ref: 3d18ea29658bfb1ac4be84d4ae0f4660199ca76d
parent: 9ae8c4986fdfe03d3e64d7ad734e418e9869b544
author: robs <robs>
date: Fri Jan 5 18:48:30 EST 2007
Removed out-of-date and redundant info; general clean-ups.
--- a/soxexam.1
+++ b/soxexam.1
@@ -1,182 +1,63 @@
-.de Sh
-.br
-.ne 5
-.PP
-\fB\\$1\fR
-.PP
-..
-.de Sp
-.if t .sp .5v
-.if n .sp
-..
-.TH SoX 1 "December 11, 2001"
+.ie n .ds EM " -
+.el .ds EM \(em
+.ds d \v'-.15m'.\v'+.15m'\" Decimal point set slightly raised
+.TH SoX 1 "January 31, 2007" "soxexam" "Sound eXchange"
.SH NAME
-soxexam - SoX Examples (CHEAT SHEET)
+soxexam\*(EMSoX Examples
.SH CONVERSIONS
-.B Introduction
+To convert from unsigned bytes to signed words:
.P
-In general, SoX will attempt to take an input sound file format and
-convert it into a new file format using a similar data type and sample
-rate. For instance, "sox monkey.au monkey.wav" would try and convert
-the mono 8000Hz u-law sample .au file that comes with SoX to a 8000Hz
-u-law .wav file.
+ sox filename.ub newfile.sw
.P
-If an output format doesn't support the same data type as the input file
-then SoX will generally select a default data type to save it in.
-You can override the default data type selection by using command line
-options. This is also useful for producing an output file with higher
-or lower precision data and/or sample rate.
-.P
-Most file formats that contain headers can automatically be read in.
-When working with header-less file formats then a user must manually
-tell SoX the data type and sample rate using command line options.
-.P
-When working with header-less files (raw files), you may take advantage of
-the pseudo-file types of .ub, .uw, .sb, .sw, .ul, and .sl. By using these
-extensions on your filenames you will not have to specify the corresponding
-options on the command line.
-.P
-.B Precision
-.P
-The following data types and formats can be represented by their total
-uncompressed bit precision. When converting from one data type to another
-care must be taken to insure it has an equal or greater precision. If not
-then the audio quality will be degraded. This is not always a bad thing
-when your working with things such as voice audio and are concerned about
-disk space or bandwidth of the audio data.
-.P
-.br
- Data Format Precision
-.br
- ___________ _________
-.br
- unsigned byte 8-bit
-.br
- signed byte 8-bit
-.br
- u-law 14-bit
-.br
- A-law 13-bit
-.br
- unsigned word 16-bit
-.br
- signed word 16-bit
-.br
- ADPCM 16-bit
-.br
- GSM 16-bit
-.br
- unsigned long 32-bit
-.br
- signed long 32-bit
-.br
- ___________ _________
-.P
-.B Examples
-.P
-Use the '-V' option on all your command lines. It makes SoX print out its
-idea of what is going on. '-V' is your friend.
-.P
-To convert from unsigned bytes at 8000 Hz to signed words at 8000 Hz:
-.P
-.br
- sox -r 8000 -c 1 filename.ub newfile.sw
-.P
To convert from Apple's AIFF format to Microsoft's WAV format:
.P
-.br
- sox filename.aiff filename.wav
+ sox filename.aiff filename.wav
.P
To convert from mono raw 8000 Hz 8-bit unsigned PCM data to a WAV file:
.P
-.br
- sox -r 8000 -u -b -c 1 filename.raw filename.wav
+ sox \-r 8000 \-u \-b \-c 1 filename.raw filename.wav
.P
SoX may even be used to convert sample rates. Downconverting will
-reduce the bandwidth of a sample, but will reduce storage space on
+reduce the bandwidth of the audio, and reduce storage space on
your disk. All such conversions are lossy and will introduce some noise.
