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CGH0S7
2025-02-06 22:24:29 +08:00
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9
externals/soundtouch/source/Android-lib/.classpath vendored Normal file
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<?xml version="1.0" encoding="UTF-8"?>
<classpath>
<classpathentry kind="con" path="com.android.ide.eclipse.adt.ANDROID_FRAMEWORK"/>
<classpathentry exported="true" kind="con" path="com.android.ide.eclipse.adt.LIBRARIES"/>
<classpathentry exported="true" kind="con" path="com.android.ide.eclipse.adt.DEPENDENCIES"/>
<classpathentry kind="src" path="src"/>
<classpathentry kind="src" path="gen"/>
<classpathentry kind="output" path="bin/classes"/>
</classpath>

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externals/soundtouch/source/Android-lib/.project vendored Normal file
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<?xml version="1.0" encoding="UTF-8"?>
<projectDescription>
<name>ExampleActivity</name>
<comment></comment>
<projects>
</projects>
<buildSpec>
<buildCommand>
<name>com.android.ide.eclipse.adt.ResourceManagerBuilder</name>
<arguments>
</arguments>
</buildCommand>
<buildCommand>
<name>com.android.ide.eclipse.adt.PreCompilerBuilder</name>
<arguments>
</arguments>
</buildCommand>
<buildCommand>
<name>org.eclipse.jdt.core.javabuilder</name>
<arguments>
</arguments>
</buildCommand>
<buildCommand>
<name>com.android.ide.eclipse.adt.ApkBuilder</name>
<arguments>
</arguments>
</buildCommand>
</buildSpec>
<natures>
<nature>com.android.ide.eclipse.adt.AndroidNature</nature>
<nature>org.eclipse.jdt.core.javanature</nature>
</natures>
</projectDescription>

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

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externals/soundtouch/source/Android-lib/README-SoundTouch-Android.html vendored Normal file
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
<title>SoundTouch in Android</title>
<meta http-equiv="Content-Type"
content="text/html; charset=windows-1252">
<meta http-equiv="Content-Language" content="en-us">
<meta name="author" content="Olli Parviainen">
<meta name="description"
content="Readme file for SoundTouch library Android compilation">
<style> <!-- .normal { font-family: Arial }
--></style>
</head>
<body class="normal">
<hr>
<h1>SoundTouch in Android</h1>
<hr>
<h2>Compiling SoundTouch for Android</h2>
<p>SoundTouch source code package contains &quot;Android-lib&quot; example project that compiles SoundTouch
source codes into Android native library, and gives an example of JNI interface
for invoking
the native SoundTouch routines from an Android application written in Java.</p>
<p style="font-weight: 700">Software prerequisites:</p>
<ul>
<li>Android SDK environment for developing your own Android application. Visit the <a href="http://developer.android.com/index.html">Android developers' site</a>
for more information about the Android SDK and developing Android applications.</li>
<li>Android NDK compiler kit for compiling native library binaries. The Android NDK
is <a href="http://developer.android.com/tools/sdk/ndk/index.html">
available for download</a> at the Android developer tools site.</li>
<li>In case you're working in Windows environment, install
<a href="http://cygwin.com/install.html">
Cygwin</a> to run the Android NDK/SDK compiler scripts</li>
<li>Latest SoundTouch source code package available at <a href="http://soundtouch.surina.net/sourcecode.html">
soundtouch.surina.net</a>.</li>
</ul>
<p><b>Hint: </b>As installing and configuring all the components for an Android SDK/NDK
environment requires fair effort, it&#39;s good idea to create a dedicated Virtual
Machine environment for the Android development environment installation.
Having the Android developer environment setup in dedicated Virtual Machine
allows keeping all these settings isolated from your other PC operations, and
eases taking backup snapshots of your full development environment.</p>
<p><b>Compiling</b></p>
<p>
To compile the SoundTouch library source codes into an Android native library,
open Cygwin/bash shell, go to directory <b>&quot;soundtouch/source/Android-lib/jni&quot;</b> and invoke the NDK
compiler with following command:</p>
<pre> $NDK/ndk-build</pre>
<p>This will build binaries for all the supported Android platforms (arm-v5, arm-v7, X86, MIPS etc) of SoundTouch library, plus the JNI wrapper interface as discussed below. The target binaries will be built into the &quot;libs&quot; subdirectory. As long as all these .so binary library versions are included in the APK Application delivery package, the target Android device can choose the correct library version to use. </p>
<p>Notice that to allow Cygwin/bash to locate the NDK compile scripts, you
need to define the location of the NDK installation defined in environment
variable &quot;NDK&quot;. That's easiest done by adding the NDK path definition at end of
your <b>~/.bash_profile</b> file, for instance as follows:</p>
<pre> NDK=/cygdrive/d/Android/android-ndk-r6</pre>
<p><b>Enabling OpenMP parallel computing mode</b></p>
<p>
SoundTouch supports OpenMP for parallel computing in multi-core
environments, and these improvements can be enabled also in the Android
build. See the SoundTouch main README.html file for generic notes about the
OpenMP implementation.</p>
<p>
To enable OpenMP mode in Android compilation, edit file <strong>Android.mk</strong>
and enable the &quot;-fopenmp&quot; flag in LOCAL_CFLAGS and LOCAL_LDFLAGS variables.
This is done by removing hash # from before the following lines in the
Android.mk file, before compiling the library:</p>
<pre> LOCAL_CFLAGS += -fopenmp
LOCAL_LDFLAGS += -fopenmp</pre>
<p><strong>OpenMP COMPATIBILITY NOTE: </strong>Android NDK has a threading issue
(at least until NDK v10) that causes the native library crash with fatal signal
11 if calling OpenMP-improved routines from a background thread. SoundTouch has
a workaround for this issue in soundtouch-jni.cpp, and this workaround requires
calling function <strong>SoundTouch.getVersionString() </strong>from the Android
application's main thread at least once before calling other SoundTouch
processing routines. See the SoundTouch Android example application and comments
in the <strong>soundtouch-jni.cpp </strong>source code file for more details.</p>
<p>
<strong>SoundTouch performance in Android</strong></p>
<p>
See external blog articles for more discussion about the
<a href="http://www.softwarecoven.com/parallel-computing-in-embedded-mobile-devices/">
SoundTouch OpenMP implementation</a> and the
<a href="http://www.softwarecoven.com/parallel-computing-with-openmp-in-android/">
SoundTouch performance benchmark tests in Android environment</a>.</p>
<hr />
<h2>
Calling SoundTouch native routines from Android application</h2>
<p>The NDK tools build the SoundTouch c++ routines into a native binary library, while
Android applications are written in Java language. To call the SoundTouch and other c/c++
routines from an Android java application code, you'll need to use Java Native
Interface (JNI).</p>
<p>
The SoundTouch source code package provides source code example how to
use JNI to call native c++ routines from a Java class, and provides source codes also for
a simple example Android application:<ul>
<li><b>ExampleActivity</b>: This is simple Android example application that
utilizes SoundTouch native routines for processing WAV audio files. To build the example
application, use Eclipse Android SDK environment to import the "ExampleActivity" project in the "Android-lib" folder into the Eclipse workspace.
<li><b>Android-lib/jni/soundtouch-jni.cpp</b>: This file contains c/c++ wrapper routines
for performing elementary audio file processing with adjusted tempo/pitch/speed parameters
from the Android application. The wrapper interface is not complete, but provides example
that is easy to extend when necessary. The NDK compiles this file along with the SoundTouch
routines into the native binary library.</li>
<li><b>Android-lib/src/net/surina/soundtouch/SoundTouch.java</b>: This file implements
the Java interface class that loasd & accesses the JNI routines in the natively compiled library.
The example Android application uses this class as interface for processing audio files
with SoundTouch.</li>
<li><b>Android-lib/build.gradle</b>: Top level build script file for Android Studio 3.1.4+</li>
</ul>
<p>
Feel free to examine and extend the provided cpp/java source code example file pair to
implement and integrate the desired SoundTouch library capabilities into your own Android application.</p>
<hr />
<h2>
Android floating-point performance considerations</h2>
<p>
The make process will build dedicated binaries for each supported Android CPU hardware platform type.
</p><p>SoundTouch uses floating-point algorithms for ideal sound quality on all other platform than in the lowest-end ARMv5. That is because lowest-end Android devices are not guaranteed to
have floating-point hardware in their CPUs, so that the ARMv5 compilation uses by default software-emulation for floating-point calculations to allow running the binary executables also in low-end devices without floating-point hardware.<p>
As floating point software-emulation is however several tens of times slower
than real hardware-level floating-point calculations, that would make running
floating-point-intensive applications such as SoundTouch infeasible in these low-end
devices. As workaround, the SoundTouch Android compilation builds the ARMv5 version using integer algorithm versions. The integer
algorithm version compromises the sound quality but provides good performance also
with low-end devices without hardware floating-point support in the CPU level.</p>
<p>When Android devices with more capable device is used, the device will automatically choose a proper library version for ideal sound quality.</p>
<hr />
<p style="text-align: center"><i>Copyright &copy; Olli Parviainen</i></p>
</body>
</html>

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externals/soundtouch/source/Android-lib/gradle/wrapper/gradle-wrapper.properties vendored Normal file
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#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

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externals/soundtouch/source/Android-lib/gradlew vendored Normal file
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#!/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" "$@"

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externals/soundtouch/source/Android-lib/gradlew.bat vendored Normal file
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@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

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externals/soundtouch/source/Android-lib/jni/Android.mk vendored Normal file
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LOCAL_PATH := $(call my-dir)
include $(CLEAR_VARS)
LOCAL_C_INCLUDES += $(LOCAL_PATH)/../../../include $(LOCAL_PATH)/../../SoundStretch
# *** Remember: Change -O0 into -O2 in add-applications.mk ***
LOCAL_MODULE := soundtouch
LOCAL_SRC_FILES := soundtouch-jni.cpp ../../SoundTouch/AAFilter.cpp ../../SoundTouch/FIFOSampleBuffer.cpp \
../../SoundTouch/FIRFilter.cpp ../../SoundTouch/cpu_detect_x86.cpp \
../../SoundTouch/sse_optimized.cpp ../../SoundStretch/WavFile.cpp \
../../SoundTouch/RateTransposer.cpp ../../SoundTouch/SoundTouch.cpp \
../../SoundTouch/InterpolateCubic.cpp ../../SoundTouch/InterpolateLinear.cpp \
../../SoundTouch/InterpolateShannon.cpp ../../SoundTouch/TDStretch.cpp \
../../SoundTouch/BPMDetect.cpp ../../SoundTouch/PeakFinder.cpp
# for native audio
LOCAL_SHARED_LIBRARIES += -lgcc
# --whole-archive -lgcc
# for logging
LOCAL_LDLIBS += -llog
# for native asset manager
#LOCAL_LDLIBS += -landroid
# Custom Flags:
# -fvisibility=hidden : don't export all symbols
LOCAL_CFLAGS += -fvisibility=hidden -fdata-sections -ffunction-sections -ffast-math
# OpenMP mode : enable these flags to enable using OpenMP for parallel computation
#LOCAL_CFLAGS += -fopenmp
#LOCAL_LDFLAGS += -fopenmp
# Use ARM instruction set instead of Thumb for improved calculation performance in ARM CPUs
LOCAL_ARM_MODE := arm
include $(BUILD_SHARED_LIBRARY)

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

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externals/soundtouch/source/Android-lib/jni/soundtouch-jni.cpp vendored Normal file
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////////////////////////////////////////////////////////////////////////////////
///
/// 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;
}

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<?xml version="1.0" encoding="UTF-8"?>
<lint>
</lint>

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

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

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

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<?xml version="1.0" encoding="utf-8"?>
<resources>
<string name="app_name">SoundTouch Example</string>
<string name="hello_world">Hello world!</string>
</resources>

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

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

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////////////////////////////////////////////////////////////////////////////////
///
/// Example class that invokes native SoundTouch routines through the JNI
/// interface.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// WWW : http://www.surina.net
///
////////////////////////////////////////////////////////////////////////////////
package net.surina.soundtouch;
public final class SoundTouch
{
// Native interface function that returns SoundTouch version string.
// This invokes the native c++ routine defined in "soundtouch-jni.cpp".
public native final static String getVersionString();
private native final void setTempo(long handle, float tempo);
private native final void setPitchSemiTones(long handle, float pitch);
private native final void setSpeed(long handle, float speed);
private native final int processFile(long handle, String inputFile, String outputFile);
public native final static String getErrorString();
private native final static long newInstance();
private native final void deleteInstance(long handle);
long handle = 0;
public SoundTouch()
{
handle = newInstance();
}
public void close()
{
deleteInstance(handle);
handle = 0;
}
public void setTempo(float tempo)
{
setTempo(handle, tempo);
}
public void setPitchSemiTones(float pitch)
{
setPitchSemiTones(handle, pitch);
}
public void setSpeed(float speed)
{
setSpeed(handle, speed);
}
public int processFile(String inputFile, String outputFile)
{
return processFile(handle, inputFile, outputFile);
}
// Load the native library upon startup
static
{
System.loadLibrary("soundtouch");
}
}

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## 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 SoundTouchDLL
# set to something if you want other stuff to be included in the distribution tarball
#EXTRA_DIST=

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## Process this file with automake to create Makefile.in
##
## 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

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////////////////////////////////////////////////////////////////////////////////
///
/// A class for parsing the 'soundstretch' application command line parameters
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// 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 = nullptr;
outFileName = nullptr;
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 = nullptr;
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);
}
}

65
externals/soundtouch/source/SoundStretch/RunParameters.h vendored Normal file
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@@ -0,0 +1,65 @@
////////////////////////////////////////////////////////////////////////////////
///
/// 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

