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SoundTouch Audio Processing Library
https://www.surina.net/soundtouch/
version 2.3.1
Some unused files (SoundStretch, Android library, VS solutions...) have been removed.
For building, premake is used to generate Visual Studio project files.
See ../build/premake/ for details.

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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
<title>SoundTouch library README</title>
<meta http-equiv="Content-Language" content="en-us">
<meta name="author" content="Olli Parviainen">
<meta name="description" content="Readme file for SoundTouch audio processing library">
<style>
body {
font-family: Arial, Helvetica;
}
</style>
</head>
<body class="normal">
<hr>
<h1>SoundTouch audio processing library v2.3.1</h1>
<p class="normal">SoundTouch library Copyright &copy; Olli Parviainen 2001-2021</p>
<hr>
<h2>1. Introduction </h2>
<p>SoundTouch is an open-source audio processing library that allows
changing the sound tempo, pitch and playback rate parameters
independently from each other, i.e.:</p>
<ul>
<li> Sound tempo can be increased or decreased while maintaining the
original pitch</li>
<li> Sound pitch can be increased or decreased while maintaining the
original tempo</li>
<li> Change playback rate that affects both tempo and pitch at the
same time</li>
<li> Choose any combination of tempo/pitch/rate</li>
</ul>
<h3>1.1 Contact information </h3>
<p>Author email: oparviai 'at' iki.fi </p>
<p>SoundTouch WWW page: <a href="http://soundtouch.surina.net">http://soundtouch.surina.net</a></p>
<p>SoundTouch git repository: <a
href="https://gitlab.com/soundtouch/soundtouch.git">https://gitlab.com/soundtouch/soundtouch.git</a></p>
<hr>
<h2>2. Compiling SoundTouch</h2>
<p>Before compiling, notice that you can choose the sample data format if it's
desirable to use 16bit integer sample data instead of floating point samples. See
section &quot;sample data format&quot; for more information.</p>
<p>Also notice that SoundTouch can use OpenMP instructions for parallel
computation to accelerate the runtime processing speed in multi-core systems,
however, these improvements need to be separately enabled before compiling. See
OpenMP notes in Chapter 3 below.</p>
<h3>2.1. Building in Microsoft Windows</h3>
<p>Project files for Microsoft Visual C++ are supplied with the source
code package. Go to Microsoft WWW page to download
<a href="http://www.visualstudio.com/en-US/products/visual-studio-express-vs">
Microsoft Visual Studio Express version for free</a>.
</p>
<p>To build the binaries with Visual C++ compiler, either run
"make-win.bat" script, or open the appropriate project files in source
code directories with Visual Studio. The final executable will appear
under the "SoundTouch\bin" directory. If using the Visual Studio IDE
instead of the make-win.bat script, directories bin and lib may need to
be created manually to the SoundTouch package root for the final
executables. The make-win.bat script creates these directories
automatically. </p>
<p><strong>C# example</strong>: The source code package includes also a C# example
application for Windows that shows how to invoke SoundTouch.dll
dynamic-load library for processing mp3 audio.
<p><strong>OpenMP NOTE</strong>: If activating the OpenMP parallel computing in
the compilation, the target program will require additional vcomp dll library to
properly run. In Visual C++ 9.0 these libraries can be found in the following
folders.</p>
<ul>
<li>x86 32bit: C:\Program Files (x86)\Microsoft Visual Studio
9.0\VC\redist\x86\Microsoft.VC90.OPENMP\vcomp90.dll</li>
<li>x64 64bit: C:\Program Files (x86)\Microsoft Visual Studio
9.0\VC\redist\amd64\Microsoft.VC90.OPENMP\vcomp90.dll</li>
</ul>
<p>In other VC++ versions the required library will be expectedly found in similar
&quot;redist&quot; location.</p>
<p>Notice that as minor demonstration of a &quot;dll hell&quot; phenomenon both the 32-bit
and 64-bit version of vcomp90.dll have the same filename but different contents,
thus choose the proper version to allow the program to start.</p>
<h3>2.2. Building in Gnu platforms</h3>
<p>The SoundTouch library compiles in practically any platform
supporting GNU compiler (GCC) tools.
<h4>2.2.1 Compiling with autotools</h4>
<p>To install build prerequisites for 'autotools' tool chain:</p>
<pre> sudo apt-get install automake autoconf libtool build-essential</pre>
<p>To build and install the binaries, run the following commands in
/soundtouch directory:</p>
<table border="0" cellpadding="0" cellspacing="4">
<tbody>
<tr>
<td style="vertical-align: top;">
<pre>./bootstrap -</pre>
</td>
<td style="vertical-align: top;">Creates "configure" file with
local autoconf/automake toolset.<br>
</td>
</tr>
<tr valign="top">
<td>
<pre>./configure -</pre>
</td>
<td>
<p>Configures the SoundTouch package for the local environment.
Notice that "configure" file is not available before running the
"./bootstrap" command as above.<br>
</p>
</td>
</tr>
<tr valign="top">
<td>
<pre>make -</pre>
</td>
<td>
<p>Builds the SoundTouch library &amp; SoundStretch utility. You can
optionally add &quot;-j&quot; switch after &quot;make&quot; to speed up the compilation in
multi-core systems.</p>
</td>
</tr>
<tr valign="top">
<td>
<pre>make install -</pre>
</td>
<td>
<p>Installs the SoundTouch &amp; BPM libraries to <b>/usr/local/lib</b>
and SoundStretch utility to <b>/usr/local/bin</b>. Please notice that
'root' privileges may be required to install the binaries to the
destination locations.</p>
</td>
</tr>
</tbody>
</table>
<b>Compiling portable Shared Library / DLL version</b>
<p> The GNU autotools compilation does not automatically create a shared-library version of
SoundTouch (.so or .dll) that features position-independent code and C-language
api that are more suitable for cross-language development than C++ libraries.</p>
<p> Use script "make-gnu-dll-sh" to build a portable dynamic library version if such is desired.</p>
<h4><b>2.2.2 Compiling with cmake</b></h4>
<p>'cmake' build scripts are provided as an alternative to the autotools toolchain.</p>
<p>To install cmake build prerequisites:</p>
<pre> sudo apt-get install libtool build-essential cmake</pre>
<p>To build:</p>
<pre>
cmake .
make -j
make install</pre>
<p>To compile the additional portable Shared Library / DLL version with the native C-language API:</p>
<pre>
cmake . -DSOUNDTOUCH_DLL=ON
make -j
make install</pre>
<h3>2.3. Building in Android</h3>
<p>Android compilation instructions are within the
source code package, see file &quot;<b>source/Android-lib/README-SoundTouch-Android.html</b>&quot;
in the source code package. </p>
<p>The Android compilation automatically builds separate .so library binaries
for ARM, X86 and MIPS processor architectures. For optimal device support,
include all these .so library binaries into the Android .apk application
package, so the target Android device can automatically choose the proper
library binary version to use.</p>
<p>The <strong>source/Android-lib</strong> folder includes also an Android
example application that processes WAV audio files using SoundTouch library in
Android devices.</p>
<h3>2.4. Building in Mac</h3>
<p>Install autoconf tool as instructed in <a
href="http://macappstore.org/autoconf/">http://macappstore.org/autoconf/</a>, or alternatively the 'cmake' toolchain.</p>
<p>Then, build as described above in section "Building in Gnu platforms".</p>
<hr>
<h2>3. About implementation &amp; Usage tips <h3>3.1. Supported sample data formats</h3>
<p>The sample data format can be chosen between 16bit signed integer
and 32bit floating point values.</p>
</p> The default sample type is 32bit floating point format,
which also provides better sound quality than integer format because
integer algorithms need to scale already intermediate calculation results to
avoid integer overflows. These early integer scalings can slightly degrade
output quality.</p>
<p> In Windows environment, the sample data format is chosen in file
"STTypes.h" by choosing one of the following defines:</p>
<ul>
<li> <span style="font-weight: bold;">#define
SOUNDTOUCH_INTEGER_SAMPLES</span> for 16bit signed integer</li>
<li> <span style="font-weight: bold;">#define </span><span style="font-weight: bold;">SOUNDTOUCH_</span><span
style="font-weight: bold;">FLOAT_SAMPLES</span> for 32bit floating
point</li>
</ul>
<p> In GNU environment, the floating sample format is used by default,
but integer sample format can be chosen by giving the following switch
to the configure script: </p>
<blockquote>
<pre>./configure --enable-integer-samples</pre>
</blockquote>
<p>The sample data can have either single (mono) or double (stereo)
audio channel. Stereo data is interleaved so that every other data
value is for left channel and every second for right channel. Notice
that while it'd be possible in theory to process stereo sound as two
separate mono channels, this isn't recommended because processing the
channels separately would result in losing the phase coherency between
the channels, which consequently would ruin the stereo effect.</p>
<p>Sample rates between 8000-48000H are supported.</p>
<h3>3.2. Processing latency</h3>
<p>The processing and latency constraints of the SoundTouch library are:</p>
<ul>
<li> Input/output processing latency for the SoundTouch processor is
around 100 ms. This is when time-stretching is used. If the rate
transposing effect alone is used, the latency requirement is much
shorter, see section 'About algorithms'.</li>
<li> Processing CD-quality sound (16bit stereo sound with 44100H
sample rate) in real-time or faster is possible starting from
processors equivalent to Intel Pentium 133Mh or better, if using the
"quick" processing algorithm. If not using the "quick" mode or if
floating point sample data are being used, several times more CPU power
is typically required.</li>
</ul>
<h3>3.3. About algorithms</h3>
<p>SoundTouch provides three seemingly independent effects: tempo,
pitch and playback rate control. These three controls are implemented
as combination of two primary effects, <em>sample rate transposing</em>
and <em>time-stretching</em>.</p>
<p><em>Sample rate transposing</em> affects both the audio stream
duration and pitch. It's implemented simply by converting the original
audio sample stream to the desired duration by interpolating from
the original audio samples. In SoundTouch, linear interpolation with
anti-alias filtering is used. Theoretically a higher-order
interpolation provide better result than 1st order linear
interpolation, but in audio application linear interpolation together
with anti-alias filtering performs subjectively about as well as
higher-order filtering would.</p>
<p><em>Time-stretching </em>means changing the audio stream duration
without affecting it's pitch. SoundTouch uses WSOLA-like
time-stretching routines that operate in the time domain. Compared to
sample rate transposing, time-stretching is a much heavier operation
and also requires a longer processing "window" of sound samples used by
the processing algorithm, thus increasing the algorithm input/output
latency. Typical i/o latency for the SoundTouch time-stretch algorithm
is around 100 ms.</p>
<p>Sample rate transposing and time-stretching are then used together
to produce the tempo, pitch and rate controls:</p>
<ul>
<li> <strong>'Tempo'</strong> control is implemented purely by
time-stretching.</li>
<li> <strong>'Rate</strong>' control is implemented purely by sample
rate transposing.</li>
<li> <strong>'Pitch</strong>' control is implemented as a
combination of time-stretching and sample rate transposing. For
example, to increase pitch the audio stream is first time-stretched to
longer duration (without affecting pitch) and then transposed back to
original duration by sample rate transposing, which simultaneously
reduces duration and increases pitch. The result is original duration
but increased pitch.</li>
</ul>
<h3>3.4 Tuning the algorithm parameters</h3>
<p>The time-stretch algorithm has few parameters that can be tuned to
optimize sound quality for certain application. The current default
parameters have been chosen by iterative if-then analysis (read: "trial
and error") to obtain best subjective sound quality in pop/rock music
processing, but in applications processing different kind of sound the
default parameter set may result into a sub-optimal result.</p>
<p>The time-stretch algorithm default parameter values are set by the
following #defines in file "TDStretch.h":</p>
<blockquote>
<pre>#define DEFAULT_SEQUENCE_MS AUTOMATIC<br>#define DEFAULT_SEEKWINDOW_MS AUTOMATIC<br>#define DEFAULT_OVERLAP_MS 8</pre>
</blockquote>
<p>These parameters affect to the time-stretch algorithm as follows:</p>
<ul>
<li> <strong>DEFAULT_SEQUENCE_MS</strong>: This is the default
length of a single processing sequence in milliseconds which determines
the how the original sound is chopped in the time-stretch algorithm.
Larger values mean fewer sequences are used in processing. In principle
a larger value sounds better when slowing down the tempo, but worse
when increasing the tempo and vice versa.<br>
<br>
By default, this setting value is calculated automatically according to
tempo value.<br>
</li>
<li> <strong>DEFAULT_SEEKWINDOW_MS</strong>: The seeking window
default length in milliseconds is for the algorithm that seeks the best
possible overlapping location. This determines from how wide a sample
"window" the algorithm can use to find an optimal mixing location when
the sound sequences are to be linked back together.<br>
<br>
The bigger this window setting is, the higher the possibility to find a
better mixing position becomes, but at the same time large values may
cause a "drifting" sound artifact because neighboring sequences can be
chosen at more uneven intervals. If there's a disturbing artifact that
sounds as if a constant frequency was drifting around, try reducing
this setting.<br>
<br>
By default, this setting value is calculated automatically according to
