUT ANY
## WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
## A PARTICULAR PURPOSE. See the GNU General Public License for more details.
##
## You should have received a copy of the GNU General Public License along with
## this program; if not, write to the Free Software Foundation, Inc., 59 Temple
## Place - Suite 330, Boston, MA 02111-1307, USA
## These are common definitions used in all Makefiles
## It is actually included when a makefile.am is coverted to Makefile.in
## by automake, so it's ok to have @MACROS@ that will be set by configure
## INCLUDES is automatically added to CXXFLAGS at compile time. The
## $(top_srcdir) macro is set by configure. It's important to use $(top_srcdir)
## in case a user decides to build in a separate directory from the base package
## directory. Using absolute, or relative paths is a bad idea.
INCLUDES=-I$(top_srcdir)/include
## CXXFLAGS is also automatically added to the $(CXX) macro at compile time, and
## is passed down to the children as well
## I list these here in case there needs to always be some flags passed to the compiler
CXXFLAGS=@CXXFLAGS@
## LDFLAGS will be added at link time
## note that the -l flags are set by the LIBS macro that is set by configure
## This line doesn't need to be here, configure will add LDFLAGS on it's own,
## but I'm leaving here as a place to add LDFLAGS if so desired
LDFLAGS=@LDFLAGS@
# doc directory
pkgdocdir=$(prefix)/doc/@PACKAGE@

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dnl SoundTouch configure.ac, by David W. Durham
dnl
dnl $Id: configure.ac 4 2008-02-10 16:07:12Z oparviai $
dnl
dnl This file is part of SoundTouch, an audio processing library for pitch/time adjustments
dnl
dnl SoundTouch is free software; you can redistribute it and/or modify it under the
dnl terms of the GNU General Public License as published by the Free Software
dnl Foundation; either version 2 of the License, or (at your option) any later
dnl version.
dnl
dnl SoundTouch is distributed in the hope that it will be useful, but WITHOUT ANY
dnl WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS
dnl FOR A PARTICULAR PURPOSE. See the GNU General Public License for more
dnl details.
dnl
dnl You should have received a copy of the GNU General Public License along with
dnl this program; if not, write to the Free Software Foundation, Inc., 59 Temple
dnl Place - Suite 330, Boston, MA 02111-1307, USA
# Process this file with autoconf to produce a configure script.
AC_INIT(SoundTouch, 1.3.1, [http://www.surina.net/soundtouch])
AC_CONFIG_AUX_DIR(config)
AM_CONFIG_HEADER([include/soundtouch_config.h])
AM_INIT_AUTOMAKE
AC_DISABLE_SHARED dnl This makes libtool only build static libs
#AC_GNU_SOURCE dnl enable posix extensions in glibc
AC_LANG(C++)
dnl ############################################################################
dnl # Checks for programs #
dnl ############################################################################
CXXFLAGS= dnl AC_PROG_CXX automatically sets CXXFLAGS to -g -O2 if g++ is the C++ compiler and CXXFLAGS hasn't been set, so by setting CXXFLAGS, we can avoid this.
AC_PROG_CXX
#AC_PROG_AWK
AC_PROG_CC
AC_PROG_CPP
AC_PROG_CXXCPP
AC_PROG_INSTALL
#AC_PROG_LN_S
AC_PROG_MAKE_SET
AM_PROG_LIBTOOL dnl turn on using libtool
dnl ############################################################################
dnl # Checks for header files #
dnl ############################################################################
AC_HEADER_STDC
#AC_HEADER_SYS_WAIT
# add any others you want to check for here
#AC_CHECK_HEADERS([fcntl.h limits.h stddef.h stdlib.h string.h sys/ioctl.h sys/vfs.h unistd.h])
dnl ############################################################################
dnl # Checks for typedefs, structures, and compiler characteristics $
dnl ############################################################################
AC_C_CONST
AC_C_INLINE
#AC_TYPE_OFF_T
#AC_TYPE_SIZE_T
AC_ARG_ENABLE(integer-samples,
[AC_HELP_STRING([--enable-integer-samples],
[use integer samples instead of floats
[default=yes]])],,
[enable_integer_samples=no])
if test "x$enable_integer_samples" = "xyes"; then
echo "****** Integer sample type enabled ******"
AC_DEFINE(INTEGER_SAMPLES,1,[Use Integer as Sample type])
else
echo "****** Float sample type enabled ******"
AC_DEFINE(FLOAT_SAMPLES,1,[Use Float as Sample type])
fi
dnl ############################################################################
dnl # Checks for library functions/classes #
dnl ############################################################################
AC_FUNC_MALLOC
AC_TYPE_SIGNAL
dnl make -lm get added to the LIBS
AC_CHECK_LIB(m, sqrt,,AC_MSG_ERROR([compatible libc math library not found]))
dnl add whatever functions you might want to check for here
#AC_CHECK_FUNCS([floor ftruncate memmove memset mkdir modf pow realpath sqrt strchr strdup strerror strrchr strstr strtol])
dnl ############################################################################
dnl # Internationalization and Localization #
dnl ############################################################################
#AM_GNU_GETTEXT_VERSION([0.11.5])
#AM_GNU_GETTEXT([external])
dnl ############################################################################
dnl # Final #
dnl ############################################################################
AC_CONFIG_FILES([
Makefile
source/Makefile
source/SoundTouch/Makefile
source/example/Makefile
source/example/SoundStretch/Makefile
source/example/bpm/Makefile
include/Makefile
])
AC_OUTPUT(
soundtouch-1.0.pc
)
dnl use 'echo' to put stuff here if you want a message to the builder

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////////////////////////////////////////////////////////////////////////////////
///
/// Beats-per-minute (BPM) detection routine.
///
/// The beat detection algorithm works as follows:
/// - Use function 'inputSamples' to input a chunks of samples to the class for
/// analysis. It's a good idea to enter a large sound file or stream in smallish
/// chunks of around few kilosamples in order not to extinguish too much RAM memory.
/// - Input sound data is decimated to approx 500 Hz to reduce calculation burden,
/// which is basically ok as low (bass) frequencies mostly determine the beat rate.
/// Simple averaging is used for anti-alias filtering because the resulting signal
/// quality isn't of that high importance.
/// - Decimated sound data is enveloped, i.e. the amplitude shape is detected by
/// taking absolute value that's smoothed by sliding average. Signal levels that
/// are below a couple of times the general RMS amplitude level are cut away to
/// leave only notable peaks there.
/// - Repeating sound patterns (e.g. beats) are detected by calculating short-term
/// autocorrelation function of the enveloped signal.
/// - After whole sound data file has been analyzed as above, the bpm level is
/// detected by function 'getBpm' that finds the highest peak of the autocorrelation
/// function, calculates it's precise location and converts this reading to bpm's.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2008-02-10 18:07:12 +0200 (su, 10 helmi 2008) $
// File revision : $Revision: 4 $
//
// $Id: BPMDetect.h 4 2008-02-10 16:07:12Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#ifndef _BPMDetect_H_
#define _BPMDetect_H_
#include "STTypes.h"
#include "FIFOSampleBuffer.h"
/// Minimum allowed BPM rate. Used to restrict accepted result above a reasonable limit.
#define MIN_BPM 45
/// Maximum allowed BPM rate. Used to restrict accepted result below a reasonable limit.
#define MAX_BPM 230
/// Class for calculating BPM rate for audio data.
class BPMDetect
{
protected:
/// Auto-correlation accumulator bins.
float *xcorr;
/// Amplitude envelope sliding average approximation level accumulator
float envelopeAccu;
/// RMS volume sliding average approximation level accumulator
float RMSVolumeAccu;
/// Sample average counter.
int decimateCount;
/// Sample average accumulator for FIFO-like decimation.
soundtouch::LONG_SAMPLETYPE decimateSum;
/// Decimate sound by this coefficient to reach approx. 500 Hz.
int decimateBy;
/// Auto-correlation window length
int windowLen;
/// Number of channels (1 = mono, 2 = stereo)
int channels;
/// sample rate
int sampleRate;
/// Beginning of auto-correlation window: Autocorrelation isn't being updated for
/// the first these many correlation bins.
int windowStart;
/// FIFO-buffer for decimated processing samples.
soundtouch::FIFOSampleBuffer *buffer;
/// Initialize the class for processing.
void init(int numChannels, int sampleRate);
/// Updates auto-correlation function for given number of decimated samples that
/// are read from the internal 'buffer' pipe (samples aren't removed from the pipe
/// though).
void updateXCorr(int process_samples /// How many samples are processed.
);
/// Decimates samples to approx. 500 Hz.
///
/// \return Number of output samples.
int decimate(soundtouch::SAMPLETYPE *dest, ///< Destination buffer
const soundtouch::SAMPLETYPE *src, ///< Source sample buffer
int numsamples ///< Number of source samples.
);
/// Calculates amplitude envelope for the buffer of samples.
/// Result is output to 'samples'.
void calcEnvelope(soundtouch::SAMPLETYPE *samples, ///< Pointer to input/output data buffer
int numsamples ///< Number of samples in buffer
);
public:
/// Constructor.
BPMDetect(int numChannels, ///< Number of channels in sample data.
int sampleRate ///< Sample rate in Hz.
);
/// Destructor.
virtual ~BPMDetect();
/// Inputs a block of samples for analyzing: Envelopes the samples and then
/// updates the autocorrelation estimation. When whole song data has been input
/// in smaller blocks using this function, read the resulting bpm with 'getBpm'
/// function.
///
/// Notice that data in 'samples' array can be disrupted in processing.
void inputSamples(soundtouch::SAMPLETYPE *samples, ///< Pointer to input/working data buffer
int numSamples ///< Number of samples in buffer
);
/// Analyzes the results and returns the BPM rate. Use this function to read result
/// after whole song data has been input to the class by consecutive calls of
/// 'inputSamples' function.
///
/// \return Beats-per-minute rate, or zero if detection failed.
float getBpm();
};
#endif // _BPMDetect_H_

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////////////////////////////////////////////////////////////////////////////////
///
/// A buffer class for temporarily storaging sound samples, operates as a
/// first-in-first-out pipe.
///
/// Samples are added to the end of the sample buffer with the 'putSamples'
/// function, and are received from the beginning of the buffer by calling
/// the 'receiveSamples' function. The class automatically removes the
/// output samples from the buffer as well as grows the storage size
/// whenever necessary.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2008-02-10 18:07:12 +0200 (su, 10 helmi 2008) $
// File revision : $Revision: 4 $
//
// $Id: FIFOSampleBuffer.h 4 2008-02-10 16:07:12Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#ifndef FIFOSampleBuffer_H
#define FIFOSampleBuffer_H
#include "FIFOSamplePipe.h"
namespace soundtouch
{
/// Sample buffer working in FIFO (first-in-first-out) principle. The class takes
/// care of storage size adjustment and data moving during input/output operations.
///
/// Notice that in case of stereo audio, one sample is considered to consist of
/// both channel data.
class FIFOSampleBuffer : public FIFOSamplePipe
{
private:
/// Sample buffer.
SAMPLETYPE *buffer;
// Raw unaligned buffer memory. 'buffer' is made aligned by pointing it to first
// 16-byte aligned location of this buffer
SAMPLETYPE *bufferUnaligned;
/// Sample buffer size in bytes
uint sizeInBytes;
/// How many samples are currently in buffer.
uint samplesInBuffer;
/// Channels, 1=mono, 2=stereo.
uint channels;
/// Current position pointer to the buffer. This pointer is increased when samples are
/// removed from the pipe so that it's necessary to actually rewind buffer (move data)
/// only new data when is put to the pipe.
uint bufferPos;
/// Rewind the buffer by moving data from position pointed by 'bufferPos' to real
/// beginning of the buffer.
void rewind();
/// Ensures that the buffer has capacity for at least this many samples.
void ensureCapacity(uint capacityRequirement);
/// Returns current capacity.
uint getCapacity() const;
public:
/// Constructor
FIFOSampleBuffer(int numChannels = 2 ///< Number of channels, 1=mono, 2=stereo.
///< Default is stereo.
);
/// destructor
~FIFOSampleBuffer();
/// 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!
///
/// When using this function to output samples, also remember to 'remove' the
/// output samples from the buffer by calling the
/// 'receiveSamples(numSamples)' function
virtual SAMPLETYPE *ptrBegin() const;
/// Returns a pointer to the end of the used part of the sample buffer (i.e.
/// where the new samples are to be inserted). This function may be used for
/// inserting new samples into the sample buffer directly. Please be careful
/// not corrupt the book-keeping!
///
/// When using this function as means for inserting new samples, also remember
/// to increase the sample count afterwards, by calling the
/// 'putSamples(numSamples)' function.
SAMPLETYPE *ptrEnd(
uint slackCapacity ///< How much free capacity (in samples) there _at least_
///< should be so that the caller can succesfully insert the
///< desired samples to the buffer. If necessary, the function
///< grows the buffer size to comply with this requirement.
);
/// Adds 'numSamples' pcs of samples from the 'samples' memory position to
/// the sample buffer.
virtual void putSamples(const SAMPLETYPE *samples, ///< Pointer to samples.
uint numSamples ///< Number of samples to insert.
);
/// Adjusts the book-keeping to increase number of samples in the buffer without
/// copying any actual samples.
///
/// This function is used to update the number of samples in the sample buffer
/// when accessing the buffer directly with 'ptrEnd' function. Please be
/// careful though!
virtual void putSamples(uint numSamples ///< Number of samples been inserted.
);
/// 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.
);
/// Returns number of samples currently available.
virtual uint numSamples() const;
/// Sets number of channels, 1 = mono, 2 = stereo.
void setChannels(int numChannels);
/// Returns nonzero if there aren't any samples available for outputting.
virtual int isEmpty() const;
/// Clears all the samples.
virtual void clear();
};
}
#endif

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////////////////////////////////////////////////////////////////////////////////
///
/// 'FIFOSamplePipe' : An abstract base class for classes that manipulate sound
/// samples by operating like a first-in-first-out pipe: New samples are fed
/// into one end of the pipe with the 'putSamples' function, and the processed
/// samples are received from the other end with the 'receiveSamples' function.
///
/// 'FIFOProcessor' : A base class for classes the do signal processing with
/// the samples while operating like a first-in-first-out pipe. When samples
/// are input with the 'putSamples' function, the class processes them
/// and moves the processed samples to the given 'output' pipe object, which
/// may be either another processing stage, or a fifo sample buffer object.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2008-02-10 18:07:12 +0200 (su, 10 helmi 2008) $
// File revision : $Revision: 4 $
//
// $Id: FIFOSamplePipe.h 4 2008-02-10 16:07:12Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#ifndef FIFOSamplePipe_H
#define FIFOSamplePipe_H
#include <assert.h>
#include <stdlib.h>
#include "STTypes.h"
namespace soundtouch
{
/// Abstract base class for FIFO (first-in-first-out) sample processing classes.
class FIFOSamplePipe
{
public:
/// 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!
///
/// When using this function to output samples, also remember to 'remove' the
/// output samples from the buffer by calling the
/// 'receiveSamples(numSamples)' function
virtual SAMPLETYPE *ptrBegin() const = 0;
/// Adds 'numSamples' pcs of samples from the 'samples' memory position to
/// the sample buffer.
virtual void putSamples(const SAMPLETYPE *samples, ///< Pointer to samples.
uint numSamples ///< Number of samples to insert.
) = 0;
// Moves samples from the 'other' pipe instance to this instance.
void moveSamples(FIFOSamplePipe &other ///< Other pipe instance where from the receive the data.
)
{
int oNumSamples = other.numSamples();
putSamples(other.ptrBegin(), oNumSamples);
other.receiveSamples(oNumSamples);
};
/// 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.
) = 0;
/// 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.
) = 0;
/// Returns number of samples currently available.
virtual uint numSamples() const = 0;
// Returns nonzero if there aren't any samples available for outputting.
virtual int isEmpty() const = 0;
/// Clears all the samples.
virtual void clear() = 0;
};
/// 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
/// all the processing stages.
///
/// When samples are input to this class, they're first processed and then put to
/// the FIFO pipe that's defined as output of this class. This output pipe can be
/// either other processing stage or a FIFO sample buffer.
class FIFOProcessor :public FIFOSamplePipe
{
protected:
/// Internal pipe where processed samples are put.
FIFOSamplePipe *output;
/// Sets output pipe.
void setOutPipe(FIFOSamplePipe *pOutput)
{
assert(output == NULL);
assert(pOutput != NULL);
output = pOutput;
}
/// Constructor. Doesn't define output pipe; it has to be set be
/// 'setOutPipe' function.
FIFOProcessor()
{
output = NULL;
}
/// Constructor. Configures output pipe.
FIFOProcessor(FIFOSamplePipe *pOutput ///< Output pipe.
)
{
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!
///
/// When using this function to output samples, also remember to 'remove' the
/// output samples from the buffer by calling the
/// 'receiveSamples(numSamples)' function
virtual SAMPLETYPE *ptrBegin() const
{
return output->ptrBegin();
}
public:
/// 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 *outBuffer, ///< Buffer where to copy output samples.
uint maxSamples ///< How many samples to receive at max.
)
{
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.
///
/// 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.
)
{
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
{
return output->isEmpty();
}
};
}
#endif

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## Process this file with automake to create Makefile.in
##
## $Id: Makefile.am 4 2008-02-10 16:07:12Z oparviai $
##
## Copyright (C) 2003 - David W. Durham
##
## This file is part of SoundTouch, an audio processing library for pitch/time adjustments
##
## SoundTouch is free software; you can redistribute it and/or modify it under the
## terms of the GNU General Public License as published by the Free Software
## Foundation; either version 2 of the License, or (at your option) any later
## version.
##
## SoundTouch is distributed in the hope that it will be useful, but WITHOUT ANY
## WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
## A PARTICULAR PURPOSE. See the GNU General Public License for more details.
##
## You should have received a copy of the GNU General Public License along with
## this program; if not, write to the Free Software Foundation, Inc., 59 Temple
## Place - Suite 330, Boston, MA 02111-1307, USA
## I used config/am_include.mk for common definitions
include $(top_srcdir)/config/am_include.mk
pkginclude_HEADERS=FIFOSampleBuffer.h FIFOSamplePipe.h SoundTouch.h STTypes.h BPMDetect.h soundtouch_config.h

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////////////////////////////////////////////////////////////////////////////////
///
/// Common type definitions for SoundTouch audio processing library.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2008-02-10 18:03:11 +0200 (su, 10 helmi 2008) $
// File revision : $Revision: 3 $
//
// $Id: STTypes.h 3 2008-02-10 16:03:11Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#ifndef STTypes_H
#define STTypes_H
typedef unsigned int uint;
typedef unsigned long ulong;
#ifdef __GNUC__
// In GCC, include soundtouch_config.h made by config scritps
#include "soundtouch_config.h"
#endif
#ifndef _WINDEF_
// if these aren't defined already by Windows headers, define now
typedef int BOOL;
#define FALSE 0
#define TRUE 1
#endif // _WINDEF_
namespace soundtouch
{
/// Activate these undef's to overrule the possible sampletype
/// setting inherited from some other header file:
//#undef INTEGER_SAMPLES
//#undef FLOAT_SAMPLES
#if !(INTEGER_SAMPLES || FLOAT_SAMPLES)
/// Choose either 32bit floating point or 16bit integer sampletype
/// by choosing one of the following defines, unless this selection
/// has already been done in some other file.
////
/// Notes:
/// - In Windows environment, choose the sample format with the
/// following defines.
/// - In GNU environment, the floating point samples are used by
/// default, but integer samples can be chosen by giving the
/// following switch to the configure script:
/// ./configure --enable-integer-samples
/// However, if you still prefer to select the sample format here
/// also in GNU environment, then please #undef the INTEGER_SAMPLE
/// and FLOAT_SAMPLE defines first as in comments above.
//#define INTEGER_SAMPLES 1 //< 16bit integer samples
#define FLOAT_SAMPLES 1 //< 32bit float samples
#endif
/// Define this to allow CPU-specific assembler optimizations. Notice that
/// having this enabled on non-x86 platforms doesn't matter; the compiler can
/// drop unsupported extensions on different platforms automatically.
/// However, if you're having difficulties getting the optimized routines
/// compiled with your compler (e.g. some gcc compiler versions may be picky),
/// you may wish to disable the optimizations to make the library compile.
#define ALLOW_OPTIMIZATIONS 1
// 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 ALLOW_NONEXACT_SIMD_OPTIMIZATION 1
#ifdef INTEGER_SAMPLES
// 16bit integer sample type
typedef short SAMPLETYPE;
// data type for sample accumulation: Use 32bit integer to prevent overflows
typedef long LONG_SAMPLETYPE;
#ifdef FLOAT_SAMPLES
// check that only one sample type is defined
#error "conflicting sample types defined"
#endif // FLOAT_SAMPLES
#ifdef ALLOW_OPTIMIZATIONS
#if (WIN32 || __i386__ || __x86_64__)
// Allow MMX optimizations
#define 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;
#ifdef ALLOW_OPTIMIZATIONS
// Allow 3DNow! and SSE optimizations
#if WIN32
#define ALLOW_3DNOW 1
#endif
#if (WIN32 || __i386__ || __x86_64__)
#define ALLOW_SSE 1
#endif
#endif
#endif // INTEGER_SAMPLES
};
#endif

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//////////////////////////////////////////////////////////////////////////////
///
/// SoundTouch - main class for tempo/pitch/rate adjusting routines.
///
/// Notes:
/// - Initialize the SoundTouch object instance by setting up the sound stream
/// parameters with functions 'setSampleRate' and 'setChannels', then set
/// desired tempo/pitch/rate settings with the corresponding functions.
///
/// - The SoundTouch class behaves like a first-in-first-out pipeline: The
/// samples that are to be processed are fed into one of the pipe by calling
/// function 'putSamples', while the ready processed samples can be read
/// from the other end of the pipeline with function 'receiveSamples'.
///
/// - The SoundTouch processing classes require certain sized 'batches' of
/// samples in order to process the sound. For this reason the classes buffer
/// incoming samples until there are enough of samples available for
/// processing, then they carry out the processing step and consequently
/// make the processed samples available for outputting.
///
/// - For the above reason, the processing routines introduce a certain
/// 'latency' between the input and output, so that the samples input to
/// SoundTouch may not be immediately available in the output, and neither
/// the amount of outputtable samples may not immediately be in direct
/// relationship with the amount of previously input samples.
///
/// - The tempo/pitch/rate control parameters can be altered during processing.
/// Please notice though that they aren't currently protected by semaphores,
/// so in multi-thread application external semaphore protection may be
/// required.
///
/// - This class utilizes classes 'TDStretch' for tempo change (without modifying
/// pitch) and 'RateTransposer' for changing the playback rate (that is, both
/// tempo and pitch in the same ratio) of the sound. The third available control
/// 'pitch' (change pitch but maintain tempo) is produced by a combination of
/// combining the two other controls.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2008-02-10 18:07:12 +0200 (su, 10 helmi 2008) $
// File revision : $Revision: 4 $
//
// $Id: SoundTouch.h 4 2008-02-10 16:07:12Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#ifndef SoundTouch_H
#define SoundTouch_H
#include "FIFOSamplePipe.h"
#include "STTypes.h"
namespace soundtouch
{
/// Soundtouch library version string
#define SOUNDTOUCH_VERSION "1.3.1"
/// SoundTouch library version id
#define SOUNDTOUCH_VERSION_ID 010301
//
// Available setting IDs for the 'setSetting' & 'get_setting' functions:
/// Enable/disable anti-alias filter in pitch transposer (0 = disable)
#define SETTING_USE_AA_FILTER 0
/// Pitch transposer anti-alias filter length (8 .. 128 taps, default = 32)
#define SETTING_AA_FILTER_LENGTH 1
/// Enable/disable quick seeking algorithm in tempo changer routine
/// (enabling quick seeking lowers CPU utilization but causes a minor sound
/// quality compromising)
#define SETTING_USE_QUICKSEEK 2
/// Time-stretch algorithm single processing sequence length in milliseconds. This determines
/// to how long sequences the original sound is chopped in the time-stretch algorithm.
/// See "STTypes.h" or README for more information.
#define SETTING_SEQUENCE_MS 3
/// Time-stretch algorithm seeking window length in milliseconds for algorithm that finds the
/// best possible overlapping location. This determines from how wide window the algorithm
/// may look for an optimal joining location when mixing the sound sequences back together.
/// See "STTypes.h" or README for more information.
#define SETTING_SEEKWINDOW_MS 4
/// Time-stretch algorithm overlap length in milliseconds. When the chopped sound sequences
/// are mixed back together, to form a continuous sound stream, this parameter defines over
/// how long period the two consecutive sequences are let to overlap each other.
/// See "STTypes.h" or README for more information.
#define SETTING_OVERLAP_MS 5
class SoundTouch : public FIFOProcessor
{
private:
/// Rate transposer class instance
class RateTransposer *pRateTransposer;
/// Time-stretch class instance
class TDStretch *pTDStretch;
/// Virtual pitch parameter. Effective rate & tempo are calculated from these parameters.
float virtualRate;
/// Virtual pitch parameter. Effective rate & tempo are calculated from these parameters.
float virtualTempo;
/// Virtual pitch parameter. Effective rate & tempo are calculated from these parameters.
float virtualPitch;
/// Flag: Has sample rate been set?
BOOL bSrateSet;
/// Calculates effective rate & tempo valuescfrom 'virtualRate', 'virtualTempo' and
/// 'virtualPitch' parameters.
void calcEffectiveRateAndTempo();
protected :
/// Number of channels
uint channels;
/// Effective 'rate' value calculated from 'virtualRate', 'virtualTempo' and 'virtualPitch'
float rate;
/// Effective 'tempo' value calculated from 'virtualRate', 'virtualTempo' and 'virtualPitch'
float tempo;
public:
SoundTouch();
virtual ~SoundTouch();
/// Get SoundTouch library version string
static const char *getVersionString();
/// Get SoundTouch library version Id
static uint getVersionId();
/// Sets new rate control value. Normal rate = 1.0, smaller values
/// represent slower rate, larger faster rates.
void setRate(float newRate);
/// Sets new tempo control value. Normal tempo = 1.0, smaller values
/// represent slower tempo, larger faster tempo.
void setTempo(float newTempo);
/// Sets new rate control value as a difference in percents compared
/// to the original rate (-50 .. +100 %)
void setRateChange(float newRate);
/// Sets new tempo control value as a difference in percents compared
/// to the original tempo (-50 .. +100 %)
void setTempoChange(float newTempo);
/// Sets new pitch control value. Original pitch = 1.0, smaller values
/// represent lower pitches, larger values higher pitch.
void setPitch(float newPitch);
/// Sets pitch change in octaves compared to the original pitch
/// (-1.00 .. +1.00)
void setPitchOctaves(float newPitch);
/// Sets pitch change in semi-tones compared to the original pitch
/// (-12 .. +12)
void setPitchSemiTones(int newPitch);
void setPitchSemiTones(float newPitch);
/// Sets the number of channels, 1 = mono, 2 = stereo
void setChannels(uint numChannels);
/// Sets sample rate.
void setSampleRate(uint srate);
/// Flushes the last samples from the processing pipeline to the output.
/// Clears also the internal processing buffers.
//
/// Note: This function is meant for extracting the last samples of a sound
/// stream. This function may introduce additional blank samples in the end
/// of the sound stream, and thus it's not recommended to call this function
/// in the middle of a sound stream.
void flush();
/// Adds 'numSamples' pcs of samples from the 'samples' memory position into
/// the input of the object. Notice that sample rate _has_to_ be set before
/// calling this function, otherwise throws a runtime_error exception.
virtual void putSamples(
const SAMPLETYPE *samples, ///< Pointer to sample buffer.
uint numSamples ///< Number of samples in buffer. Notice
///< that in case of stereo-sound a single sample
///< contains data for both channels.
);
/// Clears all the samples in the object's output and internal processing
/// buffers.
virtual void clear();
/// 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
BOOL setSetting(int settingId, ///< Setting ID number. see SETTING_... defines.
int value ///< New setting value.
);
/// Reads a setting controlling the processing system behaviour. See the
/// 'SETTING_...' defines for available setting ID's.
///
/// \return the setting value.
int getSetting(int settingId ///< Setting ID number, see SETTING_... defines.
) const;
/// Returns number of samples currently unprocessed.
virtual uint numUnprocessedSamples() const;
/// Other handy functions that are implemented in the ancestor classes (see
/// classes 'FIFOProcessor' and 'FIFOSamplePipe')
///
/// - receiveSamples() : Use this function to receive 'ready' processed samples from SoundTouch.
/// - numSamples() : Get number of 'ready' samples that can be received with
/// function 'receiveSamples()'
/// - isEmpty() : Returns nonzero if there aren't any 'ready' samples.
/// - clear() : Clears all samples from ready/processing buffers.
};
}
#endif

