cmake minimum version to 3.5 to avoid deprecation warning

Refactor pow() usage:
- use base of 0.5^k instead of 1/(2^k)
- skip pow if using INTEGER_SAMPLES

.gitignore

@@ -0,0 +1,55 @@
+bin
+lib
+
+# Win build files
+*.dll
+*.exe
+*.lib
+
+# GNU build files
+*.o
+*.so*
+*.lo
+*.P*
+*.la*
+*.a
+*.pc
+*config*
+Makefile
+Makefile.in
+.libs
+aclocal.m4
+autom4te.cache
+stamp-*
+libtool
+soundstretch
+
+# Files generated by MSVC
+*.bsc
+*.suo
+*.sdf
+*.filters
+*.user
+source/SoundTouch/Win32/
+source/SoundTouch/x64/
+source/SoundStretch/Win32/
+source/SoundStretch/x64/
+source/SoundTouchDll/Win32/
+source/SoundTouchDll/x64/
+source/SoundTouchDll/DllTest/Win32/
+source/SoundTouchDll/DllTest/x64/
+.vs
+
+# Files generated by Android Studio
+source/android-lib/.gradle
+source/android-lib/.idea
+**/*.iml
+source/android-lib/local.properties
+source/android-lib/build
+source/android-lib/.externalNativeBuild
+
+# CMake build directory
+build*
+CMakeFiles
+CMakeCache.txt
+*.cmake

.vscode/tasks.json

@@ -0,0 +1,32 @@
+{
+    // This is build task definition file for MS VisualStudio Code.
+    // See https://go.microsoft.com/fwlink/?LinkId=733558
+    // for the documentation about the tasks.json format
+    "version": "2.0.0",
+    "tasks": [
+        {
+            "label": "echo",
+            "type": "shell",
+            "command": "echo Hello"
+        },
+        {
+            "label": "configure",
+            "type": "shell",
+            "command": "./bootstrap && ./configure"
+        },
+        {
+            "label": "build",
+            "type": "shell",
+            "command": "make -j4",
+            "problemMatcher": [
+                "$gcc"
+            ]
+        },
+        {
+            "label": "clean",
+            "type": "shell",
+            "command": "make clean",
+            "problemMatcher": []
+        }
+    ]
+}

CMakeLists.txt

@@ -0,0 +1,191 @@
+cmake_minimum_required(VERSION 3.5)
+project(SoundTouch VERSION 2.3.3 LANGUAGES CXX)
+set(CMAKE_CXX_STANDARD 17)
+
+include(GNUInstallDirs)
+
+set(COMPILE_OPTIONS)
+
+if(MSVC)
+  set(COMPILE_DEFINITIONS /O2 /fp:fast)
+else()
+  list(APPEND COMPILE_OPTIONS -Wall -Wextra -Wzero-as-null-pointer-constant -Wno-unknown-pragmas)
+  if(EMSCRIPTEN)
+    list(APPEND COMPILE_OPTIONS -O3)
+  else()
+    # Apply -ffast-math to allow compiler autovectorization generate effective SIMD code for arm compilation
+    list(APPEND COMPILE_OPTIONS -O3 -ffast-math)
+  endif()
+endif()
+
+#####################
+# SoundTouch library
+
+add_library(SoundTouch
+  source/SoundTouch/AAFilter.cpp
+  source/SoundTouch/BPMDetect.cpp
+  source/SoundTouch/cpu_detect_x86.cpp
+  source/SoundTouch/FIFOSampleBuffer.cpp
+  source/SoundTouch/FIRFilter.cpp
+  source/SoundTouch/InterpolateCubic.cpp
+  source/SoundTouch/InterpolateLinear.cpp
+  source/SoundTouch/InterpolateShannon.cpp
+  source/SoundTouch/mmx_optimized.cpp
+  source/SoundTouch/PeakFinder.cpp
+  source/SoundTouch/RateTransposer.cpp
+  source/SoundTouch/SoundTouch.cpp
+  source/SoundTouch/sse_optimized.cpp
+  source/SoundTouch/TDStretch.cpp
+)
+target_include_directories(SoundTouch PUBLIC
+   $<BUILD_INTERFACE:${CMAKE_CURRENT_SOURCE_DIR}/include>
+   $<INSTALL_INTERFACE:${CMAKE_INSTALL_INCLUDEDIR}>
+)
+
+target_compile_definitions(SoundTouch PRIVATE ${COMPILE_DEFINITIONS})
+target_compile_options(SoundTouch PRIVATE ${COMPILE_OPTIONS})
+
+if(BUILD_SHARED_LIBS)
+  set_target_properties(SoundTouch PROPERTIES
+    VERSION ${CMAKE_PROJECT_VERSION}
+  )
+  if(WIN32)
+    set_target_properties(SoundTouch PROPERTIES
+      WINDOWS_EXPORT_ALL_SYMBOLS TRUE
+    )
+  else()
+    set_target_properties(SoundTouch PROPERTIES
+      SOVERSION ${PROJECT_VERSION_MAJOR}
+    )
+  endif()
+endif()
+
+option(INTEGER_SAMPLES "Use integers instead of floats for samples" OFF)
+if(INTEGER_SAMPLES)
+  target_compile_definitions(SoundTouch PRIVATE SOUNDTOUCH_INTEGER_SAMPLES)
+else()
+  target_compile_definitions(SoundTouch PRIVATE SOUNDTOUCH_FLOAT_SAMPLES)
+endif()
+
+if(CMAKE_SYSTEM_PROCESSOR MATCHES "^(armv7.*|armv8.*|aarch64.*)$")
+  set(NEON_CPU ON)
+else()
+  set(NEON_CPU OFF)
+endif()
+
+option(NEON "Use ARM Neon SIMD instructions if in ARM CPU" ON)
+if(${NEON} AND ${NEON_CPU})
+  target_compile_definitions(SoundTouch PRIVATE SOUNDTOUCH_USE_NEON)
+  if(NOT CMAKE_SYSTEM_PROCESSOR MATCHES "^aarch64.*$")
+    target_compile_options(SoundTouch PRIVATE -mfpu=neon)
+  endif()
+endif()
+
+find_package(OpenMP)
+option(OPENMP "Use parallel multicore calculation through OpenMP" OFF)
+if(OPENMP AND OPENMP_FOUND)
+  set(CMAKE_CXX_FLAGS "${CMAKE_CXX_FLAGS} ${OpenMP_CXX_FLAGS}")
+endif()
+
+install(
+  FILES
+    include/BPMDetect.h
+    include/FIFOSampleBuffer.h
+    include/FIFOSamplePipe.h
+    include/STTypes.h
+    include/SoundTouch.h
+    include/soundtouch_config.h
+  DESTINATION
+    "${CMAKE_INSTALL_INCLUDEDIR}/soundtouch"
+  COMPONENT SoundTouch
+)
+
+install(TARGETS SoundTouch
+  EXPORT SoundTouchTargets
+  ARCHIVE DESTINATION "${CMAKE_INSTALL_LIBDIR}"
+  LIBRARY DESTINATION "${CMAKE_INSTALL_LIBDIR}"
+  RUNTIME DESTINATION "${CMAKE_INSTALL_BINDIR}"
+  INCLUDES DESTINATION "${CMAKE_INSTALL_INCLUDEDIR}"
+  COMPONENT SoundTouch
+)
+
+#######################
+# soundstretch utility
+
+option(SOUNDSTRETCH "Build soundstretch command line utility." ON)
+if(SOUNDSTRETCH)
+  add_executable(soundstretch
+    source/SoundStretch/main.cpp
+    source/SoundStretch/RunParameters.cpp
+    source/SoundStretch/WavFile.cpp
+  )
+  target_include_directories(soundstretch PUBLIC $<BUILD_INTERFACE:${CMAKE_CURRENT_SOURCE_DIR}/include>)
+  target_compile_definitions(soundstretch PRIVATE ${COMPILE_DEFINITIONS})
+  target_compile_options(soundstretch PRIVATE ${COMPILE_OPTIONS})
+  target_link_libraries(soundstretch PRIVATE SoundTouch)
+  if(INTEGER_SAMPLES)
+    target_compile_definitions(soundstretch PRIVATE SOUNDTOUCH_INTEGER_SAMPLES)
+  endif()
+
+  install(TARGETS soundstretch
+    DESTINATION bin
+    COMPONENT soundstretch
+  )
+endif()
+
+########################
+# SoundTouchDll library
+
+option(SOUNDTOUCH_DLL "Build SoundTouchDLL C wrapper library" OFF)
+if(SOUNDTOUCH_DLL)
+  add_library(SoundTouchDLL SHARED
+    source/SoundTouchDLL/SoundTouchDLL.cpp
+    source/SoundTouchDLL/SoundTouchDLL.rc
+  )
+  set_target_properties(SoundTouch PROPERTIES POSITION_INDEPENDENT_CODE TRUE)
+  target_compile_options(SoundTouchDLL PRIVATE ${COMPILE_OPTIONS})
+  set_target_properties(SoundTouchDLL PROPERTIES CXX_VISIBILITY_PRESET hidden)
+  target_compile_definitions(SoundTouchDLL PRIVATE DLL_EXPORTS)
+  target_include_directories(SoundTouchDLL PUBLIC $<BUILD_INTERFACE:${CMAKE_CURRENT_SOURCE_DIR}/include>)
+  target_link_libraries(SoundTouchDLL PRIVATE SoundTouch)
+  install(FILES source/SoundTouchDLL/SoundTouchDLL.h DESTINATION "${CMAKE_INSTALL_INCLUDEDIR}/soundtouch" COMPONENT SoundTouchDLL)
+  install(TARGETS SoundTouchDLL EXPORT SoundTouchTargets COMPONENT SoundTouchDLL)
+endif()
+
+########################
+
+# pkgconfig
+set(prefix "${CMAKE_INSTALL_PREFIX}")
+set(execprefix "\${prefix}")
+set(libdir "\${prefix}/${CMAKE_INSTALL_LIBDIR}")
+set(includedir "\${prefix}/${CMAKE_INSTALL_INCLUDEDIR}")
+set(VERSION "${CMAKE_PROJECT_VERSION}")
+configure_file(soundtouch.pc.in "${CMAKE_CURRENT_BINARY_DIR}/soundtouch.pc" @ONLY)
+install(FILES "${CMAKE_CURRENT_BINARY_DIR}/soundtouch.pc" DESTINATION "${CMAKE_INSTALL_LIBDIR}/pkgconfig" COMPONENT SoundTouch)
+
+# CMake config
+include(CMakePackageConfigHelpers)
+set(SOUNDTOUCH_INSTALL_CMAKEDIR "${CMAKE_INSTALL_LIBDIR}/cmake/SoundTouch")
+install(
+  EXPORT SoundTouchTargets
+  FILE SoundTouchTargets.cmake
+  NAMESPACE SoundTouch::
+  DESTINATION "${SOUNDTOUCH_INSTALL_CMAKEDIR}"
+  COMPONENT SoundTouch
+)
+configure_package_config_file(SoundTouchConfig.cmake.in
+  "${CMAKE_CURRENT_BINARY_DIR}/SoundTouchConfig.cmake"
+  INSTALL_DESTINATION "${SOUNDTOUCH_INSTALL_CMAKEDIR}"
+)
+write_basic_package_version_file(
+  "${CMAKE_CURRENT_BINARY_DIR}/SoundTouchConfigVersion.cmake"
+  VERSION "${CMAKE_PROJECT_VERSION}"
+  COMPATIBILITY SameMajorVersion
+)
+install(
+  FILES
+    "${CMAKE_CURRENT_BINARY_DIR}/SoundTouchConfig.cmake"
+    "${CMAKE_CURRENT_BINARY_DIR}/SoundTouchConfigVersion.cmake"
+  DESTINATION "${SOUNDTOUCH_INSTALL_CMAKEDIR}"
+  COMPONENT SoundTouch
+)

COPYING.TXT

@@ -2,7 +2,7 @@
 		       Version 2.1, February 1999
 
  Copyright (C) 1991, 1999 Free Software Foundation, Inc.
-     59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+ 51 Franklin Street, Fifth Floor, Boston, MA  02110-1301  USA
  Everyone is permitted to copy and distribute verbatim copies
  of this license document, but changing it is not allowed.
 
@@ -117,7 +117,7 @@ be combined with the library in order to run.
 
   0. This License Agreement applies to any software library or other
 program which contains a notice placed by the copyright holder or
-other authoried party saying it may be distributed under the terms of
+other authorized party saying it may be distributed under the terms of
 this Lesser General Public License (also called "this License").
 Each licensee is addressed as "you".
 

Makefile.am

@@ -1,9 +1,5 @@
 ## Process this file with automake to create Makefile.in
 ##
-## $Id$
-##
-## 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
@@ -26,9 +22,8 @@ include $(top_srcdir)/config/am_include.mk
 ## subdirectories Start at frontend_fox/Makefile.am to see how everything works.
 SUBDIRS=include source
 
-
 # list files that are documentation to be packaged in a release tarball and installed
-pkgdoc_DATA=COPYING.TXT README.html
+dist_doc_DATA=COPYING.TXT README.html
 
 # extra data files that are to be pacakged in a release tarball and installed into the data directory
 #pkgdata_DATA=

SoundTouchConfig.cmake.in

@@ -0,0 +1,14 @@
+@PACKAGE_INIT@
+
+include("${CMAKE_CURRENT_LIST_DIR}/SoundTouchTargets.cmake")
+
+check_required_components(SoundTouch)
+
+get_target_property(SoundTouch_LOCATION SoundTouch::SoundTouch LOCATION)
+message(STATUS "Found SoundTouch: ${SoundTouch_LOCATION}")
+
+if(@SOUNDTOUCH_DLL@)
+  check_required_components(SoundTouchDLL)
+  get_target_property(SoundTouchDLL_LOCATION SoundTouch::SoundTouchDLL LOCATION)
+  message(STATUS "Found SoundTouchDLL: ${SoundTouchDLL_LOCATION}")
+endif()

bin/run_test.cmd

@@ -1,8 +1,8 @@
-set SOUND_DIR=d:\dev\test_sounds
+set SOUND_DIR=c:\dev\test_sounds
 set OUT_DIR=.
 set TEST_NAME=semmari
 set OUT_NAME=out
-set SS=soundstretch
+set SS=soundstretch_x64
 set TEST_PARAM=-pitch=-3 -bpm
 
 call %SS% %SOUND_DIR%\%TEST_NAME%-8b1.wav %OUT_DIR%\%OUT_NAME%-8b1.wav %TEST_PARAM%

bootstrap

@@ -1,7 +1,7 @@
-# $Id$
-
 #!/bin/sh
 
+unset ACLOCAL
+
 if [ "$1" = "--clean" ]
 then
 	if [ -a Makefile ]
@@ -16,9 +16,6 @@ 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
 
-	#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 --copy"
 	autoreconf -fisv && rm -f `find . -name "*~"` && rm -f ChangeLog

config/am_include.mk

@@ -1,8 +1,6 @@
 ## vim:tw=78
 ## Process this file with automake to create Makefile.in
 ##
-## $Id$
-##
 ## 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
@@ -19,16 +17,10 @@
 ## Place - Suite 330, Boston, MA  02111-1307, USA
 
 ## These are common definitions used in all Makefiles
-## It is actually included when a makefile.am is coverted to Makefile.in
+## It is actually included when a makefile.am is converted to Makefile.in
 ## by automake, so it's ok to have @MACROS@ that will be set by configure
 
-
+AM_CPPFLAGS=-I$(top_srcdir)/include
-## INCLUDES is automatically added to CXXFLAGS at compile time. The
-## $(top_srcdir) macro is set by configure. It's important to use $(top_srcdir)
-## in case a user decides to build in a separate directory from the base package
-## directory. Using absolute, or relative paths is a bad idea.
-INCLUDES=-I$(top_srcdir)/include
-
 
 # doc directory
 pkgdocdir=$(prefix)/doc/@PACKAGE@

configure-file-README.txt

@@ -1,3 +0,0 @@
-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.

configure.ac

@@ -1,7 +1,3 @@
-dnl SoundTouch configure.ac, by David W. Durham
-dnl
-dnl $Id$
-dnl
 dnl This file is part of SoundTouch, an audio processing library for pitch/time adjustments
 dnl
 dnl SoundTouch is free software; you can redistribute it and/or modify it under the
@@ -19,18 +15,27 @@ 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, 1.7.0, [http://www.surina.net/soundtouch])
+AC_INIT([SoundTouch],[2.3.2],[http://www.surina.net/soundtouch])
+dnl Default to libSoundTouch.so.$LIB_SONAME.0.0
+LIB_SONAME=1
+AC_SUBST(LIB_SONAME)
+
 AC_CONFIG_AUX_DIR(config)
-AM_CONFIG_HEADER([include/soundtouch_config.h])
+AC_CONFIG_MACRO_DIR([config/m4])
+AC_CONFIG_HEADERS([config.h include/soundtouch_config.h])
 AM_INIT_AUTOMAKE
+AM_SILENT_RULES([yes])
 #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_USE_SYSTEM_EXTENSIONS 		dnl enable posix extensions in glibc
 
 AC_LANG(C++)
 
+# Compiler flags. Apply -ffast-math to allow compiler autovectorization generate effective SIMD code for arm compilation
+CXXFLAGS="${CXXFLAGS} -O3 -ffast-math -Wall -Wextra -Wzero-as-null-pointer-constant -Wno-unknown-pragmas"
 
-
+# Set AR_FLAGS to avoid build warning "ar: `u' modifier ignored since `D' is the default (see `U')"
+AR_FLAGS='cr'
 
 
 dnl ############################################################################
@@ -45,7 +50,7 @@ AC_PROG_INSTALL
 #AC_PROG_LN_S
 AC_PROG_MAKE_SET
 
-AM_PROG_LIBTOOL dnl turn on using libtool
+LT_INIT dnl turn on using libtool
 
 
 
@@ -53,10 +58,11 @@ AM_PROG_LIBTOOL dnl turn on using libtool
 dnl ############################################################################
 dnl # Checks for header files                                                  #
 dnl ############################################################################
-AC_HEADER_STDC
+
 #AC_HEADER_SYS_WAIT
 #	add any others you want to check for here
 AC_CHECK_HEADERS([cpuid.h])
+AC_CHECK_HEADERS([arm_neon.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.])
@@ -75,26 +81,34 @@ AC_C_INLINE
 
 
 AC_ARG_ENABLE(integer-samples,
-              [AC_HELP_STRING([--enable-integer-samples],
+              [AS_HELP_STRING([--enable-integer-samples],[use integer samples instead of floats [default=no]])],,
-                              [use integer samples instead of floats
-[default=yes]])],,
               [enable_integer_samples=no])
 
 
+AC_ARG_ENABLE(openmp,
+              [AS_HELP_STRING([--enable-openmp],[use parallel multicore calculation through OpenMP [default=no]])],,
+              [enable_openmp=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
+                              [use MMX or SSE optimization [default=yes]])],[enable_x86_optimizations="${enableval}"],
-[default=yes]])],[enable_x86_optimizations="${enableval}"],
               [enable_x86_optimizations=yes])
 
+# Let the user enable/disable the x86 optimizations.
+# Useful when compiling on non-x86 architectures.
+AC_ARG_ENABLE([neon-optimizations],
+              [AS_HELP_STRING([--enable-neon-optimizations],
+                              [use ARM NEON optimization [default=yes]])],[enable_neon_optimizations="${enableval}"],
+              [enable_neon_optimizations=yes])
+
+
 # Tell the Makefile.am if the user wants to disable optimizations.
 # Makefile.am will enable them by default if support is available.
 # Note: We check if optimizations are supported a few lines down.
 AM_CONDITIONAL([X86_OPTIMIZATIONS], [test "x$enable_x86_optimizations" = "xyes"])
 
-
 if test "x$enable_integer_samples" = "xyes"; then
         echo "****** Integer sample type enabled ******"
         AC_DEFINE(SOUNDTOUCH_INTEGER_SAMPLES,1,[Use Integer as Sample type])
@@ -102,6 +116,14 @@ else
         echo "****** Float sample type enabled ******"
         AC_DEFINE(SOUNDTOUCH_FLOAT_SAMPLES,1,[Use Float as Sample type])
 fi
+AM_CONDITIONAL([SOUNDTOUCH_FLOAT_SAMPLES], [test "x$enable_integer_samples" != "xyes"])
+
+if test "x$enable_openmp" = "xyes"; then
+        echo "****** openmp optimizations enabled ******"
+        AM_CXXFLAGS="-fopenmp $AM_CXXFLAGS"
+else
+        echo "****** openmp optimizations disabled ******"
+fi
 
 
 # Check if optimizations are supported in the system at build time.
@@ -110,8 +132,7 @@ if test "x$enable_x86_optimizations" = "xyes" -a "x$ac_cv_header_cpuid_h" = "xye
 
 	original_saved_CXXFLAGS=$CXXFLAGS
 	have_mmx_intrinsics=no
-	OPT_CXXFLAGS="-mmmx -Winline"
+	CXXFLAGS="-mmmx -Winline $CXXFLAGS"
-	CXXFLAGS="$OPT_CXXFLAGS $CXXFLAGS"
 
 	# Check if the user can compile MMX code using intrinsics.
 	# GCC supports MMX intrinsics since version 3.3
@@ -137,14 +158,15 @@ if test "x$enable_x86_optimizations" = "xyes" -a "x$ac_cv_header_cpuid_h" = "xye
 		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])
+		        CPPFLAGS="-DSOUNDTOUCH_DISABLE_X86_OPTIMIZATIONS $CPPFLAGS"
 		fi
 	fi
 
 
+	# SSE support
+	original_saved_CXXFLAGS=$CXXFLAGS
 	have_sse_intrinsics=no
-	OPT_CXXFLAGS="-msse -Winline"
+	CXXFLAGS="-msse -Winline $CXXFLAGS"
-	CXXFLAGS="$OPT_CXXFLAGS $CXXFLAGS"
 
 	# Check if the user can compile SSE code using intrinsics.
 	# GCC supports SSE intrinsics since version 3.3
@@ -170,16 +192,69 @@ if test "x$enable_x86_optimizations" = "xyes" -a "x$ac_cv_header_cpuid_h" = "xye
 		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])
+		        CPPFLAGS="-DSOUNDTOUCH_DISABLE_X86_OPTIMIZATIONS $CPPFLAGS"
 		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])
+        CPPFLAGS="-DSOUNDTOUCH_DISABLE_X86_OPTIMIZATIONS $CPPFLAGS"
 fi
 
+
+if test "x$enable_neon_optimizations" = "xyes" -a "x$ac_cv_header_arm_neon_h" = "xyes"; then
+
+	# Check for ARM NEON support
+	original_saved_CXXFLAGS=$CXXFLAGS
+	have_neon=no
+	CXXFLAGS="-mfpu=neon -march=native $CXXFLAGS"
+
+	# Check if can compile neon code using intrinsics, require GCC >= 4.3 for autovectorization.
+	AC_COMPILE_IFELSE([AC_LANG_SOURCE([[
+	#if defined(__GNUC__) && (__GNUC__ < 4 || (__GNUC__ == 4 && __GNUC_MINOR__ < 3))
+	#error "Need GCC >= 4.3 for neon autovectorization"
+	#endif
+	#include <arm_neon.h>
+	int main () {
+		int32x4_t t = {1};
+		return vaddq_s32(t,t)[0] == 2;
+	}]])],[have_neon=yes])
+	CXXFLAGS=$original_saved_CXXFLAGS
+	if test "x$have_neon" = "xyes" ; then
+		echo "****** NEON support enabled ******"
+		CPPFLAGS="-mfpu=neon -march=native -mtune=native $CPPFLAGS"
+		AC_DEFINE(SOUNDTOUCH_USE_NEON,1,[Use ARM NEON extension])
+	fi
+fi
+
+
+AC_CANONICAL_HOST
+HOST_OS=""
+AS_CASE([$host_cpu],
+        [x86_64],
+        [
+            x86_64=true
+            x86=true
+        ],
+        [i?86],
+        [
+            x86=true
+        ])
+
+AM_CONDITIONAL([X86], [test "$x86" = true])
+AM_CONDITIONAL([X86_64], [test "$x86_64" = true])
+
+AC_SUBST([HOST_OS])
+
+
+# Set AM_CXXFLAGS
+AC_SUBST([AM_CXXFLAGS], [$AM_CXXFLAGS])
+
+# Empty default CXXFLAGS so user can set them if desirable
+#AC_SUBST([CXXFLAGS], [ ])
+
+
 # SSTypes.h by default enables optimizations. Those already got disabled if
 # the user requested for it or if the system does not support them.
 #
@@ -195,8 +270,6 @@ AM_CONDITIONAL([HAVE_SSE], [test "x$have_sse_intrinsics" = "xyes"])
 dnl ############################################################################
 dnl # Checks for library functions/classes                                     #
 dnl ############################################################################
-AC_FUNC_MALLOC
-AC_TYPE_SIGNAL
 
 dnl make -lm get added to the LIBS
 AC_CHECK_LIB(m, sqrt,,AC_MSG_ERROR([compatible libc math library not found]))
@@ -229,11 +302,12 @@ AC_CONFIG_FILES([
 	source/Makefile
 		source/SoundTouch/Makefile
 		source/SoundStretch/Makefile
+        source/SoundTouchDLL/Makefile
 	include/Makefile
 ])
 
-AC_OUTPUT(
+AC_CONFIG_FILES([soundtouch.pc
-	soundtouch.pc
+])
-)
+AC_OUTPUT
 
 dnl use 'echo' to put stuff here if you want a message to the builder

createsrcpack

@@ -1,5 +1,4 @@
 # Helper script for building a source code release package
-# $Id$
 
 rm -Rf soundtouch
 rm soundtouch.zip

include/BPMDetect.h

@@ -26,13 +26,6 @@
 ///
 ////////////////////////////////////////////////////////////////////////////////
 //
-// Last changed  : $Date$
-// File revision : $Revision: 4 $
-//
-// $Id$
-//
-////////////////////////////////////////////////////////////////////////////////
-//
 // License :
 //
 //  SoundTouch audio processing library
@@ -57,32 +50,49 @@
 #ifndef _BPMDetect_H_
 #define _BPMDetect_H_
 
+#include <vector>
 #include "STTypes.h"
 #include "FIFOSampleBuffer.h"
 
 namespace soundtouch
 {
 
-/// Minimum allowed BPM rate. Used to restrict accepted result above a reasonable limit.
+    /// Minimum allowed BPM rate. Used to restrict accepted result above a reasonable limit.
-#define MIN_BPM 29
+    #define MIN_BPM 45
 
-/// Maximum allowed BPM rate. Used to restrict accepted result below a reasonable limit.
+    /// Maximum allowed BPM rate range. Used for calculating algorithm parametrs
-#define MAX_BPM 200
+    #define MAX_BPM_RANGE 200
+
+    /// Maximum allowed BPM rate range. Used to restrict accepted result below a reasonable limit.
+    #define MAX_BPM_VALID 190
+
+////////////////////////////////////////////////////////////////////////////////
+
+    typedef struct
+    {
+        float pos;
+        float strength;
+    } BEAT;
 
 
-/// Class for calculating BPM rate for audio data.
+    class IIR2_filter
-class BPMDetect
+    {
-{
+        double coeffs[5];
-protected:
+        double prev[5];
+
+    public:
+        IIR2_filter(const double *lpf_coeffs);
+        float update(float x);
+    };
+
+
+    /// Class for calculating BPM rate for audio data.
+    class BPMDetect
+    {
+    protected:
         /// Auto-correlation accumulator bins.
         float *xcorr;
 
-    /// Amplitude envelope sliding average approximation level accumulator
-    double envelopeAccu;
-
-    /// RMS volume sliding average approximation level accumulator
-    double RMSVolumeAccu;
-
         /// Sample average counter.
         int decimateCount;
 
@@ -105,9 +115,28 @@ protected:
         /// the first these many correlation bins.
         int windowStart;
 
+        /// window functions for data preconditioning
+        float *hamw;
+        float *hamw2;
+
+        // beat detection variables
+        int pos;
+        int peakPos;
+        int beatcorr_ringbuffpos;
+        int init_scaler;
+        float peakVal;
+        float *beatcorr_ringbuff;
+
         /// FIFO-buffer for decimated processing samples.
         soundtouch::FIFOSampleBuffer *buffer;
 
+        /// Collection of detected beat positions
+        //BeatCollection beats;
+        std::vector<BEAT> beats;
+
+        // 2nd order low-pass-filter
+        IIR2_filter beat_lpf;
+
         /// 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).
@@ -131,7 +160,11 @@ protected:
         /// remove constant bias from xcorr data
         void removeBias();
 
-public:
+        // Detect individual beat positions
+        void updateBeatPos(int process_samples);
+
+
+    public:
         /// Constructor.
         BPMDetect(int numChannels,  ///< Number of channels in sample data.
             int sampleRate    ///< Sample rate in Hz.
@@ -150,15 +183,23 @@ public:
             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();
-};
 
+        /// Get beat position arrays. Note: The array includes also really low beat detection values
+        /// in absence of clear strong beats. Consumer may wish to filter low values away.
+        /// - "pos" receive array of beat positions
+        /// - "values" receive array of beat detection strengths
+        /// - max_num indicates max.size of "pos" and "values" array.
+        ///
+        /// You can query a suitable array sized by calling this with nullptr in "pos" & "values".
+        ///
+        /// \return number of beats in the arrays.
+        int getBeats(float *pos, float *strength, int max_num);
+    };
 }
-
 #endif // _BPMDetect_H_

include/FIFOSampleBuffer.h

@@ -15,13 +15,6 @@
 ///
 ////////////////////////////////////////////////////////////////////////////////
 //
-// Last changed  : $Date$
-// File revision : $Revision: 4 $
-//
-// $Id$
-//
-////////////////////////////////////////////////////////////////////////////////
-//
 // License :
 //
 //  SoundTouch audio processing library
@@ -98,7 +91,7 @@ public:
                      );
 
     /// destructor
-    ~FIFOSampleBuffer();
+    ~FIFOSampleBuffer() override;
 
     /// Returns a pointer to the beginning of the output samples.
     /// This function is provided for accessing the output samples directly.
@@ -107,7 +100,7 @@ public:
     /// 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();
+    virtual SAMPLETYPE *ptrBegin() override;
 
     /// 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
@@ -119,7 +112,7 @@ public:
     /// 'putSamples(numSamples)' function.
     SAMPLETYPE *ptrEnd(
                 uint slackCapacity   ///< How much free capacity (in samples) there _at least_
-                                     ///< should be so that the caller can succesfully insert the 
+                                     ///< should be so that the caller can successfully insert the
                                      ///< desired samples to the buffer. If necessary, the function
                                      ///< grows the buffer size to comply with this requirement.
                 );
@@ -128,7 +121,7 @@ public:
     /// the sample buffer.
     virtual void putSamples(const SAMPLETYPE *samples,  ///< Pointer to samples.
                             uint numSamples                         ///< Number of samples to insert.
-                            );
+                            ) override;
 
     /// Adjusts the book-keeping to increase number of samples in the buffer without
     /// copying any actual samples.
@@ -146,7 +139,7 @@ public:
     /// \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.
-                                );
+                                ) override;
 
     /// Adjusts book-keeping so that given number of samples are removed from beginning of the
     /// sample buffer without copying them anywhere.
@@ -154,23 +147,32 @@ public:
     /// 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.
-                                );
+                                ) override;
 
     /// Returns number of samples currently available.
-    virtual uint numSamples() const;
+    virtual uint numSamples() const override;
 
     /// Sets number of channels, 1 = mono, 2 = stereo.
     void setChannels(int numChannels);
 
+    /// Get number of channels
+    int getChannels()
+    {
+        return channels;
+    }
+
     /// Returns nonzero if there aren't any samples available for outputting.
-    virtual int isEmpty() const;
+    virtual int isEmpty() const override;
 
     /// Clears all the samples.
-    virtual void clear();
+    virtual void clear() override;
 
     /// allow trimming (downwards) amount of samples in pipeline.
     /// Returns adjusted amount of samples
-    uint adjustAmountOfSamples(uint numSamples);
+    uint adjustAmountOfSamples(uint numSamples) override;
+
+    /// Add silence to end of buffer
+    void addSilent(uint nSamples);
 };
 
 }

include/FIFOSamplePipe.h

@@ -17,13 +17,6 @@
 ///
 ////////////////////////////////////////////////////////////////////////////////
 //
-// Last changed  : $Date$
-// File revision : $Revision: 4 $
-//
-// $Id$
-//
-////////////////////////////////////////////////////////////////////////////////
-//
 // License :
 //
 //  SoundTouch audio processing library
@@ -58,6 +51,18 @@ namespace soundtouch
 /// Abstract base class for FIFO (first-in-first-out) sample processing classes.
 class FIFOSamplePipe
 {
+protected:
+
+    bool verifyNumberOfChannels(int nChannels) const
+    {
+        if ((nChannels > 0) && (nChannels <= SOUNDTOUCH_MAX_CHANNELS))
+        {
+            return true;
+        }
+        ST_THROW_RT_ERROR("Error: Illegal number of channels");
+        return false;
+    }
+
 public:
     // virtual default destructor
     virtual ~FIFOSamplePipe() {}
@@ -83,11 +88,11 @@ public:
     void moveSamples(FIFOSamplePipe &other  ///< Other pipe instance where from the receive the data.
          )
     {
-        int oNumSamples = other.numSamples();
+        const uint 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
@@ -122,7 +127,6 @@ public:
 };
 
 
-
 /// Base-class for sound processing routines working in FIFO principle. With this base
 /// class it's easy to implement sound processing stages that can be chained together,
 /// so that samples that are fed into beginning of the pipe automatically go through
@@ -140,20 +144,18 @@ protected:
     /// Sets output pipe.
     void setOutPipe(FIFOSamplePipe *pOutput)
     {
-        assert(output == NULL);
+        assert(output == nullptr);
-        assert(pOutput != NULL);
+        assert(pOutput != nullptr);
         output = pOutput;
     }
 
-
     /// Constructor. Doesn't define output pipe; it has to be set be
     /// 'setOutPipe' function.
     FIFOProcessor()
     {
-        output = NULL;
+        output = nullptr;
     }
 
-
     /// Constructor. Configures output pipe.
     FIFOProcessor(FIFOSamplePipe *pOutput   ///< Output pipe.
                  )
@@ -161,13 +163,11 @@ protected:
         output = pOutput;
     }
 
-
     /// Destructor.
-    virtual ~FIFOProcessor()
+    virtual ~FIFOProcessor() override
     {
     }
 
-
     /// 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!
@@ -175,7 +175,7 @@ protected:
     /// 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()
+    virtual SAMPLETYPE *ptrBegin() override
     {
         return output->ptrBegin();
     }
@@ -189,44 +189,40 @@ public:
     /// \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.
-                                )
+                                ) override
     {
         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.
-                                )
+                                ) override
     {
         return output->receiveSamples(maxSamples);
     }
 
-
     /// Returns number of samples currently available.
-    virtual uint numSamples() const
+    virtual uint numSamples() const override
     {
         return output->numSamples();
     }
 
-
     /// Returns nonzero if there aren't any samples available for outputting.
-    virtual int isEmpty() const
+    virtual int isEmpty() const override
     {
         return output->isEmpty();
     }
 
     /// allow trimming (downwards) amount of samples in pipeline.
     /// Returns adjusted amount of samples
-    virtual uint adjustAmountOfSamples(uint numSamples)
+    virtual uint adjustAmountOfSamples(uint numSamples) override
     {
         return output->adjustAmountOfSamples(numSamples);
     }
-
 };
 
 }

include/Makefile.am

@@ -1,9 +1,5 @@
 ## Process this file with automake to create Makefile.in
 ##
-## $Id$
-##
-## 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

include/STTypes.h

@@ -8,13 +8,6 @@
 ///
 ////////////////////////////////////////////////////////////////////////////////
 //
-// Last changed  : $Date$
-// File revision : $Revision: 3 $
-//
-// $Id$
-//
-////////////////////////////////////////////////////////////////////////////////
-//
 // License :
 //
 //  SoundTouch audio processing library
@@ -42,29 +35,50 @@
 typedef unsigned int    uint;
 typedef unsigned long   ulong;
 
-#ifdef __GNUC__
+// Patch for MinGW: on Win64 long is 32-bit
-    // In GCC, include soundtouch_config.h made by config scritps
+#ifdef _WIN64
-    #include "soundtouch_config.h"
+    typedef unsigned long long ulongptr;
+#else
+    typedef ulong ulongptr;
 #endif
 
-#ifndef _WINDEF_
-    // if these aren't defined already by Windows headers, define now
 
-    typedef int BOOL;
+// Helper macro for aligning pointer up to next 16-byte boundary
+#define SOUNDTOUCH_ALIGN_POINTER_16(x)      ( ( (ulongptr)(x) + 15 ) & ~(ulongptr)15 )
 
-    #define FALSE   0
-    #define TRUE    1
 
-#endif  // _WINDEF_
+#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
 
 
 namespace soundtouch
 {
+    /// Max allowed number of channels. This is not a hard limit but to have some
+    /// maximum value for argument sanity checks -- can be increased if necessary
+    #define SOUNDTOUCH_MAX_CHANNELS     32
+
     /// 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 following flag is defined, always uses multichannel processing
+    /// routines also for mono and stero sound. This is for routine testing
+    /// purposes; output should be same with either routines, yet disabling
+    /// the dedicated mono/stereo processing routines will result in slower
+    /// runtime performance so recommendation is to keep this off.
+    // #define USE_MULTICH_ALWAYS
+
+    #if (defined(__SOFTFP__) && defined(ANDROID))
+        // 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
+
     #if !(SOUNDTOUCH_INTEGER_SAMPLES || SOUNDTOUCH_FLOAT_SAMPLES)
 
         /// Choose either 32bit floating point or 16bit integer sampletype
@@ -108,10 +122,10 @@ namespace soundtouch
 
     #endif
 
-    // If defined, allows the SIMD-optimized routines to take minor shortcuts 
+    // If defined, allows the SIMD-optimized routines to skip unevenly aligned
-    // for improved performance. Undefine to require faithfully similar SIMD 
+    // memory offsets that can cause performance penalty in some SIMD implementations.
-    // calculations as in normal C implementation.
+    // Causes slight compromise in sound quality.
-    #define SOUNDTOUCH_ALLOW_NONEXACT_SIMD_OPTIMIZATION    1
+    // #define SOUNDTOUCH_ALLOW_NONEXACT_SIMD_OPTIMIZATION    1
 
 
     #ifdef SOUNDTOUCH_INTEGER_SAMPLES
@@ -126,16 +140,19 @@ namespace soundtouch
         #endif // SOUNDTOUCH_FLOAT_SAMPLES
 
         #ifdef SOUNDTOUCH_ALLOW_X86_OPTIMIZATIONS
-            // Allow MMX optimizations
+            // Allow MMX optimizations (not available in X64 mode)
+            #if (!_M_X64)
                 #define SOUNDTOUCH_ALLOW_MMX   1
             #endif
+        #endif
 
     #else
 
         // floating point samples
         typedef float  SAMPLETYPE;
-        // data type for sample accumulation: Use double to utilize full precision.
+        // data type for sample accumulation: Use float also here to enable
-        typedef double LONG_SAMPLETYPE;
+        // efficient autovectorization
+        typedef float LONG_SAMPLETYPE;
 
         #ifdef SOUNDTOUCH_ALLOW_X86_OPTIMIZATIONS
             // Allow SSE optimizations
@@ -144,7 +161,13 @@ namespace soundtouch
 
     #endif  // SOUNDTOUCH_INTEGER_SAMPLES
 
-};
+    #if ((SOUNDTOUCH_ALLOW_SSE) || (__SSE__) || (SOUNDTOUCH_USE_NEON))
+        #if SOUNDTOUCH_ALLOW_NONEXACT_SIMD_OPTIMIZATION
+            #define ST_SIMD_AVOID_UNALIGNED
+        #endif
+    #endif
+
+}
 
 // define ST_NO_EXCEPTION_HANDLING switch to disable throwing std exceptions:
 // #define ST_NO_EXCEPTION_HANDLING    1
@@ -155,6 +178,7 @@ namespace soundtouch
 #else
     // use c++ standard exceptions
     #include <stdexcept>
+    #include <string>
     #define ST_THROW_RT_ERROR(x)    {throw std::runtime_error(x);}
 #endif
 

include/SoundTouch.h

@@ -41,13 +41,6 @@
 ///
 ////////////////////////////////////////////////////////////////////////////////
 //
-// Last changed  : $Date$
-// File revision : $Revision: 4 $
-//
-// $Id$
-//
-////////////////////////////////////////////////////////////////////////////////
-//
 // License :
 //
 //  SoundTouch audio processing library
@@ -79,10 +72,10 @@ namespace soundtouch
 {
 
 /// Soundtouch library version string
-#define SOUNDTOUCH_VERSION          "1.7.0"
+#define SOUNDTOUCH_VERSION          "2.3.3"
 
 /// SoundTouch library version id
-#define SOUNDTOUCH_VERSION_ID       (10700)
+#define SOUNDTOUCH_VERSION_ID       (20303)
 
 //
 // Available setting IDs for the 'setSetting' & 'get_setting' functions:
@@ -116,15 +109,17 @@ namespace soundtouch
 #define SETTING_OVERLAP_MS          5
 
 
-/// Call "getSetting" with this ID to query nominal average processing sequence
+/// Call "getSetting" with this ID to query processing sequence size in samples.
-/// size in samples. This value tells approcimate value how many input samples 
+/// This value gives approximate value of how many input samples you'll need to
-/// SoundTouch needs to gather before it does DSP processing run for the sample batch.
+/// feed into SoundTouch after initial buffering to get out a new batch of
+/// output samples.
+///
+/// This value does not include initial buffering at beginning of a new processing
+/// stream, use SETTING_INITIAL_LATENCY to get the initial buffering size.
 ///
 /// Notices:
 /// - This is read-only parameter, i.e. setSetting ignores this parameter
-/// - Returned value is approximate average value, exact processing batch
+/// - This parameter value is not constant but change depending on
-///   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
 
@@ -135,12 +130,41 @@ namespace soundtouch
 ///
 /// Notices:
 /// - This is read-only parameter, i.e. setSetting ignores this parameter
-/// - Returned value is approximate average value, exact processing batch
+/// - This parameter value is not constant but change depending on
-///   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
 
+
+/// Call "getSetting" with this ID to query initial processing latency, i.e.
+/// approx. how many samples you'll need to enter to SoundTouch pipeline before
+/// you can expect to get first batch of ready output samples out.
+///
+/// After the first output batch, you can then expect to get approx.
+/// SETTING_NOMINAL_OUTPUT_SEQUENCE ready samples out for every
+/// SETTING_NOMINAL_INPUT_SEQUENCE samples that you enter into SoundTouch.
+///
+/// Example:
+///     processing with parameter -tempo=5
+///     => initial latency = 5509 samples
+///        input sequence  = 4167 samples
+///        output sequence = 3969 samples
+///
+/// Accordingly, you can expect to feed in approx. 5509 samples at beginning of
+/// the stream, and then you'll get out the first 3969 samples. After that, for
+/// every approx. 4167 samples that you'll put in, you'll receive again approx.
+/// 3969 samples out.
+///
+/// This also means that average latency during stream processing is
+/// INITIAL_LATENCY-OUTPUT_SEQUENCE/2, in the above example case 5509-3969/2
+/// = 3524 samples
+///
+/// Notices:
+/// - This is read-only parameter, i.e. setSetting ignores this parameter
+/// - This parameter value is not constant but change depending on
+///   tempo/pitch/rate/samplerate settings.
+#define SETTING_INITIAL_LATENCY             8
+
+
 class SoundTouch : public FIFOProcessor
 {
 private:
@@ -151,16 +175,23 @@ private:
     class TDStretch *pTDStretch;
 
     /// Virtual pitch parameter. Effective rate & tempo are calculated from these parameters.
-    float virtualRate;
+    double virtualRate;
 
     /// Virtual pitch parameter. Effective rate & tempo are calculated from these parameters.
-    float virtualTempo;
+    double virtualTempo;
 
     /// Virtual pitch parameter. Effective rate & tempo are calculated from these parameters.
-    float virtualPitch;
+    double virtualPitch;
 
     /// Flag: Has sample rate been set?
-    BOOL  bSrateSet;
+    bool  bSrateSet;
+
+    /// Accumulator for how many samples in total will be expected as output vs. samples put in,
+    /// considering current processing settings.
+    double samplesExpectedOut;
+
+    /// Accumulator for how many samples in total have been read out from the processing so far
+    long   samplesOutput;
 
     /// Calculates effective rate & tempo valuescfrom 'virtualRate', 'virtualTempo' and
     /// 'virtualPitch' parameters.
@@ -171,14 +202,14 @@ protected :
     uint  channels;
 
     /// Effective 'rate' value calculated from 'virtualRate', 'virtualTempo' and 'virtualPitch'
-    float rate;
+    double rate;
 
     /// Effective 'tempo' value calculated from 'virtualRate', 'virtualTempo' and 'virtualPitch'
-    float tempo;
+    double tempo;
 
 public:
     SoundTouch();
-    virtual ~SoundTouch();
+    virtual ~SoundTouch() override;
 
     /// Get SoundTouch library version string
     static const char *getVersionString();
@@ -188,32 +219,32 @@ public:
 
     /// Sets new rate control value. Normal rate = 1.0, smaller values
     /// represent slower rate, larger faster rates.
-    void setRate(float newRate);
+    void setRate(double newRate);
 
     /// Sets new tempo control value. Normal tempo = 1.0, smaller values
     /// represent slower tempo, larger faster tempo.
-    void setTempo(float newTempo);
+    void setTempo(double newTempo);
 
     /// Sets new rate control value as a difference in percents compared
     /// to the original rate (-50 .. +100 %)
-    void setRateChange(float newRate);
+    void setRateChange(double newRate);
 
     /// Sets new tempo control value as a difference in percents compared
     /// to the original tempo (-50 .. +100 %)
-    void setTempoChange(float newTempo);
+    void setTempoChange(double newTempo);
 
     /// Sets new pitch control value. Original pitch = 1.0, smaller values
     /// represent lower pitches, larger values higher pitch.
-    void setPitch(float newPitch);
+    void setPitch(double newPitch);
 
     /// Sets pitch change in octaves compared to the original pitch
     /// (-1.00 .. +1.00)
-    void setPitchOctaves(float newPitch);
+    void setPitchOctaves(double newPitch);
 
     /// Sets pitch change in semi-tones compared to the original pitch
     /// (-12 .. +12)
     void setPitchSemiTones(int newPitch);
-    void setPitchSemiTones(float newPitch);
+    void setPitchSemiTones(double newPitch);
 
     /// Sets the number of channels, 1 = mono, 2 = stereo
     void setChannels(uint numChannels);
@@ -221,6 +252,24 @@ public:
     /// Sets sample rate.
     void setSampleRate(uint srate);
 
+    /// Get ratio between input and output audio durations, useful for calculating
+    /// processed output duration: if you'll process a stream of N samples, then
+    /// you can expect to get out N * getInputOutputSampleRatio() samples.
+    ///
+    /// This ratio will give accurate target duration ratio for a full audio track,
+    /// given that the the whole track is processed with same processing parameters.
+    ///
+    /// If this ratio is applied to calculate intermediate offsets inside a processing
+    /// stream, then this ratio is approximate and can deviate +- some tens of milliseconds
+    /// from ideal offset, yet by end of the audio stream the duration ratio will become
+    /// exact.
+    ///
+    /// Example: if processing with parameters "-tempo=15 -pitch=-3", the function
+    /// will return value 0.8695652... Now, if processing an audio stream whose duration
+    /// is exactly one million audio samples, then you can expect the processed
+    /// output duration  be 0.869565 * 1000000 = 869565 samples.
+    double getInputOutputSampleRatio();
+
     /// Flushes the last samples from the processing pipeline to the output.
     /// Clears also the internal processing buffers.
     //
@@ -238,17 +287,34 @@ public:
             uint numSamples                         ///< Number of samples in buffer. Notice
                                                     ///< that in case of stereo-sound a single sample
                                                     ///< contains data for both channels.
-            );
+            ) override;
+
+    /// 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.
+        ) override;
+
+    /// 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.
+        ) override;
 
     /// Clears all the samples in the object's output and internal processing
     /// buffers.
-    virtual void clear();
+    virtual void clear() override;
 
     /// Changes a setting controlling the processing system behaviour. See the
     /// 'SETTING_...' defines for available setting ID's.
     ///
-    /// \return 'TRUE' if the setting was succesfully changed
+    /// \return 'true' if the setting was successfully changed
-    BOOL setSetting(int settingId,   ///< Setting ID number. see SETTING_... defines.
+    bool setSetting(int settingId,   ///< Setting ID number. see SETTING_... defines.
                     int value        ///< New setting value.
                     );
 
@@ -262,6 +328,11 @@ public:
     /// Returns number of samples currently unprocessed.
     virtual uint numUnprocessedSamples() const;
 
+    /// Return number of channels
+    uint numChannels() const
+    {
+        return channels;
+    }
 
     /// Other handy functions that are implemented in the ancestor classes (see
     /// classes 'FIFOProcessor' and 'FIFOSamplePipe')

include/soundtouch_config.h

@@ -0,0 +1,3 @@
+// autotools configuration step replaces this file with a configured version.
+// this empty file stub is provided to avoid error about missing include file
+// when not using autotools build

include/soundtouch_config.h.in

@@ -0,0 +1,8 @@
+/* Use Float as Sample type */
+#undef SOUNDTOUCH_FLOAT_SAMPLES
+
+/* Use Integer as Sample type */
+#undef SOUNDTOUCH_INTEGER_SAMPLES
+
+/* Use ARM NEON extension */
+#undef SOUNDTOUCH_USE_NEON

make-win.bat

@@ -5,42 +5,21 @@
 @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 $Id$ 
-
-@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
+@rem devenv source\SoundStretch\SoundStretch.sln /upgrade
-md lib
+devenv source\SoundStretch\SoundStretch.sln /build "Debug|Win32"
-devenv source\SoundTouch\SoundTouch.vcproj /upgrade
+devenv source\SoundStretch\SoundStretch.sln /build "Release|Win32"
-devenv source\SoundTouch\SoundTouch.vcproj /build debug
+devenv source\SoundStretch\SoundStretch.sln /build "Debug|x64"
-devenv source\SoundTouch\SoundTouch.vcproj /build release
+devenv source\SoundStretch\SoundStretch.sln /build "Release|x64"
 
-devenv source\SoundStretch\SoundStretch.vcproj /upgrade
+@rem devenv source\SoundTouchDll\SoundTouchDll.sln /upgrade
-devenv source\SoundStretch\SoundStretch.vcproj /build debug
+devenv source\SoundTouchDll\SoundTouchDll.sln /build "Debug|Win32"
-devenv source\SoundStretch\SoundStretch.vcproj /build release
+devenv source\SoundTouchDll\SoundTouchDll.sln /build "Release|Win32"
-
+devenv source\SoundTouchDll\SoundTouchDll.sln /build "Debug|x64"
-devenv source\SoundTouchDll\SoundTouchDll.vcproj /upgrade
+devenv source\SoundTouchDll\SoundTouchDll.sln /build "Release|x64"
-devenv source\SoundTouchDll\SoundTouchDll.vcproj /build debug
-devenv source\SoundTouchDll\SoundTouchDll.vcproj /build release
 
 @goto end
 
@@ -48,14 +27,15 @@ devenv source\SoundTouchDll\SoundTouchDll.vcproj /build release
 :nodevdir
 
 @echo off
-echo *********************************************************************
+echo ****************************************************************************
 echo **
 echo ** ERROR: Visual Studio path not set.
 echo **
-echo ** Run "vsvars32.bat" or "vcvars32.bat" from Visual Studio installation
+echo ** Open "tools"->"Developer Command Line" from Visual Studio IDE, or
-echo ** directory, e.g. "\Program Files\Microsoft Visual Studio\VC98\Bin",
+echo ** run "vcvars32.bat" from Visual Studio installation dir, e.g.
+echo ** "C:\Program Files (x86)\Microsoft Visual Studio xxx\VC\bin",
 echo ** then try again.
 echo **
-echo *********************************************************************
+echo ****************************************************************************
 
 :end

readme.md

@@ -0,0 +1,66 @@
+# SoundTouch library
+
+## About
+
+SoundTouch is an open-source audio processing library that allows changing the sound tempo, pitch and playback rate parameters independently from each other:
+* Change **tempo** while maintaining the original pitch
+* Change **pitch** while maintaining the original tempo
+* Change **playback rate** that affects both tempo and pitch at the
+same time
+* Change any combination of tempo/pitch/rate
+
+Visit [SoundTouch website](https://www.surina.net/soundtouch) and see the [README file](https://www.surina.net/soundtouch/readme.html) for more information and audio examples.
+
+### The latest stable release is 2.3.3
+
+## Example
+
+Use SoundStretch example app for modifying wav audio files, for example as follows:
+
+```
+soundstretch my_original_file.wav output_file.wav -tempo=+15 -pitch=-3
+```
+
+See the [README file](http://soundtouch.surina.net/README.html) for more usage examples and instructions how to build SoundTouch + SoundStretch.
+
+Ready [SoundStretch application executables](https://www.surina.net/soundtouch/download.html) are available for download for Windows and Mac OS.
+
+## Language & Platforms
+
+SoundTouch is written in C++ and compiles in virtually any platform:
+* Windows
+* Mac OS
+* Linux & Unices (including also Raspberry, Beaglebone, Yocto etc embedded Linux flavors)
+* Android
+* iOS
+* embedded systems
+
+The source code package includes dynamic library import modules for C#, Java and Pascal/Delphi languages.
+
+## Tarballs
+
+Source code release tarballs:
+* https://www.surina.net/soundtouch/soundtouch-2.3.3.tar.gz
+* https://www.surina.net/soundtouch/soundtouch-2.3.2.tar.gz
+* https://www.surina.net/soundtouch/soundtouch-2.3.1.tar.gz
+* https://www.surina.net/soundtouch/soundtouch-2.3.0.tar.gz
+* https://www.surina.net/soundtouch/soundtouch-2.2.0.tar.gz
+* https://www.surina.net/soundtouch/soundtouch-2.1.2.tar.gz
+* https://www.surina.net/soundtouch/soundtouch-2.1.1.tar.gz
+* https://www.surina.net/soundtouch/soundtouch-2.0.0.tar.gz
+
+## License
+
+SoundTouch is released under LGPL v2.1:
+
+This library is free software; you can redistribute it and/or modify it under the terms of the GNU Lesser General Public License version 2.1 as published by the Free Software Foundation.
+
+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
+
+See [LGPL v2.1 full license text ](https://www.gnu.org/licenses/old-licenses/lgpl-2.1.html) for details.
+
+--
+
+Also commercial license free of GPL limitations available upon request

soundtouch-1.4.pc.in

@@ -1,14 +0,0 @@
-# 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  
-

soundtouch.m4

@@ -1,7 +1,4 @@
 # m4 configure test script for the SoundTouch library
-# (c)2003 David W. Durham
-#
-# $Id$
 #
 # This file can be included with other packages that need to test
 # for libSoundTouch.