tempo value.<br>
</li>
<li> <strong>DEFAULT_OVERLAP_MS</strong>: Overlap length in
milliseconds. When the sound sequences are mixed back together to form
again a continuous sound stream, this parameter defines how much the
ends of the consecutive sequences will overlap with each other.<br>
<br>
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.
</li>
</ul>
<p>Notice that these parameters can also be set during execution time
with functions "<strong>TDStretch::setParameters()</strong>" and "<strong>SoundTouch::setSetting()</strong>".</p>
<p>The table below summaries how the parameters can be adjusted for
different applications:</p>
<table border="1">
<tbody>
<tr>
<td valign="top"><strong>Parameter name</strong></td>
<td valign="top"><strong>Default value magnitude</strong></td>
<td valign="top"><strong>Larger value affects...</strong></td>
<td valign="top"><strong>Smaller value affects...</strong></td>
<td valign="top"><strong>Effect to CPU burden</strong></td>
</tr>
<tr>
<td valign="top">
<pre>SEQUENCE_MS</pre>
</td>
<td valign="top">Default value is relatively large, chosen for
slowing down music tempo</td>
<td valign="top">Larger value is usually better for slowing down
tempo. Growing the value decelerates the "echoing" artifact when
slowing down the tempo.</td>
<td valign="top">Smaller value might be better for speeding up
tempo. Reducing the value accelerates the "echoing" artifact when
slowing down the tempo </td>
<td valign="top">Increasing the parameter value reduces
computation burden</td>
</tr>
<tr>
<td valign="top">
<pre>SEEKWINDOW_MS</pre>
</td>
<td valign="top">Default value is relatively large, chosen for
slowing down music tempo</td>
<td valign="top">Larger value eases finding a good mixing
position, but may cause a "drifting" artifact</td>
<td valign="top">Smaller reduce possibility to find a good mixing
position, but reduce the "drifting" artifact.</td>
<td valign="top">Increasing the parameter value increases
computation burden</td>
</tr>
<tr>
<td valign="top">
<pre>OVERLAP_MS</pre>
</td>
<td valign="top">Default value is relatively large, chosen to
suit with above parameters.</td>
<td valign="top"></td>
<td valign="top">If you reduce the "sequence ms" setting, you
might wish to try a smaller value.</td>
<td valign="top">Increasing the parameter value increases
computation burden</td>
</tr>
</tbody>
</table>
<h3>3.5 Performance Optimizations </h3>
<p><strong>Integer vs floating point:</strong></p>
<p>Floating point sample type is generally recommended because it provides
better sound quality.</p>
<p>However, execution speed difference between integer and floating point processing
depends on the CPU architecture. As rule of thumb,
<ul>
<li>in 32-bit x86 floating point and integer are roughly equally fast</li>
<li>in 64-bit x86/x64 floating point can be significantly faster than integer
version, because MMX integer optimizations are not available in the x64 architecture.
That depends on the compiler however, so that gcc can autovectorize integer routines
to work equally fast as floating point, where as Visual C++ (2017) does not
perform equally well and produces integer code that runs some 3x slower than
SSE-optimized floating poing code.
</li>
<li>in ARMv7 integer routines are twice as fast as floating point. Their
relative difference is roughly the same both with and without NEON; NEON
vfpu can however bring 2.4x speed improvement.
</li>
<li>in other platforms: try out if the execution time performance makes a
big difference</li>
</ul>
</p>
<p><strong>General optimizations:</strong></p>
<p>The time-stretch routine has a 'quick' mode that substantially
speeds up the algorithm but may slightly compromise the sound quality.
This mode is activated by calling SoundTouch::setSetting()
function with parameter id of SETTING_USE_QUICKSEEK and value
"1", i.e. </p>
<blockquote>
<p>setSetting(SETTING_USE_QUICKSEEK, 1);</p>
</blockquote>
<p><strong>CPU-specific optimizations:</strong></p>
<p>Intel x86 specific SIMD optimizations are implemented using compiler
intrinsics, providing about a 3x processing speedup for x86 compatible
processors vs. non-SIMD implementation:</p>
<ul>
<li> MMX optimized routines are used in 32-bit x86 build when 16bit integer
sample type is used</li>
<li> SSE optimized routines are used in 32- and 64-bit x86 CPUs when 32bit
floating point sample type is used</li>
</ul>
<p>The algorithms are tuned to utilize autovectorization efficiently
also in other CPU architectures, for example ARM cpus see approx 2.4x processing
speedup when NEON SIMD support is present.
</p>
<h3>3.5 OpenMP parallel computation</h3>
<p>SoundTouch 1.9 onwards support running the algorithms parallel in several CPU
cores. Based on benchmark the experienced multi-core processing speed-up gain
ranges between +30% (on a high-spec dual-core x86 Windows PC) to 215% (on a moderately low-spec
quad-core ARM of Raspberry Pi2). </p>
<p>See an external blog article with more detailed discussion about the
<a href="http://www.softwarecoven.com/parallel-computing-in-embedded-mobile-devices/">
SoundTouch OpenMP optimization</a>.
</p>
<p>The parallel computing support is implemented using OpenMP spec 3.0
instructions. These instructions are supported by Visual C++ 2008 and later, and
GCC v4.2 and later. Compilers that do not supporting OpenMP will ignore these
optimizations and routines will still work properly. Possible warnings about
unknown #pragmas are related to OpenMP support and can be safely ignored.</p>
<p>The OpenMP improvements are disabled by default, and need to be enabled by
developer during compile-time. Reason for this is that parallel processing adds
moderate runtime overhead in managing the multi-threading, so it may not be
necessary nor desirable in all applications. For example real-time processing
that is not constrained by CPU power will not benefit of speed-up provided by
the parallel processing, in the contrary it may increase power consumption due
to the increased overhead.</p>
<p>However, applications that run on low-spec multi-core CPUs and may otherwise
have possibly constrained performance will benefit of the OpenMP improvements.
This include for example multi-core embedded devices.</p>
<p>OpenMP parallel computation can be enabled before compiling SoundTouch
library as follows:</p>
<ul>
<li><strong>Visual Studio</strong>: Open properties for the <strong>SoundTouch
</strong>sub-project, browse to <strong>C/C++</strong> and <strong>Language
</strong>settings. Set
there &quot;<strong>OpenMP support</strong>&quot; to &quot;<strong>Yes</strong>&quot;. Alternatively add
<strong>/openmp</strong> switch to command-line
parameters
</li>
<li><strong>GNU</strong>: Run the configure script with &quot;<strong>./configure
--enable-openmp</strong>&quot; switch, then run make as usually</li>
<li><strong>Android</strong>: Add &quot;<strong>-fopenmp</strong>&quot; switches to compiler &amp; linker
options, see README-SoundTouch-Android.html in the source code package for
more detailed instructions.</li>
</ul>
<hr>
<h2><a name="SoundStretch"></a>4. SoundStretch audio processing utility
</h2>
<p>SoundStretch audio processing utility<br>
Copyright (c) Olli Parviainen 2002-2015</p>
<p>SoundStretch is a simple command-line application that can change
tempo, pitch and playback rates of WAV sound files. This program is
intended primarily to demonstrate how the "SoundTouch" library can be
used to process sound in your own program, but it can as well be used
for processing sound files.</p>
<h3>4.1. SoundStretch Usage Instructions</h3>
<p>SoundStretch Usage syntax:</p>
<blockquote>
<pre>soundstretch infilename outfilename [switches]</pre>
</blockquote>
<p>Where: </p>
<table width="100%" border="0" cellpadding="2">
<tbody>
<tr>
<td valign="top">
<pre>"infilename"</pre>
</td>
<td valign="top">Name of the input sound data file (in .WAV audio
file format). Give "stdin" as filename to use standard input pipe. </td>
</tr>
<tr>
<td valign="top">
<pre>"outfilename"</pre>
</td>
<td valign="top">Name of the output sound file where the
resulting sound is saved (in .WAV audio file format). This parameter
may be omitted if you don't want to save the output (e.g. when
only calculating BPM rate with '-bpm' switch). Give "stdout" as
filename to use standard output pipe.</td>
</tr>
<tr>
<td valign="top">
<pre>[switches]</pre>
</td>
<td valign="top">Are one or more control switches.</td>
</tr>
</tbody>
</table>
<p>Available control switches are:</p>
<table width="100%" border="0" cellpadding="2">
<tbody>
<tr>
<td valign="top">
<pre>-tempo=n </pre>
</td>
<td valign="top">Change the sound tempo by n percents (n = -95.0
.. +5000.0 %) </td>
</tr>
<tr>
<td valign="top">
<pre>-pitch=n</pre>
</td>
<td valign="top">Change the sound pitch by n semitones (n = -60.0
.. + 60.0 semitones) </td>
</tr>
<tr>
<td valign="top">
<pre>-rate=n</pre>
</td>
<td valign="top">Change the sound playback rate by n percents (n
= -95.0 .. +5000.0 %) </td>
</tr>
<tr>
<td valign="top">
<pre>-bpm=n</pre>
</td>
<td valign="top">Detect the Beats-Per-Minute (BPM) rate of the
sound and adjust the tempo to meet 'n' BPMs. When this switch is
applied, the "-tempo" switch is ignored. If "=n" is omitted, i.e.
switch "-bpm" is used alone, then the BPM rate is estimated and
displayed, but tempo not adjusted according to the BPM value. </td>
</tr>
<tr>
<td valign="top">
<pre>-quick</pre>
</td>
<td valign="top">Use quicker tempo change algorithm. Gains speed
but loses sound quality. </td>
</tr>
<tr>
<td valign="top">
<pre>-naa</pre>
</td>
<td valign="top">Don't use anti-alias filtering in sample rate
transposing. Gains speed but loses sound quality. </td>
</tr>
<tr>
<td valign="top">
<pre>-license</pre>
</td>
<td valign="top">Displays the program license text (LGPL)</td>
</tr>
</tbody>
</table>
<p>Notes:</p>
<ul>
<li> To use standard input/output pipes for processing, give "stdin"
and "stdout" as input/output filenames correspondingly. The standard
input/output pipes will still carry the audio data in .wav audio file
format.</li>
<li> The numerical switches allow both integer (e.g. "-tempo=123")
and decimal (e.g. "-tempo=123.45") numbers.</li>
<li> The "-naa" and/or "-quick" switches can be used to reduce CPU
usage while compromising some sound quality</li>
<li> The BPM detection algorithm works by detecting repeating bass or
drum patterns at low frequencies of &lt;250Hz. A lower-than-expected
BPM figure may be reported for music with uneven or complex bass
patterns.</li>
</ul>
<h3>4.2. SoundStretch usage examples </h3>
<p><strong>Example 1</strong></p>
<p>The following command increases tempo of the sound file
"originalfile.wav" by 12.5% and stores result to file
"destinationfile.wav":</p>
<blockquote>
<pre>soundstretch originalfile.wav destinationfile.wav -tempo=12.5</pre>
</blockquote>
<p><strong>Example 2</strong></p>
<p>The following command decreases the sound pitch (key) of the sound
file "orig.wav" by two semitones and stores the result to file
"dest.wav":</p>
<blockquote>
<pre>soundstretch orig.wav dest.wav -pitch=-2</pre>
</blockquote>
<p><strong>Example 3</strong></p>
<p>The following command processes the file "orig.wav" by decreasing
the sound tempo by 25.3% and increasing the sound pitch (key) by 1.5
semitones. Resulting .wav audio data is directed to standard output
pipe:</p>
<blockquote>
<pre>soundstretch orig.wav stdout -tempo=-25.3 -pitch=1.5</pre>
</blockquote>
<p><strong>Example 4</strong></p>
<p>The following command detects the BPM rate of the file "orig.wav"
and adjusts the tempo to match 100 beats per minute. Result is stored
to file "dest.wav":</p>
<blockquote>
<pre>soundstretch orig.wav dest.wav -bpm=100</pre>
</blockquote>
<p><strong>Example 5</strong></p>
<p>The following command reads .wav sound data from standard input pipe
and estimates the BPM rate:</p>
<blockquote>
<pre>soundstretch stdin -bpm</pre>
</blockquote>
<p><strong>Example 6</strong></p>
<p>The following command tunes song from original 440Hz tuning to 432Hz tuning:
this corresponds to lowering the pitch by -0.318 semitones:</p>
<blockquote>
<pre>soundstretch original.wav output.wav -pitch=-0.318</pre>
</blockquote>
<hr>
<h2>5. Change History</h2>
<h3>5.1. SoundTouch library Change History </h3>
<p><b>2.3.1:</b></p>
<ul>
<li>Adjusted cmake build settings and header files that cmake installs</li>
</ul>
<p><b>2.3.0:</b></p>
<ul>
<li>Disable setting "SOUNDTOUCH_ALLOW_NONEXACT_SIMD_OPTIMIZATION" by default. The original
purpose of this setting was to avoid performance penalty due to unaligned SIMD memory
accesses in old CPUs, but that is not any more issue in concurrent CPU SIMD implementations
and having this setting enabled can cause slight compromise in result quality.
</li>
<li>Bugfix: soundtouch.clear() to really clear whole processing pipeline state. Earlier
individual variables were left uncleared, which caused slightly different result if
the same audio stream were processed again after calling clear().
</li>
<li>Bugfix: TDstretch to align initial offset position to be in middle of correlation search
window. This ensures that with zero tempo change the output will be same as input.
</li>
<li>Bugfix: Fix a bug in TDstrectch with too small initial skipFract value that occurred
with certain processing parameter settings: Replace assert with assignment that
corrects the situation.
</li>
<li>Remove OpenMP "_init_threading" workaround from Android build as it's not needed with concurrent
Android SDKs any more.</li>
</ul>
<p><b>2.2:</b></p>
<ul>
<li>Improved source codes so that compiler can autovectorize them more effectively.
This brings remarkable improvement e.g. ARM cpus equipped with NEON vfpu: Bencmarked
2.4x improvement in execution speed in ARMv7l vs the previous SoundTouch version
for both integer and floating point sample types.
</li>
<li>Bugfix: Resolved bad sound quality when using integer sample types in non-x86 CPU</li>
<li>Bugfix: Fixed possible reading past end of array in BPM peak detection algorithm</li>
</ul>
<p><b>2.1.2:</b></p>
<ul>
