ref: 6bae11ebb8908f2836f82b52c2b3b117ffaa5191
dir: /sox.txt/
SoX(1) SoX(1)
NAME
sox - Sound eXchange : universal sound sample translator
SYNOPSIS
sox infile outfile
sox [ general options ] [ format options ] infile
-e effect [ effect options ]
sox [ general options ] [ format options ] infile
[ format options ] outfile
[ effect [ effect options ] ... ]
General options:
[ -h ] [ -p ] [ -v volume ] [ -V ]
Format options:
[ -t filetype ] [ -r rate ] [ -s/-u/-U/-A/-a/-i/-g/-f
]
[ -b/-w/-l ]
[ -c channels ] [ -x ] [ -e ]
Effects:
avg [ -l | -r | -f | -b | n,n,...,n ]
band [ -n ] center [ width ]
bandpass frequency bandwidth
bandreject frequency bandwidth
chorus gain-in gain out delay decay speed depth
-s | -t [ delay decay speed depth -s | -t ]
compand attack1,decay1[,attack2,decay2...]
in-dB1,out-dB1[,in-dB2,out-dB2...]
[ gain [ initial-volume [ delay ] ] ]
copy
dcshift shift [ limitergain ]
deemph
earwax
echo gain-in gain-out delay decay [ delay decay ... ]
echos gain-in gain-out delay decay [ delay decay ... ]
fade [ type ] fade-in-length
[ stop-time [ fade-out-length ] ]
filter [ low ]-[ high ] [ window-len [ beta ]]
flanger gain-in gain-out delay decay speed < -s | -t >
highp frequency
highpass frequency
lowp frequency
lowpass frequency
map
mask
pan direction
phaser gain-in gain-out delay decay speed < -s | -t >
pick [ -1 | -2 | -3 | -4 | -l | -r ]
pitch shift [ width interpole fade ]
polyphase [ -w < nut / ham > ]
[ -width < long / short / # > ]
[ -cutoff # ]
rate
resample [ -qs | -q | -ql ] [ rolloff [ beta ] ]
reverb gain-out reverb-time delay [ delay ... ]
reverse
silence above_periods [ duration threshold[ d | % ]
[ below_periods duration
threshold[ d | % ]]
speed [ -c ] factor
split
stat [ -s n ] [ -rms ] [ -v ] [ -d ]
stretch [ factor [ window fade shift fading ]
swap [ 1 2 | 1 2 3 4 ]
synth [ length ] type mix [ freq [ -freq2 ]
[ off ] [ ph ] [ p1 ] [ p2 ] [ p3 ]
trim start [ length ]
vibro speed [ depth ]
vol gain [ type [ limitergain ] ]
DESCRIPTION
SoX is a command line program that can convert most popu�
lar audio files to most other popular audio file formats.
It can optionally change the audio sample data type and
apply one or more sound effects to the file during this
translation.
There are two types of audio files formats that SoX can
work with. The first are self-describing file formats.
These contain a header that completely describe the char�
acteristics of the audio data that follows.
The second type are headerless data, or sometimes called
raw data. A user must pass enough information to SoX on
the command line so that it knows what type of data it
contains.
Audio data can usually be totally described by four char�
acteristics:
rate The sample rate is in samples per second. For
example, CD sample rates are at 44100.
data size The precision the data is stored in. Most popu�
lar are 8-bit bytes or 16-bit words.
data encoding
What encoding the data type uses. Examples are
u-law, ADPCM, or signed linear data.
channels How many channels are contained in the audio
data. Mono and Stereo are the two most common.
Please refer to the soxexam(1) manual page for a long
description with examples on how to use sox with various
types of file formats.
OPTIONS
The option syntax is a little grotty, but in essence:
sox file.au file.wav
translates a sound file in SUN Sparc .AU format into a
Microsoft .WAV file, while
sox -v 0.5 file.au -r 12000 file.wav mask
does the same format translation but also lowers the
amplitude by 1/2, changes the sampling rate to 12000
hertz, and applies the mask sound effect to the audio
data.