source/Android-lib/.classpath

@@ -0,0 +1,9 @@
+<?xml version="1.0" encoding="UTF-8"?>
+<classpath>
+	<classpathentry kind="con" path="com.android.ide.eclipse.adt.ANDROID_FRAMEWORK"/>
+	<classpathentry exported="true" kind="con" path="com.android.ide.eclipse.adt.LIBRARIES"/>
+	<classpathentry exported="true" kind="con" path="com.android.ide.eclipse.adt.DEPENDENCIES"/>
+	<classpathentry kind="src" path="src"/>
+	<classpathentry kind="src" path="gen"/>
+	<classpathentry kind="output" path="bin/classes"/>
+</classpath>

source/Android-lib/.project

@@ -0,0 +1,33 @@
+<?xml version="1.0" encoding="UTF-8"?>
+<projectDescription>
+	<name>ExampleActivity</name>
+	<comment></comment>
+	<projects>
+	</projects>
+	<buildSpec>
+		<buildCommand>
+			<name>com.android.ide.eclipse.adt.ResourceManagerBuilder</name>
+			<arguments>
+			</arguments>
+		</buildCommand>
+		<buildCommand>
+			<name>com.android.ide.eclipse.adt.PreCompilerBuilder</name>
+			<arguments>
+			</arguments>
+		</buildCommand>
+		<buildCommand>
+			<name>org.eclipse.jdt.core.javabuilder</name>
+			<arguments>
+			</arguments>
+		</buildCommand>
+		<buildCommand>
+			<name>com.android.ide.eclipse.adt.ApkBuilder</name>
+			<arguments>
+			</arguments>
+		</buildCommand>
+	</buildSpec>
+	<natures>
+		<nature>com.android.ide.eclipse.adt.AndroidNature</nature>
+		<nature>org.eclipse.jdt.core.javanature</nature>
+	</natures>
+</projectDescription>

source/Android-lib/AndroidManifest.xml

@@ -0,0 +1,30 @@
+<?xml version="1.0" encoding="utf-8"?>
+<manifest xmlns:android="http://schemas.android.com/apk/res/android"
+    package="net.surina.soundtouchexample"
+    android:versionCode="1"
+    android:versionName="1.0" >
+
+    <uses-sdk
+        android:minSdkVersion="11"
+        android:targetSdkVersion="21" />
+
+    <uses-permission android:name="android.permission.READ_EXTERNAL_STORAGE" />
+    <uses-permission android:name="android.permission.WRITE_EXTERNAL_STORAGE" />
+    
+    <application
+        android:allowBackup="true"
+        android:icon="@drawable/ic_launcher"
+        android:label="@string/app_name"
+        android:theme="@style/AppTheme" >
+        <activity
+            android:name="net.surina.ExampleActivity"
+            android:label="@string/app_name" >
+            <intent-filter>
+                <action android:name="android.intent.action.MAIN" />
+
+                <category android:name="android.intent.category.LAUNCHER" />
+            </intent-filter>
+        </activity>
+    </application>
+
+</manifest>

source/Android-lib/README-SoundTouch-Android.html

@@ -0,0 +1,128 @@
+<!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 binaries for all the supported Android platforms (arm-v5, arm-v7, X86, MIPS etc) of SoundTouch library, plus the JNI wrapper interface as discussed below. The target binaries will be built into the &quot;libs&quot; subdirectory. As long as all these .so binary library versions are included in the APK Application delivery package, the target Android device can choose the correct library version to use. </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>
+<p><b>Enabling OpenMP parallel computing mode</b></p>
+<p>
+    SoundTouch supports OpenMP for parallel computing in multi-core 
+	environments, and these improvements can be enabled also in the Android 
+	build. See the SoundTouch main README.html file for generic notes about the 
+	OpenMP implementation.</p>
+<p>
+    To enable OpenMP mode in Android compilation, edit file <strong>Android.mk</strong> 
+	and enable the &quot;-fopenmp&quot; flag in LOCAL_CFLAGS and LOCAL_LDFLAGS variables. 
+	This is done by removing hash # from before the following lines in the 
+	Android.mk file, before compiling the library:</p>
+<pre>    LOCAL_CFLAGS += -fopenmp
+   LOCAL_LDFLAGS += -fopenmp</pre>
+<p><strong>OpenMP COMPATIBILITY NOTE: </strong>Android NDK has a threading issue 
+(at least until NDK v10) that causes the native library crash with fatal signal 
+11 if calling OpenMP-improved routines from a background thread. SoundTouch has 
+a workaround for this issue in soundtouch-jni.cpp, and this workaround requires 
+calling function <strong>SoundTouch.getVersionString() </strong>from the Android 
+application's main thread at least once before calling other SoundTouch 
+processing routines. See the SoundTouch Android example application and comments 
+in the <strong>soundtouch-jni.cpp </strong>source code file for more details.</p>
+<p>
+    <strong>SoundTouch performance in Android</strong></p>
+<p>
+    See external blog articles for more discussion about the
+	<a href="http://www.softwarecoven.com/parallel-computing-in-embedded-mobile-devices/">
+	SoundTouch OpenMP implementation</a> and the
+	<a href="http://www.softwarecoven.com/parallel-computing-with-openmp-in-android/">
+	SoundTouch performance benchmark tests in Android environment</a>.</p>
+<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, and provides source codes also for
+	a simple example Android application:<ul>
+	<li><b>ExampleActivity</b>: This is simple Android example application that
+	utilizes SoundTouch native routines for processing WAV audio files. To build the example
+	application, use Eclipse Android SDK environment to import the "ExampleActivity" project in the 	"Android-lib" folder into the Eclipse workspace. 
+        <li><b>Android-lib/jni/soundtouch-jni.cpp</b>: This file contains c/c++ wrapper routines
+	for performing elementary audio file processing with adjusted tempo/pitch/speed parameters
+	from the Android application. The wrapper interface is not complete, but provides example
+	that is easy to extend when necessary. 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 implements
+	the Java interface class that loasd & accesses the JNI routines in the natively compiled library.
+	The example Android application uses this class as interface for processing audio files 
+	with SoundTouch.</li>
+        <li><b>Android-lib/build.gradle</b>: Top level build script file for Android Studio 3.1.4+</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 own Android application.</p>
+<hr />
+<h2>
+    Android floating-point performance considerations</h2>
+<p>
+        The make process will build dedicated binaries for each supported Android CPU hardware platform type.
+</p><p>SoundTouch uses floating-point algorithms for ideal sound quality on all other platform than in the lowest-end ARMv5. That is because lowest-end Android devices are not guaranteed to 
+        have floating-point hardware in their CPUs, so that the ARMv5 compilation uses by default software-emulation for floating-point calculations to allow running the binary executables also in low-end devices without floating-point hardware.<p>
+        As floating point software-emulation is however several tens of times slower 
+        than real hardware-level floating-point calculations, that would make running
+        floating-point-intensive applications such as SoundTouch infeasible in these low-end 
+        devices. As workaround, the SoundTouch Android compilation builds the ARMv5 version using integer algorithm versions. 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.</p>
+        <p>When Android devices with more capable device is used, the device will automatically choose a proper library version for ideal sound quality.</p>
+<hr />
+    <p style="text-align: center"><i>Copyright &copy; Olli Parviainen</i></p>
+</body>
+</html>

source/Android-lib/build.gradle

@@ -0,0 +1,55 @@
+// Top-level build file where you can add configuration options common to all sub-projects/modules.
+buildscript {
+    repositories {
+        jcenter()
+        google()
+    }
+    dependencies {
+        classpath 'com.android.tools.build:gradle:3.1.4'
+    }
+}
+
+allprojects {
+    repositories {
+        jcenter()
+        google()
+    }
+}
+
+apply plugin: 'com.android.application'
+
+android {
+    compileSdkVersion 28
+
+    defaultConfig {
+        applicationId "net.surina.soundtouchexample"
+        minSdkVersion 14
+        targetSdkVersion 21
+
+        externalNativeBuild.ndkBuild {
+            arguments "NDK_APPLICATION=jni/Application.mk",
+                    "APP_ALLOW_MISSING_DEPS:=true"
+        }
+    }
+
+    sourceSets {
+        main {
+            manifest.srcFile "./AndroidManifest.xml"
+            java.srcDirs = ["./src"]
+            res.srcDirs = ["res"]
+        }
+    }
+
+    externalNativeBuild {
+        ndkBuild {
+            path 'jni/Android.mk'
+        }
+    }
+
+
+    buildTypes {
+        release {
+            minifyEnabled false
+        }
+    }
+}

source/Android-lib/gradle/wrapper/gradle-wrapper.properties

@@ -0,0 +1,6 @@
+#Sat Jan 13 09:12:34 PST 2018
+distributionBase=GRADLE_USER_HOME
+distributionPath=wrapper/dists
+zipStoreBase=GRADLE_USER_HOME
+zipStorePath=wrapper/dists
+distributionUrl=https\://services.gradle.org/distributions/gradle-4.4.1-all.zip

source/Android-lib/gradlew

@@ -0,0 +1,172 @@
+#!/usr/bin/env sh
+
+##############################################################################
+##
+##  Gradle start up script for UN*X
+##
+##############################################################################
+
+# Attempt to set APP_HOME
+# Resolve links: $0 may be a link
+PRG="$0"
+# Need this for relative symlinks.
+while [ -h "$PRG" ] ; do
+    ls=`ls -ld "$PRG"`
+    link=`expr "$ls" : '.*-> \(.*\)$'`
+    if expr "$link" : '/.*' > /dev/null; then
+        PRG="$link"
+    else
+        PRG=`dirname "$PRG"`"/$link"
+    fi
+done
+SAVED="`pwd`"
+cd "`dirname \"$PRG\"`/" >/dev/null
+APP_HOME="`pwd -P`"
+cd "$SAVED" >/dev/null
+
+APP_NAME="Gradle"
+APP_BASE_NAME=`basename "$0"`
+
+# Add default JVM options here. You can also use JAVA_OPTS and GRADLE_OPTS to pass JVM options to this script.
+DEFAULT_JVM_OPTS=""
+
+# Use the maximum available, or set MAX_FD != -1 to use that value.
+MAX_FD="maximum"
+
+warn () {
+    echo "$*"
+}
+
+die () {
+    echo
+    echo "$*"
+    echo
+    exit 1
+}
+
+# OS specific support (must be 'true' or 'false').
+cygwin=false
+msys=false
+darwin=false
+nonstop=false
+case "`uname`" in
+  CYGWIN* )
+    cygwin=true
+    ;;
+  Darwin* )
+    darwin=true
+    ;;
+  MINGW* )
+    msys=true
+    ;;
+  NONSTOP* )
+    nonstop=true
+    ;;
+esac
+
+CLASSPATH=$APP_HOME/gradle/wrapper/gradle-wrapper.jar
+
+# Determine the Java command to use to start the JVM.
+if [ -n "$JAVA_HOME" ] ; then
+    if [ -x "$JAVA_HOME/jre/sh/java" ] ; then
+        # IBM's JDK on AIX uses strange locations for the executables
+        JAVACMD="$JAVA_HOME/jre/sh/java"
+    else
+        JAVACMD="$JAVA_HOME/bin/java"
+    fi
+    if [ ! -x "$JAVACMD" ] ; then
+        die "ERROR: JAVA_HOME is set to an invalid directory: $JAVA_HOME
+
+Please set the JAVA_HOME variable in your environment to match the
+location of your Java installation."
+    fi
+else
+    JAVACMD="java"
+    which java >/dev/null 2>&1 || die "ERROR: JAVA_HOME is not set and no 'java' command could be found in your PATH.
+
+Please set the JAVA_HOME variable in your environment to match the
+location of your Java installation."
+fi
+
+# Increase the maximum file descriptors if we can.
+if [ "$cygwin" = "false" -a "$darwin" = "false" -a "$nonstop" = "false" ] ; then
+    MAX_FD_LIMIT=`ulimit -H -n`
+    if [ $? -eq 0 ] ; then
+        if [ "$MAX_FD" = "maximum" -o "$MAX_FD" = "max" ] ; then
+            MAX_FD="$MAX_FD_LIMIT"
+        fi
+        ulimit -n $MAX_FD
+        if [ $? -ne 0 ] ; then
+            warn "Could not set maximum file descriptor limit: $MAX_FD"
+        fi
+    else
+        warn "Could not query maximum file descriptor limit: $MAX_FD_LIMIT"
+    fi
+fi
+
+# For Darwin, add options to specify how the application appears in the dock
+if $darwin; then
+    GRADLE_OPTS="$GRADLE_OPTS \"-Xdock:name=$APP_NAME\" \"-Xdock:icon=$APP_HOME/media/gradle.icns\""
+fi
+
+# For Cygwin, switch paths to Windows format before running java
+if $cygwin ; then
+    APP_HOME=`cygpath --path --mixed "$APP_HOME"`
+    CLASSPATH=`cygpath --path --mixed "$CLASSPATH"`
+    JAVACMD=`cygpath --unix "$JAVACMD"`
+
+    # We build the pattern for arguments to be converted via cygpath
+    ROOTDIRSRAW=`find -L / -maxdepth 1 -mindepth 1 -type d 2>/dev/null`
+    SEP=""
+    for dir in $ROOTDIRSRAW ; do
+        ROOTDIRS="$ROOTDIRS$SEP$dir"
+        SEP="|"
+    done
+    OURCYGPATTERN="(^($ROOTDIRS))"
+    # Add a user-defined pattern to the cygpath arguments
+    if [ "$GRADLE_CYGPATTERN" != "" ] ; then
+        OURCYGPATTERN="$OURCYGPATTERN|($GRADLE_CYGPATTERN)"
+    fi
+    # Now convert the arguments - kludge to limit ourselves to /bin/sh
+    i=0
+    for arg in "$@" ; do
+        CHECK=`echo "$arg"|egrep -c "$OURCYGPATTERN" -`
+        CHECK2=`echo "$arg"|egrep -c "^-"`                                 ### Determine if an option
+
+        if [ $CHECK -ne 0 ] && [ $CHECK2 -eq 0 ] ; then                    ### Added a condition
+            eval `echo args$i`=`cygpath --path --ignore --mixed "$arg"`
+        else
+            eval `echo args$i`="\"$arg\""
+        fi
+        i=$((i+1))
+    done
+    case $i in
+        (0) set -- ;;
+        (1) set -- "$args0" ;;
+        (2) set -- "$args0" "$args1" ;;
+        (3) set -- "$args0" "$args1" "$args2" ;;
+        (4) set -- "$args0" "$args1" "$args2" "$args3" ;;
+        (5) set -- "$args0" "$args1" "$args2" "$args3" "$args4" ;;
+        (6) set -- "$args0" "$args1" "$args2" "$args3" "$args4" "$args5" ;;
+        (7) set -- "$args0" "$args1" "$args2" "$args3" "$args4" "$args5" "$args6" ;;
+        (8) set -- "$args0" "$args1" "$args2" "$args3" "$args4" "$args5" "$args6" "$args7" ;;
+        (9) set -- "$args0" "$args1" "$args2" "$args3" "$args4" "$args5" "$args6" "$args7" "$args8" ;;
+    esac
+fi
+
+# Escape application args
+save () {
+    for i do printf %s\\n "$i" | sed "s/'/'\\\\''/g;1s/^/'/;\$s/\$/' \\\\/" ; done
+    echo " "
+}
+APP_ARGS=$(save "$@")
+
+# Collect all arguments for the java command, following the shell quoting and substitution rules
+eval set -- $DEFAULT_JVM_OPTS $JAVA_OPTS $GRADLE_OPTS "\"-Dorg.gradle.appname=$APP_BASE_NAME\"" -classpath "\"$CLASSPATH\"" org.gradle.wrapper.GradleWrapperMain "$APP_ARGS"
+
+# by default we should be in the correct project dir, but when run from Finder on Mac, the cwd is wrong
+if [ "$(uname)" = "Darwin" ] && [ "$HOME" = "$PWD" ]; then
+  cd "$(dirname "$0")"
+fi
+
+exec "$JAVACMD" "$@"

source/Android-lib/gradlew.bat

@@ -0,0 +1,84 @@
+@if "%DEBUG%" == "" @echo off
+@rem ##########################################################################
+@rem
+@rem  Gradle startup script for Windows
+@rem
+@rem ##########################################################################
+
+@rem Set local scope for the variables with windows NT shell
+if "%OS%"=="Windows_NT" setlocal
+
+set DIRNAME=%~dp0
+if "%DIRNAME%" == "" set DIRNAME=.
+set APP_BASE_NAME=%~n0
+set APP_HOME=%DIRNAME%
+
+@rem Add default JVM options here. You can also use JAVA_OPTS and GRADLE_OPTS to pass JVM options to this script.
+set DEFAULT_JVM_OPTS=
+
+@rem Find java.exe
+if defined JAVA_HOME goto findJavaFromJavaHome
+
+set JAVA_EXE=java.exe
+%JAVA_EXE% -version >NUL 2>&1
+if "%ERRORLEVEL%" == "0" goto init
+
+echo.
+echo ERROR: JAVA_HOME is not set and no 'java' command could be found in your PATH.
+echo.
+echo Please set the JAVA_HOME variable in your environment to match the
+echo location of your Java installation.
+
+goto fail
+
+:findJavaFromJavaHome
+set JAVA_HOME=%JAVA_HOME:"=%
+set JAVA_EXE=%JAVA_HOME%/bin/java.exe
+
+if exist "%JAVA_EXE%" goto init
+
+echo.
+echo ERROR: JAVA_HOME is set to an invalid directory: %JAVA_HOME%
+echo.
+echo Please set the JAVA_HOME variable in your environment to match the
+echo location of your Java installation.
+
+goto fail
+
+:init
+@rem Get command-line arguments, handling Windows variants
+
+if not "%OS%" == "Windows_NT" goto win9xME_args
+
+:win9xME_args
+@rem Slurp the command line arguments.
+set CMD_LINE_ARGS=
+set _SKIP=2
+
+:win9xME_args_slurp
+if "x%~1" == "x" goto execute
+
+set CMD_LINE_ARGS=%*
+
+:execute
+@rem Setup the command line
+
+set CLASSPATH=%APP_HOME%\gradle\wrapper\gradle-wrapper.jar
+
+@rem Execute Gradle
+"%JAVA_EXE%" %DEFAULT_JVM_OPTS% %JAVA_OPTS% %GRADLE_OPTS% "-Dorg.gradle.appname=%APP_BASE_NAME%" -classpath "%CLASSPATH%" org.gradle.wrapper.GradleWrapperMain %CMD_LINE_ARGS%
+
+:end
+@rem End local scope for the variables with windows NT shell
+if "%ERRORLEVEL%"=="0" goto mainEnd
+
+:fail
+rem Set variable GRADLE_EXIT_CONSOLE if you need the _script_ return code instead of
+rem the _cmd.exe /c_ return code!
+if  not "" == "%GRADLE_EXIT_CONSOLE%" exit 1
+exit /b 1
+
+:mainEnd
+if "%OS%"=="Windows_NT" endlocal
+
+:omega

source/Android-lib/jni/Android.mk

@@ -0,0 +1,37 @@
+LOCAL_PATH := $(call my-dir)
+
+include $(CLEAR_VARS)
+
+LOCAL_C_INCLUDES += $(LOCAL_PATH)/../../../include $(LOCAL_PATH)/../../SoundStretch
+# *** 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/sse_optimized.cpp ../../SoundStretch/WavFile.cpp \
+                ../../SoundTouch/RateTransposer.cpp ../../SoundTouch/SoundTouch.cpp \
+                ../../SoundTouch/InterpolateCubic.cpp ../../SoundTouch/InterpolateLinear.cpp \
+                ../../SoundTouch/InterpolateShannon.cpp ../../SoundTouch/TDStretch.cpp \
+                ../../SoundTouch/BPMDetect.cpp ../../SoundTouch/PeakFinder.cpp 
+
+# for native audio
+LOCAL_SHARED_LIBRARIES += -lgcc 
+# --whole-archive -lgcc 
+# for logging
+LOCAL_LDLIBS    += -llog
+# for native asset manager
+#LOCAL_LDLIBS    += -landroid
+
+# Custom Flags: 
+# -fvisibility=hidden : don't export all symbols
+LOCAL_CFLAGS += -fvisibility=hidden -fdata-sections -ffunction-sections -ffast-math
+
+# OpenMP mode : enable these flags to enable using OpenMP for parallel computation 
+#LOCAL_CFLAGS += -fopenmp
+#LOCAL_LDFLAGS += -fopenmp
+
+
+# Use ARM instruction set instead of Thumb for improved calculation performance in ARM CPUs	
+LOCAL_ARM_MODE := arm
+
+include $(BUILD_SHARED_LIBRARY)

source/Android-lib/jni/Application.mk

@@ -0,0 +1,9 @@
+#
+# Build library bilaries for all supported architectures
+#
+
+APP_ABI := all #armeabi-v7a armeabi
+APP_OPTIM := release
+APP_STL := c++_static
+APP_CPPFLAGS := -fexceptions # -D SOUNDTOUCH_DISABLE_X86_OPTIMIZATIONS
+

source/Android-lib/jni/soundtouch-jni.cpp

@@ -0,0 +1,258 @@
+////////////////////////////////////////////////////////////////////////////////
+///
+/// Example Interface class for SoundTouch native compilation
+///
+/// Author        : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai 'at' iki.fi
+/// WWW           : http://www.surina.net
+///
+////////////////////////////////////////////////////////////////////////////////
+
+#include <jni.h>
+#include <android/log.h>
+#include <stdexcept>
+#include <string>
+
+using namespace std;
+
+#include "../../../include/SoundTouch.h"
+#include "../source/SoundStretch/WavFile.h"
+
+#define LOGV(...)   __android_log_print((int)ANDROID_LOG_INFO, "SOUNDTOUCH", __VA_ARGS__)
+//#define LOGV(...)
+
+
+// String for keeping possible c++ exception error messages. Notice that this isn't
+// thread-safe but it's expected that exceptions are special situations that won't
+// occur in several threads in parallel.
+static string _errMsg = "";
+
+
+#define DLL_PUBLIC __attribute__ ((visibility ("default")))
+#define BUFF_SIZE 4096
+
+
+using namespace soundtouch;
+
+
+// Set error message to return
+static void _setErrmsg(const char *msg)
+{
+	_errMsg = msg;
+}
+
+#if 0   // apparently following workaround not needed any more with concurrent Android SDKs
+#ifdef _OPENMP
+
+#include <pthread.h>
+extern pthread_key_t gomp_tls_key;
+static void * _p_gomp_tls = nullptr;
+
+/// Function to initialize threading for OpenMP.
+///
+/// This is a workaround for bug in Android NDK v10 regarding OpenMP: OpenMP works only if
+/// called from the Android App main thread because in the main thread the gomp_tls storage is
+/// properly set, however, Android does not properly initialize gomp_tls storage for other threads.
+/// Thus if OpenMP routines are invoked from some other thread than the main thread,
+/// the OpenMP routine will crash the application due to nullptr access on uninitialized storage.
+///
+/// This workaround stores the gomp_tls storage from main thread, and copies to other threads.
+/// In order this to work, the Application main thread needws to call at least "getVersionString"
+/// routine.
+static int _init_threading(bool warn)
+{
+	void *ptr = pthread_getspecific(gomp_tls_key);
+	LOGV("JNI thread-specific TLS storage %ld", (long)ptr);
+	if (ptr == nullptr)
+	{
+		LOGV("JNI set missing TLS storage to %ld", (long)_p_gomp_tls);
+		pthread_setspecific(gomp_tls_key, _p_gomp_tls);
+	}
+	else
+	{
+		LOGV("JNI store this TLS storage");
+		_p_gomp_tls = ptr;
+	}
+	// Where critical, show warning if storage still not properly initialized
+	if ((warn) && (_p_gomp_tls == nullptr))
+	{
+		_setErrmsg("Error - OpenMP threading not properly initialized: Call SoundTouch.getVersionString() from the App main thread!");
+		return -1;
+	}
+	return 0;
+}
+
+#else
+static int _init_threading(bool warn)
+{
+	// do nothing if not OpenMP build
+	return 0;
+}
+#endif
+
+#endif
+
+// Processes the sound file
+static void _processFile(SoundTouch *pSoundTouch, const char *inFileName, const char *outFileName)
+{
+    int nSamples;
+    int nChannels;
+    int buffSizeSamples;
+    SAMPLETYPE sampleBuffer[BUFF_SIZE];
+
+    // open input file
+    WavInFile inFile(inFileName);
+    int sampleRate = inFile.getSampleRate();
+    int bits = inFile.getNumBits();
+    nChannels = inFile.getNumChannels();
+
+    // create output file
+    WavOutFile outFile(outFileName, sampleRate, bits, nChannels);
+
+    pSoundTouch->setSampleRate(sampleRate);
+    pSoundTouch->setChannels(nChannels);
+
+    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 / nChannels;
+
+        // 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);
+}
+
+
+
+extern "C" DLL_PUBLIC jstring Java_net_surina_soundtouch_SoundTouch_getVersionString(JNIEnv *env, jobject thiz)
+{
+    const char *verStr;
+
+    LOGV("JNI call SoundTouch.getVersionString");
+
+    // Call example SoundTouch routine
+    verStr = SoundTouch::getVersionString();
+
+    // gomp_tls storage bug workaround - see comments in _init_threading() function!
+    // update: apparently this is not needed any more with concurrent Android SDKs
+    // _init_threading(false);
+
+    int threads = 0;
+	#pragma omp parallel
+    {
+		#pragma omp atomic
+    	threads ++;
+    }
+    LOGV("JNI thread count %d", threads);
+
+    // return version as string
+    return env->NewStringUTF(verStr);
+}
+
+
+
+extern "C" DLL_PUBLIC jlong Java_net_surina_soundtouch_SoundTouch_newInstance(JNIEnv *env, jobject thiz)
+{
+	return (jlong)(new SoundTouch());
+}
+
+
+extern "C" DLL_PUBLIC void Java_net_surina_soundtouch_SoundTouch_deleteInstance(JNIEnv *env, jobject thiz, jlong handle)
+{
+	SoundTouch *ptr = (SoundTouch*)handle;
+	delete ptr;
+}
+
+
+extern "C" DLL_PUBLIC void Java_net_surina_soundtouch_SoundTouch_setTempo(JNIEnv *env, jobject thiz, jlong handle, jfloat tempo)
+{
+	SoundTouch *ptr = (SoundTouch*)handle;
+	ptr->setTempo(tempo);
+}
+
+
+extern "C" DLL_PUBLIC void Java_net_surina_soundtouch_SoundTouch_setPitchSemiTones(JNIEnv *env, jobject thiz, jlong handle, jfloat pitch)
+{
+	SoundTouch *ptr = (SoundTouch*)handle;
+	ptr->setPitchSemiTones(pitch);
+}
+
+
+extern "C" DLL_PUBLIC void Java_net_surina_soundtouch_SoundTouch_setSpeed(JNIEnv *env, jobject thiz, jlong handle, jfloat speed)
+{
+	SoundTouch *ptr = (SoundTouch*)handle;
+	ptr->setRate(speed);
+}
+
+
+extern "C" DLL_PUBLIC jstring Java_net_surina_soundtouch_SoundTouch_getErrorString(JNIEnv *env, jobject thiz)
+{
+	jstring result = env->NewStringUTF(_errMsg.c_str());
+	_errMsg.clear();
+
+	return result;
+}
+
+
+extern "C" DLL_PUBLIC int Java_net_surina_soundtouch_SoundTouch_processFile(JNIEnv *env, jobject thiz, jlong handle, jstring jinputFile, jstring joutputFile)
+{
+	SoundTouch *ptr = (SoundTouch*)handle;
+
+	const char *inputFile = env->GetStringUTFChars(jinputFile, 0);
+	const char *outputFile = env->GetStringUTFChars(joutputFile, 0);
+
+	LOGV("JNI process file %s", inputFile);
+
+    /// gomp_tls storage bug workaround - see comments in _init_threading() function!
+    // update: apparently this is not needed any more with concurrent Android SDKs
+    // if (_init_threading(true)) return -1;
+
+	try
+	{
+		_processFile(ptr, inputFile, outputFile);
+	}
+	catch (const runtime_error &e)
+    {
+		const char *err = e.what();
+        // An exception occurred during processing, return the error message
+    	LOGV("JNI exception in SoundTouch::processFile: %s", err);
+        _setErrmsg(err);
+        return -1;
+    }
+
+
+	env->ReleaseStringUTFChars(jinputFile, inputFile);
+	env->ReleaseStringUTFChars(joutputFile, outputFile);
+
+	return 0;
+}

source/Android-lib/lint.xml

@@ -0,0 +1,3 @@
+<?xml version="1.0" encoding="UTF-8"?>
+<lint>
+</lint>

source/Android-lib/proguard-project.txt

@@ -0,0 +1,20 @@
+# To enable ProGuard in your project, edit project.properties
+# to define the proguard.config property as described in that file.
+#
+# Add project specific ProGuard rules here.
+# By default, the flags in this file are appended to flags specified
+# in ${sdk.dir}/tools/proguard/proguard-android.txt
+# You can edit the include path and order by changing the ProGuard
+# include property in project.properties.
+#
+# For more details, see
+#   http://developer.android.com/guide/developing/tools/proguard.html
+
+# Add any project specific keep options here:
+
+# If your project uses WebView with JS, uncomment the following
+# and specify the fully qualified class name to the JavaScript interface
+# class:
+#-keepclassmembers class fqcn.of.javascript.interface.for.webview {
+#   public *;
+#}

source/Android-lib/project.properties

@@ -0,0 +1,14 @@
+# This file is automatically generated by Android Tools.
+# Do not modify this file -- YOUR CHANGES WILL BE ERASED!
+#
+# This file must be checked in Version Control Systems.
+#
+# To customize properties used by the Ant build system edit
+# "ant.properties", and override values to adapt the script to your
+# project structure.
+#
+# To enable ProGuard to shrink and obfuscate your code, uncomment this (available properties: sdk.dir, user.home):
+#proguard.config=${sdk.dir}/tools/proguard/proguard-android.txt:proguard-project.txt
+
+# Project target.
+target=android-19

source/Android-lib/res/layout/activity_example.xml

@@ -0,0 +1,140 @@
+<LinearLayout xmlns:android="http://schemas.android.com/apk/res/android"
+    xmlns:tools="http://schemas.android.com/tools"
+    android:id="@+id/linearLayout1"
+    android:layout_width="fill_parent"
+    android:layout_height="fill_parent"
+    android:orientation="vertical" >
+
+    <TableLayout
+        android:layout_width="fill_parent"
+        android:layout_height="wrap_content" >
+
+        <TableRow
+            android:id="@+id/tableRow1"
+            android:layout_width="wrap_content"
+            android:layout_height="wrap_content" >
+
+            <TextView
+                android:layout_width="wrap_content"
+                android:layout_height="wrap_content"
+                android:text="Tempo %:"
+                android:textAppearance="?android:attr/textAppearanceMedium" />
+
+            <EditText
+                android:id="@+id/editTextTempo"
+                android:layout_width="wrap_content"
+                android:layout_height="wrap_content"
+                android:ems="5"
+                android:inputType="text"
+                android:text="100" >
+            </EditText>
+        </TableRow>
+
+        <TableRow
+            android:id="@+id/tableRow2"
+            android:layout_width="wrap_content"
+            android:layout_height="wrap_content" >
+
+            <TextView
+                android:layout_width="wrap_content"
+                android:layout_height="wrap_content"
+                android:text="Pitch half-steps:"
+                android:textAppearance="?android:attr/textAppearanceMedium" />
+
+            <EditText
+                android:id="@+id/editTextPitch"
+                android:layout_width="wrap_content"
+                android:layout_height="wrap_content"
+                android:ems="5"
+                android:inputType="text"
+                android:text="-0.318" >
+            </EditText>
+        </TableRow>
+    </TableLayout>
+
+    <LinearLayout
+        android:layout_width="fill_parent"
+        android:layout_height="wrap_content"
+        android:layout_marginTop="10dp" >
+
+        <TextView
+            android:layout_width="wrap_content"
+            android:layout_height="wrap_content"
+            android:text="Source file:"
+            android:textAppearance="?android:attr/textAppearanceMedium" />
+
+        <EditText
+            android:id="@+id/editTextSrcFileName"
+            android:layout_width="fill_parent"
+            android:layout_height="wrap_content"
+            android:text="/sdcard/Download/test.wav"
+            android:layout_weight="1" />
+               
+
+        <Button
+            android:id="@+id/buttonSelectSrcFile"
+            android:layout_width="wrap_content"
+            android:layout_height="wrap_content"
+            android:text="Select" />
+    </LinearLayout>
+
+    <LinearLayout
+        android:layout_width="fill_parent"
+        android:layout_height="wrap_content" >
+
+        <TextView
+            android:layout_width="wrap_content"
+            android:layout_height="wrap_content"
+            android:text="Output file:"
+            android:textAppearance="?android:attr/textAppearanceMedium" />
+
+        <EditText
+            android:id="@+id/editTextOutFileName"
+            android:layout_width="fill_parent"
+            android:layout_height="wrap_content"
+            android:text="/sdcard/Download/soundtouch-output.wav"
+            android:layout_weight="1" />
+
+        <Button
+            android:id="@+id/buttonSelectOutFile"
+            android:layout_width="wrap_content"
+            android:layout_height="wrap_content"
+            android:text="Select" />
+    </LinearLayout>
+
+    <CheckBox
+        android:id="@+id/checkBoxPlay"
+        android:layout_width="wrap_content"
+        android:layout_height="wrap_content"
+        android:layout_marginTop="10dp"
+        android:text="Play the output file after processing!" />
+
+    <Button
+        android:id="@+id/buttonProcess"
+        android:layout_width="wrap_content"
+        android:layout_height="wrap_content"
+        android:layout_gravity="center"
+        android:text="Process file!" />
+
+    <TextView
+        android:layout_width="wrap_content"
+        android:layout_height="wrap_content"
+        android:layout_marginTop="10dp"
+        android:text="Status console:"
+        android:textAppearance="?android:attr/textAppearanceMedium" />
+
+    <ScrollView
+        android:id="@+id/scrollView1"
+        android:layout_width="match_parent"
+        android:layout_height="wrap_content" >
+
+        <TextView
+        android:id="@+id/textViewResult"
+        android:layout_width="fill_parent"
+        android:layout_height="fill_parent"
+        android:text="@string/hello_world" />
+
+
+    </ScrollView>
+
+</LinearLayout>

source/Android-lib/res/values/strings.xml

@@ -0,0 +1,7 @@
+<?xml version="1.0" encoding="utf-8"?>
+<resources>
+
+    <string name="app_name">SoundTouch Example</string>
+    <string name="hello_world">Hello world!</string>
+
+</resources>

source/Android-lib/res/values/styles.xml

@@ -0,0 +1,20 @@
+<resources>
+
+    <!--
+        Base application theme, dependent on API level. This theme is replaced
+        by AppBaseTheme from res/values-vXX/styles.xml on newer devices.
+    -->
+    <style name="AppBaseTheme" parent="android:Theme.Holo.Light">
+        <!--
+            Theme customizations available in newer API levels can go in
+            res/values-vXX/styles.xml, while customizations related to
+            backward-compatibility can go here.
+        -->
+    </style>
+
+    <!-- Application theme. -->
+    <style name="AppTheme" parent="AppBaseTheme">
+        <!-- All customizations that are NOT specific to a particular API-level can go here. -->
+    </style>
+
+</resources>

source/Android-lib/src/net/surina/ExampleActivity.java

@@ -0,0 +1,219 @@
+/////////////////////////////////////////////////////////////////////////////
+///
+/// Example Android Application/Activity that allows processing WAV 
+/// audio files with SoundTouch library
+///
+/// Copyright (c) Olli Parviainen
+///
+////////////////////////////////////////////////////////////////////////////////
+
+package net.surina;
+
+import java.io.File;
+
+import net.surina.soundtouch.SoundTouch;
+import net.surina.soundtouchexample.R;
+import android.app.Activity;
+import android.content.Intent;
+import android.net.Uri;
+import android.os.AsyncTask;
+import android.os.Bundle;
+import android.util.Log;
+import android.view.View;
+import android.view.View.OnClickListener;
+import android.widget.Button;
+import android.widget.CheckBox;
+import android.widget.EditText;
+import android.widget.TextView;
+import android.widget.Toast;
+
+public class ExampleActivity extends Activity implements OnClickListener 
+{
+	TextView textViewConsole = null;
+	EditText editSourceFile = null;
+	EditText editOutputFile = null;
+	EditText editTempo = null;
+	EditText editPitch = null;
+	CheckBox checkBoxPlay = null;
+	
+	StringBuilder consoleText = new StringBuilder();
+
+	
+	/// Called when the activity is created
+	@Override
+	protected void onCreate(Bundle savedInstanceState) 
+	{
+		super.onCreate(savedInstanceState);
+		setContentView(R.layout.activity_example);
+		
+		textViewConsole = (TextView)findViewById(R.id.textViewResult);
+		editSourceFile = (EditText)findViewById(R.id.editTextSrcFileName);
+		editOutputFile = (EditText)findViewById(R.id.editTextOutFileName);
+
+		editTempo = (EditText)findViewById(R.id.editTextTempo);
+		editPitch = (EditText)findViewById(R.id.editTextPitch);
+		
+		Button buttonFileSrc = (Button)findViewById(R.id.buttonSelectSrcFile);
+		Button buttonFileOutput = (Button)findViewById(R.id.buttonSelectOutFile);
+		Button buttonProcess = (Button)findViewById(R.id.buttonProcess);
+		buttonFileSrc.setOnClickListener(this);
+		buttonFileOutput.setOnClickListener(this);
+		buttonProcess.setOnClickListener(this);
+
+		checkBoxPlay = (CheckBox)findViewById(R.id.checkBoxPlay);
+
+		// Check soundtouch library presence & version
+		checkLibVersion();
+	}
+	
+	
+		
+	/// Function to append status text onto "console box" on the Activity
+	public void appendToConsole(final String text)
+	{
+		// run on UI thread to avoid conflicts
+		runOnUiThread(new Runnable() 
+		{
+		    public void run() 
+		    {
+				consoleText.append(text);
+				consoleText.append("\n");
+				textViewConsole.setText(consoleText);
+		    }
+		});
+	}
+	
+
+	
+	/// print SoundTouch native library version onto console
+	protected void checkLibVersion()
+	{
+		String ver = SoundTouch.getVersionString();
+		appendToConsole("SoundTouch native library version = " + ver);
+	}
+
+
+
+	/// Button click handler
+	@Override
+	public void onClick(View arg0) 
+	{
+		switch (arg0.getId())
+		{
+			case R.id.buttonSelectSrcFile:
+			case R.id.buttonSelectOutFile:
+				// one of the file select buttons clicked ... we've not just implemented them ;-)
+				Toast.makeText(this, "File selector not implemented, sorry! Enter the file path manually ;-)", Toast.LENGTH_LONG).show();
+				break;
+				
+			case R.id.buttonProcess:
+				// button "process" pushed
+				process();
+				break;						
+		}
+		
+	}
+	
+	
+	/// Play audio file
+	protected void playWavFile(String fileName)
+	{
+		File file2play = new File(fileName);
+		Intent i = new Intent();
+		i.setAction(android.content.Intent.ACTION_VIEW);
+		i.setDataAndType(Uri.fromFile(file2play), "audio/wav");
+		startActivity(i);		
+	}
+	
+				
+
+	/// Helper class that will execute the SoundTouch processing. As the processing may take
+	/// some time, run it in background thread to avoid hanging of the UI.
+	protected class ProcessTask extends AsyncTask<ProcessTask.Parameters, Integer, Long>
+	{
+		/// Helper class to store the SoundTouch file processing parameters
+		public final class Parameters
+		{
+			String inFileName;
+			String outFileName;
+			float tempo;
+			float pitch;
+		}
+
+		
+		
+		/// Function that does the SoundTouch processing
+		public final long doSoundTouchProcessing(Parameters params) 
+		{
+			
+			SoundTouch st = new SoundTouch();
+			st.setTempo(params.tempo);
+			st.setPitchSemiTones(params.pitch);
+			Log.i("SoundTouch", "process file " + params.inFileName);
+			long startTime = System.currentTimeMillis();
+			int res = st.processFile(params.inFileName, params.outFileName);
+			long endTime = System.currentTimeMillis();
+			float duration = (endTime - startTime) * 0.001f;
+			
+			Log.i("SoundTouch", "process file done, duration = " + duration);
+			appendToConsole("Processing done, duration " + duration + " sec.");
+			if (res != 0)
+			{
+				String err = SoundTouch.getErrorString();
+				appendToConsole("Failure: " + err);
+				return -1L;
+			}
+			
+			// Play file if so is desirable
+			if (checkBoxPlay.isChecked())
+			{
+				playWavFile(params.outFileName);
+			}
+			return 0L;
+		}
+
+
+		
+		/// Overloaded function that get called by the system to perform the background processing
+		@Override	
+		protected Long doInBackground(Parameters... aparams) 
+		{
+			return doSoundTouchProcessing(aparams[0]);
+		}
+		
+	}
+
+
+	/// process a file with SoundTouch. Do the processing using a background processing
+	/// task to avoid hanging of the UI
+	protected void process()
+	{
+		try 
+		{
+			ProcessTask task = new ProcessTask();
+			ProcessTask.Parameters params = task.new Parameters();
+			// parse processing parameters
+			params.inFileName = editSourceFile.getText().toString();
+			params.outFileName = editOutputFile.getText().toString();
+			params.tempo = 0.01f * Float.parseFloat(editTempo.getText().toString());
+			params.pitch = Float.parseFloat(editPitch.getText().toString());
+
+			// update UI about status
+			appendToConsole("Process audio file :" + params.inFileName +" => " + params.outFileName);
+			appendToConsole("Tempo = " + params.tempo);
+			appendToConsole("Pitch adjust = " + params.pitch);
+			
+			Toast.makeText(this, "Starting to process file " + params.inFileName + "...", Toast.LENGTH_SHORT).show();
+
+			// start SoundTouch processing in a background thread
+			task.execute(params);
+//			task.doSoundTouchProcessing(params);	// this would run processing in main thread
+			
+		}
+		catch (Exception exp)
+		{
+			exp.printStackTrace();
+		}
+	
+	}
+}

source/Android-lib/src/net/surina/soundtouch/SoundTouch.java

@@ -0,0 +1,79 @@
+////////////////////////////////////////////////////////////////////////////////
+///
+/// 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
+///
+////////////////////////////////////////////////////////////////////////////////
+
+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 static String getVersionString();
+    
+    private native final void setTempo(long handle, float tempo);
+
+    private native final void setPitchSemiTones(long handle, float pitch);
+    
+    private native final void setSpeed(long handle, float speed);
+
+    private native final int processFile(long handle, String inputFile, String outputFile);
+
+    public native final static String getErrorString();
+
+    private native final static long newInstance();
+    
+    private native final void deleteInstance(long handle);
+    
+    long handle = 0;
+    
+    
+    public SoundTouch()
+    {
+    	handle = newInstance();    	
+    }
+    
+    
+    public void close()
+    {
+    	deleteInstance(handle);
+    	handle = 0;
+    }
+
+
+    public void setTempo(float tempo)
+    {
+    	setTempo(handle, tempo);
+    }
+
+
+    public void setPitchSemiTones(float pitch)
+    {
+    	setPitchSemiTones(handle, pitch);
+    }
+
+    
+    public void setSpeed(float speed)
+    {
+    	setSpeed(handle, speed);
+    }
+
+
+    public int processFile(String inputFile, String outputFile)
+    {
+    	return processFile(handle, inputFile, outputFile);
+    }
+
+    
+    // Load the native library upon startup
+    static
+    {
+        System.loadLibrary("soundtouch");
+    }
+}

source/Makefile.am

@@ -1,9 +1,5 @@
 ## Process this file with automake to create Makefile.in
 ##
-## $Id$
-##
-## 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
@@ -21,8 +17,9 @@
 
 include $(top_srcdir)/config/am_include.mk
 
-SUBDIRS=SoundTouch SoundStretch
+if SOUNDTOUCH_FLOAT_SAMPLES
-
+    # build SoundTouchDLL only if float samples used
-# set to something if you want other stuff to be included in the distribution tarball
+    SUBDIRS=SoundTouch SoundStretch SoundTouchDLL
-#EXTRA_DIST=
+else
-
+    SUBDIRS=SoundTouch SoundStretch
+endif

source/SoundStretch/Makefile.am

@@ -1,9 +1,5 @@
 ## Process this file with automake to create Makefile.in
 ##
-## $Id$
-##
-## 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
@@ -30,8 +26,8 @@ bin_PROGRAMS=soundstretch
 
 noinst_HEADERS=RunParameters.h WavFile.h
 
-# extra files to include in distrubution tarball
+# extra files to include in distribution tarball
-EXTRA_DIST=soundstretch.dsp soundstretch.dsw soundstretch.sln soundstretch.vcproj
+EXTRA_DIST=soundstretch.sln soundstretch.vcxproj
 
 ## 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.
@@ -44,11 +40,11 @@ soundstretch_SOURCES=main.cpp RunParameters.cpp WavFile.cpp
 soundstretch_LDADD=../SoundTouch/libSoundTouch.la -lm
 
 ## linker flags.
-# OP 2011-7-17 Linker flags disabled to prevent stripping symbols by default
+# Linker flag -s disabled to prevent stripping symbols by default
-# soundstretch_LDFLAGS=-s
+#soundstretch_LDFLAGS=-s
 
 ## additional compiler flags
-soundstretch_CXXFLAGS=-O3
+soundstretch_CXXFLAGS=$(AM_CXXFLAGS)
 
 #clean-local:
 #	-rm -f additional-files-to-remove-on-make-clean

source/SoundStretch/RunParameters.cpp

@@ -8,13 +8,6 @@
 ///
 ////////////////////////////////////////////////////////////////////////////////
 //
-// Last changed  : $Date$
-// File revision : $Revision: 4 $
-//
-// $Id$
-//
-////////////////////////////////////////////////////////////////////////////////
-//
 // License :
 //
 //  SoundTouch audio processing library
@@ -37,12 +30,15 @@
 ////////////////////////////////////////////////////////////////////////////////
 
 #include <string>
-#include <stdlib.h>
+#include <cstdlib>
 
 #include "RunParameters.h"
 
 using namespace std;
 
+namespace soundstretch
+{
+
 // Program usage instructions
 
 static const char licenseText[] =
@@ -101,9 +97,8 @@ static int _toLowerCase(int c)
     return c;
 }
 
-
 // Constructor
-RunParameters::RunParameters(const int nParams, const char * const paramStr[])
+RunParameters::RunParameters(int nParams, const CHARTYPE* paramStr[])
 {
     int i;
     int nFirstParam;
@@ -119,28 +114,17 @@ RunParameters::RunParameters(const int nParams, const char * const paramStr[])
         }
         string msg = whatText;
         msg += usage;
-        ST_THROW_RT_ERROR(msg.c_str());
+        throw(msg);
     }
 
-    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];
+    inFileName = paramStr[1];
-    outFileName = (char*)paramStr[2];
+    outFileName = paramStr[2];
 
     if (outFileName[0] == '-')
     {
-        // no outputfile name was given but parameters
+        // outputfile name was omitted but other parameter switches given instead
-        outFileName = NULL;
+        outFileName = STRING_CONST("");
         nFirstParam = 2;
     }
     else
@@ -158,7 +142,6 @@ RunParameters::RunParameters(const int nParams, const char * const paramStr[])
 }
 
 
-
 // Checks parameter limits
 void RunParameters::checkLimits()
 {
@@ -190,27 +173,33 @@ void RunParameters::checkLimits()
     }
 }
 
-
+// Convert STRING to std::string. Actually needed only if STRING is std::wstring, but conversion penalty is negligible
-
+std::string convertString(const STRING& str)
-// Unknown switch parameter -- throws an exception with an error message
-void RunParameters::throwIllegalParamExp(const string &str) const
 {
-    string msg = "ERROR : Illegal parameter \"";
+    std::string res;
-    msg += str;
+    for (auto c : str)
-    msg += "\".\n\n";
+    {
-    msg += usage;
+        res += (char)c;
-    ST_THROW_RT_ERROR(msg.c_str());
+    }
+    return res;
 }
 
-
+// Unknown switch parameter -- throws an exception with an error message
+void RunParameters::throwIllegalParamExp(const STRING &str) const
+{
+    string msg = "ERROR : Illegal parameter \"";
+    msg += convertString(str);
+    msg += "\".\n\n";
+    msg += usage;
+    ST_THROW_RT_ERROR(msg);
+}
 
 void RunParameters::throwLicense() const
 {
     ST_THROW_RT_ERROR(licenseText);
 }
 
-
+double RunParameters::parseSwitchValue(const STRING& str) const
-float RunParameters::parseSwitchValue(const string &str) const
 {
     int pos;
 
@@ -222,14 +211,14 @@ float RunParameters::parseSwitchValue(const string &str) const
     }
 
     // Read numerical parameter value after '='
-    return (float)atof(str.substr(pos + 1).c_str());
+    return stof(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)
+void RunParameters::parseSwitchParam(const STRING& str)
 {
     int upS;
 
@@ -262,12 +251,12 @@ void RunParameters::parseSwitchParam(const string &str)
 
         case 'b' :
             // switch '-bpm=xx'
-            detectBPM = TRUE;
+            detectBPM = true;
             try
             {
                 goalBPM = parseSwitchValue(str);
             }
-            catch (const runtime_error)
+            catch (const runtime_error &)
             {
                 // illegal or missing bpm value => just calculate bpm
                 goalBPM = 0;
@@ -291,7 +280,7 @@ void RunParameters::parseSwitchParam(const string &str)
 
         case 's' :
             // switch '-speech'
-            speech = TRUE;
+            speech = true;
             break;
 
         default:
@@ -299,3 +288,5 @@ void RunParameters::parseSwitchParam(const string &str)
             throwIllegalParamExp(str);
     }
 }
+
+}

source/SoundStretch/RunParameters.h

@@ -8,13 +8,6 @@
 ///
 ////////////////////////////////////////////////////////////////////////////////
 //
-// Last changed  : $Date$
-// File revision : $Revision: 4 $
-//
-// $Id$
-//
-////////////////////////////////////////////////////////////////////////////////
-//
 // License :
 //
 //  SoundTouch audio processing library
@@ -39,34 +32,39 @@
 #ifndef RUNPARAMETERS_H
 #define RUNPARAMETERS_H
 
-#include "STTypes.h"
 #include <string>
+#include "STTypes.h"
+#include "SS_CharTypes.h"
+#include "WavFile.h"
 
-using namespace std;
+namespace soundstretch
+{
 
 /// Parses command line parameters into program parameters
 class RunParameters
 {
 private:
-    void throwIllegalParamExp(const string &str) const;
+    void throwIllegalParamExp(const STRING& str) const;
     void throwLicense() const;
-    void parseSwitchParam(const string &str);
+    void parseSwitchParam(const STRING& str);
     void checkLimits();
-    float parseSwitchValue(const string &str) const;
+    double parseSwitchValue(const STRING& tr) const;
 
 public:
-    char  *inFileName;
+    STRING inFileName;
-    char  *outFileName;
+    STRING outFileName;
-    float tempoDelta;
+    double tempoDelta{ 0 };
-    float pitchDelta;
+    double pitchDelta{ 0 };
-    float rateDelta;
+    double rateDelta{ 0 };
-    int   quick;
+    int   quick{ 0 };
-    int   noAntiAlias;
+    int   noAntiAlias{ 0 };
-    float goalBPM;
+    double goalBPM{ 0 };
-    BOOL  detectBPM;
+    bool  detectBPM{ false };
-    BOOL  speech;
+    bool  speech{ false };
 
-    RunParameters(const int nParams, const char * const paramStr[]);
+    RunParameters(int nParams, const CHARTYPE* paramStr[]);
 };
 
+}
+
 #endif

source/SoundStretch/SS_CharTypes.h

@@ -0,0 +1,52 @@
+////////////////////////////////////////////////////////////////////////////////
+///
+/// Char type for SoundStretch
+///
+/// Author        : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+//  SoundTouch audio processing library
+//  Copyright (c) Olli Parviainen
+//
+//  This library is free software; you can redistribute it and/or
+//  modify it under the terms of the GNU Lesser General Public
+//  License as published by the Free Software Foundation; either
+//  version 2.1 of the License, or (at your option) any later version.
+//
+//  This library is distributed in the hope that it will be useful,
+//  but WITHOUT ANY WARRANTY; without even the implied warranty of
+//  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+//  Lesser General Public License for more details.
+//
+//  You should have received a copy of the GNU Lesser General Public
+//  License along with this library; if not, write to the Free Software
+//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#ifndef SS_CHARTYPE_H
+#define SS_CHARTYPE_H
+
+#include <string>
+
+namespace soundstretch
+{
+#if _WIN32
+    // wide-char types for supporting non-latin file paths in Windows
+    using CHARTYPE = wchar_t;
+    using STRING = std::wstring;
+    #define STRING_CONST(x) (L"" x)
+#else
+    // gnu platform can natively support UTF-8 paths using "char*" set
+    using CHARTYPE = char;
+    using STRING = std::string;
+    #define STRING_CONST(x) (x)
+#endif
+}
+
+#endif //SS_CHARTYPE_H

source/SoundStretch/WavFile.cpp

@@ -17,13 +17,6 @@
 ///
 ////////////////////////////////////////////////////////////////////////////////
 //
-// Last changed  : $Date$
-// File revision : $Revision: 4 $
-//
-// $Id$
-//
-////////////////////////////////////////////////////////////////////////////////
-//
 // License :
 //
 //  SoundTouch audio processing library
@@ -49,20 +42,29 @@
 #include <string>
 #include <sstream>
 #include <cstring>
-#include <assert.h>
+#include <cassert>
-#include <limits.h>
+#include <climits>
 
 #include "WavFile.h"
 #include "STTypes.h"
 
 using namespace std;
 
+namespace soundstretch
+{
+
+#if _WIN32
+#define FOPEN(name, mode) _wfopen(name, STRING_CONST(mode))
+#else
+#define FOPEN(name, mode) fopen(name, mode)
+#endif
+
 static const char riffStr[] = "RIFF";
 static const char waveStr[] = "WAVE";
 static const char fmtStr[] = "fmt ";
+static const char factStr[] = "fact";
 static const char dataStr[] = "data";
 
-
 //////////////////////////////////////////////////////////////////////////////
 //
 // Helper functions for swapping byte order to correctly read/write WAV files
@@ -73,67 +75,67 @@ static const char dataStr[] = "data";
 // 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
+// In gcc compiler detect the byte order automatically
-    #if BYTE_ORDER == BIG_ENDIAN
+#if BYTE_ORDER == BIG_ENDIAN
-        // big-endian platform.
+// big-endian platform.
-        #define _BIG_ENDIAN_
+#define _BIG_ENDIAN_
-    #endif
+#endif
 #endif
 
 #ifdef _BIG_ENDIAN_
-    // big-endian CPU, swap bytes in 16 & 32 bit words
+// big-endian CPU, swap bytes in 16 & 32 bit words
 
-    // helper-function to swap byte-order of 32bit integer
+// helper-function to swap byte-order of 32bit integer
-    static inline int _swap32(int &dwData)
+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
+// helper-function to swap byte-order of 16bit integer
-    static inline short _swap16(short &wData)
+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
+// helper-function to swap byte-order of buffer of 16bit integers
-    static inline void _swap16Buffer(short *pData, int numWords)
+static inline void _swap16Buffer(short* pData, int numWords)
-    {
+{
     int i;
 
-        for (i = 0; i < numWords; i ++)
+    for (i = 0; i < numWords; i++)
     {
         pData[i] = _swap16(pData[i]);
     }
-    }
+}
 
 #else   // BIG_ENDIAN
-    // little-endian CPU, WAV file is ok as such
+// little-endian CPU, WAV file is ok as such
 
-    // dummy helper-function
+// dummy helper-function
-    static inline int _swap32(int &dwData)
+static inline int _swap32(int& dwData)
-    {
+{
     // do nothing
     return dwData;
-    }   
+}
 
-    // dummy helper-function
+// dummy helper-function
-    static inline short _swap16(short &wData)
+static inline short _swap16(short& wData)
-    {
+{
     // do nothing
     return wData;
-    }
+}
 
-    // dummy helper-function
+// dummy helper-function
-    static inline void _swap16Buffer(short *pData, int numBytes)
+static inline void _swap16Buffer(short*, int)
-    {
+{
     // do nothing
-    }
+}
 
 #endif  // BIG_ENDIAN
 
@@ -145,7 +147,7 @@ static const char dataStr[] = "data";
 
 WavFileBase::WavFileBase()
 {
-    convBuff = NULL;
+    convBuff = nullptr;
     convBuffSize = 0;
 }
 
@@ -158,7 +160,7 @@ WavFileBase::~WavFileBase()
 
 
 /// Get pointer to conversion buffer of at min. given size
-void *WavFileBase::getConvBuffer(int sizeBytes)
+void* WavFileBase::getConvBuffer(int sizeBytes)
 {
     if (convBuffSize < sizeBytes)
     {
@@ -176,32 +178,26 @@ void *WavFileBase::getConvBuffer(int sizeBytes)
 // Class WavInFile
 //
 
-WavInFile::WavInFile(const char *fileName)
+WavInFile::WavInFile(const STRING& fileName)
 {
     // Try to open the file for reading
-    fptr = fopen(fileName, "rb");
+    fptr = FOPEN(fileName.c_str(), "rb");
-    if (fptr == NULL) 
+    if (fptr == nullptr)
     {
-        // didn't succeed
+        ST_THROW_RT_ERROR("Error : Unable to open file for reading.");
-        string msg = "Error : Unable to open file \"";
-        msg += fileName;
-        msg += "\" for reading.";
-        ST_THROW_RT_ERROR(msg.c_str());
     }
 
     init();
 }
 
 
-WavInFile::WavInFile(FILE *file)
+WavInFile::WavInFile(FILE* file)
 {
     // Try to open the file for reading
     fptr = file;
     if (!file)
     {
-        // didn't succeed
+        ST_THROW_RT_ERROR("Error : Unable to access input stream for reading");
-        string msg = "Error : Unable to access input stream for reading";
-        ST_THROW_RT_ERROR(msg.c_str());
     }
 
     init();
@@ -220,32 +216,29 @@ void WavInFile::init()
     hdrsOk = readWavHeaders();
     if (hdrsOk != 0)
     {
-        // Something didn't match in the wav file headers 
+        ST_THROW_RT_ERROR("Input file is corrupt or not a WAV file");
-        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.
+    // sanity check for format parameters
-    if (header.format.fixed != 1)
+    if ((header.format.channel_number < 1) || (header.format.channel_number > 9) ||
+        (header.format.sample_rate < 4000) || (header.format.sample_rate > 192000) ||
+        (header.format.byte_per_sample < 1) || (header.format.byte_per_sample > 320) ||
+        (header.format.bits_per_sample < 8) || (header.format.bits_per_sample > 32))
     {
-        string msg = "Input file uses unsupported encoding.";
+        ST_THROW_RT_ERROR("Error: Illegal wav file header format parameters.");
-        ST_THROW_RT_ERROR(msg.c_str());
     }
-    */
 
     dataRead = 0;
 }
 
 
-
 WavInFile::~WavInFile()
 {
     if (fptr) fclose(fptr);
-    fptr = NULL;
+    fptr = nullptr;
 }
 
 
-
 void WavInFile::rewind()
 {
     int hdrsOk;
@@ -268,7 +261,7 @@ int WavInFile::checkCharTags() const
 }
 
 
-int WavInFile::read(unsigned char *buffer, int maxElems)
+int WavInFile::read(unsigned char* buffer, int maxElems)
 {
     int numBytes;
     uint afterDataRead;
@@ -297,7 +290,7 @@ int WavInFile::read(unsigned char *buffer, int maxElems)
 }
 
 
-int WavInFile::read(short *buffer, int maxElems)
+int WavInFile::read(short* buffer, int maxElems)
 {
     unsigned int afterDataRead;
     int numBytes;
@@ -309,12 +302,12 @@ int WavInFile::read(short *buffer, int maxElems)
     case 8:
     {
         // 8 bit format
-            unsigned char *temp = (unsigned char*)getConvBuffer(maxElems);
+        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 ++)
+        for (i = 0; i < numElems; i++)
         {
             buffer[i] = (short)(((short)temp[i] - 128) * 256);
         }
@@ -341,7 +334,7 @@ int WavInFile::read(short *buffer, int maxElems)
         numElems = numBytes / 2;
 
