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Update soundtouch to 1.7.1.

This commit is contained in:
lllucius 2013-10-24 06:36:42 +00:00
parent 8d97adecb1
commit f740766f37
107 changed files with 6813 additions and 60819 deletions
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9
lib-src/audacity-patches.txt
View File

@ -264,15 +264,8 @@ soundtouch
----------
http://www.surina.net/soundtouch/
Independant Pitch and Tempo changing library.
Version in Audacity SVN: 1.3.1
Version in Audacity SVN: 1.7.1
Patches:
PPC mac build changes to configure.in in several places
correct-const-usage.patch: makes usage of const keyword consistent so it
compiles with Sun's C++ compiler. Patch sent upstream 17/03/2008
fix-includes.patch: needed for GCC 4.3.1 to include all the headers we use
directly. In upstream SVN
Argument ([-Wall foreign]) added to AM_INIT_AUTOMAKE because autoreconf fails
otherwise
Upstream Version: 1.7.1
taglib

2
lib-src/soundtouch/COPYING.TXT
View File

@ -117,7 +117,7 @@ be combined with the library in order to run.
0. This License Agreement applies to any software library or other
program which contains a notice placed by the copyright holder or
other authorized party saying it may be distributed under the terms of
other authoried party saying it may be distributed under the terms of
this Lesser General Public License (also called "this License").
Each licensee is addressed as "you".

6
lib-src/soundtouch/Makefile.am
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@ -1,6 +1,6 @@
## Process this file with automake to create Makefile.in
##
## $Id: Makefile.am,v 1.5 2009-11-07 12:17:03 richardash1981 Exp $
## $Id: Makefile.am 89 2010-04-03 15:30:08Z oparviai $
##
## Copyright (C) 2003 - David W. Durham
##
@ -52,7 +52,7 @@ EXTRA_DIST= \
README.html
pkgconfigdir = $(libdir)/pkgconfig
pkgconfig_DATA = soundtouch-1.0.pc
pkgconfig_DATA = soundtouch.pc
## This removes stuff from the distribution which may be present
## from a cvs checkout or other build reasons
@ -67,4 +67,4 @@ dist-hook:
# flag to aclocal where to find m4 macros for tests
ACLOCAL_AMFLAGS = -I config/m4
AUTOMAKE_OPTIONS = foreign

722
lib-src/soundtouch/Makefile.in
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@ -1,722 +0,0 @@
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mkdir $(distdir)/_inst
chmod a-w $(distdir)
dc_install_base=`$(am__cd) $(distdir)/_inst && pwd | sed -e 's,^[^:\\/]:[\\/],/,'` \
&& dc_destdir="$${TMPDIR-/tmp}/am-dc-$$$$/" \
&& cd $(distdir)/_build \
&& ../configure --srcdir=.. --prefix="$$dc_install_base" \
$(DISTCHECK_CONFIGURE_FLAGS) \
&& $(MAKE) $(AM_MAKEFLAGS) \
&& $(MAKE) $(AM_MAKEFLAGS) dvi \
&& $(MAKE) $(AM_MAKEFLAGS) check \
&& $(MAKE) $(AM_MAKEFLAGS) install \
&& $(MAKE) $(AM_MAKEFLAGS) installcheck \
&& $(MAKE) $(AM_MAKEFLAGS) uninstall \
&& $(MAKE) $(AM_MAKEFLAGS) distuninstallcheck_dir="$$dc_install_base" \
distuninstallcheck \
&& chmod -R a-w "$$dc_install_base" \
&& ({ \
(cd ../.. && umask 077 && mkdir "$$dc_destdir") \
&& $(MAKE) $(AM_MAKEFLAGS) DESTDIR="$$dc_destdir" install \
&& $(MAKE) $(AM_MAKEFLAGS) DESTDIR="$$dc_destdir" uninstall \
&& $(MAKE) $(AM_MAKEFLAGS) DESTDIR="$$dc_destdir" \
distuninstallcheck_dir="$$dc_destdir" distuninstallcheck; \
} || { rm -rf "$$dc_destdir"; exit 1; }) \
&& rm -rf "$$dc_destdir" \
&& $(MAKE) $(AM_MAKEFLAGS) dist \
&& rm -rf $(DIST_ARCHIVES) \
&& $(MAKE) $(AM_MAKEFLAGS) distcleancheck
$(am__remove_distdir)
@(echo "$(distdir) archives ready for distribution: "; \
list='$(DIST_ARCHIVES)'; for i in $$list; do echo $$i; done) | \
sed -e '1{h;s/./=/g;p;x;}' -e '$${p;x;}'
distuninstallcheck:
@cd $(distuninstallcheck_dir) \
&& test `$(distuninstallcheck_listfiles) | wc -l` -le 1 \
|| { echo "ERROR: files left after uninstall:" ; \
if test -n "$(DESTDIR)"; then \
echo " (check DESTDIR support)"; \
fi ; \
$(distuninstallcheck_listfiles) ; \
exit 1; } >&2
distcleancheck: distclean
@if test '$(srcdir)' = . ; then \
echo "ERROR: distcleancheck can only run from a VPATH build" ; \
exit 1 ; \
fi
@test `$(distcleancheck_listfiles) | wc -l` -eq 0 \
|| { echo "ERROR: files left in build directory after distclean:" ; \
$(distcleancheck_listfiles) ; \
exit 1; } >&2
check-am: all-am
check: check-recursive
all-am: Makefile $(DATA)
installdirs: installdirs-recursive
installdirs-am:
for dir in "$(DESTDIR)$(m4datadir)" "$(DESTDIR)$(pkgconfigdir)" "$(DESTDIR)$(pkgdocdir)"; do \
test -z "$$dir" || $(mkdir_p) "$$dir"; \
done
install: install-recursive
install-exec: install-exec-recursive
install-data: install-data-recursive
uninstall: uninstall-recursive
install-am: all-am
@$(MAKE) $(AM_MAKEFLAGS) install-exec-am install-data-am
installcheck: installcheck-recursive
install-strip:
$(MAKE) $(AM_MAKEFLAGS) INSTALL_PROGRAM="$(INSTALL_STRIP_PROGRAM)" \
install_sh_PROGRAM="$(INSTALL_STRIP_PROGRAM)" INSTALL_STRIP_FLAG=-s \
`test -z '$(STRIP)' || \
echo "INSTALL_PROGRAM_ENV=STRIPPROG='$(STRIP)'"` install
mostlyclean-generic:
clean-generic:
distclean-generic:
-test -z "$(CONFIG_CLEAN_FILES)" || rm -f $(CONFIG_CLEAN_FILES)
maintainer-clean-generic:
@echo "This command is intended for maintainers to use"
@echo "it deletes files that may require special tools to rebuild."
clean: clean-recursive
clean-am: clean-generic clean-libtool mostlyclean-am
distclean: distclean-recursive
-rm -f $(am__CONFIG_DISTCLEAN_FILES)
-rm -f Makefile
distclean-am: clean-am distclean-generic distclean-libtool \
distclean-tags
dvi: dvi-recursive
dvi-am:
html: html-recursive
info: info-recursive
info-am:
install-data-am: install-m4dataDATA install-pkgconfigDATA \
install-pkgdocDATA
install-exec-am:
install-info: install-info-recursive
install-man:
installcheck-am:
maintainer-clean: maintainer-clean-recursive
-rm -f $(am__CONFIG_DISTCLEAN_FILES)
-rm -rf $(top_srcdir)/autom4te.cache
-rm -f Makefile
maintainer-clean-am: distclean-am maintainer-clean-generic
mostlyclean: mostlyclean-recursive
mostlyclean-am: mostlyclean-generic mostlyclean-libtool
pdf: pdf-recursive
pdf-am:
ps: ps-recursive
ps-am:
uninstall-am: uninstall-info-am uninstall-m4dataDATA \
uninstall-pkgconfigDATA uninstall-pkgdocDATA
uninstall-info: uninstall-info-recursive
.PHONY: $(RECURSIVE_TARGETS) CTAGS GTAGS all all-am am--refresh check \
check-am clean clean-generic clean-libtool clean-recursive \
ctags ctags-recursive dist dist-all dist-bzip2 dist-gzip \
dist-hook dist-shar dist-tarZ dist-zip distcheck distclean \
distclean-generic distclean-libtool distclean-recursive \
distclean-tags distcleancheck distdir distuninstallcheck dvi \
dvi-am html html-am info info-am install install-am \
install-data install-data-am install-exec install-exec-am \
install-info install-info-am install-m4dataDATA install-man \
install-pkgconfigDATA install-pkgdocDATA install-strip \
installcheck installcheck-am installdirs installdirs-am \
maintainer-clean maintainer-clean-generic \
maintainer-clean-recursive mostlyclean mostlyclean-generic \
mostlyclean-libtool mostlyclean-recursive pdf pdf-am ps ps-am \
tags tags-recursive uninstall uninstall-am uninstall-info-am \
uninstall-m4dataDATA uninstall-pkgconfigDATA \
uninstall-pkgdocDATA
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
# Tell versions [3.59,3.63) of GNU make to not export all variables.
# Otherwise a system limit (for SysV at least) may be exceeded.
.NOEXPORT:

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lib-src/soundtouch/README.html
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21
lib-src/soundtouch/bin/run_test Executable file
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@ -0,0 +1,21 @@
SOUND_DIR=~/dev/test_sounds
OUT_DIR=./test_out
TEST_NAME=semmari
OUT_NAME=out
SS=./soundstretch
TEST_PARAM="-pitch=-3 -bpm"
mkdir $OUT_DIR
$SS $SOUND_DIR/$TEST_NAME-8b1.wav $OUT_DIR/$OUT_NAME-8b1.wav $TEST_PARAM
$SS $SOUND_DIR/$TEST_NAME-8b2.wav $OUT_DIR/$OUT_NAME-8b2.wav $TEST_PARAM
$SS $SOUND_DIR/$TEST_NAME-16b1.wav $OUT_DIR/$OUT_NAME-16b1.wav $TEST_PARAM
$SS $SOUND_DIR/$TEST_NAME-16b2.wav $OUT_DIR/$OUT_NAME-16b2.wav $TEST_PARAM
$SS $SOUND_DIR/$TEST_NAME-24b1.wav $OUT_DIR/$OUT_NAME-24b1.wav $TEST_PARAM
$SS $SOUND_DIR/$TEST_NAME-24b2.wav $OUT_DIR/$OUT_NAME-24b2.wav $TEST_PARAM
$SS $SOUND_DIR/$TEST_NAME-32b1.wav $OUT_DIR/$OUT_NAME-32b1.wav $TEST_PARAM
$SS $SOUND_DIR/$TEST_NAME-32b2.wav $OUT_DIR/$OUT_NAME-32b2.wav $TEST_PARAM

19
lib-src/soundtouch/bin/run_test.cmd Normal file
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@ -0,0 +1,19 @@
set SOUND_DIR=d:\dev\test_sounds
set OUT_DIR=.
set TEST_NAME=semmari
set OUT_NAME=out
set SS=soundstretch
set TEST_PARAM=-pitch=-3 -bpm
call %SS% %SOUND_DIR%\%TEST_NAME%-8b1.wav %OUT_DIR%\%OUT_NAME%-8b1.wav %TEST_PARAM%
call %SS% %SOUND_DIR%\%TEST_NAME%-8b2.wav %OUT_DIR%\%OUT_NAME%-8b2.wav %TEST_PARAM%
call %SS% %SOUND_DIR%\%TEST_NAME%-16b1.wav %OUT_DIR%\%OUT_NAME%-16b1.wav %TEST_PARAM%
call %SS% %SOUND_DIR%\%TEST_NAME%-16b2.wav %OUT_DIR%\%OUT_NAME%-16b2.wav %TEST_PARAM%
call %SS% %SOUND_DIR%\%TEST_NAME%-24b1.wav %OUT_DIR%\%OUT_NAME%-24b1.wav %TEST_PARAM%
call %SS% %SOUND_DIR%\%TEST_NAME%-24b2.wav %OUT_DIR%\%OUT_NAME%-24b2.wav %TEST_PARAM%
call %SS% %SOUND_DIR%\%TEST_NAME%-32b1.wav %OUT_DIR%\%OUT_NAME%-32b1.wav %TEST_PARAM%
call %SS% %SOUND_DIR%\%TEST_NAME%-32b2.wav %OUT_DIR%\%OUT_NAME%-32b2.wav %TEST_PARAM%

6
lib-src/soundtouch/bootstrap
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@ -1,4 +1,4 @@
# $Id: bootstrap,v 1.2 2006-09-18 07:31:35 richardash1981 Exp $
# $Id: bootstrap 43 2008-12-25 17:54:41Z oparviai $
#!/bin/sh
@ -16,11 +16,11 @@ then
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
#gettextie 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"
export AUTOMAKE="automake --add-missing --foreign --copy"
autoreconf -fisv && rm -f `find . -name "*~"` && rm -f ChangeLog
exit $?
fi

19
lib-src/soundtouch/config/am_include.mk
View File

@ -1,10 +1,8 @@
## vim:tw=78
## Process this file with automake to create Makefile.in
##
## $Id: am_include.mk,v 1.3 2006-09-27 06:19:52 llucius Exp $
## $Id: am_include.mk 130 2011-07-17 11:46:22Z 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
@ -32,20 +30,5 @@
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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1703
lib-src/soundtouch/config/config.sub vendored
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530
lib-src/soundtouch/config/depcomp
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@ -1,530 +0,0 @@
#! /bin/sh
# depcomp - compile a program generating dependencies as side-effects
scriptversion=2005-07-09.11
# Copyright (C) 1999, 2000, 2003, 2004, 2005 Free Software Foundation, Inc.
# This program 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, or (at your option)
# any later version.
# This program 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., 51 Franklin Street, Fifth Floor, Boston, MA
# 02110-1301, USA.
# As a special exception to the GNU General Public License, if you
# distribute this file as part of a program that contains a
# configuration script generated by Autoconf, you may include it under
# the same distribution terms that you use for the rest of that program.
# Originally written by Alexandre Oliva <oliva@dcc.unicamp.br>.
case $1 in
'')
echo "$0: No command. Try \`$0 --help' for more information." 1>&2
exit 1;
;;
-h | --h*)
cat <<\EOF
Usage: depcomp [--help] [--version] PROGRAM [ARGS]
Run PROGRAMS ARGS to compile a file, generating dependencies
as side-effects.
Environment variables:
depmode Dependency tracking mode.
source Source file read by `PROGRAMS ARGS'.
object Object file output by `PROGRAMS ARGS'.
DEPDIR directory where to store dependencies.
depfile Dependency file to output.
tmpdepfile Temporary file to use when outputing dependencies.
libtool Whether libtool is used (yes/no).
Report bugs to <bug-automake@gnu.org>.
EOF
exit $?
;;
-v | --v*)
echo "depcomp $scriptversion"
exit $?
;;
esac
if test -z "$depmode" || test -z "$source" || test -z "$object"; then
echo "depcomp: Variables source, object and depmode must be set" 1>&2
exit 1
fi
# Dependencies for sub/bar.o or sub/bar.obj go into sub/.deps/bar.Po.
depfile=${depfile-`echo "$object" |
sed 's|[^\\/]*$|'${DEPDIR-.deps}'/&|;s|\.\([^.]*\)$|.P\1|;s|Pobj$|Po|'`}
tmpdepfile=${tmpdepfile-`echo "$depfile" | sed 's/\.\([^.]*\)$/.T\1/'`}
rm -f "$tmpdepfile"
# Some modes work just like other modes, but use different flags. We
# parameterize here, but still list the modes in the big case below,
# to make depend.m4 easier to write. Note that we *cannot* use a case
# here, because this file can only contain one case statement.
if test "$depmode" = hp; then
# HP compiler uses -M and no extra arg.
gccflag=-M
depmode=gcc
fi
if test "$depmode" = dashXmstdout; then
# This is just like dashmstdout with a different argument.
dashmflag=-xM
depmode=dashmstdout
fi
case "$depmode" in
gcc3)
## gcc 3 implements dependency tracking that does exactly what
## we want. Yay! Note: for some reason libtool 1.4 doesn't like
## it if -MD -MP comes after the -MF stuff. Hmm.
"$@" -MT "$object" -MD -MP -MF "$tmpdepfile"
stat=$?
if test $stat -eq 0; then :
else
rm -f "$tmpdepfile"
exit $stat
fi
mv "$tmpdepfile" "$depfile"
;;
gcc)
## There are various ways to get dependency output from gcc. Here's
## why we pick this rather obscure method:
## - Don't want to use -MD because we'd like the dependencies to end
## up in a subdir. Having to rename by hand is ugly.
## (We might end up doing this anyway to support other compilers.)
## - The DEPENDENCIES_OUTPUT environment variable makes gcc act like
## -MM, not -M (despite what the docs say).
## - Using -M directly means running the compiler twice (even worse
## than renaming).
if test -z "$gccflag"; then
gccflag=-MD,
fi
"$@" -Wp,"$gccflag$tmpdepfile"
stat=$?
if test $stat -eq 0; then :
else
rm -f "$tmpdepfile"
exit $stat
fi
rm -f "$depfile"
echo "$object : \\" > "$depfile"
alpha=ABCDEFGHIJKLMNOPQRSTUVWXYZabcdefghijklmnopqrstuvwxyz
## The second -e expression handles DOS-style file names with drive letters.
sed -e 's/^[^:]*: / /' \
-e 's/^['$alpha']:\/[^:]*: / /' < "$tmpdepfile" >> "$depfile"
## This next piece of magic avoids the `deleted header file' problem.
## The problem is that when a header file which appears in a .P file
## is deleted, the dependency causes make to die (because there is
## typically no way to rebuild the header). We avoid this by adding
## dummy dependencies for each header file. Too bad gcc doesn't do
## this for us directly.
tr ' ' '
' < "$tmpdepfile" |
## Some versions of gcc put a space before the `:'. On the theory
## that the space means something, we add a space to the output as
## well.
## Some versions of the HPUX 10.20 sed can't process this invocation
## correctly. Breaking it into two sed invocations is a workaround.
sed -e 's/^\\$//' -e '/^$/d' -e '/:$/d' | sed -e 's/$/ :/' >> "$depfile"
rm -f "$tmpdepfile"
;;
hp)
# This case exists only to let depend.m4 do its work. It works by
# looking at the text of this script. This case will never be run,
# since it is checked for above.
exit 1
;;
sgi)
if test "$libtool" = yes; then
"$@" "-Wp,-MDupdate,$tmpdepfile"
else
"$@" -MDupdate "$tmpdepfile"
fi
stat=$?
if test $stat -eq 0; then :
else
rm -f "$tmpdepfile"
exit $stat
fi
rm -f "$depfile"
if test -f "$tmpdepfile"; then # yes, the sourcefile depend on other files
echo "$object : \\" > "$depfile"
# Clip off the initial element (the dependent). Don't try to be
# clever and replace this with sed code, as IRIX sed won't handle
# lines with more than a fixed number of characters (4096 in
# IRIX 6.2 sed, 8192 in IRIX 6.5). We also remove comment lines;
# the IRIX cc adds comments like `#:fec' to the end of the
# dependency line.
tr ' ' '
' < "$tmpdepfile" \
| sed -e 's/^.*\.o://' -e 's/#.*$//' -e '/^$/ d' | \
tr '
' ' ' >> $depfile
echo >> $depfile
# The second pass generates a dummy entry for each header file.
tr ' ' '
' < "$tmpdepfile" \
| sed -e 's/^.*\.o://' -e 's/#.*$//' -e '/^$/ d' -e 's/$/:/' \
>> $depfile
else
# The sourcefile does not contain any dependencies, so just
# store a dummy comment line, to avoid errors with the Makefile
# "include basename.Plo" scheme.
echo "#dummy" > "$depfile"
fi
rm -f "$tmpdepfile"
;;
aix)
# The C for AIX Compiler uses -M and outputs the dependencies
# in a .u file. In older versions, this file always lives in the
# current directory. Also, the AIX compiler puts `$object:' at the
# start of each line; $object doesn't have directory information.
# Version 6 uses the directory in both cases.
stripped=`echo "$object" | sed 's/\(.*\)\..*$/\1/'`
tmpdepfile="$stripped.u"
if test "$libtool" = yes; then
"$@" -Wc,-M
else
"$@" -M
fi
stat=$?
if test -f "$tmpdepfile"; then :
else
stripped=`echo "$stripped" | sed 's,^.*/,,'`
tmpdepfile="$stripped.u"
fi
if test $stat -eq 0; then :
else
rm -f "$tmpdepfile"
exit $stat
fi
if test -f "$tmpdepfile"; then
outname="$stripped.o"
# Each line is of the form `foo.o: dependent.h'.
# Do two passes, one to just change these to
# `$object: dependent.h' and one to simply `dependent.h:'.
sed -e "s,^$outname:,$object :," < "$tmpdepfile" > "$depfile"
sed -e "s,^$outname: \(.*\)$,\1:," < "$tmpdepfile" >> "$depfile"
else
# The sourcefile does not contain any dependencies, so just
# store a dummy comment line, to avoid errors with the Makefile
# "include basename.Plo" scheme.
echo "#dummy" > "$depfile"
fi
rm -f "$tmpdepfile"
;;
icc)
# Intel's C compiler understands `-MD -MF file'. However on
# icc -MD -MF foo.d -c -o sub/foo.o sub/foo.c
# ICC 7.0 will fill foo.d with something like
# foo.o: sub/foo.c
# foo.o: sub/foo.h
# which is wrong. We want:
# sub/foo.o: sub/foo.c
# sub/foo.o: sub/foo.h
# sub/foo.c:
# sub/foo.h:
# ICC 7.1 will output
# foo.o: sub/foo.c sub/foo.h
# and will wrap long lines using \ :
# foo.o: sub/foo.c ... \
# sub/foo.h ... \
# ...
"$@" -MD -MF "$tmpdepfile"
stat=$?
if test $stat -eq 0; then :
else
rm -f "$tmpdepfile"
exit $stat
fi
rm -f "$depfile"
# Each line is of the form `foo.o: dependent.h',
# or `foo.o: dep1.h dep2.h \', or ` dep3.h dep4.h \'.
# Do two passes, one to just change these to
# `$object: dependent.h' and one to simply `dependent.h:'.
sed "s,^[^:]*:,$object :," < "$tmpdepfile" > "$depfile"
# Some versions of the HPUX 10.20 sed can't process this invocation
# correctly. Breaking it into two sed invocations is a workaround.
sed 's,^[^:]*: \(.*\)$,\1,;s/^\\$//;/^$/d;/:$/d' < "$tmpdepfile" |
sed -e 's/$/ :/' >> "$depfile"
rm -f "$tmpdepfile"
;;
tru64)
# The Tru64 compiler uses -MD to generate dependencies as a side
# effect. `cc -MD -o foo.o ...' puts the dependencies into `foo.o.d'.
# At least on Alpha/Redhat 6.1, Compaq CCC V6.2-504 seems to put
# dependencies in `foo.d' instead, so we check for that too.
# Subdirectories are respected.
dir=`echo "$object" | sed -e 's|/[^/]*$|/|'`
test "x$dir" = "x$object" && dir=
base=`echo "$object" | sed -e 's|^.*/||' -e 's/\.o$//' -e 's/\.lo$//'`
if test "$libtool" = yes; then
# With Tru64 cc, shared objects can also be used to make a
# static library. This mecanism is used in libtool 1.4 series to
# handle both shared and static libraries in a single compilation.
# With libtool 1.4, dependencies were output in $dir.libs/$base.lo.d.
#
# With libtool 1.5 this exception was removed, and libtool now
# generates 2 separate objects for the 2 libraries. These two
# compilations output dependencies in in $dir.libs/$base.o.d and
# in $dir$base.o.d. We have to check for both files, because
# one of the two compilations can be disabled. We should prefer
# $dir$base.o.d over $dir.libs/$base.o.d because the latter is
# automatically cleaned when .libs/ is deleted, while ignoring
# the former would cause a distcleancheck panic.
tmpdepfile1=$dir.libs/$base.lo.d # libtool 1.4
tmpdepfile2=$dir$base.o.d # libtool 1.5
tmpdepfile3=$dir.libs/$base.o.d # libtool 1.5
tmpdepfile4=$dir.libs/$base.d # Compaq CCC V6.2-504
"$@" -Wc,-MD
else
tmpdepfile1=$dir$base.o.d
tmpdepfile2=$dir$base.d
tmpdepfile3=$dir$base.d
tmpdepfile4=$dir$base.d
"$@" -MD
fi
stat=$?
if test $stat -eq 0; then :
else
rm -f "$tmpdepfile1" "$tmpdepfile2" "$tmpdepfile3" "$tmpdepfile4"
exit $stat
fi
for tmpdepfile in "$tmpdepfile1" "$tmpdepfile2" "$tmpdepfile3" "$tmpdepfile4"
do
test -f "$tmpdepfile" && break
done
if test -f "$tmpdepfile"; then
sed -e "s,^.*\.[a-z]*:,$object:," < "$tmpdepfile" > "$depfile"
# That's a tab and a space in the [].
sed -e 's,^.*\.[a-z]*:[ ]*,,' -e 's,$,:,' < "$tmpdepfile" >> "$depfile"
else
echo "#dummy" > "$depfile"
fi
rm -f "$tmpdepfile"
;;
#nosideeffect)
# This comment above is used by automake to tell side-effect
# dependency tracking mechanisms from slower ones.
dashmstdout)
# Important note: in order to support this mode, a compiler *must*
# always write the preprocessed file to stdout, regardless of -o.
"$@" || exit $?
# Remove the call to Libtool.
if test "$libtool" = yes; then
while test $1 != '--mode=compile'; do
shift
done
shift
fi
# Remove `-o $object'.
IFS=" "
for arg
do
case $arg in
-o)
shift
;;
$object)
shift
;;
*)
set fnord "$@" "$arg"
shift # fnord
shift # $arg
;;
esac
done
test -z "$dashmflag" && dashmflag=-M
# Require at least two characters before searching for `:'
# in the target name. This is to cope with DOS-style filenames:
# a dependency such as `c:/foo/bar' could be seen as target `c' otherwise.
"$@" $dashmflag |
sed 's:^[ ]*[^: ][^:][^:]*\:[ ]*:'"$object"'\: :' > "$tmpdepfile"
rm -f "$depfile"
cat < "$tmpdepfile" > "$depfile"
tr ' ' '
' < "$tmpdepfile" | \
## Some versions of the HPUX 10.20 sed can't process this invocation
## correctly. Breaking it into two sed invocations is a workaround.
sed -e 's/^\\$//' -e '/^$/d' -e '/:$/d' | sed -e 's/$/ :/' >> "$depfile"
rm -f "$tmpdepfile"
;;
dashXmstdout)
# This case only exists to satisfy depend.m4. It is never actually
# run, as this mode is specially recognized in the preamble.
exit 1
;;
makedepend)
"$@" || exit $?
# Remove any Libtool call
if test "$libtool" = yes; then
while test $1 != '--mode=compile'; do
shift
done
shift
fi
# X makedepend
shift
cleared=no
for arg in "$@"; do
case $cleared in
no)
set ""; shift
cleared=yes ;;
esac
case "$arg" in
-D*|-I*)
set fnord "$@" "$arg"; shift ;;
# Strip any option that makedepend may not understand. Remove
# the object too, otherwise makedepend will parse it as a source file.
-*|$object)
;;
*)
set fnord "$@" "$arg"; shift ;;
esac
done
obj_suffix="`echo $object | sed 's/^.*\././'`"
touch "$tmpdepfile"
${MAKEDEPEND-makedepend} -o"$obj_suffix" -f"$tmpdepfile" "$@"
rm -f "$depfile"
cat < "$tmpdepfile" > "$depfile"
sed '1,2d' "$tmpdepfile" | tr ' ' '
' | \
## Some versions of the HPUX 10.20 sed can't process this invocation
## correctly. Breaking it into two sed invocations is a workaround.
sed -e 's/^\\$//' -e '/^$/d' -e '/:$/d' | sed -e 's/$/ :/' >> "$depfile"
rm -f "$tmpdepfile" "$tmpdepfile".bak
;;
cpp)
# Important note: in order to support this mode, a compiler *must*
# always write the preprocessed file to stdout.
"$@" || exit $?
# Remove the call to Libtool.
if test "$libtool" = yes; then
while test $1 != '--mode=compile'; do
shift
done
shift
fi
# Remove `-o $object'.
IFS=" "
for arg
do
case $arg in
-o)
shift
;;
$object)
shift
;;
*)
set fnord "$@" "$arg"
shift # fnord
shift # $arg
;;
esac
done
"$@" -E |
sed -n -e '/^# [0-9][0-9]* "\([^"]*\)".*/ s:: \1 \\:p' \
-e '/^#line [0-9][0-9]* "\([^"]*\)".*/ s:: \1 \\:p' |
sed '$ s: \\$::' > "$tmpdepfile"
rm -f "$depfile"
echo "$object : \\" > "$depfile"
cat < "$tmpdepfile" >> "$depfile"
sed < "$tmpdepfile" '/^$/d;s/^ //;s/ \\$//;s/$/ :/' >> "$depfile"
rm -f "$tmpdepfile"
;;
msvisualcpp)
# Important note: in order to support this mode, a compiler *must*
# always write the preprocessed file to stdout, regardless of -o,
# because we must use -o when running libtool.
"$@" || exit $?
IFS=" "
for arg
do
case "$arg" in
"-Gm"|"/Gm"|"-Gi"|"/Gi"|"-ZI"|"/ZI")
set fnord "$@"
shift
shift
;;
*)
set fnord "$@" "$arg"
shift
shift
;;
esac
done
"$@" -E |
sed -n '/^#line [0-9][0-9]* "\([^"]*\)"/ s::echo "`cygpath -u \\"\1\\"`":p' | sort | uniq > "$tmpdepfile"
rm -f "$depfile"
echo "$object : \\" > "$depfile"
. "$tmpdepfile" | sed 's% %\\ %g' | sed -n '/^\(.*\)$/ s:: \1 \\:p' >> "$depfile"
echo " " >> "$depfile"
. "$tmpdepfile" | sed 's% %\\ %g' | sed -n '/^\(.*\)$/ s::\1\::p' >> "$depfile"
rm -f "$tmpdepfile"
;;
none)
exec "$@"
;;
*)
echo "Unknown depmode $depmode" 1>&2
exit 1
;;
esac
exit 0
# Local Variables:
# mode: shell-script
# sh-indentation: 2
# eval: (add-hook 'write-file-hooks 'time-stamp)
# time-stamp-start: "scriptversion="
# time-stamp-format: "%:y-%02m-%02d.%02H"
# time-stamp-end: "$"
# End:

519
lib-src/soundtouch/config/install-sh
View File

@ -1,519 +0,0 @@
#!/bin/sh
# install - install a program, script, or datafile
scriptversion=2006-12-25.00
# This originates from X11R5 (mit/util/scripts/install.sh), which was
# later released in X11R6 (xc/config/util/install.sh) with the
# following copyright and license.
#
# Copyright (C) 1994 X Consortium
#
# Permission is hereby granted, free of charge, to any person obtaining a copy
# of this software and associated documentation files (the "Software"), to
# deal in the Software without restriction, including without limitation the
# rights to use, copy, modify, merge, publish, distribute, sublicense, and/or
# sell copies of the Software, and to permit persons to whom the Software is
# furnished to do so, subject to the following conditions:
#
# The above copyright notice and this permission notice shall be included in
# all copies or substantial portions of the Software.
#
# THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
# IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
# FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
# X CONSORTIUM BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN
# AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNEC-
# TION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
#
# Except as contained in this notice, the name of the X Consortium shall not
# be used in advertising or otherwise to promote the sale, use or other deal-
# ings in this Software without prior written authorization from the X Consor-
# tium.
#
#
# FSF changes to this file are in the public domain.
#
# Calling this script install-sh is preferred over install.sh, to prevent
# `make' implicit rules from creating a file called install from it
# when there is no Makefile.
#
# This script is compatible with the BSD install script, but was written
# from scratch.
nl='
'
IFS=" "" $nl"
# set DOITPROG to echo to test this script
# Don't use :- since 4.3BSD and earlier shells don't like it.
doit=${DOITPROG-}
if test -z "$doit"; then
doit_exec=exec
else
doit_exec=$doit
fi
# Put in absolute file names if you don't have them in your path;
# or use environment vars.
chgrpprog=${CHGRPPROG-chgrp}
chmodprog=${CHMODPROG-chmod}
chownprog=${CHOWNPROG-chown}
cmpprog=${CMPPROG-cmp}
cpprog=${CPPROG-cp}
mkdirprog=${MKDIRPROG-mkdir}
mvprog=${MVPROG-mv}
rmprog=${RMPROG-rm}
stripprog=${STRIPPROG-strip}
posix_glob='?'
initialize_posix_glob='
test "$posix_glob" != "?" || {
if (set -f) 2>/dev/null; then
posix_glob=
else
posix_glob=:
fi
}
'
posix_mkdir=
# Desired mode of installed file.
mode=0755
chgrpcmd=
chmodcmd=$chmodprog
chowncmd=
mvcmd=$mvprog
rmcmd="$rmprog -f"
stripcmd=
src=
dst=
dir_arg=
dst_arg=
copy_on_change=false
no_target_directory=
usage="\
Usage: $0 [OPTION]... [-T] SRCFILE DSTFILE
or: $0 [OPTION]... SRCFILES... DIRECTORY
or: $0 [OPTION]... -t DIRECTORY SRCFILES...
or: $0 [OPTION]... -d DIRECTORIES...
In the 1st form, copy SRCFILE to DSTFILE.
In the 2nd and 3rd, copy all SRCFILES to DIRECTORY.
In the 4th, create DIRECTORIES.
Options:
--help display this help and exit.
--version display version info and exit.
-c (ignored)
-C install only if different (preserve the last data modification time)
-d create directories instead of installing files.
-g GROUP $chgrpprog installed files to GROUP.
-m MODE $chmodprog installed files to MODE.
-o USER $chownprog installed files to USER.
-s $stripprog installed files.
-t DIRECTORY install into DIRECTORY.
-T report an error if DSTFILE is a directory.
Environment variables override the default commands:
CHGRPPROG CHMODPROG CHOWNPROG CMPPROG CPPROG MKDIRPROG MVPROG
RMPROG STRIPPROG
"
while test $# -ne 0; do
case $1 in
-c) ;;
-C) copy_on_change=true;;
-d) dir_arg=true;;
-g) chgrpcmd="$chgrpprog $2"
shift;;
--help) echo "$usage"; exit $?;;
-m) mode=$2
case $mode in
*' '* | *' '* | *'
'* | *'*'* | *'?'* | *'['*)
echo "$0: invalid mode: $mode" >&2
exit 1;;
esac
shift;;
-o) chowncmd="$chownprog $2"
shift;;
-s) stripcmd=$stripprog;;
-t) dst_arg=$2
shift;;
-T) no_target_directory=true;;
--version) echo "$0 $scriptversion"; exit $?;;
--) shift
break;;
-*) echo "$0: invalid option: $1" >&2
exit 1;;
*) break;;
esac
shift
done
if test $# -ne 0 && test -z "$dir_arg$dst_arg"; then
# When -d is used, all remaining arguments are directories to create.
# When -t is used, the destination is already specified.
# Otherwise, the last argument is the destination. Remove it from $@.
for arg
do
if test -n "$dst_arg"; then
# $@ is not empty: it contains at least $arg.
set fnord "$@" "$dst_arg"
shift # fnord
fi
shift # arg
dst_arg=$arg
done
fi
if test $# -eq 0; then
if test -z "$dir_arg"; then
echo "$0: no input file specified." >&2
exit 1
fi
# It's OK to call `install-sh -d' without argument.
# This can happen when creating conditional directories.
exit 0
fi
if test -z "$dir_arg"; then
trap '(exit $?); exit' 1 2 13 15
# Set umask so as not to create temps with too-generous modes.
# However, 'strip' requires both read and write access to temps.
case $mode in
# Optimize common cases.
*644) cp_umask=133;;
*755) cp_umask=22;;
*[0-7])
if test -z "$stripcmd"; then
u_plus_rw=
else
u_plus_rw='% 200'
fi
cp_umask=`expr '(' 777 - $mode % 1000 ')' $u_plus_rw`;;
*)
if test -z "$stripcmd"; then
u_plus_rw=
else
u_plus_rw=,u+rw
fi
cp_umask=$mode$u_plus_rw;;
esac
fi
for src
do
# Protect names starting with `-'.
case $src in
-*) src=./$src;;
esac
if test -n "$dir_arg"; then
dst=$src
dstdir=$dst
test -d "$dstdir"
dstdir_status=$?
else
# Waiting for this to be detected by the "$cpprog $src $dsttmp" command
# might cause directories to be created, which would be especially bad
# if $src (and thus $dsttmp) contains '*'.
if test ! -f "$src" && test ! -d "$src"; then
echo "$0: $src does not exist." >&2
exit 1
fi
if test -z "$dst_arg"; then
echo "$0: no destination specified." >&2
exit 1
fi
dst=$dst_arg
# Protect names starting with `-'.
case $dst in
-*) dst=./$dst;;
esac
# If destination is a directory, append the input filename; won't work
# if double slashes aren't ignored.
if test -d "$dst"; then
if test -n "$no_target_directory"; then
echo "$0: $dst_arg: Is a directory" >&2
exit 1
fi
dstdir=$dst
dst=$dstdir/`basename "$src"`
dstdir_status=0
else
# Prefer dirname, but fall back on a substitute if dirname fails.
dstdir=`
(dirname "$dst") 2>/dev/null ||
expr X"$dst" : 'X\(.*[^/]\)//*[^/][^/]*/*$' \| \
X"$dst" : 'X\(//\)[^/]' \| \
X"$dst" : 'X\(//\)$' \| \
X"$dst" : 'X\(/\)' \| . 2>/dev/null ||
echo X"$dst" |
sed '/^X\(.*[^/]\)\/\/*[^/][^/]*\/*$/{
s//\1/
q
}
/^X\(\/\/\)[^/].*/{
s//\1/
q
}
/^X\(\/\/\)$/{
s//\1/
q
}
/^X\(\/\).*/{
s//\1/
q
}
s/.*/./; q'
`
test -d "$dstdir"
dstdir_status=$?
fi
fi
obsolete_mkdir_used=false
if test $dstdir_status != 0; then
case $posix_mkdir in
'')
# Create intermediate dirs using mode 755 as modified by the umask.
# This is like FreeBSD 'install' as of 1997-10-28.
umask=`umask`
case $stripcmd.$umask in
# Optimize common cases.
*[2367][2367]) mkdir_umask=$umask;;
.*0[02][02] | .[02][02] | .[02]) mkdir_umask=22;;
*[0-7])
mkdir_umask=`expr $umask + 22 \
- $umask % 100 % 40 + $umask % 20 \
- $umask % 10 % 4 + $umask % 2
`;;
*) mkdir_umask=$umask,go-w;;
esac
# With -d, create the new directory with the user-specified mode.
# Otherwise, rely on $mkdir_umask.
if test -n "$dir_arg"; then
mkdir_mode=-m$mode
else
mkdir_mode=
fi
posix_mkdir=false
case $umask in
*[123567][0-7][0-7])
# POSIX mkdir -p sets u+wx bits regardless of umask, which
# is incompatible with FreeBSD 'install' when (umask & 300) != 0.
;;
*)
tmpdir=${TMPDIR-/tmp}/ins$RANDOM-$$
trap 'ret=$?; rmdir "$tmpdir/d" "$tmpdir" 2>/dev/null; exit $ret' 0
if (umask $mkdir_umask &&
exec $mkdirprog $mkdir_mode -p -- "$tmpdir/d") >/dev/null 2>&1
then
if test -z "$dir_arg" || {
# Check for POSIX incompatibilities with -m.
# HP-UX 11.23 and IRIX 6.5 mkdir -m -p sets group- or
# other-writeable bit of parent directory when it shouldn't.
# FreeBSD 6.1 mkdir -m -p sets mode of existing directory.
ls_ld_tmpdir=`ls -ld "$tmpdir"`
case $ls_ld_tmpdir in
d????-?r-*) different_mode=700;;
d????-?--*) different_mode=755;;
*) false;;
esac &&
$mkdirprog -m$different_mode -p -- "$tmpdir" && {
ls_ld_tmpdir_1=`ls -ld "$tmpdir"`
test "$ls_ld_tmpdir" = "$ls_ld_tmpdir_1"
}
}
then posix_mkdir=:
fi
rmdir "$tmpdir/d" "$tmpdir"
else
# Remove any dirs left behind by ancient mkdir implementations.
rmdir ./$mkdir_mode ./-p ./-- 2>/dev/null
fi
trap '' 0;;
esac;;
esac
if
$posix_mkdir && (
umask $mkdir_umask &&
$doit_exec $mkdirprog $mkdir_mode -p -- "$dstdir"
)
then :
else
# The umask is ridiculous, or mkdir does not conform to POSIX,
# or it failed possibly due to a race condition. Create the
# directory the slow way, step by step, checking for races as we go.
case $dstdir in
/*) prefix='/';;
-*) prefix='./';;
*) prefix='';;
esac
eval "$initialize_posix_glob"
oIFS=$IFS
IFS=/
$posix_glob set -f
set fnord $dstdir
shift
$posix_glob set +f
IFS=$oIFS
prefixes=
for d
do
test -z "$d" && continue
prefix=$prefix$d
if test -d "$prefix"; then
prefixes=
else
if $posix_mkdir; then
(umask=$mkdir_umask &&
$doit_exec $mkdirprog $mkdir_mode -p -- "$dstdir") && break
# Don't fail if two instances are running concurrently.
test -d "$prefix" || exit 1
else
case $prefix in
*\'*) qprefix=`echo "$prefix" | sed "s/'/'\\\\\\\\''/g"`;;
*) qprefix=$prefix;;
esac
prefixes="$prefixes '$qprefix'"
fi
fi
prefix=$prefix/
done
if test -n "$prefixes"; then
# Don't fail if two instances are running concurrently.
(umask $mkdir_umask &&
eval "\$doit_exec \$mkdirprog $prefixes") ||
test -d "$dstdir" || exit 1
obsolete_mkdir_used=true
fi
fi
fi
if test -n "$dir_arg"; then
{ test -z "$chowncmd" || $doit $chowncmd "$dst"; } &&
{ test -z "$chgrpcmd" || $doit $chgrpcmd "$dst"; } &&
{ test "$obsolete_mkdir_used$chowncmd$chgrpcmd" = false ||
test -z "$chmodcmd" || $doit $chmodcmd $mode "$dst"; } || exit 1
else
# Make a couple of temp file names in the proper directory.
dsttmp=$dstdir/_inst.$$_
rmtmp=$dstdir/_rm.$$_
# Trap to clean up those temp files at exit.
trap 'ret=$?; rm -f "$dsttmp" "$rmtmp" && exit $ret' 0
# Copy the file name to the temp name.
(umask $cp_umask && $doit_exec $cpprog "$src" "$dsttmp") &&
# and set any options; do chmod last to preserve setuid bits.
#
# If any of these fail, we abort the whole thing. If we want to
# ignore errors from any of these, just make sure not to ignore
# errors from the above "$doit $cpprog $src $dsttmp" command.
#
{ test -z "$chowncmd" || $doit $chowncmd "$dsttmp"; } &&
{ test -z "$chgrpcmd" || $doit $chgrpcmd "$dsttmp"; } &&
{ test -z "$stripcmd" || $doit $stripcmd "$dsttmp"; } &&
{ test -z "$chmodcmd" || $doit $chmodcmd $mode "$dsttmp"; } &&
# If -C, don't bother to copy if it wouldn't change the file.
if $copy_on_change &&
old=`LC_ALL=C ls -dlL "$dst" 2>/dev/null` &&
new=`LC_ALL=C ls -dlL "$dsttmp" 2>/dev/null` &&
eval "$initialize_posix_glob" &&
$posix_glob set -f &&
set X $old && old=:$2:$4:$5:$6 &&
set X $new && new=:$2:$4:$5:$6 &&
$posix_glob set +f &&
test "$old" = "$new" &&
$cmpprog "$dst" "$dsttmp" >/dev/null 2>&1
then
rm -f "$dsttmp"
else
# Rename the file to the real destination.
$doit $mvcmd -f "$dsttmp" "$dst" 2>/dev/null ||
# The rename failed, perhaps because mv can't rename something else
# to itself, or perhaps because mv is so ancient that it does not
# support -f.
{
# Now remove or move aside any old file at destination location.
# We try this two ways since rm can't unlink itself on some
# systems and the destination file might be busy for other
# reasons. In this case, the final cleanup might fail but the new
# file should still install successfully.
{
test ! -f "$dst" ||
$doit $rmcmd -f "$dst" 2>/dev/null ||
{ $doit $mvcmd -f "$dst" "$rmtmp" 2>/dev/null &&
{ $doit $rmcmd -f "$rmtmp" 2>/dev/null; :; }
} ||
{ echo "$0: cannot unlink or rename $dst" >&2
(exit 1); exit 1
}
} &&
# Now rename the file to the real destination.
$doit $mvcmd "$dsttmp" "$dst"
}
fi || exit 1
trap '' 0
fi
done
# Local variables:
# eval: (add-hook 'write-file-hooks 'time-stamp)
# time-stamp-start: "scriptversion="
# time-stamp-format: "%:y-%02m-%02d.%02H"
# time-stamp-end: "$"
# End:

