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".
 
@@ -455,4 +455,4 @@ FAILURE OF THE LIBRARY TO OPERATE WITH ANY OTHER SOFTWARE), EVEN IF
 SUCH HOLDER OR OTHER PARTY HAS BEEN ADVISED OF THE POSSIBILITY OF SUCH
 DAMAGES.
 
-		     END OF TERMS AND CONDITIONS
+		     END OF TERMS AND CONDITIONS

Makefile.am

@@ -1,20 +1,16 @@
 ## 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
 ## terms of the GNU General Public License as published by the Free Software
 ## Foundation; either version 2 of the License, or (at your option) any later
 ## version.
-## 
+##
 ## SoundTouch is distributed in the hope that it will be useful, but WITHOUT ANY
 ## WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
 ## A PARTICULAR PURPOSE.  See the GNU General Public License for more details.
-## 
+##
 ## You should have received a copy of the GNU General Public License along with
 ## this program; if not, write to the Free Software Foundation, Inc., 59 Temple
 ## Place - Suite 330, Boston, MA  02111-1307, USA
@@ -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 ]
@@ -10,15 +10,12 @@ then
 	elif [ -a configure ]
 	then
 		configure && $0 --clean
-	else 
+	else
 		bootstrap && configure && $0 --clean
 	fi
 
 	rm -rf configure libtool aclocal.m4 `find . -name Makefile.in` autom4te*.cache config/config.guess config/config.h.in config/config.sub config/depcomp config/install-sh config/ltmain.sh config/missing config/mkinstalldirs config/stamp-h config/stamp-h.in
 
-	#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
 
-
-## 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
-
+AM_CPPFLAGS=-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

@@ -1,36 +1,41 @@
-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
 dnl SoundTouch is free software; you can redistribute it and/or modify it under the
 dnl terms of the GNU General Public License as published by the Free Software
 dnl Foundation; either version 2 of the License, or (at your option) any later
 dnl version.
-dnl 
+dnl
 dnl SoundTouch is distributed in the hope that it will be useful, but WITHOUT ANY
 dnl WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS
 dnl FOR A PARTICULAR PURPOSE.  See the GNU General Public License for more
 dnl details.
-dnl 
+dnl
 dnl You should have received a copy of the GNU General Public License along with
 dnl this program; if not, write to the Free Software Foundation, Inc., 59 Temple
 dnl Place - Suite 330, Boston, MA  02111-1307, USA
 # Process this file with autoconf to produce a configure script.
 
-AC_INIT(SoundTouch, 1.7.0, [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
-#AC_DISABLE_SHARED	dnl This makes libtool only build static libs 
+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
 
 
 
@@ -53,17 +58,18 @@ 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([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.])
+	AC_MSG_WARN([If using a x86 architecture and optimizations are desired then please install gcc (>= 4.3).])
+	AC_MSG_WARN([If using a non-x86 architecture then this is expected and can be ignored.])
+fi
 
-if test "x$ac_cv_header_cpuid_h" = "xno"; then
-	AC_MSG_WARN([The cpuid.h file was not found therefore the x86 optimizations will be disabled.])
-	AC_MSG_WARN([If using a x86 architecture and optimizations are desired then please install gcc (>= 4.3).])
-	AC_MSG_WARN([If using a non-x86 architecture then this is expected and can be ignored.])
-fi
- 
 
 dnl ############################################################################
 dnl # Checks for typedefs, structures, and compiler characteristics            $
@@ -75,26 +81,34 @@ AC_C_INLINE
 
 
 AC_ARG_ENABLE(integer-samples,
-              [AC_HELP_STRING([--enable-integer-samples],
-                              [use integer samples instead of floats
-[default=yes]])],,
+              [AS_HELP_STRING([--enable-integer-samples],[use integer samples instead of floats [default=no]])],,
               [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
-[default=yes]])],[enable_x86_optimizations="${enableval}"],
+                              [use MMX or SSE optimization [default=yes]])],[enable_x86_optimizations="${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$enable_integer_samples" = "xyes"; then
         echo "****** Integer sample type enabled ******"
         AC_DEFINE(SOUNDTOUCH_INTEGER_SAMPLES,1,[Use Integer as Sample type])
@@ -102,16 +116,23 @@ else
         echo "****** Float sample type enabled ******"
         AC_DEFINE(SOUNDTOUCH_FLOAT_SAMPLES,1,[Use Float as Sample type])
 fi
+AM_CONDITIONAL([SOUNDTOUCH_FLOAT_SAMPLES], [test "x$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.
-if test "x$enable_x86_optimizations" = "xyes" -a "x$ac_cv_header_cpuid_h" = "xyes"; then
+if test "x$enable_x86_optimizations" = "xyes" -a "x$ac_cv_header_cpuid_h" = "xyes"; then
         echo "****** x86 optimizations enabled ******"
 
 	original_saved_CXXFLAGS=$CXXFLAGS
 	have_mmx_intrinsics=no
-	OPT_CXXFLAGS="-mmmx -Winline"
-	CXXFLAGS="$OPT_CXXFLAGS $CXXFLAGS"
+	CXXFLAGS="-mmmx -Winline $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=$CXXFLAGS
 	have_sse_intrinsics=no
-	OPT_CXXFLAGS="-msse -Winline"
-	CXXFLAGS="$OPT_CXXFLAGS $CXXFLAGS"
+	CXXFLAGS="-msse -Winline $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], [$AM_CXXFLAGS])
+
+# Empty default CXXFLAGS so user can set them if desirable
+#AC_SUBST([CXXFLAGS], [ ])
+
+
 # SSTypes.h by default enables optimizations. Those already got disabled if
 # the user requested for it or if the system does not support them.
 #
@@ -195,11 +270,9 @@ AM_CONDITIONAL([HAVE_SSE], [test "x$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])) 
+AC_CHECK_LIB(m, sqrt,,AC_MSG_ERROR([compatible libc math library not found]))
 
 dnl add whatever functions you might want to check for here
 #AC_CHECK_FUNCS([floor ftruncate memmove memset mkdir modf pow realpath sqrt strchr strdup strerror strrchr strstr strtol])
@@ -229,11 +302,12 @@ AC_CONFIG_FILES([
 	source/Makefile
 		source/SoundTouch/Makefile
 		source/SoundStretch/Makefile
+        source/SoundTouchDLL/Makefile
 	include/Makefile
 ])
 
-AC_OUTPUT(
-	soundtouch.pc
-)
+AC_CONFIG_FILES([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

@@ -1,164 +1,205 @@
-////////////////////////////////////////////////////////////////////////////////
-///
-/// Beats-per-minute (BPM) detection routine.
-///
-/// The beat detection algorithm works as follows:
-/// - Use function 'inputSamples' to input a chunks of samples to the class for
-///   analysis. It's a good idea to enter a large sound file or stream in smallish
-///   chunks of around few kilosamples in order not to extinguish too much RAM memory.
-/// - Input sound data is decimated to approx 500 Hz to reduce calculation burden,
-///   which is basically ok as low (bass) frequencies mostly determine the beat rate.
-///   Simple averaging is used for anti-alias filtering because the resulting signal
-///   quality isn't of that high importance.
-/// - Decimated sound data is enveloped, i.e. the amplitude shape is detected by
-///   taking absolute value that's smoothed by sliding average. Signal levels that
-///   are below a couple of times the general RMS amplitude level are cut away to
-///   leave only notable peaks there.
-/// - Repeating sound patterns (e.g. beats) are detected by calculating short-term 
-///   autocorrelation function of the enveloped signal.
-/// - After whole sound data file has been analyzed as above, the bpm level is 
-///   detected by function 'getBpm' that finds the highest peak of the autocorrelation 
-///   function, calculates it's precise location and converts this reading to bpm's.
-///
-/// Author        : Copyright (c) Olli Parviainen
-/// Author e-mail : oparviai 'at' iki.fi
-/// SoundTouch WWW: http://www.surina.net/soundtouch
-///
-////////////////////////////////////////////////////////////////////////////////
-//
-// Last changed  : $Date$
-// File revision : $Revision: 4 $
-//
-// $Id$
-//
-////////////////////////////////////////////////////////////////////////////////
-//
-// License :
-//
-//  SoundTouch audio processing library
-//  Copyright (c) Olli Parviainen
-//
-//  This library is free software; you can redistribute it and/or
-//  modify it under the terms of the GNU Lesser General Public
-//  License as published by the Free Software Foundation; either
-//  version 2.1 of the License, or (at your option) any later version.
-//
-//  This library is distributed in the hope that it will be useful,
-//  but WITHOUT ANY WARRANTY; without even the implied warranty of
-//  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-//  Lesser General Public License for more details.
-//
-//  You should have received a copy of the GNU Lesser General Public
-//  License along with this library; if not, write to the Free Software
-//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
-//
-////////////////////////////////////////////////////////////////////////////////
-
-#ifndef _BPMDetect_H_
-#define _BPMDetect_H_
-
-#include "STTypes.h"
-#include "FIFOSampleBuffer.h"
-
-namespace soundtouch
-{
-
-/// Minimum allowed BPM rate. Used to restrict accepted result above a reasonable limit.
-#define MIN_BPM 29
-
-/// Maximum allowed BPM rate. Used to restrict accepted result below a reasonable limit.
-#define MAX_BPM 200
-
-
-/// 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;
-
-    /// Sample average accumulator for FIFO-like decimation.
-    soundtouch::LONG_SAMPLETYPE decimateSum;
-
-    /// Decimate sound by this coefficient to reach approx. 500 Hz.
-    int decimateBy;
-
-    /// Auto-correlation window length
-    int windowLen;
-
-    /// Number of channels (1 = mono, 2 = stereo)
-    int channels;
-
-    /// sample rate
-    int sampleRate;
-
-    /// Beginning of auto-correlation window: Autocorrelation isn't being updated for
-    /// the first these many correlation bins.
-    int windowStart;
- 
-    /// FIFO-buffer for decimated processing samples.
-    soundtouch::FIFOSampleBuffer *buffer;
-
-    /// Updates auto-correlation function for given number of decimated samples that 
-    /// are read from the internal 'buffer' pipe (samples aren't removed from the pipe 
-    /// though).
-    void updateXCorr(int process_samples      /// How many samples are processed.
-                     );
-
-    /// Decimates samples to approx. 500 Hz.
-    ///
-    /// \return Number of output samples.
-    int decimate(soundtouch::SAMPLETYPE *dest,      ///< Destination buffer
-                 const soundtouch::SAMPLETYPE *src, ///< Source sample buffer
-                 int numsamples                     ///< Number of source samples.
-                 );
-
-    /// Calculates amplitude envelope for the buffer of samples.
-    /// Result is output to 'samples'.
-    void calcEnvelope(soundtouch::SAMPLETYPE *samples,  ///< Pointer to input/output data buffer
-                      int numsamples                    ///< Number of samples in buffer
-                      );
-
-    /// remove constant bias from xcorr data
-    void removeBias();
-
-public:
-    /// Constructor.
-    BPMDetect(int numChannels,  ///< Number of channels in sample data.
-              int sampleRate    ///< Sample rate in Hz.
-              );
-
-    /// Destructor.
-    virtual ~BPMDetect();
-
-    /// Inputs a block of samples for analyzing: Envelopes the samples and then
-    /// updates the autocorrelation estimation. When whole song data has been input
-    /// in smaller blocks using this function, read the resulting bpm with 'getBpm' 
-    /// function. 
-    /// 
-    /// Notice that data in 'samples' array can be disrupted in processing.
-    void inputSamples(const soundtouch::SAMPLETYPE *samples,    ///< Pointer to input/working data buffer
-                      int numSamples                            ///< Number of samples in buffer
-                      );
-
-
-    /// Analyzes the results and returns the BPM rate. Use this function to read result
-    /// after whole song data has been input to the class by consecutive calls of
-    /// 'inputSamples' function.
-    ///
-    /// \return Beats-per-minute rate, or zero if detection failed.
-    float getBpm();
-};
-
-}
-
-#endif // _BPMDetect_H_
+////////////////////////////////////////////////////////////////////////////////
+///
+/// Beats-per-minute (BPM) detection routine.
+///
+/// The beat detection algorithm works as follows:
+/// - Use function 'inputSamples' to input a chunks of samples to the class for
+///   analysis. It's a good idea to enter a large sound file or stream in smallish
+///   chunks of around few kilosamples in order not to extinguish too much RAM memory.
+/// - Input sound data is decimated to approx 500 Hz to reduce calculation burden,
+///   which is basically ok as low (bass) frequencies mostly determine the beat rate.
+///   Simple averaging is used for anti-alias filtering because the resulting signal
+///   quality isn't of that high importance.
+/// - Decimated sound data is enveloped, i.e. the amplitude shape is detected by
+///   taking absolute value that's smoothed by sliding average. Signal levels that
+///   are below a couple of times the general RMS amplitude level are cut away to
+///   leave only notable peaks there.
+/// - Repeating sound patterns (e.g. beats) are detected by calculating short-term
+///   autocorrelation function of the enveloped signal.
+/// - After whole sound data file has been analyzed as above, the bpm level is
+///   detected by function 'getBpm' that finds the highest peak of the autocorrelation
+///   function, calculates it's precise location and converts this reading to bpm's.
+///
+/// Author        : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+//  SoundTouch audio processing library
+//  Copyright (c) Olli Parviainen
+//
+//  This library is free software; you can redistribute it and/or
+//  modify it under the terms of the GNU Lesser General Public
+//  License as published by the Free Software Foundation; either
+//  version 2.1 of the License, or (at your option) any later version.
+//
+//  This library is distributed in the hope that it will be useful,
+//  but WITHOUT ANY WARRANTY; without even the implied warranty of
+//  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+//  Lesser General Public License for more details.
+//
+//  You should have received a copy of the GNU Lesser General Public
+//  License along with this library; if not, write to the Free Software
+//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#ifndef _BPMDetect_H_
+#define _BPMDetect_H_
+
+#include <vector>
+#include "STTypes.h"
+#include "FIFOSampleBuffer.h"
+
+namespace soundtouch
+{
+
+    /// Minimum allowed BPM rate. Used to restrict accepted result above a reasonable limit.
+    #define MIN_BPM 45
+
+    /// Maximum allowed BPM rate range. Used for calculating algorithm parametrs
+    #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 IIR2_filter
+    {
+        double coeffs[5];
+        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;
+
+        /// Sample average counter.
+        int decimateCount;
+
+        /// Sample average accumulator for FIFO-like decimation.
+        soundtouch::LONG_SAMPLETYPE decimateSum;
+
+        /// Decimate sound by this coefficient to reach approx. 500 Hz.
+        int decimateBy;
+
+        /// Auto-correlation window length
+        int windowLen;
+
+        /// Number of channels (1 = mono, 2 = stereo)
+        int channels;
+
+        /// sample rate
+        int sampleRate;
+
+        /// Beginning of auto-correlation window: Autocorrelation isn't being updated for
+        /// the first these many correlation bins.
+        int windowStart;
+
+        /// 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).
+        void updateXCorr(int process_samples      /// How many samples are processed.
+        );
+
+        /// Decimates samples to approx. 500 Hz.
+        ///
+        /// \return Number of output samples.
+        int decimate(soundtouch::SAMPLETYPE *dest,      ///< Destination buffer
+            const soundtouch::SAMPLETYPE *src, ///< Source sample buffer
+            int numsamples                     ///< Number of source samples.
+        );
+
+        /// Calculates amplitude envelope for the buffer of samples.
+        /// Result is output to 'samples'.
+        void calcEnvelope(soundtouch::SAMPLETYPE *samples,  ///< Pointer to input/output data buffer
+            int numsamples                    ///< Number of samples in buffer
+        );
+
+        /// remove constant bias from xcorr data
+        void removeBias();
+
+        // 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.
+        );
+
+        /// Destructor.
+        virtual ~BPMDetect();
+
+        /// Inputs a block of samples for analyzing: Envelopes the samples and then
+        /// updates the autocorrelation estimation. When whole song data has been input
+        /// in smaller blocks using this function, read the resulting bpm with 'getBpm'
+        /// function.
+        ///
+        /// Notice that data in 'samples' array can be disrupted in processing.
+        void inputSamples(const soundtouch::SAMPLETYPE *samples,    ///< Pointer to input/working data buffer
+            int numSamples                            ///< Number of samples in buffer
+        );
+
+        /// Analyzes the results and returns the BPM rate. Use this function to read result
+        /// after whole song data has been input to the class by consecutive calls of
+        /// 'inputSamples' function.
+        ///
+        /// \return Beats-per-minute rate, or zero if detection failed.
+        float getBpm();
+
+        /// 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

@@ -1,178 +1,180 @@
-////////////////////////////////////////////////////////////////////////////////
-///
-/// A buffer class for temporarily storaging sound samples, operates as a 
-/// first-in-first-out pipe.
-///
-/// Samples are added to the end of the sample buffer with the 'putSamples' 
-/// function, and are received from the beginning of the buffer by calling
-/// the 'receiveSamples' function. The class automatically removes the 
-/// output samples from the buffer as well as grows the storage size 
-/// whenever necessary.
-///
-/// Author        : Copyright (c) Olli Parviainen
-/// Author e-mail : oparviai 'at' iki.fi
-/// SoundTouch WWW: http://www.surina.net/soundtouch
-///
-////////////////////////////////////////////////////////////////////////////////
-//
-// Last changed  : $Date$
-// File revision : $Revision: 4 $
-//
-// $Id$
-//
-////////////////////////////////////////////////////////////////////////////////
-//
-// License :
-//
-//  SoundTouch audio processing library
-//  Copyright (c) Olli Parviainen
-//
-//  This library is free software; you can redistribute it and/or
-//  modify it under the terms of the GNU Lesser General Public
-//  License as published by the Free Software Foundation; either
-//  version 2.1 of the License, or (at your option) any later version.
-//
-//  This library is distributed in the hope that it will be useful,
-//  but WITHOUT ANY WARRANTY; without even the implied warranty of
-//  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-//  Lesser General Public License for more details.
-//
-//  You should have received a copy of the GNU Lesser General Public
-//  License along with this library; if not, write to the Free Software
-//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
-//
-////////////////////////////////////////////////////////////////////////////////
-
-#ifndef FIFOSampleBuffer_H
-#define FIFOSampleBuffer_H
-
-#include "FIFOSamplePipe.h"
-
-namespace soundtouch
-{
-
-/// Sample buffer working in FIFO (first-in-first-out) principle. The class takes
-/// care of storage size adjustment and data moving during input/output operations.
-///
-/// Notice that in case of stereo audio, one sample is considered to consist of 
-/// both channel data.
-class FIFOSampleBuffer : public FIFOSamplePipe
-{
-private:
-    /// Sample buffer.
-    SAMPLETYPE *buffer;
-
-    // Raw unaligned buffer memory. 'buffer' is made aligned by pointing it to first
-    // 16-byte aligned location of this buffer
-    SAMPLETYPE *bufferUnaligned;
-
-    /// Sample buffer size in bytes
-    uint sizeInBytes;
-
-    /// How many samples are currently in buffer.
-    uint samplesInBuffer;
-
-    /// Channels, 1=mono, 2=stereo.
-    uint channels;
-
-    /// Current position pointer to the buffer. This pointer is increased when samples are 
-    /// removed from the pipe so that it's necessary to actually rewind buffer (move data)
-    /// only new data when is put to the pipe.
-    uint bufferPos;
-
-    /// Rewind the buffer by moving data from position pointed by 'bufferPos' to real 
-    /// beginning of the buffer.
-    void rewind();
-
-    /// Ensures that the buffer has capacity for at least this many samples.
-    void ensureCapacity(uint capacityRequirement);
-
-    /// Returns current capacity.
-    uint getCapacity() const;
-
-public:
-
-    /// Constructor
-    FIFOSampleBuffer(int numChannels = 2     ///< Number of channels, 1=mono, 2=stereo.
-                                              ///< Default is stereo.
-                     );
-
-    /// destructor
-    ~FIFOSampleBuffer();
-
-    /// Returns a pointer to the beginning of the output samples. 
-    /// This function is provided for accessing the output samples directly. 
-    /// Please be careful for not to corrupt the book-keeping!
-    ///
-    /// When using this function to output samples, also remember to 'remove' the
-    /// output samples from the buffer by calling the 
-    /// 'receiveSamples(numSamples)' function
-    virtual SAMPLETYPE *ptrBegin();
-
-    /// Returns a pointer to the end of the used part of the sample buffer (i.e. 
-    /// where the new samples are to be inserted). This function may be used for 
-    /// inserting new samples into the sample buffer directly. Please be careful
-    /// not corrupt the book-keeping!
-    ///
-    /// When using this function as means for inserting new samples, also remember 
-    /// to increase the sample count afterwards, by calling  the 
-    /// 'putSamples(numSamples)' function.
-    SAMPLETYPE *ptrEnd(
-                uint slackCapacity   ///< How much free capacity (in samples) there _at least_ 
-                                     ///< should be so that the caller can succesfully insert the 
-                                     ///< desired samples to the buffer. If necessary, the function 
-                                     ///< grows the buffer size to comply with this requirement.
-                );
-
-    /// Adds 'numSamples' pcs of samples from the 'samples' memory position to
-    /// the sample buffer.
-    virtual void putSamples(const SAMPLETYPE *samples,  ///< Pointer to samples.
-                            uint numSamples                         ///< Number of samples to insert.
-                            );
-
-    /// Adjusts the book-keeping to increase number of samples in the buffer without 
-    /// copying any actual samples.
-    ///
-    /// This function is used to update the number of samples in the sample buffer
-    /// when accessing the buffer directly with 'ptrEnd' function. Please be 
-    /// careful though!
-    virtual void putSamples(uint numSamples   ///< Number of samples been inserted.
-                            );
-
-    /// Output samples from beginning of the sample buffer. Copies requested samples to 
-    /// output buffer and removes them from the sample buffer. If there are less than 
-    /// 'numsample' samples in the buffer, returns all that available.
-    ///
-    /// \return Number of samples returned.
-    virtual uint receiveSamples(SAMPLETYPE *output, ///< Buffer where to copy output samples.
-                                uint maxSamples                 ///< How many samples to receive at max.
-                                );
-
-    /// Adjusts book-keeping so that given number of samples are removed from beginning of the 
-    /// sample buffer without copying them anywhere. 
-    ///
-    /// Used to reduce the number of samples in the buffer when accessing the sample buffer directly
-    /// with 'ptrBegin' function.
-    virtual uint receiveSamples(uint maxSamples   ///< Remove this many samples from the beginning of pipe.
-                                );
-
-    /// Returns number of samples currently available.
-    virtual uint numSamples() const;
-
-    /// Sets number of channels, 1 = mono, 2 = stereo.
-    void setChannels(int numChannels);
-
-    /// Returns nonzero if there aren't any samples available for outputting.
-    virtual int isEmpty() const;
-
-    /// Clears all the samples.
-    virtual void clear();
-
-    /// allow trimming (downwards) amount of samples in pipeline.
-    /// Returns adjusted amount of samples
-    uint adjustAmountOfSamples(uint numSamples);
-};
-
-}
-
-#endif
+////////////////////////////////////////////////////////////////////////////////
+///
+/// A buffer class for temporarily storaging sound samples, operates as a
+/// first-in-first-out pipe.
+///
+/// Samples are added to the end of the sample buffer with the 'putSamples'
+/// function, and are received from the beginning of the buffer by calling
+/// the 'receiveSamples' function. The class automatically removes the
+/// output samples from the buffer as well as grows the storage size
+/// whenever necessary.
+///
+/// Author        : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+//  SoundTouch audio processing library
+//  Copyright (c) Olli Parviainen
+//
+//  This library is free software; you can redistribute it and/or
+//  modify it under the terms of the GNU Lesser General Public
+//  License as published by the Free Software Foundation; either
+//  version 2.1 of the License, or (at your option) any later version.
+//
+//  This library is distributed in the hope that it will be useful,
+//  but WITHOUT ANY WARRANTY; without even the implied warranty of
+//  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+//  Lesser General Public License for more details.
+//
+//  You should have received a copy of the GNU Lesser General Public
+//  License along with this library; if not, write to the Free Software
+//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#ifndef FIFOSampleBuffer_H
+#define FIFOSampleBuffer_H
+
+#include "FIFOSamplePipe.h"
+
+namespace soundtouch
+{
+
+/// Sample buffer working in FIFO (first-in-first-out) principle. The class takes
+/// care of storage size adjustment and data moving during input/output operations.
+///
+/// Notice that in case of stereo audio, one sample is considered to consist of
+/// both channel data.
+class FIFOSampleBuffer : public FIFOSamplePipe
+{
+private:
+    /// Sample buffer.
+    SAMPLETYPE *buffer;
+
+    // Raw unaligned buffer memory. 'buffer' is made aligned by pointing it to first
+    // 16-byte aligned location of this buffer
+    SAMPLETYPE *bufferUnaligned;
+
+    /// Sample buffer size in bytes
+    uint sizeInBytes;
+
+    /// How many samples are currently in buffer.
+    uint samplesInBuffer;
+
+    /// Channels, 1=mono, 2=stereo.
+    uint channels;
+
+    /// Current position pointer to the buffer. This pointer is increased when samples are
+    /// removed from the pipe so that it's necessary to actually rewind buffer (move data)
+    /// only new data when is put to the pipe.
+    uint bufferPos;
+
+    /// Rewind the buffer by moving data from position pointed by 'bufferPos' to real
+    /// beginning of the buffer.
+    void rewind();
+
+    /// Ensures that the buffer has capacity for at least this many samples.
+    void ensureCapacity(uint capacityRequirement);
+
+    /// Returns current capacity.
+    uint getCapacity() const;
+
+public:
+
+    /// Constructor
+    FIFOSampleBuffer(int numChannels = 2     ///< Number of channels, 1=mono, 2=stereo.
+                                              ///< Default is stereo.
+                     );
+
+    /// destructor
+    ~FIFOSampleBuffer() 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!
+    ///
+    /// 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() 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
+    /// inserting new samples into the sample buffer directly. Please be careful
+    /// not corrupt the book-keeping!
+    ///
+    /// When using this function as means for inserting new samples, also remember
+    /// to increase the sample count afterwards, by calling  the
+    /// 'putSamples(numSamples)' function.
+    SAMPLETYPE *ptrEnd(
+                uint slackCapacity   ///< How much free capacity (in samples) there _at least_
+                                     ///< should be so that the caller can successfully insert the
+                                     ///< desired samples to the buffer. If necessary, the function
+                                     ///< grows the buffer size to comply with this requirement.
+                );
+
+    /// Adds 'numSamples' pcs of samples from the 'samples' memory position to
+    /// the sample buffer.
+    virtual void putSamples(const SAMPLETYPE *samples,  ///< Pointer to samples.
+                            uint numSamples                         ///< Number of samples to insert.
+                            ) override;
+
+    /// Adjusts the book-keeping to increase number of samples in the buffer without
+    /// copying any actual samples.
+    ///
+    /// This function is used to update the number of samples in the sample buffer
+    /// when accessing the buffer directly with 'ptrEnd' function. Please be
+    /// careful though!
+    virtual void putSamples(uint numSamples   ///< Number of samples been inserted.
+                            );
+
+    /// Output samples from beginning of the sample buffer. Copies requested samples to
+    /// output buffer and removes them from the sample buffer. If there are less than
+    /// 'numsample' samples in the buffer, returns all that available.
+    ///
+    /// \return Number of samples returned.
+    virtual uint receiveSamples(SAMPLETYPE *output, ///< Buffer where to copy output samples.
+                                uint maxSamples                 ///< How many samples to receive at max.
+                                ) 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;
+
+    /// Returns number of samples currently available.
+    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 override;
+
+    /// Clears all the samples.
+    virtual void clear() override;
+
+    /// allow trimming (downwards) amount of samples in pipeline.
+    /// Returns adjusted amount of samples
+    uint adjustAmountOfSamples(uint numSamples) override;
+
+    /// Add silence to end of buffer
+    void addSilent(uint nSamples);
+};
+
+}
+
+#endif

include/FIFOSamplePipe.h

@@ -1,234 +1,230 @@
-////////////////////////////////////////////////////////////////////////////////
-///
-/// 'FIFOSamplePipe' : An abstract base class for classes that manipulate sound
-/// samples by operating like a first-in-first-out pipe: New samples are fed
-/// into one end of the pipe with the 'putSamples' function, and the processed
-/// samples are received from the other end with the 'receiveSamples' function.
-///
-/// 'FIFOProcessor' : A base class for classes the do signal processing with 
-/// the samples while operating like a first-in-first-out pipe. When samples
-/// are input with the 'putSamples' function, the class processes them
-/// and moves the processed samples to the given 'output' pipe object, which
-/// may be either another processing stage, or a fifo sample buffer object.
-///
-/// Author        : Copyright (c) Olli Parviainen
-/// Author e-mail : oparviai 'at' iki.fi
-/// SoundTouch WWW: http://www.surina.net/soundtouch
-///
-////////////////////////////////////////////////////////////////////////////////
-//
-// Last changed  : $Date$
-// File revision : $Revision: 4 $
-//
-// $Id$
-//
-////////////////////////////////////////////////////////////////////////////////
-//
-// License :
-//
-//  SoundTouch audio processing library
-//  Copyright (c) Olli Parviainen
-//
-//  This library is free software; you can redistribute it and/or
-//  modify it under the terms of the GNU Lesser General Public
-//  License as published by the Free Software Foundation; either
-//  version 2.1 of the License, or (at your option) any later version.
-//
-//  This library is distributed in the hope that it will be useful,
-//  but WITHOUT ANY WARRANTY; without even the implied warranty of
-//  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-//  Lesser General Public License for more details.
-//
-//  You should have received a copy of the GNU Lesser General Public
-//  License along with this library; if not, write to the Free Software
-//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
-//
-////////////////////////////////////////////////////////////////////////////////
-
-#ifndef FIFOSamplePipe_H
-#define FIFOSamplePipe_H
-
-#include <assert.h>
-#include <stdlib.h>
-#include "STTypes.h"
-
-namespace soundtouch
-{
-
-/// Abstract base class for FIFO (first-in-first-out) sample processing classes.
-class FIFOSamplePipe
-{
-public:
-    // virtual default destructor
-    virtual ~FIFOSamplePipe() {}
-
-
-    /// Returns a pointer to the beginning of the output samples. 
-    /// This function is provided for accessing the output samples directly. 
-    /// Please be careful for not to corrupt the book-keeping!
-    ///
-    /// When using this function to output samples, also remember to 'remove' the
-    /// output samples from the buffer by calling the 
-    /// 'receiveSamples(numSamples)' function
-    virtual SAMPLETYPE *ptrBegin() = 0;
-
-    /// Adds 'numSamples' pcs of samples from the 'samples' memory position to
-    /// the sample buffer.
-    virtual void putSamples(const SAMPLETYPE *samples,  ///< Pointer to samples.
-                            uint numSamples             ///< Number of samples to insert.
-                            ) = 0;
-
-
-    // Moves samples from the 'other' pipe instance to this instance.
-    void moveSamples(FIFOSamplePipe &other  ///< Other pipe instance where from the receive the data.
-         )
-    {
-        int oNumSamples = other.numSamples();
-
-        putSamples(other.ptrBegin(), oNumSamples);
-        other.receiveSamples(oNumSamples);
-    };
-
-    /// Output samples from beginning of the sample buffer. Copies requested samples to 
-    /// output buffer and removes them from the sample buffer. If there are less than 
-    /// 'numsample' samples in the buffer, returns all that available.
-    ///
-    /// \return Number of samples returned.
-    virtual uint receiveSamples(SAMPLETYPE *output, ///< Buffer where to copy output samples.
-                                uint maxSamples                 ///< How many samples to receive at max.
-                                ) = 0;
-
-    /// Adjusts book-keeping so that given number of samples are removed from beginning of the 
-    /// sample buffer without copying them anywhere. 
-    ///
-    /// Used to reduce the number of samples in the buffer when accessing the sample buffer directly
-    /// with 'ptrBegin' function.
-    virtual uint receiveSamples(uint maxSamples   ///< Remove this many samples from the beginning of pipe.
-                                ) = 0;
-
-    /// Returns number of samples currently available.
-    virtual uint numSamples() const = 0;
-
-    // Returns nonzero if there aren't any samples available for outputting.
-    virtual int isEmpty() const = 0;
-
-    /// Clears all the samples.
-    virtual void clear() = 0;
-
-    /// allow trimming (downwards) amount of samples in pipeline.
-    /// Returns adjusted amount of samples
-    virtual uint adjustAmountOfSamples(uint numSamples) = 0;
-
-};
-
-
-
-/// Base-class for sound processing routines working in FIFO principle. With this base 
-/// class it's easy to implement sound processing stages that can be chained together,
-/// so that samples that are fed into beginning of the pipe automatically go through 
-/// all the processing stages.
-///
-/// When samples are input to this class, they're first processed and then put to 
-/// the FIFO pipe that's defined as output of this class. This output pipe can be
-/// either other processing stage or a FIFO sample buffer.
-class FIFOProcessor :public FIFOSamplePipe
-{
-protected:
-    /// Internal pipe where processed samples are put.
-    FIFOSamplePipe *output;
-
-    /// Sets output pipe.
-    void setOutPipe(FIFOSamplePipe *pOutput)
-    {
-        assert(output == NULL);
-        assert(pOutput != NULL);
-        output = pOutput;
-    }
-
-
-    /// Constructor. Doesn't define output pipe; it has to be set be 
-    /// 'setOutPipe' function.
-    FIFOProcessor()
-    {
-        output = NULL;
-    }
-
-
-    /// Constructor. Configures output pipe.
-    FIFOProcessor(FIFOSamplePipe *pOutput   ///< Output pipe.
-                 )
-    {
-        output = pOutput;
-    }
-
-
-    /// Destructor.
-    virtual ~FIFOProcessor()
-    {
-    }
-
-
-    /// Returns a pointer to the beginning of the output samples. 
-    /// This function is provided for accessing the output samples directly. 
-    /// Please be careful for not to corrupt the book-keeping!
-    ///
-    /// When using this function to output samples, also remember to 'remove' the
-    /// output samples from the buffer by calling the 
-    /// 'receiveSamples(numSamples)' function
-    virtual SAMPLETYPE *ptrBegin()
-    {
-        return output->ptrBegin();
-    }
-
-public:
-
-    /// Output samples from beginning of the sample buffer. Copies requested samples to 
-    /// output buffer and removes them from the sample buffer. If there are less than 
-    /// 'numsample' samples in the buffer, returns all that available.
-    ///
-    /// \return Number of samples returned.
-    virtual uint receiveSamples(SAMPLETYPE *outBuffer, ///< Buffer where to copy output samples.
-                                uint maxSamples                    ///< How many samples to receive at max.
-                                )
-    {
-        return output->receiveSamples(outBuffer, maxSamples);
-    }
-
-
-    /// Adjusts book-keeping so that given number of samples are removed from beginning of the 
-    /// sample buffer without copying them anywhere. 
-    ///
-    /// Used to reduce the number of samples in the buffer when accessing the sample buffer directly
-    /// with 'ptrBegin' function.
-    virtual uint receiveSamples(uint maxSamples   ///< Remove this many samples from the beginning of pipe.
-                                )
-    {
-        return output->receiveSamples(maxSamples);
-    }
-
-
-    /// Returns number of samples currently available.
-    virtual uint numSamples() const
-    {
-        return output->numSamples();
-    }
-
-
-    /// Returns nonzero if there aren't any samples available for outputting.
-    virtual int isEmpty() const
-    {
-        return output->isEmpty();
-    }
-
-    /// allow trimming (downwards) amount of samples in pipeline.
-    /// Returns adjusted amount of samples
-    virtual uint adjustAmountOfSamples(uint numSamples)
-    {
-        return output->adjustAmountOfSamples(numSamples);
-    }
-
-};
-
-}
-
-#endif
+////////////////////////////////////////////////////////////////////////////////
+///
+/// 'FIFOSamplePipe' : An abstract base class for classes that manipulate sound
+/// samples by operating like a first-in-first-out pipe: New samples are fed
+/// into one end of the pipe with the 'putSamples' function, and the processed
+/// samples are received from the other end with the 'receiveSamples' function.
+///
+/// 'FIFOProcessor' : A base class for classes the do signal processing with
+/// the samples while operating like a first-in-first-out pipe. When samples
+/// are input with the 'putSamples' function, the class processes them
+/// and moves the processed samples to the given 'output' pipe object, which
+/// may be either another processing stage, or a fifo sample buffer object.
+///
+/// Author        : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+//  SoundTouch audio processing library
+//  Copyright (c) Olli Parviainen
+//
+//  This library is free software; you can redistribute it and/or
+//  modify it under the terms of the GNU Lesser General Public
+//  License as published by the Free Software Foundation; either
+//  version 2.1 of the License, or (at your option) any later version.
+//
+//  This library is distributed in the hope that it will be useful,
+//  but WITHOUT ANY WARRANTY; without even the implied warranty of
+//  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+//  Lesser General Public License for more details.
+//
+//  You should have received a copy of the GNU Lesser General Public
+//  License along with this library; if not, write to the Free Software
+//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#ifndef FIFOSamplePipe_H
+#define FIFOSamplePipe_H
+
+#include <assert.h>
+#include <stdlib.h>
+#include "STTypes.h"
+
+namespace soundtouch
+{
+
+/// Abstract base class for FIFO (first-in-first-out) sample processing classes.
+class FIFOSamplePipe
+{
+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() {}
+
+
+    /// Returns a pointer to the beginning of the output samples.
+    /// This function is provided for accessing the output samples directly.
+    /// Please be careful for not to corrupt the book-keeping!
+    ///
+    /// When using this function to output samples, also remember to 'remove' the
+    /// output samples from the buffer by calling the
+    /// 'receiveSamples(numSamples)' function
+    virtual SAMPLETYPE *ptrBegin() = 0;
+
+    /// Adds 'numSamples' pcs of samples from the 'samples' memory position to
+    /// the sample buffer.
+    virtual void putSamples(const SAMPLETYPE *samples,  ///< Pointer to samples.
+                            uint numSamples             ///< Number of samples to insert.
+                            ) = 0;
+
+
+    // Moves samples from the 'other' pipe instance to this instance.
+    void moveSamples(FIFOSamplePipe &other  ///< Other pipe instance where from the receive the data.
+         )
+    {
+        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
+    /// 'numsample' samples in the buffer, returns all that available.
+    ///
+    /// \return Number of samples returned.
+    virtual uint receiveSamples(SAMPLETYPE *output, ///< Buffer where to copy output samples.
+                                uint maxSamples                 ///< How many samples to receive at max.
+                                ) = 0;
+
+    /// Adjusts book-keeping so that given number of samples are removed from beginning of the
+    /// sample buffer without copying them anywhere.
+    ///
+    /// Used to reduce the number of samples in the buffer when accessing the sample buffer directly
+    /// with 'ptrBegin' function.
+    virtual uint receiveSamples(uint maxSamples   ///< Remove this many samples from the beginning of pipe.
+                                ) = 0;
+
+    /// Returns number of samples currently available.
+    virtual uint numSamples() const = 0;
+
+    // Returns nonzero if there aren't any samples available for outputting.
+    virtual int isEmpty() const = 0;
+
+    /// Clears all the samples.
+    virtual void clear() = 0;
+
+    /// allow trimming (downwards) amount of samples in pipeline.
+    /// Returns adjusted amount of samples
+    virtual uint adjustAmountOfSamples(uint numSamples) = 0;
+
+};
+
+
+/// Base-class for sound processing routines working in FIFO principle. With this base
+/// class it's easy to implement sound processing stages that can be chained together,
+/// so that samples that are fed into beginning of the pipe automatically go through
+/// all the processing stages.
+///
+/// When samples are input to this class, they're first processed and then put to
+/// the FIFO pipe that's defined as output of this class. This output pipe can be
+/// either other processing stage or a FIFO sample buffer.
+class FIFOProcessor :public FIFOSamplePipe
+{
+protected:
+    /// Internal pipe where processed samples are put.
+    FIFOSamplePipe *output;
+
+    /// Sets output pipe.
+    void setOutPipe(FIFOSamplePipe *pOutput)
+    {
+        assert(output == nullptr);
+        assert(pOutput != nullptr);
+        output = pOutput;
+    }
+
+    /// Constructor. Doesn't define output pipe; it has to be set be
+    /// 'setOutPipe' function.
+    FIFOProcessor()
+    {
+        output = nullptr;
+    }
+
+    /// Constructor. Configures output pipe.
+    FIFOProcessor(FIFOSamplePipe *pOutput   ///< Output pipe.
+                 )
+    {
+        output = pOutput;
+    }
+
+    /// Destructor.
+    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!
+    ///
+    /// 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() override
+    {
+        return output->ptrBegin();
+    }
+
+public:
+
+    /// Output samples from beginning of the sample buffer. Copies requested samples to
+    /// output buffer and removes them from the sample buffer. If there are less than
+    /// 'numsample' samples in the buffer, returns all that available.
+    ///
+    /// \return Number of samples returned.
+    virtual uint receiveSamples(SAMPLETYPE *outBuffer, ///< Buffer where to copy output samples.
+                                uint maxSamples                    ///< How many samples to receive at max.
+                                ) 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 override
+    {
+        return output->numSamples();
+    }
+
+    /// Returns nonzero if there aren't any samples available for outputting.
+    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) override
+    {
+        return output->adjustAmountOfSamples(numSamples);
+    }
+};
+
+}
+
+#endif

include/Makefile.am

@@ -1,26 +1,22 @@
-## 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
-## terms of the GNU General Public License as published by the Free Software
-## Foundation; either version 2 of the License, or (at your option) any later
-## version.
-## 
-## SoundTouch is distributed in the hope that it will be useful, but WITHOUT ANY
-## WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
-## A PARTICULAR PURPOSE.  See the GNU General Public License for more details.
-## 
-## You should have received a copy of the GNU General Public License along with
-## this program; if not, write to the Free Software Foundation, Inc., 59 Temple
-## Place - Suite 330, Boston, MA  02111-1307, USA
-
-## I used config/am_include.mk for common definitions
-include $(top_srcdir)/config/am_include.mk
-
-pkginclude_HEADERS=FIFOSampleBuffer.h FIFOSamplePipe.h SoundTouch.h STTypes.h BPMDetect.h soundtouch_config.h
-
+## Process this file with automake to create Makefile.in
+##
+## This file is part of SoundTouch, an audio processing library for pitch/time adjustments
+##
+## SoundTouch is free software; you can redistribute it and/or modify it under the
+## terms of the GNU General Public License as published by the Free Software
+## Foundation; either version 2 of the License, or (at your option) any later
+## version.
+##
+## SoundTouch is distributed in the hope that it will be useful, but WITHOUT ANY
+## WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
+## A PARTICULAR PURPOSE.  See the GNU General Public License for more details.
+##
+## You should have received a copy of the GNU General Public License along with
+## this program; if not, write to the Free Software Foundation, Inc., 59 Temple
+## Place - Suite 330, Boston, MA  02111-1307, USA
+
+## I used config/am_include.mk for common definitions
+include $(top_srcdir)/config/am_include.mk
+
+pkginclude_HEADERS=FIFOSampleBuffer.h FIFOSamplePipe.h SoundTouch.h STTypes.h BPMDetect.h soundtouch_config.h
+