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@REM
@REM SoundTouch & SoundStretch Build script for Win32 platform
@REM
@REM You'll need Visual C++ 6.0 installed to compile - also execute the
@REM "vcvars32.bat" in VC install directotry before running this one.
@REM
@REM Copyright (c) Olli Parviainen
@REM File Created: 09/Sep/2003
@REM
@REM $Id: make-win.bat 4 2008-02-10 16:07:12Z oparviai $
@if "%MsDevDir%"=="" goto nodevdir
md bin
md lib
msdev source\SoundTouch\SoundTouch.dsw /MAKE ALL
msdev source\example\bpm\bpm.dsw /MAKE ALL
msdev source\example\SoundStretch\SoundStretch.dsw /MAKE "SoundStretch - Win32 Release"
goto end
:nodevdir
@echo off
echo *********************************************************************
echo **
echo ** ERROR: Visual Studio path not set.
echo **
echo ** Run "vcvars32.bat" from Visual Studio installation directory, e.g.
echo ** "\Program Files\Microsoft Visual Studio\VC98\Bin", then try again.
echo **
echo *********************************************************************
:end

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prefix=@prefix@
exec_prefix=@exec_prefix@
libdir=@libdir@
includedir=@includedir@
Name: SoundTouch
Description: SoundTouch is an open-source audio processing library for changing the Tempo, Pitch and Playback Rates of audio streams or files
Version: @VERSION@
Libs: -L${libdir} -lSoundTouch
Cflags: -I${includedir}/soundtouch

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# m4 configure test script for the SoundTouch library
# (c)2003 David W. Durham
#
# $Id: soundtouch.m4 4 2008-02-10 16:07:12Z oparviai $
#
# This file can be included with other packages that need to test
# for libSoundTouch.
#
# It will #define HAVE_LIBSOUNDTOUCH iff the library is found
# It will AC_SUBST SOUNDTOUCH_LIBS and SOUNDTOUCH_CXXFLAGS as well
# It also defines some flags to the configure script for specifying
# the location to search for libSoundTouch
#
# A user of libSoundTouch should add @SOUNDTOUCH_LIBS@ and
# @SOUNDTOUCH_CXXFLAGS@ to the appropriate variables in his
# Makefile.am files
#
# This script works with autoconf-2.5x and automake-1.6 but I have
# not tested it with older versions.
dnl min version not supported yet
dnl AM_PATH_SOUNDTOUCH([MINMUM-VERSION, [(additional) ACTION-IF-FOUND] [, ACTION-IF-NOT-FOUND]]])
AH_TEMPLATE([HAVE_LIBSOUNDTOUCH], [defined by $0])
SOUNDTOUCH_CXXFLAGS=""
SOUNDTOUCH_LIBS=""
AC_DEFUN([AM_PATH_SOUNDTOUCH],[
AC_ARG_WITH(soundtouch-prefix,[ --with-soundtouch-prefix=DIR Prefix where SoundTouch was installed (optional)], [soundtouch_prefix="$withval"],[soundtouch_prefix=""])
AC_ARG_ENABLE(soundtouch-check,[ --disable-soundtouch-check Do not look for the SoundTouch Library],[enable_soundtouch_check="$enableval"],[enable_soundtouch_check="yes"])
if test "$enable_soundtouch_check" = "yes"
then
saved_CPPFLAGS="$CPPFLAGS"
saved_LDFLAGS="$LDFLAGS"
CPPFLAGS="$CPPFLAGS -I$soundtouch_prefix/include"
LDFLAGS="$LDFLAGS -L$soundtouch_prefix/lib"
dnl make sure SoundTouch.h header file exists
dnl could use AC_CHECK_HEADERS to check for all of them, but the supporting .h file names may change later
AC_CHECK_HEADER([soundtouch/SoundTouch.h],[
dnl SoundTouch.h found
dnl make sure libSoundTouch is linkable
AC_CHECK_LIB([SoundTouch],[soundtouch_ac_test],[
dnl libSoundTouch found
SOUNDTOUCH_CXXFLAGS="-I$soundtouch_prefix/include"
SOUNDTOUCH_LIBS="-L$soundtouch_prefix/lib -lSoundTouch"
AC_DEFINE([HAVE_LIBSOUNDTOUCH])
dnl run action-if-found
ifelse([$2], , :, [$2])
],[
dnl run action-if-not-found
ifelse([$3], , :, [$3])
])
],[
dnl run action-if-not-found
ifelse([$3], , :, [$3])
])
CPPFLAGS="$saved_CPPFLAGS"
LDFLAGS="$saved_LDFLAGS"
fi
AC_SUBST(SOUNDTOUCH_CXXFLAGS)
AC_SUBST(SOUNDTOUCH_LIBS)
])

28
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## Process this file with automake to create Makefile.in
##
## $Id: Makefile.am 4 2008-02-10 16:07:12Z oparviai $
##
## Copyright (C) 2003 - David W. Durham
##
## This file is part of SoundTouch, an audio processing library for pitch/time adjustments
##
## SoundTouch is free software; you can redistribute it and/or modify it under the
## terms of the GNU General Public License as published by the Free Software
## Foundation; either version 2 of the License, or (at your option) any later
## version.
##
## SoundTouch is distributed in the hope that it will be useful, but WITHOUT ANY
## WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
## A PARTICULAR PURPOSE. See the GNU General Public License for more details.
##
## You should have received a copy of the GNU General Public License along with
## this program; if not, write to the Free Software Foundation, Inc., 59 Temple
## Place - Suite 330, Boston, MA 02111-1307, USA
include $(top_srcdir)/config/am_include.mk
SUBDIRS=SoundTouch example
# set to something if you want other stuff to be included in the distribution tarball
#EXTRA_DIST=

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////////////////////////////////////////////////////////////////////////////////
///
/// Win32 version of the AMD 3DNow! optimized routines for AMD K6-2/Athlon
/// processors. All 3DNow! optimized functions have been gathered into this
/// single source code file, regardless to their class or original source code
/// file, in order to ease porting the library to other compiler and processor
/// platforms.
///
/// By the way; the performance gain depends heavily on the CPU generation: On
/// K6-2 these routines provided speed-up of even 2.4 times, while on Athlon the
/// difference to the original routines stayed at unremarkable 8%! Such a small
/// improvement on Athlon is due to 3DNow can perform only two operations in
/// parallel, and obviously also the Athlon FPU is doing a very good job with
/// the standard C floating point routines! Here these routines are anyway,
/// although it might not be worth the effort to convert these to GCC platform,
/// for Athlon CPU at least. The situation is different regarding the SSE
/// optimizations though, thanks to the four parallel operations of SSE that
/// already make a difference.
///
/// This file is to be compiled in Windows platform with Microsoft Visual C++
/// Compiler. Please see '3dnow_gcc.cpp' for the gcc compiler version for all
/// GNU platforms (if file supplied).
///
/// NOTICE: If using Visual Studio 6.0, you'll need to install the "Visual C++
/// 6.0 processor pack" update to support 3DNow! instruction set. The update is
/// available for download at Microsoft Developers Network, see here:
/// http://msdn.microsoft.com/vstudio/downloads/tools/ppack/default.aspx
///
/// If the above URL is expired or removed, go to "http://msdn.microsoft.com" and
/// perform a search with keywords "processor pack".
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2008-02-10 18:07:12 +0200 (su, 10 helmi 2008) $
// File revision : $Revision: 4 $
//
// $Id: 3dnow_win.cpp 4 2008-02-10 16:07:12Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#include "cpu_detect.h"
#include "STTypes.h"
#ifndef WIN32
#error "wrong platform - this source code file is exclusively for Win32 platform"
#endif
using namespace soundtouch;
#ifdef ALLOW_3DNOW
// 3DNow! routines available only with float sample type
//////////////////////////////////////////////////////////////////////////////
//
// implementation of 3DNow! optimized functions of class 'TDStretch3DNow'
//
//////////////////////////////////////////////////////////////////////////////
#include "TDStretch.h"
//#include <limits.h>
// these are declared in 'TDStretch.cpp'
// extern int scanOffsets[4][24];
// Calculates cross correlation of two buffers
double TDStretch3DNow::calcCrossCorrStereo(const float *pV1, const float *pV2) const
{
uint overlapLengthLocal = overlapLength;
float corr;
// Calculates the cross-correlation value between 'pV1' and 'pV2' vectors
/*
c-pseudocode:
corr = 0;
for (i = 0; i < overlapLength / 4; i ++)
{
corr += pV1[0] * pV2[0];
pV1[1] * pV2[1];
pV1[2] * pV2[2];
pV1[3] * pV2[3];
pV1[4] * pV2[4];
pV1[5] * pV2[5];
pV1[6] * pV2[6];
pV1[7] * pV2[7];
pV1 += 8;
pV2 += 8;
}
*/
_asm
{
// give prefetch hints to CPU of what data are to be needed soonish.
// give more aggressive hints on pV1 as that changes more between different calls
// while pV2 stays the same.
prefetch [pV1]
prefetch [pV2]
prefetch [pV1 + 32]
mov eax, dword ptr pV2
mov ebx, dword ptr pV1
pxor mm0, mm0
mov ecx, overlapLengthLocal
shr ecx, 2 // div by four
loop1:
movq mm1, [eax]
prefetch [eax + 32] // give a prefetch hint to CPU what data are to be needed soonish
pfmul mm1, [ebx]
prefetch [ebx + 64] // give a prefetch hint to CPU what data are to be needed soonish
movq mm2, [eax + 8]
pfadd mm0, mm1
pfmul mm2, [ebx + 8]
movq mm3, [eax + 16]
pfadd mm0, mm2
pfmul mm3, [ebx + 16]
movq mm4, [eax + 24]
pfadd mm0, mm3
pfmul mm4, [ebx + 24]
add eax, 32
pfadd mm0, mm4
add ebx, 32
dec ecx
jnz loop1
// add halfs of mm0 together and return the result.
// note: mm1 is used as a dummy parameter only, we actually don't care about it's value
pfacc mm0, mm1
movd corr, mm0
femms
}
return corr;
}
//////////////////////////////////////////////////////////////////////////////
//
// implementation of 3DNow! optimized functions of class 'FIRFilter'
//
//////////////////////////////////////////////////////////////////////////////
#include "FIRFilter.h"
FIRFilter3DNow::FIRFilter3DNow() : FIRFilter()
{
filterCoeffsUnalign = NULL;
filterCoeffsAlign = NULL;
}
FIRFilter3DNow::~FIRFilter3DNow()
{
delete[] filterCoeffsUnalign;
filterCoeffsUnalign = NULL;
filterCoeffsAlign = NULL;
}
// (overloaded) Calculates filter coefficients for 3DNow! routine
void FIRFilter3DNow::setCoefficients(const float *coeffs, uint newLength, uint uResultDivFactor)
{
uint i;
float fDivider;
FIRFilter::setCoefficients(coeffs, newLength, uResultDivFactor);
// Scale the filter coefficients so that it won't be necessary to scale the filtering result
// also rearrange coefficients suitably for 3DNow!
// Ensure that filter coeffs array is aligned to 16-byte boundary
delete[] filterCoeffsUnalign;
filterCoeffsUnalign = new float[2 * newLength + 4];
filterCoeffsAlign = (float *)(((uint)filterCoeffsUnalign + 15) & (uint)-16);
fDivider = (float)resultDivider;
// rearrange the filter coefficients for mmx routines
for (i = 0; i < newLength; i ++)
{
filterCoeffsAlign[2 * i + 0] =
filterCoeffsAlign[2 * i + 1] = coeffs[i + 0] / fDivider;
}
}
// 3DNow!-optimized version of the filter routine for stereo sound
uint FIRFilter3DNow::evaluateFilterStereo(float *dest, const float *src, const uint numSamples) const
{
float *filterCoeffsLocal = filterCoeffsAlign;
uint count = (numSamples - length) & (uint)-2;
uint lengthLocal = length / 4;
assert(length != 0);
assert(count % 2 == 0);
/* original code:
double suml1, suml2;
double sumr1, sumr2;
uint i, j;
for (j = 0; j < count; j += 2)
{
const float *ptr;
suml1 = sumr1 = 0.0;
suml2 = sumr2 = 0.0;
ptr = src;
filterCoeffsLocal = filterCoeffs;
for (i = 0; i < lengthLocal; i ++)
{
// unroll loop for efficiency.
suml1 += ptr[0] * filterCoeffsLocal[0] +
ptr[2] * filterCoeffsLocal[2] +
ptr[4] * filterCoeffsLocal[4] +
ptr[6] * filterCoeffsLocal[6];
sumr1 += ptr[1] * filterCoeffsLocal[1] +
ptr[3] * filterCoeffsLocal[3] +
ptr[5] * filterCoeffsLocal[5] +
ptr[7] * filterCoeffsLocal[7];
suml2 += ptr[8] * filterCoeffsLocal[0] +
ptr[10] * filterCoeffsLocal[2] +
ptr[12] * filterCoeffsLocal[4] +
ptr[14] * filterCoeffsLocal[6];
sumr2 += ptr[9] * filterCoeffsLocal[1] +
ptr[11] * filterCoeffsLocal[3] +
ptr[13] * filterCoeffsLocal[5] +
ptr[15] * filterCoeffsLocal[7];
ptr += 16;
filterCoeffsLocal += 8;
}
dest[0] = (float)suml1;
dest[1] = (float)sumr1;
dest[2] = (float)suml2;
dest[3] = (float)sumr2;
src += 4;
dest += 4;
}
*/
_asm
{
mov eax, dword ptr dest
mov ebx, dword ptr src
mov edx, count
shr edx, 1
loop1:
// "outer loop" : during each round 2*2 output samples are calculated
prefetch [ebx] // give a prefetch hint to CPU what data are to be needed soonish
prefetch [filterCoeffsLocal] // give a prefetch hint to CPU what data are to be needed soonish
mov esi, ebx
mov edi, filterCoeffsLocal
pxor mm0, mm0
pxor mm1, mm1
mov ecx, lengthLocal
loop2:
// "inner loop" : during each round four FIR filter taps are evaluated for 2*2 output samples
movq mm2, [edi]
movq mm3, mm2
prefetch [edi + 32] // give a prefetch hint to CPU what data are to be needed soonish
pfmul mm2, [esi]
prefetch [esi + 32] // give a prefetch hint to CPU what data are to be needed soonish
pfmul mm3, [esi + 8]
movq mm4, [edi + 8]
movq mm5, mm4
pfadd mm0, mm2
pfmul mm4, [esi + 8]
pfadd mm1, mm3
pfmul mm5, [esi + 16]
movq mm2, [edi + 16]
movq mm6, mm2
pfadd mm0, mm4
pfmul mm2, [esi + 16]
pfadd mm1, mm5
pfmul mm6, [esi + 24]
movq mm3, [edi + 24]
movq mm7, mm3
pfadd mm0, mm2
pfmul mm3, [esi + 24]
pfadd mm1, mm6
pfmul mm7, [esi + 32]
add esi, 32
pfadd mm0, mm3
add edi, 32
pfadd mm1, mm7
dec ecx
jnz loop2
movq [eax], mm0
add ebx, 16
movq [eax + 8], mm1
add eax, 16
dec edx
jnz loop1
femms
}
return count;
}
#endif // ALLOW_3DNOW

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////////////////////////////////////////////////////////////////////////////////
///
/// FIR low-pass (anti-alias) filter with filter coefficient design routine and
/// MMX optimization.
///
/// Anti-alias filter is used to prevent folding of high frequencies when
/// transposing the sample rate with interpolation.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2008-02-10 18:07:12 +0200 (su, 10 helmi 2008) $
// File revision : $Revision: 4 $
//
// $Id: AAFilter.cpp 4 2008-02-10 16:07:12Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#include <memory.h>
#include <assert.h>
#include <math.h>
#include <stdlib.h>
#include "AAFilter.h"
#include "FIRFilter.h"
using namespace soundtouch;
#define PI 3.141592655357989
#define TWOPI (2 * PI)
/*****************************************************************************
*
* Implementation of the class 'AAFilter'
*
*****************************************************************************/
AAFilter::AAFilter(uint len)
{
pFIR = FIRFilter::newInstance();
cutoffFreq = 0.5;
setLength(len);
}
AAFilter::~AAFilter()
{
delete pFIR;
}
// Sets new anti-alias filter cut-off edge frequency, scaled to
// sampling frequency (nyquist frequency = 0.5).
// The filter will cut frequencies higher than the given frequency.
void AAFilter::setCutoffFreq(double newCutoffFreq)
{
cutoffFreq = newCutoffFreq;
calculateCoeffs();
}
// Sets number of FIR filter taps
void AAFilter::setLength(const uint newLength)
{
length = newLength;
calculateCoeffs();
}
// Calculates coefficients for a low-pass FIR filter using Hamming window
void AAFilter::calculateCoeffs()
{
uint i;
double cntTemp, temp, tempCoeff,h, w;
double fc2, wc;
double scaleCoeff, sum;
double *work;
SAMPLETYPE *coeffs;
assert(length >= 2);
assert(length % 4 == 0);
assert(cutoffFreq >= 0);
assert(cutoffFreq <= 0.5);
work = new double[length];
coeffs = new SAMPLETYPE[length];
fc2 = 2.0 * cutoffFreq;
wc = PI * fc2;
tempCoeff = TWOPI / (double)length;
sum = 0;
for (i = 0; i < length; i ++)
{
cntTemp = (double)i - (double)(length / 2);
temp = cntTemp * wc;
if (temp != 0)
{
h = fc2 * sin(temp) / temp; // sinc function
}
else
{
h = 1.0;
}
w = 0.54 + 0.46 * cos(tempCoeff * cntTemp); // hamming window
temp = w * h;
work[i] = temp;
// calc net sum of coefficients
sum += temp;
}
// ensure the sum of coefficients is larger than zero
assert(sum > 0);
// ensure we've really designed a lowpass filter...
assert(work[length/2] > 0);
assert(work[length/2 + 1] > -1e-6);
assert(work[length/2 - 1] > -1e-6);
// Calculate a scaling coefficient in such a way that the result can be
// divided by 16384
scaleCoeff = 16384.0f / sum;
for (i = 0; i < length; i ++)
{
// scale & round to nearest integer
temp = work[i] * scaleCoeff;
temp += (temp >= 0) ? 0.5 : -0.5;
// ensure no overfloods
assert(temp >= -32768 && temp <= 32767);
coeffs[i] = (SAMPLETYPE)temp;
}
// Set coefficients. Use divide factor 14 => divide result by 2^14 = 16384
pFIR->setCoefficients(coeffs, length, 14);
delete[] work;
delete[] coeffs;
}
// Applies the filter to the given sequence of samples.
// Note : The amount of outputted samples is by value of 'filter length'
// smaller than the amount of input samples.
uint AAFilter::evaluate(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples, uint numChannels) const
{
return pFIR->evaluate(dest, src, numSamples, numChannels);
}
uint AAFilter::getLength() const
{
return pFIR->getLength();
}

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////////////////////////////////////////////////////////////////////////////////
///
/// Sampled sound tempo changer/time stretch algorithm. Changes the sound tempo
/// while maintaining the original pitch by using a time domain WSOLA-like method
/// with several performance-increasing tweaks.
///
/// Anti-alias filter is used to prevent folding of high frequencies when
/// transposing the sample rate with interpolation.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2008-02-10 18:07:12 +0200 (su, 10 helmi 2008) $
// File revision : $Revision: 4 $
//
// $Id: AAFilter.h 4 2008-02-10 16:07:12Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#ifndef AAFilter_H
#define AAFilter_H
#include "STTypes.h"
namespace soundtouch
{
class AAFilter
{
protected:
class FIRFilter *pFIR;
/// Low-pass filter cut-off frequency, negative = invalid
double cutoffFreq;
/// num of filter taps
uint length;
/// Calculate the FIR coefficients realizing the given cutoff-frequency
void calculateCoeffs();
public:
AAFilter(uint length);
~AAFilter();
/// Sets new anti-alias filter cut-off edge frequency, scaled to sampling
/// frequency (nyquist frequency = 0.5). The filter will cut off the
/// frequencies than that.
void setCutoffFreq(double newCutoffFreq);
/// Sets number of FIR filter taps, i.e. ~filter complexity
void setLength(uint newLength);
uint getLength() const;
/// Applies the filter to the given sequence of samples.
/// Note : The amount of outputted samples is by value of 'filter length'
/// smaller than the amount of input samples.
uint evaluate(SAMPLETYPE *dest,
const SAMPLETYPE *src,
uint numSamples,
uint numChannels) const;
};
}
#endif

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////////////////////////////////////////////////////////////////////////////////
///
/// A buffer class for temporarily storaging sound samples, operates as a
/// first-in-first-out pipe.
///
/// Samples are added to the end of the sample buffer with the 'putSamples'
/// function, and are received from the beginning of the buffer by calling
/// the 'receiveSamples' function. The class automatically removes the
/// outputted samples from the buffer, as well as grows the buffer size
/// whenever necessary.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2008-02-10 18:07:12 +0200 (su, 10 helmi 2008) $
// File revision : $Revision: 4 $
//
// $Id: FIFOSampleBuffer.cpp 4 2008-02-10 16:07:12Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#include <stdlib.h>
#include <memory.h>
#include <string.h>
#include <assert.h>
#include <stdexcept>
#include "FIFOSampleBuffer.h"
using namespace soundtouch;
// Constructor
FIFOSampleBuffer::FIFOSampleBuffer(int numChannels)
{
assert(numChannels > 0);
sizeInBytes = 0; // reasonable initial value
buffer = NULL;
bufferUnaligned = NULL;
samplesInBuffer = 0;
bufferPos = 0;
channels = (uint)numChannels;
}
// destructor
FIFOSampleBuffer::~FIFOSampleBuffer()
{
delete[] bufferUnaligned;
bufferUnaligned = NULL;
buffer = NULL;
}
// Sets number of channels, 1 = mono, 2 = stereo
void FIFOSampleBuffer::setChannels(int numChannels)
{
uint usedBytes;
assert(numChannels > 0);
usedBytes = channels * samplesInBuffer;
channels = (uint)numChannels;
samplesInBuffer = usedBytes / channels;
}
// if output location pointer 'bufferPos' isn't zero, 'rewinds' the buffer and
// zeroes this pointer by copying samples from the 'bufferPos' pointer
// location on to the beginning of the buffer.
void FIFOSampleBuffer::rewind()
{
if (buffer && bufferPos)
{
memmove(buffer, ptrBegin(), sizeof(SAMPLETYPE) * channels * samplesInBuffer);
bufferPos = 0;
}
}
// Adds 'numSamples' pcs of samples from the 'samples' memory position to
// the sample buffer.
void FIFOSampleBuffer::putSamples(const SAMPLETYPE *samples, uint nSamples)
{
memcpy(ptrEnd(nSamples), samples, sizeof(SAMPLETYPE) * nSamples * channels);
samplesInBuffer += nSamples;
}
// Increases the number of samples in the buffer without copying any actual
// samples.
//
// This function is used to update the number of samples in the sample buffer
// when accessing the buffer directly with 'ptrEnd' function. Please be
// careful though!
void FIFOSampleBuffer::putSamples(uint nSamples)
{
uint req;
req = samplesInBuffer + nSamples;
ensureCapacity(req);
samplesInBuffer += nSamples;
}
// Returns a pointer to the end of the used part of the sample buffer (i.e.
// where the new samples are to be inserted). This function may be used for
// inserting new samples into the sample buffer directly. Please be careful!
//
// Parameter 'slackCapacity' tells the function how much free capacity (in
// terms of samples) there _at least_ should be, in order to the caller to
// succesfully insert all the required samples to the buffer. When necessary,
// the function grows the buffer size to comply with this requirement.
//
// When using this function as means for inserting new samples, also remember
// to increase the sample count afterwards, by calling the
// 'putSamples(numSamples)' function.
SAMPLETYPE *FIFOSampleBuffer::ptrEnd(uint slackCapacity)
{
ensureCapacity(samplesInBuffer + slackCapacity);
return buffer + samplesInBuffer * channels;
}
// Returns a pointer to the beginning of the currently non-outputted samples.
// This function is provided for accessing the output samples directly.
// Please be careful!
//
// When using this function to output samples, also remember to 'remove' the
// outputted samples from the buffer by calling the
// 'receiveSamples(numSamples)' function
SAMPLETYPE *FIFOSampleBuffer::ptrBegin() const
{
return buffer + bufferPos * channels;
}
// Ensures that the buffer has enought capacity, i.e. space for _at least_
// 'capacityRequirement' number of samples. The buffer is grown in steps of
// 4 kilobytes to eliminate the need for frequently growing up the buffer,
// as well as to round the buffer size up to the virtual memory page size.
void FIFOSampleBuffer::ensureCapacity(uint capacityRequirement)
{
SAMPLETYPE *tempUnaligned, *temp;
if (capacityRequirement > getCapacity())
{
// enlarge the buffer in 4kbyte steps (round up to next 4k boundary)
sizeInBytes = (capacityRequirement * channels * sizeof(SAMPLETYPE) + 4095) & (uint)-4096;
assert(sizeInBytes % 2 == 0);
tempUnaligned = new SAMPLETYPE[sizeInBytes / sizeof(SAMPLETYPE) + 16 / sizeof(SAMPLETYPE)];
if (tempUnaligned == NULL)
{
throw std::runtime_error("Couldn't allocate memory!\n");
}
temp = (SAMPLETYPE *)(((ulong)tempUnaligned + 15) & (ulong)-16);
memcpy(temp, ptrBegin(), samplesInBuffer * channels * sizeof(SAMPLETYPE));
delete[] bufferUnaligned;
buffer = temp;
bufferUnaligned = tempUnaligned;
bufferPos = 0;
}
else
{
// simply rewind the buffer (if necessary)
rewind();
}
}
// Returns the current buffer capacity in terms of samples
uint FIFOSampleBuffer::getCapacity() const
{
return sizeInBytes / (channels * sizeof(SAMPLETYPE));
}
// Returns the number of samples currently in the buffer
uint FIFOSampleBuffer::numSamples() const
{
return samplesInBuffer;
}
// Output samples from beginning of the sample buffer. Copies demanded number
// of samples to output and removes them from the sample buffer. If there
// are less than 'numsample' samples in the buffer, returns all available.
//
// Returns number of samples copied.
uint FIFOSampleBuffer::receiveSamples(SAMPLETYPE *output, uint maxSamples)
{
uint num;
num = (maxSamples > samplesInBuffer) ? samplesInBuffer : maxSamples;
memcpy(output, ptrBegin(), channels * sizeof(SAMPLETYPE) * num);
return receiveSamples(num);
}
// Removes samples from the beginning of the sample buffer without copying them
// anywhere. Used to reduce the number of samples in the buffer, when accessing
// the sample buffer with the 'ptrBegin' function.
uint FIFOSampleBuffer::receiveSamples(uint maxSamples)
{
if (maxSamples >= samplesInBuffer)
{
uint temp;
temp = samplesInBuffer;
samplesInBuffer = 0;
return temp;
}
samplesInBuffer -= maxSamples;
bufferPos += maxSamples;
return maxSamples;
}
// Returns nonzero if the sample buffer is empty
int FIFOSampleBuffer::isEmpty() const
{
return (samplesInBuffer == 0) ? 1 : 0;
}
// Clears the sample buffer
void FIFOSampleBuffer::clear()
{
samplesInBuffer = 0;
bufferPos = 0;
}