Format options:
Format options effect the audio samples that they
immediately preceed. 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 output data. By taking advantage of this, you can
override a input file's corrupted header or produce an
output file that is totally different style then the input
file. It is also how sox is informed about the format of
raw input data.
-t filetype
gives the type of the sound sample file. Useful
when file extension is not standard or for spec�
ifying the .auto file type.
-r rate Gives the sample rate in Hertz of the file. To
cause the output file to have a different sample
rate than the input file, include this option as
a part of the output options.
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 internally be
ran by sox using its default parameters.
-s/-u/-U/-A/-a/-i/-g/-f
The sample data encoding is signed linear (2's
complement), unsigned linear, U-law (logarith�
mic), A-law (logarithmic), ADPCM, IMA_ADPCM,
GSM, or Floating-point.
U-law (actually shorthand for mu-law) and A-law
are the U.S. and international standards for
logarithmic telephone sound compression. When
uncompressed it has roughly the precision of
12-byte PCM audio.
ADPCM is a form of sound compression that has a
good compromise between good sound quality and
fast encoding/decoding time. It is used for
telephone sound compression and places were full
fidelity is not as important. When uncompressed
it has roughly the precision of 16-bit PCM
audio. Popular version of ADPCM include G.726,
MS ADPCM, and IMA ADPCM. The -a flag has dif�
ferent meanings in different file handlers. In
.wav files it represents MS ADPCM files, in all
others it means G.726 ADPCM. IMA ADPCM is a
specific form of adpcm compression, slightly
simpler and slightly lower fidelity than
Microsoft's flavor of ADPCM. IMA ADPCM is also
called DVI ADPCM.
GSM is a standard used for telephone sound com�
pression in European countries and its gaining
popularity because of its quality. It usually
is CPU intensive to work with GSM audio data.
-b/-w/-l The sample data size is in bytes, 16-bit words,
or 32-bit long words.
-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-
format floating-point data is not portable.
-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. To cause the output file to
have a different number of channels than the
input file, include this option with the output
file options. If the input and output file have
a different number of channels then the avg
effect must be used. If the avg effect is not
specified on the command line it will be invoked
internally with default parameters.
-e When used after the input filename (so that it
applies to the output file) it allows you to
avoid giving an output filename and will not
produce an output file. It will apply any spec�
ified effects to the input file. This is mainly
useful with the stat effect but can be used with
others.
General options:
-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 than the input file. Currently, this
defaults to using the rate effect instead of the
resample effect for sample rate changes.
-v volume Change amplitude (floating point); less than 1.0
decreases, greater than 1.0 increases. May use
a negative number to invert the phase of the
audio data. It is interesting to note that we
percieve volume logarithmically but this adjusts
the amplitude linearly.
Note: see the stat effect for information on
finding the maximum value that can be used with
this option without causing audio data be be
clipped.
-V Print a description of processing phases. Use�
ful for figuring out exactly how sox is mangling
your sound samples.
FILE TYPES
SoX attempts to determine the file type of input files
automatically by looking at the header of the audio file.
When it is unable to detect the file type or if its an
output file then it uses the file extension of the file to
determine what type of file format handler to use. This
can be overridden by specifying the "-t" option on the
command line.
The input and output files may be read from standard in
and out. This is done by specifying '-' as the filename.
File formats which have headers are checked, if that
header doesn't seem right, the program exits with an
appropriate message.
The following file formats 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 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 on Com�
pact Disks. The audio data on a CD-R disk is a
raw audio file with a format of stereo 16-bit
signed samples at a 44khz sample rate. There is
a special blocking/padding oddity at the end of
the audio file and is why it needs its own han�
dler.
.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 first sample and
the sample value. Values are normalized so that
the 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 analyz�
ers 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 sup�
port 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
compression. The Macintosh has wacky file for�
mats and this format handler apparently doesn't
handle 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.
.nul Null file handler. This is a fake file hander
that act as if its reading a stream of 0's from
a while or fake writing output to a file. This
is not a very useful file handler in most cases.
It might be useful in some scripts were you do
not want to read or write from a real file but
would like to specify a filename for consis�
tency.
.ogg Ogg Vorbis Compressed Audio.