include/STTypes.h

@@ -1,167 +1,191 @@
-////////////////////////////////////////////////////////////////////////////////
-///
-/// Common type definitions for SoundTouch audio processing library.
-///
-/// Author        : Copyright (c) Olli Parviainen
-/// Author e-mail : oparviai 'at' iki.fi
-/// SoundTouch WWW: http://www.surina.net/soundtouch
-///
-////////////////////////////////////////////////////////////////////////////////
-//
-// Last changed  : $Date$
-// File revision : $Revision: 3 $
-//
-// $Id$
-//
-////////////////////////////////////////////////////////////////////////////////
-//
-// License :
-//
-//  SoundTouch audio processing library
-//  Copyright (c) Olli Parviainen
-//
-//  This library is free software; you can redistribute it and/or
-//  modify it under the terms of the GNU Lesser General Public
-//  License as published by the Free Software Foundation; either
-//  version 2.1 of the License, or (at your option) any later version.
-//
-//  This library is distributed in the hope that it will be useful,
-//  but WITHOUT ANY WARRANTY; without even the implied warranty of
-//  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-//  Lesser General Public License for more details.
-//
-//  You should have received a copy of the GNU Lesser General Public
-//  License along with this library; if not, write to the Free Software
-//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
-//
-////////////////////////////////////////////////////////////////////////////////
-
-#ifndef STTypes_H
-#define STTypes_H
-
-typedef unsigned int    uint;
-typedef unsigned long   ulong;
-
-#ifdef __GNUC__
-    // In GCC, include soundtouch_config.h made by config scritps
-    #include "soundtouch_config.h"
-#endif
-
-#ifndef _WINDEF_
-    // if these aren't defined already by Windows headers, define now
-
-    typedef int BOOL;
-
-    #define FALSE   0
-    #define TRUE    1
-
-#endif  // _WINDEF_
-
-
-namespace soundtouch
-{
-    /// Activate these undef's to overrule the possible sampletype 
-    /// setting inherited from some other header file:
-    //#undef SOUNDTOUCH_INTEGER_SAMPLES
-    //#undef SOUNDTOUCH_FLOAT_SAMPLES
-
-    #if !(SOUNDTOUCH_INTEGER_SAMPLES || SOUNDTOUCH_FLOAT_SAMPLES)
-       
-        /// Choose either 32bit floating point or 16bit integer sampletype
-        /// by choosing one of the following defines, unless this selection 
-        /// has already been done in some other file.
-        ////
-        /// Notes:
-        /// - In Windows environment, choose the sample format with the
-        ///   following defines.
-        /// - In GNU environment, the floating point samples are used by 
-        ///   default, but integer samples can be chosen by giving the 
-        ///   following switch to the configure script:
-        ///       ./configure --enable-integer-samples
-        ///   However, if you still prefer to select the sample format here 
-        ///   also in GNU environment, then please #undef the INTEGER_SAMPLE
-        ///   and FLOAT_SAMPLE defines first as in comments above.
-        //#define SOUNDTOUCH_INTEGER_SAMPLES     1    //< 16bit integer samples
-        #define SOUNDTOUCH_FLOAT_SAMPLES       1    //< 32bit float samples
-     
-    #endif
-
-    #if (_M_IX86 || __i386__ || __x86_64__ || _M_X64)
-        /// Define this to allow X86-specific assembler/intrinsic optimizations. 
-        /// Notice that library contains also usual C++ versions of each of these
-        /// these routines, so if you're having difficulties getting the optimized 
-        /// routines compiled for whatever reason, you may disable these optimizations 
-        /// to make the library compile.
-
-        #define SOUNDTOUCH_ALLOW_X86_OPTIMIZATIONS     1
-
-        /// In GNU environment, allow the user to override this setting by
-        /// giving the following switch to the configure script:
-        /// ./configure --disable-x86-optimizations
-        /// ./configure --enable-x86-optimizations=no
-        #ifdef SOUNDTOUCH_DISABLE_X86_OPTIMIZATIONS
-            #undef SOUNDTOUCH_ALLOW_X86_OPTIMIZATIONS
-        #endif
-    #else
-        /// Always disable optimizations when not using a x86 systems.
-        #undef SOUNDTOUCH_ALLOW_X86_OPTIMIZATIONS
-
-    #endif
-
-    // If defined, allows the SIMD-optimized routines to take minor shortcuts 
-    // for improved performance. Undefine to require faithfully similar SIMD 
-    // calculations as in normal C implementation.
-    #define SOUNDTOUCH_ALLOW_NONEXACT_SIMD_OPTIMIZATION    1
-
-
-    #ifdef SOUNDTOUCH_INTEGER_SAMPLES
-        // 16bit integer sample type
-        typedef short SAMPLETYPE;
-        // data type for sample accumulation: Use 32bit integer to prevent overflows
-        typedef long  LONG_SAMPLETYPE;
-
-        #ifdef SOUNDTOUCH_FLOAT_SAMPLES
-            // check that only one sample type is defined
-            #error "conflicting sample types defined"
-        #endif // SOUNDTOUCH_FLOAT_SAMPLES
-
-        #ifdef SOUNDTOUCH_ALLOW_X86_OPTIMIZATIONS
-            // Allow MMX optimizations
-            #define SOUNDTOUCH_ALLOW_MMX   1
-        #endif
-
-    #else
-
-        // floating point samples
-        typedef float  SAMPLETYPE;
-        // data type for sample accumulation: Use double to utilize full precision.
-        typedef double LONG_SAMPLETYPE;
-
-        #ifdef SOUNDTOUCH_ALLOW_X86_OPTIMIZATIONS
-            // Allow SSE optimizations
-            #define SOUNDTOUCH_ALLOW_SSE       1
-        #endif
-
-    #endif  // SOUNDTOUCH_INTEGER_SAMPLES
-
-};
-
-// define ST_NO_EXCEPTION_HANDLING switch to disable throwing std exceptions:
-// #define ST_NO_EXCEPTION_HANDLING    1
-#ifdef ST_NO_EXCEPTION_HANDLING
-    // Exceptions disabled. Throw asserts instead if enabled.
-    #include <assert.h>
-    #define ST_THROW_RT_ERROR(x)    {assert((const char *)x);}
-#else
-    // use c++ standard exceptions
-    #include <stdexcept>
-    #define ST_THROW_RT_ERROR(x)    {throw std::runtime_error(x);}
-#endif
-
-// When this #define is active, eliminates a clicking sound when the "rate" or "pitch" 
-// parameter setting crosses from value <1 to >=1 or vice versa during processing. 
-// Default is off as such crossover is untypical case and involves a slight sound 
-// quality compromise.
-//#define SOUNDTOUCH_PREVENT_CLICK_AT_RATE_CROSSOVER   1
-
-#endif
+////////////////////////////////////////////////////////////////////////////////
+///
+/// Common type definitions for SoundTouch audio processing library.
+///
+/// Author        : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+//  SoundTouch audio processing library
+//  Copyright (c) Olli Parviainen
+//
+//  This library is free software; you can redistribute it and/or
+//  modify it under the terms of the GNU Lesser General Public
+//  License as published by the Free Software Foundation; either
+//  version 2.1 of the License, or (at your option) any later version.
+//
+//  This library is distributed in the hope that it will be useful,
+//  but WITHOUT ANY WARRANTY; without even the implied warranty of
+//  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+//  Lesser General Public License for more details.
+//
+//  You should have received a copy of the GNU Lesser General Public
+//  License along with this library; if not, write to the Free Software
+//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#ifndef STTypes_H
+#define STTypes_H
+
+typedef unsigned int    uint;
+typedef unsigned long   ulong;
+
+// Patch for MinGW: on Win64 long is 32-bit
+#ifdef _WIN64
+    typedef unsigned long long ulongptr;
+#else
+    typedef ulong ulongptr;
+#endif
+
+
+// Helper macro for aligning pointer up to next 16-byte boundary
+#define SOUNDTOUCH_ALIGN_POINTER_16(x)      ( ( (ulongptr)(x) + 15 ) & ~(ulongptr)15 )
+
+
+#if (defined(__GNUC__) && !defined(ANDROID))
+    // In GCC, include soundtouch_config.h made by config scritps.
+    // Skip this in Android compilation that uses GCC but without configure scripts.
+    #include "soundtouch_config.h"
+#endif
+
+
+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
+        /// by choosing one of the following defines, unless this selection
+        /// has already been done in some other file.
+        ////
+        /// Notes:
+        /// - In Windows environment, choose the sample format with the
+        ///   following defines.
+        /// - In GNU environment, the floating point samples are used by
+        ///   default, but integer samples can be chosen by giving the
+        ///   following switch to the configure script:
+        ///       ./configure --enable-integer-samples
+        ///   However, if you still prefer to select the sample format here
+        ///   also in GNU environment, then please #undef the INTEGER_SAMPLE
+        ///   and FLOAT_SAMPLE defines first as in comments above.
+        //#define SOUNDTOUCH_INTEGER_SAMPLES     1    //< 16bit integer samples
+        #define SOUNDTOUCH_FLOAT_SAMPLES       1    //< 32bit float samples
+
+    #endif
+
+    #if (_M_IX86 || __i386__ || __x86_64__ || _M_X64)
+        /// Define this to allow X86-specific assembler/intrinsic optimizations.
+        /// Notice that library contains also usual C++ versions of each of these
+        /// these routines, so if you're having difficulties getting the optimized
+        /// routines compiled for whatever reason, you may disable these optimizations
+        /// to make the library compile.
+
+        #define SOUNDTOUCH_ALLOW_X86_OPTIMIZATIONS     1
+
+        /// In GNU environment, allow the user to override this setting by
+        /// giving the following switch to the configure script:
+        /// ./configure --disable-x86-optimizations
+        /// ./configure --enable-x86-optimizations=no
+        #ifdef SOUNDTOUCH_DISABLE_X86_OPTIMIZATIONS
+            #undef SOUNDTOUCH_ALLOW_X86_OPTIMIZATIONS
+        #endif
+    #else
+        /// Always disable optimizations when not using a x86 systems.
+        #undef SOUNDTOUCH_ALLOW_X86_OPTIMIZATIONS
+
+    #endif
+
+    // If defined, allows the SIMD-optimized routines to skip unevenly aligned
+    // memory offsets that can cause performance penalty in some SIMD implementations.
+    // Causes slight compromise in sound quality.
+    // #define SOUNDTOUCH_ALLOW_NONEXACT_SIMD_OPTIMIZATION    1
+
+
+    #ifdef SOUNDTOUCH_INTEGER_SAMPLES
+        // 16bit integer sample type
+        typedef short SAMPLETYPE;
+        // data type for sample accumulation: Use 32bit integer to prevent overflows
+        typedef long  LONG_SAMPLETYPE;
+
+        #ifdef SOUNDTOUCH_FLOAT_SAMPLES
+            // check that only one sample type is defined
+            #error "conflicting sample types defined"
+        #endif // SOUNDTOUCH_FLOAT_SAMPLES
+
+        #ifdef SOUNDTOUCH_ALLOW_X86_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 float also here to enable
+        // efficient autovectorization
+        typedef float LONG_SAMPLETYPE;
+
+        #ifdef SOUNDTOUCH_ALLOW_X86_OPTIMIZATIONS
+            // Allow SSE optimizations
+            #define SOUNDTOUCH_ALLOW_SSE       1
+        #endif
+
+    #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
+#ifdef ST_NO_EXCEPTION_HANDLING
+    // Exceptions disabled. Throw asserts instead if enabled.
+    #include <assert.h>
+    #define ST_THROW_RT_ERROR(x)    {assert((const char *)x);}
+#else
+    // use c++ standard exceptions
+    #include <stdexcept>
+    #include <string>
+    #define ST_THROW_RT_ERROR(x)    {throw std::runtime_error(x);}
+#endif
+
+// When this #define is active, eliminates a clicking sound when the "rate" or "pitch"
+// parameter setting crosses from value <1 to >=1 or vice versa during processing.
+// Default is off as such crossover is untypical case and involves a slight sound
+// quality compromise.
+//#define SOUNDTOUCH_PREVENT_CLICK_AT_RATE_CROSSOVER   1
+
+#endif

include/SoundTouch.h

@@ -1,277 +1,348 @@
-//////////////////////////////////////////////////////////////////////////////
-///
-/// SoundTouch - main class for tempo/pitch/rate adjusting routines. 
-///
-/// Notes:
-/// - Initialize the SoundTouch object instance by setting up the sound stream 
-///   parameters with functions 'setSampleRate' and 'setChannels', then set 
-///   desired tempo/pitch/rate settings with the corresponding functions.
-///
-/// - The SoundTouch class behaves like a first-in-first-out pipeline: The 
-///   samples that are to be processed are fed into one of the pipe by calling
-///   function 'putSamples', while the ready processed samples can be read 
-///   from the other end of the pipeline with function 'receiveSamples'.
-/// 
-/// - The SoundTouch processing classes require certain sized 'batches' of 
-///   samples in order to process the sound. For this reason the classes buffer 
-///   incoming samples until there are enough of samples available for 
-///   processing, then they carry out the processing step and consequently
-///   make the processed samples available for outputting.
-/// 
-/// - For the above reason, the processing routines introduce a certain 
-///   'latency' between the input and output, so that the samples input to
-///   SoundTouch may not be immediately available in the output, and neither 
-///   the amount of outputtable samples may not immediately be in direct 
-///   relationship with the amount of previously input samples.
-///
-/// - The tempo/pitch/rate control parameters can be altered during processing.
-///   Please notice though that they aren't currently protected by semaphores,
-///   so in multi-thread application external semaphore protection may be
-///   required.
-///
-/// - This class utilizes classes 'TDStretch' for tempo change (without modifying
-///   pitch) and 'RateTransposer' for changing the playback rate (that is, both 
-///   tempo and pitch in the same ratio) of the sound. The third available control 
-///   'pitch' (change pitch but maintain tempo) is produced by a combination of
-///   combining the two other controls.
-///
-/// Author        : Copyright (c) Olli Parviainen
-/// Author e-mail : oparviai 'at' iki.fi
-/// SoundTouch WWW: http://www.surina.net/soundtouch
-///
-////////////////////////////////////////////////////////////////////////////////
-//
-// Last changed  : $Date$
-// File revision : $Revision: 4 $
-//
-// $Id$
-//
-////////////////////////////////////////////////////////////////////////////////
-//
-// License :
-//
-//  SoundTouch audio processing library
-//  Copyright (c) Olli Parviainen
-//
-//  This library is free software; you can redistribute it and/or
-//  modify it under the terms of the GNU Lesser General Public
-//  License as published by the Free Software Foundation; either
-//  version 2.1 of the License, or (at your option) any later version.
-//
-//  This library is distributed in the hope that it will be useful,
-//  but WITHOUT ANY WARRANTY; without even the implied warranty of
-//  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-//  Lesser General Public License for more details.
-//
-//  You should have received a copy of the GNU Lesser General Public
-//  License along with this library; if not, write to the Free Software
-//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
-//
-////////////////////////////////////////////////////////////////////////////////
-
-#ifndef SoundTouch_H
-#define SoundTouch_H
-
-#include "FIFOSamplePipe.h"
-#include "STTypes.h"
-
-namespace soundtouch
-{
-
-/// Soundtouch library version string
-#define SOUNDTOUCH_VERSION          "1.7.0"
-
-/// SoundTouch library version id
-#define SOUNDTOUCH_VERSION_ID       (10700)
-
-//
-// Available setting IDs for the 'setSetting' & 'get_setting' functions:
-
-/// Enable/disable anti-alias filter in pitch transposer (0 = disable)
-#define SETTING_USE_AA_FILTER       0
-
-/// Pitch transposer anti-alias filter length (8 .. 128 taps, default = 32)
-#define SETTING_AA_FILTER_LENGTH    1
-
-/// Enable/disable quick seeking algorithm in tempo changer routine
-/// (enabling quick seeking lowers CPU utilization but causes a minor sound
-///  quality compromising)
-#define SETTING_USE_QUICKSEEK       2
-
-/// Time-stretch algorithm single processing sequence length in milliseconds. This determines 
-/// to how long sequences the original sound is chopped in the time-stretch algorithm. 
-/// See "STTypes.h" or README for more information.
-#define SETTING_SEQUENCE_MS         3
-
-/// Time-stretch algorithm seeking window length in milliseconds for algorithm that finds the 
-/// best possible overlapping location. This determines from how wide window the algorithm 
-/// may look for an optimal joining location when mixing the sound sequences back together. 
-/// See "STTypes.h" or README for more information.
-#define SETTING_SEEKWINDOW_MS       4
-
-/// Time-stretch algorithm overlap length in milliseconds. When the chopped sound sequences 
-/// are mixed back together, to form a continuous sound stream, this parameter defines over 
-/// how long period the two consecutive sequences are let to overlap each other. 
-/// See "STTypes.h" or README for more information.
-#define SETTING_OVERLAP_MS          5
-
-
-/// Call "getSetting" with this ID to query nominal average processing sequence
-/// size in samples. This value tells approcimate value how many input samples 
-/// SoundTouch needs to gather before it does DSP processing run for the sample batch.
-///
-/// Notices: 
-/// - This is read-only parameter, i.e. setSetting ignores this parameter
-/// - Returned value is approximate average value, exact processing batch
-///   size may wary from time to time
-/// - This parameter value is not constant but may change depending on 
-///   tempo/pitch/rate/samplerate settings.
-#define SETTING_NOMINAL_INPUT_SEQUENCE		6
-
-
-/// Call "getSetting" with this ID to query nominal average processing output 
-/// size in samples. This value tells approcimate value how many output samples 
-/// SoundTouch outputs once it does DSP processing run for a batch of input samples.
-///	
-/// Notices: 
-/// - This is read-only parameter, i.e. setSetting ignores this parameter
-/// - Returned value is approximate average value, exact processing batch
-///   size may wary from time to time
-/// - This parameter value is not constant but may change depending on 
-///   tempo/pitch/rate/samplerate settings.
-#define SETTING_NOMINAL_OUTPUT_SEQUENCE		7
-
-class SoundTouch : public FIFOProcessor
-{
-private:
-    /// Rate transposer class instance
-    class RateTransposer *pRateTransposer;
-
-    /// Time-stretch class instance
-    class TDStretch *pTDStretch;
-
-    /// Virtual pitch parameter. Effective rate & tempo are calculated from these parameters.
-    float virtualRate;
-
-    /// Virtual pitch parameter. Effective rate & tempo are calculated from these parameters.
-    float virtualTempo;
-
-    /// Virtual pitch parameter. Effective rate & tempo are calculated from these parameters.
-    float virtualPitch;
-
-    /// Flag: Has sample rate been set?
-    BOOL  bSrateSet;
-
-    /// Calculates effective rate & tempo valuescfrom 'virtualRate', 'virtualTempo' and 
-    /// 'virtualPitch' parameters.
-    void calcEffectiveRateAndTempo();
-
-protected :
-    /// Number of channels
-    uint  channels;
-
-    /// Effective 'rate' value calculated from 'virtualRate', 'virtualTempo' and 'virtualPitch'
-    float rate;
-
-    /// Effective 'tempo' value calculated from 'virtualRate', 'virtualTempo' and 'virtualPitch'
-    float tempo;
-
-public:
-    SoundTouch();
-    virtual ~SoundTouch();
-
-    /// Get SoundTouch library version string
-    static const char *getVersionString();
-
-    /// Get SoundTouch library version Id
-    static uint getVersionId();
-
-    /// Sets new rate control value. Normal rate = 1.0, smaller values
-    /// represent slower rate, larger faster rates.
-    void setRate(float newRate);
-
-    /// Sets new tempo control value. Normal tempo = 1.0, smaller values
-    /// represent slower tempo, larger faster tempo.
-    void setTempo(float newTempo);
-
-    /// Sets new rate control value as a difference in percents compared
-    /// to the original rate (-50 .. +100 %)
-    void setRateChange(float newRate);
-
-    /// Sets new tempo control value as a difference in percents compared
-    /// to the original tempo (-50 .. +100 %)
-    void setTempoChange(float newTempo);
-
-    /// Sets new pitch control value. Original pitch = 1.0, smaller values
-    /// represent lower pitches, larger values higher pitch.
-    void setPitch(float newPitch);
-
-    /// Sets pitch change in octaves compared to the original pitch  
-    /// (-1.00 .. +1.00)
-    void setPitchOctaves(float newPitch);
-
-    /// Sets pitch change in semi-tones compared to the original pitch
-    /// (-12 .. +12)
-    void setPitchSemiTones(int newPitch);
-    void setPitchSemiTones(float newPitch);
-
-    /// Sets the number of channels, 1 = mono, 2 = stereo
-    void setChannels(uint numChannels);
-
-    /// Sets sample rate.
-    void setSampleRate(uint srate);
-
-    /// Flushes the last samples from the processing pipeline to the output.
-    /// Clears also the internal processing buffers.
-    //
-    /// Note: This function is meant for extracting the last samples of a sound
-    /// stream. This function may introduce additional blank samples in the end
-    /// of the sound stream, and thus it's not recommended to call this function
-    /// in the middle of a sound stream.
-    void flush();
-
-    /// Adds 'numSamples' pcs of samples from the 'samples' memory position into
-    /// the input of the object. Notice that sample rate _has_to_ be set before
-    /// calling this function, otherwise throws a runtime_error exception.
-    virtual void putSamples(
-            const SAMPLETYPE *samples,  ///< Pointer to sample buffer.
-            uint numSamples                         ///< Number of samples in buffer. Notice
-                                                    ///< that in case of stereo-sound a single sample
-                                                    ///< contains data for both channels.
-            );
-
-    /// Clears all the samples in the object's output and internal processing
-    /// buffers.
-    virtual void clear();
-
-    /// Changes a setting controlling the processing system behaviour. See the
-    /// 'SETTING_...' defines for available setting ID's.
-    /// 
-    /// \return 'TRUE' if the setting was succesfully changed
-    BOOL setSetting(int settingId,   ///< Setting ID number. see SETTING_... defines.
-                    int value        ///< New setting value.
-                    );
-
-    /// Reads a setting controlling the processing system behaviour. See the
-    /// 'SETTING_...' defines for available setting ID's.
-    ///
-    /// \return the setting value.
-    int getSetting(int settingId    ///< Setting ID number, see SETTING_... defines.
-                   ) const;
-
-    /// Returns number of samples currently unprocessed.
-    virtual uint numUnprocessedSamples() const;
-
-
-    /// Other handy functions that are implemented in the ancestor classes (see
-    /// classes 'FIFOProcessor' and 'FIFOSamplePipe')
-    ///
-    /// - receiveSamples() : Use this function to receive 'ready' processed samples from SoundTouch.
-    /// - numSamples()     : Get number of 'ready' samples that can be received with 
-    ///                      function 'receiveSamples()'
-    /// - isEmpty()        : Returns nonzero if there aren't any 'ready' samples.
-    /// - clear()          : Clears all samples from ready/processing buffers.
-};
-
-}
-#endif
+//////////////////////////////////////////////////////////////////////////////
+///
+/// SoundTouch - main class for tempo/pitch/rate adjusting routines.
+///
+/// Notes:
+/// - Initialize the SoundTouch object instance by setting up the sound stream
+///   parameters with functions 'setSampleRate' and 'setChannels', then set
+///   desired tempo/pitch/rate settings with the corresponding functions.
+///
+/// - The SoundTouch class behaves like a first-in-first-out pipeline: The
+///   samples that are to be processed are fed into one of the pipe by calling
+///   function 'putSamples', while the ready processed samples can be read
+///   from the other end of the pipeline with function 'receiveSamples'.
+///
+/// - The SoundTouch processing classes require certain sized 'batches' of
+///   samples in order to process the sound. For this reason the classes buffer
+///   incoming samples until there are enough of samples available for
+///   processing, then they carry out the processing step and consequently
+///   make the processed samples available for outputting.
+///
+/// - For the above reason, the processing routines introduce a certain
+///   'latency' between the input and output, so that the samples input to
+///   SoundTouch may not be immediately available in the output, and neither
+///   the amount of outputtable samples may not immediately be in direct
+///   relationship with the amount of previously input samples.
+///
+/// - The tempo/pitch/rate control parameters can be altered during processing.
+///   Please notice though that they aren't currently protected by semaphores,
+///   so in multi-thread application external semaphore protection may be
+///   required.
+///
+/// - This class utilizes classes 'TDStretch' for tempo change (without modifying
+///   pitch) and 'RateTransposer' for changing the playback rate (that is, both
+///   tempo and pitch in the same ratio) of the sound. The third available control
+///   'pitch' (change pitch but maintain tempo) is produced by a combination of
+///   combining the two other controls.
+///
+/// Author        : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+//  SoundTouch audio processing library
+//  Copyright (c) Olli Parviainen
+//
+//  This library is free software; you can redistribute it and/or
+//  modify it under the terms of the GNU Lesser General Public
+//  License as published by the Free Software Foundation; either
+//  version 2.1 of the License, or (at your option) any later version.
+//
+//  This library is distributed in the hope that it will be useful,
+//  but WITHOUT ANY WARRANTY; without even the implied warranty of
+//  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+//  Lesser General Public License for more details.
+//
+//  You should have received a copy of the GNU Lesser General Public
+//  License along with this library; if not, write to the Free Software
+//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#ifndef SoundTouch_H
+#define SoundTouch_H
+
+#include "FIFOSamplePipe.h"
+#include "STTypes.h"
+
+namespace soundtouch
+{
+
+/// Soundtouch library version string
+#define SOUNDTOUCH_VERSION          "2.3.3"
+
+/// SoundTouch library version id
+#define SOUNDTOUCH_VERSION_ID       (20303)
+
+//
+// Available setting IDs for the 'setSetting' & 'get_setting' functions:
+
+/// Enable/disable anti-alias filter in pitch transposer (0 = disable)
+#define SETTING_USE_AA_FILTER       0
+
+/// Pitch transposer anti-alias filter length (8 .. 128 taps, default = 32)
+#define SETTING_AA_FILTER_LENGTH    1
+
+/// Enable/disable quick seeking algorithm in tempo changer routine
+/// (enabling quick seeking lowers CPU utilization but causes a minor sound
+///  quality compromising)
+#define SETTING_USE_QUICKSEEK       2
+
+/// Time-stretch algorithm single processing sequence length in milliseconds. This determines
+/// to how long sequences the original sound is chopped in the time-stretch algorithm.
+/// See "STTypes.h" or README for more information.
+#define SETTING_SEQUENCE_MS         3
+
+/// Time-stretch algorithm seeking window length in milliseconds for algorithm that finds the
+/// best possible overlapping location. This determines from how wide window the algorithm
+/// may look for an optimal joining location when mixing the sound sequences back together.
+/// See "STTypes.h" or README for more information.
+#define SETTING_SEEKWINDOW_MS       4
+
+/// Time-stretch algorithm overlap length in milliseconds. When the chopped sound sequences
+/// are mixed back together, to form a continuous sound stream, this parameter defines over
+/// how long period the two consecutive sequences are let to overlap each other.
+/// See "STTypes.h" or README for more information.
+#define SETTING_OVERLAP_MS          5
+
+
+/// Call "getSetting" with this ID to query processing sequence size in samples.
+/// This value gives approximate value of how many input samples you'll need to
+/// feed into SoundTouch after initial buffering to get out a new batch of
+/// output samples.
+///
+/// This value does not include initial buffering at beginning of a new processing
+/// stream, use SETTING_INITIAL_LATENCY to get the initial buffering size.
+///
+/// Notices:
+/// - This is read-only parameter, i.e. setSetting ignores this parameter
+/// - This parameter value is not constant but change depending on
+///   tempo/pitch/rate/samplerate settings.
+#define SETTING_NOMINAL_INPUT_SEQUENCE      6
+
+
+/// Call "getSetting" with this ID to query nominal average processing output
+/// size in samples. This value tells approcimate value how many output samples
+/// SoundTouch outputs once it does DSP processing run for a batch of input samples.
+///
+/// Notices:
+/// - This is read-only parameter, i.e. setSetting ignores this parameter
+/// - This parameter value is not constant but 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:
+    /// Rate transposer class instance
+    class RateTransposer *pRateTransposer;
+
+    /// Time-stretch class instance
+    class TDStretch *pTDStretch;
+
+    /// Virtual pitch parameter. Effective rate & tempo are calculated from these parameters.
+    double virtualRate;
+
+    /// Virtual pitch parameter. Effective rate & tempo are calculated from these parameters.
+    double virtualTempo;
+
+    /// Virtual pitch parameter. Effective rate & tempo are calculated from these parameters.
+    double virtualPitch;
+
+    /// Flag: Has sample rate been set?
+    bool  bSrateSet;
+
+    /// Accumulator for how many samples in total will be expected as output vs. samples put in,
+    /// considering current processing settings.
+    double samplesExpectedOut;
+
+    /// Accumulator for how many samples in total have been read out from the processing so far
+    long   samplesOutput;
+
+    /// Calculates effective rate & tempo valuescfrom 'virtualRate', 'virtualTempo' and
+    /// 'virtualPitch' parameters.
+    void calcEffectiveRateAndTempo();
+
+protected :
+    /// Number of channels
+    uint  channels;
+
+    /// Effective 'rate' value calculated from 'virtualRate', 'virtualTempo' and 'virtualPitch'
+    double rate;
+
+    /// Effective 'tempo' value calculated from 'virtualRate', 'virtualTempo' and 'virtualPitch'
+    double tempo;
+
+public:
+    SoundTouch();
+    virtual ~SoundTouch() override;
+
+    /// Get SoundTouch library version string
+    static const char *getVersionString();
+
+    /// Get SoundTouch library version Id
+    static uint getVersionId();
+
+    /// Sets new rate control value. Normal rate = 1.0, smaller values
+    /// represent slower rate, larger faster rates.
+    void setRate(double newRate);
+
+    /// Sets new tempo control value. Normal tempo = 1.0, smaller values
+    /// represent slower tempo, larger faster tempo.
+    void setTempo(double newTempo);
+
+    /// Sets new rate control value as a difference in percents compared
+    /// to the original rate (-50 .. +100 %)
+    void setRateChange(double newRate);
+
+    /// Sets new tempo control value as a difference in percents compared
+    /// to the original tempo (-50 .. +100 %)
+    void setTempoChange(double newTempo);
+
+    /// Sets new pitch control value. Original pitch = 1.0, smaller values
+    /// represent lower pitches, larger values higher pitch.
+    void setPitch(double newPitch);
+
+    /// Sets pitch change in octaves compared to the original pitch
+    /// (-1.00 .. +1.00)
+    void setPitchOctaves(double newPitch);
+
+    /// Sets pitch change in semi-tones compared to the original pitch
+    /// (-12 .. +12)
+    void setPitchSemiTones(int newPitch);
+    void setPitchSemiTones(double newPitch);
+
+    /// Sets the number of channels, 1 = mono, 2 = stereo
+    void setChannels(uint numChannels);
+
+    /// 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.
+    //
+    /// Note: This function is meant for extracting the last samples of a sound
+    /// stream. This function may introduce additional blank samples in the end
+    /// of the sound stream, and thus it's not recommended to call this function
+    /// in the middle of a sound stream.
+    void flush();
+
+    /// Adds 'numSamples' pcs of samples from the 'samples' memory position into
+    /// the input of the object. Notice that sample rate _has_to_ be set before
+    /// calling this function, otherwise throws a runtime_error exception.
+    virtual void putSamples(
+            const SAMPLETYPE *samples,  ///< Pointer to sample buffer.
+            uint numSamples                         ///< Number of samples in buffer. Notice
+                                                    ///< that in case of stereo-sound a single sample
+                                                    ///< contains data for both channels.
+            ) 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() override;
+
+    /// Changes a setting controlling the processing system behaviour. See the
+    /// 'SETTING_...' defines for available setting ID's.
+    ///
+    /// \return 'true' if the setting was successfully changed
+    bool setSetting(int settingId,   ///< Setting ID number. see SETTING_... defines.
+                    int value        ///< New setting value.
+                    );
+
+    /// Reads a setting controlling the processing system behaviour. See the
+    /// 'SETTING_...' defines for available setting ID's.
+    ///
+    /// \return the setting value.
+    int getSetting(int settingId    ///< Setting ID number, see SETTING_... defines.
+                   ) const;
+
+    /// Returns number of samples currently unprocessed.
+    virtual uint numUnprocessedSamples() const;
+
+    /// Return number of channels
+    uint numChannels() const
+    {
+        return channels;
+    }
+
+    /// Other handy functions that are implemented in the ancestor classes (see
+    /// classes 'FIFOProcessor' and 'FIFOSamplePipe')
+    ///
+    /// - receiveSamples() : Use this function to receive 'ready' processed samples from SoundTouch.
+    /// - numSamples()     : Get number of 'ready' samples that can be received with
+    ///                      function 'receiveSamples()'
+    /// - isEmpty()        : Returns nonzero if there aren't any 'ready' samples.
+    /// - clear()          : Clears all samples from ready/processing buffers.
+};
+
+}
+#endif

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
-md lib
-devenv source\SoundTouch\SoundTouch.vcproj /upgrade
-devenv source\SoundTouch\SoundTouch.vcproj /build debug
-devenv source\SoundTouch\SoundTouch.vcproj /build release
+@rem devenv source\SoundStretch\SoundStretch.sln /upgrade
+devenv source\SoundStretch\SoundStretch.sln /build "Debug|Win32"
+devenv source\SoundStretch\SoundStretch.sln /build "Release|Win32"
+devenv source\SoundStretch\SoundStretch.sln /build "Debug|x64"
+devenv source\SoundStretch\SoundStretch.sln /build "Release|x64"
 
-devenv source\SoundStretch\SoundStretch.vcproj /upgrade
-devenv source\SoundStretch\SoundStretch.vcproj /build debug
-devenv source\SoundStretch\SoundStretch.vcproj /build release
-
-devenv source\SoundTouchDll\SoundTouchDll.vcproj /upgrade
-devenv source\SoundTouchDll\SoundTouchDll.vcproj /build debug
-devenv source\SoundTouchDll\SoundTouchDll.vcproj /build release
+@rem devenv source\SoundTouchDll\SoundTouchDll.sln /upgrade
+devenv source\SoundTouchDll\SoundTouchDll.sln /build "Debug|Win32"
+devenv source\SoundTouchDll\SoundTouchDll.sln /build "Release|Win32"
+devenv source\SoundTouchDll\SoundTouchDll.sln /build "Debug|x64"
+devenv source\SoundTouchDll\SoundTouchDll.sln /build "Release|x64"
 
 @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 ** directory, e.g. "\Program Files\Microsoft Visual Studio\VC98\Bin",
+echo ** Open "tools"->"Developer Command Line" from Visual Studio IDE, or
+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.
@@ -11,7 +8,7 @@
 # It also defines some flags to the configure script for specifying
 # the location to search for libSoundTouch
 #
-# A user of libSoundTouch should add @SOUNDTOUCH_LIBS@ and 
+# A user of libSoundTouch should add @SOUNDTOUCH_LIBS@ and
 # @SOUNDTOUCH_CXXFLAGS@ to the appropriate variables in his
 # Makefile.am files
 #
@@ -35,10 +32,10 @@ AC_DEFUN([AM_PATH_SOUNDTOUCH],[
 	then
 		saved_CPPFLAGS="$CPPFLAGS"
 		saved_LDFLAGS="$LDFLAGS"
-	   
+
 		CPPFLAGS="$CPPFLAGS -I$soundtouch_prefix/include"
 		LDFLAGS="$LDFLAGS -L$soundtouch_prefix/lib"
-   
+
 		dnl make sure SoundTouch.h header file exists
 			dnl could use AC_CHECK_HEADERS to check for all of them, but the supporting .h file names may change later
 		AC_CHECK_HEADER([soundtouch/SoundTouch.h],[
@@ -52,7 +49,7 @@ AC_DEFUN([AM_PATH_SOUNDTOUCH],[
 
 					dnl run action-if-found
 					ifelse([$2], , :, [$2])
-				],[ 
+				],[
 					dnl run action-if-not-found
 					ifelse([$3], , :, [$3])
 				])

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

@@ -1,28 +1,25 @@
-## 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
-## terms of the GNU General Public License as published by the Free Software
-## Foundation; either version 2 of the License, or (at your option) any later
-## version.
-## 
-## SoundTouch is distributed in the hope that it will be useful, but WITHOUT ANY
-## WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
-## A PARTICULAR PURPOSE.  See the GNU General Public License for more details.
-## 
-## You should have received a copy of the GNU General Public License along with
-## this program; if not, write to the Free Software Foundation, Inc., 59 Temple
-## Place - Suite 330, Boston, MA  02111-1307, USA
-
-include $(top_srcdir)/config/am_include.mk
-
-SUBDIRS=SoundTouch SoundStretch
-
-# set to something if you want other stuff to be included in the distribution tarball
-#EXTRA_DIST=
-
+## Process this file with automake to create Makefile.in
+##
+## This file is part of SoundTouch, an audio processing library for pitch/time adjustments
+##
+## SoundTouch is free software; you can redistribute it and/or modify it under the
+## terms of the GNU General Public License as published by the Free Software
+## Foundation; either version 2 of the License, or (at your option) any later
+## version.
+##
+## SoundTouch is distributed in the hope that it will be useful, but WITHOUT ANY
+## WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
+## A PARTICULAR PURPOSE.  See the GNU General Public License for more details.
+##
+## You should have received a copy of the GNU General Public License along with
+## this program; if not, write to the Free Software Foundation, Inc., 59 Temple
+## Place - Suite 330, Boston, MA  02111-1307, USA
+
+include $(top_srcdir)/config/am_include.mk
+
+if SOUNDTOUCH_FLOAT_SAMPLES
+    # build SoundTouchDLL only if float samples used
+    SUBDIRS=SoundTouch SoundStretch SoundTouchDLL
+else
+    SUBDIRS=SoundTouch SoundStretch
+endif

source/SoundStretch/Makefile.am

@@ -1,54 +1,50 @@
-## 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
-## terms of the GNU General Public License as published by the Free Software
-## Foundation; either version 2 of the License, or (at your option) any later
-## version.
-## 
-## SoundTouch is distributed in the hope that it will be useful, but WITHOUT ANY
-## WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
-## A PARTICULAR PURPOSE.  See the GNU General Public License for more details.
-## 
-## You should have received a copy of the GNU General Public License along with
-## this program; if not, write to the Free Software Foundation, Inc., 59 Temple
-## Place - Suite 330, Boston, MA  02111-1307, USA
-
-include $(top_srcdir)/config/am_include.mk
-
-
-## bin_PROGRAMS is the macro that tells automake the name of the programs to
-## install in the bin directory (/usr/local/bin) by default. By setting
-## --prefix= at configure time the user can change this (eg: ./configure
-## --prefix=/usr will install soundstretch under /usr/bin/soundstretch )
-bin_PROGRAMS=soundstretch
-
-noinst_HEADERS=RunParameters.h WavFile.h
-
-# extra files to include in distrubution tarball
-EXTRA_DIST=soundstretch.dsp soundstretch.dsw soundstretch.sln soundstretch.vcproj
-
-## for every name listed under bin_PROGRAMS, you have a <prog>_SOURCES. This lists
-## all the sources in the current directory that are used to build soundstretch.
-soundstretch_SOURCES=main.cpp RunParameters.cpp WavFile.cpp 
-
-## soundstretch_LDADD is a list of extras to pass at link time. All the objects
-## created by the above soundstretch_SOURCES are automatically linked in, so here I
-## list object files from other directories as well as flags passed to the
-## linker. 
-soundstretch_LDADD=../SoundTouch/libSoundTouch.la -lm
-
-## linker flags. 
-# OP 2011-7-17 Linker flags disabled to prevent stripping symbols by default
-# soundstretch_LDFLAGS=-s
-
-## additional compiler flags
-soundstretch_CXXFLAGS=-O3
-
-#clean-local: 
-#	-rm -f additional-files-to-remove-on-make-clean
+## Process this file with automake to create Makefile.in
+##
+## This file is part of SoundTouch, an audio processing library for pitch/time adjustments
+##
+## SoundTouch is free software; you can redistribute it and/or modify it under the
+## terms of the GNU General Public License as published by the Free Software
+## Foundation; either version 2 of the License, or (at your option) any later
+## version.
+##
+## SoundTouch is distributed in the hope that it will be useful, but WITHOUT ANY
+## WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
+## A PARTICULAR PURPOSE.  See the GNU General Public License for more details.
+##
+## You should have received a copy of the GNU General Public License along with
+## this program; if not, write to the Free Software Foundation, Inc., 59 Temple
+## Place - Suite 330, Boston, MA  02111-1307, USA
+
+include $(top_srcdir)/config/am_include.mk
+
+
+## bin_PROGRAMS is the macro that tells automake the name of the programs to
+## install in the bin directory (/usr/local/bin) by default. By setting
+## --prefix= at configure time the user can change this (eg: ./configure
+## --prefix=/usr will install soundstretch under /usr/bin/soundstretch )
+bin_PROGRAMS=soundstretch
+
+noinst_HEADERS=RunParameters.h WavFile.h
+
+# extra files to include in distribution tarball
+EXTRA_DIST=soundstretch.sln soundstretch.vcxproj
+
+## for every name listed under bin_PROGRAMS, you have a <prog>_SOURCES. This lists
+## all the sources in the current directory that are used to build soundstretch.
+soundstretch_SOURCES=main.cpp RunParameters.cpp WavFile.cpp
+
+## soundstretch_LDADD is a list of extras to pass at link time. All the objects
+## created by the above soundstretch_SOURCES are automatically linked in, so here I
+## list object files from other directories as well as flags passed to the
+## linker.
+soundstretch_LDADD=../SoundTouch/libSoundTouch.la -lm
+
+## linker flags.
+# Linker flag -s disabled to prevent stripping symbols by default
+#soundstretch_LDFLAGS=-s
+
+## additional compiler flags
+soundstretch_CXXFLAGS=$(AM_CXXFLAGS)
+
+#clean-local:
+#	-rm -f additional-files-to-remove-on-make-clean

