GNU/Linux man pages

Livre :
Expressions régulières,
Syntaxe et mise en oeuvre :

ISBN : 978-2-7460-9712-4
EAN : 9782746097124
(Editions ENI)


CentOS 2.1AS







soxexam - SoX Examples (CHEAT SHEET)



In general, sox will attempt to take an input sound file format and convert it to 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 a 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 headerless file formats then a user must manually tell sox the data type and sample rate using command line options.

When working with headerless files (raw files), you may take advantage of they 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 corrisponding options on the command line.


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 12-bit

a-law 12-bit

unsigned word 16-bit

signed word 16-bit

ADPCM 16-bit

GSM 16-bit

unsigned long 32-bit

signed long 32-bit

___________ _________


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 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 in to 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 unneeded 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.

The following example is to apply 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


Special thanks goes to Juergen Mueller (jmeuller@uia.au.ac.be) for this write up on effects.


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 theorie by listening to the sound file with the added effect.

Using multiple effects in parallel or in sequel can result either in very perfect 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 less as possible. We don’t regard the composition of effects in the examples because to many combinations are possible and you really need a very fast maschine and a lot of memory to play them in real-time.

And real-time playing of sounds will speed up learning the parameter setting.

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 cryptical 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 ):

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.


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-groves, then take any other sample. (Typically, less then four different intruments and no synthesizer in the sample is suitable. Likewise, the combination vocal, drums, bass and guitar.)



An echo effect can be naturally found in the mountains, standing somewhere on a moutain and shouting a single word will result in one or more repetitions of the word ( if not, turn a bit around ant try next, 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 different delays and decays.

Very popular is using 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 doubling the number of instruments playing 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) roboter playing music:

play file.xxx echo 0.8 0.88 6.0 0.4

Longer delay will sound like a 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


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 played in a garage:

play file.xxx echos 0.8 0.7 40.0 0.25 63.0 0.3


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 sinodial 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 vocal are a bit out of tune.

The typical delay is around 40ms to 60ms, the speed of the modualtion 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


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 sinodial 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 loundspeaker system:

play file.xxx flanger 0.9 0.9 4.0 0.23 1.3 -s


The reverb effect is often used in audience hall which are to small or to many visitors disturb the reflection of sound at the walls to make the sound played more monumental. 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 relistic an doesn’t sound like music playing in a tin or overloaded feedback distroys any illusion of any big hall. To help you for much realisitc 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 the reverb time "t", in small halls 200ms in bigger one 1000ms, if you like. 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 eway for the reverb time, you won’t hear the reverb, so the nearest wall will be best "t/4" delay and the farest "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 interger 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 much applications as possible ( every interupt will consume a lot of cpu time which for bigger halls is absolutely neccessary ).


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 done sinodial or triangular, preferable is the later one for multiple instruments playing. For single instrument sounds the sinodial phaser effect will give a sharper phasing effect. The decay shouln’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 loundspeaker system ( now less alkohol ):

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

Other effects ( copy, rate, avg, stat, vibro, lowp, highp, band, reverb )

The other effects are simply 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 to be more useful in sound mixing technics coming together with a great varity of different sound effects.

Combining effects be using then in parallel or sequel on different channels needs some easy mechanism which is real-time stable.

Really missing, is the changing of the parameters, starting and stoping of effects while playing samples in real-time!

Good luck and have fun with all the effects!

Juergen Mueller



sox(1), play(1), rec(1)