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

@@ -38,3 +38,18 @@ 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

CMakeLists.txt

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

COPYING.TXT

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

Makefile.am

@@ -1,16 +1,16 @@
 ## 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

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

@@ -1,5 +1,7 @@
 #!/bin/sh
 
+unset ACLOCAL
+
 if [ "$1" = "--clean" ]
 then
 	if [ -a Makefile ]
@@ -8,15 +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

configure.ac

@@ -1,36 +1,39 @@
 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], [2.0.0], [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)
 AC_CONFIG_MACRO_DIR([config/m4])
-AM_CONFIG_HEADER([config.h include/soundtouch_config.h])
+AC_CONFIG_HEADERS([config.h include/soundtouch_config.h])
 AM_INIT_AUTOMAKE
 AM_SILENT_RULES([yes])
-#AC_DISABLE_SHARED	dnl This makes libtool only build static libs 
+#AC_DISABLE_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'
 
@@ -47,7 +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
 
 
 
@@ -55,17 +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([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
- 
+
 
 dnl ############################################################################
 dnl # Checks for typedefs, structures, and compiler characteristics            $
@@ -77,31 +81,34 @@ AC_C_INLINE
 
 
 AC_ARG_ENABLE(integer-samples,
-              [AC_HELP_STRING([--enable-integer-samples],
-                              [use integer samples instead of floats
-[default=no]])],,
+              [AS_HELP_STRING([--enable-integer-samples],[use integer samples instead of floats [default=no]])],,
               [enable_integer_samples=no])
 
 
 AC_ARG_ENABLE(openmp,
-              [AC_HELP_STRING([--enable-openmp],
-                              [use parallel multicore calculation through OpenMP [default=no]])],,
+              [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$enable_integer_samples" = "xyes"; then
         echo "****** Integer sample type enabled ******"
         AC_DEFINE(SOUNDTOUCH_INTEGER_SAMPLES,1,[Use Integer as Sample type])
@@ -109,7 +116,7 @@ else
         echo "****** Float sample type enabled ******"
         AC_DEFINE(SOUNDTOUCH_FLOAT_SAMPLES,1,[Use Float as Sample type])
 fi
-
+AM_CONDITIONAL([SOUNDTOUCH_FLOAT_SAMPLES], [test "x$enable_integer_samples" != "xyes"])
 
 if test "x$enable_openmp" = "xyes"; then
         echo "****** openmp optimizations enabled ******"
@@ -195,6 +202,52 @@ else
         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])
 
@@ -217,11 +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])
@@ -251,11 +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

include/BPMDetect.h

@@ -1,205 +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
-///
-////////////////////////////////////////////////////////////////////////////////
-//
-// 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 NULL in "pos" & "values".
-        ///
-        /// \return number of beats in the arrays.
-        int getBeats(float *pos, float *strength, int max_num);
-    };
-}
-#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

@@ -1,177 +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
-///
-////////////////////////////////////////////////////////////////////////////////
-//
-// 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 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.
-                            );
-
-    /// 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);
-
-    /// Get number of channels
-    int getChannels() 
-    {
-        return channels;
-    }
-
-    /// 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

@@ -1,230 +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
-///
-////////////////////////////////////////////////////////////////////////////////
-//
-// 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.
-         )
-    {
-        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

@@ -1,22 +1,22 @@
-## 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
-
+## 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

@@ -1,183 +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
-///
-////////////////////////////////////////////////////////////////////////////////
-//
-// 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
-    #define SOUNDTOUCH_MAX_CHANNELS     16
-
-    /// Activate these undef's to overrule the possible sampletype 
-    /// setting inherited from some other header file:
-    //#undef SOUNDTOUCH_INTEGER_SAMPLES
-    //#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 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 (not available in X64 mode)
-            #if (!_M_X64)
-                #define SOUNDTOUCH_ALLOW_MMX   1
-            #endif
-        #endif
-
-    #else
-
-        // floating point samples
-        typedef float  SAMPLETYPE;
-        // data type for sample accumulation: Use double to utilize full precision.
-        typedef double LONG_SAMPLETYPE;
-
-        #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>
-    #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
+////////////////////////////////////////////////////////////////////////////////
+///
+/// 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

@@ -1,348 +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
-///
-////////////////////////////////////////////////////////////////////////////////
-//
-// 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          "2.1pre"
-
-/// SoundTouch library version id
-#define SOUNDTOUCH_VERSION_ID       (20009)
-
-//
-// 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();
-
-    /// 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.
-            );
-
-    /// Output samples from beginning of the sample buffer. Copies requested samples to 
-    /// output buffer and removes them from the sample buffer. If there are less than 
-    /// 'numsample' samples in the buffer, returns all that available.
-    ///
-    /// \return Number of samples returned.
-    virtual uint receiveSamples(SAMPLETYPE *output, ///< Buffer where to copy output samples.
-        uint maxSamples                 ///< How many samples to receive at max.
-        );
-
-    /// Adjusts book-keeping so that given number of samples are removed from beginning of the 
-    /// sample buffer without copying them anywhere. 
-    ///
-    /// Used to reduce the number of samples in the buffer when accessing the sample buffer directly
-    /// with 'ptrBegin' function.
-    virtual uint receiveSamples(uint maxSamples   ///< Remove this many samples from the beginning of pipe.
-        );
-
-    /// Clears all the samples in the object's output and internal processing
-    /// buffers.
-    virtual void clear();
-
-    /// 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
+//////////////////////////////////////////////////////////////////////////////
+///
+/// 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

@@ -3,3 +3,6 @@
 
 /* Use Integer as Sample type */
 #undef SOUNDTOUCH_INTEGER_SAMPLES
+
+/* Use ARM NEON extension */
+#undef SOUNDTOUCH_USE_NEON

make-win.bat

@@ -31,8 +31,9 @@ echo ***************************************************************************
 echo **
 echo ** ERROR: Visual Studio path not set.
 echo **
-echo ** Run "vsvars32.bat" or "vcvars32.bat" from Visual Studio installation dir,
-echo ** e.g. "C:\Program Files (x86)\Microsoft Visual Studio 14.0\VC\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 ****************************************************************************

readme.md

@@ -1,5 +1,7 @@
 # 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
@@ -7,7 +9,9 @@ SoundTouch is an open-source audio processing library that allows changing the s
 same time
 * Change any combination of tempo/pitch/rate
 
-Visit [SoundTouch website](https://www.surina.net/soundtouch) and see the [README file](README.html) for more information and audio examples.
+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
 