<li>Bump version to 2.1.2 also in configure.ac. The earlier release had old version info for GNU autotools.</li>
</ul>
<p><b>2.1.1:</b></p>
<ul>
<li>Bugfixes: Fixed potential buffer overwrite bugs in WavFile routines. Replaced asserts with runtime exceptions.
</li>
<li>Android: Migrated the SoundTouch Android example to new Android Studio</li>
<li>Automake: unset ACLOCAL in bootstrap script in case earlier build script has set it</li>
</ul>
<p><b>2.1:</b></p>
<ul>
<li>Refactored C# interface example</li>
<li>Disable anti-alias filter when switch
SOUNDTOUCH_PREVENT_CLICK_AT_RATE_CROSSOVER defined because anti-alias
filter cause slight click if the rate change crosses zero during
processing</li>
<li>Added script for building SoundTouchDll dynamic-link-library for GNU platforms</li>
<li>Rewrote Beats-per-Minute analysis algorithm for more reliable BPM detection</li>
<li>Added BPM functions to SoundTouchDll API</li>
<li>Migrated Visual Studio project files to MSVC 201x format</li>
<li>Replaced function parameter value asserts with runtime exceptions</li>
<li>Code maintenance & style cleanup</li>
</ul>
<p><b>2.0:</b></p>
<ul>
<li>Added functions to get initial processing latency, duration ratio between the original input and processed
output tracks, and clarified reporting of input/output batch sizes</li>
<li>Fixed issue that added brief sequence of silence to beginning of output audio</li>
<li>Adjusted algorithm parameters to reduce reverberating effect at tempo slowdown</li>
<li>Bugfix: Fixed a glitch that could cause negative array indexing in quick seek algorithm</li>
<li>Bugfix: flush() didn't properly flush final samples from the pipeline on 2nd time in case that soundtouch
object instance was recycled and used for processing a second audio stream.</li>
<li>Bugfix: Pi value had incorrect 9th/10th decimals</li>
<li>Added C# example application that uses SoundTouch dll library for processing MP3 files</li>
</ul>
<p><b>1.9.2:</b></p>
<ul>
<li>Fix in GNU package configuration</li>
</ul>
<p><b>1.9.1:</b></p>
<ul>
<li>Improved SoundTouch::flush() function so that it returns precisely the desired amount of samples for exact
output duration control</li>
<li>Redesigned quickseek algorithm for improved sound quality when using the quickseek mode. The new quickseek
algorithm can find 99% as good results as the
default full-scan mode, while the quickseek algorithm is remarkable less
CPU intensive.</li>
<li>Added adaptive integer divider scaling for improved sound quality when using integer processing algorithm
</li>
</ul>
<p><b>1.9:</b></p>
<ul>
<li>Added support for parallel computation support via OpenMP primitives for better performance in multicore
systems.
Benchmarks show that achieved parallel processing speedup improvement
typically range from +30% (x86 dual-core) to +180% (ARM quad-core). The
OpenMP optimizations are disabled by default, see OpenMP notes above in this
readme file how to enabled these optimizations.</li>
<li>Android: Added support for Android devices featuring X86 and MIPS CPUs,
in addition to ARM CPUs.</li>
<li>Android: More versatile Android example application that processes WAV
audio files with SoundTouch library</li>
<li>Replaced Windows-like 'BOOL' types with native 'bool'</li>
<li>Changed documentation token to "dist_doc_DATA" in Makefile.am file</li>
<li>Miscellaneous small fixes and improvements</li>
</ul>
<p><b>1.8.0:</b></p>
<ul>
<li>Added support for multi-channel audio processing</li>
<li>Added support for <b>cubic</b> and <b>shannon</b> interpolation for rate and pitch shift effects besides
the original <b>linear</b> interpolation, to reduce aliasing at high frequencies due to interpolation.
Cubic interpolation is used as default for floating point processing, and linear interpolation for integer
processing.</li>
<li>Fixed bug in anti-alias filtering that limited stop-band attenuation to -10 dB instead of <-50dB, and
increased filter length from 32 to 64 taps to further reduce aliasing due to frequency folding.</li>
<li>Performance improvements in cross-correlation algorithm</li>
<li>Other bug and compatibility fixes</li>
</ul>
<p><b>1.7.1:</b></p>
<ul>
<li>Added files for Android compilation
</ul>
<p><b>1.7.0:</b></p>
<ul>
<li>Sound quality improvements/li>
<li>Improved flush() to adjust output sound stream duration to match better with
ideal duration</li>
<li>Rewrote x86 cpu feature check to resolve compatibility problems</li>
<li>Configure script automatically checks if CPU supports mmx & sse compatibility for GNU platform, and
the script support now "--enable-x86-optimizations" switch to allow disabling x86-specific optimizations.</li>
<li>Revised #define conditions for 32bit/64bit compatibility</li>
<li>gnu autoconf/automake script compatibility fixes</li>
<li>Tuned beat-per-minute detection algorithm</li>
</ul>
<p><b>1.6.0:</b></p>
<ul>
<li> Added automatic cutoff threshold adaptation to beat detection
routine to better adapt BPM calculation to different types of music</li>
<li> Retired 3DNow! optimization support as 3DNow! is nowadays
obsoleted and assembler code is nuisance to maintain</li>
<li>Retired "configure" file from source code package due to
autoconf/automake versio conflicts, so that it is from now on to be
generated by invoking "boostrap" script that uses locally available
toolchain version for generating the "configure" file</li>
<li>Resolved namespace/label naming conflicts with other libraries by
replacing global labels such as INTEGER_SAMPLES with more specific
SOUNDTOUCH_INTEGER_SAMPLES etc.<br>
</li>
<li>Updated windows build scripts &amp; project files for Visual
Studio 2008 support</li>
<li> Updated SoundTouch.dll API for .NET compatibility</li>
<li> Added API for querying nominal processing input &amp; output
sample batch sizes</li>
</ul>
<p><strong>1.5.0:</strong></p>
<ul>
<li> Added normalization to correlation calculation and improvement
automatic seek/sequence parameter calculation to improve sound quality</li>
<li> Bugfixes:
<ul>
<li> Fixed negative array indexing in quick seek algorithm</li>
<li> FIR autoalias filter running too far in processing buffer</li>
<li> Check against zero sample count in rate transposing</li>
<li> Fix for x86-64 support: Removed pop/push instructions from
the cpu detection algorithm.</li>
<li> Check against empty buffers in FIFOSampleBuffer</li>
<li> Other minor fixes &amp; code cleanup</li>
</ul>
</li>
<li> Fixes in compilation scripts for non-Intel platforms</li>
<li> Added Dynamic-Link-Library (DLL) version of SoundTouch library
build, provided with Delphi/Pascal wrapper for calling the dll routines
</li>
<li> Added #define PREVENT_CLICK_AT_RATE_CROSSOVER that prevents a
click artifact when crossing the nominal pitch from either positive to
negative side or vice versa</li>
</ul>
<p><strong>1.4.1:</strong></p>
<ul>
<li> Fixed a buffer overflow bug in BPM detect algorithm routines if
processing more than 2048 samples at one call</li>
</ul>
<p><strong>1.4.0:</strong></p>
<ul>
<li> Improved sound quality by automatic calculation of time stretch
algorithm processing parameters according to tempo setting</li>
<li> Moved BPM detection routines from SoundStretch application into
SoundTouch library</li>
<li> Bugfixes: Usage of uninitialied variables, GNU build scripts,
compiler errors due to 'const' keyword mismatch.</li>
<li> Source code cleanup</li>
</ul>
<p><strong>1.3.1: </strong> </p>
<ul>
<li> Changed static class declaration to GCC 4.x compiler compatible
syntax.</li>
<li> Enabled MMX/SSE-optimized routines also for GCC compilers.
Earlier the MMX/SSE-optimized routines were written in
compiler-specific inline assembler, now these routines are migrated to
use compiler intrinsic syntax which allows compiling the same
MMX/SSE-optimized source code with both Visual C++ and GCC compilers.</li>
<li> Set floating point as the default sample format and added switch
to the GNU configure script for selecting the other sample format.</li>
</ul>
<p><strong>1.3.0: </strong> </p>
<ul>
<li> Fixed tempo routine output duration inaccuracy due to rounding
error</li>
<li> Implemented separate processing routines for integer and
floating arithmetic to allow improvements to floating point routines
(earlier used algorithms mostly optimized for integer arithmetic also
for floating point samples)</li>
<li> Fixed a bug that distorts sound if sample rate changes during
the sound stream</li>
<li> Fixed a memory leak that appeared in MMX/SSE/3DNow! optimized
routines</li>
<li> Reduced redundant code pieces in MMX/SSE/3DNow! optimized
routines vs. the standard C routines.</li>
<li> MMX routine incompatibility with new gcc compiler versions</li>
<li> Other miscellaneous bug fixes</li>
</ul>
<p><strong>1.2.1: </strong> </p>
<ul>
<li> Added automake/autoconf scripts for GNU platforms (in courtesy
of David Durham)</li>
<li> Fixed SCALE overflow bug in rate transposer routine.</li>
<li> Fixed 64bit address space bugs.</li>
<li> Created a 'soundtouch' namespace for SAMPLETYPE definitions.</li>
</ul>
<p><strong>1.2.0: </strong> </p>
<ul>
<li> Added support for 32bit floating point sample data type with
SSE/3DNow! optimizations for Win32 platform (SSE/3DNow! optimizations
currently not supported in GCC environment)</li>
<li> Replaced 'make-gcc' script for GNU environment by master
Makefile</li>
<li> Added time-stretch routine configurability to SoundTouch main
class</li>
<li> Bugfixes</li>
</ul>
<p><strong>1.1.1: </strong> </p>
<ul>
<li> Moved SoundTouch under lesser GPL license (LGPL). This allows
using SoundTouch library in programs that aren't released under GPL
license.</li>
<li> Changed MMX routine organiation so that MMX optimized routines
are now implemented in classes that are derived from the basic classes
having the standard non-mmx routines.</li>
<li> MMX routines to support gcc version 3.</li>
<li> Replaced windows makefiles by script using the .dsw files</li>
</ul>
<p><strong>1.0.1: </strong> </p>
<ul>
<li> "mmx_gcc.cpp": Added "using namespace std" and removed "return
0" from a function with void return value to fix compiler errors when
compiling the library in Solaris environment.</li>
<li> Moved file "FIFOSampleBuffer.h" to "include" directory to allow
accessing the FIFOSampleBuffer class from external files.</li>
</ul>
<p><strong>1.0: </strong> </p>
<ul>
<li> Initial release</li>
</ul>
<h3>5.2. SoundStretch application Change History </h3>
<p><b>1.9:</b></p>
<ul>
<li>Added support for WAV file 'fact' information chunk.</li>
</ul>
<p><b>1.7.0:</b></p>
<ul>
<li>Bugfixes in Wavfile: exception string formatting, avoid getLengthMs() integer
precision overflow, support WAV files using 24/32bit sample format.</li>
</ul>
<p><b>1.5.0:</b></p>
<ul>
<li> Added "-speech" switch to activate algorithm parameters more
suitable for speech processing than the default parameters tuned for
music processing.</li>
</ul>
<p><strong>1.4.0:</strong></p>
<ul>
<li> Moved BPM detection routines from SoundStretch application into
SoundTouch library</li>
<li> Allow using standard input/output pipes as audio processing
input/output streams</li>
</ul>
<p><strong>1.3.0:</strong></p>
<ul>
<li> Simplified accessing WAV files with floating point sample
format.</li>
</ul>
<p><strong>1.2.1: </strong> </p>
<ul>
<li> Fixed 64bit address space bugs.</li>
</ul>
<p><strong>1.2.0: </strong> </p>
<ul>
<li> Added support for 32bit floating point sample data type</li>
<li> Restructured the BPM routines into separate library</li>
<li> Fixed big-endian conversion bugs in WAV file routines (hopefully
:)</li>
</ul>
<p><strong>1.1.1: </strong> </p>
<ul>
<li> Fixed bugs in WAV file reading &amp; added byte-order conversion
for big-endian processors.</li>
<li> Moved SoundStretch source code under 'example' directory to
highlight difference from SoundTouch stuff.</li>
<li> Replaced windows makefiles by script using the .dsw files</li>
<li> Output file name isn't required if output isn't desired (e.g. if
using the switch '-bpm' in plain format only)</li>
</ul>
<p><strong>1.1:</strong></p>
<ul>
<li> Fixed "Release" settings in Microsoft Visual C++ project file
(.dsp)</li>
<li> Added beats-per-minute (BPM) detection routine and command-line
switch "-bpm"</li>
</ul>
<p><strong>1.01: </strong> </p>
<ul>
<li> Initial release</li>
</ul>
<hr>
<h2>6. Acknowledgements </h2>
<p>Kudos for these people who have contributed to development or
submitted bugfixes:</p>
<ul>
<li> Arthur A</li>
<li> Paul Adenot</li>
<li> Richard Ash</li>
<li> Stanislav Brabec</li>
<li> Christian Budde</li>
<li> Jamie Bullock</li>
<li> Chris Bryan</li>
<li> Jacek Caban</li>
<li> Marketa Calabkova</li>
<li> Brian Cameron</li>
<li> Jason Champion</li>
<li> Giuseppe Cigala</li>
<li> David Clark</li>
<li> Patrick Colis</li>
<li> Miquel Colon</li>
<li> Jim Credland</li>
<li> Sandro Cumerlato</li>
<li> Gerry Fan</li>
<li> Justin Frankel</li>
<li> Masa H.</li>
<li> Jason Garland</li>
<li> Takashi Iwai</li>
<li> Thomas Klausner</li>
<li> Lu Zhihe</li>
<li> Luzpaz</li>
<li> Tony Mechelynck </li>
<li> Mathias M&ouml;hl</li>
<li> Yuval Naveh</li>
<li> Mats Palmgren </li>
<li> Chandni Patel</li>
<li> Paulo Pizarro</li>
<li> Andrey Ponomarenko</li>
<li> Blaise Potard</li>
<li> Michael Pruett</li>
<li> Rajeev Puran</li>
<li> RJ Ryan</li>
<li> John Sheehy</li>
<li> Tim Shuttleworth</li>
<li> Albert Sirvent</li>
<li> Tyson Smith</li>
<li> John Stumpo</li>
<li> Mario di Vece</li>
<li> Rémi Verschelde</li>
<li> Katja Vetter</li>
<li> Wu Q.</li>
</ul>
<p>Moral greetings to all other contributors and users also!</p>
<hr>
<h2>7. LICENSE </h2>
<p>SoundTouch audio processing library<br>
Copyright (c) Olli Parviainen</p>
<p>This library is free software; you can redistribute it and/or modify
it under the terms of the GNU Lesser General Public License version 2.1
as published by the Free Software Foundation.</p>
<p>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.</p>
<p>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</p>
<p>---</p>
<p>commercial license alternative also available, contact author for details.</p>
<hr>
</body>
</html>