986
externals/soundtouch/source/SoundStretch/WavFile.cpp vendored Normal file
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@@ -0,0 +1,986 @@
////////////////////////////////////////////////////////////////////////////////
///
/// 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
//
////////////////////////////////////////////////////////////////////////////////
#include <stdio.h>
#include <string>
#include <sstream>
#include <cstring>
#include <assert.h>
#include <limits.h>
#include "WavFile.h"
#include "STTypes.h"
using namespace std;
static const char riffStr[] = "RIFF";
static const char waveStr[] = "WAVE";
static const char fmtStr[] = "fmt ";
static const char factStr[] = "fact";
static const char dataStr[] = "data";
//////////////////////////////////////////////////////////////////////////////
//
// Helper functions for swapping byte order to correctly read/write WAV files
// with big-endian CPU's: Define compile-time definition _BIG_ENDIAN_ to
// turn-on the conversion if it appears necessary.
//
// For example, Intel x86 is little-endian and doesn't require conversion,
// while PowerPC of Mac's and many other RISC cpu's are big-endian.
#ifdef BYTE_ORDER
// In gcc compiler detect the byte order automatically
#if BYTE_ORDER == BIG_ENDIAN
// big-endian platform.
#define _BIG_ENDIAN_
#endif
#endif
#ifdef _BIG_ENDIAN_
// big-endian CPU, swap bytes in 16 & 32 bit words
// helper-function to swap byte-order of 32bit integer
static inline int _swap32(int &dwData)
{
dwData = ((dwData >> 24) & 0x000000FF) |
((dwData >> 8) & 0x0000FF00) |
((dwData << 8) & 0x00FF0000) |
((dwData << 24) & 0xFF000000);
return dwData;
}
// helper-function to swap byte-order of 16bit integer
static inline short _swap16(short &wData)
{
wData = ((wData >> 8) & 0x00FF) |
((wData << 8) & 0xFF00);
return wData;
}
// helper-function to swap byte-order of buffer of 16bit integers
static inline void _swap16Buffer(short *pData, int numWords)
{
int i;
for (i = 0; i < numWords; i ++)
{
pData[i] = _swap16(pData[i]);
}
}
#else // BIG_ENDIAN
// little-endian CPU, WAV file is ok as such
// dummy helper-function
static inline int _swap32(int &dwData)
{
// do nothing
return dwData;
}
// dummy helper-function
static inline short _swap16(short &wData)
{
// do nothing
return wData;
}
// dummy helper-function
static inline void _swap16Buffer(short *, int)
{
// do nothing
}
#endif // BIG_ENDIAN
//////////////////////////////////////////////////////////////////////////////
//
// Class WavFileBase
//
WavFileBase::WavFileBase()
{
convBuff = nullptr;
convBuffSize = 0;
}
WavFileBase::~WavFileBase()
{
delete[] convBuff;
convBuffSize = 0;
}
/// Get pointer to conversion buffer of at min. given size
void *WavFileBase::getConvBuffer(int sizeBytes)
{
if (convBuffSize < sizeBytes)
{
delete[] convBuff;
convBuffSize = (sizeBytes + 15) & -8; // round up to following 8-byte bounday
convBuff = new char[convBuffSize];
}
return convBuff;
}
//////////////////////////////////////////////////////////////////////////////
//
// Class WavInFile
//
WavInFile::WavInFile(const char *fileName)
{
// Try to open the file for reading
fptr = fopen(fileName, "rb");
if (fptr == nullptr)
{
// didn't succeed
string msg = "Error : Unable to open file \"";
msg += fileName;
msg += "\" for reading.";
ST_THROW_RT_ERROR(msg.c_str());
}
init();
}
WavInFile::WavInFile(FILE *file)
{
// Try to open the file for reading
fptr = file;
if (!file)
{
// didn't succeed
string msg = "Error : Unable to access input stream for reading";
ST_THROW_RT_ERROR(msg.c_str());
}
init();
}
/// Init the WAV file stream
void WavInFile::init()
{
int hdrsOk;
// assume file stream is already open
assert(fptr);
// Read the file headers
hdrsOk = readWavHeaders();
if (hdrsOk != 0)
{
// Something didn't match in the wav file headers
ST_THROW_RT_ERROR("Input file is corrupt or not a WAV file");
}
// sanity check for format parameters
if ((header.format.channel_number < 1) || (header.format.channel_number > 9) ||
(header.format.sample_rate < 4000) || (header.format.sample_rate > 192000) ||
(header.format.byte_per_sample < 1) || (header.format.byte_per_sample > 320) ||
(header.format.bits_per_sample < 8) || (header.format.bits_per_sample > 32))
{
// Something didn't match in the wav file headers
ST_THROW_RT_ERROR("Error: Illegal wav file header format parameters.");
}
dataRead = 0;
}
WavInFile::~WavInFile()
{
if (fptr) fclose(fptr);
fptr = nullptr;
}
void WavInFile::rewind()
{
int hdrsOk;
fseek(fptr, 0, SEEK_SET);
hdrsOk = readWavHeaders();
assert(hdrsOk == 0);
dataRead = 0;
}
int WavInFile::checkCharTags() const
{
// header.format.fmt should equal to 'fmt '
if (memcmp(fmtStr, header.format.fmt, 4) != 0) return -1;
// header.data.data_field should equal to 'data'
if (memcmp(dataStr, header.data.data_field, 4) != 0) return -1;
return 0;
}
int WavInFile::read(unsigned char *buffer, int maxElems)
{
int numBytes;
uint afterDataRead;
// ensure it's 8 bit format
if (header.format.bits_per_sample != 8)
{
ST_THROW_RT_ERROR("Error: WavInFile::read(char*, int) works only with 8bit samples.");
}
assert(sizeof(char) == 1);
numBytes = maxElems;
afterDataRead = dataRead + numBytes;
if (afterDataRead > header.data.data_len)
{
// Don't read more samples than are marked available in header
numBytes = (int)header.data.data_len - (int)dataRead;
assert(numBytes >= 0);
}
assert(buffer);
numBytes = (int)fread(buffer, 1, numBytes, fptr);
dataRead += numBytes;
return numBytes;
}
int WavInFile::read(short *buffer, int maxElems)
{
unsigned int afterDataRead;
int numBytes;
int numElems;
assert(buffer);
switch (header.format.bits_per_sample)
{
case 8:
{
// 8 bit format
unsigned char *temp = (unsigned char*)getConvBuffer(maxElems);
int i;
numElems = read(temp, maxElems);
// convert from 8 to 16 bit
for (i = 0; i < numElems; i ++)
{
buffer[i] = (short)(((short)temp[i] - 128) * 256);
}
break;
}
case 16:
{
// 16 bit format
assert(sizeof(short) == 2);
numBytes = maxElems * 2;
afterDataRead = dataRead + numBytes;
if (afterDataRead > header.data.data_len)
{
// Don't read more samples than are marked available in header
numBytes = (int)header.data.data_len - (int)dataRead;
assert(numBytes >= 0);
}
numBytes = (int)fread(buffer, 1, numBytes, fptr);
dataRead += numBytes;
numElems = numBytes / 2;
// 16bit samples, swap byte order if necessary
_swap16Buffer((short *)buffer, numElems);
break;
}
default:
{
stringstream ss;
ss << "\nOnly 8/16 bit sample WAV files supported in integer compilation. Can't open WAV file with ";
ss << (int)header.format.bits_per_sample;
ss << " bit sample format. ";
ST_THROW_RT_ERROR(ss.str().c_str());
}
};
return numElems;
}
/// Read data in float format. Notice that when reading in float format
/// 8/16/24/32 bit sample formats are supported
int WavInFile::read(float *buffer, int maxElems)
{
unsigned int afterDataRead;
int numBytes;
int numElems;
int bytesPerSample;
assert(buffer);
bytesPerSample = header.format.bits_per_sample / 8;
if ((bytesPerSample < 1) || (bytesPerSample > 4))
{
stringstream ss;
ss << "\nOnly 8/16/24/32 bit sample WAV files supported. Can't open WAV file with ";
ss << (int)header.format.bits_per_sample;
ss << " bit sample format. ";
ST_THROW_RT_ERROR(ss.str().c_str());
}
numBytes = maxElems * bytesPerSample;
afterDataRead = dataRead + numBytes;
if (afterDataRead > header.data.data_len)
{
// Don't read more samples than are marked available in header
numBytes = (int)header.data.data_len - (int)dataRead;
assert(numBytes >= 0);
}
// read raw data into temporary buffer
char *temp = (char*)getConvBuffer(numBytes);
numBytes = (int)fread(temp, 1, numBytes, fptr);
dataRead += numBytes;
numElems = numBytes / bytesPerSample;
// swap byte ordert & convert to float, depending on sample format
switch (bytesPerSample)
{
case 1:
{
unsigned char *temp2 = (unsigned char*)temp;
double conv = 1.0 / 128.0;
for (int i = 0; i < numElems; i ++)
{
buffer[i] = (float)(temp2[i] * conv - 1.0);
}
break;
}
case 2:
{
short *temp2 = (short*)temp;
double conv = 1.0 / 32768.0;
for (int i = 0; i < numElems; i ++)
{
short value = temp2[i];
buffer[i] = (float)(_swap16(value) * conv);
}
break;
}
case 3:
{
char *temp2 = (char *)temp;
double conv = 1.0 / 8388608.0;
for (int i = 0; i < numElems; i ++)
{
int value = *((int*)temp2);
value = _swap32(value) & 0x00ffffff; // take 24 bits
value |= (value & 0x00800000) ? 0xff000000 : 0; // extend minus sign bits
buffer[i] = (float)(value * conv);
temp2 += 3;
}
break;
}
case 4:
{
int *temp2 = (int *)temp;
double conv = 1.0 / 2147483648.0;
assert(sizeof(int) == 4);
for (int i = 0; i < numElems; i ++)
{
int value = temp2[i];
buffer[i] = (float)(_swap32(value) * conv);
}
break;
}
}
return numElems;
}
int WavInFile::eof() const
{
// return true if all data has been read or file eof has reached
return ((uint)dataRead == header.data.data_len || feof(fptr));
}
// test if character code is between a white space ' ' and little 'z'
static int isAlpha(char c)
{
return (c >= ' ' && c <= 'z') ? 1 : 0;
}
// test if all characters are between a white space ' ' and little 'z'
static int isAlphaStr(const char *str)
{
char c;
c = str[0];
while (c)
{
if (isAlpha(c) == 0) return 0;
str ++;
c = str[0];
}
return 1;
}
int WavInFile::readRIFFBlock()
{
if (fread(&(header.riff), sizeof(WavRiff), 1, fptr) != 1) return -1;
// swap 32bit data byte order if necessary
_swap32((int &)header.riff.package_len);
// header.riff.riff_char should equal to 'RIFF');
if (memcmp(riffStr, header.riff.riff_char, 4) != 0) return -1;
// header.riff.wave should equal to 'WAVE'
if (memcmp(waveStr, header.riff.wave, 4) != 0) return -1;
return 0;
}
int WavInFile::readHeaderBlock()
{
char label[5];
string sLabel;
// lead label string
if (fread(label, 1, 4, fptr) !=4) return -1;
label[4] = 0;
if (isAlphaStr(label) == 0) return -1; // not a valid label
// Decode blocks according to their label
if (strcmp(label, fmtStr) == 0)
{
int nLen, nDump;
// 'fmt ' block
memcpy(header.format.fmt, fmtStr, 4);
// read length of the format field
if (fread(&nLen, sizeof(int), 1, fptr) != 1) return -1;
// swap byte order if necessary
_swap32(nLen);
// calculate how much length differs from expected
nDump = nLen - ((int)sizeof(header.format) - 8);
// verify that header length isn't smaller than expected structure
if ((nLen < 0) || (nDump < 0)) return -1;
header.format.format_len = nLen;
// if format_len is larger than expected, read only as much data as we've space for
if (nDump > 0)
{
nLen = sizeof(header.format) - 8;
}
// read data
if (fread(&(header.format.fixed), nLen, 1, fptr) != 1) return -1;
// swap byte order if necessary
_swap16((short &)header.format.fixed); // short int fixed;
_swap16((short &)header.format.channel_number); // short int channel_number;
_swap32((int &)header.format.sample_rate); // int sample_rate;
_swap32((int &)header.format.byte_rate); // int byte_rate;
_swap16((short &)header.format.byte_per_sample); // short int byte_per_sample;
_swap16((short &)header.format.bits_per_sample); // short int bits_per_sample;
// if format_len is larger than expected, skip the extra data
if (nDump > 0)
{
fseek(fptr, nDump, SEEK_CUR);
}
return 0;
}
else if (strcmp(label, factStr) == 0)
{
int nLen, nDump;
// 'fact' block
memcpy(header.fact.fact_field, factStr, 4);
// read length of the fact field
if (fread(&nLen, sizeof(int), 1, fptr) != 1) return -1;
// swap byte order if necessary
_swap32(nLen);
// calculate how much length differs from expected
nDump = nLen - ((int)sizeof(header.fact) - 8);
// verify that fact length isn't smaller than expected structure
if ((nLen < 0) || (nDump < 0)) return -1;
header.fact.fact_len = nLen;
// if format_len is larger than expected, read only as much data as we've space for
if (nDump > 0)
{
nLen = sizeof(header.fact) - 8;
}
// read data
if (fread(&(header.fact.fact_sample_len), nLen, 1, fptr) != 1) return -1;
// swap byte order if necessary
_swap32((int &)header.fact.fact_sample_len); // int sample_length;
// if fact_len is larger than expected, skip the extra data
if (nDump > 0)
{
fseek(fptr, nDump, SEEK_CUR);
}
return 0;
}
else if (strcmp(label, dataStr) == 0)
{
// 'data' block
memcpy(header.data.data_field, dataStr, 4);
if (fread(&(header.data.data_len), sizeof(uint), 1, fptr) != 1) return -1;
// swap byte order if necessary
_swap32((int &)header.data.data_len);
return 1;
}
else
{
uint len, i;
uint temp;
// unknown block
// read length
if (fread(&len, sizeof(len), 1, fptr) != 1) return -1;
// scan through the block
for (i = 0; i < len; i ++)
{
if (fread(&temp, 1, 1, fptr) != 1) return -1;
if (feof(fptr)) return -1; // unexpected eof
}
}
return 0;
}
int WavInFile::readWavHeaders()
{
int res;
memset(&header, 0, sizeof(header));
res = readRIFFBlock();
if (res) return 1;
// read header blocks until data block is found
do
{
// read header blocks
res = readHeaderBlock();
if (res < 0) return 1; // error in file structure
} while (res == 0);
// check that all required tags are legal
return checkCharTags();
}
uint WavInFile::getNumChannels() const
{
return header.format.channel_number;
}
uint WavInFile::getNumBits() const
{
return header.format.bits_per_sample;
}
uint WavInFile::getBytesPerSample() const
{
return getNumChannels() * getNumBits() / 8;
}
uint WavInFile::getSampleRate() const
{
return header.format.sample_rate;
}
uint WavInFile::getDataSizeInBytes() const
{
return header.data.data_len;
}
uint WavInFile::getNumSamples() const
{
if (header.format.byte_per_sample == 0) return 0;
if (header.format.fixed > 1) return header.fact.fact_sample_len;
return header.data.data_len / (unsigned short)header.format.byte_per_sample;
}
uint WavInFile::getLengthMS() const
{
double numSamples;
double sampleRate;
numSamples = (double)getNumSamples();
sampleRate = (double)getSampleRate();
return (uint)(1000.0 * numSamples / sampleRate + 0.5);
}
/// Returns how many milliseconds of audio have so far been read from the file
uint WavInFile::getElapsedMS() const
{
return (uint)(1000.0 * (double)dataRead / (double)header.format.byte_rate);
}
//////////////////////////////////////////////////////////////////////////////
//
// Class WavOutFile
//
WavOutFile::WavOutFile(const char *fileName, int sampleRate, int bits, int channels)
{
bytesWritten = 0;
fptr = fopen(fileName, "wb");
if (fptr == nullptr)
{
string msg = "Error : Unable to open file \"";
msg += fileName;
msg += "\" for writing.";
//pmsg = msg.c_str;
ST_THROW_RT_ERROR(msg.c_str());
}
fillInHeader(sampleRate, bits, channels);
writeHeader();
}
WavOutFile::WavOutFile(FILE *file, int sampleRate, int bits, int channels)
{
bytesWritten = 0;
fptr = file;
if (fptr == nullptr)
{
string msg = "Error : Unable to access output file stream.";
ST_THROW_RT_ERROR(msg.c_str());
}
fillInHeader(sampleRate, bits, channels);
writeHeader();
}
WavOutFile::~WavOutFile()
{
finishHeader();
if (fptr) fclose(fptr);
fptr = nullptr;
}
void WavOutFile::fillInHeader(uint sampleRate, uint bits, uint channels)
{
// fill in the 'riff' part..
// copy string 'RIFF' to riff_char
memcpy(&(header.riff.riff_char), riffStr, 4);
// package_len unknown so far
header.riff.package_len = 0;
// copy string 'WAVE' to wave
memcpy(&(header.riff.wave), waveStr, 4);
// fill in the 'format' part..
// copy string 'fmt ' to fmt
memcpy(&(header.format.fmt), fmtStr, 4);
header.format.format_len = 0x10;
header.format.fixed = 1;
header.format.channel_number = (short)channels;
header.format.sample_rate = (int)sampleRate;
header.format.bits_per_sample = (short)bits;
header.format.byte_per_sample = (short)(bits * channels / 8);
header.format.byte_rate = header.format.byte_per_sample * (int)sampleRate;
header.format.sample_rate = (int)sampleRate;
// fill in the 'fact' part...
memcpy(&(header.fact.fact_field), factStr, 4);
header.fact.fact_len = 4;
header.fact.fact_sample_len = 0;
// fill in the 'data' part..
// copy string 'data' to data_field
memcpy(&(header.data.data_field), dataStr, 4);
// data_len unknown so far
header.data.data_len = 0;
}
void WavOutFile::finishHeader()
{
// supplement the file length into the header structure
header.riff.package_len = bytesWritten + sizeof(WavHeader) - sizeof(WavRiff) + 4;
header.data.data_len = bytesWritten;
header.fact.fact_sample_len = bytesWritten / header.format.byte_per_sample;
writeHeader();
}
void WavOutFile::writeHeader()
{
WavHeader hdrTemp;
int res;
// swap byte order if necessary
hdrTemp = header;
_swap32((int &)hdrTemp.riff.package_len);
_swap32((int &)hdrTemp.format.format_len);
_swap16((short &)hdrTemp.format.fixed);
_swap16((short &)hdrTemp.format.channel_number);
_swap32((int &)hdrTemp.format.sample_rate);
_swap32((int &)hdrTemp.format.byte_rate);
_swap16((short &)hdrTemp.format.byte_per_sample);
_swap16((short &)hdrTemp.format.bits_per_sample);
_swap32((int &)hdrTemp.data.data_len);
_swap32((int &)hdrTemp.fact.fact_len);
_swap32((int &)hdrTemp.fact.fact_sample_len);
// write the supplemented header in the beginning of the file
fseek(fptr, 0, SEEK_SET);
res = (int)fwrite(&hdrTemp, sizeof(hdrTemp), 1, fptr);
if (res != 1)
{
ST_THROW_RT_ERROR("Error while writing to a wav file.");
}
// jump back to the end of the file
fseek(fptr, 0, SEEK_END);
}
void WavOutFile::write(const unsigned char *buffer, int numElems)
{
int res;
if (header.format.bits_per_sample != 8)
{
ST_THROW_RT_ERROR("Error: WavOutFile::write(const char*, int) accepts only 8bit samples.");
}
assert(sizeof(char) == 1);
res = (int)fwrite(buffer, 1, numElems, fptr);
if (res != numElems)
{
ST_THROW_RT_ERROR("Error while writing to a wav file.");
}
bytesWritten += numElems;
}
void WavOutFile::write(const short *buffer, int numElems)
{
int res;
// 16 bit samples
if (numElems < 1) return; // nothing to do
switch (header.format.bits_per_sample)
{
case 8:
{
int i;
unsigned char *temp = (unsigned char *)getConvBuffer(numElems);
// convert from 16bit format to 8bit format
for (i = 0; i < numElems; i ++)
{
temp[i] = (unsigned char)(buffer[i] / 256 + 128);
}
// write in 8bit format
write(temp, numElems);
break;
}
case 16:
{
// 16bit format
// use temp buffer to swap byte order if necessary
short *pTemp = (short *)getConvBuffer(numElems * sizeof(short));
memcpy(pTemp, buffer, numElems * 2);
_swap16Buffer(pTemp, numElems);
res = (int)fwrite(pTemp, 2, numElems, fptr);
if (res != numElems)
{
ST_THROW_RT_ERROR("Error while writing to a wav file.");
}
bytesWritten += 2 * numElems;
break;
}
default:
{
stringstream ss;
ss << "\nOnly 8/16 bit sample WAV files supported in integer compilation. Can't open WAV file with ";
ss << (int)header.format.bits_per_sample;
ss << " bit sample format. ";
ST_THROW_RT_ERROR(ss.str().c_str());
}
}
}
/// Convert from float to integer and saturate
inline int saturate(float fvalue, float minval, float maxval)
{
if (fvalue > maxval)
{
fvalue = maxval;
}
else if (fvalue < minval)
{
fvalue = minval;
}
return (int)fvalue;
}
void WavOutFile::write(const float *buffer, int numElems)
{
int numBytes;
int bytesPerSample;
if (numElems == 0) return;
bytesPerSample = header.format.bits_per_sample / 8;
numBytes = numElems * bytesPerSample;
void *temp = getConvBuffer(numBytes + 7); // round bit up to avoid buffer overrun with 24bit-value assignment
switch (bytesPerSample)
{
case 1:
{
unsigned char *temp2 = (unsigned char *)temp;
for (int i = 0; i < numElems; i ++)
{
temp2[i] = (unsigned char)saturate(buffer[i] * 128.0f + 128.0f, 0.0f, 255.0f);
}
break;
}
case 2:
{
short *temp2 = (short *)temp;
for (int i = 0; i < numElems; i ++)
{
short value = (short)saturate(buffer[i] * 32768.0f, -32768.0f, 32767.0f);
temp2[i] = _swap16(value);
}
break;
}
case 3:
{
char *temp2 = (char *)temp;
for (int i = 0; i < numElems; i ++)
{
int value = saturate(buffer[i] * 8388608.0f, -8388608.0f, 8388607.0f);
*((int*)temp2) = _swap32(value);
temp2 += 3;
}
break;
}
case 4:
{
int *temp2 = (int *)temp;
for (int i = 0; i < numElems; i ++)
{
int value = saturate(buffer[i] * 2147483648.0f, -2147483648.0f, 2147483647.0f);
temp2[i] = _swap32(value);
}
break;
}
default:
assert(false);
}
int res = (int)fwrite(temp, 1, numBytes, fptr);
if (res != numBytes)
{
ST_THROW_RT_ERROR("Error while writing to a wav file.");
}
bytesWritten += numBytes;
}

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////////////////////////////////////////////////////////////////////////////////
///
/// Classes for easy reading & writing of WAV sound files.
///
/// For big-endian CPU, define BIG_ENDIAN during compile-time to correctly
/// parse the WAV files with such processors.
///
/// Admittingly, more complete WAV reader routines may exist in public domain, but
/// the reason for 'yet another' one is that those generic WAV reader libraries are
/// exhaustingly large and cumbersome! Wanted to have something simpler here, i.e.
/// something that's not already larger than rest of the SoundTouch/SoundStretch program...
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// 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;
/// 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

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////////////////////////////////////////////////////////////////////////////////
///
/// SoundStretch main routine.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// 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 = nullptr;
}
}
// 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 == nullptr) || (outFile == nullptr)) 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();
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)
{
int num, samples;
// Read sample data from input file
num = inFile->read(sampleBuffer, readSize);
// 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;
}

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

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<?xml version="1.0" encoding="utf-8"?>
<Project DefaultTargets="Build" ToolsVersion="14.0" xmlns="http://schemas.microsoft.com/developer/msbuild/2003">
<ItemGroup Label="ProjectConfigurations">
<ProjectConfiguration Include="Debug|Win32">
<Configuration>Debug</Configuration>
<Platform>Win32</Platform>
</ProjectConfiguration>
<ProjectConfiguration Include="Debug|x64">
<Configuration>Debug</Configuration>
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222
externals/soundtouch/source/SoundTouch/AAFilter.cpp vendored Normal file
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////////////////////////////////////////////////////////////////////////////////
///
/// FIR low-pass (anti-alias) filter with filter coefficient design routine and
/// MMX optimization.
///
/// Anti-alias filter is used to prevent folding of high frequencies when
/// transposing the sample rate with interpolation.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// 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();
}

93
externals/soundtouch/source/SoundTouch/AAFilter.h vendored Normal file
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////////////////////////////////////////////////////////////////////////////////
///
/// Sampled sound tempo changer/time stretch algorithm. Changes the sound tempo
/// while maintaining the original pitch by using a time domain WSOLA-like method
/// with several performance-increasing tweaks.
///
/// 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