@@ -17,7 +21,7 @@ Use SoundStretch example app for modifying wav audio files, for example as follo
 soundstretch my_original_file.wav output_file.wav -tempo=+15 -pitch=-3
 ```
 
-See the [README file](README.html) for more usage examples and instructions how to build SoundTouch + SoundStretch.
+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.
 
@@ -33,6 +37,18 @@ SoundTouch is written in C++ and compiles in virtually any platform:
 
 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:

soundtouch.m4

@@ -8,7 +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
 #
@@ -32,10 +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],[
@@ -49,7 +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/README-SoundTouch-Android.html

@@ -103,6 +103,7 @@ in the <strong>soundtouch-jni.cpp </strong>source code file for more details.</p
 	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 

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

@@ -1,22 +1,8 @@
-# Copyright (C) 2010 The Android Open Source Project
-#
-# Licensed under the Apache License, Version 2.0 (the "License");
-# you may not use this file except in compliance with the License.
-# You may obtain a copy of the License at
-#
-#      http://www.apache.org/licenses/LICENSE-2.0
-#
-# Unless required by applicable law or agreed to in writing, software
-# distributed under the License is distributed on an "AS IS" BASIS,
-# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-# See the License for the specific language governing permissions and
-# limitations under the License.
-#
-
 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
@@ -38,7 +24,7 @@ LOCAL_LDLIBS    += -llog
 
 # Custom Flags: 
 # -fvisibility=hidden : don't export all symbols
-LOCAL_CFLAGS += -fvisibility=hidden -I ../../../include -fdata-sections -ffunction-sections
+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

source/Android-lib/jni/Application.mk

@@ -4,6 +4,6 @@
 
 APP_ABI := all #armeabi-v7a armeabi
 APP_OPTIM := release
-APP_STL := stlport_static
+APP_STL := c++_static
 APP_CPPFLAGS := -fexceptions # -D SOUNDTOUCH_DISABLE_X86_OPTIMIZATIONS
 

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

@@ -1,255 +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;
-}
-
-
-#ifdef _OPENMP
-
-#include <pthread.h>
-extern pthread_key_t gomp_tls_key;
-static void * _p_gomp_tls = NULL;
-
-/// 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 NULL pointer 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 == NULL)
-	{
-		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 == NULL))
-	{
-		_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
-
-
-// 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!
-    _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!
-    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;
-}
+////////////////////////////////////////////////////////////////////////////////
+///
+/// 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/Makefile.am

@@ -1,24 +1,25 @@
-## 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
-
-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

@@ -1,50 +1,50 @@
-## 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. 
-# OP 2011-7-17 Linker flag -s disabled to prevent stripping symbols by default
-#soundstretch_LDFLAGS=-s
-
-## additional compiler flags
-soundstretch_CXXFLAGS=-O3 $(AM_CXXFLAGS)
-
-#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

@@ -1,291 +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
-///
-////////////////////////////////////////////////////////////////////////////////
-//
-// 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

@@ -1,65 +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
-///
-////////////////////////////////////////////////////////////////////////////////
-//
-// 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

@@ -1,277 +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
-///
-////////////////////////////////////////////////////////////////////////////////
-//
-// 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];
-    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;
-
-    /// 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

@@ -1,322 +1,321 @@
-////////////////////////////////////////////////////////////////////////////////
-///
-/// 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 <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 (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(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.vcxproj

@@ -21,32 +21,32 @@
   <PropertyGroup Label="Globals">
     <ProjectGuid>{5AACDFFA-D491-44B8-A332-DA7ACCAAF2AF}</ProjectGuid>
     <RootNamespace>soundstretch</RootNamespace>
-    <WindowsTargetPlatformVersion>8.1</WindowsTargetPlatformVersion>
+    <WindowsTargetPlatformVersion>10.0</WindowsTargetPlatformVersion>
   </PropertyGroup>
   <Import Project="$(VCTargetsPath)\Microsoft.Cpp.Default.props" />
   <PropertyGroup Condition="'$(Configuration)|$(Platform)'=='Release|Win32'" Label="Configuration">
     <ConfigurationType>Application</ConfigurationType>
-    <PlatformToolset>v140</PlatformToolset>
+    <PlatformToolset>v142</PlatformToolset>
     <UseOfMfc>false</UseOfMfc>
-    <CharacterSet>MultiByte</CharacterSet>
+    <CharacterSet>Unicode</CharacterSet>
   </PropertyGroup>
   <PropertyGroup Condition="'$(Configuration)|$(Platform)'=='Debug|Win32'" Label="Configuration">
     <ConfigurationType>Application</ConfigurationType>
-    <PlatformToolset>v140</PlatformToolset>
+    <PlatformToolset>v142</PlatformToolset>
     <UseOfMfc>false</UseOfMfc>
-    <CharacterSet>MultiByte</CharacterSet>
+    <CharacterSet>Unicode</CharacterSet>
   </PropertyGroup>
   <PropertyGroup Condition="'$(Configuration)|$(Platform)'=='Release|x64'" Label="Configuration">
     <ConfigurationType>Application</ConfigurationType>
-    <PlatformToolset>v140</PlatformToolset>
+    <PlatformToolset>v142</PlatformToolset>
     <UseOfMfc>false</UseOfMfc>
-    <CharacterSet>MultiByte</CharacterSet>
+    <CharacterSet>Unicode</CharacterSet>
   </PropertyGroup>
   <PropertyGroup Condition="'$(Configuration)|$(Platform)'=='Debug|x64'" Label="Configuration">
     <ConfigurationType>Application</ConfigurationType>
-    <PlatformToolset>v140</PlatformToolset>
+    <PlatformToolset>v142</PlatformToolset>
     <UseOfMfc>false</UseOfMfc>
-    <CharacterSet>MultiByte</CharacterSet>
+    <CharacterSet>Unicode</CharacterSet>
   </PropertyGroup>
   <Import Project="$(VCTargetsPath)\Microsoft.Cpp.props" />
   <ImportGroup Label="ExtensionSettings">
@@ -114,9 +114,10 @@
       <BrowseInformation>true</BrowseInformation>
       <WarningLevel>Level3</WarningLevel>
       <SuppressStartupBanner>true</SuppressStartupBanner>
-      <DebugInformationFormat>EditAndContinue</DebugInformationFormat>
+      <DebugInformationFormat>ProgramDatabase</DebugInformationFormat>
       <CompileAs>Default</CompileAs>
       <EnableEnhancedInstructionSet>StreamingSIMDExtensions2</EnableEnhancedInstructionSet>
+      <MultiProcessorCompilation>true</MultiProcessorCompilation>
     </ClCompile>
     <ResourceCompile>
       <PreprocessorDefinitions>_DEBUG;%(PreprocessorDefinitions)</PreprocessorDefinitions>
@@ -133,10 +134,7 @@
       <GenerateMapFile>true</GenerateMapFile>
       <MapFileName>$(OutDir)$(TargetName).map</MapFileName>
       <SubSystem>Console</SubSystem>
-      <RandomizedBaseAddress>false</RandomizedBaseAddress>
       <DataExecutionPrevention />
-      <TargetMachine>MachineX86</TargetMachine>
-      <ImageHasSafeExceptionHandlers>false</ImageHasSafeExceptionHandlers>
     </Link>
     <PostBuildEvent>
       <Command>if not exist ..\..\bin mkdir ..\..\bin
@@ -167,6 +165,7 @@ copy $(OutDir)$(TargetName)$(TargetExt) ..\..\bin\</Command>
       <DebugInformationFormat />
       <CompileAs>Default</CompileAs>
       <EnableEnhancedInstructionSet>StreamingSIMDExtensions2</EnableEnhancedInstructionSet>
+      <MultiProcessorCompilation>true</MultiProcessorCompilation>
     </ClCompile>
     <ResourceCompile>
       <PreprocessorDefinitions>NDEBUG;%(PreprocessorDefinitions)</PreprocessorDefinitions>
@@ -181,9 +180,7 @@ copy $(OutDir)$(TargetName)$(TargetExt) ..\..\bin\</Command>
       <GenerateMapFile>true</GenerateMapFile>
       <MapFileName>$(OutDir)$(TargetName).map</MapFileName>
       <SubSystem>Console</SubSystem>
-      <RandomizedBaseAddress>false</RandomizedBaseAddress>
       <DataExecutionPrevention />
-      <TargetMachine>MachineX86</TargetMachine>
     </Link>
     <PostBuildEvent>
       <Command>if not exist ..\..\bin mkdir ..\..\bin
@@ -215,6 +212,7 @@ copy $(OutDir)$(TargetName)$(TargetExt) ..\..\bin\</Command>
       <CompileAs>Default</CompileAs>
       <EnableEnhancedInstructionSet>
       </EnableEnhancedInstructionSet>
+      <MultiProcessorCompilation>true</MultiProcessorCompilation>
     </ClCompile>
     <ResourceCompile>
       <PreprocessorDefinitions>_DEBUG;%(PreprocessorDefinitions)</PreprocessorDefinitions>
@@ -231,9 +229,7 @@ copy $(OutDir)$(TargetName)$(TargetExt) ..\..\bin\</Command>
       <GenerateMapFile>true</GenerateMapFile>
       <MapFileName>$(OutDir)$(TargetName).map</MapFileName>
       <SubSystem>Console</SubSystem>
-      <RandomizedBaseAddress>false</RandomizedBaseAddress>
       <DataExecutionPrevention />
-      <TargetMachine>MachineX64</TargetMachine>
     </Link>
     <PostBuildEvent>
       <Command>if not exist ..\..\bin mkdir ..\..\bin
@@ -266,6 +262,7 @@ copy $(OutDir)$(TargetName)$(TargetExt) ..\..\bin\</Command>
       <CompileAs>Default</CompileAs>
       <EnableEnhancedInstructionSet>
       </EnableEnhancedInstructionSet>
+      <MultiProcessorCompilation>true</MultiProcessorCompilation>
     </ClCompile>
     <ResourceCompile>
       <PreprocessorDefinitions>NDEBUG;%(PreprocessorDefinitions)</PreprocessorDefinitions>
@@ -280,9 +277,7 @@ copy $(OutDir)$(TargetName)$(TargetExt) ..\..\bin\</Command>
       <GenerateMapFile>true</GenerateMapFile>
       <MapFileName>$(OutDir)$(TargetName).map</MapFileName>
       <SubSystem>Console</SubSystem>
-      <RandomizedBaseAddress>false</RandomizedBaseAddress>
       <DataExecutionPrevention />
-      <TargetMachine>MachineX64</TargetMachine>
     </Link>
     <PostBuildEvent>
       <Command>if not exist ..\..\bin mkdir ..\..\bin
@@ -323,6 +318,7 @@ copy $(OutDir)$(TargetName)$(TargetExt) ..\..\bin\</Command>
   </ItemGroup>
   <ItemGroup>
     <ClInclude Include="RunParameters.h" />
+    <ClInclude Include="SS_CharTypes.h" />
     <ClInclude Include="WavFile.h" />
   </ItemGroup>
   <ItemGroup>

source/SoundTouch/AAFilter.cpp

@@ -1,222 +1,222 @@
-////////////////////////////////////////////////////////////////////////////////
-///
-/// 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 == NULL) 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();
-}
+////////////////////////////////////////////////////////////////////////////////
+///
+/// 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

