ref: 2512ec55a1df28fc324b0678816605f07411d6a0
dir: /sox.txt/
SoX(1) SoX(1)
NAME
sox - Sound eXchange : universal sound sample translator
SYNOPSIS
sox infile outfile
sox infile outfile [ effect [ effect options ... ] ]
sox infile -e effect [ effect options ... ]
sox [ general options ] [ format options ] ifile [ for-
mat options ] ofile [ effect [ effect options ... ] ]
General options: [ -e ] [ -h ] [ -p ] [ -v volume ] [ -V ]
Format options: [ -t filetype ] [ -r rate ] [
-s/-u/-U/-A/-a/-g ] [ -b/-w/-l/-f/-d/-D ] [ -c channels ]
[ -x ]
Effects:
avg [ -l | -r ]
band [ -n ] center [ width ]
check
chorus gain-in gain out delay decay speed depth
-s | -t [ delay decay speed depth -s | -fI-t ]
compand attack1,decay1[,attack2,decay2...]
in-dB1,out-dB1[,in-dB2,out-dB2...]
[gain] [initial-volume]
copy
cut
deemph
echo gain-in gain-out delay decay [ delay decay ...]
echos gain-in gain-out delay decay [ delay decay ...]
flanger gain-in gain-out delay decay speed -s | -fI-t
highp center
lowp center
map
mask
phaser gain-in gain-out delay decay speed -s | -t
pick
polyphase [ -w < num / ham > ]
[ -width < long / short / # > ]
[ -cutoff # ]
rate
resample
reverb gain-out reverb-time delay [ delay ... ]
reverse
split
stat [ debug | -v ]
swap [ 1 2 3 4 ]
vibro speed [ depth ]
DESCRIPTION
Sox translates sound files from one format to another,
possibly doing a sound effect.
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SoX(1) SoX(1)
OPTIONS
The option syntax is a little grotty, but in essence:
sox file.au file.voc
translates a sound sample in SUN Sparc .AU format into a
SoundBlaster .VOC file, while
sox -v 0.5 file.au -r 12000 file.voc rate
does the same format translation but also lowers the
amplitude by 1/2 and changes the sampling rate from 8000
hertz to 12000 hertz via the rate sound effect loop.
Format options:
Format options effect the file that they immediately
percede. If they are placed before the input file name
then they effect the input data. If they are placed
before the output file name then they will effect the out-
put data. It is also possible to read a given file in and
output it in any supported data format by specifying out-
put format options.
-t filetype
gives the type of the sound sample file.
-r rate Give sample rate in Hertz of file. If the input
and output files have different rates then a
sample rate change effect must be ran. If a
sample rate changing effect is not specified
then a default one will be used with its default
parameters.
-s/-u/-U/-A/-a/-g
The sample data is signed linear (2's comple-
ment), unsigned linear, U-law (logarithmic), A-
law (logarithmic), ADPCM, or GSM. U-law and A-
law are the U.S. and international standards for
logarithmic telephone sound compression. ADPCM
is form of sound compression that has a good
compromise between good sound quality and fast
encoding/decoding time. GSM is a standard used
for telephone sound compression in European
countries and its gaining popularity because of
its quality.
-b/-w/-l/-f/-d/-D
The sample data is in bytes, 16-bit words,
32-bit longwords, 32-bit floats, 64-bit double
floats, or 80-bit IEEE floats. Floats and dou-
ble floats are in native machine format.
-x The sample data is in XINU format; that is, it
comes from a machine with the opposite word
order than yours and must be swapped according
to the word-size given above. Only 16-bit and
32-bit integer data may be swapped. Machine-
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format floating-point data is not portable.
IEEE floats are a fixed, portable format.
-c channels
The number of sound channels in the data file.
This may be 1, 2, or 4; for mono, stereo, or
quad sound data. If an input and output file
have a different number of channels then the
average effect must be used. If it is not spec-
ified on the command line it will be invoked
with default parameters.
General options:
-e after the input file allows you to avoid giving
an output file and just name an effect. This is
mainly useful with the stat effect but can be
used with others.
-h Print version number and usage information.
-p Run in preview mode and run fast. This will
somewhat speed up sox when the output format has
a different number of channels and a different
rate then the input file. The order that the
effects are run in will be arranged for maximum
speed and not quality.
-v volume Change amplitude (floating point); less than 1.0
decreases, greater than 1.0 increases. Note: we
perceive volume logarithmically, not linearly.
Note: see the stat effect.
-V Print a description of processing phases. Use-
ful for figuring out exactly how sox is mangling
your sound samples.