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////////////////////////////////////////////////////////////////////////////////
///
/// General FIR digital filter routines with MMX optimization.
///
/// Note : MMX optimized functions reside in a separate, platform-specific file,
/// e.g. 'mmx_win.cpp' or 'mmx_gcc.cpp'
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2008-02-10 18:07:12 +0200 (su, 10 helmi 2008) $
// File revision : $Revision: 4 $
//
// $Id: FIRFilter.cpp 4 2008-02-10 16:07:12Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#include <memory.h>
#include <assert.h>
#include <math.h>
#include <stdlib.h>
#include <stdexcept>
#include "FIRFilter.h"
#include "cpu_detect.h"
using namespace soundtouch;
/*****************************************************************************
*
* Implementation of the class 'FIRFilter'
*
*****************************************************************************/
FIRFilter::FIRFilter()
{
resultDivFactor = 0;
resultDivider = 0;
length = 0;
lengthDiv8 = 0;
filterCoeffs = NULL;
}
FIRFilter::~FIRFilter()
{
delete[] filterCoeffs;
}
// Usual C-version of the filter routine for stereo sound
uint FIRFilter::evaluateFilterStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples) const
{
uint i, j, end;
LONG_SAMPLETYPE suml, sumr;
#ifdef FLOAT_SAMPLES
// when using floating point samples, use a scaler instead of a divider
// because division is much slower operation than multiplying.
double dScaler = 1.0 / (double)resultDivider;
#endif
assert(length != 0);
assert(src != NULL);
assert(dest != NULL);
assert(filterCoeffs != NULL);
end = 2 * (numSamples - length);
for (j = 0; j < end; j += 2)
{
const SAMPLETYPE *ptr;
suml = sumr = 0;
ptr = src + j;
for (i = 0; i < length; i += 4)
{
// loop is unrolled by factor of 4 here for efficiency
suml += ptr[2 * i + 0] * filterCoeffs[i + 0] +
ptr[2 * i + 2] * filterCoeffs[i + 1] +
ptr[2 * i + 4] * filterCoeffs[i + 2] +
ptr[2 * i + 6] * filterCoeffs[i + 3];
sumr += ptr[2 * i + 1] * filterCoeffs[i + 0] +
ptr[2 * i + 3] * filterCoeffs[i + 1] +
ptr[2 * i + 5] * filterCoeffs[i + 2] +
ptr[2 * i + 7] * filterCoeffs[i + 3];
}
#ifdef INTEGER_SAMPLES
suml >>= resultDivFactor;
sumr >>= resultDivFactor;
// saturate to 16 bit integer limits
suml = (suml < -32768) ? -32768 : (suml > 32767) ? 32767 : suml;
// saturate to 16 bit integer limits
sumr = (sumr < -32768) ? -32768 : (sumr > 32767) ? 32767 : sumr;
#else
suml *= dScaler;
sumr *= dScaler;
#endif // INTEGER_SAMPLES
dest[j] = (SAMPLETYPE)suml;
dest[j + 1] = (SAMPLETYPE)sumr;
}
return numSamples - length;
}
// Usual C-version of the filter routine for mono sound
uint FIRFilter::evaluateFilterMono(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples) const
{
uint i, j, end;
LONG_SAMPLETYPE sum;
#ifdef FLOAT_SAMPLES
// when using floating point samples, use a scaler instead of a divider
// because division is much slower operation than multiplying.
double dScaler = 1.0 / (double)resultDivider;
#endif
assert(length != 0);
end = numSamples - length;
for (j = 0; j < end; j ++)
{
sum = 0;
for (i = 0; i < length; i += 4)
{
// loop is unrolled by factor of 4 here for efficiency
sum += src[i + 0] * filterCoeffs[i + 0] +
src[i + 1] * filterCoeffs[i + 1] +
src[i + 2] * filterCoeffs[i + 2] +
src[i + 3] * filterCoeffs[i + 3];
}
#ifdef INTEGER_SAMPLES
sum >>= resultDivFactor;
// saturate to 16 bit integer limits
sum = (sum < -32768) ? -32768 : (sum > 32767) ? 32767 : sum;
#else
sum *= dScaler;
#endif // INTEGER_SAMPLES
dest[j] = (SAMPLETYPE)sum;
src ++;
}
return end;
}
// Set filter coeffiecients and length.
//
// Throws an exception if filter length isn't divisible by 8
void FIRFilter::setCoefficients(const SAMPLETYPE *coeffs, uint newLength, uint uResultDivFactor)
{
assert(newLength > 0);
if (newLength % 8) throw std::runtime_error("FIR filter length not divisible by 8");
lengthDiv8 = newLength / 8;
length = lengthDiv8 * 8;
assert(length == newLength);
resultDivFactor = uResultDivFactor;
resultDivider = (SAMPLETYPE)pow(2, resultDivFactor);
delete[] filterCoeffs;
filterCoeffs = new SAMPLETYPE[length];
memcpy(filterCoeffs, coeffs, length * sizeof(SAMPLETYPE));
}
uint FIRFilter::getLength() const
{
return length;
}
// Applies the filter to the given sequence of samples.
//
// Note : The amount of outputted samples is by value of 'filter_length'
// smaller than the amount of input samples.
uint FIRFilter::evaluate(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples, uint numChannels) const
{
assert(numChannels == 1 || numChannels == 2);
assert(length > 0);
assert(lengthDiv8 * 8 == length);
if (numSamples < length) return 0;
if (numChannels == 2)
{
return evaluateFilterStereo(dest, src, numSamples);
} else {
return evaluateFilterMono(dest, src, numSamples);
}
}
// Operator 'new' is overloaded so that it automatically creates a suitable instance
// depending on if we've a MMX-capable CPU available or not.
void * FIRFilter::operator new(size_t s)
{
// Notice! don't use "new FIRFilter" directly, use "newInstance" to create a new instance instead!
throw std::runtime_error("Error in FIRFilter::new: Don't use 'new FIRFilter', use 'newInstance' member instead!");
return NULL;
}
FIRFilter * FIRFilter::newInstance()
{
uint uExtensions;
uExtensions = detectCPUextensions();
// Check if MMX/SSE/3DNow! instruction set extensions supported by CPU
#ifdef ALLOW_MMX
// MMX routines available only with integer sample types
if (uExtensions & SUPPORT_MMX)
{
return ::new FIRFilterMMX;
}
else
#endif // ALLOW_MMX
#ifdef ALLOW_SSE
if (uExtensions & SUPPORT_SSE)
{
// SSE support
return ::new FIRFilterSSE;
}
else
#endif // ALLOW_SSE
#ifdef ALLOW_3DNOW
if (uExtensions & SUPPORT_3DNOW)
{
// 3DNow! support
return ::new FIRFilter3DNow;
}
else
#endif // ALLOW_3DNOW
{
// ISA optimizations not supported, use plain C version
return ::new FIRFilter;
}
}

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////////////////////////////////////////////////////////////////////////////////
///
/// General FIR digital filter routines with MMX optimization.
///
/// Note : MMX optimized functions reside in a separate, platform-specific file,
/// e.g. 'mmx_win.cpp' or 'mmx_gcc.cpp'
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2008-02-10 18:07:12 +0200 (su, 10 helmi 2008) $
// File revision : $Revision: 4 $
//
// $Id: FIRFilter.h 4 2008-02-10 16:07:12Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#ifndef FIRFilter_H
#define FIRFilter_H
#include "STTypes.h"
namespace soundtouch
{
class FIRFilter
{
protected:
// Number of FIR filter taps
uint length;
// Number of FIR filter taps divided by 8
uint lengthDiv8;
// Result divider factor in 2^k format
uint resultDivFactor;
// Result divider value.
SAMPLETYPE resultDivider;
// Memory for filter coefficients
SAMPLETYPE *filterCoeffs;
virtual uint evaluateFilterStereo(SAMPLETYPE *dest,
const SAMPLETYPE *src,
uint numSamples) const;
virtual uint evaluateFilterMono(SAMPLETYPE *dest,
const SAMPLETYPE *src,
uint numSamples) const;
public:
FIRFilter();
virtual ~FIRFilter();
/// Operator 'new' is overloaded so that it automatically creates a suitable instance
/// depending on if we've a MMX-capable CPU available or not.
static void * operator new(size_t s);
static FIRFilter *newInstance();
/// Applies the filter to the given sequence of samples.
/// Note : The amount of outputted samples is by value of 'filter_length'
/// smaller than the amount of input samples.
///
/// \return Number of samples copied to 'dest'.
uint evaluate(SAMPLETYPE *dest,
const SAMPLETYPE *src,
uint numSamples,
uint numChannels) const;
uint getLength() const;
virtual void setCoefficients(const SAMPLETYPE *coeffs,
uint newLength,
uint uResultDivFactor);
};
// Optional subclasses that implement CPU-specific optimizations:
#ifdef ALLOW_MMX
/// Class that implements MMX optimized functions exclusive for 16bit integer samples type.
class FIRFilterMMX : public FIRFilter
{
protected:
short *filterCoeffsUnalign;
short *filterCoeffsAlign;
virtual uint evaluateFilterStereo(short *dest, const short *src, uint numSamples) const;
public:
FIRFilterMMX();
~FIRFilterMMX();
virtual void setCoefficients(const short *coeffs, uint newLength, uint uResultDivFactor);
};
#endif // ALLOW_MMX
#ifdef ALLOW_3DNOW
/// Class that implements 3DNow! optimized functions exclusive for floating point samples type.
class FIRFilter3DNow : public FIRFilter
{
protected:
float *filterCoeffsUnalign;
float *filterCoeffsAlign;
virtual uint evaluateFilterStereo(float *dest, const float *src, uint numSamples) const;
public:
FIRFilter3DNow();
~FIRFilter3DNow();
virtual void setCoefficients(const float *coeffs, uint newLength, uint uResultDivFactor);
};
#endif // ALLOW_3DNOW
#ifdef ALLOW_SSE
/// Class that implements SSE optimized functions exclusive for floating point samples type.
class FIRFilterSSE : public FIRFilter
{
protected:
float *filterCoeffsUnalign;
float *filterCoeffsAlign;
virtual uint evaluateFilterStereo(float *dest, const float *src, uint numSamples) const;
public:
FIRFilterSSE();
~FIRFilterSSE();
virtual void setCoefficients(const float *coeffs, uint newLength, uint uResultDivFactor);
};
#endif // ALLOW_SSE
}
#endif // FIRFilter_H

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## Process this file with automake to create Makefile.in
##
## $Id: Makefile.am 4 2008-02-10 16:07:12Z oparviai $
##
## Copyright (C) 2003 - David W. Durham
##
## This file is part of SoundTouch, an audio processing library for pitch/time adjustments
##
## SoundTouch is free software; you can redistribute it and/or modify it under the
## terms of the GNU General Public License as published by the Free Software
## Foundation; either version 2 of the License, or (at your option) any later
## version.
##
## SoundTouch is distributed in the hope that it will be useful, but WITHOUT ANY
## WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
## A PARTICULAR PURPOSE. See the GNU General Public License for more details.
##
## You should have received a copy of the GNU General Public License along with
## this program; if not, write to the Free Software Foundation, Inc., 59 Temple
## Place - Suite 330, Boston, MA 02111-1307, USA
include $(top_srcdir)/config/am_include.mk
# set to something if you want other stuff to be included in the distribution tarball
EXTRA_DIST=3dnow_win.cpp cpu_detect_x86_win.cpp SoundTouch.dsp SoundTouch.dsw SoundTouch.sln SoundTouch.vcproj
noinst_HEADERS=AAFilter.h cpu_detect.h cpu_detect_x86_gcc.cpp FIRFilter.h RateTransposer.h TDStretch.h
lib_LTLIBRARIES=libSoundTouch.la
# the mmx_gcc.cpp and cpu_detect_x86_gcc.cpp may need to be conditionally included here from things discovered in configure.ac
libSoundTouch_la_SOURCES=AAFilter.cpp FIRFilter.cpp FIFOSampleBuffer.cpp mmx_optimized.cpp sse_optimized.cpp RateTransposer.cpp SoundTouch.cpp TDStretch.cpp cpu_detect_x86_gcc.cpp
# ??? test for -fcheck-new in configure.ac
# other compiler flags to add
AM_CXXFLAGS=-O3 -msse -fcheck-new -I../../include
# other linking flags to add
#libSoundTouch_la_LIBADD=

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////////////////////////////////////////////////////////////////////////////////
///
/// Sample rate transposer. Changes sample rate by using linear interpolation
/// together with anti-alias filtering (first order interpolation with anti-
/// alias filtering should be quite adequate for this application)
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2008-02-10 18:07:12 +0200 (su, 10 helmi 2008) $
// File revision : $Revision: 4 $
//
// $Id: RateTransposer.cpp 4 2008-02-10 16:07:12Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#include <memory.h>
#include <assert.h>
#include <stdlib.h>
#include <stdio.h>
#include <stdexcept>
#include "RateTransposer.h"
#include "AAFilter.h"
using namespace std;
using namespace soundtouch;
/// A linear samplerate transposer class that uses integer arithmetics.
/// for the transposing.
class RateTransposerInteger : public RateTransposer
{
protected:
int iSlopeCount;
int iRate;
SAMPLETYPE sPrevSampleL, sPrevSampleR;
virtual void resetRegisters();
virtual uint transposeStereo(SAMPLETYPE *dest,
const SAMPLETYPE *src,
uint numSamples);
virtual uint transposeMono(SAMPLETYPE *dest,
const SAMPLETYPE *src,
uint numSamples);
public:
RateTransposerInteger();
virtual ~RateTransposerInteger();
/// Sets new target rate. Normal rate = 1.0, smaller values represent slower
/// rate, larger faster rates.
virtual void setRate(float newRate);
};
/// A linear samplerate transposer class that uses floating point arithmetics
/// for the transposing.
class RateTransposerFloat : public RateTransposer
{
protected:
float fSlopeCount;
SAMPLETYPE sPrevSampleL, sPrevSampleR;
virtual void resetRegisters();
virtual uint transposeStereo(SAMPLETYPE *dest,
const SAMPLETYPE *src,
uint numSamples);
virtual uint transposeMono(SAMPLETYPE *dest,
const SAMPLETYPE *src,
uint numSamples);
public:
RateTransposerFloat();
virtual ~RateTransposerFloat();
};
// Operator 'new' is overloaded so that it automatically creates a suitable instance
// depending on if we've a MMX/SSE/etc-capable CPU available or not.
void * RateTransposer::operator new(size_t s)
{
throw runtime_error("Error in RateTransoser::new: don't use \"new TDStretch\" directly, use \"newInstance\" to create a new instance instead!");
return NULL;
}
RateTransposer *RateTransposer::newInstance()
{
#ifdef INTEGER_SAMPLES
return ::new RateTransposerInteger;
#else
return ::new RateTransposerFloat;
#endif
}
// Constructor
RateTransposer::RateTransposer() : FIFOProcessor(&outputBuffer)
{
numChannels = 2;
bUseAAFilter = TRUE;
fRate = 0;
// Instantiates the anti-alias filter with default tap length
// of 32
pAAFilter = new AAFilter(32);
}
RateTransposer::~RateTransposer()
{
delete pAAFilter;
}
/// Enables/disables the anti-alias filter. Zero to disable, nonzero to enable
void RateTransposer::enableAAFilter(const BOOL newMode)
{
bUseAAFilter = newMode;
}
/// Returns nonzero if anti-alias filter is enabled.
BOOL RateTransposer::isAAFilterEnabled() const
{
return bUseAAFilter;
}
AAFilter *RateTransposer::getAAFilter() const
{
return pAAFilter;
}
// Sets new target iRate. Normal iRate = 1.0, smaller values represent slower
// iRate, larger faster iRates.
void RateTransposer::setRate(float newRate)
{
double fCutoff;
fRate = newRate;
// design a new anti-alias filter
if (newRate > 1.0f)
{
fCutoff = 0.5f / newRate;
}
else
{
fCutoff = 0.5f * newRate;
}
pAAFilter->setCutoffFreq(fCutoff);
}
// Outputs as many samples of the 'outputBuffer' as possible, and if there's
// any room left, outputs also as many of the incoming samples as possible.
// The goal is to drive the outputBuffer empty.
//
// It's allowed for 'output' and 'input' parameters to point to the same
// memory position.
/*
void RateTransposer::flushStoreBuffer()
{
if (storeBuffer.isEmpty()) return;
outputBuffer.moveSamples(storeBuffer);
}
*/
// Adds 'nSamples' pcs of samples from the 'samples' memory position into
// the input of the object.
void RateTransposer::putSamples(const SAMPLETYPE *samples, uint nSamples)
{
processSamples(samples, nSamples);
}
// Transposes up the sample rate, causing the observed playback 'rate' of the
// sound to decrease
void RateTransposer::upsample(const SAMPLETYPE *src, uint nSamples)
{
uint count, sizeTemp, num;
// If the parameter 'uRate' value is smaller than 'SCALE', first transpose
// the samples and then apply the anti-alias filter to remove aliasing.
// First check that there's enough room in 'storeBuffer'
// (+16 is to reserve some slack in the destination buffer)
sizeTemp = (uint)((float)nSamples / fRate + 16.0f);
// Transpose the samples, store the result into the end of "storeBuffer"
count = transpose(storeBuffer.ptrEnd(sizeTemp), src, nSamples);
storeBuffer.putSamples(count);
// Apply the anti-alias filter to samples in "store output", output the
// result to "dest"
num = storeBuffer.numSamples();
count = pAAFilter->evaluate(outputBuffer.ptrEnd(num),
storeBuffer.ptrBegin(), num, (uint)numChannels);
outputBuffer.putSamples(count);
// Remove the processed samples from "storeBuffer"
storeBuffer.receiveSamples(count);
}
// Transposes down the sample rate, causing the observed playback 'rate' of the
// sound to increase
void RateTransposer::downsample(const SAMPLETYPE *src, uint nSamples)
{
uint count, sizeTemp;
// If the parameter 'uRate' value is larger than 'SCALE', first apply the
// anti-alias filter to remove high frequencies (prevent them from folding
// over the lover frequencies), then transpose. */
// Add the new samples to the end of the storeBuffer */
storeBuffer.putSamples(src, nSamples);
// Anti-alias filter the samples to prevent folding and output the filtered
// data to tempBuffer. Note : because of the FIR filter length, the
// filtering routine takes in 'filter_length' more samples than it outputs.
assert(tempBuffer.isEmpty());
sizeTemp = storeBuffer.numSamples();
count = pAAFilter->evaluate(tempBuffer.ptrEnd(sizeTemp),
storeBuffer.ptrBegin(), sizeTemp, (uint)numChannels);
// Remove the filtered samples from 'storeBuffer'
storeBuffer.receiveSamples(count);
// Transpose the samples (+16 is to reserve some slack in the destination buffer)
sizeTemp = (uint)((float)nSamples / fRate + 16.0f);
count = transpose(outputBuffer.ptrEnd(sizeTemp), tempBuffer.ptrBegin(), count);
outputBuffer.putSamples(count);
}
// Transposes sample rate by applying anti-alias filter to prevent folding.
// Returns amount of samples returned in the "dest" buffer.
// The maximum amount of samples that can be returned at a time is set by
// the 'set_returnBuffer_size' function.
void RateTransposer::processSamples(const SAMPLETYPE *src, uint nSamples)
{
uint count;
uint sizeReq;
if (nSamples == 0) return;
assert(pAAFilter);
// If anti-alias filter is turned off, simply transpose without applying
// the filter
if (bUseAAFilter == FALSE)
{
sizeReq = (uint)((float)nSamples / fRate + 1.0f);
count = transpose(outputBuffer.ptrEnd(sizeReq), src, nSamples);
outputBuffer.putSamples(count);
return;
}
// Transpose with anti-alias filter
if (fRate < 1.0f)
{
upsample(src, nSamples);
}
else
{
downsample(src, nSamples);
}
}
// Transposes the sample rate of the given samples using linear interpolation.
// Returns the number of samples returned in the "dest" buffer
inline uint RateTransposer::transpose(SAMPLETYPE *dest, const SAMPLETYPE *src, uint nSamples)
{
if (numChannels == 2)
{
return transposeStereo(dest, src, nSamples);
}
else
{
return transposeMono(dest, src, nSamples);
}
}
// Sets the number of channels, 1 = mono, 2 = stereo
void RateTransposer::setChannels(int nChannels)
{
assert(nChannels > 0);
if (numChannels == nChannels) return;
assert(nChannels == 1 || nChannels == 2);
numChannels = nChannels;
storeBuffer.setChannels(numChannels);
tempBuffer.setChannels(numChannels);
outputBuffer.setChannels(numChannels);
// Inits the linear interpolation registers
resetRegisters();
}
// Clears all the samples in the object
void RateTransposer::clear()
{
outputBuffer.clear();
storeBuffer.clear();
}
// Returns nonzero if there aren't any samples available for outputting.
int RateTransposer::isEmpty() const
{
int res;
res = FIFOProcessor::isEmpty();
if (res == 0) return 0;
return storeBuffer.isEmpty();
}
//////////////////////////////////////////////////////////////////////////////
//
// RateTransposerInteger - integer arithmetic implementation
//
/// fixed-point interpolation routine precision
#define SCALE 65536
// Constructor
RateTransposerInteger::RateTransposerInteger() : RateTransposer()
{
// Notice: use local function calling syntax for sake of clarity,
// to indicate the fact that C++ constructor can't call virtual functions.
RateTransposerInteger::resetRegisters();
RateTransposerInteger::setRate(1.0f);
}
RateTransposerInteger::~RateTransposerInteger()
{
}
void RateTransposerInteger::resetRegisters()
{
iSlopeCount = 0;
sPrevSampleL =
sPrevSampleR = 0;
}
// Transposes the sample rate of the given samples using linear interpolation.
// 'Mono' version of the routine. Returns the number of samples returned in
// the "dest" buffer
uint RateTransposerInteger::transposeMono(SAMPLETYPE *dest, const SAMPLETYPE *src, uint nSamples)
{
unsigned int i, used;
LONG_SAMPLETYPE temp, vol1;
used = 0;
i = 0;
// Process the last sample saved from the previous call first...
while (iSlopeCount <= SCALE)
{
vol1 = (LONG_SAMPLETYPE)(SCALE - iSlopeCount);
temp = vol1 * sPrevSampleL + iSlopeCount * src[0];
dest[i] = (SAMPLETYPE)(temp / SCALE);
i++;
iSlopeCount += iRate;
}
// now always (iSlopeCount > SCALE)
iSlopeCount -= SCALE;
while (1)
{
while (iSlopeCount > SCALE)
{
iSlopeCount -= SCALE;
used ++;
if (used >= nSamples - 1) goto end;
}
vol1 = (LONG_SAMPLETYPE)(SCALE - iSlopeCount);
temp = src[used] * vol1 + iSlopeCount * src[used + 1];
dest[i] = (SAMPLETYPE)(temp / SCALE);
i++;
iSlopeCount += iRate;
}
end:
// Store the last sample for the next round
sPrevSampleL = src[nSamples - 1];
return i;
}
// Transposes the sample rate of the given samples using linear interpolation.
// 'Stereo' version of the routine. Returns the number of samples returned in
// the "dest" buffer
uint RateTransposerInteger::transposeStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, uint nSamples)
{
unsigned int srcPos, i, used;
LONG_SAMPLETYPE temp, vol1;
if (nSamples == 0) return 0; // no samples, no work
used = 0;
i = 0;
// Process the last sample saved from the sPrevSampleLious call first...
while (iSlopeCount <= SCALE)
{
vol1 = (LONG_SAMPLETYPE)(SCALE - iSlopeCount);
temp = vol1 * sPrevSampleL + iSlopeCount * src[0];
dest[2 * i] = (SAMPLETYPE)(temp / SCALE);
temp = vol1 * sPrevSampleR + iSlopeCount * src[1];
dest[2 * i + 1] = (SAMPLETYPE)(temp / SCALE);
i++;
iSlopeCount += iRate;
}
// now always (iSlopeCount > SCALE)
iSlopeCount -= SCALE;
while (1)
{
while (iSlopeCount > SCALE)
{
iSlopeCount -= SCALE;
used ++;
if (used >= nSamples - 1) goto end;
}
srcPos = 2 * used;
vol1 = (LONG_SAMPLETYPE)(SCALE - iSlopeCount);
temp = src[srcPos] * vol1 + iSlopeCount * src[srcPos + 2];
dest[2 * i] = (SAMPLETYPE)(temp / SCALE);
temp = src[srcPos + 1] * vol1 + iSlopeCount * src[srcPos + 3];
dest[2 * i + 1] = (SAMPLETYPE)(temp / SCALE);
i++;
iSlopeCount += iRate;
}
end:
// Store the last sample for the next round
sPrevSampleL = src[2 * nSamples - 2];
sPrevSampleR = src[2 * nSamples - 1];
return i;
}
// Sets new target iRate. Normal iRate = 1.0, smaller values represent slower
// iRate, larger faster iRates.
void RateTransposerInteger::setRate(float newRate)
{
iRate = (int)(newRate * SCALE + 0.5f);
RateTransposer::setRate(newRate);
}
//////////////////////////////////////////////////////////////////////////////
//
// RateTransposerFloat - floating point arithmetic implementation
//
//////////////////////////////////////////////////////////////////////////////
// Constructor
RateTransposerFloat::RateTransposerFloat() : RateTransposer()
{
// Notice: use local function calling syntax for sake of clarity,
// to indicate the fact that C++ constructor can't call virtual functions.
RateTransposerFloat::resetRegisters();
RateTransposerFloat::setRate(1.0f);
}
RateTransposerFloat::~RateTransposerFloat()
{
}
void RateTransposerFloat::resetRegisters()
{
fSlopeCount = 0;
sPrevSampleL =
sPrevSampleR = 0;
}
// Transposes the sample rate of the given samples using linear interpolation.
// 'Mono' version of the routine. Returns the number of samples returned in
// the "dest" buffer
uint RateTransposerFloat::transposeMono(SAMPLETYPE *dest, const SAMPLETYPE *src, uint nSamples)
{
unsigned int i, used;
used = 0;
i = 0;
// Process the last sample saved from the previous call first...
while (fSlopeCount <= 1.0f)
{
dest[i] = (SAMPLETYPE)((1.0f - fSlopeCount) * sPrevSampleL + fSlopeCount * src[0]);
i++;
fSlopeCount += fRate;
}
fSlopeCount -= 1.0f;
if (nSamples == 1) goto end;
while (1)
{
while (fSlopeCount > 1.0f)
{
fSlopeCount -= 1.0f;
used ++;
if (used >= nSamples - 1) goto end;
}
dest[i] = (SAMPLETYPE)((1.0f - fSlopeCount) * src[used] + fSlopeCount * src[used + 1]);
i++;
fSlopeCount += fRate;
}
end:
// Store the last sample for the next round
sPrevSampleL = src[nSamples - 1];
return i;
}
// Transposes the sample rate of the given samples using linear interpolation.
// 'Mono' version of the routine. Returns the number of samples returned in
// the "dest" buffer
uint RateTransposerFloat::transposeStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, uint nSamples)
{
unsigned int srcPos, i, used;
if (nSamples == 0) return 0; // no samples, no work
used = 0;
i = 0;
// Process the last sample saved from the sPrevSampleLious call first...
while (fSlopeCount <= 1.0f)
{
dest[2 * i] = (SAMPLETYPE)((1.0f - fSlopeCount) * sPrevSampleL + fSlopeCount * src[0]);
dest[2 * i + 1] = (SAMPLETYPE)((1.0f - fSlopeCount) * sPrevSampleR + fSlopeCount * src[1]);
i++;
fSlopeCount += fRate;
}
// now always (iSlopeCount > 1.0f)
fSlopeCount -= 1.0f;
if (nSamples == 1) goto end;
while (1)
{
while (fSlopeCount > 1.0f)
{
fSlopeCount -= 1.0f;
used ++;
if (used >= nSamples - 1) goto end;
}
srcPos = 2 * used;
dest[2 * i] = (SAMPLETYPE)((1.0f - fSlopeCount) * src[srcPos]
+ fSlopeCount * src[srcPos + 2]);
dest[2 * i + 1] = (SAMPLETYPE)((1.0f - fSlopeCount) * src[srcPos + 1]
+ fSlopeCount * src[srcPos + 3]);
i++;
fSlopeCount += fRate;
}
end:
// Store the last sample for the next round
sPrevSampleL = src[2 * nSamples - 2];
sPrevSampleR = src[2 * nSamples - 1];
return i;
}