@@ -1,93 +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
-///
-////////////////////////////////////////////////////////////////////////////////
-//
-// 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
+////////////////////////////////////////////////////////////////////////////////
+///
+/// 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/FIFOSampleBuffer.cpp

@@ -1,267 +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
-///
-////////////////////////////////////////////////////////////////////////////////
-//
-// 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;
-
-    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 == 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 *)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;
-}
+////////////////////////////////////////////////////////////////////////////////
+///
+/// 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

@@ -1,324 +1,314 @@
-////////////////////////////////////////////////////////////////////////////////
-///
-/// 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;
-    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
-{
-    int j, end;
-#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);
-
-    #pragma omp parallel for
-    for (j = 0; j < end; j += 2) 
-    {
-        const SAMPLETYPE *ptr;
-        LONG_SAMPLETYPE suml, sumr;
-        uint i;
-
-        suml = sumr = 0;
-        ptr = src + j;
-
-        for (i = 0; i < length; i += 4) 
-        {
-            // 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
-{
-    int j, end;
-#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;
-    #pragma omp parallel for
-    for (j = 0; j < end; j ++) 
-    {
-        const SAMPLETYPE *pSrc = src + j;
-        LONG_SAMPLETYPE sum;
-        uint i;
-
-        sum = 0;
-        for (i = 0; i < length; i += 4) 
-        {
-            // loop is unrolled by factor of 4 here for efficiency
-            sum += pSrc[i + 0] * filterCoeffs[i + 0] + 
-                   pSrc[i + 1] * filterCoeffs[i + 1] + 
-                   pSrc[i + 2] * filterCoeffs[i + 2] + 
-                   pSrc[i + 3] * filterCoeffs[i + 3];
-        }
-#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;
-    }
-    return end;
-}
-
-
-uint FIRFilter::evaluateFilterMulti(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples, uint numChannels)
-{
-    int j, end;
-
-#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);
-    assert(numChannels < 16);
-
-    end = numChannels * (numSamples - length);
-
-    #pragma omp parallel for
-    for (j = 0; j < end; j += numChannels)
-    {
-        const SAMPLETYPE *ptr;
-        LONG_SAMPLETYPE sums[16];
-        uint c, i;
-
-        for (c = 0; c < numChannels; c ++)
-        {
-            sums[c] = 0;
-        }
-
-        ptr = src + j;
-
-        for (i = 0; i < length; 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;
-#else
-            sums[c] *= dScaler;
-#endif // SOUNDTOUCH_INTEGER_SAMPLES
-            dest[j+c] = (SAMPLETYPE)sums[c];
-        }
-    }
-    return numSamples - length;
-}
-
-
-// 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) 
-{
-    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 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

@@ -1,139 +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
-///
-////////////////////////////////////////////////////////////////////////////////
-//
-// 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;
-    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;
-    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

@@ -1,196 +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;
-}
+////////////////////////////////////////////////////////////////////////////////
+///
+/// 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

@@ -1,63 +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 void resetRegisters();
-    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);
-
-    double fract;
-
-public:
-    InterpolateCubic();
-};
-
-}
-
-#endif
+////////////////////////////////////////////////////////////////////////////////
+///
+/// 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

@@ -1,296 +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 = (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;
-}
+////////////////////////////////////////////////////////////////////////////////
+///
+/// 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

@@ -1,88 +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 void resetRegisters();
-
-    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:
-    InterpolateLinearInteger();
-
-    /// Sets new target rate. Normal rate = 1.0, smaller values represent slower 
-    /// rate, larger faster rates.
-    virtual void setRate(double newRate);
-};
-
-
-/// Linear transposer class that uses floating point arithmetic
-class InterpolateLinearFloat : public TransposerBase
-{
-protected:
-    double fract;
-
-    virtual void resetRegisters();
-
-    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();
-};
-
-}
-
-#endif
+////////////////////////////////////////////////////////////////////////////////
+///
+/// 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

@@ -1,181 +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 *pdest, 
-                    const SAMPLETYPE *psrc, 
-                    int &srcSamples)
-{
-    // not implemented
-    assert(false);
-    return 0;
-}
+////////////////////////////////////////////////////////////////////////////////
+///
+/// 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

@@ -1,68 +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:
-    void resetRegisters();
-    int transposeMono(SAMPLETYPE *dest, 
-                        const SAMPLETYPE *src, 
-                        int &srcSamples);
-    int transposeStereo(SAMPLETYPE *dest, 
-                        const SAMPLETYPE *src, 
-                        int &srcSamples);
-    int transposeMulti(SAMPLETYPE *dest, 
-                        const SAMPLETYPE *src, 
-                        int &srcSamples);
-
-    double fract;
-
-public:
-    InterpolateShannon();
-};
-
-}
-
-#endif
+////////////////////////////////////////////////////////////////////////////////
+///
+/// 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

@@ -1,74 +1,74 @@
-## 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+=-O3
-
-# 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
+## 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

@@ -1,277 +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
-///
-////////////////////////////////////////////////////////////////////////////////
-//
-// 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 <EFBFBD>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);
-
-    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 <EFBFBD>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

@@ -1,90 +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
-///
-////////////////////////////////////////////////////////////////////////////////
-//
-// 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_
+////////////////////////////////////////////////////////////////////////////////
+///
+/// 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

@@ -1,307 +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
-///
-////////////////////////////////////////////////////////////////////////////////
-//
-// 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();
-}
-
-
-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;
-#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)
-{
-    uint count;
-
-    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) 
-    {
-        count = 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();
-}
-
-
-// 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 (bUseAAFilter) ? pAAFilter->getLength() : 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 NULL;
-    }
-#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)
+///
+/// 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

@@ -1,163 +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
-///
-////////////////////////////////////////////////////////////////////////////////
-//
-// 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 void resetRegisters() = 0;
-
-    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);
-
-    // 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();
-
-    /// 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);
-
-    /// Clears all the samples in the object
-    void clear();
-
-    /// Returns nonzero if there aren't any samples available for outputting.
-    int isEmpty() const;
-
-    /// Return approximate initial input-output latency
-    int getLatency() 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.vcxproj

@@ -20,32 +20,32 @@
   </ItemGroup>
   <PropertyGroup Label="Globals">
     <ProjectGuid>{68A5DD20-7057-448B-8FE0-B6AC8D205509}</ProjectGuid>
-    <WindowsTargetPlatformVersion>8.1</WindowsTargetPlatformVersion>
+    <WindowsTargetPlatformVersion>10.0</WindowsTargetPlatformVersion>
   </PropertyGroup>
   <Import Project="$(VCTargetsPath)\Microsoft.Cpp.Default.props" />
   <PropertyGroup Condition="'$(Configuration)|$(Platform)'=='Debug|Win32'" Label="Configuration">
     <ConfigurationType>StaticLibrary</ConfigurationType>
-    <PlatformToolset>v140</PlatformToolset>
+    <PlatformToolset>v142</PlatformToolset>
     <UseOfMfc>false</UseOfMfc>
-    <CharacterSet>MultiByte</CharacterSet>
+    <CharacterSet>Unicode</CharacterSet>
   </PropertyGroup>
   <PropertyGroup Condition="'$(Configuration)|$(Platform)'=='Release|Win32'" Label="Configuration">
     <ConfigurationType>StaticLibrary</ConfigurationType>
-    <PlatformToolset>v140</PlatformToolset>
+    <PlatformToolset>v142</PlatformToolset>
     <UseOfMfc>false</UseOfMfc>
-    <CharacterSet>MultiByte</CharacterSet>
+    <CharacterSet>Unicode</CharacterSet>
   </PropertyGroup>
   <PropertyGroup Condition="'$(Configuration)|$(Platform)'=='Debug|x64'" Label="Configuration">
     <ConfigurationType>StaticLibrary</ConfigurationType>
-    <PlatformToolset>v140</PlatformToolset>
+    <PlatformToolset>v142</PlatformToolset>
     <UseOfMfc>false</UseOfMfc>
-    <CharacterSet>MultiByte</CharacterSet>
+    <CharacterSet>Unicode</CharacterSet>
   </PropertyGroup>
   <PropertyGroup Condition="'$(Configuration)|$(Platform)'=='Release|x64'" Label="Configuration">
     <ConfigurationType>StaticLibrary</ConfigurationType>
-    <PlatformToolset>v140</PlatformToolset>
+    <PlatformToolset>v142</PlatformToolset>
     <UseOfMfc>false</UseOfMfc>
-    <CharacterSet>MultiByte</CharacterSet>
+    <CharacterSet>Unicode</CharacterSet>
   </PropertyGroup>
   <Import Project="$(VCTargetsPath)\Microsoft.Cpp.props" />
   <ImportGroup Label="ExtensionSettings">
@@ -112,6 +112,7 @@
       <EnableEnhancedInstructionSet>StreamingSIMDExtensions2</EnableEnhancedInstructionSet>