-You should really pass your sample through a low pass filter
+You should really pass your sample through a low pass filter
prior to downconverting as this will prevent alias signals (which
-would sound like additional noise). For example to convert from a
-sample recorded at 11025 Hz to a u-law file at 8000 Hz sample rate:
+would sound like additional noise). For example to convert from a
+sample recorded at 11025 Hz to a \(*m-law file at 8000 Hz sample rate:
.P
-.br
- sox infile.wav -t au -r 8000 -U -b -c 1 outputfile.au
+ sox input.wav \-t au \-r 8000 \-U \-b \-c 1 output.au
.P
To add a low-pass filter (note use of stdout for output of
the first stage and stdin for input on the second stage):
.P
+ sox input.wav \-t raw \-s \-w \-c 1 \- lowpass 3700 |
.br
- sox infile.wav -t raw -s -w -c 1 - lowpass 3700 |
- sox -t raw -r 11025 -s -w -c 1 - -t au -r 8000 -U -b -c 1 ofile.au
+ sox \-t raw \-r 11025 \-s \-w \-c 1 \- \-t au \-r 8000 \-U \-b \-c 1 output.au
.P
-If you hear some clicks and pops when converting to u-law or A-law,
-reduce the output level slightly, for example this will decrease
+If you hear some clicks and pops when converting to \(*m-law or A-law,
+reduce the output level slightly, for example this will decrease
it by 20%:
.P
-.br
- sox infile.wav -t au -r 8000 -U -b -c 1 -v .8 outputfile.au
+ sox input.wav \-t au \-r 8000 \-U \-b \-c 1 \-v 0\*d8 outputfile.au
.P
-
-.I SoX
-is great to use along with other command line programs by passing data
-between the programs using pipelines. The most common example is to use
-mpg123 to convert mp3 files in to wav files. The following command line will
-do this:
-.P
-.br
- mpg123 -b 10000 -s filename.mp3 | sox -t raw -r 44100 -s -w -c 2 - filename.wav
-.P
-When working with totally unknown audio data then the "auto" file format may
-be of use. It attempts to guess what the file type is and then you may
-save it into a known audio format.
-.P
-.br
- sox -V -t auto filename.snd filename.wav
-.P
-It is important to understand how the internals of
-.I SoX
-work with
-compressed audio including u-law, A-law, ADPCM, or GSM.
-.I SoX
-takes ALL input data types and converts them to uncompressed 32-bit
-signed data. It will then convert this internal version into the
-requested output format. This means additional noise can be introduced
-from decompressing data and then recompressing. If applying multiple
-effects to audio data, it is best to save the intermediate data as PCM
-data. After the final effect is performed, then you can specify it as
-a compressed output format. This will keep noise introduction to a minimum.
-.P
The following example applies various effects to an 8000 Hz ADPCM input
file and then end up with the final file as 44100 Hz ADPCM.
.P
+ sox firstfile.wav \-r 44100 \-s \-w secondfile.wav
.br
- sox firstfile.wav -r 44100 -s -w secondfile.wav
-.br
- sox secondfile.wav thirdfile.wav swap
+ sox secondfile.wav thirdfile.wav swap
.br
- sox thirdfile.wav -a -b finalfile.wav mask
-.P
-Under a DOS shell, you can convert several audio files to an new output
-format using something similar to the following command line:
-.P
-.br
- FOR %X IN (*.RAW) DO sox -r 11025 -w -s -t raw $X $X.wav
+ sox thirdfile.wav \-a \-b finalfile.wav mask
.SH EFFECTS
-Special thanks goes to Juergen Mueller (jmeuller@uia.au.ac.be) for this
-write up on effects.
-.P
-.B Introduction:
-.P
The core problem is that you need some experience in using effects
-in order to say "that any old sound file sounds with effects
-absolutely hip". There isn't any rule-based system which tell you
+in order to say `that any old sound file sounds with effects
+absolutely hip'. There isn't any rule-based system which tell you
the correct setting of all the parameters for every effect.
But after some time you will become an expert in using effects.
.P
Here are some examples which can be used with any music sample.
(For a sample where only a single instrument is playing, extreme
-parameter setting may make well-known "typically" or "classical"
+parameter setting may make well-known `typically' or `classical'
sounds. Likewise, for drums, vocals or guitars.)
.P
Single effects will be explained and some given parameter settings
@@ -192,32 +73,28 @@
are possible and you really need a very fast machine and a lot of
memory to play them in real-time.
.P
-However, real-time playing of sounds will greatly speed up learning
-and/or tuning the parameter settings for your sounds in order to
-get that "perfect" effect.
+However, real-time playing of sounds will greatly speed up learning
+and/or tuning the parameter settings for your sounds in order to
+get that `perfect' effect.