The input and output files may be standard input and out-
put. This is specified by '-'. The -t type option must
be given in this case, else sox will not know the format
of the given file. The -t, -r, -s/-u/-U/-A,
-b/-w/-l/-f/-d/-D and -x options refer to the input data
when given before the input file name. After, they refer
to the output data.
If you don't give an output file name, sox will just read
the input file. This is useful for validating structured
file formats; the stat effect may also be used via the -e
option.
FILE TYPES
Sox needs to know the formats of the input and output
files. File formats which have headers are checked, if
that header doesn't seem right, the program exits with an
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SoX(1) SoX(1)
appropriate message. Currently, raw (no header) binary
and textual data, Amiga 8SVX, Apple/SGI AIFF, SPARC .AU
(w/header), AVR, NeXT .SND, CD-R, CVSD, GSM 06.10, Mac
HCOM, Sound Tools MAUD, OSS device drivers, Turtle Beach
.SMP, Sound Blaster, Sndtool, and Sounder, Sun Audio
device driver, Yamaha TX-16W Sampler, IRCAM Sound Files,
Creative Labs VOC, Psion .WVE, and Microsoft RIFF/WAV are
supported.
.8svx Amiga 8SVX musical instrument description for-
mat.
.aiff AIFF files used on Apple IIc/IIgs and SGI.
Note: the AIFF format supports only one SSND
chunk. It does not support multiple sound
chunks, or the 8SVX musical instrument descrip-
tion format. AIFF files are multimedia archives
and and can have multiple audio and picture
chunks. You may need a separate archiver to
work with them.
.au SUN Microsystems AU files. There are apparently
many types of .au files; DEC has invented its
own with a different magic number and word
order. The .au handler can read these files but
will not write them. Some .au files have valid
AU headers and some do not. The latter are
probably original SUN u-law 8000 hz samples.
These can be dealt with using the .ul format
(see below).
.avr Audio Visual Research
The AVR format is produced by a number of com-
mercial packages on the Mac.
.cdr CD-R
CD-R files are used in mastering music Compact
Disks. The file format is, as you might expect,
raw stereo raw unsigned samples at 44khz. But,
there's some blocking/padding oddity in the for-
mat, so it needs its own handler.
.cvs Continuously Variable Slope Delta modulation
Used to compress speech audio for applications
such as voice mail.
.dat Text Data files
These files contain a textual representation of
the sample data. There is one line at the
beginning that contains the sample rate. Subse-
quent lines contain two numeric data items: the
time since the beginning of the sample and the
sample value. Values are normalized so that the
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SoX(1) SoX(1)
maximum and minimum are 1.00 and -1.00. This
file format can be used to create data files for
external programs such as FFT analyzers or graph
routines. SoX can also convert a file in this
format back into one of the other file formats.
.gsm GSM 06.10 Lossy Speech Compression
A standard for compressing speech which is used
in the Global Standard for Mobil telecommunica-
tions (GSM). Its good for its purpose, shrink-
ing audio data size, but it will introduce lots
of noise when a given sound sample is encoded
and decoded multiple times. This format is used
by some voice mail applications. It is rather
CPU intensive. GSM in sox is optional and
requires access to an external GSM library. To
see if there is support for gsm run sox -h and
look for it under the list of supported file
formats.
.hcom Macintosh HCOM files. These are (apparently)
Mac FSSD files with some variant of Huffman com-
pression. The Macintosh has wacky file formats
and this format handler apparently doesn't han-
dle all the ones it should. Mac users will need
your usual arsenal of file converters to deal
with an HCOM file under Unix or DOS.
.maud An Amiga format
An IFF-conform sound file type, registered by MS
MacroSystem Computer GmbH, published along with
the "Toccata" sound-card on the Amiga. Allows
8bit linear, 16bit linear, A-Law, u-law in mono
and stereo.
ossdsp OSS /dev/dsp device driver
This is a psuedo-file type and can be optionally
compiled into Sox. Run sox -h to see if you
have support for this file type. When this
driver is used it allows you to open up the OSS
/dev/dsp file and configure it to use the same
data type as passed in to Sox. It works for
both playing and recording sound samples. When
playing sound files it attempts to set up the
OSS driver to use the same format as the input
file. It is suggested to always override the
output values to use the highest quality samples
your sound card can handle. Example: -t ossdsp
-w -s /dev/dsp
.sf IRCAM Sound Files.
SoundFiles are used by academic music software
such as the CSound package, and the MixView
sound sample editor.