       <XMLDocumentationFileName>$(IntDir)</XMLDocumentationFileName>
       <BrowseInformationFile>$(IntDir)</BrowseInformationFile>
+      <MultiProcessorCompilation>true</MultiProcessorCompilation>
     </ClCompile>
     <ResourceCompile>
       <PreprocessorDefinitions>NDEBUG;%(PreprocessorDefinitions)</PreprocessorDefinitions>
@@ -153,6 +154,7 @@ copy $(OutDir)$(TargetName)$(TargetExt) ..\..\lib</Command>
       </EnableEnhancedInstructionSet>
       <XMLDocumentationFileName>$(IntDir)</XMLDocumentationFileName>
       <BrowseInformationFile>$(IntDir)</BrowseInformationFile>
+      <MultiProcessorCompilation>true</MultiProcessorCompilation>
     </ClCompile>
     <ResourceCompile>
       <PreprocessorDefinitions>NDEBUG;%(PreprocessorDefinitions)</PreprocessorDefinitions>
@@ -183,11 +185,12 @@ copy $(OutDir)$(TargetName)$(TargetExt) ..\..\lib</Command>
       <BrowseInformation>true</BrowseInformation>
       <WarningLevel>Level3</WarningLevel>
       <SuppressStartupBanner>true</SuppressStartupBanner>
-      <DebugInformationFormat>EditAndContinue</DebugInformationFormat>
+      <DebugInformationFormat>ProgramDatabase</DebugInformationFormat>
       <CompileAs>Default</CompileAs>
       <EnableEnhancedInstructionSet>StreamingSIMDExtensions2</EnableEnhancedInstructionSet>
       <XMLDocumentationFileName>$(IntDir)</XMLDocumentationFileName>
       <BrowseInformationFile>$(IntDir)</BrowseInformationFile>
+      <MultiProcessorCompilation>true</MultiProcessorCompilation>
     </ClCompile>
     <ResourceCompile>
       <PreprocessorDefinitions>_DEBUG;%(PreprocessorDefinitions)</PreprocessorDefinitions>
@@ -227,6 +230,7 @@ copy $(OutDir)$(TargetName)$(TargetExt) ..\..\lib</Command>
       </EnableEnhancedInstructionSet>
       <XMLDocumentationFileName>$(IntDir)</XMLDocumentationFileName>
       <BrowseInformationFile>$(IntDir)</BrowseInformationFile>
+      <MultiProcessorCompilation>true</MultiProcessorCompilation>
     </ClCompile>
     <ResourceCompile>
       <PreprocessorDefinitions>_DEBUG;%(PreprocessorDefinitions)</PreprocessorDefinitions>

source/SoundTouch/TDStretch.h

@@ -1,279 +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
-///
-////////////////////////////////////////////////////////////////////////////////
-//
-// 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();
-
-    /// 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();
-
-    /// 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;
-    }
-
-	/// 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);
-        double calcCrossCorrAccumulate(const short *mixingPos, const short *compare, double &norm);
-        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, double &norm);
-        double calcCrossCorrAccumulate(const float *mixingPos, const float *compare, double &norm);
-    };
-
-#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

@@ -1,55 +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
-///
-////////////////////////////////////////////////////////////////////////////////
-//
-// 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

@@ -1,130 +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
-///
-////////////////////////////////////////////////////////////////////////////////
-//
-// 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
-}
+////////////////////////////////////////////////////////////////////////////////
+///
+/// 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

@@ -1,396 +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
-///
-////////////////////////////////////////////////////////////////////////////////
-//
-// 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 = NULL;
-    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 *)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
+////////////////////////////////////////////////////////////////////////////////
+///
+/// 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

@@ -1,365 +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
-///
-////////////////////////////////////////////////////////////////////////////////
-//
-// 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 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 (((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 = 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 *)SOUNDTOUCH_ALIGN_POINTER_16(filterCoeffsUnalign);
-
-    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(((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
+////////////////////////////////////////////////////////////////////////////////
+///
+/// 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

@@ -1,114 +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;
-
-// DllTest main
-int main(int argc, char *argv[])
-{
-    // Check program arguments
-    if (argc < 4)
-    {
-        cout << "Too few arguments. Usage: DllTest [infile.wav] [outfile.wav] [sampletype]" << endl;
-        return -1;
-    }
-
-    const char *inFileName = argv[1];
-    const char *outFileName = argv[2];
-    string str_sampleType = argv[3];
-
-    bool floatSample;
-    if (str_sampleType.compare("float") == 0)
-    {
-        floatSample = true;
-    }
-    else if (str_sampleType.compare("short") == 0)
-    {
-        floatSample = false;
-    }
-    else
-    { 
-        cerr << "Missing or invalid sampletype '" << str_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;
-}
+////////////////////////////////////////////////////////////////////////////////
+///
+/// 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

@@ -22,32 +22,32 @@
     <ProjectGuid>{E3C0726F-28F4-4F0B-8183-B87CA60C063C}</ProjectGuid>
     <Keyword>Win32Proj</Keyword>
     <RootNamespace>DllTest</RootNamespace>
-    <WindowsTargetPlatformVersion>8.1</WindowsTargetPlatformVersion>
+    <WindowsTargetPlatformVersion>10.0</WindowsTargetPlatformVersion>
   </PropertyGroup>
   <Import Project="$(VCTargetsPath)\Microsoft.Cpp.Default.props" />
   <PropertyGroup Condition="'$(Configuration)|$(Platform)'=='Debug|Win32'" Label="Configuration">
     <ConfigurationType>Application</ConfigurationType>
     <UseDebugLibraries>true</UseDebugLibraries>
-    <PlatformToolset>v140</PlatformToolset>
+    <PlatformToolset>v142</PlatformToolset>
     <CharacterSet>Unicode</CharacterSet>
   </PropertyGroup>
   <PropertyGroup Condition="'$(Configuration)|$(Platform)'=='Release|Win32'" Label="Configuration">
     <ConfigurationType>Application</ConfigurationType>
     <UseDebugLibraries>false</UseDebugLibraries>
-    <PlatformToolset>v140</PlatformToolset>
+    <PlatformToolset>v142</PlatformToolset>
     <WholeProgramOptimization>true</WholeProgramOptimization>
     <CharacterSet>Unicode</CharacterSet>
   </PropertyGroup>
   <PropertyGroup Condition="'$(Configuration)|$(Platform)'=='Debug|x64'" Label="Configuration">
     <ConfigurationType>Application</ConfigurationType>
     <UseDebugLibraries>true</UseDebugLibraries>
-    <PlatformToolset>v140</PlatformToolset>
+    <PlatformToolset>v142</PlatformToolset>
     <CharacterSet>Unicode</CharacterSet>
   </PropertyGroup>
   <PropertyGroup Condition="'$(Configuration)|$(Platform)'=='Release|x64'" Label="Configuration">
     <ConfigurationType>Application</ConfigurationType>
     <UseDebugLibraries>false</UseDebugLibraries>
-    <PlatformToolset>v140</PlatformToolset>
+    <PlatformToolset>v142</PlatformToolset>
     <WholeProgramOptimization>true</WholeProgramOptimization>
     <CharacterSet>Unicode</CharacterSet>
   </PropertyGroup>

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

@@ -2,11 +2,8 @@ unit SoundTouchDLL;
 
 //////////////////////////////////////////////////////////////////////////////
 //
-// SoundTouch.dll wrapper for accessing SoundTouch routines from Delphi/Pascal
-//
-//  Module Author : Christian Budde
-//
-//  2014-01-12 fixes by Sandro Cumerlato <sandro.cumerlato 'at' gmail.com>
+// SoundTouch.dll / libSoundTouchDll.so wrapper for accessing SoundTouch
+// routines from Delphi/Pascal/Lazarus
 //
 ////////////////////////////////////////////////////////////////////////////////
 //
@@ -33,8 +30,8 @@ unit SoundTouchDLL;
 
 interface
 
-uses
-  Windows;
+//uses
+  //Windows;
 
 type
   TSoundTouchHandle = THandle;
@@ -50,7 +47,7 @@ type
 
   // Get SoundTouch library version string 2
   TSoundTouchGetVersionString2 = procedure(VersionString : PAnsiChar; BufferSize : Integer); cdecl;
-  
+
   // Get SoundTouch library version Id
   TSoundTouchGetVersionId = function : Cardinal; cdecl;
 
@@ -107,6 +104,13 @@ type
                                                              //< contains data for both channels.
                                     ); 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); cdecl;
@@ -131,16 +135,20 @@ type
   // Returns number of samples currently unprocessed.
   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; 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; cdecl;
 
@@ -170,6 +178,7 @@ var
   SoundTouchGetSetting            : TSoundTouchGetSetting;
   SoundTouchNumUnprocessedSamples : TSoundTouchNumUnprocessedSamples;
   SoundTouchReceiveSamples        : TSoundTouchReceiveSamples;
+  SoundTouchReceiveSamplesI16     : TSoundTouchReceiveSamplesI16;
   SoundTouchNumSamples            : TSoundTouchNumSamples;
   SoundTouchIsEmpty               : TSoundTouchIsEmpty;
 
@@ -232,6 +241,9 @@ type
     property IsEmpty: Integer read GetIsEmpty;
   end;
 
+  // list of exported functions and procedures
+function IsSoundTouchLoaded: Boolean;
+
 implementation
 
 { TSoundTouch }
@@ -416,19 +428,23 @@ begin
 end;
 
 var
-  SoundTouchLibHandle: HINST;
-  SoundTouchDLLFile: 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: CInt32; sampleRate : CInt32): THandle; cdecl;
+  bpm_createInstance: function(chan: int32; sampleRate : int32): THandle; cdecl;
   bpm_destroyInstance: procedure(h: THandle); cdecl;
-  bpm_getBpm: function(h: THandle): cfloat; cdecl;
-  bpm_putSamples: procedure(h: THandle; const samples: pcfloat;
-  numSamples: cardinal); cdecl;
+  bpm_getBpm: function(h: THandle): Single; cdecl;
+  bpm_putSamples: procedure(h: THandle; const samples: PSingle; numSamples: cardinal); cdecl;
 
 procedure InitDLL;
 begin
-  SoundTouchLibHandle := LoadLibrary(SoundTouchDLLFile);
+{$ifdef mswindows}  // Windows
+  SoundTouchLibHandle := LoadLibrary('.\SoundTouchDll.dll');
+{$else}  // Unix
+  SoundTouchLibHandle := LoadLibrary('./libSoundTouchDll.so');
+{$endif}
   if SoundTouchLibHandle <> 0 then
   try
     Pointer(SoundTouchCreateInstance)        := GetProcAddress(SoundTouchLibHandle, 'soundtouch_createInstance');
@@ -461,7 +477,7 @@ begin
     Pointer(bpm_destroyInstance)            :=GetProcAddress(SoundTouchLibHandle, 'bpm_destroyInstance');
     Pointer(bpm_getBpm)                     :=GetProcAddress(SoundTouchLibHandle, 'bpm_getBpm');
     Pointer(bpm_putSamples)                 :=GetProcAddress(SoundTouchLibHandle, 'bpm_putSamples');
-        
+
   except