8406
lib-src/soundtouch/config/ltmain.sh
View File

File diff suppressed because it is too large Load Diff

7370
lib-src/soundtouch/config/m4/libtool.m4 vendored
View File

File diff suppressed because it is too large Load Diff

368
lib-src/soundtouch/config/m4/ltoptions.m4 vendored
View File

@ -1,368 +0,0 @@
# Helper functions for option handling. -*- Autoconf -*-
#
# Copyright (C) 2004, 2005, 2007, 2008 Free Software Foundation, Inc.
# Written by Gary V. Vaughan, 2004
#
# This file is free software; the Free Software Foundation gives
# unlimited permission to copy and/or distribute it, with or without
# modifications, as long as this notice is preserved.
# serial 6 ltoptions.m4
# This is to help aclocal find these macros, as it can't see m4_define.
AC_DEFUN([LTOPTIONS_VERSION], [m4_if([1])])
# _LT_MANGLE_OPTION(MACRO-NAME, OPTION-NAME)
# ------------------------------------------
m4_define([_LT_MANGLE_OPTION],
[[_LT_OPTION_]m4_bpatsubst($1__$2, [[^a-zA-Z0-9_]], [_])])
# _LT_SET_OPTION(MACRO-NAME, OPTION-NAME)
# ---------------------------------------
# Set option OPTION-NAME for macro MACRO-NAME, and if there is a
# matching handler defined, dispatch to it. Other OPTION-NAMEs are
# saved as a flag.
m4_define([_LT_SET_OPTION],
[m4_define(_LT_MANGLE_OPTION([$1], [$2]))dnl
m4_ifdef(_LT_MANGLE_DEFUN([$1], [$2]),
_LT_MANGLE_DEFUN([$1], [$2]),
[m4_warning([Unknown $1 option `$2'])])[]dnl
])
# _LT_IF_OPTION(MACRO-NAME, OPTION-NAME, IF-SET, [IF-NOT-SET])
# ------------------------------------------------------------
# Execute IF-SET if OPTION is set, IF-NOT-SET otherwise.
m4_define([_LT_IF_OPTION],
[m4_ifdef(_LT_MANGLE_OPTION([$1], [$2]), [$3], [$4])])
# _LT_UNLESS_OPTIONS(MACRO-NAME, OPTION-LIST, IF-NOT-SET)
# -------------------------------------------------------
# Execute IF-NOT-SET unless all options in OPTION-LIST for MACRO-NAME
# are set.
m4_define([_LT_UNLESS_OPTIONS],
[m4_foreach([_LT_Option], m4_split(m4_normalize([$2])),
[m4_ifdef(_LT_MANGLE_OPTION([$1], _LT_Option),
[m4_define([$0_found])])])[]dnl
m4_ifdef([$0_found], [m4_undefine([$0_found])], [$3
])[]dnl
])
# _LT_SET_OPTIONS(MACRO-NAME, OPTION-LIST)
# ----------------------------------------
# OPTION-LIST is a space-separated list of Libtool options associated
# with MACRO-NAME. If any OPTION has a matching handler declared with
# LT_OPTION_DEFINE, dispatch to that macro; otherwise complain about
# the unknown option and exit.
m4_defun([_LT_SET_OPTIONS],
[# Set options
m4_foreach([_LT_Option], m4_split(m4_normalize([$2])),
[_LT_SET_OPTION([$1], _LT_Option)])
m4_if([$1],[LT_INIT],[
dnl
dnl Simply set some default values (i.e off) if boolean options were not
dnl specified:
_LT_UNLESS_OPTIONS([LT_INIT], [dlopen], [enable_dlopen=no
])
_LT_UNLESS_OPTIONS([LT_INIT], [win32-dll], [enable_win32_dll=no
])
dnl
dnl If no reference was made to various pairs of opposing options, then
dnl we run the default mode handler for the pair. For example, if neither
dnl `shared' nor `disable-shared' was passed, we enable building of shared
dnl archives by default:
_LT_UNLESS_OPTIONS([LT_INIT], [shared disable-shared], [_LT_ENABLE_SHARED])
_LT_UNLESS_OPTIONS([LT_INIT], [static disable-static], [_LT_ENABLE_STATIC])
_LT_UNLESS_OPTIONS([LT_INIT], [pic-only no-pic], [_LT_WITH_PIC])
_LT_UNLESS_OPTIONS([LT_INIT], [fast-install disable-fast-install],
[_LT_ENABLE_FAST_INSTALL])
])
])# _LT_SET_OPTIONS
## --------------------------------- ##
## Macros to handle LT_INIT options. ##
## --------------------------------- ##
# _LT_MANGLE_DEFUN(MACRO-NAME, OPTION-NAME)
# -----------------------------------------
m4_define([_LT_MANGLE_DEFUN],
[[_LT_OPTION_DEFUN_]m4_bpatsubst(m4_toupper([$1__$2]), [[^A-Z0-9_]], [_])])
# LT_OPTION_DEFINE(MACRO-NAME, OPTION-NAME, CODE)
# -----------------------------------------------
m4_define([LT_OPTION_DEFINE],
[m4_define(_LT_MANGLE_DEFUN([$1], [$2]), [$3])[]dnl
])# LT_OPTION_DEFINE
# dlopen
# ------
LT_OPTION_DEFINE([LT_INIT], [dlopen], [enable_dlopen=yes
])
AU_DEFUN([AC_LIBTOOL_DLOPEN],
[_LT_SET_OPTION([LT_INIT], [dlopen])
AC_DIAGNOSE([obsolete],
[$0: Remove this warning and the call to _LT_SET_OPTION when you
put the `dlopen' option into LT_INIT's first parameter.])
])
dnl aclocal-1.4 backwards compatibility:
dnl AC_DEFUN([AC_LIBTOOL_DLOPEN], [])
# win32-dll
# ---------
# Declare package support for building win32 dll's.
LT_OPTION_DEFINE([LT_INIT], [win32-dll],
[enable_win32_dll=yes
case $host in
*-*-cygwin* | *-*-mingw* | *-*-pw32* | *-cegcc*)
AC_CHECK_TOOL(AS, as, false)
AC_CHECK_TOOL(DLLTOOL, dlltool, false)
AC_CHECK_TOOL(OBJDUMP, objdump, false)
;;
esac
test -z "$AS" && AS=as
_LT_DECL([], [AS], [0], [Assembler program])dnl
test -z "$DLLTOOL" && DLLTOOL=dlltool
_LT_DECL([], [DLLTOOL], [0], [DLL creation program])dnl
test -z "$OBJDUMP" && OBJDUMP=objdump
_LT_DECL([], [OBJDUMP], [0], [Object dumper program])dnl
])# win32-dll
AU_DEFUN([AC_LIBTOOL_WIN32_DLL],
[AC_REQUIRE([AC_CANONICAL_HOST])dnl
_LT_SET_OPTION([LT_INIT], [win32-dll])
AC_DIAGNOSE([obsolete],
[$0: Remove this warning and the call to _LT_SET_OPTION when you
put the `win32-dll' option into LT_INIT's first parameter.])
])
dnl aclocal-1.4 backwards compatibility:
dnl AC_DEFUN([AC_LIBTOOL_WIN32_DLL], [])
# _LT_ENABLE_SHARED([DEFAULT])
# ----------------------------
# implement the --enable-shared flag, and supports the `shared' and
# `disable-shared' LT_INIT options.
# DEFAULT is either `yes' or `no'. If omitted, it defaults to `yes'.
m4_define([_LT_ENABLE_SHARED],
[m4_define([_LT_ENABLE_SHARED_DEFAULT], [m4_if($1, no, no, yes)])dnl
AC_ARG_ENABLE([shared],
[AS_HELP_STRING([--enable-shared@<:@=PKGS@:>@],
[build shared libraries @<:@default=]_LT_ENABLE_SHARED_DEFAULT[@:>@])],
[p=${PACKAGE-default}
case $enableval in
yes) enable_shared=yes ;;
no) enable_shared=no ;;
*)
enable_shared=no
# Look at the argument we got. We use all the common list separators.
lt_save_ifs="$IFS"; IFS="${IFS}$PATH_SEPARATOR,"
for pkg in $enableval; do
IFS="$lt_save_ifs"
if test "X$pkg" = "X$p"; then
enable_shared=yes
fi
done
IFS="$lt_save_ifs"
;;
esac],
[enable_shared=]_LT_ENABLE_SHARED_DEFAULT)
_LT_DECL([build_libtool_libs], [enable_shared], [0],
[Whether or not to build shared libraries])
])# _LT_ENABLE_SHARED
LT_OPTION_DEFINE([LT_INIT], [shared], [_LT_ENABLE_SHARED([yes])])
LT_OPTION_DEFINE([LT_INIT], [disable-shared], [_LT_ENABLE_SHARED([no])])
# Old names:
AC_DEFUN([AC_ENABLE_SHARED],
[_LT_SET_OPTION([LT_INIT], m4_if([$1], [no], [disable-])[shared])
])
AC_DEFUN([AC_DISABLE_SHARED],
[_LT_SET_OPTION([LT_INIT], [disable-shared])
])
AU_DEFUN([AM_ENABLE_SHARED], [AC_ENABLE_SHARED($@)])
AU_DEFUN([AM_DISABLE_SHARED], [AC_DISABLE_SHARED($@)])
dnl aclocal-1.4 backwards compatibility:
dnl AC_DEFUN([AM_ENABLE_SHARED], [])
dnl AC_DEFUN([AM_DISABLE_SHARED], [])
# _LT_ENABLE_STATIC([DEFAULT])
# ----------------------------
# implement the --enable-static flag, and support the `static' and
# `disable-static' LT_INIT options.
# DEFAULT is either `yes' or `no'. If omitted, it defaults to `yes'.
m4_define([_LT_ENABLE_STATIC],
[m4_define([_LT_ENABLE_STATIC_DEFAULT], [m4_if($1, no, no, yes)])dnl
AC_ARG_ENABLE([static],
[AS_HELP_STRING([--enable-static@<:@=PKGS@:>@],
[build static libraries @<:@default=]_LT_ENABLE_STATIC_DEFAULT[@:>@])],
[p=${PACKAGE-default}
case $enableval in
yes) enable_static=yes ;;
no) enable_static=no ;;
*)
enable_static=no
# Look at the argument we got. We use all the common list separators.
lt_save_ifs="$IFS"; IFS="${IFS}$PATH_SEPARATOR,"
for pkg in $enableval; do
IFS="$lt_save_ifs"
if test "X$pkg" = "X$p"; then
enable_static=yes
fi
done
IFS="$lt_save_ifs"
;;
esac],
[enable_static=]_LT_ENABLE_STATIC_DEFAULT)
_LT_DECL([build_old_libs], [enable_static], [0],
[Whether or not to build static libraries])
])# _LT_ENABLE_STATIC
LT_OPTION_DEFINE([LT_INIT], [static], [_LT_ENABLE_STATIC([yes])])
LT_OPTION_DEFINE([LT_INIT], [disable-static], [_LT_ENABLE_STATIC([no])])
# Old names:
AC_DEFUN([AC_ENABLE_STATIC],
[_LT_SET_OPTION([LT_INIT], m4_if([$1], [no], [disable-])[static])
])
AC_DEFUN([AC_DISABLE_STATIC],
[_LT_SET_OPTION([LT_INIT], [disable-static])
])
AU_DEFUN([AM_ENABLE_STATIC], [AC_ENABLE_STATIC($@)])
AU_DEFUN([AM_DISABLE_STATIC], [AC_DISABLE_STATIC($@)])
dnl aclocal-1.4 backwards compatibility:
dnl AC_DEFUN([AM_ENABLE_STATIC], [])
dnl AC_DEFUN([AM_DISABLE_STATIC], [])
# _LT_ENABLE_FAST_INSTALL([DEFAULT])
# ----------------------------------
# implement the --enable-fast-install flag, and support the `fast-install'
# and `disable-fast-install' LT_INIT options.
# DEFAULT is either `yes' or `no'. If omitted, it defaults to `yes'.
m4_define([_LT_ENABLE_FAST_INSTALL],
[m4_define([_LT_ENABLE_FAST_INSTALL_DEFAULT], [m4_if($1, no, no, yes)])dnl
AC_ARG_ENABLE([fast-install],
[AS_HELP_STRING([--enable-fast-install@<:@=PKGS@:>@],
[optimize for fast installation @<:@default=]_LT_ENABLE_FAST_INSTALL_DEFAULT[@:>@])],
[p=${PACKAGE-default}
case $enableval in
yes) enable_fast_install=yes ;;
no) enable_fast_install=no ;;
*)
enable_fast_install=no
# Look at the argument we got. We use all the common list separators.
lt_save_ifs="$IFS"; IFS="${IFS}$PATH_SEPARATOR,"
for pkg in $enableval; do
IFS="$lt_save_ifs"
if test "X$pkg" = "X$p"; then
enable_fast_install=yes
fi
done
IFS="$lt_save_ifs"
;;
esac],
[enable_fast_install=]_LT_ENABLE_FAST_INSTALL_DEFAULT)
_LT_DECL([fast_install], [enable_fast_install], [0],
[Whether or not to optimize for fast installation])dnl
])# _LT_ENABLE_FAST_INSTALL
LT_OPTION_DEFINE([LT_INIT], [fast-install], [_LT_ENABLE_FAST_INSTALL([yes])])
LT_OPTION_DEFINE([LT_INIT], [disable-fast-install], [_LT_ENABLE_FAST_INSTALL([no])])
# Old names:
AU_DEFUN([AC_ENABLE_FAST_INSTALL],
[_LT_SET_OPTION([LT_INIT], m4_if([$1], [no], [disable-])[fast-install])
AC_DIAGNOSE([obsolete],
[$0: Remove this warning and the call to _LT_SET_OPTION when you put
the `fast-install' option into LT_INIT's first parameter.])
])
AU_DEFUN([AC_DISABLE_FAST_INSTALL],
[_LT_SET_OPTION([LT_INIT], [disable-fast-install])
AC_DIAGNOSE([obsolete],
[$0: Remove this warning and the call to _LT_SET_OPTION when you put
the `disable-fast-install' option into LT_INIT's first parameter.])
])
dnl aclocal-1.4 backwards compatibility:
dnl AC_DEFUN([AC_ENABLE_FAST_INSTALL], [])
dnl AC_DEFUN([AM_DISABLE_FAST_INSTALL], [])
# _LT_WITH_PIC([MODE])
# --------------------
# implement the --with-pic flag, and support the `pic-only' and `no-pic'
# LT_INIT options.
# MODE is either `yes' or `no'. If omitted, it defaults to `both'.
m4_define([_LT_WITH_PIC],
[AC_ARG_WITH([pic],
[AS_HELP_STRING([--with-pic],
[try to use only PIC/non-PIC objects @<:@default=use both@:>@])],
[pic_mode="$withval"],
[pic_mode=default])
test -z "$pic_mode" && pic_mode=m4_default([$1], [default])
_LT_DECL([], [pic_mode], [0], [What type of objects to build])dnl
])# _LT_WITH_PIC
LT_OPTION_DEFINE([LT_INIT], [pic-only], [_LT_WITH_PIC([yes])])
LT_OPTION_DEFINE([LT_INIT], [no-pic], [_LT_WITH_PIC([no])])
# Old name:
AU_DEFUN([AC_LIBTOOL_PICMODE],
[_LT_SET_OPTION([LT_INIT], [pic-only])
AC_DIAGNOSE([obsolete],
[$0: Remove this warning and the call to _LT_SET_OPTION when you
put the `pic-only' option into LT_INIT's first parameter.])
])
dnl aclocal-1.4 backwards compatibility:
dnl AC_DEFUN([AC_LIBTOOL_PICMODE], [])
## ----------------- ##
## LTDL_INIT Options ##
## ----------------- ##
m4_define([_LTDL_MODE], [])
LT_OPTION_DEFINE([LTDL_INIT], [nonrecursive],
[m4_define([_LTDL_MODE], [nonrecursive])])
LT_OPTION_DEFINE([LTDL_INIT], [recursive],
[m4_define([_LTDL_MODE], [recursive])])
LT_OPTION_DEFINE([LTDL_INIT], [subproject],
[m4_define([_LTDL_MODE], [subproject])])
m4_define([_LTDL_TYPE], [])
LT_OPTION_DEFINE([LTDL_INIT], [installable],
[m4_define([_LTDL_TYPE], [installable])])
LT_OPTION_DEFINE([LTDL_INIT], [convenience],
[m4_define([_LTDL_TYPE], [convenience])])

123
lib-src/soundtouch/config/m4/ltsugar.m4 vendored
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@ -1,123 +0,0 @@
# ltsugar.m4 -- libtool m4 base layer. -*-Autoconf-*-
#
# Copyright (C) 2004, 2005, 2007, 2008 Free Software Foundation, Inc.
# Written by Gary V. Vaughan, 2004
#
# This file is free software; the Free Software Foundation gives
# unlimited permission to copy and/or distribute it, with or without
# modifications, as long as this notice is preserved.
# serial 6 ltsugar.m4
# This is to help aclocal find these macros, as it can't see m4_define.
AC_DEFUN([LTSUGAR_VERSION], [m4_if([0.1])])
# lt_join(SEP, ARG1, [ARG2...])
# -----------------------------
# Produce ARG1SEPARG2...SEPARGn, omitting [] arguments and their
# associated separator.
# Needed until we can rely on m4_join from Autoconf 2.62, since all earlier
# versions in m4sugar had bugs.
m4_define([lt_join],
[m4_if([$#], [1], [],
[$#], [2], [[$2]],
[m4_if([$2], [], [], [[$2]_])$0([$1], m4_shift(m4_shift($@)))])])
m4_define([_lt_join],
[m4_if([$#$2], [2], [],
[m4_if([$2], [], [], [[$1$2]])$0([$1], m4_shift(m4_shift($@)))])])
# lt_car(LIST)
# lt_cdr(LIST)
# ------------
# Manipulate m4 lists.
# These macros are necessary as long as will still need to support
# Autoconf-2.59 which quotes differently.
m4_define([lt_car], [[$1]])
m4_define([lt_cdr],
[m4_if([$#], 0, [m4_fatal([$0: cannot be called without arguments])],
[$#], 1, [],
[m4_dquote(m4_shift($@))])])
m4_define([lt_unquote], $1)
# lt_append(MACRO-NAME, STRING, [SEPARATOR])
# ------------------------------------------
# Redefine MACRO-NAME to hold its former content plus `SEPARATOR'`STRING'.
# Note that neither SEPARATOR nor STRING are expanded; they are appended
# to MACRO-NAME as is (leaving the expansion for when MACRO-NAME is invoked).
# No SEPARATOR is output if MACRO-NAME was previously undefined (different
# than defined and empty).
#
# This macro is needed until we can rely on Autoconf 2.62, since earlier
# versions of m4sugar mistakenly expanded SEPARATOR but not STRING.
m4_define([lt_append],
[m4_define([$1],
m4_ifdef([$1], [m4_defn([$1])[$3]])[$2])])
# lt_combine(SEP, PREFIX-LIST, INFIX, SUFFIX1, [SUFFIX2...])
# ----------------------------------------------------------
# Produce a SEP delimited list of all paired combinations of elements of
# PREFIX-LIST with SUFFIX1 through SUFFIXn. Each element of the list
# has the form PREFIXmINFIXSUFFIXn.
# Needed until we can rely on m4_combine added in Autoconf 2.62.
m4_define([lt_combine],
[m4_if(m4_eval([$# > 3]), [1],
[m4_pushdef([_Lt_sep], [m4_define([_Lt_sep], m4_defn([lt_car]))])]]dnl
[[m4_foreach([_Lt_prefix], [$2],
[m4_foreach([_Lt_suffix],
]m4_dquote(m4_dquote(m4_shift(m4_shift(m4_shift($@)))))[,
[_Lt_sep([$1])[]m4_defn([_Lt_prefix])[$3]m4_defn([_Lt_suffix])])])])])
# lt_if_append_uniq(MACRO-NAME, VARNAME, [SEPARATOR], [UNIQ], [NOT-UNIQ])
# -----------------------------------------------------------------------
# Iff MACRO-NAME does not yet contain VARNAME, then append it (delimited
# by SEPARATOR if supplied) and expand UNIQ, else NOT-UNIQ.
m4_define([lt_if_append_uniq],
[m4_ifdef([$1],
[m4_if(m4_index([$3]m4_defn([$1])[$3], [$3$2$3]), [-1],
[lt_append([$1], [$2], [$3])$4],
[$5])],
[lt_append([$1], [$2], [$3])$4])])
# lt_dict_add(DICT, KEY, VALUE)
# -----------------------------
m4_define([lt_dict_add],
[m4_define([$1($2)], [$3])])
# lt_dict_add_subkey(DICT, KEY, SUBKEY, VALUE)
# --------------------------------------------
m4_define([lt_dict_add_subkey],
[m4_define([$1($2:$3)], [$4])])
# lt_dict_fetch(DICT, KEY, [SUBKEY])
# ----------------------------------
m4_define([lt_dict_fetch],
[m4_ifval([$3],
m4_ifdef([$1($2:$3)], [m4_defn([$1($2:$3)])]),
m4_ifdef([$1($2)], [m4_defn([$1($2)])]))])
# lt_if_dict_fetch(DICT, KEY, [SUBKEY], VALUE, IF-TRUE, [IF-FALSE])
# -----------------------------------------------------------------
m4_define([lt_if_dict_fetch],
[m4_if(lt_dict_fetch([$1], [$2], [$3]), [$4],
[$5],
[$6])])
# lt_dict_filter(DICT, [SUBKEY], VALUE, [SEPARATOR], KEY, [...])
# --------------------------------------------------------------
m4_define([lt_dict_filter],
[m4_if([$5], [], [],
[lt_join(m4_quote(m4_default([$4], [[, ]])),
lt_unquote(m4_split(m4_normalize(m4_foreach(_Lt_key, lt_car([m4_shiftn(4, $@)]),
[lt_if_dict_fetch([$1], _Lt_key, [$2], [$3], [_Lt_key ])])))))])[]dnl
])

23
lib-src/soundtouch/config/m4/ltversion.m4 vendored
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@ -1,23 +0,0 @@
# ltversion.m4 -- version numbers -*- Autoconf -*-
#
# Copyright (C) 2004 Free Software Foundation, Inc.
# Written by Scott James Remnant, 2004
#
# This file is free software; the Free Software Foundation gives
# unlimited permission to copy and/or distribute it, with or without
# modifications, as long as this notice is preserved.
# Generated from ltversion.in.
# serial 3012 ltversion.m4
# This file is part of GNU Libtool
m4_define([LT_PACKAGE_VERSION], [2.2.6])
m4_define([LT_PACKAGE_REVISION], [1.3012])
AC_DEFUN([LTVERSION_VERSION],
[macro_version='2.2.6'
macro_revision='1.3012'
_LT_DECL(, macro_version, 0, [Which release of libtool.m4 was used?])
_LT_DECL(, macro_revision, 0)
])

92
lib-src/soundtouch/config/m4/lt~obsolete.m4 vendored
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@ -1,92 +0,0 @@
# lt~obsolete.m4 -- aclocal satisfying obsolete definitions. -*-Autoconf-*-
#
# Copyright (C) 2004, 2005, 2007 Free Software Foundation, Inc.
# Written by Scott James Remnant, 2004.
#
# This file is free software; the Free Software Foundation gives
# unlimited permission to copy and/or distribute it, with or without
# modifications, as long as this notice is preserved.
# serial 4 lt~obsolete.m4
# These exist entirely to fool aclocal when bootstrapping libtool.
#
# In the past libtool.m4 has provided macros via AC_DEFUN (or AU_DEFUN)
# which have later been changed to m4_define as they aren't part of the
# exported API, or moved to Autoconf or Automake where they belong.
#
# The trouble is, aclocal is a bit thick. It'll see the old AC_DEFUN
# in /usr/share/aclocal/libtool.m4 and remember it, then when it sees us
# using a macro with the same name in our local m4/libtool.m4 it'll
# pull the old libtool.m4 in (it doesn't see our shiny new m4_define
# and doesn't know about Autoconf macros at all.)
#
# So we provide this file, which has a silly filename so it's always
# included after everything else. This provides aclocal with the
# AC_DEFUNs it wants, but when m4 processes it, it doesn't do anything
# because those macros already exist, or will be overwritten later.
# We use AC_DEFUN over AU_DEFUN for compatibility with aclocal-1.6.
#
# Anytime we withdraw an AC_DEFUN or AU_DEFUN, remember to add it here.
# Yes, that means every name once taken will need to remain here until
# we give up compatibility with versions before 1.7, at which point
# we need to keep only those names which we still refer to.
# This is to help aclocal find these macros, as it can't see m4_define.
AC_DEFUN([LTOBSOLETE_VERSION], [m4_if([1])])
m4_ifndef([AC_LIBTOOL_LINKER_OPTION], [AC_DEFUN([AC_LIBTOOL_LINKER_OPTION])])
m4_ifndef([AC_PROG_EGREP], [AC_DEFUN([AC_PROG_EGREP])])
m4_ifndef([_LT_AC_PROG_ECHO_BACKSLASH], [AC_DEFUN([_LT_AC_PROG_ECHO_BACKSLASH])])
m4_ifndef([_LT_AC_SHELL_INIT], [AC_DEFUN([_LT_AC_SHELL_INIT])])
m4_ifndef([_LT_AC_SYS_LIBPATH_AIX], [AC_DEFUN([_LT_AC_SYS_LIBPATH_AIX])])
m4_ifndef([_LT_PROG_LTMAIN], [AC_DEFUN([_LT_PROG_LTMAIN])])
m4_ifndef([_LT_AC_TAGVAR], [AC_DEFUN([_LT_AC_TAGVAR])])
m4_ifndef([AC_LTDL_ENABLE_INSTALL], [AC_DEFUN([AC_LTDL_ENABLE_INSTALL])])
m4_ifndef([AC_LTDL_PREOPEN], [AC_DEFUN([AC_LTDL_PREOPEN])])
m4_ifndef([_LT_AC_SYS_COMPILER], [AC_DEFUN([_LT_AC_SYS_COMPILER])])
m4_ifndef([_LT_AC_LOCK], [AC_DEFUN([_LT_AC_LOCK])])
m4_ifndef([AC_LIBTOOL_SYS_OLD_ARCHIVE], [AC_DEFUN([AC_LIBTOOL_SYS_OLD_ARCHIVE])])
m4_ifndef([_LT_AC_TRY_DLOPEN_SELF], [AC_DEFUN([_LT_AC_TRY_DLOPEN_SELF])])
m4_ifndef([AC_LIBTOOL_PROG_CC_C_O], [AC_DEFUN([AC_LIBTOOL_PROG_CC_C_O])])
m4_ifndef([AC_LIBTOOL_SYS_HARD_LINK_LOCKS], [AC_DEFUN([AC_LIBTOOL_SYS_HARD_LINK_LOCKS])])
m4_ifndef([AC_LIBTOOL_OBJDIR], [AC_DEFUN([AC_LIBTOOL_OBJDIR])])
m4_ifndef([AC_LTDL_OBJDIR], [AC_DEFUN([AC_LTDL_OBJDIR])])
m4_ifndef([AC_LIBTOOL_PROG_LD_HARDCODE_LIBPATH], [AC_DEFUN([AC_LIBTOOL_PROG_LD_HARDCODE_LIBPATH])])
m4_ifndef([AC_LIBTOOL_SYS_LIB_STRIP], [AC_DEFUN([AC_LIBTOOL_SYS_LIB_STRIP])])
m4_ifndef([AC_PATH_MAGIC], [AC_DEFUN([AC_PATH_MAGIC])])
m4_ifndef([AC_PROG_LD_GNU], [AC_DEFUN([AC_PROG_LD_GNU])])
m4_ifndef([AC_PROG_LD_RELOAD_FLAG], [AC_DEFUN([AC_PROG_LD_RELOAD_FLAG])])
m4_ifndef([AC_DEPLIBS_CHECK_METHOD], [AC_DEFUN([AC_DEPLIBS_CHECK_METHOD])])
m4_ifndef([AC_LIBTOOL_PROG_COMPILER_NO_RTTI], [AC_DEFUN([AC_LIBTOOL_PROG_COMPILER_NO_RTTI])])
m4_ifndef([AC_LIBTOOL_SYS_GLOBAL_SYMBOL_PIPE], [AC_DEFUN([AC_LIBTOOL_SYS_GLOBAL_SYMBOL_PIPE])])
m4_ifndef([AC_LIBTOOL_PROG_COMPILER_PIC], [AC_DEFUN([AC_LIBTOOL_PROG_COMPILER_PIC])])
m4_ifndef([AC_LIBTOOL_PROG_LD_SHLIBS], [AC_DEFUN([AC_LIBTOOL_PROG_LD_SHLIBS])])
m4_ifndef([AC_LIBTOOL_POSTDEP_PREDEP], [AC_DEFUN([AC_LIBTOOL_POSTDEP_PREDEP])])
m4_ifndef([LT_AC_PROG_EGREP], [AC_DEFUN([LT_AC_PROG_EGREP])])
m4_ifndef([LT_AC_PROG_SED], [AC_DEFUN([LT_AC_PROG_SED])])
m4_ifndef([_LT_CC_BASENAME], [AC_DEFUN([_LT_CC_BASENAME])])
m4_ifndef([_LT_COMPILER_BOILERPLATE], [AC_DEFUN([_LT_COMPILER_BOILERPLATE])])
m4_ifndef([_LT_LINKER_BOILERPLATE], [AC_DEFUN([_LT_LINKER_BOILERPLATE])])
m4_ifndef([_AC_PROG_LIBTOOL], [AC_DEFUN([_AC_PROG_LIBTOOL])])
m4_ifndef([AC_LIBTOOL_SETUP], [AC_DEFUN([AC_LIBTOOL_SETUP])])
m4_ifndef([_LT_AC_CHECK_DLFCN], [AC_DEFUN([_LT_AC_CHECK_DLFCN])])
m4_ifndef([AC_LIBTOOL_SYS_DYNAMIC_LINKER], [AC_DEFUN([AC_LIBTOOL_SYS_DYNAMIC_LINKER])])
m4_ifndef([_LT_AC_TAGCONFIG], [AC_DEFUN([_LT_AC_TAGCONFIG])])
m4_ifndef([AC_DISABLE_FAST_INSTALL], [AC_DEFUN([AC_DISABLE_FAST_INSTALL])])
m4_ifndef([_LT_AC_LANG_CXX], [AC_DEFUN([_LT_AC_LANG_CXX])])
m4_ifndef([_LT_AC_LANG_F77], [AC_DEFUN([_LT_AC_LANG_F77])])
m4_ifndef([_LT_AC_LANG_GCJ], [AC_DEFUN([_LT_AC_LANG_GCJ])])
m4_ifndef([AC_LIBTOOL_RC], [AC_DEFUN([AC_LIBTOOL_RC])])
m4_ifndef([AC_LIBTOOL_LANG_C_CONFIG], [AC_DEFUN([AC_LIBTOOL_LANG_C_CONFIG])])
m4_ifndef([_LT_AC_LANG_C_CONFIG], [AC_DEFUN([_LT_AC_LANG_C_CONFIG])])
m4_ifndef([AC_LIBTOOL_LANG_CXX_CONFIG], [AC_DEFUN([AC_LIBTOOL_LANG_CXX_CONFIG])])
m4_ifndef([_LT_AC_LANG_CXX_CONFIG], [AC_DEFUN([_LT_AC_LANG_CXX_CONFIG])])
m4_ifndef([AC_LIBTOOL_LANG_F77_CONFIG], [AC_DEFUN([AC_LIBTOOL_LANG_F77_CONFIG])])
m4_ifndef([_LT_AC_LANG_F77_CONFIG], [AC_DEFUN([_LT_AC_LANG_F77_CONFIG])])
m4_ifndef([AC_LIBTOOL_LANG_GCJ_CONFIG], [AC_DEFUN([AC_LIBTOOL_LANG_GCJ_CONFIG])])
m4_ifndef([_LT_AC_LANG_GCJ_CONFIG], [AC_DEFUN([_LT_AC_LANG_GCJ_CONFIG])])
m4_ifndef([AC_LIBTOOL_LANG_RC_CONFIG], [AC_DEFUN([AC_LIBTOOL_LANG_RC_CONFIG])])
m4_ifndef([_LT_AC_LANG_RC_CONFIG], [AC_DEFUN([_LT_AC_LANG_RC_CONFIG])])
m4_ifndef([AC_LIBTOOL_CONFIG], [AC_DEFUN([AC_LIBTOOL_CONFIG])])
m4_ifndef([_LT_AC_FILE_LTDLL_C], [AC_DEFUN([_LT_AC_FILE_LTDLL_C])])

360
lib-src/soundtouch/config/missing
View File

@ -1,360 +0,0 @@
#! /bin/sh
# Common stub for a few missing GNU programs while installing.
scriptversion=2005-06-08.21
# Copyright (C) 1996, 1997, 1999, 2000, 2002, 2003, 2004, 2005
# Free Software Foundation, Inc.
# Originally by Fran,cois Pinard <pinard@iro.umontreal.ca>, 1996.
# This program 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, or (at your option)
# any later version.
# This program 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., 51 Franklin Street, Fifth Floor, Boston, MA
# 02110-1301, USA.
# As a special exception to the GNU General Public License, if you
# distribute this file as part of a program that contains a
# configuration script generated by Autoconf, you may include it under
# the same distribution terms that you use for the rest of that program.
if test $# -eq 0; then
echo 1>&2 "Try \`$0 --help' for more information"
exit 1
fi
run=:
# In the cases where this matters, `missing' is being run in the
# srcdir already.
if test -f configure.ac; then
configure_ac=configure.ac
else
configure_ac=configure.in
fi
msg="missing on your system"
case "$1" in
--run)
# Try to run requested program, and just exit if it succeeds.
run=
shift
"$@" && exit 0
# Exit code 63 means version mismatch. This often happens
# when the user try to use an ancient version of a tool on
# a file that requires a minimum version. In this case we
# we should proceed has if the program had been absent, or
# if --run hadn't been passed.
if test $? = 63; then
run=:
msg="probably too old"
fi
;;
-h|--h|--he|--hel|--help)
echo "\
$0 [OPTION]... PROGRAM [ARGUMENT]...
Handle \`PROGRAM [ARGUMENT]...' for when PROGRAM is missing, or return an
error status if there is no known handling for PROGRAM.
Options:
-h, --help display this help and exit
-v, --version output version information and exit
--run try to run the given command, and emulate it if it fails
Supported PROGRAM values:
aclocal touch file \`aclocal.m4'
autoconf touch file \`configure'
autoheader touch file \`config.h.in'
automake touch all \`Makefile.in' files
bison create \`y.tab.[ch]', if possible, from existing .[ch]
flex create \`lex.yy.c', if possible, from existing .c
help2man touch the output file
lex create \`lex.yy.c', if possible, from existing .c
makeinfo touch the output file
tar try tar, gnutar, gtar, then tar without non-portable flags
yacc create \`y.tab.[ch]', if possible, from existing .[ch]
Send bug reports to <bug-automake@gnu.org>."
exit $?
;;
-v|--v|--ve|--ver|--vers|--versi|--versio|--version)
echo "missing $scriptversion (GNU Automake)"
exit $?
;;
-*)
echo 1>&2 "$0: Unknown \`$1' option"
echo 1>&2 "Try \`$0 --help' for more information"
exit 1
;;
esac
# Now exit if we have it, but it failed. Also exit now if we
# don't have it and --version was passed (most likely to detect
# the program).
case "$1" in
lex|yacc)
# Not GNU programs, they don't have --version.
;;
tar)
if test -n "$run"; then
echo 1>&2 "ERROR: \`tar' requires --run"
exit 1
elif test "x$2" = "x--version" || test "x$2" = "x--help"; then
exit 1
fi
;;
*)
if test -z "$run" && ($1 --version) > /dev/null 2>&1; then
# We have it, but it failed.
exit 1
elif test "x$2" = "x--version" || test "x$2" = "x--help"; then
# Could not run --version or --help. This is probably someone
# running `$TOOL --version' or `$TOOL --help' to check whether
# $TOOL exists and not knowing $TOOL uses missing.
exit 1
fi
;;
esac
# If it does not exist, or fails to run (possibly an outdated version),
# try to emulate it.
case "$1" in
aclocal*)
echo 1>&2 "\
WARNING: \`$1' is $msg. You should only need it if
you modified \`acinclude.m4' or \`${configure_ac}'. You might want
to install the \`Automake' and \`Perl' packages. Grab them from
any GNU archive site."
touch aclocal.m4
;;
autoconf)
echo 1>&2 "\
WARNING: \`$1' is $msg. You should only need it if
you modified \`${configure_ac}'. You might want to install the
\`Autoconf' and \`GNU m4' packages. Grab them from any GNU
archive site."
touch configure
;;
autoheader)
echo 1>&2 "\
WARNING: \`$1' is $msg. You should only need it if
you modified \`acconfig.h' or \`${configure_ac}'. You might want
to install the \`Autoconf' and \`GNU m4' packages. Grab them
from any GNU archive site."
files=`sed -n 's/^[ ]*A[CM]_CONFIG_HEADER(\([^)]*\)).*/\1/p' ${configure_ac}`
test -z "$files" && files="config.h"
touch_files=
for f in $files; do
case "$f" in
*:*) touch_files="$touch_files "`echo "$f" |
sed -e 's/^[^:]*://' -e 's/:.*//'`;;
*) touch_files="$touch_files $f.in";;
esac
done
touch $touch_files
;;
automake*)
echo 1>&2 "\
WARNING: \`$1' is $msg. You should only need it if
you modified \`Makefile.am', \`acinclude.m4' or \`${configure_ac}'.
You might want to install the \`Automake' and \`Perl' packages.
Grab them from any GNU archive site."
find . -type f -name Makefile.am -print |
sed 's/\.am$/.in/' |
while read f; do touch "$f"; done
;;
autom4te)
echo 1>&2 "\
WARNING: \`$1' is needed, but is $msg.
You might have modified some files without having the
proper tools for further handling them.
You can get \`$1' as part of \`Autoconf' from any GNU
archive site."
file=`echo "$*" | sed -n 's/.*--output[ =]*\([^ ]*\).*/\1/p'`
test -z "$file" && file=`echo "$*" | sed -n 's/.*-o[ ]*\([^ ]*\).*/\1/p'`
if test -f "$file"; then
touch $file
else
test -z "$file" || exec >$file
echo "#! /bin/sh"
echo "# Created by GNU Automake missing as a replacement of"
echo "# $ $@"
echo "exit 0"
chmod +x $file
exit 1
fi
;;
bison|yacc)
echo 1>&2 "\
WARNING: \`$1' $msg. You should only need it if
you modified a \`.y' file. You may need the \`Bison' package
in order for those modifications to take effect. You can get
\`Bison' from any GNU archive site."
rm -f y.tab.c y.tab.h
if [ $# -ne 1 ]; then
eval LASTARG="\${$#}"
case "$LASTARG" in
*.y)
SRCFILE=`echo "$LASTARG" | sed 's/y$/c/'`
if [ -f "$SRCFILE" ]; then
cp "$SRCFILE" y.tab.c
fi
SRCFILE=`echo "$LASTARG" | sed 's/y$/h/'`
if [ -f "$SRCFILE" ]; then
cp "$SRCFILE" y.tab.h
fi
;;
esac
fi
if [ ! -f y.tab.h ]; then
echo >y.tab.h
fi
if [ ! -f y.tab.c ]; then
echo 'main() { return 0; }' >y.tab.c
fi
;;
lex|flex)
echo 1>&2 "\
WARNING: \`$1' is $msg. You should only need it if
you modified a \`.l' file. You may need the \`Flex' package
in order for those modifications to take effect. You can get
\`Flex' from any GNU archive site."
rm -f lex.yy.c
if [ $# -ne 1 ]; then
eval LASTARG="\${$#}"
case "$LASTARG" in
*.l)
SRCFILE=`echo "$LASTARG" | sed 's/l$/c/'`
if [ -f "$SRCFILE" ]; then
cp "$SRCFILE" lex.yy.c
fi
;;
esac
fi
if [ ! -f lex.yy.c ]; then
echo 'main() { return 0; }' >lex.yy.c
fi
;;
help2man)
echo 1>&2 "\
WARNING: \`$1' is $msg. You should only need it if
you modified a dependency of a manual page. You may need the
\`Help2man' package in order for those modifications to take
effect. You can get \`Help2man' from any GNU archive site."
file=`echo "$*" | sed -n 's/.*-o \([^ ]*\).*/\1/p'`
if test -z "$file"; then
file=`echo "$*" | sed -n 's/.*--output=\([^ ]*\).*/\1/p'`
fi
if [ -f "$file" ]; then
touch $file
else
test -z "$file" || exec >$file
echo ".ab help2man is required to generate this page"
exit 1
fi
;;
makeinfo)
echo 1>&2 "\
WARNING: \`$1' is $msg. You should only need it if
you modified a \`.texi' or \`.texinfo' file, or any other file
indirectly affecting the aspect of the manual. The spurious
call might also be the consequence of using a buggy \`make' (AIX,
DU, IRIX). You might want to install the \`Texinfo' package or
the \`GNU make' package. Grab either from any GNU archive site."
# The file to touch is that specified with -o ...
file=`echo "$*" | sed -n 's/.*-o \([^ ]*\).*/\1/p'`
if test -z "$file"; then
# ... or it is the one specified with @setfilename ...
infile=`echo "$*" | sed 's/.* \([^ ]*\) *$/\1/'`
file=`sed -n '/^@setfilename/ { s/.* \([^ ]*\) *$/\1/; p; q; }' $infile`
# ... or it is derived from the source name (dir/f.texi becomes f.info)
test -z "$file" && file=`echo "$infile" | sed 's,.*/,,;s,.[^.]*$,,'`.info
fi
# If the file does not exist, the user really needs makeinfo;
# let's fail without touching anything.
test -f $file || exit 1
touch $file
;;
tar)
shift
# We have already tried tar in the generic part.
# Look for gnutar/gtar before invocation to avoid ugly error
# messages.
if (gnutar --version > /dev/null 2>&1); then
gnutar "$@" && exit 0
fi
if (gtar --version > /dev/null 2>&1); then
gtar "$@" && exit 0
fi
firstarg="$1"
if shift; then
case "$firstarg" in
*o*)
firstarg=`echo "$firstarg" | sed s/o//`
tar "$firstarg" "$@" && exit 0
;;
esac
case "$firstarg" in
*h*)
firstarg=`echo "$firstarg" | sed s/h//`
tar "$firstarg" "$@" && exit 0
;;
esac
fi
echo 1>&2 "\
WARNING: I can't seem to be able to run \`tar' with the given arguments.
You may want to install GNU tar or Free paxutils, or check the
command line arguments."
exit 1
;;
*)
echo 1>&2 "\
WARNING: \`$1' is needed, and is $msg.
You might have modified some files without having the
proper tools for further handling them. Check the \`README' file,
it often tells you about the needed prerequisites for installing
this package. You may also peek at any GNU archive site, in case
some other package would contain this missing \`$1' program."
exit 1
;;
esac
exit 0
# Local variables:
# eval: (add-hook 'write-file-hooks 'time-stamp)
# time-stamp-start: "scriptversion="
# time-stamp-format: "%:y-%02m-%02d.%02H"
# time-stamp-end: "$"
# End:

19026
lib-src/soundtouch/configure vendored
View File

File diff suppressed because it is too large Load Diff

3
lib-src/soundtouch/configure-file-README.txt Normal file
View File

@ -0,0 +1,3 @@
Starting from SoundTouch 1.6.0, the "configure" file is removed from the source code package due to autoconf/automake version conflicts.
Instead, generate the "configure" file using local tools by invoking "./bootstrap" script, then configure & compile as usual.