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.smp Turtle Beach SampleVision files.
SMP files are for use with the PC-DOS package
SampleVision by Turtle Beach Softworks. This
package is for communication to several MIDI
samplers. All sample rates are supported by the
package, although not all are supported by the
samplers themselves. Currently loop points are
ignored.
sunau Sun /dev/audio device driver
This is a psuedo-file type and can be optionally
compiled into Sox. Run sox -h to see if you
have support for this file type. When this
driver is used it allows you to open up a Sun
/dev/audio file and configure it to use the same
data type as passed in to Sox. It works for
both playing and recording sound samples. When
playing sound files it attempts to set up the
audio driver to use the same format as the input
file. It is suggested to always override the
output values to use the highest quality samples
your hardware can handle. Example: -t sunau -w
-s /dev/audio or -t sunau -U -c 1 /dev/audio for
older sun equipment.
.txw Yamaha TX-16W sampler.
A file format from a Yamaha sampling keyboard
which wrote IBM-PC format 3.5" floppies. Han-
dles reading of files which do not have the sam-
ple rate field set to one of the expected by
looking at some other bytes in the attack/loop
length fields, and defaulting to 33kHz if the
sample rate is still unknown.
.vms More info to come.
Used to compress speech audio for applications
such as voice mail.
.voc Sound Blaster VOC files.
VOC files are multi-part and contain silence
parts, looping, and different sample rates for
different chunks. On input, the silence parts
are filled out, loops are rejected, and sample
data with a new sample rate is rejected.
Silence with a different sample rate is gener-
ated appropriately. On output, silence is not
detected, nor are impossible sample rates.
.wav Microsoft .WAV RIFF files.
These appear to be very similar to IFF files,
but not the same. They are the native sound
file format of Windows. (Obviously, Windows was
of such incredible importance to the computer
industry that it just had to have its own sound
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file format.) Normally .wav files have all for-
matting information in their headers, and so do
not need any format options specified for an
input file. If any are, they will override the
file header, and you will be warned to this
effect. You had better know what you are doing!
Output format options will cause a format con-
version, and the .wav will written appropri-
ately. Note that it is possible to write data
of a type that cannot be specified by the .wav
header, and you will be warned that you a writ-
ing a bad file ! Sox currently can read PCM,
ULAW, ALAW, MS ADPCM, and IMA (or DVI) ADPCM.
It can output all of these formats except the
ADPCM styles.
.wve Psion 8-bit alaw
These are 8-bit a-law 8khz sound files used on
the Psion palmtop portable computer.
.raw Raw files (no header).
The sample rate, size (byte, word, etc), and
style (signed, unsigned, etc.) of the sample
file must be given. The number of channels
defaults to 1.
.ub, .sb, .uw, .sw, .ul
These are several suffices which serve as a
shorthand for raw files with a given size and
style. Thus, ub, sb, uw, sw, and ul correspond
to "unsigned byte", "signed byte", "unsigned
word", "signed word", and "ulaw" (byte). The
sample rate defaults to 8000 hz if not explic-
itly set, and the number of channels (as always)
defaults to 1. There are lots of Sparc samples
floating around in u-law format with no header
and fixed at a sample rate of 8000 hz. (Certain
sound management software cheerfully ignores the
headers.) Similarly, most Mac sound files are
in unsigned byte format with a sample rate of
11025 or 22050 hz.
.auto This is a ``meta-type'': specifying this type
for an input file triggers some code that tries
to guess the real type by looking for magic
words in the header. If the type can't be
guessed, the program exits with an error mes-
sage. The input must be a plain file, not a
pipe. This type can't be used for output files.
EFFECTS
Only one effect from the palette may be applied to a sound
sample. To do multiple effects you'll need to run sox in
a pipeline.
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avg [ -l | -r ]
Reduce the number of channels by averaging the
samples, or duplicate channels to increase the
number of channels. This effect is automati-
cally used when the number of input samples dif-
fer then the number of output channels. When
reducing the number of channels it is possible
to manually specify the avg effect and use the
-l and -r options to select only the left or
right channel for the output instead of averag-
ing the two channels.
band [ -n ] center [ width ]
Apply a band-pass filter. The frequency
response drops logarithmically around the center
frequency. The width gives the slope of the
drop. The frequencies at center + width and
center - width will be half of their original
amplitudes. Band defaults to a mode oriented to
pitched signals, i.e. voice, singing, or instru-
mental music. The -n (for noise) option uses
the alternate mode for un-pitched signals. Band
introduces noise in the shape of the filter,
i.e. peaking at the center frequency and set-
tling around it.
chorus gain-in gain-out delay decay speed deptch
-s | -t [ delay decay speed depth -s | -t ... ]
Add a chorus to a sound sample. Each quadtuple
delay/decay/speed/depth gives the delay in mil-
liseconds and the decay (relative to gain-in)
with a modulation speed in Hz using depth in
milliseconds. The modulation is either sinodial
(-s) or triangular (-t). Gain-out is the volume
of the output.
compand attack1,decay1[,attack2,decay2...]
in-dB1,out-dB1[,in-dB2,out-dB2...]