571
externals/soundtouch/source/SoundTouch/BPMDetect.cpp vendored Normal file
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////////////////////////////////////////////////////////////////////////////////
///
/// Beats-per-minute (BPM) detection routine.
///
/// The beat detection algorithm works as follows:
/// - Use function 'inputSamples' to input a chunks of samples to the class for
/// analysis. It's a good idea to enter a large sound file or stream in smallish
/// chunks of around few kilosamples in order not to extinguish too much RAM memory.
/// - Inputted sound data is decimated to approx 500 Hz to reduce calculation burden,
/// which is basically ok as low (bass) frequencies mostly determine the beat rate.
/// Simple averaging is used for anti-alias filtering because the resulting signal
/// quality isn't of that high importance.
/// - Decimated sound data is enveloped, i.e. the amplitude shape is detected by
/// taking absolute value that's smoothed by sliding average. Signal levels that
/// are below a couple of times the general RMS amplitude level are cut away to
/// leave only notable peaks there.
/// - Repeating sound patterns (e.g. beats) are detected by calculating short-term
/// autocorrelation function of the enveloped signal.
/// - After whole sound data file has been analyzed as above, the bpm level is
/// detected by function 'getBpm' that finds the highest peak of the autocorrelation
/// function, calculates it's precise location and converts this reading to bpm's.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#define _USE_MATH_DEFINES
#include <math.h>
#include <assert.h>
#include <string.h>
#include <stdio.h>
#include <cfloat>
#include "FIFOSampleBuffer.h"
#include "PeakFinder.h"
#include "BPMDetect.h"
using namespace soundtouch;
// algorithm input sample block size
static const int INPUT_BLOCK_SIZE = 2048;
// decimated sample block size
static const int DECIMATED_BLOCK_SIZE = 256;
/// Target sample rate after decimation
static const int TARGET_SRATE = 1000;
/// XCorr update sequence size, update in about 200msec chunks
static const int XCORR_UPDATE_SEQUENCE = (int)(TARGET_SRATE / 5);
/// Moving average N size
static const int MOVING_AVERAGE_N = 15;
/// XCorr decay time constant, decay to half in 30 seconds
/// If it's desired to have the system adapt quicker to beat rate
/// changes within a continuing music stream, then the
/// 'xcorr_decay_time_constant' value can be reduced, yet that
/// can increase possibility of glitches in bpm detection.
static const double XCORR_DECAY_TIME_CONSTANT = 30.0;
/// Data overlap factor for beat detection algorithm
static const int OVERLAP_FACTOR = 4;
static const double TWOPI = (2 * M_PI);
////////////////////////////////////////////////////////////////////////////////
// Enable following define to create bpm analysis file:
//#define _CREATE_BPM_DEBUG_FILE
#ifdef _CREATE_BPM_DEBUG_FILE
static void _SaveDebugData(const char *name, const float *data, int minpos, int maxpos, double coeff)
{
FILE *fptr = fopen(name, "wt");
int i;
if (fptr)
{
printf("\nWriting BPM debug data into file %s\n", name);
for (i = minpos; i < maxpos; i ++)
{
fprintf(fptr, "%d\t%.1lf\t%f\n", i, coeff / (double)i, data[i]);
}
fclose(fptr);
}
}
void _SaveDebugBeatPos(const char *name, const std::vector<BEAT> &beats)
{
printf("\nWriting beat detections data into file %s\n", name);
FILE *fptr = fopen(name, "wt");
if (fptr)
{
for (uint i = 0; i < beats.size(); i++)
{
BEAT b = beats[i];
fprintf(fptr, "%lf\t%lf\n", b.pos, b.strength);
}
fclose(fptr);
}
}
#else
#define _SaveDebugData(name, a,b,c,d)
#define _SaveDebugBeatPos(name, b)
#endif
// Hamming window
void hamming(float *w, int N)
{
for (int i = 0; i < N; i++)
{
w[i] = (float)(0.54 - 0.46 * cos(TWOPI * i / (N - 1)));
}
}
////////////////////////////////////////////////////////////////////////////////
//
// IIR2_filter - 2nd order IIR filter
IIR2_filter::IIR2_filter(const double *lpf_coeffs)
{
memcpy(coeffs, lpf_coeffs, 5 * sizeof(double));
memset(prev, 0, sizeof(prev));
}
float IIR2_filter::update(float x)
{
prev[0] = x;
double y = x * coeffs[0];
for (int i = 4; i >= 1; i--)
{
y += coeffs[i] * prev[i];
prev[i] = prev[i - 1];
}
prev[3] = y;
return (float)y;
}
// IIR low-pass filter coefficients, calculated with matlab/octave cheby2(2,40,0.05)
const double _LPF_coeffs[5] = { 0.00996655391939, -0.01944529148401, 0.00996655391939, 1.96867605796247, -0.96916387431724 };
////////////////////////////////////////////////////////////////////////////////
BPMDetect::BPMDetect(int numChannels, int aSampleRate) :
beat_lpf(_LPF_coeffs)
{
beats.reserve(250); // initial reservation to prevent frequent reallocation
this->sampleRate = aSampleRate;
this->channels = numChannels;
decimateSum = 0;
decimateCount = 0;
// choose decimation factor so that result is approx. 1000 Hz
decimateBy = sampleRate / TARGET_SRATE;
if ((decimateBy <= 0) || (decimateBy * DECIMATED_BLOCK_SIZE < INPUT_BLOCK_SIZE))
{
ST_THROW_RT_ERROR("Too small samplerate");
}
// Calculate window length & starting item according to desired min & max bpms
windowLen = (60 * sampleRate) / (decimateBy * MIN_BPM);
windowStart = (60 * sampleRate) / (decimateBy * MAX_BPM_RANGE);
assert(windowLen > windowStart);
// allocate new working objects
xcorr = new float[windowLen];
memset(xcorr, 0, windowLen * sizeof(float));
pos = 0;
peakPos = 0;
peakVal = 0;
init_scaler = 1;
beatcorr_ringbuffpos = 0;
beatcorr_ringbuff = new float[windowLen];
memset(beatcorr_ringbuff, 0, windowLen * sizeof(float));
// allocate processing buffer
buffer = new FIFOSampleBuffer();
// we do processing in mono mode
buffer->setChannels(1);
buffer->clear();
// calculate hamming windows
hamw = new float[XCORR_UPDATE_SEQUENCE];
hamming(hamw, XCORR_UPDATE_SEQUENCE);
hamw2 = new float[XCORR_UPDATE_SEQUENCE / 2];
hamming(hamw2, XCORR_UPDATE_SEQUENCE / 2);
}
BPMDetect::~BPMDetect()
{
delete[] xcorr;
delete[] beatcorr_ringbuff;
delete[] hamw;
delete[] hamw2;
delete buffer;
}
/// convert to mono, low-pass filter & decimate to about 500 Hz.
/// return number of outputted samples.
///
/// Decimation is used to remove the unnecessary frequencies and thus to reduce
/// the amount of data needed to be processed as calculating autocorrelation
/// function is a very-very heavy operation.
///
/// Anti-alias filtering is done simply by averaging the samples. This is really a
/// poor-man's anti-alias filtering, but it's not so critical in this kind of application
/// (it'd also be difficult to design a high-quality filter with steep cut-off at very
/// narrow band)
int BPMDetect::decimate(SAMPLETYPE *dest, const SAMPLETYPE *src, int numsamples)
{
int count, outcount;
LONG_SAMPLETYPE out;
assert(channels > 0);
assert(decimateBy > 0);
outcount = 0;
for (count = 0; count < numsamples; count ++)
{
int j;
// convert to mono and accumulate
for (j = 0; j < channels; j ++)
{
decimateSum += src[j];
}
src += j;
decimateCount ++;
if (decimateCount >= decimateBy)
{
// Store every Nth sample only
out = (LONG_SAMPLETYPE)(decimateSum / (decimateBy * channels));
decimateSum = 0;
decimateCount = 0;
#ifdef SOUNDTOUCH_INTEGER_SAMPLES
// check ranges for sure (shouldn't actually be necessary)
if (out > 32767)
{
out = 32767;
}
else if (out < -32768)
{
out = -32768;
}
#endif // SOUNDTOUCH_INTEGER_SAMPLES
dest[outcount] = (SAMPLETYPE)out;
outcount ++;
}
}
return outcount;
}
// Calculates autocorrelation function of the sample history buffer
void BPMDetect::updateXCorr(int process_samples)
{
int offs;
SAMPLETYPE *pBuffer;
assert(buffer->numSamples() >= (uint)(process_samples + windowLen));
assert(process_samples == XCORR_UPDATE_SEQUENCE);
pBuffer = buffer->ptrBegin();
// calculate decay factor for xcorr filtering
float xcorr_decay = (float)pow(0.5, 1.0 / (XCORR_DECAY_TIME_CONSTANT * TARGET_SRATE / process_samples));
// prescale pbuffer
float tmp[XCORR_UPDATE_SEQUENCE];
for (int i = 0; i < process_samples; i++)
{
tmp[i] = hamw[i] * hamw[i] * pBuffer[i];
}
#pragma omp parallel for
for (offs = windowStart; offs < windowLen; offs ++)
{
float sum;
int i;
sum = 0;
for (i = 0; i < process_samples; i ++)
{
sum += tmp[i] * pBuffer[i + offs]; // scaling the sub-result shouldn't be necessary
}
xcorr[offs] *= xcorr_decay; // decay 'xcorr' here with suitable time constant.
xcorr[offs] += (float)fabs(sum);
}
}
// Detect individual beat positions
void BPMDetect::updateBeatPos(int process_samples)
{
SAMPLETYPE *pBuffer;
assert(buffer->numSamples() >= (uint)(process_samples + windowLen));
pBuffer = buffer->ptrBegin();
assert(process_samples == XCORR_UPDATE_SEQUENCE / 2);
// static double thr = 0.0003;
double posScale = (double)this->decimateBy / (double)this->sampleRate;
int resetDur = (int)(0.12 / posScale + 0.5);
// prescale pbuffer
float tmp[XCORR_UPDATE_SEQUENCE / 2];
for (int i = 0; i < process_samples; i++)
{
tmp[i] = hamw2[i] * hamw2[i] * pBuffer[i];
}
#pragma omp parallel for
for (int offs = windowStart; offs < windowLen; offs++)
{
float sum = 0;
for (int i = 0; i < process_samples; i++)
{
sum += tmp[i] * pBuffer[offs + i];
}
beatcorr_ringbuff[(beatcorr_ringbuffpos + offs) % windowLen] += (float)((sum > 0) ? sum : 0); // accumulate only positive correlations
}
int skipstep = XCORR_UPDATE_SEQUENCE / OVERLAP_FACTOR;
// compensate empty buffer at beginning by scaling coefficient
float scale = (float)windowLen / (float)(skipstep * init_scaler);
if (scale > 1.0f)
{
init_scaler++;
}
else
{
scale = 1.0f;
}
// detect beats
for (int i = 0; i < skipstep; i++)
{
float sum = beatcorr_ringbuff[beatcorr_ringbuffpos];
sum -= beat_lpf.update(sum);
if (sum > peakVal)
{
// found new local largest value
peakVal = sum;
peakPos = pos;
}
if (pos > peakPos + resetDur)
{
// largest value not updated for 200msec => accept as beat
peakPos += skipstep;
if (peakVal > 0)
{
// add detected beat to end of "beats" vector
BEAT temp = { (float)(peakPos * posScale), (float)(peakVal * scale) };
beats.push_back(temp);
}
peakVal = 0;
peakPos = pos;
}
beatcorr_ringbuff[beatcorr_ringbuffpos] = 0;
pos++;
beatcorr_ringbuffpos = (beatcorr_ringbuffpos + 1) % windowLen;
}
}
#define max(x,y) ((x) > (y) ? (x) : (y))
void BPMDetect::inputSamples(const SAMPLETYPE *samples, int numSamples)
{
SAMPLETYPE decimated[DECIMATED_BLOCK_SIZE];
// iterate so that max INPUT_BLOCK_SAMPLES processed per iteration
while (numSamples > 0)
{
int block;
int decSamples;
block = (numSamples > INPUT_BLOCK_SIZE) ? INPUT_BLOCK_SIZE : numSamples;
// decimate. note that converts to mono at the same time
decSamples = decimate(decimated, samples, block);
samples += block * channels;
numSamples -= block;
buffer->putSamples(decimated, decSamples);
}
// when the buffer has enough samples for processing...
int req = max(windowLen + XCORR_UPDATE_SEQUENCE, 2 * XCORR_UPDATE_SEQUENCE);
while ((int)buffer->numSamples() >= req)
{
// ... update autocorrelations...
updateXCorr(XCORR_UPDATE_SEQUENCE);
// ...update beat position calculation...
updateBeatPos(XCORR_UPDATE_SEQUENCE / 2);
// ... and remove proceessed samples from the buffer
int n = XCORR_UPDATE_SEQUENCE / OVERLAP_FACTOR;
buffer->receiveSamples(n);
}
}
void BPMDetect::removeBias()
{
int i;
// Remove linear bias: calculate linear regression coefficient
// 1. calc mean of 'xcorr' and 'i'
double mean_i = 0;
double mean_x = 0;
for (i = windowStart; i < windowLen; i++)
{
mean_x += xcorr[i];
}
mean_x /= (windowLen - windowStart);
mean_i = 0.5 * (windowLen - 1 + windowStart);
// 2. calculate linear regression coefficient
double b = 0;
double div = 0;
for (i = windowStart; i < windowLen; i++)
{
double xt = xcorr[i] - mean_x;
double xi = i - mean_i;
b += xt * xi;
div += xi * xi;
}
b /= div;
// subtract linear regression and resolve min. value bias
float minval = FLT_MAX; // arbitrary large number
for (i = windowStart; i < windowLen; i ++)
{
xcorr[i] -= (float)(b * i);
if (xcorr[i] < minval)
{
minval = xcorr[i];
}
}
// subtract min.value
for (i = windowStart; i < windowLen; i ++)
{
xcorr[i] -= minval;
}
}
// Calculate N-point moving average for "source" values
void MAFilter(float *dest, const float *source, int start, int end, int N)
{
for (int i = start; i < end; i++)
{
int i1 = i - N / 2;
int i2 = i + N / 2 + 1;
if (i1 < start) i1 = start;
if (i2 > end) i2 = end;
double sum = 0;
for (int j = i1; j < i2; j ++)
{
sum += source[j];
}
dest[i] = (float)(sum / (i2 - i1));
}
}
float BPMDetect::getBpm()
{
double peakPos;
double coeff;
PeakFinder peakFinder;
// remove bias from xcorr data
removeBias();
coeff = 60.0 * ((double)sampleRate / (double)decimateBy);
// save bpm debug data if debug data writing enabled
_SaveDebugData("soundtouch-bpm-xcorr.txt", xcorr, windowStart, windowLen, coeff);
// Smoothen by N-point moving-average
float *data = new float[windowLen];
memset(data, 0, sizeof(float) * windowLen);
MAFilter(data, xcorr, windowStart, windowLen, MOVING_AVERAGE_N);
// find peak position
peakPos = peakFinder.detectPeak(data, windowStart, windowLen);
// save bpm debug data if debug data writing enabled
_SaveDebugData("soundtouch-bpm-smoothed.txt", data, windowStart, windowLen, coeff);
delete[] data;
assert(decimateBy != 0);
if (peakPos < 1e-9) return 0.0; // detection failed.
_SaveDebugBeatPos("soundtouch-detected-beats.txt", beats);
// calculate BPM
float bpm = (float)(coeff / peakPos);
return (bpm >= MIN_BPM && bpm <= MAX_BPM_VALID) ? bpm : 0;
}
/// Get beat position arrays. Note: The array includes also really low beat detection values
/// in absence of clear strong beats. Consumer may wish to filter low values away.
/// - "pos" receive array of beat positions
/// - "values" receive array of beat detection strengths
/// - max_num indicates max.size of "pos" and "values" array.
///
/// You can query a suitable array sized by calling this with nullptr in "pos" & "values".
///
/// \return number of beats in the arrays.
int BPMDetect::getBeats(float *pos, float *values, int max_num)
{
int num = (int)beats.size();
if ((!pos) || (!values)) return num; // pos or values nullptr, return just size
for (int i = 0; (i < num) && (i < max_num); i++)
{
pos[i] = beats[i].pos;
values[i] = beats[i].strength;
}
return num;
}

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////////////////////////////////////////////////////////////////////////////////
///
/// A buffer class for temporarily storaging sound samples, operates as a
/// first-in-first-out pipe.
///
/// Samples are added to the end of the sample buffer with the 'putSamples'
/// function, and are received from the beginning of the buffer by calling
/// the 'receiveSamples' function. The class automatically removes the
/// outputted samples from the buffer, as well as grows the buffer size
/// whenever necessary.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// 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;
}

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////////////////////////////////////////////////////////////////////////////////
///
/// 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 = 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 < 16);
// 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");
#ifdef SOUNDTOUCH_FLOAT_SAMPLES
// scale coefficients already here if using floating samples
double scale = 1.0 / resultDivider;
#else
short scale = 1;
#endif
lengthDiv8 = newLength / 8;
length = lengthDiv8 * 8;
assert(length == newLength);
resultDivFactor = uResultDivFactor;
resultDivider = (SAMPLETYPE)::pow(2.0, (int)resultDivFactor);
delete[] filterCoeffs;
filterCoeffs = new SAMPLETYPE[length];
delete[] filterCoeffsStereo;
filterCoeffsStereo = new SAMPLETYPE[length*2];
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;
}
}

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////////////////////////////////////////////////////////////////////////////////
///
/// General FIR digital filter routines with MMX optimization.
///
/// Note : MMX optimized functions reside in a separate, platform-specific file,
/// e.g. 'mmx_win.cpp' or 'mmx_gcc.cpp'
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// 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;
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

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////////////////////////////////////////////////////////////////////////////////
///
/// 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;
}

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////////////////////////////////////////////////////////////////////////////////
///
/// 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

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////////////////////////////////////////////////////////////////////////////////
///
/// 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;
}

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////////////////////////////////////////////////////////////////////////////////
///
/// 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

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////////////////////////////////////////////////////////////////////////////////
///
/// 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;
}

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////////////////////////////////////////////////////////////////////////////////
///
/// 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

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## 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

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////////////////////////////////////////////////////////////////////////////////
///
/// Peak detection routine.
///
/// The routine detects highest value on an array of values and calculates the
/// precise peak location as a mass-center of the 'hump' around the peak value.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// 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;
}

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////////////////////////////////////////////////////////////////////////////////
///
/// The routine detects highest value on an array of values and calculates the
/// precise peak location as a mass-center of the 'hump' around the peak value.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// 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_

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////////////////////////////////////////////////////////////////////////////////
///
/// Sample rate transposer. Changes sample rate by using linear interpolation
/// together with anti-alias filtering (first order interpolation with anti-
/// alias filtering should be quite adequate for this application)
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// 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
}

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////////////////////////////////////////////////////////////////////////////////
///
/// Sample rate transposer. Changes sample rate by using linear interpolation
/// together with anti-alias filtering (first order interpolation with anti-
/// alias filtering should be quite adequate for this application).
///
/// Use either of the derived classes of 'RateTransposerInteger' or
/// 'RateTransposerFloat' for corresponding integer/floating point tranposing
/// algorithm implementation.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// 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