     FreeLibrary(SoundTouchLibHandle);
     SoundTouchLibHandle := 0;
@@ -473,6 +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>

source/SoundTouchDLL/LazarusTest/soundtouchtest.lpr

@@ -0,0 +1,25 @@
+program soundtouchtest;
+
+{$mode objfpc}{$H+}
+
+uses
+  {$IFDEF UNIX}
+  cthreads,
+  {$ENDIF}
+  {$IFDEF HASAMIGA}
+  athreads,
+  {$ENDIF}
+  Interfaces, // this includes the LCL widgetset
+  Forms, main
+  { you can add units after this };
+
+{$R *.res}
+
+begin
+  RequireDerivedFormResource:=True;
+  Application.Scaled:=True;
+  Application.Initialize;
+  Application.CreateForm(TForm1, Form1);
+  Application.Run;
+end.
+

source/SoundTouchDLL/LazarusTest/soundtouchtest.lps

@@ -0,0 +1,186 @@
+<?xml version="1.0" encoding="UTF-8"?>
+<CONFIG>
+  <ProjectSession>
+    <Version Value="12"/>
+    <BuildModes Active="Default"/>
+    <Units>
+      <Unit>
+        <Filename Value="soundtouchtest.lpr"/>
+        <IsPartOfProject Value="True"/>
+        <EditorIndex Value="-1"/>
+        <WindowIndex Value="-1"/>
+        <TopLine Value="-1"/>
+        <CursorPos X="-1" Y="-1"/>
+        <UsageCount Value="21"/>
+      </Unit>
+      <Unit>
+        <Filename Value="main.pas"/>
+        <IsPartOfProject Value="True"/>
+        <ComponentName Value="Form1"/>
+        <HasResources Value="True"/>
+        <ResourceBaseClass Value="Form"/>
+        <IsVisibleTab Value="True"/>
+        <CursorPos X="26" Y="43"/>
+        <UsageCount Value="21"/>
+        <Loaded Value="True"/>
+        <LoadedDesigner Value="True"/>
+      </Unit>
+      <Unit>
+        <Filename Value="../SoundTouchDLL.pas"/>
+        <EditorIndex Value="-1"/>
+        <TopLine Value="37"/>
+        <CursorPos X="19"/>
+        <UsageCount Value="10"/>
+      </Unit>
+      <Unit>
+        <Filename Value="/usr/lib/lazarus/2.2.0/lcl/interfaces/gtk2/gtk2proc.inc"/>
+        <EditorIndex Value="-1"/>
+        <TopLine Value="7149"/>
+        <CursorPos X="3" Y="7184"/>
+        <UsageCount Value="10"/>
+      </Unit>
+      <Unit>
+        <Filename Value="/usr/lib/lazarus/2.2.0/components/freetype/easylazfreetype.pas"/>
+        <UnitName Value="EasyLazFreeType"/>
+        <EditorIndex Value="-1"/>
+        <TopLine Value="539"/>
+        <CursorPos X="16" Y="574"/>
+        <UsageCount Value="10"/>
+      </Unit>
+      <Unit>
+        <Filename Value="SoundTouchDLL.pas"/>
+        <EditorIndex Value="1"/>
+        <TopLine Value="326"/>
+        <CursorPos X="127" Y="379"/>
+        <UsageCount Value="10"/>
+        <Loaded Value="True"/>
+      </Unit>
+    </Units>
+    <JumpHistory HistoryIndex="29">
+      <Position>
+        <Filename Value="SoundTouchDLL.pas"/>
+        <Caret Line="439" TopLine="403"/>
+      </Position>
+      <Position>
+        <Filename Value="SoundTouchDLL.pas"/>
+        <Caret Line="427" Column="37" TopLine="403"/>
+      </Position>
+      <Position>
+        <Filename Value="SoundTouchDLL.pas"/>
+        <Caret Line="439" TopLine="403"/>
+      </Position>
+      <Position>
+        <Filename Value="SoundTouchDLL.pas"/>
+        <Caret Line="427" Column="32" TopLine="403"/>
+      </Position>
+      <Position>
+        <Filename Value="SoundTouchDLL.pas"/>
+        <Caret Line="444" Column="48" TopLine="403"/>
+      </Position>
+      <Position>
+        <Filename Value="SoundTouchDLL.pas"/>
+        <Caret Line="439" TopLine="403"/>
+      </Position>
+      <Position>
+        <Filename Value="SoundTouchDLL.pas"/>
+        <Caret Line="416" Column="116" TopLine="403"/>
+      </Position>
+      <Position>
+        <Filename Value="SoundTouchDLL.pas"/>
+        <Caret Line="439" TopLine="403"/>
+      </Position>
+      <Position>
+        <Filename Value="main.pas"/>
+        <Caret Line="45" Column="40"/>
+      </Position>
+      <Position>
+        <Filename Value="SoundTouchDLL.pas"/>
+        <Caret Line="243" Column="38" TopLine="197"/>
+      </Position>
+      <Position>
+        <Filename Value="SoundTouchDLL.pas"/>
+        <Caret Line="487" Column="38" TopLine="429"/>
+      </Position>
+      <Position>
+        <Filename Value="main.pas"/>
+        <Caret Line="42" Column="8"/>
+      </Position>
+      <Position>
+        <Filename Value="main.pas"/>
+        <Caret Line="41" Column="44"/>
+      </Position>
+      <Position>
+        <Filename Value="main.pas"/>
+        <Caret Line="44" Column="6"/>
+      </Position>
+      <Position>
+        <Filename Value="main.pas"/>
+        <Caret Line="43" Column="39"/>
+      </Position>
+      <Position>
+        <Filename Value="main.pas"/>
+        <Caret Line="44" Column="7"/>
+      </Position>
+      <Position>
+        <Filename Value="main.pas"/>
+        <Caret Line="38" Column="7"/>
+      </Position>
+      <Position>
+        <Filename Value="SoundTouchDLL.pas"/>
+        <Caret Line="467" TopLine="423"/>
+      </Position>
+      <Position>
+        <Filename Value="SoundTouchDLL.pas"/>
+        <Caret Line="212" Column="32" TopLine="78"/>
+      </Position>
+      <Position>
+        <Filename Value="SoundTouchDLL.pas"/>
+        <Caret Line="361" Column="37" TopLine="308"/>
+      </Position>
+      <Position>
+        <Filename Value="SoundTouchDLL.pas"/>
+        <Caret Line="466" Column="37" TopLine="413"/>
+      </Position>
+      <Position>
+        <Filename Value="SoundTouchDLL.pas"/>
+        <Caret Line="467" Column="37" TopLine="413"/>
+      </Position>
+      <Position>
+        <Filename Value="SoundTouchDLL.pas"/>
+        <Caret Line="466" Column="13" TopLine="413"/>
+      </Position>
+      <Position>
+        <Filename Value="SoundTouchDLL.pas"/>
+        <Caret Line="361" Column="13" TopLine="326"/>
+      </Position>
+      <Position>
+        <Filename Value="SoundTouchDLL.pas"/>
+        <Caret Line="110" Column="3" TopLine="74"/>
+      </Position>
+      <Position>
+        <Filename Value="SoundTouchDLL.pas"/>
+        <Caret Line="180" Column="37" TopLine="145"/>
+      </Position>
+      <Position>
+        <Filename Value="SoundTouchDLL.pas"/>
+        <Caret Line="179" Column="55" TopLine="145"/>
+      </Position>
+      <Position>
+        <Filename Value="SoundTouchDLL.pas"/>
+        <Caret Line="146" Column="3" TopLine="145"/>
+      </Position>
+      <Position>
+        <Filename Value="SoundTouchDLL.pas"/>
+        <Caret Line="149" TopLine="111"/>
+      </Position>
+      <Position>
+        <Filename Value="SoundTouchDLL.pas"/>
+        <Caret Line="151" Column="31" TopLine="116"/>
+      </Position>
+    </JumpHistory>
+    <RunParams>
+      <FormatVersion Value="2"/>
+      <Modes ActiveMode=""/>
+    </RunParams>
+  </ProjectSession>
+</CONFIG>

source/SoundTouchDLL/Makefile.am

@@ -0,0 +1,47 @@
+## 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 $(top_srcdir)/config/am_include.mk
+
+noinst_HEADERS=../SoundTouch/AAFilter.h ../SoundTouch/cpu_detect.h ../SoundTouch/cpu_detect_x86.cpp ../SoundTouch/FIRFilter.h \
+    ../SoundTouch/RateTransposer.h ../SoundTouch/TDStretch.h ../SoundTouch/PeakFinder.h ../SoundTouch/InterpolateCubic.h \
+    ../SoundTouch/InterpolateLinear.h ../SoundTouch/InterpolateShannon.h
+
+include_HEADERS=SoundTouchDLL.h
+
+lib_LTLIBRARIES=libSoundTouchDll.la
+#
+libSoundTouchDll_la_SOURCES=../SoundTouch/AAFilter.cpp ../SoundTouch/FIRFilter.cpp \
+    ../SoundTouch/FIFOSampleBuffer.cpp ../SoundTouch/RateTransposer.cpp ../SoundTouch/SoundTouch.cpp \
+    ../SoundTouch/TDStretch.cpp ../SoundTouch/sse_optimized.cpp ../SoundTouch/cpu_detect_x86.cpp \
+    ../SoundTouch/BPMDetect.cpp ../SoundTouch/PeakFinder.cpp ../SoundTouch/InterpolateLinear.cpp \
+    ../SoundTouch/InterpolateCubic.cpp ../SoundTouch/InterpolateShannon.cpp SoundTouchDLL.cpp
+
+# Compiler flags
+
+# Modify the default 0.0.0 to LIB_SONAME.0.0
+AM_LDFLAGS=$(LDFLAGS) -version-info @LIB_SONAME@
+
+if X86
+CXXFLAGS1=-mstackrealign -msse
+endif
+
+if X86_64
+CXXFLAGS2=-fPIC
+endif
+
+AM_CXXFLAGS=$(CXXFLAGS) $(CXXFLAGS1) $(CXXFLAGS2) -shared -DDLL_EXPORTS -fvisibility=hidden

source/SoundTouchDLL/SoundTouchDLL.h

@@ -1,229 +1,240 @@
-//////////////////////////////////////////////////////////////////////////////
-///
-/// SoundTouch DLL wrapper - wraps SoundTouch routines into a Dynamic Load 
-/// Library interface.
-///
-/// Author        : Copyright (c) Olli Parviainen
-/// Author e-mail : oparviai 'at' iki.fi
-/// SoundTouch WWW: http://www.surina.net/soundtouch
-///
-////////////////////////////////////////////////////////////////////////////////
-//
-// License :
-//
-//  SoundTouch audio processing library
-//  Copyright (c) Olli Parviainen
-//
-//  This library is free software; you can redistribute it and/or
-//  modify it under the terms of the GNU Lesser General Public
-//  License as published by the Free Software Foundation; either
-//  version 2.1 of the License, or (at your option) any later version.
-//
-//  This library is distributed in the hope that it will be useful,
-//  but WITHOUT ANY WARRANTY; without even the implied warranty of
-//  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-//  Lesser General Public License for more details.
-//
-//  You should have received a copy of the GNU Lesser General Public
-//  License along with this library; if not, write to the Free Software
-//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
-//
-////////////////////////////////////////////////////////////////////////////////
-
-#ifndef _SoundTouchDLL_h_
-#define _SoundTouchDLL_h_
-
-#if defined(_WIN32) || defined(WIN32)
-    // Windows
-    #ifndef __cplusplus
-        #error "Expected g++"
-    #endif
-
-    #ifdef DLL_EXPORTS
-        #define SOUNDTOUCHDLL_API extern "C" __declspec(dllexport)
-    #else
-        #define SOUNDTOUCHDLL_API extern "C" __declspec(dllimport)
-    #endif
-
-#else
-    // GNU version
-
-    #ifdef DLL_EXPORTS
-        // GCC declaration for exporting functions
-        #define SOUNDTOUCHDLL_API extern "C" __attribute__((__visibility__("default")))
-    #else
-        // GCC doesn't require DLL imports
-        #define SOUNDTOUCHDLL_API
-    #endif
-
-    // Linux-replacements for Windows declarations:
-    #define __cdecl
-    typedef unsigned int DWORD;
-    #define FALSE    0
-    #define TRUE    1
-
-#endif
-
-typedef void * HANDLE;
-
-/// Create a new instance of SoundTouch processor.
-SOUNDTOUCHDLL_API HANDLE __cdecl soundtouch_createInstance();
-
-/// Destroys a SoundTouch processor instance.
-SOUNDTOUCHDLL_API void __cdecl soundtouch_destroyInstance(HANDLE h);
-
-/// Get SoundTouch library version string
-SOUNDTOUCHDLL_API const char *__cdecl soundtouch_getVersionString();
-
-/// Get SoundTouch library version string - alternative function for 
-/// environments that can't properly handle character string as return value
-SOUNDTOUCHDLL_API void __cdecl soundtouch_getVersionString2(char* versionString, int bufferSize);
-
-/// Get SoundTouch library version Id
-SOUNDTOUCHDLL_API unsigned int __cdecl soundtouch_getVersionId();
-
