ref: b3524a9a9831d9d3dd49bfa8ee1d3ad55c1ff2a2
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 ] infile [ for-
mat options ] outfile [ effect [ effect options ... ] ]
General options: [ -e ] [ -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 ]
Effects:
avg [ -l | -r ]
band [ -n ] center [ width ]
bandpass frequency bandwidth
bandreject frequency bandwidth
check
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]
copy
cut
deemph
echo gain-in gain-out delay decay [ delay decay ...]
echos gain-in gain-out delay decay [ delay decay ...]
filter [ low ]-[ high ] [ window-len [ beta ]]
flanger gain-in gain-out delay decay speed -s | -t
highp center
highpass frequency
lowp center
lowpass frequency
map
mask
pan direction
phaser gain-in gain-out delay decay speed -s | -t
pick
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
speed factor
split
stat [ debug | -v ]
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SoX(1) SoX(1)
stretch [ factor [ window fade shift fading ]
swap [ 1 2 | 1 2 3 4 ]
vibro speed [ depth ]
vol gain [ type ]
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 apply a sound effect 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 type What format the data is stored in. Most popular
are 8-bit or 16-bit words.
data format
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.voc
translates a sound file 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.
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Format options:
Format options effect the audio samples that they immedi-
ately precede. 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. 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.
-t filetype
gives the type of the sound sample file.
-r rate Give sample rate in Hertz of file. To cause the
output file to have a different sample rate than
the input file, include this option with the
appropriate rate value along with 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 be used
with its default parameters.
-s/-u/-U/-A/-a/-i/-g
The sample data format is signed linear (2's
complement), unsigned linear, U-law (logarith-
mic), A-law (logarithmic), ADPCM, IMA_ADPCM, or
GSM. U-law and A-law are the U.S. and interna-
tional 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.
IMA_ADPCM is also a 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.
-b/-w/-l/-f/-d/-D
The sample data type 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-
format floating-point data is not portable.
IEEE floats are a fixed, portable format.
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-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 appro-
priate value with the output file options. If
the input and output file have a different num-
ber of channels then the avg effect must be
used. If the avg effect is not specified on the
command line it will be invoked with default
parameters.
General options:
-e When used after the input file (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.
-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. 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.
FILE TYPES
SoX uses the file extension of the input and output file
to determine what type of file format 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.
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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 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 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.
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.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 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 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.
Sound Files are used by academic music software
such as the CSound package, and the MixView
sound sample editor.
.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
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package, although not all are supported by the
samplers themselves. Currently loop points are
ignored.
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.
.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
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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, .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),
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 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.
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 channels
differ from the number of output channels. When
reducing the number of channels it is possible
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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.
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]
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
July 24, 2000 9
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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 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.
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
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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 Nut-
tall 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 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. See filter for a
highpass effect with sharper cutoff.
highpass frequency
Butterworth highpass filter. Description com-
ming soon!
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. 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
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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 Select the left or right channel of a stereo
sample, or one of four channels in a quadro-
phonic sample.
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
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algorithm. 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 bandwidth, also know as
the Nyquist frequency. Please see the resample
effect for further information on Nyquist fre-
quency. If upsampling, then this is the frac-
tion 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 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
July 24, 2000 13
SoX(1) SoX(1)
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 1/2 of the sample rate.
This refers to the fact that an audio file can
only represent frequencies up to 1/2 of the sam-
ple rate. Therefore, when reducing the sample
rate of an audio file a filter will remove all
frequency information above the new Nyquist fre-
quency. 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 filter proper-
ties, it is common to have false frequency 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.
July 24, 2000 14
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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.
speed 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
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SoX(1) SoX(1)
higher. 0.5 halves speed thus time length dou-
bles and pitch is one octave lower.
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.
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.
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
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SoX(1) SoX(1)
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 ]
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. 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 change loga-
rithmically. 0.0 is constant while +6 doubles
the amplitude.
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. 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@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
July 24, 2000 17