.P
-Basically, we will use the "play" front-end of SoX since it is easier
+Basically, we will use the `play' front-end of SoX since it is easier
to listen sounds coming out of the speaker or earphone instead
of looking at cryptic data in sound files.
.P
-For easy listening of file.xxx ("xxx" is any sound format):+For easy listening of file.xxx (`xxx' is any sound format):
.P
-.BR
play file.xxx effect-name effect-parameters
.P
-Or more SoX-like (for "dsp" output on a UNIX/Linux computer):
+Or more SoX-like (for `dsp' output on a UNIX/Linux computer):
.P
-.BR
- sox file.xxx -t ossdsp -w -s /dev/dsp effect-name effect-parameters
+ sox file.xxx \-t ossdsp \-w \-s /dev/dsp effect-name effect-parameters
.P
-or (for "au" output):
+or (for `au' output):
.P
-.BR
- sox file.xxx -t sunau -w -s /dev/audio effect-name effect-parameters
+ sox file.xxx \-t sunau \-w \-s /dev/audio effect-name effect-parameters
.P
And for date freaks:
.P
-.BR
sox file.xxx file.yyy effect-name effect-parameters
.P
Additional options can be used. However, in this case, for real-time
@@ -225,9 +102,9 @@
.P
Notes:
.P
-I played all examples in real-time on a Pentium 100 with 32 MB and
-Linux 2.0.30 using a self-recorded sample ( 3:15 min long in "wav"
-format with 44.1 kHz sample rate and stereo 16 bit ).
+I played all examples in real-time on a Pentium 100 with 32 MB and
+Linux 2.0.30 using a self-recorded sample ( 3:15 min long in `wav'
+format with 44\*d1 kHz sample rate and stereo 16 bit ).
The sample should not contain any of the effects. However,
if you take any recording of a sound track from radio or tape or CD,
and it sounds like a live concert or ten people are playing the same
@@ -235,50 +112,41 @@
(Typically, less then four different instruments and no synthesizer
in the sample is suitable. Likewise, the combination vocal, drums, bass
and guitar.)
+.SS Echo
.P
-Effects:
-.P
-.B Echo
-.P
An echo effect can be naturally found in the mountains, standing somewhere
on a mountain and shouting a single word will result in one or more repetitions
of the word (if not, turn a bit around and try again, or climb to the next
mountain).
.P
-However, the time difference between shouting and repeating is the delay
+However, the time difference between shouting and repeating is the delay
(time), its loudness is the decay. Multiple echos can have different delays and
decays.
.P
-It is very popular to use echos to play an instrument with itself together,
+It is very popular to use echos to play an instrument with itself together,
like some guitar players (Brain May from Queen) or vocalists are doing.
For music samples of more than one instrument, echo can be used to add a
second sample shortly after the original one.
.P
-This will sound as if you are doubling the number of instruments playing
+This will sound as if you are doubling the number of instruments playing
in the same sample:
.P
-.BR
- play file.xxx echo 0.8 0.88 60.0 0.4
+ play file.xxx echo 0\*d8 0\*d88 60 0\*d4
.P
If the delay is very short, then it sound like a (metallic) robot playing
music:
.P
-.BR
- play file.xxx echo 0.8 0.88 6.0 0.4
+ play file.xxx echo 0\*d8 0\*d88 6 0\*d4
.P
Longer delay will sound like an open air concert in the mountains:
.P
-.BR
- play file.xxx echo 0.8 0.9 1000.0 0.3
+ play file.xxx echo 0\*d8 0\*d9 1000 0\*d3
.P
One mountain more, and:
.P
-.BR
- play file.xxx echo 0.8 0.9 1000.0 0.3 1800.0 0.25
-.P
-.B Echos
-.P
-Like the echo effect, echos stand for "ECHO in Sequel", that is the first echos
+ play file.xxx echo 0\*d8 0\*d9 1000 0\*d3 1800 0\*d25
+.SS Echos
+Like the echo effect, echos stand for `ECHO in Sequel', that is the first echos
takes the input, the second the input and the first echos, the third the input
and the first and the second echos, ... and so on.
