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.TH SoX 1 "December 10, 1999"
.SH NAME
soxexam - SoX Examples (CHEAT SHEET)
.SH CONVERSIONS
.B Introduction
.P
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.
.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 a 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 headerless file formats then a user must manually
tell sox the data type and sample rate using command line options.
.P
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 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 12-bit
.br
a-law 12-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
.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
.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 in to 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 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.
.P
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.
.P
.br
sox firstfile.wav -r 44100 -s -w secondfile.wav
.br
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
.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
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"
sounds. Likewise, for drums, vocals or guitars.)
.P
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.
.P
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 machine and a lot of
memory to play them in real-time.
.P
And real-time playing of sounds will speed up learning the parameter
setting.
.P
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 ):
.P
.BR
play file.xxx effect-name effect-parameters
.P
Or more SOX-like ( for "dsp" output ):
.P
.BR
sox file.xxx -t ossdsp -w -s /dev/dsp effect-name effect-parameters
.P
or ( for "au" output ):
.P
.BR
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
playing you'll need a very fast machine.
.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 ).
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 combination vocal, drums, bass
and guitar.)
.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 ant try next, or climb to the next
mountain ).
.P
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
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.
.P
This will sound as doubling the number of instruments playing the same sample:
.P
.BR
play file.xxx echo 0.8 0.88 60.0 0.4
.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
.P
Longer delay will sound like a open air concert in the mountains:
.P
.BR
play file.xxx echo 0.8 0.9 1000.0 0.3
.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
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.
.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
.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
.P
The sample will sound as 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
vocal sound like a chorus. But it can be applied to other instrument samples
too.
.P
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 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.
.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.
.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
.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
.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
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
sound varies frequently slow or a bit faster.
.P
The typical delay is around 3ms to 5ms, the speed of the modulation is best
near 0.5Hz.
.P
Now, let's groove the sample:
.P
.BR
play file.xxx flanger 0.6 0.87 3.0 0.9 0.5 -s
.P
listen carefully between the difference of sinusoidal and triangular modulation:
.P
.BR
play file.xxx flanger 0.6 0.87 3.0 0.9 0.5 -t
.P
If the decay is a bit lower, than the effect sounds more popular:
.P
.BR
play file.xxx flanger 0.8 0.88 3.0 0.4 0.5 -t
.P
The drunken loudspeaker system:
.P
.BR
play file.xxx flanger 0.9 0.9 4.0 0.23 1.3 -s
.P
.B Reverb
.P
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.
.P
The biggest problem in using the reverb effect is the correct setting of the
(wall) delays such that the sound is realistic an doesn't sound like music
playing in a tin or overloaded feedback destroys any illusion of any big hall.
To help you for much 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 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 away for the reverb time, you won't hear the
reverb, so the nearest wall will be best "t/4" delay and the farthest "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 ).
.P
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
.P
One wall more:
.P
.BR
play file.xxx reverb 1.0 600.0 180.0 200.0
.P
Next two walls:
.P
.BR
play file.xxx reverb 1.0 600.0 180.0 200.0 220.0 240.0
.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
.P
If you run out of machine power or memory, then stop as much applications
as possible ( every interrupt will consume a lot of CPU time which for
bigger halls is absolutely necessary ).
.P
.B Phaser
.P
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 sinusoidal or triangular, preferable is the
later one for multiple instruments playing. 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.
.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
.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
.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
.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
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.
.P
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
.P
The transfer function ("-90,...") says that
.I 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.
.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 to be 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 sequence on different channels
needs some easy mechanism which is real-time stable.
.P
Really missing, is the changing of the parameters, starting and 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)
.SH SEE ALSO
sox(1), play(1), rec(1)