         // 16bit samples, swap byte order if necessary
-            _swap16Buffer((short *)buffer, numElems);
+        _swap16Buffer((short*)buffer, numElems);
         break;
     }
 
@@ -361,7 +354,7 @@ int WavInFile::read(short *buffer, int maxElems)
 
 /// 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)
+int WavInFile::read(float* buffer, int maxElems)
 {
     unsigned int afterDataRead;
     int numBytes;
@@ -390,7 +383,7 @@ int WavInFile::read(float *buffer, int maxElems)
     }
 
     // read raw data into temporary buffer
-    char *temp = (char*)getConvBuffer(numBytes);
+    char* temp = (char*)getConvBuffer(numBytes);
     numBytes = (int)fread(temp, 1, numBytes, fptr);
     dataRead += numBytes;
 
@@ -401,9 +394,9 @@ int WavInFile::read(float *buffer, int maxElems)
     {
     case 1:
     {
-            unsigned char *temp2 = (unsigned char*)temp;
+        unsigned char* temp2 = (unsigned char*)temp;
         double conv = 1.0 / 128.0;
-            for (int i = 0; i < numElems; i ++)
+        for (int i = 0; i < numElems; i++)
         {
             buffer[i] = (float)(temp2[i] * conv - 1.0);
         }
@@ -412,9 +405,9 @@ int WavInFile::read(float *buffer, int maxElems)
 
     case 2:
     {
-            short *temp2 = (short*)temp;
+        short* temp2 = (short*)temp;
         double conv = 1.0 / 32768.0;
-            for (int i = 0; i < numElems; i ++)
+        for (int i = 0; i < numElems; i++)
         {
             short value = temp2[i];
             buffer[i] = (float)(_swap16(value) * conv);
@@ -424,9 +417,9 @@ int WavInFile::read(float *buffer, int maxElems)
 
     case 3:
     {
-            char *temp2 = (char *)temp;
+        char* temp2 = (char*)temp;
         double conv = 1.0 / 8388608.0;
-            for (int i = 0; i < numElems; i ++)
+        for (int i = 0; i < numElems; i++)
         {
             int value = *((int*)temp2);
             value = _swap32(value) & 0x00ffffff;             // take 24 bits
@@ -439,10 +432,10 @@ int WavInFile::read(float *buffer, int maxElems)
 
     case 4:
     {
-            int *temp2 = (int *)temp;
+        int* temp2 = (int*)temp;
         double conv = 1.0 / 2147483648.0;
         assert(sizeof(int) == 4);
-            for (int i = 0; i < numElems; i ++)
+        for (int i = 0; i < numElems; i++)
         {
             int value = temp2[i];
             buffer[i] = (float)(_swap32(value) * conv);
@@ -458,11 +451,10 @@ int WavInFile::read(float *buffer, int maxElems)
 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));
+    return ((uint)dataRead == header.data.data_len || feof(fptr));
 }
 
 
-
 // test if character code is between a white space ' ' and little 'z'
 static int isAlpha(char c)
 {
@@ -471,7 +463,7 @@ static int isAlpha(char c)
 
 
 // test if all characters are between a white space ' ' and little 'z'
-static int isAlphaStr(const char *str)
+static int isAlphaStr(const char* str)
 {
     char c;
 
@@ -479,7 +471,7 @@ static int isAlphaStr(const char *str)
     while (c)
     {
         if (isAlpha(c) == 0) return 0;
-        str ++;
+        str++;
         c = str[0];
     }
 
@@ -492,7 +484,7 @@ 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);
+    _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;
@@ -503,15 +495,13 @@ int WavInFile::readRIFFBlock()
 }
 
 
-
-
 int WavInFile::readHeaderBlock()
 {
     char label[5];
     string sLabel;
 
     // lead label string
-    if (fread(label, 1, 4, fptr) !=4) return -1;
+    if (fread(label, 1, 4, fptr) != 4) return -1;
     label[4] = 0;
 
     if (isAlphaStr(label) == 0) return -1;    // not a valid label
@@ -527,12 +517,16 @@ int WavInFile::readHeaderBlock()
         // 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;
+        _swap32(nLen);
-        header.format.format_len = nLen;
 
         // calculate how much length differs from expected
         nDump = nLen - ((int)sizeof(header.format) - 8);
 
+        // verify that header length isn't smaller than expected structure
+        if ((nLen < 0) || (nDump < 0)) return -1;
+
+        header.format.format_len = nLen;
+
         // if format_len is larger than expected, read only as much data as we've space for
         if (nDump > 0)
         {
@@ -543,12 +537,12 @@ int WavInFile::readHeaderBlock()
         if (fread(&(header.format.fixed), nLen, 1, fptr) != 1) return -1;
 
         // swap byte order if necessary
-        _swap16(header.format.fixed);            // short int fixed;
+        _swap16((short&)header.format.fixed);            // short int fixed;
-        _swap16(header.format.channel_number);   // short int channel_number;
+        _swap16((short&)header.format.channel_number);   // short int channel_number;
-        _swap32((int &)header.format.sample_rate);      // int sample_rate;
+        _swap32((int&)header.format.sample_rate);        // int sample_rate;
-        _swap32((int &)header.format.byte_rate);        // int byte_rate;
+        _swap32((int&)header.format.byte_rate);          // int byte_rate;
-        _swap16(header.format.byte_per_sample);  // short int byte_per_sample;
+        _swap16((short&)header.format.byte_per_sample);  // short int byte_per_sample;
-        _swap16(header.format.bits_per_sample);  // short int bits_per_sample;
+        _swap16((short&)header.format.bits_per_sample);  // short int bits_per_sample;
 
         // if format_len is larger than expected, skip the extra data
         if (nDump > 0)
@@ -558,6 +552,46 @@ int WavInFile::readHeaderBlock()
 
         return 0;
     }
+    else if (strcmp(label, factStr) == 0)
+    {
+        int nLen, nDump;
+
+        // 'fact' block
+        memcpy(header.fact.fact_field, factStr, 4);
+
+        // read length of the fact field
+        if (fread(&nLen, sizeof(int), 1, fptr) != 1) return -1;
+        // swap byte order if necessary
+        _swap32(nLen);
+
+        // calculate how much length differs from expected
+        nDump = nLen - ((int)sizeof(header.fact) - 8);
+
+        // verify that fact length isn't smaller than expected structure
+        if ((nLen < 0) || (nDump < 0)) return -1;
+
+        header.fact.fact_len = nLen;
+
+        // if format_len is larger than expected, read only as much data as we've space for
+        if (nDump > 0)
+        {
+            nLen = sizeof(header.fact) - 8;
+        }
+
+        // read data
+        if (fread(&(header.fact.fact_sample_len), nLen, 1, fptr) != 1) return -1;
+
+        // swap byte order if necessary
+        _swap32((int&)header.fact.fact_sample_len);    // int sample_length;
+
+        // if fact_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
@@ -565,7 +599,7 @@ int WavInFile::readHeaderBlock()
         if (fread(&(header.data.data_len), sizeof(uint), 1, fptr) != 1) return -1;
 
         // swap byte order if necessary
-        _swap32((int &)header.data.data_len);
+        _swap32((int&)header.data.data_len);
 
         return 1;
     }
@@ -578,7 +612,7 @@ int WavInFile::readHeaderBlock()
         // read length
         if (fread(&len, sizeof(len), 1, fptr) != 1) return -1;
         // scan through the block
-        for (i = 0; i < len; i ++)
+        for (i = 0; i < len; i++)
         {
             if (fread(&temp, 1, 1, fptr) != 1) return -1;
             if (feof(fptr)) return -1;   // unexpected eof
@@ -632,7 +666,6 @@ uint WavInFile::getSampleRate() const
 }
 
 
-
 uint WavInFile::getDataSizeInBytes() const
 {
     return header.data.data_len;
@@ -642,6 +675,7 @@ uint WavInFile::getDataSizeInBytes() const
 uint WavInFile::getNumSamples() const
 {
     if (header.format.byte_per_sample == 0) return 0;
+    if (header.format.fixed > 1) return header.fact.fact_sample_len;
     return header.data.data_len / (unsigned short)header.format.byte_per_sample;
 }
 
@@ -665,23 +699,18 @@ uint WavInFile::getElapsedMS() const
 }
 
 
-
 //////////////////////////////////////////////////////////////////////////////
 //
 // Class WavOutFile
 //
 
-WavOutFile::WavOutFile(const char *fileName, int sampleRate, int bits, int channels)
+WavOutFile::WavOutFile(const STRING& fileName, int sampleRate, int bits, int channels)
 {
     bytesWritten = 0;
-    fptr = fopen(fileName, "wb");
+    fptr = FOPEN(fileName.c_str(), "wb");
-    if (fptr == NULL) 
+    if (fptr == nullptr)
     {
-        string msg = "Error : Unable to open file \"";
+        ST_THROW_RT_ERROR("Error : Unable to open file for writing.");
-        msg += fileName;
-        msg += "\" for writing.";
-        //pmsg = msg.c_str;
-        ST_THROW_RT_ERROR(msg.c_str());
     }
 
     fillInHeader(sampleRate, bits, channels);
@@ -689,14 +718,13 @@ WavOutFile::WavOutFile(const char *fileName, int sampleRate, int bits, int chann
 }
 
 
-WavOutFile::WavOutFile(FILE *file, int sampleRate, int bits, int channels)
+WavOutFile::WavOutFile(FILE* file, int sampleRate, int bits, int channels)
 {
     bytesWritten = 0;
     fptr = file;
-    if (fptr == NULL) 
+    if (fptr == nullptr)
     {
-        string msg = "Error : Unable to access output file stream.";
+        ST_THROW_RT_ERROR("Error : Unable to access output file stream.");
-        ST_THROW_RT_ERROR(msg.c_str());
     }
 
     fillInHeader(sampleRate, bits, channels);
@@ -704,16 +732,14 @@ WavOutFile::WavOutFile(FILE *file, int sampleRate, int bits, int channels)
 }
 
 
-
 WavOutFile::~WavOutFile()
 {
     finishHeader();
     if (fptr) fclose(fptr);
-    fptr = NULL;
+    fptr = nullptr;
 }
 
 
-
 void WavOutFile::fillInHeader(uint sampleRate, uint bits, uint channels)
 {
     // fill in the 'riff' part..
@@ -739,6 +765,11 @@ void WavOutFile::fillInHeader(uint sampleRate, uint bits, uint channels)
     header.format.byte_rate = header.format.byte_per_sample * (int)sampleRate;
     header.format.sample_rate = (int)sampleRate;
 
+    // fill in the 'fact' part...
+    memcpy(&(header.fact.fact_field), factStr, 4);
+    header.fact.fact_len = 4;
+    header.fact.fact_sample_len = 0;
+
     // fill in the 'data' part..
 
     // copy string 'data' to data_field
@@ -751,14 +782,14 @@ void WavOutFile::fillInHeader(uint sampleRate, uint bits, uint channels)
 void WavOutFile::finishHeader()
 {
     // supplement the file length into the header structure
-    header.riff.package_len = bytesWritten + 36;
+    header.riff.package_len = bytesWritten + sizeof(WavHeader) - sizeof(WavRiff) + 4;
     header.data.data_len = bytesWritten;
+    header.fact.fact_sample_len = bytesWritten / header.format.byte_per_sample;
 
     writeHeader();
 }
 
 
-
 void WavOutFile::writeHeader()
 {
     WavHeader hdrTemp;
@@ -766,15 +797,17 @@ void WavOutFile::writeHeader()
 
     // swap byte order if necessary
     hdrTemp = header;
-    _swap32((int &)hdrTemp.riff.package_len);
+    _swap32((int&)hdrTemp.riff.package_len);
-    _swap32((int &)hdrTemp.format.format_len);
+    _swap32((int&)hdrTemp.format.format_len);
-    _swap16((short &)hdrTemp.format.fixed);
+    _swap16((short&)hdrTemp.format.fixed);
-    _swap16((short &)hdrTemp.format.channel_number);
+    _swap16((short&)hdrTemp.format.channel_number);
-    _swap32((int &)hdrTemp.format.sample_rate);
+    _swap32((int&)hdrTemp.format.sample_rate);
-    _swap32((int &)hdrTemp.format.byte_rate);
+    _swap32((int&)hdrTemp.format.byte_rate);
-    _swap16((short &)hdrTemp.format.byte_per_sample);
+    _swap16((short&)hdrTemp.format.byte_per_sample);
-    _swap16((short &)hdrTemp.format.bits_per_sample);
+    _swap16((short&)hdrTemp.format.bits_per_sample);
-    _swap32((int &)hdrTemp.data.data_len);
+    _swap32((int&)hdrTemp.data.data_len);
+    _swap32((int&)hdrTemp.fact.fact_len);
+    _swap32((int&)hdrTemp.fact.fact_sample_len);
 
     // write the supplemented header in the beginning of the file
     fseek(fptr, 0, SEEK_SET);
@@ -789,8 +822,7 @@ void WavOutFile::writeHeader()
 }
 
 
-
+void WavOutFile::write(const unsigned char* buffer, int numElems)
-void WavOutFile::write(const unsigned char *buffer, int numElems)
 {
     int res;
 
@@ -810,8 +842,7 @@ void WavOutFile::write(const unsigned char *buffer, int numElems)
 }
 
 
-
+void WavOutFile::write(const short* buffer, int numElems)
-void WavOutFile::write(const short *buffer, int numElems)
 {
     int res;
 
@@ -823,9 +854,9 @@ void WavOutFile::write(const short *buffer, int numElems)
     case 8:
     {
         int i;
-            unsigned char *temp = (unsigned char *)getConvBuffer(numElems);
+        unsigned char* temp = (unsigned char*)getConvBuffer(numElems);
         // convert from 16bit format to 8bit format
-            for (i = 0; i < numElems; i ++)
+        for (i = 0; i < numElems; i++)
         {
             temp[i] = (unsigned char)(buffer[i] / 256 + 128);
         }
@@ -839,8 +870,8 @@ void WavOutFile::write(const short *buffer, int numElems)
         // 16bit format
 
         // use temp buffer to swap byte order if necessary
-            short *pTemp = (short *)getConvBuffer(numElems * sizeof(short));
+        short* pTemp = (short*)getConvBuffer(numElems * sizeof(short));
-            memcpy(pTemp, buffer, numElems * 2);
+        memcpy(pTemp, buffer, (size_t)numElems * 2L);
         _swap16Buffer(pTemp, numElems);
 
         res = (int)fwrite(pTemp, 2, numElems, fptr);
@@ -880,7 +911,7 @@ inline int saturate(float fvalue, float minval, float maxval)
 }
 
 
-void WavOutFile::write(const float *buffer, int numElems)
+void WavOutFile::write(const float* buffer, int numElems)
 {
     int numBytes;
     int bytesPerSample;
@@ -889,14 +920,14 @@ void WavOutFile::write(const float *buffer, int numElems)
 
     bytesPerSample = header.format.bits_per_sample / 8;
     numBytes = numElems * bytesPerSample;
-    short *temp = (short*)getConvBuffer(numBytes);
+    void* temp = getConvBuffer(numBytes + 7);   // round bit up to avoid buffer overrun with 24bit-value assignment
 
     switch (bytesPerSample)
     {
     case 1:
     {
-            unsigned char *temp2 = (unsigned char *)temp;
+        unsigned char* temp2 = (unsigned char*)temp;
-            for (int i = 0; i < numElems; i ++)
+        for (int i = 0; i < numElems; i++)
         {
             temp2[i] = (unsigned char)saturate(buffer[i] * 128.0f + 128.0f, 0.0f, 255.0f);
         }
@@ -905,8 +936,8 @@ void WavOutFile::write(const float *buffer, int numElems)
 
     case 2:
     {
-            short *temp2 = (short *)temp;
+        short* temp2 = (short*)temp;
-            for (int i = 0; i < numElems; i ++)
+        for (int i = 0; i < numElems; i++)
         {
             short value = (short)saturate(buffer[i] * 32768.0f, -32768.0f, 32767.0f);
             temp2[i] = _swap16(value);
@@ -916,8 +947,8 @@ void WavOutFile::write(const float *buffer, int numElems)
 
     case 3:
     {
-            char *temp2 = (char *)temp;
+        char* temp2 = (char*)temp;
-            for (int i = 0; i < numElems; i ++)
+        for (int i = 0; i < numElems; i++)
         {
             int value = saturate(buffer[i] * 8388608.0f, -8388608.0f, 8388607.0f);
             *((int*)temp2) = _swap32(value);
@@ -928,8 +959,8 @@ void WavOutFile::write(const float *buffer, int numElems)
 
     case 4:
     {
-            int *temp2 = (int *)temp;
+        int* temp2 = (int*)temp;
-            for (int i = 0; i < numElems; i ++)
+        for (int i = 0; i < numElems; i++)
         {
             int value = saturate(buffer[i] * 2147483648.0f, -2147483648.0f, 2147483647.0f);
             temp2[i] = _swap32(value);
@@ -949,3 +980,5 @@ void WavOutFile::write(const float *buffer, int numElems)
     }
     bytesWritten += numBytes;
 }
+
+}

source/SoundStretch/WavFile.h

@@ -16,13 +16,6 @@
 ///
 ////////////////////////////////////////////////////////////////////////////////
 //
-// Last changed  : $Date$
-// File revision : $Revision: 4 $
-//
-// $Id$
-//
-////////////////////////////////////////////////////////////////////////////////
-//
 // License :
 //
 //  SoundTouch audio processing library
@@ -47,7 +40,12 @@
 #ifndef WAVFILE_H
 #define WAVFILE_H
 
-#include <stdio.h>
+#include <cstdio>
+#include <string>
+#include "SS_CharTypes.h"
+
+namespace soundstretch
+{
 
 #ifndef uint
 typedef unsigned int uint;
@@ -58,7 +56,7 @@ typedef unsigned int uint;
 typedef struct
 {
     char riff_char[4];
-    int  package_len;
+    uint package_len;
     char wave[4];
 } WavRiff;
 
@@ -66,15 +64,23 @@ typedef struct
 typedef struct
 {
     char  fmt[4];
-    int   format_len;
+    unsigned int   format_len;
-    short fixed;
+    unsigned short fixed;
-    short channel_number;
+    unsigned short channel_number;
-    int   sample_rate;
+    unsigned int   sample_rate;
-    int   byte_rate;
+    unsigned int   byte_rate;
-    short byte_per_sample;
+    unsigned short byte_per_sample;
-    short bits_per_sample;
+    unsigned short bits_per_sample;
 } WavFormat;
 
+/// WAV audio file 'fact' section header
+typedef struct
+{
+    char fact_field[4];
+    uint fact_len;
+    uint fact_sample_len;
+} WavFact;
+
 /// WAV audio file 'data' section header
 typedef struct
 {
@@ -88,6 +94,7 @@ typedef struct
 {
     WavRiff   riff;
     WavFormat format;
+    WavFact   fact;
     WavData   data;
 } WavHeader;
 
@@ -116,9 +123,6 @@ 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;
 
@@ -146,7 +150,7 @@ private:
 public:
     /// Constructor: Opens the given WAV file. If the file can't be opened,
     /// throws 'runtime_error' exception.
-    WavInFile(const char *filename);
+    WavInFile(const STRING& filename);
 
     WavInFile(FILE *file);
 
@@ -216,7 +220,6 @@ public:
 };
 
 
-
 /// Class for writing WAV audio files.
 class WavOutFile : protected WavFileBase
 {
@@ -243,7 +246,7 @@ private:
 public:
     /// Constructor: Creates a new WAV file. Throws a 'runtime_error' exception
     /// if file creation fails.
-    WavOutFile(const char *fileName,    ///< Filename
+    WavOutFile(const STRING& 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)
@@ -273,4 +276,6 @@ public:
                );
 };
 
+ }
+
 #endif

source/SoundStretch/main.cpp

@@ -8,13 +8,6 @@
 ///
 ////////////////////////////////////////////////////////////////////////////////
 //
-// Last changed  : $Date$
-// File revision : $Revision: 4 $
-//
-// $Id$
-//
-////////////////////////////////////////////////////////////////////////////////
-//
 // License :
 //
 //  SoundTouch audio processing library
@@ -36,10 +29,12 @@
 //
 ////////////////////////////////////////////////////////////////////////////////
 
+#include <iostream>
+#include <memory>
 #include <stdexcept>
-#include <stdio.h>
+#include <string>
-#include <string.h>
+#include <cstdio>
-#include <time.h>
+#include <ctime>
 #include "RunParameters.h"
 #include "WavFile.h"
 #include "SoundTouch.h"
@@ -48,25 +43,28 @@
 using namespace soundtouch;
 using namespace std;
 
-// Processing chunk size
+namespace soundstretch
-#define BUFF_SIZE           2048
+{
+
+// Processing chunk size (size chosen to be divisible by 2, 4, 6, 8, 10, 12, 14, 16 channels ...)
+#define BUFF_SIZE           6720
 
 #if _WIN32
-    #include <io.h>
+#include <io.h>
-    #include <fcntl.h>
+#include <fcntl.h>
 
-    // Macro for Win32 standard input/output stream support: Sets a file stream into binary mode
+// 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))
+#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) {}
+#define SET_STREAM_TO_BIN_MODE(f) {}
 #endif
 
 
 static const char _helloText[] =
     "\n"
-    "   SoundStretch v%s -  Written by Olli Parviainen 2001 - 2012\n"
+    "   SoundStretch v%s -  Copyright (c) Olli Parviainen\n"
-    "==================================================================\n"
+    "=========================================================\n"
     "author e-mail: <oparviai"
     "@"
     "iki.fi> - WWW: http://www.surina.net/soundtouch\n"
@@ -75,91 +73,81 @@ static const char _helloText[] =
     "more information.\n"
     "\n";
 
-static void openFiles(WavInFile **inFile, WavOutFile **outFile, const RunParameters *params)
+static void openFiles(unique_ptr<WavInFile>& inFile, unique_ptr<WavOutFile>& outFile, const RunParameters& params)
 {
-    int bits, samplerate, channels;
+    if (params.inFileName == STRING_CONST("stdin"))
-
-    if (strcmp(params->inFileName, "stdin") == 0)
     {
         // used 'stdin' as input file
         SET_STREAM_TO_BIN_MODE(stdin);
-        *inFile = new WavInFile(stdin);
+        inFile = make_unique<WavInFile>(stdin);
     }
     else
     {
         // open input file...
-        *inFile = new WavInFile(params->inFileName);
+        inFile = make_unique<WavInFile>(params.inFileName.c_str());
     }
 
     // ... open output file with same sound parameters
-    bits = (int)(*inFile)->getNumBits();
+    const int bits = (int)inFile->getNumBits();
-    samplerate = (int)(*inFile)->getSampleRate();
+    const int samplerate = (int)inFile->getSampleRate();
-    channels = (int)(*inFile)->getNumChannels();
+    const int channels = (int)inFile->getNumChannels();
 
-    if (params->outFileName)
+    if (!params.outFileName.empty())
     {
-        if (strcmp(params->outFileName, "stdout") == 0)
+        if (params.outFileName == STRING_CONST("stdout"))
         {
             SET_STREAM_TO_BIN_MODE(stdout);
-            *outFile = new WavOutFile(stdout, samplerate, bits, channels);
+            outFile = make_unique<WavOutFile>(stdout, samplerate, bits, channels);
         }
         else
         {
-            *outFile = new WavOutFile(params->outFileName, samplerate, bits, channels);
+            outFile = make_unique<WavOutFile>(params.outFileName.c_str(), 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)
+static void setup(SoundTouch& soundTouch, const WavInFile& inFile, const RunParameters& params)
 {
-    int sampleRate;
+    const int sampleRate = (int)inFile.getSampleRate();
-    int channels;
+    const int channels = (int)inFile.getNumChannels();
+    soundTouch.setSampleRate(sampleRate);
+    soundTouch.setChannels(channels);
 
-    sampleRate = (int)inFile->getSampleRate();
+    soundTouch.setTempoChange(params.tempoDelta);
-    channels = (int)inFile->getNumChannels();
+    soundTouch.setPitchSemiTones(params.pitchDelta);
-    pSoundTouch->setSampleRate(sampleRate);
+    soundTouch.setRateChange(params.rateDelta);
-    pSoundTouch->setChannels(channels);
 
-    pSoundTouch->setTempoChange(params->tempoDelta);
+    soundTouch.setSetting(SETTING_USE_QUICKSEEK, params.quick);
-    pSoundTouch->setPitchSemiTones(params->pitchDelta);
+    soundTouch.setSetting(SETTING_USE_AA_FILTER, !(params.noAntiAlias));
-    pSoundTouch->setRateChange(params->rateDelta);
 
-    pSoundTouch->setSetting(SETTING_USE_QUICKSEEK, params->quick);
+    if (params.speech)
-    pSoundTouch->setSetting(SETTING_USE_AA_FILTER, !(params->noAntiAlias));
-
-    if (params->speech)
     {
         // use settings for speech processing
-        pSoundTouch->setSetting(SETTING_SEQUENCE_MS, 40);
+        soundTouch.setSetting(SETTING_SEQUENCE_MS, 40);
-        pSoundTouch->setSetting(SETTING_SEEKWINDOW_MS, 15);
+        soundTouch.setSetting(SETTING_SEEKWINDOW_MS, 15);
-        pSoundTouch->setSetting(SETTING_OVERLAP_MS, 8);
+        soundTouch.setSetting(SETTING_OVERLAP_MS, 8);
         fprintf(stderr, "Tune processing parameters for speech processing.\n");
     }
 
     // print processing information
-    if (params->outFileName)
+    if (!params.outFileName.empty())
     {
 #ifdef SOUNDTOUCH_INTEGER_SAMPLES
         fprintf(stderr, "Uses 16bit integer sample type in processing.\n\n");
 #else
-    #ifndef SOUNDTOUCH_FLOAT_SAMPLES
+#ifndef SOUNDTOUCH_FLOAT_SAMPLES
-        #error "Sampletype not defined"
+#error "Sampletype not defined"
-    #endif
+#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, "  tempo change = %+lg %%\n", params.tempoDelta);
-        fprintf(stderr, "  pitch change = %+g semitones\n", params->pitchDelta);
+        fprintf(stderr, "  pitch change = %+lg semitones\n", params.pitchDelta);
-        fprintf(stderr, "  rate change  = %+g %%\n\n", params->rateDelta);
+        fprintf(stderr, "  rate change  = %+lg %%\n\n", params.rateDelta);
         fprintf(stderr, "Working...");
     }
     else
@@ -172,32 +160,25 @@ static void setup(SoundTouch *pSoundTouch, const WavInFile *inFile, const RunPar
 }
 
 
-
 // Processes the sound
-static void process(SoundTouch *pSoundTouch, WavInFile *inFile, WavOutFile *outFile)
+static void process(SoundTouch& soundTouch, WavInFile& inFile, WavOutFile& outFile)
 {
-    int nSamples;
-    int nChannels;
-    int buffSizeSamples;
     SAMPLETYPE sampleBuffer[BUFF_SIZE];
+    int nSamples;
 
-    if ((inFile == NULL) || (outFile == NULL)) return;  // nothing to do.
+    const int nChannels = (int)inFile.getNumChannels();
-
-    nChannels = (int)inFile->getNumChannels();
     assert(nChannels > 0);
-    buffSizeSamples = BUFF_SIZE / nChannels;
+    const int buffSizeSamples = BUFF_SIZE / nChannels;
 
     // Process samples read from the input file
-    while (inFile->eof() == 0)
+    while (inFile.eof() == 0)
     {
-        int num;
-
         // Read a chunk of samples from the input file
-        num = inFile->read(sampleBuffer, BUFF_SIZE);
+        const int num = inFile.read(sampleBuffer, BUFF_SIZE);
-        nSamples = num / (int)inFile->getNumChannels();
+        int nSamples = num / (int)inFile.getNumChannels();
 
         // Feed the samples into SoundTouch processor
-        pSoundTouch->putSamples(sampleBuffer, nSamples);
+        soundTouch.putSamples(sampleBuffer, nSamples);
 
         // Read ready samples from SoundTouch processor & write them output file.
         // NOTES:
@@ -209,62 +190,57 @@ static void process(SoundTouch *pSoundTouch, WavInFile *inFile, WavOutFile *outF
         //   outputs samples.
         do
         {
-            nSamples = pSoundTouch->receiveSamples(sampleBuffer, buffSizeSamples);
+            nSamples = soundTouch.receiveSamples(sampleBuffer, buffSizeSamples);
-            outFile->write(sampleBuffer, nSamples * nChannels);
+            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();
+    soundTouch.flush();
     do
     {
-        nSamples = pSoundTouch->receiveSamples(sampleBuffer, buffSizeSamples);
+        nSamples = soundTouch.receiveSamples(sampleBuffer, buffSizeSamples);
-        outFile->write(sampleBuffer, nSamples * nChannels);
+        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)
+static void detectBPM(WavInFile& inFile, RunParameters& params)
 {
-    float bpmValue;
+    BPMDetect bpm(inFile.getNumChannels(), inFile.getSampleRate());
-    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();
+    const int nChannels = (int)inFile.getNumChannels();
-    assert(BUFF_SIZE % nChannels == 0);
+    int readSize = BUFF_SIZE - BUFF_SIZE % nChannels;   // round read size down to multiple of num.channels
 
     // Process the 'inFile' in small blocks, repeat until whole file has
     // been processed
-    while (inFile->eof() == 0)
+    while (inFile.eof() == 0)
     {
-        int num, samples;
-
         // Read sample data from input file
-        num = inFile->read(sampleBuffer, BUFF_SIZE);
+        const int num = inFile.read(sampleBuffer, readSize);
 
         // Enter the new samples to the bpm analyzer class
-        samples = num / nChannels;
+        const int samples = num / nChannels;
         bpm.inputSamples(sampleBuffer, samples);
     }
 
     // Now the whole song data has been analyzed. Read the resulting bpm.
-    bpmValue = bpm.getBpm();
+    const float bpmValue = bpm.getBpm();
     fprintf(stderr, "Done!\n");
 
     // rewind the file after bpm detection
-    inFile->rewind();
+    inFile.rewind();
 
     if (bpmValue > 0)
     {
-        fprintf(stderr, "Detected BPM rate %.1f\n\n", bpmValue);
+        fprintf(stderr, "Detected BPM rate %.1lf\n\n", bpmValue);
     }
     else
     {
@@ -272,62 +248,74 @@ static void detectBPM(WavInFile *inFile, RunParameters *params)
         return;
     }
 
-    if (params->goalBPM > 0)
+    if (params.goalBPM > 0)
     {
         // adjust tempo to given bpm
-        params->tempoDelta = (params->goalBPM / bpmValue - 1.0f) * 100.0f;
+        params.tempoDelta = (params.goalBPM / bpmValue - 1.0f) * 100.0f;
-        fprintf(stderr, "The file will be converted to %.1f BPM\n\n", params->goalBPM);
+        fprintf(stderr, "The file will be converted to %.1lf BPM\n\n", params.goalBPM);
     }
 }
 
-
+void printHelloText()
-
-int main(const int nParams, const char * const paramStr[])
 {
-    WavInFile *inFile;
+    SoundTouch soundTouch;
-    WavOutFile *outFile;
+    fprintf(stderr, _helloText, soundTouch.getVersionString());
-    RunParameters *params;
+}
+
+void ss_main(RunParameters& params)
+{
+    unique_ptr<WavInFile> inFile;
+    unique_ptr<WavOutFile> outFile;
     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);
+    openFiles(inFile, outFile, params);
 
-        if (params->detectBPM == TRUE)
+    if (params.detectBPM == true)
     {
         // detect sound BPM (and adjust processing parameters
         //  accordingly if necessary)
-            detectBPM(inFile, params);
+        detectBPM(*inFile, params);
     }
 
     // Setup the 'SoundTouch' object for processing the sound
-        setup(&soundTouch, inFile, params);
+    setup(soundTouch, *inFile, params);
 
     // clock_t cs = clock();    // for benchmarking processing duration
     // Process the sound
-        process(&soundTouch, inFile, outFile);
+    if (inFile && outFile)
+    {
+        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) 
+
+}
+
+#if _WIN32
+int wmain(int argc, const wchar_t* args[])
+#else
+int main(int argc, const char* args[])
+#endif
+{
+    try
+    {
+        soundstretch::printHelloText();
+        soundstretch::RunParameters params(argc, args);
+        soundstretch::ss_main(params);
+    }
+    catch (const runtime_error& e)
     {
-        // An exception occurred during processing, display an error message
         fprintf(stderr, "%s\n", e.what());
         return -1;
     }
-
+    catch (const string& e)
+    {
+        fprintf(stderr, "%s\n", e.c_str());
+        return -1;
+    }
     return 0;
 }

source/SoundStretch/soundstretch.dsp

@@ -1,137 +0,0 @@
-# Microsoft Developer Studio Project File - Name="soundstretch" - Package Owner=<4>
-# Microsoft Developer Studio Generated Build File, Format Version 6.00
-# ** DO NOT EDIT **
-
-# TARGTYPE "Win32 (x86) Console Application" 0x0103
-
-CFG=soundstretch - Win32 Debug
-!MESSAGE This is not a valid makefile. To build this project using NMAKE,
-!MESSAGE use the Export Makefile command and run
-!MESSAGE 
-!MESSAGE NMAKE /f "soundstretch.mak".
-!MESSAGE 
-!MESSAGE You can specify a configuration when running NMAKE
-!MESSAGE by defining the macro CFG on the command line. For example:
-!MESSAGE 
-!MESSAGE NMAKE /f "soundstretch.mak" CFG="soundstretch - Win32 Debug"
-!MESSAGE 
-!MESSAGE Possible choices for configuration are:
-!MESSAGE 
-!MESSAGE "soundstretch - Win32 Release" (based on "Win32 (x86) Console Application")
-!MESSAGE "soundstretch - Win32 Debug" (based on "Win32 (x86) Console Application")
-!MESSAGE 
-
-# Begin Project
-# PROP AllowPerConfigDependencies 0
-# PROP Scc_ProjName ""
-# PROP Scc_LocalPath ""
-CPP=cl.exe
-RSC=rc.exe
-
-!IF  "$(CFG)" == "soundstretch - Win32 Release"
-
-# PROP BASE Use_MFC 0
-# PROP BASE Use_Debug_Libraries 0
-# PROP BASE Output_Dir "Release"
-# PROP BASE Intermediate_Dir "Release"
-# PROP BASE Target_Dir ""
-# PROP Use_MFC 0
-# PROP Use_Debug_Libraries 0
-# PROP Output_Dir "Release"
-# PROP Intermediate_Dir "Release"
-# PROP Ignore_Export_Lib 0
-# PROP Target_Dir ""
-# ADD BASE CPP /nologo /W3 /GX /O2 /D "WIN32" /D "NDEBUG" /D "_CONSOLE" /D "_MBCS" /YX /FD /c
-# ADD CPP /nologo /W3 /GX /Zi /O2 /I "..\..\include" /D "WIN32" /D "NDEBUG" /D "_CONSOLE" /D "_MBCS" /FD /c
-# SUBTRACT CPP /YX
-# ADD BASE RSC /l 0x40b /d "NDEBUG"
-# ADD RSC /l 0x40b /d "NDEBUG"
-BSC32=bscmake.exe
-# ADD BASE BSC32 /nologo
-# ADD BSC32 /nologo
-LINK32=link.exe
-# ADD BASE LINK32 kernel32.lib user32.lib gdi32.lib winspool.lib comdlg32.lib advapi32.lib shell32.lib ole32.lib oleaut32.lib uuid.lib odbc32.lib odbccp32.lib kernel32.lib user32.lib gdi32.lib winspool.lib comdlg32.lib advapi32.lib shell32.lib ole32.lib oleaut32.lib uuid.lib odbc32.lib odbccp32.lib /nologo /subsystem:console /machine:I386
-# ADD LINK32 SoundTouch.lib /nologo /subsystem:console /profile /map /debug /machine:I386 /libpath:"..\..\lib"
-# Begin Special Build Tool
-SOURCE="$(InputPath)"
-PostBuild_Cmds=copy           Release\soundstretch.exe           ..\..\bin\ 
-# End Special Build Tool
-
-!ELSEIF  "$(CFG)" == "soundstretch - Win32 Debug"
-
-# PROP BASE Use_MFC 0
-# PROP BASE Use_Debug_Libraries 1
-# PROP BASE Output_Dir "Debug"
-# PROP BASE Intermediate_Dir "Debug"
-# PROP BASE Target_Dir ""
-# PROP Use_MFC 0
-# PROP Use_Debug_Libraries 1
-# PROP Output_Dir "Debug"
-# PROP Intermediate_Dir "Debug"
-# PROP Ignore_Export_Lib 0
-# PROP Target_Dir ""
-# ADD BASE CPP /nologo /W3 /Gm /GX /ZI /Od /D "WIN32" /D "_DEBUG" /D "_CONSOLE" /D "_MBCS" /YX /FD /GZ /c
-# ADD CPP /nologo /Zp16 /W3 /Gm /GX /ZI /Od /I "..\..\include" /D "WIN32" /D "_DEBUG" /D "_CONSOLE" /D "_MBCS" /FR /FD /GZ /c
-# SUBTRACT CPP /YX
-# ADD BASE RSC /l 0x40b /d "_DEBUG"
-# ADD RSC /l 0x40b /d "_DEBUG"
-BSC32=bscmake.exe
-# ADD BASE BSC32 /nologo
-# ADD BSC32 /nologo
-LINK32=link.exe
-# ADD BASE LINK32 kernel32.lib user32.lib gdi32.lib winspool.lib comdlg32.lib advapi32.lib shell32.lib ole32.lib oleaut32.lib uuid.lib odbc32.lib odbccp32.lib kernel32.lib user32.lib gdi32.lib winspool.lib comdlg32.lib advapi32.lib shell32.lib ole32.lib oleaut32.lib uuid.lib odbc32.lib odbccp32.lib /nologo /subsystem:console /debug /machine:I386 /pdbtype:sept
-# ADD LINK32 SoundTouchD.lib /nologo /subsystem:console /map /debug /machine:I386 /nodefaultlib:"libc" /out:"Debug/soundstretchD.exe" /pdbtype:sept /libpath:"..\..\lib"
-# SUBTRACT LINK32 /pdb:none
-# Begin Special Build Tool
-SOURCE="$(InputPath)"
-PostBuild_Cmds=copy           Debug\soundstretchD.exe           ..\..\bin\ 
-# End Special Build Tool
-
-!ENDIF 
-
-# Begin Target
-
-# Name "soundstretch - Win32 Release"
-# Name "soundstretch - Win32 Debug"
-# Begin Group "Source Files"
-
-# PROP Default_Filter "cpp;c;cxx;rc;def;r;odl;idl;hpj;bat"
-# Begin Source File
-
-SOURCE=.\main.cpp
-# End Source File
-# Begin Source File
-
-SOURCE=.\RunParameters.cpp
-# End Source File
-# Begin Source File
-
-SOURCE=.\WavFile.cpp
-# End Source File
-# End Group
-# Begin Group "Header Files"
-
-# PROP Default_Filter "h;hpp;hxx;hm;inl"
-# Begin Source File
-
-SOURCE=.\RunParameters.h
-# End Source File
-# Begin Source File
-
-SOURCE=..\..\..\include\SoundTouch.h
-# End Source File
-# Begin Source File
-
-SOURCE=..\..\..\include\STTypes.h
-# End Source File
-# Begin Source File
-
-SOURCE=.\WavFile.h
-# End Source File
-# End Group
-# Begin Group "Resource Files"
-
-# PROP Default_Filter "ico;cur;bmp;dlg;rc2;rct;bin;rgs;gif;jpg;jpeg;jpe"
-# End Group
-# End Target
-# End Project

source/SoundStretch/soundstretch.dsw

@@ -1,44 +0,0 @@
-Microsoft Developer Studio Workspace File, Format Version 6.00
-# WARNING: DO NOT EDIT OR DELETE THIS WORKSPACE FILE!
-
-###############################################################################
-
-Project: "SoundTouch"=..\SoundTouch\SoundTouch.dsp - Package Owner=<4>
-
-Package=<5>
-{{{
-}}}
-
-Package=<4>
-{{{
-}}}
-
-###############################################################################
-
-Project: "soundstretch"=.\soundstretch.dsp - Package Owner=<4>
-
-Package=<5>
-{{{
-}}}
-
-Package=<4>
-{{{
-    Begin Project Dependency
-    Project_Dep_Name SoundTouch
-    End Project Dependency
-}}}
-
-###############################################################################
-
-Global:
-
-Package=<5>
-{{{
-}}}
-
-Package=<3>
-{{{
-}}}
-
-###############################################################################
-

source/SoundStretch/soundstretch.sln

@@ -1,32 +1,37 @@
-Microsoft Visual Studio Solution File, Format Version 8.00
+Microsoft Visual Studio Solution File, Format Version 12.00
-Project("{8BC9CEB8-8B4A-11D0-8D11-00A0C91BC942}") = "soundstretch", "soundstretch.vcproj", "{5AACDFFA-D491-44B8-A332-DA7ACCAAF2AF}"
+# Visual Studio 14
-	ProjectSection(ProjectDependencies) = postProject
+VisualStudioVersion = 14.0.23107.0
-		{68A5DD20-7057-448B-8FE0-B6AC8D205509} = {68A5DD20-7057-448B-8FE0-B6AC8D205509}
+MinimumVisualStudioVersion = 10.0.40219.1
-	EndProjectSection
+Project("{8BC9CEB8-8B4A-11D0-8D11-00A0C91BC942}") = "soundstretch", "soundstretch.vcxproj", "{5AACDFFA-D491-44B8-A332-DA7ACCAAF2AF}"
 EndProject
-Project("{8BC9CEB8-8B4A-11D0-8D11-00A0C91BC942}") = "SoundTouch", "..\SoundTouch\SoundTouch.vcproj", "{68A5DD20-7057-448B-8FE0-B6AC8D205509}"
+Project("{8BC9CEB8-8B4A-11D0-8D11-00A0C91BC942}") = "SoundTouch", "..\SoundTouch\SoundTouch.vcxproj", "{68A5DD20-7057-448B-8FE0-B6AC8D205509}"
-	ProjectSection(ProjectDependencies) = postProject
-	EndProjectSection
 EndProject
 Global
-	GlobalSection(SolutionConfiguration) = preSolution
+	GlobalSection(SolutionConfigurationPlatforms) = preSolution
-		Debug = Debug
+		Debug|Win32 = Debug|Win32
-		Release = Release
+		Debug|x64 = Debug|x64
+		Release|Win32 = Release|Win32
+		Release|x64 = Release|x64
 	EndGlobalSection
-	GlobalSection(ProjectDependencies) = postSolution
+	GlobalSection(ProjectConfigurationPlatforms) = postSolution
+		{5AACDFFA-D491-44B8-A332-DA7ACCAAF2AF}.Debug|Win32.ActiveCfg = Debug|Win32
+		{5AACDFFA-D491-44B8-A332-DA7ACCAAF2AF}.Debug|Win32.Build.0 = Debug|Win32
+		{5AACDFFA-D491-44B8-A332-DA7ACCAAF2AF}.Debug|x64.ActiveCfg = Debug|x64
+		{5AACDFFA-D491-44B8-A332-DA7ACCAAF2AF}.Debug|x64.Build.0 = Debug|x64
+		{5AACDFFA-D491-44B8-A332-DA7ACCAAF2AF}.Release|Win32.ActiveCfg = Release|Win32
+		{5AACDFFA-D491-44B8-A332-DA7ACCAAF2AF}.Release|Win32.Build.0 = Release|Win32
+		{5AACDFFA-D491-44B8-A332-DA7ACCAAF2AF}.Release|x64.ActiveCfg = Release|x64
+		{5AACDFFA-D491-44B8-A332-DA7ACCAAF2AF}.Release|x64.Build.0 = Release|x64
+		{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}.Debug|x64.ActiveCfg = Debug|x64
+		{68A5DD20-7057-448B-8FE0-B6AC8D205509}.Debug|x64.Build.0 = Debug|x64
+		{68A5DD20-7057-448B-8FE0-B6AC8D205509}.Release|Win32.ActiveCfg = Release|Win32
+		{68A5DD20-7057-448B-8FE0-B6AC8D205509}.Release|Win32.Build.0 = Release|Win32
+		{68A5DD20-7057-448B-8FE0-B6AC8D205509}.Release|x64.ActiveCfg = Release|x64
+		{68A5DD20-7057-448B-8FE0-B6AC8D205509}.Release|x64.Build.0 = Release|x64
 	EndGlobalSection
-	GlobalSection(ProjectConfiguration) = postSolution
+	GlobalSection(SolutionProperties) = preSolution
-		{5AACDFFA-D491-44B8-A332-DA7ACCAAF2AF}.Debug.ActiveCfg = Debug|Win32
+		HideSolutionNode = FALSE
-		{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
-	GlobalSection(ExtensibilityAddIns) = postSolution
 	EndGlobalSection
 EndGlobal

source/SoundStretch/soundstretch.vcproj

@@ -1,235 +0,0 @@
-<?xml version="1.0" encoding="Windows-1252"?>
-<VisualStudioProject
-	ProjectType="Visual C++"
-	Version="7.10"
-	Name="soundstretch"
-	SccProjectName=""
-	SccLocalPath="">
-	<Platforms>
-		<Platform
-			Name="Win32"/>
-	</Platforms>
-	<Configurations>
-		<Configuration
-			Name="Debug|Win32"
-			OutputDirectory=".\Debug"
-			IntermediateDirectory=".\Debug"
-			ConfigurationType="1"
-			UseOfMFC="0"
-			ATLMinimizesCRunTimeLibraryUsage="FALSE"
-			CharacterSet="2">
-			<Tool
-				Name="VCCLCompilerTool"
-				Optimization="0"
-				AdditionalIncludeDirectories="..\..\include"
-				PreprocessorDefinitions="WIN32;_DEBUG;_CONSOLE;_CRT_SECURE_NO_WARNINGS"
-				BasicRuntimeChecks="3"
-				RuntimeLibrary="5"
-				StructMemberAlignment="5"
-				PrecompiledHeaderFile=".\Debug/soundstretch.pch"
-				AssemblerListingLocation=".\Debug/"
-				ObjectFile=".\Debug/"
-				ProgramDataBaseFileName=".\Debug/"
-				BrowseInformation="1"
-				WarningLevel="3"
-				SuppressStartupBanner="TRUE"
-				DebugInformationFormat="4"
-				CompileAs="0"/>
-			<Tool
-				Name="VCCustomBuildTool"/>
-			<Tool
-				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"
-				TypeLibraryName=".\Debug/soundstretch.tlb"
-				HeaderFileName=""/>
-			<Tool
-				Name="VCPostBuildEventTool"
-				CommandLine="copy          Debug\soundstretchD.exe          ..\..\bin\"/>
-			<Tool
-				Name="VCPreBuildEventTool"/>
-			<Tool
-				Name="VCPreLinkEventTool"/>
-			<Tool
-				Name="VCResourceCompilerTool"
-				PreprocessorDefinitions="_DEBUG"
-				Culture="1035"/>
-			<Tool
-				Name="VCWebServiceProxyGeneratorTool"/>
-			<Tool
-				Name="VCXMLDataGeneratorTool"/>
-			<Tool
-				Name="VCWebDeploymentTool"/>
-			<Tool
-				Name="VCManagedWrapperGeneratorTool"/>
-			<Tool
-				Name="VCAuxiliaryManagedWrapperGeneratorTool"/>
-		</Configuration>
-		<Configuration
-			Name="Release|Win32"
-			OutputDirectory=".\Release"
-			IntermediateDirectory=".\Release"
-			ConfigurationType="1"
-			UseOfMFC="0"
-			ATLMinimizesCRunTimeLibraryUsage="FALSE"
-			CharacterSet="2">
-			<Tool
-				Name="VCCLCompilerTool"
-				Optimization="3"
-				GlobalOptimizations="FALSE"
-				InlineFunctionExpansion="2"
-				EnableIntrinsicFunctions="TRUE"
-				AdditionalIncludeDirectories="..\..\include"
-				PreprocessorDefinitions="WIN32;NDEBUG;_CONSOLE;_CRT_SECURE_NO_WARNINGS"
-				StringPooling="TRUE"
-				RuntimeLibrary="4"
-				EnableFunctionLevelLinking="TRUE"
-				PrecompiledHeaderFile=".\Release/soundstretch.pch"
-				AssemblerListingLocation=".\Release/"
-				ObjectFile=".\Release/"
-				ProgramDataBaseFileName=".\Release/"
-				WarningLevel="3"
-				SuppressStartupBanner="TRUE"
-				DebugInformationFormat="0"
-				CompileAs="0"/>
-			<Tool
-				Name="VCCustomBuildTool"/>
-			<Tool
-				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"
-				TypeLibraryName=".\Release/soundstretch.tlb"
-				HeaderFileName=""/>
-			<Tool
-				Name="VCPostBuildEventTool"
-				CommandLine="copy          Release\soundstretch.exe          ..\..\bin\"/>
-			<Tool
-				Name="VCPreBuildEventTool"/>
-			<Tool
-				Name="VCPreLinkEventTool"/>
-			<Tool
-				Name="VCResourceCompilerTool"
-				PreprocessorDefinitions="NDEBUG"
-				Culture="1035"/>
-			<Tool
-				Name="VCWebServiceProxyGeneratorTool"/>
-			<Tool
-				Name="VCXMLDataGeneratorTool"/>
-			<Tool
-				Name="VCWebDeploymentTool"/>
-			<Tool
-				Name="VCManagedWrapperGeneratorTool"/>
-			<Tool
-				Name="VCAuxiliaryManagedWrapperGeneratorTool"/>
-		</Configuration>
-	</Configurations>
-	<References>
-	</References>
-	<Files>
-		<Filter
-			Name="Source Files"
-			Filter="cpp;c;cxx;rc;def;r;odl;idl;hpj;bat">
-			<File
-				RelativePath="main.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="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
-						Name="VCCLCompilerTool"
-						Optimization="0"
-						AdditionalIncludeDirectories=""
-						PreprocessorDefinitions=""
-						BasicRuntimeChecks="3"
-						BrowseInformation="1"/>
-				</FileConfiguration>
-				<FileConfiguration
-					Name="Release|Win32">
-					<Tool
-						Name="VCCLCompilerTool"
-						Optimization="2"
-						AdditionalIncludeDirectories=""
-						PreprocessorDefinitions=""/>
-				</FileConfiguration>
-			</File>
-		</Filter>
-		<Filter
-			Name="Header Files"
-			Filter="h;hpp;hxx;hm;inl">
-			<File
-				RelativePath="RunParameters.h">
-			</File>
-			<File
-				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>
-</VisualStudioProject>