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////////////////////////////////////////////////////////////////////////////////
///
/// Sample rate transposer. Changes sample rate by using linear interpolation
/// together with anti-alias filtering (first order interpolation with anti-
/// alias filtering should be quite adequate for this application).
///
/// Use either of the derived classes of 'RateTransposerInteger' or
/// 'RateTransposerFloat' for corresponding integer/floating point tranposing
/// algorithm implementation.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2008-02-10 18:07:12 +0200 (su, 10 helmi 2008) $
// File revision : $Revision: 4 $
//
// $Id: RateTransposer.h 4 2008-02-10 16:07:12Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#ifndef RateTransposer_H
#define RateTransposer_H
#include "AAFilter.h"
#include "FIFOSamplePipe.h"
#include "FIFOSampleBuffer.h"
#include "STTypes.h"
namespace soundtouch
{
/// A common linear samplerate transposer class.
///
/// Note: Use function "RateTransposer::newInstance()" to create a new class
/// instance instead of the "new" operator; that function automatically
/// chooses a correct implementation depending on if integer or floating
/// arithmetics are to be used.
class RateTransposer : public FIFOProcessor
{
protected:
/// Anti-alias filter object
AAFilter *pAAFilter;
float fRate;
int numChannels;
/// Buffer for collecting samples to feed the anti-alias filter between
/// two batches
FIFOSampleBuffer storeBuffer;
/// Buffer for keeping samples between transposing & anti-alias filter
FIFOSampleBuffer tempBuffer;
/// Output sample buffer
FIFOSampleBuffer outputBuffer;
BOOL bUseAAFilter;
virtual void resetRegisters() = 0;
virtual uint transposeStereo(SAMPLETYPE *dest,
const SAMPLETYPE *src,
uint numSamples) = 0;
virtual uint transposeMono(SAMPLETYPE *dest,
const SAMPLETYPE *src,
uint numSamples) = 0;
uint transpose(SAMPLETYPE *dest,
const SAMPLETYPE *src,
uint numSamples);
void downsample(const SAMPLETYPE *src,
uint numSamples);
void upsample(const SAMPLETYPE *src,
uint numSamples);
/// Transposes sample rate by applying anti-alias filter to prevent folding.
/// Returns amount of samples returned in the "dest" buffer.
/// The maximum amount of samples that can be returned at a time is set by
/// the 'set_returnBuffer_size' function.
void processSamples(const SAMPLETYPE *src,
uint numSamples);
public:
RateTransposer();
virtual ~RateTransposer();
/// Operator 'new' is overloaded so that it automatically creates a suitable instance
/// depending on if we're to use integer or floating point arithmetics.
static void *operator new(size_t s);
/// Use this function instead of "new" operator to create a new instance of this class.
/// This function automatically chooses a correct implementation, depending on if
/// integer ot floating point arithmetics are to be used.
static RateTransposer *newInstance();
/// Returns the output buffer object
FIFOSamplePipe *getOutput() { return &outputBuffer; };
/// Returns the store buffer object
FIFOSamplePipe *getStore() { return &storeBuffer; };
/// Return anti-alias filter object
AAFilter *getAAFilter() const;
/// Enables/disables the anti-alias filter. Zero to disable, nonzero to enable
void enableAAFilter(BOOL newMode);
/// Returns nonzero if anti-alias filter is enabled.
BOOL isAAFilterEnabled() const;
/// Sets new target rate. Normal rate = 1.0, smaller values represent slower
/// rate, larger faster rates.
virtual void setRate(float newRate);
/// Sets the number of channels, 1 = mono, 2 = stereo
void setChannels(int channels);
/// Adds 'numSamples' pcs of samples from the 'samples' memory position into
/// the input of the object.
void putSamples(const SAMPLETYPE *samples, uint numSamples);
/// Clears all the samples in the object
void clear();
/// Returns nonzero if there aren't any samples available for outputting.
int isEmpty() const;
};
}
#endif

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//////////////////////////////////////////////////////////////////////////////
///
/// SoundTouch - main class for tempo/pitch/rate adjusting routines.
///
/// Notes:
/// - Initialize the SoundTouch object instance by setting up the sound stream
/// parameters with functions 'setSampleRate' and 'setChannels', then set
/// desired tempo/pitch/rate settings with the corresponding functions.
///
/// - The SoundTouch class behaves like a first-in-first-out pipeline: The
/// samples that are to be processed are fed into one of the pipe by calling
/// function 'putSamples', while the ready processed samples can be read
/// from the other end of the pipeline with function 'receiveSamples'.
///
/// - The SoundTouch processing classes require certain sized 'batches' of
/// samples in order to process the sound. For this reason the classes buffer
/// incoming samples until there are enough of samples available for
/// processing, then they carry out the processing step and consequently
/// make the processed samples available for outputting.
///
/// - For the above reason, the processing routines introduce a certain
/// 'latency' between the input and output, so that the samples input to
/// SoundTouch may not be immediately available in the output, and neither
/// the amount of outputtable samples may not immediately be in direct
/// relationship with the amount of previously input samples.
///
/// - The tempo/pitch/rate control parameters can be altered during processing.
/// Please notice though that they aren't currently protected by semaphores,
/// so in multi-thread application external semaphore protection may be
/// required.
///
/// - This class utilizes classes 'TDStretch' for tempo change (without modifying
/// pitch) and 'RateTransposer' for changing the playback rate (that is, both
/// tempo and pitch in the same ratio) of the sound. The third available control
/// 'pitch' (change pitch but maintain tempo) is produced by a combination of
/// combining the two other controls.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2008-02-10 18:07:12 +0200 (su, 10 helmi 2008) $
// File revision : $Revision: 4 $
//
// $Id: SoundTouch.cpp 4 2008-02-10 16:07:12Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#include <assert.h>
#include <stdlib.h>
#include <memory.h>
#include <math.h>
#include <stdexcept>
#include <stdio.h>
#include "SoundTouch.h"
#include "TDStretch.h"
#include "RateTransposer.h"
#include "cpu_detect.h"
using namespace soundtouch;
/// test if two floating point numbers are equal
#define TEST_FLOAT_EQUAL(a, b) (fabs(a - b) < 1e-10)
/// Print library version string
extern "C" void soundtouch_ac_test()
{
printf("SoundTouch Version: %s\n",SOUNDTOUCH_VERSION);
}
SoundTouch::SoundTouch()
{
// Initialize rate transposer and tempo changer instances
pRateTransposer = RateTransposer::newInstance();
pTDStretch = TDStretch::newInstance();
setOutPipe(pTDStretch);
rate = tempo = 0;
virtualPitch =
virtualRate =
virtualTempo = 1.0;
calcEffectiveRateAndTempo();
channels = 0;
bSrateSet = FALSE;
}
SoundTouch::~SoundTouch()
{
delete pRateTransposer;
delete pTDStretch;
}
/// Get SoundTouch library version string
const char *SoundTouch::getVersionString()
{
static const char *_version = SOUNDTOUCH_VERSION;
return _version;
}
/// Get SoundTouch library version Id
uint SoundTouch::getVersionId()
{
return SOUNDTOUCH_VERSION_ID;
}
// Sets the number of channels, 1 = mono, 2 = stereo
void SoundTouch::setChannels(uint numChannels)
{
if (numChannels != 1 && numChannels != 2)
{
throw std::runtime_error("Illegal number of channels");
}
channels = numChannels;
pRateTransposer->setChannels(numChannels);
pTDStretch->setChannels(numChannels);
}
// Sets new rate control value. Normal rate = 1.0, smaller values
// represent slower rate, larger faster rates.
void SoundTouch::setRate(float newRate)
{
virtualRate = newRate;
calcEffectiveRateAndTempo();
}
// Sets new rate control value as a difference in percents compared
// to the original rate (-50 .. +100 %)
void SoundTouch::setRateChange(float newRate)
{
virtualRate = 1.0f + 0.01f * newRate;
calcEffectiveRateAndTempo();
}
// Sets new tempo control value. Normal tempo = 1.0, smaller values
// represent slower tempo, larger faster tempo.
void SoundTouch::setTempo(float newTempo)
{
virtualTempo = newTempo;
calcEffectiveRateAndTempo();
}
// Sets new tempo control value as a difference in percents compared
// to the original tempo (-50 .. +100 %)
void SoundTouch::setTempoChange(float newTempo)
{
virtualTempo = 1.0f + 0.01f * newTempo;
calcEffectiveRateAndTempo();
}
// Sets new pitch control value. Original pitch = 1.0, smaller values
// represent lower pitches, larger values higher pitch.
void SoundTouch::setPitch(float newPitch)
{
virtualPitch = newPitch;
calcEffectiveRateAndTempo();
}
// Sets pitch change in octaves compared to the original pitch
// (-1.00 .. +1.00)
void SoundTouch::setPitchOctaves(float newPitch)
{
virtualPitch = (float)exp(0.69314718056f * newPitch);
calcEffectiveRateAndTempo();
}
// Sets pitch change in semi-tones compared to the original pitch
// (-12 .. +12)
void SoundTouch::setPitchSemiTones(int newPitch)
{
setPitchOctaves((float)newPitch / 12.0f);
}
void SoundTouch::setPitchSemiTones(float newPitch)
{
setPitchOctaves(newPitch / 12.0f);
}
// Calculates 'effective' rate and tempo values from the
// nominal control values.
void SoundTouch::calcEffectiveRateAndTempo()
{
float oldTempo = tempo;
float oldRate = rate;
tempo = virtualTempo / virtualPitch;
rate = virtualPitch * virtualRate;
if (!TEST_FLOAT_EQUAL(rate,oldRate)) pRateTransposer->setRate(rate);
if (!TEST_FLOAT_EQUAL(tempo, oldTempo)) pTDStretch->setTempo(tempo);
if (rate > 1.0f)
{
if (output != pRateTransposer)
{
FIFOSamplePipe *transOut;
assert(output == pTDStretch);
// move samples in the current output buffer to the output of pRateTransposer
transOut = pRateTransposer->getOutput();
transOut->moveSamples(*output);
// move samples in tempo changer's input to pitch transposer's input
pRateTransposer->moveSamples(*pTDStretch->getInput());
output = pRateTransposer;
}
}
else
{
if (output != pTDStretch)
{
FIFOSamplePipe *tempoOut;
assert(output == pRateTransposer);
// move samples in the current output buffer to the output of pTDStretch
tempoOut = pTDStretch->getOutput();
tempoOut->moveSamples(*output);
// move samples in pitch transposer's store buffer to tempo changer's input
pTDStretch->moveSamples(*pRateTransposer->getStore());
output = pTDStretch;
}
}
}
// Sets sample rate.
void SoundTouch::setSampleRate(uint srate)
{
bSrateSet = TRUE;
// set sample rate, leave other tempo changer parameters as they are.
pTDStretch->setParameters(srate);
}
// Adds 'numSamples' pcs of samples from the 'samples' memory position into
// the input of the object.
void SoundTouch::putSamples(const SAMPLETYPE *samples, uint nSamples)
{
if (bSrateSet == FALSE)
{
throw std::runtime_error("SoundTouch : Sample rate not defined");
}
else if (channels == 0)
{
throw std::runtime_error("SoundTouch : Number of channels not defined");
}
// Transpose the rate of the new samples if necessary
/* Bypass the nominal setting - can introduce a click in sound when tempo/pitch control crosses the nominal value...
if (rate == 1.0f)
{
// The rate value is same as the original, simply evaluate the tempo changer.
assert(output == pTDStretch);
if (pRateTransposer->isEmpty() == 0)
{
// yet flush the last samples in the pitch transposer buffer
// (may happen if 'rate' changes from a non-zero value to zero)
pTDStretch->moveSamples(*pRateTransposer);
}
pTDStretch->putSamples(samples, nSamples);
}
*/
else if (rate <= 1.0f)
{
// transpose the rate down, output the transposed sound to tempo changer buffer
assert(output == pTDStretch);
pRateTransposer->putSamples(samples, nSamples);
pTDStretch->moveSamples(*pRateTransposer);
}
else
{
assert(rate > 1.0f);
// evaluate the tempo changer, then transpose the rate up,
assert(output == pRateTransposer);
pTDStretch->putSamples(samples, nSamples);
pRateTransposer->moveSamples(*pTDStretch);
}
}
// Flushes the last samples from the processing pipeline to the output.
// Clears also the internal processing buffers.
//
// Note: This function is meant for extracting the last samples of a sound
// stream. This function may introduce additional blank samples in the end
// of the sound stream, and thus it's not recommended to call this function
// in the middle of a sound stream.
void SoundTouch::flush()
{
int i;
uint nOut;
SAMPLETYPE buff[128];
nOut = numSamples();
memset(buff, 0, 128 * sizeof(SAMPLETYPE));
// "Push" the last active samples out from the processing pipeline by
// feeding blank samples into the processing pipeline until new,
// processed samples appear in the output (not however, more than
// 8ksamples in any case)
for (i = 0; i < 128; i ++)
{
putSamples(buff, 64);
if (numSamples() != nOut) break; // new samples have appeared in the output!
}
// Clear working buffers
pRateTransposer->clear();
pTDStretch->clearInput();
// yet leave the 'tempoChanger' output intouched as that's where the
// flushed samples are!
}
// Changes a setting controlling the processing system behaviour. See the
// 'SETTING_...' defines for available setting ID's.
BOOL SoundTouch::setSetting(int settingId, int value)
{
int sampleRate, sequenceMs, seekWindowMs, overlapMs;
// read current tdstretch routine parameters
pTDStretch->getParameters(&sampleRate, &sequenceMs, &seekWindowMs, &overlapMs);
switch (settingId)
{
case SETTING_USE_AA_FILTER :
// enables / disabless anti-alias filter
pRateTransposer->enableAAFilter((value != 0) ? TRUE : FALSE);
return TRUE;
case SETTING_AA_FILTER_LENGTH :
// sets anti-alias filter length
pRateTransposer->getAAFilter()->setLength(value);
return TRUE;
case SETTING_USE_QUICKSEEK :
// enables / disables tempo routine quick seeking algorithm
pTDStretch->enableQuickSeek((value != 0) ? TRUE : FALSE);
return TRUE;
case SETTING_SEQUENCE_MS:
// change time-stretch sequence duration parameter
pTDStretch->setParameters(sampleRate, value, seekWindowMs, overlapMs);
return TRUE;
case SETTING_SEEKWINDOW_MS:
// change time-stretch seek window length parameter
pTDStretch->setParameters(sampleRate, sequenceMs, value, overlapMs);
return TRUE;
case SETTING_OVERLAP_MS:
// change time-stretch overlap length parameter
pTDStretch->setParameters(sampleRate, sequenceMs, seekWindowMs, value);
return TRUE;
default :
return FALSE;
}
}
// Reads a setting controlling the processing system behaviour. See the
// 'SETTING_...' defines for available setting ID's.
//
// Returns the setting value.
int SoundTouch::getSetting(int settingId) const
{
int temp;
switch (settingId)
{
case SETTING_USE_AA_FILTER :
return (uint)pRateTransposer->isAAFilterEnabled();
case SETTING_AA_FILTER_LENGTH :
return pRateTransposer->getAAFilter()->getLength();
case SETTING_USE_QUICKSEEK :
return (uint) pTDStretch->isQuickSeekEnabled();
case SETTING_SEQUENCE_MS:
pTDStretch->getParameters(NULL, &temp, NULL, NULL);
return temp;
case SETTING_SEEKWINDOW_MS:
pTDStretch->getParameters(NULL, NULL, &temp, NULL);
return temp;
case SETTING_OVERLAP_MS:
pTDStretch->getParameters(NULL, NULL, NULL, &temp);
return temp;
default :
return 0;
}
}
// Clears all the samples in the object's output and internal processing
// buffers.
void SoundTouch::clear()
{
pRateTransposer->clear();
pTDStretch->clear();
}
/// Returns number of samples currently unprocessed.
uint SoundTouch::numUnprocessedSamples() const
{
FIFOSamplePipe * psp;
if (pTDStretch)
{
psp = pTDStretch->getInput();
if (psp)
{
return psp->numSamples();
}
}
return 0;
}