156
lib-src/soundtouch/configure.ac
View File

@ -1,6 +1,6 @@
dnl SoundTouch configure.ac, by David W. Durham
dnl
dnl $Id: configure.ac,v 1.9 2009-11-07 12:17:03 richardash1981 Exp $
dnl $Id: configure.ac 141 2012-04-04 19:47:28Z oparviai $
dnl
dnl This file is part of SoundTouch, an audio processing library for pitch/time adjustments
dnl
@ -19,29 +19,23 @@ 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_INIT(SoundTouch, 1.7.0, [http://www.surina.net/soundtouch])
AC_CONFIG_AUX_DIR(config)
AC_CONFIG_MACRO_DIR([config/m4])
AM_CONFIG_HEADER([include/soundtouch_config.h])
AM_INIT_AUTOMAKE([-Wall foreign])
AM_MAINTAINER_MODE
AC_DISABLE_SHARED dnl This makes libtool only build static libs
AM_INIT_AUTOMAKE
#AC_DISABLE_SHARED dnl This makes libtool only build static libs
AC_DISABLE_STATIC dnl This makes libtool only build shared libs
#AC_GNU_SOURCE dnl enable posix extensions in glibc
AC_LANG(C++)
dnl ############################################################################
dnl # Checks for programs #
dnl ############################################################################
dnl dmazzoni: Sorry, SoundTouch authors/packagers, but trying to override
dnl CXXFLAGS ruins cross-compilation. Commenting this out.
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
@ -62,12 +56,14 @@ 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])
AC_CHECK_HEADERS([cpuid.h])
if test "x$ac_cv_header_cpuid_h" = "xno"; then
AC_MSG_WARN([The cpuid.h file was not found therefore the x86 optimizations will be disabled.])
AC_MSG_WARN([If using a x86 architecture and optimizations are desired then please install gcc (>= 4.3).])
AC_MSG_WARN([If using a non-x86 architecture then this is expected and can be ignored.])
fi
dnl ############################################################################
dnl # Checks for typedefs, structures, and compiler characteristics $
@ -84,24 +80,116 @@ AC_ARG_ENABLE(integer-samples,
[default=yes]])],,
[enable_integer_samples=no])
# Let the user enable/disable the x86 optimizations.
# Useful when compiling on non-x86 architectures.
AC_ARG_ENABLE([x86-optimizations],
[AS_HELP_STRING([--enable-x86-optimizations],
[use MMX or SSE optimization
[default=yes]])],[enable_x86_optimizations="${enableval}"],
[enable_x86_optimizations=yes])
# Tell the Makefile.am if the user wants to disable optimizations.
# Makefile.am will enable them by default if support is available.
# Note: We check if optimizations are supported a few lines down.
AM_CONDITIONAL([X86_OPTIMIZATIONS], [test "x$enable_x86_optimizations" = "xyes"])
if test "x$enable_integer_samples" = "xyes"; then
echo "****** Integer sample type enabled ******"
AC_DEFINE(INTEGER_SAMPLES,1,[Use Integer as Sample type])
AC_DEFINE(SOUNDTOUCH_INTEGER_SAMPLES,1,[Use Integer as Sample type])
else
echo "****** Float sample type enabled ******"
AC_DEFINE(FLOAT_SAMPLES,1,[Use Float as Sample type])
AC_DEFINE(SOUNDTOUCH_FLOAT_SAMPLES,1,[Use Float as Sample type])
fi
dnl LLL: Added to choose whether to use -msse or not
AC_SUBST(FLAG_SSE)
FLAG_SSE=
saveflags=$CXXFLAGS
CXXFLAGS="$CXXFLAGS -msse"
AC_TRY_COMPILE([],[],[FLAG_SSE="-msse"])
CXXFLAGS="$saveflags"
dnl LLL: Added to choose whether to use -msse or not
# Check if optimizations are supported in the system at build time.
if test "x$enable_x86_optimizations" = "xyes" -a "x$ac_cv_header_cpuid_h" = "xyes"; then
echo "****** x86 optimizations enabled ******"
original_saved_CXXFLAGS=$CXXFLAGS
have_mmx_intrinsics=no
OPT_CXXFLAGS="-mmmx -Winline"
CXXFLAGS="$OPT_CXXFLAGS $CXXFLAGS"
# Check if the user can compile MMX code using intrinsics.
# GCC supports MMX intrinsics since version 3.3
# A more recent GCC (>= 4.3) is recommended.
AC_COMPILE_IFELSE([AC_LANG_SOURCE([[
#if defined(__GNUC__) && (__GNUC__ < 3 || (__GNUC__ == 3 && __GNUC_MINOR__ < 3))
#error "Need GCC >= 3.3 for MMX intrinsics"
#endif
#include <mmintrin.h>
int main () {
__m64 loop = _mm_cvtsi32_si64 (1);
return _mm_cvtsi64_si32 (loop);
}]])],[have_mmx_intrinsics=yes])
CXXFLAGS=$original_saved_CXXFLAGS
# Inform the user if we did or did not find MMX support.
#
# If we enable optimization and integer samples we only require MMX.
# Disable optimizations in the SSTypes.h file if this is not the case.
if test "x$have_mmx_intrinsics" = "xyes" ; then
echo "****** MMX support found ******"
else
echo "****** No MMX support found ******"
if test "x$enable_integer_samples" = "xyes"; then
echo "****** Disabling optimizations. Using integer samples with no MMX support ******"
AC_DEFINE([SOUNDTOUCH_DISABLE_X86_OPTIMIZATIONS],[1],[Do not use x86 optimizations])
fi
fi
have_sse_intrinsics=no
OPT_CXXFLAGS="-msse -Winline"
CXXFLAGS="$OPT_CXXFLAGS $CXXFLAGS"
# Check if the user can compile SSE code using intrinsics.
# GCC supports SSE intrinsics since version 3.3
# A more recent GCC (>= 4.3) is recommended.
AC_COMPILE_IFELSE([AC_LANG_SOURCE([[
#if defined(__GNUC__) && (__GNUC__ < 3 || (__GNUC__ == 3 && __GNUC_MINOR__ < 3))
#error "Need GCC >= 3.3 for SSE intrinsics"
#endif
#include <xmmintrin.h>
int main () {
_mm_setzero_ps();
return 0;
}]])],[have_sse_intrinsics=yes])
CXXFLAGS=$original_saved_CXXFLAGS
# Inform the user if we did or did not find SSE support.
#
# If we enable optimization and float samples we only require SSE.
# Disable optimizations in the SSTypes.h file if this is not the case.
if test "x$have_sse_intrinsics" = "xyes" ; then
echo "****** SSE support found ******"
else
echo "****** No SSE support found ******"
if test "x$enable_integer_samples" != "xyes"; then
echo "****** Disabling optimizations. Using float samples with no SSE support ******"
AC_DEFINE([SOUNDTOUCH_DISABLE_X86_OPTIMIZATIONS],[1],[Do not use x86 optimizations])
fi
fi
else
# Disable optimizations in SSTypes.h since the user requested it.
echo "****** x86 optimizations disabled ******"
AC_DEFINE([SOUNDTOUCH_DISABLE_X86_OPTIMIZATIONS],[1],[Do not use x86 optimizations])
fi
# SSTypes.h by default enables optimizations. Those already got disabled if
# the user requested for it or if the system does not support them.
#
# Now tell the Makefile.am the optimizations that are supported.
# Note:
# Makefile.am already knows if the user asked for optimizations. We apply
# optimizations by default (if support is available) and then disable all of
# them if the user requested it.
AM_CONDITIONAL([HAVE_MMX], [test "x$have_mmx_intrinsics" = "xyes"])
AM_CONDITIONAL([HAVE_SSE], [test "x$have_sse_intrinsics" = "xyes"])
dnl ############################################################################
@ -123,7 +211,7 @@ dnl add whatever functions you might want to check for here
dnl ############################################################################
dnl # Internationalization and Localization #
dnl # Internationaliation and Localiation #
dnl ############################################################################
#AM_GNU_GETTEXT_VERSION([0.11.5])
#AM_GNU_GETTEXT([external])
@ -140,14 +228,12 @@ AC_CONFIG_FILES([
Makefile
source/Makefile
source/SoundTouch/Makefile
source/example/Makefile
source/example/SoundStretch/Makefile
source/example/bpm/Makefile
source/SoundStretch/Makefile
include/Makefile
])
AC_OUTPUT(
soundtouch-1.0.pc
soundtouch.pc
)
dnl use 'echo' to put stuff here if you want a message to the builder

65
lib-src/soundtouch/correct-const-usage.patch
View File

@ -1,65 +0,0 @@
Index: include/FIFOSampleBuffer.h
===================================================================
RCS file: /cvsroot/audacity/lib-src/soundtouch/include/FIFOSampleBuffer.h,v
retrieving revision 1.4
diff -u -r1.4 FIFOSampleBuffer.h
--- include/FIFOSampleBuffer.h 18 Sep 2006 22:29:21 -0000 1.4
+++ include/FIFOSampleBuffer.h 17 Mar 2008 22:27:54 -0000
@@ -160,7 +160,7 @@
virtual uint numSamples() const;
/// Sets number of channels, 1 = mono, 2 = stereo.
- void setChannels(uint numChannels);
+ void setChannels(const uint numChannels);
/// Returns nonzero if there aren't any samples available for outputting.
virtual int isEmpty() const;
Index: source/SoundTouch/AAFilter.h
===================================================================
RCS file: /cvsroot/audacity/lib-src/soundtouch/source/SoundTouch/AAFilter.h,v
retrieving revision 1.4
diff -u -r1.4 AAFilter.h
--- source/SoundTouch/AAFilter.h 18 Sep 2006 22:29:22 -0000 1.4
+++ source/SoundTouch/AAFilter.h 17 Mar 2008 22:27:54 -0000
@@ -63,14 +63,14 @@
/// Calculate the FIR coefficients realizing the given cutoff-frequency
void calculateCoeffs();
public:
- AAFilter(uint length);
+ AAFilter(const 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);
+ void setCutoffFreq(const double newCutoffFreq);
/// Sets number of FIR filter taps, i.e. ~filter complexity
void setLength(uint newLength);
Index: source/SoundTouch/RateTransposer.h
===================================================================
RCS file: /cvsroot/audacity/lib-src/soundtouch/source/SoundTouch/RateTransposer.h,v
retrieving revision 1.4
diff -u -r1.4 RateTransposer.h
--- source/SoundTouch/RateTransposer.h 18 Sep 2006 22:29:22 -0000 1.4
+++ source/SoundTouch/RateTransposer.h 17 Mar 2008 22:27:54 -0000
@@ -134,7 +134,7 @@
AAFilter *getAAFilter() const;
/// Enables/disables the anti-alias filter. Zero to disable, nonzero to enable
- void enableAAFilter(BOOL newMode);
+ void enableAAFilter(const BOOL newMode);
/// Returns nonzero if anti-alias filter is enabled.
BOOL isAAFilterEnabled() const;
@@ -144,7 +144,7 @@
virtual void setRate(float newRate);
/// Sets the number of channels, 1 = mono, 2 = stereo
- void setChannels(uint channels);
+ void setChannels(const uint channels);
/// Adds 'numSamples' pcs of samples from the 'samples' memory position into
/// the input of the object.

15
lib-src/soundtouch/fix-includes.patch
View File

@ -1,15 +0,0 @@
Index: soundtouch/source/example/SoundStretch/WavFile.cpp
===================================================================
RCS file: /cvsroot/audacity/lib-src/soundtouch/source/example/SoundStretch/WavFile.cpp,v
retrieving revision 1.3
diff -u -r1.3 WavFile.cpp
--- soundtouch/source/example/SoundStretch/WavFile.cpp 18 Sep 2006 07:31:48 -0000 1.3
+++ soundtouch/source/example/SoundStretch/WavFile.cpp 25 Jun 2008 19:56:13 -0000
@@ -48,6 +48,7 @@
#include <stdio.h>
#include <stdexcept>
#include <string>
+#include <cstring>
#include <assert.h>
#include <limits.h>

49
lib-src/soundtouch/include/BPMDetect.h
View File

@ -14,10 +14,10 @@
/// 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
/// - 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
/// - 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
@ -26,10 +26,10 @@
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2006-09-18 22:29:21 $
// File revision : $Revision: 1.3 $
// Last changed : $Date: 2012-08-30 22:53:44 +0300 (Thu, 30 Aug 2012) $
// File revision : $Revision: 4 $
//
// $Id: BPMDetect.h,v 1.3 2006-09-18 22:29:21 martynshaw Exp $
// $Id: BPMDetect.h 150 2012-08-30 19:53:44Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
@ -60,11 +60,14 @@
#include "STTypes.h"
#include "FIFOSampleBuffer.h"
namespace soundtouch
{
/// Minimum allowed BPM rate. Used to restrict accepted result above a reasonable limit.
#define MIN_BPM 45
#define MIN_BPM 29
/// Maximum allowed BPM rate. Used to restrict accepted result below a reasonable limit.
#define MAX_BPM 230
#define MAX_BPM 200
/// Class for calculating BPM rate for audio data.
@ -73,12 +76,12 @@ class BPMDetect
protected:
/// Auto-correlation accumulator bins.
float *xcorr;
/// Amplitude envelope sliding average approximation level accumulator
float envelopeAccu;
double envelopeAccu;
/// RMS volume sliding average approximation level accumulator
float RMSVolumeAccu;
double RMSVolumeAccu;
/// Sample average counter.
int decimateCount;
@ -101,15 +104,12 @@ protected:
/// 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
/// 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.
);
@ -128,6 +128,9 @@ protected:
int numsamples ///< Number of samples in buffer
);
/// remove constant bias from xcorr data
void removeBias();
public:
/// Constructor.
BPMDetect(int numChannels, ///< Number of channels in sample data.
@ -139,12 +142,12 @@ public:
/// 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.
///
/// 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
void inputSamples(const soundtouch::SAMPLETYPE *samples, ///< Pointer to input/working data buffer
int numSamples ///< Number of samples in buffer
);
@ -156,4 +159,6 @@ public:
float getBpm();
};
}
#endif // _BPMDetect_H_

64
lib-src/soundtouch/include/FIFOSampleBuffer.h
View File

@ -1,12 +1,12 @@
////////////////////////////////////////////////////////////////////////////////
///
/// A buffer class for temporarily storaging sound samples, operates as a
/// 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'
/// 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
/// 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
@ -15,10 +15,10 @@
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2008-03-17 22:29:58 $
// File revision : $Revision: 1.5 $
// Last changed : $Date: 2012-06-13 22:29:53 +0300 (Wed, 13 Jun 2012) $
// File revision : $Revision: 4 $
//
// $Id: FIFOSampleBuffer.h,v 1.5 2008-03-17 22:29:58 richardash1981 Exp $
// $Id: FIFOSampleBuffer.h 143 2012-06-13 19:29:53Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
@ -54,7 +54,7 @@ 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
/// Notice that in case of stereo audio, one sample is considered to consist of
/// both channel data.
class FIFOSampleBuffer : public FIFOSamplePipe
{
@ -75,17 +75,17 @@ private:
/// Channels, 1=mono, 2=stereo.
uint channels;
/// Current position pointer to the buffer. This pointer is increased when samples are
/// 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
/// 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(const uint capacityRequirement);
void ensureCapacity(uint capacityRequirement);
/// Returns current capacity.
uint getCapacity() const;
@ -93,34 +93,34 @@ private:
public:
/// Constructor
FIFOSampleBuffer(uint numChannels = 2 ///< Number of channels, 1=mono, 2=stereo.
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.
/// 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
/// output samples from the buffer by calling the
/// 'receiveSamples(numSamples)' function
virtual SAMPLETYPE *ptrBegin() const;
virtual SAMPLETYPE *ptrBegin();
/// 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
/// 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
/// 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
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.
);
@ -130,17 +130,17 @@ public:
uint numSamples ///< Number of samples to insert.
);
/// Adjusts the book-keeping to increase number of samples in the buffer without
/// 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
/// 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
/// 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.
@ -148,8 +148,8 @@ public:
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.
/// 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.
@ -160,13 +160,17 @@ public:
virtual uint numSamples() const;
/// Sets number of channels, 1 = mono, 2 = stereo.
void setChannels(const uint numChannels);
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();
/// allow trimming (downwards) amount of samples in pipeline.
/// Returns adjusted amount of samples
uint adjustAmountOfSamples(uint numSamples);
};
}

67
lib-src/soundtouch/include/FIFOSamplePipe.h
View File

@ -5,7 +5,7 @@
/// 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
/// '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
@ -17,10 +17,10 @@
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2006-09-18 22:29:21 $
// File revision : $Revision: 1.4 $
// Last changed : $Date: 2012-06-13 22:29:53 +0300 (Wed, 13 Jun 2012) $
// File revision : $Revision: 4 $
//
// $Id: FIFOSamplePipe.h,v 1.4 2006-09-18 22:29:21 martynshaw Exp $
// $Id: FIFOSamplePipe.h 143 2012-06-13 19:29:53Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
@ -59,19 +59,23 @@ namespace soundtouch
class FIFOSamplePipe
{
public:
/// Returns a pointer to the beginning of the output samples.
/// This function is provided for accessing the output samples directly.
// virtual default destructor
virtual ~FIFOSamplePipe() {}
/// 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
/// output samples from the buffer by calling the
/// 'receiveSamples(numSamples)' function
virtual SAMPLETYPE *ptrBegin() const = 0;
virtual SAMPLETYPE *ptrBegin() = 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.
uint numSamples ///< Number of samples to insert.
) = 0;
@ -85,8 +89,8 @@ public:
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
/// 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.
@ -94,8 +98,8 @@ public:
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.
/// 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.
@ -110,16 +114,21 @@ public:
/// Clears all the samples.
virtual void clear() = 0;
/// allow trimming (downwards) amount of samples in pipeline.
/// Returns adjusted amount of samples
virtual uint adjustAmountOfSamples(uint numSamples) = 0;
};
/// Base-class for sound processing routines working in FIFO principle. With this base
/// 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
/// 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
/// 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
@ -137,7 +146,7 @@ protected:
}
/// Constructor. Doesn't define output pipe; it has to be set be
/// Constructor. Doesn't define output pipe; it has to be set be
/// 'setOutPipe' function.
FIFOProcessor()
{
@ -159,22 +168,22 @@ protected:
}
/// Returns a pointer to the beginning of the output samples.
/// This function is provided for accessing the output samples directly.
/// 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
/// output samples from the buffer by calling the
/// 'receiveSamples(numSamples)' function
virtual SAMPLETYPE *ptrBegin() const
virtual SAMPLETYPE *ptrBegin()
{
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
/// 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.
@ -186,8 +195,8 @@ public:
}
/// Adjusts book-keeping so that given number of samples are removed from beginning of the
/// sample buffer without copying them anywhere.
/// 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.
@ -210,6 +219,14 @@ public:
{
return output->isEmpty();
}
/// allow trimming (downwards) amount of samples in pipeline.
/// Returns adjusted amount of samples
virtual uint adjustAmountOfSamples(uint numSamples)
{
return output->adjustAmountOfSamples(numSamples);
}
};
}

2
lib-src/soundtouch/include/Makefile.am
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@ -1,6 +1,6 @@
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## $Id: Makefile.am,v 1.3 2006-09-27 06:19:52 llucius Exp $
## $Id: Makefile.am 11 2008-02-10 16:26:55Z oparviai $
##
## Copyright (C) 2003 - David W. Durham
##

424
lib-src/soundtouch/include/Makefile.in
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@ -1,424 +0,0 @@
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@$(MAKE) $(AM_MAKEFLAGS) install-exec-am install-data-am
installcheck: installcheck-am
install-strip:
$(MAKE) $(AM_MAKEFLAGS) INSTALL_PROGRAM="$(INSTALL_STRIP_PROGRAM)" \
install_sh_PROGRAM="$(INSTALL_STRIP_PROGRAM)" INSTALL_STRIP_FLAG=-s \
`test -z '$(STRIP)' || \
echo "INSTALL_PROGRAM_ENV=STRIPPROG='$(STRIP)'"` install
mostlyclean-generic:
clean-generic:
distclean-generic:
-test -z "$(CONFIG_CLEAN_FILES)" || rm -f $(CONFIG_CLEAN_FILES)
maintainer-clean-generic:
@echo "This command is intended for maintainers to use"
@echo "it deletes files that may require special tools to rebuild."
clean: clean-am
clean-am: clean-generic clean-libtool mostlyclean-am
distclean: distclean-am
-rm -f Makefile
distclean-am: clean-am distclean-generic distclean-hdr \
distclean-libtool distclean-tags
dvi: dvi-am
dvi-am:
html: html-am
info: info-am
info-am:
install-data-am: install-pkgincludeHEADERS
install-exec-am:
install-info: install-info-am
install-man:
installcheck-am:
maintainer-clean: maintainer-clean-am
-rm -f Makefile
maintainer-clean-am: distclean-am maintainer-clean-generic
mostlyclean: mostlyclean-am
mostlyclean-am: mostlyclean-generic mostlyclean-libtool
pdf: pdf-am
pdf-am:
ps: ps-am
ps-am:
uninstall-am: uninstall-info-am uninstall-pkgincludeHEADERS
.PHONY: CTAGS GTAGS all all-am check check-am clean clean-generic \
clean-libtool ctags distclean distclean-generic distclean-hdr \
distclean-libtool distclean-tags distdir dvi dvi-am html \
html-am info info-am install install-am install-data \
install-data-am install-exec install-exec-am install-info \
install-info-am install-man install-pkgincludeHEADERS \
install-strip installcheck installcheck-am installdirs \
maintainer-clean maintainer-clean-generic mostlyclean \
mostlyclean-generic mostlyclean-libtool pdf pdf-am ps ps-am \
tags uninstall uninstall-am uninstall-info-am \
uninstall-pkgincludeHEADERS
# Tell versions [3.59,3.63) of GNU make to not export all variables.
# Otherwise a system limit (for SysV at least) may be exceeded.
.NOEXPORT:

154
lib-src/soundtouch/include/STTypes.h
View File

@ -8,10 +8,10 @@
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2006-09-18 22:29:21 $
// File revision : $Revision: 1.8 $
// Last changed : $Date: 2012-12-28 16:53:56 +0200 (Fri, 28 Dec 2012) $
// File revision : $Revision: 3 $
//
// $Id: STTypes.h,v 1.8 2006-09-18 22:29:21 martynshaw Exp $
// $Id: STTypes.h 162 2012-12-28 14:53:56Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
@ -42,8 +42,21 @@
typedef unsigned int uint;
typedef unsigned long ulong;
#ifdef __GNUC__
// In GCC, include soundtouch_config.h made by config scritps
// Patch for MinGW: on Win64 long is 32-bit
#ifdef _WIN64
typedef unsigned long long ulongptr;
#else
typedef ulong ulongptr;
#endif
// Helper macro for aligning pointer up to next 16-byte boundary
#define SOUNDTOUCH_ALIGN_POINTER_16(x) ( ( (ulongptr)(x) + 15 ) & ~(ulongptr)15 )
#if (defined(__GNUC__) && !defined(ANDROID))
// In GCC, include soundtouch_config.h made by config scritps.
// Skip this in Android compilation that uses GCC but without configure scripts.
#include "soundtouch_config.h"
#endif
@ -60,63 +73,81 @@ typedef unsigned long ulong;
namespace soundtouch
{
/// Activate these undef's to overrule the possible sampletype
/// setting inherited from some other header file:
//#undef INTEGER_SAMPLES
//#undef FLOAT_SAMPLES
/// Activate these undef's to overrule the possible sampletype
/// setting inherited from some other header file:
//#undef SOUNDTOUCH_INTEGER_SAMPLES
//#undef SOUNDTOUCH_FLOAT_SAMPLES
#if !(INTEGER_SAMPLES || FLOAT_SAMPLES)
#if (defined(__SOFTFP__))
// For Android compilation: Force use of Integer samples in case that
// compilation uses soft-floating point emulation - soft-fp is way too slow
#undef SOUNDTOUCH_FLOAT_SAMPLES
#define SOUNDTOUCH_INTEGER_SAMPLES 1
#endif
/// 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
#if !(SOUNDTOUCH_INTEGER_SAMPLES || SOUNDTOUCH_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 SOUNDTOUCH_INTEGER_SAMPLES 1 //< 16bit integer samples
#define SOUNDTOUCH_FLOAT_SAMPLES 1 //< 32bit float samples
#endif
#endif
#if (_M_IX86 || __i386__ || __x86_64__ || _M_X64)
/// Define this to allow X86-specific assembler/intrinsic optimizations.
/// Notice that library contains also usual C++ versions of each of these
/// these routines, so if you're having difficulties getting the optimized
/// routines compiled for whatever reason, you may disable these optimizations
/// to make the library compile.
/// 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
#define SOUNDTOUCH_ALLOW_X86_OPTIMIZATIONS 1
/// In GNU environment, allow the user to override this setting by
/// giving the following switch to the configure script:
/// ./configure --disable-x86-optimizations
/// ./configure --enable-x86-optimizations=no
#ifdef SOUNDTOUCH_DISABLE_X86_OPTIMIZATIONS
#undef SOUNDTOUCH_ALLOW_X86_OPTIMIZATIONS
#endif
#else
/// Always disable optimizations when not using a x86 systems.
#undef SOUNDTOUCH_ALLOW_X86_OPTIMIZATIONS
// If defined, allows the SIMD-optimized routines to take minor shortcuts
// for improved performance. Undefine to require faithfully similar SIMD
#endif
// If defined, allows the SIMD-optimized routines to take minor shortcuts
// for improved performance. Undefine to require faithfully similar SIMD
// calculations as in normal C implementation.
#define ALLOW_NONEXACT_SIMD_OPTIMIZATION 1
#define SOUNDTOUCH_ALLOW_NONEXACT_SIMD_OPTIMIZATION 1
#ifdef INTEGER_SAMPLES
#ifdef SOUNDTOUCH_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
#ifdef SOUNDTOUCH_FLOAT_SAMPLES
// check that only one sample type is defined
#error "conflicting sample types defined"
#endif // FLOAT_SAMPLES
#endif // SOUNDTOUCH_FLOAT_SAMPLES
#ifdef ALLOW_OPTIMIZATIONS
#if (WIN32 || __i386__ || __x86_64__)
// Allow MMX optimizations
#define ALLOW_MMX 1
#endif
#ifdef SOUNDTOUCH_ALLOW_X86_OPTIMIZATIONS
// Allow MMX optimizations
#define SOUNDTOUCH_ALLOW_MMX 1
#endif
#else
@ -126,18 +157,31 @@ namespace soundtouch
// 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
#ifdef SOUNDTOUCH_ALLOW_X86_OPTIMIZATIONS
// Allow SSE optimizations
#define SOUNDTOUCH_ALLOW_SSE 1
#endif
#endif // INTEGER_SAMPLES
#endif // SOUNDTOUCH_INTEGER_SAMPLES
};
// define ST_NO_EXCEPTION_HANDLING switch to disable throwing std exceptions:
// #define ST_NO_EXCEPTION_HANDLING 1
#ifdef ST_NO_EXCEPTION_HANDLING
// Exceptions disabled. Throw asserts instead if enabled.
#include <assert.h>
#define ST_THROW_RT_ERROR(x) {assert((const char *)x);}
#else
// use c++ standard exceptions
#include <stdexcept>
#define ST_THROW_RT_ERROR(x) {throw std::runtime_error(x);}
#endif
// When this #define is active, eliminates a clicking sound when the "rate" or "pitch"
// parameter setting crosses from value <1 to >=1 or vice versa during processing.
// Default is off as such crossover is untypical case and involves a slight sound
// quality compromise.
//#define SOUNDTOUCH_PREVENT_CLICK_AT_RATE_CROSSOVER 1
#endif

97
lib-src/soundtouch/include/SoundTouch.h
View File

@ -1,27 +1,27 @@
//////////////////////////////////////////////////////////////////////////////
///
/// SoundTouch - main class for tempo/pitch/rate adjusting routines.
/// 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
/// - 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
/// - 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
/// 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
///
/// - 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
///
/// - 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
/// 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.
@ -30,8 +30,8 @@
/// 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) 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.
///
@ -41,10 +41,10 @@
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2006-09-18 22:29:21 $
// File revision : $Revision: 1.5 $
// Last changed : $Date: 2012-12-28 21:32:59 +0200 (Fri, 28 Dec 2012) $
// File revision : $Revision: 4 $
//
// $Id: SoundTouch.h,v 1.5 2006-09-18 22:29:21 martynshaw Exp $
// $Id: SoundTouch.h 163 2012-12-28 19:32:59Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
@ -79,10 +79,10 @@ namespace soundtouch
{
/// Soundtouch library version string
#define SOUNDTOUCH_VERSION "1.3.1"
#define SOUNDTOUCH_VERSION "1.7.1"
/// SoundTouch library version id
#define SOUNDTOUCH_VERSION_ID 010301
#define SOUNDTOUCH_VERSION_ID (10701)
//
// Available setting IDs for the 'setSetting' & 'get_setting' functions:
@ -98,24 +98,49 @@ namespace soundtouch
/// 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.
/// 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.
/// 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.
/// 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
/// Call "getSetting" with this ID to query nominal average processing sequence
/// size in samples. This value tells approcimate value how many input samples
/// SoundTouch needs to gather before it does DSP processing run for the sample batch.
///
/// Notices:
/// - This is read-only parameter, i.e. setSetting ignores this parameter
/// - Returned value is approximate average value, exact processing batch
/// size may wary from time to time
/// - This parameter value is not constant but may change depending on
/// tempo/pitch/rate/samplerate settings.
#define SETTING_NOMINAL_INPUT_SEQUENCE 6
/// Call "getSetting" with this ID to query nominal average processing output
/// size in samples. This value tells approcimate value how many output samples
/// SoundTouch outputs once it does DSP processing run for a batch of input samples.
///
/// Notices:
/// - This is read-only parameter, i.e. setSetting ignores this parameter
/// - Returned value is approximate average value, exact processing batch
/// size may wary from time to time
/// - This parameter value is not constant but may change depending on
/// tempo/pitch/rate/samplerate settings.
#define SETTING_NOMINAL_OUTPUT_SEQUENCE 7
class SoundTouch : public FIFOProcessor
{
private:
@ -137,7 +162,7 @@ private:
/// Flag: Has sample rate been set?
BOOL bSrateSet;
/// Calculates effective rate & tempo valuescfrom 'virtualRate', 'virtualTempo' and
/// Calculates effective rate & tempo valuescfrom 'virtualRate', 'virtualTempo' and
/// 'virtualPitch' parameters.
void calcEffectiveRateAndTempo();
@ -181,7 +206,7 @@ public:
/// represent lower pitches, larger values higher pitch.
void setPitch(float newPitch);
/// Sets pitch change in octaves compared to the original pitch
/// Sets pitch change in octaves compared to the original pitch
/// (-1.00 .. +1.00)
void setPitchOctaves(float newPitch);
@ -221,18 +246,18 @@ public:
/// Changes a setting controlling the processing system behaviour. See the
/// 'SETTING_...' defines for available setting ID's.
///
///
/// \return 'TRUE' if the setting was succesfully changed
BOOL setSetting(uint settingId, ///< Setting ID number. see SETTING_... defines.
uint value ///< New setting value.
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.
uint getSetting(uint settingId ///< Setting ID number, see SETTING_... defines.
) const;
int getSetting(int settingId ///< Setting ID number, see SETTING_... defines.
) const;
/// Returns number of samples currently unprocessed.
virtual uint numUnprocessedSamples() const;
@ -242,7 +267,7 @@ public:
/// 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
/// - 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.

83
lib-src/soundtouch/include/soundtouch_config.h.in
View File

@ -1,83 +0,0 @@
/* include/soundtouch_config.h.in. Generated from configure.ac by autoheader. */
/* Use Float as Sample type */
#undef FLOAT_SAMPLES
/* Define to 1 if you have the <dlfcn.h> header file. */
#undef HAVE_DLFCN_H
/* Define to 1 if you have the <inttypes.h> header file. */
#undef HAVE_INTTYPES_H
/* Define to 1 if you have the `m' library (-lm). */
#undef HAVE_LIBM
/* Define to 1 if your system has a GNU libc compatible `malloc' function, and
to 0 otherwise. */
#undef HAVE_MALLOC
/* Define to 1 if you have the <memory.h> header file. */
#undef HAVE_MEMORY_H
/* Define to 1 if you have the <stdint.h> header file. */
#undef HAVE_STDINT_H
/* Define to 1 if you have the <stdlib.h> header file. */
#undef HAVE_STDLIB_H
/* Define to 1 if you have the <strings.h> header file. */
#undef HAVE_STRINGS_H
/* Define to 1 if you have the <string.h> header file. */
#undef HAVE_STRING_H
/* Define to 1 if you have the <sys/stat.h> header file. */
#undef HAVE_SYS_STAT_H
/* Define to 1 if you have the <sys/types.h> header file. */
#undef HAVE_SYS_TYPES_H
/* Define to 1 if you have the <unistd.h> header file. */
#undef HAVE_UNISTD_H
/* Use Integer as Sample type */
#undef INTEGER_SAMPLES
/* Name of package */
#undef PACKAGE
/* Define to the address where bug reports for this package should be sent. */
#undef PACKAGE_BUGREPORT
/* Define to the full name of this package. */
#undef PACKAGE_NAME
/* Define to the full name and version of this package. */
#undef PACKAGE_STRING
/* Define to the one symbol short name of this package. */
#undef PACKAGE_TARNAME
/* Define to the version of this package. */
#undef PACKAGE_VERSION
/* Define as the return type of signal handlers (`int' or `void'). */
#undef RETSIGTYPE
/* Define to 1 if you have the ANSI C header files. */
#undef STDC_HEADERS
/* Version number of package */
#undef VERSION
/* Define to empty if `const' does not conform to ANSI C. */
#undef const
/* Define to `__inline__' or `__inline' if that's what the C compiler
calls it, or to nothing if 'inline' is not supported under any name. */
#ifndef __cplusplus
#undef inline
#endif
/* Define to rpl_malloc if the replacement function should be used. */
#undef malloc

95
lib-src/soundtouch/make-win.bat
View File

@ -1,34 +1,61 @@
@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,v 1.3 2006-09-18 07:31:36 richardash1981 Exp $
@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
@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 142 2012-04-04 19:55:37Z oparviai $
@rem ****************************
@rem try first for VS6.0 support
@if "%MsDevDir%"=="" goto nomsdevdir
md bin
md lib
msdev source\SoundTouch\SoundTouch.dsw /MAKE ALL
msdev source\SoundStretch\SoundStretch.dsw /MAKE ALL
goto end
:nomsdevdir
@rem **********************************
@rem try with devenv for VS2003 support
@if "%DevEnvDir%"=="" goto nodevdir
md bin
md lib
devenv source\SoundTouch\SoundTouch.vcproj /upgrade
devenv source\SoundTouch\SoundTouch.vcproj /build debug
devenv source\SoundTouch\SoundTouch.vcproj /build release
devenv source\SoundStretch\SoundStretch.vcproj /upgrade
devenv source\SoundStretch\SoundStretch.vcproj /build debug
devenv source\SoundStretch\SoundStretch.vcproj /build release
devenv source\SoundTouchDll\SoundTouchDll.vcproj /upgrade
devenv source\SoundTouchDll\SoundTouchDll.vcproj /build debug
devenv source\SoundTouchDll\SoundTouchDll.vcproj /build release
@goto end
:nodevdir
@echo off
echo *********************************************************************
echo **
echo ** ERROR: Visual Studio path not set.
echo **
echo ** Run "vsvars32.bat" or "vcvars32.bat" from Visual Studio installation
echo ** directory, e.g. "\Program Files\Microsoft Visual Studio\VC98\Bin",
echo ** then try again.
echo **
echo *********************************************************************
:end

14
lib-src/soundtouch/soundtouch-1.4.pc.in Normal file
View File

@ -0,0 +1,14 @@
# This file is obsoleted but provided for backwards compatibility
# with legacy package environments
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

140
lib-src/soundtouch/soundtouch.m4
View File

@ -1,70 +1,70 @@
# m4 configure test script for the SoundTouch library
# (c)2003 David W. Durham
#
# $Id: soundtouch.m4,v 1.2 2006-09-18 07:31:36 richardash1981 Exp $
#
# 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)
])
# m4 configure test script for the SoundTouch library
# (c)2003 David W. Durham
#
# $Id: soundtouch.m4 20 2008-02-17 14:20:52Z 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)
])

0
lib-src/soundtouch/soundtouch-1.0.pc.in → lib-src/soundtouch/soundtouch.pc.in
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159
lib-src/soundtouch/soundtouch/BPMDetect.h
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@ -1,159 +0,0 @@
////////////////////////////////////////////////////////////////////////////////
///
/// 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: 2006-09-18 07:39:15 $
// File revision : $Revision: 1.2 $
//
// $Id: BPMDetect.h,v 1.2 2006-09-18 07:39:15 richardash1981 Exp $
//
////////////////////////////////////////////////////////////////////////////////
//
// 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_

174
lib-src/soundtouch/soundtouch/FIFOSampleBuffer.h
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@ -1,174 +0,0 @@
////////////////////////////////////////////////////////////////////////////////
///
/// 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: 2006-09-18 07:39:15 $
// File revision : $Revision: 1.2 $
//
// $Id: FIFOSampleBuffer.h,v 1.2 2006-09-18 07:39:15 richardash1981 Exp $
//
////////////////////////////////////////////////////////////////////////////////
//
// 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(const uint capacityRequirement);
/// Returns current capacity.
uint getCapacity() const;
public:
/// Constructor
FIFOSampleBuffer(uint 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(uint numChannels);
/// Returns nonzero if there aren't any samples available for outputting.
virtual int isEmpty() const;
/// Clears all the samples.
virtual void clear();
};
}
#endif

217
lib-src/soundtouch/soundtouch/FIFOSamplePipe.h
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@ -1,217 +0,0 @@
////////////////////////////////////////////////////////////////////////////////
///
/// '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: 2006-09-18 07:39:15 $
// File revision : $Revision: 1.2 $
//
// $Id: FIFOSamplePipe.h,v 1.2 2006-09-18 07:39:15 richardash1981 Exp $
//
////////////////////////////////////////////////////////////////////////////////
//
// 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

411
lib-src/soundtouch/soundtouch/Makefile
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@ -1,411 +0,0 @@
# Makefile.in generated by automake 1.9.6 from Makefile.am.
# include/Makefile. Generated from Makefile.in by configure.
# Copyright (C) 1994, 1995, 1996, 1997, 1998, 1999, 2000, 2001, 2002,
# 2003, 2004, 2005 Free Software Foundation, Inc.
# This Makefile.in is free software; the Free Software Foundation
# gives unlimited permission to copy and/or distribute it,
# with or without modifications, as long as this notice is preserved.
# This program is distributed in the hope that it will be useful,
# but WITHOUT ANY WARRANTY, to the extent permitted by law; without
# even the implied warranty of MERCHANTABILITY or FITNESS FOR A
# PARTICULAR PURPOSE.
srcdir = .
top_srcdir = ..
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pkglibdir = $(libdir)/soundtouch
pkgincludedir = $(includedir)/soundtouch
top_builddir = ..
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install_sh_SCRIPT = $(install_sh) -c
INSTALL_HEADER = $(INSTALL_DATA)
transform = $(program_transform_name)
NORMAL_INSTALL = :
PRE_INSTALL = :
POST_INSTALL = :
NORMAL_UNINSTALL = :
PRE_UNINSTALL = :
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CPPFLAGS =
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LDFLAGS =
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build_alias =
build_cpu = i686
build_os = linux-gnu
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datadir = ${prefix}/share
exec_prefix = ${prefix}
host = i686-pc-linux-gnu
host_alias =
host_cpu = i686
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libdir = ${exec_prefix}/lib
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localstatedir = ${prefix}/var
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oldincludedir = /usr/include
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## Process this file with automake to create Makefile.in
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## $Id: Makefile.am,v 1.3 2006-09-27 06:19:52 llucius Exp $
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## 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
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html-am info info-am install install-am install-data \
install-data-am install-exec install-exec-am install-info \
install-info-am install-man install-pkgincludeHEADERS \
install-strip installcheck installcheck-am installdirs \
maintainer-clean maintainer-clean-generic mostlyclean \
mostlyclean-generic mostlyclean-libtool pdf pdf-am ps ps-am \
tags uninstall uninstall-am uninstall-info-am \
uninstall-pkgincludeHEADERS
# Tell versions [3.59,3.63) of GNU make to not export all variables.
# Otherwise a system limit (for SysV at least) may be exceeded.
.NOEXPORT:

7
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This directory with links to the contents of ../include/ is here
because when installed these files go into $PREFIX/include/soundtouch
hence the #include statements have to be #include <soundtouch/SoundTouch.h>
etc. There is then no easy way to get that code to compile against this
(uninstalled) copy of soundtouch. Hence this dir so we can just put
-I ../lib-src/soundtouch in the CXXFLAGS and find the headers at
lib-src/soundtouch/soundtouch/Soundtouch.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: 2006-09-18 07:39:15 $
// File revision : $Revision: 1.2 $
//
// $Id: STTypes.h,v 1.2 2006-09-18 07:39:15 richardash1981 Exp $
//
////////////////////////////////////////////////////////////////////////////////
//
// 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: 2006-09-18 07:39:15 $
// File revision : $Revision: 1.2 $
//
// $Id: SoundTouch.h,v 1.2 2006-09-18 07:39:15 richardash1981 Exp $
//
////////////////////////////////////////////////////////////////////////////////
//
// 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(uint settingId, ///< Setting ID number. see SETTING_... defines.
uint 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.
uint getSetting(uint 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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/* include/soundtouch_config.h. Generated by configure. */
/* include/soundtouch_config.h.in. Generated from configure.ac by autoheader. */
/* Use Float as Sample type */
#define FLOAT_SAMPLES 1
/* Define to 1 if you have the <dlfcn.h> header file. */
#define HAVE_DLFCN_H 1
/* Define to 1 if you have the <inttypes.h> header file. */
#define HAVE_INTTYPES_H 1
/* Define to 1 if you have the `m' library (-lm). */
#define HAVE_LIBM 1
/* Define to 1 if your system has a GNU libc compatible `malloc' function, and
to 0 otherwise. */
#define HAVE_MALLOC 1
/* Define to 1 if you have the <memory.h> header file. */
#define HAVE_MEMORY_H 1
/* Define to 1 if you have the <stdint.h> header file. */
#define HAVE_STDINT_H 1
/* Define to 1 if you have the <stdlib.h> header file. */
#define HAVE_STDLIB_H 1
/* Define to 1 if you have the <strings.h> header file. */
#define HAVE_STRINGS_H 1
/* Define to 1 if you have the <string.h> header file. */
#define HAVE_STRING_H 1
/* Define to 1 if you have the <sys/stat.h> header file. */
#define HAVE_SYS_STAT_H 1
/* Define to 1 if you have the <sys/types.h> header file. */
#define HAVE_SYS_TYPES_H 1
/* Define to 1 if you have the <unistd.h> header file. */
#define HAVE_UNISTD_H 1
/* Use Integer as Sample type */
/* #undef INTEGER_SAMPLES */
/* Name of package */
#define PACKAGE "soundtouch"
/* Define to the address where bug reports for this package should be sent. */
#define PACKAGE_BUGREPORT "http://www.surina.net/soundtouch"
/* Define to the full name of this package. */
#define PACKAGE_NAME "SoundTouch"
/* Define to the full name and version of this package. */
#define PACKAGE_STRING "SoundTouch 1.3.1"
/* Define to the one symbol short name of this package. */
#define PACKAGE_TARNAME "soundtouch"
/* Define to the version of this package. */
#define PACKAGE_VERSION "1.3.1"
/* Define as the return type of signal handlers (`int' or `void'). */
#define RETSIGTYPE void
/* Define to 1 if you have the ANSI C header files. */
#define STDC_HEADERS 1
/* Version number of package */
#define VERSION "1.3.1"
/* Define to empty if `const' does not conform to ANSI C. */
/* #undef const */
/* Define to `__inline__' or `__inline' if that's what the C compiler
calls it, or to nothing if 'inline' is not supported under any name. */
#ifndef __cplusplus
/* #undef inline */
#endif
/* Define to rpl_malloc if the replacement function should be used. */
/* #undef malloc */

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/* include/soundtouch_config.h.in. Generated from configure.ac by autoheader. */
/* Use Float as Sample type */
#undef FLOAT_SAMPLES
/* Define to 1 if you have the <dlfcn.h> header file. */
#undef HAVE_DLFCN_H
/* Define to 1 if you have the <inttypes.h> header file. */
#undef HAVE_INTTYPES_H
/* Define to 1 if you have the `m' library (-lm). */
#undef HAVE_LIBM
/* Define to 1 if your system has a GNU libc compatible `malloc' function, and
to 0 otherwise. */
#undef HAVE_MALLOC
/* Define to 1 if you have the <memory.h> header file. */
#undef HAVE_MEMORY_H
/* Define to 1 if you have the <stdint.h> header file. */
#undef HAVE_STDINT_H
/* Define to 1 if you have the <stdlib.h> header file. */
#undef HAVE_STDLIB_H
/* Define to 1 if you have the <strings.h> header file. */
#undef HAVE_STRINGS_H
/* Define to 1 if you have the <string.h> header file. */
#undef HAVE_STRING_H
/* Define to 1 if you have the <sys/stat.h> header file. */
#undef HAVE_SYS_STAT_H
/* Define to 1 if you have the <sys/types.h> header file. */
#undef HAVE_SYS_TYPES_H
/* Define to 1 if you have the <unistd.h> header file. */
#undef HAVE_UNISTD_H
/* Use Integer as Sample type */
#undef INTEGER_SAMPLES
/* Name of package */
#undef PACKAGE
/* Define to the address where bug reports for this package should be sent. */
#undef PACKAGE_BUGREPORT
/* Define to the full name of this package. */
#undef PACKAGE_NAME
/* Define to the full name and version of this package. */
#undef PACKAGE_STRING
/* Define to the one symbol short name of this package. */
#undef PACKAGE_TARNAME
/* Define to the version of this package. */
#undef PACKAGE_VERSION
/* Define as the return type of signal handlers (`int' or `void'). */
#undef RETSIGTYPE
/* Define to 1 if you have the ANSI C header files. */
#undef STDC_HEADERS
/* Version number of package */
#undef VERSION
/* Define to empty if `const' does not conform to ANSI C. */
#undef const
/* Define to `__inline__' or `__inline' if that's what the C compiler
calls it, or to nothing if 'inline' is not supported under any name. */
#ifndef __cplusplus
#undef inline
#endif
/* Define to rpl_malloc if the replacement function should be used. */
#undef malloc

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118
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
<title>SoundTouch in Android</title>
<meta http-equiv="Content-Type"
content="text/html; charset=windows-1252">
<meta http-equiv="Content-Language" content="en-us">
<meta name="author" content="Olli Parviainen">
<meta name="description"
content="Readme file for SoundTouch library Android compilation">
<style> <!-- .normal { font-family: Arial }
--></style>
</head>
<body class="normal">
<hr>
<h1>SoundTouch in Android</h1>
<hr>
<h2>Compiling SoundTouch for Android</h2>
<p>SoundTouch source code package contains &quot;Android-lib&quot; example project that compiles SoundTouch
source codes into Android native library, and gives an example of JNI interface
for invoking
the native SoundTouch routines from an Android application written in Java.</p>
<p style="font-weight: 700">Software prerequisites:</p>
<ul>
<li>Android SDK environment for developing your own Android application. Visit the <a href="http://developer.android.com/index.html">Android developers' site</a>
for more information about the Android SDK and developing Android applications.</li>
<li>Android NDK compiler kit for compiling native library binaries. The Android NDK
is <a href="http://developer.android.com/tools/sdk/ndk/index.html">
available for download</a> at the Android developer tools site.</li>
<li>In case you're working in Windows environment, install
<a href="http://cygwin.com/install.html">
Cygwin</a> to run the Android NDK/SDK compiler scripts</li>
<li>Latest SoundTouch source code package available at <a href="http://soundtouch.surina.net/sourcecode.html">
soundtouch.surina.net</a>.</li>
</ul>
<p><b>Hint: </b>As installing and configuring all the components for an Android SDK/NDK
environment requires fair effort, it&#39;s good idea to create a dedicated Virtual
Machine environment for the Android development environment installation.
Having the Android developer environment setup in dedicated Virtual Machine
allows keeping all these settings isolated from your other PC operations, and
eases taking backup snapshots of your full development environment.</p>
<p><b>Compiling</b></p>
<p>
To compile the SoundTouch library source codes into an Android native library,
open Cygwin/bash shell, go to directory <b>&quot;soundtouch/source/Android-lib/jni&quot;</b> and invoke the NDK
compiler with following command:</p>
<pre> $NDK/ndk-build</pre>
<p>This will build the ARMv5 and ARMv7 versions of SoundTouch library (including
also the example JNI
interface, see below) into the &quot;libs&quot; subdirectory.</p>
<p>Notice that to allow Cygwin/bash to locate the NDK compile scripts, you
need to define the location of the NDK installation defined in environment
variable &quot;NDK&quot;. That's easiest done by adding the NDK path definition at end of
your <b>~/.bash_profile</b> file, for instance as follows:</p>
<pre> NDK=/cygdrive/d/Android/android-ndk-r6</pre>
<hr />
<h2>
Android floating-point performance considerations</h2>
<p>
Default build target for
Android NDK is ARMv5 CPU generation, as that works in
all ARM-based Android devices.<p>
This has a pitfall though: For ideal sound quality SoundTouch should be compiled
to use floating-point algorithms, however, all low-end Android devices do not
have floating-point hardware in their CPUs, and hence the default ARMv5 compilation uses software-emulation for floating-point calculations instead of
hardware floating-point to allow running the binary executables also in low-end devices.<p>
The floating point software-emulation is however several tens of times slower
than real hardware-level floating-point calculations, making
floating-point-intensive applications such as SoundTouch infeasible with low-end
devices.<p>
As workaround, the SoundTouch Android compilation builds two separate versions
of the library:<ul>
<li>ARMv5 version that compiles SoundTouch using integer algorithm version. The integer
algorithm version compromises the sound quality but provides good performance also
with low-end
devices without hardware floating-point support in the CPU level.</li>
<li>ARMv7 version that compiles SoundTouch using hardware floating-point algorithms.
These algorithms provide ideal sound quality yet do not work in simpler CPU
models.</li>
</ul>
<p>
These two library compilations are already defined in file &quot;<b>jni/Application.mk</b>&quot;
so that these two separate library targets are automatically built under the &quot;<b>libs</b>&quot;
directory. As far as you include both these compiled library versions into your
application delivery, the Android devices can automatically select the right
library version based on the available device&#39;s capabilities.<p>
Please yet be aware that depending on capabilities of the Android devices you
will need to provide the SoundTouch routines with samples in either integer or
floating-point format, so build your interface routines to take this into
account.<hr />
<h2>
Calling SoundTouch native routines from Android application</h2>
<p>The NDK tools build the SoundTouch c++ routines into a native binary library, while
Android applications are written in Java language. To call the SoundTouch and other c/c++
routines from an Android java application code, you'll need to use Java Native
Interface (JNI).</p>
<p>
The SoundTouch source code package provides source code example how to
use JNI to call native c++ routines from a Java class through the following
source code file pair:<ul>
<li><b>Android-lib/jni/soundtouch-jni.cpp</b>: This file contains c/c++ routine that
calls SoundTouch library routine to return the library version string to the main
Android application. The NDK compiles this file along with the SoundTouch
routines into the native binary library.</li>
<li><b>Android-lib/src/net/surina/soundtouch/SoundTouch.java</b>: This file provides
a Java interface class to load the native library and to invoke the native routine implemented in
the file <b>soundtouch-jni.cpp</b></li>
</ul>
<p>
Feel free to examine and extend the provided cpp/java source code example file pair to
implement and integrate the desired SoundTouch library capabilities into your Android application.</p>
<hr />
<p style="text-align: center"><i>Copyright &copy; Olli Parviainen</i></p>
<!--
$Id: README-SoundTouch-Android.html 165 2012-12-28 19:55:23Z oparviai $
-->
</body>
</html>

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# Copyright (C) 2010 The Android Open Source Project
#
# Licensed under the Apache License, Version 2.0 (the "License");
# you may not use this file except in compliance with the License.
# You may obtain a copy of the License at
#
# http://www.apache.org/licenses/LICENSE-2.0
#
# Unless required by applicable law or agreed to in writing, software
# distributed under the License is distributed on an "AS IS" BASIS,
# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
# See the License for the specific language governing permissions and
# limitations under the License.
#
# $Id: Android.mk 165 2012-12-28 19:55:23Z oparviai $
LOCAL_PATH := $(call my-dir)
include $(CLEAR_VARS)
# *** Remember: Change -O0 into -O2 in add-applications.mk ***
LOCAL_MODULE := soundtouch
LOCAL_SRC_FILES := soundtouch-jni.cpp ../../SoundTouch/AAFilter.cpp ../../SoundTouch/FIFOSampleBuffer.cpp \
../../SoundTouch/FIRFilter.cpp ../../SoundTouch/cpu_detect_x86.cpp \
../../SoundTouch/RateTransposer.cpp ../../SoundTouch/SoundTouch.cpp \
../../SoundTouch/TDStretch.cpp ../../SoundTouch/BPMDetect.cpp ../../SoundTouch/PeakFinder.cpp
# for native audio
LOCAL_LDLIBS += -lgcc
# --whole-archive -lgcc
# for logging
LOCAL_LDLIBS += -llog
# for native asset manager
#LOCAL_LDLIBS += -landroid
# don't export all symbols
# added "-marm" switch to use arm instruction set instead of thumb for improved calculation performance.
LOCAL_CFLAGS += -Wall -fvisibility=hidden -I ../../../include -D ST_NO_EXCEPTION_HANDLING -fdata-sections -ffunction-sections -marm
include $(BUILD_SHARED_LIBRARY)

7
lib-src/soundtouch/source/Android-lib/jni/Application.mk Normal file
View File

@ -0,0 +1,7 @@
# $Id: Application.mk 165 2012-12-28 19:55:23Z oparviai $
#
# Build both ARMv5TE and ARMv7-A machine code.
#
APP_ABI := armeabi-v7a armeabi
APP_OPTIM := release

43
lib-src/soundtouch/source/Android-lib/jni/soundtouch-jni.cpp Normal file
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@ -0,0 +1,43 @@
////////////////////////////////////////////////////////////////////////////////
///
/// Example Interface class for SoundTouch native compilation
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// WWW : http://www.surina.net
///
////////////////////////////////////////////////////////////////////////////////
//
// $Id: soundtouch-jni.cpp 165 2012-12-28 19:55:23Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
#include <jni.h>
#include <android/log.h>
//#include <string.h>
//#include <stdio.h>
//#include <dlfcn.h>
#include "../../../include/SoundTouch.h"
//#include "TimeShiftEffect.h"
#define LOGV(...) __android_log_print((int)ANDROID_LOG_INFO, "SOUNDTOUCH", __VA_ARGS__)
//#define LOGV(...)
#define DLL_PUBLIC __attribute__ ((visibility ("default")))
using namespace soundtouch;
extern "C" DLL_PUBLIC jstring Java_net_surina_soundtouch_getVersionString(JNIEnv *env, jobject thiz)
{
const char *verStr;
LOGV("JNI call soundtouch.getVersionString");
// Call example SoundTouch routine
verStr = SoundTouch::getVersionString();
// return version as string
return env->NewStringUTF(verStr);
}

29
lib-src/soundtouch/source/Android-lib/src/net/surina/soundtouch/SoundTouch.java Normal file
View File

@ -0,0 +1,29 @@
////////////////////////////////////////////////////////////////////////////////
///
/// Example class that invokes native SoundTouch routines through the JNI
/// interface.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// WWW : http://www.surina.net
///
////////////////////////////////////////////////////////////////////////////////
//
// $Id: SoundTouch.java 165 2012-12-28 19:55:23Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
package net.surina.soundtouch;
public final class SoundTouch
{
// Native interface function that returns SoundTouch version string.
// This invokes the native c++ routine defined in "soundtouch-jni.cpp".
public native final String getVersionString();
// Load the native library upon startup
static
{
System.loadLibrary("soundtouch");
}
}

4
lib-src/soundtouch/source/Makefile.am
View File

@ -1,6 +1,6 @@
## Process this file with automake to create Makefile.in
##
## $Id: Makefile.am,v 1.3 2006-09-27 06:19:52 llucius Exp $
## $Id: Makefile.am 38 2008-12-25 17:00:23Z oparviai $
##
## Copyright (C) 2003 - David W. Durham
##
@ -21,7 +21,7 @@
include $(top_srcdir)/config/am_include.mk
SUBDIRS=SoundTouch example
SUBDIRS=SoundTouch SoundStretch
# set to something if you want other stuff to be included in the distribution tarball
#EXTRA_DIST=

488
lib-src/soundtouch/source/Makefile.in
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@ -1,488 +0,0 @@
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14
lib-src/soundtouch/source/example/SoundStretch/Makefile.am → lib-src/soundtouch/source/SoundStretch/Makefile.am
View File

@ -1,6 +1,6 @@
## Process this file with automake to create Makefile.in
##
## $Id: Makefile.am,v 1.3 2006-09-27 06:19:53 llucius Exp $
## $Id: Makefile.am 128 2011-07-17 10:59:56Z oparviai $
##
## Copyright (C) 2003 - David W. Durham
##
@ -31,7 +31,7 @@ bin_PROGRAMS=soundstretch
noinst_HEADERS=RunParameters.h WavFile.h
# extra files to include in distrubution tarball
EXTRA_DIST=soundstretch.dsp soundstretch.dsw
EXTRA_DIST=soundstretch.dsp soundstretch.dsw soundstretch.sln soundstretch.vcproj
## 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.
@ -41,14 +41,14 @@ soundstretch_SOURCES=main.cpp RunParameters.cpp WavFile.cpp
## 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
soundstretch_LDADD=../SoundTouch/libSoundTouch.la -lm
## linker flags
soundstretch_LDFLAGS=-s
## linker flags.
# OP 2011-7-17 Linker flags disabled to prevent stripping symbols by default
# soundstretch_LDFLAGS=-s
## additional compiler flags
soundstretch_CXXFLAGS=-O3 -I../bpm
soundstretch_CXXFLAGS=-O3
#clean-local:
# -rm -f additional-files-to-remove-on-make-clean

594
lib-src/soundtouch/source/example/SoundStretch/RunParameters.cpp → lib-src/soundtouch/source/SoundStretch/RunParameters.cpp
View File

@ -1,293 +1,301 @@
////////////////////////////////////////////////////////////////////////////////
///
/// 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: 2006-09-18 07:31:47 $
// File revision : $Revision: 1.2 $
//
// $Id: RunParameters.cpp,v 1.2 2006-09-18 07:31:47 richardash1981 Exp $
//
////////////////////////////////////////////////////////////////////////////////
//
// 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);
}
}
////////////////////////////////////////////////////////////////////////////////
///
/// 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: 2011-09-02 21:56:11 +0300 (Fri, 02 Sep 2011) $
// File revision : $Revision: 4 $
//
// $Id: RunParameters.cpp 131 2011-09-02 18:56:11Z 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
//
////////////////////////////////////////////////////////////////////////////////
#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 version 2.1 as published by the Free Software Foundation.\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 infilename outfilename [switches]\n"
"\n"
"To use standard input/output pipes, give 'stdin' and 'stdout' as filenames.\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"
" -speech : Tune algorithm for speech processing (default is for music)\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 * const 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;
ST_THROW_RT_ERROR(msg.c_str());
}
inFileName = NULL;
outFileName = NULL;
tempoDelta = 0;
pitchDelta = 0;
rateDelta = 0;
quick = 0;
noAntiAlias = 0;
goalBPM = 0;
speech = FALSE;
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;
ST_THROW_RT_ERROR(msg.c_str());
}
void RunParameters::throwLicense() const
{
ST_THROW_RT_ERROR(licenseText);
}
float RunParameters::parseSwitchValue(const string &str) const
{
int pos;
pos = (int)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 (const 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;
case 's' :
// switch '-speech'
speech = TRUE;
break;
default:
// unknown switch
throwIllegalParamExp(str);
}
}

143
lib-src/soundtouch/source/example/SoundStretch/RunParameters.h → lib-src/soundtouch/source/SoundStretch/RunParameters.h
View File

@ -1,71 +1,72 @@
////////////////////////////////////////////////////////////////////////////////
///
/// 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: 2006-09-18 07:31:47 $
// File revision : $Revision: 1.2 $
//
// $Id: RunParameters.h,v 1.2 2006-09-18 07:31:47 richardash1981 Exp $
//
////////////////////////////////////////////////////////////////////////////////
//
// 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
////////////////////////////////////////////////////////////////////////////////
///
/// 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: 2009-05-17 19:48:30 +0300 (Sun, 17 May 2009) $
// File revision : $Revision: 4 $
//
// $Id: RunParameters.h 72 2009-05-17 16:48:30Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#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;
BOOL speech;
RunParameters(const int nParams, const char * const paramStr[]);
};
#endif

951
lib-src/soundtouch/source/SoundStretch/WavFile.cpp Normal file
View File

@ -0,0 +1,951 @@
////////////////////////////////////////////////////////////////////////////////
///
/// 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: 2012-09-01 11:03:26 +0300 (Sat, 01 Sep 2012) $
// File revision : $Revision: 4 $
//
// $Id: WavFile.cpp 154 2012-09-01 08:03:26Z 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 <string>
#include <sstream>
#include <cstring>
#include <assert.h>
#include <limits.h>
#include "WavFile.h"
#include "STTypes.h"
using namespace std;
static const char riffStr[] = "RIFF";
static const char waveStr[] = "WAVE";
static const char fmtStr[] = "fmt ";
static const 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 int _swap32(int &dwData)
{
dwData = ((dwData >> 24) & 0x000000FF) |
((dwData >> 8) & 0x0000FF00) |
((dwData << 8) & 0x00FF0000) |
((dwData << 24) & 0xFF000000);
return dwData;
}
// helper-function to swap byte-order of 16bit integer
static inline short _swap16(short &wData)
{
wData = ((wData >> 8) & 0x00FF) |
((wData << 8) & 0xFF00);
return wData;
}
// helper-function to swap byte-order of buffer of 16bit integers
static inline void _swap16Buffer(short *pData, int numWords)
{
int i;
for (i = 0; i < numWords; i ++)
{
pData[i] = _swap16(pData[i]);
}
}
#else // BIG_ENDIAN
// little-endian CPU, WAV file is ok as such
// dummy helper-function
static inline int _swap32(int &dwData)
{
// do nothing
return dwData;
}
// dummy helper-function
static inline short _swap16(short &wData)
{
// do nothing
return wData;
}
// dummy helper-function
static inline void _swap16Buffer(short *pData, int numBytes)
{
// do nothing
}
#endif // BIG_ENDIAN
//////////////////////////////////////////////////////////////////////////////
//
// Class WavFileBase
//
WavFileBase::WavFileBase()
{
convBuff = NULL;
convBuffSize = 0;
}
WavFileBase::~WavFileBase()
{
delete[] convBuff;
convBuffSize = 0;
}
/// Get pointer to conversion buffer of at min. given size
void *WavFileBase::getConvBuffer(int sizeBytes)
{
if (convBuffSize < sizeBytes)
{
delete[] convBuff;
convBuffSize = (sizeBytes + 15) & -8; // round up to following 8-byte bounday
convBuff = new char[convBuffSize];
}
return convBuff;
}
//////////////////////////////////////////////////////////////////////////////
//
// Class WavInFile
//
WavInFile::WavInFile(const char *fileName)
{
// 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.";
ST_THROW_RT_ERROR(msg.c_str());
}
init();
}
WavInFile::WavInFile(FILE *file)
{
// Try to open the file for reading
fptr = file;
if (!file)
{
// didn't succeed
string msg = "Error : Unable to access input stream for reading";
ST_THROW_RT_ERROR(msg.c_str());
}
init();
}
/// Init the WAV file stream
void WavInFile::init()
{
int hdrsOk;
// assume file stream is already open
assert(fptr);
// Read the file headers
hdrsOk = readWavHeaders();
if (hdrsOk != 0)
{
// Something didn't match in the wav file headers
string msg = "Input file is corrupt or not a WAV file";
ST_THROW_RT_ERROR(msg.c_str());
}
/* Ignore 'fixed' field value as 32bit signed linear data can have other value than 1.
if (header.format.fixed != 1)
{
string msg = "Input file uses unsupported encoding.";
ST_THROW_RT_ERROR(msg.c_str());
}
*/
dataRead = 0;
}
WavInFile::~WavInFile()
{
if (fptr) fclose(fptr);
fptr = NULL;
}
void WavInFile::rewind()
{
int hdrsOk;
fseek(fptr, 0, SEEK_SET);
hdrsOk = readWavHeaders();
assert(hdrsOk == 0);
dataRead = 0;
}
int WavInFile::checkCharTags() const
{
// 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(unsigned char *buffer, int maxElems)
{
int numBytes;
uint afterDataRead;
// ensure it's 8 bit format
if (header.format.bits_per_sample != 8)
{
ST_THROW_RT_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 = (int)header.data.data_len - (int)dataRead;
assert(numBytes >= 0);
}
assert(buffer);
numBytes = (int)fread(buffer, 1, numBytes, fptr);
dataRead += numBytes;
return numBytes;
}
int WavInFile::read(short *buffer, int maxElems)
{
unsigned int afterDataRead;
int numBytes;
int numElems;
assert(buffer);
switch (header.format.bits_per_sample)
{
case 8:
{
// 8 bit format
unsigned char *temp = (unsigned char*)getConvBuffer(maxElems);
int i;
numElems = read(temp, maxElems);
// convert from 8 to 16 bit
for (i = 0; i < numElems; i ++)
{
buffer[i] = (short)(((short)temp[i] - 128) * 256);
}
break;
}
case 16:
{
// 16 bit format
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 = (int)header.data.data_len - (int)dataRead;
assert(numBytes >= 0);
}
numBytes = (int)fread(buffer, 1, numBytes, fptr);
dataRead += numBytes;
numElems = numBytes / 2;
// 16bit samples, swap byte order if necessary
_swap16Buffer((short *)buffer, numElems);
break;
}
default:
{
stringstream ss;
ss << "\nOnly 8/16 bit sample WAV files supported in integer compilation. Can't open WAV file with ";
ss << (int)header.format.bits_per_sample;
ss << " bit sample format. ";
ST_THROW_RT_ERROR(ss.str().c_str());
}
};
return numElems;
}
/// Read data in float format. Notice that when reading in float format
/// 8/16/24/32 bit sample formats are supported
int WavInFile::read(float *buffer, int maxElems)
{
unsigned int afterDataRead;
int numBytes;
int numElems;
int bytesPerSample;
assert(buffer);
bytesPerSample = header.format.bits_per_sample / 8;
if ((bytesPerSample < 1) || (bytesPerSample > 4))
{
stringstream ss;
ss << "\nOnly 8/16/24/32 bit sample WAV files supported. Can't open WAV file with ";
ss << (int)header.format.bits_per_sample;
ss << " bit sample format. ";
ST_THROW_RT_ERROR(ss.str().c_str());
}
numBytes = maxElems * bytesPerSample;
afterDataRead = dataRead + numBytes;
if (afterDataRead > header.data.data_len)
{
// Don't read more samples than are marked available in header
numBytes = (int)header.data.data_len - (int)dataRead;
assert(numBytes >= 0);
}
// read raw data into temporary buffer
char *temp = (char*)getConvBuffer(numBytes);
numBytes = (int)fread(temp, 1, numBytes, fptr);
dataRead += numBytes;
numElems = numBytes / bytesPerSample;
// swap byte ordert & convert to float, depending on sample format
switch (bytesPerSample)
{
case 1:
{
unsigned char *temp2 = (unsigned char*)temp;
double conv = 1.0 / 128.0;
for (int i = 0; i < numElems; i ++)
{
buffer[i] = (float)(temp2[i] * conv - 1.0);
}
break;
}
case 2:
{
short *temp2 = (short*)temp;
double conv = 1.0 / 32768.0;
for (int i = 0; i < numElems; i ++)
{
short value = temp2[i];
buffer[i] = (float)(_swap16(value) * conv);
}
break;
}
case 3:
{
char *temp2 = (char *)temp;
double conv = 1.0 / 8388608.0;
for (int i = 0; i < numElems; i ++)
{
int value = *((int*)temp2);
value = _swap32(value) & 0x00ffffff; // take 24 bits
value |= (value & 0x00800000) ? 0xff000000 : 0; // extend minus sign bits
buffer[i] = (float)(value * conv);
temp2 += 3;
}
break;
}
case 4:
{
int *temp2 = (int *)temp;
double conv = 1.0 / 2147483648.0;
assert(sizeof(int) == 4);
for (int i = 0; i < numElems; i ++)
{
int value = temp2[i];
buffer[i] = (float)(_swap32(value) * conv);
}
break;
}
}
return numElems;
}
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));
}
// 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(const char *str)
{
char c;
c = str[0];
while (c)
{
if (isAlpha(c) == 0) return 0;
str ++;
c = str[0];
}
return 1;
}
int WavInFile::readRIFFBlock()
{
if (fread(&(header.riff), sizeof(WavRiff), 1, fptr) != 1) return -1;
// swap 32bit data byte order if necessary
_swap32((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
if (fread(label, 1, 4, fptr) !=4) return -1;
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
if (fread(&nLen, sizeof(int), 1, fptr) != 1) return -1;
// swap byte order if necessary
_swap32(nLen); // int format_len;
header.format.format_len = nLen;
// calculate how much length differs from expected
nDump = nLen - ((int)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
if (fread(&(header.format.fixed), nLen, 1, fptr) != 1) return -1;
// swap byte order if necessary
_swap16(header.format.fixed); // short int fixed;
_swap16(header.format.channel_number); // short int channel_number;
_swap32((int &)header.format.sample_rate); // int sample_rate;
_swap32((int &)header.format.byte_rate); // int byte_rate;
_swap16(header.format.byte_per_sample); // short int byte_per_sample;
_swap16(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);
if (fread(&(header.data.data_len), sizeof(uint), 1, fptr) != 1) return -1;
// swap byte order if necessary
_swap32((int &)header.data.data_len);
return 1;
}
else
{
uint len, i;
uint temp;
// unknown block
// read length
if (fread(&len, sizeof(len), 1, fptr) != 1) return -1;
// scan through the block
for (i = 0; i < len; i ++)
{
if (fread(&temp, 1, 1, fptr) != 1) return -1;
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
{
if (header.format.byte_per_sample == 0) return 0;
return header.data.data_len / (unsigned short)header.format.byte_per_sample;
}
uint WavInFile::getLengthMS() const
{
double numSamples;
double sampleRate;
numSamples = (double)getNumSamples();
sampleRate = (double)getSampleRate();
return (uint)(1000.0 * numSamples / sampleRate + 0.5);
}
/// Returns how many milliseconds of audio have so far been read from the file
uint WavInFile::getElapsedMS() const
{
return (uint)(1000.0 * (double)dataRead / (double)header.format.byte_rate);
}
//////////////////////////////////////////////////////////////////////////////
//
// 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;
ST_THROW_RT_ERROR(msg.c_str());
}
fillInHeader(sampleRate, bits, channels);
writeHeader();
}
WavOutFile::WavOutFile(FILE *file, int sampleRate, int bits, int channels)
{
bytesWritten = 0;
fptr = file;
if (fptr == NULL)
{
string msg = "Error : Unable to access output file stream.";
ST_THROW_RT_ERROR(msg.c_str());
}
fillInHeader(sampleRate, bits, channels);
writeHeader();
}
WavOutFile::~WavOutFile()
{
finishHeader();
if (fptr) fclose(fptr);
fptr = NULL;
}
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 = (int)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 * (int)sampleRate;
header.format.sample_rate = (int)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;
int res;
// swap byte order if necessary
hdrTemp = header;
_swap32((int &)hdrTemp.riff.package_len);
_swap32((int &)hdrTemp.format.format_len);
_swap16((short &)hdrTemp.format.fixed);
_swap16((short &)hdrTemp.format.channel_number);
_swap32((int &)hdrTemp.format.sample_rate);
_swap32((int &)hdrTemp.format.byte_rate);
_swap16((short &)hdrTemp.format.byte_per_sample);
_swap16((short &)hdrTemp.format.bits_per_sample);
_swap32((int &)hdrTemp.data.data_len);
// write the supplemented header in the beginning of the file
fseek(fptr, 0, SEEK_SET);
res = (int)fwrite(&hdrTemp, sizeof(hdrTemp), 1, fptr);
if (res != 1)
{
ST_THROW_RT_ERROR("Error while writing to a wav file.");
}
// jump back to the end of the file
fseek(fptr, 0, SEEK_END);
}
void WavOutFile::write(const unsigned char *buffer, int numElems)
{
int res;
if (header.format.bits_per_sample != 8)
{
ST_THROW_RT_ERROR("Error: WavOutFile::write(const char*, int) accepts only 8bit samples.");
}
assert(sizeof(char) == 1);
res = (int)fwrite(buffer, 1, numElems, fptr);
if (res != numElems)
{
ST_THROW_RT_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
switch (header.format.bits_per_sample)
{
case 8:
{
int i;
unsigned char *temp = (unsigned char *)getConvBuffer(numElems);
// convert from 16bit format to 8bit format
for (i = 0; i < numElems; i ++)
{
temp[i] = (unsigned char)(buffer[i] / 256 + 128);
}
// write in 8bit format
write(temp, numElems);
break;
}
case 16:
{
// 16bit format
// use temp buffer to swap byte order if necessary
short *pTemp = (short *)getConvBuffer(numElems * sizeof(short));
memcpy(pTemp, buffer, numElems * 2);
_swap16Buffer(pTemp, numElems);
res = (int)fwrite(pTemp, 2, numElems, fptr);
if (res != numElems)
{
ST_THROW_RT_ERROR("Error while writing to a wav file.");
}
bytesWritten += 2 * numElems;
break;
}
default:
{
stringstream ss;
ss << "\nOnly 8/16 bit sample WAV files supported in integer compilation. Can't open WAV file with ";
ss << (int)header.format.bits_per_sample;
ss << " bit sample format. ";
ST_THROW_RT_ERROR(ss.str().c_str());
}
}
}
/// Convert from float to integer and saturate
inline int saturate(float fvalue, float minval, float maxval)
{
if (fvalue > maxval)
{
fvalue = maxval;
}
else if (fvalue < minval)
{
fvalue = minval;
}
return (int)fvalue;
}
void WavOutFile::write(const float *buffer, int numElems)
{
int numBytes;
int bytesPerSample;
if (numElems == 0) return;
bytesPerSample = header.format.bits_per_sample / 8;
numBytes = numElems * bytesPerSample;
short *temp = (short*)getConvBuffer(numBytes);
switch (bytesPerSample)
{
case 1:
{
unsigned char *temp2 = (unsigned char *)temp;
for (int i = 0; i < numElems; i ++)
{
temp2[i] = (unsigned char)saturate(buffer[i] * 128.0f + 128.0f, 0.0f, 255.0f);
}
break;
}
case 2:
{
short *temp2 = (short *)temp;
for (int i = 0; i < numElems; i ++)
{
short value = (short)saturate(buffer[i] * 32768.0f, -32768.0f, 32767.0f);
temp2[i] = _swap16(value);
}
break;
}
case 3:
{
char *temp2 = (char *)temp;
for (int i = 0; i < numElems; i ++)
{
int value = saturate(buffer[i] * 8388608.0f, -8388608.0f, 8388607.0f);
*((int*)temp2) = _swap32(value);
temp2 += 3;
}
break;
}
case 4:
{
int *temp2 = (int *)temp;
for (int i = 0; i < numElems; i ++)
{
int value = saturate(buffer[i] * 2147483648.0f, -2147483648.0f, 2147483647.0f);
temp2[i] = _swap32(value);
}
break;
}
default:
assert(false);
}
int res = (int)fwrite(temp, 1, numBytes, fptr);
if (res != numBytes)
{
ST_THROW_RT_ERROR("Error while writing to a wav file.");
}
bytesWritten += numBytes;
}

529
lib-src/soundtouch/source/example/SoundStretch/WavFile.h → lib-src/soundtouch/source/SoundStretch/WavFile.h
View File

@ -1,253 +1,276 @@
////////////////////////////////////////////////////////////////////////////////
///
/// 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: 2006-09-18 07:31:48 $
// File revision : $Revision: 1.2 $
//
// $Id: WavFile.h,v 1.2 2006-09-18 07:31:48 richardash1981 Exp $
//
////////////////////////////////////////////////////////////////////////////////
//
// 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
////////////////////////////////////////////////////////////////////////////////
///
/// 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: 2012-09-01 10:57:22 +0300 (Sat, 01 Sep 2012) $
// File revision : $Revision: 4 $
//
// $Id: WavFile.h 153 2012-09-01 07:57:22Z 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;
/// Base class for processing WAV audio files.
class WavFileBase
{
private:
/// Conversion working buffer;
char *convBuff;
int convBuffSize;
protected:
WavFileBase();
virtual ~WavFileBase();
/// Get pointer to conversion buffer of at min. given size
void *getConvBuffer(int sizeByte);
};
/// Class for reading WAV audio files.
class WavInFile : protected WavFileBase
{
private:
/// File pointer.
FILE *fptr;
/// Position within the audio stream
long position;
/// Counter of how many bytes of sample data have been read from the file.
long dataRead;
/// WAV header information
WavHeader header;
/// Init the WAV file stream
void init();
/// 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() const;
/// 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);
WavInFile(FILE *file);
/// Destructor: Closes the file.
~WavInFile();
/// 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;
/// Returns how many milliseconds of audio have so far been read from the file
///
/// \return elapsed duration in milliseconds
uint getElapsedMS() 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(unsigned 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.
/// Notice that reading in float format supports 8/16/24/32bit sample formats.
///
/// \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 : protected WavFileBase
{
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)
);
WavOutFile(FILE *file, int sampleRate, int bits, int channels);
/// 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 unsigned 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.
);
};
#endif

621
lib-src/soundtouch/source/example/SoundStretch/main.cpp → lib-src/soundtouch/source/SoundStretch/main.cpp
View File

@ -1,288 +1,333 @@
////////////////////////////////////////////////////////////////////////////////
///
/// 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: 2006-09-18 07:31:48 $
// File revision : $Revision: 1.3 $
//
// $Id: main.cpp,v 1.3 2006-09-18 07:31:48 richardash1981 Exp $
//
////////////////////////////////////////////////////////////////////////////////
//
// 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;
}
////////////////////////////////////////////////////////////////////////////////
///
/// 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: 2012-04-04 22:47:28 +0300 (Wed, 04 Apr 2012) $
// File revision : $Revision: 4 $
//
// $Id: main.cpp 141 2012-04-04 19:47:28Z 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 <string.h>
#include <time.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
#if _WIN32
#include <io.h>
#include <fcntl.h>
// Macro for Win32 standard input/output stream support: Sets a file stream into binary mode
#define SET_STREAM_TO_BIN_MODE(f) (_setmode(_fileno(f), _O_BINARY))
#else
// Not needed for GNU environment...
#define SET_STREAM_TO_BIN_MODE(f) {}
#endif
static const char _helloText[] =
"\n"
" SoundStretch v%s - Written by Olli Parviainen 2001 - 2012\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;
if (strcmp(params->inFileName, "stdin") == 0)
{
// used 'stdin' as input file
SET_STREAM_TO_BIN_MODE(stdin);
*inFile = new WavInFile(stdin);
}
else
{
// open input file...
*inFile = new WavInFile(params->inFileName);
}
// ... open output file with same sound parameters
bits = (int)(*inFile)->getNumBits();
samplerate = (int)(*inFile)->getSampleRate();
channels = (int)(*inFile)->getNumChannels();
if (params->outFileName)
{
if (strcmp(params->outFileName, "stdout") == 0)
{
SET_STREAM_TO_BIN_MODE(stdout);
*outFile = new WavOutFile(stdout, samplerate, bits, channels);
}
else
{
*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 = (int)inFile->getSampleRate();
channels = (int)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));
if (params->speech)
{
// use settings for speech processing
pSoundTouch->setSetting(SETTING_SEQUENCE_MS, 40);
pSoundTouch->setSetting(SETTING_SEEKWINDOW_MS, 15);
pSoundTouch->setSetting(SETTING_OVERLAP_MS, 8);
fprintf(stderr, "Tune processing parameters for speech processing.\n");
}
// print processing information
if (params->outFileName)
{
#ifdef SOUNDTOUCH_INTEGER_SAMPLES
fprintf(stderr, "Uses 16bit integer sample type in processing.\n\n");
#else
#ifndef SOUNDTOUCH_FLOAT_SAMPLES
#error "Sampletype not defined"
#endif
fprintf(stderr, "Uses 32bit floating point sample type in processing.\n\n");
#endif
// print processing information only if outFileName given i.e. some processing will happen
fprintf(stderr, "Processing the file with the following changes:\n");
fprintf(stderr, " tempo change = %+g %%\n", params->tempoDelta);
fprintf(stderr, " pitch change = %+g semitones\n", params->pitchDelta);
fprintf(stderr, " rate change = %+g %%\n\n", params->rateDelta);
fprintf(stderr, "Working...");
}
else
{
// outFileName not given
fprintf(stderr, "Warning: output file name missing, won't output anything.\n\n");
}
fflush(stderr);
}
// 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 = (int)inFile->getNumChannels();
assert(nChannels > 0);
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 / (int)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
fprintf(stderr, "Detecting BPM rate...");
fflush(stderr);
nChannels = (int)inFile->getNumChannels();
assert(BUFF_SIZE % nChannels == 0);
// 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();
fprintf(stderr, "Done!\n");
// rewind the file after bpm detection
inFile->rewind();
if (bpmValue > 0)
{
fprintf(stderr, "Detected BPM rate %.1f\n\n", bpmValue);
}
else
{
fprintf(stderr, "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;
fprintf(stderr, "The file will be converted to %.1f BPM\n\n", params->goalBPM);
}
}
int main(const int nParams, const char * const paramStr[])
{
WavInFile *inFile;
WavOutFile *outFile;
RunParameters *params;
SoundTouch soundTouch;
fprintf(stderr, _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);
// clock_t cs = clock(); // for benchmarking processing duration
// Process the sound
process(&soundTouch, inFile, outFile);
// clock_t ce = clock(); // for benchmarking processing duration
// printf("duration: %lf\n", (double)(ce-cs)/CLOCKS_PER_SEC);
// Close WAV file handles & dispose of the objects
delete inFile;
delete outFile;
delete params;
fprintf(stderr, "Done!\n");
}
catch (const runtime_error &e)
{
// An exception occurred during processing, display an error message
fprintf(stderr, "%s\n", e.what());
return -1;
}
return 0;
}