source/SoundStretch/soundstretch.vcxproj

@@ -0,0 +1,333 @@
+<?xml version="1.0" encoding="utf-8"?>
+<Project DefaultTargets="Build" ToolsVersion="14.0" xmlns="http://schemas.microsoft.com/developer/msbuild/2003">
+  <ItemGroup Label="ProjectConfigurations">
+    <ProjectConfiguration Include="Debug|Win32">
+      <Configuration>Debug</Configuration>
+      <Platform>Win32</Platform>
+    </ProjectConfiguration>
+    <ProjectConfiguration Include="Debug|x64">
+      <Configuration>Debug</Configuration>
+      <Platform>x64</Platform>
+    </ProjectConfiguration>
+    <ProjectConfiguration Include="Release|Win32">
+      <Configuration>Release</Configuration>
+      <Platform>Win32</Platform>
+    </ProjectConfiguration>
+    <ProjectConfiguration Include="Release|x64">
+      <Configuration>Release</Configuration>
+      <Platform>x64</Platform>
+    </ProjectConfiguration>
+  </ItemGroup>
+  <PropertyGroup Label="Globals">
+    <ProjectGuid>{5AACDFFA-D491-44B8-A332-DA7ACCAAF2AF}</ProjectGuid>
+    <RootNamespace>soundstretch</RootNamespace>
+    <WindowsTargetPlatformVersion>10.0</WindowsTargetPlatformVersion>
+  </PropertyGroup>
+  <Import Project="$(VCTargetsPath)\Microsoft.Cpp.Default.props" />
+  <PropertyGroup Condition="'$(Configuration)|$(Platform)'=='Release|Win32'" Label="Configuration">
+    <ConfigurationType>Application</ConfigurationType>
+    <PlatformToolset>v142</PlatformToolset>
+    <UseOfMfc>false</UseOfMfc>
+    <CharacterSet>Unicode</CharacterSet>
+  </PropertyGroup>
+  <PropertyGroup Condition="'$(Configuration)|$(Platform)'=='Debug|Win32'" Label="Configuration">
+    <ConfigurationType>Application</ConfigurationType>
+    <PlatformToolset>v142</PlatformToolset>
+    <UseOfMfc>false</UseOfMfc>
+    <CharacterSet>Unicode</CharacterSet>
+  </PropertyGroup>
+  <PropertyGroup Condition="'$(Configuration)|$(Platform)'=='Release|x64'" Label="Configuration">
+    <ConfigurationType>Application</ConfigurationType>
+    <PlatformToolset>v142</PlatformToolset>
+    <UseOfMfc>false</UseOfMfc>
+    <CharacterSet>Unicode</CharacterSet>
+  </PropertyGroup>
+  <PropertyGroup Condition="'$(Configuration)|$(Platform)'=='Debug|x64'" Label="Configuration">
+    <ConfigurationType>Application</ConfigurationType>
+    <PlatformToolset>v142</PlatformToolset>
+    <UseOfMfc>false</UseOfMfc>
+    <CharacterSet>Unicode</CharacterSet>
+  </PropertyGroup>
+  <Import Project="$(VCTargetsPath)\Microsoft.Cpp.props" />
+  <ImportGroup Label="ExtensionSettings">
+  </ImportGroup>
+  <ImportGroup Condition="'$(Configuration)|$(Platform)'=='Release|Win32'" Label="PropertySheets">
+    <Import Project="$(UserRootDir)\Microsoft.Cpp.$(Platform).user.props" Condition="exists('$(UserRootDir)\Microsoft.Cpp.$(Platform).user.props')" Label="LocalAppDataPlatform" />
+    <Import Project="$(VCTargetsPath)Microsoft.CPP.UpgradeFromVC71.props" />
+  </ImportGroup>
+  <ImportGroup Condition="'$(Configuration)|$(Platform)'=='Debug|Win32'" Label="PropertySheets">
+    <Import Project="$(UserRootDir)\Microsoft.Cpp.$(Platform).user.props" Condition="exists('$(UserRootDir)\Microsoft.Cpp.$(Platform).user.props')" Label="LocalAppDataPlatform" />
+    <Import Project="$(VCTargetsPath)Microsoft.CPP.UpgradeFromVC71.props" />
+  </ImportGroup>
+  <ImportGroup Condition="'$(Configuration)|$(Platform)'=='Release|x64'" Label="PropertySheets">
+    <Import Project="$(UserRootDir)\Microsoft.Cpp.$(Platform).user.props" Condition="exists('$(UserRootDir)\Microsoft.Cpp.$(Platform).user.props')" Label="LocalAppDataPlatform" />
+    <Import Project="$(VCTargetsPath)Microsoft.CPP.UpgradeFromVC71.props" />
+  </ImportGroup>
+  <ImportGroup Condition="'$(Configuration)|$(Platform)'=='Debug|x64'" Label="PropertySheets">
+    <Import Project="$(UserRootDir)\Microsoft.Cpp.$(Platform).user.props" Condition="exists('$(UserRootDir)\Microsoft.Cpp.$(Platform).user.props')" Label="LocalAppDataPlatform" />
+    <Import Project="$(VCTargetsPath)Microsoft.CPP.UpgradeFromVC71.props" />
+  </ImportGroup>
+  <PropertyGroup Label="UserMacros" />
+  <PropertyGroup>
+    <_ProjectFileVersion>14.0.23107.0</_ProjectFileVersion>
+  </PropertyGroup>
+  <PropertyGroup Condition="'$(Configuration)|$(Platform)'=='Debug|Win32'">
+    <OutDir>$(Platform)\$(Configuration)\</OutDir>
+    <IntDir>$(Platform)\$(Configuration)\</IntDir>
+    <LinkIncremental>true</LinkIncremental>
+    <TargetName>$(ProjectName)D</TargetName>
+  </PropertyGroup>
+  <PropertyGroup Condition="'$(Configuration)|$(Platform)'=='Release|Win32'">
+    <OutDir>$(Platform)\$(Configuration)\</OutDir>
+    <IntDir>$(Platform)\$(Configuration)\</IntDir>
+    <LinkIncremental>false</LinkIncremental>
+  </PropertyGroup>
+  <PropertyGroup Condition="'$(Configuration)|$(Platform)'=='Debug|x64'">
+    <OutDir>$(Platform)\$(Configuration)\</OutDir>
+    <IntDir>$(Platform)\$(Configuration)\</IntDir>
+    <LinkIncremental>true</LinkIncremental>
+    <TargetName>$(ProjectName)D_x64</TargetName>
+  </PropertyGroup>
+  <PropertyGroup Condition="'$(Configuration)|$(Platform)'=='Release|x64'">
+    <OutDir>$(Platform)\$(Configuration)\</OutDir>
+    <IntDir>$(Platform)\$(Configuration)\</IntDir>
+    <LinkIncremental>false</LinkIncremental>
+    <TargetName>$(ProjectName)_x64</TargetName>
+  </PropertyGroup>
+  <ItemDefinitionGroup Condition="'$(Configuration)|$(Platform)'=='Debug|Win32'">
+    <Midl>
+      <TypeLibraryName>.\Debug/soundstretch.tlb</TypeLibraryName>
+      <HeaderFileName />
+    </Midl>
+    <ClCompile>
+      <Optimization>Disabled</Optimization>
+      <AdditionalIncludeDirectories>..\..\include;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>
+      <PreprocessorDefinitions>WIN32;_DEBUG;_CONSOLE;_CRT_SECURE_NO_WARNINGS;%(PreprocessorDefinitions)</PreprocessorDefinitions>
+      <BasicRuntimeChecks>EnableFastChecks</BasicRuntimeChecks>
+      <RuntimeLibrary>MultiThreadedDebug</RuntimeLibrary>
+      <StructMemberAlignment>16Bytes</StructMemberAlignment>
+      <FloatingPointModel>Fast</FloatingPointModel>
+      <PrecompiledHeaderOutputFile>$(OutDir)$(TargetName).pch</PrecompiledHeaderOutputFile>
+      <AssemblerListingLocation>$(OutDir)</AssemblerListingLocation>
+      <ObjectFileName>$(OutDir)</ObjectFileName>
+      <ProgramDataBaseFileName>$(OutDir)</ProgramDataBaseFileName>
+      <BrowseInformation>true</BrowseInformation>
+      <WarningLevel>Level3</WarningLevel>
+      <SuppressStartupBanner>true</SuppressStartupBanner>
+      <DebugInformationFormat>ProgramDatabase</DebugInformationFormat>
+      <CompileAs>Default</CompileAs>
+      <EnableEnhancedInstructionSet>StreamingSIMDExtensions2</EnableEnhancedInstructionSet>
+      <MultiProcessorCompilation>true</MultiProcessorCompilation>
+    </ClCompile>
+    <ResourceCompile>
+      <PreprocessorDefinitions>_DEBUG;%(PreprocessorDefinitions)</PreprocessorDefinitions>
+      <Culture>0x040b</Culture>
+    </ResourceCompile>
+    <Link>
+      <AdditionalDependencies>%(AdditionalDependencies)</AdditionalDependencies>
+      <OutputFile>$(OutDir)$(TargetName)$(TargetExt)</OutputFile>
+      <SuppressStartupBanner>true</SuppressStartupBanner>
+      <AdditionalLibraryDirectories>..\..\lib;%(AdditionalLibraryDirectories)</AdditionalLibraryDirectories>
+      <IgnoreSpecificDefaultLibraries>libc;%(IgnoreSpecificDefaultLibraries)</IgnoreSpecificDefaultLibraries>
+      <GenerateDebugInformation>true</GenerateDebugInformation>
+      <ProgramDatabaseFile>$(OutDir)$(TargetName).pdb</ProgramDatabaseFile>
+      <GenerateMapFile>true</GenerateMapFile>
+      <MapFileName>$(OutDir)$(TargetName).map</MapFileName>
+      <SubSystem>Console</SubSystem>
+      <DataExecutionPrevention />
+    </Link>
+    <PostBuildEvent>
+      <Command>if not exist ..\..\bin mkdir ..\..\bin
+copy $(OutDir)$(TargetName)$(TargetExt) ..\..\bin\</Command>
+    </PostBuildEvent>
+  </ItemDefinitionGroup>
+  <ItemDefinitionGroup Condition="'$(Configuration)|$(Platform)'=='Release|Win32'">
+    <Midl>
+      <TypeLibraryName>.\Release/soundstretch.tlb</TypeLibraryName>
+      <HeaderFileName />
+    </Midl>
+    <ClCompile>
+      <Optimization>Full</Optimization>
+      <InlineFunctionExpansion>AnySuitable</InlineFunctionExpansion>
+      <IntrinsicFunctions>true</IntrinsicFunctions>
+      <AdditionalIncludeDirectories>..\..\include;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>
+      <PreprocessorDefinitions>WIN32;NDEBUG;_CONSOLE;_CRT_SECURE_NO_WARNINGS;%(PreprocessorDefinitions)</PreprocessorDefinitions>
+      <StringPooling>true</StringPooling>
+      <RuntimeLibrary>MultiThreaded</RuntimeLibrary>
+      <FunctionLevelLinking>true</FunctionLevelLinking>
+      <FloatingPointModel>Fast</FloatingPointModel>
+      <PrecompiledHeaderOutputFile>$(OutDir)$(TargetName).pch</PrecompiledHeaderOutputFile>
+      <AssemblerListingLocation>$(OutDir)</AssemblerListingLocation>
+      <ObjectFileName>$(OutDir)</ObjectFileName>
+      <ProgramDataBaseFileName>$(OutDir)</ProgramDataBaseFileName>
+      <WarningLevel>Level3</WarningLevel>
+      <SuppressStartupBanner>true</SuppressStartupBanner>
+      <DebugInformationFormat />
+      <CompileAs>Default</CompileAs>
+      <EnableEnhancedInstructionSet>StreamingSIMDExtensions2</EnableEnhancedInstructionSet>
+      <MultiProcessorCompilation>true</MultiProcessorCompilation>
+    </ClCompile>
+    <ResourceCompile>
+      <PreprocessorDefinitions>NDEBUG;%(PreprocessorDefinitions)</PreprocessorDefinitions>
+      <Culture>0x040b</Culture>
+    </ResourceCompile>
+    <Link>
+      <AdditionalDependencies>%(AdditionalDependencies)</AdditionalDependencies>
+      <OutputFile>$(OutDir)$(TargetName)$(TargetExt)</OutputFile>
+      <SuppressStartupBanner>true</SuppressStartupBanner>
+      <AdditionalLibraryDirectories>..\..\lib;%(AdditionalLibraryDirectories)</AdditionalLibraryDirectories>
+      <GenerateDebugInformation>false</GenerateDebugInformation>
+      <GenerateMapFile>true</GenerateMapFile>
+      <MapFileName>$(OutDir)$(TargetName).map</MapFileName>
+      <SubSystem>Console</SubSystem>
+      <DataExecutionPrevention />
+    </Link>
+    <PostBuildEvent>
+      <Command>if not exist ..\..\bin mkdir ..\..\bin
+copy $(OutDir)$(TargetName)$(TargetExt) ..\..\bin\</Command>
+    </PostBuildEvent>
+  </ItemDefinitionGroup>
+  <ItemDefinitionGroup Condition="'$(Configuration)|$(Platform)'=='Debug|x64'">
+    <Midl>
+      <TargetEnvironment>X64</TargetEnvironment>
+      <TypeLibraryName>.\Debug/soundstretch.tlb</TypeLibraryName>
+      <HeaderFileName />
+    </Midl>
+    <ClCompile>
+      <Optimization>Disabled</Optimization>
+      <AdditionalIncludeDirectories>..\..\include;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>
+      <PreprocessorDefinitions>WIN32;_DEBUG;_CONSOLE;_CRT_SECURE_NO_WARNINGS;%(PreprocessorDefinitions)</PreprocessorDefinitions>
+      <BasicRuntimeChecks>EnableFastChecks</BasicRuntimeChecks>
+      <RuntimeLibrary>MultiThreadedDebug</RuntimeLibrary>
+      <StructMemberAlignment>16Bytes</StructMemberAlignment>
+      <FloatingPointModel>Fast</FloatingPointModel>
+      <PrecompiledHeaderOutputFile>$(OutDir)$(TargetName).pch</PrecompiledHeaderOutputFile>
+      <AssemblerListingLocation>$(OutDir)</AssemblerListingLocation>
+      <ObjectFileName>$(OutDir)</ObjectFileName>
+      <ProgramDataBaseFileName>$(OutDir)</ProgramDataBaseFileName>
+      <BrowseInformation>true</BrowseInformation>
+      <WarningLevel>Level3</WarningLevel>
+      <SuppressStartupBanner>true</SuppressStartupBanner>
+      <DebugInformationFormat>ProgramDatabase</DebugInformationFormat>
+      <CompileAs>Default</CompileAs>
+      <EnableEnhancedInstructionSet>
+      </EnableEnhancedInstructionSet>
+      <MultiProcessorCompilation>true</MultiProcessorCompilation>
+    </ClCompile>
+    <ResourceCompile>
+      <PreprocessorDefinitions>_DEBUG;%(PreprocessorDefinitions)</PreprocessorDefinitions>
+      <Culture>0x040b</Culture>
+    </ResourceCompile>
+    <Link>
+      <AdditionalDependencies>%(AdditionalDependencies)</AdditionalDependencies>
+      <OutputFile>$(OutDir)$(TargetName)$(TargetExt)</OutputFile>
+      <SuppressStartupBanner>true</SuppressStartupBanner>
+      <AdditionalLibraryDirectories>..\..\lib;%(AdditionalLibraryDirectories)</AdditionalLibraryDirectories>
+      <IgnoreSpecificDefaultLibraries>libc;%(IgnoreSpecificDefaultLibraries)</IgnoreSpecificDefaultLibraries>
+      <GenerateDebugInformation>true</GenerateDebugInformation>
+      <ProgramDatabaseFile>$(OutDir)$(TargetName).pdb</ProgramDatabaseFile>
+      <GenerateMapFile>true</GenerateMapFile>
+      <MapFileName>$(OutDir)$(TargetName).map</MapFileName>
+      <SubSystem>Console</SubSystem>
+      <DataExecutionPrevention />
+    </Link>
+    <PostBuildEvent>
+      <Command>if not exist ..\..\bin mkdir ..\..\bin
+copy $(OutDir)$(TargetName)$(TargetExt) ..\..\bin\</Command>
+    </PostBuildEvent>
+  </ItemDefinitionGroup>
+  <ItemDefinitionGroup Condition="'$(Configuration)|$(Platform)'=='Release|x64'">
+    <Midl>
+      <TargetEnvironment>X64</TargetEnvironment>
+      <TypeLibraryName>.\Release/soundstretch.tlb</TypeLibraryName>
+      <HeaderFileName />
+    </Midl>
+    <ClCompile>
+      <Optimization>Full</Optimization>
+      <InlineFunctionExpansion>AnySuitable</InlineFunctionExpansion>
+      <IntrinsicFunctions>true</IntrinsicFunctions>
+      <AdditionalIncludeDirectories>..\..\include;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>
+      <PreprocessorDefinitions>WIN32;NDEBUG;_CONSOLE;_CRT_SECURE_NO_WARNINGS;%(PreprocessorDefinitions)</PreprocessorDefinitions>
+      <StringPooling>true</StringPooling>
+      <RuntimeLibrary>MultiThreaded</RuntimeLibrary>
+      <FunctionLevelLinking>true</FunctionLevelLinking>
+      <FloatingPointModel>Fast</FloatingPointModel>
+      <PrecompiledHeaderOutputFile>$(OutDir)$(TargetName).pch</PrecompiledHeaderOutputFile>
+      <AssemblerListingLocation>$(OutDir)</AssemblerListingLocation>
+      <ObjectFileName>$(OutDir)</ObjectFileName>
+      <ProgramDataBaseFileName>$(OutDir)</ProgramDataBaseFileName>
+      <WarningLevel>Level3</WarningLevel>
+      <SuppressStartupBanner>true</SuppressStartupBanner>
+      <DebugInformationFormat />
+      <CompileAs>Default</CompileAs>
+      <EnableEnhancedInstructionSet>
+      </EnableEnhancedInstructionSet>
+      <MultiProcessorCompilation>true</MultiProcessorCompilation>
+    </ClCompile>
+    <ResourceCompile>
+      <PreprocessorDefinitions>NDEBUG;%(PreprocessorDefinitions)</PreprocessorDefinitions>
+      <Culture>0x040b</Culture>
+    </ResourceCompile>
+    <Link>
+      <AdditionalDependencies>%(AdditionalDependencies)</AdditionalDependencies>
+      <OutputFile>$(OutDir)$(TargetName)$(TargetExt)</OutputFile>
+      <SuppressStartupBanner>true</SuppressStartupBanner>
+      <AdditionalLibraryDirectories>..\..\lib;%(AdditionalLibraryDirectories)</AdditionalLibraryDirectories>
+      <GenerateDebugInformation>false</GenerateDebugInformation>
+      <GenerateMapFile>true</GenerateMapFile>
+      <MapFileName>$(OutDir)$(TargetName).map</MapFileName>
+      <SubSystem>Console</SubSystem>
+      <DataExecutionPrevention />
+    </Link>
+    <PostBuildEvent>
+      <Command>if not exist ..\..\bin mkdir ..\..\bin
+copy $(OutDir)$(TargetName)$(TargetExt) ..\..\bin\</Command>
+    </PostBuildEvent>
+  </ItemDefinitionGroup>
+  <ItemGroup>
+    <ClCompile Include="main.cpp">
+      <Optimization Condition="'$(Configuration)|$(Platform)'=='Debug|Win32'">Disabled</Optimization>
+      <BasicRuntimeChecks Condition="'$(Configuration)|$(Platform)'=='Debug|Win32'">EnableFastChecks</BasicRuntimeChecks>
+      <BrowseInformation Condition="'$(Configuration)|$(Platform)'=='Debug|Win32'">true</BrowseInformation>
+      <Optimization Condition="'$(Configuration)|$(Platform)'=='Debug|x64'">Disabled</Optimization>
+      <BasicRuntimeChecks Condition="'$(Configuration)|$(Platform)'=='Debug|x64'">EnableFastChecks</BasicRuntimeChecks>
+      <BrowseInformation Condition="'$(Configuration)|$(Platform)'=='Debug|x64'">true</BrowseInformation>
+      <Optimization Condition="'$(Configuration)|$(Platform)'=='Release|Win32'">MaxSpeed</Optimization>
+      <Optimization Condition="'$(Configuration)|$(Platform)'=='Release|x64'">MaxSpeed</Optimization>
+    </ClCompile>
+    <ClCompile Include="RunParameters.cpp">
+      <Optimization Condition="'$(Configuration)|$(Platform)'=='Debug|Win32'">Disabled</Optimization>
+      <BasicRuntimeChecks Condition="'$(Configuration)|$(Platform)'=='Debug|Win32'">EnableFastChecks</BasicRuntimeChecks>
+      <BrowseInformation Condition="'$(Configuration)|$(Platform)'=='Debug|Win32'">true</BrowseInformation>
+      <Optimization Condition="'$(Configuration)|$(Platform)'=='Debug|x64'">Disabled</Optimization>
+      <BasicRuntimeChecks Condition="'$(Configuration)|$(Platform)'=='Debug|x64'">EnableFastChecks</BasicRuntimeChecks>
+      <BrowseInformation Condition="'$(Configuration)|$(Platform)'=='Debug|x64'">true</BrowseInformation>
+      <Optimization Condition="'$(Configuration)|$(Platform)'=='Release|Win32'">MaxSpeed</Optimization>
+      <Optimization Condition="'$(Configuration)|$(Platform)'=='Release|x64'">MaxSpeed</Optimization>
+    </ClCompile>
+    <ClCompile Include="WavFile.cpp">
+      <Optimization Condition="'$(Configuration)|$(Platform)'=='Debug|Win32'">Disabled</Optimization>
+      <BasicRuntimeChecks Condition="'$(Configuration)|$(Platform)'=='Debug|Win32'">EnableFastChecks</BasicRuntimeChecks>
+      <BrowseInformation Condition="'$(Configuration)|$(Platform)'=='Debug|Win32'">true</BrowseInformation>
+      <Optimization Condition="'$(Configuration)|$(Platform)'=='Debug|x64'">Disabled</Optimization>
+      <BasicRuntimeChecks Condition="'$(Configuration)|$(Platform)'=='Debug|x64'">EnableFastChecks</BasicRuntimeChecks>
+      <BrowseInformation Condition="'$(Configuration)|$(Platform)'=='Debug|x64'">true</BrowseInformation>
+      <Optimization Condition="'$(Configuration)|$(Platform)'=='Release|Win32'">MaxSpeed</Optimization>
+      <Optimization Condition="'$(Configuration)|$(Platform)'=='Release|x64'">MaxSpeed</Optimization>
+    </ClCompile>
+  </ItemGroup>
+  <ItemGroup>
+    <ClInclude Include="RunParameters.h" />
+    <ClInclude Include="SS_CharTypes.h" />
+    <ClInclude Include="WavFile.h" />
+  </ItemGroup>
+  <ItemGroup>
+    <ProjectReference Include="..\SoundTouch\SoundTouch.vcxproj">
+      <Project>{68a5dd20-7057-448b-8fe0-b6ac8d205509}</Project>
+      <ReferenceOutputAssembly>false</ReferenceOutputAssembly>
+    </ProjectReference>
+  </ItemGroup>
+  <Import Project="$(VCTargetsPath)\Microsoft.Cpp.targets" />
+  <ImportGroup Label="ExtensionTargets">
+  </ImportGroup>
+</Project>

source/SoundTouch/AAFilter.cpp

@@ -12,13 +12,6 @@
 ///
 ////////////////////////////////////////////////////////////////////////////////
 //
-// Last changed  : $Date$
-// File revision : $Revision: 4 $
-//
-// $Id$
-//
-////////////////////////////////////////////////////////////////////////////////
-//
 // License :
 //
 //  SoundTouch audio processing library
@@ -49,9 +42,32 @@
 
 using namespace soundtouch;
 
-#define PI        3.141592655357989
+#define PI       3.14159265358979323846
 #define TWOPI    (2 * PI)
 
+// define this to save AA filter coefficients to a file
+// #define _DEBUG_SAVE_AAFILTER_COEFFICIENTS   1
+
+#ifdef _DEBUG_SAVE_AAFILTER_COEFFICIENTS
+    #include <stdio.h>
+
+    static void _DEBUG_SAVE_AAFIR_COEFFS(SAMPLETYPE *coeffs, int len)
+    {
+        FILE *fptr = fopen("aa_filter_coeffs.txt", "wt");
+        if (fptr == nullptr) return;
+
+        for (int i = 0; i < len; i ++)
+        {
+            double temp = coeffs[i];
+            fprintf(fptr, "%lf\n", temp);
+        }
+        fclose(fptr);
+    }
+
+#else
+    #define _DEBUG_SAVE_AAFIR_COEFFS(x, y)
+#endif
+
 /*****************************************************************************
  *
  * Implementation of the class 'AAFilter'
@@ -66,14 +82,12 @@ AAFilter::AAFilter(uint len)
 }
 
 
-
 AAFilter::~AAFilter()
 {
     delete pFIR;
 }
 
 
-
 // Sets new anti-alias filter cut-off edge frequency, scaled to
 // sampling frequency (nyquist frequency = 0.5).
 // The filter will cut frequencies higher than the given frequency.
@@ -84,7 +98,6 @@ void AAFilter::setCutoffFreq(double newCutoffFreq)
 }
 
 
-
 // Sets number of FIR filter taps
 void AAFilter::setLength(uint newLength)
 {
@@ -93,13 +106,12 @@ void AAFilter::setLength(uint newLength)
 }
 
 
-
 // Calculates coefficients for a low-pass FIR filter using Hamming window
 void AAFilter::calculateCoeffs()
 {
     uint i;
     double cntTemp, temp, tempCoeff,h, w;
-    double fc2, wc;
+    double wc;
     double scaleCoeff, sum;
     double *work;
     SAMPLETYPE *coeffs;
@@ -112,8 +124,7 @@ void AAFilter::calculateCoeffs()
     work = new double[length];
     coeffs = new SAMPLETYPE[length];
 
-    fc2 = 2.0 * cutoffFreq; 
+    wc = 2.0 * PI * cutoffFreq;
-    wc = PI * fc2;
     tempCoeff = TWOPI / (double)length;
 
     sum = 0;
@@ -124,7 +135,7 @@ void AAFilter::calculateCoeffs()
         temp = cntTemp * wc;
         if (temp != 0)
         {
-            h = fc2 * sin(temp) / temp;                     // sinc function
+            h = sin(temp) / temp;                     // sinc function
         }
         else
         {
@@ -153,8 +164,8 @@ void AAFilter::calculateCoeffs()
 
     for (i = 0; i < length; i ++)
     {
-        // scale & round to nearest integer
         temp = work[i] * scaleCoeff;
+        // scale & round to nearest integer
         temp += (temp >= 0) ? 0.5 : -0.5;
         // ensure no overfloods
         assert(temp >= -32768 && temp <= 32767);
@@ -164,6 +175,8 @@ void AAFilter::calculateCoeffs()
     // Set coefficients. Use divide factor 14 => divide result by 2^14 = 16384
     pFIR->setCoefficients(coeffs, length, 14);
 
+    _DEBUG_SAVE_AAFIR_COEFFS(coeffs, length);
+
     delete[] work;
     delete[] coeffs;
 }
@@ -178,6 +191,31 @@ uint AAFilter::evaluate(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples
 }
 
 
+/// Applies the filter to the given src & dest pipes, so that processed amount of
+/// samples get removed from src, and produced amount added to dest
+/// Note : The amount of outputted samples is by value of 'filter length'
+/// smaller than the amount of input samples.
+uint AAFilter::evaluate(FIFOSampleBuffer &dest, FIFOSampleBuffer &src) const
+{
+    SAMPLETYPE *pdest;
+    const SAMPLETYPE *psrc;
+    uint numSrcSamples;
+    uint result;
+    int numChannels = src.getChannels();
+
+    assert(numChannels == dest.getChannels());
+
+    numSrcSamples = src.numSamples();
+    psrc = src.ptrBegin();
+    pdest = dest.ptrEnd(numSrcSamples);
+    result = pFIR->evaluate(pdest, psrc, numSrcSamples, numChannels);
+    src.receiveSamples(result);
+    dest.putSamples(result);
+
+    return result;
+}
+
+
 uint AAFilter::getLength() const
 {
     return pFIR->getLength();

source/SoundTouch/AAFilter.h

@@ -13,13 +13,6 @@
 ///
 ////////////////////////////////////////////////////////////////////////////////
 //
-// Last changed  : $Date$
-// File revision : $Revision: 4 $
-//
-// $Id$
-//
-////////////////////////////////////////////////////////////////////////////////
-//
 // License :
 //
 //  SoundTouch audio processing library
@@ -45,6 +38,7 @@
 #define AAFilter_H
 
 #include "STTypes.h"
+#include "FIFOSampleBuffer.h"
 
 namespace soundtouch
 {
@@ -84,6 +78,14 @@ public:
                   const SAMPLETYPE *src,
                   uint numSamples,
                   uint numChannels) const;
+
+    /// Applies the filter to the given src & dest pipes, so that processed amount of
+    /// samples get removed from src, and produced amount added to dest
+    /// Note : The amount of outputted samples is by value of 'filter length'
+    /// smaller than the amount of input samples.
+    uint evaluate(FIFOSampleBuffer &dest,
+                  FIFOSampleBuffer &src) const;
+
 };
 
 }

source/SoundTouch/BPMDetect.cpp

@@ -26,13 +26,6 @@
 ///
 ////////////////////////////////////////////////////////////////////////////////
 //
-// Last changed  : $Date$
-// File revision : $Revision: 4 $
-//
-// $Id$
-//
-////////////////////////////////////////////////////////////////////////////////
-//
 // License :
 //
 //  SoundTouch audio processing library
@@ -54,45 +47,62 @@
 //
 ////////////////////////////////////////////////////////////////////////////////
 
+#define _USE_MATH_DEFINES
+
 #include <math.h>
 #include <assert.h>
 #include <string.h>
 #include <stdio.h>
+#include <cfloat>
 #include "FIFOSampleBuffer.h"
 #include "PeakFinder.h"
 #include "BPMDetect.h"
 
 using namespace soundtouch;
 
-#define INPUT_BLOCK_SAMPLES       2048
+// algorithm input sample block size
-#define DECIMATED_BLOCK_SAMPLES   256
+static const int INPUT_BLOCK_SIZE = 2048;
 
-/// decay constant for calculating RMS volume sliding average approximation 
+// decimated sample block size
-/// (time constant is about 10 sec)
+static const int DECIMATED_BLOCK_SIZE = 256;
-const float avgdecay = 0.99986f;
 
-/// Normalization coefficient for calculating RMS sliding average approximation.
+/// Target sample rate after decimation
-const float avgnorm = (1 - avgdecay);
+static const int TARGET_SRATE = 1000;
 
+/// XCorr update sequence size, update in about 200msec chunks
+static const int XCORR_UPDATE_SEQUENCE = (int)(TARGET_SRATE / 5);
+
+/// Moving average N size
+static const int MOVING_AVERAGE_N = 15;
+
+/// XCorr decay time constant, decay to half in 30 seconds
+/// If it's desired to have the system adapt quicker to beat rate
+/// changes within a continuing music stream, then the
+/// 'xcorr_decay_time_constant' value can be reduced, yet that
+/// can increase possibility of glitches in bpm detection.
+static const double XCORR_DECAY_TIME_CONSTANT = 30.0;
+
+/// Data overlap factor for beat detection algorithm
+static const int OVERLAP_FACTOR = 4;
+
+static const double TWOPI = (2 * M_PI);
 
 ////////////////////////////////////////////////////////////////////////////////
 
 // Enable following define to create bpm analysis file:
 
-// #define _CREATE_BPM_DEBUG_FILE
+//#define _CREATE_BPM_DEBUG_FILE
 
 #ifdef _CREATE_BPM_DEBUG_FILE
 
-    #define DEBUGFILE_NAME  "c:\\temp\\soundtouch-bpm-debug.txt"
+    static void _SaveDebugData(const char *name, const float *data, int minpos, int maxpos, double coeff)
-
-    static void _SaveDebugData(const float *data, int minpos, int maxpos, double coeff)
     {
-        FILE *fptr = fopen(DEBUGFILE_NAME, "wt");
+        FILE *fptr = fopen(name, "wt");
         int i;
 
         if (fptr)
         {
-            printf("\n\nWriting BPM debug data into file " DEBUGFILE_NAME "\n\n");
+            printf("\nWriting BPM debug data into file %s\n", name);
             for (i = minpos; i < maxpos; i ++)
             {
                 fprintf(fptr, "%d\t%.1lf\t%f\n", i, coeff / (double)i, data[i]);
@@ -100,42 +110,90 @@ const float avgnorm = (1 - avgdecay);
             fclose(fptr);
         }
     }
+
+    void _SaveDebugBeatPos(const char *name, const std::vector<BEAT> &beats)
+    {
+        printf("\nWriting beat detections data into file %s\n", name);
+
+        FILE *fptr = fopen(name, "wt");
+        if (fptr)
+        {
+            for (uint i = 0; i < beats.size(); i++)
+            {
+                BEAT b = beats[i];
+                fprintf(fptr, "%lf\t%lf\n", b.pos, b.strength);
+            }
+            fclose(fptr);
+        }
+    }
 #else
-    #define _SaveDebugData(a,b,c,d)
+    #define _SaveDebugData(name, a,b,c,d)
+    #define _SaveDebugBeatPos(name, b)
 #endif
 
+// Hamming window
+void hamming(float *w, int N)
+{
+    for (int i = 0; i < N; i++)
+    {
+        w[i] = (float)(0.54 - 0.46 * cos(TWOPI * i / (N - 1)));
+    }
+
+}
+
+////////////////////////////////////////////////////////////////////////////////
+//
+// IIR2_filter - 2nd order IIR filter
+
+IIR2_filter::IIR2_filter(const double *lpf_coeffs)
+{
+    memcpy(coeffs, lpf_coeffs, 5 * sizeof(double));
+    memset(prev, 0, sizeof(prev));
+}
+
+
+float IIR2_filter::update(float x)
+{
+    prev[0] = x;
+    double y = x * coeffs[0];
+
+    for (int i = 4; i >= 1; i--)
+    {
+        y += coeffs[i] * prev[i];
+        prev[i] = prev[i - 1];
+    }
+
+    prev[3] = y;
+    return (float)y;
+}
+
+
+// IIR low-pass filter coefficients, calculated with matlab/octave cheby2(2,40,0.05)
+const double _LPF_coeffs[5] = { 0.00996655391939, -0.01944529148401, 0.00996655391939, 1.96867605796247, -0.96916387431724 };
+
 ////////////////////////////////////////////////////////////////////////////////
 
-
+BPMDetect::BPMDetect(int numChannels, int aSampleRate) :
-BPMDetect::BPMDetect(int numChannels, int aSampleRate)
+    beat_lpf(_LPF_coeffs)
 {
+    beats.reserve(250); // initial reservation to prevent frequent reallocation
+
     this->sampleRate = aSampleRate;
     this->channels = numChannels;
 
     decimateSum = 0;
     decimateCount = 0;
 
-    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;
+    decimateBy = sampleRate / TARGET_SRATE;
-    assert(decimateBy > 0);
+    if ((decimateBy <= 0) || (decimateBy * DECIMATED_BLOCK_SIZE < INPUT_BLOCK_SIZE))
-    assert(INPUT_BLOCK_SAMPLES < decimateBy * DECIMATED_BLOCK_SAMPLES);
+    {
+        ST_THROW_RT_ERROR("Too small samplerate");
+    }
 
     // Calculate window length & starting item according to desired min & max bpms
     windowLen = (60 * sampleRate) / (decimateBy * MIN_BPM);
-    windowStart = (60 * sampleRate) / (decimateBy * MAX_BPM);
+    windowStart = (60 * sampleRate) / (decimateBy * MAX_BPM_RANGE);
 
     assert(windowLen > windowStart);
 
@@ -143,23 +201,38 @@ BPMDetect::BPMDetect(int numChannels, int aSampleRate)
     xcorr = new float[windowLen];
     memset(xcorr, 0, windowLen * sizeof(float));
 
+    pos = 0;
+    peakPos = 0;
+    peakVal = 0;
+    init_scaler = 1;
+    beatcorr_ringbuffpos = 0;
+    beatcorr_ringbuff = new float[windowLen];
+    memset(beatcorr_ringbuff, 0, windowLen * sizeof(float));
+
     // allocate processing buffer
     buffer = new FIFOSampleBuffer();
     // we do processing in mono mode
     buffer->setChannels(1);
     buffer->clear();
-}
 
+    // calculate hamming windows
+    hamw = new float[XCORR_UPDATE_SEQUENCE];
+    hamming(hamw, XCORR_UPDATE_SEQUENCE);
+    hamw2 = new float[XCORR_UPDATE_SEQUENCE / 2];
+    hamming(hamw2, XCORR_UPDATE_SEQUENCE / 2);
+}
 
 
 BPMDetect::~BPMDetect()
 {
     delete[] xcorr;
+    delete[] beatcorr_ringbuff;
+    delete[] hamw;
+    delete[] hamw2;
     delete buffer;
 }
 
 
-
 /// convert to mono, low-pass filter & decimate to about 500 Hz.
 /// return number of outputted samples.
 ///
@@ -216,7 +289,6 @@ int BPMDetect::decimate(SAMPLETYPE *dest, const SAMPLETYPE *src, int numsamples)
 }
 
 
-
 // Calculates autocorrelation function of the sample history buffer
 void BPMDetect::updateXCorr(int process_samples)
 {
@@ -224,71 +296,122 @@ void BPMDetect::updateXCorr(int process_samples)
     SAMPLETYPE *pBuffer;
 
     assert(buffer->numSamples() >= (uint)(process_samples + windowLen));
+    assert(process_samples == XCORR_UPDATE_SEQUENCE);
 
     pBuffer = buffer->ptrBegin();
+
+    // calculate decay factor for xcorr filtering
+    float xcorr_decay = (float)pow(0.5, process_samples / (XCORR_DECAY_TIME_CONSTANT * TARGET_SRATE));
+
+    // prescale pbuffer
+    float tmp[XCORR_UPDATE_SEQUENCE];
+    for (int i = 0; i < process_samples; i++)
+    {
+        tmp[i] = hamw[i] * hamw[i] * pBuffer[i];
+    }
+
+    #pragma omp parallel for
     for (offs = windowStart; offs < windowLen; offs ++)
     {
-        LONG_SAMPLETYPE sum;
+        float 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
+            sum += tmp[i] * pBuffer[i + offs];  // scaling the sub-result shouldn't be necessary
         }
-//        xcorr[offs] *= xcorr_decay;   // decay 'xcorr' here with suitable coefficients 
+        xcorr[offs] *= xcorr_decay;   // decay 'xcorr' here with suitable time constant.
-                                        // 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;
+        xcorr[offs] += (float)fabs(sum);
     }
 }
 
 
-// Calculates envelope of the sample data
+// Detect individual beat positions
-void BPMDetect::calcEnvelope(SAMPLETYPE *samples, int numsamples) 
+void BPMDetect::updateBeatPos(int process_samples)
 {
-    const static double decay = 0.7f;               // decay constant for smoothing the envelope
+    SAMPLETYPE *pBuffer;
-    const static double norm = (1 - decay);
 
-    int i;
+    assert(buffer->numSamples() >= (uint)(process_samples + windowLen));
-    LONG_SAMPLETYPE out;
-    double val;
 
-    for (i = 0; i < numsamples; i ++) 
+    pBuffer = buffer->ptrBegin();
+    assert(process_samples == XCORR_UPDATE_SEQUENCE / 2);
+
+    //    static double thr = 0.0003;
+    double posScale = (double)this->decimateBy / (double)this->sampleRate;
+    int resetDur = (int)(0.12 / posScale + 0.5);
+
+    // prescale pbuffer
+    float tmp[XCORR_UPDATE_SEQUENCE / 2];
+    for (int i = 0; i < process_samples; i++)
     {
-        // calc average RMS volume
+        tmp[i] = hamw2[i] * hamw2[i] * pBuffer[i];
-        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
+    #pragma omp parallel for
-        envelopeAccu *= decay;
+    for (int offs = windowStart; offs < windowLen; offs++)
-        envelopeAccu += val;
+    {
-        out = (LONG_SAMPLETYPE)(envelopeAccu * norm);
+        float sum = 0;
+        for (int i = 0; i < process_samples; i++)
+        {
+            sum += tmp[i] * pBuffer[offs + i];
+        }
+        beatcorr_ringbuff[(beatcorr_ringbuffpos + offs) % windowLen] += (float)((sum > 0) ? sum : 0); // accumulate only positive correlations
+    }
 
-#ifdef SOUNDTOUCH_INTEGER_SAMPLES
+    int skipstep = XCORR_UPDATE_SEQUENCE / OVERLAP_FACTOR;
-        // cut peaks (shouldn't be necessary though)
+
-        if (out > 32767) out = 32767;
+    // compensate empty buffer at beginning by scaling coefficient
-#endif // SOUNDTOUCH_INTEGER_SAMPLES
+    float scale = (float)windowLen / (float)(skipstep * init_scaler);
-        samples[i] = (SAMPLETYPE)out;
+    if (scale > 1.0f)
+    {
+        init_scaler++;
+    }
+    else
+    {
+        scale = 1.0f;
+    }
+
+    // detect beats
+    for (int i = 0; i < skipstep; i++)
+    {
+        float sum = beatcorr_ringbuff[beatcorr_ringbuffpos];
+        sum -= beat_lpf.update(sum);
+
+        if (sum > peakVal)
+        {
+            // found new local largest value
+            peakVal = sum;
+            peakPos = pos;
+        }
+        if (pos > peakPos + resetDur)
+        {
+            // largest value not updated for 200msec => accept as beat
+            peakPos += skipstep;
+            if (peakVal > 0)
+            {
+                // add detected beat to end of "beats" vector
+                BEAT temp = { (float)(peakPos * posScale), (float)(peakVal * scale) };
+                beats.push_back(temp);
+            }
+
+            peakVal = 0;
+            peakPos = pos;
+        }
+
+        beatcorr_ringbuff[beatcorr_ringbuffpos] = 0;
+        pos++;
+        beatcorr_ringbuffpos = (beatcorr_ringbuffpos + 1) % windowLen;
     }
 }
 
 
+#define max(x,y) ((x) > (y) ? (x) : (y))
 
 void BPMDetect::inputSamples(const SAMPLETYPE *samples, int numSamples)
 {
-    SAMPLETYPE decimated[DECIMATED_BLOCK_SAMPLES];
+    SAMPLETYPE decimated[DECIMATED_BLOCK_SIZE];
 
     // iterate so that max INPUT_BLOCK_SAMPLES processed per iteration
     while (numSamples > 0)
@@ -296,48 +419,70 @@ void BPMDetect::inputSamples(const SAMPLETYPE *samples, int numSamples)
         int block;
         int decSamples;
 
-        block = (numSamples > INPUT_BLOCK_SAMPLES) ? INPUT_BLOCK_SAMPLES : numSamples;
+        block = (numSamples > INPUT_BLOCK_SIZE) ? INPUT_BLOCK_SIZE : numSamples;
 
         // decimate. note that converts to mono at the same time
         decSamples = decimate(decimated, samples, block);
         samples += block * channels;
         numSamples -= block;
 
-        // envelope new samples and add them to buffer
-        calcEnvelope(decimated, decSamples);
         buffer->putSamples(decimated, decSamples);
     }
 
-    // when the buffer has enought samples for processing...
+    // when the buffer has enough samples for processing...
-    if ((int)buffer->numSamples() > windowLen) 
+    int req = max(windowLen + XCORR_UPDATE_SEQUENCE, 2 * XCORR_UPDATE_SEQUENCE);
+    while ((int)buffer->numSamples() >= req)
     {
-        int processLength;
+        // ... update autocorrelations...
-
+        updateXCorr(XCORR_UPDATE_SEQUENCE);
-        // how many samples are processed
+        // ...update beat position calculation...
-        processLength = (int)buffer->numSamples() - windowLen;
+        updateBeatPos(XCORR_UPDATE_SEQUENCE / 2);
-
+        // ... and remove proceessed samples from the buffer
-        // ... calculate autocorrelations for oldest samples...
+        int n = XCORR_UPDATE_SEQUENCE / OVERLAP_FACTOR;
-        updateXCorr(processLength);
+        buffer->receiveSamples(n);
-        // ... and remove them from the buffer
-        buffer->receiveSamples(processLength);
     }
 }
 
 
-
 void BPMDetect::removeBias()
 {
     int i;
-    float minval = 1e12f;   // arbitrary large number
 
+    // Remove linear bias: calculate linear regression coefficient
+    // 1. calc mean of 'xcorr' and 'i'
+    double mean_i = 0;
+    double mean_x = 0;
+    for (i = windowStart; i < windowLen; i++)
+    {
+        mean_x += xcorr[i];
+    }
+    mean_x /= (windowLen - windowStart);
+    mean_i = 0.5 * (windowLen - 1 + windowStart);
+
+    // 2. calculate linear regression coefficient
+    double b = 0;
+    double div = 0;
+    for (i = windowStart; i < windowLen; i++)
+    {
+        double xt = xcorr[i] - mean_x;
+        double xi = i - mean_i;
+        b += xt * xi;
+        div += xi * xi;
+    }
+    b /= div;
+
+    // subtract linear regression and resolve min. value bias
+    float minval = FLT_MAX;   // arbitrary large number
     for (i = windowStart; i < windowLen; i ++)
     {
+        xcorr[i] -= (float)(b * i);
         if (xcorr[i] < minval)
         {
             minval = xcorr[i];
         }
     }
 
+    // subtract min.value
     for (i = windowStart; i < windowLen; i ++)
     {
         xcorr[i] -= minval;
@@ -345,26 +490,82 @@ void BPMDetect::removeBias()
 }
 
 
+// Calculate N-point moving average for "source" values
+void MAFilter(float *dest, const float *source, int start, int end, int N)
+{
+    for (int i = start; i < end; i++)
+    {
+        int i1 = i - N / 2;
+        int i2 = i + N / 2 + 1;
+        if (i1 < start) i1 = start;
+        if (i2 > end)   i2 = end;
+
+        double sum = 0;
+        for (int j = i1; j < i2; j ++)
+        {
+            sum += source[j];
+        }
+        dest[i] = (float)(sum / (i2 - i1));
+    }
+}
+
+
 float BPMDetect::getBpm()
 {
     double peakPos;
     double coeff;
     PeakFinder peakFinder;
 
-    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();
 
+    coeff = 60.0 * ((double)sampleRate / (double)decimateBy);
+
+    // save bpm debug data if debug data writing enabled
+    _SaveDebugData("soundtouch-bpm-xcorr.txt", xcorr, windowStart, windowLen, coeff);
+
+    // Smoothen by N-point moving-average
+    float *data = new float[windowLen];
+    memset(data, 0, sizeof(float) * windowLen);
+    MAFilter(data, xcorr, windowStart, windowLen, MOVING_AVERAGE_N);
+
     // find peak position
-    peakPos = peakFinder.detectPeak(xcorr, windowStart, windowLen);
+    peakPos = peakFinder.detectPeak(data, windowStart, windowLen);
+
+    // save bpm debug data if debug data writing enabled
+    _SaveDebugData("soundtouch-bpm-smoothed.txt", data, windowStart, windowLen, coeff);
+
+    delete[] data;
 
     assert(decimateBy != 0);
     if (peakPos < 1e-9) return 0.0; // detection failed.
 
+    _SaveDebugBeatPos("soundtouch-detected-beats.txt", beats);
+
     // calculate BPM
-    return (float) (coeff / peakPos);
+    float bpm = (float)(coeff / peakPos);
+    return (bpm >= MIN_BPM && bpm <= MAX_BPM_VALID) ? bpm : 0;
+}
+
+
+/// Get beat position arrays. Note: The array includes also really low beat detection values
+/// in absence of clear strong beats. Consumer may wish to filter low values away.
+/// - "pos" receive array of beat positions
+/// - "values" receive array of beat detection strengths
+/// - max_num indicates max.size of "pos" and "values" array.
+///
+/// You can query a suitable array sized by calling this with nullptr in "pos" & "values".
+///
+/// \return number of beats in the arrays.
+int BPMDetect::getBeats(float *pos, float *values, int max_num)
+{
+    int num = (int)beats.size();
+    if ((!pos) || (!values)) return num;    // pos or values nullptr, return just size
+
+    for (int i = 0; (i < num) && (i < max_num); i++)
+    {
+        pos[i] = beats[i].pos;
+        values[i] = beats[i].strength;
+    }
+    return num;
 }

source/SoundTouch/FIFOSampleBuffer.cpp

@@ -15,13 +15,6 @@
 ///
 ////////////////////////////////////////////////////////////////////////////////
 //
-// Last changed  : $Date$
-// File revision : $Revision: 4 $
-//
-// $Id$
-//
-////////////////////////////////////////////////////////////////////////////////
-//
 // License :
 //
 //  SoundTouch audio processing library
@@ -57,8 +50,8 @@ FIFOSampleBuffer::FIFOSampleBuffer(int numChannels)
 {
     assert(numChannels > 0);
     sizeInBytes = 0; // reasonable initial value
-    buffer = NULL;
+    buffer = nullptr;
-    bufferUnaligned = NULL;
+    bufferUnaligned = nullptr;
     samplesInBuffer = 0;
     bufferPos = 0;
     channels = (uint)numChannels;
@@ -70,8 +63,8 @@ FIFOSampleBuffer::FIFOSampleBuffer(int numChannels)
 FIFOSampleBuffer::~FIFOSampleBuffer()
 {
     delete[] bufferUnaligned;
-    bufferUnaligned = NULL;
+    bufferUnaligned = nullptr;
-    buffer = NULL;
+    buffer = nullptr;
 }
 
 
@@ -80,7 +73,8 @@ void FIFOSampleBuffer::setChannels(int numChannels)
 {
     uint usedBytes;
 
-    assert(numChannels > 0);
+    if (!verifyNumberOfChannels(numChannels)) return;
+
     usedBytes = channels * samplesInBuffer;
     channels = (uint)numChannels;
     samplesInBuffer = usedBytes / channels;
@@ -131,7 +125,7 @@ void FIFOSampleBuffer::putSamples(uint nSamples)
 //
 // Parameter 'slackCapacity' tells the function how much free capacity (in
 // terms of samples) there _at least_ should be, in order to the caller to
-// succesfully insert all the required samples to the buffer. When necessary, 
+// successfully insert all the required samples to the buffer. When necessary,
 // the function grows the buffer size to comply with this requirement.
 //
 // When using this function as means for inserting new samples, also remember
@@ -158,7 +152,7 @@ SAMPLETYPE *FIFOSampleBuffer::ptrBegin()
 }
 
 
-// Ensures that the buffer has enought capacity, i.e. space for _at least_
+// Ensures that the buffer has enough capacity, i.e. space for _at least_
 // 'capacityRequirement' number of samples. The buffer is grown in steps of
 // 4 kilobytes to eliminate the need for frequently growing up the buffer,
 // as well as to round the buffer size up to the virtual memory page size.
@@ -172,12 +166,12 @@ void FIFOSampleBuffer::ensureCapacity(uint capacityRequirement)
         sizeInBytes = (capacityRequirement * channels * sizeof(SAMPLETYPE) + 4095) & (uint)-4096;
         assert(sizeInBytes % 2 == 0);
         tempUnaligned = new SAMPLETYPE[sizeInBytes / sizeof(SAMPLETYPE) + 16 / sizeof(SAMPLETYPE)];
-        if (tempUnaligned == NULL)
+        if (tempUnaligned == nullptr)
         {
             ST_THROW_RT_ERROR("Couldn't allocate memory!\n");
         }
         // Align the buffer to begin at 16byte cache line boundary for optimal performance
-        temp = (SAMPLETYPE *)(((ulong)tempUnaligned + 15) & (ulong)-16);
+        temp = (SAMPLETYPE *)SOUNDTOUCH_ALIGN_POINTER_16(tempUnaligned);
         if (samplesInBuffer)
         {
             memcpy(temp, ptrBegin(), samplesInBuffer * channels * sizeof(SAMPLETYPE));
@@ -272,3 +266,10 @@ uint FIFOSampleBuffer::adjustAmountOfSamples(uint numSamples)
     return samplesInBuffer;
 }
 
+
+/// Add silence to end of buffer
+void FIFOSampleBuffer::addSilent(uint nSamples)
+{
+    memset(ptrEnd(nSamples), 0, sizeof(SAMPLETYPE) * nSamples * channels);
+    samplesInBuffer += nSamples;
+}

source/SoundTouch/FIRFilter.cpp

@@ -2,22 +2,21 @@
 ///
 /// General FIR digital filter routines with MMX optimization.
 ///
-/// Note : MMX optimized functions reside in a separate, platform-specific file, 
+/// Notes : MMX optimized functions reside in a separate, platform-specific file,
 /// e.g. 'mmx_win.cpp' or 'mmx_gcc.cpp'
 ///
+/// This source file contains OpenMP optimizations that allow speeding up the
+/// corss-correlation algorithm by executing it in several threads / CPU cores
+/// in parallel. See the following article link for more detailed discussion
+/// about SoundTouch OpenMP optimizations:
+/// http://www.softwarecoven.com/parallel-computing-in-embedded-mobile-devices
+///
 /// Author        : Copyright (c) Olli Parviainen
 /// Author e-mail : oparviai 'at' iki.fi
 /// SoundTouch WWW: http://www.surina.net/soundtouch
 ///
 ////////////////////////////////////////////////////////////////////////////////
 //
-// Last changed  : $Date$
-// File revision : $Revision: 4 $
-//
-// $Id$
-//
-////////////////////////////////////////////////////////////////////////////////
-//
 // License :
 //
 //  SoundTouch audio processing library
@@ -57,54 +56,45 @@ using namespace soundtouch;
 FIRFilter::FIRFilter()
 {
     resultDivFactor = 0;
-    resultDivider = 0;
     length = 0;
     lengthDiv8 = 0;
-    filterCoeffs = NULL;
+    filterCoeffs = nullptr;
+    filterCoeffsStereo = nullptr;
 }
 
 
 FIRFilter::~FIRFilter()
 {
     delete[] filterCoeffs;
+    delete[] filterCoeffsStereo;
 }
 
+
 // Usual C-version of the filter routine for stereo sound
 uint FIRFilter::evaluateFilterStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples) const
 {
-    uint i, j, end;
+    int j, end;
-    LONG_SAMPLETYPE suml, sumr;
+    // hint compiler autovectorization that loop length is divisible by 8
-#ifdef SOUNDTOUCH_FLOAT_SAMPLES
+    uint ilength = length & -8;
-    // 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((length != 0) && (length == ilength) && (src != nullptr) && (dest != nullptr) && (filterCoeffs != nullptr));
-    assert(src != NULL);
+    assert(numSamples > ilength);
-    assert(dest != NULL);
-    assert(filterCoeffs != NULL);
 
-    end = 2 * (numSamples - length);
+    end = 2 * (numSamples - ilength);
 
+    #pragma omp parallel for
     for (j = 0; j < end; j += 2)
     {
         const SAMPLETYPE *ptr;
+        LONG_SAMPLETYPE suml, sumr;
 
         suml = sumr = 0;
         ptr = src + j;
 
-        for (i = 0; i < length; i += 4) 
+        for (uint i = 0; i < ilength; i ++)
         {
-            // loop is unrolled by factor of 4 here for efficiency
+            suml += ptr[2 * i] * filterCoeffsStereo[2 * i];
-            suml += ptr[2 * i + 0] * filterCoeffs[i + 0] +
+            sumr += ptr[2 * i + 1] * filterCoeffsStereo[2 * i + 1];
-                    ptr[2 * i + 2] * filterCoeffs[i + 1] +
-                    ptr[2 * i + 4] * filterCoeffs[i + 2] +
-                    ptr[2 * i + 6] * filterCoeffs[i + 3];
-            sumr += ptr[2 * i + 1] * filterCoeffs[i + 0] +
-                    ptr[2 * i + 3] * filterCoeffs[i + 1] +
-                    ptr[2 * i + 5] * filterCoeffs[i + 2] +
-                    ptr[2 * i + 7] * filterCoeffs[i + 3];
         }
 
 #ifdef SOUNDTOUCH_INTEGER_SAMPLES
@@ -114,59 +104,100 @@ uint FIRFilter::evaluateFilterStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, ui
         suml = (suml < -32768) ? -32768 : (suml > 32767) ? 32767 : suml;
         // saturate to 16 bit integer limits
         sumr = (sumr < -32768) ? -32768 : (sumr > 32767) ? 32767 : sumr;
-#else
-        suml *= dScaler;
-        sumr *= dScaler;
 #endif // SOUNDTOUCH_INTEGER_SAMPLES
         dest[j] = (SAMPLETYPE)suml;
         dest[j + 1] = (SAMPLETYPE)sumr;
     }
-    return numSamples - length;
+    return numSamples - ilength;
 }
 
 
-
-
 // Usual C-version of the filter routine for mono sound
 uint FIRFilter::evaluateFilterMono(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples) const
 {
-    uint i, j, end;
+    int j, end;
-    LONG_SAMPLETYPE sum;
-#ifdef SOUNDTOUCH_FLOAT_SAMPLES
-    // when using floating point samples, use a scaler instead of a divider
-    // because division is much slower operation than multiplying.
-    double dScaler = 1.0 / (double)resultDivider;
-#endif
 
+    // hint compiler autovectorization that loop length is divisible by 8
+    int ilength = length & -8;
 
-    assert(length != 0);
+    assert(ilength != 0);
 
-    end = numSamples - length;
+    end = numSamples - ilength;
+    #pragma omp parallel for
     for (j = 0; j < end; j ++)
     {
+        const SAMPLETYPE *pSrc = src + j;
+        LONG_SAMPLETYPE sum;
+        int i;
+
         sum = 0;
-        for (i = 0; i < length; i += 4) 
+        for (i = 0; i < ilength; i ++)
         {
-            // loop is unrolled by factor of 4 here for efficiency
+            sum += pSrc[i] * filterCoeffs[i];
-            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 SOUNDTOUCH_INTEGER_SAMPLES
         sum >>= resultDivFactor;
         // saturate to 16 bit integer limits
         sum = (sum < -32768) ? -32768 : (sum > 32767) ? 32767 : sum;
-#else
-        sum *= dScaler;
 #endif // SOUNDTOUCH_INTEGER_SAMPLES
         dest[j] = (SAMPLETYPE)sum;
-        src ++;
     }
     return end;
 }
 
 
+uint FIRFilter::evaluateFilterMulti(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples, uint numChannels)
+{
+    int j, end;
+
+    assert(length != 0);
+    assert(src != nullptr);
+    assert(dest != nullptr);
+    assert(filterCoeffs != nullptr);
+    assert(numChannels <= SOUNDTOUCH_MAX_CHANNELS);
+
+    // hint compiler autovectorization that loop length is divisible by 8
+    int ilength = length & -8;
+
+    end = numChannels * (numSamples - ilength);
+
+    #pragma omp parallel for
+    for (j = 0; j < end; j += numChannels)
+    {
+        const SAMPLETYPE *ptr;
+        LONG_SAMPLETYPE sums[16];
+        uint c;
+        int i;
+
+        for (c = 0; c < numChannels; c ++)
+        {
+            sums[c] = 0;
+        }
+
+        ptr = src + j;
+
+        for (i = 0; i < ilength; i ++)
+        {
+            SAMPLETYPE coef=filterCoeffs[i];
+            for (c = 0; c < numChannels; c ++)
+            {
+                sums[c] += ptr[0] * coef;
+                ptr ++;
+            }
+        }
+
+        for (c = 0; c < numChannels; c ++)
+        {
+#ifdef SOUNDTOUCH_INTEGER_SAMPLES
+            sums[c] >>= resultDivFactor;
+#endif // SOUNDTOUCH_INTEGER_SAMPLES
+            dest[j+c] = (SAMPLETYPE)sums[c];
+        }
+    }
+    return numSamples - ilength;
+}
+
+
 // Set filter coeffiecients and length.
 //
 // Throws an exception if filter length isn't divisible by 8
@@ -180,11 +211,27 @@ void FIRFilter::setCoefficients(const SAMPLETYPE *coeffs, uint newLength, uint u
     assert(length == newLength);
 
     resultDivFactor = uResultDivFactor;
-    resultDivider = (SAMPLETYPE)::pow(2.0, (int)resultDivFactor);
 
     delete[] filterCoeffs;
     filterCoeffs = new SAMPLETYPE[length];
-    memcpy(filterCoeffs, coeffs, length * sizeof(SAMPLETYPE));
+    delete[] filterCoeffsStereo;
+    filterCoeffsStereo = new SAMPLETYPE[length*2];
+
+#ifdef SOUNDTOUCH_FLOAT_SAMPLES
+    // scale coefficients already here if using floating samples
+    const double scale = ::pow(0.5, (int)resultDivFactor);;
+#else
+    const short scale = 1;
+#endif
+
+    for (uint i = 0; i < length; i ++)
+    {
+        filterCoeffs[i] = (SAMPLETYPE)(coeffs[i] * scale);
+        // create also stereo set of filter coefficients: this allows compiler
+        // to autovectorize filter evaluation much more efficiently
+        filterCoeffsStereo[2 * i] = (SAMPLETYPE)(coeffs[i] * scale);
+        filterCoeffsStereo[2 * i + 1] = (SAMPLETYPE)(coeffs[i] * scale);
+    }
 }
 
 
@@ -194,31 +241,38 @@ uint FIRFilter::getLength() const
 }
 
 
-
 // Applies the filter to the given sequence of samples.
 //
 // Note : The amount of outputted samples is by value of 'filter_length'
 // smaller than the amount of input samples.
-uint FIRFilter::evaluate(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples, uint numChannels) const
+uint FIRFilter::evaluate(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples, uint numChannels)
 {
-    assert(numChannels == 1 || numChannels == 2);
-
     assert(length > 0);
     assert(lengthDiv8 * 8 == length);
+
     if (numSamples < length) return 0;
-    if (numChannels == 2) 
+
+#ifndef USE_MULTICH_ALWAYS
+    if (numChannels == 1)
+    {
+        return evaluateFilterMono(dest, src, numSamples);
+    }
+    else if (numChannels == 2)
     {
         return evaluateFilterStereo(dest, src, numSamples);
-    } else {
+    }
-        return evaluateFilterMono(dest, src, numSamples);
+    else
+#endif // USE_MULTICH_ALWAYS
+    {
+        assert(numChannels > 0);
+        return evaluateFilterMulti(dest, src, numSamples, numChannels);
     }
 }
 
 
-
 // Operator 'new' is overloaded so that it automatically creates a suitable instance
 // depending on if we've a MMX-capable CPU available or not.
-void * FIRFilter::operator new(size_t s)
+void * FIRFilter::operator new(size_t)
 {
     // Notice! don't use "new FIRFilter" directly, use "newInstance" to create a new instance instead!
     ST_THROW_RT_ERROR("Error in FIRFilter::new: Don't use 'new FIRFilter', use 'newInstance' member instead!");
@@ -231,6 +285,7 @@ FIRFilter * FIRFilter::newInstance()
     uint uExtensions;
 
     uExtensions = detectCPUextensions();
+    (void)uExtensions;
 
     // Check if MMX/SSE instruction set extensions supported by CPU
 

source/SoundTouch/FIRFilter.h

@@ -11,13 +11,6 @@
 ///
 ////////////////////////////////////////////////////////////////////////////////
 //
-// Last changed  : $Date$
-// File revision : $Revision: 4 $
-//
-// $Id$
-//
-////////////////////////////////////////////////////////////////////////////////
-//
 // License :
 //
 //  SoundTouch audio processing library
@@ -59,11 +52,9 @@ protected:
     // Result divider factor in 2^k format
     uint resultDivFactor;
 
-    // Result divider value.
-    SAMPLETYPE resultDivider;
-
     // Memory for filter coefficients
     SAMPLETYPE *filterCoeffs;
+    SAMPLETYPE *filterCoeffsStereo;
 
     virtual uint evaluateFilterStereo(SAMPLETYPE *dest,
                                       const SAMPLETYPE *src,
@@ -71,6 +62,7 @@ protected:
     virtual uint evaluateFilterMono(SAMPLETYPE *dest,
                                     const SAMPLETYPE *src,
                                     uint numSamples) const;
+    virtual uint evaluateFilterMulti(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples, uint numChannels);
 
 public:
     FIRFilter();
@@ -90,7 +82,7 @@ public:
     uint evaluate(SAMPLETYPE *dest,
                   const SAMPLETYPE *src,
                   uint numSamples,
-                  uint numChannels) const;
+                  uint numChannels);
 
     uint getLength() const;
 