source/SoundStretch/RunParameters.cpp

@@ -1,301 +1,292 @@
-////////////////////////////////////////////////////////////////////////////////
-///
-/// A class for parsing the 'soundstretch' application command line parameters
-///
-/// Author        : Copyright (c) Olli Parviainen
-/// Author e-mail : oparviai 'at' iki.fi
-/// SoundTouch WWW: http://www.surina.net/soundtouch
-///
-////////////////////////////////////////////////////////////////////////////////
-//
-// Last changed  : $Date$
-// File revision : $Revision: 4 $
-//
-// $Id$
-//
-////////////////////////////////////////////////////////////////////////////////
-//
-// License :
-//
-//  SoundTouch audio processing library
-//  Copyright (c) Olli Parviainen
-//
-//  This library is free software; you can redistribute it and/or
-//  modify it under the terms of the GNU Lesser General Public
-//  License as published by the Free Software Foundation; either
-//  version 2.1 of the License, or (at your option) any later version.
-//
-//  This library is distributed in the hope that it will be useful,
-//  but WITHOUT ANY WARRANTY; without even the implied warranty of
-//  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-//  Lesser General Public License for more details.
-//
-//  You should have received a copy of the GNU Lesser General Public
-//  License along with this library; if not, write to the Free Software
-//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
-//
-////////////////////////////////////////////////////////////////////////////////
-
-#include <string>
-#include <stdlib.h>
-
-#include "RunParameters.h"
-
-using namespace std;
-
-// Program usage instructions 
-
-static const char licenseText[] = 
-    "    LICENSE:\n"
-    "    ========\n"
-    "    \n"
-    "    SoundTouch sound processing library\n"
-    "    Copyright (c) Olli Parviainen\n"
-    "    \n"
-    "    This library is free software; you can redistribute it and/or\n"
-    "    modify it under the terms of the GNU Lesser General Public\n"
-    "    License version 2.1 as published by the Free Software Foundation.\n"
-    "    \n"
-    "    This library is distributed in the hope that it will be useful,\n"
-    "    but WITHOUT ANY WARRANTY; without even the implied warranty of\n"
-    "    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU\n"
-    "    Lesser General Public License for more details.\n"
-    "    \n"
-    "    You should have received a copy of the GNU Lesser General Public\n"
-    "    License along with this library; if not, write to the Free Software\n"
-    "    Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA\n"
-    "    \n"
-    "This application is distributed with full source codes; however, if you\n"
-    "didn't receive them, please visit the author's homepage (see the link above).";
-
-static const char whatText[] = 
-    "This application processes WAV audio files by modifying the sound tempo,\n"
-    "pitch and playback rate properties independently from each other.\n"
-    "\n";
-
-static const char usage[] = 
-    "Usage :\n"
-    "    soundstretch infilename outfilename [switches]\n"
-    "\n"
-    "To use standard input/output pipes, give 'stdin' and 'stdout' as filenames.\n"
-    "\n"
-    "Available switches are:\n"
-    "  -tempo=n : Change sound tempo by n percents  (n=-95..+5000 %)\n"
-    "  -pitch=n : Change sound pitch by n semitones (n=-60..+60 semitones)\n"
-    "  -rate=n  : Change sound rate by n percents   (n=-95..+5000 %)\n"
-    "  -bpm=n   : Detect the BPM rate of sound and adjust tempo to meet 'n' BPMs.\n"
-    "             If '=n' is omitted, just detects the BPM rate.\n"
-    "  -quick   : Use quicker tempo change algorithm (gain speed, lose quality)\n"
-    "  -naa     : Don't use anti-alias filtering (gain speed, lose quality)\n"
-    "  -speech  : Tune algorithm for speech processing (default is for music)\n"
-    "  -license : Display the program license text (LGPL)\n";
-
-
-// Converts a char into lower case
-static int _toLowerCase(int c)
-{
-    if (c >= 'A' && c <= 'Z') 
-    {
-        c += 'a' - 'A';
-    }
-    return c;
-}
-
-
-// Constructor
-RunParameters::RunParameters(const int nParams, const char * const paramStr[])
-{
-    int i;
-    int nFirstParam;
-
-    if (nParams < 3) 
-    {
-        // Too few parameters
-        if (nParams > 1 && paramStr[1][0] == '-' && 
-            _toLowerCase(paramStr[1][1]) == 'l') 
-        {
-            // '-license' switch
-            throwLicense();
-        }
-        string msg = whatText;
-        msg += usage;
-        ST_THROW_RT_ERROR(msg.c_str());
-    }
-
-    inFileName = NULL;
-    outFileName = NULL;
-    tempoDelta = 0;
-    pitchDelta = 0;
-    rateDelta = 0;
-    quick = 0;
-    noAntiAlias = 0;
-    goalBPM = 0;
-    speech = FALSE;
-    detectBPM = FALSE;
-
-    // Get input & output file names
-    inFileName = (char*)paramStr[1];
-    outFileName = (char*)paramStr[2];
-
-    if (outFileName[0] == '-')
-    {
-        // no outputfile name was given but parameters
-        outFileName = NULL;
-        nFirstParam = 2;
-    }
-    else
-    {
-        nFirstParam = 3;
-    }
-
-    // parse switch parameters
-    for (i = nFirstParam; i < nParams; i ++) 
-    {
-        parseSwitchParam(paramStr[i]);
-    }
-
-    checkLimits();
-}
-
-
-
-// Checks parameter limits
-void RunParameters::checkLimits()
-{
-    if (tempoDelta < -95.0f) 
-    {
-        tempoDelta = -95.0f;
-    } 
-    else if (tempoDelta > 5000.0f) 
-    {
-        tempoDelta = 5000.0f;
-    }
-
-    if (pitchDelta < -60.0f) 
-    {
-        pitchDelta = -60.0f;
-    } 
-    else if (pitchDelta > 60.0f) 
-    {
-        pitchDelta = 60.0f;
-    }
-
-    if (rateDelta < -95.0f) 
-    {
-        rateDelta = -95.0f;
-    } 
-    else if (rateDelta > 5000.0f) 
-    {
-        rateDelta = 5000.0f;
-    }
-}
-
-
-
-// Unknown switch parameter -- throws an exception with an error message
-void RunParameters::throwIllegalParamExp(const string &str) const
-{
-    string msg = "ERROR : Illegal parameter \"";
-    msg += str;
-    msg += "\".\n\n";
-    msg += usage;
-    ST_THROW_RT_ERROR(msg.c_str());
-}
-
-
-
-void RunParameters::throwLicense() const
-{
-    ST_THROW_RT_ERROR(licenseText);
-}
-
-
-float RunParameters::parseSwitchValue(const string &str) const
-{
-    int pos;
-
-    pos = (int)str.find_first_of('=');
-    if (pos < 0) 
-    {
-        // '=' missing
-        throwIllegalParamExp(str);
-    }
-
-    // Read numerical parameter value after '='
-    return (float)atof(str.substr(pos + 1).c_str());
-}
-
-
-// Interprets a single switch parameter string of format "-switch=xx"
-// Valid switches are "-tempo=xx", "-pitch=xx" and "-rate=xx". Stores
-// switch values into 'params' structure.
-void RunParameters::parseSwitchParam(const string &str)
-{
-    int upS;
-
-    if (str[0] != '-') 
-    {
-        // leading hyphen missing => not a valid parameter
-        throwIllegalParamExp(str);
-    }
-
-    // Take the first character of switch name & change to lower case
-    upS = _toLowerCase(str[1]);
-
-    // interpret the switch name & operate accordingly
-    switch (upS) 
-    {
-        case 't' :
-            // switch '-tempo=xx'
-            tempoDelta = parseSwitchValue(str);
-            break;
-
-        case 'p' :
-            // switch '-pitch=xx'
-            pitchDelta = parseSwitchValue(str);
-            break;
-
-        case 'r' :
-            // switch '-rate=xx'
-            rateDelta = parseSwitchValue(str);
-            break;
-
-        case 'b' :
-            // switch '-bpm=xx'
-            detectBPM = TRUE;
-            try
-            {
-                goalBPM = parseSwitchValue(str);
-            } 
-            catch (const runtime_error)
-            {
-                // illegal or missing bpm value => just calculate bpm
-                goalBPM = 0;
-            }
-            break;
-
-        case 'q' :
-            // switch '-quick'
-            quick = 1;
-            break;
-
-        case 'n' :
-            // switch '-naa'
-            noAntiAlias = 1;
-            break;
-
-        case 'l' :
-            // switch '-license'
-            throwLicense();
-            break;
-
-        case 's' :
-            // switch '-speech'
-            speech = TRUE;
-            break;
-
-        default:
-            // unknown switch
-            throwIllegalParamExp(str);
-    }
-}
+////////////////////////////////////////////////////////////////////////////////
+///
+/// A class for parsing the 'soundstretch' application command line parameters
+///
+/// Author        : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+//  SoundTouch audio processing library
+//  Copyright (c) Olli Parviainen
+//
+//  This library is free software; you can redistribute it and/or
+//  modify it under the terms of the GNU Lesser General Public
+//  License as published by the Free Software Foundation; either
+//  version 2.1 of the License, or (at your option) any later version.
+//
+//  This library is distributed in the hope that it will be useful,
+//  but WITHOUT ANY WARRANTY; without even the implied warranty of
+//  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+//  Lesser General Public License for more details.
+//
+//  You should have received a copy of the GNU Lesser General Public
+//  License along with this library; if not, write to the Free Software
+//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#include <string>
+#include <cstdlib>
+
+#include "RunParameters.h"
+
+using namespace std;
+
+namespace soundstretch
+{
+
+// Program usage instructions
+
+static const char licenseText[] =
+    "    LICENSE:\n"
+    "    ========\n"
+    "    \n"
+    "    SoundTouch sound processing library\n"
+    "    Copyright (c) Olli Parviainen\n"
+    "    \n"
+    "    This library is free software; you can redistribute it and/or\n"
+    "    modify it under the terms of the GNU Lesser General Public\n"
+    "    License version 2.1 as published by the Free Software Foundation.\n"
+    "    \n"
+    "    This library is distributed in the hope that it will be useful,\n"
+    "    but WITHOUT ANY WARRANTY; without even the implied warranty of\n"
+    "    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU\n"
+    "    Lesser General Public License for more details.\n"
+    "    \n"
+    "    You should have received a copy of the GNU Lesser General Public\n"
+    "    License along with this library; if not, write to the Free Software\n"
+    "    Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA\n"
+    "    \n"
+    "This application is distributed with full source codes; however, if you\n"
+    "didn't receive them, please visit the author's homepage (see the link above).";
+
+static const char whatText[] =
+    "This application processes WAV audio files by modifying the sound tempo,\n"
+    "pitch and playback rate properties independently from each other.\n"
+    "\n";
+
+static const char usage[] =
+    "Usage :\n"
+    "    soundstretch infilename outfilename [switches]\n"
+    "\n"
+    "To use standard input/output pipes, give 'stdin' and 'stdout' as filenames.\n"
+    "\n"
+    "Available switches are:\n"
+    "  -tempo=n : Change sound tempo by n percents  (n=-95..+5000 %)\n"
+    "  -pitch=n : Change sound pitch by n semitones (n=-60..+60 semitones)\n"
+    "  -rate=n  : Change sound rate by n percents   (n=-95..+5000 %)\n"
+    "  -bpm=n   : Detect the BPM rate of sound and adjust tempo to meet 'n' BPMs.\n"
+    "             If '=n' is omitted, just detects the BPM rate.\n"
+    "  -quick   : Use quicker tempo change algorithm (gain speed, lose quality)\n"
+    "  -naa     : Don't use anti-alias filtering (gain speed, lose quality)\n"
+    "  -speech  : Tune algorithm for speech processing (default is for music)\n"
+    "  -license : Display the program license text (LGPL)\n";
+
+
+// Converts a char into lower case
+static int _toLowerCase(int c)
+{
+    if (c >= 'A' && c <= 'Z')
+    {
+        c += 'a' - 'A';
+    }
+    return c;
+}
+
+// Constructor
+RunParameters::RunParameters(int nParams, const CHARTYPE* paramStr[])
+{
+    int i;
+    int nFirstParam;
+
+    if (nParams < 3)
+    {
+        // Too few parameters
+        if (nParams > 1 && paramStr[1][0] == '-' &&
+            _toLowerCase(paramStr[1][1]) == 'l')
+        {
+            // '-license' switch
+            throwLicense();
+        }
+        string msg = whatText;
+        msg += usage;
+        throw(msg);
+    }
+
+    // Get input & output file names
+    inFileName = paramStr[1];
+    outFileName = paramStr[2];
+
+    if (outFileName[0] == '-')
+    {
+        // outputfile name was omitted but other parameter switches given instead
+        outFileName = STRING_CONST("");
+        nFirstParam = 2;
+    }
+    else
+    {
+        nFirstParam = 3;
+    }
+
+    // parse switch parameters
+    for (i = nFirstParam; i < nParams; i ++)
+    {
+        parseSwitchParam(paramStr[i]);
+    }
+
+    checkLimits();
+}
+
+
+// Checks parameter limits
+void RunParameters::checkLimits()
+{
+    if (tempoDelta < -95.0f)
+    {
+        tempoDelta = -95.0f;
+    }
+    else if (tempoDelta > 5000.0f)
+    {
+        tempoDelta = 5000.0f;
+    }
+
+    if (pitchDelta < -60.0f)
+    {
+        pitchDelta = -60.0f;
+    }
+    else if (pitchDelta > 60.0f)
+    {
+        pitchDelta = 60.0f;
+    }
+
+    if (rateDelta < -95.0f)
+    {
+        rateDelta = -95.0f;
+    }
+    else if (rateDelta > 5000.0f)
+    {
+        rateDelta = 5000.0f;
+    }
+}
+
+// Convert STRING to std::string. Actually needed only if STRING is std::wstring, but conversion penalty is negligible
+std::string convertString(const STRING& str)
+{
+    std::string res;
+    for (auto c : str)
+    {
+        res += (char)c;
+    }
+    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
+{
+    int pos;
+
+    pos = (int)str.find_first_of('=');
+    if (pos < 0)
+    {
+        // '=' missing
+        throwIllegalParamExp(str);
+    }
+
+    // Read numerical parameter value after '='
+    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)
+{
+    int upS;
+
+    if (str[0] != '-')
+    {
+        // leading hyphen missing => not a valid parameter
+        throwIllegalParamExp(str);
+    }
+
+    // Take the first character of switch name & change to lower case
+    upS = _toLowerCase(str[1]);
+
+    // interpret the switch name & operate accordingly
+    switch (upS)
+    {
+        case 't' :
+            // switch '-tempo=xx'
+            tempoDelta = parseSwitchValue(str);
+            break;
+
+        case 'p' :
+            // switch '-pitch=xx'
+            pitchDelta = parseSwitchValue(str);
+            break;
+
+        case 'r' :
+            // switch '-rate=xx'
+            rateDelta = parseSwitchValue(str);
+            break;
+
+        case 'b' :
+            // switch '-bpm=xx'
+            detectBPM = true;
+            try
+            {
+                goalBPM = parseSwitchValue(str);
+            }
+            catch (const runtime_error &)
+            {
+                // illegal or missing bpm value => just calculate bpm
+                goalBPM = 0;
+            }
+            break;
+
+        case 'q' :
+            // switch '-quick'
+            quick = 1;
+            break;
+
+        case 'n' :
+            // switch '-naa'
+            noAntiAlias = 1;
+            break;
+
+        case 'l' :
+            // switch '-license'
+            throwLicense();
+            break;
+
+        case 's' :
+            // switch '-speech'
+            speech = true;
+            break;
+
+        default:
+            // unknown switch
+            throwIllegalParamExp(str);
+    }
+}
+
+}

source/SoundStretch/RunParameters.h

@@ -1,72 +1,70 @@
-////////////////////////////////////////////////////////////////////////////////
-///
-/// A class for parsing the 'soundstretch' application command line parameters
-///
-/// Author        : Copyright (c) Olli Parviainen
-/// Author e-mail : oparviai 'at' iki.fi
-/// SoundTouch WWW: http://www.surina.net/soundtouch
-///
-////////////////////////////////////////////////////////////////////////////////
-//
-// Last changed  : $Date$
-// File revision : $Revision: 4 $
-//
-// $Id$
-//
-////////////////////////////////////////////////////////////////////////////////
-//
-// License :
-//
-//  SoundTouch audio processing library
-//  Copyright (c) Olli Parviainen
-//
-//  This library is free software; you can redistribute it and/or
-//  modify it under the terms of the GNU Lesser General Public
-//  License as published by the Free Software Foundation; either
-//  version 2.1 of the License, or (at your option) any later version.
-//
-//  This library is distributed in the hope that it will be useful,
-//  but WITHOUT ANY WARRANTY; without even the implied warranty of
-//  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-//  Lesser General Public License for more details.
-//
-//  You should have received a copy of the GNU Lesser General Public
-//  License along with this library; if not, write to the Free Software
-//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
-//
-////////////////////////////////////////////////////////////////////////////////
-
-#ifndef RUNPARAMETERS_H
-#define RUNPARAMETERS_H
-
-#include "STTypes.h"
-#include <string>
-
-using namespace std;
-
-/// Parses command line parameters into program parameters
-class RunParameters
-{
-private:
-    void throwIllegalParamExp(const string &str) const;
-    void throwLicense() const;
-    void parseSwitchParam(const string &str);
-    void checkLimits();
-    float parseSwitchValue(const string &str) const;
-
-public:
-    char  *inFileName;
-    char  *outFileName;
-    float tempoDelta;
-    float pitchDelta;
-    float rateDelta;
-    int   quick;
-    int   noAntiAlias;
-    float goalBPM;
-    BOOL  detectBPM;
-    BOOL  speech;
-
-    RunParameters(const int nParams, const char * const paramStr[]);
-};
-
-#endif
+////////////////////////////////////////////////////////////////////////////////
+///
+/// A class for parsing the 'soundstretch' application command line parameters
+///
+/// Author        : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+//  SoundTouch audio processing library
+//  Copyright (c) Olli Parviainen
+//
+//  This library is free software; you can redistribute it and/or
+//  modify it under the terms of the GNU Lesser General Public
+//  License as published by the Free Software Foundation; either
+//  version 2.1 of the License, or (at your option) any later version.
+//
+//  This library is distributed in the hope that it will be useful,
+//  but WITHOUT ANY WARRANTY; without even the implied warranty of
+//  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+//  Lesser General Public License for more details.
+//
+//  You should have received a copy of the GNU Lesser General Public
+//  License along with this library; if not, write to the Free Software
+//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#ifndef RUNPARAMETERS_H
+#define RUNPARAMETERS_H
+
+#include <string>
+#include "STTypes.h"
+#include "SS_CharTypes.h"
+#include "WavFile.h"
+
+namespace soundstretch
+{
+
+/// Parses command line parameters into program parameters
+class RunParameters
+{
+private:
+    void throwIllegalParamExp(const STRING& str) const;
+    void throwLicense() const;
+    void parseSwitchParam(const STRING& str);
+    void checkLimits();
+    double parseSwitchValue(const STRING& tr) const;
+
+public:
+    STRING inFileName;
+    STRING outFileName;
+    double tempoDelta{ 0 };
+    double pitchDelta{ 0 };
+    double rateDelta{ 0 };
+    int   quick{ 0 };
+    int   noAntiAlias{ 0 };
+    double goalBPM{ 0 };
+    bool  detectBPM{ false };
+    bool  speech{ false };
+
+    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.h

@@ -1,276 +1,281 @@
-////////////////////////////////////////////////////////////////////////////////
-///
-/// Classes for easy reading & writing of WAV sound files.
-///
-/// For big-endian CPU, define BIG_ENDIAN during compile-time to correctly
-/// parse the WAV files with such processors.
-/// 
-/// Admittingly, more complete WAV reader routines may exist in public domain, but 
-/// the reason for 'yet another' one is that those generic WAV reader libraries are
-/// exhaustingly large and cumbersome! Wanted to have something simpler here, i.e. 
-/// something that's not already larger than rest of the SoundTouch/SoundStretch program...
-///
-/// Author        : Copyright (c) Olli Parviainen
-/// Author e-mail : oparviai 'at' iki.fi
-/// SoundTouch WWW: http://www.surina.net/soundtouch
-///
-////////////////////////////////////////////////////////////////////////////////
-//
-// Last changed  : $Date$
-// File revision : $Revision: 4 $
-//
-// $Id$
-//
-////////////////////////////////////////////////////////////////////////////////
-//
-// License :
-//
-//  SoundTouch audio processing library
-//  Copyright (c) Olli Parviainen
-//
-//  This library is free software; you can redistribute it and/or
-//  modify it under the terms of the GNU Lesser General Public
-//  License as published by the Free Software Foundation; either
-//  version 2.1 of the License, or (at your option) any later version.
-//
-//  This library is distributed in the hope that it will be useful,
-//  but WITHOUT ANY WARRANTY; without even the implied warranty of
-//  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-//  Lesser General Public License for more details.
-//
-//  You should have received a copy of the GNU Lesser General Public
-//  License along with this library; if not, write to the Free Software
-//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
-//
-////////////////////////////////////////////////////////////////////////////////
-
-#ifndef WAVFILE_H
-#define WAVFILE_H
-
-#include <stdio.h>
-
-#ifndef uint
-typedef unsigned int uint;
-#endif           
-
-
-/// WAV audio file 'riff' section header
-typedef struct 
-{
-    char riff_char[4];
-    int  package_len;
-    char wave[4];
-} WavRiff;
-
-/// WAV audio file 'format' section header
-typedef struct 
-{
-    char  fmt[4];
-    int   format_len;
-    short fixed;
-    short channel_number;
-    int   sample_rate;
-    int   byte_rate;
-    short byte_per_sample;
-    short bits_per_sample;
-} WavFormat;
-
-/// WAV audio file 'data' section header
-typedef struct 
-{
-    char  data_field[4];
-    uint  data_len;
-} WavData;
-
-
-/// WAV audio file header
-typedef struct 
-{
-    WavRiff   riff;
-    WavFormat format;
-    WavData   data;
-} WavHeader;
-
-
-/// Base class for processing WAV audio files.
-class WavFileBase
-{
-private:
-    /// Conversion working buffer;
-    char *convBuff;
-    int convBuffSize;
-
-protected:
-    WavFileBase();
-    virtual ~WavFileBase();
-
-    /// Get pointer to conversion buffer of at min. given size
-    void *getConvBuffer(int sizeByte);
-};
-
-
-/// Class for reading WAV audio files.
-class WavInFile : protected WavFileBase
-{
-private:
-    /// File pointer.
-    FILE *fptr;
-
-    /// Position within the audio stream
-    long position;
-
-    /// Counter of how many bytes of sample data have been read from the file.
-    long dataRead;
-
-    /// WAV header information
-    WavHeader header;
-
-    /// Init the WAV file stream
-    void init();
-
-    /// Read WAV file headers.
-    /// \return zero if all ok, nonzero if file format is invalid.
-    int readWavHeaders();
-
-    /// Checks WAV file header tags.
-    /// \return zero if all ok, nonzero if file format is invalid.
-    int checkCharTags() const;
-
-    /// Reads a single WAV file header block.
-    /// \return zero if all ok, nonzero if file format is invalid.
-    int readHeaderBlock();
-
-    /// Reads WAV file 'riff' block
-    int readRIFFBlock();
-
-public:
-    /// Constructor: Opens the given WAV file. If the file can't be opened,
-    /// throws 'runtime_error' exception.
-    WavInFile(const char *filename);
-
-    WavInFile(FILE *file);
-
-    /// Destructor: Closes the file.
-    ~WavInFile();
-
-    /// Rewind to beginning of the file
-    void rewind();
-
-    /// Get sample rate.
-    uint getSampleRate() const;
-
-    /// Get number of bits per sample, i.e. 8 or 16.
-    uint getNumBits() const;
-
-    /// Get sample data size in bytes. Ahem, this should return same information as 
-    /// 'getBytesPerSample'...
-    uint getDataSizeInBytes() const;
-
-    /// Get total number of samples in file.
-    uint getNumSamples() const;
-
-    /// Get number of bytes per audio sample (e.g. 16bit stereo = 4 bytes/sample)
-    uint getBytesPerSample() const;
-    
-    /// Get number of audio channels in the file (1=mono, 2=stereo)
-    uint getNumChannels() const;
-
-    /// Get the audio file length in milliseconds
-    uint getLengthMS() const;
-
-    /// Returns how many milliseconds of audio have so far been read from the file
-    ///
-    /// \return elapsed duration in milliseconds
-    uint getElapsedMS() const;
-
-    /// Reads audio samples from the WAV file. This routine works only for 8 bit samples.
-    /// Reads given number of elements from the file or if end-of-file reached, as many 
-    /// elements as are left in the file.
-    ///
-    /// \return Number of 8-bit integers read from the file.
-    int read(unsigned char *buffer, int maxElems);
-
-    /// Reads audio samples from the WAV file to 16 bit integer format. Reads given number 
-    /// of elements from the file or if end-of-file reached, as many elements as are 
-    /// left in the file.
-    ///
-    /// \return Number of 16-bit integers read from the file.
-    int read(short *buffer,     ///< Pointer to buffer where to read data.
-             int maxElems       ///< Size of 'buffer' array (number of array elements).
-             );
-
-    /// Reads audio samples from the WAV file to floating point format, converting 
-    /// sample values to range [-1,1[. Reads given number of elements from the file
-    /// or if end-of-file reached, as many elements as are left in the file.
-    /// Notice that reading in float format supports 8/16/24/32bit sample formats.
-    ///
-    /// \return Number of elements read from the file.
-    int read(float *buffer,     ///< Pointer to buffer where to read data.
-             int maxElems       ///< Size of 'buffer' array (number of array elements).
-             );
-
-    /// Check end-of-file.
-    ///
-    /// \return Nonzero if end-of-file reached.
-    int eof() const;
-};
-
-
-
-/// Class for writing WAV audio files.
-class WavOutFile : protected WavFileBase
-{
-private:
-    /// Pointer to the WAV file
-    FILE *fptr;
-
-    /// WAV file header data.
-    WavHeader header;
-
-    /// Counter of how many bytes have been written to the file so far.
-    int bytesWritten;
-
-    /// Fills in WAV file header information.
-    void fillInHeader(const uint sampleRate, const uint bits, const uint channels);
-
-    /// Finishes the WAV file header by supplementing information of amount of
-    /// data written to file etc
-    void finishHeader();
-
-    /// Writes the WAV file header.
-    void writeHeader();
-
-public:
-    /// Constructor: Creates a new WAV file. Throws a 'runtime_error' exception 
-    /// if file creation fails.
-    WavOutFile(const char *fileName,    ///< Filename
-               int sampleRate,          ///< Sample rate (e.g. 44100 etc)
-               int bits,                ///< Bits per sample (8 or 16 bits)
-               int channels             ///< Number of channels (1=mono, 2=stereo)
-               );
-
-    WavOutFile(FILE *file, int sampleRate, int bits, int channels);
-
-    /// Destructor: Finalizes & closes the WAV file.
-    ~WavOutFile();
-
-    /// Write data to WAV file. This function works only with 8bit samples. 
-    /// Throws a 'runtime_error' exception if writing to file fails.
-    void write(const unsigned char *buffer, ///< Pointer to sample data buffer.
-               int numElems                 ///< How many array items are to be written to file.
-               );
-
-    /// Write data to WAV file. Throws a 'runtime_error' exception if writing to
-    /// file fails.
-    void write(const short *buffer,     ///< Pointer to sample data buffer.
-               int numElems             ///< How many array items are to be written to file.
-               );
-
-    /// Write data to WAV file in floating point format, saturating sample values to range
-    /// [-1..+1[. Throws a 'runtime_error' exception if writing to file fails.
-    void write(const float *buffer,     ///< Pointer to sample data buffer.
-               int numElems             ///< How many array items are to be written to file.
-               );
-};
-
-#endif
+////////////////////////////////////////////////////////////////////////////////
+///
+/// Classes for easy reading & writing of WAV sound files.
+///
+/// For big-endian CPU, define BIG_ENDIAN during compile-time to correctly
+/// parse the WAV files with such processors.
+///
+/// Admittingly, more complete WAV reader routines may exist in public domain, but
+/// the reason for 'yet another' one is that those generic WAV reader libraries are
+/// exhaustingly large and cumbersome! Wanted to have something simpler here, i.e.
+/// something that's not already larger than rest of the SoundTouch/SoundStretch program...
+///
+/// Author        : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+//  SoundTouch audio processing library
+//  Copyright (c) Olli Parviainen
+//
+//  This library is free software; you can redistribute it and/or
+//  modify it under the terms of the GNU Lesser General Public
+//  License as published by the Free Software Foundation; either
+//  version 2.1 of the License, or (at your option) any later version.
+//
+//  This library is distributed in the hope that it will be useful,
+//  but WITHOUT ANY WARRANTY; without even the implied warranty of
+//  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+//  Lesser General Public License for more details.
+//
+//  You should have received a copy of the GNU Lesser General Public
+//  License along with this library; if not, write to the Free Software
+//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#ifndef WAVFILE_H
+#define WAVFILE_H
+
+#include <cstdio>
+#include <string>
+#include "SS_CharTypes.h"
+
+namespace soundstretch
+{
+
+#ifndef uint
+typedef unsigned int uint;
+#endif
+
+
+/// WAV audio file 'riff' section header
+typedef struct
+{
+    char riff_char[4];
+    uint package_len;
+    char wave[4];
+} WavRiff;
+
+/// WAV audio file 'format' section header
+typedef struct
+{
+    char  fmt[4];
+    unsigned int   format_len;
+    unsigned short fixed;
+    unsigned short channel_number;
+    unsigned int   sample_rate;
+    unsigned int   byte_rate;
+    unsigned short byte_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
+{
+    char  data_field[4];
+    uint  data_len;
+} WavData;
+
+
+/// WAV audio file header
+typedef struct
+{
+    WavRiff   riff;
+    WavFormat format;
+    WavFact   fact;
+    WavData   data;
+} WavHeader;
+
+
+/// Base class for processing WAV audio files.
+class WavFileBase
+{
+private:
+    /// Conversion working buffer;
+    char *convBuff;
+    int convBuffSize;
+
+protected:
+    WavFileBase();
+    virtual ~WavFileBase();
+
+    /// Get pointer to conversion buffer of at min. given size
+    void *getConvBuffer(int sizeByte);
+};
+
+
+/// Class for reading WAV audio files.
+class WavInFile : protected WavFileBase
+{
+private:
+    /// File pointer.
+    FILE *fptr;
+
+    /// Counter of how many bytes of sample data have been read from the file.
+    long dataRead;
+
+    /// WAV header information
+    WavHeader header;
+
+    /// Init the WAV file stream
+    void init();
+
+    /// Read WAV file headers.
+    /// \return zero if all ok, nonzero if file format is invalid.
+    int readWavHeaders();
+
+    /// Checks WAV file header tags.
+    /// \return zero if all ok, nonzero if file format is invalid.
+    int checkCharTags() const;
+
+    /// Reads a single WAV file header block.
+    /// \return zero if all ok, nonzero if file format is invalid.
+    int readHeaderBlock();
+
+    /// Reads WAV file 'riff' block
+    int readRIFFBlock();
+
+public:
+    /// Constructor: Opens the given WAV file. If the file can't be opened,
+    /// throws 'runtime_error' exception.
+    WavInFile(const STRING& filename);
+
+    WavInFile(FILE *file);
+
+    /// Destructor: Closes the file.
+    ~WavInFile();
+
+    /// Rewind to beginning of the file
+    void rewind();
+
+    /// Get sample rate.
+    uint getSampleRate() const;
+
+    /// Get number of bits per sample, i.e. 8 or 16.
+    uint getNumBits() const;
+
+    /// Get sample data size in bytes. Ahem, this should return same information as
+    /// 'getBytesPerSample'...
+    uint getDataSizeInBytes() const;
+
+    /// Get total number of samples in file.
+    uint getNumSamples() const;
+
+    /// Get number of bytes per audio sample (e.g. 16bit stereo = 4 bytes/sample)
+    uint getBytesPerSample() const;
+
+    /// Get number of audio channels in the file (1=mono, 2=stereo)
+    uint getNumChannels() const;
+
+    /// Get the audio file length in milliseconds
+    uint getLengthMS() const;
+
+    /// Returns how many milliseconds of audio have so far been read from the file
+    ///
+    /// \return elapsed duration in milliseconds
+    uint getElapsedMS() const;
+
+    /// Reads audio samples from the WAV file. This routine works only for 8 bit samples.
+    /// Reads given number of elements from the file or if end-of-file reached, as many
+    /// elements as are left in the file.
+    ///
+    /// \return Number of 8-bit integers read from the file.
+    int read(unsigned char *buffer, int maxElems);
+
+    /// Reads audio samples from the WAV file to 16 bit integer format. Reads given number
+    /// of elements from the file or if end-of-file reached, as many elements as are
+    /// left in the file.
+    ///
+    /// \return Number of 16-bit integers read from the file.
+    int read(short *buffer,     ///< Pointer to buffer where to read data.
+             int maxElems       ///< Size of 'buffer' array (number of array elements).
+             );
+
+    /// Reads audio samples from the WAV file to floating point format, converting
+    /// sample values to range [-1,1[. Reads given number of elements from the file
+    /// or if end-of-file reached, as many elements as are left in the file.
+    /// Notice that reading in float format supports 8/16/24/32bit sample formats.
+    ///
+    /// \return Number of elements read from the file.
+    int read(float *buffer,     ///< Pointer to buffer where to read data.
+             int maxElems       ///< Size of 'buffer' array (number of array elements).
+             );
+
+    /// Check end-of-file.
+    ///
+    /// \return Nonzero if end-of-file reached.
+    int eof() const;
+};
+
+
+/// Class for writing WAV audio files.
+class WavOutFile : protected WavFileBase
+{
+private:
+    /// Pointer to the WAV file
+    FILE *fptr;
+
+    /// WAV file header data.
+    WavHeader header;
+
+    /// Counter of how many bytes have been written to the file so far.
+    int bytesWritten;
+
+    /// Fills in WAV file header information.
+    void fillInHeader(const uint sampleRate, const uint bits, const uint channels);
+
+    /// Finishes the WAV file header by supplementing information of amount of
+    /// data written to file etc
+    void finishHeader();
+
+    /// Writes the WAV file header.
+    void writeHeader();
+
+public:
+    /// Constructor: Creates a new WAV file. Throws a 'runtime_error' exception
+    /// if file creation fails.
+    WavOutFile(const 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)
+               );
+
+    WavOutFile(FILE *file, int sampleRate, int bits, int channels);
+
+    /// Destructor: Finalizes & closes the WAV file.
+    ~WavOutFile();
+
+    /// Write data to WAV file. This function works only with 8bit samples.
+    /// Throws a 'runtime_error' exception if writing to file fails.
+    void write(const unsigned char *buffer, ///< Pointer to sample data buffer.
+               int numElems                 ///< How many array items are to be written to file.
+               );
+
+    /// Write data to WAV file. Throws a 'runtime_error' exception if writing to
+    /// file fails.
+    void write(const short *buffer,     ///< Pointer to sample data buffer.
+               int numElems             ///< How many array items are to be written to file.
+               );
+
+    /// Write data to WAV file in floating point format, saturating sample values to range
+    /// [-1..+1[. Throws a 'runtime_error' exception if writing to file fails.
+    void write(const float *buffer,     ///< Pointer to sample data buffer.
+               int numElems             ///< How many array items are to be written to file.
+               );
+};
+
+ }
+
+#endif

source/SoundStretch/main.cpp

@@ -1,333 +1,321 @@
-////////////////////////////////////////////////////////////////////////////////
-///
-/// SoundStretch main routine.
-///
-/// Author        : Copyright (c) Olli Parviainen
-/// Author e-mail : oparviai 'at' iki.fi
-/// SoundTouch WWW: http://www.surina.net/soundtouch
-///
-////////////////////////////////////////////////////////////////////////////////
-//
-// Last changed  : $Date$
-// File revision : $Revision: 4 $
-//
-// $Id$
-//
-////////////////////////////////////////////////////////////////////////////////
-//
-// License :
-//
-//  SoundTouch audio processing library
-//  Copyright (c) Olli Parviainen
-//
-//  This library is free software; you can redistribute it and/or
-//  modify it under the terms of the GNU Lesser General Public
-//  License as published by the Free Software Foundation; either
-//  version 2.1 of the License, or (at your option) any later version.
-//
-//  This library is distributed in the hope that it will be useful,
-//  but WITHOUT ANY WARRANTY; without even the implied warranty of
-//  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-//  Lesser General Public License for more details.
-//
-//  You should have received a copy of the GNU Lesser General Public
-//  License along with this library; if not, write to the Free Software
-//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
-//
-////////////////////////////////////////////////////////////////////////////////
-
-#include <stdexcept>
-#include <stdio.h>
-#include <string.h>
-#include <time.h>
-#include "RunParameters.h"
-#include "WavFile.h"
-#include "SoundTouch.h"
-#include "BPMDetect.h"
-
-using namespace soundtouch;
-using namespace std;
-
-// Processing chunk size
-#define BUFF_SIZE           2048
-
-#if _WIN32
-    #include <io.h>
-    #include <fcntl.h>
-
-    // Macro for Win32 standard input/output stream support: Sets a file stream into binary mode
-    #define SET_STREAM_TO_BIN_MODE(f) (_setmode(_fileno(f), _O_BINARY))
-#else
-    // Not needed for GNU environment... 
-    #define SET_STREAM_TO_BIN_MODE(f) {}
-#endif
-
-
-static const char _helloText[] = 
-    "\n"
-    "   SoundStretch v%s -  Written by Olli Parviainen 2001 - 2012\n"
-    "==================================================================\n"
-    "author e-mail: <oparviai"
-    "@"
-    "iki.fi> - WWW: http://www.surina.net/soundtouch\n"
-    "\n"
-    "This program is subject to (L)GPL license. Run \"soundstretch -license\" for\n"
-    "more information.\n"
-    "\n";
-
-static void openFiles(WavInFile **inFile, WavOutFile **outFile, const RunParameters *params)
-{
-    int bits, samplerate, channels;
-
-    if (strcmp(params->inFileName, "stdin") == 0)
-    {
-        // used 'stdin' as input file
-        SET_STREAM_TO_BIN_MODE(stdin);
-        *inFile = new WavInFile(stdin);
-    }
-    else
-    {
-        // open input file...
-        *inFile = new WavInFile(params->inFileName);
-    }
-
-    // ... open output file with same sound parameters
-    bits = (int)(*inFile)->getNumBits();
-    samplerate = (int)(*inFile)->getSampleRate();
-    channels = (int)(*inFile)->getNumChannels();
-
-    if (params->outFileName)
-    {
-        if (strcmp(params->outFileName, "stdout") == 0)
-        {
-            SET_STREAM_TO_BIN_MODE(stdout);
-            *outFile = new WavOutFile(stdout, samplerate, bits, channels);
-        }
-        else
-        {
-            *outFile = new WavOutFile(params->outFileName, samplerate, bits, channels);
-        }
-    }
-    else
-    {
-        *outFile = NULL;
-    }
-}
-
-
-
-// Sets the 'SoundTouch' object up according to input file sound format & 
-// command line parameters
-static void setup(SoundTouch *pSoundTouch, const WavInFile *inFile, const RunParameters *params)
-{
-    int sampleRate;
-    int channels;
-
-    sampleRate = (int)inFile->getSampleRate();
-    channels = (int)inFile->getNumChannels();
-    pSoundTouch->setSampleRate(sampleRate);
-    pSoundTouch->setChannels(channels);
-
-    pSoundTouch->setTempoChange(params->tempoDelta);
-    pSoundTouch->setPitchSemiTones(params->pitchDelta);
-    pSoundTouch->setRateChange(params->rateDelta);
-
-    pSoundTouch->setSetting(SETTING_USE_QUICKSEEK, params->quick);
-    pSoundTouch->setSetting(SETTING_USE_AA_FILTER, !(params->noAntiAlias));
-
-    if (params->speech)
-    {
-        // use settings for speech processing
-        pSoundTouch->setSetting(SETTING_SEQUENCE_MS, 40);
-        pSoundTouch->setSetting(SETTING_SEEKWINDOW_MS, 15);
-        pSoundTouch->setSetting(SETTING_OVERLAP_MS, 8);
-        fprintf(stderr, "Tune processing parameters for speech processing.\n");
-    }
-
-    // print processing information
-    if (params->outFileName)
-    {
-#ifdef SOUNDTOUCH_INTEGER_SAMPLES
-        fprintf(stderr, "Uses 16bit integer sample type in processing.\n\n");
-#else
-    #ifndef SOUNDTOUCH_FLOAT_SAMPLES
-        #error "Sampletype not defined"
-    #endif
-        fprintf(stderr, "Uses 32bit floating point sample type in processing.\n\n");
-#endif
-        // print processing information only if outFileName given i.e. some processing will happen
-        fprintf(stderr, "Processing the file with the following changes:\n");
-        fprintf(stderr, "  tempo change = %+g %%\n", params->tempoDelta);
-        fprintf(stderr, "  pitch change = %+g semitones\n", params->pitchDelta);
-        fprintf(stderr, "  rate change  = %+g %%\n\n", params->rateDelta);
-        fprintf(stderr, "Working...");
-    }
-    else
-    {
-        // outFileName not given
-        fprintf(stderr, "Warning: output file name missing, won't output anything.\n\n");
-    }
-
-    fflush(stderr);
-}
-
-
-
-// Processes the sound
-static void process(SoundTouch *pSoundTouch, WavInFile *inFile, WavOutFile *outFile)
-{
-    int nSamples;
-    int nChannels;
-    int buffSizeSamples;
-    SAMPLETYPE sampleBuffer[BUFF_SIZE];
-
-    if ((inFile == NULL) || (outFile == NULL)) return;  // nothing to do.
-
-    nChannels = (int)inFile->getNumChannels();
-    assert(nChannels > 0);
-    buffSizeSamples = BUFF_SIZE / nChannels;
-
-    // Process samples read from the input file
-    while (inFile->eof() == 0)
-    {
-        int num;
-
-        // Read a chunk of samples from the input file
-        num = inFile->read(sampleBuffer, BUFF_SIZE);
-        nSamples = num / (int)inFile->getNumChannels();
-
-        // Feed the samples into SoundTouch processor
-        pSoundTouch->putSamples(sampleBuffer, nSamples);
-
-        // Read ready samples from SoundTouch processor & write them output file.
-        // NOTES:
-        // - 'receiveSamples' doesn't necessarily return any samples at all
-        //   during some rounds!
-        // - On the other hand, during some round 'receiveSamples' may have more
-        //   ready samples than would fit into 'sampleBuffer', and for this reason 
-        //   the 'receiveSamples' call is iterated for as many times as it
-        //   outputs samples.
-        do 
-        {
-            nSamples = pSoundTouch->receiveSamples(sampleBuffer, buffSizeSamples);
-            outFile->write(sampleBuffer, nSamples * nChannels);
-        } while (nSamples != 0);
-    }
-
-    // Now the input file is processed, yet 'flush' few last samples that are
-    // hiding in the SoundTouch's internal processing pipeline.
-    pSoundTouch->flush();
-    do 
-    {
-        nSamples = pSoundTouch->receiveSamples(sampleBuffer, buffSizeSamples);
-        outFile->write(sampleBuffer, nSamples * nChannels);
-    } while (nSamples != 0);
-}
-
-
-
-// Detect BPM rate of inFile and adjust tempo setting accordingly if necessary
-static void detectBPM(WavInFile *inFile, RunParameters *params)
-{
-    float bpmValue;
-    int nChannels;
-    BPMDetect bpm(inFile->getNumChannels(), inFile->getSampleRate());
-    SAMPLETYPE sampleBuffer[BUFF_SIZE];
-
-    // detect bpm rate
-    fprintf(stderr, "Detecting BPM rate...");
-    fflush(stderr);
-
-    nChannels = (int)inFile->getNumChannels();
-    assert(BUFF_SIZE % nChannels == 0);
-
-    // Process the 'inFile' in small blocks, repeat until whole file has 
-    // been processed
-    while (inFile->eof() == 0)
-    {
-        int num, samples;
-
-        // Read sample data from input file
-        num = inFile->read(sampleBuffer, BUFF_SIZE);
-
-        // Enter the new samples to the bpm analyzer class
-        samples = num / nChannels;
-        bpm.inputSamples(sampleBuffer, samples);
-    }
-
-    // Now the whole song data has been analyzed. Read the resulting bpm.
-    bpmValue = bpm.getBpm();
-    fprintf(stderr, "Done!\n");
-
-    // rewind the file after bpm detection
-    inFile->rewind();
-
-    if (bpmValue > 0)
-    {
-        fprintf(stderr, "Detected BPM rate %.1f\n\n", bpmValue);
-    }
-    else
-    {
-        fprintf(stderr, "Couldn't detect BPM rate.\n\n");
-        return;
-    }
-
-    if (params->goalBPM > 0)
-    {
-        // adjust tempo to given bpm
-        params->tempoDelta = (params->goalBPM / bpmValue - 1.0f) * 100.0f;
-        fprintf(stderr, "The file will be converted to %.1f BPM\n\n", params->goalBPM);
-    }
-}
-
-
-
-int main(const int nParams, const char * const paramStr[])
-{
-    WavInFile *inFile;
-    WavOutFile *outFile;
-    RunParameters *params;
-    SoundTouch soundTouch;
-
-    fprintf(stderr, _helloText, SoundTouch::getVersionString());
-
-    try 
-    {
-        // Parse command line parameters
-        params = new RunParameters(nParams, paramStr);
-
-        // Open input & output files
-        openFiles(&inFile, &outFile, params);
-
-        if (params->detectBPM == TRUE)
-        {
-            // detect sound BPM (and adjust processing parameters
-            //  accordingly if necessary)
-            detectBPM(inFile, params);
-        }
-
-        // Setup the 'SoundTouch' object for processing the sound
-        setup(&soundTouch, inFile, params);
-
-        // clock_t cs = clock();    // for benchmarking processing duration
-        // Process the sound
-        process(&soundTouch, inFile, outFile);
-        // clock_t ce = clock();    // for benchmarking processing duration
-        // printf("duration: %lf\n", (double)(ce-cs)/CLOCKS_PER_SEC);
-
-        // Close WAV file handles & dispose of the objects
-        delete inFile;
-        delete outFile;
-        delete params;
-
-        fprintf(stderr, "Done!\n");
-    } 
-    catch (const runtime_error &e) 
-    {
-        // An exception occurred during processing, display an error message
-        fprintf(stderr, "%s\n", e.what());
-        return -1;
-    }
-
-    return 0;
-}
+////////////////////////////////////////////////////////////////////////////////
+///
+/// SoundStretch main 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 <iostream>
+#include <memory>
+#include <stdexcept>
+#include <string>
+#include <cstdio>
+#include <ctime>
+#include "RunParameters.h"
+#include "WavFile.h"
+#include "SoundTouch.h"
+#include "BPMDetect.h"
+
+using namespace soundtouch;
+using namespace std;
+
+namespace soundstretch
+{
+
+// 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 <fcntl.h>
+
+// Macro for Win32 standard input/output stream support: Sets a file stream into binary mode
+#define SET_STREAM_TO_BIN_MODE(f) (_setmode(_fileno(f), _O_BINARY))
+#else
+    // Not needed for GNU environment...
+#define SET_STREAM_TO_BIN_MODE(f) {}
+#endif
+
+
+static const char _helloText[] =
+    "\n"
+    "   SoundStretch v%s -  Copyright (c) Olli Parviainen\n"
+    "=========================================================\n"
+    "author e-mail: <oparviai"
+    "@"
+    "iki.fi> - WWW: http://www.surina.net/soundtouch\n"
+    "\n"
+    "This program is subject to (L)GPL license. Run \"soundstretch -license\" for\n"
+    "more information.\n"
+    "\n";
+
+static void openFiles(unique_ptr<WavInFile>& inFile, unique_ptr<WavOutFile>& outFile, const RunParameters& params)
+{
+    if (params.inFileName == STRING_CONST("stdin"))
+    {
+        // used 'stdin' as input file
+        SET_STREAM_TO_BIN_MODE(stdin);
+        inFile = make_unique<WavInFile>(stdin);
+    }
+    else
+    {
+        // open input file...
+        inFile = make_unique<WavInFile>(params.inFileName.c_str());
+    }
+
+    // ... open output file with same sound parameters
+    const int bits = (int)inFile->getNumBits();
+    const int samplerate = (int)inFile->getSampleRate();
+    const int channels = (int)inFile->getNumChannels();
+
+    if (!params.outFileName.empty())
+    {
+        if (params.outFileName == STRING_CONST("stdout"))
+        {
+            SET_STREAM_TO_BIN_MODE(stdout);
+            outFile = make_unique<WavOutFile>(stdout, samplerate, bits, channels);
+        }
+        else
+        {
+            outFile = make_unique<WavOutFile>(params.outFileName.c_str(), samplerate, bits, channels);
+        }
+    }
+}
+
+
+// Sets the 'SoundTouch' object up according to input file sound format &
+// command line parameters
+static void setup(SoundTouch& soundTouch, const WavInFile& inFile, const RunParameters& params)
+{
+    const int sampleRate = (int)inFile.getSampleRate();
+    const int channels = (int)inFile.getNumChannels();
+    soundTouch.setSampleRate(sampleRate);
+    soundTouch.setChannels(channels);
+
+    soundTouch.setTempoChange(params.tempoDelta);
+    soundTouch.setPitchSemiTones(params.pitchDelta);
+    soundTouch.setRateChange(params.rateDelta);
+
+    soundTouch.setSetting(SETTING_USE_QUICKSEEK, params.quick);
+    soundTouch.setSetting(SETTING_USE_AA_FILTER, !(params.noAntiAlias));
+
+    if (params.speech)
+    {
+        // use settings for speech processing
+        soundTouch.setSetting(SETTING_SEQUENCE_MS, 40);
+        soundTouch.setSetting(SETTING_SEEKWINDOW_MS, 15);
+        soundTouch.setSetting(SETTING_OVERLAP_MS, 8);
+        fprintf(stderr, "Tune processing parameters for speech processing.\n");
+    }
+
+    // print processing information
+    if (!params.outFileName.empty())
+    {
+#ifdef SOUNDTOUCH_INTEGER_SAMPLES
+        fprintf(stderr, "Uses 16bit integer sample type in processing.\n\n");
+#else
+#ifndef SOUNDTOUCH_FLOAT_SAMPLES
+#error "Sampletype not defined"
+#endif
+        fprintf(stderr, "Uses 32bit floating point sample type in processing.\n\n");
+#endif
+        // print processing information only if outFileName given i.e. some processing will happen
+        fprintf(stderr, "Processing the file with the following changes:\n");
+        fprintf(stderr, "  tempo change = %+lg %%\n", params.tempoDelta);
+        fprintf(stderr, "  pitch change = %+lg semitones\n", params.pitchDelta);
+        fprintf(stderr, "  rate change  = %+lg %%\n\n", params.rateDelta);
+        fprintf(stderr, "Working...");
+    }
+    else
+    {
+        // outFileName not given
+        fprintf(stderr, "Warning: output file name missing, won't output anything.\n\n");
+    }
+
+    fflush(stderr);
+}
+
+
+// Processes the sound
+static void process(SoundTouch& soundTouch, WavInFile& inFile, WavOutFile& outFile)
+{
+    SAMPLETYPE sampleBuffer[BUFF_SIZE];
+    int nSamples;
+
+    const int nChannels = (int)inFile.getNumChannels();
+    assert(nChannels > 0);
+    const int buffSizeSamples = BUFF_SIZE / nChannels;
+
+    // Process samples read from the input file
+    while (inFile.eof() == 0)
+    {
+        // Read a chunk of samples from the input file
+        const int num = inFile.read(sampleBuffer, BUFF_SIZE);
+        int nSamples = num / (int)inFile.getNumChannels();
+
+        // Feed the samples into SoundTouch processor
+        soundTouch.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 = soundTouch.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.
+    soundTouch.flush();
+    do
+    {
+        nSamples = soundTouch.receiveSamples(sampleBuffer, buffSizeSamples);
+        outFile.write(sampleBuffer, nSamples * nChannels);
+    } while (nSamples != 0);
+}
+
+
+// Detect BPM rate of inFile and adjust tempo setting accordingly if necessary
+static void detectBPM(WavInFile& inFile, RunParameters& params)
+{
+    BPMDetect bpm(inFile.getNumChannels(), inFile.getSampleRate());
+    SAMPLETYPE sampleBuffer[BUFF_SIZE];
+
+    // detect bpm rate
+    fprintf(stderr, "Detecting BPM rate...");
+    fflush(stderr);
+
+    const int nChannels = (int)inFile.getNumChannels();
+    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)
+    {
+        // Read sample data from input file
+        const int num = inFile.read(sampleBuffer, readSize);
+
+        // Enter the new samples to the bpm analyzer class
+        const int samples = num / nChannels;
+        bpm.inputSamples(sampleBuffer, samples);
+    }
+
+    // Now the whole song data has been analyzed. Read the resulting bpm.
+    const float bpmValue = bpm.getBpm();
+    fprintf(stderr, "Done!\n");
+
+    // rewind the file after bpm detection
+    inFile.rewind();
+
+    if (bpmValue > 0)
+    {
+        fprintf(stderr, "Detected BPM rate %.1lf\n\n", bpmValue);
+    }
+    else
+    {
+        fprintf(stderr, "Couldn't detect BPM rate.\n\n");
+        return;
+    }
+
+    if (params.goalBPM > 0)
+    {
+        // adjust tempo to given bpm
+        params.tempoDelta = (params.goalBPM / bpmValue - 1.0f) * 100.0f;
+        fprintf(stderr, "The file will be converted to %.1lf BPM\n\n", params.goalBPM);
+    }
+}
+
+void printHelloText()
+{
+    SoundTouch soundTouch;
+    fprintf(stderr, _helloText, soundTouch.getVersionString());
+}
+
+void ss_main(RunParameters& params)
+{
+    unique_ptr<WavInFile> inFile;
+    unique_ptr<WavOutFile> outFile;
+    SoundTouch soundTouch;
+
+    // Open input & output files
+    openFiles(inFile, outFile, params);
+
+    if (params.detectBPM == true)
+    {
+        // detect sound BPM (and adjust processing parameters
+        //  accordingly if necessary)
+        detectBPM(*inFile, params);
+    }
+
+    // Setup the 'SoundTouch' object for processing the sound
+    setup(soundTouch, *inFile, params);
+
+    // clock_t cs = clock();    // for benchmarking processing duration
+    // Process the sound
+    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);
+
+    fprintf(stderr, "Done!\n");
+}
+
+}
+
+#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)
+    {
+        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