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939
source/SoundTouch/TDStretch.cpp Normal file
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////////////////////////////////////////////////////////////////////////////////
///
/// Sampled sound tempo changer/time stretch algorithm. Changes the sound tempo
/// while maintaining the original pitch by using a time domain WSOLA-like
/// method with several performance-increasing tweaks.
///
/// Note : MMX optimized functions reside in a separate, platform-specific
/// file, e.g. 'mmx_win.cpp' or 'mmx_gcc.cpp'
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2008-02-10 18:07:12 +0200 (su, 10 helmi 2008) $
// File revision : $Revision: 4 $
//
// $Id: TDStretch.cpp 4 2008-02-10 16:07:12Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#include <string.h>
#include <limits.h>
#include <assert.h>
#include <stdexcept>
#include "STTypes.h"
#include "cpu_detect.h"
#include "TDStretch.h"
using namespace soundtouch;
#ifndef min
//#define min(a,b) (((a) > (b)) ? (b) : (a))
#define max(a,b) (((a) < (b)) ? (b) : (a))
#endif
/*****************************************************************************
*
* Constant definitions
*
*****************************************************************************/
// Table for the hierarchical mixing position seeking algorithm
static int _scanOffsets[4][24]={
{ 124, 186, 248, 310, 372, 434, 496, 558, 620, 682, 744, 806,
868, 930, 992, 1054, 1116, 1178, 1240, 1302, 1364, 1426, 1488, 0},
{-100, -75, -50, -25, 25, 50, 75, 100, 0, 0, 0, 0,
0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0},
{ -20, -15, -10, -5, 5, 10, 15, 20, 0, 0, 0, 0,
0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0},
{ -4, -3, -2, -1, 1, 2, 3, 4, 0, 0, 0, 0,
0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0}};
/*****************************************************************************
*
* Implementation of the class 'TDStretch'
*
*****************************************************************************/
TDStretch::TDStretch() : FIFOProcessor(&outputBuffer)
{
bQuickseek = FALSE;
channels = 2;
bMidBufferDirty = FALSE;
pMidBuffer = NULL;
pRefMidBufferUnaligned = NULL;
overlapLength = 0;
setParameters(44100, DEFAULT_SEQUENCE_MS, DEFAULT_SEEKWINDOW_MS, DEFAULT_OVERLAP_MS);
setTempo(1.0f);
}
TDStretch::~TDStretch()
{
delete[] pMidBuffer;
delete[] pRefMidBufferUnaligned;
}
// Sets routine control parameters. These control are certain time constants
// defining how the sound is stretched to the desired duration.
//
// 'sampleRate' = sample rate of the sound
// 'sequenceMS' = one processing sequence length in milliseconds (default = 82 ms)
// 'seekwindowMS' = seeking window length for scanning the best overlapping
// position (default = 28 ms)
// 'overlapMS' = overlapping length (default = 12 ms)
void TDStretch::setParameters(int aSampleRate, int aSequenceMS,
int aSeekWindowMS, int aOverlapMS)
{
assert(aSampleRate >= 0);
assert(aSequenceMS >= 0);
assert(aSeekWindowMS >= 0);
assert(aOverlapMS >= 0);
this->sampleRate = aSampleRate;
this->sequenceMs = aSequenceMS;
this->seekWindowMs = aSeekWindowMS;
this->overlapMs = aOverlapMS;
seekLength = (sampleRate * seekWindowMs) / 1000;
seekWindowLength = (sampleRate * sequenceMs) / 1000;
maxOffset = seekLength;
calculateOverlapLength(overlapMs);
// set tempo to recalculate 'sampleReq'
setTempo(tempo);
}
/// Get routine control parameters, see setParameters() function.
/// Any of the parameters to this function can be NULL, in such case corresponding parameter
/// value isn't returned.
void TDStretch::getParameters(int *pSampleRate, int *pSequenceMs, int *pSeekWindowMs, int *pOverlapMs) const
{
if (pSampleRate)
{
*pSampleRate = sampleRate;
}
if (pSequenceMs)
{
*pSequenceMs = sequenceMs;
}
if (pSeekWindowMs)
{
*pSeekWindowMs = seekWindowMs;
}
if (pOverlapMs)
{
*pOverlapMs = overlapMs;
}
}
// Overlaps samples in 'midBuffer' with the samples in 'pInput'
void TDStretch::overlapMono(SAMPLETYPE *pOutput, const SAMPLETYPE *pInput) const
{
int i, itemp;
for (i = 0; i < overlapLength ; i ++)
{
itemp = overlapLength - i;
pOutput[i] = (pInput[i] * i + pMidBuffer[i] * itemp ) / overlapLength; // >> overlapDividerBits;
}
}
void TDStretch::clearMidBuffer()
{
if (bMidBufferDirty)
{
memset(pMidBuffer, 0, 2 * sizeof(SAMPLETYPE) * overlapLength);
bMidBufferDirty = FALSE;
}
}
void TDStretch::clearInput()
{
inputBuffer.clear();
clearMidBuffer();
}
// Clears the sample buffers
void TDStretch::clear()
{
outputBuffer.clear();
inputBuffer.clear();
clearMidBuffer();
}
// Enables/disables the quick position seeking algorithm. Zero to disable, nonzero
// to enable
void TDStretch::enableQuickSeek(BOOL enable)
{
bQuickseek = enable;
}
// Returns nonzero if the quick seeking algorithm is enabled.
BOOL TDStretch::isQuickSeekEnabled() const
{
return bQuickseek;
}
// Seeks for the optimal overlap-mixing position.
int TDStretch::seekBestOverlapPosition(const SAMPLETYPE *refPos)
{
if (channels == 2)
{
// stereo sound
if (bQuickseek)
{
return seekBestOverlapPositionStereoQuick(refPos);
}
else
{
return seekBestOverlapPositionStereo(refPos);
}
}
else
{
// mono sound
if (bQuickseek)
{
return seekBestOverlapPositionMonoQuick(refPos);
}
else
{
return seekBestOverlapPositionMono(refPos);
}
}
}
// Overlaps samples in 'midBuffer' with the samples in 'pInputBuffer' at position
// of 'ovlPos'.
inline void TDStretch::overlap(SAMPLETYPE *pOutput, const SAMPLETYPE *pInput, uint ovlPos) const
{
if (channels == 2)
{
// stereo sound
overlapStereo(pOutput, pInput + 2 * ovlPos);
} else {
// mono sound.
overlapMono(pOutput, pInput + ovlPos);
}
}
// Seeks for the optimal overlap-mixing position. The 'stereo' version of the
// routine
//
// The best position is determined as the position where the two overlapped
// sample sequences are 'most alike', in terms of the highest cross-correlation
// value over the overlapping period
int TDStretch::seekBestOverlapPositionStereo(const SAMPLETYPE *refPos)
{
int bestOffs;
LONG_SAMPLETYPE bestCorr, corr;
int i;
// Slopes the amplitudes of the 'midBuffer' samples
precalcCorrReferenceStereo();
bestCorr = INT_MIN;
bestOffs = 0;
// Scans for the best correlation value by testing each possible position
// over the permitted range.
for (i = 0; i < seekLength; i ++)
{
// Calculates correlation value for the mixing position corresponding
// to 'i'
corr = calcCrossCorrStereo(refPos + 2 * i, pRefMidBuffer);
// Checks for the highest correlation value
if (corr > bestCorr)
{
bestCorr = corr;
bestOffs = i;
}
}
// clear cross correlation routine state if necessary (is so e.g. in MMX routines).
clearCrossCorrState();
return bestOffs;
}
// Seeks for the optimal overlap-mixing position. The 'stereo' version of the
// routine
//
// The best position is determined as the position where the two overlapped
// sample sequences are 'most alike', in terms of the highest cross-correlation
// value over the overlapping period
int TDStretch::seekBestOverlapPositionStereoQuick(const SAMPLETYPE *refPos)
{
int j;
int bestOffs;
LONG_SAMPLETYPE bestCorr, corr;
int scanCount, corrOffset, tempOffset;
// Slopes the amplitude of the 'midBuffer' samples
precalcCorrReferenceStereo();
bestCorr = INT_MIN;
bestOffs = 0;
corrOffset = 0;
tempOffset = 0;
// Scans for the best correlation value using four-pass hierarchical search.
//
// The look-up table 'scans' has hierarchical position adjusting steps.
// In first pass the routine searhes for the highest correlation with
// relatively coarse steps, then rescans the neighbourhood of the highest
// correlation with better resolution and so on.
for (scanCount = 0;scanCount < 4; scanCount ++)
{
j = 0;
while (_scanOffsets[scanCount][j])
{
tempOffset = corrOffset + _scanOffsets[scanCount][j];
if (tempOffset >= seekLength) break;
// Calculates correlation value for the mixing position corresponding
// to 'tempOffset'
corr = calcCrossCorrStereo(refPos + 2 * tempOffset, pRefMidBuffer);
// Checks for the highest correlation value
if (corr > bestCorr)
{
bestCorr = corr;
bestOffs = tempOffset;
}
j ++;
}
corrOffset = bestOffs;
}
// clear cross correlation routine state if necessary (is so e.g. in MMX routines).
clearCrossCorrState();
return bestOffs;
}
// Seeks for the optimal overlap-mixing position. The 'mono' version of the
// routine
//
// The best position is determined as the position where the two overlapped
// sample sequences are 'most alike', in terms of the highest cross-correlation
// value over the overlapping period
int TDStretch::seekBestOverlapPositionMono(const SAMPLETYPE *refPos)
{
int bestOffs;
LONG_SAMPLETYPE bestCorr, corr;
int tempOffset;
const SAMPLETYPE *compare;
// Slopes the amplitude of the 'midBuffer' samples
precalcCorrReferenceMono();
bestCorr = INT_MIN;
bestOffs = 0;
// Scans for the best correlation value by testing each possible position
// over the permitted range.
for (tempOffset = 0; tempOffset < seekLength; tempOffset ++)
{
compare = refPos + tempOffset;
// Calculates correlation value for the mixing position corresponding
// to 'tempOffset'
corr = calcCrossCorrMono(pRefMidBuffer, compare);
// Checks for the highest correlation value
if (corr > bestCorr)
{
bestCorr = corr;
bestOffs = tempOffset;
}
}
// clear cross correlation routine state if necessary (is so e.g. in MMX routines).
clearCrossCorrState();
return bestOffs;
}
// Seeks for the optimal overlap-mixing position. The 'mono' version of the
// routine
//
// The best position is determined as the position where the two overlapped
// sample sequences are 'most alike', in terms of the highest cross-correlation
// value over the overlapping period
int TDStretch::seekBestOverlapPositionMonoQuick(const SAMPLETYPE *refPos)
{
int j;
int bestOffs;
LONG_SAMPLETYPE bestCorr, corr;
int scanCount, corrOffset, tempOffset;
// Slopes the amplitude of the 'midBuffer' samples
precalcCorrReferenceMono();
bestCorr = INT_MIN;
bestOffs = 0;
corrOffset = 0;
tempOffset = 0;
// Scans for the best correlation value using four-pass hierarchical search.
//
// The look-up table 'scans' has hierarchical position adjusting steps.
// In first pass the routine searhes for the highest correlation with
// relatively coarse steps, then rescans the neighbourhood of the highest
// correlation with better resolution and so on.
for (scanCount = 0;scanCount < 4; scanCount ++)
{
j = 0;
while (_scanOffsets[scanCount][j])
{
tempOffset = corrOffset + _scanOffsets[scanCount][j];
if (tempOffset >= seekLength) break;
// Calculates correlation value for the mixing position corresponding
// to 'tempOffset'
corr = calcCrossCorrMono(refPos + tempOffset, pRefMidBuffer);
// Checks for the highest correlation value
if (corr > bestCorr)
{
bestCorr = corr;
bestOffs = tempOffset;
}
j ++;
}
corrOffset = bestOffs;
}
// clear cross correlation routine state if necessary (is so e.g. in MMX routines).
clearCrossCorrState();
return bestOffs;
}
/// clear cross correlation routine state if necessary
void TDStretch::clearCrossCorrState()
{
// default implementation is empty.
}
// Sets new target tempo. Normal tempo = 'SCALE', smaller values represent slower
// tempo, larger faster tempo.
void TDStretch::setTempo(float newTempo)
{
int intskip;
tempo = newTempo;
// Calculate ideal skip length (according to tempo value)
nominalSkip = tempo * (seekWindowLength - overlapLength);
skipFract = 0;
intskip = (int)(nominalSkip + 0.5f);
// Calculate how many samples are needed in the 'inputBuffer' to
// process another batch of samples
sampleReq = max(intskip + overlapLength, seekWindowLength) + maxOffset;
}
// Sets the number of channels, 1 = mono, 2 = stereo
void TDStretch::setChannels(int numChannels)
{
assert(numChannels > 0);
if (channels == numChannels) return;
assert(numChannels == 1 || numChannels == 2);
channels = numChannels;
inputBuffer.setChannels(channels);
outputBuffer.setChannels(channels);
}
// nominal tempo, no need for processing, just pass the samples through
// to outputBuffer
/*
void TDStretch::processNominalTempo()
{
assert(tempo == 1.0f);
if (bMidBufferDirty)
{
// If there are samples in pMidBuffer waiting for overlapping,
// do a single sliding overlapping with them in order to prevent a
// clicking distortion in the output sound
if (inputBuffer.numSamples() < overlapLength)
{
// wait until we've got overlapLength input samples
return;
}
// Mix the samples in the beginning of 'inputBuffer' with the
// samples in 'midBuffer' using sliding overlapping
overlap(outputBuffer.ptrEnd(overlapLength), inputBuffer.ptrBegin(), 0);
outputBuffer.putSamples(overlapLength);
inputBuffer.receiveSamples(overlapLength);
clearMidBuffer();
// now we've caught the nominal sample flow and may switch to
// bypass mode
}
// Simply bypass samples from input to output
outputBuffer.moveSamples(inputBuffer);
}
*/
// Processes as many processing frames of the samples 'inputBuffer', store
// the result into 'outputBuffer'
void TDStretch::processSamples()
{
int ovlSkip, offset;
int temp;
/* Removed this small optimization - can introduce a click to sound when tempo setting
crosses the nominal value
if (tempo == 1.0f)
{
// tempo not changed from the original, so bypass the processing
processNominalTempo();
return;
}
*/
if (bMidBufferDirty == FALSE)
{
// if midBuffer is empty, move the first samples of the input stream
// into it
if ((int)inputBuffer.numSamples() < overlapLength)
{
// wait until we've got overlapLength samples
return;
}
memcpy(pMidBuffer, inputBuffer.ptrBegin(), channels * overlapLength * sizeof(SAMPLETYPE));
inputBuffer.receiveSamples((uint)overlapLength);
bMidBufferDirty = TRUE;
}
// Process samples as long as there are enough samples in 'inputBuffer'
// 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());
// Mix the samples in the 'inputBuffer' at position of 'offset' with the
// samples in 'midBuffer' using sliding overlapping
// ... first partially overlap with the end of the previous sequence
// (that's in 'midBuffer')
overlap(outputBuffer.ptrEnd((uint)overlapLength), inputBuffer.ptrBegin(), (uint)offset);
outputBuffer.putSamples((uint)overlapLength);
// ... then copy sequence samples from 'inputBuffer' to output
temp = (seekWindowLength - 2 * overlapLength);// & 0xfffffffe;
if (temp > 0)
{
outputBuffer.putSamples(inputBuffer.ptrBegin() + channels * (offset + overlapLength), (uint)temp);
}
// Copies the end of the current sequence from 'inputBuffer' to
// 'midBuffer' for being mixed with the beginning of the next
// processing sequence and so on
assert(offset + seekWindowLength <= (int)inputBuffer.numSamples());
memcpy(pMidBuffer, inputBuffer.ptrBegin() + channels * (offset + seekWindowLength - overlapLength),
channels * sizeof(SAMPLETYPE) * overlapLength);
bMidBufferDirty = TRUE;
// Remove the processed samples from the input buffer. Update
// the difference between integer & nominal skip step to 'skipFract'
// in order to prevent the error from accumulating over time.
skipFract += nominalSkip; // real skip size
ovlSkip = (int)skipFract; // rounded to integer skip
skipFract -= ovlSkip; // maintain the fraction part, i.e. real vs. integer skip
inputBuffer.receiveSamples((uint)ovlSkip);
}
}
// Adds 'numsamples' pcs of samples from the 'samples' memory position into
// the input of the object.
void TDStretch::putSamples(const SAMPLETYPE *samples, uint nSamples)
{
// Add the samples into the input buffer
inputBuffer.putSamples(samples, nSamples);
// Process the samples in input buffer
processSamples();
}
/// Set new overlap length parameter & reallocate RefMidBuffer if necessary.
void TDStretch::acceptNewOverlapLength(int newOverlapLength)
{
int prevOvl;
assert(newOverlapLength >= 0);
prevOvl = overlapLength;
overlapLength = newOverlapLength;
if (overlapLength > prevOvl)
{
delete[] pMidBuffer;
delete[] pRefMidBufferUnaligned;
pMidBuffer = new SAMPLETYPE[overlapLength * 2];
bMidBufferDirty = TRUE;
clearMidBuffer();
pRefMidBufferUnaligned = new SAMPLETYPE[2 * overlapLength + 16 / sizeof(SAMPLETYPE)];
// ensure that 'pRefMidBuffer' is aligned to 16 byte boundary for efficiency
pRefMidBuffer = (SAMPLETYPE *)((((ulong)pRefMidBufferUnaligned) + 15) & (ulong)-16);
}
}
// Operator 'new' is overloaded so that it automatically creates a suitable instance
// depending on if we've a MMX/SSE/etc-capable CPU available or not.
void * TDStretch::operator new(size_t s)
{
// Notice! don't use "new TDStretch" directly, use "newInstance" to create a new instance instead!
throw std::runtime_error("Error in TDStretch::new: Don't use 'new TDStretch' directly, use 'newInstance' member instead!");
return NULL;
}
TDStretch * TDStretch::newInstance()
{
uint uExtensions;
uExtensions = detectCPUextensions();
// Check if MMX/SSE/3DNow! instruction set extensions supported by CPU
#ifdef ALLOW_MMX
// MMX routines available only with integer sample types
if (uExtensions & SUPPORT_MMX)
{
return ::new TDStretchMMX;
}
else
#endif // ALLOW_MMX
#ifdef ALLOW_SSE
if (uExtensions & SUPPORT_SSE)
{
// SSE support
return ::new TDStretchSSE;
}
else
#endif // ALLOW_SSE
#ifdef ALLOW_3DNOW
if (uExtensions & SUPPORT_3DNOW)
{
// 3DNow! support
return ::new TDStretch3DNow;
}
else
#endif // ALLOW_3DNOW
{
// ISA optimizations not supported, use plain C version
return ::new TDStretch;
}
}
//////////////////////////////////////////////////////////////////////////////
//
// Integer arithmetics specific algorithm implementations.
//
//////////////////////////////////////////////////////////////////////////////
#ifdef INTEGER_SAMPLES
// Slopes the amplitude of the 'midBuffer' samples so that cross correlation
// is faster to calculate
void TDStretch::precalcCorrReferenceStereo()
{
int i, cnt2;
int temp, temp2;
for (i=0 ; i < (int)overlapLength ;i ++)
{
temp = i * (overlapLength - i);
cnt2 = i * 2;
temp2 = (pMidBuffer[cnt2] * temp) / slopingDivider;
pRefMidBuffer[cnt2] = (short)(temp2);
temp2 = (pMidBuffer[cnt2 + 1] * temp) / slopingDivider;
pRefMidBuffer[cnt2 + 1] = (short)(temp2);
}
}
// Slopes the amplitude of the 'midBuffer' samples so that cross correlation
// is faster to calculate
void TDStretch::precalcCorrReferenceMono()
{
int i;
long temp;
long temp2;
for (i=0 ; i < (int)overlapLength ;i ++)
{
temp = i * (overlapLength - i);
temp2 = (pMidBuffer[i] * temp) / slopingDivider;
pRefMidBuffer[i] = (short)temp2;
}
}
// Overlaps samples in 'midBuffer' with the samples in 'input'. The 'Stereo'
// version of the routine.
void TDStretch::overlapStereo(short *output, const short *input) const
{
int i;
short temp;
uint cnt2;
for (i = 0; i < (int)overlapLength ; i ++)
{
temp = (short)(overlapLength - i);
cnt2 = 2 * i;
output[cnt2] = (input[cnt2] * i + pMidBuffer[cnt2] * temp ) / overlapLength;
output[cnt2 + 1] = (input[cnt2 + 1] * i + pMidBuffer[cnt2 + 1] * temp ) / overlapLength;
}
}
/// Calculates overlap period length in samples.
/// Integer version rounds overlap length to closest power of 2
/// for a divide scaling operation.
void TDStretch::calculateOverlapLength(int overlapMs)
{
int newOvl;
assert(overlapMs >= 0);
overlapDividerBits = _getClosest2Power((sampleRate * overlapMs) / 1000.0);
if (overlapDividerBits > 9) overlapDividerBits = 9;
if (overlapDividerBits < 4) overlapDividerBits = 4;
newOvl = (int)pow(2, overlapDividerBits);
acceptNewOverlapLength(newOvl);
// calculate sloping divider so that crosscorrelation operation won't
// overflow 32-bit register. Max. sum of the crosscorrelation sum without
// divider would be 2^30*(N^3-N)/3, where N = overlap length
slopingDivider = (newOvl * newOvl - 1) / 3;
}
long TDStretch::calcCrossCorrMono(const short *mixingPos, const short *compare) const
{
long corr;
uint i;
corr = 0;
for (i = 1; i < overlapLength; i ++)
{
corr += (mixingPos[i] * compare[i]) >> overlapDividerBits;
}
return corr;
}
long TDStretch::calcCrossCorrStereo(const short *mixingPos, const short *compare) const
{
long corr;
uint i;
corr = 0;
for (i = 2; i < 2 * overlapLength; i += 2)
{
corr += (mixingPos[i] * compare[i] +
mixingPos[i + 1] * compare[i + 1]) >> overlapDividerBits;
}
return corr;
}
#endif // INTEGER_SAMPLES
//////////////////////////////////////////////////////////////////////////////
//
// Floating point arithmetics specific algorithm implementations.
//
#ifdef FLOAT_SAMPLES
// Slopes the amplitude of the 'midBuffer' samples so that cross correlation
// is faster to calculate
void TDStretch::precalcCorrReferenceStereo()
{
int i, cnt2;
float temp;
for (i=0 ; i < (int)overlapLength ;i ++)
{
temp = (float)i * (float)(overlapLength - i);
cnt2 = i * 2;
pRefMidBuffer[cnt2] = (float)(pMidBuffer[cnt2] * temp);
pRefMidBuffer[cnt2 + 1] = (float)(pMidBuffer[cnt2 + 1] * temp);
}
}
// Slopes the amplitude of the 'midBuffer' samples so that cross correlation
// is faster to calculate
void TDStretch::precalcCorrReferenceMono()
{
int i;
float temp;
for (i=0 ; i < (int)overlapLength ;i ++)
{
temp = (float)i * (float)(overlapLength - i);
pRefMidBuffer[i] = (float)(pMidBuffer[i] * temp);
}
}
// SSE-optimized version of the function overlapStereo
void TDStretch::overlapStereo(float *pOutput, const float *pInput) const
{
int i;
int cnt2;
float fTemp;
float fScale;
float fi;
fScale = 1.0f / (float)overlapLength;
for (i = 0; i < (int)overlapLength ; i ++)
{
fTemp = (float)(overlapLength - i) * fScale;
fi = (float)i * fScale;
cnt2 = 2 * i;
pOutput[cnt2 + 0] = pInput[cnt2 + 0] * fi + pMidBuffer[cnt2 + 0] * fTemp;
pOutput[cnt2 + 1] = pInput[cnt2 + 1] * fi + pMidBuffer[cnt2 + 1] * fTemp;
}
}
/// Calculates overlapInMsec period length in samples.
void TDStretch::calculateOverlapLength(int overlapInMsec)
{
int newOvl;
assert(overlapInMsec >= 0);
newOvl = (sampleRate * overlapInMsec) / 1000;
if (newOvl < 16) newOvl = 16;
// must be divisible by 8
newOvl -= newOvl % 8;
acceptNewOverlapLength(newOvl);
}
double TDStretch::calcCrossCorrMono(const float *mixingPos, const float *compare) const
{
double corr;
int i;
corr = 0;
for (i = 1; i < overlapLength; i ++)
{
corr += mixingPos[i] * compare[i];
}
return corr;
}
double TDStretch::calcCrossCorrStereo(const float *mixingPos, const float *compare) const
{
double corr;
int i;
corr = 0;
for (i = 2; i < 2 * overlapLength; i += 2)
{
corr += mixingPos[i] * compare[i] +
mixingPos[i + 1] * compare[i + 1];
}
return corr;
}
#endif // FLOAT_SAMPLES

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@ -0,0 +1,252 @@
////////////////////////////////////////////////////////////////////////////////
///
/// Sampled sound tempo changer/time stretch algorithm. Changes the sound tempo
/// while maintaining the original pitch by using a time domain WSOLA-like method
/// with several performance-increasing tweaks.
///
/// Note : MMX optimized functions reside in a separate, platform-specific file,
/// e.g. 'mmx_win.cpp' or 'mmx_gcc.cpp'
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2008-02-10 18:07:12 +0200 (su, 10 helmi 2008) $
// File revision : $Revision: 4 $
//
// $Id: TDStretch.h 4 2008-02-10 16:07:12Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#ifndef TDStretch_H
#define TDStretch_H
#include "STTypes.h"
#include "RateTransposer.h"
#include "FIFOSamplePipe.h"
namespace soundtouch
{
// Default values for sound processing parameters:
/// Default length of a single processing sequence, in milliseconds. This determines to how
/// long sequences the original sound is chopped in the time-stretch algorithm.
///
/// The larger this value is, the lesser sequences are used in processing. In principle
/// a bigger value sounds better when slowing down tempo, but worse when increasing tempo
/// and vice versa.
///
/// Increasing this value reduces computational burden & vice versa.
#define DEFAULT_SEQUENCE_MS 82
/// Seeking window default length in milliseconds for algorithm that finds the best possible
/// overlapping location. This determines from how wide window the algorithm may look for an
/// optimal joining location when mixing the sound sequences back together.
///
/// The bigger this window setting is, the higher the possibility to find a better mixing
/// position will become, but at the same time large values may cause a "drifting" artifact
/// because consequent sequences will be taken at more uneven intervals.
///
/// If there's a disturbing artifact that sounds as if a constant frequency was drifting
/// around, try reducing this setting.
///
/// Increasing this value increases computational burden & vice versa.
#define DEFAULT_SEEKWINDOW_MS 14
/// Overlap length in milliseconds. When the chopped sound sequences are mixed back together,
/// to form a continuous sound stream, this parameter defines over how long period the two
/// consecutive sequences are let to overlap each other.
///
/// This shouldn't be that critical parameter. If you reduce the DEFAULT_SEQUENCE_MS setting
/// by a large amount, you might wish to try a smaller value on this.
///
/// Increasing this value increases computational burden & vice versa.
#define DEFAULT_OVERLAP_MS 12
/// Class that does the time-stretch (tempo change) effect for the processed
/// sound.
class TDStretch : public FIFOProcessor
{
protected:
int channels;
int sampleReq;
float tempo;
SAMPLETYPE *pMidBuffer;
SAMPLETYPE *pRefMidBuffer;
SAMPLETYPE *pRefMidBufferUnaligned;
int overlapLength;
int seekLength;
int seekWindowLength;
int maxOffset;
float nominalSkip;
float skipFract;
FIFOSampleBuffer outputBuffer;
FIFOSampleBuffer inputBuffer;
BOOL bQuickseek;
BOOL bMidBufferDirty;
int sampleRate;
int sequenceMs;
int seekWindowMs;
int overlapMs;
void acceptNewOverlapLength(int newOverlapLength);
virtual void clearCrossCorrState();
void calculateOverlapLength(int overlapMs);
virtual LONG_SAMPLETYPE calcCrossCorrStereo(const SAMPLETYPE *mixingPos, const SAMPLETYPE *compare) const;
virtual LONG_SAMPLETYPE calcCrossCorrMono(const SAMPLETYPE *mixingPos, const SAMPLETYPE *compare) const;
virtual int seekBestOverlapPositionStereo(const SAMPLETYPE *refPos);
virtual int seekBestOverlapPositionStereoQuick(const SAMPLETYPE *refPos);
virtual int seekBestOverlapPositionMono(const SAMPLETYPE *refPos);
virtual int seekBestOverlapPositionMonoQuick(const SAMPLETYPE *refPos);
int seekBestOverlapPosition(const SAMPLETYPE *refPos);
virtual void overlapStereo(SAMPLETYPE *output, const SAMPLETYPE *input) const;
virtual void overlapMono(SAMPLETYPE *output, const SAMPLETYPE *input) const;
void clearMidBuffer();
void overlap(SAMPLETYPE *output, const SAMPLETYPE *input, uint ovlPos) const;
void precalcCorrReferenceMono();
void precalcCorrReferenceStereo();
/// Changes the tempo of the given sound samples.
/// Returns amount of samples returned in the "output" buffer.
/// The maximum amount of samples that can be returned at a time is set by
/// the 'set_returnBuffer_size' function.
void processSamples();
public:
TDStretch();
virtual ~TDStretch();
/// Operator 'new' is overloaded so that it automatically creates a suitable instance
/// depending on if we've a MMX/SSE/etc-capable CPU available or not.
static void *operator new(size_t s);
/// Use this function instead of "new" operator to create a new instance of this class.
/// This function automatically chooses a correct feature set depending on if the CPU
/// supports MMX/SSE/etc extensions.
static TDStretch *newInstance();
/// Returns the output buffer object
FIFOSamplePipe *getOutput() { return &outputBuffer; };
/// Returns the input buffer object
FIFOSamplePipe *getInput() { return &inputBuffer; };
/// Sets new target tempo. Normal tempo = 'SCALE', smaller values represent slower
/// tempo, larger faster tempo.
void setTempo(float newTempo);
/// Returns nonzero if there aren't any samples available for outputting.
virtual void clear();
/// Clears the input buffer
void clearInput();
/// Sets the number of channels, 1 = mono, 2 = stereo
void setChannels(int numChannels);
/// Enables/disables the quick position seeking algorithm. Zero to disable,
/// nonzero to enable
void enableQuickSeek(BOOL enable);
/// Returns nonzero if the quick seeking algorithm is enabled.
BOOL isQuickSeekEnabled() const;
/// Sets routine control parameters. These control are certain time constants
/// defining how the sound is stretched to the desired duration.
//
/// 'sampleRate' = sample rate of the sound
/// 'sequenceMS' = one processing sequence length in milliseconds
/// 'seekwindowMS' = seeking window length for scanning the best overlapping
/// position
/// 'overlapMS' = overlapping length
void setParameters(int sampleRate, ///< Samplerate of sound being processed (Hz)
int sequenceMS = DEFAULT_SEQUENCE_MS, ///< Single processing sequence length (ms)
int seekwindowMS = DEFAULT_SEEKWINDOW_MS, ///< Offset seeking window length (ms)
int overlapMS = DEFAULT_OVERLAP_MS ///< Sequence overlapping length (ms)
);
/// Get routine control parameters, see setParameters() function.
/// Any of the parameters to this function can be NULL, in such case corresponding parameter
/// value isn't returned.
void getParameters(int *pSampleRate, int *pSequenceMs, int *pSeekWindowMs, int *pOverlapMs) const;
/// Adds 'numsamples' pcs of samples from the 'samples' memory position into
/// the input of the object.
virtual void putSamples(
const SAMPLETYPE *samples, ///< Input sample data
uint numSamples ///< Number of samples in 'samples' so that one sample
///< contains both channels if stereo
);
};
// Implementation-specific class declarations:
#ifdef ALLOW_MMX
/// Class that implements MMX optimized routines for 16bit integer samples type.
class TDStretchMMX : public TDStretch
{
protected:
long calcCrossCorrStereo(const short *mixingPos, const short *compare) const;
virtual void overlapStereo(short *output, const short *input) const;
virtual void clearCrossCorrState();
};
#endif /// ALLOW_MMX
#ifdef ALLOW_3DNOW
/// Class that implements 3DNow! optimized routines for floating point samples type.
class TDStretch3DNow : public TDStretch
{
protected:
double calcCrossCorrStereo(const float *mixingPos, const float *compare) const;
};
#endif /// ALLOW_3DNOW
#ifdef ALLOW_SSE
/// Class that implements SSE optimized routines for floating point samples type.
class TDStretchSSE : public TDStretch
{
protected:
double calcCrossCorrStereo(const float *mixingPos, const float *compare) const;
};
#endif /// ALLOW_SSE
}
#endif /// TDStretch_H

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////////////////////////////////////////////////////////////////////////////////
///
/// A header file for detecting the Intel MMX instructions set extension.
///
/// Please see 'mmx_win.cpp', 'mmx_cpp.cpp' and 'mmx_non_x86.cpp' for the
/// routine implementations for x86 Windows, x86 gnu version and non-x86
/// platforms, respectively.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2008-02-10 18:07:12 +0200 (su, 10 helmi 2008) $
// File revision : $Revision: 4 $
//
// $Id: cpu_detect.h 4 2008-02-10 16:07:12Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#ifndef _CPU_DETECT_H_
#define _CPU_DETECT_H_
#include "STTypes.h"
#define SUPPORT_MMX 0x0001
#define SUPPORT_3DNOW 0x0002
#define SUPPORT_ALTIVEC 0x0004
#define SUPPORT_SSE 0x0008
#define SUPPORT_SSE2 0x0010
/// Checks which instruction set extensions are supported by the CPU.
///
/// \return A bitmask of supported extensions, see SUPPORT_... defines.
uint detectCPUextensions(void);
/// Disables given set of instruction extensions. See SUPPORT_... defines.
void disableExtensions(uint wDisableMask);
#endif // _CPU_DETECT_H_

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////////////////////////////////////////////////////////////////////////////////
///
/// gcc version of the x86 CPU detect routine.
///
/// This file is to be compiled on any platform with the GNU C compiler.
/// Compiler. Please see 'cpu_detect_x86_win.cpp' for the x86 Windows version
/// of this file.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2008-02-10 18:07:12 +0200 (su, 10 helmi 2008) $
// File revision : $Revision: 4 $
//
// $Id: cpu_detect_x86_gcc.cpp 4 2008-02-10 16:07:12Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#include <stdexcept>
#include <string>
#include "cpu_detect.h"
#ifndef __GNUC__
#error wrong platform - this source code file is for the GNU C compiler.
#endif
using namespace std;
#include <stdio.h>
//////////////////////////////////////////////////////////////////////////////
//
// processor instructions extension detection routines
//
//////////////////////////////////////////////////////////////////////////////
// Flag variable indicating whick ISA extensions are disabled (for debugging)
static uint _dwDisabledISA = 0x00; // 0xffffffff; //<- use this to disable all extensions
// Disables given set of instruction extensions. See SUPPORT_... defines.
void disableExtensions(uint dwDisableMask)
{
_dwDisabledISA = dwDisableMask;
}
/// Checks which instruction set extensions are supported by the CPU.
uint detectCPUextensions(void)
{
#ifndef __i386__
return 0; // always disable extensions on non-x86 platforms.
#else
uint res = 0;
if (_dwDisabledISA == 0xffffffff) return 0;
asm volatile(
"\n\txor %%esi, %%esi" // clear %%esi = result register
// check if 'cpuid' instructions is available by toggling eflags bit 21
"\n\tpushf" // save eflags to stack
"\n\tpop %%eax" // load eax from stack (with eflags)
"\n\tmovl %%eax, %%ecx" // save the original eflags values to ecx
"\n\txor $0x00200000, %%eax" // toggle bit 21
"\n\tpush %%eax" // store toggled eflags to stack
"\n\tpopf" // load eflags from stack
"\n\tpushf" // save updated eflags to stack
"\n\tpop %%eax" // load from stack
"\n\txor %%edx, %%edx" // clear edx for defaulting no mmx
"\n\tcmp %%ecx, %%eax" // compare to original eflags values
"\n\tjz end" // jumps to 'end' if cpuid not present
// cpuid instruction available, test for presence of mmx instructions
"\n\tmovl $1, %%eax"
"\n\tcpuid"
// movl $0x00800000, %edx // force enable MMX
"\n\ttest $0x00800000, %%edx"
"\n\tjz end" // branch if MMX not available
"\n\tor $0x01, %%esi" // otherwise add MMX support bit
"\n\ttest $0x02000000, %%edx"
"\n\tjz test3DNow" // branch if SSE not available
"\n\tor $0x08, %%esi" // otherwise add SSE support bit
"\n\ttest3DNow:"
// test for precense of AMD extensions
"\n\tmov $0x80000000, %%eax"
"\n\tcpuid"
"\n\tcmp $0x80000000, %%eax"
"\n\tjbe end" // branch if no AMD extensions detected
// test for precense of 3DNow! extension
"\n\tmov $0x80000001, %%eax"
"\n\tcpuid"
"\n\ttest $0x80000000, %%edx"
"\n\tjz end" // branch if 3DNow! not detected
"\n\tor $0x02, %%esi" // otherwise add 3DNow support bit
"\n\tend:"
"\n\tmov %%esi, %0"
: "=r" (res)
: /* no inputs */
: "%edx", "%eax", "%ecx", "%esi" );
return res & ~_dwDisabledISA;
#endif
}