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//////////////////////////////////////////////////////////////////////////////
///
/// SoundTouch - main class for tempo/pitch/rate adjusting routines.
///
/// Notes:
/// - Initialize the SoundTouch object instance by setting up the sound stream
/// parameters with functions 'setSampleRate' and 'setChannels', then set
/// desired tempo/pitch/rate settings with the corresponding functions.
///
/// - The SoundTouch class behaves like a first-in-first-out pipeline: The
/// samples that are to be processed are fed into one of the pipe by calling
/// function 'putSamples', while the ready processed samples can be read
/// from the other end of the pipeline with function 'receiveSamples'.
///
/// - The SoundTouch processing classes require certain sized 'batches' of
/// samples in order to process the sound. For this reason the classes buffer
/// incoming samples until there are enough of samples available for
/// processing, then they carry out the processing step and consequently
/// make the processed samples available for outputting.
///
/// - For the above reason, the processing routines introduce a certain
/// 'latency' between the input and output, so that the samples input to
/// SoundTouch may not be immediately available in the output, and neither
/// the amount of outputtable samples may not immediately be in direct
/// relationship with the amount of previously input samples.
///
/// - The tempo/pitch/rate control parameters can be altered during processing.
/// Please notice though that they aren't currently protected by semaphores,
/// so in multi-thread application external semaphore protection may be
/// required.
///
/// - This class utilizes classes 'TDStretch' for tempo change (without modifying
/// pitch) and 'RateTransposer' for changing the playback rate (that is, both
/// tempo and pitch in the same ratio) of the sound. The third available control
/// 'pitch' (change pitch but maintain tempo) is produced by a combination of
/// combining the two other controls.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#include <assert.h>
#include <stdlib.h>
#include <memory.h>
#include <math.h>
#include <stdio.h>
#include "SoundTouch.h"
#include "TDStretch.h"
#include "RateTransposer.h"
#include "cpu_detect.h"
using namespace soundtouch;
/// test if two floating point numbers are equal
#define TEST_FLOAT_EQUAL(a, b) (fabs(a - b) < 1e-10)
/// Print library version string for autoconf
extern "C" void soundtouch_ac_test()
{
printf("SoundTouch Version: %s\n",SOUNDTOUCH_VERSION);
}
SoundTouch::SoundTouch()
{
// Initialize rate transposer and tempo changer instances
pRateTransposer = new RateTransposer();
pTDStretch = TDStretch::newInstance();
setOutPipe(pTDStretch);
rate = tempo = 0;
virtualPitch =
virtualRate =
virtualTempo = 1.0;
calcEffectiveRateAndTempo();
samplesExpectedOut = 0;
samplesOutput = 0;
channels = 0;
bSrateSet = false;
}
SoundTouch::~SoundTouch()
{
delete pRateTransposer;
delete pTDStretch;
}
/// Get SoundTouch library version string
const char *SoundTouch::getVersionString()
{
static const char *_version = SOUNDTOUCH_VERSION;
return _version;
}
/// Get SoundTouch library version Id
uint SoundTouch::getVersionId()
{
return SOUNDTOUCH_VERSION_ID;
}
// Sets the number of channels, 1 = mono, 2 = stereo
void SoundTouch::setChannels(uint numChannels)
{
if (!verifyNumberOfChannels(numChannels)) return;
channels = numChannels;
pRateTransposer->setChannels((int)numChannels);
pTDStretch->setChannels((int)numChannels);
}
// Sets new rate control value. Normal rate = 1.0, smaller values
// represent slower rate, larger faster rates.
void SoundTouch::setRate(double newRate)
{
virtualRate = newRate;
calcEffectiveRateAndTempo();
}
// Sets new rate control value as a difference in percents compared
// to the original rate (-50 .. +100 %)
void SoundTouch::setRateChange(double newRate)
{
virtualRate = 1.0 + 0.01 * newRate;
calcEffectiveRateAndTempo();
}
// Sets new tempo control value. Normal tempo = 1.0, smaller values
// represent slower tempo, larger faster tempo.
void SoundTouch::setTempo(double newTempo)
{
virtualTempo = newTempo;
calcEffectiveRateAndTempo();
}
// Sets new tempo control value as a difference in percents compared
// to the original tempo (-50 .. +100 %)
void SoundTouch::setTempoChange(double newTempo)
{
virtualTempo = 1.0 + 0.01 * newTempo;
calcEffectiveRateAndTempo();
}
// Sets new pitch control value. Original pitch = 1.0, smaller values
// represent lower pitches, larger values higher pitch.
void SoundTouch::setPitch(double newPitch)
{
virtualPitch = newPitch;
calcEffectiveRateAndTempo();
}
// Sets pitch change in octaves compared to the original pitch
// (-1.00 .. +1.00)
void SoundTouch::setPitchOctaves(double newPitch)
{
virtualPitch = exp(0.69314718056 * newPitch);
calcEffectiveRateAndTempo();
}
// Sets pitch change in semi-tones compared to the original pitch
// (-12 .. +12)
void SoundTouch::setPitchSemiTones(int newPitch)
{
setPitchOctaves((double)newPitch / 12.0);
}
void SoundTouch::setPitchSemiTones(double newPitch)
{
setPitchOctaves(newPitch / 12.0);
}
// Calculates 'effective' rate and tempo values from the
// nominal control values.
void SoundTouch::calcEffectiveRateAndTempo()
{
double oldTempo = tempo;
double oldRate = rate;
tempo = virtualTempo / virtualPitch;
rate = virtualPitch * virtualRate;
if (!TEST_FLOAT_EQUAL(rate,oldRate)) pRateTransposer->setRate(rate);
if (!TEST_FLOAT_EQUAL(tempo, oldTempo)) pTDStretch->setTempo(tempo);
#ifndef SOUNDTOUCH_PREVENT_CLICK_AT_RATE_CROSSOVER
if (rate <= 1.0f)
{
if (output != pTDStretch)
{
FIFOSamplePipe *tempoOut;
assert(output == pRateTransposer);
// move samples in the current output buffer to the output of pTDStretch
tempoOut = pTDStretch->getOutput();
tempoOut->moveSamples(*output);
// move samples in pitch transposer's store buffer to tempo changer's input
// deprecated : pTDStretch->moveSamples(*pRateTransposer->getStore());
output = pTDStretch;
}
}
else
#endif
{
if (output != pRateTransposer)
{
FIFOSamplePipe *transOut;
assert(output == pTDStretch);
// move samples in the current output buffer to the output of pRateTransposer
transOut = pRateTransposer->getOutput();
transOut->moveSamples(*output);
// move samples in tempo changer's input to pitch transposer's input
pRateTransposer->moveSamples(*pTDStretch->getInput());
output = pRateTransposer;
}
}
}
// Sets sample rate.
void SoundTouch::setSampleRate(uint srate)
{
// set sample rate, leave other tempo changer parameters as they are.
pTDStretch->setParameters((int)srate);
bSrateSet = true;
}
// Adds 'numSamples' pcs of samples from the 'samples' memory position into
// the input of the object.
void SoundTouch::putSamples(const SAMPLETYPE *samples, uint nSamples)
{
if (bSrateSet == false)
{
ST_THROW_RT_ERROR("SoundTouch : Sample rate not defined");
}
else if (channels == 0)
{
ST_THROW_RT_ERROR("SoundTouch : Number of channels not defined");
}
// accumulate how many samples are expected out from processing, given the current
// processing setting
samplesExpectedOut += (double)nSamples / ((double)rate * (double)tempo);
#ifndef SOUNDTOUCH_PREVENT_CLICK_AT_RATE_CROSSOVER
if (rate <= 1.0f)
{
// transpose the rate down, output the transposed sound to tempo changer buffer
assert(output == pTDStretch);
pRateTransposer->putSamples(samples, nSamples);
pTDStretch->moveSamples(*pRateTransposer);
}
else
#endif
{
// evaluate the tempo changer, then transpose the rate up,
assert(output == pRateTransposer);
pTDStretch->putSamples(samples, nSamples);
pRateTransposer->moveSamples(*pTDStretch);
}
}
// Flushes the last samples from the processing pipeline to the output.
// Clears also the internal processing buffers.
//
// Note: This function is meant for extracting the last samples of a sound
// stream. This function may introduce additional blank samples in the end
// of the sound stream, and thus it's not recommended to call this function
// in the middle of a sound stream.
void SoundTouch::flush()
{
int i;
int numStillExpected;
SAMPLETYPE *buff = new SAMPLETYPE[128 * channels];
// how many samples are still expected to output
numStillExpected = (int)((long)(samplesExpectedOut + 0.5) - samplesOutput);
if (numStillExpected < 0) numStillExpected = 0;
memset(buff, 0, 128 * channels * sizeof(SAMPLETYPE));
// "Push" the last active samples out from the processing pipeline by
// feeding blank samples into the processing pipeline until new,
// processed samples appear in the output (not however, more than
// 24ksamples in any case)
for (i = 0; (numStillExpected > (int)numSamples()) && (i < 200); i ++)
{
putSamples(buff, 128);
}
adjustAmountOfSamples(numStillExpected);
delete[] buff;
// Clear input buffers
pTDStretch->clearInput();
// yet leave the output intouched as that's where the
// flushed samples are!
}
// Changes a setting controlling the processing system behaviour. See the
// 'SETTING_...' defines for available setting ID's.
bool SoundTouch::setSetting(int settingId, int value)
{
int sampleRate, sequenceMs, seekWindowMs, overlapMs;
// read current tdstretch routine parameters
pTDStretch->getParameters(&sampleRate, &sequenceMs, &seekWindowMs, &overlapMs);
switch (settingId)
{
case SETTING_USE_AA_FILTER :
// enables / disabless anti-alias filter
pRateTransposer->enableAAFilter((value != 0) ? true : false);
return true;
case SETTING_AA_FILTER_LENGTH :
// sets anti-alias filter length
pRateTransposer->getAAFilter()->setLength(value);
return true;
case SETTING_USE_QUICKSEEK :
// enables / disables tempo routine quick seeking algorithm
pTDStretch->enableQuickSeek((value != 0) ? true : false);
return true;
case SETTING_SEQUENCE_MS:
// change time-stretch sequence duration parameter
pTDStretch->setParameters(sampleRate, value, seekWindowMs, overlapMs);
return true;
case SETTING_SEEKWINDOW_MS:
// change time-stretch seek window length parameter
pTDStretch->setParameters(sampleRate, sequenceMs, value, overlapMs);
return true;
case SETTING_OVERLAP_MS:
// change time-stretch overlap length parameter
pTDStretch->setParameters(sampleRate, sequenceMs, seekWindowMs, value);
return true;
default :
return false;
}
}
// Reads a setting controlling the processing system behaviour. See the
// 'SETTING_...' defines for available setting ID's.
//
// Returns the setting value.
int SoundTouch::getSetting(int settingId) const
{
int temp;
switch (settingId)
{
case SETTING_USE_AA_FILTER :
return (uint)pRateTransposer->isAAFilterEnabled();
case SETTING_AA_FILTER_LENGTH :
return pRateTransposer->getAAFilter()->getLength();
case SETTING_USE_QUICKSEEK :
return (uint)pTDStretch->isQuickSeekEnabled();
case SETTING_SEQUENCE_MS:
pTDStretch->getParameters(nullptr, &temp, nullptr, nullptr);
return temp;
case SETTING_SEEKWINDOW_MS:
pTDStretch->getParameters(nullptr, nullptr, &temp, nullptr);
return temp;
case SETTING_OVERLAP_MS:
pTDStretch->getParameters(nullptr, nullptr, nullptr, &temp);
return temp;
case SETTING_NOMINAL_INPUT_SEQUENCE :
{
int size = pTDStretch->getInputSampleReq();
#ifndef SOUNDTOUCH_PREVENT_CLICK_AT_RATE_CROSSOVER
if (rate <= 1.0)
{
// transposing done before timestretch, which impacts latency
return (int)(size * rate + 0.5);
}
#endif
return size;
}
case SETTING_NOMINAL_OUTPUT_SEQUENCE :
{
int size = pTDStretch->getOutputBatchSize();
if (rate > 1.0)
{
// transposing done after timestretch, which impacts latency
return (int)(size / rate + 0.5);
}
return size;
}
case SETTING_INITIAL_LATENCY:
{
double latency = pTDStretch->getLatency();
int latency_tr = pRateTransposer->getLatency();
#ifndef SOUNDTOUCH_PREVENT_CLICK_AT_RATE_CROSSOVER
if (rate <= 1.0)
{
// transposing done before timestretch, which impacts latency
latency = (latency + latency_tr) * rate;
}
else
#endif
{
latency += (double)latency_tr / rate;
}
return (int)(latency + 0.5);
}
default :
return 0;
}
}
// Clears all the samples in the object's output and internal processing
// buffers.
void SoundTouch::clear()
{
samplesExpectedOut = 0;
samplesOutput = 0;
pRateTransposer->clear();
pTDStretch->clear();
}
/// Returns number of samples currently unprocessed.
uint SoundTouch::numUnprocessedSamples() const
{
FIFOSamplePipe * psp;
if (pTDStretch)
{
psp = pTDStretch->getInput();
if (psp)
{
return psp->numSamples();
}
}
return 0;
}
/// Output samples from beginning of the sample buffer. Copies requested samples to
/// output buffer and removes them from the sample buffer. If there are less than
/// 'numsample' samples in the buffer, returns all that available.
///
/// \return Number of samples returned.
uint SoundTouch::receiveSamples(SAMPLETYPE *output, uint maxSamples)
{
uint ret = FIFOProcessor::receiveSamples(output, maxSamples);
samplesOutput += (long)ret;
return ret;
}
/// Adjusts book-keeping so that given number of samples are removed from beginning of the
/// sample buffer without copying them anywhere.
///
/// Used to reduce the number of samples in the buffer when accessing the sample buffer directly
/// with 'ptrBegin' function.
uint SoundTouch::receiveSamples(uint maxSamples)
{
uint ret = FIFOProcessor::receiveSamples(maxSamples);
samplesOutput += (long)ret;
return ret;
}
/// Get ratio between input and output audio durations, useful for calculating
/// processed output duration: if you'll process a stream of N samples, then
/// you can expect to get out N * getInputOutputSampleRatio() samples.
double SoundTouch::getInputOutputSampleRatio()
{
return 1.0 / (tempo * rate);
}

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Microsoft Visual Studio Solution File, Format Version 12.00
# Visual Studio 14
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////////////////////////////////////////////////////////////////////////////////
///
/// Sampled sound tempo changer/time stretch algorithm. Changes the sound tempo
/// while maintaining the original pitch by using a time domain WSOLA-like method
/// with several performance-increasing tweaks.
///
/// Note : MMX/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

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

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////////////////////////////////////////////////////////////////////////////////
///
/// 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
}

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////////////////////////////////////////////////////////////////////////////////
///
/// MMX optimized routines. All MMX optimized functions have been gathered into
/// this single source code file, regardless to their class or original source
/// code file, in order to ease porting the library to other compiler and
/// processor platforms.
///
/// The MMX-optimizations are programmed using MMX compiler intrinsics that
/// are supported both by Microsoft Visual C++ and GCC compilers, so this file
/// should compile with both toolsets.
///
/// NOTICE: If using Visual Studio 6.0, you'll need to install the "Visual C++
/// 6.0 processor pack" update to support compiler intrinsic syntax. The update
/// is available for download at Microsoft Developers Network, see here:
/// http://msdn.microsoft.com/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

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////////////////////////////////////////////////////////////////////////////////
///
/// SSE optimized routines for Pentium-III, Athlon-XP and later CPUs. All SSE
/// optimized functions have been gathered into this single source
/// code file, regardless to their class or original source code file, in order
/// to ease porting the library to other compiler and processor platforms.
///
/// The SSE-optimizations are programmed using SSE compiler intrinsics that
/// are supported both by Microsoft Visual C++ and GCC compilers, so this file
/// should compile with both toolsets.
///
/// NOTICE: If using Visual Studio 6.0, you'll need to install the "Visual C++
/// 6.0 processor pack" update to support SSE instruction set. The update is
/// available for download at Microsoft Developers Network, see here:
/// http://msdn.microsoft.com/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)
{
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 != 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

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externals/soundtouch/source/SoundTouchDLL/DllTest/DllTest.cpp vendored Normal file
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////////////////////////////////////////////////////////////////////////////////
///
/// 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;
}

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externals/soundtouch/source/SoundTouchDLL/LazarusTest/README.txt vendored Normal file
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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.