12
lib-src/soundtouch/source/example/SoundStretch/soundstretch.dsp → lib-src/soundtouch/source/SoundStretch/soundstretch.dsp
View File

@ -42,7 +42,7 @@ RSC=rc.exe
# PROP Ignore_Export_Lib 0
# PROP Target_Dir ""
# ADD BASE CPP /nologo /W3 /GX /O2 /D "WIN32" /D "NDEBUG" /D "_CONSOLE" /D "_MBCS" /YX /FD /c
# ADD CPP /nologo /W3 /GX /Zi /O2 /I "..\..\..\include" /I "..\bpm" /D "WIN32" /D "NDEBUG" /D "_CONSOLE" /D "_MBCS" /FD /c
# ADD CPP /nologo /W3 /GX /Zi /O2 /I "..\..\include" /D "WIN32" /D "NDEBUG" /D "_CONSOLE" /D "_MBCS" /FD /c
# SUBTRACT CPP /YX
# ADD BASE RSC /l 0x40b /d "NDEBUG"
# ADD RSC /l 0x40b /d "NDEBUG"
@ -51,10 +51,10 @@ BSC32=bscmake.exe
# ADD BSC32 /nologo
LINK32=link.exe
# ADD BASE LINK32 kernel32.lib user32.lib gdi32.lib winspool.lib comdlg32.lib advapi32.lib shell32.lib ole32.lib oleaut32.lib uuid.lib odbc32.lib odbccp32.lib kernel32.lib user32.lib gdi32.lib winspool.lib comdlg32.lib advapi32.lib shell32.lib ole32.lib oleaut32.lib uuid.lib odbc32.lib odbccp32.lib /nologo /subsystem:console /machine:I386
# ADD LINK32 SoundTouch.lib bpm.lib /nologo /subsystem:console /profile /map /debug /machine:I386 /libpath:"..\..\..\lib"
# ADD LINK32 SoundTouch.lib /nologo /subsystem:console /profile /map /debug /machine:I386 /libpath:"..\..\lib"
# Begin Special Build Tool
SOURCE="$(InputPath)"
PostBuild_Cmds=copy Release\soundstretch.exe ..\..\..\bin\
PostBuild_Cmds=copy Release\soundstretch.exe ..\..\bin\
# End Special Build Tool
!ELSEIF "$(CFG)" == "soundstretch - Win32 Debug"
@ -71,7 +71,7 @@ PostBuild_Cmds=copy Release\soundstretch.exe ..\..\..\bin\
# PROP Ignore_Export_Lib 0
# PROP Target_Dir ""
# ADD BASE CPP /nologo /W3 /Gm /GX /ZI /Od /D "WIN32" /D "_DEBUG" /D "_CONSOLE" /D "_MBCS" /YX /FD /GZ /c
# ADD CPP /nologo /Zp16 /W3 /Gm /GX /ZI /Od /I "..\..\..\include" /I "..\bpm" /D "WIN32" /D "_DEBUG" /D "_CONSOLE" /D "_MBCS" /FR /FD /GZ /c
# ADD CPP /nologo /Zp16 /W3 /Gm /GX /ZI /Od /I "..\..\include" /D "WIN32" /D "_DEBUG" /D "_CONSOLE" /D "_MBCS" /FR /FD /GZ /c
# SUBTRACT CPP /YX
# ADD BASE RSC /l 0x40b /d "_DEBUG"
# ADD RSC /l 0x40b /d "_DEBUG"
@ -80,11 +80,11 @@ BSC32=bscmake.exe
# ADD BSC32 /nologo
LINK32=link.exe
# ADD BASE LINK32 kernel32.lib user32.lib gdi32.lib winspool.lib comdlg32.lib advapi32.lib shell32.lib ole32.lib oleaut32.lib uuid.lib odbc32.lib odbccp32.lib kernel32.lib user32.lib gdi32.lib winspool.lib comdlg32.lib advapi32.lib shell32.lib ole32.lib oleaut32.lib uuid.lib odbc32.lib odbccp32.lib /nologo /subsystem:console /debug /machine:I386 /pdbtype:sept
# ADD LINK32 SoundTouchD.lib bpmD.lib /nologo /subsystem:console /map /debug /machine:I386 /nodefaultlib:"libc" /out:"Debug/soundstretchD.exe" /pdbtype:sept /libpath:"..\..\..\lib"
# ADD LINK32 SoundTouchD.lib /nologo /subsystem:console /map /debug /machine:I386 /nodefaultlib:"libc" /out:"Debug/soundstretchD.exe" /pdbtype:sept /libpath:"..\..\lib"
# SUBTRACT LINK32 /pdb:none
# Begin Special Build Tool
SOURCE="$(InputPath)"
PostBuild_Cmds=copy Debug\soundstretchD.exe ..\..\..\bin\
PostBuild_Cmds=copy Debug\soundstretchD.exe ..\..\bin\
# End Special Build Tool
!ENDIF

19
lib-src/soundtouch/source/example/SoundStretch/soundstretch.dsw → lib-src/soundtouch/source/SoundStretch/soundstretch.dsw
View File

@ -1,23 +1,9 @@
Microsoft Developer Studio Workspace File, Format Version 6.00
# WARNING: DO NOT EDIT OR DELETE THIS WORKSPACE FILE!
#
# $Id: soundstretch.dsw,v 1.2 2006-09-18 07:31:48 richardash1981 Exp $
###############################################################################
Project: "SoundTouch"=..\..\SoundTouch\SoundTouch.dsp - Package Owner=<4>
Package=<5>
{{{
}}}
Package=<4>
{{{
}}}
###############################################################################
Project: "bpm"=..\bpm\bpm.dsp - Package Owner=<4>
Project: "SoundTouch"=..\SoundTouch\SoundTouch.dsp - Package Owner=<4>
Package=<5>
{{{
@ -40,9 +26,6 @@ Package=<4>
Begin Project Dependency
Project_Dep_Name SoundTouch
End Project Dependency
Begin Project Dependency
Project_Dep_Name bpm
End Project Dependency
}}}
###############################################################################

29
lib-src/soundtouch/source/example/SoundStretch/soundstretch.sln → lib-src/soundtouch/source/SoundStretch/soundstretch.sln
View File

@ -1,16 +1,11 @@
Microsoft Visual Studio Solution File, Format Version 8.00
Project("{8BC9CEB8-8B4A-11D0-8D11-00A0C91BC942}") = "SoundTouch", "..\..\SoundTouch\SoundTouch.vcproj", "{121A2AA5-7B1F-4871-8436-FB9C837F11D4}"
ProjectSection(ProjectDependencies) = postProject
EndProjectSection
EndProject
Project("{8BC9CEB8-8B4A-11D0-8D11-00A0C91BC942}") = "bpm", "..\bpm\bpm.vcproj", "{DA262114-26F8-4F52-BC51-111307A6C776}"
ProjectSection(ProjectDependencies) = postProject
EndProjectSection
EndProject
Project("{8BC9CEB8-8B4A-11D0-8D11-00A0C91BC942}") = "soundstretch", "soundstretch.vcproj", "{5AACDFFA-D491-44B8-A332-DA7ACCAAF2AF}"
ProjectSection(ProjectDependencies) = postProject
{DA262114-26F8-4F52-BC51-111307A6C776} = {DA262114-26F8-4F52-BC51-111307A6C776}
{121A2AA5-7B1F-4871-8436-FB9C837F11D4} = {121A2AA5-7B1F-4871-8436-FB9C837F11D4}
{68A5DD20-7057-448B-8FE0-B6AC8D205509} = {68A5DD20-7057-448B-8FE0-B6AC8D205509}
EndProjectSection
EndProject
Project("{8BC9CEB8-8B4A-11D0-8D11-00A0C91BC942}") = "SoundTouch", "..\SoundTouch\SoundTouch.vcproj", "{68A5DD20-7057-448B-8FE0-B6AC8D205509}"
ProjectSection(ProjectDependencies) = postProject
EndProjectSection
EndProject
Global
@ -18,19 +13,17 @@ Global
Debug = Debug
Release = Release
EndGlobalSection
GlobalSection(ProjectDependencies) = postSolution
EndGlobalSection
GlobalSection(ProjectConfiguration) = postSolution
{121A2AA5-7B1F-4871-8436-FB9C837F11D4}.Debug.ActiveCfg = Debug|Win32
{121A2AA5-7B1F-4871-8436-FB9C837F11D4}.Debug.Build.0 = Debug|Win32
{121A2AA5-7B1F-4871-8436-FB9C837F11D4}.Release.ActiveCfg = Release|Win32
{121A2AA5-7B1F-4871-8436-FB9C837F11D4}.Release.Build.0 = Release|Win32
{DA262114-26F8-4F52-BC51-111307A6C776}.Debug.ActiveCfg = Debug|Win32
{DA262114-26F8-4F52-BC51-111307A6C776}.Debug.Build.0 = Debug|Win32
{DA262114-26F8-4F52-BC51-111307A6C776}.Release.ActiveCfg = Release|Win32
{DA262114-26F8-4F52-BC51-111307A6C776}.Release.Build.0 = Release|Win32
{5AACDFFA-D491-44B8-A332-DA7ACCAAF2AF}.Debug.ActiveCfg = Debug|Win32
{5AACDFFA-D491-44B8-A332-DA7ACCAAF2AF}.Debug.Build.0 = Debug|Win32
{5AACDFFA-D491-44B8-A332-DA7ACCAAF2AF}.Release.ActiveCfg = Release|Win32
{5AACDFFA-D491-44B8-A332-DA7ACCAAF2AF}.Release.Build.0 = Release|Win32
{68A5DD20-7057-448B-8FE0-B6AC8D205509}.Debug.ActiveCfg = Debug|Win32
{68A5DD20-7057-448B-8FE0-B6AC8D205509}.Debug.Build.0 = Debug|Win32
{68A5DD20-7057-448B-8FE0-B6AC8D205509}.Release.ActiveCfg = Release|Win32
{68A5DD20-7057-448B-8FE0-B6AC8D205509}.Release.Build.0 = Release|Win32
EndGlobalSection
GlobalSection(ExtensibilityGlobals) = postSolution
EndGlobalSection

117
lib-src/soundtouch/source/example/bpm/bpm.vcproj → lib-src/soundtouch/source/SoundStretch/soundstretch.vcproj
View File

@ -2,7 +2,7 @@
<VisualStudioProject
ProjectType="Visual C++"
Version="7.10"
Name="bpm"
Name="soundstretch"
SccProjectName=""
SccLocalPath="">
<Platforms>
@ -14,19 +14,19 @@
Name="Debug|Win32"
OutputDirectory=".\Debug"
IntermediateDirectory=".\Debug"
ConfigurationType="4"
ConfigurationType="1"
UseOfMFC="0"
ATLMinimizesCRunTimeLibraryUsage="FALSE"
CharacterSet="2">
<Tool
Name="VCCLCompilerTool"
Optimization="0"
AdditionalIncludeDirectories="..\..\..\include"
PreprocessorDefinitions="WIN32;_DEBUG;_LIB"
AdditionalIncludeDirectories="..\..\include"
PreprocessorDefinitions="WIN32;_DEBUG;_CONSOLE;_CRT_SECURE_NO_WARNINGS"
BasicRuntimeChecks="3"
RuntimeLibrary="5"
UsePrecompiledHeader="2"
PrecompiledHeaderFile=".\Debug/bpm.pch"
StructMemberAlignment="5"
PrecompiledHeaderFile=".\Debug/soundstretch.pch"
AssemblerListingLocation=".\Debug/"
ObjectFile=".\Debug/"
ProgramDataBaseFileName=".\Debug/"
@ -38,14 +38,26 @@
<Tool
Name="VCCustomBuildTool"/>
<Tool
Name="VCLibrarianTool"
OutputFile="Debug\bpmD.lib"
SuppressStartupBanner="TRUE"/>
Name="VCLinkerTool"
AdditionalDependencies="SoundTouchD.lib"
OutputFile="Debug/soundstretchD.exe"
LinkIncremental="2"
SuppressStartupBanner="TRUE"
AdditionalLibraryDirectories="..\..\lib"
IgnoreDefaultLibraryNames="libc"
GenerateDebugInformation="TRUE"
ProgramDatabaseFile=".\Debug/soundstretchD.pdb"
GenerateMapFile="TRUE"
MapFileName=".\Debug/soundstretchD.map"
SubSystem="1"
TargetMachine="1"/>
<Tool
Name="VCMIDLTool"/>
Name="VCMIDLTool"
TypeLibraryName=".\Debug/soundstretch.tlb"
HeaderFileName=""/>
<Tool
Name="VCPostBuildEventTool"
CommandLine="copy .\Debug\bpmD.lib ..\..\..\lib\"/>
CommandLine="copy Debug\soundstretchD.exe ..\..\bin\"/>
<Tool
Name="VCPreBuildEventTool"/>
<Tool
@ -58,6 +70,8 @@
Name="VCWebServiceProxyGeneratorTool"/>
<Tool
Name="VCXMLDataGeneratorTool"/>
<Tool
Name="VCWebDeploymentTool"/>
<Tool
Name="VCManagedWrapperGeneratorTool"/>
<Tool
@ -67,39 +81,50 @@
Name="Release|Win32"
OutputDirectory=".\Release"
IntermediateDirectory=".\Release"
ConfigurationType="4"
ConfigurationType="1"
UseOfMFC="0"
ATLMinimizesCRunTimeLibraryUsage="FALSE"
CharacterSet="2">
<Tool
Name="VCCLCompilerTool"
Optimization="2"
InlineFunctionExpansion="1"
AdditionalIncludeDirectories="..\..\..\include"
PreprocessorDefinitions="WIN32;NDEBUG;_LIB"
Optimization="3"
GlobalOptimizations="FALSE"
InlineFunctionExpansion="2"
EnableIntrinsicFunctions="TRUE"
AdditionalIncludeDirectories="..\..\include"
PreprocessorDefinitions="WIN32;NDEBUG;_CONSOLE;_CRT_SECURE_NO_WARNINGS"
StringPooling="TRUE"
RuntimeLibrary="4"
EnableFunctionLevelLinking="TRUE"
UsePrecompiledHeader="2"
PrecompiledHeaderFile=".\Release/bpm.pch"
PrecompiledHeaderFile=".\Release/soundstretch.pch"
AssemblerListingLocation=".\Release/"
ObjectFile=".\Release/"
ProgramDataBaseFileName=".\Release/"
WarningLevel="3"
SuppressStartupBanner="TRUE"
DebugInformationFormat="3"
DebugInformationFormat="0"
CompileAs="0"/>
<Tool
Name="VCCustomBuildTool"/>
<Tool
Name="VCLibrarianTool"
OutputFile=".\Release\bpm.lib"
SuppressStartupBanner="TRUE"/>
Name="VCLinkerTool"
AdditionalDependencies="SoundTouch.lib"
OutputFile=".\Release/soundstretch.exe"
LinkIncremental="1"
SuppressStartupBanner="TRUE"
AdditionalLibraryDirectories="..\..\lib"
GenerateDebugInformation="FALSE"
GenerateMapFile="TRUE"
MapFileName=".\Release/soundstretch.map"
SubSystem="1"
TargetMachine="1"/>
<Tool
Name="VCMIDLTool"/>
Name="VCMIDLTool"
TypeLibraryName=".\Release/soundstretch.tlb"
HeaderFileName=""/>
<Tool
Name="VCPostBuildEventTool"
CommandLine="copy .\Release\bpm.lib ..\..\..\lib\"/>
CommandLine="copy Release\soundstretch.exe ..\..\bin\"/>
<Tool
Name="VCPreBuildEventTool"/>
<Tool
@ -112,6 +137,8 @@
Name="VCWebServiceProxyGeneratorTool"/>
<Tool
Name="VCXMLDataGeneratorTool"/>
<Tool
Name="VCWebDeploymentTool"/>
<Tool
Name="VCManagedWrapperGeneratorTool"/>
<Tool
@ -125,7 +152,7 @@
Name="Source Files"
Filter="cpp;c;cxx;rc;def;r;odl;idl;hpj;bat">
<File
RelativePath="BPMDetect.cpp">
RelativePath="main.cpp">
<FileConfiguration
Name="Debug|Win32">
<Tool
@ -146,7 +173,28 @@
</FileConfiguration>
</File>
<File
RelativePath="PeakFinder.cpp">
RelativePath="RunParameters.cpp">
<FileConfiguration
Name="Debug|Win32">
<Tool
Name="VCCLCompilerTool"
Optimization="0"
AdditionalIncludeDirectories=""
PreprocessorDefinitions=""
BasicRuntimeChecks="3"
BrowseInformation="1"/>
</FileConfiguration>
<FileConfiguration
Name="Release|Win32">
<Tool
Name="VCCLCompilerTool"
Optimization="2"
AdditionalIncludeDirectories=""
PreprocessorDefinitions=""/>
</FileConfiguration>
</File>
<File
RelativePath="WavFile.cpp">
<FileConfiguration
Name="Debug|Win32">
<Tool
@ -171,21 +219,16 @@
Name="Header Files"
Filter="h;hpp;hxx;hm;inl">
<File
RelativePath="..\..\..\include\BPMDetect.h">
RelativePath="RunParameters.h">
</File>
<File
RelativePath="..\..\..\include\FIFOSampleBuffer.h">
</File>
<File
RelativePath="..\..\..\include\FIFOSamplePipe.h">
</File>
<File
RelativePath="PeakFinder.h">
</File>
<File
RelativePath="..\..\..\include\STTypes.h">
RelativePath="WavFile.h">
</File>
</Filter>
<Filter
Name="Resource Files"
Filter="ico;cur;bmp;dlg;rc2;rct;bin;rgs;gif;jpg;jpeg;jpe">
</Filter>
</Files>
<Globals>
</Globals>

350
lib-src/soundtouch/source/SoundTouch/3dnow_win.cpp
View File

@ -1,350 +0,0 @@
////////////////////////////////////////////////////////////////////////////////
///
/// 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: 2006-09-18 22:29:22 $
// File revision : $Revision: 1.4 $
//
// $Id: 3dnow_win.cpp,v 1.4 2006-09-18 22:29:22 martynshaw Exp $
//
////////////////////////////////////////////////////////////////////////////////
//
// 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;
}
FIRFilter3DNow::~FIRFilter3DNow()
{
delete[] filterCoeffsUnalign;
}
// (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) & -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) & -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

40
lib-src/soundtouch/source/SoundTouch/AAFilter.cpp
View File

@ -1,9 +1,9 @@
////////////////////////////////////////////////////////////////////////////////
///
/// 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
/// 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
@ -12,10 +12,10 @@
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2006-09-18 22:29:22 $
// File revision : $Revision: 1.4 $
// Last changed : $Date: 2009-01-11 13:34:24 +0200 (Sun, 11 Jan 2009) $
// File revision : $Revision: 4 $
//
// $Id: AAFilter.cpp,v 1.4 2006-09-18 22:29:22 martynshaw Exp $
// $Id: AAFilter.cpp 45 2009-01-11 11:34:24Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
@ -58,11 +58,11 @@ using namespace soundtouch;
*
*****************************************************************************/
AAFilter::AAFilter(const uint length)
AAFilter::AAFilter(uint len)
{
pFIR = FIRFilter::newInstance();
cutoffFreq = 0.5;
setLength(length);
setLength(len);
}
@ -77,7 +77,7 @@ AAFilter::~AAFilter()
// 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(const double newCutoffFreq)
void AAFilter::setCutoffFreq(double newCutoffFreq)
{
cutoffFreq = newCutoffFreq;
calculateCoeffs();
@ -86,7 +86,7 @@ void AAFilter::setCutoffFreq(const double newCutoffFreq)
// Sets number of FIR filter taps
void AAFilter::setLength(const uint newLength)
void AAFilter::setLength(uint newLength)
{
length = newLength;
calculateCoeffs();
@ -104,7 +104,7 @@ void AAFilter::calculateCoeffs()
double *work;
SAMPLETYPE *coeffs;
assert(length > 0);
assert(length >= 2);
assert(length % 4 == 0);
assert(cutoffFreq >= 0);
assert(cutoffFreq <= 0.5);
@ -112,21 +112,21 @@ void AAFilter::calculateCoeffs()
work = new double[length];
coeffs = new SAMPLETYPE[length];
fc2 = 2.0 * cutoffFreq;
fc2 = 2.0 * cutoffFreq;
wc = PI * fc2;
tempCoeff = TWOPI / (double)length;
sum = 0;
for (i = 0; i < length; i ++)
for (i = 0; i < length; i ++)
{
cntTemp = (double)i - (double)(length / 2);
temp = cntTemp * wc;
if (temp != 0)
if (temp != 0)
{
h = fc2 * sin(temp) / temp; // sinc function
}
else
}
else
{
h = 1.0;
}
@ -135,7 +135,7 @@ void AAFilter::calculateCoeffs()
temp = w * h;
work[i] = temp;
// calc net sum of coefficients
// calc net sum of coefficients
sum += temp;
}
@ -151,7 +151,7 @@ void AAFilter::calculateCoeffs()
// divided by 16384
scaleCoeff = 16384.0f / sum;
for (i = 0; i < length; i ++)
for (i = 0; i < length; i ++)
{
// scale & round to nearest integer
temp = work[i] * scaleCoeff;
@ -169,8 +169,8 @@ void AAFilter::calculateCoeffs()
}
// Applies the filter to the given sequence of samples.
// Note : The amount of outputted samples is by value of 'filter 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 AAFilter::evaluate(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples, uint numChannels) const
{

30
lib-src/soundtouch/source/SoundTouch/AAFilter.h
View File

@ -1,10 +1,10 @@
////////////////////////////////////////////////////////////////////////////////
///
/// Sampled sound tempo changer/time stretch algorithm. Changes the sound tempo
/// while maintaining the original pitch by using a time domain WSOLA-like method
/// 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
/// Anti-alias filter is used to prevent folding of high frequencies when
/// transposing the sample rate with interpolation.
///
/// Author : Copyright (c) Olli Parviainen
@ -13,10 +13,10 @@
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2008-03-17 22:29:58 $
// File revision : $Revision: 1.5 $
// Last changed : $Date: 2008-02-10 18:26:55 +0200 (Sun, 10 Feb 2008) $
// File revision : $Revision: 4 $
//
// $Id: AAFilter.h,v 1.5 2008-03-17 22:29:58 richardash1981 Exp $
// $Id: AAFilter.h 11 2008-02-10 16:26:55Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
@ -63,26 +63,26 @@ protected:
/// Calculate the FIR coefficients realizing the given cutoff-frequency
void calculateCoeffs();
public:
AAFilter(const uint length);
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
/// 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(const double newCutoffFreq);
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'
/// 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 evaluate(SAMPLETYPE *dest,
const SAMPLETYPE *src,
uint numSamples,
uint numChannels) const;
};

681
lib-src/soundtouch/source/example/bpm/BPMDetect.cpp → lib-src/soundtouch/source/SoundTouch/BPMDetect.cpp
View File

@ -1,311 +1,370 @@
////////////////////////////////////////////////////////////////////////////////
///
/// 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: 2006-09-18 07:31:48 $
// File revision : $Revision: 1.2 $
//
// $Id: BPMDetect.cpp,v 1.2 2006-09-18 07:31:48 richardash1981 Exp $
//
////////////////////////////////////////////////////////////////////////////////
//
// 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);
}
////////////////////////////////////////////////////////////////////////////////
///
/// 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: 2012-08-30 22:45:25 +0300 (Thu, 30 Aug 2012) $
// File revision : $Revision: 4 $
//
// $Id: BPMDetect.cpp 149 2012-08-30 19:45:25Z 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 <stdio.h>
#include "FIFOSampleBuffer.h"
#include "PeakFinder.h"
#include "BPMDetect.h"
using namespace soundtouch;
#define INPUT_BLOCK_SAMPLES 2048
#define DECIMATED_BLOCK_SAMPLES 256
/// decay constant for calculating RMS volume sliding average approximation
/// (time constant is about 10 sec)
const float avgdecay = 0.99986f;
/// Normalization coefficient for calculating RMS sliding average approximation.
const float avgnorm = (1 - avgdecay);
////////////////////////////////////////////////////////////////////////////////
// Enable following define to create bpm analysis file:
// #define _CREATE_BPM_DEBUG_FILE
#ifdef _CREATE_BPM_DEBUG_FILE
#define DEBUGFILE_NAME "c:\\temp\\soundtouch-bpm-debug.txt"
static void _SaveDebugData(const float *data, int minpos, int maxpos, double coeff)
{
FILE *fptr = fopen(DEBUGFILE_NAME, "wt");
int i;
if (fptr)
{
printf("\n\nWriting BPM debug data into file " DEBUGFILE_NAME "\n\n");
for (i = minpos; i < maxpos; i ++)
{
fprintf(fptr, "%d\t%.1lf\t%f\n", i, coeff / (double)i, data[i]);
}
fclose(fptr);
}
}
#else
#define _SaveDebugData(a,b,c,d)
#endif
////////////////////////////////////////////////////////////////////////////////
BPMDetect::BPMDetect(int numChannels, int aSampleRate)
{
this->sampleRate = aSampleRate;
this->channels = numChannels;
decimateSum = 0;
decimateCount = 0;
envelopeAccu = 0;
// Initialize RMS volume accumulator to RMS level of 1500 (out of 32768) that's
// safe initial RMS signal level value for song data. This value is then adapted
// to the actual level during processing.
#ifdef SOUNDTOUCH_INTEGER_SAMPLES
// integer samples
RMSVolumeAccu = (1500 * 1500) / avgnorm;
#else
// float samples, scaled to range [-1..+1[
RMSVolumeAccu = (0.045f * 0.045f) / avgnorm;
#endif
// choose decimation factor so that result is approx. 1000 Hz
decimateBy = sampleRate / 1000;
assert(decimateBy > 0);
assert(INPUT_BLOCK_SAMPLES < decimateBy * DECIMATED_BLOCK_SAMPLES);
// Calculate window length & starting item according to desired min & max bpms
windowLen = (60 * sampleRate) / (decimateBy * MIN_BPM);
windowStart = (60 * sampleRate) / (decimateBy * MAX_BPM);
assert(windowLen > windowStart);
// allocate new working objects
xcorr = new float[windowLen];
memset(xcorr, 0, windowLen * sizeof(float));
// allocate processing buffer
buffer = new FIFOSampleBuffer();
// we do processing in mono mode
buffer->setChannels(1);
buffer->clear();
}
BPMDetect::~BPMDetect()
{
delete[] xcorr;
delete buffer;
}
/// convert to mono, low-pass filter & decimate to about 500 Hz.
/// return number of outputted samples.
///
/// Decimation is used to remove the unnecessary frequencies and thus to reduce
/// the amount of data needed to be processed as calculating autocorrelation
/// function is a very-very heavy operation.
///
/// Anti-alias filtering is done simply by averaging the samples. This is really a
/// poor-man's anti-alias filtering, but it's not so critical in this kind of application
/// (it'd also be difficult to design a high-quality filter with steep cut-off at very
/// narrow band)
int BPMDetect::decimate(SAMPLETYPE *dest, const SAMPLETYPE *src, int numsamples)
{
int count, outcount;
LONG_SAMPLETYPE out;
assert(channels > 0);
assert(decimateBy > 0);
outcount = 0;
for (count = 0; count < numsamples; count ++)
{
int j;
// convert to mono and accumulate
for (j = 0; j < channels; j ++)
{
decimateSum += src[j];
}
src += j;
decimateCount ++;
if (decimateCount >= decimateBy)
{
// Store every Nth sample only
out = (LONG_SAMPLETYPE)(decimateSum / (decimateBy * channels));
decimateSum = 0;
decimateCount = 0;
#ifdef SOUNDTOUCH_INTEGER_SAMPLES
// check ranges for sure (shouldn't actually be necessary)
if (out > 32767)
{
out = 32767;
}
else if (out < -32768)
{
out = -32768;
}
#endif // SOUNDTOUCH_INTEGER_SAMPLES
dest[outcount] = (SAMPLETYPE)out;
outcount ++;
}
}
return outcount;
}
// Calculates autocorrelation function of the sample history buffer
void BPMDetect::updateXCorr(int process_samples)
{
int offs;
SAMPLETYPE *pBuffer;
assert(buffer->numSamples() >= (uint)(process_samples + windowLen));
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 static double decay = 0.7f; // decay constant for smoothing the envelope
const static double norm = (1 - decay);
int i;
LONG_SAMPLETYPE out;
double 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 cutoff ~2 times RMS volume
// (we're interested in peak values, not the silent moments)
if (val < 0.5 * sqrt(RMSVolumeAccu * avgnorm))
{
val = 0;
}
// smooth amplitude envelope
envelopeAccu *= decay;
envelopeAccu += val;
out = (LONG_SAMPLETYPE)(envelopeAccu * norm);
#ifdef SOUNDTOUCH_INTEGER_SAMPLES
// cut peaks (shouldn't be necessary though)
if (out > 32767) out = 32767;
#endif // SOUNDTOUCH_INTEGER_SAMPLES
samples[i] = (SAMPLETYPE)out;
}
}
void BPMDetect::inputSamples(const SAMPLETYPE *samples, int numSamples)
{
SAMPLETYPE decimated[DECIMATED_BLOCK_SAMPLES];
// iterate so that max INPUT_BLOCK_SAMPLES processed per iteration
while (numSamples > 0)
{
int block;
int decSamples;
block = (numSamples > INPUT_BLOCK_SAMPLES) ? INPUT_BLOCK_SAMPLES : numSamples;
// decimate. note that converts to mono at the same time
decSamples = decimate(decimated, samples, block);
samples += block * channels;
numSamples -= block;
// envelope new samples and add them to buffer
calcEnvelope(decimated, decSamples);
buffer->putSamples(decimated, decSamples);
}
// when the buffer has enought samples for processing...
if ((int)buffer->numSamples() > windowLen)
{
int processLength;
// how many samples are processed
processLength = (int)buffer->numSamples() - windowLen;
// ... calculate autocorrelations for oldest samples...
updateXCorr(processLength);
// ... and remove them from the buffer
buffer->receiveSamples(processLength);
}
}
void BPMDetect::removeBias()
{
int i;
float minval = 1e12f; // arbitrary large number
for (i = windowStart; i < windowLen; i ++)
{
if (xcorr[i] < minval)
{
minval = xcorr[i];
}
}
for (i = windowStart; i < windowLen; i ++)
{
xcorr[i] -= minval;
}
}
float BPMDetect::getBpm()
{
double peakPos;
double coeff;
PeakFinder peakFinder;
coeff = 60.0 * ((double)sampleRate / (double)decimateBy);
// save bpm debug analysis data if debug data enabled
_SaveDebugData(xcorr, windowStart, windowLen, coeff);
// remove bias from xcorr data
removeBias();
// find peak position
peakPos = peakFinder.detectPeak(xcorr, windowStart, windowLen);
assert(decimateBy != 0);
if (peakPos < 1e-9) return 0.0; // detection failed.
// calculate BPM
return (float) (coeff / peakPos);
}

104
lib-src/soundtouch/source/SoundTouch/FIFOSampleBuffer.cpp
View File

@ -1,12 +1,12 @@
////////////////////////////////////////////////////////////////////////////////
///
/// A buffer class for temporarily storaging sound samples, operates as a
/// 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'
/// 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
/// 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
@ -15,10 +15,10 @@
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2006-09-18 22:29:22 $
// File revision : $Revision: 1.4 $
// Last changed : $Date: 2012-11-08 20:53:01 +0200 (Thu, 08 Nov 2012) $
// File revision : $Revision: 4 $
//
// $Id: FIFOSampleBuffer.cpp,v 1.4 2006-09-18 22:29:22 martynshaw Exp $
// $Id: FIFOSampleBuffer.cpp 160 2012-11-08 18:53:01Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
@ -47,21 +47,22 @@
#include <memory.h>
#include <string.h>
#include <assert.h>
#include <stdexcept>
#include "FIFOSampleBuffer.h"
using namespace soundtouch;
// Constructor
FIFOSampleBuffer::FIFOSampleBuffer(uint numChannels)
FIFOSampleBuffer::FIFOSampleBuffer(int numChannels)
{
assert(numChannels > 0);
sizeInBytes = 0; // reasonable initial value
buffer = NULL; //new SAMPLETYPE[sizeInBytes / sizeof(SAMPLETYPE)];
buffer = NULL;
bufferUnaligned = NULL;
samplesInBuffer = 0;
bufferPos = 0;
channels = numChannels;
channels = (uint)numChannels;
ensureCapacity(32); // allocate initial capacity
}
@ -69,26 +70,29 @@ FIFOSampleBuffer::FIFOSampleBuffer(uint numChannels)
FIFOSampleBuffer::~FIFOSampleBuffer()
{
delete[] bufferUnaligned;
bufferUnaligned = NULL;
buffer = NULL;
}
// Sets number of channels, 1 = mono, 2 = stereo
void FIFOSampleBuffer::setChannels(const uint numChannels)
void FIFOSampleBuffer::setChannels(int numChannels)
{
uint usedBytes;
assert(numChannels > 0);
usedBytes = channels * samplesInBuffer;
channels = numChannels;
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
// zeroes this pointer by copying samples from the 'bufferPos' pointer
// location on to the beginning of the buffer.
void FIFOSampleBuffer::rewind()
{
if (bufferPos)
if (buffer && bufferPos)
{
memmove(buffer, ptrBegin(), sizeof(SAMPLETYPE) * channels * samplesInBuffer);
bufferPos = 0;
@ -96,12 +100,12 @@ void FIFOSampleBuffer::rewind()
}
// Adds 'numSamples' pcs of samples from the 'samples' memory position to
// Adds 'numSamples' pcs of samples from the 'samples' memory position to
// the sample buffer.
void FIFOSampleBuffer::putSamples(const SAMPLETYPE *samples, uint numSamples)
void FIFOSampleBuffer::putSamples(const SAMPLETYPE *samples, uint nSamples)
{
memcpy(ptrEnd(numSamples), samples, sizeof(SAMPLETYPE) * numSamples * channels);
samplesInBuffer += numSamples;
memcpy(ptrEnd(nSamples), samples, sizeof(SAMPLETYPE) * nSamples * channels);
samplesInBuffer += nSamples;
}
@ -109,46 +113,47 @@ void FIFOSampleBuffer::putSamples(const SAMPLETYPE *samples, uint numSamples)
// 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
// when accessing the buffer directly with 'ptrEnd' function. Please be
// careful though!
void FIFOSampleBuffer::putSamples(uint numSamples)
void FIFOSampleBuffer::putSamples(uint nSamples)
{
uint req;
req = samplesInBuffer + numSamples;
req = samplesInBuffer + nSamples;
ensureCapacity(req);
samplesInBuffer += numSamples;
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!
// 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,
// 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
// 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)
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.
// 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
// outputted samples from the buffer by calling the
// 'receiveSamples(numSamples)' function
SAMPLETYPE *FIFOSampleBuffer::ptrBegin() const
SAMPLETYPE *FIFOSampleBuffer::ptrBegin()
{
assert(buffer);
return buffer + bufferPos * channels;
}
@ -161,24 +166,28 @@ void FIFOSampleBuffer::ensureCapacity(uint capacityRequirement)
{
SAMPLETYPE *tempUnaligned, *temp;
if (capacityRequirement > getCapacity())
if (capacityRequirement > getCapacity())
{
// enlarge the buffer in 4kbyte steps (round up to next 4k boundary)
sizeInBytes = (capacityRequirement * channels * sizeof(SAMPLETYPE) + 4095) & -4096;
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");
ST_THROW_RT_ERROR("Couldn't allocate memory!\n");
}
// Align the buffer to begin at 16byte cache line boundary for optimal performance
temp = (SAMPLETYPE *)SOUNDTOUCH_ALIGN_POINTER_16(tempUnaligned);
if (samplesInBuffer)
{
memcpy(temp, ptrBegin(), samplesInBuffer * channels * sizeof(SAMPLETYPE));
}
temp = (SAMPLETYPE *)(((ulong)tempUnaligned + 15) & -16);
memcpy(temp, ptrBegin(), samplesInBuffer * channels * sizeof(SAMPLETYPE));
delete[] bufferUnaligned;
buffer = temp;
bufferUnaligned = tempUnaligned;
bufferPos = 0;
}
else
}
else
{
// simply rewind the buffer (if necessary)
rewind();
@ -250,3 +259,16 @@ void FIFOSampleBuffer::clear()
samplesInBuffer = 0;
bufferPos = 0;
}
/// allow trimming (downwards) amount of samples in pipeline.
/// Returns adjusted amount of samples
uint FIFOSampleBuffer::adjustAmountOfSamples(uint numSamples)
{
if (numSamples < samplesInBuffer)
{
samplesInBuffer = numSamples;
}
return samplesInBuffer;
}

77
lib-src/soundtouch/source/SoundTouch/FIRFilter.cpp
View File

@ -1,8 +1,8 @@
////////////////////////////////////////////////////////////////////////////////
///
/// General FIR digital filter routines with MMX optimization.
/// General FIR digital filter routines with MMX optimization.
///
/// Note : MMX optimized functions reside in a separate, platform-specific file,
/// 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
@ -11,10 +11,10 @@
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2006-09-18 22:29:22 $
// File revision : $Revision: 1.4 $
// Last changed : $Date: 2011-09-02 21:56:11 +0300 (Fri, 02 Sep 2011) $
// File revision : $Revision: 4 $
//
// $Id: FIRFilter.cpp,v 1.4 2006-09-18 22:29:22 martynshaw Exp $
// $Id: FIRFilter.cpp 131 2011-09-02 18:56:11Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
@ -43,7 +43,6 @@
#include <assert.h>
#include <math.h>
#include <stdlib.h>
#include <stdexcept>
#include "FIRFilter.h"
#include "cpu_detect.h"
@ -58,6 +57,7 @@ using namespace soundtouch;
FIRFilter::FIRFilter()
{
resultDivFactor = 0;
resultDivider = 0;
length = 0;
lengthDiv8 = 0;
filterCoeffs = NULL;
@ -74,24 +74,27 @@ uint FIRFilter::evaluateFilterStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, ui
{
uint i, j, end;
LONG_SAMPLETYPE suml, sumr;
#ifdef FLOAT_SAMPLES
#ifdef SOUNDTOUCH_FLOAT_SAMPLES
// when using floating point samples, use a scaler instead of a divider
// because division is much slower operation than multiplying.
double dScaler = 1.0 / (double)resultDivider;
#endif
assert(length != 0);
assert(src != NULL);
assert(dest != NULL);
assert(filterCoeffs != NULL);
end = 2 * (numSamples - length);
for (j = 0; j < end; j += 2)
for (j = 0; j < end; j += 2)
{
const SAMPLETYPE *ptr;
suml = sumr = 0;
ptr = src + j;
for (i = 0; i < length; i += 4)
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] +
@ -104,7 +107,7 @@ uint FIRFilter::evaluateFilterStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, ui
ptr[2 * i + 7] * filterCoeffs[i + 3];
}
#ifdef INTEGER_SAMPLES
#ifdef SOUNDTOUCH_INTEGER_SAMPLES
suml >>= resultDivFactor;
sumr >>= resultDivFactor;
// saturate to 16 bit integer limits
@ -114,7 +117,7 @@ uint FIRFilter::evaluateFilterStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, ui
#else
suml *= dScaler;
sumr *= dScaler;
#endif // INTEGER_SAMPLES
#endif // SOUNDTOUCH_INTEGER_SAMPLES
dest[j] = (SAMPLETYPE)suml;
dest[j + 1] = (SAMPLETYPE)sumr;
}
@ -129,7 +132,7 @@ uint FIRFilter::evaluateFilterMono(SAMPLETYPE *dest, const SAMPLETYPE *src, uint
{
uint i, j, end;
LONG_SAMPLETYPE sum;
#ifdef FLOAT_SAMPLES
#ifdef SOUNDTOUCH_FLOAT_SAMPLES
// when using floating point samples, use a scaler instead of a divider
// because division is much slower operation than multiplying.
double dScaler = 1.0 / (double)resultDivider;
@ -139,24 +142,24 @@ uint FIRFilter::evaluateFilterMono(SAMPLETYPE *dest, const SAMPLETYPE *src, uint
assert(length != 0);
end = numSamples - length;
for (j = 0; j < end; j ++)
for (j = 0; j < end; j ++)
{
sum = 0;
for (i = 0; i < length; i += 4)
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] +
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
#ifdef SOUNDTOUCH_INTEGER_SAMPLES
sum >>= resultDivFactor;
// saturate to 16 bit integer limits
sum = (sum < -32768) ? -32768 : (sum > 32767) ? 32767 : sum;
#else
sum *= dScaler;
#endif // INTEGER_SAMPLES
#endif // SOUNDTOUCH_INTEGER_SAMPLES
dest[j] = (SAMPLETYPE)sum;
src ++;
}
@ -170,14 +173,14 @@ uint FIRFilter::evaluateFilterMono(SAMPLETYPE *dest, const SAMPLETYPE *src, uint
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");
if (newLength % 8) ST_THROW_RT_ERROR("FIR filter length not divisible by 8");
lengthDiv8 = newLength / 8;
length = lengthDiv8 * 8;
assert(length == newLength);
resultDivFactor = uResultDivFactor;
resultDivider = (SAMPLETYPE)pow(2., (double)resultDivFactor);
resultDivider = (SAMPLETYPE)::pow(2.0, (int)resultDivFactor);
delete[] filterCoeffs;
filterCoeffs = new SAMPLETYPE[length];
@ -192,9 +195,9 @@ uint FIRFilter::getLength() const
// Applies the filter to the given sequence of samples.
// Applies the filter to the given sequence of samples.
//
// Note : The amount of outputted samples is by value of 'filter_length'
// 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
{
@ -203,8 +206,7 @@ uint FIRFilter::evaluate(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSample
assert(length > 0);
assert(lengthDiv8 * 8 == length);
if (numSamples < length) return 0;
assert(resultDivFactor >= 0);
if (numChannels == 2)
if (numChannels == 2)
{
return evaluateFilterStereo(dest, src, numSamples);
} else {
@ -214,13 +216,13 @@ uint FIRFilter::evaluate(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSample
// Operator 'new' is overloaded so that it automatically creates a suitable instance
// 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("Don't use 'new FIRFilter', use 'newInstance' member instead!");
return NULL;
ST_THROW_RT_ERROR("Error in FIRFilter::new: Don't use 'new FIRFilter', use 'newInstance' member instead!");
return newInstance();
}
@ -230,34 +232,25 @@ FIRFilter * FIRFilter::newInstance()
uExtensions = detectCPUextensions();
// Check if MMX/SSE/3DNow! instruction set extensions supported by CPU
// Check if MMX/SSE instruction set extensions supported by CPU
#ifdef ALLOW_MMX
#ifdef SOUNDTOUCH_ALLOW_MMX
// MMX routines available only with integer sample types
if (uExtensions & SUPPORT_MMX)
{
return ::new FIRFilterMMX;
}
else
#endif // ALLOW_MMX
#endif // SOUNDTOUCH_ALLOW_MMX
#ifdef ALLOW_SSE
#ifdef SOUNDTOUCH_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
#endif // SOUNDTOUCH_ALLOW_SSE
{
// ISA optimizations not supported, use plain C version