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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.
/// - Input sound data is decimated to approx 500 Hz to reduce calculation burden,
/// which is basically ok as low (bass) frequencies mostly determine the beat rate.
/// Simple averaging is used for anti-alias filtering because the resulting signal
/// quality isn't of that high importance.
/// - Decimated sound data is enveloped, i.e. the amplitude shape is detected by
/// taking absolute value that's smoothed by sliding average. Signal levels that
/// are below a couple of times the general RMS amplitude level are cut away to
/// leave only notable peaks there.
/// - Repeating sound patterns (e.g. beats) are detected by calculating short-term
/// autocorrelation function of the enveloped signal.
/// - After whole sound data file has been analyzed as above, the bpm level is
/// detected by function 'getBpm' that finds the highest peak of the autocorrelation
/// function, calculates it's precise location and converts this reading to bpm's.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#ifndef _BPMDetect_H_
#define _BPMDetect_H_
#include <vector>
#include "STTypes.h"
#include "FIFOSampleBuffer.h"
namespace soundtouch
{
/// Minimum allowed BPM rate. Used to restrict accepted result above a reasonable limit.
#define MIN_BPM 45
/// Maximum allowed BPM rate range. Used for calculating algorithm parametrs
#define MAX_BPM_RANGE 200
/// Maximum allowed BPM rate range. Used to restrict accepted result below a reasonable limit.
#define MAX_BPM_VALID 190
////////////////////////////////////////////////////////////////////////////////
typedef struct
{
float pos;
float strength;
} BEAT;
class IIR2_filter
{
double coeffs[5];
double prev[5];
public:
IIR2_filter(const double *lpf_coeffs);
float update(float x);
};
/// Class for calculating BPM rate for audio data.
class BPMDetect
{
protected:
/// Auto-correlation accumulator bins.
float *xcorr;
/// Sample average counter.
int decimateCount;
/// Sample average accumulator for FIFO-like decimation.
soundtouch::LONG_SAMPLETYPE decimateSum;
/// Decimate sound by this coefficient to reach approx. 500 Hz.
int decimateBy;
/// Auto-correlation window length
int windowLen;
/// Number of channels (1 = mono, 2 = stereo)
int channels;
/// sample rate
int sampleRate;
/// Beginning of auto-correlation window: Autocorrelation isn't being updated for
/// the first these many correlation bins.
int windowStart;
/// window functions for data preconditioning
float *hamw;
float *hamw2;
// beat detection variables
int pos;
int peakPos;
int beatcorr_ringbuffpos;
int init_scaler;
float peakVal;
float *beatcorr_ringbuff;
/// FIFO-buffer for decimated processing samples.
soundtouch::FIFOSampleBuffer *buffer;
/// Collection of detected beat positions
//BeatCollection beats;
std::vector<BEAT> beats;
// 2nd order low-pass-filter
IIR2_filter beat_lpf;
/// Updates auto-correlation function for given number of decimated samples that
/// are read from the internal 'buffer' pipe (samples aren't removed from the pipe
/// though).
void updateXCorr(int process_samples /// How many samples are processed.
);
/// Decimates samples to approx. 500 Hz.
///
/// \return Number of output samples.
int decimate(soundtouch::SAMPLETYPE *dest, ///< Destination buffer
const soundtouch::SAMPLETYPE *src, ///< Source sample buffer
int numsamples ///< Number of source samples.
);
/// Calculates amplitude envelope for the buffer of samples.
/// Result is output to 'samples'.
void calcEnvelope(soundtouch::SAMPLETYPE *samples, ///< Pointer to input/output data buffer
int numsamples ///< Number of samples in buffer
);
/// remove constant bias from xcorr data
void removeBias();
// Detect individual beat positions
void updateBeatPos(int process_samples);
public:
/// Constructor.
BPMDetect(int numChannels, ///< Number of channels in sample data.
int sampleRate ///< Sample rate in Hz.
);
/// Destructor.
virtual ~BPMDetect();
/// Inputs a block of samples for analyzing: Envelopes the samples and then
/// updates the autocorrelation estimation. When whole song data has been input
/// in smaller blocks using this function, read the resulting bpm with 'getBpm'
/// function.
///
/// Notice that data in 'samples' array can be disrupted in processing.
void inputSamples(const soundtouch::SAMPLETYPE *samples, ///< Pointer to input/working data buffer
int numSamples ///< Number of samples in buffer
);
/// Analyzes the results and returns the BPM rate. Use this function to read result
/// after whole song data has been input to the class by consecutive calls of
/// 'inputSamples' function.
///
/// \return Beats-per-minute rate, or zero if detection failed.
float getBpm();
/// Get beat position arrays. Note: The array includes also really low beat detection values
/// in absence of clear strong beats. Consumer may wish to filter low values away.
/// - "pos" receive array of beat positions
/// - "values" receive array of beat detection strengths
/// - max_num indicates max.size of "pos" and "values" array.
///
/// You can query a suitable array sized by calling this with NULL in "pos" & "values".
///
/// \return number of beats in the arrays.
int getBeats(float *pos, float *strength, int max_num);
};
}
#endif // _BPMDetect_H_