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source/SoundStretch/soundstretch.dsw

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source/SoundStretch/soundstretch.sln

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 Global
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source/SoundStretch/soundstretch.vcproj

@@ -1,235 +0,0 @@
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source/SoundTouch/AAFilter.cpp

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-////////////////////////////////////////////////////////////////////////////////
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-/// MMX optimization. 
-/// 
-/// Anti-alias filter is used to prevent folding of high frequencies when 
-/// transposing the sample rate with interpolation.
-///
-/// Author        : Copyright (c) Olli Parviainen
-/// Author e-mail : oparviai 'at' iki.fi
-/// SoundTouch WWW: http://www.surina.net/soundtouch
-///
-////////////////////////////////////////////////////////////////////////////////
-//
-// Last changed  : $Date$
-// File revision : $Revision: 4 $
-//
-// $Id$
-//
-////////////////////////////////////////////////////////////////////////////////
-//
-// License :
-//
-//  SoundTouch audio processing library
-//  Copyright (c) Olli Parviainen
-//
-//  This library is free software; you can redistribute it and/or
-//  modify it under the terms of the GNU Lesser General Public
-//  License as published by the Free Software Foundation; either
-//  version 2.1 of the License, or (at your option) any later version.
-//
-//  This library is distributed in the hope that it will be useful,
-//  but WITHOUT ANY WARRANTY; without even the implied warranty of
-//  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-//  Lesser General Public License for more details.
-//
-//  You should have received a copy of the GNU Lesser General Public
-//  License along with this library; if not, write to the Free Software
-//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
-//
-////////////////////////////////////////////////////////////////////////////////
-
-#include <memory.h>
-#include <assert.h>
-#include <math.h>
-#include <stdlib.h>
-#include "AAFilter.h"
-#include "FIRFilter.h"
-
-using namespace soundtouch;
-
-#define PI        3.141592655357989
-#define TWOPI    (2 * PI)
-
-/*****************************************************************************
- *
- * Implementation of the class 'AAFilter'
- *
- *****************************************************************************/
-
-AAFilter::AAFilter(uint len)
-{
-    pFIR = FIRFilter::newInstance();
-    cutoffFreq = 0.5;
-    setLength(len);
-}
-
-
-
-AAFilter::~AAFilter()
-{
-    delete pFIR;
-}
-
-
-
-// Sets new anti-alias filter cut-off edge frequency, scaled to
-// sampling frequency (nyquist frequency = 0.5).
-// The filter will cut frequencies higher than the given frequency.
-void AAFilter::setCutoffFreq(double newCutoffFreq)
-{
-    cutoffFreq = newCutoffFreq;
-    calculateCoeffs();
-}
-
-
-
-// Sets number of FIR filter taps
-void AAFilter::setLength(uint newLength)
-{
-    length = newLength;
-    calculateCoeffs();
-}
-
-
-
-// Calculates coefficients for a low-pass FIR filter using Hamming window
-void AAFilter::calculateCoeffs()
-{
-    uint i;
-    double cntTemp, temp, tempCoeff,h, w;
-    double fc2, wc;
-    double scaleCoeff, sum;
-    double *work;
-    SAMPLETYPE *coeffs;
-
-    assert(length >= 2);
-    assert(length % 4 == 0);
-    assert(cutoffFreq >= 0);
-    assert(cutoffFreq <= 0.5);
-
-    work = new double[length];
-    coeffs = new SAMPLETYPE[length];
-
-    fc2 = 2.0 * cutoffFreq; 
-    wc = PI * fc2;
-    tempCoeff = TWOPI / (double)length;
-
-    sum = 0;
-    for (i = 0; i < length; i ++) 
-    {
-        cntTemp = (double)i - (double)(length / 2);
-
-        temp = cntTemp * wc;
-        if (temp != 0) 
-        {
-            h = fc2 * sin(temp) / temp;                     // sinc function
-        } 
-        else 
-        {
-            h = 1.0;
-        }
-        w = 0.54 + 0.46 * cos(tempCoeff * cntTemp);       // hamming window
-
-        temp = w * h;
-        work[i] = temp;
-
-        // calc net sum of coefficients 
-        sum += temp;
-    }
-
-    // ensure the sum of coefficients is larger than zero
-    assert(sum > 0);
-
-    // ensure we've really designed a lowpass filter...
-    assert(work[length/2] > 0);
-    assert(work[length/2 + 1] > -1e-6);
-    assert(work[length/2 - 1] > -1e-6);
-
-    // Calculate a scaling coefficient in such a way that the result can be
-    // divided by 16384
-    scaleCoeff = 16384.0f / sum;
-
-    for (i = 0; i < length; i ++) 
-    {
-        // scale & round to nearest integer
-        temp = work[i] * scaleCoeff;
-        temp += (temp >= 0) ? 0.5 : -0.5;
-        // ensure no overfloods
-        assert(temp >= -32768 && temp <= 32767);
-        coeffs[i] = (SAMPLETYPE)temp;
-    }
-
-    // Set coefficients. Use divide factor 14 => divide result by 2^14 = 16384
-    pFIR->setCoefficients(coeffs, length, 14);
-
-    delete[] work;
-    delete[] coeffs;
-}
-
-
-// Applies the filter to the given sequence of samples. 
-// Note : The amount of outputted samples is by value of 'filter length' 
-// smaller than the amount of input samples.
-uint AAFilter::evaluate(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples, uint numChannels) const
-{
-    return pFIR->evaluate(dest, src, numSamples, numChannels);
-}
-
-
-uint AAFilter::getLength() const
-{
-    return pFIR->getLength();
-}
+////////////////////////////////////////////////////////////////////////////////
+///
+/// FIR low-pass (anti-alias) filter with filter coefficient design routine and
+/// MMX optimization.
+///
+/// Anti-alias filter is used to prevent folding of high frequencies when
+/// transposing the sample rate with interpolation.
+///
+/// Author        : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+//  SoundTouch audio processing library
+//  Copyright (c) Olli Parviainen
+//
+//  This library is free software; you can redistribute it and/or
+//  modify it under the terms of the GNU Lesser General Public
+//  License as published by the Free Software Foundation; either
+//  version 2.1 of the License, or (at your option) any later version.
+//
+//  This library is distributed in the hope that it will be useful,
+//  but WITHOUT ANY WARRANTY; without even the implied warranty of
+//  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+//  Lesser General Public License for more details.
+//
+//  You should have received a copy of the GNU Lesser General Public
+//  License along with this library; if not, write to the Free Software
+//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#include <memory.h>
+#include <assert.h>
+#include <math.h>
+#include <stdlib.h>
+#include "AAFilter.h"
+#include "FIRFilter.h"
+
+using namespace soundtouch;
+
+#define PI       3.14159265358979323846
+#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'
+ *
+ *****************************************************************************/
+
+AAFilter::AAFilter(uint len)
+{
+    pFIR = FIRFilter::newInstance();
+    cutoffFreq = 0.5;
+    setLength(len);
+}
+
+
+AAFilter::~AAFilter()
+{
+    delete pFIR;
+}
+
+
+// Sets new anti-alias filter cut-off edge frequency, scaled to
+// sampling frequency (nyquist frequency = 0.5).
+// The filter will cut frequencies higher than the given frequency.
+void AAFilter::setCutoffFreq(double newCutoffFreq)
+{
+    cutoffFreq = newCutoffFreq;
+    calculateCoeffs();
+}
+
+
+// Sets number of FIR filter taps
+void AAFilter::setLength(uint newLength)
+{
+    length = newLength;
+    calculateCoeffs();
+}
+
+
+// Calculates coefficients for a low-pass FIR filter using Hamming window
+void AAFilter::calculateCoeffs()
+{
+    uint i;
+    double cntTemp, temp, tempCoeff,h, w;
+    double wc;
+    double scaleCoeff, sum;
+    double *work;
+    SAMPLETYPE *coeffs;
+
+    assert(length >= 2);
+    assert(length % 4 == 0);
+    assert(cutoffFreq >= 0);
+    assert(cutoffFreq <= 0.5);
+
+    work = new double[length];
+    coeffs = new SAMPLETYPE[length];
+
+    wc = 2.0 * PI * cutoffFreq;
+    tempCoeff = TWOPI / (double)length;
+
+    sum = 0;
+    for (i = 0; i < length; i ++)
+    {
+        cntTemp = (double)i - (double)(length / 2);
+
+        temp = cntTemp * wc;
+        if (temp != 0)
+        {
+            h = sin(temp) / temp;                     // sinc function
+        }
+        else
+        {
+            h = 1.0;
+        }
+        w = 0.54 + 0.46 * cos(tempCoeff * cntTemp);       // hamming window
+
+        temp = w * h;
+        work[i] = temp;
+
+        // calc net sum of coefficients
+        sum += temp;
+    }
+
+    // ensure the sum of coefficients is larger than zero
+    assert(sum > 0);
+
+    // ensure we've really designed a lowpass filter...
+    assert(work[length/2] > 0);
+    assert(work[length/2 + 1] > -1e-6);
+    assert(work[length/2 - 1] > -1e-6);
+
+    // Calculate a scaling coefficient in such a way that the result can be
+    // divided by 16384
+    scaleCoeff = 16384.0f / sum;
+
+    for (i = 0; i < length; i ++)
+    {
+        temp = work[i] * scaleCoeff;
+        // scale & round to nearest integer
+        temp += (temp >= 0) ? 0.5 : -0.5;
+        // ensure no overfloods
+        assert(temp >= -32768 && temp <= 32767);
+        coeffs[i] = (SAMPLETYPE)temp;
+    }
+
+    // Set coefficients. Use divide factor 14 => divide result by 2^14 = 16384
+    pFIR->setCoefficients(coeffs, length, 14);
+
+    _DEBUG_SAVE_AAFIR_COEFFS(coeffs, length);
+
+    delete[] work;
+    delete[] coeffs;
+}
+
+
+// Applies the filter to the given sequence of samples.
+// Note : The amount of outputted samples is by value of 'filter length'
+// smaller than the amount of input samples.
+uint AAFilter::evaluate(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples, uint numChannels) const
+{
+    return pFIR->evaluate(dest, src, numSamples, numChannels);
+}
+
+
+/// 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

@@ -1,91 +1,93 @@
-////////////////////////////////////////////////////////////////////////////////
-///
-/// Sampled sound tempo changer/time stretch algorithm. Changes the sound tempo 
-/// while maintaining the original pitch by using a time domain WSOLA-like method 
-/// with several performance-increasing tweaks.
-///
-/// Anti-alias filter is used to prevent folding of high frequencies when 
-/// transposing the sample rate with interpolation.
-///
-/// Author        : Copyright (c) Olli Parviainen
-/// Author e-mail : oparviai 'at' iki.fi
-/// SoundTouch WWW: http://www.surina.net/soundtouch
-///
-////////////////////////////////////////////////////////////////////////////////
-//
-// Last changed  : $Date$
-// File revision : $Revision: 4 $
-//
-// $Id$
-//
-////////////////////////////////////////////////////////////////////////////////
-//
-// License :
-//
-//  SoundTouch audio processing library
-//  Copyright (c) Olli Parviainen
-//
-//  This library is free software; you can redistribute it and/or
-//  modify it under the terms of the GNU Lesser General Public
-//  License as published by the Free Software Foundation; either
-//  version 2.1 of the License, or (at your option) any later version.
-//
-//  This library is distributed in the hope that it will be useful,
-//  but WITHOUT ANY WARRANTY; without even the implied warranty of
-//  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-//  Lesser General Public License for more details.
-//
-//  You should have received a copy of the GNU Lesser General Public
-//  License along with this library; if not, write to the Free Software
-//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
-//
-////////////////////////////////////////////////////////////////////////////////
-
-#ifndef AAFilter_H
-#define AAFilter_H
-
-#include "STTypes.h"
-
-namespace soundtouch
-{
-
-class AAFilter
-{
-protected:
-    class FIRFilter *pFIR;
-
-    /// Low-pass filter cut-off frequency, negative = invalid
-    double cutoffFreq;
-
-    /// num of filter taps
-    uint length;
-
-    /// Calculate the FIR coefficients realizing the given cutoff-frequency
-    void calculateCoeffs();
-public:
-    AAFilter(uint length);
-
-    ~AAFilter();
-
-    /// Sets new anti-alias filter cut-off edge frequency, scaled to sampling 
-    /// frequency (nyquist frequency = 0.5). The filter will cut off the 
-    /// frequencies than that.
-    void setCutoffFreq(double newCutoffFreq);
-
-    /// Sets number of FIR filter taps, i.e. ~filter complexity
-    void setLength(uint newLength);
-
-    uint getLength() const;
-
-    /// Applies the filter to the given sequence of samples. 
-    /// Note : The amount of outputted samples is by value of 'filter length' 
-    /// smaller than the amount of input samples.
-    uint evaluate(SAMPLETYPE *dest, 
-                  const SAMPLETYPE *src, 
-                  uint numSamples, 
-                  uint numChannels) const;
-};
-
-}
-
-#endif
+////////////////////////////////////////////////////////////////////////////////
+///
+/// Sampled sound tempo changer/time stretch algorithm. Changes the sound tempo
+/// while maintaining the original pitch by using a time domain WSOLA-like method
+/// with several performance-increasing tweaks.
+///
+/// Anti-alias filter is used to prevent folding of high frequencies when
+/// transposing the sample rate with interpolation.
+///
+/// Author        : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+//  SoundTouch audio processing library
+//  Copyright (c) Olli Parviainen
+//
+//  This library is free software; you can redistribute it and/or
+//  modify it under the terms of the GNU Lesser General Public
+//  License as published by the Free Software Foundation; either
+//  version 2.1 of the License, or (at your option) any later version.
+//
+//  This library is distributed in the hope that it will be useful,
+//  but WITHOUT ANY WARRANTY; without even the implied warranty of
+//  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+//  Lesser General Public License for more details.
+//
+//  You should have received a copy of the GNU Lesser General Public
+//  License along with this library; if not, write to the Free Software
+//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#ifndef AAFilter_H
+#define AAFilter_H
+
+#include "STTypes.h"
+#include "FIFOSampleBuffer.h"
+
+namespace soundtouch
+{
+
+class AAFilter
+{
+protected:
+    class FIRFilter *pFIR;
+
+    /// Low-pass filter cut-off frequency, negative = invalid
+    double cutoffFreq;
+
+    /// num of filter taps
+    uint length;
+
+    /// Calculate the FIR coefficients realizing the given cutoff-frequency
+    void calculateCoeffs();
+public:
+    AAFilter(uint length);
+
+    ~AAFilter();
+
+    /// Sets new anti-alias filter cut-off edge frequency, scaled to sampling
+    /// frequency (nyquist frequency = 0.5). The filter will cut off the
+    /// frequencies than that.
+    void setCutoffFreq(double newCutoffFreq);
+
+    /// Sets number of FIR filter taps, i.e. ~filter complexity
+    void setLength(uint newLength);
+
+    uint getLength() const;
+
+    /// Applies the filter to the given sequence of samples.
+    /// Note : The amount of outputted samples is by value of 'filter length'
+    /// smaller than the amount of input samples.
+    uint evaluate(SAMPLETYPE *dest,
+                  const SAMPLETYPE *src,
+                  uint numSamples,
+                  uint numChannels) const;
+
+    /// 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;
+
+};
+
+}
+
+#endif

source/SoundTouch/BPMDetect.cpp

@@ -1,370 +1,571 @@
-////////////////////////////////////////////////////////////////////////////////
-///
-/// Beats-per-minute (BPM) detection routine.
-///
-/// The beat detection algorithm works as follows:
-/// - Use function 'inputSamples' to input a chunks of samples to the class for
-///   analysis. It's a good idea to enter a large sound file or stream in smallish
-///   chunks of around few kilosamples in order not to extinguish too much RAM memory.
-/// - Inputted sound data is decimated to approx 500 Hz to reduce calculation burden,
-///   which is basically ok as low (bass) frequencies mostly determine the beat rate.
-///   Simple averaging is used for anti-alias filtering because the resulting signal
-///   quality isn't of that high importance.
-/// - Decimated sound data is enveloped, i.e. the amplitude shape is detected by
-///   taking absolute value that's smoothed by sliding average. Signal levels that
-///   are below a couple of times the general RMS amplitude level are cut away to
-///   leave only notable peaks there.
-/// - Repeating sound patterns (e.g. beats) are detected by calculating short-term 
-///   autocorrelation function of the enveloped signal.
-/// - After whole sound data file has been analyzed as above, the bpm level is 
-///   detected by function 'getBpm' that finds the highest peak of the autocorrelation 
-///   function, calculates it's precise location and converts this reading to bpm's.
-///
-/// Author        : Copyright (c) Olli Parviainen
-/// Author e-mail : oparviai 'at' iki.fi
-/// SoundTouch WWW: http://www.surina.net/soundtouch
-///
-////////////////////////////////////////////////////////////////////////////////
-//
-// Last changed  : $Date$
-// File revision : $Revision: 4 $
-//
-// $Id$
-//
-////////////////////////////////////////////////////////////////////////////////
-//
-// License :
-//
-//  SoundTouch audio processing library
-//  Copyright (c) Olli Parviainen
-//
-//  This library is free software; you can redistribute it and/or
-//  modify it under the terms of the GNU Lesser General Public
-//  License as published by the Free Software Foundation; either
-//  version 2.1 of the License, or (at your option) any later version.
-//
-//  This library is distributed in the hope that it will be useful,
-//  but WITHOUT ANY WARRANTY; without even the implied warranty of
-//  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-//  Lesser General Public License for more details.
-//
-//  You should have received a copy of the GNU Lesser General Public
-//  License along with this library; if not, write to the Free Software
-//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
-//
-////////////////////////////////////////////////////////////////////////////////
-
-#include <math.h>
-#include <assert.h>
-#include <string.h>
-#include <stdio.h>
-#include "FIFOSampleBuffer.h"
-#include "PeakFinder.h"
-#include "BPMDetect.h"
-
-using namespace soundtouch;
-
-#define INPUT_BLOCK_SAMPLES       2048
-#define DECIMATED_BLOCK_SAMPLES   256
-
-/// decay constant for calculating RMS volume sliding average approximation 
-/// (time constant is about 10 sec)
-const float avgdecay = 0.99986f;
-
-/// Normalization coefficient for calculating RMS sliding average approximation.
-const float avgnorm = (1 - avgdecay);
-
-
-////////////////////////////////////////////////////////////////////////////////
-
-// Enable following define to create bpm analysis file:
-
-// #define _CREATE_BPM_DEBUG_FILE
-
-#ifdef _CREATE_BPM_DEBUG_FILE
-
-    #define DEBUGFILE_NAME  "c:\\temp\\soundtouch-bpm-debug.txt"
-
-    static void _SaveDebugData(const float *data, int minpos, int maxpos, double coeff)
-    {
-        FILE *fptr = fopen(DEBUGFILE_NAME, "wt");
-        int i;
-
-        if (fptr)
-        {
-            printf("\n\nWriting BPM debug data into file " DEBUGFILE_NAME "\n\n");
-            for (i = minpos; i < maxpos; i ++)
-            {
-                fprintf(fptr, "%d\t%.1lf\t%f\n", i, coeff / (double)i, data[i]);
-            }
-            fclose(fptr);
-        }
-    }
-#else
-    #define _SaveDebugData(a,b,c,d)
-#endif
-
-////////////////////////////////////////////////////////////////////////////////
-
-
-BPMDetect::BPMDetect(int numChannels, int aSampleRate)
-{
-    this->sampleRate = aSampleRate;
-    this->channels = numChannels;
-
-    decimateSum = 0;
-    decimateCount = 0;
-
-    envelopeAccu = 0;
-
-    // Initialize RMS volume accumulator to RMS level of 1500 (out of 32768) that's
-    // safe initial RMS signal level value for song data. This value is then adapted
-    // to the actual level during processing.
-#ifdef SOUNDTOUCH_INTEGER_SAMPLES
-    // integer samples
-    RMSVolumeAccu = (1500 * 1500) / avgnorm;
-#else
-    // float samples, scaled to range [-1..+1[
-    RMSVolumeAccu = (0.045f * 0.045f) / avgnorm;
-#endif
-
-    // choose decimation factor so that result is approx. 1000 Hz
-    decimateBy = sampleRate / 1000;
-    assert(decimateBy > 0);
-    assert(INPUT_BLOCK_SAMPLES < decimateBy * DECIMATED_BLOCK_SAMPLES);
-
-    // Calculate window length & starting item according to desired min & max bpms
-    windowLen = (60 * sampleRate) / (decimateBy * MIN_BPM);
-    windowStart = (60 * sampleRate) / (decimateBy * MAX_BPM);
-
-    assert(windowLen > windowStart);
-
-    // allocate new working objects
-    xcorr = new float[windowLen];
-    memset(xcorr, 0, windowLen * sizeof(float));
-
-    // allocate processing buffer
-    buffer = new FIFOSampleBuffer();
-    // we do processing in mono mode
-    buffer->setChannels(1);
-    buffer->clear();
-}
-
-
-
-BPMDetect::~BPMDetect()
-{
-    delete[] xcorr;
-    delete buffer;
-}
-
-
-
-/// convert to mono, low-pass filter & decimate to about 500 Hz. 
-/// return number of outputted samples.
-///
-/// Decimation is used to remove the unnecessary frequencies and thus to reduce 
-/// the amount of data needed to be processed as calculating autocorrelation 
-/// function is a very-very heavy operation.
-///
-/// Anti-alias filtering is done simply by averaging the samples. This is really a 
-/// poor-man's anti-alias filtering, but it's not so critical in this kind of application
-/// (it'd also be difficult to design a high-quality filter with steep cut-off at very 
-/// narrow band)
-int BPMDetect::decimate(SAMPLETYPE *dest, const SAMPLETYPE *src, int numsamples)
-{
-    int count, outcount;
-    LONG_SAMPLETYPE out;
-
-    assert(channels > 0);
-    assert(decimateBy > 0);
-    outcount = 0;
-    for (count = 0; count < numsamples; count ++) 
-    {
-        int j;
-
-        // convert to mono and accumulate
-        for (j = 0; j < channels; j ++)
-        {
-            decimateSum += src[j];
-        }
-        src += j;
-
-        decimateCount ++;
-        if (decimateCount >= decimateBy) 
-        {
-            // Store every Nth sample only
-            out = (LONG_SAMPLETYPE)(decimateSum / (decimateBy * channels));
-            decimateSum = 0;
-            decimateCount = 0;
-#ifdef SOUNDTOUCH_INTEGER_SAMPLES
-            // check ranges for sure (shouldn't actually be necessary)
-            if (out > 32767) 
-            {
-                out = 32767;
-            } 
-            else if (out < -32768) 
-            {
-                out = -32768;
-            }
-#endif // SOUNDTOUCH_INTEGER_SAMPLES
-            dest[outcount] = (SAMPLETYPE)out;
-            outcount ++;
-        }
-    }
-    return outcount;
-}
-
-
-
-// Calculates autocorrelation function of the sample history buffer
-void BPMDetect::updateXCorr(int process_samples)
-{
-    int offs;
-    SAMPLETYPE *pBuffer;
-    
-    assert(buffer->numSamples() >= (uint)(process_samples + windowLen));
-
-    pBuffer = buffer->ptrBegin();
-    for (offs = windowStart; offs < windowLen; offs ++) 
-    {
-        LONG_SAMPLETYPE sum;
-        int i;
-
-        sum = 0;
-        for (i = 0; i < process_samples; i ++) 
-        {
-            sum += pBuffer[i] * pBuffer[i + offs];    // scaling the sub-result shouldn't be necessary
-        }
-//        xcorr[offs] *= xcorr_decay;   // decay 'xcorr' here with suitable coefficients 
-                                        // if it's desired that the system adapts automatically to
-                                        // various bpms, e.g. in processing continouos music stream.
-                                        // The 'xcorr_decay' should be a value that's smaller than but 
-                                        // close to one, and should also depend on 'process_samples' value.
-
-        xcorr[offs] += (float)sum;
-    }
-}
-
-
-// Calculates envelope of the sample data
-void BPMDetect::calcEnvelope(SAMPLETYPE *samples, int numsamples) 
-{
-    const static double decay = 0.7f;               // decay constant for smoothing the envelope
-    const static double norm = (1 - decay);
-
-    int i;
-    LONG_SAMPLETYPE out;
-    double val;
-
-    for (i = 0; i < numsamples; i ++) 
-    {
-        // calc average RMS volume
-        RMSVolumeAccu *= avgdecay;
-        val = (float)fabs((float)samples[i]);
-        RMSVolumeAccu += val * val;
-
-        // cut amplitudes that are below cutoff ~2 times RMS volume
-        // (we're interested in peak values, not the silent moments)
-        if (val < 0.5 * sqrt(RMSVolumeAccu * avgnorm))
-        {
-            val = 0;
-        }
-
-        // smooth amplitude envelope
-        envelopeAccu *= decay;
-        envelopeAccu += val;
-        out = (LONG_SAMPLETYPE)(envelopeAccu * norm);
-
-#ifdef SOUNDTOUCH_INTEGER_SAMPLES
-        // cut peaks (shouldn't be necessary though)
-        if (out > 32767) out = 32767;
-#endif // SOUNDTOUCH_INTEGER_SAMPLES
-        samples[i] = (SAMPLETYPE)out;
-    }
-}
-
-
-
-void BPMDetect::inputSamples(const SAMPLETYPE *samples, int numSamples)
-{
-    SAMPLETYPE decimated[DECIMATED_BLOCK_SAMPLES];
-
-    // iterate so that max INPUT_BLOCK_SAMPLES processed per iteration
-    while (numSamples > 0)
-    {
-        int block;
-        int decSamples;
-
-        block = (numSamples > INPUT_BLOCK_SAMPLES) ? INPUT_BLOCK_SAMPLES : numSamples;
-
-        // decimate. note that converts to mono at the same time
-        decSamples = decimate(decimated, samples, block);
-        samples += block * channels;
-        numSamples -= block;
-
-        // envelope new samples and add them to buffer
-        calcEnvelope(decimated, decSamples);
-        buffer->putSamples(decimated, decSamples);
-    }
-
-    // when the buffer has enought samples for processing...
-    if ((int)buffer->numSamples() > windowLen) 
-    {
-        int processLength;
-
-        // how many samples are processed
-        processLength = (int)buffer->numSamples() - windowLen;
-
-        // ... calculate autocorrelations for oldest samples...
-        updateXCorr(processLength);
-        // ... and remove them from the buffer
-        buffer->receiveSamples(processLength);
-    }
-}
-
-
-
-void BPMDetect::removeBias()
-{
-    int i;
-    float minval = 1e12f;   // arbitrary large number
-
-    for (i = windowStart; i < windowLen; i ++)
-    {
-        if (xcorr[i] < minval)
-        {
-            minval = xcorr[i];
-        }
-    }
-
-    for (i = windowStart; i < windowLen; i ++)
-    {
-        xcorr[i] -= minval;
-    }
-}
-
-
-float BPMDetect::getBpm()
-{
-    double peakPos;
-    double coeff;
-    PeakFinder peakFinder;
-
-    coeff = 60.0 * ((double)sampleRate / (double)decimateBy);
-
-    // save bpm debug analysis data if debug data enabled
-    _SaveDebugData(xcorr, windowStart, windowLen, coeff);
-
-    // remove bias from xcorr data
-    removeBias();
-
-    // find peak position
-    peakPos = peakFinder.detectPeak(xcorr, windowStart, windowLen);
-
-    assert(decimateBy != 0);
-    if (peakPos < 1e-9) return 0.0; // detection failed.
-
-    // calculate BPM
-    return (float) (coeff / peakPos);
-}
+////////////////////////////////////////////////////////////////////////////////
+///
+/// Beats-per-minute (BPM) detection routine.
+///
+/// The beat detection algorithm works as follows:
+/// - Use function 'inputSamples' to input a chunks of samples to the class for
+///   analysis. It's a good idea to enter a large sound file or stream in smallish
+///   chunks of around few kilosamples in order not to extinguish too much RAM memory.
+/// - Inputted sound data is decimated to approx 500 Hz to reduce calculation burden,
+///   which is basically ok as low (bass) frequencies mostly determine the beat rate.
+///   Simple averaging is used for anti-alias filtering because the resulting signal
+///   quality isn't of that high importance.
+/// - Decimated sound data is enveloped, i.e. the amplitude shape is detected by
+///   taking absolute value that's smoothed by sliding average. Signal levels that
+///   are below a couple of times the general RMS amplitude level are cut away to
+///   leave only notable peaks there.
+/// - Repeating sound patterns (e.g. beats) are detected by calculating short-term
+///   autocorrelation function of the enveloped signal.
+/// - After whole sound data file has been analyzed as above, the bpm level is
+///   detected by function 'getBpm' that finds the highest peak of the autocorrelation
+///   function, calculates it's precise location and converts this reading to bpm's.
+///
+/// Author        : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// 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
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#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;
+
+// algorithm input sample block size
+static const int INPUT_BLOCK_SIZE = 2048;
+
+// decimated sample block size
+static const int DECIMATED_BLOCK_SIZE = 256;
+
+/// Target sample rate after decimation
+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
+
+#ifdef _CREATE_BPM_DEBUG_FILE
+
+    static void _SaveDebugData(const char *name, const float *data, int minpos, int maxpos, double coeff)
+    {
+        FILE *fptr = fopen(name, "wt");
+        int i;
+
+        if (fptr)
+        {
+            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]);
+            }
+            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(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) :
+    beat_lpf(_LPF_coeffs)
+{
+    beats.reserve(250); // initial reservation to prevent frequent reallocation
+
+    this->sampleRate = aSampleRate;
+    this->channels = numChannels;
+
+    decimateSum = 0;
+    decimateCount = 0;
+
+    // choose decimation factor so that result is approx. 1000 Hz
+    decimateBy = sampleRate / TARGET_SRATE;
+    if ((decimateBy <= 0) || (decimateBy * DECIMATED_BLOCK_SIZE < INPUT_BLOCK_SIZE))
+    {
+        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_RANGE);
+
+    assert(windowLen > windowStart);
+
+    // allocate new working objects
+    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.
+///
+/// Decimation is used to remove the unnecessary frequencies and thus to reduce
+/// the amount of data needed to be processed as calculating autocorrelation
+/// function is a very-very heavy operation.
+///
+/// Anti-alias filtering is done simply by averaging the samples. This is really a
+/// poor-man's anti-alias filtering, but it's not so critical in this kind of application
+/// (it'd also be difficult to design a high-quality filter with steep cut-off at very
+/// narrow band)
+int BPMDetect::decimate(SAMPLETYPE *dest, const SAMPLETYPE *src, int numsamples)
+{
+    int count, outcount;
+    LONG_SAMPLETYPE out;
+
+    assert(channels > 0);
+    assert(decimateBy > 0);
+    outcount = 0;
+    for (count = 0; count < numsamples; count ++)
+    {
+        int j;
+
+        // convert to mono and accumulate
+        for (j = 0; j < channels; j ++)
+        {
+            decimateSum += src[j];
+        }
+        src += j;
+
+        decimateCount ++;
+        if (decimateCount >= decimateBy)
+        {
+            // Store every Nth sample only
+            out = (LONG_SAMPLETYPE)(decimateSum / (decimateBy * channels));
+            decimateSum = 0;
+            decimateCount = 0;
+#ifdef SOUNDTOUCH_INTEGER_SAMPLES
+            // check ranges for sure (shouldn't actually be necessary)
+            if (out > 32767)
+            {
+                out = 32767;
+            }
+            else if (out < -32768)
+            {
+                out = -32768;
+            }
+#endif // SOUNDTOUCH_INTEGER_SAMPLES
+            dest[outcount] = (SAMPLETYPE)out;
+            outcount ++;
+        }
+    }
+    return outcount;
+}
+
+
+// Calculates autocorrelation function of the sample history buffer
+void BPMDetect::updateXCorr(int process_samples)
+{
+    int offs;
+    SAMPLETYPE *pBuffer;
+
+    assert(buffer->numSamples() >= (uint)(process_samples + windowLen));
+    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 ++)
+    {
+        float sum;
+        int i;
+
+        sum = 0;
+        for (i = 0; i < process_samples; i ++)
+        {
+            sum += tmp[i] * pBuffer[i + offs];  // scaling the sub-result shouldn't be necessary
+        }
+        xcorr[offs] *= xcorr_decay;   // decay 'xcorr' here with suitable time constant.
+
+        xcorr[offs] += (float)fabs(sum);
+    }
+}
+
+
+// Detect individual beat positions
+void BPMDetect::updateBeatPos(int process_samples)
+{
+    SAMPLETYPE *pBuffer;
+
+    assert(buffer->numSamples() >= (uint)(process_samples + windowLen));
+
+    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++)
+    {
+        tmp[i] = hamw2[i] * hamw2[i] * pBuffer[i];
+    }
+
+    #pragma omp parallel for
+    for (int offs = windowStart; offs < windowLen; offs++)
+    {
+        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
+    }
+
+    int skipstep = XCORR_UPDATE_SEQUENCE / OVERLAP_FACTOR;
+
+    // compensate empty buffer at beginning by scaling coefficient
+    float scale = (float)windowLen / (float)(skipstep * init_scaler);
+    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_SIZE];
+
+    // iterate so that max INPUT_BLOCK_SAMPLES processed per iteration
+    while (numSamples > 0)
+    {
+        int block;
+        int decSamples;
+
+        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;
+
+        buffer->putSamples(decimated, decSamples);
+    }
+
+    // when the buffer has enough samples for processing...
+    int req = max(windowLen + XCORR_UPDATE_SEQUENCE, 2 * XCORR_UPDATE_SEQUENCE);
+    while ((int)buffer->numSamples() >= req)
+    {
+        // ... update autocorrelations...
+        updateXCorr(XCORR_UPDATE_SEQUENCE);
+        // ...update beat position calculation...
+        updateBeatPos(XCORR_UPDATE_SEQUENCE / 2);
+        // ... and remove proceessed samples from the buffer
+        int n = XCORR_UPDATE_SEQUENCE / OVERLAP_FACTOR;
+        buffer->receiveSamples(n);
+    }
+}
+
+
+void BPMDetect::removeBias()
+{
+    int i;
+
+    // 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;
+    }
+}
+
+
+// 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;
+
+    // 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(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
+    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

@@ -1,274 +1,275 @@
-////////////////////////////////////////////////////////////////////////////////
-///
-/// A buffer class for temporarily storaging sound samples, operates as a 
-/// first-in-first-out pipe.
-///
-/// Samples are added to the end of the sample buffer with the 'putSamples' 
-/// function, and are received from the beginning of the buffer by calling
-/// the 'receiveSamples' function. The class automatically removes the 
-/// outputted samples from the buffer, as well as grows the buffer size 
-/// whenever necessary.
-///
-/// Author        : Copyright (c) Olli Parviainen
-/// Author e-mail : oparviai 'at' iki.fi
-/// SoundTouch WWW: http://www.surina.net/soundtouch
-///
-////////////////////////////////////////////////////////////////////////////////
-//
-// Last changed  : $Date$
-// File revision : $Revision: 4 $
-//
-// $Id$
-//
-////////////////////////////////////////////////////////////////////////////////
-//
-// License :
-//
-//  SoundTouch audio processing library
-//  Copyright (c) Olli Parviainen
-//
-//  This library is free software; you can redistribute it and/or
-//  modify it under the terms of the GNU Lesser General Public
-//  License as published by the Free Software Foundation; either
-//  version 2.1 of the License, or (at your option) any later version.
-//
-//  This library is distributed in the hope that it will be useful,
-//  but WITHOUT ANY WARRANTY; without even the implied warranty of
-//  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-//  Lesser General Public License for more details.
-//
-//  You should have received a copy of the GNU Lesser General Public
-//  License along with this library; if not, write to the Free Software
-//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
-//
-////////////////////////////////////////////////////////////////////////////////
-
-#include <stdlib.h>
-#include <memory.h>
-#include <string.h>
-#include <assert.h>
-
-#include "FIFOSampleBuffer.h"
-
-using namespace soundtouch;
-
-// Constructor
-FIFOSampleBuffer::FIFOSampleBuffer(int numChannels)
-{
-    assert(numChannels > 0);
-    sizeInBytes = 0; // reasonable initial value
-    buffer = NULL;
-    bufferUnaligned = NULL;
-    samplesInBuffer = 0;
-    bufferPos = 0;
-    channels = (uint)numChannels;
-    ensureCapacity(32);     // allocate initial capacity 
-}
-
-
-// destructor
-FIFOSampleBuffer::~FIFOSampleBuffer()
-{
-    delete[] bufferUnaligned;
-    bufferUnaligned = NULL;
-    buffer = NULL;
-}
-
-
-// Sets number of channels, 1 = mono, 2 = stereo
-void FIFOSampleBuffer::setChannels(int numChannels)
-{
-    uint usedBytes;
-
-    assert(numChannels > 0);
-    usedBytes = channels * samplesInBuffer;
-    channels = (uint)numChannels;
-    samplesInBuffer = usedBytes / channels;
-}
-
-
-// if output location pointer 'bufferPos' isn't zero, 'rewinds' the buffer and
-// zeroes this pointer by copying samples from the 'bufferPos' pointer 
-// location on to the beginning of the buffer.
-void FIFOSampleBuffer::rewind()
-{
-    if (buffer && bufferPos) 
-    {
-        memmove(buffer, ptrBegin(), sizeof(SAMPLETYPE) * channels * samplesInBuffer);
-        bufferPos = 0;
-    }
-}
-
-
-// Adds 'numSamples' pcs of samples from the 'samples' memory position to 
-// the sample buffer.
-void FIFOSampleBuffer::putSamples(const SAMPLETYPE *samples, uint nSamples)
-{
-    memcpy(ptrEnd(nSamples), samples, sizeof(SAMPLETYPE) * nSamples * channels);
-    samplesInBuffer += nSamples;
-}
-
-
-// Increases the number of samples in the buffer without copying any actual
-// samples.
-//
-// This function is used to update the number of samples in the sample buffer
-// when accessing the buffer directly with 'ptrEnd' function. Please be 
-// careful though!
-void FIFOSampleBuffer::putSamples(uint nSamples)
-{
-    uint req;
-
-    req = samplesInBuffer + nSamples;
-    ensureCapacity(req);
-    samplesInBuffer += nSamples;
-}
-
-
-// Returns a pointer to the end of the used part of the sample buffer (i.e. 
-// where the new samples are to be inserted). This function may be used for 
-// inserting new samples into the sample buffer directly. Please be careful! 
-//
-// Parameter 'slackCapacity' tells the function how much free capacity (in
-// terms of samples) there _at least_ should be, in order to the caller to
-// succesfully insert all the required samples to the buffer. When necessary, 
-// the function grows the buffer size to comply with this requirement.
-//
-// When using this function as means for inserting new samples, also remember 
-// to increase the sample count afterwards, by calling  the 
-// 'putSamples(numSamples)' function.
-SAMPLETYPE *FIFOSampleBuffer::ptrEnd(uint slackCapacity) 
-{
-    ensureCapacity(samplesInBuffer + slackCapacity);
-    return buffer + samplesInBuffer * channels;
-}
-
-
-// Returns a pointer to the beginning of the currently non-outputted samples. 
-// This function is provided for accessing the output samples directly. 
-// Please be careful!
-//
-// When using this function to output samples, also remember to 'remove' the
-// outputted samples from the buffer by calling the 
-// 'receiveSamples(numSamples)' function
-SAMPLETYPE *FIFOSampleBuffer::ptrBegin()
-{
-    assert(buffer);
-    return buffer + bufferPos * channels;
-}
-
-
-// Ensures that the buffer has enought capacity, i.e. space for _at least_
-// 'capacityRequirement' number of samples. The buffer is grown in steps of
-// 4 kilobytes to eliminate the need for frequently growing up the buffer,
-// as well as to round the buffer size up to the virtual memory page size.
-void FIFOSampleBuffer::ensureCapacity(uint capacityRequirement)
-{
-    SAMPLETYPE *tempUnaligned, *temp;
-
-    if (capacityRequirement > getCapacity()) 
-    {
-        // enlarge the buffer in 4kbyte steps (round up to next 4k boundary)
-        sizeInBytes = (capacityRequirement * channels * sizeof(SAMPLETYPE) + 4095) & (uint)-4096;
-        assert(sizeInBytes % 2 == 0);
-        tempUnaligned = new SAMPLETYPE[sizeInBytes / sizeof(SAMPLETYPE) + 16 / sizeof(SAMPLETYPE)];
-        if (tempUnaligned == NULL)
-        {
-            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);
-        if (samplesInBuffer)
-        {
-            memcpy(temp, ptrBegin(), samplesInBuffer * channels * sizeof(SAMPLETYPE));
-        }
-        delete[] bufferUnaligned;
-        buffer = temp;
-        bufferUnaligned = tempUnaligned;
-        bufferPos = 0;
-    } 
-    else 
-    {
-        // simply rewind the buffer (if necessary)
-        rewind();
-    }
-}
-
-
-// Returns the current buffer capacity in terms of samples
-uint FIFOSampleBuffer::getCapacity() const
-{
-    return sizeInBytes / (channels * sizeof(SAMPLETYPE));
-}
-
-
-// Returns the number of samples currently in the buffer
-uint FIFOSampleBuffer::numSamples() const
-{
-    return samplesInBuffer;
-}
-
-
-// Output samples from beginning of the sample buffer. Copies demanded number
-// of samples to output and removes them from the sample buffer. If there
-// are less than 'numsample' samples in the buffer, returns all available.
-//
-// Returns number of samples copied.
-uint FIFOSampleBuffer::receiveSamples(SAMPLETYPE *output, uint maxSamples)
-{
-    uint num;
-
-    num = (maxSamples > samplesInBuffer) ? samplesInBuffer : maxSamples;
-
-    memcpy(output, ptrBegin(), channels * sizeof(SAMPLETYPE) * num);
-    return receiveSamples(num);
-}
-
-
-// Removes samples from the beginning of the sample buffer without copying them
-// anywhere. Used to reduce the number of samples in the buffer, when accessing
-// the sample buffer with the 'ptrBegin' function.
-uint FIFOSampleBuffer::receiveSamples(uint maxSamples)
-{
-    if (maxSamples >= samplesInBuffer)
-    {
-        uint temp;
-
-        temp = samplesInBuffer;
-        samplesInBuffer = 0;
-        return temp;
-    }
-
-    samplesInBuffer -= maxSamples;
-    bufferPos += maxSamples;
-
-    return maxSamples;
-}
-
-
-// Returns nonzero if the sample buffer is empty
-int FIFOSampleBuffer::isEmpty() const
-{
-    return (samplesInBuffer == 0) ? 1 : 0;
-}
-
-
-// Clears the sample buffer
-void FIFOSampleBuffer::clear()
-{
-    samplesInBuffer = 0;
-    bufferPos = 0;
-}
-
-
-/// allow trimming (downwards) amount of samples in pipeline.
-/// Returns adjusted amount of samples
-uint FIFOSampleBuffer::adjustAmountOfSamples(uint numSamples)
-{
-    if (numSamples < samplesInBuffer)
-    {
-        samplesInBuffer = numSamples;
-    }
-    return samplesInBuffer;
-}
-
+////////////////////////////////////////////////////////////////////////////////
+///
+/// A buffer class for temporarily storaging sound samples, operates as a
+/// first-in-first-out pipe.
+///
+/// Samples are added to the end of the sample buffer with the 'putSamples'
+/// function, and are received from the beginning of the buffer by calling
+/// the 'receiveSamples' function. The class automatically removes the
+/// outputted samples from the buffer, as well as grows the buffer size
+/// whenever necessary.
+///
+/// Author        : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+//  SoundTouch audio processing library
+//  Copyright (c) Olli Parviainen
+//
+//  This library is free software; you can redistribute it and/or
+//  modify it under the terms of the GNU Lesser General Public
+//  License as published by the Free Software Foundation; either
+//  version 2.1 of the License, or (at your option) any later version.
+//
+//  This library is distributed in the hope that it will be useful,
+//  but WITHOUT ANY WARRANTY; without even the implied warranty of
+//  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+//  Lesser General Public License for more details.
+//
+//  You should have received a copy of the GNU Lesser General Public
+//  License along with this library; if not, write to the Free Software
+//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#include <stdlib.h>
+#include <memory.h>
+#include <string.h>
+#include <assert.h>
+
+#include "FIFOSampleBuffer.h"
+
+using namespace soundtouch;
+
+// Constructor
+FIFOSampleBuffer::FIFOSampleBuffer(int numChannels)
+{
+    assert(numChannels > 0);
+    sizeInBytes = 0; // reasonable initial value
+    buffer = nullptr;
+    bufferUnaligned = nullptr;
+    samplesInBuffer = 0;
+    bufferPos = 0;
+    channels = (uint)numChannels;
+    ensureCapacity(32);     // allocate initial capacity
+}
+
+
+// destructor
+FIFOSampleBuffer::~FIFOSampleBuffer()
+{
+    delete[] bufferUnaligned;
+    bufferUnaligned = nullptr;
+    buffer = nullptr;
+}
+
+
+// Sets number of channels, 1 = mono, 2 = stereo
+void FIFOSampleBuffer::setChannels(int numChannels)
+{
+    uint usedBytes;
+
+    if (!verifyNumberOfChannels(numChannels)) return;
+
+    usedBytes = channels * samplesInBuffer;
+    channels = (uint)numChannels;
+    samplesInBuffer = usedBytes / channels;
+}
+
+
+// if output location pointer 'bufferPos' isn't zero, 'rewinds' the buffer and
+// zeroes this pointer by copying samples from the 'bufferPos' pointer
+// location on to the beginning of the buffer.
+void FIFOSampleBuffer::rewind()
+{
+    if (buffer && bufferPos)
+    {
+        memmove(buffer, ptrBegin(), sizeof(SAMPLETYPE) * channels * samplesInBuffer);
+        bufferPos = 0;
+    }
+}
+
+
+// Adds 'numSamples' pcs of samples from the 'samples' memory position to
+// the sample buffer.
+void FIFOSampleBuffer::putSamples(const SAMPLETYPE *samples, uint nSamples)
+{
+    memcpy(ptrEnd(nSamples), samples, sizeof(SAMPLETYPE) * nSamples * channels);
+    samplesInBuffer += nSamples;
+}
+
+
+// Increases the number of samples in the buffer without copying any actual
+// samples.
+//
+// This function is used to update the number of samples in the sample buffer
+// when accessing the buffer directly with 'ptrEnd' function. Please be
+// careful though!
+void FIFOSampleBuffer::putSamples(uint nSamples)
+{
+    uint req;
+
+    req = samplesInBuffer + nSamples;
+    ensureCapacity(req);
+    samplesInBuffer += nSamples;
+}
+
+
+// Returns a pointer to the end of the used part of the sample buffer (i.e.
+// where the new samples are to be inserted). This function may be used for
+// inserting new samples into the sample buffer directly. Please be careful!
+//
+// Parameter 'slackCapacity' tells the function how much free capacity (in
+// terms of samples) there _at least_ should be, in order to the caller to
+// 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
+// to increase the sample count afterwards, by calling  the
+// 'putSamples(numSamples)' function.
+SAMPLETYPE *FIFOSampleBuffer::ptrEnd(uint slackCapacity)
+{
+    ensureCapacity(samplesInBuffer + slackCapacity);
+    return buffer + samplesInBuffer * channels;
+}
+
+
+// Returns a pointer to the beginning of the currently non-outputted samples.
+// This function is provided for accessing the output samples directly.
+// Please be careful!
+//
+// When using this function to output samples, also remember to 'remove' the
+// outputted samples from the buffer by calling the
+// 'receiveSamples(numSamples)' function
+SAMPLETYPE *FIFOSampleBuffer::ptrBegin()
+{
+    assert(buffer);
+    return buffer + bufferPos * channels;
+}
+
+
+// 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.
+void FIFOSampleBuffer::ensureCapacity(uint capacityRequirement)
+{
+    SAMPLETYPE *tempUnaligned, *temp;
+
+    if (capacityRequirement > getCapacity())
+    {
+        // enlarge the buffer in 4kbyte steps (round up to next 4k boundary)
+        sizeInBytes = (capacityRequirement * channels * sizeof(SAMPLETYPE) + 4095) & (uint)-4096;
+        assert(sizeInBytes % 2 == 0);
+        tempUnaligned = new SAMPLETYPE[sizeInBytes / sizeof(SAMPLETYPE) + 16 / sizeof(SAMPLETYPE)];
+        if (tempUnaligned == 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 *)SOUNDTOUCH_ALIGN_POINTER_16(tempUnaligned);
+        if (samplesInBuffer)
+        {
+            memcpy(temp, ptrBegin(), samplesInBuffer * channels * sizeof(SAMPLETYPE));
+        }
+        delete[] bufferUnaligned;
+        buffer = temp;
+        bufferUnaligned = tempUnaligned;
+        bufferPos = 0;
+    }
+    else
+    {
+        // simply rewind the buffer (if necessary)
+        rewind();
+    }
+}
+
+
+// Returns the current buffer capacity in terms of samples
+uint FIFOSampleBuffer::getCapacity() const
+{
+    return sizeInBytes / (channels * sizeof(SAMPLETYPE));
+}
+
+
+// Returns the number of samples currently in the buffer
+uint FIFOSampleBuffer::numSamples() const
+{
+    return samplesInBuffer;
+}
+
+
+// Output samples from beginning of the sample buffer. Copies demanded number
+// of samples to output and removes them from the sample buffer. If there
+// are less than 'numsample' samples in the buffer, returns all available.
+//
+// Returns number of samples copied.
+uint FIFOSampleBuffer::receiveSamples(SAMPLETYPE *output, uint maxSamples)
+{
+    uint num;
+
+    num = (maxSamples > samplesInBuffer) ? samplesInBuffer : maxSamples;
+
+    memcpy(output, ptrBegin(), channels * sizeof(SAMPLETYPE) * num);
+    return receiveSamples(num);
+}
+
+
+// Removes samples from the beginning of the sample buffer without copying them
+// anywhere. Used to reduce the number of samples in the buffer, when accessing
+// the sample buffer with the 'ptrBegin' function.
+uint FIFOSampleBuffer::receiveSamples(uint maxSamples)
+{
+    if (maxSamples >= samplesInBuffer)
+    {
+        uint temp;
+
+        temp = samplesInBuffer;
+        samplesInBuffer = 0;
+        return temp;
+    }
+
+    samplesInBuffer -= maxSamples;
+    bufferPos += maxSamples;
+
+    return maxSamples;
+}
+
+
+// Returns nonzero if the sample buffer is empty
+int FIFOSampleBuffer::isEmpty() const
+{
+    return (samplesInBuffer == 0) ? 1 : 0;
+}
+
+
+// Clears the sample buffer
+void FIFOSampleBuffer::clear()
+{
+    samplesInBuffer = 0;
+    bufferPos = 0;
+}
+
+
+/// allow trimming (downwards) amount of samples in pipeline.
+/// Returns adjusted amount of samples
+uint FIFOSampleBuffer::adjustAmountOfSamples(uint numSamples)
+{
+    if (numSamples < samplesInBuffer)
+    {
+        samplesInBuffer = numSamples;
+    }
+    return samplesInBuffer;
+}
+
+
+/// 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