70
lib-src/soundtouch/source/SoundTouch/FIRFilter.h
View File

@ -1,8 +1,8 @@
////////////////////////////////////////////////////////////////////////////////
///
/// General FIR digital filter routines with MMX optimization.
/// General FIR digital filter routines with MMX optimization.
///
/// Note : MMX optimized functions reside in a separate, platform-specific file,
/// 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
@ -11,10 +11,10 @@
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2006-09-18 22:29:22 $
// File revision : $Revision: 1.4 $
// Last changed : $Date: 2011-02-13 21:13:57 +0200 (Sun, 13 Feb 2011) $
// File revision : $Revision: 4 $
//
// $Id: FIRFilter.h,v 1.4 2006-09-18 22:29:22 martynshaw Exp $
// $Id: FIRFilter.h 104 2011-02-13 19:13:57Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
@ -42,16 +42,17 @@
#ifndef FIRFilter_H
#define FIRFilter_H
#include <stddef.h>
#include "STTypes.h"
namespace soundtouch
{
class FIRFilter
class FIRFilter
{
protected:
// Number of FIR filter taps
uint length;
uint length;
// Number of FIR filter taps divided by 8
uint lengthDiv8;
@ -64,46 +65,46 @@ protected:
// Memory for filter coefficients
SAMPLETYPE *filterCoeffs;
virtual uint evaluateFilterStereo(SAMPLETYPE *dest,
const SAMPLETYPE *src,
virtual uint evaluateFilterStereo(SAMPLETYPE *dest,
const SAMPLETYPE *src,
uint numSamples) const;
virtual uint evaluateFilterMono(SAMPLETYPE *dest,
const SAMPLETYPE *src,
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
/// 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 * operator new(size_t s);
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'
/// 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 evaluate(SAMPLETYPE *dest,
const SAMPLETYPE *src,
uint numSamples,
uint numChannels) const;
uint getLength() const;
virtual void setCoefficients(const SAMPLETYPE *coeffs,
uint newLength,
virtual void setCoefficients(const SAMPLETYPE *coeffs,
uint newLength,
uint uResultDivFactor);
};
// Optional subclasses that implement CPU-specific optimizations:
#ifdef ALLOW_MMX
#ifdef SOUNDTOUCH_ALLOW_MMX
/// Class that implements MMX optimized functions exclusive for 16bit integer samples type.
/// Class that implements MMX optimized functions exclusive for 16bit integer samples type.
class FIRFilterMMX : public FIRFilter
{
protected:
@ -118,29 +119,10 @@ public:
virtual void setCoefficients(const short *coeffs, uint newLength, uint uResultDivFactor);
};
#endif // ALLOW_MMX
#endif // SOUNDTOUCH_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
#ifdef SOUNDTOUCH_ALLOW_SSE
/// Class that implements SSE optimized functions exclusive for floating point samples type.
class FIRFilterSSE : public FIRFilter
{
@ -156,7 +138,7 @@ public:
virtual void setCoefficients(const float *coeffs, uint newLength, uint uResultDivFactor);
};
#endif // ALLOW_SSE
#endif // SOUNDTOUCH_ALLOW_SSE
}

53
lib-src/soundtouch/source/SoundTouch/Makefile.am
View File

@ -1,9 +1,7 @@
## Process this file with automake to create Makefile.in
##
## $Id: Makefile.am,v 1.5 2007-06-13 02:51:36 llucius Exp $
## $Id: Makefile.am 138 2012-04-01 20:00:09Z 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
@ -24,19 +22,50 @@ 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
EXTRA_DIST=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
noinst_HEADERS=AAFilter.h cpu_detect.h cpu_detect_x86.cpp FIRFilter.h RateTransposer.h TDStretch.h PeakFinder.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
#
libSoundTouch_la_SOURCES=AAFilter.cpp FIRFilter.cpp FIFOSampleBuffer.cpp RateTransposer.cpp SoundTouch.cpp TDStretch.cpp cpu_detect_x86.cpp BPMDetect.cpp PeakFinder.cpp
# ??? test for -fcheck-new in configure.ac
# other compiler flags to add
AM_CXXFLAGS=-O3 @FLAG_SSE@ -fcheck-new -I../../include
# Compiler flags
AM_CXXFLAGS=-O3 -fcheck-new -I../../include
# Compile the files that need MMX and SSE individually.
libSoundTouch_la_LIBADD=libSoundTouchMMX.la libSoundTouchSSE.la
noinst_LTLIBRARIES=libSoundTouchMMX.la libSoundTouchSSE.la
libSoundTouchMMX_la_SOURCES=mmx_optimized.cpp
libSoundTouchSSE_la_SOURCES=sse_optimized.cpp
# We enable optimizations by default.
# If MMX is supported compile with -mmmx.
# Do not assume -msse is also supported.
if HAVE_MMX
libSoundTouchMMX_la_CXXFLAGS = -mmmx $(AM_CXXFLAGS)
else
libSoundTouchMMX_la_CXXFLAGS = $(AM_CXXFLAGS)
endif
# We enable optimizations by default.
# If SSE is supported compile with -msse.
if HAVE_SSE
libSoundTouchSSE_la_CXXFLAGS = -msse $(AM_CXXFLAGS)
else
libSoundTouchSSE_la_CXXFLAGS = $(AM_CXXFLAGS)
endif
# Let the user disable optimizations if he wishes to.
if !X86_OPTIMIZATIONS
libSoundTouchMMX_la_CXXFLAGS = $(AM_CXXFLAGS)
libSoundTouchSSE_la_CXXFLAGS = $(AM_CXXFLAGS)
endif
# other linking flags to add
#libSoundTouch_la_LIBADD=
# noinst_LTLIBRARIES = libSoundTouchOpt.la
# libSoundTouch_la_LIBADD = libSoundTouchOpt.la
# libSoundTouchOpt_la_SOURCES = mmx_optimized.cpp sse_optimized.cpp
# libSoundTouchOpt_la_CXXFLAGS = -O3 -msse -fcheck-new -I../../include

492
lib-src/soundtouch/source/SoundTouch/Makefile.in
View File

@ -1,492 +0,0 @@
# Makefile.in generated by automake 1.9.6 from Makefile.am.
# @configure_input@
# Copyright (C) 1994, 1995, 1996, 1997, 1998, 1999, 2000, 2001, 2002,
# 2003, 2004, 2005 Free Software Foundation, Inc.
# This Makefile.in is free software; the Free Software Foundation
# gives unlimited permission to copy and/or distribute it,
# with or without modifications, as long as this notice is preserved.
# This program is distributed in the hope that it will be useful,
# but WITHOUT ANY WARRANTY, to the extent permitted by law; without
# even the implied warranty of MERCHANTABILITY or FITNESS FOR A
# PARTICULAR PURPOSE.
@SET_MAKE@
srcdir = @srcdir@
top_srcdir = @top_srcdir@
VPATH = @srcdir@
pkgdatadir = $(datadir)/@PACKAGE@
pkglibdir = $(libdir)/@PACKAGE@
pkgincludedir = $(includedir)/@PACKAGE@
top_builddir = ../..
am__cd = CDPATH="$${ZSH_VERSION+.}$(PATH_SEPARATOR)" && cd
INSTALL = @INSTALL@
install_sh_DATA = $(install_sh) -c -m 644
install_sh_PROGRAM = $(install_sh) -c
install_sh_SCRIPT = $(install_sh) -c
INSTALL_HEADER = $(INSTALL_DATA)
transform = $(program_transform_name)
NORMAL_INSTALL = :
PRE_INSTALL = :
POST_INSTALL = :
NORMAL_UNINSTALL = :
PRE_UNINSTALL = :
POST_UNINSTALL = :
build_triplet = @build@
host_triplet = @host@
DIST_COMMON = $(noinst_HEADERS) $(srcdir)/Makefile.am \
$(srcdir)/Makefile.in $(top_srcdir)/config/am_include.mk
subdir = source/SoundTouch
ACLOCAL_M4 = $(top_srcdir)/aclocal.m4
am__aclocal_m4_deps = $(top_srcdir)/config/m4/libtool.m4 \
$(top_srcdir)/configure.ac
am__configure_deps = $(am__aclocal_m4_deps) $(CONFIGURE_DEPENDENCIES) \
$(ACLOCAL_M4)
mkinstalldirs = $(install_sh) -d
CONFIG_HEADER = $(top_builddir)/include/soundtouch_config.h
CONFIG_CLEAN_FILES =
am__vpath_adj_setup = srcdirstrip=`echo "$(srcdir)" | sed 's|.|.|g'`;
am__vpath_adj = case $$p in \
$(srcdir)/*) f=`echo "$$p" | sed "s|^$$srcdirstrip/||"`;; \
*) f=$$p;; \
esac;
am__strip_dir = `echo $$p | sed -e 's|^.*/||'`;
am__installdirs = "$(DESTDIR)$(libdir)"
libLTLIBRARIES_INSTALL = $(INSTALL)
LTLIBRARIES = $(lib_LTLIBRARIES)
libSoundTouch_la_LIBADD =
am_libSoundTouch_la_OBJECTS = AAFilter.lo FIRFilter.lo \
FIFOSampleBuffer.lo mmx_optimized.lo sse_optimized.lo \
RateTransposer.lo SoundTouch.lo TDStretch.lo \
cpu_detect_x86_gcc.lo
libSoundTouch_la_OBJECTS = $(am_libSoundTouch_la_OBJECTS)
DEFAULT_INCLUDES = -I. -I$(srcdir) -I$(top_builddir)/include
depcomp = $(SHELL) $(top_srcdir)/config/depcomp
am__depfiles_maybe = depfiles
CXXCOMPILE = $(CXX) $(DEFS) $(DEFAULT_INCLUDES) $(INCLUDES) \
$(AM_CPPFLAGS) $(CPPFLAGS) $(AM_CXXFLAGS) $(CXXFLAGS)
LTCXXCOMPILE = $(LIBTOOL) --tag=CXX --mode=compile $(CXX) $(DEFS) \
$(DEFAULT_INCLUDES) $(INCLUDES) $(AM_CPPFLAGS) $(CPPFLAGS) \
$(AM_CXXFLAGS) $(CXXFLAGS)
CXXLD = $(CXX)
CXXLINK = $(LIBTOOL) --tag=CXX --mode=link $(CXXLD) $(AM_CXXFLAGS) \
$(CXXFLAGS) $(AM_LDFLAGS) $(LDFLAGS) -o $@
SOURCES = $(libSoundTouch_la_SOURCES)
DIST_SOURCES = $(libSoundTouch_la_SOURCES)
HEADERS = $(noinst_HEADERS)
ETAGS = etags
CTAGS = ctags
DISTFILES = $(DIST_COMMON) $(DIST_SOURCES) $(TEXINFOS) $(EXTRA_DIST)
ACLOCAL = @ACLOCAL@
AMDEP_FALSE = @AMDEP_FALSE@
AMDEP_TRUE = @AMDEP_TRUE@
AMTAR = @AMTAR@
AR = @AR@
AUTOCONF = @AUTOCONF@
AUTOHEADER = @AUTOHEADER@
AUTOMAKE = @AUTOMAKE@
AWK = @AWK@
CC = @CC@
CCDEPMODE = @CCDEPMODE@
CFLAGS = @CFLAGS@
CPP = @CPP@
CPPFLAGS = @CPPFLAGS@
CXX = @CXX@
CXXCPP = @CXXCPP@
CXXDEPMODE = @CXXDEPMODE@
CXXFLAGS = @CXXFLAGS@
CYGPATH_W = @CYGPATH_W@
DEFS = @DEFS@
DEPDIR = @DEPDIR@
DSYMUTIL = @DSYMUTIL@
ECHO = @ECHO@
ECHO_C = @ECHO_C@
ECHO_N = @ECHO_N@
ECHO_T = @ECHO_T@
EGREP = @EGREP@
EXEEXT = @EXEEXT@
F77 = @F77@
FFLAGS = @FFLAGS@
FLAG_SSE = @FLAG_SSE@
GREP = @GREP@
INSTALL_DATA = @INSTALL_DATA@
INSTALL_PROGRAM = @INSTALL_PROGRAM@
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# other linking flags to add
#libSoundTouch_la_LIBADD=
# Tell versions [3.59,3.63) of GNU make to not export all variables.
# Otherwise a system limit (for SysV at least) may be exceeded.
.NOEXPORT:

467
lib-src/soundtouch/source/example/bpm/PeakFinder.cpp → lib-src/soundtouch/source/SoundTouch/PeakFinder.cpp
View File

@ -1,191 +1,276 @@
////////////////////////////////////////////////////////////////////////////////
///
/// 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: 2006-09-18 07:31:48 $
// File revision : $Revision: 1.2 $
//
// $Id: PeakFinder.cpp,v 1.2 2006-09-18 07:31:48 richardash1981 Exp $
//
////////////////////////////////////////////////////////////////////////////////
//
// 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);
}
////////////////////////////////////////////////////////////////////////////////
///
/// 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: 2012-12-28 21:52:47 +0200 (Fri, 28 Dec 2012) $
// File revision : $Revision: 4 $
//
// $Id: PeakFinder.cpp 164 2012-12-28 19:52:47Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#include <math.h>
#include <assert.h>
#include "PeakFinder.h"
using namespace soundtouch;
#define max(x, y) (((x) > (y)) ? (x) : (y))
PeakFinder::PeakFinder()
{
minPos = maxPos = 0;
}
// Finds real 'top' of a peak hump from neighnourhood of the given 'peakpos'.
int PeakFinder::findTop(const float *data, int peakpos) const
{
int i;
int start, end;
float refvalue;
refvalue = data[peakpos];
// seek within ±10 points
start = peakpos - 10;
if (start < minPos) start = minPos;
end = peakpos + 10;
if (end > maxPos) end = maxPos;
for (i = start; i <= end; i ++)
{
if (data[i] > refvalue)
{
peakpos = i;
refvalue = data[i];
}
}
// failure if max value is at edges of seek range => it's not peak, it's at slope.
if ((peakpos == start) || (peakpos == end)) return 0;
return peakpos;
}
// Finds 'ground level' of a peak hump by starting from 'peakpos' and proceeding
// to direction defined by 'direction' until next 'hump' after minimum value will
// begin
int PeakFinder::findGround(const float *data, int peakpos, int direction) const
{
int lowpos;
int pos;
int climb_count;
float refvalue;
float delta;
climb_count = 0;
refvalue = data[peakpos];
lowpos = peakpos;
pos = peakpos;
while ((pos > minPos+1) && (pos < maxPos-1))
{
int prevpos;
prevpos = pos;
pos += direction;
// calculate derivate
delta = data[pos] - data[prevpos];
if (delta <= 0)
{
// going downhill, ok
if (climb_count)
{
climb_count --; // decrease climb count
}
// check if new minimum found
if (data[pos] < refvalue)
{
// new minimum found
lowpos = pos;
refvalue = data[pos];
}
}
else
{
// going uphill, increase climbing counter
climb_count ++;
if (climb_count > 5) break; // we've been climbing too long => it's next uphill => quit
}
}
return lowpos;
}
// Find offset where the value crosses the given level, when starting from 'peakpos' and
// proceeds to direction defined in 'direction'
int PeakFinder::findCrossingLevel(const float *data, float level, int peakpos, int direction) const
{
float peaklevel;
int pos;
peaklevel = data[peakpos];
assert(peaklevel >= level);
pos = peakpos;
while ((pos >= minPos) && (pos < maxPos))
{
if (data[pos + direction] < level) return pos; // crossing found
pos += direction;
}
return -1; // not found
}
// Calculates the center of mass location of 'data' array items between 'firstPos' and 'lastPos'
double PeakFinder::calcMassCenter(const float *data, int firstPos, int lastPos) const
{
int i;
float sum;
float wsum;
sum = 0;
wsum = 0;
for (i = firstPos; i <= lastPos; i ++)
{
sum += (float)i * data[i];
wsum += data[i];
}
if (wsum < 1e-6) return 0;
return sum / wsum;
}
/// get exact center of peak near given position by calculating local mass of center
double PeakFinder::getPeakCenter(const float *data, int peakpos) const
{
float peakLevel; // peak level
int crosspos1, crosspos2; // position where the peak 'hump' crosses cutting level
float cutLevel; // cutting value
float groundLevel; // ground level of the peak
int gp1, gp2; // bottom positions of the peak 'hump'
// find ground positions.
gp1 = findGround(data, peakpos, -1);
gp2 = findGround(data, peakpos, 1);
groundLevel = 0.5f * (data[gp1] + data[gp2]);
peakLevel = data[peakpos];
// calculate 70%-level of the peak
cutLevel = 0.70f * peakLevel + 0.30f * groundLevel;
// find mid-level crossings
crosspos1 = findCrossingLevel(data, cutLevel, peakpos, -1);
crosspos2 = findCrossingLevel(data, cutLevel, peakpos, 1);
if ((crosspos1 < 0) || (crosspos2 < 0)) return 0; // no crossing, no peak..
// calculate mass center of the peak surroundings
return calcMassCenter(data, crosspos1, crosspos2);
}
double PeakFinder::detectPeak(const float *data, int aminPos, int amaxPos)
{
int i;
int peakpos; // position of peak level
double highPeak, peak;
this->minPos = aminPos;
this->maxPos = amaxPos;
// find absolute peak
peakpos = minPos;
peak = data[minPos];
for (i = minPos + 1; i < maxPos; i ++)
{
if (data[i] > peak)
{
peak = data[i];
peakpos = i;
}
}
// Calculate exact location of the highest peak mass center
highPeak = getPeakCenter(data, peakpos);
peak = highPeak;
// Now check if the highest peak were in fact harmonic of the true base beat peak
// - sometimes the highest peak can be Nth harmonic of the true base peak yet
// just a slightly higher than the true base
for (i = 3; i < 10; i ++)
{
double peaktmp, harmonic;
int i1,i2;
harmonic = (double)i * 0.5;
peakpos = (int)(highPeak / harmonic + 0.5f);
if (peakpos < minPos) break;
peakpos = findTop(data, peakpos); // seek true local maximum index
if (peakpos == 0) continue; // no local max here
// calculate mass-center of possible harmonic peak
peaktmp = getPeakCenter(data, peakpos);
// accept harmonic peak if
// (a) it is found
// (b) is within ±4% of the expected harmonic interval
// (c) has at least half x-corr value of the max. peak
double diff = harmonic * peaktmp / highPeak;
if ((diff < 0.96) || (diff > 1.04)) continue; // peak too afar from expected
// now compare to highest detected peak
i1 = (int)(highPeak + 0.5);
i2 = (int)(peaktmp + 0.5);
if (data[i2] >= 0.4*data[i1])
{
// The harmonic is at least half as high primary peak,
// thus use the harmonic peak instead
peak = peaktmp;
}
}
return peak;
}

182
lib-src/soundtouch/source/example/bpm/PeakFinder.h → lib-src/soundtouch/source/SoundTouch/PeakFinder.h
View File

@ -1,85 +1,97 @@
////////////////////////////////////////////////////////////////////////////////
///
/// 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: 2006-09-18 07:31:49 $
// File revision : $Revision: 1.2 $
//
// $Id: PeakFinder.h,v 1.2 2006-09-18 07:31:49 richardash1981 Exp $
//
////////////////////////////////////////////////////////////////////////////////
//
// 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_
////////////////////////////////////////////////////////////////////////////////
///
/// 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: 2011-12-30 22:33:46 +0200 (Fri, 30 Dec 2011) $
// File revision : $Revision: 4 $
//
// $Id: PeakFinder.h 132 2011-12-30 20:33:46Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#ifndef _PeakFinder_H_
#define _PeakFinder_H_
namespace soundtouch
{
class PeakFinder
{
protected:
/// Min, max allowed peak positions within the data vector
int minPos, maxPos;
/// Calculates the mass center between given vector items.
double calcMassCenter(const float *data, ///< Data vector.
int firstPos, ///< Index of first vector item beloging to the peak.
int lastPos ///< Index of last vector item beloging to the peak.
) const;
/// Finds the data vector index where the monotoniously decreasing signal crosses the
/// given level.
int findCrossingLevel(const float *data, ///< Data vector.
float level, ///< Goal crossing level.
int peakpos, ///< Peak position index within the data vector.
int direction /// Direction where to proceed from the peak: 1 = right, -1 = left.
) const;
// Finds real 'top' of a peak hump from neighnourhood of the given 'peakpos'.
int findTop(const float *data, int peakpos) const;
/// Finds the 'ground' level, i.e. smallest level between two neighbouring peaks, to right-
/// or left-hand side of the given peak position.
int findGround(const float *data, /// Data vector.
int peakpos, /// Peak position index within the data vector.
int direction /// Direction where to proceed from the peak: 1 = right, -1 = left.
) const;
/// get exact center of peak near given position by calculating local mass of center
double getPeakCenter(const float *data, int peakpos) const;
public:
/// Constructor.
PeakFinder();
/// Detect exact peak position of the data vector by finding the largest peak 'hump'
/// and calculating the mass-center location of the peak hump.
///
/// \return The location of the largest base harmonic peak hump.
double detectPeak(const float *data, /// Data vector to be analyzed. The data vector has
/// to be at least 'maxPos' items long.
int minPos, ///< Min allowed peak location within the vector data.
int maxPos ///< Max allowed peak location within the vector data.
);
};
}
#endif // _PeakFinder_H_

320
lib-src/soundtouch/source/SoundTouch/RateTransposer.cpp
View File

@ -1,6 +1,6 @@
////////////////////////////////////////////////////////////////////////////////
///
/// Sample rate transposer. Changes sample rate by using linear interpolation
///
/// 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)
///
@ -10,10 +10,10 @@
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2006-09-18 22:29:22 $
// File revision : $Revision: 1.5 $
// Last changed : $Date: 2011-09-02 21:56:11 +0300 (Fri, 02 Sep 2011) $
// File revision : $Revision: 4 $
//
// $Id: RateTransposer.cpp,v 1.5 2006-09-18 22:29:22 martynshaw Exp $
// $Id: RateTransposer.cpp 131 2011-09-02 18:56:11Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
@ -42,7 +42,6 @@
#include <assert.h>
#include <stdlib.h>
#include <stdio.h>
#include <limits.h>
#include "RateTransposer.h"
#include "AAFilter.h"
@ -55,23 +54,23 @@ class RateTransposerInteger : public RateTransposer
{
protected:
int iSlopeCount;
uint uRate;
int iRate;
SAMPLETYPE sPrevSampleL, sPrevSampleR;
virtual void resetRegisters();
virtual uint transposeStereo(SAMPLETYPE *dest,
const SAMPLETYPE *src,
virtual uint transposeStereo(SAMPLETYPE *dest,
const SAMPLETYPE *src,
uint numSamples);
virtual uint transposeMono(SAMPLETYPE *dest,
const SAMPLETYPE *src,
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
/// Sets new target rate. Normal rate = 1.0, smaller values represent slower
/// rate, larger faster rates.
virtual void setRate(float newRate);
@ -84,16 +83,15 @@ class RateTransposerFloat : public RateTransposer
{
protected:
float fSlopeCount;
float fRateStep;
SAMPLETYPE sPrevSampleL, sPrevSampleR;
virtual void resetRegisters();
virtual uint transposeStereo(SAMPLETYPE *dest,
const SAMPLETYPE *src,
virtual uint transposeStereo(SAMPLETYPE *dest,
const SAMPLETYPE *src,
uint numSamples);
virtual uint transposeMono(SAMPLETYPE *dest,
const SAMPLETYPE *src,
virtual uint transposeMono(SAMPLETYPE *dest,
const SAMPLETYPE *src,
uint numSamples);
public:
@ -103,25 +101,19 @@ public:
#ifndef min
#define min(a,b) ((a > b) ? b : a)
#define max(a,b) ((a < b) ? b : a)
#endif
// Operator 'new' is overloaded so that it automatically creates a suitable instance
// 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)
{
// Notice! don't use "new TDStretch" directly, use "newInstance" to create a new instance instead!
assert(FALSE);
return NULL;
ST_THROW_RT_ERROR("Error in RateTransoser::new: don't use \"new TDStretch\" directly, use \"newInstance\" to create a new instance instead!");
return newInstance();
}
RateTransposer *RateTransposer::newInstance()
{
#ifdef INTEGER_SAMPLES
#ifdef SOUNDTOUCH_INTEGER_SAMPLES
return ::new RateTransposerInteger;
#else
return ::new RateTransposerFloat;
@ -132,8 +124,9 @@ RateTransposer *RateTransposer::newInstance()
// Constructor
RateTransposer::RateTransposer() : FIFOProcessor(&outputBuffer)
{
uChannels = 2;
numChannels = 2;
bUseAAFilter = TRUE;
fRate = 0;
// Instantiates the anti-alias filter with default tap length
// of 32
@ -150,7 +143,7 @@ RateTransposer::~RateTransposer()
/// Enables/disables the anti-alias filter. Zero to disable, nonzero to enable
void RateTransposer::enableAAFilter(const BOOL newMode)
void RateTransposer::enableAAFilter(BOOL newMode)
{
bUseAAFilter = newMode;
}
@ -163,27 +156,27 @@ BOOL RateTransposer::isAAFilterEnabled() const
}
AAFilter *RateTransposer::getAAFilter() const
AAFilter *RateTransposer::getAAFilter()
{
return pAAFilter;
}
// Sets new target uRate. Normal uRate = 1.0, smaller values represent slower
// uRate, larger faster uRates.
// Sets new target iRate. Normal iRate = 1.0, smaller values represent slower
// iRate, larger faster iRates.
void RateTransposer::setRate(float newRate)
{
float fCutoff;
double fCutoff;
fRate = newRate;
// design a new anti-alias filter
if (newRate > 1.0f)
if (newRate > 1.0f)
{
fCutoff = 0.5f / newRate;
}
else
}
else
{
fCutoff = 0.5f * newRate;
}
@ -197,45 +190,47 @@ void RateTransposer::setRate(float newRate)
//
// 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 'numSamples' pcs of samples from the 'samples' memory position into
// Adds 'nSamples' pcs of samples from the 'samples' memory position into
// the input of the object.
void RateTransposer::putSamples(const SAMPLETYPE *samples, uint numSamples)
void RateTransposer::putSamples(const SAMPLETYPE *samples, uint nSamples)
{
processSamples(samples, numSamples);
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 numSamples)
void RateTransposer::upsample(const SAMPLETYPE *src, uint nSamples)
{
int count, sizeTemp, num;
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'
// First check that there's enough room in 'storeBuffer'
// (+16 is to reserve some slack in the destination buffer)
sizeTemp = (int)((float)numSamples / fRate + 16.0f);
sizeTemp = (uint)((float)nSamples / fRate + 16.0f);
// Transpose the samples, store the result into the end of "storeBuffer"
count = transpose(storeBuffer.ptrEnd(sizeTemp), src, numSamples);
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, uChannels);
count = pAAFilter->evaluate(outputBuffer.ptrEnd(num),
storeBuffer.ptrBegin(), num, (uint)numChannels);
outputBuffer.putSamples(count);
// Remove the processed samples from "storeBuffer"
@ -245,96 +240,99 @@ void RateTransposer::upsample(const SAMPLETYPE *src, uint numSamples)
// Transposes down the sample rate, causing the observed playback 'rate' of the
// sound to increase
void RateTransposer::downsample(const SAMPLETYPE *src, uint numSamples)
void RateTransposer::downsample(const SAMPLETYPE *src, uint nSamples)
{
int count, sizeTemp;
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. */
// over the lover frequencies), then transpose.
// Add the new samples to the end of the storeBuffer */
storeBuffer.putSamples(src, numSamples);
// 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
// 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, uChannels);
count = pAAFilter->evaluate(tempBuffer.ptrEnd(sizeTemp),
storeBuffer.ptrBegin(), sizeTemp, (uint)numChannels);
if (count == 0) return;
// Remove the filtered samples from 'storeBuffer'
storeBuffer.receiveSamples(count);
// Transpose the samples (+16 is to reserve some slack in the destination buffer)
sizeTemp = (int)((float)numSamples / fRate + 16.0f);
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.
// 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 numSamples)
void RateTransposer::processSamples(const SAMPLETYPE *src, uint nSamples)
{
uint count;
uint sizeReq;
if (numSamples == 0) return;
if (nSamples == 0) return;
assert(pAAFilter);
// If anti-alias filter is turned off, simply transpose without applying
// the filter
if (bUseAAFilter == FALSE)
if (bUseAAFilter == FALSE)
{
sizeReq = (int)((float)numSamples / fRate + 1.0f);
count = transpose(outputBuffer.ptrEnd(sizeReq), src, numSamples);
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)
if (fRate < 1.0f)
{
upsample(src, numSamples);
}
else
upsample(src, nSamples);
}
else
{
downsample(src, numSamples);
downsample(src, nSamples);
}
}
// Transposes the sample rate of the given samples using linear interpolation.
// 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 numSamples)
inline uint RateTransposer::transpose(SAMPLETYPE *dest, const SAMPLETYPE *src, uint nSamples)
{
if (uChannels == 2)
if (numChannels == 2)
{
return transposeStereo(dest, src, numSamples);
}
else
return transposeStereo(dest, src, nSamples);
}
else
{
return transposeMono(dest, src, numSamples);
return transposeMono(dest, src, nSamples);
}
}
// Sets the number of channels, 1 = mono, 2 = stereo
void RateTransposer::setChannels(const uint numchannels)
void RateTransposer::setChannels(int nChannels)
{
if (uChannels == numchannels) return;
assert(nChannels > 0);
if (numChannels == nChannels) return;
assert(numchannels == 1 || numchannels == 2);
uChannels = numchannels;
assert(nChannels == 1 || nChannels == 2);
numChannels = nChannels;
storeBuffer.setChannels(uChannels);
tempBuffer.setChannels(uChannels);
outputBuffer.setChannels(uChannels);
storeBuffer.setChannels(numChannels);
tempBuffer.setChannels(numChannels);
outputBuffer.setChannels(numChannels);
// Inits the linear interpolation registers
resetRegisters();
@ -350,7 +348,7 @@ void RateTransposer::clear()
// Returns nonzero if there aren't any samples available for outputting.
uint RateTransposer::isEmpty()
int RateTransposer::isEmpty() const
{
int res;
@ -363,7 +361,7 @@ uint RateTransposer::isEmpty()
//////////////////////////////////////////////////////////////////////////////
//
// RateTransposerInteger - integer arithmetic implementation
//
//
/// fixed-point interpolation routine precision
#define SCALE 65536
@ -371,11 +369,10 @@ uint RateTransposer::isEmpty()
// Constructor
RateTransposerInteger::RateTransposerInteger() : RateTransposer()
{
// call these here as these are virtual functions; calling these
// from the base class constructor wouldn't execute the overloaded
// versions (<master yoda>peculiar C++ can be</my>).
resetRegisters();
setRate(1.0f);
// 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);
}
@ -387,73 +384,75 @@ RateTransposerInteger::~RateTransposerInteger()
void RateTransposerInteger::resetRegisters()
{
iSlopeCount = 0;
sPrevSampleL =
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
// 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 numSamples)
uint RateTransposerInteger::transposeMono(SAMPLETYPE *dest, const SAMPLETYPE *src, uint nSamples)
{
unsigned int i, used;
LONG_SAMPLETYPE temp, vol1;
used = 0;
if (nSamples == 0) return 0; // no samples, no work
used = 0;
i = 0;
// Process the last sample saved from the previous call first...
while (iSlopeCount <= SCALE)
while (iSlopeCount <= SCALE)
{
vol1 = (LONG_SAMPLETYPE)(SCALE - iSlopeCount);
temp = vol1 * sPrevSampleL + iSlopeCount * src[0];
dest[i] = (SAMPLETYPE)(temp / SCALE);
i++;
iSlopeCount += uRate;
iSlopeCount += iRate;
}
// now always (iSlopeCount > SCALE)
iSlopeCount -= SCALE;
while (1)
{
while (iSlopeCount > SCALE)
while (iSlopeCount > SCALE)
{
iSlopeCount -= SCALE;
used ++;
if (used >= numSamples - 1) goto end;
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 += uRate;
iSlopeCount += iRate;
}
end:
// Store the last sample for the next round
sPrevSampleL = src[numSamples - 1];
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
// 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 numSamples)
uint RateTransposerInteger::transposeStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, uint nSamples)
{
unsigned int srcPos, i, used;
LONG_SAMPLETYPE temp, vol1;
if (numSamples == 0) return 0; // no samples, no work
if (nSamples == 0) return 0; // no samples, no work
used = 0;
used = 0;
i = 0;
// Process the last sample saved from the sPrevSampleLious call first...
while (iSlopeCount <= SCALE)
while (iSlopeCount <= SCALE)
{
vol1 = (LONG_SAMPLETYPE)(SCALE - iSlopeCount);
temp = vol1 * sPrevSampleL + iSlopeCount * src[0];
@ -461,18 +460,18 @@ uint RateTransposerInteger::transposeStereo(SAMPLETYPE *dest, const SAMPLETYPE *
temp = vol1 * sPrevSampleR + iSlopeCount * src[1];
dest[2 * i + 1] = (SAMPLETYPE)(temp / SCALE);
i++;
iSlopeCount += uRate;
iSlopeCount += iRate;
}
// now always (iSlopeCount > SCALE)
iSlopeCount -= SCALE;
while (1)
{
while (iSlopeCount > SCALE)
while (iSlopeCount > SCALE)
{
iSlopeCount -= SCALE;
used ++;
if (used >= numSamples - 1) goto end;
if (used >= nSamples - 1) goto end;
}
srcPos = 2 * used;
vol1 = (LONG_SAMPLETYPE)(SCALE - iSlopeCount);
@ -482,22 +481,22 @@ uint RateTransposerInteger::transposeStereo(SAMPLETYPE *dest, const SAMPLETYPE *
dest[2 * i + 1] = (SAMPLETYPE)(temp / SCALE);
i++;
iSlopeCount += uRate;
iSlopeCount += iRate;
}
end:
// Store the last sample for the next round
sPrevSampleL = src[2 * numSamples - 2];
sPrevSampleR = src[2 * numSamples - 1];
sPrevSampleL = src[2 * nSamples - 2];
sPrevSampleR = src[2 * nSamples - 1];
return i;
}
// Sets new target uRate. Normal uRate = 1.0, smaller values represent slower
// uRate, larger faster uRates.
// Sets new target iRate. Normal iRate = 1.0, smaller values represent slower
// iRate, larger faster iRates.
void RateTransposerInteger::setRate(float newRate)
{
uRate = (int)(newRate * SCALE + 0.5f);
iRate = (int)(newRate * SCALE + 0.5f);
RateTransposer::setRate(newRate);
}
@ -505,17 +504,16 @@ void RateTransposerInteger::setRate(float newRate)
//////////////////////////////////////////////////////////////////////////////
//
// RateTransposerFloat - floating point arithmetic implementation
//
//
//////////////////////////////////////////////////////////////////////////////
// Constructor
RateTransposerFloat::RateTransposerFloat() : RateTransposer()
{
// call these here as these are virtual functions; calling these
// from the base class constructor wouldn't execute the overloaded
// versions (<master yoda>peculiar C++ can be</my>).
resetRegisters();
setRate(1.0f);
// 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);
}
@ -527,24 +525,24 @@ RateTransposerFloat::~RateTransposerFloat()
void RateTransposerFloat::resetRegisters()
{
fSlopeCount = 0;
sPrevSampleL =
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
// 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 numSamples)
uint RateTransposerFloat::transposeMono(SAMPLETYPE *dest, const SAMPLETYPE *src, uint nSamples)
{
unsigned int i, used;
used = 0;
used = 0;
i = 0;
// Process the last sample saved from the previous call first...
while (fSlopeCount <= 1.0f)
while (fSlopeCount <= 1.0f)
{
dest[i] = (SAMPLETYPE)((1.0f - fSlopeCount) * sPrevSampleL + fSlopeCount * src[0]);
i++;
@ -552,42 +550,43 @@ uint RateTransposerFloat::transposeMono(SAMPLETYPE *dest, const SAMPLETYPE *src,
}
fSlopeCount -= 1.0f;
if (numSamples == 1) goto end;
while (1)
if (nSamples > 1)
{
while (fSlopeCount > 1.0f)
while (1)
{
fSlopeCount -= 1.0f;
used ++;
if (used >= numSamples - 1) goto end;
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;
}
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[numSamples - 1];
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
// 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 numSamples)
uint RateTransposerFloat::transposeStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, uint nSamples)
{
unsigned int srcPos, i, used;
if (numSamples == 0) return 0; // no samples, no work
if (nSamples == 0) return 0; // no samples, no work
used = 0;
used = 0;
i = 0;
// Process the last sample saved from the sPrevSampleLious call first...
while (fSlopeCount <= 1.0f)
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]);
@ -597,30 +596,31 @@ uint RateTransposerFloat::transposeStereo(SAMPLETYPE *dest, const SAMPLETYPE *sr
// now always (iSlopeCount > 1.0f)
fSlopeCount -= 1.0f;
if (numSamples == 1) goto end;
while (1)
if (nSamples > 1)
{
while (fSlopeCount > 1.0f)
while (1)
{
fSlopeCount -= 1.0f;
used ++;
if (used >= numSamples - 1) goto end;
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;
}
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 * numSamples - 2];
sPrevSampleR = src[2 * numSamples - 1];
sPrevSampleL = src[2 * nSamples - 2];
sPrevSampleR = src[2 * nSamples - 1];
return i;
}

63
lib-src/soundtouch/source/SoundTouch/RateTransposer.h
View File

@ -1,10 +1,10 @@
////////////////////////////////////////////////////////////////////////////////
///
/// Sample rate transposer. Changes sample rate by using linear interpolation
///
/// 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
/// Use either of the derived classes of 'RateTransposerInteger' or
/// 'RateTransposerFloat' for corresponding integer/floating point tranposing
/// algorithm implementation.
///
@ -14,10 +14,10 @@
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2008-03-17 22:29:58 $
// File revision : $Revision: 1.5 $
// Last changed : $Date: 2009-02-21 18:00:14 +0200 (Sat, 21 Feb 2009) $
// File revision : $Revision: 4 $
//
// $Id: RateTransposer.h,v 1.5 2008-03-17 22:29:58 richardash1981 Exp $
// $Id: RateTransposer.h 63 2009-02-21 16:00:14Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
@ -45,6 +45,7 @@
#ifndef RateTransposer_H
#define RateTransposer_H
#include <stddef.h>
#include "AAFilter.h"
#include "FIFOSamplePipe.h"
#include "FIFOSampleBuffer.h"
@ -56,9 +57,9 @@ 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
/// 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
{
@ -68,7 +69,7 @@ protected:
float fRate;
uint uChannels;
int numChannels;
/// Buffer for collecting samples to feed the anti-alias filter between
/// two batches
@ -82,32 +83,28 @@ protected:
BOOL bUseAAFilter;
void init();
virtual void resetRegisters() = 0;
virtual uint transposeStereo(SAMPLETYPE *dest,
const SAMPLETYPE *src,
virtual uint transposeStereo(SAMPLETYPE *dest,
const SAMPLETYPE *src,
uint numSamples) = 0;
virtual uint transposeMono(SAMPLETYPE *dest,
const SAMPLETYPE *src,
virtual uint transposeMono(SAMPLETYPE *dest,
const SAMPLETYPE *src,
uint numSamples) = 0;
uint transpose(SAMPLETYPE *dest,
const SAMPLETYPE *src,
inline uint transpose(SAMPLETYPE *dest,
const SAMPLETYPE *src,
uint numSamples);
void flushStoreBuffer();
void downsample(const SAMPLETYPE *src,
void downsample(const SAMPLETYPE *src,
uint numSamples);
void upsample(const SAMPLETYPE *src,
void upsample(const SAMPLETYPE *src,
uint numSamples);
/// Transposes sample rate by applying anti-alias filter to prevent folding.
/// 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,
void processSamples(const SAMPLETYPE *src,
uint numSamples);
@ -115,12 +112,12 @@ public:
RateTransposer();
virtual ~RateTransposer();
/// Operator 'new' is overloaded so that it automatically creates a suitable instance
/// Operator 'new' is overloaded so that it automatically creates a suitable instance
/// depending on if we're to use integer or floating point arithmetics.
void *operator new(size_t s);
static void *operator new(size_t s);
/// Use this function instead of "new" operator to create a new instance of this class.
/// This function automatically chooses a correct implementation, depending on if
/// 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();
@ -131,20 +128,20 @@ public:
FIFOSamplePipe *getStore() { return &storeBuffer; };
/// Return anti-alias filter object
AAFilter *getAAFilter() const;
AAFilter *getAAFilter();
/// Enables/disables the anti-alias filter. Zero to disable, nonzero to enable
void enableAAFilter(const BOOL newMode);
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
/// 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(const uint channels);
void setChannels(int channels);
/// Adds 'numSamples' pcs of samples from the 'samples' memory position into
/// the input of the object.
@ -154,7 +151,7 @@ public:
void clear();
/// Returns nonzero if there aren't any samples available for outputting.
uint isEmpty();
int isEmpty() const;
};
}