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////////////////////////////////////////////////////////////////////////////////
///
/// Win32 version of the x86 CPU detect routine.
///
/// This file is to be compiled in Windows platform with Microsoft Visual C++
/// Compiler. Please see 'cpu_detect_x86_gcc.cpp' for the gcc compiler version
/// for all GNU platforms.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2008-02-10 18:07:12 +0200 (su, 10 helmi 2008) $
// File revision : $Revision: 4 $
//
// $Id: cpu_detect_x86_win.cpp 4 2008-02-10 16:07:12Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#include "cpu_detect.h"
#ifndef WIN32
#error wrong platform - this source code file is exclusively for Win32 platform
#endif
//////////////////////////////////////////////////////////////////////////////
//
// processor instructions extension detection routines
//
//////////////////////////////////////////////////////////////////////////////
// Flag variable indicating whick ISA extensions are disabled (for debugging)
static uint _dwDisabledISA = 0x00; // 0xffffffff; //<- use this to disable all extensions
// Disables given set of instruction extensions. See SUPPORT_... defines.
void disableExtensions(uint dwDisableMask)
{
_dwDisabledISA = dwDisableMask;
}
/// Checks which instruction set extensions are supported by the CPU.
uint detectCPUextensions(void)
{
uint res = 0;
if (_dwDisabledISA == 0xffffffff) return 0;
_asm
{
; check if 'cpuid' instructions is available by toggling eflags bit 21
;
xor esi, esi ; clear esi = result register
pushfd ; save eflags to stack
pop eax ; load eax from stack (with eflags)
mov ecx, eax ; save the original eflags values to ecx
xor eax, 0x00200000 ; toggle bit 21
push eax ; store toggled eflags to stack
popfd ; load eflags from stack
pushfd ; save updated eflags to stack
pop eax ; load from stack
xor edx, edx ; clear edx for defaulting no mmx
cmp eax, ecx ; compare to original eflags values
jz end ; jumps to 'end' if cpuid not present
; cpuid instruction available, test for presence of mmx instructions
mov eax, 1
cpuid
test edx, 0x00800000
jz end ; branch if MMX not available
or esi, SUPPORT_MMX ; otherwise add MMX support bit
test edx, 0x02000000
jz test3DNow ; branch if SSE not available
or esi, SUPPORT_SSE ; otherwise add SSE support bit
test3DNow:
; test for precense of AMD extensions
mov eax, 0x80000000
cpuid
cmp eax, 0x80000000
jbe end ; branch if no AMD extensions detected
; test for precense of 3DNow! extension
mov eax, 0x80000001
cpuid
test edx, 0x80000000
jz end ; branch if 3DNow! not detected
or esi, SUPPORT_3DNOW ; otherwise add 3DNow support bit
end:
mov res, esi
}
return res & ~_dwDisabledISA;
}

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////////////////////////////////////////////////////////////////////////////////
///
/// MMX optimized routines. All MMX optimized functions have been gathered into
/// this single source code file, regardless to their class or original source
/// code file, in order to ease porting the library to other compiler and
/// processor platforms.
///
/// The MMX-optimizations are programmed using MMX compiler intrinsics that
/// are supported both by Microsoft Visual C++ and GCC compilers, so this file
/// should compile with both toolsets.
///
/// NOTICE: If using Visual Studio 6.0, you'll need to install the "Visual C++
/// 6.0 processor pack" update to support compiler intrinsic syntax. The update
/// is available for download at Microsoft Developers Network, see here:
/// http://msdn.microsoft.com/vstudio/downloads/tools/ppack/default.aspx
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2008-02-10 18:07:12 +0200 (su, 10 helmi 2008) $
// File revision : $Revision: 4 $
//
// $Id: mmx_optimized.cpp 4 2008-02-10 16:07:12Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#include "STTypes.h"
#ifdef ALLOW_MMX
// MMX routines available only with integer sample type
#if !(WIN32 || __i386__ || __x86_64__)
#error "wrong platform - this source code file is exclusively for x86 platforms"
#endif
using namespace soundtouch;
//////////////////////////////////////////////////////////////////////////////
//
// implementation of MMX optimized functions of class 'TDStretchMMX'
//
//////////////////////////////////////////////////////////////////////////////
#include "TDStretch.h"
#include <mmintrin.h>
#include <limits.h>
// Calculates cross correlation of two buffers
long TDStretchMMX::calcCrossCorrStereo(const short *pV1, const short *pV2) const
{
const __m64 *pVec1, *pVec2;
__m64 shifter;
__m64 accu;
long corr;
uint i;
pVec1 = (__m64*)pV1;
pVec2 = (__m64*)pV2;
shifter = _m_from_int(overlapDividerBits);
accu = _mm_setzero_si64();
// Process 4 parallel sets of 2 * stereo samples each during each
// round to improve CPU-level parallellization.
for (i = 0; i < overlapLength / 8; i ++)
{
__m64 temp;
// dictionary of instructions:
// _m_pmaddwd : 4*16bit multiply-add, resulting two 32bits = [a0*b0+a1*b1 ; a2*b2+a3*b3]
// _mm_add_pi32 : 2*32bit add
// _m_psrad : 32bit right-shift
temp = _mm_add_pi32(_mm_madd_pi16(pVec1[0], pVec2[0]),
_mm_madd_pi16(pVec1[1], pVec2[1]));
accu = _mm_add_pi32(accu, _mm_sra_pi32(temp, shifter));
temp = _mm_add_pi32(_mm_madd_pi16(pVec1[2], pVec2[2]),
_mm_madd_pi16(pVec1[3], pVec2[3]));
accu = _mm_add_pi32(accu, _mm_sra_pi32(temp, shifter));
pVec1 += 4;
pVec2 += 4;
}
// copy hi-dword of mm0 to lo-dword of mm1, then sum mmo+mm1
// and finally store the result into the variable "corr"
accu = _mm_add_pi32(accu, _mm_srli_si64(accu, 32));
corr = _m_to_int(accu);
// Clear MMS state
_m_empty();
return corr;
// Note: Warning about the missing EMMS instruction is harmless
// as it'll be called elsewhere.
}
void TDStretchMMX::clearCrossCorrState()
{
// Clear MMS state
_m_empty();
//_asm EMMS;
}
// MMX-optimized version of the function overlapStereo
void TDStretchMMX::overlapStereo(short *output, const short *input) const
{
const __m64 *pVinput, *pVMidBuf;
__m64 *pVdest;
__m64 mix1, mix2, adder, shifter;
uint i;
pVinput = (const __m64*)input;
pVMidBuf = (const __m64*)pMidBuffer;
pVdest = (__m64*)output;
// mix1 = mixer values for 1st stereo sample
// mix1 = mixer values for 2nd stereo sample
// adder = adder for updating mixer values after each round
mix1 = _mm_set_pi16(0, overlapLength, 0, overlapLength);
adder = _mm_set_pi16(1, -1, 1, -1);
mix2 = _mm_add_pi16(mix1, adder);
adder = _mm_add_pi16(adder, adder);
shifter = _m_from_int(overlapDividerBits);
for (i = 0; i < overlapLength / 4; i ++)
{
__m64 temp1, temp2;
// load & shuffle data so that input & mixbuffer data samples are paired
temp1 = _mm_unpacklo_pi16(pVMidBuf[0], pVinput[0]); // = i0l m0l i0r m0r
temp2 = _mm_unpackhi_pi16(pVMidBuf[0], pVinput[0]); // = i1l m1l i1r m1r
// temp = (temp .* mix) >> shifter
temp1 = _mm_sra_pi32(_mm_madd_pi16(temp1, mix1), shifter);
temp2 = _mm_sra_pi32(_mm_madd_pi16(temp2, mix2), shifter);
pVdest[0] = _mm_packs_pi32(temp1, temp2); // pack 2*2*32bit => 4*16bit
// update mix += adder
mix1 = _mm_add_pi16(mix1, adder);
mix2 = _mm_add_pi16(mix2, adder);
// --- second round begins here ---
// load & shuffle data so that input & mixbuffer data samples are paired
temp1 = _mm_unpacklo_pi16(pVMidBuf[1], pVinput[1]); // = i2l m2l i2r m2r
temp2 = _mm_unpackhi_pi16(pVMidBuf[1], pVinput[1]); // = i3l m3l i3r m3r
// temp = (temp .* mix) >> shifter
temp1 = _mm_sra_pi32(_mm_madd_pi16(temp1, mix1), shifter);
temp2 = _mm_sra_pi32(_mm_madd_pi16(temp2, mix2), shifter);
pVdest[1] = _mm_packs_pi32(temp1, temp2); // pack 2*2*32bit => 4*16bit
// update mix += adder
mix1 = _mm_add_pi16(mix1, adder);
mix2 = _mm_add_pi16(mix2, adder);
pVinput += 2;
pVMidBuf += 2;
pVdest += 2;
}
_m_empty(); // clear MMS state
}
//////////////////////////////////////////////////////////////////////////////
//
// implementation of MMX optimized functions of class 'FIRFilter'
//
//////////////////////////////////////////////////////////////////////////////
#include "FIRFilter.h"
FIRFilterMMX::FIRFilterMMX() : FIRFilter()
{
filterCoeffsUnalign = NULL;
}
FIRFilterMMX::~FIRFilterMMX()
{
delete[] filterCoeffsUnalign;
}
// (overloaded) Calculates filter coefficients for MMX routine
void FIRFilterMMX::setCoefficients(const short *coeffs, uint newLength, uint uResultDivFactor)
{
uint i;
FIRFilter::setCoefficients(coeffs, newLength, uResultDivFactor);
// Ensure that filter coeffs array is aligned to 16-byte boundary
delete[] filterCoeffsUnalign;
filterCoeffsUnalign = new short[2 * newLength + 8];
filterCoeffsAlign = (short *)(((uint)filterCoeffsUnalign + 15) & -16);
// rearrange the filter coefficients for mmx routines
for (i = 0;i < length; i += 4)
{
filterCoeffsAlign[2 * i + 0] = coeffs[i + 0];
filterCoeffsAlign[2 * i + 1] = coeffs[i + 2];
filterCoeffsAlign[2 * i + 2] = coeffs[i + 0];
filterCoeffsAlign[2 * i + 3] = coeffs[i + 2];
filterCoeffsAlign[2 * i + 4] = coeffs[i + 1];
filterCoeffsAlign[2 * i + 5] = coeffs[i + 3];
filterCoeffsAlign[2 * i + 6] = coeffs[i + 1];
filterCoeffsAlign[2 * i + 7] = coeffs[i + 3];
}
}
// mmx-optimized version of the filter routine for stereo sound
uint FIRFilterMMX::evaluateFilterStereo(short *dest, const short *src, const uint numSamples) const
{
// Create stack copies of the needed member variables for asm routines :
uint i, j;
__m64 *pVdest = (__m64*)dest;
if (length < 2) return 0;
for (i = 0; i < numSamples / 2; i ++)
{
__m64 accu1;
__m64 accu2;
const __m64 *pVsrc = (const __m64*)src;
const __m64 *pVfilter = (const __m64*)filterCoeffsAlign;
accu1 = accu2 = _mm_setzero_si64();
for (j = 0; j < lengthDiv8 * 2; j ++)
{
__m64 temp1, temp2;
temp1 = _mm_unpacklo_pi16(pVsrc[0], pVsrc[1]); // = l2 l0 r2 r0
temp2 = _mm_unpackhi_pi16(pVsrc[0], pVsrc[1]); // = l3 l1 r3 r1
accu1 = _mm_add_pi32(accu1, _mm_madd_pi16(temp1, pVfilter[0])); // += l2*f2+l0*f0 r2*f2+r0*f0
accu1 = _mm_add_pi32(accu1, _mm_madd_pi16(temp2, pVfilter[1])); // += l3*f3+l1*f1 r3*f3+r1*f1
temp1 = _mm_unpacklo_pi16(pVsrc[1], pVsrc[2]); // = l4 l2 r4 r2
accu2 = _mm_add_pi32(accu2, _mm_madd_pi16(temp2, pVfilter[0])); // += l3*f2+l1*f0 r3*f2+r1*f0
accu2 = _mm_add_pi32(accu2, _mm_madd_pi16(temp1, pVfilter[1])); // += l4*f3+l2*f1 r4*f3+r2*f1
// accu1 += l2*f2+l0*f0 r2*f2+r0*f0
// += l3*f3+l1*f1 r3*f3+r1*f1
// accu2 += l3*f2+l1*f0 r3*f2+r1*f0
// l4*f3+l2*f1 r4*f3+r2*f1
pVfilter += 2;
pVsrc += 2;
}
// accu >>= resultDivFactor
accu1 = _mm_srai_pi32(accu1, resultDivFactor);
accu2 = _mm_srai_pi32(accu2, resultDivFactor);
// pack 2*2*32bits => 4*16 bits
pVdest[0] = _mm_packs_pi32(accu1, accu2);
src += 4;
pVdest ++;
}
_m_empty(); // clear emms state
return (numSamples & 0xfffffffe) - length;
}
#endif // ALLOW_MMX

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////////////////////////////////////////////////////////////////////////////////
///
/// SSE optimized routines for Pentium-III, Athlon-XP and later CPUs. All SSE
/// optimized functions have been gathered into this single source
/// code file, regardless to their class or original source code file, in order
/// to ease porting the library to other compiler and processor platforms.
///
/// The SSE-optimizations are programmed using SSE compiler intrinsics that
/// are supported both by Microsoft Visual C++ and GCC compilers, so this file
/// should compile with both toolsets.
///
/// NOTICE: If using Visual Studio 6.0, you'll need to install the "Visual C++
/// 6.0 processor pack" update to support SSE instruction set. The update is
/// available for download at Microsoft Developers Network, see here:
/// http://msdn.microsoft.com/vstudio/downloads/tools/ppack/default.aspx
///
/// If the above URL is expired or removed, go to "http://msdn.microsoft.com" and
/// perform a search with keywords "processor pack".
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2008-02-10 18:07:12 +0200 (su, 10 helmi 2008) $
// File revision : $Revision: 4 $
//
// $Id: sse_optimized.cpp 4 2008-02-10 16:07:12Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#include "cpu_detect.h"
#include "STTypes.h"
using namespace soundtouch;
#ifdef ALLOW_SSE
// SSE routines available only with float sample type
//////////////////////////////////////////////////////////////////////////////
//
// implementation of SSE optimized functions of class 'TDStretchSSE'
//
//////////////////////////////////////////////////////////////////////////////
#include "TDStretch.h"
#include <xmmintrin.h>
// Calculates cross correlation of two buffers
double TDStretchSSE::calcCrossCorrStereo(const float *pV1, const float *pV2) const
{
int i;
__m128 vSum, *pVec2;
// Note. It means a major slow-down if the routine needs to tolerate
// unaligned __m128 memory accesses. It's way faster if we can skip
// unaligned slots and use _mm_load_ps instruction instead of _mm_loadu_ps.
// This can mean up to ~ 10-fold difference (incl. part of which is
// due to skipping every second round for stereo sound though).
//
// Compile-time define ALLOW_NONEXACT_SIMD_OPTIMIZATION is provided
// for choosing if this little cheating is allowed.
#ifdef ALLOW_NONEXACT_SIMD_OPTIMIZATION
// Little cheating allowed, return valid correlation only for
// aligned locations, meaning every second round for stereo sound.
#define _MM_LOAD _mm_load_ps
if (((ulong)pV1) & 15) return -1e50; // skip unaligned locations
#else
// No cheating allowed, use unaligned load & take the resulting
// performance hit.
#define _MM_LOAD _mm_loadu_ps
#endif
// ensure overlapLength is divisible by 8
assert((overlapLength % 8) == 0);
// Calculates the cross-correlation value between 'pV1' and 'pV2' vectors
// Note: pV2 _must_ be aligned to 16-bit boundary, pV1 need not.
pVec2 = (__m128*)pV2;
vSum = _mm_setzero_ps();
// Unroll the loop by factor of 4 * 4 operations
for (i = 0; i < overlapLength / 8; i ++)
{
// vSum += pV1[0..3] * pV2[0..3]
vSum = _mm_add_ps(vSum, _mm_mul_ps(_MM_LOAD(pV1),pVec2[0]));
// vSum += pV1[4..7] * pV2[4..7]
vSum = _mm_add_ps(vSum, _mm_mul_ps(_MM_LOAD(pV1 + 4), pVec2[1]));
// vSum += pV1[8..11] * pV2[8..11]
vSum = _mm_add_ps(vSum, _mm_mul_ps(_MM_LOAD(pV1 + 8), pVec2[2]));
// vSum += pV1[12..15] * pV2[12..15]
vSum = _mm_add_ps(vSum, _mm_mul_ps(_MM_LOAD(pV1 + 12), pVec2[3]));
pV1 += 16;
pVec2 += 4;
}
// return value = vSum[0] + vSum[1] + vSum[2] + vSum[3]
float *pvSum = (float*)&vSum;
return (double)(pvSum[0] + pvSum[1] + pvSum[2] + pvSum[3]);
/* This is approximately corresponding routine in C-language:
double corr;
uint i;
// Calculates the cross-correlation value between 'pV1' and 'pV2' vectors
corr = 0.0;
for (i = 0; i < overlapLength / 8; i ++)
{
corr += pV1[0] * pV2[0] +
pV1[1] * pV2[1] +
pV1[2] * pV2[2] +
pV1[3] * pV2[3] +
pV1[4] * pV2[4] +
pV1[5] * pV2[5] +
pV1[6] * pV2[6] +
pV1[7] * pV2[7] +
pV1[8] * pV2[8] +
pV1[9] * pV2[9] +
pV1[10] * pV2[10] +
pV1[11] * pV2[11] +
pV1[12] * pV2[12] +
pV1[13] * pV2[13] +
pV1[14] * pV2[14] +
pV1[15] * pV2[15];
pV1 += 16;
pV2 += 16;
}
*/
/* This is corresponding routine in assembler. This may be teeny-weeny bit faster
than intrinsic version, but more difficult to maintain & get compiled on multiple
platforms.
uint overlapLengthLocal = overlapLength;
float corr;
_asm
{
// Very important note: data in 'pV2' _must_ be aligned to
// 16-byte boundary!
// give prefetch hints to CPU of what data are to be needed soonish
// give more aggressive hints on pV1 as that changes while pV2 stays
// same between runs
prefetcht0 [pV1]
prefetcht0 [pV2]
prefetcht0 [pV1 + 32]
mov eax, dword ptr pV1
mov ebx, dword ptr pV2
xorps xmm0, xmm0
mov ecx, overlapLengthLocal
shr ecx, 3 // div by eight
loop1:
prefetcht0 [eax + 64] // give a prefetch hint to CPU what data are to be needed soonish
prefetcht0 [ebx + 32] // give a prefetch hint to CPU what data are to be needed soonish
movups xmm1, [eax]
mulps xmm1, [ebx]
addps xmm0, xmm1
movups xmm2, [eax + 16]
mulps xmm2, [ebx + 16]
addps xmm0, xmm2
prefetcht0 [eax + 96] // give a prefetch hint to CPU what data are to be needed soonish
prefetcht0 [ebx + 64] // give a prefetch hint to CPU what data are to be needed soonish
movups xmm3, [eax + 32]
mulps xmm3, [ebx + 32]
addps xmm0, xmm3
movups xmm4, [eax + 48]
mulps xmm4, [ebx + 48]
addps xmm0, xmm4
add eax, 64
add ebx, 64
dec ecx
jnz loop1
// add the four floats of xmm0 together and return the result.
movhlps xmm1, xmm0 // move 3 & 4 of xmm0 to 1 & 2 of xmm1
addps xmm1, xmm0
movaps xmm2, xmm1
shufps xmm2, xmm2, 0x01 // move 2 of xmm2 as 1 of xmm2
addss xmm2, xmm1
movss corr, xmm2
}
return (double)corr;
*/
}
//////////////////////////////////////////////////////////////////////////////
//
// implementation of SSE optimized functions of class 'FIRFilter'
//
//////////////////////////////////////////////////////////////////////////////
#include "FIRFilter.h"
FIRFilterSSE::FIRFilterSSE() : FIRFilter()
{
filterCoeffsAlign = NULL;
filterCoeffsUnalign = NULL;
}
FIRFilterSSE::~FIRFilterSSE()
{
delete[] filterCoeffsUnalign;
filterCoeffsAlign = NULL;
filterCoeffsUnalign = NULL;
}
// (overloaded) Calculates filter coefficients for SSE routine
void FIRFilterSSE::setCoefficients(const float *coeffs, uint newLength, uint uResultDivFactor)
{
uint i;
float fDivider;
FIRFilter::setCoefficients(coeffs, newLength, uResultDivFactor);
// Scale the filter coefficients so that it won't be necessary to scale the filtering result
// also rearrange coefficients suitably for 3DNow!
// Ensure that filter coeffs array is aligned to 16-byte boundary
delete[] filterCoeffsUnalign;
filterCoeffsUnalign = new float[2 * newLength + 4];
filterCoeffsAlign = (float *)(((unsigned long)filterCoeffsUnalign + 15) & (ulong)-16);
fDivider = (float)resultDivider;
// rearrange the filter coefficients for mmx routines
for (i = 0; i < newLength; i ++)
{
filterCoeffsAlign[2 * i + 0] =
filterCoeffsAlign[2 * i + 1] = coeffs[i + 0] / fDivider;
}
}
// SSE-optimized version of the filter routine for stereo sound
uint FIRFilterSSE::evaluateFilterStereo(float *dest, const float *source, uint numSamples) const
{
int count = (int)((numSamples - length) & (uint)-2);
int j;
assert(count % 2 == 0);
if (count < 2) return 0;
assert(source != NULL);
assert(dest != NULL);
assert((length % 8) == 0);
assert(filterCoeffsAlign != NULL);
assert(((ulong)filterCoeffsAlign) % 16 == 0);
// filter is evaluated for two stereo samples with each iteration, thus use of 'j += 2'
for (j = 0; j < count; j += 2)
{
const float *pSrc;
const __m128 *pFil;
__m128 sum1, sum2;
uint i;
pSrc = source; // source audio data
pFil = (__m128*)filterCoeffsAlign; // filter coefficients. NOTE: Assumes coefficients
// are aligned to 16-byte boundary
sum1 = sum2 = _mm_setzero_ps();
for (i = 0; i < length / 8; i ++)
{
// Unroll loop for efficiency & calculate filter for 2*2 stereo samples
// at each pass
// sum1 is accu for 2*2 filtered stereo sound data at the primary sound data offset
// sum2 is accu for 2*2 filtered stereo sound data for the next sound sample offset.
sum1 = _mm_add_ps(sum1, _mm_mul_ps(_mm_loadu_ps(pSrc) , pFil[0]));
sum2 = _mm_add_ps(sum2, _mm_mul_ps(_mm_loadu_ps(pSrc + 2), pFil[0]));
sum1 = _mm_add_ps(sum1, _mm_mul_ps(_mm_loadu_ps(pSrc + 4), pFil[1]));
sum2 = _mm_add_ps(sum2, _mm_mul_ps(_mm_loadu_ps(pSrc + 6), pFil[1]));
sum1 = _mm_add_ps(sum1, _mm_mul_ps(_mm_loadu_ps(pSrc + 8) , pFil[2]));
sum2 = _mm_add_ps(sum2, _mm_mul_ps(_mm_loadu_ps(pSrc + 10), pFil[2]));
sum1 = _mm_add_ps(sum1, _mm_mul_ps(_mm_loadu_ps(pSrc + 12), pFil[3]));
sum2 = _mm_add_ps(sum2, _mm_mul_ps(_mm_loadu_ps(pSrc + 14), pFil[3]));
pSrc += 16;
pFil += 4;
}
// Now sum1 and sum2 both have a filtered 2-channel sample each, but we still need
// to sum the two hi- and lo-floats of these registers together.
// post-shuffle & add the filtered values and store to dest.
_mm_storeu_ps(dest, _mm_add_ps(
_mm_shuffle_ps(sum1, sum2, _MM_SHUFFLE(1,0,3,2)), // s2_1 s2_0 s1_3 s1_2
_mm_shuffle_ps(sum1, sum2, _MM_SHUFFLE(3,2,1,0)) // s2_3 s2_2 s1_1 s1_0
));
source += 4;
dest += 4;
}
// Ideas for further improvement:
// 1. If it could be guaranteed that 'source' were always aligned to 16-byte
// boundary, a faster aligned '_mm_load_ps' instruction could be used.
// 2. If it could be guaranteed that 'dest' were always aligned to 16-byte
// boundary, a faster '_mm_store_ps' instruction could be used.
return (uint)count;
/* original routine in C-language. please notice the C-version has differently
organized coefficients though.
double suml1, suml2;
double sumr1, sumr2;
uint i, j;
for (j = 0; j < count; j += 2)
{
const float *ptr;
const float *pFil;
suml1 = sumr1 = 0.0;
suml2 = sumr2 = 0.0;
ptr = src;
pFil = filterCoeffs;
for (i = 0; i < lengthLocal; i ++)
{
// unroll loop for efficiency.
suml1 += ptr[0] * pFil[0] +
ptr[2] * pFil[2] +
ptr[4] * pFil[4] +
ptr[6] * pFil[6];
sumr1 += ptr[1] * pFil[1] +
ptr[3] * pFil[3] +
ptr[5] * pFil[5] +
ptr[7] * pFil[7];
suml2 += ptr[8] * pFil[0] +
ptr[10] * pFil[2] +
ptr[12] * pFil[4] +
ptr[14] * pFil[6];
sumr2 += ptr[9] * pFil[1] +
ptr[11] * pFil[3] +
ptr[13] * pFil[5] +
ptr[15] * pFil[7];
ptr += 16;
pFil += 8;
}
dest[0] = (float)suml1;
dest[1] = (float)sumr1;
dest[2] = (float)suml2;
dest[3] = (float)sumr2;
src += 4;
dest += 4;
}
*/
/* Similar routine in assembly, again obsoleted due to maintainability
_asm
{
// Very important note: data in 'src' _must_ be aligned to
// 16-byte boundary!
mov edx, count
mov ebx, dword ptr src
mov eax, dword ptr dest
shr edx, 1
loop1:
// "outer loop" : during each round 2*2 output samples are calculated
// give prefetch hints to CPU of what data are to be needed soonish
prefetcht0 [ebx]
prefetcht0 [filterCoeffsLocal]
mov esi, ebx
mov edi, filterCoeffsLocal
xorps xmm0, xmm0
xorps xmm1, xmm1
mov ecx, lengthLocal
loop2:
// "inner loop" : during each round eight FIR filter taps are evaluated for 2*2 samples
prefetcht0 [esi + 32] // give a prefetch hint to CPU what data are to be needed soonish
prefetcht0 [edi + 32] // give a prefetch hint to CPU what data are to be needed soonish
movups xmm2, [esi] // possibly unaligned load
movups xmm3, [esi + 8] // possibly unaligned load
mulps xmm2, [edi]
mulps xmm3, [edi]
addps xmm0, xmm2
addps xmm1, xmm3
movups xmm4, [esi + 16] // possibly unaligned load
movups xmm5, [esi + 24] // possibly unaligned load
mulps xmm4, [edi + 16]
mulps xmm5, [edi + 16]
addps xmm0, xmm4
addps xmm1, xmm5
prefetcht0 [esi + 64] // give a prefetch hint to CPU what data are to be needed soonish
prefetcht0 [edi + 64] // give a prefetch hint to CPU what data are to be needed soonish
movups xmm6, [esi + 32] // possibly unaligned load
movups xmm7, [esi + 40] // possibly unaligned load
mulps xmm6, [edi + 32]
mulps xmm7, [edi + 32]
addps xmm0, xmm6
addps xmm1, xmm7
movups xmm4, [esi + 48] // possibly unaligned load
movups xmm5, [esi + 56] // possibly unaligned load
mulps xmm4, [edi + 48]
mulps xmm5, [edi + 48]
addps xmm0, xmm4
addps xmm1, xmm5
add esi, 64
add edi, 64
dec ecx
jnz loop2
// Now xmm0 and xmm1 both have a filtered 2-channel sample each, but we still need
// to sum the two hi- and lo-floats of these registers together.
movhlps xmm2, xmm0 // xmm2 = xmm2_3 xmm2_2 xmm0_3 xmm0_2
movlhps xmm2, xmm1 // xmm2 = xmm1_1 xmm1_0 xmm0_3 xmm0_2
shufps xmm0, xmm1, 0xe4 // xmm0 = xmm1_3 xmm1_2 xmm0_1 xmm0_0
addps xmm0, xmm2
movaps [eax], xmm0
add ebx, 16
add eax, 16
dec edx
jnz loop1
}
*/
}
#endif // ALLOW_SSE