Ogg Vorbis is a open, patent-free codec designed
for compressing music and streaming audio. It
is similar to MP3, VQF, AAC, and other lossy
formats. sox can decode all types of Ogg Vorbis
files, but can only encode at 128 kbps. Decod�
ing is somewhat CPU intensive and encoding is
very CPU intensive.
Ogg Vorbis in sox is optional and requires
access to external Ogg Vorbis libraries. To see
if there is support for Ogg Vorbis run sox -h
and look for it under the list of supported file
formats as "vorbis".
ossdsp OSS /dev/dsp device driver
This is a pseudo-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 format as passed in to /fBSoX. 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.
Sound Files are used by academic music software
such as the CSound package, and the MixView
sound sample editor.
.sph
SPHERE (SPeech HEader Resources) is a file for�
mat defined by NIST (National Institute of Stan�
dards and Technology) and is used with speech
audio. SoX can read these files when they con�
tain ulaw and PCM data. It will ignore any
header information that says the data is com�
pressed using shorten compression and will treat
the data as either ulaw or PCM. This will allow
SoX and the command line shorten program to be
ran together using pipes to uncompress the data
and then pass the result to SoX for processing.
.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.
.snd
Under DOS this file format is the same as the
.sndt format. Under all other platforms it is
the same as the .au format.
.sndt SoundTool files.
This is an older DOS file format.
sunau Sun /dev/audio device driver
This is a pseudo-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.
vorbis See .ogg format.
.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
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. Sox currently can read PCM, ULAW, ALAW,
MS ADPCM, and IMA (or DVI) ADPCM. It can write
all of these formats including (NEW!) the ADPCM
encoding.
.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
encoding (signed, unsigned, etc.) of the sample
file must be given. The number of channels
defaults to 1.
.ub, .sb, .uw, .sw, .ul, .al, .sl
These are several suffices which serve as a
shorthand for raw files with a given size and
encoding. Thus, ub, sb, uw, sw, ul and sl cor�
respond to "unsigned byte", "signed byte",
"unsigned word", "signed word", "ulaw" (byte),
"alaw" (byte), and "signed long". The sample
rate defaults to 8000 hz if not explicitly 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 head�
ers.) 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
Multiple effects may be applied to the audio data by spec�
ifying them one after another at the end of the command
line.
avg [ -l | -r | -f | -b | n,n,...,n ]
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 channels
differ from the number of output channels. When
reducing the number of channels it is possible
to manually specify the avg effect and use the
-l, -r, -f, or -b options to select only the
left, right, front, or back channel(s) for the
output instead of averaging the channels. The
-f and -b options maintain left/right stereo
separation; use the avg effect twice to select a
single channel.
The avg effect can also be invoked with up to 16
double-precision numbers, which specify the pro�
portion of each input channel that is to be
mixed into each output channel. In two-channel
mode, 4 numbers are given: l->l, l->r, r->l, and
r->r, respectively. In four-channel mode, the
first 4 numbers give the proportions for the
left-front output channel, as follows: lf->lf,
rf->lf, lb->lf, and rb->rf. The next 4 give the
right-front output in the same order, then left-
back and right-back.
It is also possible to use the 16 numbers to
expand or reduce the channel count; just specify
0 for unused channels. Finally, if fewer than 4
numbers are given, certain special abbreviations
may be invoked; see the source code for details.
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.
Warning: -n introduces a power-gain of about
11dB in the filter, so beware of output clip�
ping. Band introduces noise in the shape of the
filter, i.e. peaking at the center frequency and
settling around it. See filter for a bandpass
effect with steeper shoulders.
bandpass frequency bandwidth
Butterworth bandpass filter. Description coming
soon!
bandreject frequency bandwidth
Butterworth bandreject filter. Description com�
ing soon!
chorus gain-in gain-out delay decay speed depth
-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 [delay ] ] ]
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; attacks refer to increases
in volume and decays refer to decreases. Where
more than one pair of attack/decay parameters
are specified, each channel is treated sepa�
rately and the number of pairs must agree with
the number of input channels. The second param�
eter 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 strictly increasing
order but the transfer function does not have to
be monotonically rising. The special value -inf
may be used to indicate that the input volume
should be associated output 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 postprocess�
ing gain in dB which is applied after the com�
pression has taken place; the fourth (optional)
parameter is an initial volume to be assumed for
each channel when the effect starts. This per�
mits the user to supply a nominal level ini�
tially, 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 com�
pander gain properly adjusts itself.