@@ -111,12 +103,12 @@ public:
         short *filterCoeffsUnalign;
         short *filterCoeffsAlign;
 
-        virtual uint evaluateFilterStereo(short *dest, const short *src, uint numSamples) const;
+        virtual uint evaluateFilterStereo(short *dest, const short *src, uint numSamples) const override;
     public:
         FIRFilterMMX();
         ~FIRFilterMMX();
 
-        virtual void setCoefficients(const short *coeffs, uint newLength, uint uResultDivFactor);
+        virtual void setCoefficients(const short *coeffs, uint newLength, uint uResultDivFactor) override;
     };
 
 #endif // SOUNDTOUCH_ALLOW_MMX
@@ -130,12 +122,12 @@ public:
         float *filterCoeffsUnalign;
         float *filterCoeffsAlign;
 
-        virtual uint evaluateFilterStereo(float *dest, const float *src, uint numSamples) const;
+        virtual uint evaluateFilterStereo(float *dest, const float *src, uint numSamples) const override;
     public:
         FIRFilterSSE();
         ~FIRFilterSSE();
 
-        virtual void setCoefficients(const float *coeffs, uint newLength, uint uResultDivFactor);
+        virtual void setCoefficients(const float *coeffs, uint newLength, uint uResultDivFactor) override;
     };
 
 #endif // SOUNDTOUCH_ALLOW_SSE

source/SoundTouch/InterpolateCubic.cpp

@@ -0,0 +1,196 @@
+////////////////////////////////////////////////////////////////////////////////
+///
+/// Cubic interpolation routine.
+///
+/// Author        : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+//  SoundTouch audio processing library
+//  Copyright (c) Olli Parviainen
+//
+//  This library is free software; you can redistribute it and/or
+//  modify it under the terms of the GNU Lesser General Public
+//  License as published by the Free Software Foundation; either
+//  version 2.1 of the License, or (at your option) any later version.
+//
+//  This library is distributed in the hope that it will be useful,
+//  but WITHOUT ANY WARRANTY; without even the implied warranty of
+//  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+//  Lesser General Public License for more details.
+//
+//  You should have received a copy of the GNU Lesser General Public
+//  License along with this library; if not, write to the Free Software
+//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#include <stddef.h>
+#include <math.h>
+#include "InterpolateCubic.h"
+#include "STTypes.h"
+
+using namespace soundtouch;
+
+// cubic interpolation coefficients
+static const float _coeffs[]=
+{ -0.5f,  1.0f, -0.5f, 0.0f,
+   1.5f, -2.5f,  0.0f, 1.0f,
+  -1.5f,  2.0f,  0.5f, 0.0f,
+   0.5f, -0.5f,  0.0f, 0.0f};
+
+
+InterpolateCubic::InterpolateCubic()
+{
+    fract = 0;
+}
+
+
+void InterpolateCubic::resetRegisters()
+{
+    fract = 0;
+}
+
+
+/// Transpose mono audio. Returns number of produced output samples, and
+/// updates "srcSamples" to amount of consumed source samples
+int InterpolateCubic::transposeMono(SAMPLETYPE *pdest,
+                    const SAMPLETYPE *psrc,
+                    int &srcSamples)
+{
+    int i;
+    int srcSampleEnd = srcSamples - 4;
+    int srcCount = 0;
+
+    i = 0;
+    while (srcCount < srcSampleEnd)
+    {
+        float out;
+        const float x3 = 1.0f;
+        const float x2 = (float)fract;    // x
+        const float x1 = x2*x2;           // x^2
+        const float x0 = x1*x2;           // x^3
+        float y0, y1, y2, y3;
+
+        assert(fract < 1.0);
+
+        y0 =  _coeffs[0] * x0 +  _coeffs[1] * x1 +  _coeffs[2] * x2 +  _coeffs[3] * x3;
+        y1 =  _coeffs[4] * x0 +  _coeffs[5] * x1 +  _coeffs[6] * x2 +  _coeffs[7] * x3;
+        y2 =  _coeffs[8] * x0 +  _coeffs[9] * x1 + _coeffs[10] * x2 + _coeffs[11] * x3;
+        y3 = _coeffs[12] * x0 + _coeffs[13] * x1 + _coeffs[14] * x2 + _coeffs[15] * x3;
+
+        out = y0 * psrc[0] + y1 * psrc[1] + y2 * psrc[2] + y3 * psrc[3];
+
+        pdest[i] = (SAMPLETYPE)out;
+        i ++;
+
+        // update position fraction
+        fract += rate;
+        // update whole positions
+        int whole = (int)fract;
+        fract -= whole;
+        psrc += whole;
+        srcCount += whole;
+    }
+    srcSamples = srcCount;
+    return i;
+}
+
+
+/// Transpose stereo audio. Returns number of produced output samples, and
+/// updates "srcSamples" to amount of consumed source samples
+int InterpolateCubic::transposeStereo(SAMPLETYPE *pdest,
+                    const SAMPLETYPE *psrc,
+                    int &srcSamples)
+{
+    int i;
+    int srcSampleEnd = srcSamples - 4;
+    int srcCount = 0;
+
+    i = 0;
+    while (srcCount < srcSampleEnd)
+    {
+        const float x3 = 1.0f;
+        const float x2 = (float)fract;    // x
+        const float x1 = x2*x2;           // x^2
+        const float x0 = x1*x2;           // x^3
+        float y0, y1, y2, y3;
+        float out0, out1;
+
+        assert(fract < 1.0);
+
+        y0 =  _coeffs[0] * x0 +  _coeffs[1] * x1 +  _coeffs[2] * x2 +  _coeffs[3] * x3;
+        y1 =  _coeffs[4] * x0 +  _coeffs[5] * x1 +  _coeffs[6] * x2 +  _coeffs[7] * x3;
+        y2 =  _coeffs[8] * x0 +  _coeffs[9] * x1 + _coeffs[10] * x2 + _coeffs[11] * x3;
+        y3 = _coeffs[12] * x0 + _coeffs[13] * x1 + _coeffs[14] * x2 + _coeffs[15] * x3;
+
+        out0 = y0 * psrc[0] + y1 * psrc[2] + y2 * psrc[4] + y3 * psrc[6];
+        out1 = y0 * psrc[1] + y1 * psrc[3] + y2 * psrc[5] + y3 * psrc[7];
+
+        pdest[2*i]   = (SAMPLETYPE)out0;
+        pdest[2*i+1] = (SAMPLETYPE)out1;
+        i ++;
+
+        // update position fraction
+        fract += rate;
+        // update whole positions
+        int whole = (int)fract;
+        fract -= whole;
+        psrc += 2*whole;
+        srcCount += whole;
+    }
+    srcSamples = srcCount;
+    return i;
+}
+
+
+/// Transpose multi-channel audio. Returns number of produced output samples, and
+/// updates "srcSamples" to amount of consumed source samples
+int InterpolateCubic::transposeMulti(SAMPLETYPE *pdest,
+                    const SAMPLETYPE *psrc,
+                    int &srcSamples)
+{
+    int i;
+    int srcSampleEnd = srcSamples - 4;
+    int srcCount = 0;
+
+    i = 0;
+    while (srcCount < srcSampleEnd)
+    {
+        const float x3 = 1.0f;
+        const float x2 = (float)fract;    // x
+        const float x1 = x2*x2;           // x^2
+        const float x0 = x1*x2;           // x^3
+        float y0, y1, y2, y3;
+
+        assert(fract < 1.0);
+
+        y0 =  _coeffs[0] * x0 +  _coeffs[1] * x1 +  _coeffs[2] * x2 +  _coeffs[3] * x3;
+        y1 =  _coeffs[4] * x0 +  _coeffs[5] * x1 +  _coeffs[6] * x2 +  _coeffs[7] * x3;
+        y2 =  _coeffs[8] * x0 +  _coeffs[9] * x1 + _coeffs[10] * x2 + _coeffs[11] * x3;
+        y3 = _coeffs[12] * x0 + _coeffs[13] * x1 + _coeffs[14] * x2 + _coeffs[15] * x3;
+
+        for (int c = 0; c < numChannels; c ++)
+        {
+            float out;
+            out = y0 * psrc[c] + y1 * psrc[c + numChannels] + y2 * psrc[c + 2 * numChannels] + y3 * psrc[c + 3 * numChannels];
+            pdest[0] = (SAMPLETYPE)out;
+            pdest ++;
+        }
+        i ++;
+
+        // update position fraction
+        fract += rate;
+        // update whole positions
+        int whole = (int)fract;
+        fract -= whole;
+        psrc += numChannels*whole;
+        srcCount += whole;
+    }
+    srcSamples = srcCount;
+    return i;
+}

source/SoundTouch/InterpolateCubic.h

@@ -0,0 +1,69 @@
+////////////////////////////////////////////////////////////////////////////////
+///
+/// Cubic interpolation routine.
+///
+/// Author        : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+//  SoundTouch audio processing library
+//  Copyright (c) Olli Parviainen
+//
+//  This library is free software; you can redistribute it and/or
+//  modify it under the terms of the GNU Lesser General Public
+//  License as published by the Free Software Foundation; either
+//  version 2.1 of the License, or (at your option) any later version.
+//
+//  This library is distributed in the hope that it will be useful,
+//  but WITHOUT ANY WARRANTY; without even the implied warranty of
+//  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+//  Lesser General Public License for more details.
+//
+//  You should have received a copy of the GNU Lesser General Public
+//  License along with this library; if not, write to the Free Software
+//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#ifndef _InterpolateCubic_H_
+#define _InterpolateCubic_H_
+
+#include "RateTransposer.h"
+#include "STTypes.h"
+
+namespace soundtouch
+{
+
+class InterpolateCubic : public TransposerBase
+{
+protected:
+    virtual int transposeMono(SAMPLETYPE *dest,
+                        const SAMPLETYPE *src,
+                        int &srcSamples) override;
+    virtual int transposeStereo(SAMPLETYPE *dest,
+                        const SAMPLETYPE *src,
+                        int &srcSamples) override;
+    virtual int transposeMulti(SAMPLETYPE *dest,
+                        const SAMPLETYPE *src,
+                        int &srcSamples) override;
+
+    double fract;
+
+public:
+    InterpolateCubic();
+
+    virtual void resetRegisters() override;
+
+    virtual int getLatency() const override
+    {
+        return 1;
+    }
+};
+
+}
+
+#endif

source/SoundTouch/InterpolateLinear.cpp

@@ -0,0 +1,296 @@
+////////////////////////////////////////////////////////////////////////////////
+///
+/// Linear interpolation algorithm.
+///
+/// Author        : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+//  SoundTouch audio processing library
+//  Copyright (c) Olli Parviainen
+//
+//  This library is free software; you can redistribute it and/or
+//  modify it under the terms of the GNU Lesser General Public
+//  License as published by the Free Software Foundation; either
+//  version 2.1 of the License, or (at your option) any later version.
+//
+//  This library is distributed in the hope that it will be useful,
+//  but WITHOUT ANY WARRANTY; without even the implied warranty of
+//  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+//  Lesser General Public License for more details.
+//
+//  You should have received a copy of the GNU Lesser General Public
+//  License along with this library; if not, write to the Free Software
+//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#include <assert.h>
+#include <stdlib.h>
+#include "InterpolateLinear.h"
+
+using namespace soundtouch;
+
+//////////////////////////////////////////////////////////////////////////////
+//
+// InterpolateLinearInteger - integer arithmetic implementation
+//
+
+/// fixed-point interpolation routine precision
+#define SCALE    65536
+
+
+// Constructor
+InterpolateLinearInteger::InterpolateLinearInteger() : TransposerBase()
+{
+    // Notice: use local function calling syntax for sake of clarity,
+    // to indicate the fact that C++ constructor can't call virtual functions.
+    resetRegisters();
+    setRate(1.0f);
+}
+
+
+void InterpolateLinearInteger::resetRegisters()
+{
+    iFract = 0;
+}
+
+
+// Transposes the sample rate of the given samples using linear interpolation.
+// 'Mono' version of the routine. Returns the number of samples returned in
+// the "dest" buffer
+int InterpolateLinearInteger::transposeMono(SAMPLETYPE *dest, const SAMPLETYPE *src, int &srcSamples)
+{
+    int i;
+    int srcSampleEnd = srcSamples - 1;
+    int srcCount = 0;
+
+    i = 0;
+    while (srcCount < srcSampleEnd)
+    {
+        LONG_SAMPLETYPE temp;
+
+        assert(iFract < SCALE);
+
+        temp = (SCALE - iFract) * src[0] + iFract * src[1];
+        dest[i] = (SAMPLETYPE)(temp / SCALE);
+        i++;
+
+        iFract += iRate;
+
+        int iWhole = iFract / SCALE;
+        iFract -= iWhole * SCALE;
+        srcCount += iWhole;
+        src += iWhole;
+    }
+    srcSamples = srcCount;
+
+    return i;
+}
+
+
+// Transposes the sample rate of the given samples using linear interpolation.
+// 'Stereo' version of the routine. Returns the number of samples returned in
+// the "dest" buffer
+int InterpolateLinearInteger::transposeStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, int &srcSamples)
+{
+    int i;
+    int srcSampleEnd = srcSamples - 1;
+    int srcCount = 0;
+
+    i = 0;
+    while (srcCount < srcSampleEnd)
+    {
+        LONG_SAMPLETYPE temp0;
+        LONG_SAMPLETYPE temp1;
+
+        assert(iFract < SCALE);
+
+        temp0 = (SCALE - iFract) * src[0] + iFract * src[2];
+        temp1 = (SCALE - iFract) * src[1] + iFract * src[3];
+        dest[0] = (SAMPLETYPE)(temp0 / SCALE);
+        dest[1] = (SAMPLETYPE)(temp1 / SCALE);
+        dest += 2;
+        i++;
+
+        iFract += iRate;
+
+        int iWhole = iFract / SCALE;
+        iFract -= iWhole * SCALE;
+        srcCount += iWhole;
+        src += 2*iWhole;
+    }
+    srcSamples = srcCount;
+
+    return i;
+}
+
+
+int InterpolateLinearInteger::transposeMulti(SAMPLETYPE *dest, const SAMPLETYPE *src, int &srcSamples)
+{
+    int i;
+    int srcSampleEnd = srcSamples - 1;
+    int srcCount = 0;
+
+    i = 0;
+    while (srcCount < srcSampleEnd)
+    {
+        LONG_SAMPLETYPE temp, vol1;
+
+        assert(iFract < SCALE);
+        vol1 = (LONG_SAMPLETYPE)(SCALE - iFract);
+        for (int c = 0; c < numChannels; c ++)
+        {
+            temp = vol1 * src[c] + iFract * src[c + numChannels];
+            dest[0] = (SAMPLETYPE)(temp / SCALE);
+            dest ++;
+        }
+        i++;
+
+        iFract += iRate;
+
+        int iWhole = iFract / SCALE;
+        iFract -= iWhole * SCALE;
+        srcCount += iWhole;
+        src += iWhole * numChannels;
+    }
+    srcSamples = srcCount;
+
+    return i;
+}
+
+
+// Sets new target iRate. Normal iRate = 1.0, smaller values represent slower
+// iRate, larger faster iRates.
+void InterpolateLinearInteger::setRate(double newRate)
+{
+    iRate = (int)(newRate * SCALE + 0.5);
+    TransposerBase::setRate(newRate);
+}
+
+
+//////////////////////////////////////////////////////////////////////////////
+//
+// InterpolateLinearFloat - floating point arithmetic implementation
+//
+//////////////////////////////////////////////////////////////////////////////
+
+
+// Constructor
+InterpolateLinearFloat::InterpolateLinearFloat() : TransposerBase()
+{
+    // Notice: use local function calling syntax for sake of clarity,
+    // to indicate the fact that C++ constructor can't call virtual functions.
+    resetRegisters();
+    setRate(1.0);
+}
+
+
+void InterpolateLinearFloat::resetRegisters()
+{
+    fract = 0;
+}
+
+
+// Transposes the sample rate of the given samples using linear interpolation.
+// 'Mono' version of the routine. Returns the number of samples returned in
+// the "dest" buffer
+int InterpolateLinearFloat::transposeMono(SAMPLETYPE *dest, const SAMPLETYPE *src, int &srcSamples)
+{
+    int i;
+    int srcSampleEnd = srcSamples - 1;
+    int srcCount = 0;
+
+    i = 0;
+    while (srcCount < srcSampleEnd)
+    {
+        double out;
+        assert(fract < 1.0);
+
+        out = (1.0 - fract) * src[0] + fract * src[1];
+        dest[i] = (SAMPLETYPE)out;
+        i ++;
+
+        // update position fraction
+        fract += rate;
+        // update whole positions
+        int whole = (int)fract;
+        fract -= whole;
+        src += whole;
+        srcCount += whole;
+    }
+    srcSamples = srcCount;
+    return i;
+}
+
+
+// Transposes the sample rate of the given samples using linear interpolation.
+// 'Mono' version of the routine. Returns the number of samples returned in
+// the "dest" buffer
+int InterpolateLinearFloat::transposeStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, int &srcSamples)
+{
+    int i;
+    int srcSampleEnd = srcSamples - 1;
+    int srcCount = 0;
+
+    i = 0;
+    while (srcCount < srcSampleEnd)
+    {
+        double out0, out1;
+        assert(fract < 1.0);
+
+        out0 = (1.0 - fract) * src[0] + fract * src[2];
+        out1 = (1.0 - fract) * src[1] + fract * src[3];
+        dest[2*i]   = (SAMPLETYPE)out0;
+        dest[2*i+1] = (SAMPLETYPE)out1;
+        i ++;
+
+        // update position fraction
+        fract += rate;
+        // update whole positions
+        int whole = (int)fract;
+        fract -= whole;
+        src += 2*whole;
+        srcCount += whole;
+    }
+    srcSamples = srcCount;
+    return i;
+}
+
+
+int InterpolateLinearFloat::transposeMulti(SAMPLETYPE *dest, const SAMPLETYPE *src, int &srcSamples)
+{
+    int i;
+    int srcSampleEnd = srcSamples - 1;
+    int srcCount = 0;
+
+    i = 0;
+    while (srcCount < srcSampleEnd)
+    {
+        float temp, vol1, fract_float;
+
+        vol1 = (float)(1.0 - fract);
+		fract_float = (float)fract;
+        for (int c = 0; c < numChannels; c ++)
+        {
+			temp = vol1 * src[c] + fract_float * src[c + numChannels];
+            *dest = (SAMPLETYPE)temp;
+            dest ++;
+        }
+        i++;
+
+        fract += rate;
+
+        int iWhole = (int)fract;
+        fract -= iWhole;
+        srcCount += iWhole;
+        src += iWhole * numChannels;
+    }
+    srcSamples = srcCount;
+
+    return i;
+}

source/SoundTouch/InterpolateLinear.h

@@ -0,0 +1,98 @@
+////////////////////////////////////////////////////////////////////////////////
+///
+/// Linear interpolation routine.
+///
+/// Author        : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+//  SoundTouch audio processing library
+//  Copyright (c) Olli Parviainen
+//
+//  This library is free software; you can redistribute it and/or
+//  modify it under the terms of the GNU Lesser General Public
+//  License as published by the Free Software Foundation; either
+//  version 2.1 of the License, or (at your option) any later version.
+//
+//  This library is distributed in the hope that it will be useful,
+//  but WITHOUT ANY WARRANTY; without even the implied warranty of
+//  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+//  Lesser General Public License for more details.
+//
+//  You should have received a copy of the GNU Lesser General Public
+//  License along with this library; if not, write to the Free Software
+//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#ifndef _InterpolateLinear_H_
+#define _InterpolateLinear_H_
+
+#include "RateTransposer.h"
+#include "STTypes.h"
+
+namespace soundtouch
+{
+
+/// Linear transposer class that uses integer arithmetic
+class InterpolateLinearInteger : public TransposerBase
+{
+protected:
+    int iFract;
+    int iRate;
+
+    virtual int transposeMono(SAMPLETYPE *dest,
+                       const SAMPLETYPE *src,
+                       int &srcSamples) override;
+    virtual int transposeStereo(SAMPLETYPE *dest,
+                         const SAMPLETYPE *src,
+                         int &srcSamples) override;
+    virtual int transposeMulti(SAMPLETYPE *dest, const SAMPLETYPE *src, int &srcSamples) override;
+public:
+    InterpolateLinearInteger();
+
+    /// Sets new target rate. Normal rate = 1.0, smaller values represent slower
+    /// rate, larger faster rates.
+    virtual void setRate(double newRate) override;
+
+    virtual void resetRegisters() override;
+
+    virtual int getLatency() const override
+    {
+        return 0;
+    }
+};
+
+
+/// Linear transposer class that uses floating point arithmetic
+class InterpolateLinearFloat : public TransposerBase
+{
+protected:
+    double fract;
+
+    virtual int transposeMono(SAMPLETYPE *dest,
+                       const SAMPLETYPE *src,
+                       int &srcSamples);
+    virtual int transposeStereo(SAMPLETYPE *dest,
+                         const SAMPLETYPE *src,
+                         int &srcSamples);
+    virtual int transposeMulti(SAMPLETYPE *dest, const SAMPLETYPE *src, int &srcSamples);
+
+public:
+    InterpolateLinearFloat();
+
+    virtual void resetRegisters();
+
+    int getLatency() const
+    {
+        return 0;
+    }
+};
+
+}
+
+#endif

source/SoundTouch/InterpolateShannon.cpp

@@ -0,0 +1,181 @@
+////////////////////////////////////////////////////////////////////////////////
+///
+/// Sample interpolation routine using 8-tap band-limited Shannon interpolation
+/// with kaiser window.
+///
+/// Notice. This algorithm is remarkably much heavier than linear or cubic
+/// interpolation, and not remarkably better than cubic algorithm. Thus mostly
+/// for experimental purposes
+///
+/// Author        : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+//  SoundTouch audio processing library
+//  Copyright (c) Olli Parviainen
+//
+//  This library is free software; you can redistribute it and/or
+//  modify it under the terms of the GNU Lesser General Public
+//  License as published by the Free Software Foundation; either
+//  version 2.1 of the License, or (at your option) any later version.
+//
+//  This library is distributed in the hope that it will be useful,
+//  but WITHOUT ANY WARRANTY; without even the implied warranty of
+//  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+//  Lesser General Public License for more details.
+//
+//  You should have received a copy of the GNU Lesser General Public
+//  License along with this library; if not, write to the Free Software
+//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#include <math.h>
+#include "InterpolateShannon.h"
+#include "STTypes.h"
+
+using namespace soundtouch;
+
+
+/// Kaiser window with beta = 2.0
+/// Values scaled down by 5% to avoid overflows
+static const double _kaiser8[8] =
+{
+   0.41778693317814,
+   0.64888025049173,
+   0.83508562409944,
+   0.93887857733412,
+   0.93887857733412,
+   0.83508562409944,
+   0.64888025049173,
+   0.41778693317814
+};
+
+
+InterpolateShannon::InterpolateShannon()
+{
+    fract = 0;
+}
+
+
+void InterpolateShannon::resetRegisters()
+{
+    fract = 0;
+}
+
+
+#define PI 3.1415926536
+#define sinc(x) (sin(PI * (x)) / (PI * (x)))
+
+/// Transpose mono audio. Returns number of produced output samples, and
+/// updates "srcSamples" to amount of consumed source samples
+int InterpolateShannon::transposeMono(SAMPLETYPE *pdest,
+                    const SAMPLETYPE *psrc,
+                    int &srcSamples)
+{
+    int i;
+    int srcSampleEnd = srcSamples - 8;
+    int srcCount = 0;
+
+    i = 0;
+    while (srcCount < srcSampleEnd)
+    {
+        double out;
+        assert(fract < 1.0);
+
+        out  = psrc[0] * sinc(-3.0 - fract) * _kaiser8[0];
+        out += psrc[1] * sinc(-2.0 - fract) * _kaiser8[1];
+        out += psrc[2] * sinc(-1.0 - fract) * _kaiser8[2];
+        if (fract < 1e-6)
+        {
+            out += psrc[3] * _kaiser8[3];     // sinc(0) = 1
+        }
+        else
+        {
+            out += psrc[3] * sinc(- fract) * _kaiser8[3];
+        }
+        out += psrc[4] * sinc( 1.0 - fract) * _kaiser8[4];
+        out += psrc[5] * sinc( 2.0 - fract) * _kaiser8[5];
+        out += psrc[6] * sinc( 3.0 - fract) * _kaiser8[6];
+        out += psrc[7] * sinc( 4.0 - fract) * _kaiser8[7];
+
+        pdest[i] = (SAMPLETYPE)out;
+        i ++;
+
+        // update position fraction
+        fract += rate;
+        // update whole positions
+        int whole = (int)fract;
+        fract -= whole;
+        psrc += whole;
+        srcCount += whole;
+    }
+    srcSamples = srcCount;
+    return i;
+}
+
+
+/// Transpose stereo audio. Returns number of produced output samples, and
+/// updates "srcSamples" to amount of consumed source samples
+int InterpolateShannon::transposeStereo(SAMPLETYPE *pdest,
+                    const SAMPLETYPE *psrc,
+                    int &srcSamples)
+{
+    int i;
+    int srcSampleEnd = srcSamples - 8;
+    int srcCount = 0;
+
+    i = 0;
+    while (srcCount < srcSampleEnd)
+    {
+        double out0, out1, w;
+        assert(fract < 1.0);
+
+        w = sinc(-3.0 - fract) * _kaiser8[0];
+        out0 = psrc[0] * w; out1 = psrc[1] * w;
+        w = sinc(-2.0 - fract) * _kaiser8[1];
+        out0 += psrc[2] * w; out1 += psrc[3] * w;
+        w = sinc(-1.0 - fract) * _kaiser8[2];
+        out0 += psrc[4] * w; out1 += psrc[5] * w;
+        w = _kaiser8[3] * ((fract < 1e-5) ? 1.0 : sinc(- fract));   // sinc(0) = 1
+        out0 += psrc[6] * w; out1 += psrc[7] * w;
+        w = sinc( 1.0 - fract) * _kaiser8[4];
+        out0 += psrc[8] * w; out1 += psrc[9] * w;
+        w = sinc( 2.0 - fract) * _kaiser8[5];
+        out0 += psrc[10] * w; out1 += psrc[11] * w;
+        w = sinc( 3.0 - fract) * _kaiser8[6];
+        out0 += psrc[12] * w; out1 += psrc[13] * w;
+        w = sinc( 4.0 - fract) * _kaiser8[7];
+        out0 += psrc[14] * w; out1 += psrc[15] * w;
+
+        pdest[2*i]   = (SAMPLETYPE)out0;
+        pdest[2*i+1] = (SAMPLETYPE)out1;
+        i ++;
+
+        // update position fraction
+        fract += rate;
+        // update whole positions
+        int whole = (int)fract;
+        fract -= whole;
+        psrc += 2*whole;
+        srcCount += whole;
+    }
+    srcSamples = srcCount;
+    return i;
+}
+
+
+/// Transpose stereo audio. Returns number of produced output samples, and
+/// updates "srcSamples" to amount of consumed source samples
+int InterpolateShannon::transposeMulti(SAMPLETYPE *,
+                    const SAMPLETYPE *,
+                    int &)
+{
+    // not implemented
+    assert(false);
+    return 0;
+}

source/SoundTouch/InterpolateShannon.h

@@ -0,0 +1,74 @@
+////////////////////////////////////////////////////////////////////////////////
+///
+/// Sample interpolation routine using 8-tap band-limited Shannon interpolation
+/// with kaiser window.
+///
+/// Notice. This algorithm is remarkably much heavier than linear or cubic
+/// interpolation, and not remarkably better than cubic algorithm. Thus mostly
+/// for experimental purposes
+///
+/// Author        : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+//  SoundTouch audio processing library
+//  Copyright (c) Olli Parviainen
+//
+//  This library is free software; you can redistribute it and/or
+//  modify it under the terms of the GNU Lesser General Public
+//  License as published by the Free Software Foundation; either
+//  version 2.1 of the License, or (at your option) any later version.
+//
+//  This library is distributed in the hope that it will be useful,
+//  but WITHOUT ANY WARRANTY; without even the implied warranty of
+//  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+//  Lesser General Public License for more details.
+//
+//  You should have received a copy of the GNU Lesser General Public
+//  License along with this library; if not, write to the Free Software
+//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#ifndef _InterpolateShannon_H_
+#define _InterpolateShannon_H_
+
+#include "RateTransposer.h"
+#include "STTypes.h"
+
+namespace soundtouch
+{
+
+class InterpolateShannon : public TransposerBase
+{
+protected:
+    int transposeMono(SAMPLETYPE *dest,
+                        const SAMPLETYPE *src,
+                        int &srcSamples) override;
+    int transposeStereo(SAMPLETYPE *dest,
+                        const SAMPLETYPE *src,
+                        int &srcSamples) override;
+    int transposeMulti(SAMPLETYPE *dest,
+                        const SAMPLETYPE *src,
+                        int &srcSamples) override;
+
+    double fract;
+
+public:
+    InterpolateShannon();
+
+    void resetRegisters() override;
+
+    virtual int getLatency() const override
+    {
+        return 3;
+    }
+};
+
+}
+
+#endif

source/SoundTouch/Makefile.am

@@ -1,7 +1,5 @@
 ## Process this file with automake to create Makefile.in
 ##
-## $Id$
-##
 ## 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
@@ -22,17 +20,20 @@ include $(top_srcdir)/config/am_include.mk
 
 
 # set to something if you want other stuff to be included in the distribution tarball
-EXTRA_DIST=SoundTouch.dsp SoundTouch.dsw SoundTouch.sln SoundTouch.vcproj
+EXTRA_DIST=SoundTouch.sln SoundTouch.vcxproj
 
-noinst_HEADERS=AAFilter.h cpu_detect.h cpu_detect_x86.cpp FIRFilter.h RateTransposer.h TDStretch.h PeakFinder.h
+noinst_HEADERS=AAFilter.h cpu_detect.h cpu_detect_x86.cpp FIRFilter.h RateTransposer.h TDStretch.h PeakFinder.h \
+    InterpolateCubic.h InterpolateLinear.h InterpolateShannon.h
 
 lib_LTLIBRARIES=libSoundTouch.la
 #
-libSoundTouch_la_SOURCES=AAFilter.cpp FIRFilter.cpp FIFOSampleBuffer.cpp RateTransposer.cpp SoundTouch.cpp TDStretch.cpp cpu_detect_x86.cpp BPMDetect.cpp PeakFinder.cpp
+libSoundTouch_la_SOURCES=AAFilter.cpp FIRFilter.cpp FIFOSampleBuffer.cpp    \
-
+    RateTransposer.cpp SoundTouch.cpp TDStretch.cpp cpu_detect_x86.cpp      \
+    BPMDetect.cpp PeakFinder.cpp InterpolateLinear.cpp InterpolateCubic.cpp \
+    InterpolateShannon.cpp
 
 # Compiler flags
-AM_CXXFLAGS=-O3 -fcheck-new -I../../include
+#AM_CXXFLAGS+=
 
 # Compile the files that need MMX and SSE individually.
 libSoundTouch_la_LIBADD=libSoundTouchMMX.la libSoundTouchSSE.la
@@ -63,6 +64,8 @@ libSoundTouchMMX_la_CXXFLAGS = $(AM_CXXFLAGS)
 libSoundTouchSSE_la_CXXFLAGS = $(AM_CXXFLAGS)
 endif
 
+# Modify the default 0.0.0 to LIB_SONAME.0.0
+libSoundTouch_la_LDFLAGS=-version-info @LIB_SONAME@
 
 # other linking flags to add
 # noinst_LTLIBRARIES = libSoundTouchOpt.la

source/SoundTouch/PeakFinder.cpp

@@ -11,13 +11,6 @@
 ///
 ////////////////////////////////////////////////////////////////////////////////
 //
-// Last changed  : $Date$
-// File revision : $Revision: 4 $
-//
-// $Id$
-//
-////////////////////////////////////////////////////////////////////////////////
-//
 // License :
 //
 //  SoundTouch audio processing library
@@ -64,7 +57,7 @@ int PeakFinder::findTop(const float *data, int peakpos) const
 
     refvalue = data[peakpos];
 
-    // seek within ±10 points
+    // seek within ±10 points
     start = peakpos - 10;
     if (start < minPos) start = minPos;
     end = peakpos + 10;
@@ -149,7 +142,7 @@ int PeakFinder::findCrossingLevel(const float *data, float level, int peakpos, i
     peaklevel = data[peakpos];
     assert(peaklevel >= level);
     pos = peakpos;
-    while ((pos >= minPos) && (pos < maxPos))
+    while ((pos >= minPos) && (pos + direction < maxPos))
     {
         if (data[pos + direction] < level) return pos;   // crossing found
         pos += direction;
@@ -178,7 +171,6 @@ double PeakFinder::calcMassCenter(const float *data, int firstPos, int lastPos)
 }
 
 
-
 /// get exact center of peak near given position by calculating local mass of center
 double PeakFinder::getPeakCenter(const float *data, int peakpos) const
 {
@@ -192,11 +184,21 @@ double PeakFinder::getPeakCenter(const float *data, int peakpos) const
     gp1 = findGround(data, peakpos, -1);
     gp2 = findGround(data, peakpos, 1);
 
-    groundLevel = 0.5f * (data[gp1] + data[gp2]);
     peakLevel = data[peakpos];
 
+    if (gp1 == gp2)
+    {
+        // avoid rounding errors when all are equal
+        assert(gp1 == peakpos);
+        cutLevel = groundLevel = peakLevel;
+    } else {
+        // get average of the ground levels
+        groundLevel = 0.5f * (data[gp1] + data[gp2]);
+
         // calculate 70%-level of the peak
         cutLevel = 0.70f * peakLevel + 0.30f * groundLevel;
+    }
+
     // find mid-level crossings
     crosspos1 = findCrossingLevel(data, cutLevel, peakpos, -1);
     crosspos2 = findCrossingLevel(data, cutLevel, peakpos, 1);
@@ -208,7 +210,6 @@ double PeakFinder::getPeakCenter(const float *data, int peakpos) const
 }
 
 
-
 double PeakFinder::detectPeak(const float *data, int aminPos, int amaxPos)
 {
 
@@ -239,14 +240,12 @@ double PeakFinder::detectPeak(const float *data, int aminPos, int amaxPos)
     // - sometimes the highest peak can be Nth harmonic of the true base peak yet
     // just a slightly higher than the true base
 
-    int hp = (int)(highPeak + 0.5);
+    for (i = 1; i < 3; i ++)
-
-    for (i = 3; i < 10; i ++)
     {
         double peaktmp, harmonic;
         int i1,i2;
 
-        harmonic = (double)i * 0.5;
+        harmonic = (double)pow(2.0, i);
         peakpos = (int)(highPeak / harmonic + 0.5f);
         if (peakpos < minPos) break;
         peakpos = findTop(data, peakpos);   // seek true local maximum index
@@ -257,7 +256,7 @@ double PeakFinder::detectPeak(const float *data, int aminPos, int amaxPos)
 
         // accept harmonic peak if
         // (a) it is found
-        // (b) is within ±4% of the expected harmonic interval
+        // (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;

source/SoundTouch/PeakFinder.h

@@ -9,13 +9,6 @@
 ///
 ////////////////////////////////////////////////////////////////////////////////
 //
-// Last changed  : $Date$
-// File revision : $Revision: 4 $
-//
-// $Id$
-//
-////////////////////////////////////////////////////////////////////////////////
-//
 // License :
 //
 //  SoundTouch audio processing library
@@ -51,8 +44,8 @@ protected:
 
     /// 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 firstPos,      ///< Index of first vector item belonging to the peak.
-                         int lastPos        ///< Index of last vector item beloging to the peak.
+                         int lastPos        ///< Index of last vector item belonging to the peak.
                          ) const;
 
     /// Finds the data vector index where the monotoniously decreasing signal crosses the

source/SoundTouch/RateTransposer.cpp

@@ -10,13 +10,6 @@
 ///
 ////////////////////////////////////////////////////////////////////////////////
 //
-// Last changed  : $Date$
-// File revision : $Revision: 4 $
-//
-// $Id$
-//
-////////////////////////////////////////////////////////////////////////////////
-//
 // License :
 //
 //  SoundTouch audio processing library
@@ -43,114 +36,55 @@
 #include <stdlib.h>
 #include <stdio.h>
 #include "RateTransposer.h"
+#include "InterpolateLinear.h"
+#include "InterpolateCubic.h"
+#include "InterpolateShannon.h"
 #include "AAFilter.h"
 
 using namespace soundtouch;
 
-
+// Define default interpolation algorithm here
-/// A linear samplerate transposer class that uses integer arithmetics.
+TransposerBase::ALGORITHM TransposerBase::algorithm = TransposerBase::CUBIC;
-/// for the transposing.
-class RateTransposerInteger : public RateTransposer
-{
-protected:
-    int iSlopeCount;
-    int iRate;
-    SAMPLETYPE sPrevSampleL, sPrevSampleR;
-
-    virtual void resetRegisters();
-
-    virtual uint transposeStereo(SAMPLETYPE *dest, 
-                         const SAMPLETYPE *src, 
-                         uint numSamples);
-    virtual uint transposeMono(SAMPLETYPE *dest, 
-                       const SAMPLETYPE *src, 
-                       uint numSamples);
-
-public:
-    RateTransposerInteger();
-    virtual ~RateTransposerInteger();
-
-    /// Sets new target rate. Normal rate = 1.0, smaller values represent slower 
-    /// rate, larger faster rates.
-    virtual void setRate(float newRate);
-
-};
-
-
-/// A linear samplerate transposer class that uses floating point arithmetics
-/// for the transposing.
-class RateTransposerFloat : public RateTransposer
-{
-protected:
-    float fSlopeCount;
-    SAMPLETYPE sPrevSampleL, sPrevSampleR;
-
-    virtual void resetRegisters();
-
-    virtual uint transposeStereo(SAMPLETYPE *dest, 
-                         const SAMPLETYPE *src, 
-                         uint numSamples);
-    virtual uint transposeMono(SAMPLETYPE *dest, 
-                       const SAMPLETYPE *src, 
-                       uint numSamples);
-
-public:
-    RateTransposerFloat();
-    virtual ~RateTransposerFloat();
-};
-
-
-
-
-// Operator 'new' is overloaded so that it automatically creates a suitable instance 
-// depending on if we've a MMX/SSE/etc-capable CPU available or not.
-void * RateTransposer::operator new(size_t s)
-{
-    ST_THROW_RT_ERROR("Error in RateTransoser::new: don't use \"new TDStretch\" directly, use \"newInstance\" to create a new instance instead!");
-    return newInstance();
-}
-
-
-RateTransposer *RateTransposer::newInstance()
-{
-#ifdef SOUNDTOUCH_INTEGER_SAMPLES
-    return ::new RateTransposerInteger;
-#else
-    return ::new RateTransposerFloat;
-#endif
-}
 
 
 // Constructor
 RateTransposer::RateTransposer() : FIFOProcessor(&outputBuffer)
 {
-    numChannels = 2;
+    bUseAAFilter =
-    bUseAAFilter = TRUE;
+#ifndef SOUNDTOUCH_PREVENT_CLICK_AT_RATE_CROSSOVER
-    fRate = 0;
+        true;
+#else
+        // Disable Anti-alias filter if desirable to avoid click at rate change zero value crossover
+        false;
+#endif
 
-    // Instantiates the anti-alias filter with default tap length
+    // Instantiates the anti-alias filter
-    // of 32
+    pAAFilter = new AAFilter(64);
-    pAAFilter = new AAFilter(32);
+    pTransposer = TransposerBase::newInstance();
+    clear();
 }
 
 
-
 RateTransposer::~RateTransposer()
 {
     delete pAAFilter;
+    delete pTransposer;
 }
 
 
-
 /// Enables/disables the anti-alias filter. Zero to disable, nonzero to enable
-void RateTransposer::enableAAFilter(BOOL newMode)
+void RateTransposer::enableAAFilter(bool newMode)
 {
+#ifndef SOUNDTOUCH_PREVENT_CLICK_AT_RATE_CROSSOVER
+    // Disable Anti-alias filter if desirable to avoid click at rate change zero value crossover
     bUseAAFilter = newMode;
+    clear();
+#endif
 }
 
 
 /// Returns nonzero if anti-alias filter is enabled.
-BOOL RateTransposer::isAAFilterEnabled() const
+bool RateTransposer::isAAFilterEnabled() const
 {
     return bUseAAFilter;
 }
@@ -162,44 +96,27 @@ AAFilter *RateTransposer::getAAFilter()
 }
 
 
-
 // Sets new target iRate. Normal iRate = 1.0, smaller values represent slower
 // iRate, larger faster iRates.
-void RateTransposer::setRate(float newRate)
+void RateTransposer::setRate(double newRate)
 {
     double fCutoff;
 
-    fRate = newRate;
+    pTransposer->setRate(newRate);
 
     // design a new anti-alias filter
-    if (newRate > 1.0f) 
+    if (newRate > 1.0)
     {
-        fCutoff = 0.5f / newRate;
+        fCutoff = 0.5 / newRate;
     }
     else
     {
-        fCutoff = 0.5f * newRate;
+        fCutoff = 0.5 * newRate;
     }
     pAAFilter->setCutoffFreq(fCutoff);
 }
 
 
-// Outputs as many samples of the 'outputBuffer' as possible, and if there's
-// any room left, outputs also as many of the incoming samples as possible.
-// The goal is to drive the outputBuffer empty.
-//
-// It's allowed for 'output' and 'input' parameters to point to the same
-// memory position.
-/*
-void RateTransposer::flushStoreBuffer()
-{
-    if (storeBuffer.isEmpty()) return;
-
-    outputBuffer.moveSamples(storeBuffer);
-}
-*/
-
-
 // Adds 'nSamples' pcs of samples from the 'samples' memory position into
 // the input of the object.
 void RateTransposer::putSamples(const SAMPLETYPE *samples, uint nSamples)
@@ -208,115 +125,50 @@ void RateTransposer::putSamples(const SAMPLETYPE *samples, uint nSamples)
 }
 
 
-
-// Transposes up the sample rate, causing the observed playback 'rate' of the
-// sound to decrease
-void RateTransposer::upsample(const SAMPLETYPE *src, uint nSamples)
-{
-    uint count, sizeTemp, num;
-
-    // If the parameter 'uRate' value is smaller than 'SCALE', first transpose
-    // the samples and then apply the anti-alias filter to remove aliasing.
-
-    // First check that there's enough room in 'storeBuffer' 
-    // (+16 is to reserve some slack in the destination buffer)
-    sizeTemp = (uint)((float)nSamples / fRate + 16.0f);
-
-    // Transpose the samples, store the result into the end of "storeBuffer"
-    count = transpose(storeBuffer.ptrEnd(sizeTemp), src, nSamples);
-    storeBuffer.putSamples(count);
-
-    // Apply the anti-alias filter to samples in "store output", output the
-    // result to "dest"
-    num = storeBuffer.numSamples();
-    count = pAAFilter->evaluate(outputBuffer.ptrEnd(num), 
-        storeBuffer.ptrBegin(), num, (uint)numChannels);
-    outputBuffer.putSamples(count);
-
-    // Remove the processed samples from "storeBuffer"
-    storeBuffer.receiveSamples(count);
-}
-
-
-// Transposes down the sample rate, causing the observed playback 'rate' of the
-// sound to increase
-void RateTransposer::downsample(const SAMPLETYPE *src, uint nSamples)
-{
-    uint count, sizeTemp;
-
-    // If the parameter 'uRate' value is larger than 'SCALE', first apply the
-    // anti-alias filter to remove high frequencies (prevent them from folding
-    // over the lover frequencies), then transpose.
-
-    // Add the new samples to the end of the storeBuffer
-    storeBuffer.putSamples(src, nSamples);
-
-    // Anti-alias filter the samples to prevent folding and output the filtered 
-    // data to tempBuffer. Note : because of the FIR filter length, the
-    // filtering routine takes in 'filter_length' more samples than it outputs.
-    assert(tempBuffer.isEmpty());
-    sizeTemp = storeBuffer.numSamples();
-
-    count = pAAFilter->evaluate(tempBuffer.ptrEnd(sizeTemp), 
-        storeBuffer.ptrBegin(), sizeTemp, (uint)numChannels);
-
-	if (count == 0) return;
-
-    // Remove the filtered samples from 'storeBuffer'
-    storeBuffer.receiveSamples(count);
-
-    // Transpose the samples (+16 is to reserve some slack in the destination buffer)
-    sizeTemp = (uint)((float)nSamples / fRate + 16.0f);
-    count = transpose(outputBuffer.ptrEnd(sizeTemp), tempBuffer.ptrBegin(), count);
-    outputBuffer.putSamples(count);
-}
-
-
 // Transposes sample rate by applying anti-alias filter to prevent folding.
 // Returns amount of samples returned in the "dest" buffer.
 // The maximum amount of samples that can be returned at a time is set by
 // the 'set_returnBuffer_size' function.
 void RateTransposer::processSamples(const SAMPLETYPE *src, uint nSamples)
 {
-    uint count;
-    uint sizeReq;
-
     if (nSamples == 0) return;
-    assert(pAAFilter);
+
+    // Store samples to input buffer
+    inputBuffer.putSamples(src, nSamples);
 
     // If anti-alias filter is turned off, simply transpose without applying
     // the filter
-    if (bUseAAFilter == FALSE) 
+    if (bUseAAFilter == false)
     {
-        sizeReq = (uint)((float)nSamples / fRate + 1.0f);
+        (void)pTransposer->transpose(outputBuffer, inputBuffer);
-        count = transpose(outputBuffer.ptrEnd(sizeReq), src, nSamples);
-        outputBuffer.putSamples(count);
         return;
     }
 
+    assert(pAAFilter);
+
     // Transpose with anti-alias filter
-    if (fRate < 1.0f) 
+    if (pTransposer->rate < 1.0f)
     {
-        upsample(src, nSamples);
+        // If the parameter 'Rate' value is smaller than 1, first transpose
+        // the samples and then apply the anti-alias filter to remove aliasing.
+
+        // Transpose the samples, store the result to end of "midBuffer"
+        pTransposer->transpose(midBuffer, inputBuffer);
+
+        // Apply the anti-alias filter for transposed samples in midBuffer
+        pAAFilter->evaluate(outputBuffer, midBuffer);
     }
     else
     {
-        downsample(src, nSamples);
+        // If the parameter 'Rate' value is larger than 1, first apply the
-    }
+        // anti-alias filter to remove high frequencies (prevent them from folding
-}
+        // over the lover frequencies), then transpose.
 
+        // Apply the anti-alias filter for samples in inputBuffer
+        pAAFilter->evaluate(midBuffer, inputBuffer);
 
-// Transposes the sample rate of the given samples using linear interpolation. 
+        // Transpose the AA-filtered samples in "midBuffer"
-// Returns the number of samples returned in the "dest" buffer
+        pTransposer->transpose(outputBuffer, midBuffer);
-inline uint RateTransposer::transpose(SAMPLETYPE *dest, const SAMPLETYPE *src, uint nSamples)
-{
-    if (numChannels == 2) 
-    {
-        return transposeStereo(dest, src, nSamples);
-    } 
-    else 
-    {
-        return transposeMono(dest, src, nSamples);
     }
 }
 
@@ -324,18 +176,13 @@ inline uint RateTransposer::transpose(SAMPLETYPE *dest, const SAMPLETYPE *src, u
 // Sets the number of channels, 1 = mono, 2 = stereo
 void RateTransposer::setChannels(int nChannels)
 {
-    assert(nChannels > 0);
+    if (!verifyNumberOfChannels(nChannels) ||
-    if (numChannels == nChannels) return;
+        (pTransposer->numChannels == nChannels)) return;
 
-    assert(nChannels == 1 || nChannels == 2);
+    pTransposer->setChannels(nChannels);
-    numChannels = nChannels;
+    inputBuffer.setChannels(nChannels);
-
+    midBuffer.setChannels(nChannels);
-    storeBuffer.setChannels(numChannels);
+    outputBuffer.setChannels(nChannels);
-    tempBuffer.setChannels(numChannels);
-    outputBuffer.setChannels(numChannels);
-
-    // Inits the linear interpolation registers
-    resetRegisters();
 }
 
 
@@ -343,7 +190,13 @@ void RateTransposer::setChannels(int nChannels)
 void RateTransposer::clear()
 {
     outputBuffer.clear();
-    storeBuffer.clear();
+    midBuffer.clear();
+    inputBuffer.clear();
+    pTransposer->resetRegisters();
+
+    // prefill buffer to avoid losing first samples at beginning of stream
+    int prefill = getLatency();
+    inputBuffer.addSilent(prefill);
 }
 
 
@@ -354,273 +207,107 @@ int RateTransposer::isEmpty() const
 
     res = FIFOProcessor::isEmpty();
     if (res == 0) return 0;
-    return storeBuffer.isEmpty();
+    return inputBuffer.isEmpty();
+}
+
+
+/// Return approximate initial input-output latency
+int RateTransposer::getLatency() const
+{
+    return pTransposer->getLatency() +
+        ((bUseAAFilter) ? (pAAFilter->getLength() / 2) : 0);
 }
 
 
 //////////////////////////////////////////////////////////////////////////////
 //
-// RateTransposerInteger - integer arithmetic implementation
+// TransposerBase - Base class for interpolation
 //
 
-/// fixed-point interpolation routine precision
+// static function to set interpolation algorithm
-#define SCALE    65536
+void TransposerBase::setAlgorithm(TransposerBase::ALGORITHM a)
-
-// Constructor
-RateTransposerInteger::RateTransposerInteger() : RateTransposer()
 {
-    // Notice: use local function calling syntax for sake of clarity, 
+    TransposerBase::algorithm = a;
-    // to indicate the fact that C++ constructor can't call virtual functions.
-    RateTransposerInteger::resetRegisters();
-    RateTransposerInteger::setRate(1.0f);
-}
-
-
-RateTransposerInteger::~RateTransposerInteger()
-{
-}
-
-
-void RateTransposerInteger::resetRegisters()
-{
-    iSlopeCount = 0;
-    sPrevSampleL = 
-    sPrevSampleR = 0;
-}
-
-
-
-// Transposes the sample rate of the given samples using linear interpolation. 
-// 'Mono' version of the routine. Returns the number of samples returned in 
-// the "dest" buffer
-uint RateTransposerInteger::transposeMono(SAMPLETYPE *dest, const SAMPLETYPE *src, uint nSamples)
-{
-    unsigned int i, used;
-    LONG_SAMPLETYPE temp, vol1;
-
-    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) 
-    {
-        vol1 = (LONG_SAMPLETYPE)(SCALE - iSlopeCount);
-        temp = vol1 * sPrevSampleL + iSlopeCount * src[0];
-        dest[i] = (SAMPLETYPE)(temp / SCALE);
-        i++;
-        iSlopeCount += iRate;
-    }
-    // now always (iSlopeCount > SCALE)
-    iSlopeCount -= SCALE;
-
-    while (1)
-    {
-        while (iSlopeCount > SCALE) 
-        {
-            iSlopeCount -= SCALE;
-            used ++;
-            if (used >= nSamples - 1) goto end;
-        }
-        vol1 = (LONG_SAMPLETYPE)(SCALE - iSlopeCount);
-        temp = src[used] * vol1 + iSlopeCount * src[used + 1];
-        dest[i] = (SAMPLETYPE)(temp / SCALE);
-
-        i++;
-        iSlopeCount += iRate;
-    }
-end:
-    // Store the last sample for the next round
-    sPrevSampleL = src[nSamples - 1];
-
-    return i;
 }
 
 
 // Transposes the sample rate of the given samples using linear interpolation.
-// 'Stereo' version of the routine. Returns the number of samples returned in 
+// Returns the number of samples returned in the "dest" buffer
-// the "dest" buffer
+int TransposerBase::transpose(FIFOSampleBuffer &dest, FIFOSampleBuffer &src)
-uint RateTransposerInteger::transposeStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, uint nSamples)
 {
-    unsigned int srcPos, i, used;
+    int numSrcSamples = src.numSamples();
-    LONG_SAMPLETYPE temp, vol1;
+    int sizeDemand = (int)((double)numSrcSamples / rate) + 8;
+    int numOutput;
+    SAMPLETYPE *psrc = src.ptrBegin();
+    SAMPLETYPE *pdest = dest.ptrEnd(sizeDemand);
 
-    if (nSamples == 0) return 0;  // no samples, no work
+#ifndef USE_MULTICH_ALWAYS
-
+    if (numChannels == 1)
-    used = 0;    
-    i = 0;
-
-    // Process the last sample saved from the sPrevSampleLious call first...
-    while (iSlopeCount <= SCALE) 
     {
-        vol1 = (LONG_SAMPLETYPE)(SCALE - iSlopeCount);
+        numOutput = transposeMono(pdest, psrc, numSrcSamples);
-        temp = vol1 * sPrevSampleL + iSlopeCount * src[0];
-        dest[2 * i] = (SAMPLETYPE)(temp / SCALE);
-        temp = vol1 * sPrevSampleR + iSlopeCount * src[1];
-        dest[2 * i + 1] = (SAMPLETYPE)(temp / SCALE);
-        i++;
-        iSlopeCount += iRate;
     }
-    // now always (iSlopeCount > SCALE)
+    else if (numChannels == 2)
-    iSlopeCount -= SCALE;
-
-    while (1)
     {
-        while (iSlopeCount > SCALE) 
+        numOutput = transposeStereo(pdest, psrc, numSrcSamples);
+    }
+    else
+#endif // USE_MULTICH_ALWAYS
     {
-            iSlopeCount -= SCALE;
+        assert(numChannels > 0);
-            used ++;
+        numOutput = transposeMulti(pdest, psrc, numSrcSamples);
-            if (used >= nSamples - 1) goto end;
     }
-        srcPos = 2 * used;
+    dest.putSamples(numOutput);
-        vol1 = (LONG_SAMPLETYPE)(SCALE - iSlopeCount);
+    src.receiveSamples(numSrcSamples);
-        temp = src[srcPos] * vol1 + iSlopeCount * src[srcPos + 2];
+    return numOutput;
-        dest[2 * i] = (SAMPLETYPE)(temp / SCALE);
-        temp = src[srcPos + 1] * vol1 + iSlopeCount * src[srcPos + 3];
-        dest[2 * i + 1] = (SAMPLETYPE)(temp / SCALE);
-
-        i++;
-        iSlopeCount += iRate;
-    }
-end:
-    // Store the last sample for the next round
-    sPrevSampleL = src[2 * nSamples - 2];
-    sPrevSampleR = src[2 * nSamples - 1];
-
-    return i;
 }
 
 
-// Sets new target iRate. Normal iRate = 1.0, smaller values represent slower 
+TransposerBase::TransposerBase()
-// iRate, larger faster iRates.
-void RateTransposerInteger::setRate(float newRate)
 {
-    iRate = (int)(newRate * SCALE + 0.5f);
+    numChannels = 0;
-    RateTransposer::setRate(newRate);
+    rate = 1.0f;
 }
 
 
-//////////////////////////////////////////////////////////////////////////////
+TransposerBase::~TransposerBase()
-//
-// RateTransposerFloat - floating point arithmetic implementation
-// 
-//////////////////////////////////////////////////////////////////////////////
-
-// Constructor
-RateTransposerFloat::RateTransposerFloat() : RateTransposer()
-{
-    // Notice: use local function calling syntax for sake of clarity, 
-    // to indicate the fact that C++ constructor can't call virtual functions.
-    RateTransposerFloat::resetRegisters();
-    RateTransposerFloat::setRate(1.0f);
-}
-
-
-RateTransposerFloat::~RateTransposerFloat()
 {
 }
 
 
-void RateTransposerFloat::resetRegisters()
+void TransposerBase::setChannels(int channels)
 {
-    fSlopeCount = 0;
+    numChannels = channels;
-    sPrevSampleL = 
+    resetRegisters();
-    sPrevSampleR = 0;
 }
 
 
-
+void TransposerBase::setRate(double newRate)
-// Transposes the sample rate of the given samples using linear interpolation. 
-// 'Mono' version of the routine. Returns the number of samples returned in 
-// the "dest" buffer
-uint RateTransposerFloat::transposeMono(SAMPLETYPE *dest, const SAMPLETYPE *src, uint nSamples)
 {
-    unsigned int i, used;
+    rate = newRate;
-
-    used = 0;    
-    i = 0;
-
-    // Process the last sample saved from the previous call first...
-    while (fSlopeCount <= 1.0f) 
-    {
-        dest[i] = (SAMPLETYPE)((1.0f - fSlopeCount) * sPrevSampleL + fSlopeCount * src[0]);
-        i++;
-        fSlopeCount += fRate;
-    }
-    fSlopeCount -= 1.0f;
-
-    if (nSamples > 1)
-    {
-        while (1)
-        {
-            while (fSlopeCount > 1.0f) 
-            {
-                fSlopeCount -= 1.0f;
-                used ++;
-                if (used >= nSamples - 1) goto end;
-            }
-            dest[i] = (SAMPLETYPE)((1.0f - fSlopeCount) * src[used] + fSlopeCount * src[used + 1]);
-            i++;
-            fSlopeCount += fRate;
-        }
-    }
-end:
-    // Store the last sample for the next round
-    sPrevSampleL = src[nSamples - 1];
-
-    return i;
 }
 