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## Process this file with automake to create Makefile.in
##
## $Id: Makefile.am 4 2008-02-10 16:07:12Z oparviai $
##
## Copyright (C) 2003 - David W. Durham
##
## This file is part of SoundTouch, an audio processing library for pitch/time adjustments
##
## SoundTouch is free software; you can redistribute it and/or modify it under the
## terms of the GNU General Public License as published by the Free Software
## Foundation; either version 2 of the License, or (at your option) any later
## version.
##
## SoundTouch is distributed in the hope that it will be useful, but WITHOUT ANY
## WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
## A PARTICULAR PURPOSE. See the GNU General Public License for more details.
##
## You should have received a copy of the GNU General Public License along with
## this program; if not, write to the Free Software Foundation, Inc., 59 Temple
## Place - Suite 330, Boston, MA 02111-1307, USA
include $(top_srcdir)/config/am_include.mk
SUBDIRS=bpm SoundStretch
# set to something if you want other stuff to be included in the distribution tarball
#EXTRA_DIST=

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## Process this file with automake to create Makefile.in
##
## $Id: Makefile.am 4 2008-02-10 16:07:12Z oparviai $
##
## Copyright (C) 2003 - David W. Durham
##
## This file is part of SoundTouch, an audio processing library for pitch/time adjustments
##
## SoundTouch is free software; you can redistribute it and/or modify it under the
## terms of the GNU General Public License as published by the Free Software
## Foundation; either version 2 of the License, or (at your option) any later
## version.
##
## SoundTouch is distributed in the hope that it will be useful, but WITHOUT ANY
## WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
## A PARTICULAR PURPOSE. See the GNU General Public License for more details.
##
## You should have received a copy of the GNU General Public License along with
## this program; if not, write to the Free Software Foundation, Inc., 59 Temple
## Place - Suite 330, Boston, MA 02111-1307, USA
include $(top_srcdir)/config/am_include.mk
## bin_PROGRAMS is the macro that tells automake the name of the programs to
## install in the bin directory (/usr/local/bin) by default. By setting
## --prefix= at configure time the user can change this (eg: ./configure
## --prefix=/usr will install soundstretch under /usr/bin/soundstretch )
bin_PROGRAMS=soundstretch
noinst_HEADERS=RunParameters.h WavFile.h
# extra files to include in distrubution tarball
EXTRA_DIST=soundstretch.dsp soundstretch.dsw
## for every name listed under bin_PROGRAMS, you have a <prog>_SOURCES. This lists
## all the sources in the current directory that are used to build soundstretch.
soundstretch_SOURCES=main.cpp RunParameters.cpp WavFile.cpp
## soundstretch_LDADD is a list of extras to pass at link time. All the objects
## created by the above soundstretch_SOURCES are automatically linked in, so here I
## list object files from other directories as well as flags passed to the
## linker.
soundstretch_LDADD=../../SoundTouch/libSoundTouch.la ../bpm/libBPM.la -lm
## linker flags
soundstretch_LDFLAGS=-s
## additional compiler flags
soundstretch_CXXFLAGS=-O3 -I../bpm
#clean-local:
# -rm -f additional-files-to-remove-on-make-clean

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////////////////////////////////////////////////////////////////////////////////
///
/// A class for parsing the 'soundstretch' application command line parameters
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2008-02-10 18:07:12 +0200 (su, 10 helmi 2008) $
// File revision : $Revision: 4 $
//
// $Id: RunParameters.cpp 4 2008-02-10 16:07:12Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#include <stdexcept>
#include <string>
#include <stdlib.h>
#include "RunParameters.h"
using namespace std;
// Program usage instructions
static const char licenseText[] =
" LICENSE:\n"
" ========\n"
" \n"
" SoundTouch sound processing library\n"
" Copyright (c) Olli Parviainen\n"
" \n"
" This library is free software; you can redistribute it and/or\n"
" modify it under the terms of the GNU Lesser General Public\n"
" License as published by the Free Software Foundation; either\n"
" version 2.1 of the License, or (at your option) any later version.\n"
" \n"
" This library is distributed in the hope that it will be useful,\n"
" but WITHOUT ANY WARRANTY; without even the implied warranty of\n"
" MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU\n"
" Lesser General Public License for more details.\n"
" \n"
" You should have received a copy of the GNU Lesser General Public\n"
" License along with this library; if not, write to the Free Software\n"
" Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA\n"
" \n"
"This application is distributed with full source codes; however, if you\n"
"didn't receive them, please visit the author's homepage (see the link above).";
static const char whatText[] =
"This application processes WAV audio files by modifying the sound tempo,\n"
"pitch and playback rate properties independently from each other.\n"
"\n";
static const char usage[] =
"Usage :\n"
" soundstretch infile.wav outfile.wav [switches]\n\n"
"Available switches are:\n"
" -tempo=n : Change sound tempo by n percents (n=-95..+5000 %)\n"
" -pitch=n : Change sound pitch by n semitones (n=-60..+60 semitones)\n"
" -rate=n : Change sound rate by n percents (n=-95..+5000 %)\n"
" -bpm=n : Detect the BPM rate of sound and adjust tempo to meet 'n' BPMs.\n"
" If '=n' is omitted, just detects the BPM rate.\n"
" -quick : Use quicker tempo change algorithm (gain speed, lose quality)\n"
" -naa : Don't use anti-alias filtering (gain speed, lose quality)\n"
" -license : Display the program license text (LGPL)\n";
// Converts a char into lower case
static int _toLowerCase(int c)
{
if (c >= 'A' && c <= 'Z')
{
c += 'a' - 'A';
}
return c;
}
// Constructor
RunParameters::RunParameters(const int nParams, const char *paramStr[])
{
int i;
int nFirstParam;
if (nParams < 3)
{
// Too few parameters
if (nParams > 1 && paramStr[1][0] == '-' &&
_toLowerCase(paramStr[1][1]) == 'l')
{
// '-license' switch
throwLicense();
}
string msg = whatText;
msg += usage;
throw runtime_error(msg.c_str());
}
inFileName = NULL;
outFileName = NULL;
tempoDelta = 0;
pitchDelta = 0;
rateDelta = 0;
quick = 0;
noAntiAlias = 0;
goalBPM = 0;
detectBPM = FALSE;
// Get input & output file names
inFileName = (char*)paramStr[1];
outFileName = (char*)paramStr[2];
if (outFileName[0] == '-')
{
// no outputfile name was given but parameters
outFileName = NULL;
nFirstParam = 2;
}
else
{
nFirstParam = 3;
}
// parse switch parameters
for (i = nFirstParam; i < nParams; i ++)
{
parseSwitchParam(paramStr[i]);
}
checkLimits();
}
// Checks parameter limits
void RunParameters::checkLimits()
{
if (tempoDelta < -95.0f)
{
tempoDelta = -95.0f;
}
else if (tempoDelta > 5000.0f)
{
tempoDelta = 5000.0f;
}
if (pitchDelta < -60.0f)
{
pitchDelta = -60.0f;
}
else if (pitchDelta > 60.0f)
{
pitchDelta = 60.0f;
}
if (rateDelta < -95.0f)
{
rateDelta = -95.0f;
}
else if (rateDelta > 5000.0f)
{
rateDelta = 5000.0f;
}
}
// Unknown switch parameter -- throws an exception with an error message
void RunParameters::throwIllegalParamExp(const string &str) const
{
string msg = "ERROR : Illegal parameter \"";
msg += str;
msg += "\".\n\n";
msg += usage;
throw runtime_error(msg.c_str());
}
void RunParameters::throwLicense() const
{
throw runtime_error(licenseText);
}
float RunParameters::parseSwitchValue(const string &str) const
{
int pos;
pos = str.find_first_of('=');
if (pos < 0)
{
// '=' missing
throwIllegalParamExp(str);
}
// Read numerical parameter value after '='
return (float)atof(str.substr(pos + 1).c_str());
}
// Interprets a single switch parameter string of format "-switch=xx"
// Valid switches are "-tempo=xx", "-pitch=xx" and "-rate=xx". Stores
// switch values into 'params' structure.
void RunParameters::parseSwitchParam(const string &str)
{
int upS;
if (str[0] != '-')
{
// leading hyphen missing => not a valid parameter
throwIllegalParamExp(str);
}
// Take the first character of switch name & change to lower case
upS = _toLowerCase(str[1]);
// interpret the switch name & operate accordingly
switch (upS)
{
case 't' :
// switch '-tempo=xx'
tempoDelta = parseSwitchValue(str);
break;
case 'p' :
// switch '-pitch=xx'
pitchDelta = parseSwitchValue(str);
break;
case 'r' :
// switch '-rate=xx'
rateDelta = parseSwitchValue(str);
break;
case 'b' :
// switch '-bpm=xx'
detectBPM = TRUE;
try
{
goalBPM = parseSwitchValue(str);
}
catch (runtime_error)
{
// illegal or missing bpm value => just calculate bpm
goalBPM = 0;
}
break;
case 'q' :
// switch '-quick'
quick = 1;
break;
case 'n' :
// switch '-naa'
noAntiAlias = 1;
break;
case 'l' :
// switch '-license'
throwLicense();
break;
default:
// unknown switch
throwIllegalParamExp(str);
}
}

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////////////////////////////////////////////////////////////////////////////////
///
/// A class for parsing the 'soundstretch' application command line parameters
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2008-02-10 18:07:12 +0200 (su, 10 helmi 2008) $
// File revision : $Revision: 4 $
//
// $Id: RunParameters.h 4 2008-02-10 16:07:12Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#ifndef RUNPARAMETERS_H
#define RUNPARAMETERS_H
#include "STTypes.h"
#include <string>
using namespace std;
/// Parses command line parameters into program parameters
class RunParameters
{
private:
void throwIllegalParamExp(const string &str) const;
void throwLicense() const;
void parseSwitchParam(const string &str);
void checkLimits();
float parseSwitchValue(const string &str) const;
public:
char *inFileName;
char *outFileName;
float tempoDelta;
float pitchDelta;
float rateDelta;
int quick;
int noAntiAlias;
float goalBPM;
BOOL detectBPM;
RunParameters(const int nParams, const char *paramStr[]);
};
#endif

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////////////////////////////////////////////////////////////////////////////////
///
/// Classes for easy reading & writing of WAV sound files.
///
/// For big-endian CPU, define _BIG_ENDIAN_ during compile-time to correctly
/// parse the WAV files with such processors.
///
/// Admittingly, more complete WAV reader routines may exist in public domain,
/// but the reason for 'yet another' one is that those generic WAV reader
/// libraries are exhaustingly large and cumbersome! Wanted to have something
/// simpler here, i.e. something that's not already larger than rest of the
/// SoundTouch/SoundStretch program...
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2008-02-10 18:07:12 +0200 (su, 10 helmi 2008) $
// File revision : $Revision: 4 $
//
// $Id: WavFile.cpp 4 2008-02-10 16:07:12Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#include <stdio.h>
#include <stdexcept>
#include <string>
#include <assert.h>
#include <limits.h>
#include "WavFile.h"
using namespace std;
const static char riffStr[] = "RIFF";
const static char waveStr[] = "WAVE";
const static char fmtStr[] = "fmt ";
const static char dataStr[] = "data";
//////////////////////////////////////////////////////////////////////////////
//
// Helper functions for swapping byte order to correctly read/write WAV files
// with big-endian CPU's: Define compile-time definition _BIG_ENDIAN_ to
// turn-on the conversion if it appears necessary.
//
// For example, Intel x86 is little-endian and doesn't require conversion,
// while PowerPC of Mac's and many other RISC cpu's are big-endian.
#ifdef BYTE_ORDER
// In gcc compiler detect the byte order automatically
#if BYTE_ORDER == BIG_ENDIAN
// big-endian platform.
#define _BIG_ENDIAN_
#endif
#endif
#ifdef _BIG_ENDIAN_
// big-endian CPU, swap bytes in 16 & 32 bit words
// helper-function to swap byte-order of 32bit integer
static inline void _swap32(unsigned int &dwData)
{
dwData = ((dwData >> 24) & 0x000000FF) |
((dwData >> 8) & 0x0000FF00) |
((dwData << 8) & 0x00FF0000) |
((dwData << 24) & 0xFF000000);
}
// helper-function to swap byte-order of 16bit integer
static inline void _swap16(unsigned short &wData)
{
wData = ((wData >> 8) & 0x00FF) |
((wData << 8) & 0xFF00);
}
// helper-function to swap byte-order of buffer of 16bit integers
static inline void _swap16Buffer(unsigned short *pData, unsigned int dwNumWords)
{
unsigned long i;
for (i = 0; i < dwNumWords; i ++)
{
_swap16(pData[i]);
}
}
#else // BIG_ENDIAN
// little-endian CPU, WAV file is ok as such
// dummy helper-function
static inline void _swap32(unsigned int &dwData)
{
// do nothing
}
// dummy helper-function
static inline void _swap16(unsigned short &wData)
{
// do nothing
}
// dummy helper-function
static inline void _swap16Buffer(unsigned short *pData, unsigned int dwNumBytes)
{
// do nothing
}
#endif // BIG_ENDIAN
//////////////////////////////////////////////////////////////////////////////
//
// Class WavInFile
//
WavInFile::WavInFile(const char *fileName)
{
int hdrsOk;
// Try to open the file for reading
fptr = fopen(fileName, "rb");
if (fptr == NULL)
{
// didn't succeed
string msg = "Error : Unable to open file \"";
msg += fileName;
msg += "\" for reading.";
throw runtime_error(msg);
}
// Read the file headers
hdrsOk = readWavHeaders();
if (hdrsOk != 0)
{
// Something didn't match in the wav file headers
string msg = "File \"";
msg += fileName;
msg += "\" is corrupt or not a WAV file";
throw runtime_error(msg);
}
if (header.format.fixed != 1)
{
string msg = "File \"";
msg += fileName;
msg += "\" uses unsupported encoding.";
throw runtime_error(msg);
}
dataRead = 0;
}
WavInFile::~WavInFile()
{
close();
}
void WavInFile::rewind()
{
int hdrsOk;
fseek(fptr, 0, SEEK_SET);
hdrsOk = readWavHeaders();
assert(hdrsOk == 0);
dataRead = 0;
}
int WavInFile::checkCharTags()
{
// header.format.fmt should equal to 'fmt '
if (memcmp(fmtStr, header.format.fmt, 4) != 0) return -1;
// header.data.data_field should equal to 'data'
if (memcmp(dataStr, header.data.data_field, 4) != 0) return -1;
return 0;
}
int WavInFile::read(char *buffer, int maxElems)
{
int numBytes;
uint afterDataRead;
// ensure it's 8 bit format
if (header.format.bits_per_sample != 8)
{
throw runtime_error("Error: WavInFile::read(char*, int) works only with 8bit samples.");
}
assert(sizeof(char) == 1);
numBytes = maxElems;
afterDataRead = dataRead + numBytes;
if (afterDataRead > header.data.data_len)
{
// Don't read more samples than are marked available in header
numBytes = header.data.data_len - dataRead;
assert(numBytes >= 0);
}
numBytes = fread(buffer, 1, numBytes, fptr);
dataRead += numBytes;
return numBytes;
}
int WavInFile::read(short *buffer, int maxElems)
{
unsigned int afterDataRead;
int numBytes;
int numElems;
if (header.format.bits_per_sample == 8)
{
// 8 bit format
char *temp = new char[maxElems];
int i;
numElems = read(temp, maxElems);
// convert from 8 to 16 bit
for (i = 0; i < numElems; i ++)
{
buffer[i] = temp[i] << 8;
}
delete[] temp;
}
else
{
// 16 bit format
assert(header.format.bits_per_sample == 16);
assert(sizeof(short) == 2);
numBytes = maxElems * 2;
afterDataRead = dataRead + numBytes;
if (afterDataRead > header.data.data_len)
{
// Don't read more samples than are marked available in header
numBytes = header.data.data_len - dataRead;
assert(numBytes >= 0);
}
numBytes = fread(buffer, 1, numBytes, fptr);
dataRead += numBytes;
numElems = numBytes / 2;
// 16bit samples, swap byte order if necessary
_swap16Buffer((unsigned short *)buffer, numElems);
}
return numElems;
}
int WavInFile::read(float *buffer, int maxElems)
{
short *temp = new short[maxElems];
int num;
int i;
double fscale;
num = read(temp, maxElems);
fscale = 1.0 / 32768.0;
// convert to floats, scale to range [-1..+1[
for (i = 0; i < num; i ++)
{
buffer[i] = (float)(fscale * (double)temp[i]);
}
delete[] temp;
return num;
}
int WavInFile::eof() const
{
// return true if all data has been read or file eof has reached
return (dataRead == header.data.data_len || feof(fptr));
}
void WavInFile::close()
{
fclose(fptr);
fptr = NULL;
}
// test if character code is between a white space ' ' and little 'z'
static int isAlpha(char c)
{
return (c >= ' ' && c <= 'z') ? 1 : 0;
}
// test if all characters are between a white space ' ' and little 'z'
static int isAlphaStr(char *str)
{
int c;
c = str[0];
while (c)
{
if (isAlpha(c) == 0) return 0;
str ++;
c = str[0];
}
return 1;
}
int WavInFile::readRIFFBlock()
{
fread(&(header.riff), sizeof(WavRiff), 1, fptr);
// swap 32bit data byte order if necessary
_swap32((unsigned int &)header.riff.package_len);
// header.riff.riff_char should equal to 'RIFF');
if (memcmp(riffStr, header.riff.riff_char, 4) != 0) return -1;
// header.riff.wave should equal to 'WAVE'
if (memcmp(waveStr, header.riff.wave, 4) != 0) return -1;
return 0;
}
int WavInFile::readHeaderBlock()
{
char label[5];
string sLabel;
// lead label string
fread(label, 1, 4, fptr);
label[4] = 0;
if (isAlphaStr(label) == 0) return -1; // not a valid label
// Decode blocks according to their label
if (strcmp(label, fmtStr) == 0)
{
int nLen, nDump;
// 'fmt ' block
memcpy(header.format.fmt, fmtStr, 4);
// read length of the format field
fread(&nLen, sizeof(int), 1, fptr);
// swap byte order if necessary
_swap32((unsigned int &)nLen); // int format_len;
header.format.format_len = nLen;
// calculate how much length differs from expected
nDump = nLen - (sizeof(header.format) - 8);
// if format_len is larger than expected, read only as much data as we've space for
if (nDump > 0)
{
nLen = sizeof(header.format) - 8;
}
// read data
fread(&(header.format.fixed), nLen, 1, fptr);
// swap byte order if necessary
_swap16((unsigned short &)header.format.fixed); // short int fixed;
_swap16((unsigned short &)header.format.channel_number); // short int channel_number;
_swap32((unsigned int &)header.format.sample_rate); // int sample_rate;
_swap32((unsigned int &)header.format.byte_rate); // int byte_rate;
_swap16((unsigned short &)header.format.byte_per_sample); // short int byte_per_sample;
_swap16((unsigned short &)header.format.bits_per_sample); // short int bits_per_sample;
// if format_len is larger than expected, skip the extra data
if (nDump > 0)
{
fseek(fptr, nDump, SEEK_CUR);
}
return 0;
}
else if (strcmp(label, dataStr) == 0)
{
// 'data' block
memcpy(header.data.data_field, dataStr, 4);
fread(&(header.data.data_len), sizeof(uint), 1, fptr);
// swap byte order if necessary
_swap32((unsigned int &)header.data.data_len);
return 1;
}
else
{
uint len, i;
uint temp;
// unknown block
// read length
fread(&len, sizeof(len), 1, fptr);
// scan through the block
for (i = 0; i < len; i ++)
{
fread(&temp, 1, 1, fptr);
if (feof(fptr)) return -1; // unexpected eof
}
}
return 0;
}
int WavInFile::readWavHeaders()
{
int res;
memset(&header, 0, sizeof(header));
res = readRIFFBlock();
if (res) return 1;
// read header blocks until data block is found
do
{
// read header blocks
res = readHeaderBlock();
if (res < 0) return 1; // error in file structure
} while (res == 0);
// check that all required tags are legal
return checkCharTags();
}
uint WavInFile::getNumChannels() const
{
return header.format.channel_number;
}
uint WavInFile::getNumBits() const
{
return header.format.bits_per_sample;
}
uint WavInFile::getBytesPerSample() const
{
return getNumChannels() * getNumBits() / 8;
}
uint WavInFile::getSampleRate() const
{
return header.format.sample_rate;
}
uint WavInFile::getDataSizeInBytes() const
{
return header.data.data_len;
}
uint WavInFile::getNumSamples() const
{
return header.data.data_len / header.format.byte_per_sample;
}
uint WavInFile::getLengthMS() const
{
uint numSamples;
uint sampleRate;
numSamples = getNumSamples();
sampleRate = getSampleRate();
assert(numSamples < UINT_MAX / 1000);
return (1000 * numSamples / sampleRate);
}
//////////////////////////////////////////////////////////////////////////////
//
// Class WavOutFile
//
WavOutFile::WavOutFile(const char *fileName, int sampleRate, int bits, int channels)
{
bytesWritten = 0;
fptr = fopen(fileName, "wb");
if (fptr == NULL)
{
string msg = "Error : Unable to open file \"";
msg += fileName;
msg += "\" for writing.";
//pmsg = msg.c_str;
throw runtime_error(msg);
}
fillInHeader(sampleRate, bits, channels);
writeHeader();
}
WavOutFile::~WavOutFile()
{
close();
}
void WavOutFile::fillInHeader(uint sampleRate, uint bits, uint channels)
{
// fill in the 'riff' part..
// copy string 'RIFF' to riff_char
memcpy(&(header.riff.riff_char), riffStr, 4);
// package_len unknown so far
header.riff.package_len = 0;
// copy string 'WAVE' to wave
memcpy(&(header.riff.wave), waveStr, 4);
// fill in the 'format' part..
// copy string 'fmt ' to fmt
memcpy(&(header.format.fmt), fmtStr, 4);
header.format.format_len = 0x10;
header.format.fixed = 1;
header.format.channel_number = (short)channels;
header.format.sample_rate = sampleRate;
header.format.bits_per_sample = (short)bits;
header.format.byte_per_sample = (short)(bits * channels / 8);
header.format.byte_rate = header.format.byte_per_sample * sampleRate;
header.format.sample_rate = sampleRate;
// fill in the 'data' part..
// copy string 'data' to data_field
memcpy(&(header.data.data_field), dataStr, 4);
// data_len unknown so far
header.data.data_len = 0;
}
void WavOutFile::finishHeader()
{
// supplement the file length into the header structure
header.riff.package_len = bytesWritten + 36;
header.data.data_len = bytesWritten;
writeHeader();
}
void WavOutFile::writeHeader()
{
WavHeader hdrTemp;
// swap byte order if necessary
hdrTemp = header;
_swap32((unsigned int &)hdrTemp.riff.package_len);
_swap32((unsigned int &)hdrTemp.format.format_len);
_swap16((unsigned short &)hdrTemp.format.fixed);
_swap16((unsigned short &)hdrTemp.format.channel_number);
_swap32((unsigned int &)hdrTemp.format.sample_rate);
_swap32((unsigned int &)hdrTemp.format.byte_rate);
_swap16((unsigned short &)hdrTemp.format.byte_per_sample);
_swap16((unsigned short &)hdrTemp.format.bits_per_sample);
_swap32((unsigned int &)hdrTemp.data.data_len);
// write the supplemented header in the beginning of the file
fseek(fptr, 0, SEEK_SET);
fwrite(&hdrTemp, sizeof(hdrTemp), 1, fptr);
// jump back to the end of the file
fseek(fptr, 0, SEEK_END);
}
void WavOutFile::close()
{
finishHeader();
fclose(fptr);
fptr = NULL;
}
void WavOutFile::write(const char *buffer, int numElems)
{
int res;
if (header.format.bits_per_sample != 8)
{
throw runtime_error("Error: WavOutFile::write(const char*, int) accepts only 8bit samples.");
}
assert(sizeof(char) == 1);
res = fwrite(buffer, 1, numElems, fptr);
if (res != numElems)
{
throw runtime_error("Error while writing to a wav file.");
}
bytesWritten += numElems;
}
void WavOutFile::write(const short *buffer, int numElems)
{
int res;
// 16 bit samples
if (numElems < 1) return; // nothing to do
if (header.format.bits_per_sample == 8)
{
int i;
char *temp = new char[numElems];
// convert from 16bit format to 8bit format
for (i = 0; i < numElems; i ++)
{
temp[i] = buffer[i] >> 8;
}
// write in 8bit format
write(temp, numElems);
delete[] temp;
}
else
{
// 16bit format
unsigned short *pTemp = new unsigned short[numElems];
assert(header.format.bits_per_sample == 16);
// allocate temp buffer to swap byte order if necessary
memcpy(pTemp, buffer, numElems * 2);
_swap16Buffer(pTemp, numElems);
res = fwrite(pTemp, 2, numElems, fptr);
delete[] pTemp;
if (res != numElems)
{
throw runtime_error("Error while writing to a wav file.");
}
bytesWritten += 2 * numElems;
}
}
void WavOutFile::write(const float *buffer, int numElems)
{
int i;
short *temp = new short[numElems];
int iTemp;
// convert to 16 bit integer
for (i = 0; i < numElems; i ++)
{
// convert to integer
iTemp = (int)(32768.0f * buffer[i]);
// saturate
if (iTemp < -32768) iTemp = -32768;
if (iTemp > 32767) iTemp = 32767;
temp[i] = (short)iTemp;
}
write(temp, numElems);
delete[] temp;
}