The fifth (optional) parameter is a delay in
seconds. The input signal is analyzed immedi�
ately to control the compander, but it is
delayed before being fed to the volume adjuster.
Specifying a delay approximately equal to the
attack/decay times allows the compander to
effectively operate in a "predictive" rather
than a reactive mode.
copy Copy the input file to the output file. This is
the default effect if both files have the same
sampling rate.
dcshift shift [ limitergain ]
DC Shift the audio data, with basic linear
amplitudate formula. This is most useful if
your audio data tends to not be centered around
a value of 0. Shifting it back will allow you
to get the most volume adjustments without clip�
ping audio data.
The first option is the dcshift value. It is a
floating point number that indicates the amount
to shift.
An option limtergain value can be specified as
well. It should have a value much less then 1.0
and is used only on peaks to prevent clipping.
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.
earwax Makes sound easier to listen to on headphones.
Adds audio-cues to samples in audio cd format so
that when listened to on headphones the stereo
image is moved from inside your head (standard
for headphones) to outside and in front of the
listener (standard for speakers). See
www.geocities.com/beinges for a full explana�
tion.
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.
fade [ type ] fade-in-length
[ stop-time [ fade-out-length ] ]
Add a fade effect to the beginning, end, or both
of the audio data.
For fade-ins, this starts from the first sample
and ramps the volume of the audio from 0 to full
volume over fade-in-length seconds. Specify 0
seconds if no fade-in is wanted.
For fade-outs, the audio data will be truncated
at the stop-time and the volume will be ramped
from full volume down to 0 starting at fade-out-
length seconds before the stop-time. No fade-
out is performed if these options are not speci�
fied.
All times can be specified in either periods of
time or sample counts. To specify time periods
use the format hh:mm:ss.frac format. To specify
using sample counts, specify the number of sam�
ples and append the letter 's' to the sample
count (for example 8000s).
An optional type can be specified to change the
type of envelope. Choices are q for quarter of
a sinewave, h for half a sinewave, t for linear
slope, l for logarithmic, and p for inverted
parabola. The default is a linear slope.
filter [ low ]-[ high ] [ window-len [ beta ] ]
Apply a Sinc-windowed lowpass, highpass, or
bandpass filter of given window length to the
signal. low refers to the frequency of the
lower 6dB corner of the filter. high refers to
the frequency of the upper 6dB corner of the
filter.
A lowpass filter is obtained by leaving low
unspecified, or 0. A highpass filter is
obtained by leaving high unspecified, or 0, or
greater than or equal to the Nyquist frequency.
The window-len, if unspecified, defaults to 128.
Longer windows give a sharper cutoff, smaller
windows a more gradual cutoff.
The beta, if unspecified, defaults to 16. This
selects a Kaiser window. You can select a
Nuttall window by specifying anything <= 2.0
here. For more discussion of beta, look under
the resample effect.
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.
highp frequency
Apply a single pole recursive high-pass filter.
The frequency response drops logarithmically
with I frequency in the middle of the drop. The
slope of the filter is quite gentle. See filter
for a highpass effect with sharper cutoff.
highpass frequency
Butterworth highpass filter. Description com�
ming soon!
lowp frequency
Apply a single pole recursive low-pass filter.
The frequency response drops logarithmically
with frequency in the middle of the drop. The
slope of the filter is quite gentle. See filter
for a lowpass effect with sharper cutoff.
lowpass frequency
Butterworth lowpass filter. Description coming
soon!
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.
pan direction
Pan the sound of an audio file from one channel
to another. This is done by changing the volume
of the input channels so that it fades out on
one channel and fades-in on another. If the
number of input channels is different then the
number of output channels then this effect tries
to intelligently handle this. For instance, if
the input contains 1 channel and the output con�
tains 2 channels, then it will create the miss�
ing channel itself. The direction is a value
from -1.0 to 1.0. -1.0 represents far left and
1.0 represents far right. Numbers in between
will start the pan effect without totally muting
the opposite channel.