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

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

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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
## I used config/am_include.mk for common definitions
include $(top_srcdir)/config/am_include.mk
pkginclude_HEADERS=FIFOSampleBuffer.h FIFOSamplePipe.h SoundTouch.h STTypes.h BPMDetect.h soundtouch_config.h

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////////////////////////////////////////////////////////////////////////////////
///
/// Common type definitions for SoundTouch audio processing library.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#ifndef STTypes_H
#define STTypes_H
typedef unsigned int uint;
typedef unsigned long ulong;
// Patch for MinGW: on Win64 long is 32-bit
#ifdef _WIN64
typedef unsigned long long ulongptr;
#else
typedef ulong ulongptr;
#endif
// Helper macro for aligning pointer up to next 16-byte boundary
#define SOUNDTOUCH_ALIGN_POINTER_16(x) ( ( (ulongptr)(x) + 15 ) & ~(ulongptr)15 )
#if (defined(__GNUC__) && !defined(ANDROID))
// In GCC, include soundtouch_config.h made by config scritps.
// Skip this in Android compilation that uses GCC but without configure scripts.
#include "soundtouch_config.h"
#endif
namespace soundtouch
{
/// Max allowed number of channels
#define SOUNDTOUCH_MAX_CHANNELS 16
/// Activate these undef's to overrule the possible sampletype
/// setting inherited from some other header file:
//#undef SOUNDTOUCH_INTEGER_SAMPLES
//#undef SOUNDTOUCH_FLOAT_SAMPLES
/// If following flag is defined, always uses multichannel processing
/// routines also for mono and stero sound. This is for routine testing
/// purposes; output should be same with either routines, yet disabling
/// the dedicated mono/stereo processing routines will result in slower
/// runtime performance so recommendation is to keep this off.
// #define USE_MULTICH_ALWAYS
#if (defined(__SOFTFP__) && defined(ANDROID))
// For Android compilation: Force use of Integer samples in case that
// compilation uses soft-floating point emulation - soft-fp is way too slow
#undef SOUNDTOUCH_FLOAT_SAMPLES
#define SOUNDTOUCH_INTEGER_SAMPLES 1
#endif
#if !(SOUNDTOUCH_INTEGER_SAMPLES || SOUNDTOUCH_FLOAT_SAMPLES)
/// Choose either 32bit floating point or 16bit integer sampletype
/// by choosing one of the following defines, unless this selection
/// has already been done in some other file.
////
/// Notes:
/// - In Windows environment, choose the sample format with the
/// following defines.
/// - In GNU environment, the floating point samples are used by
/// default, but integer samples can be chosen by giving the
/// following switch to the configure script:
/// ./configure --enable-integer-samples
/// However, if you still prefer to select the sample format here
/// also in GNU environment, then please #undef the INTEGER_SAMPLE
/// and FLOAT_SAMPLE defines first as in comments above.
//#define SOUNDTOUCH_INTEGER_SAMPLES 1 //< 16bit integer samples
#define SOUNDTOUCH_FLOAT_SAMPLES 1 //< 32bit float samples
#endif
#if (_M_IX86 || __i386__ || __x86_64__ || _M_X64)
/// Define this to allow X86-specific assembler/intrinsic optimizations.
/// Notice that library contains also usual C++ versions of each of these
/// these routines, so if you're having difficulties getting the optimized
/// routines compiled for whatever reason, you may disable these optimizations
/// to make the library compile.
#define SOUNDTOUCH_ALLOW_X86_OPTIMIZATIONS 1
/// In GNU environment, allow the user to override this setting by
/// giving the following switch to the configure script:
/// ./configure --disable-x86-optimizations
/// ./configure --enable-x86-optimizations=no
#ifdef SOUNDTOUCH_DISABLE_X86_OPTIMIZATIONS
#undef SOUNDTOUCH_ALLOW_X86_OPTIMIZATIONS
#endif
#else
/// Always disable optimizations when not using a x86 systems.
#undef SOUNDTOUCH_ALLOW_X86_OPTIMIZATIONS
#endif
// If defined, allows the SIMD-optimized routines to skip unevenly aligned
// memory offsets that can cause performance penalty in some SIMD implementations.
// Causes slight compromise in sound quality.
// #define SOUNDTOUCH_ALLOW_NONEXACT_SIMD_OPTIMIZATION 1
#ifdef SOUNDTOUCH_INTEGER_SAMPLES
// 16bit integer sample type
typedef short SAMPLETYPE;
// data type for sample accumulation: Use 32bit integer to prevent overflows
typedef long LONG_SAMPLETYPE;
#ifdef SOUNDTOUCH_FLOAT_SAMPLES
// check that only one sample type is defined
#error "conflicting sample types defined"
#endif // SOUNDTOUCH_FLOAT_SAMPLES
#ifdef SOUNDTOUCH_ALLOW_X86_OPTIMIZATIONS
// Allow MMX optimizations (not available in X64 mode)
#if (!_M_X64)
#define SOUNDTOUCH_ALLOW_MMX 1
#endif
#endif
#else
// floating point samples
typedef float SAMPLETYPE;
// data type for sample accumulation: Use float also here to enable
// efficient autovectorization
typedef float LONG_SAMPLETYPE;
#ifdef SOUNDTOUCH_ALLOW_X86_OPTIMIZATIONS
// Allow SSE optimizations
#define SOUNDTOUCH_ALLOW_SSE 1
#endif
#endif // SOUNDTOUCH_INTEGER_SAMPLES
#if ((SOUNDTOUCH_ALLOW_SSE) || (__SSE__) || (SOUNDTOUCH_USE_NEON))
#if SOUNDTOUCH_ALLOW_NONEXACT_SIMD_OPTIMIZATION
#define ST_SIMD_AVOID_UNALIGNED
#endif
#endif
}
// define ST_NO_EXCEPTION_HANDLING switch to disable throwing std exceptions:
// #define ST_NO_EXCEPTION_HANDLING 1
#ifdef ST_NO_EXCEPTION_HANDLING
// Exceptions disabled. Throw asserts instead if enabled.
#include <assert.h>
#define ST_THROW_RT_ERROR(x) {assert((const char *)x);}
#else
// use c++ standard exceptions
#include <stdexcept>
#include <string>
#define ST_THROW_RT_ERROR(x) {throw std::runtime_error(x);}
#endif
// When this #define is active, eliminates a clicking sound when the "rate" or "pitch"
// parameter setting crosses from value <1 to >=1 or vice versa during processing.
// Default is off as such crossover is untypical case and involves a slight sound
// quality compromise.
//#define SOUNDTOUCH_PREVENT_CLICK_AT_RATE_CROSSOVER 1
#endif

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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
//
////////////////////////////////////////////////////////////////////////////////
#ifndef SoundTouch_H
#define SoundTouch_H
#include "FIFOSamplePipe.h"
#include "STTypes.h"
namespace soundtouch
{
/// Soundtouch library version string
#define SOUNDTOUCH_VERSION "2.3.1"
/// SoundTouch library version id
#define SOUNDTOUCH_VERSION_ID (20301)
//
// Available setting IDs for the 'setSetting' & 'get_setting' functions:
/// Enable/disable anti-alias filter in pitch transposer (0 = disable)
#define SETTING_USE_AA_FILTER 0
/// Pitch transposer anti-alias filter length (8 .. 128 taps, default = 32)
#define SETTING_AA_FILTER_LENGTH 1
/// Enable/disable quick seeking algorithm in tempo changer routine
/// (enabling quick seeking lowers CPU utilization but causes a minor sound
/// quality compromising)
#define SETTING_USE_QUICKSEEK 2
/// Time-stretch algorithm single processing sequence length in milliseconds. This determines
/// to how long sequences the original sound is chopped in the time-stretch algorithm.
/// See "STTypes.h" or README for more information.
#define SETTING_SEQUENCE_MS 3
/// Time-stretch algorithm seeking window length in milliseconds for algorithm that finds the
/// best possible overlapping location. This determines from how wide window the algorithm
/// may look for an optimal joining location when mixing the sound sequences back together.
/// See "STTypes.h" or README for more information.
#define SETTING_SEEKWINDOW_MS 4
/// Time-stretch algorithm overlap length in milliseconds. When the chopped sound sequences
/// are mixed back together, to form a continuous sound stream, this parameter defines over
/// how long period the two consecutive sequences are let to overlap each other.
/// See "STTypes.h" or README for more information.
#define SETTING_OVERLAP_MS 5
/// Call "getSetting" with this ID to query processing sequence size in samples.
/// This value gives approximate value of how many input samples you'll need to
/// feed into SoundTouch after initial buffering to get out a new batch of
/// output samples.
///
/// This value does not include initial buffering at beginning of a new processing
/// stream, use SETTING_INITIAL_LATENCY to get the initial buffering size.
///
/// Notices:
/// - This is read-only parameter, i.e. setSetting ignores this parameter
/// - This parameter value is not constant but change depending on
/// tempo/pitch/rate/samplerate settings.
#define SETTING_NOMINAL_INPUT_SEQUENCE 6
/// Call "getSetting" with this ID to query nominal average processing output
/// size in samples. This value tells approcimate value how many output samples
/// SoundTouch outputs once it does DSP processing run for a batch of input samples.
///
/// Notices:
/// - This is read-only parameter, i.e. setSetting ignores this parameter
/// - This parameter value is not constant but change depending on
/// tempo/pitch/rate/samplerate settings.
#define SETTING_NOMINAL_OUTPUT_SEQUENCE 7
/// Call "getSetting" with this ID to query initial processing latency, i.e.
/// approx. how many samples you'll need to enter to SoundTouch pipeline before
/// you can expect to get first batch of ready output samples out.
///
/// After the first output batch, you can then expect to get approx.
/// SETTING_NOMINAL_OUTPUT_SEQUENCE ready samples out for every
/// SETTING_NOMINAL_INPUT_SEQUENCE samples that you enter into SoundTouch.
///
/// Example:
/// processing with parameter -tempo=5
/// => initial latency = 5509 samples
/// input sequence = 4167 samples
/// output sequence = 3969 samples
///
/// Accordingly, you can expect to feed in approx. 5509 samples at beginning of
/// the stream, and then you'll get out the first 3969 samples. After that, for
/// every approx. 4167 samples that you'll put in, you'll receive again approx.
/// 3969 samples out.
///
/// This also means that average latency during stream processing is
/// INITIAL_LATENCY-OUTPUT_SEQUENCE/2, in the above example case 5509-3969/2
/// = 3524 samples
///
/// Notices:
/// - This is read-only parameter, i.e. setSetting ignores this parameter
/// - This parameter value is not constant but change depending on
/// tempo/pitch/rate/samplerate settings.
#define SETTING_INITIAL_LATENCY 8
class SoundTouch : public FIFOProcessor
{
private:
/// Rate transposer class instance
class RateTransposer *pRateTransposer;
/// Time-stretch class instance
class TDStretch *pTDStretch;
/// Virtual pitch parameter. Effective rate & tempo are calculated from these parameters.
double virtualRate;
/// Virtual pitch parameter. Effective rate & tempo are calculated from these parameters.
double virtualTempo;
/// Virtual pitch parameter. Effective rate & tempo are calculated from these parameters.
double virtualPitch;
/// Flag: Has sample rate been set?
bool bSrateSet;
/// Accumulator for how many samples in total will be expected as output vs. samples put in,
/// considering current processing settings.
double samplesExpectedOut;
/// Accumulator for how many samples in total have been read out from the processing so far
long samplesOutput;
/// Calculates effective rate & tempo valuescfrom 'virtualRate', 'virtualTempo' and
/// 'virtualPitch' parameters.
void calcEffectiveRateAndTempo();
protected :
/// Number of channels
uint channels;
/// Effective 'rate' value calculated from 'virtualRate', 'virtualTempo' and 'virtualPitch'
double rate;
/// Effective 'tempo' value calculated from 'virtualRate', 'virtualTempo' and 'virtualPitch'
double tempo;
public:
SoundTouch();
virtual ~SoundTouch();
/// Get SoundTouch library version string
static const char *getVersionString();
/// Get SoundTouch library version Id
static uint getVersionId();
/// Sets new rate control value. Normal rate = 1.0, smaller values
/// represent slower rate, larger faster rates.
void setRate(double newRate);
/// Sets new tempo control value. Normal tempo = 1.0, smaller values
/// represent slower tempo, larger faster tempo.
void setTempo(double newTempo);
/// Sets new rate control value as a difference in percents compared
/// to the original rate (-50 .. +100 %)
void setRateChange(double newRate);
/// Sets new tempo control value as a difference in percents compared
/// to the original tempo (-50 .. +100 %)
void setTempoChange(double newTempo);
/// Sets new pitch control value. Original pitch = 1.0, smaller values
/// represent lower pitches, larger values higher pitch.
void setPitch(double newPitch);
/// Sets pitch change in octaves compared to the original pitch
/// (-1.00 .. +1.00)
void setPitchOctaves(double newPitch);
/// Sets pitch change in semi-tones compared to the original pitch
/// (-12 .. +12)
void setPitchSemiTones(int newPitch);
void setPitchSemiTones(double newPitch);
/// Sets the number of channels, 1 = mono, 2 = stereo
void setChannels(uint numChannels);
/// Sets sample rate.
void setSampleRate(uint srate);
/// Get ratio between input and output audio durations, useful for calculating
/// processed output duration: if you'll process a stream of N samples, then
/// you can expect to get out N * getInputOutputSampleRatio() samples.
///
/// This ratio will give accurate target duration ratio for a full audio track,
/// given that the the whole track is processed with same processing parameters.
///
/// If this ratio is applied to calculate intermediate offsets inside a processing
/// stream, then this ratio is approximate and can deviate +- some tens of milliseconds
/// from ideal offset, yet by end of the audio stream the duration ratio will become
/// exact.
///
/// Example: if processing with parameters "-tempo=15 -pitch=-3", the function
/// will return value 0.8695652... Now, if processing an audio stream whose duration
/// is exactly one million audio samples, then you can expect the processed
/// output duration be 0.869565 * 1000000 = 869565 samples.
double getInputOutputSampleRatio();
/// Flushes the last samples from the processing pipeline to the output.
/// Clears also the internal processing buffers.
//
/// Note: This function is meant for extracting the last samples of a sound
/// stream. This function may introduce additional blank samples in the end
/// of the sound stream, and thus it's not recommended to call this function
/// in the middle of a sound stream.
void flush();
/// Adds 'numSamples' pcs of samples from the 'samples' memory position into
/// the input of the object. Notice that sample rate _has_to_ be set before
/// calling this function, otherwise throws a runtime_error exception.
virtual void putSamples(
const SAMPLETYPE *samples, ///< Pointer to sample buffer.
uint numSamples ///< Number of samples in buffer. Notice
///< that in case of stereo-sound a single sample
///< contains data for both channels.
);
/// Output samples from beginning of the sample buffer. Copies requested samples to
/// output buffer and removes them from the sample buffer. If there are less than
/// 'numsample' samples in the buffer, returns all that available.
///
/// \return Number of samples returned.
virtual uint receiveSamples(SAMPLETYPE *output, ///< Buffer where to copy output samples.
uint maxSamples ///< How many samples to receive at max.
);
/// Adjusts book-keeping so that given number of samples are removed from beginning of the
/// sample buffer without copying them anywhere.
///
/// Used to reduce the number of samples in the buffer when accessing the sample buffer directly
/// with 'ptrBegin' function.
virtual uint receiveSamples(uint maxSamples ///< Remove this many samples from the beginning of pipe.
);
/// Clears all the samples in the object's output and internal processing
/// buffers.
virtual void clear();
/// Changes a setting controlling the processing system behaviour. See the
/// 'SETTING_...' defines for available setting ID's.
///
/// \return 'true' if the setting was successfully changed
bool setSetting(int settingId, ///< Setting ID number. see SETTING_... defines.
int value ///< New setting value.
);
/// Reads a setting controlling the processing system behaviour. See the
/// 'SETTING_...' defines for available setting ID's.
///
/// \return the setting value.
int getSetting(int settingId ///< Setting ID number, see SETTING_... defines.
) const;
/// Returns number of samples currently unprocessed.
virtual uint numUnprocessedSamples() const;
/// Return number of channels
uint numChannels() const
{
return channels;
}
/// Other handy functions that are implemented in the ancestor classes (see
/// classes 'FIFOProcessor' and 'FIFOSamplePipe')
///
/// - receiveSamples() : Use this function to receive 'ready' processed samples from SoundTouch.
/// - numSamples() : Get number of 'ready' samples that can be received with
/// function 'receiveSamples()'
/// - isEmpty() : Returns nonzero if there aren't any 'ready' samples.
/// - clear() : Clears all samples from ready/processing buffers.
};
}
#endif

