SoX(1) SoX(1)
July 24, 2000
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 ]
[ -b/-w/-l/-f/-d/-D ]
[ -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 ]
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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 | % | s]
[ below_periods duration
threshold[ d | % | s ]]
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 popular 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 characteristics 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 characteristics:
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 popular 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.
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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 specifying 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
The sample data encoding is signed linear (2's complement),
unsigned linear, U-law (logarithmic), A-law (logarithmic),
ADPCM, IMA_ADPCM, or GSM.
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 form of sound compression that has a good
compromise between good sound quality and fast
encoding/decoding time. It is used for telephone sound
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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 different
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 compression 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/-f/-d/-D
The sample data size is in bytes, 16-bit words, 32-bit
longwords, 32-bit floats, 64-bit double floats, or 80-bit
IEEE floats. Floats and double 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-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. 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 specified 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.
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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. Useful 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 format.
.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
description 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 commercial
packages on the Mac.
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.cdr CD-R
CD-R files are used in mastering music on Compact 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 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. Subsequent 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
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 telecommunications (GSM). Its
good for its purpose, shrinking 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 compression. The Macintosh has
wacky file formats 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
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specify a filename for consistency.
.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.
Decoding 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 format defined by
NIST (National Institute of Standards and Technology) and is
used with speech audio. SoX can read these files when they
contain ulaw and PCM data. It will ignore any header
information that says the data is compressed 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.
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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. Handles reading of files which
do not have the sample 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 generated 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 formatting
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
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format options will cause a format conversion, and the .wav
will written appropriately. 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 correspond 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 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 message. 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 specifying 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 automatically 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.
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The avg effect can also be invoked with up to 16 double-
precision numbers, which specify the proportion 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 instrumental 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 clipping. 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 coming 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 milliseconds 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...]
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[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 separately and the number of pairs must agree
with the number of input channels. 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 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 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 nominal 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.
The fifth (optional) parameter is a delay in seconds. The
input signal is analyzed immediately 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
clipping 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.
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deemph Apply a treble attenuation shelving filter to samples in
audio cd format. The frequency response of pre-emphasized
recordings is rectified. 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 explanation.
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 specified.
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 samples 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
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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 milliseconds 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 comming 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 miscellaneous loop
info.
mask Add "masking noise" to signal. This effect deliberately
adds white noise to a sound in order to mask quantization
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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 contains 2 channels, then it will create the
missing 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 milliseconds 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 quadrophonic 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 contain 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 algorithm. This method is
slow and uses lots of RAM, but gives much better results
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than rate.
-w < nut / ham > : select either a Nuttal (~90 dB stopband)
or Hamming (~43 dB stopband) window. Default is nut.
-width long / short / # : specify the (approximate) width of
the filter. long is 1024 samples; 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 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 Common Multiple of the two
sampling rates. This is the default effect if the two files
have different 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-
dresden.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 better 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
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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 converter 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 (frequencies 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 better. When increasing
the sample rate of an audio file you would not expect to
have any frequencies exist that are past the original
Nyquist frequency. Because of resampling properties, it is
common to have alaising data created 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
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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 interpolation 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
reverberation. 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 | % | s]
[ below_periods duration
threshold[ d | % | s ]]
Removes silence from the beginning or end of a sound file.
Silence is anything below a specified threshold.
When trimming silence from the beginning of a sound file,
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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 s to indicated the
value is in decibels, percent, or an exact signed long
interger sample value. A value of '0s' 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 doubles 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 copying 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 Adjustment:"
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.
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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 duration. 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 specified. 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)
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<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 samples 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 parameter 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 volume
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 logarithmically.
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 prevent 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.
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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/
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