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 [ -1 | -2 | -3 | -4 | -l | -r ]
Select the left or right channel of a stereo
sample, or one of four channels in a quadro�
phonic sample. The -l and -r options represent
either the left or right channel. It is
required that you use the -c 1 command line
option in order to force the output file to con�
tain only 1 channel.
pitch shift [ width interpole fade ]
Change the pitch of file without affecting its
duration by cross-fading shifted samples. shift
is given in cents. Use a positive value to shift
to treble, negative value to shift to bass.
Default shift is 0. width of window is in ms.
Default width is 20ms. Try 30ms to lower pitch,
and 10ms to raise pitch. interpole option, can
be "cubic" or "linear". Default is "cubic". The
fade option, can be "cos", "hamming", "linear"
or "trapezoid". Default is "cos".
polyphase [ -w < nut / 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 than rate.
-w < nut / ham > : select either a Nuttal (~90
dB stopband) or Hamming (~43 dB stopband) win�
dow. Default is nut.
-width long / short / # : specify the (approxi�
mate) width of the filter. long is 1024 sam�
ples; short is 128 samples. Alternatively, an
exact number can be used. Default is long. The
short option is not recommended, as it produces
poor quality results.
-cutoff # : specify the filter cutoff frequency
in terms of fraction of frequency bandwidth,
also know as the Nyquist frequency. Please see
the resample effect for further information on
Nyquist frequency. If upsampling, then this is
the fraction of the original signal that should
go through. If downsampling, this is the frac�
tion 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 rate changing effects to
use, you will want to read a detailed analysis
of all of them at http://eakaw2.et.tu-dres�
den.de/~wilde/resample/resample.html
resample [ -qs | -q | -ql ] [ 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.
By default, linear interpolation is used, with a
window width about 45 samples at the lower of
the two rate. This gives an accuracy of about
16 bits, but insufficient stopband rejection in
the case that you want to have rolloff greater
than about 0.80 of the Nyquist frequency.
The -q* options will change the default values
for rolloff and beta as well as use quadratic
interpolation of filter coefficients, resulting
in about 24 bits precision. The -qs, -q, or -ql
options specify increased accuracy at the cost
of lower execution speed. It is optional to
specify rolloff and beta parameters when using
the -q* options.
Following is a table of the reasonable defaults
which are built-in to sox:
Option Window rolloff beta interpolation
------ ------ ------- ---- -------------
(none) 45 0.80 16 linear
-qs 45 0.80 16 quadratic
-q 75 0.875 16 quadratic
-ql 149 0.94 16 quadratic
------ ------ ------- ---- -------------
-qs, -q, or -ql use window lengths of 45, 75, or
149 samples, respectively, at the lower sample-
rate of the two files. This means progressively
sharper stop-band rejection, at proportionally
slower execution times.
rolloff refers to the cut-off frequency of the
low pass filter and is given in terms of the
Nyquist frequency for the lower sample rate.
rolloff therefore should be something between
0.0 and 1.0, in practice 0.8-0.95. The defaults
are indicated above.
The Nyquist frequency is equal to (sample rate /
2). Logically, this is because the A/D con�
verter needs at least 2 samples to detect 1
cycle at the Nyquist frequency. Frequencies
higher then the Nyquist will actually appear as
lower frequencies to the A/D converter and is
called aliasing. Normally, A/D converts run the
signal through a highpass filter first to avoid
these problems.
Similar problems will happen in software when
reducing the sample rate of an audio file (fre�
quencies above the new Nyquist frequency can be
aliased to lower frequencies). Therefore, a
good resample effect will remove all frequency
information above the new Nyquist frequency.
The rolloff refers to how close to the Nyquist
frequency this cutoff is, with closer being bet�
ter. When increasing the sample rate of an
audio file you would not expect to have any fre�
quencies exist that are past the original
Nyquist frequency. Because of resampling prop�
erties, it is common to have alaising data cre�
ated that is above the old Nyquist frequency.