[gain] [initial-volume]
Compand (compress or expand) the dynamic range
of a sample. The attack and decay time specify
the integration time over which the absolute
value of the input signal is integrated to
determine its volume. Where more than one pair
of attack/decay parameters are specified, each
channel is treated separately and the number of
pairs must agree with the number of input chan-
nels. The second parameter is a list of points
on the compander's transfer function specified
in dB relative to the maximum possible signal
amplitude. The input values must be in a
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strictly increasing order but the transfer func-
tion does not have to be monotonically rising.
The special value -inf may be used to indicate
that the input volume should be associated out-
put volume. The points -inf,-inf and 0,0 are
assumed; the latter may be overridden, but the
former may not. The third (optional) parameter
is a postprocessing gain in dB which is applied
after the compression has taken place; the
fourth (optional) parameter is an initial volume
to be assumed for each channel when the effect
starts. This permits the user to supply a nomi-
nal level initially, so that, for example, a
very large gain is not applied to initial signal
levels before the companding action has begun to
operate: it is quite probable that in such an
event, the output would be severely clipped
while the compander gain properly adjusts
itself.
copy Copy the input file to the output file. This is
the default effect if both files have the same
sampling rate.
cut loopnumber
Extract loop #N from a sample.
deemph Apply a treble attenuation shelving filter to
samples in audio cd format. The frequency
response of pre-emphasized recordings is recti-
fied. The filtering is defined in the standard
document ISO 908.
echo gain-in gain-out delay decay [ delay decay ... ]
Add echoing to a sound sample. Each delay/decay
part gives the delay in milliseconds and the
decay (relative to gain-in) of that echo. Gain-
out is the volume of the output.
echos gain-in gain-out delay decay [ delay decay ... ]
Add a sequence of echos to a sound sample. Each
delay/decay part gives the delay in milliseconds
and the decay (relative to gain-in) of that
echo. Gain-out is the volume of the output.
flanger gain-in gain-out delay decay speed -s | -t
Add a flanger to a sound sample. Each triple
delay/decay/speed gives the delay in millisec-
onds and the decay (relative to gain-in) with a
modulation speed in Hz. The modulation is
either sinodial (-s) or triangular (-t). Gain-
out is the volume of the output.
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highp center
Apply a high-pass filter. The frequency
response drops logarithmically with center fre-
quency in the middle of the drop. The slope of
the filter is quite gentle.
lowp center
Apply a low-pass filter. The frequency response
drops logarithmically with center frequency in
the middle of the drop. The slope of the filter
is quite gentle.
map Display a list of loops in a sample, and miscel-
laneous loop info.
mask Add "masking noise" to signal. This effect
deliberately adds white noise to a sound in
order to mask quantization effects, created by
the process of playing a sound digitally. It
tends to mask buzzing voices, for example. It
adds 1/2 bit of noise to the sound file at the
output bit depth.
phaser gain-in gain-out delay decay speed -s | -t
Add a phaser to a sound sample. Each triple
delay/decay/speed gives the delay in millisec-
onds and the decay (relative to gain-in) with a
modulation speed in Hz. The modulation is
either sinodial (-s) or triangular (-t). The
decay should be less than 0.5 to avoid feedback.
Gain-out is the volume of the output.
pick Select the left or right channel of a stereo
sample, or one of four channels in a quadro-
phonic sample.
polyphase [ -w < num / ham > ]
[ -width < long / short / # > ]
[ -cutoff # ]
Translate input sampling rate to output sampling
rate via polyphase interpolation, a DSP algo-
rithm. This method is slow and uses lots of
RAM, but gives much better results then rate.
-w < nut / ham > : select either a Nuttal (~90
dB stopband) or Hamming (~43 dB stopband) win-
dow. Warning: Nuttall windows require 2x length
than Hamming windows. Default is nut.
-width long / short / # : specify the width of
the filter. long is 1024 samples; short is 128
samples. Alternatively, an exact number can be
used. Default is long.