 
-// Transposes the sample rate of the given samples using linear interpolation. 
+// static factory function
-// 'Mono' version of the routine. Returns the number of samples returned in 
+TransposerBase *TransposerBase::newInstance()
-// the "dest" buffer
-uint RateTransposerFloat::transposeStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, uint nSamples)
 {
-    unsigned int srcPos, i, used;
+#ifdef SOUNDTOUCH_INTEGER_SAMPLES
-
+    // Notice: For integer arithmetic support only linear algorithm (due to simplest calculus)
-    if (nSamples == 0) return 0;  // no samples, no work
+    return ::new InterpolateLinearInteger;
-
+#else
-    used = 0;    
+    switch (algorithm)
-    i = 0;
-
-    // Process the last sample saved from the sPrevSampleLious call first...
-    while (fSlopeCount <= 1.0f) 
     {
-        dest[2 * i] = (SAMPLETYPE)((1.0f - fSlopeCount) * sPrevSampleL + fSlopeCount * src[0]);
+        case LINEAR:
-        dest[2 * i + 1] = (SAMPLETYPE)((1.0f - fSlopeCount) * sPrevSampleR + fSlopeCount * src[1]);
+            return new InterpolateLinearFloat;
-        i++;
-        fSlopeCount += fRate;
-    }
-    // now always (iSlopeCount > 1.0f)
-    fSlopeCount -= 1.0f;
 
-    if (nSamples > 1)
+        case CUBIC:
-    {
+            return new InterpolateCubic;
-        while (1)
-        {
-            while (fSlopeCount > 1.0f) 
-            {
-                fSlopeCount -= 1.0f;
-                used ++;
-                if (used >= nSamples - 1) goto end;
-            }
-            srcPos = 2 * used;
 
-            dest[2 * i] = (SAMPLETYPE)((1.0f - fSlopeCount) * src[srcPos] 
+        case SHANNON:
-                + fSlopeCount * src[srcPos + 2]);
+            return new InterpolateShannon;
-            dest[2 * i + 1] = (SAMPLETYPE)((1.0f - fSlopeCount) * src[srcPos + 1] 
-                + fSlopeCount * src[srcPos + 3]);
 
-            i++;
+        default:
-            fSlopeCount += fRate;
+            assert(false);
+            return nullptr;
     }
-    }
+#endif
-end:
-    // Store the last sample for the next round
-    sPrevSampleL = src[2 * nSamples - 2];
-    sPrevSampleR = src[2 * nSamples - 1];
-
-    return i;
 }

source/SoundTouch/RateTransposer.h

@@ -14,13 +14,6 @@
 ///
 ////////////////////////////////////////////////////////////////////////////////
 //
-// Last changed  : $Date$
-// File revision : $Revision: 4 $
-//
-// $Id$
-//
-////////////////////////////////////////////////////////////////////////////////
-//
 // License :
 //
 //  SoundTouch audio processing library
@@ -55,50 +48,72 @@
 namespace soundtouch
 {
 
+/// Abstract base class for transposer implementations (linear, advanced vs integer, float etc)
+class TransposerBase
+{
+public:
+        enum ALGORITHM {
+        LINEAR = 0,
+        CUBIC,
+        SHANNON
+    };
+
+protected:
+    virtual int transposeMono(SAMPLETYPE *dest,
+                        const SAMPLETYPE *src,
+                        int &srcSamples)  = 0;
+    virtual int transposeStereo(SAMPLETYPE *dest,
+                        const SAMPLETYPE *src,
+                        int &srcSamples) = 0;
+    virtual int transposeMulti(SAMPLETYPE *dest,
+                        const SAMPLETYPE *src,
+                        int &srcSamples) = 0;
+
+    static ALGORITHM algorithm;
+
+public:
+    double rate;
+    int numChannels;
+
+    TransposerBase();
+    virtual ~TransposerBase();
+
+    virtual int transpose(FIFOSampleBuffer &dest, FIFOSampleBuffer &src);
+    virtual void setRate(double newRate);
+    virtual void setChannels(int channels);
+    virtual int getLatency() const = 0;
+
+    virtual void resetRegisters() = 0;
+
+    // static factory function
+    static TransposerBase *newInstance();
+
+    // static function to set interpolation algorithm
+    static void setAlgorithm(ALGORITHM a);
+};
+
+
 /// A common linear samplerate transposer class.
 ///
-/// Note: Use function "RateTransposer::newInstance()" to create a new class 
-/// instance instead of the "new" operator; that function automatically 
-/// chooses a correct implementation depending on if integer or floating 
-/// arithmetics are to be used.
 class RateTransposer : public FIFOProcessor
 {
 protected:
     /// Anti-alias filter object
     AAFilter *pAAFilter;
-
+    TransposerBase *pTransposer;
-    float fRate;
-
-    int numChannels;
 
     /// Buffer for collecting samples to feed the anti-alias filter between
     /// two batches
-    FIFOSampleBuffer storeBuffer;
+    FIFOSampleBuffer inputBuffer;
 
     /// Buffer for keeping samples between transposing & anti-alias filter
-    FIFOSampleBuffer tempBuffer;
+    FIFOSampleBuffer midBuffer;
 
     /// Output sample buffer
     FIFOSampleBuffer outputBuffer;
 
-    BOOL bUseAAFilter;
+    bool bUseAAFilter;
 
-    virtual void resetRegisters() = 0;
-
-    virtual uint transposeStereo(SAMPLETYPE *dest, 
-                         const SAMPLETYPE *src, 
-                         uint numSamples) = 0;
-    virtual uint transposeMono(SAMPLETYPE *dest, 
-                       const SAMPLETYPE *src, 
-                       uint numSamples) = 0;
-    inline uint transpose(SAMPLETYPE *dest, 
-                   const SAMPLETYPE *src, 
-                   uint numSamples);
-
-    void downsample(const SAMPLETYPE *src, 
-                    uint numSamples);
-    void upsample(const SAMPLETYPE *src, 
-                 uint numSamples);
 
     /// Transposes sample rate by applying anti-alias filter to prevent folding.
     /// Returns amount of samples returned in the "dest" buffer.
@@ -107,51 +122,41 @@ protected:
     void processSamples(const SAMPLETYPE *src,
                         uint numSamples);
 
-
 public:
     RateTransposer();
-    virtual ~RateTransposer();
+    virtual ~RateTransposer() override;
-
-    /// Operator 'new' is overloaded so that it automatically creates a suitable instance 
-    /// depending on if we're to use integer or floating point arithmetics.
-    static void *operator new(size_t s);
-
-    /// Use this function instead of "new" operator to create a new instance of this class. 
-    /// This function automatically chooses a correct implementation, depending on if 
-    /// integer ot floating point arithmetics are to be used.
-    static RateTransposer *newInstance();
 
     /// Returns the output buffer object
     FIFOSamplePipe *getOutput() { return &outputBuffer; };
 
-    /// Returns the store buffer object
-    FIFOSamplePipe *getStore() { return &storeBuffer; };
-
     /// Return anti-alias filter object
     AAFilter *getAAFilter();
 
     /// Enables/disables the anti-alias filter. Zero to disable, nonzero to enable
-    void enableAAFilter(BOOL newMode);
+    void enableAAFilter(bool newMode);
 
     /// Returns nonzero if anti-alias filter is enabled.
-    BOOL isAAFilterEnabled() const;
+    bool isAAFilterEnabled() const;
 
     /// Sets new target rate. Normal rate = 1.0, smaller values represent slower
     /// rate, larger faster rates.
-    virtual void setRate(float newRate);
+    virtual void setRate(double newRate);
 
     /// Sets the number of channels, 1 = mono, 2 = stereo
     void setChannels(int channels);
 
     /// Adds 'numSamples' pcs of samples from the 'samples' memory position into
     /// the input of the object.
-    void putSamples(const SAMPLETYPE *samples, uint numSamples);
+    void putSamples(const SAMPLETYPE *samples, uint numSamples) override;
 
     /// Clears all the samples in the object
-    void clear();
+    void clear() override;
 
     /// Returns nonzero if there aren't any samples available for outputting.
-    int isEmpty() const;
+    int isEmpty() const override;
+
+    /// Return approximate initial input-output latency
+    int getLatency() const;
 };
 
 }

source/SoundTouch/SoundTouch.cpp

@@ -41,13 +41,6 @@
 ///
 ////////////////////////////////////////////////////////////////////////////////
 //
-// Last changed  : $Date$
-// File revision : $Revision: 4 $
-//
-// $Id$
-//
-////////////////////////////////////////////////////////////////////////////////
-//
 // License :
 //
 //  SoundTouch audio processing library
@@ -97,7 +90,7 @@ SoundTouch::SoundTouch()
 {
     // Initialize rate transposer and tempo changer instances
 
-    pRateTransposer = RateTransposer::newInstance();
+    pRateTransposer = new RateTransposer();
     pTDStretch = TDStretch::newInstance();
 
     setOutPipe(pTDStretch);
@@ -110,10 +103,12 @@ SoundTouch::SoundTouch()
 
     calcEffectiveRateAndTempo();
 
-    channels = 0;
+    samplesExpectedOut = 0;
-    bSrateSet = FALSE;
+    samplesOutput = 0;
-}
 
+    channels = 0;
+    bSrateSet = false;
+}
 
 
 SoundTouch::~SoundTouch()
@@ -123,7 +118,6 @@ SoundTouch::~SoundTouch()
 }
 
 
-
 /// Get SoundTouch library version string
 const char *SoundTouch::getVersionString()
 {
@@ -143,89 +137,79 @@ uint SoundTouch::getVersionId()
 // Sets the number of channels, 1 = mono, 2 = stereo
 void SoundTouch::setChannels(uint numChannels)
 {
-    if (numChannels != 1 && numChannels != 2) 
+    if (!verifyNumberOfChannels(numChannels)) return;
-    {
+
-        ST_THROW_RT_ERROR("Illegal number of channels");
-    }
     channels = numChannels;
     pRateTransposer->setChannels((int)numChannels);
     pTDStretch->setChannels((int)numChannels);
 }
 
 
-
 // Sets new rate control value. Normal rate = 1.0, smaller values
 // represent slower rate, larger faster rates.
-void SoundTouch::setRate(float newRate)
+void SoundTouch::setRate(double newRate)
 {
     virtualRate = newRate;
     calcEffectiveRateAndTempo();
 }
 
 
-
 // Sets new rate control value as a difference in percents compared
 // to the original rate (-50 .. +100 %)
-void SoundTouch::setRateChange(float newRate)
+void SoundTouch::setRateChange(double newRate)
 {
-    virtualRate = 1.0f + 0.01f * newRate;
+    virtualRate = 1.0 + 0.01 * newRate;
     calcEffectiveRateAndTempo();
 }
 
 
-
 // Sets new tempo control value. Normal tempo = 1.0, smaller values
 // represent slower tempo, larger faster tempo.
-void SoundTouch::setTempo(float newTempo)
+void SoundTouch::setTempo(double newTempo)
 {
     virtualTempo = newTempo;
     calcEffectiveRateAndTempo();
 }
 
 
-
 // Sets new tempo control value as a difference in percents compared
 // to the original tempo (-50 .. +100 %)
-void SoundTouch::setTempoChange(float newTempo)
+void SoundTouch::setTempoChange(double newTempo)
 {
-    virtualTempo = 1.0f + 0.01f * newTempo;
+    virtualTempo = 1.0 + 0.01 * newTempo;
     calcEffectiveRateAndTempo();
 }
 
 
-
 // Sets new pitch control value. Original pitch = 1.0, smaller values
 // represent lower pitches, larger values higher pitch.
-void SoundTouch::setPitch(float newPitch)
+void SoundTouch::setPitch(double newPitch)
 {
     virtualPitch = newPitch;
     calcEffectiveRateAndTempo();
 }
 
 
-
 // Sets pitch change in octaves compared to the original pitch
 // (-1.00 .. +1.00)
-void SoundTouch::setPitchOctaves(float newPitch)
+void SoundTouch::setPitchOctaves(double newPitch)
 {
-    virtualPitch = (float)exp(0.69314718056f * newPitch);
+    virtualPitch = exp(0.69314718056 * newPitch);
     calcEffectiveRateAndTempo();
 }
 
 
-
 // Sets pitch change in semi-tones compared to the original pitch
 // (-12 .. +12)
 void SoundTouch::setPitchSemiTones(int newPitch)
 {
-    setPitchOctaves((float)newPitch / 12.0f);
+    setPitchOctaves((double)newPitch / 12.0);
 }
 
 
-
+void SoundTouch::setPitchSemiTones(double newPitch)
-void SoundTouch::setPitchSemiTones(float newPitch)
 {
-    setPitchOctaves(newPitch / 12.0f);
+    setPitchOctaves(newPitch / 12.0);
 }
 
 
@@ -233,8 +217,8 @@ void SoundTouch::setPitchSemiTones(float newPitch)
 // nominal control values.
 void SoundTouch::calcEffectiveRateAndTempo()
 {
-    float oldTempo = tempo;
+    double oldTempo = tempo;
-    float oldRate = rate;
+    double oldRate = rate;
 
     tempo = virtualTempo / virtualPitch;
     rate = virtualPitch * virtualRate;
@@ -254,7 +238,7 @@ void SoundTouch::calcEffectiveRateAndTempo()
             tempoOut = pTDStretch->getOutput();
             tempoOut->moveSamples(*output);
             // move samples in pitch transposer's store buffer to tempo changer's input
-            pTDStretch->moveSamples(*pRateTransposer->getStore());
+            // deprecated : pTDStretch->moveSamples(*pRateTransposer->getStore());
 
             output = pTDStretch;
         }
@@ -282,9 +266,9 @@ void SoundTouch::calcEffectiveRateAndTempo()
 // Sets sample rate.
 void SoundTouch::setSampleRate(uint srate)
 {
-    bSrateSet = TRUE;
     // set sample rate, leave other tempo changer parameters as they are.
     pTDStretch->setParameters((int)srate);
+    bSrateSet = true;
 }
 
 
@@ -292,7 +276,7 @@ void SoundTouch::setSampleRate(uint srate)
 // the input of the object.
 void SoundTouch::putSamples(const SAMPLETYPE *samples, uint nSamples)
 {
-    if (bSrateSet == FALSE) 
+    if (bSrateSet == false)
     {
         ST_THROW_RT_ERROR("SoundTouch : Sample rate not defined");
     }
@@ -301,23 +285,12 @@ void SoundTouch::putSamples(const SAMPLETYPE *samples, uint nSamples)
         ST_THROW_RT_ERROR("SoundTouch : Number of channels not defined");
     }
 
-    // Transpose the rate of the new samples if necessary
+    // accumulate how many samples are expected out from processing, given the current
-    /* Bypass the nominal setting - can introduce a click in sound when tempo/pitch control crosses the nominal value...
+    // processing setting
-    if (rate == 1.0f) 
+    samplesExpectedOut += (double)nSamples / ((double)rate * (double)tempo);
-    {
+
-        // The rate value is same as the original, simply evaluate the tempo changer. 
-        assert(output == pTDStretch);
-        if (pRateTransposer->isEmpty() == 0) 
-        {
-            // yet flush the last samples in the pitch transposer buffer
-            // (may happen if 'rate' changes from a non-zero value to zero)
-            pTDStretch->moveSamples(*pRateTransposer);
-        }
-        pTDStretch->putSamples(samples, nSamples);
-    } 
-    */
 #ifndef SOUNDTOUCH_PREVENT_CLICK_AT_RATE_CROSSOVER
-    else if (rate <= 1.0f) 
+    if (rate <= 1.0f)
     {
         // transpose the rate down, output the transposed sound to tempo changer buffer
         assert(output == pTDStretch);
@@ -345,49 +318,37 @@ void SoundTouch::putSamples(const SAMPLETYPE *samples, uint nSamples)
 void SoundTouch::flush()
 {
     int i;
-    int nUnprocessed;
+    int numStillExpected;
-    int nOut;
+    SAMPLETYPE *buff = new SAMPLETYPE[128 * channels];
-    SAMPLETYPE buff[64*2];   // note: allocate 2*64 to cater 64 sample frames of stereo sound
 
-    // check how many samples still await processing, and scale
+    // how many samples are still expected to output
-    // that by tempo & rate to get expected output sample count
+    numStillExpected = (int)((long)(samplesExpectedOut + 0.5) - samplesOutput);
-    nUnprocessed = numUnprocessedSamples();
+    if (numStillExpected < 0) numStillExpected = 0;
-    nUnprocessed = (int)((double)nUnprocessed / (tempo * rate) + 0.5);
 
-    nOut = numSamples();        // ready samples currently in buffer ...
+    memset(buff, 0, 128 * channels * sizeof(SAMPLETYPE));
-    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
-    // 8ksamples in any case)
+    // 24ksamples in any case)
-    for (i = 0; i < 128; i ++) 
+    for (i = 0; (numStillExpected > (int)numSamples()) && (i < 200); i ++)
     {
-        putSamples(buff, 64);
+        putSamples(buff, 128);
-        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
+    adjustAmountOfSamples(numStillExpected);
-    pRateTransposer->clear();
+
+    delete[] buff;
+
+    // Clear input buffers
     pTDStretch->clearInput();
-    // yet leave the 'tempoChanger' output intouched as that's where the
+    // yet leave the output intouched as that's where the
     // flushed samples are!
 }
 
 
 // Changes a setting controlling the processing system behaviour. See the
 // 'SETTING_...' defines for available setting ID's.
-BOOL SoundTouch::setSetting(int settingId, int value)
+bool SoundTouch::setSetting(int settingId, int value)
 {
     int sampleRate, sequenceMs, seekWindowMs, overlapMs;
 
@@ -398,36 +359,36 @@ BOOL SoundTouch::setSetting(int settingId, int value)
     {
         case SETTING_USE_AA_FILTER :
             // enables / disabless anti-alias filter
-            pRateTransposer->enableAAFilter((value != 0) ? TRUE : FALSE);
+            pRateTransposer->enableAAFilter((value != 0) ? true : false);
-            return TRUE;
+            return true;
 
         case SETTING_AA_FILTER_LENGTH :
             // sets anti-alias filter length
             pRateTransposer->getAAFilter()->setLength(value);
-            return TRUE;
+            return true;
 
         case SETTING_USE_QUICKSEEK :
             // enables / disables tempo routine quick seeking algorithm
-            pTDStretch->enableQuickSeek((value != 0) ? TRUE : FALSE);
+            pTDStretch->enableQuickSeek((value != 0) ? true : false);
-            return TRUE;
+            return true;
 
         case SETTING_SEQUENCE_MS:
             // change time-stretch sequence duration parameter
             pTDStretch->setParameters(sampleRate, value, seekWindowMs, overlapMs);
-            return TRUE;
+            return true;
 
         case SETTING_SEEKWINDOW_MS:
             // change time-stretch seek window length parameter
             pTDStretch->setParameters(sampleRate, sequenceMs, value, overlapMs);
-            return TRUE;
+            return true;
 
         case SETTING_OVERLAP_MS:
             // change time-stretch overlap length parameter
             pTDStretch->setParameters(sampleRate, sequenceMs, seekWindowMs, value);
-            return TRUE;
+            return true;
 
         default :
-            return FALSE;
+            return false;
     }
 }
 
@@ -449,25 +410,65 @@ int SoundTouch::getSetting(int settingId) const
             return pRateTransposer->getAAFilter()->getLength();
 
         case SETTING_USE_QUICKSEEK :
-            return (uint)   pTDStretch->isQuickSeekEnabled();
+            return (uint)pTDStretch->isQuickSeekEnabled();
 
         case SETTING_SEQUENCE_MS:
-            pTDStretch->getParameters(NULL, &temp, NULL, NULL);
+            pTDStretch->getParameters(nullptr, &temp, nullptr, nullptr);
             return temp;
 
         case SETTING_SEEKWINDOW_MS:
-            pTDStretch->getParameters(NULL, NULL, &temp, NULL);
+            pTDStretch->getParameters(nullptr, nullptr, &temp, nullptr);
             return temp;
 
         case SETTING_OVERLAP_MS:
-            pTDStretch->getParameters(NULL, NULL, NULL, &temp);
+            pTDStretch->getParameters(nullptr, nullptr, nullptr, &temp);
             return temp;
 
         case SETTING_NOMINAL_INPUT_SEQUENCE :
-			return pTDStretch->getInputSampleReq();
+        {
+            int size = pTDStretch->getInputSampleReq();
+
+#ifndef SOUNDTOUCH_PREVENT_CLICK_AT_RATE_CROSSOVER
+            if (rate <= 1.0)
+            {
+                // transposing done before timestretch, which impacts latency
+                return (int)(size * rate + 0.5);
+            }
+#endif
+            return size;
+        }
 
         case SETTING_NOMINAL_OUTPUT_SEQUENCE :
-			return pTDStretch->getOutputBatchSize();
+        {
+            int size = pTDStretch->getOutputBatchSize();
+
+            if (rate > 1.0)
+            {
+                // transposing done after timestretch, which impacts latency
+                return (int)(size / rate + 0.5);
+            }
+            return size;
+        }
+
+        case SETTING_INITIAL_LATENCY:
+        {
+            double latency = pTDStretch->getLatency();
+            int latency_tr = pRateTransposer->getLatency();
+
+#ifndef SOUNDTOUCH_PREVENT_CLICK_AT_RATE_CROSSOVER
+            if (rate <= 1.0)
+            {
+                // transposing done before timestretch, which impacts latency
+                latency = (latency + latency_tr) * rate;
+            }
+            else
+#endif
+            {
+                latency += (double)latency_tr / rate;
+            }
+
+            return (int)(latency + 0.5);
+        }
 
         default :
             return 0;
@@ -479,12 +480,13 @@ int SoundTouch::getSetting(int settingId) const
 // buffers.
 void SoundTouch::clear()
 {
+    samplesExpectedOut = 0;
+    samplesOutput = 0;
     pRateTransposer->clear();
     pTDStretch->clear();
 }
 
 
-
 /// Returns number of samples currently unprocessed.
 uint SoundTouch::numUnprocessedSamples() const
 {
@@ -499,3 +501,38 @@ uint SoundTouch::numUnprocessedSamples() const
     }
     return 0;
 }
+
+
+/// Output samples from beginning of the sample buffer. Copies requested samples to
+/// output buffer and removes them from the sample buffer. If there are less than
+/// 'numsample' samples in the buffer, returns all that available.
+///
+/// \return Number of samples returned.
+uint SoundTouch::receiveSamples(SAMPLETYPE *output, uint maxSamples)
+{
+    uint ret = FIFOProcessor::receiveSamples(output, maxSamples);
+    samplesOutput += (long)ret;
+    return ret;
+}
+
+
+/// Adjusts book-keeping so that given number of samples are removed from beginning of the
+/// sample buffer without copying them anywhere.
+///
+/// Used to reduce the number of samples in the buffer when accessing the sample buffer directly
+/// with 'ptrBegin' function.
+uint SoundTouch::receiveSamples(uint maxSamples)
+{
+    uint ret = FIFOProcessor::receiveSamples(maxSamples);
+    samplesOutput += (long)ret;
+    return ret;
+}
+
+
+/// Get ratio between input and output audio durations, useful for calculating
+/// processed output duration: if you'll process a stream of N samples, then
+/// you can expect to get out N * getInputOutputSampleRatio() samples.
+double SoundTouch::getInputOutputSampleRatio()
+{
+    return 1.0 / (tempo * rate);
+}

source/SoundTouch/SoundTouch.dsp

@@ -1,188 +0,0 @@
-# Microsoft Developer Studio Project File - Name="SoundTouch" - Package Owner=<4>
-# Microsoft Developer Studio Generated Build File, Format Version 6.00
-# ** DO NOT EDIT **
-
-# TARGTYPE "Win32 (x86) Static Library" 0x0104
-
-CFG=SoundTouch - Win32 Debug
-!MESSAGE This is not a valid makefile. To build this project using NMAKE,
-!MESSAGE use the Export Makefile command and run
-!MESSAGE 
-!MESSAGE NMAKE /f "SoundTouch.mak".
-!MESSAGE 
-!MESSAGE You can specify a configuration when running NMAKE
-!MESSAGE by defining the macro CFG on the command line. For example:
-!MESSAGE 
-!MESSAGE NMAKE /f "SoundTouch.mak" CFG="SoundTouch - Win32 Debug"
-!MESSAGE 
-!MESSAGE Possible choices for configuration are:
-!MESSAGE 
-!MESSAGE "SoundTouch - Win32 Release" (based on "Win32 (x86) Static Library")
-!MESSAGE "SoundTouch - Win32 Debug" (based on "Win32 (x86) Static Library")
-!MESSAGE 
-
-# Begin Project
-# PROP AllowPerConfigDependencies 0
-# PROP Scc_ProjName ""
-# PROP Scc_LocalPath ""
-CPP=cl.exe
-RSC=rc.exe
-
-!IF  "$(CFG)" == "SoundTouch - Win32 Release"
-
-# PROP BASE Use_MFC 0
-# PROP BASE Use_Debug_Libraries 0
-# PROP BASE Output_Dir "Release"
-# PROP BASE Intermediate_Dir "Release"
-# PROP BASE Target_Dir ""
-# PROP Use_MFC 0
-# PROP Use_Debug_Libraries 0
-# PROP Output_Dir "Release"
-# PROP Intermediate_Dir "Release"
-# PROP Target_Dir ""
-# ADD BASE CPP /nologo /W3 /GX /O2 /D "WIN32" /D "NDEBUG" /D "_MBCS" /D "_LIB" /YX /FD /c
-# ADD CPP /nologo /W3 /GX /Zi /O2 /I "..\..\include" /D "WIN32" /D "NDEBUG" /D "_MBCS" /D "_LIB" /YX /FD /c
-# ADD BASE RSC /l 0x40b /d "NDEBUG"
-# ADD RSC /l 0x40b /d "NDEBUG"
-BSC32=bscmake.exe
-# ADD BASE BSC32 /nologo
-# ADD BSC32 /nologo
-LIB32=link.exe -lib
-# ADD BASE LIB32 /nologo
-# ADD LIB32 /nologo
-# Begin Special Build Tool
-SOURCE="$(InputPath)"
-PostBuild_Cmds=copy            .\Release\SoundTouch.lib            ..\..\lib\ 
-# End Special Build Tool
-
-!ELSEIF  "$(CFG)" == "SoundTouch - Win32 Debug"
-
-# PROP BASE Use_MFC 0
-# PROP BASE Use_Debug_Libraries 1
-# PROP BASE Output_Dir "Debug"
-# PROP BASE Intermediate_Dir "Debug"
-# PROP BASE Target_Dir ""
-# PROP Use_MFC 0
-# PROP Use_Debug_Libraries 1
-# PROP Output_Dir "Debug"
-# PROP Intermediate_Dir "Debug"
-# PROP Target_Dir ""
-# ADD BASE CPP /nologo /W3 /Gm /GX /ZI /Od /D "WIN32" /D "_DEBUG" /D "_MBCS" /D "_LIB" /YX /FD /GZ /c
-# ADD CPP /nologo /W3 /Gm /GX /ZI /Od /I "..\..\include" /D "WIN32" /D "_DEBUG" /D "_MBCS" /D "_LIB" /FR /YX /FD /GZ /c
-# ADD BASE RSC /l 0x40b /d "_DEBUG"
-# ADD RSC /l 0x40b /d "_DEBUG"
-BSC32=bscmake.exe
-# ADD BASE BSC32 /nologo
-# ADD BSC32 /nologo
-LIB32=link.exe -lib
-# ADD BASE LIB32 /nologo
-# ADD LIB32 /nologo /out:"Debug\SoundTouchD.lib"
-# Begin Special Build Tool
-SOURCE="$(InputPath)"
-PostBuild_Cmds=copy            .\Debug\SoundTouchD.lib            ..\..\lib\ 
-# End Special Build Tool
-
-!ENDIF 
-
-# Begin Target
-
-# Name "SoundTouch - Win32 Release"
-# Name "SoundTouch - Win32 Debug"
-# 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
-# End Source File
-# Begin Source File
-
-SOURCE=.\AAFilter.cpp
-# End Source File
-# Begin Source File
-
-SOURCE=.\cpu_detect_x86_win.cpp
-# End Source File
-# Begin Source File
-
-SOURCE=.\FIFOSampleBuffer.cpp
-# End Source File
-# Begin Source File
-
-SOURCE=.\FIRFilter.cpp
-# End Source File
-# Begin Source File
-
-SOURCE=.\mmx_optimized.cpp
-# End Source File
-# Begin Source File
-
-SOURCE=.\RateTransposer.cpp
-# End Source File
-# Begin Source File
-
-SOURCE=.\SoundTouch.cpp
-# End Source File
-# Begin Source File
-
-SOURCE=.\sse_optimized.cpp
-# End Source File
-# Begin Source File
-
-SOURCE=.\TDStretch.cpp
-# End Source File
-# End Group
-# Begin Group "Header Files"
-
-# PROP Default_Filter "h;hpp;hxx;hm;inl"
-# Begin Source File
-
-SOURCE=.\AAFilter.h
-# End Source File
-# Begin Source File
-
-SOURCE=.\cpu_detect.h
-# End Source File
-# Begin Source File
-
-SOURCE=..\..\include\FIFOSampleBuffer.h
-# End Source File
-# Begin Source File
-
-SOURCE=..\..\include\FIFOSamplePipe.h
-# End Source File
-# Begin Source File
-
-SOURCE=.\FIRFilter.h
-# End Source File
-# Begin Source File
-
-SOURCE=.\RateTransposer.h
-# End Source File
-# Begin Source File
-
-SOURCE=..\..\include\SoundTouch.h
-# End Source File
-# Begin Source File
-
-SOURCE=..\..\include\STTypes.h
-# End Source File
-# Begin Source File
-
-SOURCE=.\TDStretch.h
-# End Source File
-# End Group
-# End Target
-# End Project

source/SoundTouch/SoundTouch.dsw

@@ -1,31 +0,0 @@
-Microsoft Developer Studio Workspace File, Format Version 6.00
-# WARNING: DO NOT EDIT OR DELETE THIS WORKSPACE FILE!
-#
-# $Id$
-
-###############################################################################
-
-Project: "SoundTouch"=.\SoundTouch.dsp - Package Owner=<4>
-
-Package=<5>
-{{{
-}}}
-
-Package=<4>
-{{{
-}}}
-
-###############################################################################
-
-Global:
-
-Package=<5>
-{{{
-}}}
-
-Package=<3>
-{{{
-}}}
-
-###############################################################################
-

source/SoundTouch/SoundTouch.sln

@@ -1,21 +1,29 @@
-Microsoft Visual Studio Solution File, Format Version 8.00
+Microsoft Visual Studio Solution File, Format Version 12.00
-Project("{8BC9CEB8-8B4A-11D0-8D11-00A0C91BC942}") = "SoundTouch", "SoundTouch.vcproj", "{0A626C77-0515-4131-AA80-E0BFFC479FEB}"
+# Visual Studio 14
-	ProjectSection(ProjectDependencies) = postProject
+VisualStudioVersion = 14.0.23107.0
-	EndProjectSection
+MinimumVisualStudioVersion = 10.0.40219.1
+Project("{8BC9CEB8-8B4A-11D0-8D11-00A0C91BC942}") = "SoundTouch", "SoundTouch.vcxproj", "{68A5DD20-7057-448B-8FE0-B6AC8D205509}"
 EndProject
 Global
-	GlobalSection(SolutionConfiguration) = preSolution
+	GlobalSection(SolutionConfigurationPlatforms) = preSolution
-		Debug = Debug
+		Debug|Win32 = Debug|Win32
-		Release = Release
+		Debug|x64 = Debug|x64
+		Release|Win32 = Release|Win32
+		Release|x64 = Release|x64
+		ReleaseX64|Win32 = ReleaseX64|Win32
+		ReleaseX64|x64 = ReleaseX64|x64
 	EndGlobalSection
-	GlobalSection(ProjectConfiguration) = postSolution
+	GlobalSection(ProjectConfigurationPlatforms) = postSolution
-		{0A626C77-0515-4131-AA80-E0BFFC479FEB}.Debug.ActiveCfg = Debug|Win32
+		{68A5DD20-7057-448B-8FE0-B6AC8D205509}.Debug|Win32.ActiveCfg = Debug|Win32
-		{0A626C77-0515-4131-AA80-E0BFFC479FEB}.Debug.Build.0 = Debug|Win32
+		{68A5DD20-7057-448B-8FE0-B6AC8D205509}.Debug|x64.ActiveCfg = Debug|x64
-		{0A626C77-0515-4131-AA80-E0BFFC479FEB}.Release.ActiveCfg = Release|Win32
+		{68A5DD20-7057-448B-8FE0-B6AC8D205509}.Release|Win32.ActiveCfg = Release|Win32
-		{0A626C77-0515-4131-AA80-E0BFFC479FEB}.Release.Build.0 = Release|Win32
+		{68A5DD20-7057-448B-8FE0-B6AC8D205509}.Release|x64.ActiveCfg = Release|x64
+		{68A5DD20-7057-448B-8FE0-B6AC8D205509}.ReleaseX64|Win32.ActiveCfg = ReleaseX64|Win32
+		{68A5DD20-7057-448B-8FE0-B6AC8D205509}.ReleaseX64|Win32.Build.0 = ReleaseX64|Win32
+		{68A5DD20-7057-448B-8FE0-B6AC8D205509}.ReleaseX64|x64.ActiveCfg = ReleaseX64|x64
+		{68A5DD20-7057-448B-8FE0-B6AC8D205509}.ReleaseX64|x64.Build.0 = ReleaseX64|x64
 	EndGlobalSection
-	GlobalSection(ExtensibilityGlobals) = postSolution
+	GlobalSection(SolutionProperties) = preSolution
-	EndGlobalSection
+		HideSolutionNode = FALSE
-	GlobalSection(ExtensibilityAddIns) = postSolution
 	EndGlobalSection
 EndGlobal

source/SoundTouch/SoundTouch.vcproj

@@ -1,315 +0,0 @@
-<?xml version="1.0" encoding="Windows-1252"?>
-<VisualStudioProject
-	ProjectType="Visual C++"
-	Version="7.10"
-	Name="SoundTouch"
-	ProjectGUID="{68A5DD20-7057-448B-8FE0-B6AC8D205509}"
-	SccProjectName=""
-	SccLocalPath="">
-	<Platforms>
-		<Platform
-			Name="Win32"/>
-	</Platforms>
-	<Configurations>
-		<Configuration
-			Name="Release|Win32"
-			OutputDirectory=".\Release"
-			IntermediateDirectory=".\Release"
-			ConfigurationType="4"
-			UseOfMFC="0"
-			ATLMinimizesCRunTimeLibraryUsage="FALSE"
-			CharacterSet="2">
-			<Tool
-				Name="VCCLCompilerTool"
-				Optimization="3"
-				InlineFunctionExpansion="2"
-				EnableIntrinsicFunctions="TRUE"
-				AdditionalIncludeDirectories="..\..\include"
-				PreprocessorDefinitions="WIN32;NDEBUG;_LIB"
-				StringPooling="TRUE"
-				RuntimeLibrary="4"
-				EnableFunctionLevelLinking="TRUE"
-				UsePrecompiledHeader="2"
-				PrecompiledHeaderFile=".\Release/SoundTouch.pch"
-				AssemblerListingLocation=".\Release/"
-				ObjectFile=".\Release/"
-				ProgramDataBaseFileName=".\Release/"
-				WarningLevel="3"
-				SuppressStartupBanner="TRUE"
-				DebugInformationFormat="0"
-				CompileAs="0"/>
-			<Tool
-				Name="VCCustomBuildTool"/>
-			<Tool
-				Name="VCLibrarianTool"
-				OutputFile=".\Release\SoundTouch.lib"
-				SuppressStartupBanner="TRUE"/>
-			<Tool
-				Name="VCMIDLTool"/>
-			<Tool
-				Name="VCPostBuildEventTool"
-				CommandLine="copy         .\Release\SoundTouch.lib         ..\..\lib\"/>
-			<Tool
-				Name="VCPreBuildEventTool"/>
-			<Tool
-				Name="VCPreLinkEventTool"/>
-			<Tool
-				Name="VCResourceCompilerTool"
-				PreprocessorDefinitions="NDEBUG"
-				Culture="1035"/>
-			<Tool
-				Name="VCWebServiceProxyGeneratorTool"/>
-			<Tool
-				Name="VCXMLDataGeneratorTool"/>
-			<Tool
-				Name="VCManagedWrapperGeneratorTool"/>
-			<Tool
-				Name="VCAuxiliaryManagedWrapperGeneratorTool"/>
-		</Configuration>
-		<Configuration
-			Name="Debug|Win32"
-			OutputDirectory=".\Debug"
-			IntermediateDirectory=".\Debug"
-			ConfigurationType="4"
-			UseOfMFC="0"
-			ATLMinimizesCRunTimeLibraryUsage="FALSE"
-			CharacterSet="2">
-			<Tool
-				Name="VCCLCompilerTool"
-				Optimization="0"
-				AdditionalIncludeDirectories="..\..\include"
-				PreprocessorDefinitions="WIN32;_DEBUG;_LIB"
-				BasicRuntimeChecks="3"
-				RuntimeLibrary="5"
-				UsePrecompiledHeader="2"
-				PrecompiledHeaderFile=".\Debug/SoundTouch.pch"
-				AssemblerListingLocation=".\Debug/"
-				ObjectFile=".\Debug/"
-				ProgramDataBaseFileName=".\Debug/"
-				BrowseInformation="1"
-				WarningLevel="3"
-				SuppressStartupBanner="TRUE"
-				DebugInformationFormat="4"
-				CompileAs="0"/>
-			<Tool
-				Name="VCCustomBuildTool"/>
-			<Tool
-				Name="VCLibrarianTool"
-				OutputFile="Debug\SoundTouchD.lib"
-				SuppressStartupBanner="TRUE"/>
-			<Tool
-				Name="VCMIDLTool"/>
-			<Tool
-				Name="VCPostBuildEventTool"
-				CommandLine="copy .\Debug\SoundTouchD.lib ..\..\lib\"/>
-			<Tool
-				Name="VCPreBuildEventTool"/>
-			<Tool
-				Name="VCPreLinkEventTool"/>
-			<Tool
-				Name="VCResourceCompilerTool"
-				PreprocessorDefinitions="_DEBUG"
-				Culture="1035"/>
-			<Tool
-				Name="VCWebServiceProxyGeneratorTool"/>
-			<Tool
-				Name="VCXMLDataGeneratorTool"/>
-			<Tool
-				Name="VCManagedWrapperGeneratorTool"/>
-			<Tool
-				Name="VCAuxiliaryManagedWrapperGeneratorTool"/>
-		</Configuration>
-	</Configurations>
-	<References>
-	</References>
-	<Files>
-		<Filter
-			Name="Source Files"
-			Filter="cpp;c;cxx;rc;def;r;odl;idl;hpj;bat">
-			<File
-				RelativePath="AAFilter.cpp">
-				<FileConfiguration
-					Name="Release|Win32">
-					<Tool
-						Name="VCCLCompilerTool"
-						Optimization="2"
-						AdditionalIncludeDirectories=""
-						PreprocessorDefinitions=""/>
-				</FileConfiguration>
-				<FileConfiguration
-					Name="Debug|Win32">
-					<Tool
-						Name="VCCLCompilerTool"
-						Optimization="0"
-						AdditionalIncludeDirectories=""
-						PreprocessorDefinitions=""
-						BasicRuntimeChecks="3"
-						BrowseInformation="1"/>
-				</FileConfiguration>
-			</File>
-			<File
-				RelativePath=".\cpu_detect_x86.cpp">
-			</File>
-			<File
-				RelativePath="FIFOSampleBuffer.cpp">
-				<FileConfiguration
-					Name="Release|Win32">
-					<Tool
-						Name="VCCLCompilerTool"
-						Optimization="2"
-						AdditionalIncludeDirectories=""
-						PreprocessorDefinitions=""/>
-				</FileConfiguration>
-				<FileConfiguration
-					Name="Debug|Win32">
-					<Tool
-						Name="VCCLCompilerTool"
-						Optimization="0"
-						AdditionalIncludeDirectories=""
-						PreprocessorDefinitions=""
-						BasicRuntimeChecks="3"
-						BrowseInformation="1"/>
-				</FileConfiguration>
-			</File>
-			<File
-				RelativePath="FIRFilter.cpp">
-				<FileConfiguration
-					Name="Release|Win32">
-					<Tool
-						Name="VCCLCompilerTool"
-						Optimization="2"
-						AdditionalIncludeDirectories=""
-						PreprocessorDefinitions=""/>
-				</FileConfiguration>
-				<FileConfiguration
-					Name="Debug|Win32">
-					<Tool
-						Name="VCCLCompilerTool"
-						Optimization="0"
-						AdditionalIncludeDirectories=""
-						PreprocessorDefinitions=""
-						BasicRuntimeChecks="3"
-						BrowseInformation="1"/>
-				</FileConfiguration>
-			</File>
-			<File
-				RelativePath=".\mmx_optimized.cpp">
-			</File>
-			<File
-				RelativePath="RateTransposer.cpp">
-				<FileConfiguration
-					Name="Release|Win32">
-					<Tool
-						Name="VCCLCompilerTool"
-						Optimization="2"
-						AdditionalIncludeDirectories=""
-						PreprocessorDefinitions=""/>
-				</FileConfiguration>
-				<FileConfiguration
-					Name="Debug|Win32">
-					<Tool
-						Name="VCCLCompilerTool"
-						Optimization="0"
-						AdditionalIncludeDirectories=""
-						PreprocessorDefinitions=""
-						BasicRuntimeChecks="3"
-						BrowseInformation="1"/>
-				</FileConfiguration>
-			</File>
-			<File
-				RelativePath="SoundTouch.cpp">
-				<FileConfiguration
-					Name="Release|Win32">
-					<Tool
-						Name="VCCLCompilerTool"
-						Optimization="2"
-						AdditionalIncludeDirectories=""
-						PreprocessorDefinitions=""/>
-				</FileConfiguration>
-				<FileConfiguration
-					Name="Debug|Win32">
-					<Tool
-						Name="VCCLCompilerTool"
-						Optimization="0"
-						AdditionalIncludeDirectories=""
-						PreprocessorDefinitions=""
-						BasicRuntimeChecks="3"
-						BrowseInformation="1"/>
-				</FileConfiguration>
-			</File>
-			<File
-				RelativePath=".\sse_optimized.cpp">
-			</File>
-			<File
-				RelativePath="TDStretch.cpp">
-				<FileConfiguration
-					Name="Release|Win32">
-					<Tool
-						Name="VCCLCompilerTool"
-						Optimization="2"
-						AdditionalIncludeDirectories=""
-						PreprocessorDefinitions=""/>
-				</FileConfiguration>
-				<FileConfiguration
-					Name="Debug|Win32">
-					<Tool
-						Name="VCCLCompilerTool"
-						Optimization="0"
-						AdditionalIncludeDirectories=""
-						PreprocessorDefinitions=""
-						BasicRuntimeChecks="3"
-						BrowseInformation="1"/>
-				</FileConfiguration>
-			</File>
-			<Filter
-				Name="bpm"
-				Filter="">
-				<File
-					RelativePath=".\BPMDetect.cpp">
-				</File>
-				<File
-					RelativePath=".\PeakFinder.cpp">
-				</File>
-			</Filter>
-		</Filter>
-		<Filter
-			Name="Header Files"
-			Filter="h;hpp;hxx;hm;inl">
-			<File
-				RelativePath="AAFilter.h">
-			</File>
-			<File
-				RelativePath="..\..\include\BPMDetect.h">
-			</File>
-			<File
-				RelativePath="cpu_detect.h">
-			</File>
-			<File
-				RelativePath="..\..\include\FIFOSampleBuffer.h">
-			</File>
-			<File
-				RelativePath="..\..\include\FIFOSamplePipe.h">
-			</File>
-			<File
-				RelativePath="FIRFilter.h">
-			</File>
-			<File
-				RelativePath=".\PeakFinder.h">
-			</File>
-			<File
-				RelativePath="RateTransposer.h">
-			</File>
-			<File
-				RelativePath="..\..\include\SoundTouch.h">
-			</File>
-			<File
-				RelativePath="..\..\include\STTypes.h">
-			</File>
-			<File
-				RelativePath="TDStretch.h">
-			</File>
-		</Filter>
-	</Files>
-	<Globals>
-	</Globals>
-</VisualStudioProject>

source/SoundTouch/SoundTouch.vcxproj

@@ -0,0 +1,342 @@
+<?xml version="1.0" encoding="utf-8"?>
+<Project DefaultTargets="Build" ToolsVersion="14.0" xmlns="http://schemas.microsoft.com/developer/msbuild/2003">
+  <ItemGroup Label="ProjectConfigurations">
+    <ProjectConfiguration Include="Debug|Win32">
+      <Configuration>Debug</Configuration>
+      <Platform>Win32</Platform>
+    </ProjectConfiguration>
+    <ProjectConfiguration Include="Debug|x64">
+      <Configuration>Debug</Configuration>
+      <Platform>x64</Platform>
+    </ProjectConfiguration>
+    <ProjectConfiguration Include="Release|Win32">
+      <Configuration>Release</Configuration>
+      <Platform>Win32</Platform>
+    </ProjectConfiguration>
+    <ProjectConfiguration Include="Release|x64">
+      <Configuration>Release</Configuration>
+      <Platform>x64</Platform>
+    </ProjectConfiguration>
+  </ItemGroup>
+  <PropertyGroup Label="Globals">
+    <ProjectGuid>{68A5DD20-7057-448B-8FE0-B6AC8D205509}</ProjectGuid>
+    <WindowsTargetPlatformVersion>10.0</WindowsTargetPlatformVersion>
+  </PropertyGroup>
+  <Import Project="$(VCTargetsPath)\Microsoft.Cpp.Default.props" />
+  <PropertyGroup Condition="'$(Configuration)|$(Platform)'=='Debug|Win32'" Label="Configuration">
+    <ConfigurationType>StaticLibrary</ConfigurationType>
+    <PlatformToolset>v142</PlatformToolset>
+    <UseOfMfc>false</UseOfMfc>
+    <CharacterSet>Unicode</CharacterSet>
+  </PropertyGroup>
+  <PropertyGroup Condition="'$(Configuration)|$(Platform)'=='Release|Win32'" Label="Configuration">
+    <ConfigurationType>StaticLibrary</ConfigurationType>
+    <PlatformToolset>v142</PlatformToolset>
+    <UseOfMfc>false</UseOfMfc>
+    <CharacterSet>Unicode</CharacterSet>
+  </PropertyGroup>
+  <PropertyGroup Condition="'$(Configuration)|$(Platform)'=='Debug|x64'" Label="Configuration">
+    <ConfigurationType>StaticLibrary</ConfigurationType>
+    <PlatformToolset>v142</PlatformToolset>
+    <UseOfMfc>false</UseOfMfc>
+    <CharacterSet>Unicode</CharacterSet>
+  </PropertyGroup>
+  <PropertyGroup Condition="'$(Configuration)|$(Platform)'=='Release|x64'" Label="Configuration">
+    <ConfigurationType>StaticLibrary</ConfigurationType>
+    <PlatformToolset>v142</PlatformToolset>
+    <UseOfMfc>false</UseOfMfc>
+    <CharacterSet>Unicode</CharacterSet>
+  </PropertyGroup>
+  <Import Project="$(VCTargetsPath)\Microsoft.Cpp.props" />
+  <ImportGroup Label="ExtensionSettings">
+  </ImportGroup>
+  <ImportGroup Condition="'$(Configuration)|$(Platform)'=='Debug|Win32'" Label="PropertySheets">
+    <Import Project="$(UserRootDir)\Microsoft.Cpp.$(Platform).user.props" Condition="exists('$(UserRootDir)\Microsoft.Cpp.$(Platform).user.props')" Label="LocalAppDataPlatform" />
+    <Import Project="$(VCTargetsPath)Microsoft.CPP.UpgradeFromVC71.props" />
+  </ImportGroup>
+  <ImportGroup Condition="'$(Configuration)|$(Platform)'=='Release|Win32'" Label="PropertySheets">
+    <Import Project="$(UserRootDir)\Microsoft.Cpp.$(Platform).user.props" Condition="exists('$(UserRootDir)\Microsoft.Cpp.$(Platform).user.props')" Label="LocalAppDataPlatform" />
+    <Import Project="$(VCTargetsPath)Microsoft.CPP.UpgradeFromVC71.props" />
+  </ImportGroup>
+  <ImportGroup Condition="'$(Configuration)|$(Platform)'=='Debug|x64'" Label="PropertySheets">
+    <Import Project="$(UserRootDir)\Microsoft.Cpp.$(Platform).user.props" Condition="exists('$(UserRootDir)\Microsoft.Cpp.$(Platform).user.props')" Label="LocalAppDataPlatform" />
+    <Import Project="$(VCTargetsPath)Microsoft.CPP.UpgradeFromVC71.props" />
+  </ImportGroup>
+  <ImportGroup Condition="'$(Configuration)|$(Platform)'=='Release|x64'" Label="PropertySheets">
+    <Import Project="$(UserRootDir)\Microsoft.Cpp.$(Platform).user.props" Condition="exists('$(UserRootDir)\Microsoft.Cpp.$(Platform).user.props')" Label="LocalAppDataPlatform" />
+    <Import Project="$(VCTargetsPath)Microsoft.CPP.UpgradeFromVC71.props" />
+  </ImportGroup>
+  <PropertyGroup Label="UserMacros" />
+  <PropertyGroup>
+    <_ProjectFileVersion>14.0.23107.0</_ProjectFileVersion>
+  </PropertyGroup>
+  <PropertyGroup Condition="'$(Configuration)|$(Platform)'=='Release|Win32'">
+    <OutDir>$(Platform)\$(Configuration)\</OutDir>
+    <IntDir>$(Platform)\$(Configuration)\</IntDir>
+  </PropertyGroup>
+  <PropertyGroup Condition="'$(Configuration)|$(Platform)'=='Release|x64'">
+    <OutDir>$(Platform)\$(Configuration)\</OutDir>
+    <IntDir>$(Platform)\$(Configuration)\</IntDir>
+    <TargetName>$(ProjectName)_x64</TargetName>
+  </PropertyGroup>
+  <PropertyGroup Condition="'$(Configuration)|$(Platform)'=='Debug|Win32'">
+    <OutDir>$(Platform)\$(Configuration)\</OutDir>
+    <IntDir>$(Platform)\$(Configuration)\</IntDir>
+    <TargetName>$(ProjectName)D</TargetName>
+  </PropertyGroup>
+  <PropertyGroup Condition="'$(Configuration)|$(Platform)'=='Debug|x64'">
+    <OutDir>$(Platform)\$(Configuration)\</OutDir>
+    <IntDir>$(Platform)\$(Configuration)\</IntDir>
+    <TargetName>$(ProjectName)D_x64</TargetName>
+  </PropertyGroup>
+  <ItemDefinitionGroup Condition="'$(Configuration)|$(Platform)'=='Release|Win32'">
+    <ClCompile>
+      <Optimization>Full</Optimization>
+      <InlineFunctionExpansion>AnySuitable</InlineFunctionExpansion>
+      <IntrinsicFunctions>true</IntrinsicFunctions>
+      <AdditionalIncludeDirectories>..\..\include;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>
+      <PreprocessorDefinitions>WIN32;NDEBUG;_LIB;%(PreprocessorDefinitions)</PreprocessorDefinitions>
+      <StringPooling>true</StringPooling>
+      <RuntimeLibrary>MultiThreaded</RuntimeLibrary>
+      <FunctionLevelLinking>true</FunctionLevelLinking>
+      <FloatingPointModel>Fast</FloatingPointModel>
+      <PrecompiledHeader />
+      <PrecompiledHeaderOutputFile>$(OutDir)$(TargetName).pch</PrecompiledHeaderOutputFile>
+      <AssemblerListingLocation>$(OutDir)</AssemblerListingLocation>
+      <ObjectFileName>$(OutDir)</ObjectFileName>
+      <ProgramDataBaseFileName>$(OutDir)</ProgramDataBaseFileName>
+      <WarningLevel>Level3</WarningLevel>
+      <SuppressStartupBanner>true</SuppressStartupBanner>
+      <DebugInformationFormat />
+      <CompileAs>Default</CompileAs>
+      <EnableEnhancedInstructionSet>StreamingSIMDExtensions2</EnableEnhancedInstructionSet>
+      <XMLDocumentationFileName>$(IntDir)</XMLDocumentationFileName>
+      <BrowseInformationFile>$(IntDir)</BrowseInformationFile>
+      <MultiProcessorCompilation>true</MultiProcessorCompilation>
+    </ClCompile>
+    <ResourceCompile>
+      <PreprocessorDefinitions>NDEBUG;%(PreprocessorDefinitions)</PreprocessorDefinitions>
+      <Culture>0x040b</Culture>
+    </ResourceCompile>
+    <Lib>
+      <OutputFile>$(OutDir)$(TargetName)$(TargetExt)</OutputFile>
+      <SuppressStartupBanner>true</SuppressStartupBanner>
+    </Lib>
+    <PostBuildEvent>
+      <Command>if not exist ..\..\lib mkdir ..\..\lib
+copy $(OutDir)$(TargetName)$(TargetExt) ..\..\lib</Command>
+    </PostBuildEvent>
+  </ItemDefinitionGroup>
+  <ItemDefinitionGroup Condition="'$(Configuration)|$(Platform)'=='Release|x64'">
+    <Midl>
+      <TargetEnvironment>X64</TargetEnvironment>
+    </Midl>
+    <ClCompile>
+      <Optimization>Full</Optimization>
+      <InlineFunctionExpansion>AnySuitable</InlineFunctionExpansion>
+      <IntrinsicFunctions>true</IntrinsicFunctions>
+      <AdditionalIncludeDirectories>..\..\include;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>
+      <PreprocessorDefinitions>WIN32;NDEBUG;_LIB;%(PreprocessorDefinitions)</PreprocessorDefinitions>
+      <StringPooling>true</StringPooling>
+      <RuntimeLibrary>MultiThreaded</RuntimeLibrary>
+      <FunctionLevelLinking>true</FunctionLevelLinking>
+      <FloatingPointModel>Fast</FloatingPointModel>
+      <PrecompiledHeader />
+      <PrecompiledHeaderOutputFile>$(OutDir)$(TargetName).pch</PrecompiledHeaderOutputFile>
+      <AssemblerListingLocation>$(OutDir)</AssemblerListingLocation>
+      <ObjectFileName>$(OutDir)</ObjectFileName>
+      <ProgramDataBaseFileName>$(OutDir)</ProgramDataBaseFileName>
+      <WarningLevel>Level3</WarningLevel>
+      <SuppressStartupBanner>true</SuppressStartupBanner>
+      <DebugInformationFormat />
+      <CompileAs>Default</CompileAs>
+      <EnableEnhancedInstructionSet>
+      </EnableEnhancedInstructionSet>
+      <XMLDocumentationFileName>$(IntDir)</XMLDocumentationFileName>
+      <BrowseInformationFile>$(IntDir)</BrowseInformationFile>
+      <MultiProcessorCompilation>true</MultiProcessorCompilation>
+    </ClCompile>
+    <ResourceCompile>
+      <PreprocessorDefinitions>NDEBUG;%(PreprocessorDefinitions)</PreprocessorDefinitions>
+      <Culture>0x040b</Culture>
+    </ResourceCompile>
+    <Lib>
+      <OutputFile>$(OutDir)$(TargetName)$(TargetExt)</OutputFile>
+      <SuppressStartupBanner>true</SuppressStartupBanner>
+    </Lib>
+    <PostBuildEvent>
+      <Command>if not exist ..\..\lib mkdir ..\..\lib
+copy $(OutDir)$(TargetName)$(TargetExt) ..\..\lib</Command>
+    </PostBuildEvent>
+  </ItemDefinitionGroup>
+  <ItemDefinitionGroup Condition="'$(Configuration)|$(Platform)'=='Debug|Win32'">
+    <ClCompile>
+      <Optimization>Disabled</Optimization>
+      <AdditionalIncludeDirectories>..\..\include;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>
+      <PreprocessorDefinitions>WIN32;_DEBUG;_LIB;%(PreprocessorDefinitions)</PreprocessorDefinitions>
+      <BasicRuntimeChecks>EnableFastChecks</BasicRuntimeChecks>
+      <RuntimeLibrary>MultiThreadedDebug</RuntimeLibrary>
+      <FloatingPointModel>Fast</FloatingPointModel>
+      <PrecompiledHeader />
+      <PrecompiledHeaderOutputFile>$(OutDir)$(TargetName).pch</PrecompiledHeaderOutputFile>
+      <AssemblerListingLocation>$(OutDir)</AssemblerListingLocation>
+      <ObjectFileName>$(OutDir)</ObjectFileName>
+      <ProgramDataBaseFileName>$(OutDir)</ProgramDataBaseFileName>
+      <BrowseInformation>true</BrowseInformation>
+      <WarningLevel>Level3</WarningLevel>
+      <SuppressStartupBanner>true</SuppressStartupBanner>
+      <DebugInformationFormat>ProgramDatabase</DebugInformationFormat>
+      <CompileAs>Default</CompileAs>
+      <EnableEnhancedInstructionSet>StreamingSIMDExtensions2</EnableEnhancedInstructionSet>
+      <XMLDocumentationFileName>$(IntDir)</XMLDocumentationFileName>
+      <BrowseInformationFile>$(IntDir)</BrowseInformationFile>
+      <MultiProcessorCompilation>true</MultiProcessorCompilation>
+    </ClCompile>
+    <ResourceCompile>
+      <PreprocessorDefinitions>_DEBUG;%(PreprocessorDefinitions)</PreprocessorDefinitions>
+      <Culture>0x040b</Culture>
+    </ResourceCompile>
+    <Lib>
+      <OutputFile>$(OutDir)$(TargetName)$(TargetExt)</OutputFile>
+      <SuppressStartupBanner>true</SuppressStartupBanner>
+    </Lib>
+    <PostBuildEvent>
+      <Command>if not exist ..\..\lib mkdir ..\..\lib
+copy $(OutDir)$(TargetName)$(TargetExt) ..\..\lib</Command>
+    </PostBuildEvent>
+  </ItemDefinitionGroup>
+  <ItemDefinitionGroup Condition="'$(Configuration)|$(Platform)'=='Debug|x64'">
+    <Midl>
+      <TargetEnvironment>X64</TargetEnvironment>
+    </Midl>
+    <ClCompile>
+      <Optimization>Disabled</Optimization>
+      <AdditionalIncludeDirectories>..\..\include;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>
+      <PreprocessorDefinitions>WIN32;_DEBUG;_LIB;%(PreprocessorDefinitions)</PreprocessorDefinitions>
+      <BasicRuntimeChecks>EnableFastChecks</BasicRuntimeChecks>
+      <RuntimeLibrary>MultiThreadedDebug</RuntimeLibrary>
+      <FloatingPointModel>Fast</FloatingPointModel>
+      <PrecompiledHeader />
+      <PrecompiledHeaderOutputFile>$(OutDir)$(TargetName).pch</PrecompiledHeaderOutputFile>
+      <AssemblerListingLocation>$(OutDir)</AssemblerListingLocation>
+      <ObjectFileName>$(OutDir)</ObjectFileName>
+      <ProgramDataBaseFileName>$(OutDir)</ProgramDataBaseFileName>
+      <BrowseInformation>true</BrowseInformation>
+      <WarningLevel>Level3</WarningLevel>
+      <SuppressStartupBanner>true</SuppressStartupBanner>
+      <DebugInformationFormat>ProgramDatabase</DebugInformationFormat>
+      <CompileAs>Default</CompileAs>
+      <EnableEnhancedInstructionSet>
+      </EnableEnhancedInstructionSet>
+      <XMLDocumentationFileName>$(IntDir)</XMLDocumentationFileName>
+      <BrowseInformationFile>$(IntDir)</BrowseInformationFile>
+      <MultiProcessorCompilation>true</MultiProcessorCompilation>
+    </ClCompile>
+    <ResourceCompile>
+      <PreprocessorDefinitions>_DEBUG;%(PreprocessorDefinitions)</PreprocessorDefinitions>
+      <Culture>0x040b</Culture>
+    </ResourceCompile>
+    <Lib>
+      <OutputFile>$(OutDir)$(TargetName)$(TargetExt)</OutputFile>
+      <SuppressStartupBanner>true</SuppressStartupBanner>
+    </Lib>
+    <PostBuildEvent>
+      <Command>if not exist ..\..\lib mkdir ..\..\lib
+copy $(OutDir)$(TargetName)$(TargetExt) ..\..\lib</Command>
+    </PostBuildEvent>
+  </ItemDefinitionGroup>
+  <ItemGroup>
+    <ClCompile Include="AAFilter.cpp">
+      <Optimization Condition="'$(Configuration)|$(Platform)'=='Debug|Win32'">Disabled</Optimization>
+      <BasicRuntimeChecks Condition="'$(Configuration)|$(Platform)'=='Debug|Win32'">EnableFastChecks</BasicRuntimeChecks>
+      <BrowseInformation Condition="'$(Configuration)|$(Platform)'=='Debug|Win32'">true</BrowseInformation>
+      <Optimization Condition="'$(Configuration)|$(Platform)'=='Debug|x64'">Disabled</Optimization>
+      <BasicRuntimeChecks Condition="'$(Configuration)|$(Platform)'=='Debug|x64'">EnableFastChecks</BasicRuntimeChecks>
+      <BrowseInformation Condition="'$(Configuration)|$(Platform)'=='Debug|x64'">true</BrowseInformation>
+      <Optimization Condition="'$(Configuration)|$(Platform)'=='Release|Win32'">MaxSpeed</Optimization>
+      <Optimization Condition="'$(Configuration)|$(Platform)'=='Release|x64'">MaxSpeed</Optimization>
+    </ClCompile>
+    <ClCompile Include="BPMDetect.cpp">
+      <DisableSpecificWarnings Condition="'$(Configuration)|$(Platform)'=='Debug|Win32'">4996</DisableSpecificWarnings>
+      <DisableSpecificWarnings Condition="'$(Configuration)|$(Platform)'=='Release|Win32'">4996</DisableSpecificWarnings>
+      <DisableSpecificWarnings Condition="'$(Configuration)|$(Platform)'=='Debug|x64'">4996</DisableSpecificWarnings>
+      <DisableSpecificWarnings Condition="'$(Configuration)|$(Platform)'=='Release|x64'">4996</DisableSpecificWarnings>
+    </ClCompile>
+    <ClCompile Include="cpu_detect_x86.cpp" />
+    <ClCompile Include="FIFOSampleBuffer.cpp">
+      <Optimization Condition="'$(Configuration)|$(Platform)'=='Debug|Win32'">Disabled</Optimization>
+      <BasicRuntimeChecks Condition="'$(Configuration)|$(Platform)'=='Debug|Win32'">EnableFastChecks</BasicRuntimeChecks>
+      <BrowseInformation Condition="'$(Configuration)|$(Platform)'=='Debug|Win32'">true</BrowseInformation>
+      <Optimization Condition="'$(Configuration)|$(Platform)'=='Debug|x64'">Disabled</Optimization>
+      <BasicRuntimeChecks Condition="'$(Configuration)|$(Platform)'=='Debug|x64'">EnableFastChecks</BasicRuntimeChecks>
+      <BrowseInformation Condition="'$(Configuration)|$(Platform)'=='Debug|x64'">true</BrowseInformation>
+      <Optimization Condition="'$(Configuration)|$(Platform)'=='Release|Win32'">MaxSpeed</Optimization>
+      <Optimization Condition="'$(Configuration)|$(Platform)'=='Release|x64'">MaxSpeed</Optimization>
+    </ClCompile>
+    <ClCompile Include="FIRFilter.cpp">
+      <Optimization Condition="'$(Configuration)|$(Platform)'=='Debug|Win32'">Disabled</Optimization>
+      <BasicRuntimeChecks Condition="'$(Configuration)|$(Platform)'=='Debug|Win32'">EnableFastChecks</BasicRuntimeChecks>
+      <BrowseInformation Condition="'$(Configuration)|$(Platform)'=='Debug|Win32'">true</BrowseInformation>
+      <Optimization Condition="'$(Configuration)|$(Platform)'=='Debug|x64'">Disabled</Optimization>
+      <BasicRuntimeChecks Condition="'$(Configuration)|$(Platform)'=='Debug|x64'">EnableFastChecks</BasicRuntimeChecks>
+      <BrowseInformation Condition="'$(Configuration)|$(Platform)'=='Debug|x64'">true</BrowseInformation>
+      <Optimization Condition="'$(Configuration)|$(Platform)'=='Release|Win32'">MaxSpeed</Optimization>
+      <Optimization Condition="'$(Configuration)|$(Platform)'=='Release|x64'">MaxSpeed</Optimization>
+    </ClCompile>
+    <ClCompile Include="InterpolateCubic.cpp" />
+    <ClCompile Include="InterpolateLinear.cpp" />
+    <ClCompile Include="InterpolateShannon.cpp" />
+    <ClCompile Include="mmx_optimized.cpp" />
+    <ClCompile Include="PeakFinder.cpp" />
+    <ClCompile Include="RateTransposer.cpp">
+      <Optimization Condition="'$(Configuration)|$(Platform)'=='Debug|Win32'">Disabled</Optimization>
+      <BasicRuntimeChecks Condition="'$(Configuration)|$(Platform)'=='Debug|Win32'">EnableFastChecks</BasicRuntimeChecks>
+      <BrowseInformation Condition="'$(Configuration)|$(Platform)'=='Debug|Win32'">true</BrowseInformation>
+      <Optimization Condition="'$(Configuration)|$(Platform)'=='Debug|x64'">Disabled</Optimization>
+      <BasicRuntimeChecks Condition="'$(Configuration)|$(Platform)'=='Debug|x64'">EnableFastChecks</BasicRuntimeChecks>
+      <BrowseInformation Condition="'$(Configuration)|$(Platform)'=='Debug|x64'">true</BrowseInformation>
+      <Optimization Condition="'$(Configuration)|$(Platform)'=='Release|Win32'">MaxSpeed</Optimization>
+      <Optimization Condition="'$(Configuration)|$(Platform)'=='Release|x64'">MaxSpeed</Optimization>
+    </ClCompile>
+    <ClCompile Include="SoundTouch.cpp">
+      <Optimization Condition="'$(Configuration)|$(Platform)'=='Debug|Win32'">Disabled</Optimization>
+      <BasicRuntimeChecks Condition="'$(Configuration)|$(Platform)'=='Debug|Win32'">EnableFastChecks</BasicRuntimeChecks>
+      <BrowseInformation Condition="'$(Configuration)|$(Platform)'=='Debug|Win32'">true</BrowseInformation>
+      <Optimization Condition="'$(Configuration)|$(Platform)'=='Debug|x64'">Disabled</Optimization>
+      <BasicRuntimeChecks Condition="'$(Configuration)|$(Platform)'=='Debug|x64'">EnableFastChecks</BasicRuntimeChecks>
+      <BrowseInformation Condition="'$(Configuration)|$(Platform)'=='Debug|x64'">true</BrowseInformation>
+      <Optimization Condition="'$(Configuration)|$(Platform)'=='Release|Win32'">MaxSpeed</Optimization>
+      <Optimization Condition="'$(Configuration)|$(Platform)'=='Release|x64'">MaxSpeed</Optimization>
+    </ClCompile>
+    <ClCompile Include="sse_optimized.cpp" />
+    <ClCompile Include="TDStretch.cpp">
+      <Optimization Condition="'$(Configuration)|$(Platform)'=='Debug|Win32'">Disabled</Optimization>
+      <BasicRuntimeChecks Condition="'$(Configuration)|$(Platform)'=='Debug|Win32'">EnableFastChecks</BasicRuntimeChecks>
+      <BrowseInformation Condition="'$(Configuration)|$(Platform)'=='Debug|Win32'">true</BrowseInformation>
+      <Optimization Condition="'$(Configuration)|$(Platform)'=='Debug|x64'">Disabled</Optimization>
+      <BasicRuntimeChecks Condition="'$(Configuration)|$(Platform)'=='Debug|x64'">EnableFastChecks</BasicRuntimeChecks>
+      <BrowseInformation Condition="'$(Configuration)|$(Platform)'=='Debug|x64'">true</BrowseInformation>
+      <Optimization Condition="'$(Configuration)|$(Platform)'=='Release|Win32'">MaxSpeed</Optimization>
+      <Optimization Condition="'$(Configuration)|$(Platform)'=='Release|x64'">MaxSpeed</Optimization>
+    </ClCompile>
+  </ItemGroup>
+  <ItemGroup>
+    <ClInclude Include="..\..\include\BPMDetect.h" />
+    <ClInclude Include="..\..\include\FIFOSampleBuffer.h" />
+    <ClInclude Include="..\..\include\FIFOSamplePipe.h" />
+    <ClInclude Include="..\..\include\SoundTouch.h" />
+    <ClInclude Include="..\..\include\STTypes.h" />
+    <ClInclude Include="AAFilter.h" />
+    <ClInclude Include="cpu_detect.h" />
+    <ClInclude Include="FIRFilter.h" />
+    <ClInclude Include="InterpolateCubic.h" />
+    <ClInclude Include="InterpolateLinear.h" />
+    <ClInclude Include="InterpolateShannon.h" />
+    <ClInclude Include="PeakFinder.h" />
+    <ClInclude Include="RateTransposer.h" />
+    <ClInclude Include="TDStretch.h" />
+  </ItemGroup>
+  <Import Project="$(VCTargetsPath)\Microsoft.Cpp.targets" />
+  <ImportGroup Label="ExtensionTargets">
+  </ImportGroup>
+</Project>

