ref: af7e8d1b71257abdfc1caf93cce47946a11b606c
dir: /soxexam.txt/
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 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.
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 sample 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 preci-
sion. 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. Downconverting 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 convert 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 outputfile.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 com-
mand 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 for-
mat 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 com-
pressed 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 decompress-
ing 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 out-
put 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 following 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 set-
ting 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. Like-
wise, for drums, vocals or guitars.)
Single effects will be explained and some given parameter settings that
can be used to understand the theory by listening 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 unsat-
isfied. Hence, for the first time using effects try to compose them as
minimally 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 look-
ing at cryptic data in sound files.
For easy listening of file.xxx ("xxx" is any sound format):
play file.xxx effect-name effect-parameters
Or more SoX-like (for "dsp" output on a UNIX/Linux computer):
sox file.xxx -t ossdsp -w -s /dev/dsp effect-name effect-parame-
ters
or (for "au" output):
sox file.xxx -t sunau -w -s /dev/audio effect-name effect-parame-
ters
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" for-
mat 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 dif-
ferent instruments and no synthesizer in the sample is suitable. Like-
wise, the combination vocal, drums, bass and guitar.)
Effects:
Echo
An echo effect can be naturally found in the mountains, standing some-
where 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 repeating is the
delay (time), its loudness is the decay. Multiple echos can have dif-
ferent 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 sample shortly after the original one.
This will sound as if you are doubling the number of instruments play-
ing 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 (metallic) robot play-
ing 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 sinusoidal or triangular modula-
tion. 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 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 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 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 some-
thing 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 applica-
tions as possible (every interrupt will consume a lot of CPU time which
for bigger halls is absolutely 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 modulation can be sinusoidal or triangular, preferable is the
later for multiple instruments. For single instrument sounds, the sinu-
soidal 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 com-
pressed 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 sample 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
Reducing noise in a recording
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
sox file.wav -t nul /dev/null trim 0 1.5 noiseprof /tmp/profile
Next, use the noisered effect to actually reduce the noise:
play file.wav noisered /tmp/profile
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, 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.
Combining effects by using them in parallel or serially on different
channels needs some easy mechanism which is stable for use in real-
time.
Really missing are the the changing of the parameters and start-
ing/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)