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////////////////////////////////////////////////////////////////////////////////
///
/// Classes for easy reading & writing of WAV sound files.
///
/// For big-endian CPU, define BIG_ENDIAN during compile-time to correctly
/// parse the WAV files with such processors.
///
/// Admittingly, more complete WAV reader routines may exist in public domain, but
/// the reason for 'yet another' one is that those generic WAV reader libraries are
/// exhaustingly large and cumbersome! Wanted to have something simpler here, i.e.
/// something that's not already larger than rest of the SoundTouch/SoundStretch program...
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2008-02-10 18:07:12 +0200 (su, 10 helmi 2008) $
// File revision : $Revision: 4 $
//
// $Id: WavFile.h 4 2008-02-10 16:07:12Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#ifndef WAVFILE_H
#define WAVFILE_H
#include <stdio.h>
#ifndef uint
typedef unsigned int uint;
#endif
/// WAV audio file 'riff' section header
typedef struct
{
char riff_char[4];
int package_len;
char wave[4];
} WavRiff;
/// WAV audio file 'format' section header
typedef struct
{
char fmt[4];
int format_len;
short fixed;
short channel_number;
int sample_rate;
int byte_rate;
short byte_per_sample;
short bits_per_sample;
} WavFormat;
/// WAV audio file 'data' section header
typedef struct
{
char data_field[4];
uint data_len;
} WavData;
/// WAV audio file header
typedef struct
{
WavRiff riff;
WavFormat format;
WavData data;
} WavHeader;
/// Class for reading WAV audio files.
class WavInFile
{
private:
/// File pointer.
FILE *fptr;
/// Counter of how many bytes of sample data have been read from the file.
uint dataRead;
/// WAV header information
WavHeader header;
/// Read WAV file headers.
/// \return zero if all ok, nonzero if file format is invalid.
int readWavHeaders();
/// Checks WAV file header tags.
/// \return zero if all ok, nonzero if file format is invalid.
int checkCharTags();
/// Reads a single WAV file header block.
/// \return zero if all ok, nonzero if file format is invalid.
int readHeaderBlock();
/// Reads WAV file 'riff' block
int readRIFFBlock();
public:
/// Constructor: Opens the given WAV file. If the file can't be opened,
/// throws 'runtime_error' exception.
WavInFile(const char *filename);
/// Destructor: Closes the file.
~WavInFile();
/// Close the file. Notice that file is automatically closed also when the
/// class instance is deleted.
void close();
/// Rewind to beginning of the file
void rewind();
/// Get sample rate.
uint getSampleRate() const;
/// Get number of bits per sample, i.e. 8 or 16.
uint getNumBits() const;
/// Get sample data size in bytes. Ahem, this should return same information as
/// 'getBytesPerSample'...
uint getDataSizeInBytes() const;
/// Get total number of samples in file.
uint getNumSamples() const;
/// Get number of bytes per audio sample (e.g. 16bit stereo = 4 bytes/sample)
uint getBytesPerSample() const;
/// Get number of audio channels in the file (1=mono, 2=stereo)
uint getNumChannels() const;
/// Get the audio file length in milliseconds
uint getLengthMS() const;
/// Reads audio samples from the WAV file. This routine works only for 8 bit samples.
/// Reads given number of elements from the file or if end-of-file reached, as many
/// elements as are left in the file.
///
/// \return Number of 8-bit integers read from the file.
int read(char *buffer, int maxElems);
/// Reads audio samples from the WAV file to 16 bit integer format. Reads given number
/// of elements from the file or if end-of-file reached, as many elements as are
/// left in the file.
///
/// \return Number of 16-bit integers read from the file.
int read(short *buffer, ///< Pointer to buffer where to read data.
int maxElems ///< Size of 'buffer' array (number of array elements).
);
/// Reads audio samples from the WAV file to floating point format, converting
/// sample values to range [-1,1[. Reads given number of elements from the file
/// or if end-of-file reached, as many elements as are left in the file.
///
/// \return Number of elements read from the file.
int read(float *buffer, ///< Pointer to buffer where to read data.
int maxElems ///< Size of 'buffer' array (number of array elements).
);
/// Check end-of-file.
///
/// \return Nonzero if end-of-file reached.
int eof() const;
};
/// Class for writing WAV audio files.
class WavOutFile
{
private:
/// Pointer to the WAV file
FILE *fptr;
/// WAV file header data.
WavHeader header;
/// Counter of how many bytes have been written to the file so far.
int bytesWritten;
/// Fills in WAV file header information.
void fillInHeader(const uint sampleRate, const uint bits, const uint channels);
/// Finishes the WAV file header by supplementing information of amount of
/// data written to file etc
void finishHeader();
/// Writes the WAV file header.
void writeHeader();
public:
/// Constructor: Creates a new WAV file. Throws a 'runtime_error' exception
/// if file creation fails.
WavOutFile(const char *fileName, ///< Filename
int sampleRate, ///< Sample rate (e.g. 44100 etc)
int bits, ///< Bits per sample (8 or 16 bits)
int channels ///< Number of channels (1=mono, 2=stereo)
);
/// Destructor: Finalizes & closes the WAV file.
~WavOutFile();
/// Write data to WAV file. This function works only with 8bit samples.
/// Throws a 'runtime_error' exception if writing to file fails.
void write(const char *buffer, ///< Pointer to sample data buffer.
int numElems ///< How many array items are to be written to file.
);
/// Write data to WAV file. Throws a 'runtime_error' exception if writing to
/// file fails.
void write(const short *buffer, ///< Pointer to sample data buffer.
int numElems ///< How many array items are to be written to file.
);
/// Write data to WAV file in floating point format, saturating sample values to range
/// [-1..+1[. Throws a 'runtime_error' exception if writing to file fails.
void write(const float *buffer, ///< Pointer to sample data buffer.
int numElems ///< How many array items are to be written to file.
);
/// Finalize & close the WAV file. Automatically supplements the WAV file header
/// information according to written data etc.
///
/// Notice that file is automatically closed also when the class instance is deleted.
void close();
};
#endif

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////////////////////////////////////////////////////////////////////////////////
///
/// SoundStretch main routine.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2008-02-10 18:07:12 +0200 (su, 10 helmi 2008) $
// File revision : $Revision: 4 $
//
// $Id: main.cpp 4 2008-02-10 16:07:12Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#include <stdexcept>
#include <stdio.h>
#include "RunParameters.h"
#include "WavFile.h"
#include "SoundTouch.h"
#include "BPMDetect.h"
using namespace soundtouch;
using namespace std;
// Processing chunk size
#define BUFF_SIZE 2048
static const char _helloText[] =
"\n"
" SoundStretch v%s - Written by Olli Parviainen 2001 - 2006\n"
"==================================================================\n"
"author e-mail: <oparviai@iki.fi> - WWW: http://www.surina.net/soundtouch\n"
"\n"
"This program is subject to (L)GPL license. Run \"soundstretch -license\" for\n"
"more information.\n"
"\n";
static void openFiles(WavInFile **inFile, WavOutFile **outFile, const RunParameters *params)
{
int bits, samplerate, channels;
// open input file...
*inFile = new WavInFile(params->inFileName);
// ... open output file with same sound parameters
bits = (*inFile)->getNumBits();
samplerate = (*inFile)->getSampleRate();
channels = (*inFile)->getNumChannels();
if (params->outFileName)
{
*outFile = new WavOutFile(params->outFileName, samplerate, bits, channels);
}
else
{
*outFile = NULL;
}
}
// Sets the 'SoundTouch' object up according to input file sound format &
// command line parameters
static void setup(SoundTouch *pSoundTouch, const WavInFile *inFile, const RunParameters *params)
{
int sampleRate;
int channels;
sampleRate = inFile->getSampleRate();
channels = inFile->getNumChannels();
pSoundTouch->setSampleRate(sampleRate);
pSoundTouch->setChannels(channels);
pSoundTouch->setTempoChange(params->tempoDelta);
pSoundTouch->setPitchSemiTones(params->pitchDelta);
pSoundTouch->setRateChange(params->rateDelta);
pSoundTouch->setSetting(SETTING_USE_QUICKSEEK, params->quick);
pSoundTouch->setSetting(SETTING_USE_AA_FILTER, !params->noAntiAlias);
// print processing information
if (params->outFileName)
{
#ifdef INTEGER_SAMPLES
printf("Uses 16bit integer sample type in processing.\n\n");
#else
#ifndef FLOAT_SAMPLES
#error "Sampletype not defined"
#endif
printf("Uses 32bit floating point sample type in processing.\n\n");
#endif
// print processing information only if outFileName given i.e. some processing will happen
printf("Processing the file with the following changes:\n");
printf(" tempo change = %+g %%\n", params->tempoDelta);
printf(" pitch change = %+g semitones\n", params->pitchDelta);
printf(" rate change = %+g %%\n\n", params->rateDelta);
printf("Working...");
}
else
{
// outFileName not given
printf("Warning: output file name missing, won't output anything.\n\n");
}
fflush(stdout);
}
// Processes the sound
static void process(SoundTouch *pSoundTouch, WavInFile *inFile, WavOutFile *outFile)
{
int nSamples;
int nChannels;
int buffSizeSamples;
SAMPLETYPE sampleBuffer[BUFF_SIZE];
if ((inFile == NULL) || (outFile == NULL)) return; // nothing to do.
nChannels = inFile->getNumChannels();
buffSizeSamples = BUFF_SIZE / nChannels;
// Process samples read from the input file
while (inFile->eof() == 0)
{
int num;
// Read a chunk of samples from the input file
num = inFile->read(sampleBuffer, BUFF_SIZE);
nSamples = num / inFile->getNumChannels();
// Feed the samples into SoundTouch processor
pSoundTouch->putSamples(sampleBuffer, nSamples);
// Read ready samples from SoundTouch processor & write them output file.
// NOTES:
// - 'receiveSamples' doesn't necessarily return any samples at all
// during some rounds!
// - On the other hand, during some round 'receiveSamples' may have more
// ready samples than would fit into 'sampleBuffer', and for this reason
// the 'receiveSamples' call is iterated for as many times as it
// outputs samples.
do
{
nSamples = pSoundTouch->receiveSamples(sampleBuffer, buffSizeSamples);
outFile->write(sampleBuffer, nSamples * nChannels);
} while (nSamples != 0);
}
// Now the input file is processed, yet 'flush' few last samples that are
// hiding in the SoundTouch's internal processing pipeline.
pSoundTouch->flush();
do
{
nSamples = pSoundTouch->receiveSamples(sampleBuffer, buffSizeSamples);
outFile->write(sampleBuffer, nSamples * nChannels);
} while (nSamples != 0);
}
// Detect BPM rate of inFile and adjust tempo setting accordingly if necessary
static void detectBPM(WavInFile *inFile, RunParameters *params)
{
float bpmValue;
int nChannels;
BPMDetect bpm(inFile->getNumChannels(), inFile->getSampleRate());
SAMPLETYPE sampleBuffer[BUFF_SIZE];
// detect bpm rate
printf("Detecting BPM rate...");
fflush(stdout);
nChannels = inFile->getNumChannels();
// Process the 'inFile' in small blocks, repeat until whole file has
// been processed
while (inFile->eof() == 0)
{
int num, samples;
// Read sample data from input file
num = inFile->read(sampleBuffer, BUFF_SIZE);
// Enter the new samples to the bpm analyzer class
samples = num / nChannels;
bpm.inputSamples(sampleBuffer, samples);
}
// Now the whole song data has been analyzed. Read the resulting bpm.
bpmValue = bpm.getBpm();
printf("Done!\n");
// rewind the file after bpm detection
inFile->rewind();
if (bpmValue > 0)
{
printf("Detected BPM rate %.1f\n\n", bpmValue);
}
else
{
printf("Couldn't detect BPM rate.\n\n");
return;
}
if (params->goalBPM > 0)
{
// adjust tempo to given bpm
params->tempoDelta = (params->goalBPM / bpmValue - 1.0f) * 100.0f;
printf("The file will be converted to %.1f BPM\n\n", params->goalBPM);
}
}
int main(const int nParams, const char *paramStr[])
{
WavInFile *inFile;
WavOutFile *outFile;
RunParameters *params;
SoundTouch SoundTouch;
printf(_helloText, SoundTouch::getVersionString());
try
{
// Parse command line parameters
params = new RunParameters(nParams, paramStr);
// Open input & output files
openFiles(&inFile, &outFile, params);
if (params->detectBPM == TRUE)
{
// detect sound BPM (and adjust processing parameters
// accordingly if necessary)
detectBPM(inFile, params);
}
// Setup the 'SoundTouch' object for processing the sound
setup(&SoundTouch, inFile, params);
// Process the sound
process(&SoundTouch, inFile, outFile);
// Close WAV file handles & dispose of the objects
delete inFile;
delete outFile;
delete params;
printf("Done!\n");
}
catch (runtime_error &e)
{
// An exception occurred during processing, display an error message
printf("%s\n", e.what());
return -1;
}
return 0;
}

137
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////////////////////////////////////////////////////////////////////////////////
///
/// Beats-per-minute (BPM) detection routine.
///
/// The beat detection algorithm works as follows:
/// - Use function 'inputSamples' to input a chunks of samples to the class for
/// analysis. It's a good idea to enter a large sound file or stream in smallish
/// chunks of around few kilosamples in order not to extinguish too much RAM memory.
/// - Inputted sound data is decimated to approx 500 Hz to reduce calculation burden,
/// which is basically ok as low (bass) frequencies mostly determine the beat rate.
/// Simple averaging is used for anti-alias filtering because the resulting signal
/// quality isn't of that high importance.
/// - Decimated sound data is enveloped, i.e. the amplitude shape is detected by
/// taking absolute value that's smoothed by sliding average. Signal levels that
/// are below a couple of times the general RMS amplitude level are cut away to
/// leave only notable peaks there.
/// - Repeating sound patterns (e.g. beats) are detected by calculating short-term
/// autocorrelation function of the enveloped signal.
/// - After whole sound data file has been analyzed as above, the bpm level is
/// detected by function 'getBpm' that finds the highest peak of the autocorrelation
/// function, calculates it's precise location and converts this reading to bpm's.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2008-02-10 18:07:12 +0200 (su, 10 helmi 2008) $
// File revision : $Revision: 4 $
//
// $Id: BPMDetect.cpp 4 2008-02-10 16:07:12Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#include <math.h>
#include <assert.h>
#include <string.h>
#include "FIFOSampleBuffer.h"
#include "PeakFinder.h"
#include "BPMDetect.h"
using namespace soundtouch;
#define INPUT_BLOCK_SAMPLES 2048
#define DECIMATED_BLOCK_SAMPLES 256
typedef unsigned short ushort;
/// decay constant for calculating RMS volume sliding average approximation
/// (time constant is about 10 sec)
const float avgdecay = 0.99986f;
/// Normalization coefficient for calculating RMS sliding average approximation.
const float avgnorm = (1 - avgdecay);
BPMDetect::BPMDetect(int numChannels, int sampleRate)
{
xcorr = NULL;
buffer = new FIFOSampleBuffer();
decimateSum = 0;
decimateCount = 0;
decimateBy = 0;
this->sampleRate = sampleRate;
this->channels = numChannels;
envelopeAccu = 0;
// Initialize RMS volume accumulator to RMS level of 3000 (out of 32768) that's
// a typical RMS signal level value for song data. This value is then adapted
// to the actual level during processing.
#ifdef INTEGER_SAMPLES
// integer samples
RMSVolumeAccu = (3000 * 3000) / avgnorm;
#else
// float samples, scaled to range [-1..+1[
RMSVolumeAccu = (0.092f * 0.092f) / avgnorm;
#endif
init(numChannels, sampleRate);
}
BPMDetect::~BPMDetect()
{
delete[] xcorr;
delete buffer;
}
/// low-pass filter & decimate to about 500 Hz. return number of outputted samples.
///
/// Decimation is used to remove the unnecessary frequencies and thus to reduce
/// the amount of data needed to be processed as calculating autocorrelation
/// function is a very-very heavy operation.
///
/// Anti-alias filtering is done simply by averaging the samples. This is really a
/// poor-man's anti-alias filtering, but it's not so critical in this kind of application
/// (it'd also be difficult to design a high-quality filter with steep cut-off at very
/// narrow band)
int BPMDetect::decimate(SAMPLETYPE *dest, const SAMPLETYPE *src, int numsamples)
{
int count, outcount;
LONG_SAMPLETYPE out;
assert(decimateBy != 0);
outcount = 0;
for (count = 0; count < numsamples; count ++)
{
decimateSum += src[count];
decimateCount ++;
if (decimateCount >= decimateBy)
{
// Store every Nth sample only
out = (LONG_SAMPLETYPE)(decimateSum / decimateBy);
decimateSum = 0;
decimateCount = 0;
#ifdef INTEGER_SAMPLES
// check ranges for sure (shouldn't actually be necessary)
if (out > 32767)
{
out = 32767;
}
else if (out < -32768)
{
out = -32768;
}
#endif // INTEGER_SAMPLES
dest[outcount] = (SAMPLETYPE)out;
outcount ++;
}
}
return outcount;
}
// Calculates autocorrelation function of the sample history buffer
void BPMDetect::updateXCorr(int process_samples)
{
int offs;
SAMPLETYPE *pBuffer;
assert(buffer->numSamples() >= (uint)(process_samples + windowLen));
pBuffer = buffer->ptrBegin();
for (offs = windowStart; offs < windowLen; offs ++)
{
LONG_SAMPLETYPE sum;
int i;
sum = 0;
for (i = 0; i < process_samples; i ++)
{
sum += pBuffer[i] * pBuffer[i + offs]; // scaling the sub-result shouldn't be necessary
}
// xcorr[offs] *= xcorr_decay; // decay 'xcorr' here with suitable coefficients
// if it's desired that the system adapts automatically to
// various bpms, e.g. in processing continouos music stream.
// The 'xcorr_decay' should be a value that's smaller than but
// close to one, and should also depend on 'process_samples' value.
xcorr[offs] += (float)sum;
}
}
// Calculates envelope of the sample data
void BPMDetect::calcEnvelope(SAMPLETYPE *samples, int numsamples)
{
const float decay = 0.7f; // decay constant for smoothing the envelope
const float norm = (1 - decay);
int i;
LONG_SAMPLETYPE out;
float val;
for (i = 0; i < numsamples; i ++)
{
// calc average RMS volume
RMSVolumeAccu *= avgdecay;
val = (float)fabs((float)samples[i]);
RMSVolumeAccu += val * val;
// cut amplitudes that are below 2 times average RMS volume
// (we're interested in peak values, not the silent moments)
val -= 2 * (float)sqrt(RMSVolumeAccu * avgnorm);
val = (val > 0) ? val : 0;
// smooth amplitude envelope
envelopeAccu *= decay;
envelopeAccu += val;
out = (LONG_SAMPLETYPE)(envelopeAccu * norm);
#ifdef INTEGER_SAMPLES
// cut peaks (shouldn't be necessary though)
if (out > 32767) out = 32767;
#endif // INTEGER_SAMPLES
samples[i] = (SAMPLETYPE)out;
}
}
void BPMDetect::inputSamples(SAMPLETYPE *samples, int numSamples)
{
SAMPLETYPE decimated[DECIMATED_BLOCK_SAMPLES];
// convert from stereo to mono if necessary
if (channels == 2)
{
int i;
for (i = 0; i < numSamples; i ++)
{
samples[i] = (samples[i * 2] + samples[i * 2 + 1]) / 2;
}
}
// decimate
numSamples = decimate(decimated, samples, numSamples);
// envelope new samples and add them to buffer
calcEnvelope(decimated, numSamples);
buffer->putSamples(decimated, numSamples);
// when the buffer has enought samples for processing...
if ((int)buffer->numSamples() > windowLen)
{
int processLength;
// how many samples are processed
processLength = buffer->numSamples() - windowLen;
// ... calculate autocorrelations for oldest samples...
updateXCorr(processLength);
// ... and remove them from the buffer
buffer->receiveSamples(processLength);
}
}
void BPMDetect::init(int numChannels, int sampleRate)
{
this->sampleRate = sampleRate;
// choose decimation factor so that result is approx. 500 Hz
decimateBy = sampleRate / 500;
assert(decimateBy > 0);
assert(INPUT_BLOCK_SAMPLES < decimateBy * DECIMATED_BLOCK_SAMPLES);
// Calculate window length & starting item according to desired min & max bpms
windowLen = (60 * sampleRate) / (decimateBy * MIN_BPM);
windowStart = (60 * sampleRate) / (decimateBy * MAX_BPM);
assert(windowLen > windowStart);
// allocate new working objects
xcorr = new float[windowLen];
memset(xcorr, 0, windowLen * sizeof(float));
// we do processing in mono mode
buffer->setChannels(1);
buffer->clear();
}
float BPMDetect::getBpm()
{
float peakPos;
PeakFinder peakFinder;
// find peak position
peakPos = peakFinder.detectPeak(xcorr, windowStart, windowLen);
assert(decimateBy != 0);
if (peakPos < 1e-6) return 0.0; // detection failed.
// calculate BPM
return 60.0f * (((float)sampleRate / (float)decimateBy) / peakPos);
}

40
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## Process this file with automake to create Makefile.in
##
## $Id: Makefile.am 4 2008-02-10 16:07:12Z oparviai $
##
## Copyright (C) 2003 - David W. Durham
##
## This file is part of SoundTouch, an audio processing library for pitch/time adjustments
##
## SoundTouch is free software; you can redistribute it and/or modify it under the
## terms of the GNU General Public License as published by the Free Software
## Foundation; either version 2 of the License, or (at your option) any later
## version.
##
## SoundTouch is distributed in the hope that it will be useful, but WITHOUT ANY
## WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
## A PARTICULAR PURPOSE. See the GNU General Public License for more details.
##
## You should have received a copy of the GNU General Public License along with
## this program; if not, write to the Free Software Foundation, Inc., 59 Temple
## Place - Suite 330, Boston, MA 02111-1307, USA
include $(top_srcdir)/config/am_include.mk
# set to something if you want other stuff to be included in the distribution tarball
EXTRA_DIST=bpm.dsp bpm.dsw
noinst_HEADERS=PeakFinder.h
lib_LTLIBRARIES=libBPM.la
libBPM_la_SOURCES=BPMDetect.cpp PeakFinder.cpp
# other compiler flags to add
AM_CXXFLAGS=-O3 -I../../../include
# other linking flags to add
#libBPM_la_LIBADD=

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////////////////////////////////////////////////////////////////////////////////
///
/// Peak detection routine.
///
/// The routine detects highest value on an array of values and calculates the
/// precise peak location as a mass-center of the 'hump' around the peak value.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2008-02-10 18:07:12 +0200 (su, 10 helmi 2008) $
// File revision : $Revision: 4 $
//
// $Id: PeakFinder.cpp 4 2008-02-10 16:07:12Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#include <math.h>
#include <assert.h>
#include "PeakFinder.h"
PeakFinder::PeakFinder()
{
}
// Finds 'ground level' of a peak hump by starting from 'peakpos' and proceeding
// to direction defined by 'direction' until next 'hump' after minimum value will
// begin
int PeakFinder::findGround(const float *data, int peakpos, int direction) const
{
float refvalue;
int lowpos;
int pos;
int climb_count;
float delta;
climb_count = 0;
refvalue = data[peakpos];
lowpos = peakpos;
pos = peakpos;
while ((pos > minPos) && (pos < maxPos))
{
int prevpos;
prevpos = pos;
pos += direction;
// calculate derivate
delta = data[pos] - data[prevpos];
if (delta <= 0)
{
// going downhill, ok
if (climb_count)
{
climb_count --; // decrease climb count
}
// check if new minimum found
if (data[pos] < refvalue)
{
// new minimum found
lowpos = pos;
refvalue = data[pos];
}
}
else
{
// going uphill, increase climbing counter
climb_count ++;
if (climb_count > 5) break; // we've been climbing too long => it's next uphill => quit
}
}
return lowpos;
}
// Find offset where the value crosses the given level, when starting from 'peakpos' and
// proceeds to direction defined in 'direction'
int PeakFinder::findCrossingLevel(const float *data, float level, int peakpos, int direction) const
{
float peaklevel;
int pos;
peaklevel = data[peakpos];
assert(peaklevel >= level);
pos = peakpos;
while ((pos >= minPos) && (pos < maxPos))
{
if (data[pos + direction] < level) return pos; // crossing found
pos += direction;
}
return -1; // not found
}
// Calculates the center of mass location of 'data' array items between 'firstPos' and 'lastPos'
float PeakFinder::calcMassCenter(const float *data, int firstPos, int lastPos) const
{
int i;
float sum;
float wsum;
sum = 0;
wsum = 0;
for (i = firstPos; i <= lastPos; i ++)
{
sum += (float)i * data[i];
wsum += data[i];
}
return sum / wsum;
}
float PeakFinder::detectPeak(const float *data, int minPos, int maxPos)
{
#define max(x, y) (((x) > (y)) ? (x) : (y))
int i;
int peakpos; // position of peak level
float peakLevel; // peak level
int crosspos1, crosspos2; // position where the peak 'hump' crosses cutting level
float cutLevel; // cutting value
float groundLevel; // ground level of the peak
int gp1, gp2; // bottom positions of the peak 'hump'
this->minPos = minPos;
this->maxPos = maxPos;
// find absolute peak
peakpos = minPos;
peakLevel = data[minPos];
for (i = minPos + 1; i < maxPos; i ++)
{
if (data[i] > peakLevel)
{
peakLevel = data[i];
peakpos = i;
}
}
// find ground positions.
gp1 = findGround(data, peakpos, -1);
gp2 = findGround(data, peakpos, 1);
groundLevel = max(data[gp1], data[gp2]);
if (groundLevel < 1e-6) return 0; // ground level too small => detection failed
if ((peakLevel / groundLevel) < 1.3) return 0; // peak less than 30% of the ground level => no good peak detected
// calculate 70%-level of the peak
cutLevel = 0.70f * peakLevel + 0.30f * groundLevel;
// find mid-level crossings
crosspos1 = findCrossingLevel(data, cutLevel, peakpos, -1);
crosspos2 = findCrossingLevel(data, cutLevel, peakpos, 1);
if ((crosspos1 < 0) || (crosspos2 < 0)) return 0; // no crossing, no peak..
// calculate mass center of the peak surroundings
return calcMassCenter(data, crosspos1, crosspos2);
}

85
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////////////////////////////////////////////////////////////////////////////////
///
/// The routine detects highest value on an array of values and calculates the
/// precise peak location as a mass-center of the 'hump' around the peak value.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2008-02-10 18:07:12 +0200 (su, 10 helmi 2008) $
// File revision : $Revision: 4 $
//
// $Id: PeakFinder.h 4 2008-02-10 16:07:12Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#ifndef _PeakFinder_H_
#define _PeakFinder_H_
class PeakFinder
{
protected:
/// Min, max allowed peak positions within the data vector
int minPos, maxPos;
/// Calculates the mass center between given vector items.
float calcMassCenter(const float *data, ///< Data vector.
int firstPos, ///< Index of first vector item beloging to the peak.
int lastPos ///< Index of last vector item beloging to the peak.
) const;
/// Finds the data vector index where the monotoniously decreasing signal crosses the
/// given level.
int findCrossingLevel(const float *data, ///< Data vector.
float level, ///< Goal crossing level.
int peakpos, ///< Peak position index within the data vector.
int direction /// Direction where to proceed from the peak: 1 = right, -1 = left.
) const;
/// Finds the 'ground' level, i.e. smallest level between two neighbouring peaks, to right-
/// or left-hand side of the given peak position.
int findGround(const float *data, /// Data vector.
int peakpos, /// Peak position index within the data vector.
int direction /// Direction where to proceed from the peak: 1 = right, -1 = left.
) const;
public:
/// Constructor.
PeakFinder();
/// Detect exact peak position of the data vector by finding the largest peak 'hump'
/// and calculating the mass-center location of the peak hump.
///
/// \return The exact mass-center location of the largest peak hump.
float detectPeak(const float *data, /// Data vector to be analyzed. The data vector has
/// to be at least 'maxPos' items long.
int minPos, ///< Min allowed peak location within the vector data.
int maxPos ///< Max allowed peak location within the vector data.
);
};
#endif // _PeakFinder_H_

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