-/// Sets new rate control value. Normal rate = 1.0, smaller values
-/// represent slower rate, larger faster rates.
-SOUNDTOUCHDLL_API void __cdecl soundtouch_setRate(HANDLE h, float newRate);
-
-/// Sets new tempo control value. Normal tempo = 1.0, smaller values
-/// represent slower tempo, larger faster tempo.
-SOUNDTOUCHDLL_API void __cdecl soundtouch_setTempo(HANDLE h, float newTempo);
-
-/// Sets new rate control value as a difference in percents compared
-/// to the original rate (-50 .. +100 %);
-SOUNDTOUCHDLL_API void __cdecl soundtouch_setRateChange(HANDLE h, float newRate);
-
-/// Sets new tempo control value as a difference in percents compared
-/// to the original tempo (-50 .. +100 %);
-SOUNDTOUCHDLL_API void __cdecl soundtouch_setTempoChange(HANDLE h, float newTempo);
-
-/// Sets new pitch control value. Original pitch = 1.0, smaller values
-/// represent lower pitches, larger values higher pitch.
-SOUNDTOUCHDLL_API void __cdecl soundtouch_setPitch(HANDLE h, float newPitch);
-
-/// Sets pitch change in octaves compared to the original pitch  
-/// (-1.00 .. +1.00);
-SOUNDTOUCHDLL_API void __cdecl soundtouch_setPitchOctaves(HANDLE h, float newPitch);
-
-/// Sets pitch change in semi-tones compared to the original pitch
-/// (-12 .. +12);
-SOUNDTOUCHDLL_API void __cdecl soundtouch_setPitchSemiTones(HANDLE h, float newPitch);
-
-
-/// Sets the number of channels, 1 = mono, 2 = stereo, n = multichannel
-SOUNDTOUCHDLL_API void __cdecl soundtouch_setChannels(HANDLE h, unsigned int numChannels);
-
-/// Sets sample rate.
-SOUNDTOUCHDLL_API void __cdecl soundtouch_setSampleRate(HANDLE h, unsigned int srate);
-
-/// Flushes the last samples from the processing pipeline to the output.
-/// Clears also the internal processing buffers.
-//
-/// Note: This function is meant for extracting the last samples of a sound
-/// stream. This function may introduce additional blank samples in the end
-/// of the sound stream, and thus it's not recommended to call this function
-/// in the middle of a sound stream.
-SOUNDTOUCHDLL_API void __cdecl soundtouch_flush(HANDLE h);
-
-/// Adds 'numSamples' pcs of samples from the 'samples' memory position into
-/// the input of the object. Notice that sample rate _has_to_ be set before
-/// calling this function, otherwise throws a runtime_error exception.
-SOUNDTOUCHDLL_API void __cdecl soundtouch_putSamples(HANDLE h, 
-        const float *samples,       ///< Pointer to sample buffer.
-        unsigned int numSamples     ///< Number of sample frames in buffer. Notice
-                                    ///< that in case of multi-channel sound a single 
-                                    ///< sample frame contains data for all channels.
-);
-
-/// int16 version of soundtouch_putSamples(): This accept int16 (short) sample data
-/// and internally converts it to float format before processing
-SOUNDTOUCHDLL_API void __cdecl soundtouch_putSamples_i16(HANDLE h,
-        const short *samples,       ///< Pointer to sample buffer.
-        unsigned int numSamples     ///< Number of sample frames in buffer. Notice
-                                    ///< that in case of multi-channel sound a single 
-                                    ///< sample frame contains data for all channels.
-);
-
-
-/// Clears all the samples in the object's output and internal processing
-/// buffers.
-SOUNDTOUCHDLL_API void __cdecl soundtouch_clear(HANDLE h);
-
-/// Changes a setting controlling the processing system behaviour. See the
-/// 'SETTING_...' defines for available setting ID's.
-/// 
-/// \return 'nonzero' if the setting was successfully changed, otherwise zero
-SOUNDTOUCHDLL_API int __cdecl soundtouch_setSetting(HANDLE h, 
-        int settingId,   ///< Setting ID number. see SETTING_... defines.
-        int value        ///< New setting value.
-);
-
-/// Reads a setting controlling the processing system behaviour. See the
-/// 'SETTING_...' defines for available setting ID's.
-///
-/// \return the setting value.
-SOUNDTOUCHDLL_API int __cdecl soundtouch_getSetting(HANDLE h, 
-        int settingId    ///< Setting ID number, see SETTING_... defines.
-);
-
-
-/// Returns number of samples currently unprocessed.
-SOUNDTOUCHDLL_API unsigned int __cdecl soundtouch_numUnprocessedSamples(HANDLE h);
-
-/// Adjusts book-keeping so that given number of samples are removed from beginning of the 
-/// sample buffer without copying them anywhere. 
-///
-/// Used to reduce the number of samples in the buffer when accessing the sample buffer directly
-/// with 'ptrBegin' function.
-SOUNDTOUCHDLL_API unsigned int __cdecl soundtouch_receiveSamples(HANDLE h, 
-        float *outBuffer,           ///< Buffer where to copy output samples.
-        unsigned int maxSamples     ///< How many samples to receive at max.
-);
-
-
-/// int16 version of soundtouch_receiveSamples(): This converts internal float samples
-/// into int16 (short) return data type
-SOUNDTOUCHDLL_API unsigned int __cdecl soundtouch_receiveSamples_i16(HANDLE h,
-        short *outBuffer,           ///< Buffer where to copy output samples.
-        unsigned int maxSamples     ///< How many samples to receive at max.
-);
-
-/// Returns number of samples currently available.
-SOUNDTOUCHDLL_API unsigned int __cdecl soundtouch_numSamples(HANDLE h);
-
-/// Returns nonzero if there aren't any samples available for outputting.
-SOUNDTOUCHDLL_API int __cdecl soundtouch_isEmpty(HANDLE h);
-
-/// Create a new instance of BPM detector
-SOUNDTOUCHDLL_API HANDLE __cdecl bpm_createInstance(int numChannels, int sampleRate);
-
-/// Destroys a BPM detector instance.
-SOUNDTOUCHDLL_API void __cdecl bpm_destroyInstance(HANDLE h);
-
-/// Feed 'numSamples' sample frames from 'samples' into the BPM detector.
-SOUNDTOUCHDLL_API void __cdecl bpm_putSamples(HANDLE h, 
-        const float *samples,           ///< Pointer to sample buffer.
-        unsigned int numSamples         ///< Number of samples in buffer. Notice
-                                        ///< that in case of stereo-sound a single sample
-                                        ///< contains data for both channels.
-        );
-
-/// Feed 'numSamples' sample frames from 'samples' into the BPM detector.
-/// 16bit int sample format version.
-SOUNDTOUCHDLL_API void __cdecl bpm_putSamples_i16(HANDLE h, 
-        const short *samples,           ///< Pointer to sample buffer.
-        unsigned int numSamples         ///< Number of samples in buffer. Notice
-                                        ///< that in case of stereo-sound a single sample
-                                        ///< contains data for both channels.
-        );
-
-/// 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.
-SOUNDTOUCHDLL_API float __cdecl bpm_getBpm(HANDLE h);
-
-#endif  // _SoundTouchDLL_h_
-
+//////////////////////////////////////////////////////////////////////////////
+///
+/// SoundTouch DLL wrapper - wraps SoundTouch routines into a Dynamic Load
+/// Library interface.
+///
+/// Author        : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+//  SoundTouch audio processing library
+//  Copyright (c) Olli Parviainen
+//
+//  This library is free software; you can redistribute it and/or
+//  modify it under the terms of the GNU Lesser General Public
+//  License as published by the Free Software Foundation; either
+//  version 2.1 of the License, or (at your option) any later version.
+//
+//  This library is distributed in the hope that it will be useful,
+//  but WITHOUT ANY WARRANTY; without even the implied warranty of
+//  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+//  Lesser General Public License for more details.
+//
+//  You should have received a copy of the GNU Lesser General Public
+//  License along with this library; if not, write to the Free Software
+//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#ifndef _SoundTouchDLL_h_
+#define _SoundTouchDLL_h_
+
+#if defined(_WIN32) || defined(WIN32)
+    // Windows
+    #ifndef __cplusplus
+        #error "Expected g++"
+    #endif
+
+    #ifdef DLL_EXPORTS
+        #define SOUNDTOUCHDLL_API extern "C" __declspec(dllexport)
+    #else
+        #define SOUNDTOUCHDLL_API extern "C" __declspec(dllimport)
+    #endif
+
+#else
+    // GNU version
+
+    #if defined(DLL_EXPORTS) || defined(SoundTouchDLL_EXPORTS)
+        // GCC declaration for exporting functions
+        #define SOUNDTOUCHDLL_API extern "C" __attribute__((__visibility__("default")))
+    #else
+        // import function
+        #define SOUNDTOUCHDLL_API extern "C"
+    #endif
+
+    // Linux-replacements for Windows declarations:
+    #define __cdecl
+    typedef unsigned int DWORD;
+    #define FALSE    0
+    #define TRUE    1
+
+#endif
+
+typedef void * HANDLE;
+
+/// Create a new instance of SoundTouch processor.
+SOUNDTOUCHDLL_API HANDLE __cdecl soundtouch_createInstance();
+
+/// Destroys a SoundTouch processor instance.
+SOUNDTOUCHDLL_API void __cdecl soundtouch_destroyInstance(HANDLE h);
+
+/// Get SoundTouch library version string
+SOUNDTOUCHDLL_API const char *__cdecl soundtouch_getVersionString();
+
+/// Get SoundTouch library version string - alternative function for
+/// environments that can't properly handle character string as return value
+SOUNDTOUCHDLL_API void __cdecl soundtouch_getVersionString2(char* versionString, int bufferSize);
+
+/// Get SoundTouch library version Id
+SOUNDTOUCHDLL_API unsigned int __cdecl soundtouch_getVersionId();
+
+/// Sets new rate control value. Normal rate = 1.0, smaller values
+/// represent slower rate, larger faster rates.
+SOUNDTOUCHDLL_API void __cdecl soundtouch_setRate(HANDLE h, float newRate);
+
+/// Sets new tempo control value. Normal tempo = 1.0, smaller values
+/// represent slower tempo, larger faster tempo.
+SOUNDTOUCHDLL_API void __cdecl soundtouch_setTempo(HANDLE h, float newTempo);
+
+/// Sets new rate control value as a difference in percents compared
+/// to the original rate (-50 .. +100 %);
+SOUNDTOUCHDLL_API void __cdecl soundtouch_setRateChange(HANDLE h, float newRate);
+
+/// Sets new tempo control value as a difference in percents compared
+/// to the original tempo (-50 .. +100 %);
+SOUNDTOUCHDLL_API void __cdecl soundtouch_setTempoChange(HANDLE h, float newTempo);
+
+/// Sets new pitch control value. Original pitch = 1.0, smaller values
+/// represent lower pitches, larger values higher pitch.
+SOUNDTOUCHDLL_API void __cdecl soundtouch_setPitch(HANDLE h, float newPitch);
+
+/// Sets pitch change in octaves compared to the original pitch
+/// (-1.00 .. +1.00);
+SOUNDTOUCHDLL_API void __cdecl soundtouch_setPitchOctaves(HANDLE h, float newPitch);
+
+/// Sets pitch change in semi-tones compared to the original pitch
+/// (-12 .. +12);
+SOUNDTOUCHDLL_API void __cdecl soundtouch_setPitchSemiTones(HANDLE h, float newPitch);
+
+
+/// Sets the number of channels, 1 = mono, 2 = stereo, n = multichannel
+SOUNDTOUCHDLL_API int __cdecl soundtouch_setChannels(HANDLE h, unsigned int numChannels);