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

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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 == NULL) return;
for (int i = 0; i < len; i ++)
{
double temp = coeffs[i];
fprintf(fptr, "%lf\n", temp);
}
fclose(fptr);
}
#else
#define _DEBUG_SAVE_AAFIR_COEFFS(x, y)
#endif
/*****************************************************************************
*
* Implementation of the class 'AAFilter'
*
*****************************************************************************/
AAFilter::AAFilter(uint len)
{
pFIR = FIRFilter::newInstance();
cutoffFreq = 0.5;
setLength(len);
}
AAFilter::~AAFilter()
{
delete pFIR;
}
// Sets new anti-alias filter cut-off edge frequency, scaled to
// sampling frequency (nyquist frequency = 0.5).
// The filter will cut frequencies higher than the given frequency.
void AAFilter::setCutoffFreq(double newCutoffFreq)
{
cutoffFreq = newCutoffFreq;
calculateCoeffs();
}
// Sets number of FIR filter taps
void AAFilter::setLength(uint newLength)
{
length = newLength;
calculateCoeffs();
}
// Calculates coefficients for a low-pass FIR filter using Hamming window
void AAFilter::calculateCoeffs()
{
uint i;
double cntTemp, temp, tempCoeff,h, w;
double wc;
double scaleCoeff, sum;
double *work;
SAMPLETYPE *coeffs;
assert(length >= 2);
assert(length % 4 == 0);
assert(cutoffFreq >= 0);
assert(cutoffFreq <= 0.5);
work = new double[length];
coeffs = new SAMPLETYPE[length];
wc = 2.0 * PI * cutoffFreq;
tempCoeff = TWOPI / (double)length;
sum = 0;
for (i = 0; i < length; i ++)
{
cntTemp = (double)i - (double)(length / 2);
temp = cntTemp * wc;
if (temp != 0)
{
h = sin(temp) / temp; // sinc function
}
else
{
h = 1.0;
}
w = 0.54 + 0.46 * cos(tempCoeff * cntTemp); // hamming window
temp = w * h;
work[i] = temp;
// calc net sum of coefficients
sum += temp;
}
// ensure the sum of coefficients is larger than zero
assert(sum > 0);
// ensure we've really designed a lowpass filter...
assert(work[length/2] > 0);
assert(work[length/2 + 1] > -1e-6);
assert(work[length/2 - 1] > -1e-6);
// Calculate a scaling coefficient in such a way that the result can be
// divided by 16384
scaleCoeff = 16384.0f / sum;
for (i = 0; i < length; i ++)
{
temp = work[i] * scaleCoeff;
// scale & round to nearest integer
temp += (temp >= 0) ? 0.5 : -0.5;
// ensure no overfloods
assert(temp >= -32768 && temp <= 32767);
coeffs[i] = (SAMPLETYPE)temp;
}
// Set coefficients. Use divide factor 14 => divide result by 2^14 = 16384
pFIR->setCoefficients(coeffs, length, 14);
_DEBUG_SAVE_AAFIR_COEFFS(coeffs, length);
delete[] work;
delete[] coeffs;
}
// Applies the filter to the given sequence of samples.
// Note : The amount of outputted samples is by value of 'filter length'
// smaller than the amount of input samples.
uint AAFilter::evaluate(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples, uint numChannels) const
{
return pFIR->evaluate(dest, src, numSamples, numChannels);
}
/// Applies the filter to the given src & dest pipes, so that processed amount of
/// samples get removed from src, and produced amount added to dest
/// Note : The amount of outputted samples is by value of 'filter length'
/// smaller than the amount of input samples.
uint AAFilter::evaluate(FIFOSampleBuffer &dest, FIFOSampleBuffer &src) const
{
SAMPLETYPE *pdest;
const SAMPLETYPE *psrc;
uint numSrcSamples;
uint result;
int numChannels = src.getChannels();
assert(numChannels == dest.getChannels());
numSrcSamples = src.numSamples();
psrc = src.ptrBegin();
pdest = dest.ptrEnd(numSrcSamples);
result = pFIR->evaluate(pdest, psrc, numSrcSamples, numChannels);
src.receiveSamples(result);
dest.putSamples(result);
return result;
}
uint AAFilter::getLength() const
{
return pFIR->getLength();
}

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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

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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++)
{
LONG_SAMPLETYPE max = 0;
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 NULL 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 NULL, 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 = NULL;
bufferUnaligned = NULL;
samplesInBuffer = 0;
bufferPos = 0;
channels = (uint)numChannels;
ensureCapacity(32); // allocate initial capacity
}
// destructor
FIFOSampleBuffer::~FIFOSampleBuffer()
{
delete[] bufferUnaligned;
bufferUnaligned = NULL;
buffer = NULL;
}
// Sets number of channels, 1 = mono, 2 = stereo
void FIFOSampleBuffer::setChannels(int numChannels)
{
uint usedBytes;
if (!verifyNumberOfChannels(numChannels)) return;
usedBytes = channels * samplesInBuffer;
channels = (uint)numChannels;
samplesInBuffer = usedBytes / channels;
}
// if output location pointer 'bufferPos' isn't zero, 'rewinds' the buffer and
// zeroes this pointer by copying samples from the 'bufferPos' pointer
// location on to the beginning of the buffer.
void FIFOSampleBuffer::rewind()
{
if (buffer && bufferPos)
{
memmove(buffer, ptrBegin(), sizeof(SAMPLETYPE) * channels * samplesInBuffer);
bufferPos = 0;
}
}
// Adds 'numSamples' pcs of samples from the 'samples' memory position to
// the sample buffer.
void FIFOSampleBuffer::putSamples(const SAMPLETYPE *samples, uint nSamples)
{
memcpy(ptrEnd(nSamples), samples, sizeof(SAMPLETYPE) * nSamples * channels);
samplesInBuffer += nSamples;
}
// Increases the number of samples in the buffer without copying any actual
// samples.
//
// This function is used to update the number of samples in the sample buffer
// when accessing the buffer directly with 'ptrEnd' function. Please be
// careful though!
void FIFOSampleBuffer::putSamples(uint nSamples)
{
uint req;
req = samplesInBuffer + nSamples;
ensureCapacity(req);
samplesInBuffer += nSamples;
}
// Returns a pointer to the end of the used part of the sample buffer (i.e.
// where the new samples are to be inserted). This function may be used for
// inserting new samples into the sample buffer directly. Please be careful!
//
// Parameter 'slackCapacity' tells the function how much free capacity (in
// terms of samples) there _at least_ should be, in order to the caller to
// successfully insert all the required samples to the buffer. When necessary,
// the function grows the buffer size to comply with this requirement.
//
// When using this function as means for inserting new samples, also remember
// to increase the sample count afterwards, by calling the
// 'putSamples(numSamples)' function.
SAMPLETYPE *FIFOSampleBuffer::ptrEnd(uint slackCapacity)
{
ensureCapacity(samplesInBuffer + slackCapacity);
return buffer + samplesInBuffer * channels;
}
// Returns a pointer to the beginning of the currently non-outputted samples.
// This function is provided for accessing the output samples directly.
// Please be careful!
//
// When using this function to output samples, also remember to 'remove' the
// outputted samples from the buffer by calling the
// 'receiveSamples(numSamples)' function
SAMPLETYPE *FIFOSampleBuffer::ptrBegin()
{
assert(buffer);
return buffer + bufferPos * channels;
}
// Ensures that the buffer has enough capacity, i.e. space for _at least_
// 'capacityRequirement' number of samples. The buffer is grown in steps of
// 4 kilobytes to eliminate the need for frequently growing up the buffer,
// as well as to round the buffer size up to the virtual memory page size.
void FIFOSampleBuffer::ensureCapacity(uint capacityRequirement)
{
SAMPLETYPE *tempUnaligned, *temp;
if (capacityRequirement > getCapacity())
{
// enlarge the buffer in 4kbyte steps (round up to next 4k boundary)
sizeInBytes = (capacityRequirement * channels * sizeof(SAMPLETYPE) + 4095) & (uint)-4096;
assert(sizeInBytes % 2 == 0);
tempUnaligned = new SAMPLETYPE[sizeInBytes / sizeof(SAMPLETYPE) + 16 / sizeof(SAMPLETYPE)];
if (tempUnaligned == NULL)
{
ST_THROW_RT_ERROR("Couldn't allocate memory!\n");
}
// Align the buffer to begin at 16byte cache line boundary for optimal performance
temp = (SAMPLETYPE *)SOUNDTOUCH_ALIGN_POINTER_16(tempUnaligned);
if (samplesInBuffer)
{
memcpy(temp, ptrBegin(), samplesInBuffer * channels * sizeof(SAMPLETYPE));
}
delete[] bufferUnaligned;
buffer = temp;
bufferUnaligned = tempUnaligned;
bufferPos = 0;
}
else
{
// simply rewind the buffer (if necessary)
rewind();
}
}
// Returns the current buffer capacity in terms of samples
uint FIFOSampleBuffer::getCapacity() const
{
return sizeInBytes / (channels * sizeof(SAMPLETYPE));
}
// Returns the number of samples currently in the buffer
uint FIFOSampleBuffer::numSamples() const
{
return samplesInBuffer;
}
// Output samples from beginning of the sample buffer. Copies demanded number
// of samples to output and removes them from the sample buffer. If there
// are less than 'numsample' samples in the buffer, returns all available.
//
// Returns number of samples copied.
uint FIFOSampleBuffer::receiveSamples(SAMPLETYPE *output, uint maxSamples)
{
uint num;
num = (maxSamples > samplesInBuffer) ? samplesInBuffer : maxSamples;
memcpy(output, ptrBegin(), channels * sizeof(SAMPLETYPE) * num);
return receiveSamples(num);
}
// Removes samples from the beginning of the sample buffer without copying them
// anywhere. Used to reduce the number of samples in the buffer, when accessing
// the sample buffer with the 'ptrBegin' function.
uint FIFOSampleBuffer::receiveSamples(uint maxSamples)
{
if (maxSamples >= samplesInBuffer)
{
uint temp;
temp = samplesInBuffer;
samplesInBuffer = 0;
return temp;
}
samplesInBuffer -= maxSamples;
bufferPos += maxSamples;
return maxSamples;
}
// Returns nonzero if the sample buffer is empty
int FIFOSampleBuffer::isEmpty() const
{
return (samplesInBuffer == 0) ? 1 : 0;
}
// Clears the sample buffer
void FIFOSampleBuffer::clear()
{
samplesInBuffer = 0;
bufferPos = 0;
}
/// allow trimming (downwards) amount of samples in pipeline.
/// Returns adjusted amount of samples
uint FIFOSampleBuffer::adjustAmountOfSamples(uint numSamples)
{
if (numSamples < samplesInBuffer)
{
samplesInBuffer = numSamples;
}
return samplesInBuffer;
}
/// 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 = NULL;
filterCoeffsStereo = NULL;
}
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;
#ifdef SOUNDTOUCH_FLOAT_SAMPLES
// when using floating point samples, use a scaler instead of a divider
// because division is much slower operation than multiplying.
double dScaler = 1.0 / (double)resultDivider;
#endif
// hint compiler autovectorization that loop length is divisible by 8
int ilength = length & -8;
assert((length != 0) && (length == ilength) && (src != NULL) && (dest != NULL) && (filterCoeffs != NULL));
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 (int 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;
#ifdef SOUNDTOUCH_FLOAT_SAMPLES
// when using floating point samples, use a scaler instead of a divider
// because division is much slower operation than multiplying.
double dScaler = 1.0 / (double)resultDivider;
#endif
// hint compiler autovectorization that loop length is divisible by 8
int ilength = length & -8;
assert(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;
#ifdef SOUNDTOUCH_FLOAT_SAMPLES
// when using floating point samples, use a scaler instead of a divider
// because division is much slower operation than multiplying.
double dScaler = 1.0 / (double)resultDivider;
#endif
assert(length != 0);
assert(src != NULL);
assert(dest != NULL);
assert(filterCoeffs != NULL);
assert(numChannels < 16);
// 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 s)
{
// Notice! don't use "new FIRFilter" directly, use "newInstance" to create a new instance instead!
ST_THROW_RT_ERROR("Error in FIRFilter::new: Don't use 'new FIRFilter', use 'newInstance' member instead!");
return newInstance();
}
FIRFilter * FIRFilter::newInstance()
{
uint uExtensions;
uExtensions = detectCPUextensions();
// Check if MMX/SSE instruction set extensions supported by CPU
#ifdef SOUNDTOUCH_ALLOW_MMX
// MMX routines available only with integer sample types
if (uExtensions & SUPPORT_MMX)
{
return ::new FIRFilterMMX;
}
else
#endif // SOUNDTOUCH_ALLOW_MMX
#ifdef SOUNDTOUCH_ALLOW_SSE
if (uExtensions & SUPPORT_SSE)
{
// SSE support
return ::new FIRFilterSSE;
}
else
#endif // SOUNDTOUCH_ALLOW_SSE
{
// ISA optimizations not supported, use plain C version
return ::new FIRFilter;
}
}