source/SoundTouch/TDStretch.cpp

@@ -1,11 +1,17 @@
-////////////////////////////////////////////////////////////////////////////////
+///////////////////////////////////////////////////////////////////////////////
 ///
 /// Sampled sound tempo changer/time stretch algorithm. Changes the sound tempo
 /// while maintaining the original pitch by using a time domain WSOLA-like
 /// method with several performance-increasing tweaks.
 ///
-/// Note : MMX optimized functions reside in a separate, platform-specific 
+/// Notes : MMX optimized functions reside in a separate, platform-specific
-/// file, e.g. 'mmx_win.cpp' or 'mmx_gcc.cpp'
+/// file, e.g. 'mmx_win.cpp' or 'mmx_gcc.cpp'.
+///
+/// This source file contains OpenMP optimizations that allow speeding up the
+/// corss-correlation algorithm by executing it in several threads / CPU cores
+/// in parallel. See the following article link for more detailed discussion
+/// about SoundTouch OpenMP optimizations:
+/// http://www.softwarecoven.com/parallel-computing-in-embedded-mobile-devices
 ///
 /// Author        : Copyright (c) Olli Parviainen
 /// Author e-mail : oparviai 'at' iki.fi
@@ -13,13 +19,6 @@
 ///
 ////////////////////////////////////////////////////////////////////////////////
 //
-// Last changed  : $Date$
-// File revision : $Revision: 1.12 $
-//
-// $Id$
-//
-////////////////////////////////////////////////////////////////////////////////
-//
 // License :
 //
 //  SoundTouch audio processing library
@@ -51,32 +50,10 @@
 #include "cpu_detect.h"
 #include "TDStretch.h"
 
-#include <stdio.h>
-
 using namespace soundtouch;
 
 #define max(x, y) (((x) > (y)) ? (x) : (y))
 
-
-/*****************************************************************************
- *
- * Constant definitions
- *
- *****************************************************************************/
-
-// Table for the hierarchical mixing position seeking algorithm
-static const short _scanOffsets[5][24]={
-    { 124,  186,  248,  310,  372,  434,  496,  558,  620,  682,  744, 806,
-      868,  930,  992, 1054, 1116, 1178, 1240, 1302, 1364, 1426, 1488,   0},
-    {-100,  -75,  -50,  -25,   25,   50,   75,  100,    0,    0,    0,   0,
-        0,    0,    0,    0,    0,    0,    0,    0,    0,    0,    0,   0},
-    { -20,  -15,  -10,   -5,    5,   10,   15,   20,    0,    0,    0,   0,
-        0,    0,    0,    0,    0,    0,    0,    0,    0,    0,    0,   0},
-    {  -4,   -3,   -2,   -1,    1,    2,    3,    4,    0,    0,    0,   0,
-        0,    0,    0,    0,    0,    0,    0,    0,    0,    0,    0,   0},
-    { 121,  114,   97,  114,   98,  105,  108,   32,  104,   99,  117,  111,
-      116,  100,  110,  117,  111,  115,    0,    0,    0,    0,    0,   0}};
-
 /*****************************************************************************
  *
  * Implementation of the class 'TDStretch'
@@ -86,18 +63,15 @@ static const short _scanOffsets[5][24]={
 
 TDStretch::TDStretch() : FIFOProcessor(&outputBuffer)
 {
-    bQuickSeek = FALSE;
+    bQuickSeek = false;
     channels = 2;
 
-    pMidBuffer = NULL;
+    pMidBuffer = nullptr;
-    pMidBufferUnaligned = NULL;
+    pMidBufferUnaligned = nullptr;
     overlapLength = 0;
 
-    bAutoSeqSetting = TRUE;
+    bAutoSeqSetting = true;
-    bAutoSeekSetting = TRUE;
+    bAutoSeekSetting = true;
-
-//    outDebt = 0;
-    skipFract = 0;
 
     tempo = 1.0f;
     setParameters(44100, DEFAULT_SEQUENCE_MS, DEFAULT_SEEKWINDOW_MS, DEFAULT_OVERLAP_MS);
@@ -128,29 +102,34 @@ void TDStretch::setParameters(int aSampleRate, int aSequenceMS,
                               int aSeekWindowMS, int aOverlapMS)
 {
     // accept only positive parameter values - if zero or negative, use old values instead
-    if (aSampleRate > 0)   this->sampleRate = aSampleRate;
+    if (aSampleRate > 0)
+    {
+        if (aSampleRate > 192000) ST_THROW_RT_ERROR("Error: Excessive samplerate");
+        this->sampleRate = aSampleRate;
+    }
+
     if (aOverlapMS > 0) this->overlapMs = aOverlapMS;
 
     if (aSequenceMS > 0)
     {
         this->sequenceMs = aSequenceMS;
-        bAutoSeqSetting = FALSE;
+        bAutoSeqSetting = false;
     }
     else if (aSequenceMS == 0)
     {
         // if zero, use automatic setting
-        bAutoSeqSetting = TRUE;
+        bAutoSeqSetting = true;
     }
 
     if (aSeekWindowMS > 0)
     {
         this->seekWindowMs = aSeekWindowMS;
-        bAutoSeekSetting = FALSE;
+        bAutoSeekSetting = false;
     }
     else if (aSeekWindowMS == 0)
     {
         // if zero, use automatic setting
-        bAutoSeekSetting = TRUE;
+        bAutoSeekSetting = true;
     }
 
     calcSeqParameters();
@@ -159,13 +138,12 @@ void TDStretch::setParameters(int aSampleRate, int aSequenceMS,
 
     // set tempo to recalculate 'sampleReq'
     setTempo(tempo);
-
 }
 
 
 
 /// Get routine control parameters, see setParameters() function.
-/// Any of the parameters to this function can be NULL, in such case corresponding parameter
+/// Any of the parameters to this function can be nullptr, in such case corresponding parameter
 /// value isn't returned.
 void TDStretch::getParameters(int *pSampleRate, int *pSequenceMs, int *pSeekWindowMs, int *pOverlapMs) const
 {
@@ -212,7 +190,7 @@ void TDStretch::overlapMono(SAMPLETYPE *pOutput, const SAMPLETYPE *pInput) const
 
 void TDStretch::clearMidBuffer()
 {
-    memset(pMidBuffer, 0, 2 * sizeof(SAMPLETYPE) * overlapLength);
+    memset(pMidBuffer, 0, channels * sizeof(SAMPLETYPE) * overlapLength);
 }
 
 
@@ -220,6 +198,10 @@ void TDStretch::clearInput()
 {
     inputBuffer.clear();
     clearMidBuffer();
+    isBeginning = true;
+    maxnorm = 0;
+    maxnormf = 1e8;
+    skipFract = 0;
 }
 
 
@@ -234,14 +216,14 @@ void TDStretch::clear()
 
 // Enables/disables the quick position seeking algorithm. Zero to disable, nonzero
 // to enable
-void TDStretch::enableQuickSeek(BOOL enable)
+void TDStretch::enableQuickSeek(bool enable)
 {
     bQuickSeek = enable;
 }
 
 
 // Returns nonzero if the quick seeking algorithm is enabled.
-BOOL TDStretch::isQuickSeekEnabled() const
+bool TDStretch::isQuickSeekEnabled() const
 {
     return bQuickSeek;
 }
@@ -265,18 +247,26 @@ int TDStretch::seekBestOverlapPosition(const SAMPLETYPE *refPos)
 // of 'ovlPos'.
 inline void TDStretch::overlap(SAMPLETYPE *pOutput, const SAMPLETYPE *pInput, uint ovlPos) const
 {
-    if (channels == 2) 
+#ifndef USE_MULTICH_ALWAYS
+    if (channels == 1)
     {
-        // stereo sound
-        overlapStereo(pOutput, pInput + 2 * ovlPos);
-    } else {
         // mono sound.
         overlapMono(pOutput, pInput + ovlPos);
     }
+    else if (channels == 2)
+    {
+        // stereo sound
+        overlapStereo(pOutput, pInput + 2 * ovlPos);
+    }
+    else
+#endif // USE_MULTICH_ALWAYS
+    {
+        assert(channels > 0);
+        overlapMulti(pOutput, pInput + channels * ovlPos);
+    }
 }
 
 
-
 // Seeks for the optimal overlap-mixing position. The 'stereo' version of the
 // routine
 //
@@ -286,23 +276,144 @@ inline void TDStretch::overlap(SAMPLETYPE *pOutput, const SAMPLETYPE *pInput, ui
 int TDStretch::seekBestOverlapPositionFull(const SAMPLETYPE *refPos)
 {
     int bestOffs;
-    double bestCorr, corr;
+    double bestCorr;
     int i;
+    double norm;
 
-    bestCorr = FLT_MIN;
+    bestCorr = -FLT_MAX;
     bestOffs = 0;
 
     // Scans for the best correlation value by testing each possible position
     // over the permitted range.
-    for (i = 0; i < seekLength; i ++) 
+    bestCorr = calcCrossCorr(refPos, pMidBuffer, norm);
+    bestCorr = (bestCorr + 0.1) * 0.75;
+
+    #pragma omp parallel for
+    for (i = 1; i < seekLength; i ++)
     {
-        // Calculates correlation value for the mixing position corresponding
+        double corr;
-        // to 'i'
+        // Calculates correlation value for the mixing position corresponding to 'i'
-        corr = calcCrossCorr(refPos + channels * i, pMidBuffer);
+#if defined(_OPENMP) || defined(ST_SIMD_AVOID_UNALIGNED)
+        // in parallel OpenMP mode, can't use norm accumulator version as parallel executor won't
+        // iterate the loop in sequential order
+        // in SIMD mode, avoid accumulator version to allow avoiding unaligned positions
+        corr = calcCrossCorr(refPos + channels * i, pMidBuffer, norm);
+#else
+        // In non-parallel version call "calcCrossCorrAccumulate" that is otherwise same
+        // as "calcCrossCorr", but saves time by reusing & updating previously stored
+        // "norm" value
+        corr = calcCrossCorrAccumulate(refPos + channels * i, pMidBuffer, norm);
+#endif
         // heuristic rule to slightly favour values close to mid of the range
         double tmp = (double)(2 * i - seekLength) / (double)seekLength;
         corr = ((corr + 0.1) * (1.0 - 0.25 * tmp * tmp));
 
+        // Checks for the highest correlation value
+        if (corr > bestCorr)
+        {
+            // For optimal performance, enter critical section only in case that best value found.
+            // in such case repeat 'if' condition as it's possible that parallel execution may have
+            // updated the bestCorr value in the mean time
+            #pragma omp critical
+            if (corr > bestCorr)
+            {
+                bestCorr = corr;
+                bestOffs = i;
+            }
+        }
+    }
+
+#ifdef SOUNDTOUCH_INTEGER_SAMPLES
+    adaptNormalizer();
+#endif
+
+    // clear cross correlation routine state if necessary (is so e.g. in MMX routines).
+    clearCrossCorrState();
+
+    return bestOffs;
+}
+
+
+// Quick seek algorithm for improved runtime-performance: First roughly scans through the
+// correlation area, and then scan surroundings of two best preliminary correlation candidates
+// with improved precision
+//
+// Based on testing:
+// - This algorithm gives on average 99% as good match as the full algorithm
+// - this quick seek algorithm finds the best match on ~90% of cases
+// - on those 10% of cases when this algorithm doesn't find best match,
+//   it still finds on average ~90% match vs. the best possible match
+int TDStretch::seekBestOverlapPositionQuick(const SAMPLETYPE *refPos)
+{
+#define _MIN(a, b)   (((a) < (b)) ? (a) : (b))
+#define SCANSTEP    16
+#define SCANWIND    8
+
+    int bestOffs;
+    int i;
+    int bestOffs2;
+    float bestCorr, corr;
+    float bestCorr2;
+    double norm;
+
+    // note: 'float' types used in this function in case that the platform would need to use software-fp
+
+    bestCorr =
+    bestCorr2 = -FLT_MAX;
+    bestOffs =
+    bestOffs2 = SCANWIND;
+
+    // Scans for the best correlation value by testing each possible position
+    // over the permitted range. Look for two best matches on the first pass to
+    // increase possibility of ideal match.
+    //
+    // Begin from "SCANSTEP" instead of SCANWIND to make the calculation
+    // catch the 'middlepoint' of seekLength vector as that's the a-priori
+    // expected best match position
+    //
+    // Roughly:
+    // - 15% of cases find best result directly on the first round,
+    // - 75% cases find better match on 2nd round around the best match from 1st round
+    // - 10% cases find better match on 2nd round around the 2nd-best-match from 1st round
+    for (i = SCANSTEP; i < seekLength - SCANWIND - 1; i += SCANSTEP)
+    {
+        // Calculates correlation value for the mixing position corresponding
+        // to 'i'
+        corr = (float)calcCrossCorr(refPos + channels*i, pMidBuffer, norm);
+        // heuristic rule to slightly favour values close to mid of the seek range
+        float tmp = (float)(2 * i - seekLength - 1) / (float)seekLength;
+        corr = ((corr + 0.1f) * (1.0f - 0.25f * tmp * tmp));
+
+        // Checks for the highest correlation value
+        if (corr > bestCorr)
+        {
+            // found new best match. keep the previous best as 2nd best match
+            bestCorr2 = bestCorr;
+            bestOffs2 = bestOffs;
+            bestCorr = corr;
+            bestOffs = i;
+        }
+        else if (corr > bestCorr2)
+        {
+            // not new best, but still new 2nd best match
+            bestCorr2 = corr;
+            bestOffs2 = i;
+        }
+    }
+
+    // Scans surroundings of the found best match with small stepping
+    int end = _MIN(bestOffs + SCANWIND + 1, seekLength);
+    for (i = bestOffs - SCANWIND; i < end; i++)
+    {
+        if (i == bestOffs) continue;    // this offset already calculated, thus skip
+
+        // Calculates correlation value for the mixing position corresponding
+        // to 'i'
+        corr = (float)calcCrossCorr(refPos + channels*i, pMidBuffer, norm);
+        // heuristic rule to slightly favour values close to mid of the range
+        float tmp = (float)(2 * i - seekLength - 1) / (float)seekLength;
+        corr = ((corr + 0.1f) * (1.0f - 0.25f * tmp * tmp));
+
         // Checks for the highest correlation value
         if (corr > bestCorr)
         {
@@ -310,70 +421,70 @@ int TDStretch::seekBestOverlapPositionFull(const SAMPLETYPE *refPos)
             bestOffs = i;
         }
     }
-    // clear cross correlation routine state if necessary (is so e.g. in MMX routines).
-    clearCrossCorrState();
 
-    return bestOffs;
+    // Scans surroundings of the 2nd best match with small stepping
-}
+    end = _MIN(bestOffs2 + SCANWIND + 1, seekLength);
-
+    for (i = bestOffs2 - SCANWIND; i < end; i++)
-
-// Seeks for the optimal overlap-mixing position. The 'stereo' version of the
-// routine
-//
-// The best position is determined as the position where the two overlapped
-// sample sequences are 'most alike', in terms of the highest cross-correlation
-// value over the overlapping period
-int TDStretch::seekBestOverlapPositionQuick(const SAMPLETYPE *refPos) 
-{
-    int j;
-    int bestOffs;
-    double bestCorr, corr;
-    int scanCount, corrOffset, tempOffset;
-
-    bestCorr = FLT_MIN;
-    bestOffs = _scanOffsets[0][0];
-    corrOffset = 0;
-    tempOffset = 0;
-
-    // Scans for the best correlation value using four-pass hierarchical search.
-    //
-    // The look-up table 'scans' has hierarchical position adjusting steps.
-    // In first pass the routine searhes for the highest correlation with 
-    // relatively coarse steps, then rescans the neighbourhood of the highest
-    // correlation with better resolution and so on.
-    for (scanCount = 0;scanCount < 4; scanCount ++) 
     {
-        j = 0;
+        if (i == bestOffs2) continue;    // this offset already calculated, thus skip
-        while (_scanOffsets[scanCount][j]) 
-        {
-            tempOffset = corrOffset + _scanOffsets[scanCount][j];
-            if (tempOffset >= seekLength) break;
 
         // Calculates correlation value for the mixing position corresponding
-            // to 'tempOffset'
+        // to 'i'
-            corr = (double)calcCrossCorr(refPos + channels * tempOffset, pMidBuffer);
+        corr = (float)calcCrossCorr(refPos + channels*i, pMidBuffer, norm);
         // heuristic rule to slightly favour values close to mid of the range
-            double tmp = (double)(2 * tempOffset - seekLength) / seekLength;
+        float tmp = (float)(2 * i - seekLength - 1) / (float)seekLength;
-            corr = ((corr + 0.1) * (1.0 - 0.25 * tmp * tmp));
+        corr = ((corr + 0.1f) * (1.0f - 0.25f * tmp * tmp));
 
         // Checks for the highest correlation value
         if (corr > bestCorr)
         {
             bestCorr = corr;
-                bestOffs = tempOffset;
+            bestOffs = i;
         }
-            j ++;
-        }
-        corrOffset = bestOffs;
     }
+
     // clear cross correlation routine state if necessary (is so e.g. in MMX routines).
     clearCrossCorrState();
 
+#ifdef SOUNDTOUCH_INTEGER_SAMPLES
+    adaptNormalizer();
+#endif
+
     return bestOffs;
 }
 
 
 
+
+/// For integer algorithm: adapt normalization factor divider with music so that
+/// it'll not be pessimistically restrictive that can degrade quality on quieter sections
+/// yet won't cause integer overflows either
+void TDStretch::adaptNormalizer()
+{
+    // Do not adapt normalizer over too silent sequences to avoid averaging filter depleting to
+    // too low values during pauses in music
+    if ((maxnorm > 1000) || (maxnormf > 40000000))
+    {
+        //norm averaging filter
+        maxnormf = 0.9f * maxnormf + 0.1f * (float)maxnorm;
+
+        if ((maxnorm > 800000000) && (overlapDividerBitsNorm < 16))
+        {
+            // large values, so increase divider
+            overlapDividerBitsNorm++;
+            if (maxnorm > 1600000000) overlapDividerBitsNorm++; // extra large value => extra increase
+        }
+        else if ((maxnormf < 1000000) && (overlapDividerBitsNorm > 0))
+        {
+            // extra small values, decrease divider
+            overlapDividerBitsNorm--;
+        }
+    }
+
+    maxnorm = 0;
+}
+
+
 /// clear cross correlation routine state if necessary
 void TDStretch::clearCrossCorrState()
 {
@@ -385,18 +496,18 @@ void TDStretch::clearCrossCorrState()
 void TDStretch::calcSeqParameters()
 {
     // Adjust tempo param according to tempo, so that variating processing sequence length is used
-    // at varius tempo settings, between the given low...top limits
+    // at various tempo settings, between the given low...top limits
     #define AUTOSEQ_TEMPO_LOW   0.5     // auto setting low tempo range (-50%)
     #define AUTOSEQ_TEMPO_TOP   2.0     // auto setting top tempo range (+100%)
 
     // sequence-ms setting values at above low & top tempo
-    #define AUTOSEQ_AT_MIN      125.0
+    #define AUTOSEQ_AT_MIN      90.0
-    #define AUTOSEQ_AT_MAX      50.0
+    #define AUTOSEQ_AT_MAX      40.0
     #define AUTOSEQ_K           ((AUTOSEQ_AT_MAX - AUTOSEQ_AT_MIN) / (AUTOSEQ_TEMPO_TOP - AUTOSEQ_TEMPO_LOW))
     #define AUTOSEQ_C           (AUTOSEQ_AT_MIN - (AUTOSEQ_K) * (AUTOSEQ_TEMPO_LOW))
 
-    // seek-window-ms setting values at above low & top tempo
+    // seek-window-ms setting values at above low & top tempoq
-    #define AUTOSEEK_AT_MIN     25.0
+    #define AUTOSEEK_AT_MIN     20.0
     #define AUTOSEEK_AT_MAX     15.0
     #define AUTOSEEK_K          ((AUTOSEEK_AT_MAX - AUTOSEEK_AT_MIN) / (AUTOSEQ_TEMPO_TOP - AUTOSEQ_TEMPO_LOW))
     #define AUTOSEEK_C          (AUTOSEEK_AT_MIN - (AUTOSEEK_K) * (AUTOSEQ_TEMPO_LOW))
@@ -432,7 +543,7 @@ void TDStretch::calcSeqParameters()
 
 // Sets new target tempo. Normal tempo = 'SCALE', smaller values represent slower
 // tempo, larger faster tempo.
-void TDStretch::setTempo(float newTempo)
+void TDStretch::setTempo(double newTempo)
 {
     int intskip;
 
@@ -443,7 +554,7 @@ void TDStretch::setTempo(float newTempo)
 
     // Calculate ideal skip length (according to tempo value)
     nominalSkip = tempo * (seekWindowLength - overlapLength);
-    intskip = (int)(nominalSkip + 0.5f);
+    intskip = (int)(nominalSkip + 0.5);
 
     // Calculate how many samples are needed in the 'inputBuffer' to
     // process another batch of samples
@@ -456,13 +567,16 @@ void TDStretch::setTempo(float newTempo)
 // Sets the number of channels, 1 = mono, 2 = stereo
 void TDStretch::setChannels(int numChannels)
 {
-    assert(numChannels > 0);
+    if (!verifyNumberOfChannels(numChannels) ||
-    if (channels == numChannels) return;
+        (channels == numChannels)) return;
-    assert(numChannels == 1 || numChannels == 2);
 
     channels = numChannels;
     inputBuffer.setChannels(channels);
     outputBuffer.setChannels(channels);
+
+    // re-init overlap/buffer
+    overlapLength=0;
+    setParameters(sampleRate);
 }
 
 
@@ -498,13 +612,13 @@ void TDStretch::processNominalTempo()
 }
 */
 
-#include <stdio.h>
 
 // Processes as many processing frames of the samples 'inputBuffer', store
 // the result into 'outputBuffer'
 void TDStretch::processSamples()
 {
-    int ovlSkip, offset;
+    int ovlSkip;
+    int offset = 0;
     int temp;
 
     /* Removed this small optimization - can introduce a click to sound when tempo setting
@@ -521,8 +635,10 @@ void TDStretch::processSamples()
     // to form a processing frame.
     while ((int)inputBuffer.numSamples() >= sampleReq)
     {
-        // If tempo differs from the normal ('SCALE'), scan for the best overlapping
+        if (isBeginning == false)
-        // position
+        {
+            // apart from the very beginning of the track,
+            // scan for the best overlapping position & do overlap-add
             offset = seekBestOverlapPosition(inputBuffer.ptrBegin());
 
             // Mix the samples in the 'inputBuffer' at position of 'offset' with the
@@ -531,25 +647,50 @@ void TDStretch::processSamples()
             // (that's in 'midBuffer')
             overlap(outputBuffer.ptrEnd((uint)overlapLength), inputBuffer.ptrBegin(), (uint)offset);
             outputBuffer.putSamples((uint)overlapLength);
+            offset += overlapLength;
+        }
+        else
+        {
+            // Adjust processing offset at beginning of track by not perform initial overlapping
+            // and compensating that in the 'input buffer skip' calculation
+            isBeginning = false;
+            int skip = (int)(tempo * overlapLength + 0.5 * seekLength + 0.5);
+
+            #ifdef ST_SIMD_AVOID_UNALIGNED
+            // in SIMD mode, round the skip amount to value corresponding to aligned memory address
+            if (channels == 1)
+            {
+                skip &= -4;
+            }
+            else if (channels == 2)
+            {
+                skip &= -2;
+            }
+            #endif
+            skipFract -= skip;
+            if (skipFract <= -nominalSkip)
+            {
+                skipFract = -nominalSkip;
+            }
+        }
 
         // ... then copy sequence samples from 'inputBuffer' to output:
 
-        // length of sequence
-        temp = (seekWindowLength - 2 * overlapLength);
-
         // crosscheck that we don't have buffer overflow...
-        if ((int)inputBuffer.numSamples() < (offset + temp + overlapLength * 2))
+        if ((int)inputBuffer.numSamples() < (offset + seekWindowLength - overlapLength))
         {
             continue;    // just in case, shouldn't really happen
         }
 
-        outputBuffer.putSamples(inputBuffer.ptrBegin() + channels * (offset + overlapLength), (uint)temp);
+        // length of sequence
+        temp = (seekWindowLength - 2 * overlapLength);
+        outputBuffer.putSamples(inputBuffer.ptrBegin() + channels * offset, (uint)temp);
 
         // Copies the end of the current sequence from 'inputBuffer' to
         // 'midBuffer' for being mixed with the beginning of the next
         // processing sequence and so on
-        assert((offset + temp + overlapLength * 2) <= (int)inputBuffer.numSamples());
+        assert((offset + temp + overlapLength) <= (int)inputBuffer.numSamples());
-        memcpy(pMidBuffer, inputBuffer.ptrBegin() + channels * (offset + temp + overlapLength), 
+        memcpy(pMidBuffer, inputBuffer.ptrBegin() + channels * (offset + temp),
             channels * sizeof(SAMPLETYPE) * overlapLength);
 
         // Remove the processed samples from the input buffer. Update
@@ -588,9 +729,9 @@ void TDStretch::acceptNewOverlapLength(int newOverlapLength)
     {
         delete[] pMidBufferUnaligned;
 
-        pMidBufferUnaligned = new SAMPLETYPE[overlapLength * 2 + 16 / sizeof(SAMPLETYPE)];
+        pMidBufferUnaligned = new SAMPLETYPE[overlapLength * channels + 16 / sizeof(SAMPLETYPE)];
         // ensure that 'pMidBuffer' is aligned to 16 byte boundary for efficiency
-        pMidBuffer = (SAMPLETYPE *)((((ulong)pMidBufferUnaligned) + 15) & (ulong)-16);
+        pMidBuffer = (SAMPLETYPE *)SOUNDTOUCH_ALIGN_POINTER_16(pMidBufferUnaligned);
 
         clearMidBuffer();
     }
@@ -599,7 +740,7 @@ void TDStretch::acceptNewOverlapLength(int newOverlapLength)
 
 // Operator 'new' is overloaded so that it automatically creates a suitable instance
 // depending on if we've a MMX/SSE/etc-capable CPU available or not.
-void * TDStretch::operator new(size_t s)
+void * TDStretch::operator new(size_t)
 {
     // Notice! don't use "new TDStretch" directly, use "newInstance" to create a new instance instead!
     ST_THROW_RT_ERROR("Error in TDStretch::new: Don't use 'new TDStretch' directly, use 'newInstance' member instead!");
@@ -612,6 +753,7 @@ TDStretch * TDStretch::newInstance()
     uint uExtensions;
 
     uExtensions = detectCPUextensions();
+    (void)uExtensions;
 
     // Check if MMX/SSE instruction set extensions supported by CPU
 
@@ -643,7 +785,7 @@ TDStretch * TDStretch::newInstance()
 
 //////////////////////////////////////////////////////////////////////////////
 //
-// Integer arithmetics specific algorithm implementations.
+// Integer arithmetic specific algorithm implementations.
 //
 //////////////////////////////////////////////////////////////////////////////
 
@@ -666,6 +808,25 @@ void TDStretch::overlapStereo(short *poutput, const short *input) const
     }
 }
 
+
+// Overlaps samples in 'midBuffer' with the samples in 'input'. The 'Multi'
+// version of the routine.
+void TDStretch::overlapMulti(short *poutput, const short *input) const
+{
+    short m1;
+    int i = 0;
+
+    for (m1 = 0; m1 < overlapLength; m1 ++)
+    {
+        short m2 = (short)(overlapLength - m1);
+        for (int c = 0; c < channels; c ++)
+        {
+            poutput[i] = (input[i] * m1 + pMidBuffer[i] * m2)  / overlapLength;
+            i++;
+        }
+    }
+}
+
 // Calculates the x having the closest 2^x value for the given value
 static int _getClosest2Power(double value)
 {
@@ -685,13 +846,15 @@ void TDStretch::calculateOverlapLength(int aoverlapMs)
     // calculate overlap length so that it's power of 2 - thus it's easy to do
     // integer division by right-shifting. Term "-1" at end is to account for
     // the extra most significatnt bit left unused in result by signed multiplication
-    overlapDividerBits = _getClosest2Power((sampleRate * aoverlapMs) / 1000.0) - 1;
+    overlapDividerBitsPure = _getClosest2Power((sampleRate * aoverlapMs) / 1000.0) - 1;
-    if (overlapDividerBits > 9) overlapDividerBits = 9;
+    if (overlapDividerBitsPure > 9) overlapDividerBitsPure = 9;
-    if (overlapDividerBits < 3) overlapDividerBits = 3;
+    if (overlapDividerBitsPure < 3) overlapDividerBitsPure = 3;
-    newOvl = (int)pow(2.0, (int)overlapDividerBits + 1);    // +1 => account for -1 above
+    newOvl = (int)pow(2.0, (int)overlapDividerBitsPure + 1);    // +1 => account for -1 above
 
     acceptNewOverlapLength(newOvl);
 
+    overlapDividerBitsNorm = overlapDividerBitsPure;
+
     // calculate sloping divider so that crosscorrelation operation won't
     // overflow 32-bit register. Max. sum of the crosscorrelation sum without
     // divider would be 2^30*(N^3-N)/3, where N = overlap length
@@ -699,39 +862,95 @@ void TDStretch::calculateOverlapLength(int aoverlapMs)
 }
 
 
-double TDStretch::calcCrossCorr(const short *mixingPos, const short *compare) const
+double TDStretch::calcCrossCorr(const short *mixingPos, const short *compare, double &norm)
 {
     long corr;
-    long norm;
+    unsigned long lnorm;
     int i;
 
-    corr = norm = 0;
+    #ifdef ST_SIMD_AVOID_UNALIGNED
-    // Same routine for stereo and mono. For stereo, unroll loop for better
+        // in SIMD mode skip 'mixingPos' positions that aren't aligned to 16-byte boundary
-    // efficiency and gives slightly better resolution against rounding. 
+        if (((ulongptr)mixingPos) & 15) return -1e50;
-    // For mono it same routine, just  unrolls loop by factor of 4
+    #endif
-    for (i = 0; i < channels * overlapLength; i += 4) 
+
+    // hint compiler autovectorization that loop length is divisible by 8
+    int ilength = (channels * overlapLength) & -8;
+
+    corr = lnorm = 0;
+    // Same routine for stereo and mono
+    for (i = 0; i < ilength; i += 2)
     {
         corr += (mixingPos[i] * compare[i] +
-                 mixingPos[i + 1] * compare[i + 1] +
+                 mixingPos[i + 1] * compare[i + 1]) >> overlapDividerBitsNorm;
-                 mixingPos[i + 2] * compare[i + 2] + 
+        lnorm += (mixingPos[i] * mixingPos[i] +
-                 mixingPos[i + 3] * compare[i + 3]) >> overlapDividerBits;
+                  mixingPos[i + 1] * mixingPos[i + 1]) >> overlapDividerBitsNorm;
-        norm += (mixingPos[i] * mixingPos[i] + 
+        // do intermediate scalings to avoid integer overflow
-                 mixingPos[i + 1] * mixingPos[i + 1] +
+    }
-                 mixingPos[i + 2] * mixingPos[i + 2] + 
+
-                 mixingPos[i + 3] * mixingPos[i + 3]) >> overlapDividerBits;
+    if (lnorm > maxnorm)
+    {
+        // modify 'maxnorm' inside critical section to avoid multi-access conflict if in OpenMP mode
+        #pragma omp critical
+        if (lnorm > maxnorm)
+        {
+            maxnorm = lnorm;
+        }
+    }
+    // Normalize result by dividing by sqrt(norm) - this step is easiest
+    // done using floating point operation
+    norm = (double)lnorm;
+    return (double)corr / sqrt((norm < 1e-9) ? 1.0 : norm);
+}
+
+
+/// Update cross-correlation by accumulating "norm" coefficient by previously calculated value
+double TDStretch::calcCrossCorrAccumulate(const short *mixingPos, const short *compare, double &norm)
+{
+    long corr;
+    long lnorm;
+    int i;
+
+    // hint compiler autovectorization that loop length is divisible by 8
+    int ilength = (channels * overlapLength) & -8;
+
+    // cancel first normalizer tap from previous round
+    lnorm = 0;
+    for (i = 1; i <= channels; i ++)
+    {
+        lnorm -= (mixingPos[-i] * mixingPos[-i]) >> overlapDividerBitsNorm;
+    }
+
+    corr = 0;
+    // Same routine for stereo and mono.
+    for (i = 0; i < ilength; i += 2)
+    {
+        corr += (mixingPos[i] * compare[i] +
+                 mixingPos[i + 1] * compare[i + 1]) >> overlapDividerBitsNorm;
+    }
+
+    // update normalizer with last samples of this round
+    for (int j = 0; j < channels; j ++)
+    {
+        i --;
+        lnorm += (mixingPos[i] * mixingPos[i]) >> overlapDividerBitsNorm;
+    }
+
+    norm += (double)lnorm;
+    if (norm > maxnorm)
+    {
+        maxnorm = (unsigned long)norm;
     }
 
     // Normalize result by dividing by sqrt(norm) - this step is easiest
     // done using floating point operation
-    if (norm == 0) norm = 1;    // to avoid div by zero
+    return (double)corr / sqrt((norm < 1e-9) ? 1.0 : norm);
-    return (double)corr / sqrt((double)norm);
 }
 
 #endif // SOUNDTOUCH_INTEGER_SAMPLES
 
 //////////////////////////////////////////////////////////////////////////////
 //
-// Floating point arithmetics specific algorithm implementations.
+// Floating point arithmetic specific algorithm implementations.
 //
 
 #ifdef SOUNDTOUCH_FLOAT_SAMPLES
@@ -760,6 +979,34 @@ void TDStretch::overlapStereo(float *pOutput, const float *pInput) const
 }
 
 
+// Overlaps samples in 'midBuffer' with the samples in 'input'.
+void TDStretch::overlapMulti(float *pOutput, const float *pInput) const
+{
+    int i;
+    float fScale;
+    float f1;
+    float f2;
+
+    fScale = 1.0f / (float)overlapLength;
+
+    f1 = 0;
+    f2 = 1.0f;
+
+    i=0;
+    for (int i2 = 0; i2 < overlapLength; i2 ++)
+    {
+        // note: Could optimize this slightly by taking into account that always channels > 2
+        for (int c = 0; c < channels; c ++)
+        {
+            pOutput[i] = pInput[i] * f1 + pMidBuffer[i] * f2;
+            i++;
+        }
+        f1 += fScale;
+        f2 -= fScale;
+    }
+}
+
+
 /// Calculates overlapInMsec period length in samples.
 void TDStretch::calculateOverlapLength(int overlapInMsec)
 {
@@ -776,33 +1023,66 @@ void TDStretch::calculateOverlapLength(int overlapInMsec)
 }
 
 
-double TDStretch::calcCrossCorr(const float *mixingPos, const float *compare) const
+/// Calculate cross-correlation
+double TDStretch::calcCrossCorr(const float *mixingPos, const float *compare, double &anorm)
 {
-    double corr;
+    float corr;
-    double norm;
+    float norm;
     int i;
 
+    #ifdef ST_SIMD_AVOID_UNALIGNED
+        // in SIMD mode skip 'mixingPos' positions that aren't aligned to 16-byte boundary
+        if (((ulongptr)mixingPos) & 15) return -1e50;
+    #endif
+
+    // hint compiler autovectorization that loop length is divisible by 8
+    int ilength = (channels * overlapLength) & -8;
+
     corr = norm = 0;
-    // Same routine for stereo and mono. For Stereo, unroll by factor of 2.
+    // Same routine for stereo and mono
-    // For mono it's same routine yet unrollsd by factor of 4.
+    for (i = 0; i < ilength; i ++)
-    for (i = 0; i < channels * overlapLength; i += 4) 
     {
-        corr += mixingPos[i] * compare[i] +
+        corr += mixingPos[i] * compare[i];
-                mixingPos[i + 1] * compare[i + 1];
+        norm += mixingPos[i] * mixingPos[i];
-
-        norm += mixingPos[i] * mixingPos[i] + 
-                mixingPos[i + 1] * mixingPos[i + 1];
-
-        // unroll the loop for better CPU efficiency:
-        corr += mixingPos[i + 2] * compare[i + 2] +
-                mixingPos[i + 3] * compare[i + 3];
-
-        norm += mixingPos[i + 2] * mixingPos[i + 2] +
-                mixingPos[i + 3] * mixingPos[i + 3];
     }
 
-    if (norm < 1e-9) norm = 1.0;    // to avoid div by zero
+    anorm = norm;
-    return corr / sqrt(norm);
+    return corr / sqrt((norm < 1e-9 ? 1.0 : norm));
 }
 
+
+/// Update cross-correlation by accumulating "norm" coefficient by previously calculated value
+double TDStretch::calcCrossCorrAccumulate(const float *mixingPos, const float *compare, double &norm)
+{
+    float corr;
+    int i;
+
+    corr = 0;
+
+    // cancel first normalizer tap from previous round
+    for (i = 1; i <= channels; i ++)
+    {
+        norm -= mixingPos[-i] * mixingPos[-i];
+    }
+
+    // hint compiler autovectorization that loop length is divisible by 8
+    int ilength = (channels * overlapLength) & -8;
+
+    // Same routine for stereo and mono
+    for (i = 0; i < ilength; i ++)
+    {
+        corr += mixingPos[i] * compare[i];
+    }
+
+    // update normalizer with last samples of this round
+    for (int j = 0; j < channels; j ++)
+    {
+        i --;
+        norm += mixingPos[i] * mixingPos[i];
+    }
+
+    return corr / sqrt((norm < 1e-9 ? 1.0 : norm));
+}
+
+
 #endif // SOUNDTOUCH_FLOAT_SAMPLES

source/SoundTouch/TDStretch.h

@@ -13,13 +13,6 @@
 ///
 ////////////////////////////////////////////////////////////////////////////////
 //
-// Last changed  : $Date$
-// File revision : $Revision: 4 $
-//
-// $Id$
-//
-////////////////////////////////////////////////////////////////////////////////
-//
 // License :
 //
 //  SoundTouch audio processing library
@@ -112,46 +105,57 @@ class TDStretch : public FIFOProcessor
 protected:
     int channels;
     int sampleReq;
-    float tempo;
 
-    SAMPLETYPE *pMidBuffer;
-    SAMPLETYPE *pMidBufferUnaligned;
     int overlapLength;
     int seekLength;
     int seekWindowLength;
-    int overlapDividerBits;
+    int overlapDividerBitsNorm;
+    int overlapDividerBitsPure;
     int slopingDivider;
-    float nominalSkip;
-    float skipFract;
-    FIFOSampleBuffer outputBuffer;
-    FIFOSampleBuffer inputBuffer;
-    BOOL bQuickSeek;
-
     int sampleRate;
     int sequenceMs;
     int seekWindowMs;
     int overlapMs;
-    BOOL bAutoSeqSetting;
+
-    BOOL bAutoSeekSetting;
+    unsigned long maxnorm;
+    float maxnormf;
+
+    double tempo;
+    double nominalSkip;
+    double skipFract;
+
+    bool bQuickSeek;
+    bool bAutoSeqSetting;
+    bool bAutoSeekSetting;
+    bool isBeginning;
+
+    SAMPLETYPE *pMidBuffer;
+    SAMPLETYPE *pMidBufferUnaligned;
+
+    FIFOSampleBuffer outputBuffer;
+    FIFOSampleBuffer inputBuffer;
 
     void acceptNewOverlapLength(int newOverlapLength);
 
     virtual void clearCrossCorrState();
     void calculateOverlapLength(int overlapMs);
 
-    virtual double calcCrossCorr(const SAMPLETYPE *mixingPos, const SAMPLETYPE *compare) const;
+    virtual double calcCrossCorr(const SAMPLETYPE *mixingPos, const SAMPLETYPE *compare, double &norm);
+    virtual double calcCrossCorrAccumulate(const SAMPLETYPE *mixingPos, const SAMPLETYPE *compare, double &norm);
 
     virtual int seekBestOverlapPositionFull(const SAMPLETYPE *refPos);
     virtual int seekBestOverlapPositionQuick(const SAMPLETYPE *refPos);
-    int seekBestOverlapPosition(const SAMPLETYPE *refPos);
+    virtual int seekBestOverlapPosition(const SAMPLETYPE *refPos);
 
     virtual void overlapStereo(SAMPLETYPE *output, const SAMPLETYPE *input) const;
     virtual void overlapMono(SAMPLETYPE *output, const SAMPLETYPE *input) const;
+    virtual void overlapMulti(SAMPLETYPE *output, const SAMPLETYPE *input) const;
 
     void clearMidBuffer();
     void overlap(SAMPLETYPE *output, const SAMPLETYPE *input, uint ovlPos) const;
 
     void calcSeqParameters();
+    void adaptNormalizer();
 
     /// Changes the tempo of the given sound samples.
     /// Returns amount of samples returned in the "output" buffer.
@@ -161,7 +165,7 @@ protected:
 
 public:
     TDStretch();
-    virtual ~TDStretch();
+    virtual ~TDStretch() override;
 
     /// 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.
@@ -180,10 +184,10 @@ public:
 
     /// Sets new target tempo. Normal tempo = 'SCALE', smaller values represent slower
     /// tempo, larger faster tempo.
-    void setTempo(float newTempo);
+    void setTempo(double newTempo);
 
     /// Returns nonzero if there aren't any samples available for outputting.
-    virtual void clear();
+    virtual void clear() override;
 
     /// Clears the input buffer
     void clearInput();
@@ -193,10 +197,10 @@ public:
 
     /// Enables/disables the quick position seeking algorithm. Zero to disable,
     /// nonzero to enable
-    void enableQuickSeek(BOOL enable);
+    void enableQuickSeek(bool enable);
 
     /// Returns nonzero if the quick seeking algorithm is enabled.
-    BOOL isQuickSeekEnabled() const;
+    bool isQuickSeekEnabled() const;
 
     /// Sets routine control parameters. These control are certain time constants
     /// defining how the sound is stretched to the desired duration.
@@ -213,7 +217,7 @@ public:
                        );
 
     /// Get routine control parameters, see setParameters() function.
-    /// Any of the parameters to this function can be NULL, in such case corresponding parameter
+    /// Any of the parameters to this function can be nullptr, in such case corresponding parameter
     /// value isn't returned.
     void getParameters(int *pSampleRate, int *pSequenceMs, int *pSeekWindowMs, int *pOverlapMs) const;
 
@@ -223,7 +227,7 @@ public:
             const SAMPLETYPE *samples,  ///< Input sample data
             uint numSamples                         ///< Number of samples in 'samples' so that one sample
                                                     ///< contains both channels if stereo
-            );
+            ) override;
 
     /// return nominal input sample requirement for triggering a processing batch
     int getInputSampleReq() const
@@ -236,8 +240,13 @@ public:
     {
         return seekWindowLength - overlapLength;
     }
-};
 
+	/// return approximate initial input-output latency
+	int getLatency() const
+	{
+		return sampleReq;
+	}
+};
 
 
 // Implementation-specific class declarations:
@@ -247,9 +256,10 @@ public:
     class TDStretchMMX : public TDStretch
     {
     protected:
-        double calcCrossCorr(const short *mixingPos, const short *compare) const;
+        double calcCrossCorr(const short *mixingPos, const short *compare, double &norm) override;
-        virtual void overlapStereo(short *output, const short *input) const;
+        double calcCrossCorrAccumulate(const short *mixingPos, const short *compare, double &norm) override;
-        virtual void clearCrossCorrState();
+        virtual void overlapStereo(short *output, const short *input) const override;
+        virtual void clearCrossCorrState() override;
     };
 #endif /// SOUNDTOUCH_ALLOW_MMX
 
@@ -259,7 +269,8 @@ public:
     class TDStretchSSE : public TDStretch
     {
     protected:
-        double calcCrossCorr(const float *mixingPos, const float *compare) const;
+        double calcCrossCorr(const float *mixingPos, const float *compare, double &norm) override;
+        double calcCrossCorrAccumulate(const float *mixingPos, const float *compare, double &norm) override;
     };
 
 #endif /// SOUNDTOUCH_ALLOW_SSE

source/SoundTouch/cpu_detect.h

@@ -12,13 +12,6 @@
 ///
 ////////////////////////////////////////////////////////////////////////////////
 //
-// Last changed  : $Date$
-// File revision : $Revision: 4 $
-//
-// $Id$
-//
-////////////////////////////////////////////////////////////////////////////////
-//
 // License :
 //
 //  SoundTouch audio processing library

source/SoundTouch/cpu_detect_x86.cpp

@@ -11,13 +11,6 @@
 ///
 ////////////////////////////////////////////////////////////////////////////////
 //
-// Last changed  : $Date$
-// File revision : $Revision: 4 $
-//
-// $Id$
-//
-////////////////////////////////////////////////////////////////////////////////
-//
 // License :
 //
 //  SoundTouch audio processing library
@@ -42,21 +35,20 @@
 #include "cpu_detect.h"
 #include "STTypes.h"
 
+
 #if defined(SOUNDTOUCH_ALLOW_X86_OPTIMIZATIONS)
 
    #if defined(__GNUC__) && defined(__i386__)
        // gcc
        #include "cpuid.h"
+   #elif defined(_M_IX86)
+       // windows non-gcc
+       #include <intrin.h>
    #endif
 
-    #if defined(_M_IX86)
-        // windows
-        #include <intrin.h>
    #define bit_MMX     (1 << 23)
    #define bit_SSE     (1 << 25)
    #define bit_SSE2    (1 << 26)
-    #endif
-
 #endif
 