197
lib-src/soundtouch/source/SoundTouch/SoundTouch.cpp
View File

@ -1,27 +1,27 @@
//////////////////////////////////////////////////////////////////////////////
///
/// SoundTouch - main class for tempo/pitch/rate adjusting routines.
/// 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
/// - 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
/// - 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
/// 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
///
/// - 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
///
/// - 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
/// 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.
@ -30,8 +30,8 @@
/// 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) 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.
///
@ -41,10 +41,10 @@
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2006-09-18 22:29:22 $
// File revision : $Revision: 1.4 $
// Last changed : $Date: 2012-06-13 22:29:53 +0300 (Wed, 13 Jun 2012) $
// File revision : $Revision: 4 $
//
// $Id: SoundTouch.cpp,v 1.4 2006-09-18 22:29:22 martynshaw Exp $
// $Id: SoundTouch.cpp 143 2012-06-13 19:29:53Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
@ -73,7 +73,6 @@
#include <stdlib.h>
#include <memory.h>
#include <math.h>
#include <stdexcept>
#include <stdio.h>
#include "SoundTouch.h"
@ -82,12 +81,16 @@
#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
/// Print library version string for autoconf
extern "C" void soundtouch_ac_test()
{
printf("SoundTouch Version: %s\n",SOUNDTOUCH_VERSION);
}
}
SoundTouch::SoundTouch()
@ -101,8 +104,8 @@ SoundTouch::SoundTouch()
rate = tempo = 0;
virtualPitch =
virtualRate =
virtualPitch =
virtualRate =
virtualTempo = 1.0;
calcEffectiveRateAndTempo();
@ -140,13 +143,13 @@ uint SoundTouch::getVersionId()
// Sets the number of channels, 1 = mono, 2 = stereo
void SoundTouch::setChannels(uint numChannels)
{
if (numChannels != 1 && numChannels != 2)
if (numChannels != 1 && numChannels != 2)
{
throw std::runtime_error("Illegal number of channels");
ST_THROW_RT_ERROR("Illegal number of channels");
}
channels = numChannels;
pRateTransposer->setChannels(numChannels);
pTDStretch->setChannels(numChannels);
pRateTransposer->setChannels((int)numChannels);
pTDStretch->setChannels((int)numChannels);
}
@ -236,12 +239,30 @@ void SoundTouch::calcEffectiveRateAndTempo()
tempo = virtualTempo / virtualPitch;
rate = virtualPitch * virtualRate;
if (rate != oldRate) pRateTransposer->setRate(rate);
if (tempo != oldTempo) pTDStretch->setTempo(tempo);
if (!TEST_FLOAT_EQUAL(rate,oldRate)) pRateTransposer->setRate(rate);
if (!TEST_FLOAT_EQUAL(tempo, oldTempo)) pTDStretch->setTempo(tempo);
if (rate > 1.0f)
#ifndef SOUNDTOUCH_PREVENT_CLICK_AT_RATE_CROSSOVER
if (rate <= 1.0f)
{
if (output != pRateTransposer)
if (output != pTDStretch)
{
FIFOSamplePipe *tempoOut;
assert(output == pRateTransposer);
// move samples in the current output buffer to the output of pTDStretch
tempoOut = pTDStretch->getOutput();
tempoOut->moveSamples(*output);
// move samples in pitch transposer's store buffer to tempo changer's input
pTDStretch->moveSamples(*pRateTransposer->getStore());
output = pTDStretch;
}
}
else
#endif
{
if (output != pRateTransposer)
{
FIFOSamplePipe *transOut;
@ -254,24 +275,7 @@ void SoundTouch::calcEffectiveRateAndTempo()
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;
}
}
}
}
@ -280,51 +284,52 @@ void SoundTouch::setSampleRate(uint srate)
{
bSrateSet = TRUE;
// set sample rate, leave other tempo changer parameters as they are.
pTDStretch->setParameters(srate);
pTDStretch->setParameters((int)srate);
}
// Adds 'numSamples' pcs of samples from the 'samples' memory position into
// the input of the object.
void SoundTouch::putSamples(const SAMPLETYPE *samples, uint numSamples)
void SoundTouch::putSamples(const SAMPLETYPE *samples, uint nSamples)
{
if (bSrateSet == FALSE)
if (bSrateSet == FALSE)
{
throw std::runtime_error("SoundTouch : Sample rate not defined");
}
else if (channels == 0)
ST_THROW_RT_ERROR("SoundTouch : Sample rate not defined");
}
else if (channels == 0)
{
throw std::runtime_error("SoundTouch : Number of channels not defined");
ST_THROW_RT_ERROR("SoundTouch : Number of channels not defined");
}
// Transpose the rate of the new samples if necessary
/* Bypass the nominal setting - can introduce a click in sound when tempo/pitch control crosses the nominal value...
if (rate == 1.0f)
if (rate == 1.0f)
{
// The rate value is same as the original, simply evaluate the tempo changer.
// The rate value is same as the original, simply evaluate the tempo changer.
assert(output == pTDStretch);
if (pRateTransposer->isEmpty() == 0)
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, numSamples);
}
pTDStretch->putSamples(samples, nSamples);
}
*/
else if (rate <= 1.0f)
#ifndef SOUNDTOUCH_PREVENT_CLICK_AT_RATE_CROSSOVER
else if (rate <= 1.0f)
{
// transpose the rate down, output the transposed sound to tempo changer buffer
assert(output == pTDStretch);
pRateTransposer->putSamples(samples, numSamples);
pRateTransposer->putSamples(samples, nSamples);
pTDStretch->moveSamples(*pRateTransposer);
}
else
}
else
#endif
{
assert(rate > 1.0f);
// evaluate the tempo changer, then transpose the rate up,
// evaluate the tempo changer, then transpose the rate up,
assert(output == pRateTransposer);
pTDStretch->putSamples(samples, numSamples);
pTDStretch->putSamples(samples, nSamples);
pRateTransposer->moveSamples(*pTDStretch);
}
}
@ -340,20 +345,36 @@ void SoundTouch::putSamples(const SAMPLETYPE *samples, uint numSamples)
void SoundTouch::flush()
{
int i;
uint nOut;
SAMPLETYPE buff[128];
int nUnprocessed;
int nOut;
SAMPLETYPE buff[64*2]; // note: allocate 2*64 to cater 64 sample frames of stereo sound
nOut = numSamples();
// check how many samples still await processing, and scale
// that by tempo & rate to get expected output sample count
nUnprocessed = numUnprocessedSamples();
nUnprocessed = (int)((double)nUnprocessed / (tempo * rate) + 0.5);
memset(buff, 0, 128 * sizeof(SAMPLETYPE));
nOut = numSamples(); // ready samples currently in buffer ...
nOut += nUnprocessed; // ... and how many we expect there to be in the end
memset(buff, 0, 64 * channels * sizeof(SAMPLETYPE));
// "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
// 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 ++)
for (i = 0; i < 128; i ++)
{
putSamples(buff, 64);
if (numSamples() != nOut) break; // new samples have appeared in the output!
if ((int)numSamples() >= nOut)
{
// Enough new samples have appeared into the output!
// As samples come from processing with bigger chunks, now truncate it
// back to maximum "nOut" samples to improve duration accuracy
adjustAmountOfSamples(nOut);
// finish
break;
}
}
// Clear working buffers
@ -366,14 +387,14 @@ void SoundTouch::flush()
// Changes a setting controlling the processing system behaviour. See the
// 'SETTING_...' defines for available setting ID's.
BOOL SoundTouch::setSetting(uint settingId, uint value)
BOOL SoundTouch::setSetting(int settingId, int value)
{
uint sampleRate, sequenceMs, seekWindowMs, overlapMs;
int sampleRate, sequenceMs, seekWindowMs, overlapMs;
// read current tdstretch routine parameters
pTDStretch->getParameters(&sampleRate, &sequenceMs, &seekWindowMs, &overlapMs);
switch (settingId)
switch (settingId)
{
case SETTING_USE_AA_FILTER :
// enables / disabless anti-alias filter
@ -415,20 +436,20 @@ BOOL SoundTouch::setSetting(uint settingId, uint value)
// 'SETTING_...' defines for available setting ID's.
//
// Returns the setting value.
uint SoundTouch::getSetting(uint settingId) const
int SoundTouch::getSetting(int settingId) const
{
uint temp;
int temp;
switch (settingId)
switch (settingId)
{
case SETTING_USE_AA_FILTER :
return pRateTransposer->isAAFilterEnabled();
return (uint)pRateTransposer->isAAFilterEnabled();
case SETTING_AA_FILTER_LENGTH :
return pRateTransposer->getAAFilter()->getLength();
case SETTING_USE_QUICKSEEK :
return pTDStretch->isQuickSeekEnabled();
return (uint) pTDStretch->isQuickSeekEnabled();
case SETTING_SEQUENCE_MS:
pTDStretch->getParameters(NULL, &temp, NULL, NULL);
@ -442,7 +463,13 @@ uint SoundTouch::getSetting(uint settingId) const
pTDStretch->getParameters(NULL, NULL, NULL, &temp);
return temp;
default :
case SETTING_NOMINAL_INPUT_SEQUENCE :
return pTDStretch->getInputSampleReq();
case SETTING_NOMINAL_OUTPUT_SEQUENCE :
return pTDStretch->getOutputBatchSize();
default :
return 0;
}
}

16
lib-src/soundtouch/source/SoundTouch/SoundTouch.dsp
View File

@ -52,7 +52,7 @@ LIB32=link.exe -lib
# ADD LIB32 /nologo
# Begin Special Build Tool
SOURCE="$(InputPath)"
PostBuild_Cmds=copy .\Release\SoundTouch.lib ..\..\lib\
PostBuild_Cmds=copy .\Release\SoundTouch.lib ..\..\lib\
# End Special Build Tool
!ELSEIF "$(CFG)" == "SoundTouch - Win32 Debug"
@ -79,7 +79,7 @@ LIB32=link.exe -lib
# ADD LIB32 /nologo /out:"Debug\SoundTouchD.lib"
# Begin Special Build Tool
SOURCE="$(InputPath)"
PostBuild_Cmds=copy .\Debug\SoundTouchD.lib ..\..\lib\
PostBuild_Cmds=copy .\Debug\SoundTouchD.lib ..\..\lib\
# End Special Build Tool
!ENDIF
@ -91,6 +91,18 @@ PostBuild_Cmds=copy .\Debug\SoundTouchD.lib ..\..\lib\
# Begin Group "Source Files"
# PROP Default_Filter "cpp;c;cxx;rc;def;r;odl;idl;hpj;bat"
# Begin Group "bpm"
# PROP Default_Filter ""
# Begin Source File
SOURCE=.\BPMDetect.cpp
# End Source File
# Begin Source File
SOURCE=.\PeakFinder.cpp
# End Source File
# End Group
# Begin Source File
SOURCE=.\3dnow_win.cpp

2
lib-src/soundtouch/source/SoundTouch/SoundTouch.dsw
View File

@ -1,7 +1,7 @@
Microsoft Developer Studio Workspace File, Format Version 6.00
# WARNING: DO NOT EDIT OR DELETE THIS WORKSPACE FILE!
#
# $Id: SoundTouch.dsw,v 1.2 2006-09-18 07:31:41 richardash1981 Exp $
# $Id: SoundTouch.dsw 11 2008-02-10 16:26:55Z oparviai $
###############################################################################

1128
lib-src/soundtouch/source/SoundTouch/SoundTouch.vcproj
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File diff suppressed because it is too large Load Diff

740
lib-src/soundtouch/source/SoundTouch/TDStretch.cpp
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File diff suppressed because it is too large Load Diff

172
lib-src/soundtouch/source/SoundTouch/TDStretch.h
View File

@ -1,11 +1,11 @@
////////////////////////////////////////////////////////////////////////////////
///
/// Sampled sound tempo changer/time stretch algorithm. Changes the sound tempo
/// while maintaining the original pitch by using a time domain WSOLA-like method
///
/// 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'
/// Note : MMX/SSE optimized functions reside in separate, platform-specific files
/// 'mmx_optimized.cpp' and 'sse_optimized.cpp'
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
@ -13,10 +13,10 @@
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2006-09-18 22:29:22 $
// File revision : $Revision: 1.4 $
// Last changed : $Date: 2012-04-01 22:49:30 +0300 (Sun, 01 Apr 2012) $
// File revision : $Revision: 4 $
//
// $Id: TDStretch.h,v 1.4 2006-09-18 22:29:22 martynshaw Exp $
// $Id: TDStretch.h 137 2012-04-01 19:49:30Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
@ -44,6 +44,7 @@
#ifndef TDStretch_H
#define TDStretch_H
#include <stddef.h>
#include "STTypes.h"
#include "RateTransposer.h"
#include "FIFOSamplePipe.h"
@ -51,9 +52,15 @@
namespace soundtouch
{
// Default values for sound processing parameters:
/// Default values for sound processing parameters:
/// Notice that the default parameters are tuned for contemporary popular music
/// processing. For speech processing applications these parameters suit better:
/// #define DEFAULT_SEQUENCE_MS 40
/// #define DEFAULT_SEEKWINDOW_MS 15
/// #define DEFAULT_OVERLAP_MS 8
///
/// Default length of a single processing sequence, in milliseconds. This determines to how
/// 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
@ -61,31 +68,41 @@ namespace soundtouch
/// and vice versa.
///
/// Increasing this value reduces computational burden & vice versa.
#define DEFAULT_SEQUENCE_MS 82
//#define DEFAULT_SEQUENCE_MS 40
#define DEFAULT_SEQUENCE_MS USE_AUTO_SEQUENCE_LEN
/// 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.
/// Giving this value for the sequence length sets automatic parameter value
/// according to tempo setting (recommended)
#define USE_AUTO_SEQUENCE_LEN 0
/// Seeking window default length in milliseconds for algorithm that finds the best possible
/// overlapping location. This determines from how wide window the algorithm may look for an
/// optimal joining location when mixing the sound sequences back together.
///
/// The bigger this window setting is, the higher the possibility to find a better mixing
/// position will become, but at the same time large values may cause a "drifting" artifact
/// because consequent sequences will be taken at more uneven intervals.
///
/// If there's a disturbing artifact that sounds as if a constant frequency was drifting
/// 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
//#define DEFAULT_SEEKWINDOW_MS 15
#define DEFAULT_SEEKWINDOW_MS USE_AUTO_SEEKWINDOW_LEN
/// 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.
/// Giving this value for the seek window length sets automatic parameter value
/// according to tempo setting (recommended)
#define USE_AUTO_SEEKWINDOW_LEN 0
/// Overlap length in milliseconds. When the chopped sound sequences are mixed back together,
/// to form a continuous sound stream, this parameter defines over how long period the two
/// consecutive sequences are let to overlap each other.
///
/// This shouldn't be that critical parameter. If you reduce the DEFAULT_SEQUENCE_MS setting
/// 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
#define DEFAULT_OVERLAP_MS 8
/// Class that does the time-stretch (tempo change) effect for the processed
@ -93,44 +110,40 @@ namespace soundtouch
class TDStretch : public FIFOProcessor
{
protected:
uint channels;
uint sampleReq;
int channels;
int sampleReq;
float tempo;
SAMPLETYPE *pMidBuffer;
SAMPLETYPE *pRefMidBuffer;
SAMPLETYPE *pRefMidBufferUnaligned;
uint overlapLength;
uint overlapDividerBits;
uint slopingDivider;
uint seekLength;
uint seekWindowLength;
uint maxOffset;
SAMPLETYPE *pMidBufferUnaligned;
int overlapLength;
int seekLength;
int seekWindowLength;
int overlapDividerBits;
int slopingDivider;
float nominalSkip;
float skipFract;
FIFOSampleBuffer outputBuffer;
FIFOSampleBuffer inputBuffer;
BOOL bQuickseek;
BOOL bMidBufferDirty;
BOOL bQuickSeek;
uint sampleRate;
uint sequenceMs;
uint seekWindowMs;
uint overlapMs;
int sampleRate;
int sequenceMs;
int seekWindowMs;
int overlapMs;
BOOL bAutoSeqSetting;
BOOL bAutoSeekSetting;
void acceptNewOverlapLength(uint newOverlapLength);
void acceptNewOverlapLength(int newOverlapLength);
virtual void clearCrossCorrState();
void calculateOverlapLength(uint overlapMs);
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 double calcCrossCorr(const SAMPLETYPE *mixingPos, const SAMPLETYPE *compare) const;
virtual uint seekBestOverlapPositionStereo(const SAMPLETYPE *refPos);
virtual uint seekBestOverlapPositionStereoQuick(const SAMPLETYPE *refPos);
virtual uint seekBestOverlapPositionMono(const SAMPLETYPE *refPos);
virtual uint seekBestOverlapPositionMonoQuick(const SAMPLETYPE *refPos);
uint seekBestOverlapPosition(const SAMPLETYPE *refPos);
virtual int seekBestOverlapPositionFull(const SAMPLETYPE *refPos);
virtual int seekBestOverlapPositionQuick(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;
@ -138,37 +151,34 @@ protected:
void clearMidBuffer();
void overlap(SAMPLETYPE *output, const SAMPLETYPE *input, uint ovlPos) const;
void precalcCorrReferenceMono();
void precalcCorrReferenceStereo();
void processNominalTempo();
void calcSeqParameters();
/// Changes the tempo of the given sound samples.
/// Returns amount of samples returned in the "output" buffer.
/// The maximum amount of samples that can be returned at a time is set by
/// the 'set_returnBuffer_size' function.
void processSamples();
public:
TDStretch();
virtual ~TDStretch();
/// Operator 'new' is overloaded so that it automatically creates a suitable instance
/// 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 *operator new(size_t s);
static void *operator new(size_t s);
/// Use this function instead of "new" operator to create a new instance of this class.
/// 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
/// Sets new target tempo. Normal tempo = 'SCALE', smaller values represent slower
/// tempo, larger faster tempo.
void setTempo(float newTempo);
@ -179,9 +189,9 @@ public:
void clearInput();
/// Sets the number of channels, 1 = mono, 2 = stereo
void setChannels(uint numChannels);
void setChannels(int numChannels);
/// Enables/disables the quick position seeking algorithm. Zero to disable,
/// Enables/disables the quick position seeking algorithm. Zero to disable,
/// nonzero to enable
void enableQuickSeek(BOOL enable);
@ -193,19 +203,19 @@ public:
//
/// 'sampleRate' = sample rate of the sound
/// 'sequenceMS' = one processing sequence length in milliseconds
/// 'seekwindowMS' = seeking window length for scanning the best overlapping
/// 'seekwindowMS' = seeking window length for scanning the best overlapping
/// position
/// 'overlapMS' = overlapping length
void setParameters(uint sampleRate, ///< Samplerate of sound being processed (Hz)
uint sequenceMS = DEFAULT_SEQUENCE_MS, ///< Single processing sequence length (ms)
uint seekwindowMS = DEFAULT_SEEKWINDOW_MS, ///< Offset seeking window length (ms)
uint overlapMS = DEFAULT_OVERLAP_MS ///< Sequence overlapping length (ms)
void setParameters(int sampleRate, ///< Samplerate of sound being processed (Hz)
int sequenceMS = -1, ///< Single processing sequence length (ms)
int seekwindowMS = -1, ///< Offset seeking window length (ms)
int overlapMS = -1 ///< Sequence overlapping length (ms)
);
/// Get routine control parameters, see setParameters() function.
/// Any of the parameters to this function can be NULL, in such case corresponding parameter
/// value isn't returned.
void getParameters(uint *pSampleRate, uint *pSequenceMs, uint *pSeekWindowMs, uint *pOverlapMs);
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.
@ -214,43 +224,45 @@ public:
uint numSamples ///< Number of samples in 'samples' so that one sample
///< contains both channels if stereo
);
/// return nominal input sample requirement for triggering a processing batch
int getInputSampleReq() const
{
return (int)(nominalSkip + 0.5);
}
/// return nominal output sample amount when running a processing batch
int getOutputBatchSize() const
{
return seekWindowLength - overlapLength;
}
};
// Implementation-specific class declarations:
#ifdef ALLOW_MMX
#ifdef SOUNDTOUCH_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;
double calcCrossCorr(const short *mixingPos, const short *compare) const;
virtual void overlapStereo(short *output, const short *input) const;
virtual void clearCrossCorrState();
};
#endif /// ALLOW_MMX
#endif /// SOUNDTOUCH_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
#ifdef SOUNDTOUCH_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;
double calcCrossCorr(const float *mixingPos, const float *compare) const;
};
#endif /// ALLOW_SSE
#endif /// SOUNDTOUCH_ALLOW_SSE
}
#endif /// TDStretch_H

10
lib-src/soundtouch/source/SoundTouch/cpu_detect.h
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@ -2,8 +2,8 @@
///
/// 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
/// 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
@ -12,10 +12,10 @@
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2006-09-18 22:29:22 $
// File revision : $Revision: 1.4 $
// Last changed : $Date: 2008-02-10 18:26:55 +0200 (Sun, 10 Feb 2008) $
// File revision : $Revision: 4 $
//
// $Id: cpu_detect.h,v 1.4 2006-09-18 22:29:22 martynshaw Exp $
// $Id: cpu_detect.h 11 2008-02-10 16:26:55Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//

137
lib-src/soundtouch/source/SoundTouch/cpu_detect_x86.cpp Normal file
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@ -0,0 +1,137 @@
////////////////////////////////////////////////////////////////////////////////
///
/// Generic version of the x86 CPU extension detection routine.
///
/// This file is for GNU & other non-Windows compilers, see 'cpu_detect_x86_win.cpp'
/// for the Microsoft compiler version.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2012-11-08 20:44:37 +0200 (Thu, 08 Nov 2012) $
// File revision : $Revision: 4 $
//
// $Id: cpu_detect_x86.cpp 159 2012-11-08 18:44:37Z 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"
#if defined(SOUNDTOUCH_ALLOW_X86_OPTIMIZATIONS)
#if defined(__GNUC__) && defined(__i386__)
// gcc
#include "cpuid.h"
#elif defined(_M_IX86)
// windows non-gcc
#include <intrin.h>
#define bit_MMX (1 << 23)
#define bit_SSE (1 << 25)
#define bit_SSE2 (1 << 26)
#endif
#endif
//////////////////////////////////////////////////////////////////////////////
//
// processor instructions extension detection routines
//
//////////////////////////////////////////////////////////////////////////////
// Flag variable indicating whick ISA extensions are disabled (for debugging)
static uint _dwDisabledISA = 0x00; // 0xffffffff; //<- use this to disable all extensions
// Disables given set of instruction extensions. See SUPPORT_... defines.
void disableExtensions(uint dwDisableMask)
{
_dwDisabledISA = dwDisableMask;
}
/// Checks which instruction set extensions are supported by the CPU.
uint detectCPUextensions(void)
{
/// If building for a 64bit system (no Itanium) and the user wants optimizations.
/// Return the OR of SUPPORT_{MMX,SSE,SSE2}. 11001 or 0x19.
/// Keep the _dwDisabledISA test (2 more operations, could be eliminated).
#if ((defined(__GNUC__) && defined(__x86_64__)) \
|| defined(_M_X64)) \
&& defined(SOUNDTOUCH_ALLOW_X86_OPTIMIZATIONS)
return 0x19 & ~_dwDisabledISA;
/// If building for a 32bit system and the user wants optimizations.
/// Keep the _dwDisabledISA test (2 more operations, could be eliminated).
#elif ((defined(__GNUC__) && defined(__i386__)) \
|| defined(_M_IX86)) \
&& defined(SOUNDTOUCH_ALLOW_X86_OPTIMIZATIONS)
if (_dwDisabledISA == 0xffffffff) return 0;
uint res = 0;
#if defined(__GNUC__)
// GCC version of cpuid. Requires GCC 4.3.0 or later for __cpuid intrinsic support.
uint eax, ebx, ecx, edx; // unsigned int is the standard type. uint is defined by the compiler and not guaranteed to be portable.
// Check if no cpuid support.
if (!__get_cpuid (1, &eax, &ebx, &ecx, &edx)) return 0; // always disable extensions.
if (edx & bit_MMX) res = res | SUPPORT_MMX;
if (edx & bit_SSE) res = res | SUPPORT_SSE;
if (edx & bit_SSE2) res = res | SUPPORT_SSE2;
#else
// Window / VS version of cpuid. Notice that Visual Studio 2005 or later required
// for __cpuid intrinsic support.
int reg[4] = {-1};
// Check if no cpuid support.
__cpuid(reg,0);
if ((unsigned int)reg[0] == 0) return 0; // always disable extensions.
__cpuid(reg,1);
if ((unsigned int)reg[3] & bit_MMX) res = res | SUPPORT_MMX;
if ((unsigned int)reg[3] & bit_SSE) res = res | SUPPORT_SSE;
if ((unsigned int)reg[3] & bit_SSE2) res = res | SUPPORT_SSE2;
#endif
return res & ~_dwDisabledISA;
#else
/// One of these is true:
/// 1) We don't want optimizations.
/// 2) Using an unsupported compiler.
/// 3) Running on a non-x86 platform.
return 0;
#endif
}

138
lib-src/soundtouch/source/SoundTouch/cpu_detect_x86_gcc.cpp
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@ -1,138 +0,0 @@
////////////////////////////////////////////////////////////////////////////////
///
/// 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: 2006-09-18 07:31:42 $
// File revision : $Revision: 1.3 $
//
// $Id: cpu_detect_x86_gcc.cpp,v 1.3 2006-09-18 07:31:42 richardash1981 Exp $
//
////////////////////////////////////////////////////////////////////////////////
//
// 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
}

126
lib-src/soundtouch/source/SoundTouch/cpu_detect_x86_win.cpp
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@ -1,126 +0,0 @@
////////////////////////////////////////////////////////////////////////////////
///
/// 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: 2006-09-18 22:29:22 $
// File revision : $Revision: 1.4 $
//
// $Id: cpu_detect_x86_win.cpp,v 1.4 2006-09-18 22:29:22 martynshaw Exp $
//
////////////////////////////////////////////////////////////////////////////////
//
// 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;
}

80
lib-src/soundtouch/source/SoundTouch/mmx_optimized.cpp
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@ -1,18 +1,18 @@
////////////////////////////////////////////////////////////////////////////////
///
/// 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
/// 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++
/// 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
/// http://msdn.microsoft.com/en-us/vstudio/aa718349.aspx
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
@ -20,10 +20,10 @@
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2006-09-18 22:29:22 $
// File revision : $Revision: 1.3 $
// Last changed : $Date: 2012-11-08 20:53:01 +0200 (Thu, 08 Nov 2012) $
// File revision : $Revision: 4 $
//
// $Id: mmx_optimized.cpp,v 1.3 2006-09-18 22:29:22 martynshaw Exp $
// $Id: mmx_optimized.cpp 160 2012-11-08 18:53:01Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
@ -50,13 +50,9 @@
#include "STTypes.h"
#ifdef ALLOW_MMX
#ifdef SOUNDTOUCH_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;
//////////////////////////////////////////////////////////////////////////////
@ -68,28 +64,29 @@ using namespace soundtouch;
#include "TDStretch.h"
#include <mmintrin.h>
#include <limits.h>
#include <math.h>
// Calculates cross correlation of two buffers
long TDStretchMMX::calcCrossCorrStereo(const short *pV1, const short *pV2) const
double TDStretchMMX::calcCrossCorr(const short *pV1, const short *pV2) const
{
const __m64 *pVec1, *pVec2;
__m64 shifter;
__m64 accu;
long corr;
uint i;
__m64 accu, normaccu;
long corr, norm;
int i;
pVec1 = (__m64*)pV1;
pVec2 = (__m64*)pV2;
shifter = _m_from_int(overlapDividerBits);
accu = _mm_setzero_si64();
normaccu = accu = _mm_setzero_si64();
// Process 4 parallel sets of 2 * stereo samples each during each
// round to improve CPU-level parallellization.
for (i = 0; i < overlapLength / 8; i ++)
// Process 4 parallel sets of 2 * stereo samples or 4 * mono samples
// during each round for improved CPU-level parallellization.
for (i = 0; i < channels * overlapLength / 16; i ++)
{
__m64 temp;
__m64 temp, temp2;
// dictionary of instructions:
// _m_pmaddwd : 4*16bit multiply-add, resulting two 32bits = [a0*b0+a1*b1 ; a2*b2+a3*b3]
@ -98,11 +95,17 @@ long TDStretchMMX::calcCrossCorrStereo(const short *pV1, const short *pV2) const
temp = _mm_add_pi32(_mm_madd_pi16(pVec1[0], pVec2[0]),
_mm_madd_pi16(pVec1[1], pVec2[1]));
temp2 = _mm_add_pi32(_mm_madd_pi16(pVec1[0], pVec1[0]),
_mm_madd_pi16(pVec1[1], pVec1[1]));
accu = _mm_add_pi32(accu, _mm_sra_pi32(temp, shifter));
normaccu = _mm_add_pi32(normaccu, _mm_sra_pi32(temp2, shifter));
temp = _mm_add_pi32(_mm_madd_pi16(pVec1[2], pVec2[2]),
_mm_madd_pi16(pVec1[3], pVec2[3]));
temp2 = _mm_add_pi32(_mm_madd_pi16(pVec1[2], pVec1[2]),
_mm_madd_pi16(pVec1[3], pVec1[3]));
accu = _mm_add_pi32(accu, _mm_sra_pi32(temp, shifter));
normaccu = _mm_add_pi32(normaccu, _mm_sra_pi32(temp2, shifter));
pVec1 += 4;
pVec2 += 4;
@ -114,10 +117,17 @@ long TDStretchMMX::calcCrossCorrStereo(const short *pV1, const short *pV2) const
accu = _mm_add_pi32(accu, _mm_srli_si64(accu, 32));
corr = _m_to_int(accu);
normaccu = _mm_add_pi32(normaccu, _mm_srli_si64(normaccu, 32));
norm = _m_to_int(normaccu);
// Clear MMS state
_m_empty();
return corr;
// Normalize result by dividing by sqrt(norm) - this step is easiest
// done using floating point operation
if (norm == 0) norm = 1; // to avoid div by zero
return (double)corr / sqrt((double)norm);
// Note: Warning about the missing EMMS instruction is harmless
// as it'll be called elsewhere.
}
@ -139,7 +149,7 @@ void TDStretchMMX::overlapStereo(short *output, const short *input) const
const __m64 *pVinput, *pVMidBuf;
__m64 *pVdest;
__m64 mix1, mix2, adder, shifter;
uint i;
int i;
pVinput = (const __m64*)input;
pVMidBuf = (const __m64*)pMidBuffer;
@ -148,18 +158,20 @@ void TDStretchMMX::overlapStereo(short *output, const short *input) const
// 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);
// Overlaplength-division by shifter. "+1" is to account for "-1" deduced in
// overlapDividerBits calculation earlier.
shifter = _m_from_int(overlapDividerBits + 1);
for (i = 0; i < overlapLength / 4; i ++)
{
__m64 temp1, temp2;
// load & shuffle data so that input & mixbuffer data samples are paired
temp1 = _mm_unpacklo_pi16(pVMidBuf[0], pVinput[0]); // = i0l m0l i0r m0r
temp2 = _mm_unpackhi_pi16(pVMidBuf[0], pVinput[0]); // = i1l m1l i1r m1r
@ -227,10 +239,10 @@ void FIRFilterMMX::setCoefficients(const short *coeffs, uint newLength, uint uRe
// 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);
filterCoeffsAlign = (short *)SOUNDTOUCH_ALIGN_POINTER_16(filterCoeffsUnalign);
// rearrange the filter coefficients for mmx routines
for (i = 0;i < length; i += 4)
// 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];
@ -247,7 +259,7 @@ void FIRFilterMMX::setCoefficients(const short *coeffs, uint newLength, uint uRe
// mmx-optimized version of the filter routine for stereo sound
uint FIRFilterMMX::evaluateFilterStereo(short *dest, const short *src, const uint numSamples) const
uint FIRFilterMMX::evaluateFilterStereo(short *dest, const short *src, uint numSamples) const
{
// Create stack copies of the needed member variables for asm routines :
uint i, j;
@ -255,7 +267,7 @@ uint FIRFilterMMX::evaluateFilterStereo(short *dest, const short *src, const uin
if (length < 2) return 0;
for (i = 0; i < numSamples / 2; i ++)
for (i = 0; i < (numSamples - length) / 2; i ++)
{
__m64 accu1;
__m64 accu2;
@ -302,4 +314,4 @@ uint FIRFilterMMX::evaluateFilterStereo(short *dest, const short *src, const uin
return (numSamples & 0xfffffffe) - length;
}
#endif // ALLOW_MMX
#endif // SOUNDTOUCH_ALLOW_MMX

289
lib-src/soundtouch/source/SoundTouch/sse_optimized.cpp
View File

@ -1,20 +1,20 @@
////////////////////////////////////////////////////////////////////////////////
///
/// 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
/// 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
/// 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
/// http://msdn.microsoft.com/en-us/vstudio/aa718349.aspx
///
/// If the above URL is expired or removed, go to "http://msdn.microsoft.com" and
/// 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
@ -23,10 +23,10 @@
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2006-09-18 22:29:23 $
// File revision : $Revision: 1.3 $
// Last changed : $Date: 2012-11-08 20:53:01 +0200 (Thu, 08 Nov 2012) $
// File revision : $Revision: 4 $
//
// $Id: sse_optimized.cpp,v 1.3 2006-09-18 22:29:23 martynshaw Exp $
// $Id: sse_optimized.cpp 160 2012-11-08 18:53:01Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
@ -56,9 +56,9 @@
using namespace soundtouch;
#ifdef ALLOW_SSE
#ifdef SOUNDTOUCH_ALLOW_SSE
// SSE routines available only with float sample type
// SSE routines available only with float sample type
//////////////////////////////////////////////////////////////////////////////
//
@ -68,74 +68,92 @@ using namespace soundtouch;
#include "TDStretch.h"
#include <xmmintrin.h>
#include <math.h>
// Calculates cross correlation of two buffers
double TDStretchSSE::calcCrossCorrStereo(const float *pV1, const float *pV2) const
double TDStretchSSE::calcCrossCorr(const float *pV1, const float *pV2) const
{
uint i;
__m128 vSum, *pVec2;
int i;
const float *pVec1;
const __m128 *pVec2;
__m128 vSum, vNorm;
// Note. It means a major slow-down if the routine needs to tolerate
// unaligned __m128 memory accesses. It's way faster if we can skip
// 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
// Compile-time define SOUNDTOUCH_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
#ifdef SOUNDTOUCH_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
if (((ulongptr)pV1) & 15) return -1e50; // skip unaligned locations
#else
// No cheating allowed, use unaligned load & take the resulting
// performance hit.
#define _MM_LOAD _mm_loadu_ps
#endif
#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();
pVec1 = (const float*)pV1;
pVec2 = (const __m128*)pV2;
vSum = vNorm = _mm_setzero_ps();
// Unroll the loop by factor of 4 * 4 operations
for (i = 0; i < overlapLength / 8; i ++)
// Unroll the loop by factor of 4 * 4 operations. Use same routine for
// stereo & mono, for mono it just means twice the amount of unrolling.
for (i = 0; i < channels * overlapLength / 16; i ++)
{
__m128 vTemp;
// vSum += pV1[0..3] * pV2[0..3]
vSum = _mm_add_ps(vSum, _mm_mul_ps(_MM_LOAD(pV1),pVec2[0]));
vTemp = _MM_LOAD(pVec1);
vSum = _mm_add_ps(vSum, _mm_mul_ps(vTemp ,pVec2[0]));
vNorm = _mm_add_ps(vNorm, _mm_mul_ps(vTemp ,vTemp));
// vSum += pV1[4..7] * pV2[4..7]
vSum = _mm_add_ps(vSum, _mm_mul_ps(_MM_LOAD(pV1 + 4), pVec2[1]));
vTemp = _MM_LOAD(pVec1 + 4);
vSum = _mm_add_ps(vSum, _mm_mul_ps(vTemp, pVec2[1]));
vNorm = _mm_add_ps(vNorm, _mm_mul_ps(vTemp ,vTemp));
// vSum += pV1[8..11] * pV2[8..11]
vSum = _mm_add_ps(vSum, _mm_mul_ps(_MM_LOAD(pV1 + 8), pVec2[2]));
vTemp = _MM_LOAD(pVec1 + 8);
vSum = _mm_add_ps(vSum, _mm_mul_ps(vTemp, pVec2[2]));
vNorm = _mm_add_ps(vNorm, _mm_mul_ps(vTemp ,vTemp));
// vSum += pV1[12..15] * pV2[12..15]
vSum = _mm_add_ps(vSum, _mm_mul_ps(_MM_LOAD(pV1 + 12), pVec2[3]));
vTemp = _MM_LOAD(pVec1 + 12);
vSum = _mm_add_ps(vSum, _mm_mul_ps(vTemp, pVec2[3]));
vNorm = _mm_add_ps(vNorm, _mm_mul_ps(vTemp ,vTemp));
pV1 += 16;
pVec1 += 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]);
float *pvNorm = (float*)&vNorm;
double norm = sqrt(pvNorm[0] + pvNorm[1] + pvNorm[2] + pvNorm[3]);
if (norm < 1e-9) norm = 1.0; // to avoid div by zero
/* This is approximately corresponding routine in C-language:
double corr;
float *pvSum = (float*)&vSum;
return (double)(pvSum[0] + pvSum[1] + pvSum[2] + pvSum[3]) / norm;
/* This is approximately corresponding routine in C-language yet without normalization:
double corr, norm;
uint i;
// Calculates the cross-correlation value between 'pV1' and 'pV2' vectors
corr = 0.0;
for (i = 0; i < overlapLength / 8; i ++)
corr = norm = 0.0;
for (i = 0; i < channels * overlapLength / 16; i ++)
{
corr += pV1[0] * pV2[0] +
pV1[1] * pV2[1] +
@ -154,77 +172,12 @@ double TDStretchSSE::calcCrossCorrStereo(const float *pV1, const float *pV2) con
pV1[14] * pV2[14] +
pV1[15] * pV2[15];
for (j = 0; j < 15; j ++) norm += pV1[j] * pV1[j];
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;
return corr / sqrt(norm);
*/
}
@ -239,6 +192,7 @@ double TDStretchSSE::calcCrossCorrStereo(const float *pV1, const float *pV2) con
FIRFilterSSE::FIRFilterSSE() : FIRFilter()
{
filterCoeffsAlign = NULL;
filterCoeffsUnalign = NULL;
}
@ -246,6 +200,8 @@ FIRFilterSSE::FIRFilterSSE() : FIRFilter()
FIRFilterSSE::~FIRFilterSSE()
{
delete[] filterCoeffsUnalign;
filterCoeffsAlign = NULL;
filterCoeffsUnalign = NULL;
}
@ -258,15 +214,15 @@ void FIRFilterSSE::setCoefficients(const float *coeffs, uint newLength, uint uRe
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!
// also rearrange coefficients suitably for SSE
// Ensure that filter coeffs array is aligned to 16-byte boundary
delete[] filterCoeffsUnalign;
filterCoeffsUnalign = new float[2 * newLength + 4];
filterCoeffsAlign = (float *)(((unsigned long)filterCoeffsUnalign + 15) & -16);
filterCoeffsAlign = (float *)SOUNDTOUCH_ALIGN_POINTER_16(filterCoeffsUnalign);
fDivider = (float)resultDivider;
// rearrange the filter coefficients for mmx routines
// rearrange the filter coefficients for mmx routines
for (i = 0; i < newLength; i ++)
{
filterCoeffsAlign[2 * i + 0] =
@ -279,15 +235,18 @@ void FIRFilterSSE::setCoefficients(const float *coeffs, uint newLength, uint uRe
// SSE-optimized version of the filter routine for stereo sound
uint FIRFilterSSE::evaluateFilterStereo(float *dest, const float *source, uint numSamples) const
{
int count = (numSamples - length) & -2;
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(((unsigned long)filterCoeffsAlign) % 16 == 0);
assert(filterCoeffsAlign != NULL);
assert(((ulongptr)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)
@ -297,14 +256,14 @@ uint FIRFilterSSE::evaluateFilterStereo(float *dest, const float *source, uint n
__m128 sum1, sum2;
uint i;
pSrc = source; // source audio data
pFil = (__m128*)filterCoeffsAlign; // filter coefficients. NOTE: Assumes coefficients
// are aligned to 16-byte boundary
pSrc = (const float*)source; // source audio data
pFil = (const __m128*)filterCoeffsAlign; // filter coefficients. NOTE: Assumes coefficients
// are aligned to 16-byte boundary
sum1 = sum2 = _mm_setzero_ps();
for (i = 0; i < length / 8; i ++)
for (i = 0; i < length / 8; i ++)
{
// Unroll loop for efficiency & calculate filter for 2*2 stereo samples
// 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
@ -339,14 +298,14 @@ uint FIRFilterSSE::evaluateFilterStereo(float *dest, const float *source, uint n
}
// Ideas for further improvement:
// 1. If it could be guaranteed that 'source' were always aligned to 16-byte
// 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
// 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
/* original routine in C-language. please notice the C-version has differently
organized coefficients though.
double suml1, suml2;
double sumr1, sumr2;
@ -361,26 +320,26 @@ uint FIRFilterSSE::evaluateFilterStereo(float *dest, const float *source, uint n
suml2 = sumr2 = 0.0;
ptr = src;
pFil = filterCoeffs;
for (i = 0; i < lengthLocal; i ++)
for (i = 0; i < lengthLocal; i ++)
{
// unroll loop for efficiency.
suml1 += ptr[0] * pFil[0] +
suml1 += ptr[0] * pFil[0] +
ptr[2] * pFil[2] +
ptr[4] * pFil[4] +
ptr[6] * pFil[6];
sumr1 += ptr[1] * pFil[1] +
sumr1 += ptr[1] * pFil[1] +
ptr[3] * pFil[3] +
ptr[5] * pFil[5] +
ptr[7] * pFil[7];
suml2 += ptr[8] * pFil[0] +
suml2 += ptr[8] * pFil[0] +
ptr[10] * pFil[2] +
ptr[12] * pFil[4] +
ptr[14] * pFil[6];
sumr2 += ptr[9] * pFil[1] +
sumr2 += ptr[9] * pFil[1] +
ptr[11] * pFil[3] +
ptr[13] * pFil[5] +
ptr[15] * pFil[7];
@ -397,88 +356,6 @@ uint FIRFilterSSE::evaluateFilterStereo(float *dest, const float *source, uint n
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
#endif // SOUNDTOUCH_ALLOW_SSE

338
lib-src/soundtouch/source/SoundTouchDLL/SoundTouchDLL.cpp Normal file
View File