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externals/soundtouch/source/SoundTouchDLL/LazarusTest/SoundTouchDLL.pas vendored Normal file
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unit SoundTouchDLL;
//////////////////////////////////////////////////////////////////////////////
//
// SoundTouch.dll / libSoundTouchDll.so wrapper for accessing SoundTouch
// routines from Delphi/Pascal/Lazarus
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
interface
//uses
//Windows;
type
TSoundTouchHandle = THandle;
// Create a new instance of SoundTouch processor.
TSoundTouchCreateInstance = function : TSoundTouchHandle; cdecl;
// Destroys a SoundTouch processor instance.
TSoundTouchDestroyInstance = procedure (Handle: TSoundTouchHandle); cdecl;
// Get SoundTouch library version string
TSoundTouchGetVersionString = function : PAnsiChar; cdecl;
// Get SoundTouch library version string 2
TSoundTouchGetVersionString2 = procedure(VersionString : PAnsiChar; BufferSize : Integer); cdecl;
// Get SoundTouch library version Id
TSoundTouchGetVersionId = function : Cardinal; cdecl;
// Sets new rate control value. Normal rate = 1.0, smaller values
// represent slower rate, larger faster rates.
TSoundTouchSetRate = procedure (Handle: TSoundTouchHandle; NewRate: Single); cdecl;
// Sets new tempo control value. Normal tempo = 1.0, smaller values
// represent slower tempo, larger faster tempo.
TSoundTouchSetTempo = procedure (Handle: TSoundTouchHandle; NewTempo: Single); cdecl;
// Sets new rate control value as a difference in percents compared
// to the original rate (-50 .. +100 %);
TSoundTouchSetRateChange = procedure (Handle: TSoundTouchHandle; NewRate: Single); cdecl;
// Sets new tempo control value as a difference in percents compared
// to the original tempo (-50 .. +100 %);
TSoundTouchSetTempoChange = procedure (Handle: TSoundTouchHandle; NewTempo: Single); cdecl;
// Sets new pitch control value. Original pitch = 1.0, smaller values
// represent lower pitches, larger values higher pitch.
TSoundTouchSetPitch = procedure (Handle: TSoundTouchHandle; NewPitch: Single); cdecl;
// Sets pitch change in octaves compared to the original pitch
// (-1.00 .. +1.00);
TSoundTouchSetPitchOctaves = procedure (Handle: TSoundTouchHandle; NewPitch: Single); cdecl;
// Sets pitch change in semi-tones compared to the original pitch
// (-12 .. +12);
TSoundTouchSetPitchSemiTones = procedure (Handle: TSoundTouchHandle; NewPitch: Single); cdecl;
// Sets the number of channels, 1 = mono, 2 = stereo
TSoundTouchSetChannels = procedure (Handle: TSoundTouchHandle; NumChannels: Cardinal); cdecl;
// Sets sample rate.
TSoundTouchSetSampleRate = procedure (Handle: TSoundTouchHandle; SampleRate: Cardinal); cdecl;
// Flushes the last samples from the processing pipeline to the output.
// Clears also the internal processing buffers.
//
// Note: This function is meant for extracting the last samples of a sound
// stream. This function may introduce additional blank samples in the end
// of the sound stream, and thus it
// in the middle of a sound stream.
TSoundTouchFlush = procedure (Handle: TSoundTouchHandle); cdecl;
// Adds 'numSamples' pcs of samples from the 'samples' memory position into
// the input of the object. Notice that sample rate _has_to_ be set before
// calling this function, otherwise throws a runtime_error exception.
TSoundTouchPutSamples = procedure (Handle: TSoundTouchHandle;
const Samples: PSingle; //< Pointer to sample buffer.
NumSamples: Cardinal //< Number of samples in buffer. Notice
//< that in case of stereo-sound a single sample
//< contains data for both channels.
); 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;
// 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
TSoundTouchSetSetting = function (Handle: TSoundTouchHandle;
SettingId: Integer; //< Setting ID number. see SETTING_... defines.
Value: Integer //< New setting value.
): Boolean; cdecl;
// Reads a setting controlling the processing system behaviour. See the
// 'SETTING_...' defines for available setting ID's.
//
// \return the setting value.
TSoundTouchGetSetting = function (Handle: TSoundTouchHandle;
SettingId: Integer //< Setting ID number, see SETTING_... defines.
): Integer; cdecl;
// Returns number of samples currently unprocessed.
TSoundTouchNumUnprocessedSamples = function (Handle: TSoundTouchHandle): Cardinal; cdecl;
/// 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;
// Returns nonzero if there aren't any samples available for outputting.
TSoundTouchIsEmpty = function (Handle: TSoundTouchHandle): Integer; cdecl;
var
SoundTouchCreateInstance : TSoundTouchCreateInstance;
SoundTouchDestroyInstance : TSoundTouchDestroyInstance;
SoundTouchGetVersionString : TSoundTouchGetVersionString;
SoundTouchGetVersionString2 : TSoundTouchGetVersionString2;
SoundTouchGetVersionId : TSoundTouchGetVersionId;
SoundTouchSetRate : TSoundTouchSetRate;
SoundTouchSetTempo : TSoundTouchSetTempo;
SoundTouchSetRateChange : TSoundTouchSetRateChange;
SoundTouchSetTempoChange : TSoundTouchSetTempoChange;
SoundTouchSetPitch : TSoundTouchSetPitch;
SoundTouchSetPitchOctaves : TSoundTouchSetPitchOctaves;
SoundTouchSetPitchSemiTones : TSoundTouchSetPitchSemiTones;
SoundTouchSetChannels : TSoundTouchSetChannels;
SoundTouchSetSampleRate : TSoundTouchSetSampleRate;
SoundTouchFlush : TSoundTouchFlush;
SoundTouchPutSamples : TSoundTouchPutSamples;
SoundTouchPutSamplesI16 : TSoundTouchPutSamplesI16;
SoundTouchClear : TSoundTouchClear;
SoundTouchSetSetting : TSoundTouchSetSetting;
SoundTouchGetSetting : TSoundTouchGetSetting;
SoundTouchNumUnprocessedSamples : TSoundTouchNumUnprocessedSamples;
SoundTouchReceiveSamples : TSoundTouchReceiveSamples;
SoundTouchReceiveSamplesI16 : TSoundTouchReceiveSamplesI16;
SoundTouchNumSamples : TSoundTouchNumSamples;
SoundTouchIsEmpty : TSoundTouchIsEmpty;
type
TSoundTouch = class
private
FHandle : TSoundTouchHandle;
FRate : Single;
FPitch : Single;
FTempo : Single;
FSampleRate : Single;
FChannels : Cardinal;
function GetNumSamples: Cardinal;
function GetNumUnprocessedSamples: Cardinal;
function GetIsEmpty: Integer;
function GetPitchChange: Single;
function GetRateChange: Single;
function GetTempoChange: Single;
procedure SetRate(const Value: Single);
procedure SetPitch(const Value: Single);
procedure SetTempo(const Value: Single);
procedure SetPitchChange(const Value: Single);
procedure SetRateChange(const Value: Single);
procedure SetTempoChange(const Value: Single);
procedure SetChannels(const Value: Cardinal);
procedure SetSampleRate(const Value: Single);
protected
procedure SamplerateChanged; virtual;
procedure ChannelsChanged; virtual;
procedure PitchChanged; virtual;
procedure TempoChanged; virtual;
procedure RateChanged; virtual;
public
class function GetVersionString: string;
class function GetVersionId: Cardinal;
constructor Create; virtual;
destructor Destroy; override;
procedure Flush; virtual;
procedure Clear; virtual;
procedure PutSamples(const Samples: PSingle; const NumSamples: Cardinal);
function ReceiveSamples(const OutBuffer: PSingle; const MaxSamples: Integer): Cardinal;
function SetSetting(const SettingId: Integer; const Value: Integer): Boolean;
function GetSetting(const SettingId: Integer): Integer;
property VersionString: string read GetVersionString;
property VersionID: Cardinal read GetVersionId;
property Channels: Cardinal read FChannels write SetChannels;
property Rate: Single read FRate write SetRate;
property RateChange: Single read GetRateChange write SetRateChange;
property Tempo: Single read FTempo write SetTempo;
property TempoChange: Single read GetTempoChange write SetTempoChange;
property Pitch: Single read FPitch write SetPitch;
property PitchChange: Single read GetPitchChange write SetPitchChange;
property SampleRate: Single read FSampleRate write SetSampleRate;
property NumSamples: Cardinal read GetNumSamples;
property NumUnprocessedSamples: Cardinal read GetNumUnprocessedSamples;
property IsEmpty: Integer read GetIsEmpty;
end;
// list of exported functions and procedures
function IsSoundTouchLoaded: Boolean;
implementation
{ TSoundTouch }
constructor TSoundTouch.Create;
begin
inherited;
FHandle := SoundTouchCreateInstance();
FRate := 1;
FTempo := 1;
FPitch := 1;
FChannels := 1;
FSampleRate := 44100;
SamplerateChanged;
ChannelsChanged;
end;
destructor TSoundTouch.Destroy;
begin
SoundTouchDestroyInstance(FHandle);
inherited;
end;
procedure TSoundTouch.Flush;
begin
SoundTouchFlush(FHandle);
end;
procedure TSoundTouch.Clear;
begin
SoundTouchClear(FHandle);
end;
function TSoundTouch.GetIsEmpty: Integer;
begin
result := SoundTouchIsEmpty(FHandle);
end;
function TSoundTouch.GetNumSamples: Cardinal;
begin
result := SoundTouchNumSamples(FHandle);
end;
function TSoundTouch.GetNumUnprocessedSamples: Cardinal;
begin
result := SoundTouchNumUnprocessedSamples(FHandle);
end;
function TSoundTouch.GetPitchChange: Single;
begin
result := 100 * (FPitch - 1.0);
end;
function TSoundTouch.GetRateChange: Single;
begin
result := 100 * (FRate - 1.0);
end;
function TSoundTouch.GetTempoChange: Single;
begin
result := 100 * (FTempo - 1.0);
end;
class function TSoundTouch.GetVersionId: Cardinal;
begin
result := SoundTouchGetVersionId();
end;
class function TSoundTouch.GetVersionString: string;
begin
result := StrPas(SoundTouchGetVersionString());
end;
procedure TSoundTouch.SetChannels(const Value: Cardinal);
begin
if FChannels <> Value then
begin
FChannels := Value;
ChannelsChanged;
end;
end;
procedure TSoundTouch.ChannelsChanged;
begin
assert(FChannels in [1, 2]);
SoundTouchSetChannels(FHandle, FChannels);
end;
procedure TSoundTouch.SetPitch(const Value: Single);
begin
if FPitch <> Value then
begin
FPitch := Value;
PitchChanged;
end;
end;
procedure TSoundTouch.PitchChanged;
begin
SoundTouchSetPitch(FHandle, FPitch);
end;
procedure TSoundTouch.putSamples(const Samples: PSingle;
const NumSamples: Cardinal);
begin
SoundTouchPutSamples(FHandle, Samples, NumSamples);
end;
procedure TSoundTouch.RateChanged;
begin
SoundTouchSetRate(FHandle, FRate);
end;
function TSoundTouch.ReceiveSamples(const OutBuffer: PSingle;
const MaxSamples: Integer): Cardinal;
begin
result := SoundTouchReceiveSamples(FHandle, OutBuffer, MaxSamples);
end;
procedure TSoundTouch.SetPitchChange(const Value: Single);
begin
Pitch := 1.0 + 0.01 * Value;
end;
procedure TSoundTouch.SetRate(const Value: Single);
begin
if FRate <> Value then
begin
FRate := Value;
RateChanged;
end;
end;
procedure TSoundTouch.SetRateChange(const Value: Single);
begin
Rate := 1.0 + 0.01 * Value;
end;
procedure TSoundTouch.SetSampleRate(const Value: Single);
begin
if FSampleRate <> Value then
begin
FSampleRate := Value;
SamplerateChanged;
end;
end;
procedure TSoundTouch.SamplerateChanged;
begin
assert(FSampleRate > 0);
SoundTouchsetSampleRate(FHandle, round(FSampleRate));
end;
procedure TSoundTouch.SetTempo(const Value: Single);
begin
if FTempo <> Value then
begin
FTempo := Value;
TempoChanged;
end;
end;
procedure TSoundTouch.SetTempoChange(const Value: Single);
begin
Tempo := 1.0 + 0.01 * Value;
end;
function TSoundTouch.GetSetting(const SettingId: Integer): Integer;
begin
result := SoundTouchGetSetting(FHandle, SettingId);
end;
function TSoundTouch.SetSetting(const SettingId: Integer;
const Value: Integer): Boolean;
begin
result := SoundTouchSetSetting(FHandle, SettingId, Value);
end;
procedure TSoundTouch.TempoChanged;
begin
SoundTouchsetTempo(FHandle, FTempo);
end;
var
SoundTouchLibHandle: THandle;
SoundTouchDLLFile: AnsiString = 'libSoundTouchDll.so';
//SoundTouchDLLFile: AnsiString = 'SoundTouch.dll';
// bpm detect functions. untested -- if these don't work then remove:
bpm_createInstance: function(chan: int32; sampleRate : int32): THandle; cdecl;
bpm_destroyInstance: procedure(h: THandle); cdecl;
bpm_getBpm: function(h: THandle): Single; cdecl;
bpm_putSamples: procedure(h: THandle; const samples: PSingle; numSamples: cardinal); cdecl;
procedure InitDLL;
begin
{$ifdef mswindows} // Windows
SoundTouchLibHandle := LoadLibrary('.\SoundTouchDll.dll');
{$else} // Unix
SoundTouchLibHandle := LoadLibrary('./libSoundTouchDll.so');
{$endif}
if SoundTouchLibHandle <> 0 then
try
Pointer(SoundTouchCreateInstance) := GetProcAddress(SoundTouchLibHandle, 'soundtouch_createInstance');
Pointer(SoundTouchDestroyInstance) := GetProcAddress(SoundTouchLibHandle, 'soundtouch_destroyInstance');
Pointer(SoundTouchGetVersionString) := GetProcAddress(SoundTouchLibHandle, 'soundtouch_getVersionString');
Pointer(SoundTouchGetVersionString2) := GetProcAddress(SoundTouchLibHandle, 'soundtouch_getVersionString2');
Pointer(SoundTouchGetVersionId) := GetProcAddress(SoundTouchLibHandle, 'soundtouch_getVersionId');
Pointer(SoundTouchSetRate) := GetProcAddress(SoundTouchLibHandle, 'soundtouch_setRate');
Pointer(SoundTouchSetTempo) := GetProcAddress(SoundTouchLibHandle, 'soundtouch_setTempo');
Pointer(SoundTouchSetRateChange) := GetProcAddress(SoundTouchLibHandle, 'soundtouch_setRateChange');
Pointer(SoundTouchSetTempoChange) := GetProcAddress(SoundTouchLibHandle, 'soundtouch_setTempoChange');
Pointer(SoundTouchSetPitch) := GetProcAddress(SoundTouchLibHandle, 'soundtouch_setPitch');
Pointer(SoundTouchSetPitchOctaves) := GetProcAddress(SoundTouchLibHandle, 'soundtouch_setPitchOctaves');
Pointer(SoundTouchSetPitchSemiTones) := GetProcAddress(SoundTouchLibHandle, 'soundtouch_setPitchSemiTones');
Pointer(SoundTouchSetChannels) := GetProcAddress(SoundTouchLibHandle, 'soundtouch_setChannels');
Pointer(SoundTouchSetSampleRate) := GetProcAddress(SoundTouchLibHandle, 'soundtouch_setSampleRate');
Pointer(SoundTouchFlush) := GetProcAddress(SoundTouchLibHandle, 'soundtouch_flush');
Pointer(SoundTouchPutSamples) := GetProcAddress(SoundTouchLibHandle, 'soundtouch_putSamples');
Pointer(SoundTouchPutSamplesI16) := GetProcAddress(SoundTouchLibHandle, 'soundtouch_putSamples_i16');
Pointer(SoundTouchClear) := GetProcAddress(SoundTouchLibHandle, 'soundtouch_clear');
Pointer(SoundTouchSetSetting) := GetProcAddress(SoundTouchLibHandle, 'soundtouch_SetSetting');
Pointer(SoundTouchGetSetting) := GetProcAddress(SoundTouchLibHandle, 'soundtouch_setSetting');
Pointer(SoundTouchNumUnprocessedSamples) := GetProcAddress(SoundTouchLibHandle, 'soundtouch_numUnprocessedSamples');
Pointer(SoundTouchReceiveSamples) := GetProcAddress(SoundTouchLibHandle, 'soundtouch_receiveSamples');
Pointer(SoundTouchReceiveSamplesI16) := GetProcAddress(SoundTouchLibHandle, 'soundtouch_receiveSamples_i16');
Pointer(SoundTouchNumSamples) := GetProcAddress(SoundTouchLibHandle, 'soundtouch_numSamples');
Pointer(SoundTouchIsEmpty) := GetProcAddress(SoundTouchLibHandle, 'soundtouch_isEmpty');
Pointer(bpm_createInstance) :=GetProcAddress(SoundTouchLibHandle, 'bpm_createInstance');
Pointer(bpm_destroyInstance) :=GetProcAddress(SoundTouchLibHandle, 'bpm_destroyInstance');
Pointer(bpm_getBpm) :=GetProcAddress(SoundTouchLibHandle, 'bpm_getBpm');
Pointer(bpm_putSamples) :=GetProcAddress(SoundTouchLibHandle, 'bpm_putSamples');
except
FreeLibrary(SoundTouchLibHandle);
SoundTouchLibHandle := 0;
end;
end;
procedure FreeDLL;
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;
finalization
FreeDLL;
end.

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externals/soundtouch/source/SoundTouchDLL/LazarusTest/main.lfm vendored Normal file
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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

49
externals/soundtouch/source/SoundTouchDLL/LazarusTest/main.pas vendored Normal file
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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.

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externals/soundtouch/source/SoundTouchDLL/LazarusTest/soundtouchtest.lpi vendored Normal file
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<?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>

25
externals/soundtouch/source/SoundTouchDLL/LazarusTest/soundtouchtest.lpr vendored Normal file
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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.

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<?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>
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</JumpHistory>
<RunParams>
<FormatVersion Value="2"/>
<Modes ActiveMode=""/>
</RunParams>
</ProjectSession>
</CONFIG>

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

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externals/soundtouch/source/SoundTouchDLL/SoundTouchDLL.cpp vendored Normal file
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//////////////////////////////////////////////////////////////////////////////
///
/// SoundTouch DLL wrapper - wraps SoundTouch routines into a Dynamic Load
/// Library interface.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// 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
//
////////////////////////////////////////////////////////////////////////////////
#if defined(_WIN32) || defined(WIN32)
#include <windows.h>
// DLL main in Windows compilation
BOOL APIENTRY DllMain( HANDLE hModule,
DWORD ul_reason_for_call,
LPVOID lpReserved
)
{
switch (ul_reason_for_call)
{
case DLL_PROCESS_ATTACH:
case DLL_THREAD_ATTACH:
case DLL_THREAD_DETACH:
case DLL_PROCESS_DETACH:
break;
}
return TRUE;
}
#endif
#include <limits.h>
#include <string.h>
#include "SoundTouchDLL.h"
#include "SoundTouch.h"
#include "BPMDetect.h"
using namespace soundtouch;
#ifdef SOUNDTOUCH_INTEGER_SAMPLES
#error "error - compile the dll version with float samples"
#endif // SOUNDTOUCH_INTEGER_SAMPLES
//////////////
typedef struct
{
DWORD dwMagic;
SoundTouch *pst;
} STHANDLE;
typedef struct
{
DWORD dwMagic;
BPMDetect *pbpm;
uint numChannels;
} BPMHANDLE;
#define STMAGIC 0x1770C001
#define BPMMAGIC 0x1771C10a
SOUNDTOUCHDLL_API HANDLE __cdecl soundtouch_createInstance()
{
STHANDLE *tmp = new STHANDLE;
if (tmp)
{
tmp->dwMagic = STMAGIC;
tmp->pst = new SoundTouch();
if (tmp->pst == nullptr)
{
delete tmp;
tmp = nullptr;
}
}
return (HANDLE)tmp;
}
SOUNDTOUCHDLL_API void __cdecl soundtouch_destroyInstance(HANDLE h)
{
STHANDLE *sth = (STHANDLE*)h;
if (sth->dwMagic != STMAGIC) return;
sth->dwMagic = 0;
if (sth->pst) delete sth->pst;
sth->pst = nullptr;
delete sth;
}
/// Get SoundTouch library version string
SOUNDTOUCHDLL_API const char *__cdecl soundtouch_getVersionString()
{
return SoundTouch::getVersionString();
}
/// Get SoundTouch library version string - alternative function for
/// environments that can't properly handle character string as return value
SOUNDTOUCHDLL_API void __cdecl soundtouch_getVersionString2(char* versionString, int bufferSize)
{
strncpy(versionString, SoundTouch::getVersionString(), bufferSize - 1);
versionString[bufferSize - 1] = 0;
}
/// Get SoundTouch library version Id
SOUNDTOUCHDLL_API uint __cdecl soundtouch_getVersionId()
{
return SoundTouch::getVersionId();
}
/// Sets new rate control value. Normal rate = 1.0, smaller values
/// represent slower rate, larger faster rates.
SOUNDTOUCHDLL_API void __cdecl soundtouch_setRate(HANDLE h, float newRate)
{
STHANDLE *sth = (STHANDLE*)h;
if (sth->dwMagic != STMAGIC) return;
sth->pst->setRate(newRate);
}
/// Sets new tempo control value. Normal tempo = 1.0, smaller values
/// represent slower tempo, larger faster tempo.
SOUNDTOUCHDLL_API void __cdecl soundtouch_setTempo(HANDLE h, float newTempo)
{
STHANDLE *sth = (STHANDLE*)h;
if (sth->dwMagic != STMAGIC) return;
sth->pst->setTempo(newTempo);
}
/// Sets new rate control value as a difference in percents compared
/// to the original rate (-50 .. +100 %)
SOUNDTOUCHDLL_API void __cdecl soundtouch_setRateChange(HANDLE h, float newRate)
{
STHANDLE *sth = (STHANDLE*)h;
if (sth->dwMagic != STMAGIC) return;
sth->pst->setRateChange(newRate);
}
/// Sets new tempo control value as a difference in percents compared
/// to the original tempo (-50 .. +100 %)
SOUNDTOUCHDLL_API void __cdecl soundtouch_setTempoChange(HANDLE h, float newTempo)
{
STHANDLE *sth = (STHANDLE*)h;
if (sth->dwMagic != STMAGIC) return;
sth->pst->setTempoChange(newTempo);
}
/// Sets new pitch control value. Original pitch = 1.0, smaller values
/// represent lower pitches, larger values higher pitch.
SOUNDTOUCHDLL_API void __cdecl soundtouch_setPitch(HANDLE h, float newPitch)
{
STHANDLE *sth = (STHANDLE*)h;
if (sth->dwMagic != STMAGIC) return;
sth->pst->setPitch(newPitch);
}
/// Sets pitch change in octaves compared to the original pitch
/// (-1.00 .. +1.00)
SOUNDTOUCHDLL_API void __cdecl soundtouch_setPitchOctaves(HANDLE h, float newPitch)
{
STHANDLE *sth = (STHANDLE*)h;
if (sth->dwMagic != STMAGIC) return;
sth->pst->setPitchOctaves(newPitch);
}
/// Sets pitch change in semi-tones compared to the original pitch
/// (-12 .. +12)
SOUNDTOUCHDLL_API void __cdecl soundtouch_setPitchSemiTones(HANDLE h, float newPitch)
{
STHANDLE *sth = (STHANDLE*)h;
if (sth->dwMagic != STMAGIC) return;
sth->pst->setPitchSemiTones(newPitch);
}
/// Sets the number of channels, 1 = mono, 2 = stereo
SOUNDTOUCHDLL_API int __cdecl soundtouch_setChannels(HANDLE h, uint numChannels)
{
STHANDLE *sth = (STHANDLE*)h;
if (sth->dwMagic != STMAGIC) return 0;
try
{
sth->pst->setChannels(numChannels);
}
catch (const std::exception&)
{
return 0;
}
return 1;
}
/// Sets sample rate.
SOUNDTOUCHDLL_API int __cdecl soundtouch_setSampleRate(HANDLE h, uint srate)
{
STHANDLE *sth = (STHANDLE*)h;
if (sth->dwMagic != STMAGIC) return 0;
try
{
sth->pst->setSampleRate(srate);
}
catch (const std::exception&)
{
return 0;
}
return 1;
}
/// 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)
{
STHANDLE *sth = (STHANDLE*)h;
if (sth->dwMagic != STMAGIC) return 0;
try
{
sth->pst->flush();
}
catch (const std::exception&)
{
return 0;
}
return 1;
}
/// 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 SAMPLETYPE *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.
)
{
STHANDLE *sth = (STHANDLE*)h;
if (sth->dwMagic != STMAGIC) return 0;
try
{
sth->pst->putSamples(samples, numSamples);
}
catch (const std::exception&)
{
return 0;
}
return 1;
}
/// 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
///< contains data for all channels.
)
{
STHANDLE *sth = (STHANDLE*)h;
if (sth->dwMagic != STMAGIC) return;
uint numChannels = sth->pst->numChannels();
// iterate until all samples converted & put to SoundTouch object
while (numSamples > 0)
{
float convert[8192]; // allocate temporary conversion buffer from stack
// how many multichannel samples fit into 'convert' buffer:
uint convSamples = 8192 / numChannels;
// convert max 'nround' values at a time to guarantee that these fit in the 'convert' buffer
uint n = (numSamples > convSamples) ? convSamples : numSamples;
for (uint i = 0; i < n * numChannels; i++)
{
convert[i] = samples[i];
}
// put the converted samples into SoundTouch
sth->pst->putSamples(convert, n);
numSamples -= n;
samples += n * numChannels;
}
}
/// Clears all the samples in the object's output and internal processing
/// buffers.
SOUNDTOUCHDLL_API void __cdecl soundtouch_clear(HANDLE h)
{
STHANDLE *sth = (STHANDLE*)h;
if (sth->dwMagic != STMAGIC) return;
sth->pst->clear();
}
/// Changes a setting controlling the processing system behaviour. See the
/// 'SETTING_...' defines for available setting ID's.
///
/// \return 'nonzero' if the setting was successfully changed
SOUNDTOUCHDLL_API int __cdecl soundtouch_setSetting(HANDLE h,
int settingId, ///< Setting ID number. see SETTING_... defines.
int value ///< New setting value.
)
{
STHANDLE *sth = (STHANDLE*)h;
if (sth->dwMagic != STMAGIC) return FALSE;
return sth->pst->setSetting(settingId, value);
}
/// Reads a setting controlling the processing system behaviour. See the
/// 'SETTING_...' defines for available setting ID's.
///
/// \return the setting value.
SOUNDTOUCHDLL_API int __cdecl soundtouch_getSetting(HANDLE h,
int settingId ///< Setting ID number, see SETTING_... defines.
)
{
STHANDLE *sth = (STHANDLE*)h;
if (sth->dwMagic != STMAGIC) return -1;
return sth->pst->getSetting(settingId);
}
/// Returns number of samples currently unprocessed.
SOUNDTOUCHDLL_API uint __cdecl soundtouch_numUnprocessedSamples(HANDLE h)
{
STHANDLE *sth = (STHANDLE*)h;
if (sth->dwMagic != STMAGIC) return 0;
return sth->pst->numUnprocessedSamples();
}
/// Receive ready samples from the processing pipeline.
///
/// if called with outBuffer=nullptr, just reduces amount of ready samples within the pipeline.
SOUNDTOUCHDLL_API uint __cdecl soundtouch_receiveSamples(HANDLE h,
SAMPLETYPE *outBuffer, ///< Buffer where to copy output samples.
unsigned int maxSamples ///< How many samples to receive at max.
)
{
STHANDLE *sth = (STHANDLE*)h;
if (sth->dwMagic != STMAGIC) return 0;
if (outBuffer)
{
return sth->pst->receiveSamples(outBuffer, maxSamples);
}
else
{
return sth->pst->receiveSamples(maxSamples);
}
}
/// int16 version of soundtouch_receiveSamples(): This converts internal float samples
/// into int16 (short) return data type
SOUNDTOUCHDLL_API uint __cdecl soundtouch_receiveSamples_i16(HANDLE h,
short *outBuffer, ///< Buffer where to copy output samples.
unsigned int maxSamples ///< How many samples to receive at max.
)
{
STHANDLE *sth = (STHANDLE*)h;
if (sth->dwMagic != STMAGIC) return 0;
uint outTotal = 0;
if (outBuffer == nullptr)
{
// only reduce sample count, not receive samples
return sth->pst->receiveSamples(maxSamples);
}
uint numChannels = sth->pst->numChannels();
// iterate until all samples converted & put to SoundTouch object
while (maxSamples > 0)
{
float convert[8192]; // allocate temporary conversion buffer from stack
// how many multichannel samples fit into 'convert' buffer:
uint convSamples = 8192 / numChannels;
// request max 'nround' values at a time to guarantee that these fit in the 'convert' buffer
uint n = (maxSamples > convSamples) ? convSamples : maxSamples;
uint out = sth->pst->receiveSamples(convert, n);
// convert & saturate received samples to int16
for (uint i = 0; i < out * numChannels; i++)
{
// first convert value to int32, then saturate to int16 min/max limits
int value = (int)convert[i];
value = (value < SHRT_MIN) ? SHRT_MIN : (value > SHRT_MAX) ? SHRT_MAX : value;
outBuffer[i] = (short)value;
}
outTotal += out;
if (out < n) break; // didn't get as many as asked => no more samples available => break here
maxSamples -= n;
outBuffer += out * numChannels;
}
// return number of processed samples
return outTotal;
}
/// Returns number of samples currently available.
SOUNDTOUCHDLL_API uint __cdecl soundtouch_numSamples(HANDLE h)
{
STHANDLE *sth = (STHANDLE*)h;
if (sth->dwMagic != STMAGIC) return 0;
return sth->pst->numSamples();
}
/// Returns nonzero if there aren't any samples available for outputting.
SOUNDTOUCHDLL_API int __cdecl soundtouch_isEmpty(HANDLE h)
{
STHANDLE *sth = (STHANDLE*)h;
if (sth->dwMagic != STMAGIC) return -1;
return sth->pst->isEmpty();
}
SOUNDTOUCHDLL_API HANDLE __cdecl bpm_createInstance(int numChannels, int sampleRate)
{
BPMHANDLE *tmp = new BPMHANDLE;
if (tmp)
{
tmp->dwMagic = BPMMAGIC;
try
{
tmp->pbpm = new BPMDetect(numChannels, sampleRate);
}
catch (const std::exception&)
{
tmp->pbpm = nullptr;
}
if (tmp->pbpm == nullptr)
{
delete tmp;
tmp = nullptr;
}
}
return (HANDLE)tmp;
}
SOUNDTOUCHDLL_API void __cdecl bpm_destroyInstance(HANDLE h)
{
BPMHANDLE *sth = (BPMHANDLE*)h;
if (sth->dwMagic != BPMMAGIC) return;
sth->dwMagic = 0;
if (sth->pbpm) delete sth->pbpm;
sth->pbpm = nullptr;
delete sth;
}
/// Feed 'numSamples' sample frames from 'samples' into the BPM detection handler
SOUNDTOUCHDLL_API void __cdecl bpm_putSamples(HANDLE h,
const float *samples,
unsigned int numSamples)
{
BPMHANDLE *bpmh = (BPMHANDLE*)h;
if (bpmh->dwMagic != BPMMAGIC) return;
bpmh->pbpm->inputSamples(samples, numSamples);
}
/// Feed 'numSamples' sample frames from 'samples' into the BPM detection handler.
/// 16bit int sample format version.
SOUNDTOUCHDLL_API void __cdecl bpm_putSamples_i16(HANDLE h,
const short *samples,
unsigned int numSamples)
{
BPMHANDLE *bpmh = (BPMHANDLE*)h;
if (bpmh->dwMagic != BPMMAGIC) return;
uint numChannels = bpmh->numChannels;
// iterate until all samples converted & put to SoundTouch object
while (numSamples > 0)
{
float convert[8192]; // allocate temporary conversion buffer from stack
// how many multichannel samples fit into 'convert' buffer:
uint convSamples = 8192 / numChannels;
// convert max 'nround' values at a time to guarantee that these fit in the 'convert' buffer
uint n = (numSamples > convSamples) ? convSamples : numSamples;
for (uint i = 0; i < n * numChannels; i++)
{
convert[i] = samples[i];
}
// put the converted samples into SoundTouch
bpmh->pbpm->inputSamples(convert, n);
numSamples -= n;
samples += n * numChannels;
}
}
/// 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)
{
BPMHANDLE *bpmh = (BPMHANDLE*)h;
if (bpmh->dwMagic != BPMMAGIC) return 0;
return bpmh->pbpm->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.
SOUNDTOUCHDLL_API int __cdecl bpm_getBeats(HANDLE h, float* pos, float* strength, int count)
{
BPMHANDLE *bpmh = (BPMHANDLE *)h;
if (bpmh->dwMagic != BPMMAGIC) return 0;
return bpmh->pbpm->getBeats(pos, strength, count);
}