@@ -1,259 +1,314 @@
-////////////////////////////////////////////////////////////////////////////////
-///
-/// General FIR digital filter routines with MMX optimization. 
-///
-/// Note : MMX optimized functions reside in a separate, platform-specific file, 
-/// e.g. 'mmx_win.cpp' or 'mmx_gcc.cpp'
-///
-/// Author        : Copyright (c) Olli Parviainen
-/// Author e-mail : oparviai 'at' iki.fi
-/// SoundTouch WWW: http://www.surina.net/soundtouch
-///
-////////////////////////////////////////////////////////////////////////////////
-//
-// Last changed  : $Date$
-// File revision : $Revision: 4 $
-//
-// $Id$
-//
-////////////////////////////////////////////////////////////////////////////////
-//
-// License :
-//
-//  SoundTouch audio processing library
-//  Copyright (c) Olli Parviainen
-//
-//  This library is free software; you can redistribute it and/or
-//  modify it under the terms of the GNU Lesser General Public
-//  License as published by the Free Software Foundation; either
-//  version 2.1 of the License, or (at your option) any later version.
-//
-//  This library is distributed in the hope that it will be useful,
-//  but WITHOUT ANY WARRANTY; without even the implied warranty of
-//  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-//  Lesser General Public License for more details.
-//
-//  You should have received a copy of the GNU Lesser General Public
-//  License along with this library; if not, write to the Free Software
-//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
-//
-////////////////////////////////////////////////////////////////////////////////
-
-#include <memory.h>
-#include <assert.h>
-#include <math.h>
-#include <stdlib.h>
-#include "FIRFilter.h"
-#include "cpu_detect.h"
-
-using namespace soundtouch;
-
-/*****************************************************************************
- *
- * Implementation of the class 'FIRFilter'
- *
- *****************************************************************************/
-
-FIRFilter::FIRFilter()
-{
-    resultDivFactor = 0;
-    resultDivider = 0;
-    length = 0;
-    lengthDiv8 = 0;
-    filterCoeffs = NULL;
-}
-
-
-FIRFilter::~FIRFilter()
-{
-    delete[] filterCoeffs;
-}
-
-// Usual C-version of the filter routine for stereo sound
-uint FIRFilter::evaluateFilterStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples) const
-{
-    uint i, j, end;
-    LONG_SAMPLETYPE suml, sumr;
-#ifdef SOUNDTOUCH_FLOAT_SAMPLES
-    // when using floating point samples, use a scaler instead of a divider
-    // because division is much slower operation than multiplying.
-    double dScaler = 1.0 / (double)resultDivider;
-#endif
-
-    assert(length != 0);
-    assert(src != NULL);
-    assert(dest != NULL);
-    assert(filterCoeffs != NULL);
-
-    end = 2 * (numSamples - length);
-
-    for (j = 0; j < end; j += 2) 
-    {
-        const SAMPLETYPE *ptr;
-
-        suml = sumr = 0;
-        ptr = src + j;
-
-        for (i = 0; i < length; i += 4) 
-        {
-            // loop is unrolled by factor of 4 here for efficiency
-            suml += ptr[2 * i + 0] * filterCoeffs[i + 0] +
-                    ptr[2 * i + 2] * filterCoeffs[i + 1] +
-                    ptr[2 * i + 4] * filterCoeffs[i + 2] +
-                    ptr[2 * i + 6] * filterCoeffs[i + 3];
-            sumr += ptr[2 * i + 1] * filterCoeffs[i + 0] +
-                    ptr[2 * i + 3] * filterCoeffs[i + 1] +
-                    ptr[2 * i + 5] * filterCoeffs[i + 2] +
-                    ptr[2 * i + 7] * filterCoeffs[i + 3];
-        }
-
-#ifdef SOUNDTOUCH_INTEGER_SAMPLES
-        suml >>= resultDivFactor;
-        sumr >>= resultDivFactor;
-        // saturate to 16 bit integer limits
-        suml = (suml < -32768) ? -32768 : (suml > 32767) ? 32767 : suml;
-        // saturate to 16 bit integer limits
-        sumr = (sumr < -32768) ? -32768 : (sumr > 32767) ? 32767 : sumr;
-#else
-        suml *= dScaler;
-        sumr *= dScaler;
-#endif // SOUNDTOUCH_INTEGER_SAMPLES
-        dest[j] = (SAMPLETYPE)suml;
-        dest[j + 1] = (SAMPLETYPE)sumr;
-    }
-    return numSamples - length;
-}
-
-
-
-
-// Usual C-version of the filter routine for mono sound
-uint FIRFilter::evaluateFilterMono(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples) const
-{
-    uint i, j, end;
-    LONG_SAMPLETYPE sum;
-#ifdef SOUNDTOUCH_FLOAT_SAMPLES
-    // when using floating point samples, use a scaler instead of a divider
-    // because division is much slower operation than multiplying.
-    double dScaler = 1.0 / (double)resultDivider;
-#endif
-
-
-    assert(length != 0);
-
-    end = numSamples - length;
-    for (j = 0; j < end; j ++) 
-    {
-        sum = 0;
-        for (i = 0; i < length; i += 4) 
-        {
-            // loop is unrolled by factor of 4 here for efficiency
-            sum += src[i + 0] * filterCoeffs[i + 0] + 
-                   src[i + 1] * filterCoeffs[i + 1] + 
-                   src[i + 2] * filterCoeffs[i + 2] + 
-                   src[i + 3] * filterCoeffs[i + 3];
-        }
-#ifdef 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;
-}
-
-
-// Set filter coeffiecients and length.
-//
-// Throws an exception if filter length isn't divisible by 8
-void FIRFilter::setCoefficients(const SAMPLETYPE *coeffs, uint newLength, uint uResultDivFactor)
-{
-    assert(newLength > 0);
-    if (newLength % 8) ST_THROW_RT_ERROR("FIR filter length not divisible by 8");
-
-    lengthDiv8 = newLength / 8;
-    length = lengthDiv8 * 8;
-    assert(length == newLength);
-
-    resultDivFactor = uResultDivFactor;
-    resultDivider = (SAMPLETYPE)::pow(2.0, (int)resultDivFactor);
-
-    delete[] filterCoeffs;
-    filterCoeffs = new SAMPLETYPE[length];
-    memcpy(filterCoeffs, coeffs, length * sizeof(SAMPLETYPE));
-}
-
-
-uint FIRFilter::getLength() const
-{
-    return length;
-}
-
-
-
-// Applies the filter to the given sequence of samples. 
-//
-// Note : The amount of outputted samples is by value of 'filter_length' 
-// smaller than the amount of input samples.
-uint FIRFilter::evaluate(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples, uint numChannels) const
-{
-    assert(numChannels == 1 || numChannels == 2);
-
-    assert(length > 0);
-    assert(lengthDiv8 * 8 == length);
-    if (numSamples < length) return 0;
-    if (numChannels == 2) 
-    {
-        return evaluateFilterStereo(dest, src, numSamples);
-    } else {
-        return evaluateFilterMono(dest, src, numSamples);
-    }
-}
-
-
-
-// Operator 'new' is overloaded so that it automatically creates a suitable instance 
-// depending on if we've a MMX-capable CPU available or not.
-void * FIRFilter::operator new(size_t s)
-{
-    // Notice! don't use "new FIRFilter" directly, use "newInstance" to create a new instance instead!
-    ST_THROW_RT_ERROR("Error in FIRFilter::new: Don't use 'new FIRFilter', use 'newInstance' member instead!");
-    return newInstance();
-}
-
-
-FIRFilter * FIRFilter::newInstance()
-{
-    uint uExtensions;
-
-    uExtensions = detectCPUextensions();
-
-    // Check if MMX/SSE instruction set extensions supported by CPU
-
-#ifdef SOUNDTOUCH_ALLOW_MMX
-    // MMX routines available only with integer sample types
-    if (uExtensions & SUPPORT_MMX)
-    {
-        return ::new FIRFilterMMX;
-    }
-    else
-#endif // SOUNDTOUCH_ALLOW_MMX
-
-#ifdef SOUNDTOUCH_ALLOW_SSE
-    if (uExtensions & SUPPORT_SSE)
-    {
-        // SSE support
-        return ::new FIRFilterSSE;
-    }
-    else
-#endif // SOUNDTOUCH_ALLOW_SSE
-
-    {
-        // ISA optimizations not supported, use plain C version
-        return ::new FIRFilter;
-    }
-}
+////////////////////////////////////////////////////////////////////////////////
+///
+/// General FIR digital filter routines with MMX optimization.
+///
+/// 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
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+//  SoundTouch audio processing library
+//  Copyright (c) Olli Parviainen
+//
+//  This library is free software; you can redistribute it and/or
+//  modify it under the terms of the GNU Lesser General Public
+//  License as published by the Free Software Foundation; either
+//  version 2.1 of the License, or (at your option) any later version.
+//
+//  This library is distributed in the hope that it will be useful,
+//  but WITHOUT ANY WARRANTY; without even the implied warranty of
+//  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+//  Lesser General Public License for more details.
+//
+//  You should have received a copy of the GNU Lesser General Public
+//  License along with this library; if not, write to the Free Software
+//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#include <memory.h>
+#include <assert.h>
+#include <math.h>
+#include <stdlib.h>
+#include "FIRFilter.h"
+#include "cpu_detect.h"
+
+using namespace soundtouch;
+
+/*****************************************************************************
+ *
+ * Implementation of the class 'FIRFilter'
+ *
+ *****************************************************************************/
+
+FIRFilter::FIRFilter()
+{
+    resultDivFactor = 0;
+    length = 0;
+    lengthDiv8 = 0;
+    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
+{
+    int j, end;
+    // hint compiler autovectorization that loop length is divisible by 8
+    uint ilength = length & -8;
+
+    assert((length != 0) && (length == ilength) && (src != nullptr) && (dest != nullptr) && (filterCoeffs != nullptr));
+    assert(numSamples > ilength);
+
+    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 (uint i = 0; i < ilength; i ++)
+        {
+            suml += ptr[2 * i] * filterCoeffsStereo[2 * i];
+            sumr += ptr[2 * i + 1] * filterCoeffsStereo[2 * i + 1];
+        }
+
+#ifdef SOUNDTOUCH_INTEGER_SAMPLES
+        suml >>= resultDivFactor;
+        sumr >>= resultDivFactor;
+        // saturate to 16 bit integer limits
+        suml = (suml < -32768) ? -32768 : (suml > 32767) ? 32767 : suml;
+        // saturate to 16 bit integer limits
+        sumr = (sumr < -32768) ? -32768 : (sumr > 32767) ? 32767 : sumr;
+#endif // SOUNDTOUCH_INTEGER_SAMPLES
+        dest[j] = (SAMPLETYPE)suml;
+        dest[j + 1] = (SAMPLETYPE)sumr;
+    }
+    return numSamples - ilength;
+}
+
+
+// Usual C-version of the filter routine for mono sound
+uint FIRFilter::evaluateFilterMono(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples) const
+{
+    int j, end;
+
+    // hint compiler autovectorization that loop length is divisible by 8
+    int ilength = length & -8;
+
+    assert(ilength != 0);
+
+    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 < ilength; i ++)
+        {
+            sum += pSrc[i] * filterCoeffs[i];
+        }
+#ifdef SOUNDTOUCH_INTEGER_SAMPLES
+        sum >>= resultDivFactor;
+        // saturate to 16 bit integer limits
+        sum = (sum < -32768) ? -32768 : (sum > 32767) ? 32767 : sum;
+#endif // SOUNDTOUCH_INTEGER_SAMPLES
+        dest[j] = (SAMPLETYPE)sum;
+    }
+    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
+void FIRFilter::setCoefficients(const SAMPLETYPE *coeffs, uint newLength, uint uResultDivFactor)
+{
+    assert(newLength > 0);
+    if (newLength % 8) ST_THROW_RT_ERROR("FIR filter length not divisible by 8");
+
+    lengthDiv8 = newLength / 8;
+    length = lengthDiv8 * 8;
+    assert(length == newLength);
+
+    resultDivFactor = uResultDivFactor;
+
+    delete[] filterCoeffs;
+    filterCoeffs = new SAMPLETYPE[length];
+    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);
+    }
+}
+
+
+uint FIRFilter::getLength() const
+{
+    return length;
+}
+
+
+// Applies the filter to the given sequence of samples.
+//
+// Note : The amount of outputted samples is by value of 'filter_length'
+// smaller than the amount of input samples.
+uint FIRFilter::evaluate(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples, uint numChannels)
+{
+    assert(length > 0);
+    assert(lengthDiv8 * 8 == length);
+
+    if (numSamples < length) return 0;
+
+#ifndef USE_MULTICH_ALWAYS
+    if (numChannels == 1)
+    {
+        return evaluateFilterMono(dest, src, numSamples);
+    }
+    else if (numChannels == 2)
+    {
+        return evaluateFilterStereo(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)
+{
+    // 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!");
+    return newInstance();
+}
+
+
+FIRFilter * FIRFilter::newInstance()
+{
+    uint uExtensions;
+
+    uExtensions = detectCPUextensions();
+    (void)uExtensions;
+
+    // Check if MMX/SSE instruction set extensions supported by CPU
+
+#ifdef SOUNDTOUCH_ALLOW_MMX
+    // MMX routines available only with integer sample types
+    if (uExtensions & SUPPORT_MMX)
+    {
+        return ::new FIRFilterMMX;
+    }
+    else
+#endif // SOUNDTOUCH_ALLOW_MMX
+
+#ifdef SOUNDTOUCH_ALLOW_SSE
+    if (uExtensions & SUPPORT_SSE)
+    {
+        // SSE support
+        return ::new FIRFilterSSE;
+    }
+    else
+#endif // SOUNDTOUCH_ALLOW_SSE
+
+    {
+        // ISA optimizations not supported, use plain C version
+        return ::new FIRFilter;
+    }
+}

source/SoundTouch/FIRFilter.h

@@ -1,145 +1,137 @@
-////////////////////////////////////////////////////////////////////////////////
-///
-/// General FIR digital filter routines with MMX optimization. 
-///
-/// Note : MMX optimized functions reside in a separate, platform-specific file, 
-/// e.g. 'mmx_win.cpp' or 'mmx_gcc.cpp'
-///
-/// Author        : Copyright (c) Olli Parviainen
-/// Author e-mail : oparviai 'at' iki.fi
-/// SoundTouch WWW: http://www.surina.net/soundtouch
-///
-////////////////////////////////////////////////////////////////////////////////
-//
-// Last changed  : $Date$
-// File revision : $Revision: 4 $
-//
-// $Id$
-//
-////////////////////////////////////////////////////////////////////////////////
-//
-// License :
-//
-//  SoundTouch audio processing library
-//  Copyright (c) Olli Parviainen
-//
-//  This library is free software; you can redistribute it and/or
-//  modify it under the terms of the GNU Lesser General Public
-//  License as published by the Free Software Foundation; either
-//  version 2.1 of the License, or (at your option) any later version.
-//
-//  This library is distributed in the hope that it will be useful,
-//  but WITHOUT ANY WARRANTY; without even the implied warranty of
-//  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-//  Lesser General Public License for more details.
-//
-//  You should have received a copy of the GNU Lesser General Public
-//  License along with this library; if not, write to the Free Software
-//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
-//
-////////////////////////////////////////////////////////////////////////////////
-
-#ifndef FIRFilter_H
-#define FIRFilter_H
-
-#include <stddef.h>
-#include "STTypes.h"
-
-namespace soundtouch
-{
-
-class FIRFilter 
-{
-protected:
-    // Number of FIR filter taps
-    uint length;    
-    // Number of FIR filter taps divided by 8
-    uint lengthDiv8;
-
-    // Result divider factor in 2^k format
-    uint resultDivFactor;
-
-    // Result divider value.
-    SAMPLETYPE resultDivider;
-
-    // Memory for filter coefficients
-    SAMPLETYPE *filterCoeffs;
-
-    virtual uint evaluateFilterStereo(SAMPLETYPE *dest, 
-                                      const SAMPLETYPE *src, 
-                                      uint numSamples) const;
-    virtual uint evaluateFilterMono(SAMPLETYPE *dest, 
-                                    const SAMPLETYPE *src, 
-                                    uint numSamples) const;
-
-public:
-    FIRFilter();
-    virtual ~FIRFilter();
-
-    /// Operator 'new' is overloaded so that it automatically creates a suitable instance 
-    /// depending on if we've a MMX-capable CPU available or not.
-    static void * operator new(size_t s);
-
-    static FIRFilter *newInstance();
-
-    /// Applies the filter to the given sequence of samples. 
-    /// Note : The amount of outputted samples is by value of 'filter_length' 
-    /// smaller than the amount of input samples.
-    ///
-    /// \return Number of samples copied to 'dest'.
-    uint evaluate(SAMPLETYPE *dest, 
-                  const SAMPLETYPE *src, 
-                  uint numSamples, 
-                  uint numChannels) const;
-
-    uint getLength() const;
-
-    virtual void setCoefficients(const SAMPLETYPE *coeffs, 
-                                 uint newLength, 
-                                 uint uResultDivFactor);
-};
-
-
-// Optional subclasses that implement CPU-specific optimizations:
-
-#ifdef SOUNDTOUCH_ALLOW_MMX
-
-/// Class that implements MMX optimized functions exclusive for 16bit integer samples type.
-    class FIRFilterMMX : public FIRFilter
-    {
-    protected:
-        short *filterCoeffsUnalign;
-        short *filterCoeffsAlign;
-
-        virtual uint evaluateFilterStereo(short *dest, const short *src, uint numSamples) const;
-    public:
-        FIRFilterMMX();
-        ~FIRFilterMMX();
-
-        virtual void setCoefficients(const short *coeffs, uint newLength, uint uResultDivFactor);
-    };
-
-#endif // SOUNDTOUCH_ALLOW_MMX
-
-
-#ifdef SOUNDTOUCH_ALLOW_SSE
-    /// Class that implements SSE optimized functions exclusive for floating point samples type.
-    class FIRFilterSSE : public FIRFilter
-    {
-    protected:
-        float *filterCoeffsUnalign;
-        float *filterCoeffsAlign;
-
-        virtual uint evaluateFilterStereo(float *dest, const float *src, uint numSamples) const;
-    public:
-        FIRFilterSSE();
-        ~FIRFilterSSE();
-
-        virtual void setCoefficients(const float *coeffs, uint newLength, uint uResultDivFactor);
-    };
-
-#endif // SOUNDTOUCH_ALLOW_SSE
-
-}
-
-#endif  // FIRFilter_H
+////////////////////////////////////////////////////////////////////////////////
+///
+/// General FIR digital filter routines with MMX optimization.
+///
+/// Note : MMX optimized functions reside in a separate, platform-specific file,
+/// e.g. 'mmx_win.cpp' or 'mmx_gcc.cpp'
+///
+/// Author        : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+//  SoundTouch audio processing library
+//  Copyright (c) Olli Parviainen
+//
+//  This library is free software; you can redistribute it and/or
+//  modify it under the terms of the GNU Lesser General Public
+//  License as published by the Free Software Foundation; either
+//  version 2.1 of the License, or (at your option) any later version.
+//
+//  This library is distributed in the hope that it will be useful,
+//  but WITHOUT ANY WARRANTY; without even the implied warranty of
+//  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+//  Lesser General Public License for more details.
+//
+//  You should have received a copy of the GNU Lesser General Public
+//  License along with this library; if not, write to the Free Software
+//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#ifndef FIRFilter_H
+#define FIRFilter_H
+
+#include <stddef.h>
+#include "STTypes.h"
+
+namespace soundtouch
+{
+
+class FIRFilter
+{
+protected:
+    // Number of FIR filter taps
+    uint length;
+    // Number of FIR filter taps divided by 8
+    uint lengthDiv8;
+
+    // Result divider factor in 2^k format
+    uint resultDivFactor;
+
+    // Memory for filter coefficients
+    SAMPLETYPE *filterCoeffs;
+    SAMPLETYPE *filterCoeffsStereo;
+
+    virtual uint evaluateFilterStereo(SAMPLETYPE *dest,
+                                      const SAMPLETYPE *src,
+                                      uint numSamples) const;
+    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();
+    virtual ~FIRFilter();
+
+    /// Operator 'new' is overloaded so that it automatically creates a suitable instance
+    /// depending on if we've a MMX-capable CPU available or not.
+    static void * operator new(size_t s);
+
+    static FIRFilter *newInstance();
+
+    /// Applies the filter to the given sequence of samples.
+    /// Note : The amount of outputted samples is by value of 'filter_length'
+    /// smaller than the amount of input samples.
+    ///
+    /// \return Number of samples copied to 'dest'.
+    uint evaluate(SAMPLETYPE *dest,
+                  const SAMPLETYPE *src,
+                  uint numSamples,
+                  uint numChannels);
+
+    uint getLength() const;
+
+    virtual void setCoefficients(const SAMPLETYPE *coeffs,
+                                 uint newLength,
+                                 uint uResultDivFactor);
+};
+
+
+// Optional subclasses that implement CPU-specific optimizations:
+
+#ifdef SOUNDTOUCH_ALLOW_MMX
+
+/// Class that implements MMX optimized functions exclusive for 16bit integer samples type.
+    class FIRFilterMMX : public FIRFilter
+    {
+    protected:
+        short *filterCoeffsUnalign;
+        short *filterCoeffsAlign;
+
+        virtual uint evaluateFilterStereo(short *dest, const short *src, uint numSamples) const override;
+    public:
+        FIRFilterMMX();
+        ~FIRFilterMMX();
+
+        virtual void setCoefficients(const short *coeffs, uint newLength, uint uResultDivFactor) override;
+    };
+
+#endif // SOUNDTOUCH_ALLOW_MMX
+
+
+#ifdef SOUNDTOUCH_ALLOW_SSE
+    /// Class that implements SSE optimized functions exclusive for floating point samples type.
+    class FIRFilterSSE : public FIRFilter
+    {
+    protected:
+        float *filterCoeffsUnalign;
+        float *filterCoeffsAlign;
+
+        virtual uint evaluateFilterStereo(float *dest, const float *src, uint numSamples) const override;
+    public:
+        FIRFilterSSE();
+        ~FIRFilterSSE();
+
+        virtual void setCoefficients(const float *coeffs, uint newLength, uint uResultDivFactor) override;
+    };
+
+#endif // SOUNDTOUCH_ALLOW_SSE
+
+}
+
+#endif  // FIRFilter_H

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

@@ -1,71 +1,74 @@
-## 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
-## terms of the GNU General Public License as published by the Free Software
-## Foundation; either version 2 of the License, or (at your option) any later
-## version.
-## 
-## SoundTouch is distributed in the hope that it will be useful, but WITHOUT ANY
-## WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
-## A PARTICULAR PURPOSE.  See the GNU General Public License for more details.
-## 
-## You should have received a copy of the GNU General Public License along with
-## this program; if not, write to the Free Software Foundation, Inc., 59 Temple
-## Place - Suite 330, Boston, MA  02111-1307, USA
-
-
-include $(top_srcdir)/config/am_include.mk
-
-
-# set to something if you want other stuff to be included in the distribution tarball
-EXTRA_DIST=SoundTouch.dsp SoundTouch.dsw SoundTouch.sln SoundTouch.vcproj
-
-noinst_HEADERS=AAFilter.h cpu_detect.h cpu_detect_x86.cpp FIRFilter.h RateTransposer.h TDStretch.h PeakFinder.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
-
-
-# Compiler flags
-AM_CXXFLAGS=-O3 -fcheck-new -I../../include
-
-# Compile the files that need MMX and SSE individually.
-libSoundTouch_la_LIBADD=libSoundTouchMMX.la libSoundTouchSSE.la
-noinst_LTLIBRARIES=libSoundTouchMMX.la libSoundTouchSSE.la
-libSoundTouchMMX_la_SOURCES=mmx_optimized.cpp
-libSoundTouchSSE_la_SOURCES=sse_optimized.cpp
-
-# We enable optimizations by default.
-# If MMX is supported compile with -mmmx.
-# Do not assume -msse is also supported.
-if HAVE_MMX
-libSoundTouchMMX_la_CXXFLAGS = -mmmx $(AM_CXXFLAGS)
-else
-libSoundTouchMMX_la_CXXFLAGS = $(AM_CXXFLAGS)
-endif
-
-# We enable optimizations by default.
-# If SSE is supported compile with -msse.
-if HAVE_SSE
-libSoundTouchSSE_la_CXXFLAGS = -msse $(AM_CXXFLAGS)
-else
-libSoundTouchSSE_la_CXXFLAGS = $(AM_CXXFLAGS)
-endif
-
-# Let the user disable optimizations if he wishes to.
-if !X86_OPTIMIZATIONS
-libSoundTouchMMX_la_CXXFLAGS = $(AM_CXXFLAGS)
-libSoundTouchSSE_la_CXXFLAGS = $(AM_CXXFLAGS)
-endif
-
-
-# other linking flags to add
-# noinst_LTLIBRARIES = libSoundTouchOpt.la
-# libSoundTouch_la_LIBADD = libSoundTouchOpt.la
-# libSoundTouchOpt_la_SOURCES = mmx_optimized.cpp sse_optimized.cpp 
-# libSoundTouchOpt_la_CXXFLAGS = -O3 -msse -fcheck-new -I../../include
+## Process this file with automake to create Makefile.in
+##
+## This file is part of SoundTouch, an audio processing library for pitch/time adjustments
+##
+## SoundTouch is free software; you can redistribute it and/or modify it under the
+## terms of the GNU General Public License as published by the Free Software
+## Foundation; either version 2 of the License, or (at your option) any later
+## version.
+##
+## SoundTouch is distributed in the hope that it will be useful, but WITHOUT ANY
+## WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
+## A PARTICULAR PURPOSE.  See the GNU General Public License for more details.
+##
+## You should have received a copy of the GNU General Public License along with
+## this program; if not, write to the Free Software Foundation, Inc., 59 Temple
+## Place - Suite 330, Boston, MA  02111-1307, USA
+
+
+include $(top_srcdir)/config/am_include.mk
+
+
+# set to something if you want other stuff to be included in the distribution tarball
+EXTRA_DIST=SoundTouch.sln SoundTouch.vcxproj
+
+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 InterpolateLinear.cpp InterpolateCubic.cpp \
+    InterpolateShannon.cpp
+
+# Compiler flags
+#AM_CXXFLAGS+=
+
+# Compile the files that need MMX and SSE individually.
+libSoundTouch_la_LIBADD=libSoundTouchMMX.la libSoundTouchSSE.la
+noinst_LTLIBRARIES=libSoundTouchMMX.la libSoundTouchSSE.la
+libSoundTouchMMX_la_SOURCES=mmx_optimized.cpp
+libSoundTouchSSE_la_SOURCES=sse_optimized.cpp
+
+# We enable optimizations by default.
+# If MMX is supported compile with -mmmx.
+# Do not assume -msse is also supported.
+if HAVE_MMX
+libSoundTouchMMX_la_CXXFLAGS = -mmmx $(AM_CXXFLAGS)
+else
+libSoundTouchMMX_la_CXXFLAGS = $(AM_CXXFLAGS)
+endif
+
+# We enable optimizations by default.
+# If SSE is supported compile with -msse.
+if HAVE_SSE
+libSoundTouchSSE_la_CXXFLAGS = -msse $(AM_CXXFLAGS)
+else
+libSoundTouchSSE_la_CXXFLAGS = $(AM_CXXFLAGS)
+endif
+
+# Let the user disable optimizations if he wishes to.
+if !X86_OPTIMIZATIONS
+libSoundTouchMMX_la_CXXFLAGS = $(AM_CXXFLAGS)
+libSoundTouchSSE_la_CXXFLAGS = $(AM_CXXFLAGS)
+endif
+
+# 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
+# libSoundTouch_la_LIBADD = libSoundTouchOpt.la
+# libSoundTouchOpt_la_SOURCES = mmx_optimized.cpp sse_optimized.cpp
+# libSoundTouchOpt_la_CXXFLAGS = -O3 -msse -fcheck-new -I../../include

source/SoundTouch/PeakFinder.cpp

@@ -1,278 +1,277 @@
-////////////////////////////////////////////////////////////////////////////////
-///
-/// Peak detection routine. 
-///
-/// The routine detects highest value on an array of values and calculates the 
-/// precise peak location as a mass-center of the 'hump' around the peak value.
-///
-/// Author        : Copyright (c) Olli Parviainen
-/// Author e-mail : oparviai 'at' iki.fi
-/// SoundTouch WWW: http://www.surina.net/soundtouch
-///
-////////////////////////////////////////////////////////////////////////////////
-//
-// Last changed  : $Date$
-// File revision : $Revision: 4 $
-//
-// $Id$
-//
-////////////////////////////////////////////////////////////////////////////////
-//
-// License :
-//
-//  SoundTouch audio processing library
-//  Copyright (c) Olli Parviainen
-//
-//  This library is free software; you can redistribute it and/or
-//  modify it under the terms of the GNU Lesser General Public
-//  License as published by the Free Software Foundation; either
-//  version 2.1 of the License, or (at your option) any later version.
-//
-//  This library is distributed in the hope that it will be useful,
-//  but WITHOUT ANY WARRANTY; without even the implied warranty of
-//  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-//  Lesser General Public License for more details.
-//
-//  You should have received a copy of the GNU Lesser General Public
-//  License along with this library; if not, write to the Free Software
-//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
-//
-////////////////////////////////////////////////////////////////////////////////
-
-#include <math.h>
-#include <assert.h>
-
-#include "PeakFinder.h"
-
-using namespace soundtouch;
-
-#define max(x, y) (((x) > (y)) ? (x) : (y))
-
-
-PeakFinder::PeakFinder()
-{
-    minPos = maxPos = 0;
-}
-
-
-// Finds real 'top' of a peak hump from neighnourhood of the given 'peakpos'.
-int PeakFinder::findTop(const float *data, int peakpos) const
-{
-    int i;
-    int start, end;
-    float refvalue;
-
-    refvalue = data[peakpos];
-
-    // seek within ±10 points
-    start = peakpos - 10;
-    if (start < minPos) start = minPos;
-    end = peakpos + 10;
-    if (end > maxPos) end = maxPos;
-
-    for (i = start; i <= end; i ++)
-    {
-        if (data[i] > refvalue)
-        {
-            peakpos = i;
-            refvalue = data[i];
-        }
-    }
-
-    // failure if max value is at edges of seek range => it's not peak, it's at slope.
-    if ((peakpos == start) || (peakpos == end)) return 0;
-
-    return peakpos;
-}
-
-
-// Finds 'ground level' of a peak hump by starting from 'peakpos' and proceeding
-// to direction defined by 'direction' until next 'hump' after minimum value will 
-// begin
-int PeakFinder::findGround(const float *data, int peakpos, int direction) const
-{
-    int lowpos;
-    int pos;
-    int climb_count;
-    float refvalue;
-    float delta;
-
-    climb_count = 0;
-    refvalue = data[peakpos];
-    lowpos = peakpos;
-
-    pos = peakpos;
-
-    while ((pos > minPos+1) && (pos < maxPos-1))
-    {
-        int prevpos;
-
-        prevpos = pos;
-        pos += direction;
-
-        // calculate derivate
-        delta = data[pos] - data[prevpos];
-        if (delta <= 0)
-        {
-            // going downhill, ok
-            if (climb_count)
-            {
-                climb_count --;  // decrease climb count
-            }
-
-            // check if new minimum found
-            if (data[pos] < refvalue)
-            {
-                // new minimum found
-                lowpos = pos;
-                refvalue = data[pos];
-            }
-        }
-        else
-        {
-            // going uphill, increase climbing counter
-            climb_count ++;
-            if (climb_count > 5) break;    // we've been climbing too long => it's next uphill => quit
-        }
-    }
-    return lowpos;
-}
-
-
-// Find offset where the value crosses the given level, when starting from 'peakpos' and
-// proceeds to direction defined in 'direction'
-int PeakFinder::findCrossingLevel(const float *data, float level, int peakpos, int direction) const
-{
-    float peaklevel;
-    int pos;
-
-    peaklevel = data[peakpos];
-    assert(peaklevel >= level);
-    pos = peakpos;
-    while ((pos >= minPos) && (pos < maxPos))
-    {
-        if (data[pos + direction] < level) return pos;   // crossing found
-        pos += direction;
-    }
-    return -1;  // not found
-}
-
-
-// Calculates the center of mass location of 'data' array items between 'firstPos' and 'lastPos'
-double PeakFinder::calcMassCenter(const float *data, int firstPos, int lastPos) const
-{
-    int i;
-    float sum;
-    float wsum;
-
-    sum = 0;
-    wsum = 0;
-    for (i = firstPos; i <= lastPos; i ++)
-    {
-        sum += (float)i * data[i];
-        wsum += data[i];
-    }
-
-    if (wsum < 1e-6) return 0;
-    return sum / wsum;
-}
-
-
-
-/// get exact center of peak near given position by calculating local mass of center
-double PeakFinder::getPeakCenter(const float *data, int peakpos) const
-{
-    float peakLevel;            // peak level
-    int crosspos1, crosspos2;   // position where the peak 'hump' crosses cutting level
-    float cutLevel;             // cutting value
-    float groundLevel;          // ground level of the peak
-    int gp1, gp2;               // bottom positions of the peak 'hump'
-
-    // find ground positions.
-    gp1 = findGround(data, peakpos, -1);
-    gp2 = findGround(data, peakpos, 1);
-
-    groundLevel = 0.5f * (data[gp1] + data[gp2]);
-    peakLevel = data[peakpos];
-
-    // calculate 70%-level of the peak
-    cutLevel = 0.70f * peakLevel + 0.30f * groundLevel;
-    // find mid-level crossings
-    crosspos1 = findCrossingLevel(data, cutLevel, peakpos, -1);
-    crosspos2 = findCrossingLevel(data, cutLevel, peakpos, 1);
-
-    if ((crosspos1 < 0) || (crosspos2 < 0)) return 0;   // no crossing, no peak..
-
-    // calculate mass center of the peak surroundings
-    return calcMassCenter(data, crosspos1, crosspos2);
-}
-
-
-
-double PeakFinder::detectPeak(const float *data, int aminPos, int amaxPos) 
-{
-
-    int i;
-    int peakpos;                // position of peak level
-    double highPeak, peak;
-
-    this->minPos = aminPos;
-    this->maxPos = amaxPos;
-
-    // find absolute peak
-    peakpos = minPos;
-    peak = data[minPos];
-    for (i = minPos + 1; i < maxPos; i ++)
-    {
-        if (data[i] > peak) 
-        {
-            peak = data[i];
-            peakpos = i;
-        }
-    }
-    
-    // Calculate exact location of the highest peak mass center
-    highPeak = getPeakCenter(data, peakpos);
-    peak = highPeak;
-
-    // Now check if the highest peak were in fact harmonic of the true base beat peak 
-    // - sometimes the highest peak can be Nth harmonic of the true base peak yet 
-    // just a slightly higher than the true base
-
-    int hp = (int)(highPeak + 0.5);
-
-    for (i = 3; i < 10; i ++)
-    {
-        double peaktmp, harmonic;
-        int i1,i2;
-
-        harmonic = (double)i * 0.5;
-        peakpos = (int)(highPeak / harmonic + 0.5f);
-        if (peakpos < minPos) break;
-        peakpos = findTop(data, peakpos);   // seek true local maximum index
-        if (peakpos == 0) continue;         // no local max here
-
-        // calculate mass-center of possible harmonic peak
-        peaktmp = getPeakCenter(data, peakpos);
-
-        // accept harmonic peak if 
-        // (a) it is found
-        // (b) is within ±4% of the expected harmonic interval
-        // (c) has at least half x-corr value of the max. peak
-
-        double diff = harmonic * peaktmp / highPeak;
-        if ((diff < 0.96) || (diff > 1.04)) continue;   // peak too afar from expected
-
-        // now compare to highest detected peak
-        i1 = (int)(highPeak + 0.5);
-        i2 = (int)(peaktmp + 0.5);
-        if (data[i2] >= 0.4*data[i1])
-        {
-            // The harmonic is at least half as high primary peak,
-            // thus use the harmonic peak instead
-            peak = peaktmp;
-        }
-    }
-
-    return peak;
-}
+////////////////////////////////////////////////////////////////////////////////
+///
+/// Peak detection routine.
+///
+/// The routine detects highest value on an array of values and calculates the
+/// precise peak location as a mass-center of the 'hump' around the peak value.
+///
+/// Author        : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+//  SoundTouch audio processing library
+//  Copyright (c) Olli Parviainen
+//
+//  This library is free software; you can redistribute it and/or
+//  modify it under the terms of the GNU Lesser General Public
+//  License as published by the Free Software Foundation; either
+//  version 2.1 of the License, or (at your option) any later version.
+//
+//  This library is distributed in the hope that it will be useful,
+//  but WITHOUT ANY WARRANTY; without even the implied warranty of
+//  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+//  Lesser General Public License for more details.
+//
+//  You should have received a copy of the GNU Lesser General Public
+//  License along with this library; if not, write to the Free Software
+//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#include <math.h>
+#include <assert.h>
+
+#include "PeakFinder.h"
+
+using namespace soundtouch;
+
+#define max(x, y) (((x) > (y)) ? (x) : (y))
+
+
+PeakFinder::PeakFinder()
+{
+    minPos = maxPos = 0;
+}
+
+
+// Finds real 'top' of a peak hump from neighnourhood of the given 'peakpos'.
+int PeakFinder::findTop(const float *data, int peakpos) const
+{
+    int i;
+    int start, end;
+    float refvalue;
+
+    refvalue = data[peakpos];
+
+    // seek within ±10 points
+    start = peakpos - 10;
+    if (start < minPos) start = minPos;
+    end = peakpos + 10;
+    if (end > maxPos) end = maxPos;
+
+    for (i = start; i <= end; i ++)
+    {
+        if (data[i] > refvalue)
+        {
+            peakpos = i;
+            refvalue = data[i];
+        }
+    }
+
+    // failure if max value is at edges of seek range => it's not peak, it's at slope.
+    if ((peakpos == start) || (peakpos == end)) return 0;
+
+    return peakpos;
+}
+
+
+// Finds 'ground level' of a peak hump by starting from 'peakpos' and proceeding
+// to direction defined by 'direction' until next 'hump' after minimum value will
+// begin
+int PeakFinder::findGround(const float *data, int peakpos, int direction) const
+{
+    int lowpos;
+    int pos;
+    int climb_count;
+    float refvalue;
+    float delta;
+
+    climb_count = 0;
+    refvalue = data[peakpos];
+    lowpos = peakpos;
+
+    pos = peakpos;
+
+    while ((pos > minPos+1) && (pos < maxPos-1))
+    {
+        int prevpos;
+
+        prevpos = pos;
+        pos += direction;
+
+        // calculate derivate
+        delta = data[pos] - data[prevpos];
+        if (delta <= 0)
+        {
+            // going downhill, ok
+            if (climb_count)
+            {
+                climb_count --;  // decrease climb count
+            }
+
+            // check if new minimum found
+            if (data[pos] < refvalue)
+            {
+                // new minimum found
+                lowpos = pos;
+                refvalue = data[pos];
+            }
+        }
+        else
+        {
+            // going uphill, increase climbing counter
+            climb_count ++;
+            if (climb_count > 5) break;    // we've been climbing too long => it's next uphill => quit
+        }
+    }
+    return lowpos;
+}
+
+
+// Find offset where the value crosses the given level, when starting from 'peakpos' and
+// proceeds to direction defined in 'direction'
+int PeakFinder::findCrossingLevel(const float *data, float level, int peakpos, int direction) const
+{
+    float peaklevel;
+    int pos;
+
+    peaklevel = data[peakpos];
+    assert(peaklevel >= level);
+    pos = peakpos;
+    while ((pos >= minPos) && (pos + direction < maxPos))
+    {
+        if (data[pos + direction] < level) return pos;   // crossing found
+        pos += direction;
+    }
+    return -1;  // not found
+}
+
+
+// Calculates the center of mass location of 'data' array items between 'firstPos' and 'lastPos'
+double PeakFinder::calcMassCenter(const float *data, int firstPos, int lastPos) const
+{
+    int i;
+    float sum;
+    float wsum;
+
+    sum = 0;
+    wsum = 0;
+    for (i = firstPos; i <= lastPos; i ++)
+    {
+        sum += (float)i * data[i];
+        wsum += data[i];
+    }
+
+    if (wsum < 1e-6) return 0;
+    return sum / wsum;
+}
+
+
+/// get exact center of peak near given position by calculating local mass of center
+double PeakFinder::getPeakCenter(const float *data, int peakpos) const
+{
+    float peakLevel;            // peak level
+    int crosspos1, crosspos2;   // position where the peak 'hump' crosses cutting level
+    float cutLevel;             // cutting value
+    float groundLevel;          // ground level of the peak
+    int gp1, gp2;               // bottom positions of the peak 'hump'
+
+    // find ground positions.
+    gp1 = findGround(data, peakpos, -1);
+    gp2 = findGround(data, peakpos, 1);
+
+    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);
+
+    if ((crosspos1 < 0) || (crosspos2 < 0)) return 0;   // no crossing, no peak..
+
+    // calculate mass center of the peak surroundings
+    return calcMassCenter(data, crosspos1, crosspos2);
+}
+
+
+double PeakFinder::detectPeak(const float *data, int aminPos, int amaxPos)
+{
+
+    int i;
+    int peakpos;                // position of peak level
+    double highPeak, peak;
+
+    this->minPos = aminPos;
+    this->maxPos = amaxPos;
+
+    // find absolute peak
+    peakpos = minPos;
+    peak = data[minPos];
+    for (i = minPos + 1; i < maxPos; i ++)
+    {
+        if (data[i] > peak)
+        {
+            peak = data[i];
+            peakpos = i;
+        }
+    }
+
+    // Calculate exact location of the highest peak mass center
+    highPeak = getPeakCenter(data, peakpos);
+    peak = highPeak;
+
+    // Now check if the highest peak were in fact harmonic of the true base beat peak
+    // - sometimes the highest peak can be Nth harmonic of the true base peak yet
+    // just a slightly higher than the true base
+
+    for (i = 1; i < 3; i ++)
+    {
+        double peaktmp, harmonic;
+        int i1,i2;
+
+        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
+        if (peakpos == 0) continue;         // no local max here
+
+        // calculate mass-center of possible harmonic peak
+        peaktmp = getPeakCenter(data, peakpos);
+
+        // accept harmonic peak if
+        // (a) it is found
+        // (b) is within ±4% of the expected harmonic interval
+        // (c) has at least half x-corr value of the max. peak
+
+        double diff = harmonic * peaktmp / highPeak;
+        if ((diff < 0.96) || (diff > 1.04)) continue;   // peak too afar from expected
+
+        // now compare to highest detected peak
+        i1 = (int)(highPeak + 0.5);
+        i2 = (int)(peaktmp + 0.5);
+        if (data[i2] >= 0.4*data[i1])
+        {
+            // The harmonic is at least half as high primary peak,
+            // thus use the harmonic peak instead
+            peak = peaktmp;
+        }
+    }
+
+    return peak;
+}