+
+/// Sets sample rate.
+SOUNDTOUCHDLL_API int __cdecl soundtouch_setSampleRate(HANDLE h, unsigned int srate);
+
+/// Flushes the last samples from the processing pipeline to the output.
+/// Clears also the internal processing buffers.
+//
+/// Note: This function is meant for extracting the last samples of a sound
+/// stream. This function may introduce additional blank samples in the end
+/// of the sound stream, and thus it's not recommended to call this function
+/// in the middle of a sound stream.
+SOUNDTOUCHDLL_API int __cdecl soundtouch_flush(HANDLE h);
+
+/// Adds 'numSamples' pcs of samples from the 'samples' memory position into
+/// the input of the object. Notice that sample rate _has_to_ be set before
+/// calling this function, otherwise throws a runtime_error exception.
+SOUNDTOUCHDLL_API int __cdecl soundtouch_putSamples(HANDLE h,
+        const float *samples,       ///< Pointer to sample buffer.
+        unsigned int numSamples     ///< Number of sample frames in buffer. Notice
+                                    ///< that in case of multi-channel sound a single
+                                    ///< sample frame contains data for all channels.
+);
+
+/// int16 version of soundtouch_putSamples(): This accept int16 (short) sample data
+/// and internally converts it to float format before processing
+SOUNDTOUCHDLL_API void __cdecl soundtouch_putSamples_i16(HANDLE h,
+        const short *samples,       ///< Pointer to sample buffer.
+        unsigned int numSamples     ///< Number of sample frames in buffer. Notice
+                                    ///< that in case of multi-channel sound a single
+                                    ///< sample frame contains data for all channels.
+);
+
+
+/// Clears all the samples in the object's output and internal processing
+/// buffers.
+SOUNDTOUCHDLL_API void __cdecl soundtouch_clear(HANDLE h);
+
+/// Changes a setting controlling the processing system behaviour. See the
+/// 'SETTING_...' defines for available setting ID's.
+///
+/// \return 'nonzero' if the setting was successfully changed, otherwise zero
+SOUNDTOUCHDLL_API int __cdecl soundtouch_setSetting(HANDLE h,
+        int settingId,   ///< Setting ID number. see SETTING_... defines.
+        int value        ///< New setting value.
+);
+
+/// Reads a setting controlling the processing system behaviour. See the
+/// 'SETTING_...' defines for available setting ID's.
+///
+/// \return the setting value.
+SOUNDTOUCHDLL_API int __cdecl soundtouch_getSetting(HANDLE h,
+        int settingId    ///< Setting ID number, see SETTING_... defines.
+);
+
+
+/// Returns number of samples currently unprocessed.
+SOUNDTOUCHDLL_API unsigned int __cdecl soundtouch_numUnprocessedSamples(HANDLE h);
+
+/// Adjusts book-keeping so that given number of samples are removed from beginning of the
+/// sample buffer without copying them anywhere.
+///
+/// Used to reduce the number of samples in the buffer when accessing the sample buffer directly
+/// with 'ptrBegin' function.
+SOUNDTOUCHDLL_API unsigned int __cdecl soundtouch_receiveSamples(HANDLE h,
+        float *outBuffer,           ///< Buffer where to copy output samples.
+        unsigned int maxSamples     ///< How many samples to receive at max.
+);
+
+
+/// int16 version of soundtouch_receiveSamples(): This converts internal float samples
+/// into int16 (short) return data type
+SOUNDTOUCHDLL_API unsigned int __cdecl soundtouch_receiveSamples_i16(HANDLE h,
+        short *outBuffer,           ///< Buffer where to copy output samples.
+        unsigned int maxSamples     ///< How many samples to receive at max.
+);
+
+/// Returns number of samples currently available.
+SOUNDTOUCHDLL_API unsigned int __cdecl soundtouch_numSamples(HANDLE h);
+
+/// Returns nonzero if there aren't any samples available for outputting.
+SOUNDTOUCHDLL_API int __cdecl soundtouch_isEmpty(HANDLE h);
+
+/// Create a new instance of BPM detector
+SOUNDTOUCHDLL_API HANDLE __cdecl bpm_createInstance(int numChannels, int sampleRate);
+
+/// Destroys a BPM detector instance.
+SOUNDTOUCHDLL_API void __cdecl bpm_destroyInstance(HANDLE h);
+
+/// Feed 'numSamples' sample frames from 'samples' into the BPM detector.
+SOUNDTOUCHDLL_API void __cdecl bpm_putSamples(HANDLE h,
+        const float *samples,           ///< Pointer to sample buffer.
+        unsigned int numSamples         ///< Number of samples in buffer. Notice
+                                        ///< that in case of stereo-sound a single sample
+                                        ///< contains data for both channels.
+        );
+
+/// Feed 'numSamples' sample frames from 'samples' into the BPM detector.
+/// 16bit int sample format version.
+SOUNDTOUCHDLL_API void __cdecl bpm_putSamples_i16(HANDLE h,
+        const short *samples,           ///< Pointer to sample buffer.
+        unsigned int numSamples         ///< Number of samples in buffer. Notice
+                                        ///< that in case of stereo-sound a single sample
+                                        ///< contains data for both channels.
+        );
+
+/// 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.
+SOUNDTOUCHDLL_API float __cdecl bpm_getBpm(HANDLE h);
+
+/// 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.
+SOUNDTOUCHDLL_API int __cdecl bpm_getBeats(HANDLE h, float *pos, float *strength, int count);
+
+#endif  // _SoundTouchDLL_h_
+

source/SoundTouchDLL/SoundTouchDLL.rc

@@ -25,18 +25,18 @@ LANGUAGE LANG_ENGLISH, SUBLANG_ENGLISH_US
 // TEXTINCLUDE
 //
 
-1 TEXTINCLUDE 
+1 TEXTINCLUDE
 BEGIN
     "resource.h\0"
 END
 
-2 TEXTINCLUDE 
+2 TEXTINCLUDE
 BEGIN
     "#include ""afxres.h""\r\n"
     "\0"
 END
 
-3 TEXTINCLUDE 
+3 TEXTINCLUDE
 BEGIN
     "\r\n"
     "\0"
@@ -51,8 +51,8 @@ END
 //
 
 VS_VERSION_INFO VERSIONINFO
- FILEVERSION 2,0,0,0
- PRODUCTVERSION 2,0,0,0
+ FILEVERSION 2,3,2,0
+ PRODUCTVERSION 2,3,2,0
  FILEFLAGSMASK 0x17L
 #ifdef _DEBUG
  FILEFLAGS 0x1L
@@ -69,12 +69,12 @@ BEGIN
         BEGIN
             VALUE "Comments", "SoundTouch Library licensed for 3rd party applications subject to LGPL license v2.1. Visit http://www.surina.net/soundtouch for more information about the SoundTouch library."
             VALUE "FileDescription", "SoundTouch Dynamic Link Library"
-            VALUE "FileVersion", "2.0.0.0"
+            VALUE "FileVersion", "2.3.3.0"
             VALUE "InternalName", "SoundTouch"
-            VALUE "LegalCopyright", "Copyright (C) Olli Parviainen 2017"
+            VALUE "LegalCopyright", "Copyright (C) Olli Parviainen 2024"
             VALUE "OriginalFilename", "SoundTouch.dll"
             VALUE "ProductName", " SoundTouch Dynamic Link Library"
-            VALUE "ProductVersion", "2.0.0.0"
+            VALUE "ProductVersion", "2.3.3.0"
         END
     END
     BLOCK "VarFileInfo"

source/SoundTouchDLL/SoundTouchDLL.vcxproj

@@ -21,28 +21,28 @@
   <PropertyGroup Label="Globals">
     <ProjectGuid>{164DE61D-6391-4265-8273-30740117D356}</ProjectGuid>
     <Keyword>Win32Proj</Keyword>
-    <WindowsTargetPlatformVersion>8.1</WindowsTargetPlatformVersion>
+    <WindowsTargetPlatformVersion>10.0</WindowsTargetPlatformVersion>
   </PropertyGroup>
   <Import Project="$(VCTargetsPath)\Microsoft.Cpp.Default.props" />
   <PropertyGroup Condition="'$(Configuration)|$(Platform)'=='Release|Win32'" Label="Configuration">
     <ConfigurationType>DynamicLibrary</ConfigurationType>
-    <PlatformToolset>v140</PlatformToolset>
-    <CharacterSet>MultiByte</CharacterSet>
+    <PlatformToolset>v142</PlatformToolset>
+    <CharacterSet>Unicode</CharacterSet>
   </PropertyGroup>
   <PropertyGroup Condition="'$(Configuration)|$(Platform)'=='Debug|Win32'" Label="Configuration">
     <ConfigurationType>DynamicLibrary</ConfigurationType>
-    <PlatformToolset>v140</PlatformToolset>
-    <CharacterSet>MultiByte</CharacterSet>
+    <PlatformToolset>v142</PlatformToolset>
+    <CharacterSet>Unicode</CharacterSet>
   </PropertyGroup>
   <PropertyGroup Condition="'$(Configuration)|$(Platform)'=='Release|x64'" Label="Configuration">
     <ConfigurationType>DynamicLibrary</ConfigurationType>
-    <PlatformToolset>v140</PlatformToolset>
-    <CharacterSet>MultiByte</CharacterSet>
+    <PlatformToolset>v142</PlatformToolset>
+    <CharacterSet>Unicode</CharacterSet>
   </PropertyGroup>
   <PropertyGroup Condition="'$(Configuration)|$(Platform)'=='Debug|x64'" Label="Configuration">
     <ConfigurationType>DynamicLibrary</ConfigurationType>
-    <PlatformToolset>v140</PlatformToolset>
-    <CharacterSet>MultiByte</CharacterSet>
+    <PlatformToolset>v142</PlatformToolset>
+    <CharacterSet>Unicode</CharacterSet>
   </PropertyGroup>
   <Import Project="$(VCTargetsPath)\Microsoft.Cpp.props" />
   <ImportGroup Label="ExtensionSettings">
@@ -95,7 +95,8 @@
       <Optimization>Disabled</Optimization>
       <AdditionalIncludeDirectories>..\..\include;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>
       <PreprocessorDefinitions>WIN32;_DEBUG;_WINDOWS;_USRDLL;_CRT_SECURE_NO_WARNINGS;DLL_EXPORTS;%(PreprocessorDefinitions)</PreprocessorDefinitions>
-      <MinimalRebuild>true</MinimalRebuild>
+      <MinimalRebuild>
+      </MinimalRebuild>
       <BasicRuntimeChecks>EnableFastChecks</BasicRuntimeChecks>
       <RuntimeLibrary>MultiThreadedDebug</RuntimeLibrary>
       <PrecompiledHeader />
@@ -106,6 +107,7 @@
       <ObjectFileName>$(OutDir)</ObjectFileName>
       <ProgramDataBaseFileName>$(OutDir)</ProgramDataBaseFileName>
       <EnableEnhancedInstructionSet>StreamingSIMDExtensions2</EnableEnhancedInstructionSet>
+      <FloatingPointModel>Fast</FloatingPointModel>
     </ClCompile>
     <Link>
       <OutputFile>$(OutDir)$(TargetName)$(TargetExt)</OutputFile>
@@ -134,7 +136,8 @@ copy $(OutDir)$(TargetName).lib ..\..\lib
       <Optimization>Disabled</Optimization>
       <AdditionalIncludeDirectories>..\..\include;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>
       <PreprocessorDefinitions>WIN32;_DEBUG;_WINDOWS;_USRDLL;_CRT_SECURE_NO_WARNINGS;DLL_EXPORTS;%(PreprocessorDefinitions)</PreprocessorDefinitions>
-      <MinimalRebuild>true</MinimalRebuild>
+      <MinimalRebuild>
+      </MinimalRebuild>
       <BasicRuntimeChecks>EnableFastChecks</BasicRuntimeChecks>
       <RuntimeLibrary>MultiThreadedDebug</RuntimeLibrary>
       <PrecompiledHeader />
@@ -144,6 +147,7 @@ copy $(OutDir)$(TargetName).lib ..\..\lib
       <AssemblerListingLocation>$(OutDir)</AssemblerListingLocation>
       <ObjectFileName>$(OutDir)</ObjectFileName>
       <ProgramDataBaseFileName>$(OutDir)</ProgramDataBaseFileName>
+      <FloatingPointModel>Fast</FloatingPointModel>
     </ClCompile>
     <Link>
       <OutputFile>$(OutDir)$(TargetName)$(TargetExt)</OutputFile>
@@ -182,6 +186,8 @@ copy $(OutDir)$(TargetName).lib ..\..\lib
       <ObjectFileName>$(OutDir)</ObjectFileName>
       <ProgramDataBaseFileName>$(OutDir)</ProgramDataBaseFileName>
