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SoX(1) SoX(1)
NAME
soxexam - SoX Examples (CHEAT SHEET)
CONVERSIONS
Introduction
In general, SoX will attempt to take an input sound file
format and convert it into a new file format using a simi�
lar data type and sample rate. For instance, "sox mon�
key.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.
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.
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 sam�
ple rate using command line options.
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.
Precision
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.
Data Format Precision
___________ _________
unsigned byte 8-bit
signed byte 8-bit
u-law 14-bit
A-law 13-bit
unsigned word 16-bit
signed word 16-bit
ADPCM 16-bit
GSM 16-bit
unsigned long 32-bit
signed long 32-bit
___________ _________
Examples
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.
To convert from unsigned bytes at 8000 Hz to signed words
at 8000 Hz:
sox -r 8000 -c 1 filename.ub newfile.sw
To convert from Apple's AIFF format to Microsoft's WAV
format:
sox filename.aiff filename.wav
To convert from mono raw 8000 Hz 8-bit unsigned PCM data
to a WAV file:
sox -r 8000 -u -b -c 1 filename.raw filename.wav
SoX may even be used to convert sample rates. Downcon�
verting will reduce the bandwidth of a sample, but will
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 prior to
downconverting as this will prevent alias signals (which
would sound like additional noise). For example to con�
vert from a sample recorded at 11025 Hz to a u-law file at
8000 Hz sample rate:
sox infile.wav -t au -r 8000 -U -b -c 1 outputfile.au
To add a low-pass filter (note use of stdout for output of
the first stage and stdin for input on the second stage):
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
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 it by 20%:
sox infile.wav -t au -r 8000 -U -b -c 1 -v .8 output�
file.au
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:
mpg123 -b 10000 -s filename.mp3 | sox -t raw -r 44100 -s
-w -c 2 - filename.wav
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.
sox -V -t auto filename.snd filename.wav
It is important to understand how the internals of SoX
work with compressed audio including u-law, A-law, ADPCM,
or GSM. 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 decom�
pressing data and then recompressing. If applying multi�
ple effects to audio data, it is best to save the interme�
diate 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.
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.
sox firstfile.wav -r 44100 -s -w secondfile.wav
sox secondfile.wav thirdfile.wav swap
sox thirdfile.wav -a -b finalfile.wav mask
Under a DOS shell, you can convert several audio files to
an new output format using something similar to the fol�
lowing command line:
FOR %X IN (*.RAW) DO sox -r 11025 -w -s -t raw $X $X.wav
EFFECTS
Special thanks goes to Juergen Mueller
(jmeuller@uia.au.ac.be) for this write up on effects.
Introduction:
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 the correct setting of all the
parameters for every effect. But after some time you will
become an expert in using effects.
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" sounds. Likewise, for drums,
vocals or guitars.)
Single effects will be explained and some given parameter
settings that can be used to understand the theory by lis�
tening to the sound file with the added effect.
Using multiple effects in parallel or in series can result
either in a very nice sound or (mostly) in a dramatic
overloading in variations of sounds such that your ear may
follow the sound but you will feel unsatisfied. Hence, for
the first time using effects try to compose them as mini�
mally as possible. We don't regard the composition of
effects in the examples because too many combinations are
possible and you really need a very fast machine and a lot
of memory to play them in real-time.
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.
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.
For easy listening of file.xxx ("xxx" is any sound for�
mat):
play file.xxx effect-name effect-parameters
Or more SoX-like (for "dsp" output on a UNIX/Linux com�
puter):
sox file.xxx -t ossdsp -w -s /dev/dsp effect-name
effect-parameters
or (for "au" output):
sox file.xxx -t sunau -w -s /dev/audio effect-name
effect-parameters
And for date freaks:
sox file.xxx file.yyy effect-name effect-parameters
Additional options can be used. However, in this case, for
real-time playing you'll need a very fast machine.
Notes:
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 ). 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 rhythm with
their drums or funky-grooves, then take any other sample.
(Typically, less then four different instruments and no
synthesizer in the sample is suitable. Likewise, the com�
bination vocal, drums, bass and guitar.)
Effects:
Echo
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).
However, the time difference between shouting and repeat�
ing is the delay (time), its loudness is the decay. Multi�
ple echos can have different delays and decays.
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 sam�
ple shortly after the original one.
This will sound as if you are doubling the number of
instruments playing in the same sample:
play file.xxx echo 0.8 0.88 60.0 0.4
If the delay is very short, then it sound like a (metal�
lic) robot playing music:
play file.xxx echo 0.8 0.88 6.0 0.4
Longer delay will sound like an open air concert in the
mountains:
play file.xxx echo 0.8 0.9 1000.0 0.3
One mountain more, and:
play file.xxx echo 0.8 0.9 1000.0 0.3 1800.0 0.25
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
has the same effect as a single echo.