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//////////////////////////////////////////////////////////////////////////////
///
/// SoundTouch DLL wrapper - wraps SoundTouch routines into a Dynamic Load
/// Library interface.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// 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
// 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 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_

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externals/soundtouch/source/SoundTouchDLL/SoundTouchDLL.rc vendored Normal file
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@@ -0,0 +1,100 @@
// Microsoft Visual C++ generated resource script.
//
#include "resource.h"
#define APSTUDIO_READONLY_SYMBOLS
/////////////////////////////////////////////////////////////////////////////
//
// Generated from the TEXTINCLUDE 2 resource.
//
#include "afxres.h"
/////////////////////////////////////////////////////////////////////////////
#undef APSTUDIO_READONLY_SYMBOLS
/////////////////////////////////////////////////////////////////////////////
// English (United States) resources
#if !defined(AFX_RESOURCE_DLL) || defined(AFX_TARG_ENU)
LANGUAGE LANG_ENGLISH, SUBLANG_ENGLISH_US
#pragma code_page(1252)
#ifdef APSTUDIO_INVOKED
/////////////////////////////////////////////////////////////////////////////
//
// TEXTINCLUDE
//
1 TEXTINCLUDE
BEGIN
"resource.h\0"
END
2 TEXTINCLUDE
BEGIN
"#include ""afxres.h""\r\n"
"\0"
END
3 TEXTINCLUDE
BEGIN
"\r\n"
"\0"
END
#endif // APSTUDIO_INVOKED
/////////////////////////////////////////////////////////////////////////////
//
// Version
//
VS_VERSION_INFO VERSIONINFO
FILEVERSION 2,2,0,0
PRODUCTVERSION 2,2,0,0
FILEFLAGSMASK 0x17L
#ifdef _DEBUG
FILEFLAGS 0x1L
#else
FILEFLAGS 0x0L
#endif
FILEOS 0x4L
FILETYPE 0x2L
FILESUBTYPE 0x0L
BEGIN
BLOCK "StringFileInfo"
BEGIN
BLOCK "000004b0"
BEGIN
VALUE "Comments", "SoundTouch Library licensed for 3rd party applications subject to LGPL license v2.1. Visit http://www.surina.net/soundtouch for more information about the SoundTouch library."
VALUE "FileDescription", "SoundTouch Dynamic Link Library"
VALUE "FileVersion", "2.3.2.0"
VALUE "InternalName", "SoundTouch"
VALUE "LegalCopyright", "Copyright (C) Olli Parviainen 2022"
VALUE "OriginalFilename", "SoundTouch.dll"
VALUE "ProductName", " SoundTouch Dynamic Link Library"
VALUE "ProductVersion", "2.3.2.0"
END
END
BLOCK "VarFileInfo"
BEGIN
VALUE "Translation", 0x0, 1200
END
END
#endif // English (United States) resources
/////////////////////////////////////////////////////////////////////////////
#ifndef APSTUDIO_INVOKED
/////////////////////////////////////////////////////////////////////////////
//
// Generated from the TEXTINCLUDE 3 resource.
//
/////////////////////////////////////////////////////////////////////////////
#endif // not APSTUDIO_INVOKED

50
externals/soundtouch/source/SoundTouchDLL/SoundTouchDLL.sln vendored Normal file
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@@ -0,0 +1,50 @@
Microsoft Visual Studio Solution File, Format Version 12.00
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MinimumVisualStudioVersion = 10.0.40219.1
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ProjectSection(ProjectDependencies) = postProject
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GlobalSection(SolutionConfigurationPlatforms) = preSolution
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Debug|x64 = Debug|x64
Release|Win32 = Release|Win32
Release|x64 = Release|x64
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{68A5DD20-7057-448B-8FE0-B6AC8D205509}.Release|Win32.Build.0 = Release|Win32
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{68A5DD20-7057-448B-8FE0-B6AC8D205509}.Release|x64.Build.0 = Release|x64
{E3C0726F-28F4-4F0B-8183-B87CA60C063C}.Debug|Win32.ActiveCfg = Debug|Win32
{E3C0726F-28F4-4F0B-8183-B87CA60C063C}.Debug|Win32.Build.0 = Debug|Win32
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{E3C0726F-28F4-4F0B-8183-B87CA60C063C}.Debug|x64.Build.0 = Debug|x64
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{E3C0726F-28F4-4F0B-8183-B87CA60C063C}.Release|Win32.Build.0 = Release|Win32
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{E3C0726F-28F4-4F0B-8183-B87CA60C063C}.Release|x64.Build.0 = Release|x64
EndGlobalSection
GlobalSection(SolutionProperties) = preSolution
HideSolutionNode = FALSE
EndGlobalSection
EndGlobal

273
externals/soundtouch/source/SoundTouchDLL/SoundTouchDLL.vcxproj vendored Normal file
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</ItemGroup>
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<Text Include="ReadMe.txt" />
</ItemGroup>
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29
externals/soundtouch/source/SoundTouchDLL/make-gnu-dll.sh vendored Normal file
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#!/bin/bash
#
# 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
#
arch=$(uname -m)
flags=""
if [[ $arch == *"86"* ]]; then
# Intel x86/x64 architecture
flags="$flags -mstackrealign -msse"
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 -ffast-math -shared $flags -DDLL_EXPORTS -fvisibility=hidden -I../../include \
-I../SoundTouch -o SoundTouchDll.so SoundTouchDLL.cpp ../SoundTouch/*.cpp

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

9
externals/soundtouch/source/csharp-example/App.xaml vendored Normal file
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<Application x:Class="csharp_example.App"
xmlns="http://schemas.microsoft.com/winfx/2006/xaml/presentation"
xmlns:x="http://schemas.microsoft.com/winfx/2006/xaml"
xmlns:local="clr-namespace:csharp_example"
StartupUri="MainWindow.xaml">
<Application.Resources>
</Application.Resources>
</Application>

25
externals/soundtouch/source/csharp-example/App.xaml.cs vendored Normal file
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@@ -0,0 +1,25 @@
//////////////////////////////////////////////////////////////////////////////
///
/// C# example that manipulates mp3 audio files with SoundTouch library.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// License for this source code file: Microsoft Public License(Ms-PL)
//
////////////////////////////////////////////////////////////////////////////////
using System.Windows;
namespace csharp_example
{
/// <summary>
/// Interaction logic for App.xaml
/// </summary>
public partial class App : Application
{
}
}

31
externals/soundtouch/source/csharp-example/MainWindow.xaml vendored Normal file
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@@ -0,0 +1,31 @@
<Window x:Class="csharp_example.MainWindow"
xmlns="http://schemas.microsoft.com/winfx/2006/xaml/presentation"
xmlns:x="http://schemas.microsoft.com/winfx/2006/xaml"
xmlns:d="http://schemas.microsoft.com/expression/blend/2008"
xmlns:mc="http://schemas.openxmlformats.org/markup-compatibility/2006"
xmlns:local="clr-namespace:csharp_example"
mc:Ignorable="d"
Title="C# SoundTouch example" Height="250" Width="400" AllowDrop="True" Drop="Window_Drop">
<Grid Margin="0,0,0,-3">
<TextBlock HorizontalAlignment="Left" Margin="10,21,0,0" Text="Input audio file:" VerticalAlignment="Top"/>
<TextBox x:Name="textBox_filename" Height="23" Margin="107,20,0,0" VerticalAlignment="Top" HorizontalAlignment="Left" Width="180" IsEnabled="False"/>
<Button x:Name="button_browse" Content="_Browse" HorizontalAlignment="Left" Margin="292,19,0,0" VerticalAlignment="Top" Width="75" Height="24" Click="button_browse_Click"/>
<TextBlock HorizontalAlignment="Left" Margin="54,55,0,0" Text="Tempo:" VerticalAlignment="Top" TextAlignment="Right"/>
<TextBox x:Name="textBox_tempo" HorizontalAlignment="Left" Height="23" Margin="107,55,0,0" Text="0" VerticalAlignment="Top" Width="75" TextAlignment="Center" LostFocus="textBox_tempo_LostFocus" KeyDown="textBox_tempo_KeyDown"/>
<TextBlock HorizontalAlignment="Left" Margin="185,55,0,0" Text="%" VerticalAlignment="Top" TextAlignment="Right"/>
<TextBlock HorizontalAlignment="Left" Margin="64,85,0,0" Text="Pitch:" VerticalAlignment="Top" TextAlignment="Right"/>
<TextBox x:Name="textBox_pitch" HorizontalAlignment="Left" Height="23" Margin="107,85,0,0" Text="0" VerticalAlignment="Top" Width="75" TextAlignment="Center" LostFocus="textBox_pitch_LostFocus" KeyDown="textBox_pitch_KeyDown"/>
<TextBlock HorizontalAlignment="Left" Margin="185,85,0,0" Text="semitones" VerticalAlignment="Top" TextAlignment="Right"/>
<TextBlock HorizontalAlignment="Left" Margin="66,116,0,0" Text="Rate:" VerticalAlignment="Top" TextAlignment="Right"/>
<TextBox x:Name="textBox_rate" HorizontalAlignment="Left" Height="23" Margin="107,116,0,0" Text="0" VerticalAlignment="Top" Width="75" TextAlignment="Center" LostFocus="textBox_rate_LostFocus" KeyDown="textBox_rate_KeyDown"/>
<TextBlock HorizontalAlignment="Left" Margin="185,116,0,0" Text="%" VerticalAlignment="Top" TextAlignment="Right"/>
<Button x:Name="button_play" Content="_Play" Margin="107,150,0,0" VerticalAlignment="Top" Height="24" HorizontalAlignment="Left" Width="75" Click="button_play_Click" IsEnabled="False"/>
<Button x:Name="button_stop" Content="_Stop" Margin="200,150,0,0" VerticalAlignment="Top" Height="24" HorizontalAlignment="Left" Width="75" Click="button_stop_Click" IsEnabled="False"/>
<StatusBar VerticalAlignment="Bottom">
<TextBlock x:Name="text_status" HorizontalAlignment="Left" Margin="2,0,0,2"/>
</StatusBar>
</Grid>
</Window>

258
externals/soundtouch/source/csharp-example/MainWindow.xaml.cs vendored Normal file
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@@ -0,0 +1,258 @@
//////////////////////////////////////////////////////////////////////////////
///
/// C# example that manipulates mp3 audio files with SoundTouch library.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// License for this source code file: Microsoft Public License(Ms-PL)
//
////////////////////////////////////////////////////////////////////////////////
using soundtouch;
using System;
using System.IO;
using System.Windows;
using System.Windows.Controls;
using System.Windows.Input;
namespace csharp_example
{
/// <summary>
/// Interaction logic for MainWindow.xaml
/// </summary>
public partial class MainWindow : Window
{
protected SoundProcessor processor = new SoundProcessor();
public MainWindow()
{
InitializeComponent();
StatusMessage.statusEvent += StatusEventHandler;
processor.PlaybackStopped += EventHandler_playbackStopped;
DisplaySoundTouchVersion();
}
/// <summary>
/// Display SoundTouch library version string in status bar. This also indicates whether the DLL was loaded successfully or not ...
/// </summary>
private void DisplaySoundTouchVersion()
{
string status;
try
{
status = String.Format("SoundTouch version: {0}", SoundTouch.Version);
}
catch (Exception exp)
{
status = exp.Message;
}
text_status.Text = status;
}
private void StatusEventHandler(object sender, string msg)
{
text_status.Text = msg;
}
// Open mp3 file for playback
private void OpenFile(string fileName)
{
Stop();
if (processor.OpenMp3File(fileName) == true)
{
textBox_filename.Text = fileName;
button_play.IsEnabled = true;
button_stop.IsEnabled = true;
// Parse adjustment settings
ParseTempoTextBox();
ParsePitchTextBox();
ParseRateTextBox();
}
else
{
textBox_filename.Text = "";
button_play.IsEnabled = false;
button_stop.IsEnabled = false;
MessageBox.Show("Coudln't open audio file " + fileName);
}
}
private void button_browse_Click(object sender, RoutedEventArgs e)
{
// Show file selection dialog
Microsoft.Win32.OpenFileDialog openDialog = new Microsoft.Win32.OpenFileDialog();
if (string.IsNullOrEmpty(textBox_filename.Text) == false)
{
// if an audio file is open, set directory to same as with the file
openDialog.InitialDirectory = Path.GetDirectoryName(textBox_filename.Text);
}
openDialog.Filter = "MP3 files (*.mp3)|*.mp3";
if (openDialog.ShowDialog() == true)
{
OpenFile(openDialog.FileName);
}
}
private void setPlayButtonMode(bool play)
{
button_play.Content = play ? "_Play" : "_Pause";
}
private void EventHandler_playbackStopped(object sender, bool hasReachedEnd)
{
if (hasReachedEnd)
{
text_status.Text = "Stopped";
} // otherwise paused
setPlayButtonMode(true);
}
private void button_play_Click(object sender, RoutedEventArgs e)
{
if ((string)button_play.Content == "_Pause")
{
// Pause
if (processor.Pause())
{
text_status.Text = "Paused";
}
setPlayButtonMode(true);
}
else
{
// Play
if (processor.Play())
{
text_status.Text = "Playing";
setPlayButtonMode(false);
}
}
}
private void Stop()
{
if (processor.Stop())
{
text_status.Text = "Stopped";
}
setPlayButtonMode(true);
}
private void button_stop_Click(object sender, RoutedEventArgs e)
{
Stop();
}
private bool parse_percentValue(TextBox box, out double value)
{
if (double.TryParse(box.Text, out value) == false) return false;
if (value < -99.0) value = -99.0; // don't allow more than -100% slowdown ... :)
box.Text = value.ToString();
return true;
}
private void ParsePitchTextBox()
{
double pitchValue;
if (double.TryParse(textBox_pitch.Text, out pitchValue))
{
if (processor.streamProcessor != null) processor.streamProcessor.st.PitchSemiTones = (float)pitchValue;
}
}
private void ParseTempoTextBox()
{
double tempoValue;
if (parse_percentValue(textBox_tempo, out tempoValue))
{
if (processor.streamProcessor != null) processor.streamProcessor.st.TempoChange = (float)tempoValue;
}
}
private void ParseRateTextBox()
{
double rateValue;
if (parse_percentValue(textBox_rate, out rateValue))
{
if (processor.streamProcessor != null) processor.streamProcessor.st.RateChange = (float)rateValue;
}
}
private void textBox_tempo_LostFocus(object sender, RoutedEventArgs e)
{
ParseTempoTextBox();
}
private void textBox_tempo_KeyDown(object sender, KeyEventArgs e)
{
if (e.Key == Key.Enter)
{
// enter pressed -- parse value
ParseTempoTextBox();
}
}
private void textBox_pitch_LostFocus(object sender, RoutedEventArgs e)
{
ParsePitchTextBox();
}
private void textBox_pitch_KeyDown(object sender, KeyEventArgs e)
{
if (e.Key == Key.Enter)
{
// enter pressed -- parse value
ParsePitchTextBox();
}
}
private void textBox_rate_LostFocus(object sender, RoutedEventArgs e)
{
ParseRateTextBox();
}
private void textBox_rate_KeyDown(object sender, KeyEventArgs e)
{
if (e.Key == Key.Enter)
{
// enter pressed -- parse value
ParseRateTextBox();
}
}
// Handler for file drag & drop over the window
private void Window_Drop(object sender, DragEventArgs e)
{
string[] files = (string[])e.Data.GetData(DataFormats.FileDrop);
// open 1st of the chosen files
OpenFile(files[0]);
}
}
}