       <EnableEnhancedInstructionSet>StreamingSIMDExtensions2</EnableEnhancedInstructionSet>
+      <MinimalRebuild />
+      <FloatingPointModel>Fast</FloatingPointModel>
     </ClCompile>
     <Link>
       <OutputFile>$(OutDir)$(TargetName)$(TargetExt)</OutputFile>
@@ -223,6 +229,8 @@ copy $(OutDir)$(TargetName).lib ..\..\lib
       <AssemblerListingLocation>$(OutDir)</AssemblerListingLocation>
       <ObjectFileName>$(OutDir)</ObjectFileName>
       <ProgramDataBaseFileName>$(OutDir)</ProgramDataBaseFileName>
+      <MinimalRebuild />
+      <FloatingPointModel>Fast</FloatingPointModel>
     </ClCompile>
     <Link>
       <OutputFile>$(OutDir)$(TargetName)$(TargetExt)</OutputFile>

source/SoundTouchDLL/make-gnu-dll.sh

@@ -1,6 +1,9 @@
 #!/bin/bash
 #
-# This script compiles SoundTouch dynamic-link library for GNU environment 
+# This script is deprecated. Don't use this, the makefile can now compile
+# the dynamic-link library 'libSoundTouchDLL.so' automatically.
+#
+# This script compiles SoundTouch dynamic-link library for GNU environment
 # with wrapper functions that are easier to import to Java / Mono / etc
 #
 
@@ -11,12 +14,16 @@ if [[ $arch == *"86"* ]]; then
     # Intel x86/x64 architecture
     flags="$flags -mstackrealign -msse"
 
-    if [[ $arch == *"_64" ]]; then	
+    if [[ $arch == *"_64" ]]; then
         flags="$flags -fPIC"
     fi
 fi
 
+echo "*************************************************************************"
+echo "NOTE: Rather use the makefile that can now build the dynamic-link library"
+echo "*************************************************************************"
+echo ""
 echo "Building SoundTouchDLL for $arch with flags:$flags"
 
-g++ -O3 -shared $flags -DDLL_EXPORTS -fvisibility=hidden -I../../include \
+g++ -O3 -ffast-math -shared $flags -DDLL_EXPORTS -fvisibility=hidden -I../../include \
     -I../SoundTouch -o SoundTouchDll.so SoundTouchDLL.cpp ../SoundTouch/*.cpp

source/SoundTouchDLL/resource.h

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

source/csharp-example/App.config

@@ -1,6 +1,6 @@
-<?xml version="1.0" encoding="utf-8" ?>
+<?xml version="1.0" encoding="utf-8"?>
 <configuration>
     <startup> 
-        <supportedRuntime version="v4.0" sku=".NETFramework,Version=v4.5.2" />
+        <supportedRuntime version="v4.0" sku=".NETFramework,Version=v4.8.1"/>
     </startup>
-</configuration>
+</configuration>

source/csharp-example/Properties/AssemblyInfo.cs

@@ -4,7 +4,7 @@ using System.Runtime.CompilerServices;
 using System.Runtime.InteropServices;
 using System.Windows;
 
-// General Information about an assembly is controlled through the following 
+// General Information about an assembly is controlled through the following
 // set of attributes. Change these attribute values to modify the information
 // associated with an assembly.
 [assembly: AssemblyTitle("csharp-example")]
@@ -12,16 +12,16 @@ using System.Windows;
 [assembly: AssemblyConfiguration("")]
 [assembly: AssemblyCompany("")]
 [assembly: AssemblyProduct("csharp-example")]
-[assembly: AssemblyCopyright("Copyright Olli Parviainen ©  2017")]
+[assembly: AssemblyCopyright("Copyright © Olli Parviainen")]
 [assembly: AssemblyTrademark("")]
 [assembly: AssemblyCulture("")]
 
-// Setting ComVisible to false makes the types in this assembly not visible 
-// to COM components.  If you need to access a type in this assembly from 
+// Setting ComVisible to false makes the types in this assembly not visible
+// to COM components.  If you need to access a type in this assembly from
 // COM, set the ComVisible attribute to true on that type.
 [assembly: ComVisible(false)]
 
-//In order to begin building localizable applications, set 
+//In order to begin building localizable applications, set
 //<UICulture>CultureYouAreCodingWith</UICulture> in your .csproj file
 //inside a <PropertyGroup>.  For example, if you are using US english
 //in your source files, set the <UICulture> to en-US.  Then uncomment
@@ -33,10 +33,10 @@ using System.Windows;
 
 [assembly: ThemeInfo(
     ResourceDictionaryLocation.None, //where theme specific resource dictionaries are located
-                                     //(used if a resource is not found in the page, 
+                                     //(used if a resource is not found in the page,
                                      // or application resource dictionaries)
     ResourceDictionaryLocation.SourceAssembly //where the generic resource dictionary is located
-                                              //(used if a resource is not found in the page, 
+                                              //(used if a resource is not found in the page,
                                               // app, or any theme specific resource dictionaries)
 )]
 
@@ -44,11 +44,11 @@ using System.Windows;
 // Version information for an assembly consists of the following four values:
 //
 //      Major Version
-//      Minor Version 
+//      Minor Version
 //      Build Number
 //      Revision
 //
-// You can specify all the values or you can default the Build and Revision Numbers 
+// You can specify all the values or you can default the Build and Revision Numbers
 // by using the '*' as shown below:
 // [assembly: AssemblyVersion("1.0.*")]
 [assembly: AssemblyVersion("1.0.0.0")]

source/csharp-example/Properties/Resources.Designer.cs

@@ -8,10 +8,10 @@
 // </auto-generated>
 //------------------------------------------------------------------------------
 
-namespace csharp_example.Properties
-{
-
-
+namespace csharp_example.Properties {
+    using System;
+    
+    
     /// <summary>
     ///   A strongly-typed resource class, for looking up localized strings, etc.
     /// </summary>
@@ -19,51 +19,43 @@ namespace csharp_example.Properties
     // class via a tool like ResGen or Visual Studio.
     // To add or remove a member, edit your .ResX file then rerun ResGen
     // with the /str option, or rebuild your VS project.
-    [global::System.CodeDom.Compiler.GeneratedCodeAttribute("System.Resources.Tools.StronglyTypedResourceBuilder", "4.0.0.0")]
+    [global::System.CodeDom.Compiler.GeneratedCodeAttribute("System.Resources.Tools.StronglyTypedResourceBuilder", "16.0.0.0")]
     [global::System.Diagnostics.DebuggerNonUserCodeAttribute()]
     [global::System.Runtime.CompilerServices.CompilerGeneratedAttribute()]
-    internal class Resources
-    {
-
+    internal class Resources {
+        
         private static global::System.Resources.ResourceManager resourceMan;
-
+        
         private static global::System.Globalization.CultureInfo resourceCulture;
-
+        
         [global::System.Diagnostics.CodeAnalysis.SuppressMessageAttribute("Microsoft.Performance", "CA1811:AvoidUncalledPrivateCode")]
-        internal Resources()
-        {
+        internal Resources() {
         }
-
+        
         /// <summary>
         ///   Returns the cached ResourceManager instance used by this class.
         /// </summary>
         [global::System.ComponentModel.EditorBrowsableAttribute(global::System.ComponentModel.EditorBrowsableState.Advanced)]
-        internal static global::System.Resources.ResourceManager ResourceManager
-        {
-            get
-            {
-                if ((resourceMan == null))
-                {
+        internal static global::System.Resources.ResourceManager ResourceManager {
+            get {
+                if (object.ReferenceEquals(resourceMan, null)) {
                     global::System.Resources.ResourceManager temp = new global::System.Resources.ResourceManager("csharp_example.Properties.Resources", typeof(Resources).Assembly);
                     resourceMan = temp;
                 }
                 return resourceMan;
             }
         }
-
+        
         /// <summary>
         ///   Overrides the current thread's CurrentUICulture property for all
         ///   resource lookups using this strongly typed resource class.
         /// </summary>
         [global::System.ComponentModel.EditorBrowsableAttribute(global::System.ComponentModel.EditorBrowsableState.Advanced)]
-        internal static global::System.Globalization.CultureInfo Culture
-        {
-            get
-            {
+        internal static global::System.Globalization.CultureInfo Culture {
+            get {
                 return resourceCulture;
             }
-            set
-            {
+            set {
                 resourceCulture = value;
             }
         }

source/csharp-example/Properties/Settings.Designer.cs

@@ -8,21 +8,17 @@
 // </auto-generated>
 //------------------------------------------------------------------------------
 
-namespace csharp_example.Properties
-{
-
-
+namespace csharp_example.Properties {
+    
+    
     [global::System.Runtime.CompilerServices.CompilerGeneratedAttribute()]
-    [global::System.CodeDom.Compiler.GeneratedCodeAttribute("Microsoft.VisualStudio.Editors.SettingsDesigner.SettingsSingleFileGenerator", "11.0.0.0")]
-    internal sealed partial class Settings : global::System.Configuration.ApplicationSettingsBase
-    {
-
+    [global::System.CodeDom.Compiler.GeneratedCodeAttribute("Microsoft.VisualStudio.Editors.SettingsDesigner.SettingsSingleFileGenerator", "16.10.0.0")]
+    internal sealed partial class Settings : global::System.Configuration.ApplicationSettingsBase {
+        
         private static Settings defaultInstance = ((Settings)(global::System.Configuration.ApplicationSettingsBase.Synchronized(new Settings())));
-
-        public static Settings Default
-        {
-            get
-            {
+        
+        public static Settings Default {
+            get {
                 return defaultInstance;
             }
         }

source/csharp-example/csharp-example.csproj

@@ -9,14 +9,15 @@
     <AppDesignerFolder>Properties</AppDesignerFolder>
     <RootNamespace>csharp_example</RootNamespace>
     <AssemblyName>csharp-example</AssemblyName>
-    <TargetFrameworkVersion>v4.5.2</TargetFrameworkVersion>
+    <TargetFrameworkVersion>v4.8.1</TargetFrameworkVersion>
     <FileAlignment>512</FileAlignment>
     <ProjectTypeGuids>{60dc8134-eba5-43b8-bcc9-bb4bc16c2548};{FAE04EC0-301F-11D3-BF4B-00C04F79EFBC}</ProjectTypeGuids>
     <WarningLevel>4</WarningLevel>
     <AutoGenerateBindingRedirects>true</AutoGenerateBindingRedirects>
+    <TargetFrameworkProfile />
   </PropertyGroup>
   <PropertyGroup Condition=" '$(Configuration)|$(Platform)' == 'Debug|AnyCPU' ">
-    <PlatformTarget>AnyCPU</PlatformTarget>
+    <PlatformTarget>x64</PlatformTarget>
     <DebugSymbols>true</DebugSymbols>
     <DebugType>full</DebugType>
     <Optimize>false</Optimize>
@@ -24,6 +25,7 @@
     <DefineConstants>DEBUG;TRACE</DefineConstants>
     <ErrorReport>prompt</ErrorReport>
     <WarningLevel>4</WarningLevel>
+    <Prefer32Bit>false</Prefer32Bit>
   </PropertyGroup>
   <PropertyGroup Condition=" '$(Configuration)|$(Platform)' == 'Release|AnyCPU' ">
     <PlatformTarget>AnyCPU</PlatformTarget>