Care should be taken using many echos (see introduction); a single echos
@@ -286,22 +154,17 @@
.P
The sample will be bounced twice in symmetric echos:
.P
-.BR
- play file.xxx echos 0.8 0.7 700.0 0.25 700.0 0.3
+ play file.xxx echos 0\*d8 0\*d7 700 0\*d25 700 0\*d3
.P
The sample will be bounced twice in asymmetric echos:
.P
-.BR
- play file.xxx echos 0.8 0.7 700.0 0.25 900.0 0.3
+ play file.xxx echos 0\*d8 0\*d7 700 0\*d25 900 0\*d3
.P
The sample will sound as if played in a garage:
.P
-.BR
- play file.xxx echos 0.8 0.7 40.0 0.25 63.0 0.3
-.P
-.B Chorus
-.P
-The chorus effect has its name because it will often be used to make a single
+ play file.xxx echos 0\*d8 0\*d7 40 0\*d25 63 0\*d3
+.SS Chorus
+The chorus effect has its name because it will often be used to make a single
vocal sound like a chorus. But it can be applied to other instrument samples
too.
.P
@@ -313,50 +176,38 @@
a bit out of tune.
.P
The typical delay is around 40ms to 60ms, the speed of the modulation is best
-near 0.25Hz and the modulation depth around 2ms.
+near 0\*d25Hz and the modulation depth around 2ms.
.P
A single delay will make the sample more overloaded:
.P
-.BR
- play file.xxx chorus 0.7 0.9 55.0 0.4 0.25 2.0 -t
+ play file.xxx chorus 0\*d7 0\*d9 55 0\*d4 0\*d25 2 \-t
.P
Two delays of the original samples sound like this:
.P
-.BR
- play file.xxx chorus 0.6 0.9 50.0 0.4 0.25 2.0 -t 60.0 0.32 0.4 1.3 -s
+ play file.xxx chorus 0\*d6 0\*d9 50 0\*d4 0\*d25 2 \-t 60 0\*d32 0\*d4 1\*d3 \-s
.P
A big chorus of the sample is (three additional samples):
.P
-.BR
- play file.xxx chorus 0.5 0.9 50.0 0.4 0.25 2.0 -t 60.0 0.32 0.4 2.3 -t \
- 40.0 0.3 0.3 1.3 -s
-.P
-.B Flanger
-.P
+ play file.xxx chorus 0\*d5 0\*d9 50 0\*d4 0\*d25 2 \-t 60 0\*d32 0\*d4 2\*d3 \-t 40 0\*d3 0\*d3 1\*d3 \-s
+.SS Flanger
The flanger effect is like the chorus effect, but the delay varies between
0ms and maximal 5ms. It sound like wind blowing, sometimes faster or slower
including changes of the speed.
.P
-The flanger effect is widely used in funk and soul music, where the guitar
+The flanger effect is widely used in funk and soul music, where the guitar
sound varies frequently slow or a bit faster.
.P
-Enter \fIsox --help-effect flanger\fR to see the default settings.
-.P
Now, let's groove the sample:
.P
-.BR
play file.xxx flanger
.P
listen carefully between the difference of sinusoidal and triangular modulation:
.P
-.BR
play file.xxx flanger triangle
-.P
-.B Reverb
-.P
+.SS Reverb
The reverb effect is often used in audience hall which are to small or contain
-too many many visitors which disturb (dampen) the reflection of sound at
-the walls. Reverb will make the sound be perceived as if it were in
+too many many visitors which disturb (dampen) the reflection of sound at
+the walls. Reverb will make the sound be perceived as if it were in
a large hall. You can try the reverb effect in your bathroom or garage or
sport halls by shouting loud some words. You'll hear the words reflected from
the walls.
@@ -363,48 +214,42 @@
.P
The biggest problem in using the reverb effect is the correct setting of the
(wall) delays such that the sound is realistic and doesn't sound like music
-playing in a tin can or has overloaded feedback which destroys any illusion
+playing in a tin can or has overloaded feedback which destroys any illusion
of playing in a big hall.
To help you obtain realistic reverb effects, you should decide first how
long the reverb should take place until it is not loud enough to be registered
-by your ears. This is be done by varying the reverb time "t". To simulate
-small halls, use 200ms. To simulate large halls, use 1000ms. Clearly,
+by your ears. This is be done by varying the reverb time `t'. To simulate
+small halls, use 200ms. To simulate large halls, use 1000ms. Clearly,
the walls of such a hall aren't far
away, so you should define its setting be given every wall its delay time.