source/SoundTouch/PeakFinder.h

@@ -1,97 +1,90 @@
-////////////////////////////////////////////////////////////////////////////////
-///
-/// The routine detects highest value on an array of values and calculates the 
-/// precise peak location as a mass-center of the 'hump' around the peak value.
-///
-/// Author        : Copyright (c) Olli Parviainen
-/// Author e-mail : oparviai 'at' iki.fi
-/// SoundTouch WWW: http://www.surina.net/soundtouch
-///
-////////////////////////////////////////////////////////////////////////////////
-//
-// Last changed  : $Date$
-// File revision : $Revision: 4 $
-//
-// $Id$
-//
-////////////////////////////////////////////////////////////////////////////////
-//
-// License :
-//
-//  SoundTouch audio processing library
-//  Copyright (c) Olli Parviainen
-//
-//  This library is free software; you can redistribute it and/or
-//  modify it under the terms of the GNU Lesser General Public
-//  License as published by the Free Software Foundation; either
-//  version 2.1 of the License, or (at your option) any later version.
-//
-//  This library is distributed in the hope that it will be useful,
-//  but WITHOUT ANY WARRANTY; without even the implied warranty of
-//  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-//  Lesser General Public License for more details.
-//
-//  You should have received a copy of the GNU Lesser General Public
-//  License along with this library; if not, write to the Free Software
-//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
-//
-////////////////////////////////////////////////////////////////////////////////
-
-#ifndef _PeakFinder_H_
-#define _PeakFinder_H_
-
-namespace soundtouch
-{
-
-class PeakFinder
-{
-protected:
-    /// Min, max allowed peak positions within the data vector
-    int minPos, maxPos;
-
-    /// Calculates the mass center between given vector items.
-    double calcMassCenter(const float *data, ///< Data vector.
-                         int firstPos,      ///< Index of first vector item beloging to the peak.
-                         int lastPos        ///< Index of last vector item beloging to the peak.
-                         ) const;
-
-    /// Finds the data vector index where the monotoniously decreasing signal crosses the
-    /// given level.
-    int   findCrossingLevel(const float *data,  ///< Data vector.
-                            float level,        ///< Goal crossing level.
-                            int peakpos,        ///< Peak position index within the data vector.
-                            int direction       /// Direction where to proceed from the peak: 1 = right, -1 = left.
-                            ) const;
-
-    // Finds real 'top' of a peak hump from neighnourhood of the given 'peakpos'.
-    int findTop(const float *data, int peakpos) const;
-
-
-    /// Finds the 'ground' level, i.e. smallest level between two neighbouring peaks, to right- 
-    /// or left-hand side of the given peak position.
-    int   findGround(const float *data,     /// Data vector.
-                     int peakpos,           /// Peak position index within the data vector.
-                     int direction          /// Direction where to proceed from the peak: 1 = right, -1 = left.
-                     ) const;
-
-    /// get exact center of peak near given position by calculating local mass of center
-    double getPeakCenter(const float *data, int peakpos) const;
-
-public:
-    /// Constructor. 
-    PeakFinder();
-
-    /// Detect exact peak position of the data vector by finding the largest peak 'hump'
-    /// and calculating the mass-center location of the peak hump.
-    ///
-    /// \return The location of the largest base harmonic peak hump.
-    double detectPeak(const float *data, /// Data vector to be analyzed. The data vector has
-                                        /// to be at least 'maxPos' items long.
-                     int minPos,        ///< Min allowed peak location within the vector data.
-                     int maxPos         ///< Max allowed peak location within the vector data.
-                     );
-};
-
-}
-
-#endif // _PeakFinder_H_
+////////////////////////////////////////////////////////////////////////////////
+///
+/// The routine detects highest value on an array of values and calculates the
+/// precise peak location as a mass-center of the 'hump' around the peak value.
+///
+/// Author        : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+//  SoundTouch audio processing library
+//  Copyright (c) Olli Parviainen
+//
+//  This library is free software; you can redistribute it and/or
+//  modify it under the terms of the GNU Lesser General Public
+//  License as published by the Free Software Foundation; either
+//  version 2.1 of the License, or (at your option) any later version.
+//
+//  This library is distributed in the hope that it will be useful,
+//  but WITHOUT ANY WARRANTY; without even the implied warranty of
+//  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+//  Lesser General Public License for more details.
+//
+//  You should have received a copy of the GNU Lesser General Public
+//  License along with this library; if not, write to the Free Software
+//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#ifndef _PeakFinder_H_
+#define _PeakFinder_H_
+
+namespace soundtouch
+{
+
+class PeakFinder
+{
+protected:
+    /// Min, max allowed peak positions within the data vector
+    int minPos, maxPos;
+
+    /// Calculates the mass center between given vector items.
+    double calcMassCenter(const float *data, ///< Data vector.
+                         int firstPos,      ///< Index of first vector item belonging 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
+    /// given level.
+    int   findCrossingLevel(const float *data,  ///< Data vector.
+                            float level,        ///< Goal crossing level.
+                            int peakpos,        ///< Peak position index within the data vector.
+                            int direction       /// Direction where to proceed from the peak: 1 = right, -1 = left.
+                            ) const;
+
+    // Finds real 'top' of a peak hump from neighnourhood of the given 'peakpos'.
+    int findTop(const float *data, int peakpos) const;
+
+
+    /// Finds the 'ground' level, i.e. smallest level between two neighbouring peaks, to right-
+    /// or left-hand side of the given peak position.
+    int   findGround(const float *data,     /// Data vector.
+                     int peakpos,           /// Peak position index within the data vector.
+                     int direction          /// Direction where to proceed from the peak: 1 = right, -1 = left.
+                     ) const;
+
+    /// get exact center of peak near given position by calculating local mass of center
+    double getPeakCenter(const float *data, int peakpos) const;
+
+public:
+    /// Constructor.
+    PeakFinder();
+
+    /// Detect exact peak position of the data vector by finding the largest peak 'hump'
+    /// and calculating the mass-center location of the peak hump.
+    ///
+    /// \return The location of the largest base harmonic peak hump.
+    double detectPeak(const float *data, /// Data vector to be analyzed. The data vector has
+                                        /// to be at least 'maxPos' items long.
+                     int minPos,        ///< Min allowed peak location within the vector data.
+                     int maxPos         ///< Max allowed peak location within the vector data.
+                     );
+};
+
+}
+
+#endif // _PeakFinder_H_

source/SoundTouch/RateTransposer.cpp

@@ -1,626 +1,313 @@
-////////////////////////////////////////////////////////////////////////////////
-/// 
-/// Sample rate transposer. Changes sample rate by using linear interpolation 
-/// together with anti-alias filtering (first order interpolation with anti-
-/// alias filtering should be quite adequate for this application)
-///
-/// Author        : Copyright (c) Olli Parviainen
-/// Author e-mail : oparviai 'at' iki.fi
-/// SoundTouch WWW: http://www.surina.net/soundtouch
-///
-////////////////////////////////////////////////////////////////////////////////
-//
-// Last changed  : $Date$
-// File revision : $Revision: 4 $
-//
-// $Id$
-//
-////////////////////////////////////////////////////////////////////////////////
-//
-// License :
-//
-//  SoundTouch audio processing library
-//  Copyright (c) Olli Parviainen
-//
-//  This library is free software; you can redistribute it and/or
-//  modify it under the terms of the GNU Lesser General Public
-//  License as published by the Free Software Foundation; either
-//  version 2.1 of the License, or (at your option) any later version.
-//
-//  This library is distributed in the hope that it will be useful,
-//  but WITHOUT ANY WARRANTY; without even the implied warranty of
-//  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-//  Lesser General Public License for more details.
-//
-//  You should have received a copy of the GNU Lesser General Public
-//  License along with this library; if not, write to the Free Software
-//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
-//
-////////////////////////////////////////////////////////////////////////////////
-
-#include <memory.h>
-#include <assert.h>
-#include <stdlib.h>
-#include <stdio.h>
-#include "RateTransposer.h"
-#include "AAFilter.h"
-
-using namespace soundtouch;
-
-
-/// A linear samplerate transposer class that uses integer arithmetics.
-/// for the transposing.
-class RateTransposerInteger : public RateTransposer
-{
-protected:
-    int iSlopeCount;
-    int iRate;
-    SAMPLETYPE sPrevSampleL, sPrevSampleR;
-
-    virtual void resetRegisters();
-
-    virtual uint transposeStereo(SAMPLETYPE *dest, 
-                         const SAMPLETYPE *src, 
-                         uint numSamples);
-    virtual uint transposeMono(SAMPLETYPE *dest, 
-                       const SAMPLETYPE *src, 
-                       uint numSamples);
-
-public:
-    RateTransposerInteger();
-    virtual ~RateTransposerInteger();
-
-    /// Sets new target rate. Normal rate = 1.0, smaller values represent slower 
-    /// rate, larger faster rates.
-    virtual void setRate(float newRate);
-
-};
-
-
-/// A linear samplerate transposer class that uses floating point arithmetics
-/// for the transposing.
-class RateTransposerFloat : public RateTransposer
-{
-protected:
-    float fSlopeCount;
-    SAMPLETYPE sPrevSampleL, sPrevSampleR;
-
-    virtual void resetRegisters();
-
-    virtual uint transposeStereo(SAMPLETYPE *dest, 
-                         const SAMPLETYPE *src, 
-                         uint numSamples);
-    virtual uint transposeMono(SAMPLETYPE *dest, 
-                       const SAMPLETYPE *src, 
-                       uint numSamples);
-
-public:
-    RateTransposerFloat();
-    virtual ~RateTransposerFloat();
-};
-
-
-
-
-// Operator 'new' is overloaded so that it automatically creates a suitable instance 
-// depending on if we've a MMX/SSE/etc-capable CPU available or not.
-void * RateTransposer::operator new(size_t s)
-{
-    ST_THROW_RT_ERROR("Error in RateTransoser::new: don't use \"new TDStretch\" directly, use \"newInstance\" to create a new instance instead!");
-    return newInstance();
-}
-
-
-RateTransposer *RateTransposer::newInstance()
-{
-#ifdef SOUNDTOUCH_INTEGER_SAMPLES
-    return ::new RateTransposerInteger;
-#else
-    return ::new RateTransposerFloat;
-#endif
-}
-
-
-// Constructor
-RateTransposer::RateTransposer() : FIFOProcessor(&outputBuffer)
-{
-    numChannels = 2;
-    bUseAAFilter = TRUE;
-    fRate = 0;
-
-    // Instantiates the anti-alias filter with default tap length
-    // of 32
-    pAAFilter = new AAFilter(32);
-}
-
-
-
-RateTransposer::~RateTransposer()
-{
-    delete pAAFilter;
-}
-
-
-
-/// Enables/disables the anti-alias filter. Zero to disable, nonzero to enable
-void RateTransposer::enableAAFilter(BOOL newMode)
-{
-    bUseAAFilter = newMode;
-}
-
-
-/// Returns nonzero if anti-alias filter is enabled.
-BOOL RateTransposer::isAAFilterEnabled() const
-{
-    return bUseAAFilter;
-}
-
-
-AAFilter *RateTransposer::getAAFilter()
-{
-    return pAAFilter;
-}
-
-
-
-// Sets new target iRate. Normal iRate = 1.0, smaller values represent slower 
-// iRate, larger faster iRates.
-void RateTransposer::setRate(float newRate)
-{
-    double fCutoff;
-
-    fRate = newRate;
-
-    // design a new anti-alias filter
-    if (newRate > 1.0f) 
-    {
-        fCutoff = 0.5f / newRate;
-    } 
-    else 
-    {
-        fCutoff = 0.5f * newRate;
-    }
-    pAAFilter->setCutoffFreq(fCutoff);
-}
-
-
-// Outputs as many samples of the 'outputBuffer' as possible, and if there's
-// any room left, outputs also as many of the incoming samples as possible.
-// The goal is to drive the outputBuffer empty.
-//
-// It's allowed for 'output' and 'input' parameters to point to the same
-// memory position.
-/*
-void RateTransposer::flushStoreBuffer()
-{
-    if (storeBuffer.isEmpty()) return;
-
-    outputBuffer.moveSamples(storeBuffer);
-}
-*/
-
-
-// Adds 'nSamples' pcs of samples from the 'samples' memory position into
-// the input of the object.
-void RateTransposer::putSamples(const SAMPLETYPE *samples, uint nSamples)
-{
-    processSamples(samples, nSamples);
-}
-
-
-
-// Transposes up the sample rate, causing the observed playback 'rate' of the
-// sound to decrease
-void RateTransposer::upsample(const SAMPLETYPE *src, uint nSamples)
-{
-    uint count, sizeTemp, num;
-
-    // If the parameter 'uRate' value is smaller than 'SCALE', first transpose
-    // the samples and then apply the anti-alias filter to remove aliasing.
-
-    // First check that there's enough room in 'storeBuffer' 
-    // (+16 is to reserve some slack in the destination buffer)
-    sizeTemp = (uint)((float)nSamples / fRate + 16.0f);
-
-    // Transpose the samples, store the result into the end of "storeBuffer"
-    count = transpose(storeBuffer.ptrEnd(sizeTemp), src, nSamples);
-    storeBuffer.putSamples(count);
-
-    // Apply the anti-alias filter to samples in "store output", output the
-    // result to "dest"
-    num = storeBuffer.numSamples();
-    count = pAAFilter->evaluate(outputBuffer.ptrEnd(num), 
-        storeBuffer.ptrBegin(), num, (uint)numChannels);
-    outputBuffer.putSamples(count);
-
-    // Remove the processed samples from "storeBuffer"
-    storeBuffer.receiveSamples(count);
-}
-
-
-// Transposes down the sample rate, causing the observed playback 'rate' of the
-// sound to increase
-void RateTransposer::downsample(const SAMPLETYPE *src, uint nSamples)
-{
-    uint count, sizeTemp;
-
-    // If the parameter 'uRate' value is larger than 'SCALE', first apply the
-    // anti-alias filter to remove high frequencies (prevent them from folding
-    // over the lover frequencies), then transpose.
-
-    // Add the new samples to the end of the storeBuffer
-    storeBuffer.putSamples(src, nSamples);
-
-    // Anti-alias filter the samples to prevent folding and output the filtered 
-    // data to tempBuffer. Note : because of the FIR filter length, the
-    // filtering routine takes in 'filter_length' more samples than it outputs.
-    assert(tempBuffer.isEmpty());
-    sizeTemp = storeBuffer.numSamples();
-
-    count = pAAFilter->evaluate(tempBuffer.ptrEnd(sizeTemp), 
-        storeBuffer.ptrBegin(), sizeTemp, (uint)numChannels);
-
-	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);
-
-    // If anti-alias filter is turned off, simply transpose without applying
-    // the filter
-    if (bUseAAFilter == FALSE) 
-    {
-        sizeReq = (uint)((float)nSamples / fRate + 1.0f);
-        count = transpose(outputBuffer.ptrEnd(sizeReq), src, nSamples);
-        outputBuffer.putSamples(count);
-        return;
-    }
-
-    // Transpose with anti-alias filter
-    if (fRate < 1.0f) 
-    {
-        upsample(src, nSamples);
-    } 
-    else  
-    {
-        downsample(src, nSamples);
-    }
-}
-
-
-// Transposes the sample rate of the given samples using linear interpolation. 
-// Returns the number of samples returned in the "dest" buffer
-inline uint RateTransposer::transpose(SAMPLETYPE *dest, const SAMPLETYPE *src, uint nSamples)
-{
-    if (numChannels == 2) 
-    {
-        return transposeStereo(dest, src, nSamples);
-    } 
-    else 
-    {
-        return transposeMono(dest, src, nSamples);
-    }
-}
-
-
-// Sets the number of channels, 1 = mono, 2 = stereo
-void RateTransposer::setChannels(int nChannels)
-{
-    assert(nChannels > 0);
-    if (numChannels == nChannels) return;
-
-    assert(nChannels == 1 || nChannels == 2);
-    numChannels = nChannels;
-
-    storeBuffer.setChannels(numChannels);
-    tempBuffer.setChannels(numChannels);
-    outputBuffer.setChannels(numChannels);
-
-    // Inits the linear interpolation registers
-    resetRegisters();
-}
-
-
-// Clears all the samples in the object
-void RateTransposer::clear()
-{
-    outputBuffer.clear();
-    storeBuffer.clear();
-}
-
-
-// Returns nonzero if there aren't any samples available for outputting.
-int RateTransposer::isEmpty() const
-{
-    int res;
-
-    res = FIFOProcessor::isEmpty();
-    if (res == 0) return 0;
-    return storeBuffer.isEmpty();
-}
-
-
-//////////////////////////////////////////////////////////////////////////////
-//
-// RateTransposerInteger - integer arithmetic implementation
-// 
-
-/// fixed-point interpolation routine precision
-#define SCALE    65536
-
-// Constructor
-RateTransposerInteger::RateTransposerInteger() : RateTransposer()
-{
-    // Notice: use local function calling syntax for sake of clarity, 
-    // to indicate the fact that C++ constructor can't call virtual functions.
-    RateTransposerInteger::resetRegisters();
-    RateTransposerInteger::setRate(1.0f);
-}
-
-
-RateTransposerInteger::~RateTransposerInteger()
-{
-}
-
-
-void RateTransposerInteger::resetRegisters()
-{
-    iSlopeCount = 0;
-    sPrevSampleL = 
-    sPrevSampleR = 0;
-}
-
-
-
-// Transposes the sample rate of the given samples using linear interpolation. 
-// 'Mono' version of the routine. Returns the number of samples returned in 
-// the "dest" buffer
-uint RateTransposerInteger::transposeMono(SAMPLETYPE *dest, const SAMPLETYPE *src, uint nSamples)
-{
-    unsigned int i, used;
-    LONG_SAMPLETYPE temp, vol1;
-
-    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 
-// the "dest" buffer
-uint RateTransposerInteger::transposeStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, uint nSamples)
-{
-    unsigned int srcPos, i, used;
-    LONG_SAMPLETYPE temp, vol1;
-
-    if (nSamples == 0) return 0;  // no samples, no work
-
-    used = 0;    
-    i = 0;
-
-    // Process the last sample saved from the sPrevSampleLious call first...
-    while (iSlopeCount <= SCALE) 
-    {
-        vol1 = (LONG_SAMPLETYPE)(SCALE - iSlopeCount);
-        temp = vol1 * sPrevSampleL + iSlopeCount * src[0];
-        dest[2 * i] = (SAMPLETYPE)(temp / SCALE);
-        temp = vol1 * sPrevSampleR + iSlopeCount * src[1];
-        dest[2 * i + 1] = (SAMPLETYPE)(temp / SCALE);
-        i++;
-        iSlopeCount += iRate;
-    }
-    // now always (iSlopeCount > SCALE)
-    iSlopeCount -= SCALE;
-
-    while (1)
-    {
-        while (iSlopeCount > SCALE) 
-        {
-            iSlopeCount -= SCALE;
-            used ++;
-            if (used >= nSamples - 1) goto end;
-        }
-        srcPos = 2 * used;
-        vol1 = (LONG_SAMPLETYPE)(SCALE - iSlopeCount);
-        temp = src[srcPos] * vol1 + iSlopeCount * src[srcPos + 2];
-        dest[2 * i] = (SAMPLETYPE)(temp / SCALE);
-        temp = src[srcPos + 1] * vol1 + iSlopeCount * src[srcPos + 3];
-        dest[2 * i + 1] = (SAMPLETYPE)(temp / SCALE);
-
-        i++;
-        iSlopeCount += iRate;
-    }
-end:
-    // Store the last sample for the next round
-    sPrevSampleL = src[2 * nSamples - 2];
-    sPrevSampleR = src[2 * nSamples - 1];
-
-    return i;
-}
-
-
-// Sets new target iRate. Normal iRate = 1.0, smaller values represent slower 
-// iRate, larger faster iRates.
-void RateTransposerInteger::setRate(float newRate)
-{
-    iRate = (int)(newRate * SCALE + 0.5f);
-    RateTransposer::setRate(newRate);
-}
-
-
-//////////////////////////////////////////////////////////////////////////////
-//
-// RateTransposerFloat - floating point arithmetic implementation
-// 
-//////////////////////////////////////////////////////////////////////////////
-
-// Constructor
-RateTransposerFloat::RateTransposerFloat() : RateTransposer()
-{
-    // Notice: use local function calling syntax for sake of clarity, 
-    // to indicate the fact that C++ constructor can't call virtual functions.
-    RateTransposerFloat::resetRegisters();
-    RateTransposerFloat::setRate(1.0f);
-}
-
-
-RateTransposerFloat::~RateTransposerFloat()
-{
-}
-
-
-void RateTransposerFloat::resetRegisters()
-{
-    fSlopeCount = 0;
-    sPrevSampleL = 
-    sPrevSampleR = 0;
-}
-
-
-
-// Transposes the sample rate of the given samples using linear interpolation. 
-// 'Mono' version of the routine. Returns the number of samples returned in 
-// the "dest" buffer
-uint RateTransposerFloat::transposeMono(SAMPLETYPE *dest, const SAMPLETYPE *src, uint nSamples)
-{
-    unsigned int i, used;
-
-    used = 0;    
-    i = 0;
-
-    // Process the last sample saved from the previous call first...
-    while (fSlopeCount <= 1.0f) 
-    {
-        dest[i] = (SAMPLETYPE)((1.0f - fSlopeCount) * sPrevSampleL + fSlopeCount * src[0]);
-        i++;
-        fSlopeCount += fRate;
-    }
-    fSlopeCount -= 1.0f;
-
-    if (nSamples > 1)
-    {
-        while (1)
-        {
-            while (fSlopeCount > 1.0f) 
-            {
-                fSlopeCount -= 1.0f;
-                used ++;
-                if (used >= nSamples - 1) goto end;
-            }
-            dest[i] = (SAMPLETYPE)((1.0f - fSlopeCount) * src[used] + fSlopeCount * src[used + 1]);
-            i++;
-            fSlopeCount += fRate;
-        }
-    }
-end:
-    // Store the last sample for the next round
-    sPrevSampleL = src[nSamples - 1];
-
-    return i;
-}
-
-
-// Transposes the sample rate of the given samples using linear interpolation. 
-// 'Mono' version of the routine. Returns the number of samples returned in 
-// the "dest" buffer
-uint RateTransposerFloat::transposeStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, uint nSamples)
-{
-    unsigned int srcPos, i, used;
-
-    if (nSamples == 0) return 0;  // no samples, no work
-
-    used = 0;    
-    i = 0;
-
-    // Process the last sample saved from the sPrevSampleLious call first...
-    while (fSlopeCount <= 1.0f) 
-    {
-        dest[2 * i] = (SAMPLETYPE)((1.0f - fSlopeCount) * sPrevSampleL + fSlopeCount * src[0]);
-        dest[2 * i + 1] = (SAMPLETYPE)((1.0f - fSlopeCount) * sPrevSampleR + fSlopeCount * src[1]);
-        i++;
-        fSlopeCount += fRate;
-    }
-    // now always (iSlopeCount > 1.0f)
-    fSlopeCount -= 1.0f;
-
-    if (nSamples > 1)
-    {
-        while (1)
-        {
-            while (fSlopeCount > 1.0f) 
-            {
-                fSlopeCount -= 1.0f;
-                used ++;
-                if (used >= nSamples - 1) goto end;
-            }
-            srcPos = 2 * used;
-
-            dest[2 * i] = (SAMPLETYPE)((1.0f - fSlopeCount) * src[srcPos] 
-                + fSlopeCount * src[srcPos + 2]);
-            dest[2 * i + 1] = (SAMPLETYPE)((1.0f - fSlopeCount) * src[srcPos + 1] 
-                + fSlopeCount * src[srcPos + 3]);
-
-            i++;
-            fSlopeCount += fRate;
-        }
-    }
-end:
-    // Store the last sample for the next round
-    sPrevSampleL = src[2 * nSamples - 2];
-    sPrevSampleR = src[2 * nSamples - 1];
-
-    return i;
-}
+////////////////////////////////////////////////////////////////////////////////
+///
+/// Sample rate transposer. Changes sample rate by using linear interpolation
+/// together with anti-alias filtering (first order interpolation with anti-
+/// alias filtering should be quite adequate for this application)
+///
+/// Author        : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+//  SoundTouch audio processing library
+//  Copyright (c) Olli Parviainen
+//
+//  This library is free software; you can redistribute it and/or
+//  modify it under the terms of the GNU Lesser General Public
+//  License as published by the Free Software Foundation; either
+//  version 2.1 of the License, or (at your option) any later version.
+//
+//  This library is distributed in the hope that it will be useful,
+//  but WITHOUT ANY WARRANTY; without even the implied warranty of
+//  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+//  Lesser General Public License for more details.
+//
+//  You should have received a copy of the GNU Lesser General Public
+//  License along with this library; if not, write to the Free Software
+//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#include <memory.h>
+#include <assert.h>
+#include <stdlib.h>
+#include <stdio.h>
+#include "RateTransposer.h"
+#include "InterpolateLinear.h"
+#include "InterpolateCubic.h"
+#include "InterpolateShannon.h"
+#include "AAFilter.h"
+
+using namespace soundtouch;
+
+// Define default interpolation algorithm here
+TransposerBase::ALGORITHM TransposerBase::algorithm = TransposerBase::CUBIC;
+
+
+// Constructor
+RateTransposer::RateTransposer() : FIFOProcessor(&outputBuffer)
+{
+    bUseAAFilter =
+#ifndef SOUNDTOUCH_PREVENT_CLICK_AT_RATE_CROSSOVER
+        true;
+#else
+        // Disable Anti-alias filter if desirable to avoid click at rate change zero value crossover
+        false;
+#endif
+
+    // Instantiates the anti-alias filter
+    pAAFilter = new AAFilter(64);
+    pTransposer = TransposerBase::newInstance();
+    clear();
+}
+
+
+RateTransposer::~RateTransposer()
+{
+    delete pAAFilter;
+    delete pTransposer;
+}
+
+
+/// Enables/disables the anti-alias filter. Zero to disable, nonzero to enable
+void RateTransposer::enableAAFilter(bool newMode)
+{
+#ifndef SOUNDTOUCH_PREVENT_CLICK_AT_RATE_CROSSOVER
+    // Disable Anti-alias filter if desirable to avoid click at rate change zero value crossover
+    bUseAAFilter = newMode;
+    clear();
+#endif
+}
+
+
+/// Returns nonzero if anti-alias filter is enabled.
+bool RateTransposer::isAAFilterEnabled() const
+{
+    return bUseAAFilter;
+}
+
+
+AAFilter *RateTransposer::getAAFilter()
+{
+    return pAAFilter;
+}
+
+
+// Sets new target iRate. Normal iRate = 1.0, smaller values represent slower
+// iRate, larger faster iRates.
+void RateTransposer::setRate(double newRate)
+{
+    double fCutoff;
+
+    pTransposer->setRate(newRate);
+
+    // design a new anti-alias filter
+    if (newRate > 1.0)
+    {
+        fCutoff = 0.5 / newRate;
+    }
+    else
+    {
+        fCutoff = 0.5 * newRate;
+    }
+    pAAFilter->setCutoffFreq(fCutoff);
+}
+
+
+// Adds 'nSamples' pcs of samples from the 'samples' memory position into
+// the input of the object.
+void RateTransposer::putSamples(const SAMPLETYPE *samples, uint nSamples)
+{
+    processSamples(samples, nSamples);
+}
+
+
+// Transposes 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)
+{
+    if (nSamples == 0) return;
+
+    // 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)
+    {
+        (void)pTransposer->transpose(outputBuffer, inputBuffer);
+        return;
+    }
+
+    assert(pAAFilter);
+
+    // Transpose with anti-alias filter
+    if (pTransposer->rate < 1.0f)
+    {
+        // 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
+    {
+        // 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);
+
+        // Transpose the AA-filtered samples in "midBuffer"
+        pTransposer->transpose(outputBuffer, midBuffer);
+    }
+}
+
+
+// Sets the number of channels, 1 = mono, 2 = stereo
+void RateTransposer::setChannels(int nChannels)
+{
+    if (!verifyNumberOfChannels(nChannels) ||
+        (pTransposer->numChannels == nChannels)) return;
+
+    pTransposer->setChannels(nChannels);
+    inputBuffer.setChannels(nChannels);
+    midBuffer.setChannels(nChannels);
+    outputBuffer.setChannels(nChannels);
+}
+
+
+// Clears all the samples in the object
+void RateTransposer::clear()
+{
+    outputBuffer.clear();
+    midBuffer.clear();
+    inputBuffer.clear();
+    pTransposer->resetRegisters();
+
+    // prefill buffer to avoid losing first samples at beginning of stream
+    int prefill = getLatency();
+    inputBuffer.addSilent(prefill);
+}
+
+
+// Returns nonzero if there aren't any samples available for outputting.
+int RateTransposer::isEmpty() const
+{
+    int res;
+
+    res = FIFOProcessor::isEmpty();
+    if (res == 0) return 0;
+    return inputBuffer.isEmpty();
+}
+
+
+/// Return approximate initial input-output latency
+int RateTransposer::getLatency() const
+{
+    return pTransposer->getLatency() +
+        ((bUseAAFilter) ? (pAAFilter->getLength() / 2) : 0);
+}
+
+
+//////////////////////////////////////////////////////////////////////////////
+//
+// TransposerBase - Base class for interpolation
+//
+
+// static function to set interpolation algorithm
+void TransposerBase::setAlgorithm(TransposerBase::ALGORITHM a)
+{
+    TransposerBase::algorithm = a;
+}
+
+
+// Transposes the sample rate of the given samples using linear interpolation.
+// Returns the number of samples returned in the "dest" buffer
+int TransposerBase::transpose(FIFOSampleBuffer &dest, FIFOSampleBuffer &src)
+{
+    int numSrcSamples = src.numSamples();
+    int sizeDemand = (int)((double)numSrcSamples / rate) + 8;
+    int numOutput;
+    SAMPLETYPE *psrc = src.ptrBegin();
+    SAMPLETYPE *pdest = dest.ptrEnd(sizeDemand);
+
+#ifndef USE_MULTICH_ALWAYS
+    if (numChannels == 1)
+    {
+        numOutput = transposeMono(pdest, psrc, numSrcSamples);
+    }
+    else if (numChannels == 2)
+    {
+        numOutput = transposeStereo(pdest, psrc, numSrcSamples);
+    }
+    else
+#endif // USE_MULTICH_ALWAYS
+    {
+        assert(numChannels > 0);
+        numOutput = transposeMulti(pdest, psrc, numSrcSamples);
+    }
+    dest.putSamples(numOutput);
+    src.receiveSamples(numSrcSamples);
+    return numOutput;
+}
+
+
+TransposerBase::TransposerBase()
+{
+    numChannels = 0;
+    rate = 1.0f;
+}
+
+
+TransposerBase::~TransposerBase()
+{
+}
+
+
+void TransposerBase::setChannels(int channels)
+{
+    numChannels = channels;
+    resetRegisters();
+}
+
+
+void TransposerBase::setRate(double newRate)
+{
+    rate = newRate;
+}
+
+
+// static factory function
+TransposerBase *TransposerBase::newInstance()
+{
+#ifdef SOUNDTOUCH_INTEGER_SAMPLES
+    // Notice: For integer arithmetic support only linear algorithm (due to simplest calculus)
+    return ::new InterpolateLinearInteger;
+#else
+    switch (algorithm)
+    {
+        case LINEAR:
+            return new InterpolateLinearFloat;
+
+        case CUBIC:
+            return new InterpolateCubic;
+
+        case SHANNON:
+            return new InterpolateShannon;
+
+        default:
+            assert(false);
+            return nullptr;
+    }
+#endif
+}

source/SoundTouch/RateTransposer.h

@@ -1,159 +1,164 @@
-////////////////////////////////////////////////////////////////////////////////
-/// 
-/// Sample rate transposer. Changes sample rate by using linear interpolation 
-/// together with anti-alias filtering (first order interpolation with anti-
-/// alias filtering should be quite adequate for this application).
-///
-/// Use either of the derived classes of 'RateTransposerInteger' or 
-/// 'RateTransposerFloat' for corresponding integer/floating point tranposing
-/// algorithm implementation.
-///
-/// Author        : Copyright (c) Olli Parviainen
-/// Author e-mail : oparviai 'at' iki.fi
-/// SoundTouch WWW: http://www.surina.net/soundtouch
-///
-////////////////////////////////////////////////////////////////////////////////
-//
-// Last changed  : $Date$
-// File revision : $Revision: 4 $
-//
-// $Id$
-//
-////////////////////////////////////////////////////////////////////////////////
-//
-// License :
-//
-//  SoundTouch audio processing library
-//  Copyright (c) Olli Parviainen
-//
-//  This library is free software; you can redistribute it and/or
-//  modify it under the terms of the GNU Lesser General Public
-//  License as published by the Free Software Foundation; either
-//  version 2.1 of the License, or (at your option) any later version.
-//
-//  This library is distributed in the hope that it will be useful,
-//  but WITHOUT ANY WARRANTY; without even the implied warranty of
-//  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-//  Lesser General Public License for more details.
-//
-//  You should have received a copy of the GNU Lesser General Public
-//  License along with this library; if not, write to the Free Software
-//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
-//
-////////////////////////////////////////////////////////////////////////////////
-
-#ifndef RateTransposer_H
-#define RateTransposer_H
-
-#include <stddef.h>
-#include "AAFilter.h"
-#include "FIFOSamplePipe.h"
-#include "FIFOSampleBuffer.h"
-
-#include "STTypes.h"
-
-namespace soundtouch
-{
-
-/// A common linear samplerate transposer class.
-///
-/// Note: Use function "RateTransposer::newInstance()" to create a new class 
-/// instance instead of the "new" operator; that function automatically 
-/// chooses a correct implementation depending on if integer or floating 
-/// arithmetics are to be used.
-class RateTransposer : public FIFOProcessor
-{
-protected:
-    /// Anti-alias filter object
-    AAFilter *pAAFilter;
-
-    float fRate;
-
-    int numChannels;
-
-    /// Buffer for collecting samples to feed the anti-alias filter between
-    /// two batches
-    FIFOSampleBuffer storeBuffer;
-
-    /// Buffer for keeping samples between transposing & anti-alias filter
-    FIFOSampleBuffer tempBuffer;
-
-    /// Output sample buffer
-    FIFOSampleBuffer outputBuffer;
-
-    BOOL bUseAAFilter;
-
-    virtual void resetRegisters() = 0;
-
-    virtual uint transposeStereo(SAMPLETYPE *dest, 
-                         const SAMPLETYPE *src, 
-                         uint numSamples) = 0;
-    virtual uint transposeMono(SAMPLETYPE *dest, 
-                       const SAMPLETYPE *src, 
-                       uint numSamples) = 0;
-    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.
-    /// The maximum amount of samples that can be returned at a time is set by
-    /// the 'set_returnBuffer_size' function.
-    void processSamples(const SAMPLETYPE *src, 
-                        uint numSamples);
-
-
-public:
-    RateTransposer();
-    virtual ~RateTransposer();
-
-    /// Operator 'new' is overloaded so that it automatically creates a suitable instance 
-    /// depending on if we're to use integer or floating point arithmetics.
-    static void *operator new(size_t s);
-
-    /// Use this function instead of "new" operator to create a new instance of this class. 
-    /// This function automatically chooses a correct implementation, depending on if 
-    /// integer ot floating point arithmetics are to be used.
-    static RateTransposer *newInstance();
-
-    /// Returns the output buffer object
-    FIFOSamplePipe *getOutput() { return &outputBuffer; };
-
-    /// Returns the store buffer object
-    FIFOSamplePipe *getStore() { return &storeBuffer; };
-
-    /// Return anti-alias filter object
-    AAFilter *getAAFilter();
-
-    /// Enables/disables the anti-alias filter. Zero to disable, nonzero to enable
-    void enableAAFilter(BOOL newMode);
-
-    /// Returns nonzero if anti-alias filter is enabled.
-    BOOL isAAFilterEnabled() const;
-
-    /// Sets new target rate. Normal rate = 1.0, smaller values represent slower 
-    /// rate, larger faster rates.
-    virtual void setRate(float newRate);
-
-    /// Sets the number of channels, 1 = mono, 2 = stereo
-    void setChannels(int channels);
-
-    /// Adds 'numSamples' pcs of samples from the 'samples' memory position into
-    /// the input of the object.
-    void putSamples(const SAMPLETYPE *samples, uint numSamples);
-
-    /// Clears all the samples in the object
-    void clear();
-
-    /// Returns nonzero if there aren't any samples available for outputting.
-    int isEmpty() const;
-};
-
-}
-
-#endif
+////////////////////////////////////////////////////////////////////////////////
+///
+/// Sample rate transposer. Changes sample rate by using linear interpolation
+/// together with anti-alias filtering (first order interpolation with anti-
+/// alias filtering should be quite adequate for this application).
+///
+/// Use either of the derived classes of 'RateTransposerInteger' or
+/// 'RateTransposerFloat' for corresponding integer/floating point tranposing
+/// algorithm implementation.
+///
+/// Author        : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+//  SoundTouch audio processing library
+//  Copyright (c) Olli Parviainen
+//
+//  This library is free software; you can redistribute it and/or
+//  modify it under the terms of the GNU Lesser General Public
+//  License as published by the Free Software Foundation; either
+//  version 2.1 of the License, or (at your option) any later version.
+//
+//  This library is distributed in the hope that it will be useful,
+//  but WITHOUT ANY WARRANTY; without even the implied warranty of
+//  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+//  Lesser General Public License for more details.
+//
+//  You should have received a copy of the GNU Lesser General Public
+//  License along with this library; if not, write to the Free Software
+//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#ifndef RateTransposer_H
+#define RateTransposer_H
+
+#include <stddef.h>
+#include "AAFilter.h"
+#include "FIFOSamplePipe.h"
+#include "FIFOSampleBuffer.h"
+
+#include "STTypes.h"
+
+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.
+///
+class RateTransposer : public FIFOProcessor
+{
+protected:
+    /// Anti-alias filter object
+    AAFilter *pAAFilter;
+    TransposerBase *pTransposer;
+
+    /// Buffer for collecting samples to feed the anti-alias filter between
+    /// two batches
+    FIFOSampleBuffer inputBuffer;
+
+    /// Buffer for keeping samples between transposing & anti-alias filter
+    FIFOSampleBuffer midBuffer;
+
+    /// Output sample buffer
+    FIFOSampleBuffer outputBuffer;
+
+    bool bUseAAFilter;
+
+
+    /// Transposes sample rate by applying anti-alias filter to prevent folding.
+    /// Returns amount of samples returned in the "dest" buffer.
+    /// The maximum amount of samples that can be returned at a time is set by
+    /// the 'set_returnBuffer_size' function.
+    void processSamples(const SAMPLETYPE *src,
+                        uint numSamples);
+
+public:
+    RateTransposer();
+    virtual ~RateTransposer() override;
+
+    /// Returns the output buffer object
+    FIFOSamplePipe *getOutput() { return &outputBuffer; };
+
+    /// Return anti-alias filter object
+    AAFilter *getAAFilter();
+
+    /// Enables/disables the anti-alias filter. Zero to disable, nonzero to enable
+    void enableAAFilter(bool newMode);
+
+    /// Returns nonzero if anti-alias filter is enabled.
+    bool isAAFilterEnabled() const;
+
+    /// Sets new target rate. Normal rate = 1.0, smaller values represent slower
+    /// rate, larger faster rates.
+    virtual void setRate(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) override;
+
+    /// Clears all the samples in the object
+    void clear() override;
+
+    /// Returns nonzero if there aren't any samples available for outputting.
+    int isEmpty() const override;
+
+    /// Return approximate initial input-output latency
+    int getLatency() const;
+};
+
+}
+
+#endif

source/SoundTouch/SoundTouch.dsp

@@ -1,188 +0,0 @@
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 EndProject
 Global
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source/SoundTouch/SoundTouch.vcproj

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source/SoundTouch/TDStretch.h