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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;
public:
FIRFilterMMX();
~FIRFilterMMX();
virtual void setCoefficients(const short *coeffs, uint newLength, uint uResultDivFactor);
};
#endif // SOUNDTOUCH_ALLOW_MMX
#ifdef SOUNDTOUCH_ALLOW_SSE
/// Class that implements SSE optimized functions exclusive for floating point samples type.
class FIRFilterSSE : public FIRFilter
{
protected:
float *filterCoeffsUnalign;
float *filterCoeffsAlign;
virtual uint evaluateFilterStereo(float *dest, const float *src, uint numSamples) const;
public:
FIRFilterSSE();
~FIRFilterSSE();
virtual void setCoefficients(const float *coeffs, uint newLength, uint uResultDivFactor);
};
#endif // SOUNDTOUCH_ALLOW_SSE
}
#endif // FIRFilter_H

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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);
virtual int transposeStereo(SAMPLETYPE *dest,
const SAMPLETYPE *src,
int &srcSamples);
virtual int transposeMulti(SAMPLETYPE *dest,
const SAMPLETYPE *src,
int &srcSamples);
double fract;
public:
InterpolateCubic();
virtual void resetRegisters();
int getLatency() const
{
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);
virtual int transposeStereo(SAMPLETYPE *dest,
const SAMPLETYPE *src,
int &srcSamples);
virtual int transposeMulti(SAMPLETYPE *dest, const SAMPLETYPE *src, int &srcSamples);
public:
InterpolateLinearInteger();
/// Sets new target rate. Normal rate = 1.0, smaller values represent slower
/// rate, larger faster rates.
virtual void setRate(double newRate);
virtual void resetRegisters();
int getLatency() const
{
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 *pdest,
const SAMPLETYPE *psrc,
int &srcSamples)
{
// 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);
int transposeStereo(SAMPLETYPE *dest,
const SAMPLETYPE *src,
int &srcSamples);
int transposeMulti(SAMPLETYPE *dest,
const SAMPLETYPE *src,
int &srcSamples);
double fract;
public:
InterpolateShannon();
void resetRegisters();
int getLatency() const
{
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)
{
uint count;
if (nSamples == 0) return;
// Store samples to input buffer
inputBuffer.putSamples(src, nSamples);
// If anti-alias filter is turned off, simply transpose without applying
// the filter
if (bUseAAFilter == false)
{
count = pTransposer->transpose(outputBuffer, inputBuffer);
return;
}
assert(pAAFilter);
// Transpose with anti-alias filter
if (pTransposer->rate < 1.0f)
{
// If the parameter 'Rate' value is smaller than 1, first transpose
// the samples and then apply the anti-alias filter to remove aliasing.
// Transpose the samples, store the result to end of "midBuffer"
pTransposer->transpose(midBuffer, inputBuffer);
// Apply the anti-alias filter for transposed samples in midBuffer
pAAFilter->evaluate(outputBuffer, midBuffer);
}
else
{
// If the parameter 'Rate' value is larger than 1, first apply the
// anti-alias filter to remove high frequencies (prevent them from folding
// over the lover frequencies), then transpose.
// Apply the anti-alias filter for samples in inputBuffer
pAAFilter->evaluate(midBuffer, inputBuffer);
// Transpose the AA-filtered samples in "midBuffer"
pTransposer->transpose(outputBuffer, midBuffer);
}
}
// Sets the number of channels, 1 = mono, 2 = stereo
void RateTransposer::setChannels(int nChannels)
{
if (!verifyNumberOfChannels(nChannels) ||
(pTransposer->numChannels == nChannels)) return;
pTransposer->setChannels(nChannels);
inputBuffer.setChannels(nChannels);
midBuffer.setChannels(nChannels);
outputBuffer.setChannels(nChannels);
}
// Clears all the samples in the object
void RateTransposer::clear()
{
outputBuffer.clear();
midBuffer.clear();
inputBuffer.clear();
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 NULL;
}
#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();
/// Returns the output buffer object
FIFOSamplePipe *getOutput() { return &outputBuffer; };
/// Return anti-alias filter object
AAFilter *getAAFilter();
/// Enables/disables the anti-alias filter. Zero to disable, nonzero to enable
void enableAAFilter(bool newMode);
/// Returns nonzero if anti-alias filter is enabled.
bool isAAFilterEnabled() const;
/// Sets new target rate. Normal rate = 1.0, smaller values represent slower
/// rate, larger faster rates.
virtual void setRate(double newRate);
/// Sets the number of channels, 1 = mono, 2 = stereo
void setChannels(int channels);
/// Adds 'numSamples' pcs of samples from the 'samples' memory position into
/// the input of the object.
void putSamples(const SAMPLETYPE *samples, uint numSamples);
/// Clears all the samples in the object
void clear();
/// Returns nonzero if there aren't any samples available for outputting.
int isEmpty() const;
/// Return approximate initial input-output latency
int getLatency() const;
};
}
#endif

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

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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 = NULL;
filterCoeffsUnalign = NULL;
}
FIRFilterMMX::~FIRFilterMMX()
{
delete[] filterCoeffsUnalign;
}
// (overloaded) Calculates filter coefficients for MMX routine
void FIRFilterMMX::setCoefficients(const short *coeffs, uint newLength, uint uResultDivFactor)
{
uint i;
FIRFilter::setCoefficients(coeffs, newLength, uResultDivFactor);
// Ensure that filter coeffs array is aligned to 16-byte boundary
delete[] filterCoeffsUnalign;
filterCoeffsUnalign = new short[2 * newLength + 8];
filterCoeffsAlign = (short *)SOUNDTOUCH_ALIGN_POINTER_16(filterCoeffsUnalign);
// rearrange the filter coefficients for mmx routines
for (i = 0;i < length; i += 4)
{
filterCoeffsAlign[2 * i + 0] = coeffs[i + 0];
filterCoeffsAlign[2 * i + 1] = coeffs[i + 2];
filterCoeffsAlign[2 * i + 2] = coeffs[i + 0];
filterCoeffsAlign[2 * i + 3] = coeffs[i + 2];
filterCoeffsAlign[2 * i + 4] = coeffs[i + 1];
filterCoeffsAlign[2 * i + 5] = coeffs[i + 3];
filterCoeffsAlign[2 * i + 6] = coeffs[i + 1];
filterCoeffsAlign[2 * i + 7] = coeffs[i + 3];
}
}
// mmx-optimized version of the filter routine for stereo sound
uint FIRFilterMMX::evaluateFilterStereo(short *dest, const short *src, uint numSamples) const
{
// Create stack copies of the needed member variables for asm routines :
uint i, j;
__m64 *pVdest = (__m64*)dest;
if (length < 2) return 0;
for (i = 0; i < (numSamples - length) / 2; i ++)
{
__m64 accu1;
__m64 accu2;
const __m64 *pVsrc = (const __m64*)src;
const __m64 *pVfilter = (const __m64*)filterCoeffsAlign;
accu1 = accu2 = _mm_setzero_si64();
for (j = 0; j < lengthDiv8 * 2; j ++)
{
__m64 temp1, temp2;
temp1 = _mm_unpacklo_pi16(pVsrc[0], pVsrc[1]); // = l2 l0 r2 r0
temp2 = _mm_unpackhi_pi16(pVsrc[0], pVsrc[1]); // = l3 l1 r3 r1
accu1 = _mm_add_pi32(accu1, _mm_madd_pi16(temp1, pVfilter[0])); // += l2*f2+l0*f0 r2*f2+r0*f0
accu1 = _mm_add_pi32(accu1, _mm_madd_pi16(temp2, pVfilter[1])); // += l3*f3+l1*f1 r3*f3+r1*f1
temp1 = _mm_unpacklo_pi16(pVsrc[1], pVsrc[2]); // = l4 l2 r4 r2
accu2 = _mm_add_pi32(accu2, _mm_madd_pi16(temp2, pVfilter[0])); // += l3*f2+l1*f0 r3*f2+r1*f0
accu2 = _mm_add_pi32(accu2, _mm_madd_pi16(temp1, pVfilter[1])); // += l4*f3+l2*f1 r4*f3+r2*f1
// accu1 += l2*f2+l0*f0 r2*f2+r0*f0
// += l3*f3+l1*f1 r3*f3+r1*f1
// accu2 += l3*f2+l1*f0 r3*f2+r1*f0
// l4*f3+l2*f1 r4*f3+r2*f1
pVfilter += 2;
pVsrc += 2;
}
// accu >>= resultDivFactor
accu1 = _mm_srai_pi32(accu1, resultDivFactor);
accu2 = _mm_srai_pi32(accu2, resultDivFactor);
// pack 2*2*32bits => 4*16 bits
pVdest[0] = _mm_packs_pi32(accu1, accu2);
src += 4;
pVdest ++;
}
_m_empty(); // clear emms state
return (numSamples & 0xfffffffe) - length;
}
#else
// workaround to not complain about empty module
bool _dontcomplain_mmx_empty;
#endif // SOUNDTOUCH_ALLOW_MMX