However, if the wall is to far away for the reverb time, you won't hear the
-reverb, so the nearest wall will be best at "t/4" delay and the farthest
-at "t/2". You can try other distances as well, but it won't sound very realistic.
+reverb, so the nearest wall will be best at `t/4' delay and the farthest
+at `t/2'. You can try other distances as well, but it won't sound very realistic.
The walls shouldn't stand to close to each other and not in a multiple integer
-distance to each other ( so avoid wall like: 200.0 and 202.0, or something
-like 100.0 and 200.0 ).
+distance to each other ( so avoid wall like: 200 and 202, or something
+like 100 and 200 ).
.P
-Since audience halls do have a lot of walls, we will start designing one
+Since audience halls do have a lot of walls, we will start designing one
beginning with one wall:
.P
-.BR
- play file.xxx reverb 1.0 600.0 180.0
+ play file.xxx reverb 1 600 180
.P
One wall more:
.P
-.BR
- play file.xxx reverb 1.0 600.0 180.0 200.0
+ play file.xxx reverb 1 600 180 200
.P
Next two walls:
.P
-.BR
- play file.xxx reverb 1.0 600.0 180.0 200.0 220.0 240.0
+ play file.xxx reverb 1 600 180 200 220 240
.P
Now, why not a futuristic hall with six walls:
.P
-.BR
- play file.xxx reverb 1.0 600.0 180.0 200.0 220.0 240.0 280.0 300.0
+ play file.xxx reverb 1 600 180 200 220 240 280 300
.P
If you run out of machine power or memory, then stop as many applications
as possible (every interrupt will consume a lot of CPU time which for
bigger halls is absolutely necessary).
-.P
-.B Phaser
-.P
+.SS Phaser
The phaser effect is like the flanger effect, but it uses a reverb instead of
an echo and does phase shifting. You'll hear the difference in the examples
comparing both effects (simply change the effect name).
@@ -411,32 +256,26 @@
The delay modulation can be sinusoidal or triangular, preferable is the
later for multiple instruments. For single instrument sounds,
the sinusoidal phaser effect will give a sharper phasing effect.
-The decay shouldn't be to close to 1.0 which will cause dramatic feedback.
-A good range is about 0.5 to 0.1 for the decay.
+The decay shouldn't be to close to 1 which will cause dramatic feedback.
+A good range is about 0\*d5 to 0\*d1 for the decay.
.P
We will take a parameter setting as for the flanger before (gain-out is
lower since feedback can raise the output dramatically):
.P
-.BR
- play file.xxx phaser 0.8 0.74 3.0 0.4 0.5 -t
+ play file.xxx phaser 0\*d8 0\*d74 3 0\*d4 0\*d5 \-t
.P
The drunken loudspeaker system (now less alcohol):
.P
-.BR
- play file.xxx phaser 0.9 0.85 4.0 0.23 1.3 -s
+ play file.xxx phaser 0\*d9 0\*d85 4 0\*d23 1\*d3 \-s
.P
A popular sound of the sample is as follows:
.P
-.BR
- play file.xxx phaser 0.89 0.85 1.0 0.24 2.0 -t
+ play file.xxx phaser 0\*d89 0\*d85 1 0\*d24 2 \-t
.P
The sample sounds if ten springs are in your ears:
.P
-.BR
- play file.xxx phaser 0.6 0.66 3.0 0.6 2.0 -t
-.P
-.B Compander
-.P
+ play file.xxx phaser 0\*d6 0\*d66 3 0\*d6 2 \-t
+.SS Compander
The compander effect allows the dynamic range of a signal to be
compressed or expanded.
For most situations, the attack time (response to the music getting
@@ -443,100 +282,66 @@
louder) should be shorter than the decay time because our ears are more
sensitive to suddenly loud music than to suddenly soft music.
.P
-For example, suppose you are listening to Strauss' "Also Sprach
-Zarathustra" in a noisy environment such as a car.
+For example, suppose you are listening to Strauss' `Also Sprach
+Zarathustra' in a noisy environment such as a car.
If you turn up the volume enough to hear the soft passages over the
road noise, the loud sections will be too loud.