31
externals/soundtouch/source/csharp-example/NAudio-license.txt vendored Normal file
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Microsoft Public License (Ms-PL)
This license governs use of the accompanying software. If you use the software, you accept this license. If you do not accept the license, do not use the software.
1. Definitions
The terms "reproduce," "reproduction," "derivative works," and "distribution" have the same meaning here as under U.S. copyright law.
A "contribution" is the original software, or any additions or changes to the software.
A "contributor" is any person that distributes its contribution under this license.
"Licensed patents" are a contributor's patent claims that read directly on its contribution.
2. Grant of Rights
(A) Copyright Grant- Subject to the terms of this license, including the license conditions and limitations in section 3, each contributor grants you a non-exclusive, worldwide, royalty-free copyright license to reproduce its contribution, prepare derivative works of its contribution, and distribute its contribution or any derivative works that you create.
(B) Patent Grant- Subject to the terms of this license, including the license conditions and limitations in section 3, each contributor grants you a non-exclusive, worldwide, royalty-free license under its licensed patents to make, have made, use, sell, offer for sale, import, and/or otherwise dispose of its contribution in the software or derivative works of the contribution in the software.
3. Conditions and Limitations
(A) No Trademark License- This license does not grant you rights to use any contributors' name, logo, or trademarks.
(B) If you bring a patent claim against any contributor over patents that you claim are infringed by the software, your patent license from such contributor to the software ends automatically.
(C) If you distribute any portion of the software, you must retain all copyright, patent, trademark, and attribution notices that are present in the software.
(D) If you distribute any portion of the software in source code form, you may do so only under this license by including a complete copy of this license with your distribution. If you distribute any portion of the software in compiled or object code form, you may only do so under a license that complies with this license.
(E) The software is licensed "as-is." You bear the risk of using it. The contributors give no express warranties, guarantees or conditions. You may have additional consumer rights under your local laws which this license cannot change. To the extent permitted under your local laws, the contributors exclude the implied warranties of merchantability, fitness for a particular purpose and non-infringement.

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externals/soundtouch/source/csharp-example/NAudio-readme.txt vendored Normal file
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NAudio is an open source .NET audio library written by Mark Heath (mark.heath@gmail.com)
For more information, visit http://naudio.codeplex.com
NAudio is now being hosted on GitHub http://github.com/naudio/NAudio
THANKS
======
The following list includes some of the people who have contributed in various ways to NAudio, such as code contributions,
bug fixes, documentation, helping out on the forums and even donations. I haven't finished compiling this list yet, so
if your name should be on it but isn't please let me know and I will include it. Also, some people I only know by their forum
id, so if you want me to put your full name here, please also get in touch.
in alphabetical order:
Alan Jordan
Alexandre Mutel
Alexander Binkert
AmandaTarafaMas
balistof
biermeester
borman11
bradb
Brandon Hansen (kg6ypi)
csechet
ChunkWare Music Software
CKing
DaMacc
Dirk Eckhardt
Du10
eejake52
Florian Rosmann (filoe)
Freefall
Giawa
Harald Petrovitsch
Hfuy
Iain McCowan
Idael Cardaso
ioctlLR
Ivan Kochurkin (KvanTTT)
Jamie Michael Ewins
jannera
jbaker8935
jcameron23
JoeGaggler
jonahoffmann
jontdelorme
Jospin Software
Justin Frankel
K24A3
Kamen Lichev
Kassoul
kevinxxx
kzych
LionCash
Lustild
Lucian Wischik (ljw1004)
ManuN
MeelMarcel
Michael Chadwick
Michael Feld
Michael J
Michael Lehenbauer
milligan22963
myrkle
nelsonkidd
Nigel Redmon
Nikolaos Georgiou
Owen Skriloff
owoudenb
painmailer
PPavan
Pygmy
Ray Molenkamp
Roadz
Robert Bristow-Johnson
Scott Fleischman
Simon Clark
Sirish Bajpai
sporn
Steve Underwood
Ted Murphy
Tiny Simple Tools
Tobias Fleming
TomBogle
Tony Cabello
Tony Sistemas
TuneBlade
topher3683
volmart
Vladimir Rokovanov
Ville Koskinen
Wyatt Rice
Yuval Naveh
Zsb

55
externals/soundtouch/source/csharp-example/Properties/AssemblyInfo.cs vendored Normal file
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using System.Reflection;
using System.Resources;
using System.Runtime.CompilerServices;
using System.Runtime.InteropServices;
using System.Windows;
// 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")]
[assembly: AssemblyDescription("")]
[assembly: AssemblyConfiguration("")]
[assembly: AssemblyCompany("")]
[assembly: AssemblyProduct("csharp-example")]
[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
// COM, set the ComVisible attribute to true on that type.
[assembly: ComVisible(false)]
//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
//the NeutralResourceLanguage attribute below. Update the "en-US" in
//the line below to match the UICulture setting in the project file.
//[assembly: NeutralResourcesLanguage("en-US", UltimateResourceFallbackLocation.Satellite)]
[assembly: ThemeInfo(
ResourceDictionaryLocation.None, //where theme specific resource dictionaries are located
//(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,
// app, or any theme specific resource dictionaries)
)]
// Version information for an assembly consists of the following four values:
//
// Major Version
// Minor Version
// Build Number
// Revision
//
// 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")]
[assembly: AssemblyFileVersion("1.0.0.0")]

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externals/soundtouch/source/csharp-example/Properties/Resources.Designer.cs generated vendored Normal file
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//------------------------------------------------------------------------------
// <auto-generated>
// This code was generated by a tool.
// Runtime Version:4.0.30319.42000
//
// Changes to this file may cause incorrect behavior and will be lost if
// the code is regenerated.
// </auto-generated>
//------------------------------------------------------------------------------
namespace csharp_example.Properties
{
/// <summary>
/// A strongly-typed resource class, for looking up localized strings, etc.
/// </summary>
// This class was auto-generated by the StronglyTypedResourceBuilder
// 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.Diagnostics.DebuggerNonUserCodeAttribute()]
[global::System.Runtime.CompilerServices.CompilerGeneratedAttribute()]
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()
{
}
/// <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))
{
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
{
return resourceCulture;
}
set
{
resourceCulture = value;
}
}
}
}

117
externals/soundtouch/source/csharp-example/Properties/Resources.resx vendored Normal file
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<?xml version="1.0" encoding="utf-8"?>
<root>
<!--
Microsoft ResX Schema
Version 2.0
The primary goals of this format is to allow a simple XML format
that is mostly human readable. The generation and parsing of the
various data types are done through the TypeConverter classes
associated with the data types.
Example:
... ado.net/XML headers & schema ...
<resheader name="resmimetype">text/microsoft-resx</resheader>
<resheader name="version">2.0</resheader>
<resheader name="reader">System.Resources.ResXResourceReader, System.Windows.Forms, ...</resheader>
<resheader name="writer">System.Resources.ResXResourceWriter, System.Windows.Forms, ...</resheader>
<data name="Name1"><value>this is my long string</value><comment>this is a comment</comment></data>
<data name="Color1" type="System.Drawing.Color, System.Drawing">Blue</data>
<data name="Bitmap1" mimetype="application/x-microsoft.net.object.binary.base64">
<value>[base64 mime encoded serialized .NET Framework object]</value>
</data>
<data name="Icon1" type="System.Drawing.Icon, System.Drawing" mimetype="application/x-microsoft.net.object.bytearray.base64">
<value>[base64 mime encoded string representing a byte array form of the .NET Framework object]</value>
<comment>This is a comment</comment>
</data>
There are any number of "resheader" rows that contain simple
name/value pairs.
Each data row contains a name, and value. The row also contains a
type or mimetype. Type corresponds to a .NET class that support
text/value conversion through the TypeConverter architecture.
Classes that don't support this are serialized and stored with the
mimetype set.
The mimetype is used for serialized objects, and tells the
ResXResourceReader how to depersist the object. This is currently not
extensible. For a given mimetype the value must be set accordingly:
Note - application/x-microsoft.net.object.binary.base64 is the format
that the ResXResourceWriter will generate, however the reader can
read any of the formats listed below.
mimetype: application/x-microsoft.net.object.binary.base64
value : The object must be serialized with
: System.Serialization.Formatters.Binary.BinaryFormatter
: and then encoded with base64 encoding.
mimetype: application/x-microsoft.net.object.soap.base64
value : The object must be serialized with
: System.Runtime.Serialization.Formatters.Soap.SoapFormatter
: and then encoded with base64 encoding.
mimetype: application/x-microsoft.net.object.bytearray.base64
value : The object must be serialized into a byte array
: using a System.ComponentModel.TypeConverter
: and then encoded with base64 encoding.
-->
<xsd:schema id="root" xmlns="" xmlns:xsd="http://www.w3.org/2001/XMLSchema" xmlns:msdata="urn:schemas-microsoft-com:xml-msdata">
<xsd:element name="root" msdata:IsDataSet="true">
<xsd:complexType>
<xsd:choice maxOccurs="unbounded">
<xsd:element name="metadata">
<xsd:complexType>
<xsd:sequence>
<xsd:element name="value" type="xsd:string" minOccurs="0" />
</xsd:sequence>
<xsd:attribute name="name" type="xsd:string" />
<xsd:attribute name="type" type="xsd:string" />
<xsd:attribute name="mimetype" type="xsd:string" />
</xsd:complexType>
</xsd:element>
<xsd:element name="assembly">
<xsd:complexType>
<xsd:attribute name="alias" type="xsd:string" />
<xsd:attribute name="name" type="xsd:string" />
</xsd:complexType>
</xsd:element>
<xsd:element name="data">
<xsd:complexType>
<xsd:sequence>
<xsd:element name="value" type="xsd:string" minOccurs="0" msdata:Ordinal="1" />
<xsd:element name="comment" type="xsd:string" minOccurs="0" msdata:Ordinal="2" />
</xsd:sequence>
<xsd:attribute name="name" type="xsd:string" msdata:Ordinal="1" />
<xsd:attribute name="type" type="xsd:string" msdata:Ordinal="3" />
<xsd:attribute name="mimetype" type="xsd:string" msdata:Ordinal="4" />
</xsd:complexType>
</xsd:element>
<xsd:element name="resheader">
<xsd:complexType>
<xsd:sequence>
<xsd:element name="value" type="xsd:string" minOccurs="0" msdata:Ordinal="1" />
</xsd:sequence>
<xsd:attribute name="name" type="xsd:string" use="required" />
</xsd:complexType>
</xsd:element>
</xsd:choice>
</xsd:complexType>
</xsd:element>
</xsd:schema>
<resheader name="resmimetype">
<value>text/microsoft-resx</value>
</resheader>
<resheader name="version">
<value>2.0</value>
</resheader>
<resheader name="reader">
<value>System.Resources.ResXResourceReader, System.Windows.Forms, Version=2.0.0.0, Culture=neutral, PublicKeyToken=b77a5c561934e089</value>
</resheader>
<resheader name="writer">
<value>System.Resources.ResXResourceWriter, System.Windows.Forms, Version=2.0.0.0, Culture=neutral, PublicKeyToken=b77a5c561934e089</value>
</resheader>
</root>

30
externals/soundtouch/source/csharp-example/Properties/Settings.Designer.cs generated vendored Normal file
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//------------------------------------------------------------------------------
// <auto-generated>
// This code was generated by a tool.
// Runtime Version:4.0.30319.42000
//
// Changes to this file may cause incorrect behavior and will be lost if
// the code is regenerated.
// </auto-generated>
//------------------------------------------------------------------------------
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
{
private static Settings defaultInstance = ((Settings)(global::System.Configuration.ApplicationSettingsBase.Synchronized(new Settings())));
public static Settings Default
{
get
{
return defaultInstance;
}
}
}
}

7
externals/soundtouch/source/csharp-example/Properties/Settings.settings vendored Normal file
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@@ -0,0 +1,7 @@
<?xml version='1.0' encoding='utf-8'?>
<SettingsFile xmlns="uri:settings" CurrentProfile="(Default)">
<Profiles>
<Profile Name="(Default)" />
</Profiles>
<Settings />
</SettingsFile>

9
externals/soundtouch/source/csharp-example/README.txt vendored Normal file
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csharp-example
==============
Copyright (c) Olli Parviainen
This is a C# example application that plays MP3 audio file and
uses SoundTouch.dll library for adjusting the sound in real-time.
The example uses NAudio library for MP3 file reading and audio playback.
See NAudio-readme.txt and NAudio-license.txt for NAudio information.

292
externals/soundtouch/source/csharp-example/SoundProcessor.cs vendored Normal file
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//////////////////////////////////////////////////////////////////////////////
///
/// WaveStream processor class for manipulating audio stream in C# with
/// SoundTouch library.
///
/// This module uses NAudio library for C# audio file input / output
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// License for this source code file: Microsoft Public License(Ms-PL)
//
////////////////////////////////////////////////////////////////////////////////
using NAudio.Wave;
using soundtouch;
using System;
namespace csharp_example
{
/// <summary>
/// Helper class that allow writing status texts to the host application
/// </summary>
public class StatusMessage
{
/// <summary>
/// Handler for status message events. Subscribe this from the host application
/// </summary>
public static event EventHandler<string> statusEvent;
/// <summary>
/// Pass a status message to the host application
/// </summary>
public static void Write(string msg)
{
if (statusEvent != null)
{
statusEvent(null, msg);
}
}
}
/// <summary>
/// NAudui WaveStream class for processing audio stream with SoundTouch effects
/// </summary>
public class WaveStreamProcessor : WaveStream
{
private WaveChannel32 inputStr;
public SoundTouch st;
private byte[] bytebuffer = new byte[4096];
private float[] floatbuffer = new float[1024];
bool endReached = false;
/// <summary>
/// Constructor
/// </summary>
/// <param name="input">WaveChannel32 stream used for processor stream input</param>
public WaveStreamProcessor(WaveChannel32 input)
{
inputStr = input;
st = new SoundTouch();
st.Channels = (uint)input.WaveFormat.Channels;
st.SampleRate = (uint)input.WaveFormat.SampleRate;
}
/// <summary>
/// True if end of stream reached
/// </summary>
public bool EndReached
{
get { return endReached; }
}
public override long Length
{
get
{
return inputStr.Length;
}
}
public override long Position
{
get
{
return inputStr.Position;
}
set
{
inputStr.Position = value;
}
}
public override WaveFormat WaveFormat
{
get
{
return inputStr.WaveFormat;
}
}
/// <summary>
/// Overridden Read function that returns samples processed with SoundTouch. Returns data in same format as
/// WaveChannel32 i.e. stereo float samples.
/// </summary>
/// <param name="buffer">Buffer where to return sample data</param>
/// <param name="offset">Offset from beginning of the buffer</param>
/// <param name="count">Number of bytes to return</param>
/// <returns>Number of bytes copied to buffer</returns>
public override int Read(byte[] buffer, int offset, int count)
{
try
{
// Iterate until enough samples available for output:
// - read samples from input stream
// - put samples to SoundStretch processor
while (st.AvailableSampleCount < count)
{
int nbytes = inputStr.Read(bytebuffer, 0, bytebuffer.Length);
if (nbytes == 0)
{
// end of stream. flush final samples from SoundTouch buffers to output
if (endReached == false)
{
endReached = true; // do only once to avoid continuous flushing
st.Flush();
}
break;
}
// binary copy data from "byte[]" to "float[]" buffer
Buffer.BlockCopy(bytebuffer, 0, floatbuffer, 0, nbytes);
st.PutSamples(floatbuffer, (uint)(nbytes / 8));
}
// ensure that buffer is large enough to receive desired amount of data out
if (floatbuffer.Length < count / 4)
{
floatbuffer = new float[count / 4];
}
// get processed output samples from SoundTouch
int numsamples = (int)st.ReceiveSamples(floatbuffer, (uint)(count / 8));
// binary copy data from "float[]" to "byte[]" buffer
Buffer.BlockCopy(floatbuffer, 0, buffer, offset, numsamples * 8);
return numsamples * 8; // number of bytes
}
catch (Exception exp)
{
StatusMessage.Write("exception in WaveStreamProcessor.Read: " + exp.Message);
return 0;
}
}
/// <summary>
/// Clear the internal processor buffers. Call this if seeking or rewinding to new position within the stream.
/// </summary>
public void Clear()
{
st.Clear();
endReached = false;
}
}
/// <summary>
/// Class that opens & plays MP3 file and allows real-time audio processing with SoundTouch
/// while playing
/// </summary>
public class SoundProcessor
{
Mp3FileReader mp3File;
WaveOut waveOut;
public WaveStreamProcessor streamProcessor;
/// <summary>
/// Start / resume playback
/// </summary>
/// <returns>true if successful, false if audio file not open</returns>
public bool Play()
{
if (waveOut == null) return false;
if (waveOut.PlaybackState != PlaybackState.Playing)
{
waveOut.Play();
}
return true;
}
/// <summary>
/// Pause playback
/// </summary>
/// <returns>true if successful, false if audio not playing</returns>
public bool Pause()
{
if (waveOut == null) return false;
if (waveOut.PlaybackState == PlaybackState.Playing)
{
waveOut.Stop();
return true;
}
return false;
}
/// <summary>
/// Stop playback
/// </summary>
/// <returns>true if successful, false if audio file not open</returns>
public bool Stop()
{
if (waveOut == null) return false;
waveOut.Stop();
mp3File.Position = 0;
streamProcessor.Clear();
return true;
}
/// <summary>
/// Event for "playback stopped" event. 'bool' argument is true if playback has reached end of stream.
/// </summary>
public event EventHandler<bool> PlaybackStopped;
/// <summary>
/// Proxy event handler for receiving playback stopped event from WaveOut
/// </summary>
protected void EventHandler_stopped(object sender, StoppedEventArgs args)
{
bool isEnd = streamProcessor.EndReached;
if (isEnd)
{
Stop();
}
if (PlaybackStopped != null)
{
PlaybackStopped(sender, isEnd);
}
}
/// <summary>
/// Open MP3 file
/// </summary>
/// <param name="filePath">Path to file to open</param>
/// <returns>true if successful</returns>
public bool OpenMp3File(string filePath)
{
try
{
mp3File = new Mp3FileReader(filePath);
WaveChannel32 inputStream = new WaveChannel32(mp3File);
inputStream.PadWithZeroes = false; // don't pad, otherwise the stream never ends
streamProcessor = new WaveStreamProcessor(inputStream);
waveOut = new WaveOut()
{
DesiredLatency = 100
};
waveOut.Init(streamProcessor); // inputStream);
waveOut.PlaybackStopped += EventHandler_stopped;
StatusMessage.Write("Opened file " + filePath);
return true;
}
catch (Exception exp)
{
// Error in opening file
waveOut = null;
StatusMessage.Write("Can't open file: " + exp.Message);
return false;
}
}
}
}