@@ -1,268 +1,279 @@
-////////////////////////////////////////////////////////////////////////////////
-/// 
-/// 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/SSE optimized functions reside in separate, platform-specific files 
-/// 'mmx_optimized.cpp' and 'sse_optimized.cpp'
-///
-/// Author        : Copyright (c) Olli Parviainen
-/// Author e-mail : oparviai 'at' iki.fi
-/// SoundTouch WWW: http://www.surina.net/soundtouch
-///
-////////////////////////////////////////////////////////////////////////////////
-//
-// Last changed  : $Date$
-// File revision : $Revision: 4 $
-//
-// $Id$
-//
-////////////////////////////////////////////////////////////////////////////////
-//
-// License :
-//
-//  SoundTouch audio processing library
-//  Copyright (c) Olli Parviainen
-//
-//  This library is free software; you can redistribute it and/or
-//  modify it under the terms of the GNU Lesser General Public
-//  License as published by the Free Software Foundation; either
-//  version 2.1 of the License, or (at your option) any later version.
-//
-//  This library is distributed in the hope that it will be useful,
-//  but WITHOUT ANY WARRANTY; without even the implied warranty of
-//  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-//  Lesser General Public License for more details.
-//
-//  You should have received a copy of the GNU Lesser General Public
-//  License along with this library; if not, write to the Free Software
-//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
-//
-////////////////////////////////////////////////////////////////////////////////
-
-#ifndef TDStretch_H
-#define TDStretch_H
-
-#include <stddef.h>
-#include "STTypes.h"
-#include "RateTransposer.h"
-#include "FIFOSamplePipe.h"
-
-namespace soundtouch
-{
-
-/// Default values for sound processing parameters:
-/// Notice that the default parameters are tuned for contemporary popular music 
-/// processing. For speech processing applications these parameters suit better:
-///     #define DEFAULT_SEQUENCE_MS     40
-///     #define DEFAULT_SEEKWINDOW_MS   15
-///     #define DEFAULT_OVERLAP_MS      8
-///
-
-/// Default length of a single processing sequence, in milliseconds. This determines to how 
-/// long sequences the original sound is chopped in the time-stretch algorithm.
-///
-/// The larger this value is, the lesser sequences are used in processing. In principle
-/// a bigger value sounds better when slowing down tempo, but worse when increasing tempo
-/// and vice versa.
-///
-/// Increasing this value reduces computational burden & vice versa.
-//#define DEFAULT_SEQUENCE_MS         40
-#define DEFAULT_SEQUENCE_MS         USE_AUTO_SEQUENCE_LEN
-
-/// Giving this value for the sequence length sets automatic parameter value
-/// according to tempo setting (recommended)
-#define USE_AUTO_SEQUENCE_LEN       0
-
-/// Seeking window default length in milliseconds for algorithm that finds the best possible 
-/// overlapping location. This determines from how wide window the algorithm may look for an 
-/// optimal joining location when mixing the sound sequences back together. 
-///
-/// The bigger this window setting is, the higher the possibility to find a better mixing
-/// position will become, but at the same time large values may cause a "drifting" artifact
-/// because consequent sequences will be taken at more uneven intervals.
-///
-/// If there's a disturbing artifact that sounds as if a constant frequency was drifting 
-/// around, try reducing this setting.
-///
-/// Increasing this value increases computational burden & vice versa.
-//#define DEFAULT_SEEKWINDOW_MS       15
-#define DEFAULT_SEEKWINDOW_MS       USE_AUTO_SEEKWINDOW_LEN
-
-/// Giving this value for the seek window length sets automatic parameter value
-/// according to tempo setting (recommended)
-#define USE_AUTO_SEEKWINDOW_LEN     0
-
-/// Overlap length in milliseconds. When the chopped sound sequences are mixed back together, 
-/// to form a continuous sound stream, this parameter defines over how long period the two 
-/// consecutive sequences are let to overlap each other. 
-///
-/// This shouldn't be that critical parameter. If you reduce the DEFAULT_SEQUENCE_MS setting 
-/// by a large amount, you might wish to try a smaller value on this.
-///
-/// Increasing this value increases computational burden & vice versa.
-#define DEFAULT_OVERLAP_MS      8
-
-
-/// Class that does the time-stretch (tempo change) effect for the processed
-/// sound.
-class TDStretch : public FIFOProcessor
-{
-protected:
-    int channels;
-    int sampleReq;
-    float tempo;
-
-    SAMPLETYPE *pMidBuffer;
-    SAMPLETYPE *pMidBufferUnaligned;
-    int overlapLength;
-    int seekLength;
-    int seekWindowLength;
-    int overlapDividerBits;
-    int slopingDivider;
-    float nominalSkip;
-    float skipFract;
-    FIFOSampleBuffer outputBuffer;
-    FIFOSampleBuffer inputBuffer;
-    BOOL bQuickSeek;
-
-    int sampleRate;
-    int sequenceMs;
-    int seekWindowMs;
-    int overlapMs;
-    BOOL bAutoSeqSetting;
-    BOOL bAutoSeekSetting;
-
-    void acceptNewOverlapLength(int newOverlapLength);
-
-    virtual void clearCrossCorrState();
-    void calculateOverlapLength(int overlapMs);
-
-    virtual double calcCrossCorr(const SAMPLETYPE *mixingPos, const SAMPLETYPE *compare) const;
-
-    virtual int seekBestOverlapPositionFull(const SAMPLETYPE *refPos);
-    virtual int seekBestOverlapPositionQuick(const SAMPLETYPE *refPos);
-    int seekBestOverlapPosition(const SAMPLETYPE *refPos);
-
-    virtual void overlapStereo(SAMPLETYPE *output, const SAMPLETYPE *input) const;
-    virtual void overlapMono(SAMPLETYPE *output, const SAMPLETYPE *input) const;
-
-    void clearMidBuffer();
-    void overlap(SAMPLETYPE *output, const SAMPLETYPE *input, uint ovlPos) const;
-
-    void calcSeqParameters();
-
-    /// Changes the tempo of the given sound samples.
-    /// Returns amount of samples returned in the "output" buffer.
-    /// The maximum amount of samples that can be returned at a time is set by
-    /// the 'set_returnBuffer_size' function.
-    void processSamples();
-    
-public:
-    TDStretch();
-    virtual ~TDStretch();
-
-    /// Operator 'new' is overloaded so that it automatically creates a suitable instance 
-    /// depending on if we've a MMX/SSE/etc-capable CPU available or not.
-    static void *operator new(size_t s);
-
-    /// Use this function instead of "new" operator to create a new instance of this class. 
-    /// This function automatically chooses a correct feature set depending on if the CPU
-    /// supports MMX/SSE/etc extensions.
-    static TDStretch *newInstance();
-    
-    /// Returns the output buffer object
-    FIFOSamplePipe *getOutput() { return &outputBuffer; };
-
-    /// Returns the input buffer object
-    FIFOSamplePipe *getInput() { return &inputBuffer; };
-
-    /// Sets new target tempo. Normal tempo = 'SCALE', smaller values represent slower 
-    /// tempo, larger faster tempo.
-    void setTempo(float newTempo);
-
-    /// Returns nonzero if there aren't any samples available for outputting.
-    virtual void clear();
-
-    /// Clears the input buffer
-    void clearInput();
-
-    /// Sets the number of channels, 1 = mono, 2 = stereo
-    void setChannels(int numChannels);
-
-    /// Enables/disables the quick position seeking algorithm. Zero to disable, 
-    /// nonzero to enable
-    void enableQuickSeek(BOOL enable);
-
-    /// Returns nonzero if the quick seeking algorithm is enabled.
-    BOOL isQuickSeekEnabled() const;
-
-    /// Sets routine control parameters. These control are certain time constants
-    /// defining how the sound is stretched to the desired duration.
-    //
-    /// 'sampleRate' = sample rate of the sound
-    /// 'sequenceMS' = one processing sequence length in milliseconds
-    /// 'seekwindowMS' = seeking window length for scanning the best overlapping 
-    ///      position
-    /// 'overlapMS' = overlapping length
-    void setParameters(int sampleRate,          ///< Samplerate of sound being processed (Hz)
-                       int sequenceMS = -1,     ///< Single processing sequence length (ms)
-                       int seekwindowMS = -1,   ///< Offset seeking window length (ms)
-                       int overlapMS = -1       ///< Sequence overlapping length (ms)
-                       );
-
-    /// Get routine control parameters, see setParameters() function.
-    /// Any of the parameters to this function can be NULL, in such case corresponding parameter
-    /// value isn't returned.
-    void getParameters(int *pSampleRate, int *pSequenceMs, int *pSeekWindowMs, int *pOverlapMs) const;
-
-    /// Adds 'numsamples' pcs of samples from the 'samples' memory position into
-    /// the input of the object.
-    virtual void putSamples(
-            const SAMPLETYPE *samples,  ///< Input sample data
-            uint numSamples                         ///< Number of samples in 'samples' so that one sample
-                                                    ///< contains both channels if stereo
-            );
-
-    /// return nominal input sample requirement for triggering a processing batch
-    int getInputSampleReq() const
-    {
-        return (int)(nominalSkip + 0.5);
-    }
-
-    /// return nominal output sample amount when running a processing batch
-    int getOutputBatchSize() const
-    {
-        return seekWindowLength - overlapLength;
-    }
-};
-
-
-
-// Implementation-specific class declarations:
-
-#ifdef SOUNDTOUCH_ALLOW_MMX
-    /// Class that implements MMX optimized routines for 16bit integer samples type.
-    class TDStretchMMX : public TDStretch
-    {
-    protected:
-        double calcCrossCorr(const short *mixingPos, const short *compare) const;
-        virtual void overlapStereo(short *output, const short *input) const;
-        virtual void clearCrossCorrState();
-    };
-#endif /// SOUNDTOUCH_ALLOW_MMX
-
-
-#ifdef SOUNDTOUCH_ALLOW_SSE
-    /// Class that implements SSE optimized routines for floating point samples type.
-    class TDStretchSSE : public TDStretch
-    {
-    protected:
-        double calcCrossCorr(const float *mixingPos, const float *compare) const;
-    };
-
-#endif /// SOUNDTOUCH_ALLOW_SSE
-
-}
-#endif  /// TDStretch_H
+////////////////////////////////////////////////////////////////////////////////
+///
+/// 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/SSE optimized functions reside in separate, platform-specific files
+/// 'mmx_optimized.cpp' and 'sse_optimized.cpp'
+///
+/// Author        : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai 'at' iki.fi
+/// 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 TDStretch_H
+#define TDStretch_H
+
+#include <stddef.h>
+#include "STTypes.h"
+#include "RateTransposer.h"
+#include "FIFOSamplePipe.h"
+
+namespace soundtouch
+{
+
+/// Default values for sound processing parameters:
+/// Notice that the default parameters are tuned for contemporary popular music
+/// processing. For speech processing applications these parameters suit better:
+///     #define DEFAULT_SEQUENCE_MS     40
+///     #define DEFAULT_SEEKWINDOW_MS   15
+///     #define DEFAULT_OVERLAP_MS      8
+///
+
+/// Default length of a single processing sequence, in milliseconds. This determines to how
+/// long sequences the original sound is chopped in the time-stretch algorithm.
+///
+/// The larger this value is, the lesser sequences are used in processing. In principle
+/// a bigger value sounds better when slowing down tempo, but worse when increasing tempo
+/// and vice versa.
+///
+/// Increasing this value reduces computational burden & vice versa.
+//#define DEFAULT_SEQUENCE_MS         40
+#define DEFAULT_SEQUENCE_MS         USE_AUTO_SEQUENCE_LEN
+
+/// Giving this value for the sequence length sets automatic parameter value
+/// according to tempo setting (recommended)
+#define USE_AUTO_SEQUENCE_LEN       0
+
+/// Seeking window default length in milliseconds for algorithm that finds the best possible
+/// overlapping location. This determines from how wide window the algorithm may look for an
+/// optimal joining location when mixing the sound sequences back together.
+///
+/// The bigger this window setting is, the higher the possibility to find a better mixing
+/// position will become, but at the same time large values may cause a "drifting" artifact
+/// because consequent sequences will be taken at more uneven intervals.
+///
+/// If there's a disturbing artifact that sounds as if a constant frequency was drifting
+/// around, try reducing this setting.
+///
+/// Increasing this value increases computational burden & vice versa.
+//#define DEFAULT_SEEKWINDOW_MS       15
+#define DEFAULT_SEEKWINDOW_MS       USE_AUTO_SEEKWINDOW_LEN
+
+/// Giving this value for the seek window length sets automatic parameter value
+/// according to tempo setting (recommended)
+#define USE_AUTO_SEEKWINDOW_LEN     0
+
+/// Overlap length in milliseconds. When the chopped sound sequences are mixed back together,
+/// to form a continuous sound stream, this parameter defines over how long period the two
+/// consecutive sequences are let to overlap each other.
+///
+/// This shouldn't be that critical parameter. If you reduce the DEFAULT_SEQUENCE_MS setting
+/// by a large amount, you might wish to try a smaller value on this.
+///
+/// Increasing this value increases computational burden & vice versa.
+#define DEFAULT_OVERLAP_MS      8
+
+
+/// Class that does the time-stretch (tempo change) effect for the processed
+/// sound.
+class TDStretch : public FIFOProcessor
+{
+protected:
+    int channels;
+    int sampleReq;
+
+    int overlapLength;
+    int seekLength;
+    int seekWindowLength;
+    int overlapDividerBitsNorm;
+    int overlapDividerBitsPure;
+    int slopingDivider;
+    int sampleRate;
+    int sequenceMs;
+    int seekWindowMs;
+    int overlapMs;
+
+    unsigned long maxnorm;
+    float maxnormf;
+
+    double tempo;
+    double nominalSkip;
+    double skipFract;
+
+    bool bQuickSeek;
+    bool bAutoSeqSetting;
+    bool bAutoSeekSetting;
+    bool isBeginning;
+
+    SAMPLETYPE *pMidBuffer;
+    SAMPLETYPE *pMidBufferUnaligned;
+
+    FIFOSampleBuffer outputBuffer;
+    FIFOSampleBuffer inputBuffer;
+
+    void acceptNewOverlapLength(int newOverlapLength);
+
+    virtual void clearCrossCorrState();
+    void calculateOverlapLength(int overlapMs);
+
+    virtual double calcCrossCorr(const SAMPLETYPE *mixingPos, const SAMPLETYPE *compare, double &norm);
+    virtual double calcCrossCorrAccumulate(const SAMPLETYPE *mixingPos, const SAMPLETYPE *compare, double &norm);
+
+    virtual int seekBestOverlapPositionFull(const SAMPLETYPE *refPos);
+    virtual int seekBestOverlapPositionQuick(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.
+    /// The maximum amount of samples that can be returned at a time is set by
+    /// the 'set_returnBuffer_size' function.
+    void processSamples();
+
+public:
+    TDStretch();
+    virtual ~TDStretch() 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.
+    static void *operator new(size_t s);
+
+    /// Use this function instead of "new" operator to create a new instance of this class.
+    /// This function automatically chooses a correct feature set depending on if the CPU
+    /// supports MMX/SSE/etc extensions.
+    static TDStretch *newInstance();
+
+    /// Returns the output buffer object
+    FIFOSamplePipe *getOutput() { return &outputBuffer; };
+
+    /// Returns the input buffer object
+    FIFOSamplePipe *getInput() { return &inputBuffer; };
+
+    /// Sets new target tempo. Normal tempo = 'SCALE', smaller values represent slower
+    /// tempo, larger faster tempo.
+    void setTempo(double newTempo);
+
+    /// Returns nonzero if there aren't any samples available for outputting.
+    virtual void clear() override;
+
+    /// Clears the input buffer
+    void clearInput();
+
+    /// Sets the number of channels, 1 = mono, 2 = stereo
+    void setChannels(int numChannels);
+
+    /// Enables/disables the quick position seeking algorithm. Zero to disable,
+    /// nonzero to enable
+    void enableQuickSeek(bool enable);
+
+    /// Returns nonzero if the quick seeking algorithm is enabled.
+    bool isQuickSeekEnabled() const;
+
+    /// Sets routine control parameters. These control are certain time constants
+    /// defining how the sound is stretched to the desired duration.
+    //
+    /// 'sampleRate' = sample rate of the sound
+    /// 'sequenceMS' = one processing sequence length in milliseconds
+    /// 'seekwindowMS' = seeking window length for scanning the best overlapping
+    ///      position
+    /// 'overlapMS' = overlapping length
+    void setParameters(int sampleRate,          ///< Samplerate of sound being processed (Hz)
+                       int sequenceMS = -1,     ///< Single processing sequence length (ms)
+                       int seekwindowMS = -1,   ///< Offset seeking window length (ms)
+                       int overlapMS = -1       ///< Sequence overlapping length (ms)
+                       );
+
+    /// Get routine control parameters, see setParameters() function.
+    /// Any of the parameters to this function can be nullptr, in such case corresponding parameter
+    /// value isn't returned.
+    void getParameters(int *pSampleRate, int *pSequenceMs, int *pSeekWindowMs, int *pOverlapMs) const;
+
+    /// Adds 'numsamples' pcs of samples from the 'samples' memory position into
+    /// the input of the object.
+    virtual void putSamples(
+            const SAMPLETYPE *samples,  ///< Input sample data
+            uint numSamples                         ///< Number of samples in 'samples' so that one sample
+                                                    ///< contains both channels if stereo
+            ) override;
+
+    /// return nominal input sample requirement for triggering a processing batch
+    int getInputSampleReq() const
+    {
+        return (int)(nominalSkip + 0.5);
+    }
+
+    /// return nominal output sample amount when running a processing batch
+    int getOutputBatchSize() const
+    {
+        return seekWindowLength - overlapLength;
+    }
+
+	/// return approximate initial input-output latency
+	int getLatency() const
+	{
+		return sampleReq;
+	}
+};
+
+
+// Implementation-specific class declarations:
+
+#ifdef SOUNDTOUCH_ALLOW_MMX
+    /// Class that implements MMX optimized routines for 16bit integer samples type.
+    class TDStretchMMX : public TDStretch
+    {
+    protected:
+        double calcCrossCorr(const short *mixingPos, const short *compare, double &norm) override;
+        double calcCrossCorrAccumulate(const short *mixingPos, const short *compare, double &norm) override;
+        virtual void overlapStereo(short *output, const short *input) const override;
+        virtual void clearCrossCorrState() override;
+    };
+#endif /// SOUNDTOUCH_ALLOW_MMX
+
+
+#ifdef SOUNDTOUCH_ALLOW_SSE
+    /// Class that implements SSE optimized routines for floating point samples type.
+    class TDStretchSSE : public TDStretch
+    {
+    protected:
+        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
+
+}
+#endif  /// TDStretch_H

source/SoundTouch/cpu_detect.h

@@ -1,62 +1,55 @@
-////////////////////////////////////////////////////////////////////////////////
-///
-/// A header file for detecting the Intel MMX instructions set extension.
-///
-/// Please see 'mmx_win.cpp', 'mmx_cpp.cpp' and 'mmx_non_x86.cpp' for the 
-/// routine implementations for x86 Windows, x86 gnu version and non-x86 
-/// platforms, respectively.
-///
-/// Author        : Copyright (c) Olli Parviainen
-/// Author e-mail : oparviai 'at' iki.fi
-/// SoundTouch WWW: http://www.surina.net/soundtouch
-///
-////////////////////////////////////////////////////////////////////////////////
-//
-// Last changed  : $Date$
-// File revision : $Revision: 4 $
-//
-// $Id$
-//
-////////////////////////////////////////////////////////////////////////////////
-//
-// License :
-//
-//  SoundTouch audio processing library
-//  Copyright (c) Olli Parviainen
-//
-//  This library is free software; you can redistribute it and/or
-//  modify it under the terms of the GNU Lesser General Public
-//  License as published by the Free Software Foundation; either
-//  version 2.1 of the License, or (at your option) any later version.
-//
-//  This library is distributed in the hope that it will be useful,
-//  but WITHOUT ANY WARRANTY; without even the implied warranty of
-//  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-//  Lesser General Public License for more details.
-//
-//  You should have received a copy of the GNU Lesser General Public
-//  License along with this library; if not, write to the Free Software
-//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
-//
-////////////////////////////////////////////////////////////////////////////////
-
-#ifndef _CPU_DETECT_H_
-#define _CPU_DETECT_H_
-
-#include "STTypes.h"
-
-#define SUPPORT_MMX         0x0001
-#define SUPPORT_3DNOW       0x0002
-#define SUPPORT_ALTIVEC     0x0004
-#define SUPPORT_SSE         0x0008
-#define SUPPORT_SSE2        0x0010
-
-/// Checks which instruction set extensions are supported by the CPU.
-///
-/// \return A bitmask of supported extensions, see SUPPORT_... defines.
-uint detectCPUextensions(void);
-
-/// Disables given set of instruction extensions. See SUPPORT_... defines.
-void disableExtensions(uint wDisableMask);
-
-#endif  // _CPU_DETECT_H_
+////////////////////////////////////////////////////////////////////////////////
+///
+/// A header file for detecting the Intel MMX instructions set extension.
+///
+/// Please see 'mmx_win.cpp', 'mmx_cpp.cpp' and 'mmx_non_x86.cpp' for the
+/// routine implementations for x86 Windows, x86 gnu version and non-x86
+/// platforms, respectively.
+///
+/// Author        : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+//  SoundTouch audio processing library
+//  Copyright (c) Olli Parviainen
+//
+//  This library is free software; you can redistribute it and/or
+//  modify it under the terms of the GNU Lesser General Public
+//  License as published by the Free Software Foundation; either
+//  version 2.1 of the License, or (at your option) any later version.
+//
+//  This library is distributed in the hope that it will be useful,
+//  but WITHOUT ANY WARRANTY; without even the implied warranty of
+//  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+//  Lesser General Public License for more details.
+//
+//  You should have received a copy of the GNU Lesser General Public
+//  License along with this library; if not, write to the Free Software
+//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#ifndef _CPU_DETECT_H_
+#define _CPU_DETECT_H_
+
+#include "STTypes.h"
+
+#define SUPPORT_MMX         0x0001
+#define SUPPORT_3DNOW       0x0002
+#define SUPPORT_ALTIVEC     0x0004
+#define SUPPORT_SSE         0x0008
+#define SUPPORT_SSE2        0x0010
+
+/// Checks which instruction set extensions are supported by the CPU.
+///
+/// \return A bitmask of supported extensions, see SUPPORT_... defines.
+uint detectCPUextensions(void);
+
+/// Disables given set of instruction extensions. See SUPPORT_... defines.
+void disableExtensions(uint wDisableMask);
+
+#endif  // _CPU_DETECT_H_

source/SoundTouch/cpu_detect_x86.cpp

@@ -1,139 +1,130 @@
-////////////////////////////////////////////////////////////////////////////////
-///
-/// Generic version of the x86 CPU extension detection routine.
-///
-/// This file is for GNU & other non-Windows compilers, see 'cpu_detect_x86_win.cpp' 
-/// for the Microsoft compiler version.
-///
-/// Author        : Copyright (c) Olli Parviainen
-/// Author e-mail : oparviai 'at' iki.fi
-/// SoundTouch WWW: http://www.surina.net/soundtouch
-///
-////////////////////////////////////////////////////////////////////////////////
-//
-// Last changed  : $Date$
-// File revision : $Revision: 4 $
-//
-// $Id$
-//
-////////////////////////////////////////////////////////////////////////////////
-//
-// License :
-//
-//  SoundTouch audio processing library
-//  Copyright (c) Olli Parviainen
-//
-//  This library is free software; you can redistribute it and/or
-//  modify it under the terms of the GNU Lesser General Public
-//  License as published by the Free Software Foundation; either
-//  version 2.1 of the License, or (at your option) any later version.
-//
-//  This library is distributed in the hope that it will be useful,
-//  but WITHOUT ANY WARRANTY; without even the implied warranty of
-//  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-//  Lesser General Public License for more details.
-//
-//  You should have received a copy of the GNU Lesser General Public
-//  License along with this library; if not, write to the Free Software
-//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
-//
-////////////////////////////////////////////////////////////////////////////////
-
-#include "cpu_detect.h"
-#include "STTypes.h"
-
-#if defined(SOUNDTOUCH_ALLOW_X86_OPTIMIZATIONS)
-
-    #if defined(__GNUC__) && defined(__i386__)
-        // gcc
-        #include "cpuid.h"
-    #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
-
-
-//////////////////////////////////////////////////////////////////////////////
-//
-// processor instructions extension detection routines
-//
-//////////////////////////////////////////////////////////////////////////////
-
-// Flag variable indicating whick ISA extensions are disabled (for debugging)
-static uint _dwDisabledISA = 0x00;      // 0xffffffff; //<- use this to disable all extensions
-
-// Disables given set of instruction extensions. See SUPPORT_... defines.
-void disableExtensions(uint dwDisableMask)
-{
-    _dwDisabledISA = dwDisableMask;
-}
-
-
-
-/// Checks which instruction set extensions are supported by the CPU.
-uint detectCPUextensions(void)
-{
-/// If building for a 64bit system (no Itanium) and the user wants optimizations.
-/// Return the OR of SUPPORT_{MMX,SSE,SSE2}. 11001 or 0x19.
-/// Keep the _dwDisabledISA test (2 more operations, could be eliminated).
-#if ((defined(__GNUC__) && defined(__x86_64__)) \
-    || defined(_M_X64))  \
-    && defined(SOUNDTOUCH_ALLOW_X86_OPTIMIZATIONS)
-    return 0x19 & ~_dwDisabledISA;
-
-/// If building for a 32bit system and the user wants optimizations.
-/// Keep the _dwDisabledISA test (2 more operations, could be eliminated).
-#elif ((defined(__GNUC__) && defined(__i386__)) \
-    || defined(_M_IX86))  \
-    && defined(SOUNDTOUCH_ALLOW_X86_OPTIMIZATIONS)
-
-    if (_dwDisabledISA == 0xffffffff) return 0;
- 
-    uint res = 0;
- 
-#if defined(__GNUC__)
-    // GCC version of cpuid. Requires GCC 4.3.0 or later for __cpuid intrinsic support.
-    uint eax, ebx, ecx, edx;  // unsigned int is the standard type. uint is defined by the compiler and not guaranteed to be portable.
-
-    // Check if no cpuid support.
-    if (!__get_cpuid (1, &eax, &ebx, &ecx, &edx)) return 0; // always disable extensions.
-
-    if (edx & bit_MMX)  res = res | SUPPORT_MMX;
-    if (edx & bit_SSE)  res = res | SUPPORT_SSE;
-    if (edx & bit_SSE2) res = res | SUPPORT_SSE2;
-
-#else
-    // Window / VS version of cpuid. Notice that Visual Studio 2005 or later required 
-    // for __cpuid intrinsic support.
-    int reg[4] = {-1};
-
-    // Check if no cpuid support.
-    __cpuid(reg,0);
-    if ((unsigned int)reg[0] == 0) return 0; // always disable extensions.
-
-    __cpuid(reg,1);
-    if ((unsigned int)reg[3] & bit_MMX)  res = res | SUPPORT_MMX;
-    if ((unsigned int)reg[3] & bit_SSE)  res = res | SUPPORT_SSE;
-    if ((unsigned int)reg[3] & bit_SSE2) res = res | SUPPORT_SSE2;
-
-#endif
-
-    return res & ~_dwDisabledISA;
-
-#else
-
-/// One of these is true:
-/// 1) We don't want optimizations.
-/// 2) Using an unsupported compiler.
-/// 3) Running on a non-x86 platform.
-    return 0;
-
-#endif
-}
+////////////////////////////////////////////////////////////////////////////////
+///
+/// Generic version of the x86 CPU extension detection routine.
+///
+/// This file is for GNU & other non-Windows compilers, see 'cpu_detect_x86_win.cpp'
+/// for the Microsoft compiler version.
+///
+/// Author        : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+//  SoundTouch audio processing library
+//  Copyright (c) Olli Parviainen
+//
+//  This library is free software; you can redistribute it and/or
+//  modify it under the terms of the GNU Lesser General Public
+//  License as published by the Free Software Foundation; either
+//  version 2.1 of the License, or (at your option) any later version.
+//
+//  This library is distributed in the hope that it will be useful,
+//  but WITHOUT ANY WARRANTY; without even the implied warranty of
+//  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+//  Lesser General Public License for more details.
+//
+//  You should have received a copy of the GNU Lesser General Public
+//  License along with this library; if not, write to the Free Software
+//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#include "cpu_detect.h"
+#include "STTypes.h"
+
+
+#if defined(SOUNDTOUCH_ALLOW_X86_OPTIMIZATIONS)
+
+   #if defined(__GNUC__) && defined(__i386__)
+       // gcc
+       #include "cpuid.h"
+   #elif defined(_M_IX86)
+       // windows non-gcc
+       #include <intrin.h>
+   #endif
+
+   #define bit_MMX     (1 << 23)
+   #define bit_SSE     (1 << 25)
+   #define bit_SSE2    (1 << 26)
+#endif
+
+
+//////////////////////////////////////////////////////////////////////////////
+//
+// processor instructions extension detection routines
+//
+//////////////////////////////////////////////////////////////////////////////
+
+// Flag variable indicating whick ISA extensions are disabled (for debugging)
+static uint _dwDisabledISA = 0x00;      // 0xffffffff; //<- use this to disable all extensions
+
+// Disables given set of instruction extensions. See SUPPORT_... defines.
+void disableExtensions(uint dwDisableMask)
+{
+    _dwDisabledISA = dwDisableMask;
+}
+
+
+/// Checks which instruction set extensions are supported by the CPU.
+uint detectCPUextensions(void)
+{
+/// If building for a 64bit system (no Itanium) and the user wants optimizations.
+/// Return the OR of SUPPORT_{MMX,SSE,SSE2}. 11001 or 0x19.
+/// Keep the _dwDisabledISA test (2 more operations, could be eliminated).
+#if ((defined(__GNUC__) && defined(__x86_64__)) \
+    || defined(_M_X64))  \
+    && defined(SOUNDTOUCH_ALLOW_X86_OPTIMIZATIONS)
+    return 0x19 & ~_dwDisabledISA;
+
+/// If building for a 32bit system and the user wants optimizations.
+/// Keep the _dwDisabledISA test (2 more operations, could be eliminated).
+#elif ((defined(__GNUC__) && defined(__i386__)) \
+    || defined(_M_IX86))  \
+    && defined(SOUNDTOUCH_ALLOW_X86_OPTIMIZATIONS)
+
+    if (_dwDisabledISA == 0xffffffff) return 0;
+
+    uint res = 0;
+
+#if defined(__GNUC__)
+    // GCC version of cpuid. Requires GCC 4.3.0 or later for __cpuid intrinsic support.
+    uint eax, ebx, ecx, edx;  // unsigned int is the standard type. uint is defined by the compiler and not guaranteed to be portable.
+
+    // Check if no cpuid support.
+    if (!__get_cpuid (1, &eax, &ebx, &ecx, &edx)) return 0; // always disable extensions.
+
+    if (edx & bit_MMX)  res = res | SUPPORT_MMX;
+    if (edx & bit_SSE)  res = res | SUPPORT_SSE;
+    if (edx & bit_SSE2) res = res | SUPPORT_SSE2;
+
+#else
+    // Window / VS version of cpuid. Notice that Visual Studio 2005 or later required
+    // for __cpuid intrinsic support.
+    int reg[4] = {-1};
+
+    // Check if no cpuid support.
+    __cpuid(reg,0);
+    if ((unsigned int)reg[0] == 0) return 0; // always disable extensions.
+
+    __cpuid(reg,1);
+    if ((unsigned int)reg[3] & bit_MMX)  res = res | SUPPORT_MMX;
+    if ((unsigned int)reg[3] & bit_SSE)  res = res | SUPPORT_SSE;
+    if ((unsigned int)reg[3] & bit_SSE2) res = res | SUPPORT_SSE2;
+
+#endif
+
+    return res & ~_dwDisabledISA;
+
+#else
+
+/// One of these is true:
+/// 1) We don't want optimizations.
+/// 2) Using an unsupported compiler.
+/// 3) Running on a non-x86 platform.
+    return 0;
+
+#endif
+}

source/SoundTouch/mmx_optimized.cpp

@@ -1,317 +1,396 @@
-////////////////////////////////////////////////////////////////////////////////
-///
-/// MMX optimized routines. All MMX optimized functions have been gathered into 
-/// this single source code file, regardless to their class or original source 
-/// code file, in order to ease porting the library to other compiler and 
-/// processor platforms.
-///
-/// The MMX-optimizations are programmed using MMX compiler intrinsics that
-/// are supported both by Microsoft Visual C++ and GCC compilers, so this file
-/// should compile with both toolsets.
-///
-/// NOTICE: If using Visual Studio 6.0, you'll need to install the "Visual C++ 
-/// 6.0 processor pack" update to support compiler intrinsic syntax. The update
-/// is available for download at Microsoft Developers Network, see here:
-/// http://msdn.microsoft.com/en-us/vstudio/aa718349.aspx
-///
-/// 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
-//  Copyright (c) Olli Parviainen
-//
-//  This library is free software; you can redistribute it and/or
-//  modify it under the terms of the GNU Lesser General Public
-//  License as published by the Free Software Foundation; either
-//  version 2.1 of the License, or (at your option) any later version.
-//
-//  This library is distributed in the hope that it will be useful,
-//  but WITHOUT ANY WARRANTY; without even the implied warranty of
-//  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-//  Lesser General Public License for more details.
-//
-//  You should have received a copy of the GNU Lesser General Public
-//  License along with this library; if not, write to the Free Software
-//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
-//
-////////////////////////////////////////////////////////////////////////////////
-
-#include "STTypes.h"
-
-#ifdef SOUNDTOUCH_ALLOW_MMX
-// MMX routines available only with integer sample type
-
-using namespace soundtouch;
-
-//////////////////////////////////////////////////////////////////////////////
-//
-// implementation of MMX optimized functions of class 'TDStretchMMX'
-//
-//////////////////////////////////////////////////////////////////////////////
-
-#include "TDStretch.h"
-#include <mmintrin.h>
-#include <limits.h>
-#include <math.h>
-
-
-// Calculates cross correlation of two buffers
-double TDStretchMMX::calcCrossCorr(const short *pV1, const short *pV2) const
-{
-    const __m64 *pVec1, *pVec2;
-    __m64 shifter;
-    __m64 accu, normaccu;
-    long corr, norm;
-    int i;
-   
-    pVec1 = (__m64*)pV1;
-    pVec2 = (__m64*)pV2;
-
-    shifter = _m_from_int(overlapDividerBits);
-    normaccu = 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, temp2;
-
-        // dictionary of instructions:
-        // _m_pmaddwd   : 4*16bit multiply-add, resulting two 32bits = [a0*b0+a1*b1 ; a2*b2+a3*b3]
-        // _mm_add_pi32 : 2*32bit add
-        // _m_psrad     : 32bit right-shift
-
-        temp = _mm_add_pi32(_mm_madd_pi16(pVec1[0], pVec2[0]),
-                            _mm_madd_pi16(pVec1[1], pVec2[1]));
-        temp2 = _mm_add_pi32(_mm_madd_pi16(pVec1[0], pVec1[0]),
-                             _mm_madd_pi16(pVec1[1], pVec1[1]));
-        accu = _mm_add_pi32(accu, _mm_sra_pi32(temp, shifter));
-        normaccu = _mm_add_pi32(normaccu, _mm_sra_pi32(temp2, shifter));
-
-        temp = _mm_add_pi32(_mm_madd_pi16(pVec1[2], pVec2[2]),
-                            _mm_madd_pi16(pVec1[3], pVec2[3]));
-        temp2 = _mm_add_pi32(_mm_madd_pi16(pVec1[2], pVec1[2]),
-                             _mm_madd_pi16(pVec1[3], pVec1[3]));
-        accu = _mm_add_pi32(accu, _mm_sra_pi32(temp, shifter));
-        normaccu = _mm_add_pi32(normaccu, _mm_sra_pi32(temp2, shifter));
-
-        pVec1 += 4;
-        pVec2 += 4;
-    }
-
-    // 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);
-
-    normaccu = _mm_add_pi32(normaccu, _mm_srli_si64(normaccu, 32));
-    norm = _m_to_int(normaccu);
-
-    // Clear MMS state
-    _m_empty();
-
-    // Normalize result by dividing by sqrt(norm) - this step is easiest 
-    // done using floating point operation
-    if (norm == 0) norm = 1;    // to avoid div by zero
-
-    return (double)corr / sqrt((double)norm);
-    // Note: Warning about the missing EMMS instruction is harmless
-    // as it'll be called elsewhere.
-}
-
-
-
-void TDStretchMMX::clearCrossCorrState()
-{
-    // Clear MMS state
-    _m_empty();
-    //_asm EMMS;
-}
-
-
-
-// MMX-optimized version of the function overlapStereo
-void TDStretchMMX::overlapStereo(short *output, const short *input) const
-{
-    const __m64 *pVinput, *pVMidBuf;
-    __m64 *pVdest;
-    __m64 mix1, mix2, adder, shifter;
-    int i;
-
-    pVinput  = (const __m64*)input;
-    pVMidBuf = (const __m64*)pMidBuffer;
-    pVdest   = (__m64*)output;
-
-    // mix1  = mixer values for 1st stereo sample
-    // mix1  = mixer values for 2nd stereo sample
-    // adder = adder for updating mixer values after each round
-    
-    mix1  = _mm_set_pi16(0, overlapLength,   0, overlapLength);
-    adder = _mm_set_pi16(1, -1, 1, -1);
-    mix2  = _mm_add_pi16(mix1, adder);
-    adder = _mm_add_pi16(adder, adder);
-
-    // Overlaplength-division by shifter. "+1" is to account for "-1" deduced in
-    // overlapDividerBits calculation earlier.
-    shifter = _m_from_int(overlapDividerBits + 1);
-
-    for (i = 0; i < overlapLength / 4; i ++)
-    {
-        __m64 temp1, temp2;
-                
-        // load & shuffle data so that input & mixbuffer data samples are paired
-        temp1 = _mm_unpacklo_pi16(pVMidBuf[0], pVinput[0]);     // = i0l m0l i0r m0r
-        temp2 = _mm_unpackhi_pi16(pVMidBuf[0], pVinput[0]);     // = i1l m1l i1r m1r
-
-        // temp = (temp .* mix) >> shifter
-        temp1 = _mm_sra_pi32(_mm_madd_pi16(temp1, mix1), shifter);
-        temp2 = _mm_sra_pi32(_mm_madd_pi16(temp2, mix2), shifter);
-        pVdest[0] = _mm_packs_pi32(temp1, temp2); // pack 2*2*32bit => 4*16bit
-
-        // update mix += adder
-        mix1 = _mm_add_pi16(mix1, adder);
-        mix2 = _mm_add_pi16(mix2, adder);
-
-        // --- second round begins here ---
-
-        // load & shuffle data so that input & mixbuffer data samples are paired
-        temp1 = _mm_unpacklo_pi16(pVMidBuf[1], pVinput[1]);       // = i2l m2l i2r m2r
-        temp2 = _mm_unpackhi_pi16(pVMidBuf[1], pVinput[1]);       // = i3l m3l i3r m3r
-
-        // temp = (temp .* mix) >> shifter
-        temp1 = _mm_sra_pi32(_mm_madd_pi16(temp1, mix1), shifter);
-        temp2 = _mm_sra_pi32(_mm_madd_pi16(temp2, mix2), shifter);
-        pVdest[1] = _mm_packs_pi32(temp1, temp2); // pack 2*2*32bit => 4*16bit
-
-        // update mix += adder
-        mix1 = _mm_add_pi16(mix1, adder);
-        mix2 = _mm_add_pi16(mix2, adder);
-
-        pVinput  += 2;
-        pVMidBuf += 2;
-        pVdest   += 2;
-    }
-
-    _m_empty(); // clear MMS state
-}
-
-
-//////////////////////////////////////////////////////////////////////////////
-//
-// implementation of MMX optimized functions of class 'FIRFilter'
-//
-//////////////////////////////////////////////////////////////////////////////
-
-#include "FIRFilter.h"
-
-
-FIRFilterMMX::FIRFilterMMX() : FIRFilter()
-{
-    filterCoeffsUnalign = NULL;
-}
-
-
-FIRFilterMMX::~FIRFilterMMX()
-{
-    delete[] filterCoeffsUnalign;
-}
-
-
-// (overloaded) Calculates filter coefficients for MMX routine
-void FIRFilterMMX::setCoefficients(const short *coeffs, uint newLength, uint uResultDivFactor)
-{
-    uint i;
-    FIRFilter::setCoefficients(coeffs, newLength, uResultDivFactor);
-
-    // Ensure that filter coeffs array is aligned to 16-byte boundary
-    delete[] filterCoeffsUnalign;
-    filterCoeffsUnalign = new short[2 * newLength + 8];
-    filterCoeffsAlign = (short *)(((ulong)filterCoeffsUnalign + 15) & -16);
-
-    // rearrange the filter coefficients for mmx routines 
-    for (i = 0;i < length; i += 4) 
-    {
-        filterCoeffsAlign[2 * i + 0] = coeffs[i + 0];
-        filterCoeffsAlign[2 * i + 1] = coeffs[i + 2];
-        filterCoeffsAlign[2 * i + 2] = coeffs[i + 0];
-        filterCoeffsAlign[2 * i + 3] = coeffs[i + 2];
-
-        filterCoeffsAlign[2 * i + 4] = coeffs[i + 1];
-        filterCoeffsAlign[2 * i + 5] = coeffs[i + 3];
-        filterCoeffsAlign[2 * i + 6] = coeffs[i + 1];
-        filterCoeffsAlign[2 * i + 7] = coeffs[i + 3];
-    }
-}
-
-
-
-// mmx-optimized version of the filter routine for stereo sound
-uint FIRFilterMMX::evaluateFilterStereo(short *dest, const short *src, uint numSamples) const
-{
-    // Create stack copies of the needed member variables for asm routines :
-    uint i, j;
-    __m64 *pVdest = (__m64*)dest;
-
-    if (length < 2) return 0;
-
-    for (i = 0; i < (numSamples - length) / 2; i ++)
-    {
-        __m64 accu1;
-        __m64 accu2;
-        const __m64 *pVsrc = (const __m64*)src;
-        const __m64 *pVfilter = (const __m64*)filterCoeffsAlign;
-
-        accu1 = accu2 = _mm_setzero_si64();
-        for (j = 0; j < lengthDiv8 * 2; j ++)
-        {
-            __m64 temp1, temp2;
-
-            temp1 = _mm_unpacklo_pi16(pVsrc[0], pVsrc[1]);  // = l2 l0 r2 r0
-            temp2 = _mm_unpackhi_pi16(pVsrc[0], pVsrc[1]);  // = l3 l1 r3 r1
-
-            accu1 = _mm_add_pi32(accu1, _mm_madd_pi16(temp1, pVfilter[0]));  // += l2*f2+l0*f0 r2*f2+r0*f0
-            accu1 = _mm_add_pi32(accu1, _mm_madd_pi16(temp2, pVfilter[1]));  // += l3*f3+l1*f1 r3*f3+r1*f1
-
-            temp1 = _mm_unpacklo_pi16(pVsrc[1], pVsrc[2]);  // = l4 l2 r4 r2
-
-            accu2 = _mm_add_pi32(accu2, _mm_madd_pi16(temp2, pVfilter[0]));  // += l3*f2+l1*f0 r3*f2+r1*f0
-            accu2 = _mm_add_pi32(accu2, _mm_madd_pi16(temp1, pVfilter[1]));  // += l4*f3+l2*f1 r4*f3+r2*f1
-
-            // accu1 += l2*f2+l0*f0 r2*f2+r0*f0
-            //       += l3*f3+l1*f1 r3*f3+r1*f1
-
-            // accu2 += l3*f2+l1*f0 r3*f2+r1*f0
-            //          l4*f3+l2*f1 r4*f3+r2*f1
-
-            pVfilter += 2;
-            pVsrc += 2;
-        }
-        // accu >>= resultDivFactor
-        accu1 = _mm_srai_pi32(accu1, resultDivFactor);
-        accu2 = _mm_srai_pi32(accu2, resultDivFactor);
-
-        // pack 2*2*32bits => 4*16 bits
-        pVdest[0] = _mm_packs_pi32(accu1, accu2);
-        src += 4;
-        pVdest ++;
-    }
-
-   _m_empty();  // clear emms state
-
-    return (numSamples & 0xfffffffe) - length;
-}
-
-#endif  // SOUNDTOUCH_ALLOW_MMX
+////////////////////////////////////////////////////////////////////////////////
+///
+/// MMX optimized routines. All MMX optimized functions have been gathered into
+/// this single source code file, regardless to their class or original source
+/// code file, in order to ease porting the library to other compiler and
+/// processor platforms.
+///
+/// The MMX-optimizations are programmed using MMX compiler intrinsics that
+/// are supported both by Microsoft Visual C++ and GCC compilers, so this file
+/// should compile with both toolsets.
+///
+/// NOTICE: If using Visual Studio 6.0, you'll need to install the "Visual C++
+/// 6.0 processor pack" update to support compiler intrinsic syntax. The update
+/// is available for download at Microsoft Developers Network, see here:
+/// http://msdn.microsoft.com/en-us/vstudio/aa718349.aspx
+///
+/// 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 "STTypes.h"
+
+#ifdef SOUNDTOUCH_ALLOW_MMX
+// MMX routines available only with integer sample type
+
+using namespace soundtouch;
+
+//////////////////////////////////////////////////////////////////////////////
+//
+// implementation of MMX optimized functions of class 'TDStretchMMX'
+//
+//////////////////////////////////////////////////////////////////////////////
+
+#include "TDStretch.h"
+#include <mmintrin.h>
+#include <limits.h>
+#include <math.h>
+
+
+// Calculates cross correlation of two buffers
+double TDStretchMMX::calcCrossCorr(const short *pV1, const short *pV2, double &dnorm)
+{
+    const __m64 *pVec1, *pVec2;
+    __m64 shifter;
+    __m64 accu, normaccu;
+    long corr, norm;
+    int i;
+
+    pVec1 = (__m64*)pV1;
+    pVec2 = (__m64*)pV2;
+
+    shifter = _m_from_int(overlapDividerBitsNorm);
+    normaccu = 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, temp2;
+
+        // 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));
+        temp2 = _mm_add_pi32(_mm_sra_pi32(_mm_madd_pi16(pVec1[0], pVec1[0]), shifter),
+                            _mm_sra_pi32(_mm_madd_pi16(pVec1[1], pVec1[1]), shifter));
+        accu = _mm_add_pi32(accu, temp);
+        normaccu = _mm_add_pi32(normaccu, temp2);
+
+        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));
+        temp2 = _mm_add_pi32(_mm_sra_pi32(_mm_madd_pi16(pVec1[2], pVec1[2]), shifter),
+                            _mm_sra_pi32(_mm_madd_pi16(pVec1[3], pVec1[3]), shifter));
+        accu = _mm_add_pi32(accu, temp);
+        normaccu = _mm_add_pi32(normaccu, temp2);
+
+        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);
+
+    normaccu = _mm_add_pi32(normaccu, _mm_srli_si64(normaccu, 32));
+    norm = _m_to_int(normaccu);
+
+    // Clear MMS state
+    _m_empty();
+
+    if (norm > (long)maxnorm)
+    {
+        // modify 'maxnorm' inside critical section to avoid multi-access conflict if in OpenMP mode
+        #pragma omp critical
+        if (norm > (long)maxnorm)
+        {
+            maxnorm = norm;
+        }
+    }
+
+    // Normalize result by dividing by sqrt(norm) - this step is easiest
+    // done using floating point operation
+    dnorm = (double)norm;
+
+    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()
+{
+    // Clear MMS state
+    _m_empty();
+    //_asm EMMS;
+}
+
+
+// MMX-optimized version of the function overlapStereo
+void TDStretchMMX::overlapStereo(short *output, const short *input) const
+{
+    const __m64 *pVinput, *pVMidBuf;
+    __m64 *pVdest;
+    __m64 mix1, mix2, adder, shifter;
+    int i;
+
+    pVinput  = (const __m64*)input;
+    pVMidBuf = (const __m64*)pMidBuffer;
+    pVdest   = (__m64*)output;
+
+    // mix1  = mixer values for 1st stereo sample
+    // mix1  = mixer values for 2nd stereo sample
+    // adder = adder for updating mixer values after each round
+
+    mix1  = _mm_set_pi16(0, overlapLength,   0, overlapLength);
+    adder = _mm_set_pi16(1, -1, 1, -1);
+    mix2  = _mm_add_pi16(mix1, adder);
+    adder = _mm_add_pi16(adder, adder);
+
+    // Overlaplength-division by shifter. "+1" is to account for "-1" deduced in
+    // overlapDividerBits calculation earlier.
+    shifter = _m_from_int(overlapDividerBitsPure + 1);
+
+    for (i = 0; i < overlapLength / 4; i ++)
+    {
+        __m64 temp1, temp2;
+
+        // load & shuffle data so that input & mixbuffer data samples are paired
+        temp1 = _mm_unpacklo_pi16(pVMidBuf[0], pVinput[0]);     // = i0l m0l i0r m0r
+        temp2 = _mm_unpackhi_pi16(pVMidBuf[0], pVinput[0]);     // = i1l m1l i1r m1r
+
+        // temp = (temp .* mix) >> shifter
+        temp1 = _mm_sra_pi32(_mm_madd_pi16(temp1, mix1), shifter);
+        temp2 = _mm_sra_pi32(_mm_madd_pi16(temp2, mix2), shifter);
+        pVdest[0] = _mm_packs_pi32(temp1, temp2); // pack 2*2*32bit => 4*16bit
+
+        // update mix += adder
+        mix1 = _mm_add_pi16(mix1, adder);
+        mix2 = _mm_add_pi16(mix2, adder);
+
+        // --- second round begins here ---
+
+        // load & shuffle data so that input & mixbuffer data samples are paired
+        temp1 = _mm_unpacklo_pi16(pVMidBuf[1], pVinput[1]);       // = i2l m2l i2r m2r
+        temp2 = _mm_unpackhi_pi16(pVMidBuf[1], pVinput[1]);       // = i3l m3l i3r m3r
+
+        // temp = (temp .* mix) >> shifter
+        temp1 = _mm_sra_pi32(_mm_madd_pi16(temp1, mix1), shifter);
+        temp2 = _mm_sra_pi32(_mm_madd_pi16(temp2, mix2), shifter);
+        pVdest[1] = _mm_packs_pi32(temp1, temp2); // pack 2*2*32bit => 4*16bit
+
+        // update mix += adder
+        mix1 = _mm_add_pi16(mix1, adder);
+        mix2 = _mm_add_pi16(mix2, adder);
+
+        pVinput  += 2;
+        pVMidBuf += 2;
+        pVdest   += 2;
+    }
+
+    _m_empty(); // clear MMS state
+}
+
+
+//////////////////////////////////////////////////////////////////////////////
+//
+// implementation of MMX optimized functions of class 'FIRFilter'
+//
+//////////////////////////////////////////////////////////////////////////////
+
+#include "FIRFilter.h"
+
+
+FIRFilterMMX::FIRFilterMMX() : FIRFilter()
+{
+    filterCoeffsAlign = nullptr;
+    filterCoeffsUnalign = nullptr;
+}
+
+
+FIRFilterMMX::~FIRFilterMMX()
+{
+    delete[] filterCoeffsUnalign;
+}
+
+
+// (overloaded) Calculates filter coefficients for MMX routine
+void FIRFilterMMX::setCoefficients(const short *coeffs, uint newLength, uint uResultDivFactor)
+{
+    uint i;
+    FIRFilter::setCoefficients(coeffs, newLength, uResultDivFactor);
+
+    // Ensure that filter coeffs array is aligned to 16-byte boundary
+    delete[] filterCoeffsUnalign;
+    filterCoeffsUnalign = new short[2 * newLength + 8];
+    filterCoeffsAlign = (short *)SOUNDTOUCH_ALIGN_POINTER_16(filterCoeffsUnalign);
+
+    // rearrange the filter coefficients for mmx routines
+    for (i = 0;i < length; i += 4)
+    {
+        filterCoeffsAlign[2 * i + 0] = coeffs[i + 0];
+        filterCoeffsAlign[2 * i + 1] = coeffs[i + 2];
+        filterCoeffsAlign[2 * i + 2] = coeffs[i + 0];
+        filterCoeffsAlign[2 * i + 3] = coeffs[i + 2];
+
+        filterCoeffsAlign[2 * i + 4] = coeffs[i + 1];
+        filterCoeffsAlign[2 * i + 5] = coeffs[i + 3];
+        filterCoeffsAlign[2 * i + 6] = coeffs[i + 1];
+        filterCoeffsAlign[2 * i + 7] = coeffs[i + 3];
+    }
+}
+
+
+// mmx-optimized version of the filter routine for stereo sound
+uint FIRFilterMMX::evaluateFilterStereo(short *dest, const short *src, uint numSamples) const
+{
+    // Create stack copies of the needed member variables for asm routines :
+    uint i, j;
+    __m64 *pVdest = (__m64*)dest;
+
+    if (length < 2) return 0;
+
+    for (i = 0; i < (numSamples - length) / 2; i ++)
+    {
+        __m64 accu1;
+        __m64 accu2;
+        const __m64 *pVsrc = (const __m64*)src;
+        const __m64 *pVfilter = (const __m64*)filterCoeffsAlign;
+
+        accu1 = accu2 = _mm_setzero_si64();
+        for (j = 0; j < lengthDiv8 * 2; j ++)
+        {
+            __m64 temp1, temp2;
+
+            temp1 = _mm_unpacklo_pi16(pVsrc[0], pVsrc[1]);  // = l2 l0 r2 r0
+            temp2 = _mm_unpackhi_pi16(pVsrc[0], pVsrc[1]);  // = l3 l1 r3 r1
+
+            accu1 = _mm_add_pi32(accu1, _mm_madd_pi16(temp1, pVfilter[0]));  // += l2*f2+l0*f0 r2*f2+r0*f0
+            accu1 = _mm_add_pi32(accu1, _mm_madd_pi16(temp2, pVfilter[1]));  // += l3*f3+l1*f1 r3*f3+r1*f1
+
+            temp1 = _mm_unpacklo_pi16(pVsrc[1], pVsrc[2]);  // = l4 l2 r4 r2
+
+            accu2 = _mm_add_pi32(accu2, _mm_madd_pi16(temp2, pVfilter[0]));  // += l3*f2+l1*f0 r3*f2+r1*f0
+            accu2 = _mm_add_pi32(accu2, _mm_madd_pi16(temp1, pVfilter[1]));  // += l4*f3+l2*f1 r4*f3+r2*f1
+
+            // accu1 += l2*f2+l0*f0 r2*f2+r0*f0
+            //       += l3*f3+l1*f1 r3*f3+r1*f1
+
+            // accu2 += l3*f2+l1*f0 r3*f2+r1*f0
+            //          l4*f3+l2*f1 r4*f3+r2*f1
+
+            pVfilter += 2;
+            pVsrc += 2;
+        }
+        // accu >>= resultDivFactor
+        accu1 = _mm_srai_pi32(accu1, resultDivFactor);
+        accu2 = _mm_srai_pi32(accu2, resultDivFactor);
+
+        // pack 2*2*32bits => 4*16 bits
+        pVdest[0] = _mm_packs_pi32(accu1, accu2);
+        src += 4;
+        pVdest ++;
+    }
+
+   _m_empty();  // clear emms state
+
+    return (numSamples & 0xfffffffe) - length;
+}
+
+#else
+
+// workaround to not complain about empty module
+bool _dontcomplain_mmx_empty;
+
+#endif  // SOUNDTOUCH_ALLOW_MMX