In that case the rolloff refers to how close to
the original Nyquist frequency to use a highpass
filter to remove this false data, with closer
also being better.
The beta parameter determines the type of filter
window used. Any value greater than 2.0 is the
beta for a Kaiser window. Beta <= 2.0 selects a
Nuttall window. If unspecified, the default is
a Kaiser window with beta 16.
In the case of Kaiser window (beta > 2.0), lower
betas produce a somewhat faster transition from
passband to stopband, at the cost of noticeable
artifacts. A beta of 16 is the default, beta
less than 10 is not recommended. If you want a
sharper cutoff, don't use low beta's, use a
longer sample window. A Nuttall window is
selected by specifying any 'beta' <= 2, and the
Nuttall window has somewhat steeper cutoff than
the default Kaiser window. You will probably
not need to use the beta parameter at all,
unless you are just curious about comparing the
effects of Nuttall vs. Kaiser windows.
This is the default effect if the two files have
different sampling rates. Default parameters
are, as indicated above, Kaiser window of length
45, rolloff 0.80, beta 16, linear interpolation.
NOTE: -qs is only slightly slower, but more
accurate for 16-bit or higher precision.
NOTE: In many cases of up-sampling, no interpo�
lation is needed, as exact filter coefficients
can be computed in a reasonable amount of space.
To be precise, this is done when
input_rate < output_rate
&&
output_rate/gcd(input_rate,output_rate) <= 511
reverb gain-out delay [ delay ... ]
Add reverberation 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�
beration. Gain-out is the volume of the output.
reverse Reverse the sound sample completely. Included
for finding Satanic subliminals.
silence above_periods [ duration threshold[ d | % ]
[ below_periods duration
threshold[ d | % ]]
Removes silence from the beginning or end of a
sound file. Silence is anything below a speci�
fied threshold.
When trimming silence from the beginning of a
sound file, you specify a duration of audio that
is above a given silence threshold before audio
data is processed. You can also specify the
count of periods of none silence you want to
detect before processing audio data. Specify a
period of 0 if you do not want to trim data from
the front of the sound file.
When optionally trimming silence form the end of
a sound file, you specify the duration of audio
that must be below a given threshold before
stopping to process audio data. A count of
periods that occur below the threshold may also
be speficied. If this options are not specified
then data is not trimmed from the end of the
audio file.
Duration counts may be in the format of time,
hh:mm:ss.frac, or in the exact count of samples.
Threshold may be suffixed with d, or % to indi�
cated the value is in decibels or a percentage
of max value of the sample value. A value of
'0%' will look for total silence.
speed [ -c ] factor
Speed up or down the sound, as a magnetic tape
with a speed control. It affects both pitch and
time. A factor of 1.0 means no change, and is
the default. 2.0 doubles speed, thus time
length is cut by a half and pitch is one octave
higher. 0.5 halves speed thus time length dou�
bles and pitch is one octave lower. If the
optional -c parameter is used then the factor is
specified in "cents".
split Turn a mono sample into a stereo sample by copy�
ing the input channel to the left and right
channels.
stat [ -s n ] [-rms ] [ -v ] [ -d ]
Do a statistical check on the input file, and
print results on the standard error file. Audio
data is passed unmodified from input to output
file unless used along with the -e option.
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.
The option -v will print out the "Volume Adjust�
ment:" field's value only and return. This
could be of use in scripts to auto convert the
volume.
The -s n option is used to scale the input data
by a given factor. The default value of n is
the max value of a signed long variable
(0x7fffffff). Internal effects always work with
signed long PCM data and so the value should
relate to this fact.
The -rms option will convert all output average
values to root mean square format.
There is also an optional parameter -d that will
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.
stretch factor [window fade shift fading]
Time stretch file by a given factor. Change
duration without affecting the pitch. factor of
stretching: >1.0 lengthen, <1.0 shorten dura�
tion. window size is in ms. Default is 20ms.
The fade option, can be "lin". shift ratio, in
[0.0 1.0]. Default depends on stretch factor.