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externals/soundtouch/source/csharp-example/SoundTouch.cs vendored Normal file
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//////////////////////////////////////////////////////////////////////////////
///
/// C# wrapper to access SoundTouch APIs from an external SoundTouch.dll library
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
/// The C# wrapper improved by Mario Di Vece
///
////////////////////////////////////////////////////////////////////////////////
//
// 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
//
////////////////////////////////////////////////////////////////////////////////
using System;
using System.Runtime.InteropServices;
namespace soundtouch
{
public sealed class SoundTouch : IDisposable
{
#region Internal Members
internal const string SoundTouchLibrary = "SoundTouch.dll";
#endregion
#region Private Members // hahaha what a curious region
private readonly object SyncRoot = new object();
private bool IsDisposed = false;
private IntPtr handle;
#endregion
#region Constructor
/// <summary>
/// Initializes a new instance of the <see cref="SoundTouch"/> class.
/// </summary>
public SoundTouch()
{
handle = NativeMethods.CreateInstance();
}
/// <summary>
/// Finalizes an instance of the <see cref="SoundTouch"/> class.
/// </summary>
~SoundTouch()
{
// Do not change this code. Put cleanup code in Dispose(bool disposing) above.
Dispose(false);
}
/// <summary>
/// Settings as defined in SoundTouch.h
/// </summary>
public enum Setting
{
/// <summary>
/// Enable/disable anti-alias filter in pitch transposer (0 = disable)
/// </summary>
UseAntiAliasFilter = 0,
/// <summary>
/// Pitch transposer anti-alias filter length (8 .. 128 taps, default = 32)
/// </summary>
AntiAliasFilterLength = 1,
/// <summary>
/// Enable/disable quick seeking algorithm in tempo changer routine
/// (enabling quick seeking lowers CPU utilization but causes a minor sound
/// quality compromising)
/// </summary>
UseQuickSeek = 2,
/// <summary>
/// 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.
/// </summary>
SequenceMilliseconds = 3,
/// <summary>
/// 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.
/// </summary>
SeekWindowMilliseconds = 4,
/// <summary>
/// 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.
/// </summary>
OverlapMilliseconds = 5,
/// <summary>
/// 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.
/// </summary>
NominalInputSequence = 6,
/// <summary>
/// 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.
/// </summary>
NominalOutputSequence = 7,
/// <summary>
/// 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.
/// </summary>
InitialLatency = 8,
}
#endregion
#region Properties
/// <summary>
/// Get SoundTouch version string
/// </summary>
public static string Version
{
get
{
// convert "char *" data to c# string
return Marshal.PtrToStringAnsi(NativeMethods.GetVersionString());
}
}
/// <summary>
/// Gets a value indicating whether the SoundTouch Library (dll) is available
/// </summary>
public static bool IsAvailable
{
get
{
try
{
var versionId = NativeMethods.GetVersionId();
return versionId != 0;
}
catch
{
return false;
}
}
}
/// <summary>
/// Returns number of processed samples currently available in SoundTouch for immediate output.
/// </summary>
public uint AvailableSampleCount
{
get { lock (SyncRoot) { return NativeMethods.NumSamples(handle); } }
}
/// <summary>
/// Returns number of samples currently unprocessed in SoundTouch internal buffer
/// </summary>
/// <returns>Number of sample frames</returns>
public uint UnprocessedSampleCount
{
get { lock (SyncRoot) { return NativeMethods.NumUnprocessedSamples(handle); } }
}
/// <summary>
/// Check if there aren't any samples available for outputting.
/// </summary>
/// <returns>nonzero if there aren't any samples available for outputting</returns>
public int IsEmpty
{
get { lock (SyncRoot) { return NativeMethods.IsEmpty(handle); } }
}
/// <summary>
/// Sets the number of channels
///
/// Value: 1 = mono, 2 = stereo, n = multichannel
/// </summary>
public uint Channels
{
set { lock (SyncRoot) { NativeMethods.SetChannels(handle, value); } }
}
/// <summary>
/// Sets sample rate.
/// Value: Sample rate, e.g. 44100
/// </summary>
public uint SampleRate
{
set { lock (SyncRoot) { NativeMethods.SetSampleRate(handle, value); } }
}
/// <summary>
/// Sets new tempo control value.
///
/// Value: Tempo setting. Normal tempo = 1.0, smaller values
/// represent slower tempo, larger faster tempo.
/// </summary>
public float Tempo
{
set { lock (SyncRoot) { NativeMethods.SetTempo(handle, value); } }
}
/// <summary>
/// Sets new tempo control value as a difference in percents compared
/// to the original tempo (-50 .. +100 %);
/// </summary>
public float TempoChange
{
set { lock (SyncRoot) { NativeMethods.SetTempoChange(handle, value); } }
}
/// <summary>
/// Sets new rate control value.
/// Rate setting. Normal rate = 1.0, smaller values
/// represent slower rate, larger faster rate.
/// </summary>
public float Rate
{
set { lock (SyncRoot) { NativeMethods.SetTempo(handle, value); } }
}
/// <summary>
/// Sets new rate control value as a difference in percents compared
/// to the original rate (-50 .. +100 %);
///
/// Value: Rate setting is in %
/// </summary>
public float RateChange
{
set { lock (SyncRoot) { NativeMethods.SetRateChange(handle, value); } }
}
/// <summary>
/// Sets new pitch control value.
///
/// Value: Pitch setting. Original pitch = 1.0, smaller values
/// represent lower pitches, larger values higher pitch.
/// </summary>
public float Pitch
{
set { lock (SyncRoot) { NativeMethods.SetPitch(handle, value); } }
}
/// <summary>
/// Sets pitch change in octaves compared to the original pitch
/// (-1.00 .. +1.00 for +- one octave);
///
/// Value: Pitch setting in octaves
/// </summary>
public float PitchOctaves
{
set { lock (SyncRoot) { NativeMethods.SetPitchOctaves(handle, value); } }
}
/// <summary>
/// Sets pitch change in semi-tones compared to the original pitch
/// (-12 .. +12 for +- one octave);
///
/// Value: Pitch setting in semitones
/// </summary>
public float PitchSemiTones
{
set { lock (SyncRoot) { NativeMethods.SetPitchSemiTones(handle, value); } }
}
/// <summary>
/// Changes or gets a setting controlling the processing system behaviour. See the
/// 'SETTING_...' defines for available setting ID's.
/// </summary>
/// <value>
/// The <see cref="System.Int32"/>.
/// </value>
/// <param name="settingId">The setting identifier.</param>
/// <returns>The value of the setting</returns>
public int this[Setting settingId]
{
get
{
lock (SyncRoot) { return NativeMethods.GetSetting(handle, (int)settingId); }
}
set
{
lock (SyncRoot) { NativeMethods.SetSetting(handle, (int)settingId, value); }
}
}
#endregion
#region Sample Stream Methods
/// <summary>
/// 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.
/// </summary>
public void Flush()
{
lock (SyncRoot) { NativeMethods.Flush(handle); }
}
/// <summary>
/// Clears all the samples in the object's output and internal processing
/// buffers.
/// </summary>
public void Clear()
{
lock (SyncRoot) { NativeMethods.Clear(handle); }
}
/// <summary>
/// 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.
/// </summary>
/// <param name="samples">Sample buffer to input</param>
/// <param name="numSamples">Number of sample frames in buffer. Notice
/// that in case of multi-channel sound a single sample frame contains
/// data for all channels</param>
public void PutSamples(float[] samples, uint numSamples)
{
lock (SyncRoot) { NativeMethods.PutSamples(handle, samples, numSamples); }
}
/// <summary>
/// int16 version of putSamples(): This accept int16 (short) sample data
/// and internally converts it to float format before processing
/// </summary>
/// <param name="samples">Sample input buffer.</param>
/// <param name="numSamples">Number of sample frames in buffer. Notice
/// that in case of multi-channel sound a single
/// sample frame contains data for all channels.</param>
public void PutSamplesI16(short[] samples, uint numSamples)
{
lock (SyncRoot) { NativeMethods.PutSamples_i16(handle, samples, numSamples); }
}
/// <summary>
/// Receive processed samples from the processor.
/// </summary>
/// <param name="outBuffer">Buffer where to copy output samples</param>
/// <param name="maxSamples">Max number of sample frames to receive</param>
/// <returns>The number of samples received</returns>
public uint ReceiveSamples(float[] outBuffer, uint maxSamples)
{
lock (SyncRoot) { return NativeMethods.ReceiveSamples(handle, outBuffer, maxSamples); }
}
/// <summary>
/// int16 version of receiveSamples(): This converts internal float samples
/// into int16 (short) return data type
/// </summary>
/// <param name="outBuffer">Buffer where to copy output samples.</param>
/// <param name="maxSamples">How many samples to receive at max.</param>
/// <returns>Number of received sample frames</returns>
public uint ReceiveSamplesI16(short[] outBuffer, uint maxSamples)
{
lock (SyncRoot) { return NativeMethods.ReceiveSamples_i16(handle, outBuffer, maxSamples); }
}
#endregion
#region IDisposable Support
/// <summary>
/// Performs application-defined tasks associated with freeing, releasing, or resetting unmanaged resources.
/// </summary>
public void Dispose()
{
Dispose(true);
GC.SuppressFinalize(this);
}
/// <summary>
/// Releases unmanaged and - optionally - managed resources.
/// </summary>
/// <param name="alsoManaged"><c>true</c> to release both managed and unmanaged resources; <c>false</c> to release only unmanaged resources.</param>
private void Dispose(bool alsoManaged)
{
if (!IsDisposed)
{
if (alsoManaged)
{
// NOTE: Placeholder, dispose managed state (managed objects).
// At this point, nothing managed to dispose
}
NativeMethods.DestroyInstance(handle);
handle = IntPtr.Zero;
IsDisposed = true;
}
}
#endregion
#region Native Methods
/// <summary>
/// Provides direct access to mapped DLL methods
/// </summary>
private static class NativeMethods
{
[DllImport(SoundTouchLibrary, CallingConvention = CallingConvention.Cdecl, EntryPoint = "soundtouch_getVersionId")]
public static extern int GetVersionId();
[DllImport(SoundTouchLibrary, CallingConvention = CallingConvention.Cdecl, EntryPoint = "soundtouch_createInstance")]
public static extern IntPtr CreateInstance();
[DllImport(SoundTouchLibrary, CallingConvention = CallingConvention.Cdecl, EntryPoint = "soundtouch_destroyInstance")]
public static extern void DestroyInstance(IntPtr h);
[DllImport(SoundTouchLibrary, CallingConvention = CallingConvention.Cdecl, EntryPoint = "soundtouch_getVersionString")]
public static extern IntPtr GetVersionString();
[DllImport(SoundTouchLibrary, CallingConvention = CallingConvention.Cdecl, EntryPoint = "soundtouch_setRate")]
public static extern void SetRate(IntPtr h, float newRate);
[DllImport(SoundTouchLibrary, CallingConvention = CallingConvention.Cdecl, EntryPoint = "soundtouch_setTempo")]
public static extern void SetTempo(IntPtr h, float newTempo);
[DllImport(SoundTouchLibrary, CallingConvention = CallingConvention.Cdecl, EntryPoint = "soundtouch_setRateChange")]
public static extern void SetRateChange(IntPtr h, float newRate);
[DllImport(SoundTouchLibrary, CallingConvention = CallingConvention.Cdecl, EntryPoint = "soundtouch_setTempoChange")]
public static extern void SetTempoChange(IntPtr h, float newTempo);
[DllImport(SoundTouchLibrary, CallingConvention = CallingConvention.Cdecl, EntryPoint = "soundtouch_setPitch")]
public static extern void SetPitch(IntPtr h, float newPitch);
[DllImport(SoundTouchLibrary, CallingConvention = CallingConvention.Cdecl, EntryPoint = "soundtouch_setPitchOctaves")]
public static extern void SetPitchOctaves(IntPtr h, float newPitch);
[DllImport(SoundTouchLibrary, CallingConvention = CallingConvention.Cdecl, EntryPoint = "soundtouch_setPitchSemiTones")]
public static extern void SetPitchSemiTones(IntPtr h, float newPitch);
[DllImport(SoundTouchLibrary, CallingConvention = CallingConvention.Cdecl, EntryPoint = "soundtouch_setChannels")]
public static extern void SetChannels(IntPtr h, uint numChannels);
[DllImport(SoundTouchLibrary, CallingConvention = CallingConvention.Cdecl, EntryPoint = "soundtouch_setSampleRate")]
public static extern void SetSampleRate(IntPtr h, uint srate);
[DllImport(SoundTouchLibrary, CallingConvention = CallingConvention.Cdecl, EntryPoint = "soundtouch_flush")]
public static extern void Flush(IntPtr h);
[DllImport(SoundTouchLibrary, CallingConvention = CallingConvention.Cdecl, EntryPoint = "soundtouch_putSamples")]
public static extern void PutSamples(IntPtr h, float[] samples, uint numSamples);
[DllImport(SoundTouchLibrary, CallingConvention = CallingConvention.Cdecl, EntryPoint = "soundtouch_putSamples_i16")]
public static extern void PutSamples_i16(IntPtr h, short[] samples, uint numSamples);
[DllImport(SoundTouchLibrary, CallingConvention = CallingConvention.Cdecl, EntryPoint = "soundtouch_clear")]
public static extern void Clear(IntPtr h);
[DllImport(SoundTouchLibrary, CallingConvention = CallingConvention.Cdecl, EntryPoint = "soundtouch_setSetting")]
public static extern int SetSetting(IntPtr h, int settingId, int value);
[DllImport(SoundTouchLibrary, CallingConvention = CallingConvention.Cdecl, EntryPoint = "soundtouch_getSetting")]
public static extern int GetSetting(IntPtr h, int settingId);
[DllImport(SoundTouchLibrary, CallingConvention = CallingConvention.Cdecl, EntryPoint = "soundtouch_numUnprocessedSamples")]
public static extern uint NumUnprocessedSamples(IntPtr h);
[DllImport(SoundTouchLibrary, CallingConvention = CallingConvention.Cdecl, EntryPoint = "soundtouch_receiveSamples")]
public static extern uint ReceiveSamples(IntPtr h, float[] outBuffer, uint maxSamples);
[DllImport(SoundTouchLibrary, CallingConvention = CallingConvention.Cdecl, EntryPoint = "soundtouch_receiveSamples_i16")]
public static extern uint ReceiveSamples_i16(IntPtr h, short[] outBuffer, uint maxSamples);
[DllImport(SoundTouchLibrary, CallingConvention = CallingConvention.Cdecl, EntryPoint = "soundtouch_numSamples")]
public static extern uint NumSamples(IntPtr h);
[DllImport(SoundTouchLibrary, CallingConvention = CallingConvention.Cdecl, EntryPoint = "soundtouch_isEmpty")]
public static extern int IsEmpty(IntPtr h);
}
#endregion
}
public sealed class BPMDetect : IDisposable
{
#region Private Members
private readonly object SyncRoot = new object();
private bool IsDisposed = false;
private IntPtr handle;
#endregion
#region Constructor
/// <summary>
/// Initializes a new instance of the <see cref="BPMDetect"/> class.
/// </summary>
public BPMDetect(int numChannels, int sampleRate)
{
handle = NativeMethods.BpmCreateInstance(numChannels, sampleRate);
}
/// <summary>
/// Finalizes an instance of the <see cref="BPMDetect"/> class.
/// </summary>
~BPMDetect()
{
// Do not change this code. Put cleanup code in Dispose(bool disposing) above.
Dispose(false);
}
#endregion
#region Properties
/// <summary>
/// Returns the analysed BPM rate.
/// </summary>
public float Bpm
{
get { lock (SyncRoot) { return NativeMethods.BpmGet(handle); } }
}
#endregion
#region Sample Stream Methods
/// <summary>
/// Feed 'numSamples' sample into the BPM detector
/// </summary>
/// <param name="samples">Sample buffer to input</param>
/// <param name="numSamples">Number of sample frames in buffer. Notice
/// that in case of multi-channel sound a single sample frame contains
/// data for all channels</param>
public void PutSamples(float[] samples, uint numSamples)
{
lock (SyncRoot) { NativeMethods.BpmPutSamples(handle, samples, numSamples); }
}
/// <summary>
/// int16 version of putSamples(): This accept int16 (short) sample data
/// and internally converts it to float format before processing
/// </summary>
/// <param name="samples">Sample input buffer.</param>
/// <param name="numSamples">Number of sample frames in buffer. Notice
/// that in case of multi-channel sound a single
/// sample frame contains data for all channels.</param>
public void PutSamplesI16(short[] samples, uint numSamples)
{
lock (SyncRoot) { NativeMethods.BpmPutSamples_i16(handle, samples, numSamples); }
}
#endregion
#region IDisposable Support
/// <summary>
/// Performs application-defined tasks associated with freeing, releasing, or resetting unmanaged resources.
/// </summary>
public void Dispose()
{
Dispose(true);
GC.SuppressFinalize(this);
}
/// <summary>
/// Releases unmanaged and - optionally - managed resources.
/// </summary>
/// <param name="alsoManaged"><c>true</c> to release both managed and unmanaged resources; <c>false</c> to release only unmanaged resources.</param>
private void Dispose(bool alsoManaged)
{
if (!IsDisposed)
{
if (alsoManaged)
{
// NOTE: Placeholder, dispose managed state (managed objects).
// At this point, nothing managed to dispose
}
NativeMethods.BpmDestroyInstance(handle);
handle = IntPtr.Zero;
IsDisposed = true;
}
}
#endregion
#region Native Methods
/// <summary>
/// Provides direct access to mapped DLL methods
/// </summary>
private static class NativeMethods
{
[DllImport(SoundTouch.SoundTouchLibrary, CallingConvention = CallingConvention.Cdecl, EntryPoint = "bpm_createInstance")]
public static extern IntPtr BpmCreateInstance(int numChannels, int sampleRate);
[DllImport(SoundTouch.SoundTouchLibrary, CallingConvention = CallingConvention.Cdecl, EntryPoint = "bpm_destroyInstance")]
public static extern void BpmDestroyInstance(IntPtr h);
[DllImport(SoundTouch.SoundTouchLibrary, CallingConvention = CallingConvention.Cdecl, EntryPoint = "bpm_putSamples")]
public static extern void BpmPutSamples(IntPtr h, float[] samples, uint numSamples);
[DllImport(SoundTouch.SoundTouchLibrary, CallingConvention = CallingConvention.Cdecl, EntryPoint = "bpm_putSamples_i16")]
public static extern void BpmPutSamples_i16(IntPtr h, short[] samples, uint numSamples);
[DllImport(SoundTouch.SoundTouchLibrary, CallingConvention = CallingConvention.Cdecl, EntryPoint = "bpm_getBpm")]
public static extern float BpmGet(IntPtr h);
}
#endregion
}
}

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externals/soundtouch/source/csharp-example/csharp-example.csproj vendored Normal file
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<CopyToOutputDirectory>Always</CopyToOutputDirectory>
</Content>
</ItemGroup>
<ItemGroup>
<WCFMetadata Include="Service References\" />
</ItemGroup>
<Import Project="$(MSBuildToolsPath)\Microsoft.CSharp.targets" />
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Other similar extension points exist, see Microsoft.Common.targets.
<Target Name="BeforeBuild">
</Target>
<Target Name="AfterBuild">
</Target>
-->
</Project>

22
externals/soundtouch/source/csharp-example/csharp-example.sln vendored Normal file
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{1BE6EF4F-6A62-447B-A863-E918D68BDAD7}.Release|Any CPU.ActiveCfg = Release|Any CPU
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GlobalSection(SolutionProperties) = preSolution
HideSolutionNode = FALSE
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