@ -0,0 +1,338 @@
//////////////////////////////////////////////////////////////////////////////
///
/// SoundTouch DLL wrapper - wraps SoundTouch routines into a Dynamic Load
/// Library interface.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// $Id: SoundTouchDLL.cpp 96 2010-12-12 19:12:12Z 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 <windows.h>
#include <string.h>
#include "SoundTouchDLL.h"
#include "soundtouch.h"
using namespace soundtouch;
BOOL APIENTRY DllMain( HANDLE hModule,
DWORD ul_reason_for_call,
LPVOID lpReserved
)
{
switch (ul_reason_for_call)
{
case DLL_PROCESS_ATTACH:
case DLL_THREAD_ATTACH:
case DLL_THREAD_DETACH:
case DLL_PROCESS_DETACH:
break;
}
return TRUE;
}
//////////////
typedef struct
{
DWORD dwMagic;
SoundTouch *pst;
} STHANDLE;
#define STMAGIC 0x1770C001
SOUNDTOUCHDLL_API HANDLE __stdcall soundtouch_createInstance()
{
STHANDLE *tmp = new STHANDLE;
if (tmp)
{
tmp->dwMagic = STMAGIC;
tmp->pst = new SoundTouch();
if (tmp->pst == NULL)
{
delete tmp;
tmp = NULL;
}
}
return (HANDLE)tmp;
}
SOUNDTOUCHDLL_API void __stdcall soundtouch_destroyInstance(HANDLE h)
{
/*
STHANDLE *sth = (STHANDLE*)h;
if (sth->dwMagic != STMAGIC) return;
*/
/*
sth->dwMagic = 0;
delete sth->pst;
sth->pst = NULL;
delete sth;
*/
}
/// Get SoundTouch library version string
SOUNDTOUCHDLL_API const char *__stdcall soundtouch_getVersionString()
{
return SoundTouch::getVersionString();
}
/// Get SoundTouch library version string - alternative function for
/// environments that can't properly handle character string as return value
SOUNDTOUCHDLL_API void __stdcall soundtouch_getVersionString2(char* versionString, int bufferSize)
{
strncpy(versionString, SoundTouch::getVersionString(), bufferSize - 1);
versionString[bufferSize - 1] = 0;
}
/// Get SoundTouch library version Id
SOUNDTOUCHDLL_API uint __stdcall soundtouch_getVersionId()
{
return SoundTouch::getVersionId();
}
/// Sets new rate control value. Normal rate = 1.0, smaller values
/// represent slower rate, larger faster rates.
SOUNDTOUCHDLL_API void __stdcall soundtouch_setRate(HANDLE h, float newRate)
{
STHANDLE *sth = (STHANDLE*)h;
if (sth->dwMagic != STMAGIC) return;
sth->pst->setRate(newRate);
}
/// Sets new tempo control value. Normal tempo = 1.0, smaller values
/// represent slower tempo, larger faster tempo.
SOUNDTOUCHDLL_API void __stdcall soundtouch_setTempo(HANDLE h, float newTempo)
{
STHANDLE *sth = (STHANDLE*)h;
if (sth->dwMagic != STMAGIC) return;
sth->pst->setTempo(newTempo);
}
/// Sets new rate control value as a difference in percents compared
/// to the original rate (-50 .. +100 %)
SOUNDTOUCHDLL_API void __stdcall soundtouch_setRateChange(HANDLE h, float newRate)
{
STHANDLE *sth = (STHANDLE*)h;
if (sth->dwMagic != STMAGIC) return;
sth->pst->setRateChange(newRate);
}
/// Sets new tempo control value as a difference in percents compared
/// to the original tempo (-50 .. +100 %)
SOUNDTOUCHDLL_API void __stdcall soundtouch_setTempoChange(HANDLE h, float newTempo)
{
STHANDLE *sth = (STHANDLE*)h;
if (sth->dwMagic != STMAGIC) return;
sth->pst->setTempoChange(newTempo);
}
/// Sets new pitch control value. Original pitch = 1.0, smaller values
/// represent lower pitches, larger values higher pitch.
SOUNDTOUCHDLL_API void __stdcall soundtouch_setPitch(HANDLE h, float newPitch)
{
STHANDLE *sth = (STHANDLE*)h;
if (sth->dwMagic != STMAGIC) return;
sth->pst->setPitch(newPitch);
}
/// Sets pitch change in octaves compared to the original pitch
/// (-1.00 .. +1.00)
SOUNDTOUCHDLL_API void __stdcall soundtouch_setPitchOctaves(HANDLE h, float newPitch)
{
STHANDLE *sth = (STHANDLE*)h;
if (sth->dwMagic != STMAGIC) return;
sth->pst->setPitchOctaves(newPitch);
}
/// Sets pitch change in semi-tones compared to the original pitch
/// (-12 .. +12)
SOUNDTOUCHDLL_API void __stdcall soundtouch_setPitchSemiTones(HANDLE h, float newPitch)
{
STHANDLE *sth = (STHANDLE*)h;
if (sth->dwMagic != STMAGIC) return;
sth->pst->setPitchSemiTones(newPitch);
}
/// Sets the number of channels, 1 = mono, 2 = stereo
SOUNDTOUCHDLL_API void __stdcall soundtouch_setChannels(HANDLE h, uint numChannels)
{
STHANDLE *sth = (STHANDLE*)h;
if (sth->dwMagic != STMAGIC) return;
sth->pst->setChannels(numChannels);
}
/// Sets sample rate.
SOUNDTOUCHDLL_API void __stdcall soundtouch_setSampleRate(HANDLE h, uint srate)
{
STHANDLE *sth = (STHANDLE*)h;
if (sth->dwMagic != STMAGIC) return;
sth->pst->setSampleRate(srate);
}
/// Flushes the last samples from the processing pipeline to the output.
/// Clears also the internal processing buffers.
//
/// Note: This function is meant for extracting the last samples of a sound
/// stream. This function may introduce additional blank samples in the end
/// of the sound stream, and thus it's not recommended to call this function
/// in the middle of a sound stream.
SOUNDTOUCHDLL_API void __stdcall soundtouch_flush(HANDLE h)
{
STHANDLE *sth = (STHANDLE*)h;
if (sth->dwMagic != STMAGIC) return;
sth->pst->flush();
}
/// Adds 'numSamples' pcs of samples from the 'samples' memory position into
/// the input of the object. Notice that sample rate _has_to_ be set before
/// calling this function, otherwise throws a runtime_error exception.
SOUNDTOUCHDLL_API void __stdcall soundtouch_putSamples(HANDLE h,
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.
)
{
STHANDLE *sth = (STHANDLE*)h;
if (sth->dwMagic != STMAGIC) return;
sth->pst->putSamples(samples, numSamples);
}
/// Clears all the samples in the object's output and internal processing
/// buffers.
SOUNDTOUCHDLL_API void __stdcall soundtouch_clear(HANDLE h)
{
STHANDLE *sth = (STHANDLE*)h;
if (sth->dwMagic != STMAGIC) return;
sth->pst->clear();
}
/// Changes a setting controlling the processing system behaviour. See the
/// 'SETTING_...' defines for available setting ID's.
///
/// \return 'TRUE' if the setting was succesfully changed
SOUNDTOUCHDLL_API BOOL __stdcall soundtouch_setSetting(HANDLE h,
int settingId, ///< Setting ID number. see SETTING_... defines.
int value ///< New setting value.
)
{
STHANDLE *sth = (STHANDLE*)h;
if (sth->dwMagic != STMAGIC) return FALSE;
return sth->pst->setSetting(settingId, value);
}
/// Reads a setting controlling the processing system behaviour. See the
/// 'SETTING_...' defines for available setting ID's.
///
/// \return the setting value.
SOUNDTOUCHDLL_API int __stdcall soundtouch_getSetting(HANDLE h,
int settingId ///< Setting ID number, see SETTING_... defines.
)
{
STHANDLE *sth = (STHANDLE*)h;
if (sth->dwMagic != STMAGIC) return -1;
return sth->pst->getSetting(settingId);
}
/// Returns number of samples currently unprocessed.
SOUNDTOUCHDLL_API uint __stdcall soundtouch_numUnprocessedSamples(HANDLE h)
{
STHANDLE *sth = (STHANDLE*)h;
if (sth->dwMagic != STMAGIC) return 0;
return sth->pst->numUnprocessedSamples();
}
/// Adjusts book-keeping so that given number of samples are removed from beginning of the
/// sample buffer without copying them anywhere.
///
/// Used to reduce the number of samples in the buffer when accessing the sample buffer directly
/// with 'ptrBegin' function.
SOUNDTOUCHDLL_API uint __stdcall soundtouch_receiveSamples(HANDLE h,
SAMPLETYPE *outBuffer, ///< Buffer where to copy output samples.
uint maxSamples ///< How many samples to receive at max.
)
{
STHANDLE *sth = (STHANDLE*)h;
if (sth->dwMagic != STMAGIC) return 0;
if (outBuffer)
{
return sth->pst->receiveSamples(outBuffer, maxSamples);
}
else
{
return sth->pst->receiveSamples(maxSamples);
}
}
/// Returns number of samples currently available.
SOUNDTOUCHDLL_API uint __stdcall soundtouch_numSamples(HANDLE h)
{
STHANDLE *sth = (STHANDLE*)h;
if (sth->dwMagic != STMAGIC) return 0;
return sth->pst->numSamples();
}
/// Returns nonzero if there aren't any samples available for outputting.
SOUNDTOUCHDLL_API int __stdcall soundtouch_isEmpty(HANDLE h)
{
STHANDLE *sth = (STHANDLE*)h;
if (sth->dwMagic != STMAGIC) return -1;
return sth->pst->isEmpty();
}

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//////////////////////////////////////////////////////////////////////////////
///
/// SoundTouch DLL wrapper - wraps SoundTouch routines into a Dynamic Load
/// Library interface.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// $Id: SoundTouchDLL.h 94 2010-12-12 18:28:49Z 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 _SoundTouchDLL_h_
#define _SoundTouchDLL_h_
#ifdef __cplusplus
extern "C" {
#endif
#ifdef DLL_EXPORTS
#define SOUNDTOUCHDLL_API __declspec(dllexport)
#else
#define SOUNDTOUCHDLL_API __declspec(dllimport)
#endif
typedef void * HANDLE;
/// Create a new instance of SoundTouch processor.
SOUNDTOUCHDLL_API HANDLE __stdcall soundtouch_createInstance();
/// Destroys a SoundTouch processor instance.
SOUNDTOUCHDLL_API void __stdcall soundtouch_destroyInstance(HANDLE h);
/// Get SoundTouch library version string
SOUNDTOUCHDLL_API const char *__stdcall soundtouch_getVersionString();
/// Get SoundTouch library version string - alternative function for
/// environments that can't properly handle character string as return value
SOUNDTOUCHDLL_API void __stdcall soundtouch_getVersionString2(char* versionString, int bufferSize);
/// Get SoundTouch library version Id
SOUNDTOUCHDLL_API unsigned int __stdcall soundtouch_getVersionId();
/// Sets new rate control value. Normal rate = 1.0, smaller values
/// represent slower rate, larger faster rates.
SOUNDTOUCHDLL_API void __stdcall soundtouch_setRate(HANDLE h, float newRate);
/// Sets new tempo control value. Normal tempo = 1.0, smaller values
/// represent slower tempo, larger faster tempo.
SOUNDTOUCHDLL_API void __stdcall soundtouch_setTempo(HANDLE h, float newTempo);
/// Sets new rate control value as a difference in percents compared
/// to the original rate (-50 .. +100 %);
SOUNDTOUCHDLL_API void __stdcall soundtouch_setRateChange(HANDLE h, float newRate);
/// Sets new tempo control value as a difference in percents compared
/// to the original tempo (-50 .. +100 %);
SOUNDTOUCHDLL_API void __stdcall soundtouch_setTempoChange(HANDLE h, float newTempo);
/// Sets new pitch control value. Original pitch = 1.0, smaller values
/// represent lower pitches, larger values higher pitch.
SOUNDTOUCHDLL_API void __stdcall soundtouch_setPitch(HANDLE h, float newPitch);
/// Sets pitch change in octaves compared to the original pitch
/// (-1.00 .. +1.00);
SOUNDTOUCHDLL_API void __stdcall soundtouch_setPitchOctaves(HANDLE h, float newPitch);
/// Sets pitch change in semi-tones compared to the original pitch
/// (-12 .. +12);
SOUNDTOUCHDLL_API void __stdcall soundtouch_setPitchSemiTones(HANDLE h, float newPitch);
/// Sets the number of channels, 1 = mono, 2 = stereo
SOUNDTOUCHDLL_API void __stdcall soundtouch_setChannels(HANDLE h, unsigned int numChannels);
/// Sets sample rate.
SOUNDTOUCHDLL_API void __stdcall soundtouch_setSampleRate(HANDLE h, unsigned int srate);
/// Flushes the last samples from the processing pipeline to the output.
/// Clears also the internal processing buffers.
//
/// Note: This function is meant for extracting the last samples of a sound
/// stream. This function may introduce additional blank samples in the end
/// of the sound stream, and thus it's not recommended to call this function
/// in the middle of a sound stream.
SOUNDTOUCHDLL_API void __stdcall soundtouch_flush(HANDLE h);
/// Adds 'numSamples' pcs of samples from the 'samples' memory position into
/// the input of the object. Notice that sample rate _has_to_ be set before
/// calling this function, otherwise throws a runtime_error exception.
SOUNDTOUCHDLL_API void __stdcall soundtouch_putSamples(HANDLE h,
const float *samples, ///< Pointer to sample buffer.
unsigned int numSamples ///< Number of samples in buffer. Notice
///< that in case of stereo-sound a single sample
///< contains data for both channels.
);
/// Clears all the samples in the object's output and internal processing
/// buffers.
SOUNDTOUCHDLL_API void __stdcall soundtouch_clear(HANDLE h);
/// 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
SOUNDTOUCHDLL_API BOOL __stdcall soundtouch_setSetting(HANDLE h,
int settingId, ///< Setting ID number. see SETTING_... defines.
int value ///< New setting value.
);
/// Reads a setting controlling the processing system behaviour. See the
/// 'SETTING_...' defines for available setting ID's.
///
/// \return the setting value.
SOUNDTOUCHDLL_API int __stdcall soundtouch_getSetting(HANDLE h,
int settingId ///< Setting ID number, see SETTING_... defines.
);
/// Returns number of samples currently unprocessed.
SOUNDTOUCHDLL_API unsigned int __stdcall soundtouch_numUnprocessedSamples(HANDLE h);
/// Adjusts book-keeping so that given number of samples are removed from beginning of the
/// sample buffer without copying them anywhere.
///
/// Used to reduce the number of samples in the buffer when accessing the sample buffer directly
/// with 'ptrBegin' function.
SOUNDTOUCHDLL_API unsigned int __stdcall soundtouch_receiveSamples(HANDLE h,
float *outBuffer, ///< Buffer where to copy output samples.
unsigned int maxSamples ///< How many samples to receive at max.
);
/// Returns number of samples currently available.
SOUNDTOUCHDLL_API unsigned int __stdcall soundtouch_numSamples(HANDLE h);
/// Returns nonzero if there aren't any samples available for outputting.
SOUNDTOUCHDLL_API int __stdcall soundtouch_isEmpty(HANDLE h);
#ifdef __cplusplus
}
#endif
#endif // _SoundTouchDLL_h_

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unit SoundTouchDLL;
//////////////////////////////////////////////////////////////////////////////
//
// SoundTouch.dll wrapper for accessing SoundTouch routines from Delphi/Pascal
//
// Module Author : Christian Budde
//
////////////////////////////////////////////////////////////////////////////////
//
// $Id: SoundTouchDLL.pas 66 2009-02-24 14:32:44Z 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
//
////////////////////////////////////////////////////////////////////////////////
interface
uses
Windows;
type
TSoundTouchHandle = THandle;
// Create a new instance of SoundTouch processor.
TSoundTouchCreateInstance = function : TSoundTouchHandle; stdcall;
// Destroys a SoundTouch processor instance.
TSoundTouchDestroyInstance = procedure (Handle: TSoundTouchHandle); stdcall;
// Get SoundTouch library version string
TSoundTouchGetVersionString = function : PChar; stdcall;
// Get SoundTouch library version Id
TSoundTouchGetVersionId = function : Cardinal; stdcall;
// Sets new rate control value. Normal rate = 1.0, smaller values
// represent slower rate, larger faster rates.
TSoundTouchSetRate = procedure (Handle: TSoundTouchHandle; newRate: Single); stdcall;
// Sets new tempo control value. Normal tempo = 1.0, smaller values
// represent slower tempo, larger faster tempo.
TSoundTouchSetTempo = procedure (Handle: TSoundTouchHandle; newTempo: Single); stdcall;
// Sets new rate control value as a difference in percents compared
// to the original rate (-50 .. +100 %);
TSoundTouchSetRateChange = procedure (Handle: TSoundTouchHandle; newRate: Single); stdcall;
// Sets new tempo control value as a difference in percents compared
// to the original tempo (-50 .. +100 %);
TSoundTouchSetTempoChange = procedure (Handle: TSoundTouchHandle; newTempo: Single); stdcall;
// Sets new pitch control value. Original pitch = 1.0, smaller values
// represent lower pitches, larger values higher pitch.
TSoundTouchSetPitch = procedure (Handle: TSoundTouchHandle; newPitch: Single); stdcall;
// Sets pitch change in octaves compared to the original pitch
// (-1.00 .. +1.00);
TSoundTouchSetPitchOctaves = procedure (Handle: TSoundTouchHandle; newPitch: Single); stdcall;
// Sets pitch change in semi-tones compared to the original pitch
// (-12 .. +12);
TSoundTouchSetPitchSemiTones = procedure (Handle: TSoundTouchHandle; newPitch: Single); stdcall;
// Sets the number of channels, 1 = mono, 2 = stereo
TSoundTouchSetChannels = procedure (Handle: TSoundTouchHandle; numChannels: Cardinal); stdcall;
// Sets sample rate.
TSoundTouchSetSampleRate = procedure (Handle: TSoundTouchHandle; SampleRate: Cardinal); stdcall;
// 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
// in the middle of a sound stream.
TSoundTouchFlush = procedure (Handle: TSoundTouchHandle); stdcall;
// 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.
TSoundTouchPutSamples = procedure (Handle: TSoundTouchHandle;
const Samples: PSingle; //< Pointer to sample buffer.
NumSamples: Cardinal //< Number of samples in buffer. Notice
//< that in case of stereo-sound a single sample
//< contains data for both channels.
); stdcall;
// Clears all the samples in the object's output and internal processing
// buffers.
TSoundTouchClear = procedure (Handle: TSoundTouchHandle); stdcall;
// 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
TSoundTouchSetSetting = function (Handle: TSoundTouchHandle;
SettingId: Integer; //< Setting ID number. see SETTING_... defines.
Value: Integer //< New setting value.
): Boolean; stdcall;
// Reads a setting controlling the processing system behaviour. See the
// 'SETTING_...' defines for available setting ID's.
//
// \return the setting value.
TSoundTouchGetSetting = function (Handle: TSoundTouchHandle;
settingId: Integer //< Setting ID number, see SETTING_... defines.
): Integer; stdcall;
// Returns number of samples currently unprocessed.
TSoundTouchNumUnprocessedSamples = function (Handle: TSoundTouchHandle): Cardinal; stdcall;
// 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.
TSoundTouchReceiveSamples = function (Handle: TSoundTouchHandle;
outBuffer: PSingle; //< Buffer where to copy output samples.
maxSamples: Integer //< How many samples to receive at max.
): Cardinal; stdcall;
// Returns number of samples currently available.
TSoundTouchNumSamples = function (Handle: TSoundTouchHandle): Cardinal; stdcall;
// Returns nonzero if there aren't any samples available for outputting.
TSoundTouchIsEmpty = function (Handle: TSoundTouchHandle): Integer; stdcall;
var
SoundTouchCreateInstance : TSoundTouchCreateInstance;
SoundTouchDestroyInstance : TSoundTouchDestroyInstance;
SoundTouchGetVersionString : TSoundTouchGetVersionString;
SoundTouchGetVersionId : TSoundTouchGetVersionId;
SoundTouchSetRate : TSoundTouchSetRate;
SoundTouchSetTempo : TSoundTouchSetTempo;
SoundTouchSetRateChange : TSoundTouchSetRateChange;
SoundTouchSetTempoChange : TSoundTouchSetTempoChange;
SoundTouchSetPitch : TSoundTouchSetPitch;
SoundTouchSetPitchOctaves : TSoundTouchSetPitchOctaves;
SoundTouchSetPitchSemiTones : TSoundTouchSetPitchSemiTones;
SoundTouchSetChannels : TSoundTouchSetChannels;
SoundTouchSetSampleRate : TSoundTouchSetSampleRate;
SoundTouchFlush : TSoundTouchFlush;
SoundTouchPutSamples : TSoundTouchPutSamples;
SoundTouchClear : TSoundTouchClear;
SoundTouchSetSetting : TSoundTouchSetSetting;
SoundTouchGetSetting : TSoundTouchGetSetting;
SoundTouchNumUnprocessedSamples : TSoundTouchNumUnprocessedSamples;
SoundTouchReceiveSamples : TSoundTouchReceiveSamples;
SoundTouchNumSamples : TSoundTouchNumSamples;
SoundTouchIsEmpty : TSoundTouchIsEmpty;
type
TSoundTouch = class
private
FHandle : TSoundTouchHandle;
FRate : Single;
FPitch : Single;
FTempo : Single;
FSampleRate : Single;
FChannels : Cardinal;
function GetNumSamples: Cardinal;
function GetNumUnprocessedSamples: Cardinal;
function GetIsEmpty: Integer;
function GetPitchChange: Single;
function GetRateChange: Single;
function GetTempoChange: Single;
procedure SetRate(const Value: Single);
procedure SetPitch(const Value: Single);
procedure SetTempo(const Value: Single);
procedure SetPitchChange(const Value: Single);
procedure SetRateChange(const Value: Single);
procedure SetTempoChange(const Value: Single);
procedure SetChannels(const Value: Cardinal);
procedure SetSampleRate(const Value: Single);
protected
procedure SamplerateChanged; virtual;
procedure ChannelsChanged; virtual;
procedure PitchChanged; virtual;
procedure TempoChanged; virtual;
procedure RateChanged; virtual;
public
class function GetVersionString: string;
class function GetVersionId: Cardinal;
constructor Create; virtual;
destructor Destroy; override;
procedure Flush; virtual;
procedure Clear; virtual;
procedure PutSamples(const Samples: PSingle; const NumSamples: Cardinal);
function ReceiveSamples(const outBuffer: PSingle; const maxSamples: Integer): Cardinal;
function SetSetting(const SettingId: Integer; const Value: Integer): Boolean;
function GetSetting(const settingId: Integer): Integer;
property VersionString: string read GetVersionString;
property VersionID: Cardinal read GetVersionId;
property Channels: Cardinal read FChannels write SetChannels;
property Rate: Single read FRate write SetRate;
property RateChange: Single read GetRateChange write SetRateChange;
property Tempo: Single read FTempo write SetTempo;
property TempoChange: Single read GetTempoChange write SetTempoChange;
property Pitch: Single read FPitch write SetPitch;
property PitchChange: Single read GetPitchChange write SetPitchChange;
property SampleRate: Single read FSampleRate write SetSampleRate;
property NumSamples: Cardinal read GetNumSamples;
property NumUnprocessedSamples: Cardinal read GetNumUnprocessedSamples;
property IsEmpty: Integer read GetIsEmpty;
end;
implementation
uses
SysUtils;
{ TSoundTouch }
constructor TSoundTouch.Create;
begin
inherited;
FHandle := SoundTouchCreateInstance;
FRate := 1;
FTempo := 1;
FPitch := 1;
FChannels := 1;
FSampleRate := 44100;
SamplerateChanged;
ChannelsChanged;
end;
destructor TSoundTouch.Destroy;
begin
SoundTouchDestroyInstance(FHandle);
inherited;
end;
procedure TSoundTouch.Flush;
begin
SoundTouchFlush(FHandle);
end;
procedure TSoundTouch.Clear;
begin
SoundTouchClear(FHandle);
end;
function TSoundTouch.GetIsEmpty: Integer;
begin
result := SoundTouchIsEmpty(FHandle);
end;
function TSoundTouch.GetNumSamples: Cardinal;
begin
result := SoundTouchNumSamples(FHandle);
end;
function TSoundTouch.GetNumUnprocessedSamples: Cardinal;
begin
result := SoundTouchNumUnprocessedSamples(FHandle);
end;
function TSoundTouch.GetPitchChange: Single;
begin
result := 100 * (FPitch - 1.0);
end;
function TSoundTouch.GetRateChange: Single;
begin
result := 100 * (FRate - 1.0);
end;
function TSoundTouch.GetTempoChange: Single;
begin
result := 100 * (FTempo - 1.0);
end;
class function TSoundTouch.GetVersionId: Cardinal;
begin
result := SoundTouchGetVersionId;
end;
class function TSoundTouch.GetVersionString: string;
begin
result := StrPas(SoundTouchGetVersionString);
end;
procedure TSoundTouch.SetChannels(const Value: Cardinal);
begin
if FChannels <> Value then
begin
FChannels := Value;
ChannelsChanged;
end;
end;
procedure TSoundTouch.ChannelsChanged;
begin
assert(FChannels in [1, 2]);
SoundTouchSetChannels(FHandle, FChannels);
end;
procedure TSoundTouch.SetPitch(const Value: Single);
begin
if FPitch <> Value then
begin
FPitch := Value;
PitchChanged;
end;
end;
procedure TSoundTouch.PitchChanged;
begin
SoundTouchSetPitch(FHandle, FPitch);
end;
procedure TSoundTouch.putSamples(const Samples: PSingle;
const NumSamples: Cardinal);
begin
SoundTouchPutSamples(FHandle, Samples, NumSamples);
end;
procedure TSoundTouch.RateChanged;
begin
SoundTouchSetRate(FHandle, FRate);
end;
function TSoundTouch.ReceiveSamples(const outBuffer: PSingle;
const maxSamples: Integer): Cardinal;
begin
result := SoundTouchReceiveSamples(FHandle, outBuffer, maxSamples);
end;
procedure TSoundTouch.SetPitchChange(const Value: Single);
begin
Pitch := 1.0 + 0.01 * Value;
end;
procedure TSoundTouch.SetRate(const Value: Single);
begin
if FRate <> Value then
begin
FRate := Value;
RateChanged;
end;
end;
procedure TSoundTouch.SetRateChange(const Value: Single);
begin
Rate := 1.0 + 0.01 * Value;
end;
procedure TSoundTouch.SetSampleRate(const Value: Single);
begin
if FSampleRate <> Value then
begin
FSampleRate := Value;
SamplerateChanged;
end;
end;
procedure TSoundTouch.SamplerateChanged;
begin
assert(FSampleRate > 0);
SoundTouchsetSampleRate(FHandle, round(FSampleRate));
end;
procedure TSoundTouch.SetTempo(const Value: Single);
begin
if FTempo <> Value then
begin
FTempo := Value;
TempoChanged;
end;
end;
procedure TSoundTouch.SetTempoChange(const Value: Single);
begin
Tempo := 1.0 + 0.01 * Value;
end;
function TSoundTouch.GetSetting(const SettingId: Integer): Integer;
begin
result := SoundTouchGetSetting(FHandle, SettingId);
end;
function TSoundTouch.SetSetting(const SettingId: Integer;
const Value: Integer): Boolean;
begin
result := SoundTouchSetSetting(FHandle, SettingId, Value);
end;
procedure TSoundTouch.TempoChanged;
begin
SoundTouchsetTempo(FHandle, FTempo);
end;
var
SoundTouchLibHandle: HINST;
SoundTouchDLL: PAnsiChar = 'SoundTouch.DLL';
procedure InitDLL;
begin
SoundTouchLibHandle := LoadLibrary(SoundTouchDLL);
if SoundTouchLibHandle <> 0 then
try
SoundTouchCreateInstance := GetProcAddress(SoundTouchLibHandle, PAnsiChar( 2)); //'soundtouch_createInstance');
SoundTouchDestroyInstance := GetProcAddress(SoundTouchLibHandle, PAnsiChar( 3)); //'soundtouch_destroyInstance');
SoundTouchGetVersionString := GetProcAddress(SoundTouchLibHandle, PAnsiChar( 7)); //'soundtouch_getVersionString');
SoundTouchGetVersionId := GetProcAddress(SoundTouchLibHandle, PAnsiChar( 6)); //'soundtouch_getVersionId');
SoundTouchSetRate := GetProcAddress(SoundTouchLibHandle, PAnsiChar(17)); //'soundtouch_setRate');
SoundTouchSetTempo := GetProcAddress(SoundTouchLibHandle, PAnsiChar(21)); //'soundtouch_setTempo');
SoundTouchSetRateChange := GetProcAddress(SoundTouchLibHandle, PAnsiChar(18)); //'soundtouch_setRateChange');
SoundTouchSetTempoChange := GetProcAddress(SoundTouchLibHandle, PAnsiChar(22)); //'soundtouch_setTempoChange');
SoundTouchSetPitch := GetProcAddress(SoundTouchLibHandle, PAnsiChar(14)); //'soundtouch_setPitch');
SoundTouchSetPitchOctaves := GetProcAddress(SoundTouchLibHandle, PAnsiChar(15)); //'soundtouch_setPitchOctaves');
SoundTouchSetPitchSemiTones := GetProcAddress(SoundTouchLibHandle, PAnsiChar(16)); //'soundtouch_setPitchSemiTones');
SoundTouchSetChannels := GetProcAddress(SoundTouchLibHandle, PAnsiChar(13)); //'soundtouch_setChannels');
SoundTouchSetSampleRate := GetProcAddress(SoundTouchLibHandle, PAnsiChar(19)); //'soundtouch_setSampleRate');
SoundTouchFlush := GetProcAddress(SoundTouchLibHandle, PAnsiChar(4)); //'soundtouch_flush');
SoundTouchPutSamples := GetProcAddress(SoundTouchLibHandle, PAnsiChar(11)); //'soundtouch_putSamples');
SoundTouchClear := GetProcAddress(SoundTouchLibHandle, PAnsiChar(1)); //'soundtouch_clear');
SoundTouchSetSetting := GetProcAddress(SoundTouchLibHandle, PAnsiChar(20)); //'soundtouch_SetSetting');
SoundTouchGetSetting := GetProcAddress(SoundTouchLibHandle, PAnsiChar(5)); //'soundtouch_setSetting');
SoundTouchNumUnprocessedSamples := GetProcAddress(SoundTouchLibHandle, PAnsiChar(10)); //'soundtouch_numUnprocessedSamples');
SoundTouchReceiveSamples := GetProcAddress(SoundTouchLibHandle, PAnsiChar(12)); //'soundtouch_receiveSamples');
SoundTouchNumSamples := GetProcAddress(SoundTouchLibHandle, PAnsiChar(9)); //'soundtouch_numSamples');
SoundTouchIsEmpty := GetProcAddress(SoundTouchLibHandle, PAnsiChar(8)); //'soundtouch_isEmpty');
except
FreeLibrary(SoundTouchLibHandle);
SoundTouchLibHandle := 0;
end;
end;
procedure FreeDLL;
begin
if SoundTouchLibHandle <> 0 then FreeLibrary(SoundTouchLibHandle);
end;
initialization
InitDLL;
finalization
FreeDLL;
end.

114
lib-src/soundtouch/source/SoundTouchDLL/SoundTouchDLL.rc Normal file
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@ -0,0 +1,114 @@
// Microsoft Visual C++ generated resource script.
//
#include "resource.h"
#define APSTUDIO_READONLY_SYMBOLS
/////////////////////////////////////////////////////////////////////////////
//
// Generated from the TEXTINCLUDE 2 resource.
//
#include "afxres.h"
/////////////////////////////////////////////////////////////////////////////
#undef APSTUDIO_READONLY_SYMBOLS
/////////////////////////////////////////////////////////////////////////////
// English (U.S.) resources
#if !defined(AFX_RESOURCE_DLL) || defined(AFX_TARG_ENU)
#ifdef _WIN32
LANGUAGE LANG_ENGLISH, SUBLANG_ENGLISH_US
#pragma code_page(1252)
#endif //_WIN32
#ifdef APSTUDIO_INVOKED
/////////////////////////////////////////////////////////////////////////////
//
// TEXTINCLUDE
//
1 TEXTINCLUDE
BEGIN
"resource.h\0"
END
2 TEXTINCLUDE
BEGIN
"#include ""afxres.h""\r\n"
"\0"
END
3 TEXTINCLUDE
BEGIN
"\r\n"
"\0"
END
#endif // APSTUDIO_INVOKED
#endif // English (U.S.) resources
/////////////////////////////////////////////////////////////////////////////
/////////////////////////////////////////////////////////////////////////////
// Finnish resources
#if !defined(AFX_RESOURCE_DLL) || defined(AFX_TARG_FIN)
#ifdef _WIN32
LANGUAGE LANG_FINNISH, SUBLANG_DEFAULT
#pragma code_page(1252)
#endif //_WIN32
/////////////////////////////////////////////////////////////////////////////
//
// Version
//
VS_VERSION_INFO VERSIONINFO
FILEVERSION 1,7,0,0
PRODUCTVERSION 1,7,0,0
FILEFLAGSMASK 0x17L
#ifdef _DEBUG
FILEFLAGS 0x1L
#else
FILEFLAGS 0x0L
#endif
FILEOS 0x4L
FILETYPE 0x2L
FILESUBTYPE 0x0L
BEGIN
BLOCK "StringFileInfo"
BEGIN
BLOCK "000004b0"
BEGIN
VALUE "Comments", "SoundTouch Library licensed for 3rd party applications subject to LGPL license v2.1. Visit http://www.surina.net/soundtouch for more information about the SoundTouch library."
VALUE "FileDescription", "SoundTouch Dynamic Link Library"
VALUE "FileVersion", "1, 7, 0, 0"
VALUE "InternalName", "SoundTouch"
VALUE "LegalCopyright", "Copyright (C) Olli Parviainen 1999-2012"
VALUE "OriginalFilename", "SoundTouch.dll"
VALUE "ProductName", " SoundTouch Dynamic Link Library"
VALUE "ProductVersion", "1, 7, 0, 0"
END
END
BLOCK "VarFileInfo"
BEGIN
VALUE "Translation", 0x0, 1200
END
END
#endif // Finnish resources
/////////////////////////////////////////////////////////////////////////////
#ifndef APSTUDIO_INVOKED
/////////////////////////////////////////////////////////////////////////////
//
// Generated from the TEXTINCLUDE 3 resource.
//
/////////////////////////////////////////////////////////////////////////////
#endif // not APSTUDIO_INVOKED

28
lib-src/soundtouch/source/SoundTouchDLL/SoundTouchDLL.sln Normal file
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@ -0,0 +1,28 @@
Microsoft Visual Studio Solution File, Format Version 10.00
# Visual Studio 2008
Project("{8BC9CEB8-8B4A-11D0-8D11-00A0C91BC942}") = "SoundTouchDLL", "SoundTouchDLL.vcproj", "{164DE61D-6391-4265-8273-30740117D356}"
ProjectSection(ProjectDependencies) = postProject
{68A5DD20-7057-448B-8FE0-B6AC8D205509} = {68A5DD20-7057-448B-8FE0-B6AC8D205509}
EndProjectSection
EndProject
Project("{8BC9CEB8-8B4A-11D0-8D11-00A0C91BC942}") = "SoundTouch", "..\SoundTouch\SoundTouch.vcproj", "{68A5DD20-7057-448B-8FE0-B6AC8D205509}"
EndProject
Global
GlobalSection(SolutionConfigurationPlatforms) = preSolution
Debug|Win32 = Debug|Win32
Release|Win32 = Release|Win32
EndGlobalSection
GlobalSection(ProjectConfigurationPlatforms) = postSolution
{164DE61D-6391-4265-8273-30740117D356}.Debug|Win32.ActiveCfg = Debug|Win32
{164DE61D-6391-4265-8273-30740117D356}.Debug|Win32.Build.0 = Debug|Win32
{164DE61D-6391-4265-8273-30740117D356}.Release|Win32.ActiveCfg = Release|Win32
{164DE61D-6391-4265-8273-30740117D356}.Release|Win32.Build.0 = Release|Win32
{68A5DD20-7057-448B-8FE0-B6AC8D205509}.Debug|Win32.ActiveCfg = Debug|Win32
{68A5DD20-7057-448B-8FE0-B6AC8D205509}.Debug|Win32.Build.0 = Debug|Win32
{68A5DD20-7057-448B-8FE0-B6AC8D205509}.Release|Win32.ActiveCfg = Release|Win32
{68A5DD20-7057-448B-8FE0-B6AC8D205509}.Release|Win32.Build.0 = Release|Win32
EndGlobalSection
GlobalSection(SolutionProperties) = preSolution
HideSolutionNode = FALSE
EndGlobalSection
EndGlobal

229
lib-src/soundtouch/source/SoundTouchDLL/SoundTouchDLL.vcproj Normal file
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@ -0,0 +1,229 @@
<?xml version="1.0" encoding="Windows-1252"?>
<VisualStudioProject
ProjectType="Visual C++"
Version="9,00"
Name="SoundTouchDLL"
ProjectGUID="{164DE61D-6391-4265-8273-30740117D356}"
Keyword="Win32Proj"
TargetFrameworkVersion="131072"
>
<Platforms>
<Platform
Name="Win32"
/>
</Platforms>
<ToolFiles>
</ToolFiles>
<Configurations>
<Configuration
Name="Debug|Win32"
OutputDirectory="Debug"
IntermediateDirectory="Debug"
ConfigurationType="2"
InheritedPropertySheets="$(VCInstallDir)VCProjectDefaults\UpgradeFromVC71.vsprops"
CharacterSet="2"
>
<Tool
Name="VCPreBuildEventTool"
/>
<Tool
Name="VCCustomBuildTool"
/>
<Tool
Name="VCXMLDataGeneratorTool"
/>
<Tool
Name="VCWebServiceProxyGeneratorTool"
/>
<Tool
Name="VCMIDLTool"
/>
<Tool
Name="VCCLCompilerTool"
Optimization="0"
AdditionalIncludeDirectories="..\..\include"
PreprocessorDefinitions="WIN32;_DEBUG;_WINDOWS;_USRDLL;_CRT_SECURE_NO_WARNINGS;DLL_EXPORTS"
MinimalRebuild="true"
BasicRuntimeChecks="3"
RuntimeLibrary="1"
UsePrecompiledHeader="0"
WarningLevel="3"
Detect64BitPortabilityProblems="true"
DebugInformationFormat="3"
/>
<Tool
Name="VCManagedResourceCompilerTool"
/>
<Tool
Name="VCResourceCompilerTool"
/>
<Tool
Name="VCPreLinkEventTool"
/>
<Tool
Name="VCLinkerTool"
OutputFile="$(OutDir)/SoundTouchD.dll"
LinkIncremental="2"
IgnoreDefaultLibraryNames="libcd"
GenerateDebugInformation="true"
ProgramDatabaseFile="$(OutDir)/SoundTouchD.pdb"
SubSystem="2"
RandomizedBaseAddress="1"
DataExecutionPrevention="0"
ImportLibrary="$(OutDir)/SoundTouchDllD.lib"
TargetMachine="1"
/>
<Tool
Name="VCALinkTool"
/>
<Tool
Name="VCManifestTool"
/>
<Tool
Name="VCXDCMakeTool"
/>
<Tool
Name="VCBscMakeTool"
/>
<Tool
Name="VCFxCopTool"
/>
<Tool
Name="VCAppVerifierTool"
/>
<Tool
Name="VCPostBuildEventTool"
CommandLine="copy $(OutDir)\*.dll ..\..\lib&#x0D;&#x0A;copy $(OutDir)\*.lib ..\..\lib&#x0D;&#x0A;"
/>
</Configuration>
<Configuration
Name="Release|Win32"
OutputDirectory="Release"
IntermediateDirectory="Release"
ConfigurationType="2"
InheritedPropertySheets="$(VCInstallDir)VCProjectDefaults\UpgradeFromVC71.vsprops"
CharacterSet="2"
>
<Tool
Name="VCPreBuildEventTool"
/>
<Tool
Name="VCCustomBuildTool"
/>
<Tool
Name="VCXMLDataGeneratorTool"
/>
<Tool
Name="VCWebServiceProxyGeneratorTool"
/>
<Tool
Name="VCMIDLTool"
/>
<Tool
Name="VCCLCompilerTool"
Optimization="3"
InlineFunctionExpansion="2"
EnableIntrinsicFunctions="true"
OmitFramePointers="false"
AdditionalIncludeDirectories="..\..\include"
PreprocessorDefinitions="WIN32;NDEBUG;_WINDOWS;_USRDLL;_CRT_SECURE_NO_WARNINGS;DLL_EXPORTS"
RuntimeLibrary="0"
UsePrecompiledHeader="0"
WarningLevel="3"
Detect64BitPortabilityProblems="true"
DebugInformationFormat="0"
CallingConvention="0"
/>
<Tool
Name="VCManagedResourceCompilerTool"
/>
<Tool
Name="VCResourceCompilerTool"
/>
<Tool
Name="VCPreLinkEventTool"
/>
<Tool
Name="VCLinkerTool"
OutputFile="$(OutDir)/SoundTouch.dll"
LinkIncremental="1"
IgnoreDefaultLibraryNames="libc"
GenerateDebugInformation="false"
GenerateMapFile="true"
SubSystem="2"
OptimizeReferences="2"
EnableCOMDATFolding="2"
RandomizedBaseAddress="1"
DataExecutionPrevention="0"
ImportLibrary="$(OutDir)/SoundTouchDll.lib"
TargetMachine="1"
/>
<Tool
Name="VCALinkTool"
/>
<Tool
Name="VCManifestTool"
/>
<Tool
Name="VCXDCMakeTool"
/>
<Tool
Name="VCBscMakeTool"
/>
<Tool
Name="VCFxCopTool"
/>
<Tool
Name="VCAppVerifierTool"
/>
<Tool
Name="VCPostBuildEventTool"
CommandLine="copy $(OutDir)\*.dll ..\..\lib&#x0D;&#x0A;copy $(OutDir)\*.lib ..\..\lib&#x0D;&#x0A;"
/>
</Configuration>
</Configurations>
<References>
</References>
<Files>
<Filter
Name="Source Files"
Filter="cpp;c;cxx;def;odl;idl;hpj;bat;asm;asmx"
UniqueIdentifier="{4FC737F1-C7A5-4376-A066-2A32D752A2FF}"
>
<File
RelativePath=".\SoundTouchDLL.cpp"
>
</File>
</Filter>
<Filter
Name="Header Files"
Filter="h;hpp;hxx;hm;inl;inc;xsd"
UniqueIdentifier="{93995380-89BD-4b04-88EB-625FBE52EBFB}"
>
<File
RelativePath=".\resource.h"
>
</File>
<File
RelativePath=".\SoundTouchDLL.h"
>
</File>
</Filter>
<Filter
Name="Resource Files"
Filter="rc;ico;cur;bmp;dlg;rc2;rct;bin;rgs;gif;jpg;jpeg;jpe;resx"
UniqueIdentifier="{67DA6AB6-F800-4c08-8B7A-83BB121AAD01}"
>
<File
RelativePath=".\SoundTouchDLL.rc"
>
</File>
</Filter>
<File
RelativePath=".\ReadMe.txt"
>
</File>
</Files>
<Globals>
</Globals>
</VisualStudioProject>

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

28
lib-src/soundtouch/source/example/Makefile.am
View File

@ -1,28 +0,0 @@
## Process this file with automake to create Makefile.in
##
## $Id: Makefile.am,v 1.3 2006-09-27 06:19:53 llucius Exp $
##
## 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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