1.0 to shorten, 0.8 to lengthen. The fading
ratio, in [0.0 0.5]. The amount of a fade's
default depends on factor and shift.
swap [ 1 2 | 1 2 3 4 ]
Swap channels in multi-channel sound files.
Optionally, you may specify the channel order
you would like the output in. This defaults to
output channel 2 and then 1 for stereo and 2, 1,
4, 3 for quad-channels. An interesting feature
is that you may duplicate a given channel by
overwriting another. This is done by repeating
an output channel on the command line. For
example, swap 2 2 will overwrite channel 1 with
channel 2's data; creating a stereo file with
both channels containing the same audio data.
synth [ length ] type mix [ freq [ -freq2 ]
[ off ] [ ph ] [ p1 ] [ p2 ] [ p3 ]
The synth effect will generate various types of
audio data. Although this effect is used to
generate audio data, an input file must be spec�
ified. The length of the input audio file
determines the length of the output audio file.
<length> length in sec or hh:mm:ss.frac,
0=inputlength, default=0
<type> is sine, square, triangle, sawtooth,
trapetz, exp, whitenoise, pinknoise, brownnoise,
default=sine
<mix> is create, mix, amod, default=create
<freq> frequency at beginning in Hz, not used
for noise..
<freq2> frequency at end in Hz, not used for
noise.. <freq/2> can be given as %%n, where 'n'
is the number of half notes in respect to A
(440Hz)
<off> Bias (DC-offset) of signal in percent,
default=0
<ph> phase shift 0..100 shift phase 0..2*Pi, not
used for noise..
<p1> square: Ton/Toff, triangle+trapetz: rising
slope time (0..100)
<p2> trapetz: ON time (0..100)
<p3> trapetz: falling slope position (0..100)
trim start [ length ]
Trim can trim off unwanted audio data from the
beginning and end of the audio file. Audio sam�
ples are not sent to the output stream until the
start location is reached.
The optional length parameter tells the number
of samples to output after the start sample and
is used to trim off the back side of the audio
data. Using a value of 0 for the start parame�
ter will allow trimming off the back side only.
Both options can be specified using either an
amount of time and an exact count of samples.
The format for specifying lengths in time is
hh:mm:ss.frac. A start value of 1:30.5 will not
start until 1 minute, thirty and 1/2 seconds
into the audio data. The format for specifying
sample counts is the number of samples with the
letter 's' appended to it. A value of 8000s
will wait until 8000 samples are read before
starting to process audio data.
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.
vol gain [ type [ limitergain ] ]
The vol effect is much like the command line
option -v. It allows you to adjust the volume
of an input file and allows you to specify the
adjustment in relation to amplitude, power, or
dB. If type is not specified then it defaults
to amplitude.
When type is amplitude then a linear change of
the amplitude is performed based on the gain.
Therefore, a value of 1.0 will keep the volume
the same, 0.0 to < 1.0 will cause the volume to
decrease and values of > 1.0 will cause the vol�
ume to increase. Beware of clipping audio data
when the gain is greater then 1.0. A negative
value performs the same adjustment while also
changing the phase.
When type is power then a value of 1.0 also
means no change in volume.
When type is dB the amplitude is changed loga�
rithmically. 0.0 is constant while +6 doubles
the amplitude.
An optional limitergain value can be specified
and should be a value much less then 1.0 (ie
0.05 or 0.02) and is used only on peaks to pre�
vent clipping. Not specifying this parameter
will cause no limiter to be used. In verbose
mode, this effect will display the percentage of
audio data that needed to be limited.
BUGS
The syntax is horrific. Thats the breaks when trying to
handle all things from the command line.
Please report any bugs found in this version of sox to
Chris Bagwell (cbagwell@sprynet.com)
FILES
SEE ALSO
play(1), rec(1), soxexam(1)
NOTICES
The version of Sox that accompanies this manual page is
support by Chris Bagwell (cbagwell@users.sourceforge.net).
Please refer any questions regarding it to this address.
You may obtain the latest version at the the web site
http://sox.sourceforge.net/
July 24, 2000 SoX(1)