You could try this:
.P
-.BR
- play file.xxx compand 0.3,1 -90,-90,-70,-70,-60,-20,0,0 -5 0 0.2
+ play file.xxx compand 0\*d3,1 \-90,\-90,\-70,\-70,\-60,\-20,0,0 \-5 0 0\*d2
.P
-The transfer function ("-90,...") says that+The transfer function (`\-90,...') says that
.I very
-soft sounds between -90 and -70 decibels (-90 is about the limit of
+soft sounds between \-90 and \-70 decibels (\-90 is about the limit of
16-bit encoding) will remain unchanged.
-That keeps the compander from boosting the volume on "silent" passages
+That keeps the compander from boosting the volume on `silent' passages
such as between movements.
-However, sounds in the range -60 decibels to 0 decibels (maximum
-volume) will be boosted so that the 60-dB dynamic range of the
-original music will be compressed 3-to-1 into a 20-dB range, which is
+However, sounds in the range \-60 decibels to 0 decibels (maximum
+volume) will be boosted so that the 60dB dynamic range of the
+original music will be compressed 3-to-1 into a 20dB range, which is
wide enough to enjoy the music but narrow enough to get around the
road noise.
-The -5 dB output gain is needed to avoid clipping (the number is
+The \-5 dB output gain is needed to avoid clipping (the number is
inexact, and was derived by experimentation).
The 0 for the initial volume will work fine for a clip that starts
-with a bit of silence, and the delay of 0.2 has the effect of causing
+with a bit of silence, and the delay of 0\*d2 has the effect of causing
the compander to react a bit more quickly to sudden volume changes.
-.P
-.B Changing the Rate of Playback
-.P
+.SS Changing the Rate of Playback
You can use stretch to change the rate of playback of an audio sample
-while preserving the pitch. For example to play at 1/2 the speed:
+while preserving the pitch. For example to play at half the speed:
.P
-.BR
play file.wav stretch 2
.P
To play a file at twice the speed:
.P
-.BR
- play file.wav stretch .5
+ play file.wav stretch 0\*d5
.P
-Other related options are "speed" to change the speed of play
-(and changing the pitch accordingly), and pitch, to alter the
+Other related options are `speed' to change the speed of play
+(and changing the pitch accordingly), and pitch, to alter the
pitch of a sample. For example to speed a sample so it plays in
-1/2 the time (for those Mickey Mouse voices):
+half the time (for those Mickey Mouse voices):
.P
-.BR
play file.wav speed 2
.P
To raise the pitch of a sample 1 while note (100 cents):
.P
-.BR
play file.wav pitch 100
.P
-
+.SS Reducing noise in a recording
.P
-.B Reducing noise in a recording
-.P
First find a period of silence in your recording, such as the beginning or
-end of a piece. If the first 1.5 seconds of the recording are silent, do
+end of a piece. If the first 1\*d5 seconds of the recording are silent, do
.P
-.BR
- sox file.wav -t nul /dev/null trim 0 1.5 noiseprof /tmp/profile
+ sox file.wav \-n trim 0 1\*d5 noiseprof /tmp/profile
.P
Next, use the noisered effect to actually reduce the noise:
.P
-.BR
- play file.wav noisered /tmp/profile
-.P
-
-.P
-.B Other effects (copy, rate, avg, stat, vibro, lowp, highp, band, reverb)
-.P
-The other effects are simple to use. However, an "easy to use manual" should
-be given here.
-.P
-.B More effects (to do !)
-.P
-There are a lot of effects around like noise gates, compressors, waw-waw,
-stereo effects and so on. They should be implemented, making SoX more
-useful in sound mixing techniques coming together with a great variety of
-different sound effects.
-.P
-Combining effects by using them in parallel or serially on different channels
-needs some easy mechanism which is stable for use in real-time.
-.P
-Really missing are the the changing of the parameters and starting/stopping of
-effects while playing samples in real-time!
-.P
-Good luck and have fun with all the effects!
-
- Juergen Mueller (jmueller@uia.ua.ac.be)
-
+ play file.wav noisered /tmp/profile
.SH SEE ALSO
-sox(1), play(1), rec(1)
+.BR sox (1),
+.BR play (1),
+.BR rec (1)
.SH AUTHOR
-Juergen Mueller (jmueller@uia.ua.ac.be)
-.P
-Updates by Anonymous.
+Juergen Mueller (jmueller@uia.ua.ac.be).
+Additional authors and contributors are listed in the AUTHORS file that
+is distributed with the source code.