 
@@ -76,7 +68,6 @@ void disableExtensions(uint dwDisableMask)
 }
 
 
-
 /// Checks which instruction set extensions are supported by the CPU.
 uint detectCPUextensions(void)
 {

source/SoundTouch/mmx_optimized.cpp

@@ -20,13 +20,6 @@
 ///
 ////////////////////////////////////////////////////////////////////////////////
 //
-// Last changed  : $Date$
-// File revision : $Revision: 4 $
-//
-// $Id$
-//
-////////////////////////////////////////////////////////////////////////////////
-//
 // License :
 //
 //  SoundTouch audio processing library
@@ -68,7 +61,7 @@ using namespace soundtouch;
 
 
 // Calculates cross correlation of two buffers
-double TDStretchMMX::calcCrossCorr(const short *pV1, const short *pV2) const
+double TDStretchMMX::calcCrossCorr(const short *pV1, const short *pV2, double &dnorm)
 {
     const __m64 *pVec1, *pVec2;
     __m64 shifter;
@@ -79,7 +72,7 @@ double TDStretchMMX::calcCrossCorr(const short *pV1, const short *pV2) const
     pVec1 = (__m64*)pV1;
     pVec2 = (__m64*)pV2;
 
-    shifter = _m_from_int(overlapDividerBits);
+    shifter = _m_from_int(overlapDividerBitsNorm);
     normaccu = accu = _mm_setzero_si64();
 
     // Process 4 parallel sets of 2 * stereo samples or 4 * mono samples
@@ -93,19 +86,19 @@ double TDStretchMMX::calcCrossCorr(const short *pV1, const short *pV2) const
         // _mm_add_pi32 : 2*32bit add
         // _m_psrad     : 32bit right-shift
 
-        temp = _mm_add_pi32(_mm_madd_pi16(pVec1[0], pVec2[0]),
+        temp = _mm_add_pi32(_mm_sra_pi32(_mm_madd_pi16(pVec1[0], pVec2[0]), shifter),
-                            _mm_madd_pi16(pVec1[1], pVec2[1]));
+                            _mm_sra_pi32(_mm_madd_pi16(pVec1[1], pVec2[1]), shifter));
-        temp2 = _mm_add_pi32(_mm_madd_pi16(pVec1[0], pVec1[0]),
+        temp2 = _mm_add_pi32(_mm_sra_pi32(_mm_madd_pi16(pVec1[0], pVec1[0]), shifter),
-                             _mm_madd_pi16(pVec1[1], pVec1[1]));
+                            _mm_sra_pi32(_mm_madd_pi16(pVec1[1], pVec1[1]), shifter));
-        accu = _mm_add_pi32(accu, _mm_sra_pi32(temp, shifter));
+        accu = _mm_add_pi32(accu, temp);
-        normaccu = _mm_add_pi32(normaccu, _mm_sra_pi32(temp2, shifter));
+        normaccu = _mm_add_pi32(normaccu, temp2);
 
-        temp = _mm_add_pi32(_mm_madd_pi16(pVec1[2], pVec2[2]),
+        temp = _mm_add_pi32(_mm_sra_pi32(_mm_madd_pi16(pVec1[2], pVec2[2]), shifter),
-                            _mm_madd_pi16(pVec1[3], pVec2[3]));
+                            _mm_sra_pi32(_mm_madd_pi16(pVec1[3], pVec2[3]), shifter));
-        temp2 = _mm_add_pi32(_mm_madd_pi16(pVec1[2], pVec1[2]),
+        temp2 = _mm_add_pi32(_mm_sra_pi32(_mm_madd_pi16(pVec1[2], pVec1[2]), shifter),
-                             _mm_madd_pi16(pVec1[3], pVec1[3]));
+                            _mm_sra_pi32(_mm_madd_pi16(pVec1[3], pVec1[3]), shifter));
-        accu = _mm_add_pi32(accu, _mm_sra_pi32(temp, shifter));
+        accu = _mm_add_pi32(accu, temp);
-        normaccu = _mm_add_pi32(normaccu, _mm_sra_pi32(temp2, shifter));
+        normaccu = _mm_add_pi32(normaccu, temp2);
 
         pVec1 += 4;
         pVec2 += 4;
@@ -123,16 +116,98 @@ double TDStretchMMX::calcCrossCorr(const short *pV1, const short *pV2) const
     // Clear MMS state
     _m_empty();
 
+    if (norm > (long)maxnorm)
+    {
+        // modify 'maxnorm' inside critical section to avoid multi-access conflict if in OpenMP mode
+        #pragma omp critical
+        if (norm > (long)maxnorm)
+        {
+            maxnorm = norm;
+        }
+    }
+
     // Normalize result by dividing by sqrt(norm) - this step is easiest
     // done using floating point operation
-    if (norm == 0) norm = 1;    // to avoid div by zero
+    dnorm = (double)norm;
 
-    return (double)corr / sqrt((double)norm);
+    return (double)corr / sqrt(dnorm < 1e-9 ? 1.0 : dnorm);
     // Note: Warning about the missing EMMS instruction is harmless
     // as it'll be called elsewhere.
 }
 
 
+/// Update cross-correlation by accumulating "norm" coefficient by previously calculated value
+double TDStretchMMX::calcCrossCorrAccumulate(const short *pV1, const short *pV2, double &dnorm)
+{
+    const __m64 *pVec1, *pVec2;
+    __m64 shifter;
+    __m64 accu;
+    long corr, lnorm;
+    int i;
+
+    // cancel first normalizer tap from previous round
+    lnorm = 0;
+    for (i = 1; i <= channels; i ++)
+    {
+        lnorm -= (pV1[-i] * pV1[-i]) >> overlapDividerBitsNorm;
+    }
+
+    pVec1 = (__m64*)pV1;
+    pVec2 = (__m64*)pV2;
+
+    shifter = _m_from_int(overlapDividerBitsNorm);
+    accu = _mm_setzero_si64();
+
+    // Process 4 parallel sets of 2 * stereo samples or 4 * mono samples
+    // during each round for improved CPU-level parallellization.
+    for (i = 0; i < channels * overlapLength / 16; i ++)
+    {
+        __m64 temp;
+
+        // dictionary of instructions:
+        // _m_pmaddwd   : 4*16bit multiply-add, resulting two 32bits = [a0*b0+a1*b1 ; a2*b2+a3*b3]
+        // _mm_add_pi32 : 2*32bit add
+        // _m_psrad     : 32bit right-shift
+
+        temp = _mm_add_pi32(_mm_sra_pi32(_mm_madd_pi16(pVec1[0], pVec2[0]), shifter),
+                            _mm_sra_pi32(_mm_madd_pi16(pVec1[1], pVec2[1]), shifter));
+        accu = _mm_add_pi32(accu, temp);
+
+        temp = _mm_add_pi32(_mm_sra_pi32(_mm_madd_pi16(pVec1[2], pVec2[2]), shifter),
+                            _mm_sra_pi32(_mm_madd_pi16(pVec1[3], pVec2[3]), shifter));
+        accu = _mm_add_pi32(accu, temp);
+
+        pVec1 += 4;
+        pVec2 += 4;
+    }
+
+    // copy hi-dword of mm0 to lo-dword of mm1, then sum mmo+mm1
+    // and finally store the result into the variable "corr"
+
+    accu = _mm_add_pi32(accu, _mm_srli_si64(accu, 32));
+    corr = _m_to_int(accu);
+
+    // Clear MMS state
+    _m_empty();
+
+    // update normalizer with last samples of this round
+    pV1 = (short *)pVec1;
+    for (int j = 1; j <= channels; j ++)
+    {
+        lnorm += (pV1[-j] * pV1[-j]) >> overlapDividerBitsNorm;
+    }
+    dnorm += (double)lnorm;
+
+    if (lnorm > (long)maxnorm)
+    {
+        maxnorm = lnorm;
+    }
+
+    // Normalize result by dividing by sqrt(norm) - this step is easiest
+    // done using floating point operation
+    return (double)corr / sqrt((dnorm < 1e-9) ? 1.0 : dnorm);
+}
+
 
 void TDStretchMMX::clearCrossCorrState()
 {
@@ -142,7 +217,6 @@ void TDStretchMMX::clearCrossCorrState()
 }
 
 
-
 // MMX-optimized version of the function overlapStereo
 void TDStretchMMX::overlapStereo(short *output, const short *input) const
 {
@@ -166,7 +240,7 @@ void TDStretchMMX::overlapStereo(short *output, const short *input) const
 
     // Overlaplength-division by shifter. "+1" is to account for "-1" deduced in
     // overlapDividerBits calculation earlier.
-    shifter = _m_from_int(overlapDividerBits + 1);
+    shifter = _m_from_int(overlapDividerBitsPure + 1);
 
     for (i = 0; i < overlapLength / 4; i ++)
     {
@@ -220,7 +294,8 @@ void TDStretchMMX::overlapStereo(short *output, const short *input) const
 
 FIRFilterMMX::FIRFilterMMX() : FIRFilter()
 {
-    filterCoeffsUnalign = NULL;
+    filterCoeffsAlign = nullptr;
+    filterCoeffsUnalign = nullptr;
 }
 
 
@@ -239,7 +314,7 @@ 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 *)(((ulong)filterCoeffsUnalign + 15) & -16);
+    filterCoeffsAlign = (short *)SOUNDTOUCH_ALIGN_POINTER_16(filterCoeffsUnalign);
 
     // rearrange the filter coefficients for mmx routines
     for (i = 0;i < length; i += 4)
@@ -257,7 +332,6 @@ void FIRFilterMMX::setCoefficients(const short *coeffs, uint newLength, uint uRe
 }
 
 
-
 // mmx-optimized version of the filter routine for stereo sound
 uint FIRFilterMMX::evaluateFilterStereo(short *dest, const short *src, uint numSamples) const
 {
@@ -314,4 +388,9 @@ uint FIRFilterMMX::evaluateFilterStereo(short *dest, const short *src, uint numS
     return (numSamples & 0xfffffffe) - length;
 }
 
+#else
+
+// workaround to not complain about empty module
+bool _dontcomplain_mmx_empty;
+
 #endif  // SOUNDTOUCH_ALLOW_MMX

source/SoundTouch/sse_optimized.cpp

@@ -23,13 +23,6 @@
 ///
 ////////////////////////////////////////////////////////////////////////////////
 //
-// Last changed  : $Date$
-// File revision : $Revision: 4 $
-//
-// $Id$
-//
-////////////////////////////////////////////////////////////////////////////////
-//
 // License :
 //
 //  SoundTouch audio processing library
@@ -71,7 +64,7 @@ using namespace soundtouch;
 #include <math.h>
 
 // Calculates cross correlation of two buffers
-double TDStretchSSE::calcCrossCorr(const float *pV1, const float *pV2) const
+double TDStretchSSE::calcCrossCorr(const float *pV1, const float *pV2, double &anorm)
 {
     int i;
     const float *pVec1;
@@ -87,13 +80,13 @@ double TDStretchSSE::calcCrossCorr(const float *pV1, const float *pV2) const
     // Compile-time define SOUNDTOUCH_ALLOW_NONEXACT_SIMD_OPTIMIZATION is provided
     // for choosing if this little cheating is allowed.
 
-#ifdef SOUNDTOUCH_ALLOW_NONEXACT_SIMD_OPTIMIZATION
+#ifdef ST_SIMD_AVOID_UNALIGNED
     // Little cheating allowed, return valid correlation only for
     // aligned locations, meaning every second round for stereo sound.
 
     #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
@@ -141,11 +134,11 @@ double TDStretchSSE::calcCrossCorr(const float *pV1, const float *pV2) const
 
     // return value = vSum[0] + vSum[1] + vSum[2] + vSum[3]
     float *pvNorm = (float*)&vNorm;
-    double norm = sqrt(pvNorm[0] + pvNorm[1] + pvNorm[2] + pvNorm[3]);
+    float norm = (pvNorm[0] + pvNorm[1] + pvNorm[2] + pvNorm[3]);
-    if (norm < 1e-9) norm = 1.0;    // to avoid div by zero
+    anorm = norm;
 
     float *pvSum = (float*)&vSum;
-    return (double)(pvSum[0] + pvSum[1] + pvSum[2] + pvSum[3]) / norm;
+    return (double)(pvSum[0] + pvSum[1] + pvSum[2] + pvSum[3]) / sqrt(norm < 1e-9 ? 1.0 : norm);
 
     /* This is approximately corresponding routine in C-language yet without normalization:
     double corr, norm;
@@ -182,6 +175,16 @@ double TDStretchSSE::calcCrossCorr(const float *pV1, const float *pV2) const
 }
 
 
+
+double TDStretchSSE::calcCrossCorrAccumulate(const float *pV1, const float *pV2, double &norm)
+{
+    // call usual calcCrossCorr function because SSE does not show big benefit of
+    // accumulating "norm" value, and also the "norm" rolling algorithm would get
+    // complicated due to SSE-specific alignment-vs-nonexact correlation rules.
+    return calcCrossCorr(pV1, pV2, norm);
+}
+
+
 //////////////////////////////////////////////////////////////////////////////
 //
 // implementation of SSE optimized functions of class 'FIRFilter'
@@ -192,25 +195,22 @@ double TDStretchSSE::calcCrossCorr(const float *pV1, const float *pV2) const
 
 FIRFilterSSE::FIRFilterSSE() : FIRFilter()
 {
-    filterCoeffsAlign = NULL;
+    filterCoeffsAlign = nullptr;
-    filterCoeffsUnalign = NULL;
+    filterCoeffsUnalign = nullptr;
 }
 
 
 FIRFilterSSE::~FIRFilterSSE()
 {
     delete[] filterCoeffsUnalign;
-    filterCoeffsAlign = NULL;
+    filterCoeffsAlign = nullptr;
-    filterCoeffsUnalign = NULL;
+    filterCoeffsUnalign = nullptr;
 }
 
 
 // (overloaded) Calculates filter coefficients for SSE routine
 void FIRFilterSSE::setCoefficients(const float *coeffs, uint newLength, uint uResultDivFactor)
 {
-    uint i;
-    float fDivider;
-
     FIRFilter::setCoefficients(coeffs, newLength, uResultDivFactor);
 
     // Scale the filter coefficients so that it won't be necessary to scale the filtering result
@@ -218,15 +218,15 @@ void FIRFilterSSE::setCoefficients(const float *coeffs, uint newLength, uint uRe
     // Ensure that filter coeffs array is aligned to 16-byte boundary
     delete[] filterCoeffsUnalign;
     filterCoeffsUnalign = new float[2 * newLength + 4];
-    filterCoeffsAlign = (float *)(((unsigned long)filterCoeffsUnalign + 15) & (ulong)-16);
+    filterCoeffsAlign = (float *)SOUNDTOUCH_ALIGN_POINTER_16(filterCoeffsUnalign);
 
-    fDivider = (float)resultDivider;
+    const float scale = ::pow(0.5, (int)resultDivFactor);
 
-    // rearrange the filter coefficients for mmx routines 
+    // rearrange the filter coefficients for sse routines
-    for (i = 0; i < newLength; i ++)
+    for (auto i = 0U; i < newLength; i ++)
     {
         filterCoeffsAlign[2 * i + 0] =
-        filterCoeffsAlign[2 * i + 1] = coeffs[i + 0] / fDivider;
+        filterCoeffsAlign[2 * i + 1] = coeffs[i] * scale;
     }
 }
 
@@ -242,21 +242,24 @@ uint FIRFilterSSE::evaluateFilterStereo(float *dest, const float *source, uint n
 
     if (count < 2) return 0;
 
-    assert(source != NULL);
+    assert(source != nullptr);
-    assert(dest != NULL);
+    assert(dest != nullptr);
     assert((length % 8) == 0);
-    assert(filterCoeffsAlign != NULL);
+    assert(filterCoeffsAlign != nullptr);
-    assert(((ulong)filterCoeffsAlign) % 16 == 0);
+    assert(((ulongptr)filterCoeffsAlign) % 16 == 0);
 
     // filter is evaluated for two stereo samples with each iteration, thus use of 'j += 2'
+    #pragma omp parallel for
     for (j = 0; j < count; j += 2)
     {
         const float *pSrc;
+        float *pDest;
         const __m128 *pFil;
         __m128 sum1, sum2;
         uint i;
 
-        pSrc = (const float*)source;              // source audio data
+        pSrc = (const float*)source + j * 2;      // source audio data
+        pDest = dest + j * 2;                     // destination audio data
         pFil = (const __m128*)filterCoeffsAlign;  // filter coefficients. NOTE: Assumes coefficients
                                                   // are aligned to 16-byte boundary
         sum1 = sum2 = _mm_setzero_ps();
@@ -289,12 +292,10 @@ uint FIRFilterSSE::evaluateFilterStereo(float *dest, const float *source, uint n
         // to sum the two hi- and lo-floats of these registers together.
 
         // post-shuffle & add the filtered values and store to dest.
-        _mm_storeu_ps(dest, _mm_add_ps(
+        _mm_storeu_ps(pDest, _mm_add_ps(
                     _mm_shuffle_ps(sum1, sum2, _MM_SHUFFLE(1,0,3,2)),   // s2_1 s2_0 s1_3 s1_2
                     _mm_shuffle_ps(sum1, sum2, _MM_SHUFFLE(3,2,1,0))    // s2_3 s2_2 s1_1 s1_0
                     ));
-        source += 4;
-        dest += 4;
     }
 
     // Ideas for further improvement:

source/SoundTouchDLL/DllTest/DllTest.cpp

@@ -0,0 +1,115 @@
+////////////////////////////////////////////////////////////////////////////////
+///
+/// DllTest.cpp : This is small app main routine used for testing sound processing
+/// with SoundTouch.dll API
+///
+/// Author        : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+
+#include <string>
+#include <iostream>
+#include <fstream>
+#include "../SoundTouchDLL.h"
+#include "../../SoundStretch/WavFile.h"
+
+using namespace std;
+using namespace soundstretch;
+
+// DllTest main
+int wmain(int argc, const wchar_t *argv[])
+{
+    // Check program arguments
+    if (argc < 4)
+    {
+        cout << "Too few arguments. Usage: DllTest [infile.wav] [outfile.wav] [sampletype]" << endl;
+        return -1;
+    }
+
+    wstring inFileName = argv[1];
+    wstring outFileName = argv[2];
+    wstring str_sampleType = argv[3];
+
+    bool floatSample;
+    if (str_sampleType == L"float")
+    {
+        floatSample = true;
+    }
+    else if (str_sampleType == L"short")
+    {
+        floatSample = false;
+    }
+    else
+    { 
+        cerr << "Missing or invalid sampletype. Expected either short or float" << endl;
+        return -1;
+    }
+
+    try
+    {
+        // Open input & output WAV files
+        WavInFile inFile(inFileName);
+        int numChannels = inFile.getNumChannels();
+        int sampleRate = inFile.getSampleRate();
+        WavOutFile outFile(outFileName, sampleRate, inFile.getNumBits(), numChannels);
+
+        // Create SoundTouch DLL instance
+        HANDLE st = soundtouch_createInstance();
+        soundtouch_setChannels(st, numChannels);
+        soundtouch_setSampleRate(st, sampleRate);
+        soundtouch_setPitchSemiTones(st, 2);
+
+        cout << "processing with soundtouch.dll routines";
+
+        if (floatSample)
+        {
+            // Process file with SoundTouch.DLL float sample (default) API
+            float fbuffer[2048];
+            int nmax = 2048 / numChannels;
+
+            cout << " using float api ..." << endl;
+            while (inFile.eof() == false)
+            {
+                int n = inFile.read(fbuffer, nmax * numChannels) / numChannels;
+                soundtouch_putSamples(st, fbuffer, n);
+                do
+                {
+                    n = soundtouch_receiveSamples(st, fbuffer, nmax);
+                    outFile.write(fbuffer, n * numChannels);
+                } while (n > 0);
+            }
+        }
+        else
+        {
+            // Process file with SoundTouch.DLL int16 (short) sample API.
+            // Notice that SoundTouch.dll does internally processing using floating
+            // point routines so the int16 API is not any faster, but provided for 
+            // convenience.
+            short i16buffer[2048];
+            int nmax = 2048 / numChannels;
+
+            cout << " using i16 api ..." << endl;
+            while (inFile.eof() == false)
+            {
+                int n = inFile.read(i16buffer, nmax * numChannels) / numChannels;
+                soundtouch_putSamples_i16(st, i16buffer, n);
+                do
+                {
+                    n = soundtouch_receiveSamples_i16(st, i16buffer, nmax);
+                    outFile.write(i16buffer, n * numChannels);
+                } while (n > 0);
+            }
+        }
+
+        soundtouch_destroyInstance(st);
+        cout << "done." << endl;
+    }
+    catch (const runtime_error &e)
+    {
+        cerr << e.what() << endl;
+    }
+
+    return 0;
+}

source/SoundTouchDLL/DllTest/DllTest.vcxproj

@@ -0,0 +1,180 @@
+<?xml version="1.0" encoding="utf-8"?>
+<Project DefaultTargets="Build" ToolsVersion="14.0" xmlns="http://schemas.microsoft.com/developer/msbuild/2003">
+  <ItemGroup Label="ProjectConfigurations">
+    <ProjectConfiguration Include="Debug|Win32">
+      <Configuration>Debug</Configuration>
+      <Platform>Win32</Platform>
+    </ProjectConfiguration>
+    <ProjectConfiguration Include="Release|Win32">
+      <Configuration>Release</Configuration>
+      <Platform>Win32</Platform>
+    </ProjectConfiguration>
+    <ProjectConfiguration Include="Debug|x64">
+      <Configuration>Debug</Configuration>
+      <Platform>x64</Platform>
+    </ProjectConfiguration>
+    <ProjectConfiguration Include="Release|x64">
+      <Configuration>Release</Configuration>
+      <Platform>x64</Platform>
+    </ProjectConfiguration>
+  </ItemGroup>
+  <PropertyGroup Label="Globals">
+    <ProjectGuid>{E3C0726F-28F4-4F0B-8183-B87CA60C063C}</ProjectGuid>
+    <Keyword>Win32Proj</Keyword>
+    <RootNamespace>DllTest</RootNamespace>
+    <WindowsTargetPlatformVersion>10.0</WindowsTargetPlatformVersion>
+  </PropertyGroup>
+  <Import Project="$(VCTargetsPath)\Microsoft.Cpp.Default.props" />
+  <PropertyGroup Condition="'$(Configuration)|$(Platform)'=='Debug|Win32'" Label="Configuration">
+    <ConfigurationType>Application</ConfigurationType>
+    <UseDebugLibraries>true</UseDebugLibraries>
+    <PlatformToolset>v142</PlatformToolset>
+    <CharacterSet>Unicode</CharacterSet>
+  </PropertyGroup>
+  <PropertyGroup Condition="'$(Configuration)|$(Platform)'=='Release|Win32'" Label="Configuration">
+    <ConfigurationType>Application</ConfigurationType>
+    <UseDebugLibraries>false</UseDebugLibraries>
+    <PlatformToolset>v142</PlatformToolset>
+    <WholeProgramOptimization>true</WholeProgramOptimization>
+    <CharacterSet>Unicode</CharacterSet>
+  </PropertyGroup>
+  <PropertyGroup Condition="'$(Configuration)|$(Platform)'=='Debug|x64'" Label="Configuration">
+    <ConfigurationType>Application</ConfigurationType>
+    <UseDebugLibraries>true</UseDebugLibraries>
+    <PlatformToolset>v142</PlatformToolset>
+    <CharacterSet>Unicode</CharacterSet>
+  </PropertyGroup>
+  <PropertyGroup Condition="'$(Configuration)|$(Platform)'=='Release|x64'" Label="Configuration">
+    <ConfigurationType>Application</ConfigurationType>
+    <UseDebugLibraries>false</UseDebugLibraries>
+    <PlatformToolset>v142</PlatformToolset>
+    <WholeProgramOptimization>true</WholeProgramOptimization>
+    <CharacterSet>Unicode</CharacterSet>
+  </PropertyGroup>
+  <Import Project="$(VCTargetsPath)\Microsoft.Cpp.props" />
+  <ImportGroup Label="ExtensionSettings">
+  </ImportGroup>
+  <ImportGroup Label="Shared">
+  </ImportGroup>
+  <ImportGroup Label="PropertySheets" Condition="'$(Configuration)|$(Platform)'=='Debug|Win32'">
+    <Import Project="$(UserRootDir)\Microsoft.Cpp.$(Platform).user.props" Condition="exists('$(UserRootDir)\Microsoft.Cpp.$(Platform).user.props')" Label="LocalAppDataPlatform" />
+  </ImportGroup>
+  <ImportGroup Label="PropertySheets" Condition="'$(Configuration)|$(Platform)'=='Release|Win32'">
+    <Import Project="$(UserRootDir)\Microsoft.Cpp.$(Platform).user.props" Condition="exists('$(UserRootDir)\Microsoft.Cpp.$(Platform).user.props')" Label="LocalAppDataPlatform" />
+  </ImportGroup>
+  <ImportGroup Label="PropertySheets" Condition="'$(Configuration)|$(Platform)'=='Debug|x64'">
+    <Import Project="$(UserRootDir)\Microsoft.Cpp.$(Platform).user.props" Condition="exists('$(UserRootDir)\Microsoft.Cpp.$(Platform).user.props')" Label="LocalAppDataPlatform" />
+  </ImportGroup>
+  <ImportGroup Label="PropertySheets" Condition="'$(Configuration)|$(Platform)'=='Release|x64'">
+    <Import Project="$(UserRootDir)\Microsoft.Cpp.$(Platform).user.props" Condition="exists('$(UserRootDir)\Microsoft.Cpp.$(Platform).user.props')" Label="LocalAppDataPlatform" />
+  </ImportGroup>
+  <PropertyGroup Label="UserMacros" />
+  <PropertyGroup Condition="'$(Configuration)|$(Platform)'=='Debug|Win32'">
+    <LinkIncremental>true</LinkIncremental>
+    <OutDir>$(Platform)\$(Configuration)\</OutDir>
+    <IntDir>$(Platform)\$(Configuration)\</IntDir>
+    <TargetName>$(ProjectName)D</TargetName>
+  </PropertyGroup>
+  <PropertyGroup Condition="'$(Configuration)|$(Platform)'=='Debug|x64'">
+    <LinkIncremental>true</LinkIncremental>
+    <OutDir>$(Platform)\$(Configuration)\</OutDir>
+    <TargetName>$(ProjectName)D_x64</TargetName>
+  </PropertyGroup>
+  <PropertyGroup Condition="'$(Configuration)|$(Platform)'=='Release|Win32'">
+    <LinkIncremental>false</LinkIncremental>
+    <OutDir>$(Platform)\$(Configuration)\</OutDir>
+    <IntDir>$(Platform)\$(Configuration)\</IntDir>
+  </PropertyGroup>
+  <PropertyGroup Condition="'$(Configuration)|$(Platform)'=='Release|x64'">
+    <LinkIncremental>false</LinkIncremental>
+    <OutDir>$(Platform)\$(Configuration)\</OutDir>
+    <TargetName>$(ProjectName)_x64</TargetName>
+  </PropertyGroup>
+  <ItemDefinitionGroup Condition="'$(Configuration)|$(Platform)'=='Debug|Win32'">
+    <ClCompile>
+      <PrecompiledHeader>
+      </PrecompiledHeader>
+      <WarningLevel>Level3</WarningLevel>
+      <Optimization>Disabled</Optimization>
+      <PreprocessorDefinitions>WIN32;_DEBUG;_CONSOLE;%(PreprocessorDefinitions)</PreprocessorDefinitions>
+      <AdditionalIncludeDirectories>../../../include</AdditionalIncludeDirectories>
+      <DisableSpecificWarnings>4996</DisableSpecificWarnings>
+    </ClCompile>
+    <Link>
+      <SubSystem>Console</SubSystem>
+      <GenerateDebugInformation>true</GenerateDebugInformation>
+      <AdditionalDependencies>..\..\..\lib\SoundTouchDllD.lib</AdditionalDependencies>
+      <OutputFile>$(OutDir)$(TargetName)$(TargetExt)</OutputFile>
+    </Link>
+  </ItemDefinitionGroup>
+  <ItemDefinitionGroup Condition="'$(Configuration)|$(Platform)'=='Debug|x64'">
+    <ClCompile>
+      <PrecompiledHeader>
+      </PrecompiledHeader>
+      <WarningLevel>Level3</WarningLevel>
+      <Optimization>Disabled</Optimization>
+      <PreprocessorDefinitions>_DEBUG;_CONSOLE;%(PreprocessorDefinitions)</PreprocessorDefinitions>
+      <AdditionalIncludeDirectories>../../../include</AdditionalIncludeDirectories>
+      <DisableSpecificWarnings>4996</DisableSpecificWarnings>
+    </ClCompile>
+    <Link>
+      <SubSystem>Console</SubSystem>
+      <GenerateDebugInformation>true</GenerateDebugInformation>
+      <AdditionalDependencies>..\..\..\lib\SoundTouchDllD_x64.lib</AdditionalDependencies>
+    </Link>
+  </ItemDefinitionGroup>
+  <ItemDefinitionGroup Condition="'$(Configuration)|$(Platform)'=='Release|Win32'">
+    <ClCompile>
+      <WarningLevel>Level3</WarningLevel>
+      <PrecompiledHeader>
+      </PrecompiledHeader>
+      <Optimization>MaxSpeed</Optimization>
+      <FunctionLevelLinking>true</FunctionLevelLinking>
+      <IntrinsicFunctions>true</IntrinsicFunctions>
+      <PreprocessorDefinitions>WIN32;NDEBUG;_CONSOLE;%(PreprocessorDefinitions)</PreprocessorDefinitions>
+      <AdditionalIncludeDirectories>../../../include</AdditionalIncludeDirectories>
+      <DisableSpecificWarnings>4996</DisableSpecificWarnings>
+    </ClCompile>
+    <Link>
+      <SubSystem>Console</SubSystem>
+      <EnableCOMDATFolding>true</EnableCOMDATFolding>
+      <OptimizeReferences>true</OptimizeReferences>
+      <GenerateDebugInformation>true</GenerateDebugInformation>
+      <AdditionalDependencies>..\..\..\lib\SoundTouchDll.lib</AdditionalDependencies>
+    </Link>
+  </ItemDefinitionGroup>
+  <ItemDefinitionGroup Condition="'$(Configuration)|$(Platform)'=='Release|x64'">
+    <ClCompile>
+      <WarningLevel>Level3</WarningLevel>
+      <PrecompiledHeader>
+      </PrecompiledHeader>
+      <Optimization>MaxSpeed</Optimization>
+      <FunctionLevelLinking>true</FunctionLevelLinking>
+      <IntrinsicFunctions>true</IntrinsicFunctions>
+      <PreprocessorDefinitions>NDEBUG;_CONSOLE;%(PreprocessorDefinitions)</PreprocessorDefinitions>
+      <AdditionalIncludeDirectories>../../../include</AdditionalIncludeDirectories>
+      <DisableSpecificWarnings>4996</DisableSpecificWarnings>
+    </ClCompile>
+    <Link>
+      <SubSystem>Console</SubSystem>
+      <EnableCOMDATFolding>true</EnableCOMDATFolding>
+      <OptimizeReferences>true</OptimizeReferences>
+      <GenerateDebugInformation>true</GenerateDebugInformation>
+      <AdditionalDependencies>..\..\..\lib\SoundTouchDll_x64.lib</AdditionalDependencies>
+    </Link>
+  </ItemDefinitionGroup>
+  <ItemGroup>
+    <Text Include="ReadMe.txt" />
+  </ItemGroup>
+  <ItemGroup>
+    <ClInclude Include="stdafx.h" />
+    <ClInclude Include="targetver.h" />
+  </ItemGroup>
+  <ItemGroup>
+    <ClCompile Include="..\..\SoundStretch\WavFile.cpp" />
+    <ClCompile Include="DllTest.cpp" />
+  </ItemGroup>
+  <Import Project="$(VCTargetsPath)\Microsoft.Cpp.targets" />
+  <ImportGroup Label="ExtensionTargets">
+  </ImportGroup>
+</Project>

source/SoundTouchDLL/LazarusTest/README.txt

@@ -0,0 +1,10 @@
+This is Lazarus Pascal example that loads the SoundTouch dynamic-load library
+and queries the library version as a simple example how to load SoundTouch from
+Pascal / Lazarus.
+
+Set the SoundTouch dynamic library file name in the 'InitDLL' procedure of
+file 'SoundTouchDLL.pas' depending on if you're building for Windows or Linux.
+
+The example expects the the 'libSoundTouchDll.so' (linux) or 'SoundTouch.dll' (Windows)
+library binary files is found within this project directory, either via soft-link
+(in Linux) or as a copied file.

source/SoundTouchDLL/LazarusTest/SoundTouchDLL.pas

@@ -2,13 +2,8 @@ unit SoundTouchDLL;
 
 //////////////////////////////////////////////////////////////////////////////
 //
-// SoundTouch.dll wrapper for accessing SoundTouch routines from Delphi/Pascal
+// SoundTouch.dll / libSoundTouchDll.so wrapper for accessing SoundTouch
-//
+// routines from Delphi/Pascal/Lazarus
-//  Module Author : Christian Budde
-//  
-////////////////////////////////////////////////////////////////////////////////
-//
-// $Id$
 //
 ////////////////////////////////////////////////////////////////////////////////
 //
@@ -35,58 +30,60 @@ unit SoundTouchDLL;
 
 interface
 
-uses
+//uses
-  Windows;
+  //Windows;
 
 type
   TSoundTouchHandle = THandle;
 
   // Create a new instance of SoundTouch processor.
-  TSoundTouchCreateInstance = function : TSoundTouchHandle; stdcall;
+  TSoundTouchCreateInstance = function : TSoundTouchHandle; cdecl;
 
   // Destroys a SoundTouch processor instance.
-  TSoundTouchDestroyInstance = procedure (Handle: TSoundTouchHandle); stdcall;
+  TSoundTouchDestroyInstance = procedure (Handle: TSoundTouchHandle); cdecl;
 
   // Get SoundTouch library version string
-  TSoundTouchGetVersionString = function : PChar; stdcall;
+  TSoundTouchGetVersionString = function : PAnsiChar; cdecl;
+
+  // Get SoundTouch library version string 2
+  TSoundTouchGetVersionString2 = procedure(VersionString : PAnsiChar; BufferSize : Integer); cdecl;
 
   // Get SoundTouch library version Id
-  TSoundTouchGetVersionId = function : Cardinal; stdcall;
+  TSoundTouchGetVersionId = function : Cardinal; cdecl;
 
   // Sets new rate control value. Normal rate = 1.0, smaller values
   // represent slower rate, larger faster rates.
-  TSoundTouchSetRate = procedure (Handle: TSoundTouchHandle; newRate: Single); stdcall;
+  TSoundTouchSetRate = procedure (Handle: TSoundTouchHandle; NewRate: Single); cdecl;
 
   // Sets new tempo control value. Normal tempo = 1.0, smaller values
   // represent slower tempo, larger faster tempo.
-  TSoundTouchSetTempo = procedure (Handle: TSoundTouchHandle; newTempo: Single); stdcall;
+  TSoundTouchSetTempo = procedure (Handle: TSoundTouchHandle; NewTempo: Single); cdecl;
 
   // Sets new rate control value as a difference in percents compared
   // to the original rate (-50 .. +100 %);
-  TSoundTouchSetRateChange = procedure (Handle: TSoundTouchHandle; newRate: Single); stdcall;
+  TSoundTouchSetRateChange = procedure (Handle: TSoundTouchHandle; NewRate: Single); cdecl;
 
   // Sets new tempo control value as a difference in percents compared
   // to the original tempo (-50 .. +100 %);
-  TSoundTouchSetTempoChange = procedure (Handle: TSoundTouchHandle; newTempo: Single); stdcall;
+  TSoundTouchSetTempoChange = procedure (Handle: TSoundTouchHandle; NewTempo: Single); cdecl;
 
   // 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;
+  TSoundTouchSetPitch = procedure (Handle: TSoundTouchHandle; NewPitch: Single); cdecl;
 
   // Sets pitch change in octaves compared to the original pitch
   // (-1.00 .. +1.00);
-  TSoundTouchSetPitchOctaves = procedure (Handle: TSoundTouchHandle; newPitch: Single); stdcall;
+  TSoundTouchSetPitchOctaves = procedure (Handle: TSoundTouchHandle; NewPitch: Single); cdecl;
 
   // Sets pitch change in semi-tones compared to the original pitch
   // (-12 .. +12);
-  TSoundTouchSetPitchSemiTones = procedure (Handle: TSoundTouchHandle; newPitch: Single); stdcall;
+  TSoundTouchSetPitchSemiTones = procedure (Handle: TSoundTouchHandle; NewPitch: Single); cdecl;
-
 
   // Sets the number of channels, 1 = mono, 2 = stereo
-  TSoundTouchSetChannels = procedure (Handle: TSoundTouchHandle; numChannels: Cardinal); stdcall;
+  TSoundTouchSetChannels = procedure (Handle: TSoundTouchHandle; NumChannels: Cardinal); cdecl;
 
   // Sets sample rate.
-  TSoundTouchSetSampleRate = procedure (Handle: TSoundTouchHandle; SampleRate: Cardinal); stdcall;
+  TSoundTouchSetSampleRate = procedure (Handle: TSoundTouchHandle; SampleRate: Cardinal); cdecl;
 
   // Flushes the last samples from the processing pipeline to the output.
   // Clears also the internal processing buffers.
@@ -95,7 +92,7 @@ type
   // 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;
+  TSoundTouchFlush = procedure (Handle: TSoundTouchHandle); cdecl;
 
   // 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
@@ -105,53 +102,64 @@ type
                                      NumSamples: Cardinal    //< Number of samples in buffer. Notice
                                                              //< that in case of stereo-sound a single sample
                                                              //< contains data for both channels.
-                                    ); stdcall;
+                                    ); cdecl;
+
+  TSoundTouchPutSamplesI16 = procedure (Handle: TSoundTouchHandle;
+                                     const Samples: Pint16; //< 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.
+                                    ); cdecl;
 
   // Clears all the samples in the object's output and internal processing
   // buffers.
-  TSoundTouchClear = procedure (Handle: TSoundTouchHandle); stdcall;
+  TSoundTouchClear = procedure (Handle: TSoundTouchHandle); cdecl;
 
   // Changes a setting controlling the processing system behaviour. See the
   // 'SETTING_...' defines for available setting ID's.
   //
-  // \return 'TRUE' if the setting was succesfully changed
+  // \return 'TRUE' if the setting was successfully changed
   TSoundTouchSetSetting = function (Handle: TSoundTouchHandle;
                                  SettingId: Integer;   //< Setting ID number. see SETTING_... defines.
                                  Value: Integer        //< New setting value.
-                                ): Boolean; stdcall;
+                                ): Boolean; cdecl;
 
   // 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.
+                                 SettingId: Integer     //< Setting ID number, see SETTING_... defines.
-                                ): Integer; stdcall;
+                                ): Integer; cdecl;
-
 
   // Returns number of samples currently unprocessed.
-  TSoundTouchNumUnprocessedSamples = function (Handle: TSoundTouchHandle): Cardinal; stdcall;
+  TSoundTouchNumUnprocessedSamples = function (Handle: TSoundTouchHandle): Cardinal; cdecl;
 
-  // Adjusts book-keeping so that given number of samples are removed from beginning of the
+  /// Receive ready samples from the processing pipeline.
-  // sample buffer without copying them anywhere.
+  ///
-  //
+  /// if called with outBuffer=nullptr, just reduces amount of ready samples within the pipeline.
-  // 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.
+                                     OutBuffer: PSingle;           //< Buffer where to copy output samples.
-                                     maxSamples: Integer      //< How many samples to receive at max.
+                                     MaxSamples: Integer      //< How many samples to receive at max.
-                                    ): Cardinal; stdcall;
+                                    ): Cardinal; cdecl;
 
+  /// int16 version of soundtouch_receiveSamples(): This converts internal float samples
+  /// into int16 (short) return data type
+  TSoundTouchReceiveSamplesI16 = function (Handle: TSoundTouchHandle;
+                                     OutBuffer: int16;        //< Buffer where to copy output samples.
+                                     MaxSamples: Integer      //< How many samples to receive at max.
+                                    ): Cardinal; cdecl;
   // Returns number of samples currently available.
-  TSoundTouchNumSamples = function (Handle: TSoundTouchHandle): Cardinal; stdcall;
+  TSoundTouchNumSamples = function (Handle: TSoundTouchHandle): Cardinal; cdecl;
 
   // Returns nonzero if there aren't any samples available for outputting.
-  TSoundTouchIsEmpty = function (Handle: TSoundTouchHandle): Integer; stdcall;
+  TSoundTouchIsEmpty = function (Handle: TSoundTouchHandle): Integer; cdecl;
 
 var
   SoundTouchCreateInstance        : TSoundTouchCreateInstance;
   SoundTouchDestroyInstance       : TSoundTouchDestroyInstance;
   SoundTouchGetVersionString      : TSoundTouchGetVersionString;
+  SoundTouchGetVersionString2     : TSoundTouchGetVersionString2;
   SoundTouchGetVersionId          : TSoundTouchGetVersionId;
   SoundTouchSetRate               : TSoundTouchSetRate;
   SoundTouchSetTempo              : TSoundTouchSetTempo;
@@ -164,11 +172,13 @@ var
   SoundTouchSetSampleRate         : TSoundTouchSetSampleRate;
   SoundTouchFlush                 : TSoundTouchFlush;
   SoundTouchPutSamples            : TSoundTouchPutSamples;
+  SoundTouchPutSamplesI16         : TSoundTouchPutSamplesI16;
   SoundTouchClear                 : TSoundTouchClear;
   SoundTouchSetSetting            : TSoundTouchSetSetting;
   SoundTouchGetSetting            : TSoundTouchGetSetting;
   SoundTouchNumUnprocessedSamples : TSoundTouchNumUnprocessedSamples;
   SoundTouchReceiveSamples        : TSoundTouchReceiveSamples;
+  SoundTouchReceiveSamplesI16     : TSoundTouchReceiveSamplesI16;
   SoundTouchNumSamples            : TSoundTouchNumSamples;
   SoundTouchIsEmpty               : TSoundTouchIsEmpty;
 
@@ -210,10 +220,10 @@ type
     procedure Clear; virtual;
 
     procedure PutSamples(const Samples: PSingle; const NumSamples: Cardinal);
-    function ReceiveSamples(const outBuffer: PSingle; const maxSamples: Integer): 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;
+    function GetSetting(const SettingId: Integer): Integer;
 
     property VersionString: string read GetVersionString;
     property VersionID: Cardinal read GetVersionId;
@@ -231,17 +241,17 @@ type
     property IsEmpty: Integer read GetIsEmpty;
   end;
 
-implementation
+  // list of exported functions and procedures
+function IsSoundTouchLoaded: Boolean;
 
-uses
+implementation
-  SysUtils;
 
 { TSoundTouch }
 
 constructor TSoundTouch.Create;
 begin
   inherited;
-  FHandle := SoundTouchCreateInstance;
+  FHandle := SoundTouchCreateInstance();
   FRate := 1;
   FTempo := 1;
   FPitch := 1;
@@ -299,12 +309,12 @@ end;
 
 class function TSoundTouch.GetVersionId: Cardinal;
 begin
-  result := SoundTouchGetVersionId;
+  result := SoundTouchGetVersionId();
 end;
 
 class function TSoundTouch.GetVersionString: string;
 begin
-  result := StrPas(SoundTouchGetVersionString);
+  result := StrPas(SoundTouchGetVersionString());
 end;
 
 procedure TSoundTouch.SetChannels(const Value: Cardinal);
@@ -347,10 +357,10 @@ begin
   SoundTouchSetRate(FHandle, FRate);
 end;
 
-function TSoundTouch.ReceiveSamples(const outBuffer: PSingle;
+function TSoundTouch.ReceiveSamples(const OutBuffer: PSingle;
-  const maxSamples: Integer): Cardinal;
+  const MaxSamples: Integer): Cardinal;
 begin
-  result := SoundTouchReceiveSamples(FHandle, outBuffer, maxSamples);
+  result := SoundTouchReceiveSamples(FHandle, OutBuffer, MaxSamples);
 end;
 
 procedure TSoundTouch.SetPitchChange(const Value: Single);
@@ -418,36 +428,55 @@ begin
 end;
 
 var
-  SoundTouchLibHandle: HINST;
+  SoundTouchLibHandle: THandle;
-  SoundTouchDLL: PAnsiChar = 'SoundTouch.DLL';
+  SoundTouchDLLFile: AnsiString = 'libSoundTouchDll.so';
+  //SoundTouchDLLFile: AnsiString = 'SoundTouch.dll';
+
+  // bpm detect functions. untested -- if these don't work then remove:
+  bpm_createInstance: function(chan: int32; sampleRate : int32): THandle; cdecl;
+  bpm_destroyInstance: procedure(h: THandle); cdecl;
+  bpm_getBpm: function(h: THandle): Single; cdecl;
+  bpm_putSamples: procedure(h: THandle; const samples: PSingle; numSamples: cardinal); cdecl;
 
 procedure InitDLL;
 begin
-  SoundTouchLibHandle := LoadLibrary(SoundTouchDLL);
+{$ifdef mswindows}  // Windows
+  SoundTouchLibHandle := LoadLibrary('.\SoundTouchDll.dll');
+{$else}  // Unix
+  SoundTouchLibHandle := LoadLibrary('./libSoundTouchDll.so');
+{$endif}
   if SoundTouchLibHandle <> 0 then
   try
-    SoundTouchCreateInstance        := GetProcAddress(SoundTouchLibHandle, PAnsiChar( 2)); //'soundtouch_createInstance');
+    Pointer(SoundTouchCreateInstance)        := GetProcAddress(SoundTouchLibHandle, 'soundtouch_createInstance');
-    SoundTouchDestroyInstance       := GetProcAddress(SoundTouchLibHandle, PAnsiChar( 3)); //'soundtouch_destroyInstance');
+    Pointer(SoundTouchDestroyInstance)       := GetProcAddress(SoundTouchLibHandle, 'soundtouch_destroyInstance');
-    SoundTouchGetVersionString      := GetProcAddress(SoundTouchLibHandle, PAnsiChar( 7)); //'soundtouch_getVersionString');
+    Pointer(SoundTouchGetVersionString)      := GetProcAddress(SoundTouchLibHandle, 'soundtouch_getVersionString');
-    SoundTouchGetVersionId          := GetProcAddress(SoundTouchLibHandle, PAnsiChar( 6)); //'soundtouch_getVersionId');
+    Pointer(SoundTouchGetVersionString2)     := GetProcAddress(SoundTouchLibHandle, 'soundtouch_getVersionString2');
-    SoundTouchSetRate               := GetProcAddress(SoundTouchLibHandle, PAnsiChar(17)); //'soundtouch_setRate');
+    Pointer(SoundTouchGetVersionId)          := GetProcAddress(SoundTouchLibHandle, 'soundtouch_getVersionId');
-    SoundTouchSetTempo              := GetProcAddress(SoundTouchLibHandle, PAnsiChar(21)); //'soundtouch_setTempo');
+    Pointer(SoundTouchSetRate)               := GetProcAddress(SoundTouchLibHandle, 'soundtouch_setRate');
-    SoundTouchSetRateChange         := GetProcAddress(SoundTouchLibHandle, PAnsiChar(18)); //'soundtouch_setRateChange');
+    Pointer(SoundTouchSetTempo)              := GetProcAddress(SoundTouchLibHandle, 'soundtouch_setTempo');
-    SoundTouchSetTempoChange        := GetProcAddress(SoundTouchLibHandle, PAnsiChar(22)); //'soundtouch_setTempoChange');
+    Pointer(SoundTouchSetRateChange)         := GetProcAddress(SoundTouchLibHandle, 'soundtouch_setRateChange');
-    SoundTouchSetPitch              := GetProcAddress(SoundTouchLibHandle, PAnsiChar(14)); //'soundtouch_setPitch');
+    Pointer(SoundTouchSetTempoChange)        := GetProcAddress(SoundTouchLibHandle, 'soundtouch_setTempoChange');
-    SoundTouchSetPitchOctaves       := GetProcAddress(SoundTouchLibHandle, PAnsiChar(15)); //'soundtouch_setPitchOctaves');
+    Pointer(SoundTouchSetPitch)              := GetProcAddress(SoundTouchLibHandle, 'soundtouch_setPitch');
-    SoundTouchSetPitchSemiTones     := GetProcAddress(SoundTouchLibHandle, PAnsiChar(16)); //'soundtouch_setPitchSemiTones');
+    Pointer(SoundTouchSetPitchOctaves)       := GetProcAddress(SoundTouchLibHandle, 'soundtouch_setPitchOctaves');
-    SoundTouchSetChannels           := GetProcAddress(SoundTouchLibHandle, PAnsiChar(13)); //'soundtouch_setChannels');
+    Pointer(SoundTouchSetPitchSemiTones)     := GetProcAddress(SoundTouchLibHandle, 'soundtouch_setPitchSemiTones');
-    SoundTouchSetSampleRate         := GetProcAddress(SoundTouchLibHandle, PAnsiChar(19)); //'soundtouch_setSampleRate');
+    Pointer(SoundTouchSetChannels)           := GetProcAddress(SoundTouchLibHandle, 'soundtouch_setChannels');
-    SoundTouchFlush                 := GetProcAddress(SoundTouchLibHandle, PAnsiChar(4)); //'soundtouch_flush');
+    Pointer(SoundTouchSetSampleRate)         := GetProcAddress(SoundTouchLibHandle, 'soundtouch_setSampleRate');
-    SoundTouchPutSamples            := GetProcAddress(SoundTouchLibHandle, PAnsiChar(11)); //'soundtouch_putSamples');
+    Pointer(SoundTouchFlush)                 := GetProcAddress(SoundTouchLibHandle, 'soundtouch_flush');
-    SoundTouchClear                 := GetProcAddress(SoundTouchLibHandle, PAnsiChar(1)); //'soundtouch_clear');
+    Pointer(SoundTouchPutSamples)            := GetProcAddress(SoundTouchLibHandle, 'soundtouch_putSamples');
-    SoundTouchSetSetting            := GetProcAddress(SoundTouchLibHandle, PAnsiChar(20)); //'soundtouch_SetSetting');
+    Pointer(SoundTouchPutSamplesI16)         := GetProcAddress(SoundTouchLibHandle, 'soundtouch_putSamples_i16');
-    SoundTouchGetSetting            := GetProcAddress(SoundTouchLibHandle, PAnsiChar(5)); //'soundtouch_setSetting');
+    Pointer(SoundTouchClear)                 := GetProcAddress(SoundTouchLibHandle, 'soundtouch_clear');
-    SoundTouchNumUnprocessedSamples := GetProcAddress(SoundTouchLibHandle, PAnsiChar(10)); //'soundtouch_numUnprocessedSamples');
+    Pointer(SoundTouchSetSetting)            := GetProcAddress(SoundTouchLibHandle, 'soundtouch_SetSetting');
-    SoundTouchReceiveSamples        := GetProcAddress(SoundTouchLibHandle, PAnsiChar(12)); //'soundtouch_receiveSamples');
+    Pointer(SoundTouchGetSetting)            := GetProcAddress(SoundTouchLibHandle, 'soundtouch_setSetting');
-    SoundTouchNumSamples            := GetProcAddress(SoundTouchLibHandle, PAnsiChar(9)); //'soundtouch_numSamples');
+    Pointer(SoundTouchNumUnprocessedSamples) := GetProcAddress(SoundTouchLibHandle, 'soundtouch_numUnprocessedSamples');
-    SoundTouchIsEmpty               := GetProcAddress(SoundTouchLibHandle, PAnsiChar(8)); //'soundtouch_isEmpty');
+    Pointer(SoundTouchReceiveSamples)        := GetProcAddress(SoundTouchLibHandle, 'soundtouch_receiveSamples');
+    Pointer(SoundTouchReceiveSamplesI16)     := GetProcAddress(SoundTouchLibHandle, 'soundtouch_receiveSamples_i16');
+    Pointer(SoundTouchNumSamples)            := GetProcAddress(SoundTouchLibHandle, 'soundtouch_numSamples');
+    Pointer(SoundTouchIsEmpty)               := GetProcAddress(SoundTouchLibHandle, 'soundtouch_isEmpty');
+
+    Pointer(bpm_createInstance)             :=GetProcAddress(SoundTouchLibHandle, 'bpm_createInstance');
+    Pointer(bpm_destroyInstance)            :=GetProcAddress(SoundTouchLibHandle, 'bpm_destroyInstance');
+    Pointer(bpm_getBpm)                     :=GetProcAddress(SoundTouchLibHandle, 'bpm_getBpm');
+    Pointer(bpm_putSamples)                 :=GetProcAddress(SoundTouchLibHandle, 'bpm_putSamples');
 
   except
     FreeLibrary(SoundTouchLibHandle);
@@ -460,6 +489,12 @@ begin
   if SoundTouchLibHandle <> 0 then FreeLibrary(SoundTouchLibHandle);
 end;
 
+// returns 'true' if SoundTouch dynamic library has been successfully loaded, otherwise 'false'
+function IsSoundTouchLoaded: Boolean;
+begin;
+  result := SoundTouchLibHandle <> 0
+end;
+
 initialization
   InitDLL;
 

source/SoundTouchDLL/LazarusTest/libSoundTouchDll.so

@@ -0,0 +1 @@
+../.libs/libSoundTouchDll.so

source/SoundTouchDLL/LazarusTest/main.lfm

@@ -0,0 +1,36 @@
+object Form1: TForm1
+  Left = 2237
+  Height = 128
+  Top = 242
+  Width = 381
+  Caption = 'SoundTouch test'
+  ClientHeight = 128
+  ClientWidth = 381
+  LCLVersion = '2.2.0.4'
+  object Load: TButton
+    Left = 19
+    Height = 50
+    Top = 16
+    Width = 144
+    Caption = 'Load SoundTouch'
+    OnClick = LoadClick
+    TabOrder = 0
+  end
+  object EditVersion: TEdit
+    Left = 184
+    Height = 34
+    Top = 80
+    Width = 184
+    TabOrder = 1
+    Text = 'n/a'
+    TextHint = 'click to populate'
+  end
+  object Label1: TLabel
+    Left = 19
+    Height = 17
+    Top = 90
+    Width = 156
+    Caption = 'Soundtouch lib version:'
+    WordWrap = True
+  end
+end

source/SoundTouchDLL/LazarusTest/main.pas

@@ -0,0 +1,49 @@
+unit main;
+
+{$mode objfpc}{$H+}
+
+interface
+
+uses
+  Classes, SysUtils, Forms, Controls, Graphics, Dialogs, StdCtrls, SoundTouchDLL;
+
+
+type
+
+  { TForm1 }
+
+  TForm1 = class(TForm)
+    EditVersion: TEdit;
+    Label1: TLabel;
+    Load: TButton;
+
+    procedure LoadClick(Sender: TObject);
+  private
+
+  public
+
+  end;
+
+var
+  Form1: TForm1;
+
+implementation
+
+{$R *.lfm}
+
+{ TForm1 }
+
+procedure TForm1.LoadClick(Sender: TObject);
+var
+  version:string;
+begin
+  if IsSoundTouchLoaded() then
+    version := SoundTouchGetVersionString()
+  else
+    version := '<library loading failed>';
+
+  EditVersion.Text:= version;
+end;
+
+end.
+

source/SoundTouchDLL/LazarusTest/soundtouchtest.lpi

@@ -0,0 +1,78 @@
+<?xml version="1.0" encoding="UTF-8"?>
+<CONFIG>
+  <ProjectOptions>
+    <Version Value="12"/>
+    <General>
+      <SessionStorage Value="InProjectDir"/>
+      <Title Value="soundtouchtest"/>
+      <Scaled Value="True"/>
+      <ResourceType Value="res"/>
+      <UseXPManifest Value="True"/>
+      <XPManifest>
+        <DpiAware Value="True"/>
+      </XPManifest>
+      <Icon Value="0"/>
+    </General>
+    <BuildModes>
+      <Item Name="Default" Default="True"/>
+    </BuildModes>
+    <PublishOptions>
+      <Version Value="2"/>
+      <UseFileFilters Value="True"/>
+    </PublishOptions>
+    <RunParams>
+      <FormatVersion Value="2"/>
+    </RunParams>
+    <RequiredPackages>
+      <Item>
+        <PackageName Value="LCL"/>
+      </Item>
+    </RequiredPackages>
+    <Units>
+      <Unit>
+        <Filename Value="soundtouchtest.lpr"/>
+        <IsPartOfProject Value="True"/>
+      </Unit>
+      <Unit>
+        <Filename Value="main.pas"/>
+        <IsPartOfProject Value="True"/>
+        <ComponentName Value="Form1"/>
+        <HasResources Value="True"/>
+        <ResourceBaseClass Value="Form"/>
+      </Unit>
+    </Units>
+  </ProjectOptions>
+  <CompilerOptions>
+    <Version Value="11"/>
+    <Target>
+      <Filename Value="soundtouchtest"/>
+    </Target>
+    <SearchPaths>
+      <IncludeFiles Value="$(ProjOutDir)"/>
+      <UnitOutputDirectory Value="lib/$(TargetCPU)-$(TargetOS)"/>
+    </SearchPaths>
+    <Linking>
+      <Debugging>
+        <DebugInfoType Value="dsDwarf3"/>
+      </Debugging>
+      <Options>
+        <Win32>
+          <GraphicApplication Value="True"/>
+        </Win32>
+      </Options>
+    </Linking>
+  </CompilerOptions>
+  <Debugging>
+    <Exceptions>
+      <Item>
+        <Name Value="EAbort"/>
+      </Item>
+      <Item>
+        <Name Value="ECodetoolError"/>
+      </Item>
+      <Item>
+        <Name Value="EFOpenError"/>
+      </Item>
+    </Exceptions>
+  </Debugging>
+</CONFIG>