source/SoundTouch/sse_optimized.cpp

@@ -1,361 +1,362 @@
-////////////////////////////////////////////////////////////////////////////////
-///
-/// SSE optimized routines for Pentium-III, Athlon-XP and later CPUs. All SSE 
-/// optimized functions have been gathered into this single source 
-/// code file, regardless to their class or original source code file, in order 
-/// to ease porting the library to other compiler and processor platforms.
-///
-/// The SSE-optimizations are programmed using SSE compiler intrinsics that
-/// are supported both by Microsoft Visual C++ and GCC compilers, so this file
-/// should compile with both toolsets.
-///
-/// NOTICE: If using Visual Studio 6.0, you'll need to install the "Visual C++ 
-/// 6.0 processor pack" update to support SSE instruction set. The update is 
-/// available for download at Microsoft Developers Network, see here:
-/// http://msdn.microsoft.com/en-us/vstudio/aa718349.aspx
-///
-/// If the above URL is expired or removed, go to "http://msdn.microsoft.com" and 
-/// perform a search with keywords "processor pack".
-///
-/// Author        : Copyright (c) Olli Parviainen
-/// Author e-mail : oparviai 'at' iki.fi
-/// SoundTouch WWW: http://www.surina.net/soundtouch
-///
-////////////////////////////////////////////////////////////////////////////////
-//
-// Last changed  : $Date$
-// File revision : $Revision: 4 $
-//
-// $Id$
-//
-////////////////////////////////////////////////////////////////////////////////
-//
-// License :
-//
-//  SoundTouch audio processing library
-//  Copyright (c) Olli Parviainen
-//
-//  This library is free software; you can redistribute it and/or
-//  modify it under the terms of the GNU Lesser General Public
-//  License as published by the Free Software Foundation; either
-//  version 2.1 of the License, or (at your option) any later version.
-//
-//  This library is distributed in the hope that it will be useful,
-//  but WITHOUT ANY WARRANTY; without even the implied warranty of
-//  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-//  Lesser General Public License for more details.
-//
-//  You should have received a copy of the GNU Lesser General Public
-//  License along with this library; if not, write to the Free Software
-//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
-//
-////////////////////////////////////////////////////////////////////////////////
-
-#include "cpu_detect.h"
-#include "STTypes.h"
-
-using namespace soundtouch;
-
-#ifdef SOUNDTOUCH_ALLOW_SSE
-
-// SSE routines available only with float sample type    
-
-//////////////////////////////////////////////////////////////////////////////
-//
-// implementation of SSE optimized functions of class 'TDStretchSSE'
-//
-//////////////////////////////////////////////////////////////////////////////
-
-#include "TDStretch.h"
-#include <xmmintrin.h>
-#include <math.h>
-
-// Calculates cross correlation of two buffers
-double TDStretchSSE::calcCrossCorr(const float *pV1, const float *pV2) const
-{
-    int i;
-    const float *pVec1;
-    const __m128 *pVec2;
-    __m128 vSum, vNorm;
-
-    // Note. It means a major slow-down if the routine needs to tolerate 
-    // unaligned __m128 memory accesses. It's way faster if we can skip 
-    // unaligned slots and use _mm_load_ps instruction instead of _mm_loadu_ps.
-    // This can mean up to ~ 10-fold difference (incl. part of which is
-    // due to skipping every second round for stereo sound though).
-    //
-    // Compile-time define SOUNDTOUCH_ALLOW_NONEXACT_SIMD_OPTIMIZATION is provided
-    // for choosing if this little cheating is allowed.
-
-#ifdef SOUNDTOUCH_ALLOW_NONEXACT_SIMD_OPTIMIZATION
-    // Little cheating allowed, return valid correlation only for 
-    // aligned locations, meaning every second round for stereo sound.
-
-    #define _MM_LOAD    _mm_load_ps
-
-    if (((ulong)pV1) & 15) return -1e50;    // skip unaligned locations
-
-#else
-    // No cheating allowed, use unaligned load & take the resulting
-    // performance hit.
-    #define _MM_LOAD    _mm_loadu_ps
-#endif 
-
-    // ensure overlapLength is divisible by 8
-    assert((overlapLength % 8) == 0);
-
-    // Calculates the cross-correlation value between 'pV1' and 'pV2' vectors
-    // Note: pV2 _must_ be aligned to 16-bit boundary, pV1 need not.
-    pVec1 = (const float*)pV1;
-    pVec2 = (const __m128*)pV2;
-    vSum = vNorm = _mm_setzero_ps();
-
-    // Unroll the loop by factor of 4 * 4 operations. Use same routine for
-    // stereo & mono, for mono it just means twice the amount of unrolling.
-    for (i = 0; i < channels * overlapLength / 16; i ++) 
-    {
-        __m128 vTemp;
-        // vSum += pV1[0..3] * pV2[0..3]
-        vTemp = _MM_LOAD(pVec1);
-        vSum  = _mm_add_ps(vSum,  _mm_mul_ps(vTemp ,pVec2[0]));
-        vNorm = _mm_add_ps(vNorm, _mm_mul_ps(vTemp ,vTemp));
-
-        // vSum += pV1[4..7] * pV2[4..7]
-        vTemp = _MM_LOAD(pVec1 + 4);
-        vSum  = _mm_add_ps(vSum, _mm_mul_ps(vTemp, pVec2[1]));
-        vNorm = _mm_add_ps(vNorm, _mm_mul_ps(vTemp ,vTemp));
-
-        // vSum += pV1[8..11] * pV2[8..11]
-        vTemp = _MM_LOAD(pVec1 + 8);
-        vSum  = _mm_add_ps(vSum, _mm_mul_ps(vTemp, pVec2[2]));
-        vNorm = _mm_add_ps(vNorm, _mm_mul_ps(vTemp ,vTemp));
-
-        // vSum += pV1[12..15] * pV2[12..15]
-        vTemp = _MM_LOAD(pVec1 + 12);
-        vSum  = _mm_add_ps(vSum, _mm_mul_ps(vTemp, pVec2[3]));
-        vNorm = _mm_add_ps(vNorm, _mm_mul_ps(vTemp ,vTemp));
-
-        pVec1 += 16;
-        pVec2 += 4;
-    }
-
-    // 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]);
-    if (norm < 1e-9) norm = 1.0;    // to avoid div by zero
-
-    float *pvSum = (float*)&vSum;
-    return (double)(pvSum[0] + pvSum[1] + pvSum[2] + pvSum[3]) / norm;
-
-    /* This is approximately corresponding routine in C-language yet without normalization:
-    double corr, norm;
-    uint i;
-
-    // Calculates the cross-correlation value between 'pV1' and 'pV2' vectors
-    corr = norm = 0.0;
-    for (i = 0; i < channels * overlapLength / 16; i ++) 
-    {
-        corr += pV1[0] * pV2[0] +
-                pV1[1] * pV2[1] +
-                pV1[2] * pV2[2] +
-                pV1[3] * pV2[3] +
-                pV1[4] * pV2[4] +
-                pV1[5] * pV2[5] +
-                pV1[6] * pV2[6] +
-                pV1[7] * pV2[7] +
-                pV1[8] * pV2[8] +
-                pV1[9] * pV2[9] +
-                pV1[10] * pV2[10] +
-                pV1[11] * pV2[11] +
-                pV1[12] * pV2[12] +
-                pV1[13] * pV2[13] +
-                pV1[14] * pV2[14] +
-                pV1[15] * pV2[15];
-
-    for (j = 0; j < 15; j ++) norm += pV1[j] * pV1[j];
-
-        pV1 += 16;
-        pV2 += 16;
-    }
-    return corr / sqrt(norm);
-    */
-}
-
-
-//////////////////////////////////////////////////////////////////////////////
-//
-// implementation of SSE optimized functions of class 'FIRFilter'
-//
-//////////////////////////////////////////////////////////////////////////////
-
-#include "FIRFilter.h"
-
-FIRFilterSSE::FIRFilterSSE() : FIRFilter()
-{
-    filterCoeffsAlign = NULL;
-    filterCoeffsUnalign = NULL;
-}
-
-
-FIRFilterSSE::~FIRFilterSSE()
-{
-    delete[] filterCoeffsUnalign;
-    filterCoeffsAlign = NULL;
-    filterCoeffsUnalign = NULL;
-}
-
-
-// (overloaded) Calculates filter coefficients for SSE routine
-void FIRFilterSSE::setCoefficients(const float *coeffs, uint newLength, uint uResultDivFactor)
-{
-    uint i;
-    float fDivider;
-
-    FIRFilter::setCoefficients(coeffs, newLength, uResultDivFactor);
-
-    // Scale the filter coefficients so that it won't be necessary to scale the filtering result
-    // also rearrange coefficients suitably for SSE
-    // Ensure that filter coeffs array is aligned to 16-byte boundary
-    delete[] filterCoeffsUnalign;
-    filterCoeffsUnalign = new float[2 * newLength + 4];
-    filterCoeffsAlign = (float *)(((unsigned long)filterCoeffsUnalign + 15) & (ulong)-16);
-
-    fDivider = (float)resultDivider;
-
-    // rearrange the filter coefficients for mmx routines 
-    for (i = 0; i < newLength; i ++)
-    {
-        filterCoeffsAlign[2 * i + 0] =
-        filterCoeffsAlign[2 * i + 1] = coeffs[i + 0] / fDivider;
-    }
-}
-
-
-
-// SSE-optimized version of the filter routine for stereo sound
-uint FIRFilterSSE::evaluateFilterStereo(float *dest, const float *source, uint numSamples) const
-{
-    int count = (int)((numSamples - length) & (uint)-2);
-    int j;
-
-    assert(count % 2 == 0);
-
-    if (count < 2) return 0;
-
-    assert(source != NULL);
-    assert(dest != NULL);
-    assert((length % 8) == 0);
-    assert(filterCoeffsAlign != NULL);
-    assert(((ulong)filterCoeffsAlign) % 16 == 0);
-
-    // filter is evaluated for two stereo samples with each iteration, thus use of 'j += 2'
-    for (j = 0; j < count; j += 2)
-    {
-        const float *pSrc;
-        const __m128 *pFil;
-        __m128 sum1, sum2;
-        uint i;
-
-        pSrc = (const float*)source;              // source audio data
-        pFil = (const __m128*)filterCoeffsAlign;  // filter coefficients. NOTE: Assumes coefficients 
-                                                  // are aligned to 16-byte boundary
-        sum1 = sum2 = _mm_setzero_ps();
-
-        for (i = 0; i < length / 8; i ++) 
-        {
-            // Unroll loop for efficiency & calculate filter for 2*2 stereo samples 
-            // at each pass
-
-            // sum1 is accu for 2*2 filtered stereo sound data at the primary sound data offset
-            // sum2 is accu for 2*2 filtered stereo sound data for the next sound sample offset.
-
-            sum1 = _mm_add_ps(sum1, _mm_mul_ps(_mm_loadu_ps(pSrc)    , pFil[0]));
-            sum2 = _mm_add_ps(sum2, _mm_mul_ps(_mm_loadu_ps(pSrc + 2), pFil[0]));
-
-            sum1 = _mm_add_ps(sum1, _mm_mul_ps(_mm_loadu_ps(pSrc + 4), pFil[1]));
-            sum2 = _mm_add_ps(sum2, _mm_mul_ps(_mm_loadu_ps(pSrc + 6), pFil[1]));
-
-            sum1 = _mm_add_ps(sum1, _mm_mul_ps(_mm_loadu_ps(pSrc + 8) ,  pFil[2]));
-            sum2 = _mm_add_ps(sum2, _mm_mul_ps(_mm_loadu_ps(pSrc + 10), pFil[2]));
-
-            sum1 = _mm_add_ps(sum1, _mm_mul_ps(_mm_loadu_ps(pSrc + 12), pFil[3]));
-            sum2 = _mm_add_ps(sum2, _mm_mul_ps(_mm_loadu_ps(pSrc + 14), pFil[3]));
-
-            pSrc += 16;
-            pFil += 4;
-        }
-
-        // Now sum1 and sum2 both have a filtered 2-channel sample each, but we still need
-        // to sum the two hi- and lo-floats of these registers together.
-
-        // post-shuffle & add the filtered values and store to dest.
-        _mm_storeu_ps(dest, _mm_add_ps(
-                    _mm_shuffle_ps(sum1, sum2, _MM_SHUFFLE(1,0,3,2)),   // s2_1 s2_0 s1_3 s1_2
-                    _mm_shuffle_ps(sum1, sum2, _MM_SHUFFLE(3,2,1,0))    // s2_3 s2_2 s1_1 s1_0
-                    ));
-        source += 4;
-        dest += 4;
-    }
-
-    // Ideas for further improvement:
-    // 1. If it could be guaranteed that 'source' were always aligned to 16-byte 
-    //    boundary, a faster aligned '_mm_load_ps' instruction could be used.
-    // 2. If it could be guaranteed that 'dest' were always aligned to 16-byte 
-    //    boundary, a faster '_mm_store_ps' instruction could be used.
-
-    return (uint)count;
-
-    /* original routine in C-language. please notice the C-version has differently 
-       organized coefficients though.
-    double suml1, suml2;
-    double sumr1, sumr2;
-    uint i, j;
-
-    for (j = 0; j < count; j += 2)
-    {
-        const float *ptr;
-        const float *pFil;
-
-        suml1 = sumr1 = 0.0;
-        suml2 = sumr2 = 0.0;
-        ptr = src;
-        pFil = filterCoeffs;
-        for (i = 0; i < lengthLocal; i ++) 
-        {
-            // unroll loop for efficiency.
-
-            suml1 += ptr[0] * pFil[0] + 
-                     ptr[2] * pFil[2] +
-                     ptr[4] * pFil[4] +
-                     ptr[6] * pFil[6];
-
-            sumr1 += ptr[1] * pFil[1] + 
-                     ptr[3] * pFil[3] +
-                     ptr[5] * pFil[5] +
-                     ptr[7] * pFil[7];
-
-            suml2 += ptr[8] * pFil[0] + 
-                     ptr[10] * pFil[2] +
-                     ptr[12] * pFil[4] +
-                     ptr[14] * pFil[6];
-
-            sumr2 += ptr[9] * pFil[1] + 
-                     ptr[11] * pFil[3] +
-                     ptr[13] * pFil[5] +
-                     ptr[15] * pFil[7];
-
-            ptr += 16;
-            pFil += 8;
-        }
-        dest[0] = (float)suml1;
-        dest[1] = (float)sumr1;
-        dest[2] = (float)suml2;
-        dest[3] = (float)sumr2;
-
-        src += 4;
-        dest += 4;
-    }
-    */
-}
-
-#endif  // SOUNDTOUCH_ALLOW_SSE
+////////////////////////////////////////////////////////////////////////////////
+///
+/// SSE optimized routines for Pentium-III, Athlon-XP and later CPUs. All SSE
+/// optimized functions have been gathered into this single source
+/// code file, regardless to their class or original source code file, in order
+/// to ease porting the library to other compiler and processor platforms.
+///
+/// The SSE-optimizations are programmed using SSE compiler intrinsics that
+/// are supported both by Microsoft Visual C++ and GCC compilers, so this file
+/// should compile with both toolsets.
+///
+/// NOTICE: If using Visual Studio 6.0, you'll need to install the "Visual C++
+/// 6.0 processor pack" update to support SSE instruction set. The update is
+/// available for download at Microsoft Developers Network, see here:
+/// http://msdn.microsoft.com/en-us/vstudio/aa718349.aspx
+///
+/// If the above URL is expired or removed, go to "http://msdn.microsoft.com" and
+/// perform a search with keywords "processor pack".
+///
+/// Author        : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+//  SoundTouch audio processing library
+//  Copyright (c) Olli Parviainen
+//
+//  This library is free software; you can redistribute it and/or
+//  modify it under the terms of the GNU Lesser General Public
+//  License as published by the Free Software Foundation; either
+//  version 2.1 of the License, or (at your option) any later version.
+//
+//  This library is distributed in the hope that it will be useful,
+//  but WITHOUT ANY WARRANTY; without even the implied warranty of
+//  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+//  Lesser General Public License for more details.
+//
+//  You should have received a copy of the GNU Lesser General Public
+//  License along with this library; if not, write to the Free Software
+//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#include "cpu_detect.h"
+#include "STTypes.h"
+
+using namespace soundtouch;
+
+#ifdef SOUNDTOUCH_ALLOW_SSE
+
+// SSE routines available only with float sample type
+
+//////////////////////////////////////////////////////////////////////////////
+//
+// implementation of SSE optimized functions of class 'TDStretchSSE'
+//
+//////////////////////////////////////////////////////////////////////////////
+
+#include "TDStretch.h"
+#include <xmmintrin.h>
+#include <math.h>
+
+// Calculates cross correlation of two buffers
+double TDStretchSSE::calcCrossCorr(const float *pV1, const float *pV2, double &anorm)
+{
+    int i;
+    const float *pVec1;
+    const __m128 *pVec2;
+    __m128 vSum, vNorm;
+
+    // Note. It means a major slow-down if the routine needs to tolerate
+    // unaligned __m128 memory accesses. It's way faster if we can skip
+    // unaligned slots and use _mm_load_ps instruction instead of _mm_loadu_ps.
+    // This can mean up to ~ 10-fold difference (incl. part of which is
+    // due to skipping every second round for stereo sound though).
+    //
+    // Compile-time define SOUNDTOUCH_ALLOW_NONEXACT_SIMD_OPTIMIZATION is provided
+    // for choosing if this little cheating is allowed.
+
+#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 (((ulongptr)pV1) & 15) return -1e50;    // skip unaligned locations
+
+#else
+    // No cheating allowed, use unaligned load & take the resulting
+    // performance hit.
+    #define _MM_LOAD    _mm_loadu_ps
+#endif
+
+    // ensure overlapLength is divisible by 8
+    assert((overlapLength % 8) == 0);
+
+    // Calculates the cross-correlation value between 'pV1' and 'pV2' vectors
+    // Note: pV2 _must_ be aligned to 16-bit boundary, pV1 need not.
+    pVec1 = (const float*)pV1;
+    pVec2 = (const __m128*)pV2;
+    vSum = vNorm = _mm_setzero_ps();
+
+    // Unroll the loop by factor of 4 * 4 operations. Use same routine for
+    // stereo & mono, for mono it just means twice the amount of unrolling.
+    for (i = 0; i < channels * overlapLength / 16; i ++)
+    {
+        __m128 vTemp;
+        // vSum += pV1[0..3] * pV2[0..3]
+        vTemp = _MM_LOAD(pVec1);
+        vSum  = _mm_add_ps(vSum,  _mm_mul_ps(vTemp ,pVec2[0]));
+        vNorm = _mm_add_ps(vNorm, _mm_mul_ps(vTemp ,vTemp));
+
+        // vSum += pV1[4..7] * pV2[4..7]
+        vTemp = _MM_LOAD(pVec1 + 4);
+        vSum  = _mm_add_ps(vSum, _mm_mul_ps(vTemp, pVec2[1]));
+        vNorm = _mm_add_ps(vNorm, _mm_mul_ps(vTemp ,vTemp));
+
+        // vSum += pV1[8..11] * pV2[8..11]
+        vTemp = _MM_LOAD(pVec1 + 8);
+        vSum  = _mm_add_ps(vSum, _mm_mul_ps(vTemp, pVec2[2]));
+        vNorm = _mm_add_ps(vNorm, _mm_mul_ps(vTemp ,vTemp));
+
+        // vSum += pV1[12..15] * pV2[12..15]
+        vTemp = _MM_LOAD(pVec1 + 12);
+        vSum  = _mm_add_ps(vSum, _mm_mul_ps(vTemp, pVec2[3]));
+        vNorm = _mm_add_ps(vNorm, _mm_mul_ps(vTemp ,vTemp));
+
+        pVec1 += 16;
+        pVec2 += 4;
+    }
+
+    // return value = vSum[0] + vSum[1] + vSum[2] + vSum[3]
+    float *pvNorm = (float*)&vNorm;
+    float norm = (pvNorm[0] + pvNorm[1] + pvNorm[2] + pvNorm[3]);
+    anorm = norm;
+
+    float *pvSum = (float*)&vSum;
+    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;
+    uint i;
+
+    // Calculates the cross-correlation value between 'pV1' and 'pV2' vectors
+    corr = norm = 0.0;
+    for (i = 0; i < channels * overlapLength / 16; i ++)
+    {
+        corr += pV1[0] * pV2[0] +
+                pV1[1] * pV2[1] +
+                pV1[2] * pV2[2] +
+                pV1[3] * pV2[3] +
+                pV1[4] * pV2[4] +
+                pV1[5] * pV2[5] +
+                pV1[6] * pV2[6] +
+                pV1[7] * pV2[7] +
+                pV1[8] * pV2[8] +
+                pV1[9] * pV2[9] +
+                pV1[10] * pV2[10] +
+                pV1[11] * pV2[11] +
+                pV1[12] * pV2[12] +
+                pV1[13] * pV2[13] +
+                pV1[14] * pV2[14] +
+                pV1[15] * pV2[15];
+
+    for (j = 0; j < 15; j ++) norm += pV1[j] * pV1[j];
+
+        pV1 += 16;
+        pV2 += 16;
+    }
+    return corr / sqrt(norm);
+    */
+}
+
+
+
+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'
+//
+//////////////////////////////////////////////////////////////////////////////
+
+#include "FIRFilter.h"
+
+FIRFilterSSE::FIRFilterSSE() : FIRFilter()
+{
+    filterCoeffsAlign = nullptr;
+    filterCoeffsUnalign = nullptr;
+}
+
+
+FIRFilterSSE::~FIRFilterSSE()
+{
+    delete[] filterCoeffsUnalign;
+    filterCoeffsAlign = nullptr;
+    filterCoeffsUnalign = nullptr;
+}
+
+
+// (overloaded) Calculates filter coefficients for SSE routine
+void FIRFilterSSE::setCoefficients(const float *coeffs, uint newLength, uint uResultDivFactor)
+{
+    FIRFilter::setCoefficients(coeffs, newLength, uResultDivFactor);
+
+    // Scale the filter coefficients so that it won't be necessary to scale the filtering result
+    // also rearrange coefficients suitably for SSE
+    // Ensure that filter coeffs array is aligned to 16-byte boundary
+    delete[] filterCoeffsUnalign;
+    filterCoeffsUnalign = new float[2 * newLength + 4];
+    filterCoeffsAlign = (float *)SOUNDTOUCH_ALIGN_POINTER_16(filterCoeffsUnalign);
+
+    const float scale = ::pow(0.5, (int)resultDivFactor);
+
+    // rearrange the filter coefficients for sse routines
+    for (auto i = 0U; i < newLength; i ++)
+    {
+        filterCoeffsAlign[2 * i + 0] =
+        filterCoeffsAlign[2 * i + 1] = coeffs[i] * scale;
+    }
+}
+
+
+
+// SSE-optimized version of the filter routine for stereo sound
+uint FIRFilterSSE::evaluateFilterStereo(float *dest, const float *source, uint numSamples) const
+{
+    int count = (int)((numSamples - length) & (uint)-2);
+    int j;
+
+    assert(count % 2 == 0);
+
+    if (count < 2) return 0;
+
+    assert(source != nullptr);
+    assert(dest != nullptr);
+    assert((length % 8) == 0);
+    assert(filterCoeffsAlign != nullptr);
+    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 + 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();
+
+        for (i = 0; i < length / 8; i ++)
+        {
+            // Unroll loop for efficiency & calculate filter for 2*2 stereo samples
+            // at each pass
+
+            // sum1 is accu for 2*2 filtered stereo sound data at the primary sound data offset
+            // sum2 is accu for 2*2 filtered stereo sound data for the next sound sample offset.
+
+            sum1 = _mm_add_ps(sum1, _mm_mul_ps(_mm_loadu_ps(pSrc)    , pFil[0]));
+            sum2 = _mm_add_ps(sum2, _mm_mul_ps(_mm_loadu_ps(pSrc + 2), pFil[0]));
+
+            sum1 = _mm_add_ps(sum1, _mm_mul_ps(_mm_loadu_ps(pSrc + 4), pFil[1]));
+            sum2 = _mm_add_ps(sum2, _mm_mul_ps(_mm_loadu_ps(pSrc + 6), pFil[1]));
+
+            sum1 = _mm_add_ps(sum1, _mm_mul_ps(_mm_loadu_ps(pSrc + 8) ,  pFil[2]));
+            sum2 = _mm_add_ps(sum2, _mm_mul_ps(_mm_loadu_ps(pSrc + 10), pFil[2]));
+
+            sum1 = _mm_add_ps(sum1, _mm_mul_ps(_mm_loadu_ps(pSrc + 12), pFil[3]));
+            sum2 = _mm_add_ps(sum2, _mm_mul_ps(_mm_loadu_ps(pSrc + 14), pFil[3]));
+
+            pSrc += 16;
+            pFil += 4;
+        }
+
+        // Now sum1 and sum2 both have a filtered 2-channel sample each, but we still need
+        // to sum the two hi- and lo-floats of these registers together.
+
+        // post-shuffle & add the filtered values and store to dest.
+        _mm_storeu_ps(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
+                    ));
+    }
+
+    // Ideas for further improvement:
+    // 1. If it could be guaranteed that 'source' were always aligned to 16-byte
+    //    boundary, a faster aligned '_mm_load_ps' instruction could be used.
+    // 2. If it could be guaranteed that 'dest' were always aligned to 16-byte
+    //    boundary, a faster '_mm_store_ps' instruction could be used.
+
+    return (uint)count;
+
+    /* original routine in C-language. please notice the C-version has differently
+       organized coefficients though.
+    double suml1, suml2;
+    double sumr1, sumr2;
+    uint i, j;
+
+    for (j = 0; j < count; j += 2)
+    {
+        const float *ptr;
+        const float *pFil;
+
+        suml1 = sumr1 = 0.0;
+        suml2 = sumr2 = 0.0;
+        ptr = src;
+        pFil = filterCoeffs;
+        for (i = 0; i < lengthLocal; i ++)
+        {
+            // unroll loop for efficiency.
+
+            suml1 += ptr[0] * pFil[0] +
+                     ptr[2] * pFil[2] +
+                     ptr[4] * pFil[4] +
+                     ptr[6] * pFil[6];
+
+            sumr1 += ptr[1] * pFil[1] +
+                     ptr[3] * pFil[3] +
+                     ptr[5] * pFil[5] +
+                     ptr[7] * pFil[7];
+
+            suml2 += ptr[8] * pFil[0] +
+                     ptr[10] * pFil[2] +
+                     ptr[12] * pFil[4] +
+                     ptr[14] * pFil[6];
+
+            sumr2 += ptr[9] * pFil[1] +
+                     ptr[11] * pFil[3] +
+                     ptr[13] * pFil[5] +
+                     ptr[15] * pFil[7];
+
+            ptr += 16;
+            pFil += 8;
+        }
+        dest[0] = (float)suml1;
+        dest[1] = (float)sumr1;
+        dest[2] = (float)suml2;
+        dest[3] = (float)sumr2;
+
+        src += 4;
+        dest += 4;
+    }
+    */
+}
+
+#endif  // SOUNDTOUCH_ALLOW_SSE

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 @@
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+    <Keyword>Win32Proj</Keyword>
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+      <PrecompiledHeader>
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+      <AdditionalIncludeDirectories>../../../include</AdditionalIncludeDirectories>
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+    </ClCompile>
+    <Link>
+      <SubSystem>Console</SubSystem>
+      <GenerateDebugInformation>true</GenerateDebugInformation>
+      <AdditionalDependencies>..\..\..\lib\SoundTouchDllD.lib</AdditionalDependencies>
+      <OutputFile>$(OutDir)$(TargetName)$(TargetExt)</OutputFile>
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+      </PrecompiledHeader>
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+      <AdditionalIncludeDirectories>../../../include</AdditionalIncludeDirectories>
+      <DisableSpecificWarnings>4996</DisableSpecificWarnings>
+    </ClCompile>
+    <Link>
+      <SubSystem>Console</SubSystem>
+      <GenerateDebugInformation>true</GenerateDebugInformation>
+      <AdditionalDependencies>..\..\..\lib\SoundTouchDllD_x64.lib</AdditionalDependencies>
+    </Link>
+  </ItemDefinitionGroup>
+  <ItemDefinitionGroup Condition="'$(Configuration)|$(Platform)'=='Release|Win32'">
+    <ClCompile>
+      <WarningLevel>Level3</WarningLevel>
+      <PrecompiledHeader>
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+      <IntrinsicFunctions>true</IntrinsicFunctions>
+      <PreprocessorDefinitions>WIN32;NDEBUG;_CONSOLE;%(PreprocessorDefinitions)</PreprocessorDefinitions>
+      <AdditionalIncludeDirectories>../../../include</AdditionalIncludeDirectories>
+      <DisableSpecificWarnings>4996</DisableSpecificWarnings>
+    </ClCompile>
+    <Link>
+      <SubSystem>Console</SubSystem>
+      <EnableCOMDATFolding>true</EnableCOMDATFolding>
+      <OptimizeReferences>true</OptimizeReferences>
+      <GenerateDebugInformation>true</GenerateDebugInformation>
+      <AdditionalDependencies>..\..\..\lib\SoundTouchDll.lib</AdditionalDependencies>
+    </Link>
+  </ItemDefinitionGroup>
+  <ItemDefinitionGroup Condition="'$(Configuration)|$(Platform)'=='Release|x64'">
+    <ClCompile>
+      <WarningLevel>Level3</WarningLevel>
+      <PrecompiledHeader>
+      </PrecompiledHeader>
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+      <IntrinsicFunctions>true</IntrinsicFunctions>
+      <PreprocessorDefinitions>NDEBUG;_CONSOLE;%(PreprocessorDefinitions)</PreprocessorDefinitions>
+      <AdditionalIncludeDirectories>../../../include</AdditionalIncludeDirectories>
+      <DisableSpecificWarnings>4996</DisableSpecificWarnings>
+    </ClCompile>
+    <Link>
+      <SubSystem>Console</SubSystem>
+      <EnableCOMDATFolding>true</EnableCOMDATFolding>
+      <OptimizeReferences>true</OptimizeReferences>
+      <GenerateDebugInformation>true</GenerateDebugInformation>
+      <AdditionalDependencies>..\..\..\lib\SoundTouchDll_x64.lib</AdditionalDependencies>
+    </Link>
+  </ItemDefinitionGroup>
+  <ItemGroup>
+    <Text Include="ReadMe.txt" />
+  </ItemGroup>
+  <ItemGroup>
+    <ClInclude Include="stdafx.h" />
+    <ClInclude Include="targetver.h" />
+  </ItemGroup>
+  <ItemGroup>
+    <ClCompile Include="..\..\SoundStretch\WavFile.cpp" />
+    <ClCompile Include="DllTest.cpp" />
+  </ItemGroup>
+  <Import Project="$(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
-//
-//  Module Author : Christian Budde
-//  
-////////////////////////////////////////////////////////////////////////////////
-//
-// $Id$
+// SoundTouch.dll / libSoundTouchDll.so wrapper for accessing SoundTouch
+// routines from Delphi/Pascal/Lazarus
 //
 ////////////////////////////////////////////////////////////////////////////////
 //
@@ -35,58 +30,60 @@ unit SoundTouchDLL;
 
 interface
 
-uses
-  Windows;
+//uses
+  //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.
-                                ): Integer; stdcall;
-
+                                 SettingId: Integer     //< Setting ID number, see SETTING_... defines.
+                                ): 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
-  // 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.
+  /// Receive ready samples from the processing pipeline.
+  ///
+  /// if called with outBuffer=nullptr, just reduces amount of ready samples within the pipeline.
   TSoundTouchReceiveSamples = function (Handle: TSoundTouchHandle;
-                                     outBuffer: PSingle;           //< Buffer where to copy output samples.
-                                     maxSamples: Integer      //< How many samples to receive at max.
-                                    ): Cardinal; stdcall;
+                                     OutBuffer: PSingle;           //< Buffer where to copy output samples.
+                                     MaxSamples: Integer      //< How many samples to receive at max.
+                                    ): 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
-  SysUtils;
+implementation
 
 { 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;
-  const maxSamples: Integer): Cardinal;
+function TSoundTouch.ReceiveSamples(const OutBuffer: PSingle;
+  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;
-  SoundTouchDLL: PAnsiChar = 'SoundTouch.DLL';
+  SoundTouchLibHandle: THandle;
+  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');
-    SoundTouchDestroyInstance       := GetProcAddress(SoundTouchLibHandle, PAnsiChar( 3)); //'soundtouch_destroyInstance');
-    SoundTouchGetVersionString      := GetProcAddress(SoundTouchLibHandle, PAnsiChar( 7)); //'soundtouch_getVersionString');
-    SoundTouchGetVersionId          := GetProcAddress(SoundTouchLibHandle, PAnsiChar( 6)); //'soundtouch_getVersionId');
-    SoundTouchSetRate               := GetProcAddress(SoundTouchLibHandle, PAnsiChar(17)); //'soundtouch_setRate');
-    SoundTouchSetTempo              := GetProcAddress(SoundTouchLibHandle, PAnsiChar(21)); //'soundtouch_setTempo');
-    SoundTouchSetRateChange         := GetProcAddress(SoundTouchLibHandle, PAnsiChar(18)); //'soundtouch_setRateChange');
-    SoundTouchSetTempoChange        := GetProcAddress(SoundTouchLibHandle, PAnsiChar(22)); //'soundtouch_setTempoChange');
-    SoundTouchSetPitch              := GetProcAddress(SoundTouchLibHandle, PAnsiChar(14)); //'soundtouch_setPitch');
-    SoundTouchSetPitchOctaves       := GetProcAddress(SoundTouchLibHandle, PAnsiChar(15)); //'soundtouch_setPitchOctaves');
-    SoundTouchSetPitchSemiTones     := GetProcAddress(SoundTouchLibHandle, PAnsiChar(16)); //'soundtouch_setPitchSemiTones');
-    SoundTouchSetChannels           := GetProcAddress(SoundTouchLibHandle, PAnsiChar(13)); //'soundtouch_setChannels');
-    SoundTouchSetSampleRate         := GetProcAddress(SoundTouchLibHandle, PAnsiChar(19)); //'soundtouch_setSampleRate');
-    SoundTouchFlush                 := GetProcAddress(SoundTouchLibHandle, PAnsiChar(4)); //'soundtouch_flush');
-    SoundTouchPutSamples            := GetProcAddress(SoundTouchLibHandle, PAnsiChar(11)); //'soundtouch_putSamples');
-    SoundTouchClear                 := GetProcAddress(SoundTouchLibHandle, PAnsiChar(1)); //'soundtouch_clear');
-    SoundTouchSetSetting            := GetProcAddress(SoundTouchLibHandle, PAnsiChar(20)); //'soundtouch_SetSetting');
-    SoundTouchGetSetting            := GetProcAddress(SoundTouchLibHandle, PAnsiChar(5)); //'soundtouch_setSetting');
-    SoundTouchNumUnprocessedSamples := GetProcAddress(SoundTouchLibHandle, PAnsiChar(10)); //'soundtouch_numUnprocessedSamples');
-    SoundTouchReceiveSamples        := GetProcAddress(SoundTouchLibHandle, PAnsiChar(12)); //'soundtouch_receiveSamples');
-    SoundTouchNumSamples            := GetProcAddress(SoundTouchLibHandle, PAnsiChar(9)); //'soundtouch_numSamples');
-    SoundTouchIsEmpty               := GetProcAddress(SoundTouchLibHandle, PAnsiChar(8)); //'soundtouch_isEmpty');
+    Pointer(SoundTouchCreateInstance)        := GetProcAddress(SoundTouchLibHandle, 'soundtouch_createInstance');
+    Pointer(SoundTouchDestroyInstance)       := GetProcAddress(SoundTouchLibHandle, 'soundtouch_destroyInstance');
+    Pointer(SoundTouchGetVersionString)      := GetProcAddress(SoundTouchLibHandle, 'soundtouch_getVersionString');
+    Pointer(SoundTouchGetVersionString2)     := GetProcAddress(SoundTouchLibHandle, 'soundtouch_getVersionString2');
+    Pointer(SoundTouchGetVersionId)          := GetProcAddress(SoundTouchLibHandle, 'soundtouch_getVersionId');
+    Pointer(SoundTouchSetRate)               := GetProcAddress(SoundTouchLibHandle, 'soundtouch_setRate');
+    Pointer(SoundTouchSetTempo)              := GetProcAddress(SoundTouchLibHandle, 'soundtouch_setTempo');
+    Pointer(SoundTouchSetRateChange)         := GetProcAddress(SoundTouchLibHandle, 'soundtouch_setRateChange');
+    Pointer(SoundTouchSetTempoChange)        := GetProcAddress(SoundTouchLibHandle, 'soundtouch_setTempoChange');
+    Pointer(SoundTouchSetPitch)              := GetProcAddress(SoundTouchLibHandle, 'soundtouch_setPitch');
+    Pointer(SoundTouchSetPitchOctaves)       := GetProcAddress(SoundTouchLibHandle, 'soundtouch_setPitchOctaves');
+    Pointer(SoundTouchSetPitchSemiTones)     := GetProcAddress(SoundTouchLibHandle, 'soundtouch_setPitchSemiTones');
+    Pointer(SoundTouchSetChannels)           := GetProcAddress(SoundTouchLibHandle, 'soundtouch_setChannels');
+    Pointer(SoundTouchSetSampleRate)         := GetProcAddress(SoundTouchLibHandle, 'soundtouch_setSampleRate');
+    Pointer(SoundTouchFlush)                 := GetProcAddress(SoundTouchLibHandle, 'soundtouch_flush');
+    Pointer(SoundTouchPutSamples)            := GetProcAddress(SoundTouchLibHandle, 'soundtouch_putSamples');
+    Pointer(SoundTouchPutSamplesI16)         := GetProcAddress(SoundTouchLibHandle, 'soundtouch_putSamples_i16');
+    Pointer(SoundTouchClear)                 := GetProcAddress(SoundTouchLibHandle, 'soundtouch_clear');
+    Pointer(SoundTouchSetSetting)            := GetProcAddress(SoundTouchLibHandle, 'soundtouch_SetSetting');
+    Pointer(SoundTouchGetSetting)            := GetProcAddress(SoundTouchLibHandle, 'soundtouch_setSetting');
+    Pointer(SoundTouchNumUnprocessedSamples) := GetProcAddress(SoundTouchLibHandle, 'soundtouch_numUnprocessedSamples');
+    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>