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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 = NULL;
filterCoeffsUnalign = NULL;
}
FIRFilterSSE::~FIRFilterSSE()
{
delete[] filterCoeffsUnalign;
filterCoeffsAlign = NULL;
filterCoeffsUnalign = NULL;
}
// (overloaded) Calculates filter coefficients for SSE routine
void FIRFilterSSE::setCoefficients(const float *coeffs, uint newLength, uint uResultDivFactor)
{
uint i;
float fDivider;
FIRFilter::setCoefficients(coeffs, newLength, uResultDivFactor);
// Scale the filter coefficients so that it won't be necessary to scale the filtering result
// also rearrange coefficients suitably for SSE
// Ensure that filter coeffs array is aligned to 16-byte boundary
delete[] filterCoeffsUnalign;
filterCoeffsUnalign = new float[2 * newLength + 4];
filterCoeffsAlign = (float *)SOUNDTOUCH_ALIGN_POINTER_16(filterCoeffsUnalign);
fDivider = (float)resultDivider;
// rearrange the filter coefficients for mmx routines
for (i = 0; i < newLength; i ++)
{
filterCoeffsAlign[2 * i + 0] =
filterCoeffsAlign[2 * i + 1] = coeffs[i + 0] / fDivider;
}
}
// SSE-optimized version of the filter routine for stereo sound
uint FIRFilterSSE::evaluateFilterStereo(float *dest, const float *source, uint numSamples) const
{
int count = (int)((numSamples - length) & (uint)-2);
int j;
assert(count % 2 == 0);
if (count < 2) return 0;
assert(source != NULL);
assert(dest != NULL);
assert((length % 8) == 0);
assert(filterCoeffsAlign != NULL);
assert(((ulongptr)filterCoeffsAlign) % 16 == 0);
// filter is evaluated for two stereo samples with each iteration, thus use of 'j += 2'
#pragma omp parallel for
for (j = 0; j < count; j += 2)
{
const float *pSrc;
float *pDest;
const __m128 *pFil;
__m128 sum1, sum2;
uint i;
pSrc = (const float*)source + j * 2; // source audio data
pDest = dest + j * 2; // destination audio data
pFil = (const __m128*)filterCoeffsAlign; // filter coefficients. NOTE: Assumes coefficients
// are aligned to 16-byte boundary
sum1 = sum2 = _mm_setzero_ps();
for (i = 0; i < length / 8; i ++)
{
// Unroll loop for efficiency & calculate filter for 2*2 stereo samples
// at each pass
// sum1 is accu for 2*2 filtered stereo sound data at the primary sound data offset
// sum2 is accu for 2*2 filtered stereo sound data for the next sound sample offset.
sum1 = _mm_add_ps(sum1, _mm_mul_ps(_mm_loadu_ps(pSrc) , pFil[0]));
sum2 = _mm_add_ps(sum2, _mm_mul_ps(_mm_loadu_ps(pSrc + 2), pFil[0]));
sum1 = _mm_add_ps(sum1, _mm_mul_ps(_mm_loadu_ps(pSrc + 4), pFil[1]));
sum2 = _mm_add_ps(sum2, _mm_mul_ps(_mm_loadu_ps(pSrc + 6), pFil[1]));
sum1 = _mm_add_ps(sum1, _mm_mul_ps(_mm_loadu_ps(pSrc + 8) , pFil[2]));
sum2 = _mm_add_ps(sum2, _mm_mul_ps(_mm_loadu_ps(pSrc + 10), pFil[2]));
sum1 = _mm_add_ps(sum1, _mm_mul_ps(_mm_loadu_ps(pSrc + 12), pFil[3]));
sum2 = _mm_add_ps(sum2, _mm_mul_ps(_mm_loadu_ps(pSrc + 14), pFil[3]));
pSrc += 16;
pFil += 4;
}
// Now sum1 and sum2 both have a filtered 2-channel sample each, but we still need
// to sum the two hi- and lo-floats of these registers together.
// post-shuffle & add the filtered values and store to dest.
_mm_storeu_ps(pDest, _mm_add_ps(
_mm_shuffle_ps(sum1, sum2, _MM_SHUFFLE(1,0,3,2)), // s2_1 s2_0 s1_3 s1_2
_mm_shuffle_ps(sum1, sum2, _MM_SHUFFLE(3,2,1,0)) // s2_3 s2_2 s1_1 s1_0
));
}
// Ideas for further improvement:
// 1. If it could be guaranteed that 'source' were always aligned to 16-byte
// boundary, a faster aligned '_mm_load_ps' instruction could be used.
// 2. If it could be guaranteed that 'dest' were always aligned to 16-byte
// boundary, a faster '_mm_store_ps' instruction could be used.
return (uint)count;
/* original routine in C-language. please notice the C-version has differently
organized coefficients though.
double suml1, suml2;
double sumr1, sumr2;
uint i, j;
for (j = 0; j < count; j += 2)
{
const float *ptr;
const float *pFil;
suml1 = sumr1 = 0.0;
suml2 = sumr2 = 0.0;
ptr = src;
pFil = filterCoeffs;
for (i = 0; i < lengthLocal; i ++)
{
// unroll loop for efficiency.
suml1 += ptr[0] * pFil[0] +
ptr[2] * pFil[2] +
ptr[4] * pFil[4] +
ptr[6] * pFil[6];
sumr1 += ptr[1] * pFil[1] +
ptr[3] * pFil[3] +
ptr[5] * pFil[5] +
ptr[7] * pFil[7];
suml2 += ptr[8] * pFil[0] +
ptr[10] * pFil[2] +
ptr[12] * pFil[4] +
ptr[14] * pFil[6];
sumr2 += ptr[9] * pFil[1] +
ptr[11] * pFil[3] +
ptr[13] * pFil[5] +
ptr[15] * pFil[7];
ptr += 16;
pFil += 8;
}
dest[0] = (float)suml1;
dest[1] = (float)sumr1;
dest[2] = (float)suml2;
dest[3] = (float)sumr2;
src += 4;
dest += 4;
}
*/
}
#endif // SOUNDTOUCH_ALLOW_SSE

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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 == NULL)
{
delete tmp;
tmp = NULL;
}
}
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 = NULL;
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 void __cdecl soundtouch_setChannels(HANDLE h, uint numChannels)
{
STHANDLE *sth = (STHANDLE*)h;
if (sth->dwMagic != STMAGIC) return;
sth->pst->setChannels(numChannels);
}
/// Sets sample rate.
SOUNDTOUCHDLL_API void __cdecl soundtouch_setSampleRate(HANDLE h, uint srate)
{
STHANDLE *sth = (STHANDLE*)h;
if (sth->dwMagic != STMAGIC) return;
sth->pst->setSampleRate(srate);
}
/// Flushes the last samples from the processing pipeline to the output.
/// Clears also the internal processing buffers.
//
/// Note: This function is meant for extracting the last samples of a sound
/// stream. This function may introduce additional blank samples in the end
/// of the sound stream, and thus it's not recommended to call this function
/// in the middle of a sound stream.
SOUNDTOUCHDLL_API void __cdecl soundtouch_flush(HANDLE h)
{
STHANDLE *sth = (STHANDLE*)h;
if (sth->dwMagic != STMAGIC) return;
sth->pst->flush();
}
/// Adds 'numSamples' pcs of samples from the 'samples' memory position into
/// the input of the object. Notice that sample rate _has_to_ be set before
/// calling this function, otherwise throws a runtime_error exception.
SOUNDTOUCHDLL_API void __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;
sth->pst->putSamples(samples, numSamples);
}
/// 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();
}
/// Adjusts book-keeping so that given number of samples are removed from beginning of the
/// sample buffer without copying them anywhere.
///
/// Used to reduce the number of samples in the buffer when accessing the sample buffer directly
/// with 'ptrBegin' function.
SOUNDTOUCHDLL_API uint __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 == NULL)
{
// 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;
tmp->pbpm = new BPMDetect(numChannels, sampleRate);
if (tmp->pbpm == NULL)
{
delete tmp;
tmp = NULL;
}
}
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 = NULL;
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();
}

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Src/external_dependencies/openmpt-trunk/include/soundtouch/source/SoundTouchDLL/SoundTouchDLL.h 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
//
////////////////////////////////////////////////////////////////////////////////
#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 void __cdecl soundtouch_setChannels(HANDLE h, unsigned int numChannels);
/// Sets sample rate.
SOUNDTOUCHDLL_API void __cdecl soundtouch_setSampleRate(HANDLE h, unsigned int srate);
/// Flushes the last samples from the processing pipeline to the output.
/// Clears also the internal processing buffers.
//
/// Note: This function is meant for extracting the last samples of a sound
/// stream. This function may introduce additional blank samples in the end
/// of the sound stream, and thus it's not recommended to call this function
/// in the middle of a sound stream.
SOUNDTOUCHDLL_API void __cdecl soundtouch_flush(HANDLE h);
/// Adds 'numSamples' pcs of samples from the 'samples' memory position into
/// the input of the object. Notice that sample rate _has_to_ be set before
/// calling this function, otherwise throws a runtime_error exception.
SOUNDTOUCHDLL_API void __cdecl soundtouch_putSamples(HANDLE h,
const float *samples, ///< Pointer to sample buffer.
unsigned int numSamples ///< Number of sample frames in buffer. Notice
///< that in case of multi-channel sound a single
///< sample frame contains data for all channels.
);
/// int16 version of soundtouch_putSamples(): This accept int16 (short) sample data
/// and internally converts it to float format before processing
SOUNDTOUCHDLL_API void __cdecl soundtouch_putSamples_i16(HANDLE h,
const short *samples, ///< Pointer to sample buffer.
unsigned int numSamples ///< Number of sample frames in buffer. Notice
///< that in case of multi-channel sound a single
///< sample frame contains data for all channels.
);
/// Clears all the samples in the object's output and internal processing
/// buffers.
SOUNDTOUCHDLL_API void __cdecl soundtouch_clear(HANDLE h);
/// Changes a setting controlling the processing system behaviour. See the
/// 'SETTING_...' defines for available setting ID's.
///
/// \return 'nonzero' if the setting was successfully changed, otherwise zero
SOUNDTOUCHDLL_API int __cdecl soundtouch_setSetting(HANDLE h,
int settingId, ///< Setting ID number. see SETTING_... defines.
int value ///< New setting value.
);
/// Reads a setting controlling the processing system behaviour. See the
/// 'SETTING_...' defines for available setting ID's.
///
/// \return the setting value.
SOUNDTOUCHDLL_API int __cdecl soundtouch_getSetting(HANDLE h,
int settingId ///< Setting ID number, see SETTING_... defines.
);
/// Returns number of samples currently unprocessed.
SOUNDTOUCHDLL_API unsigned int __cdecl soundtouch_numUnprocessedSamples(HANDLE h);
/// Adjusts book-keeping so that given number of samples are removed from beginning of the
/// sample buffer without copying them anywhere.
///
/// Used to reduce the number of samples in the buffer when accessing the sample buffer directly
/// with 'ptrBegin' function.
SOUNDTOUCHDLL_API unsigned int __cdecl soundtouch_receiveSamples(HANDLE h,
float *outBuffer, ///< Buffer where to copy output samples.
unsigned int maxSamples ///< How many samples to receive at max.
);
/// int16 version of soundtouch_receiveSamples(): This converts internal float samples
/// into int16 (short) return data type
SOUNDTOUCHDLL_API unsigned int __cdecl soundtouch_receiveSamples_i16(HANDLE h,
short *outBuffer, ///< Buffer where to copy output samples.
unsigned int maxSamples ///< How many samples to receive at max.
);
/// Returns number of samples currently available.
SOUNDTOUCHDLL_API unsigned int __cdecl soundtouch_numSamples(HANDLE h);
/// Returns nonzero if there aren't any samples available for outputting.
SOUNDTOUCHDLL_API int __cdecl soundtouch_isEmpty(HANDLE h);
/// Create a new instance of BPM detector
SOUNDTOUCHDLL_API HANDLE __cdecl bpm_createInstance(int numChannels, int sampleRate);
/// Destroys a BPM detector instance.
SOUNDTOUCHDLL_API void __cdecl bpm_destroyInstance(HANDLE h);
/// Feed 'numSamples' sample frames from 'samples' into the BPM detector.
SOUNDTOUCHDLL_API void __cdecl bpm_putSamples(HANDLE h,
const float *samples, ///< Pointer to sample buffer.
unsigned int numSamples ///< Number of samples in buffer. Notice
///< that in case of stereo-sound a single sample
///< contains data for both channels.
);
/// Feed 'numSamples' sample frames from 'samples' into the BPM detector.
/// 16bit int sample format version.
SOUNDTOUCHDLL_API void __cdecl bpm_putSamples_i16(HANDLE h,
const short *samples, ///< Pointer to sample buffer.
unsigned int numSamples ///< Number of samples in buffer. Notice
///< that in case of stereo-sound a single sample
///< contains data for both channels.
);
/// Analyzes the results and returns the BPM rate. Use this function to read result
/// after whole song data has been input to the class by consecutive calls of
/// 'inputSamples' function.
///
/// \return Beats-per-minute rate, or zero if detection failed.
SOUNDTOUCHDLL_API float __cdecl bpm_getBpm(HANDLE h);
#endif // _SoundTouchDLL_h_

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Src/external_dependencies/openmpt-trunk/include/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", "2.3.1.0"
VALUE "InternalName", "SoundTouch"
VALUE "LegalCopyright", "Copyright (C) Olli Parviainen 2021"
VALUE "OriginalFilename", "SoundTouch.dll"
VALUE "ProductName", " SoundTouch Dynamic Link Library"
VALUE "ProductVersion", "2.3.1.0"
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

15
Src/external_dependencies/openmpt-trunk/include/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