-cutoff # : specify the filter cutoff frequency
June 28, 1999 10
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in terms of fraction of bandwidth. If upsam-
pling, then this is the fraction of the orignal
signal that should go through. If downsampling,
this is the fraction of the signal left after
downsampling. Default is 0.95. Remember that
this is a float.
rate Translate input sampling rate to output sampling
rate via linear interpolation to the Least Com-
mon Multiple of the two sampling rates. This is
the default effect if the two files have differ-
ent sampling rates and the preview options was
specified. This is fast but noisy: the spectrum
of the original sound will be shifted upwards
and duplicated faintly when up-translating by a
multiple. Lerp-ing is acceptable for cheap
8-bit sound hardware, but for CD-quality sound
you should instead use either resample or
polyphase. If you are wondering which of Sox's
rate changing effects to ues, you will want to
read a detailed analysis of all of them at
http://eakaw2.et.tu-dresden.de/~andreas/resam-
ple/resample.html
resample [ rolloff [ beta ] ]
Translate input sampling rate to output sampling
rate via simulated analog filtration. This
method is slower than rate, but gives much bet-
ter results. rolloff refers to the cut-off fre-
quency of the low pass filter and is given in
terms of the Nyquist frequency for the lower
sample rate. rolloff therefor should be some-
thing between 0. and 1., in practice 0.8-0.95.
beta trades stop band rejection against transi-
tion width from passband to stop band. Larger
beta means a slower transition and greater stop-
band rejection. beta should be at least greater
than 2. The default is rollof 0.8, beta 17.5,
which is rather conservative with respect to
aliasing. Lower beta and higher rolloff values
preserve more high frequency signal energy, but
introduce measurable artifacts. This is the
default effect if the two files have different
sampling rates.
reverb gain-out delay [ delay ... ]
Add reverbation to a sound sample. Each delay
is given in milliseconds and its feedback is
depending on the reverb-time in milliseconds.
Each delay should be in the range of half to
quarter of reverb-time to get a realistic rever-
bation. Gain-out is the volume of the output.
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reverse Reverse the sound sample completely. Included
for finding Satanic subliminals.
split Turn a mono sample into a stereo sample by copy-
ing the input channel to the left and right
channels.
stat [ debug | -v ]
Do a statistical check on the input file, and
print results on the standard error file. stat
may copy the file untouched from input to out-
put, if you select an output file. The "Volume
Adjustment:" field in the statistics gives you
the argument to the -v number which will make
the sample as loud as possible without clipping.
There is an optional parameter -v that will
print out the "Volume Adjustment:" field's value
and return. This could be of use in scripts to
auto convert the volume. There is an also an
optional parameter debug that will place sox
into debug mode and print out a hex dump of the
sound file from the internal buffer that is in
32-bit signed PCM data. This is mainly only of
use in tracking down endian problems that creep
in to sox on cross-platform versions.
swap [ 1 2 3 4 ]
Swap channels in multi-channel sound files. In
files with more than 2 channels you may specify
the order that the channels should be rearranged
in.
vibro speed [ depth ]
Add the world-famous Fender Vibro-Champ sound
effect to a sound sample by using a sine wave as
the volume knob. Speed gives the Hertz value of
the wave. This must be under 30. Depth gives
the amount the volume is cut into by the sine
wave, ranging 0.0 to 1.0 and defaulting to 0.5.
Sox enforces certain effects. If the two files have dif-
ferent sampling rates, the requested effect must be one of
copy, or rate, If the two files have different numbers of
channels, the avg effect must be requested.
BUGS
The syntax is horrific. It's very tempting to include a
default system that allows an effect name as the program
name and just pipes a sound sample from standard input to
standard output, but the problem of inputting the sample
rates makes this unworkable.
Please report any bugs found in this version of sox to
Chris Bagwell (cbagwell@sprynet.com)
June 28, 1999 12
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FILES
SEE ALSO
play(1), rec(1)
NOTICES
The echoplex effect is: Copyright (C) 1989 by Jef
Poskanzer.
Permission to use, copy, modify, and distribute this soft-
ware and its documentation for any purpose and without fee
is hereby granted, provided that the above copyright
notice appear in all copies and that both that copyright
notice and this permission notice appear in supporting
documentation. This software is provided "as is" without
express or implied warranty.
The version of Sox that accompanies this manual page is
support by Chris Bagwell (cbagwell@sprynet.com). Please
refer any questions regarding it to this address. You may
obtain the latest version at the the web site
http://home.sprynet.com/~cbagwell/sox.html
June 28, 1999 13