The sample will be bounced twice in symmetric echos:
play file.xxx echos 0.8 0.7 700.0 0.25 700.0 0.3
The sample will be bounced twice in asymmetric echos:
play file.xxx echos 0.8 0.7 700.0 0.25 900.0 0.3
The sample will sound as if played in a garage:
play file.xxx echos 0.8 0.7 40.0 0.25 63.0 0.3
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.
It works like the echo effect with a short delay, but the
delay isn't constant. The delay is varied using a sinu�
soidal or triangular modulation. The modulation depth
defines the range the modulated delay is played before or
after the delay. Hence the delayed sound will sound slower
or faster, that is the delayed sound tuned around the
original one, like in a chorus where some vocals are a bit
out of tune.
The typical delay is around 40ms to 60ms, the speed of the
modulation is best near 0.25Hz and the modulation depth
around 2ms.
A single delay will make the sample more overloaded:
play file.xxx chorus 0.7 0.9 55.0 0.4 0.25 2.0 -t
Two delays of the original samples sound like this:
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
A big chorus of the sample is (three additional samples):
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
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.
The flanger effect is widely used in funk and soul music,
where the guitar sound varies frequently slow or a bit
faster.
The typical delay is around 3ms to 5ms, the speed of the
modulation is best near 0.5Hz.
Now, let's groove the sample:
play file.xxx flanger 0.6 0.87 3.0 0.9 0.5 -s
listen carefully between the difference of sinusoidal and
triangular modulation:
play file.xxx flanger 0.6 0.87 3.0 0.9 0.5 -t
If the decay is a bit lower, than the effect sounds more
popular:
play file.xxx flanger 0.8 0.88 3.0 0.4 0.5 -t
The drunken loudspeaker system:
play file.xxx flanger 0.9 0.9 4.0 0.23 1.3 -s
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 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.
The biggest problem in using the reverb effect is the cor�
rect 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
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, the walls of
such a hall aren't far away, so you should define its set�
ting 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 dis�
tances 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
).
Since audience halls do have a lot of walls, we will start
designing one beginning with one wall:
play file.xxx reverb 1.0 600.0 180.0
One wall more:
play file.xxx reverb 1.0 600.0 180.0 200.0
Next two walls:
play file.xxx reverb 1.0 600.0 180.0 200.0 220.0
240.0
Now, why not a futuristic hall with six walls:
play file.xxx reverb 1.0 600.0 180.0 200.0 220.0
240.0 280.0 300.0
If you run out of machine power or memory, then stop as
many applications as possible (every interrupt will con�
sume a lot of CPU time which for bigger halls is abso�
lutely necessary).
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). The delay modu�
lation 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.
We will take a parameter setting as for the flanger before
(gain-out is lower since feedback can raise the output
dramatically):
play file.xxx phaser 0.8 0.74 3.0 0.4 0.5 -t
The drunken loudspeaker system (now less alcohol):
play file.xxx phaser 0.9 0.85 4.0 0.23 1.3 -s
A popular sound of the sample is as follows:
play file.xxx phaser 0.89 0.85 1.0 0.24 2.0 -t
The sample sounds if ten springs are in your ears:
play file.xxx phaser 0.6 0.66 3.0 0.6 2.0 -t
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 louder) should
be shorter than the decay time because our ears are more
sensitive to suddenly loud music than to suddenly soft
music.
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:
play file.xxx compand 0.3,1
-90,-90,-70,-70,-60,-20,0,0 -5 0 0.2
The transfer function ("-90,...") says that very 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 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 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 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 the compander to
react a bit more quickly to sudden volume changes.
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:
play file.wav stretch 2
To play a file at twice the speed:
play file.wav stretch .5
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 sam�
ple so it plays in 1/2 the time (for those Mickey Mouse
voices):
play file.wav speed 2
To raise the pitch of a sample 1 while note (100 cents):
play file.wav pitch 100
Other effects (copy, rate, avg, stat, vibro, lowp, highp,
band, reverb)
The other effects are simple to use. However, an "easy to
use manual" should be given here.
More effects (to do !)
There are a lot of effects around like noise gates, com�
pressors, 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 differ�
ent sound effects.
Combining effects by using them in parallel or serially on
different channels needs some easy mechanism which is sta�
ble for use in real-time.
Really missing are the the changing of the parameters and
starting/stopping of effects while playing samples in
real-time!
Good luck and have fun with all the effects!
Juergen Mueller (jmueller@uia.ua.ac.be)
SEE ALSO
sox(1), play(1), rec(1)
AUTHOR
Juergen Mueller (jmueller@uia.ua.ac.be)
Updates by Anonymous.
December 11, 2001 SoX(1)