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.de Sh
.br
.ne 5
.PP
\fB\\$1\fR
.PP
..
.de Sp
.if t .sp .5v
.if n .sp
..
.TH SoX 1 "June 28, 1999"
.SH NAME
sox \- Sound eXchange : universal sound sample translator
.SH SYNOPSIS
.B sox \fIinfile outfile \fB
.br
.B sox \fIinfile outfile \fB[ \fIeffect\fR 
.B [ \fIeffect options ...\fB ] ]
.br
.B sox \fIinfile \fB-e \fIeffect\fR 
.B [ \fIeffect options ...\fB ]
.br
.B sox
[\fI general options \fB ]
[ \fIformat options \fB ]
\fIifile\fB 
[ \fIformat options \fB ]
\fIofile\fB 
[ \fIeffect\fR [ \fIeffect options ...\fB ] ]
.P
\fIGeneral options:\fB
[ -e ]
[ -h ]
[ -p ]
[ -v \fIvolume\fB ]
[ -V ]
.P
\fIFormat options:\fB
[ \fB-t \fIfiletype\fB ]
[ -r \fIrate\fB ]
[ -s/-u/-U/-A/-a/-g ]
[ -b/-w/-l/-f/-d/-D ]
[ -c \fIchannels\fB ]
[ -x ]
.P
\fIEffects:\fB
.br
	avg [ \fI-l\fB | \fI-r\fB ]
.br
	band \fB[ \fI-n \fB] \fIcenter \fB[ \fIwidth\fB ]
.br
 	check 
.br
	chorus \fIgain-in gain out delay decay speed depth
		-s\fB | \fI-t\fB [ \fIdelay decay speed depth -s\fB | -fI-t\fB ]
.br
	copy
.br
	cut
.br
	deemph
.br
	echo \fIgain-in gain-out delay decay\fB [ \fIdelay decay ...\fB]
.br
	echos \fIgain-in gain-out delay decay\fB [ \fIdelay decay ...\fB]
.br
	flanger \fIgain-in gain-out delay decay speed -s\fB | -fI-t\fB
.br
	highp \fIcenter\fB
.br
	lowp \fIcenter\fB
.br
	map
.br
	mask
.br
	phaser \fIgain-in gain-out delay decay speed -s\fB | \fI-t\fB
.br
	pick
.br
	polyphase [ \fI-w \fR< \fInum\fR / \fIham\fR > ] 
               [ \fI -width \fR< \fI long \fR / \fIshort \fR / \fI# \fR> ] 
               [ \fI-cutoff # \fR ]
.br
	\fBrate 
.br
	resample
.br
	reverb \fIgain-out reverb-time delay\fB [ \fIdelay ... \fB]
.br
	reverse
.br
	split
.br
	stat [ \fIdebug\fB | \fI-v\fB ]
.br
	swap [ \fI1 2 3 4\fB ]
.br
	vibro \fIspeed \fB[ \fIdepth\fB ]
.SH DESCRIPTION
.I Sox
translates sound files from one format to another,
possibly doing a sound effect.
.SH OPTIONS
The option syntax is a little grotty, but in essence:
.br
	sox file.au file.voc 
.br
translates a sound sample in SUN Sparc .AU format 
into a SoundBlaster .VOC file, while
.br
	sox -v 0.5 file.au -r 12000 file.voc rate
.br
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
.B rate
\fIsound effect\fR loop.
.PP
File type options:
.TP 10
\fB-t\fI filetype
gives the type of the sound sample file.
.TP 10
\fB-r \fIrate\fR
Give sample rate in Hertz of file.
.TP 10
\fB-s/-u/-U/-A/-a/-g\fR
The sample data is signed linear (2's complement),
unsigned linear, U-law (logarithmic), A-law (logarithmic),
ADPCM, or GSM.
U-law and A-law are the U.S. and international
standards for logarithmic telephone sound compression.
ADPCM is form of sound compression that has a good
compromise between good sound quality and fast encoding/decoding
time.
GSM is a standard used for telephone sound compression in
European countries and its gaining popularity because of its
quality.
.TP 10
\fB-b/-w/-l/-f/-d/-D\fR
The sample data is in bytes, 16-bit words, 32-bit longwords,
32-bit floats, 64-bit double floats, or 80-bit IEEE floats.
Floats and double floats are in native machine format.
.TP 10
\fB-x\fR
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. ???
.TP 10
\fB-c \fIchannels\fR
The number of sound channels in the data file.
This may be 1, 2, or 4; for mono, stereo, or quad sound data.
.PP
General options:
.TP 10
\fB-e\fR
after the input file allows you to avoid giving
an output file and just name an effect.
This is mainly useful with the 
.B stat
effect but can be used with others.
.TP 10
\fB-h\fR
Print version number and usage information.
.TP 10
\fB-p\fR
Run in preview mode and run fast.  This will somewhat speed up
sox when the output format has a different number of channels and
a different rate then the input file.  The order that the effects 
are run in will be arranged for maximum speed and not quality.
.TP 10
\fB-v \fIvolume\fR
Change amplitude (floating point); 
less than 1.0 decreases, greater than 1.0 increases.
Note: we perceive volume logarithmically, not linearly.
Note: see the
.B stat
effect.
.TP 10
\fB-V\fR
Print a description of processing phases.
Useful for figuring out exactly how
.I sox
is mangling your sound samples.
.PP
The input and output files may be standard input and output.
This is specified by '-'.
The 
.B -t\ \fItype
option must be given in this case,
else 
.I sox 
will not know the format of the given file.
The
.B -t,
.B -r,
.B -s/-u/-U/-A,
.B -b/-w/-l/-f/-d/-D
and
.B -x
options refer to the input data when given before the
input file name.  After, they refer to the output data.
.PP
If you don't give an output file name,
.I sox
will just read the input file.
This is useful for validating structured file formats;
the 
.B stat 
effect may also be used
via the 
.B -e
option.
.SH FILE TYPES
.I Sox
needs to know the formats of the input and output files.
File formats which have headers are checked, 
if that header doesn't seem right,
the program exits with an appropriate message.
Currently, raw (no header) binary and textual data, 
Amiga 8SVX, Apple/SGI AIFF, SPARC .AU (w/header), NeXT .SND,
CD-R, CVSD, GSM 06.10, Mac HCOM, Sound Tools MAUD, OSS device drivers,
Turtle Beach .SMP, Sound Blaster, Sndtool, and Sounder,
Sun Audio device driver,
Yamaha TX-16W Sampler, IRCAM Sound Files,  Creative Labs VOC,
Psion .WVE, and Microsoft RIFF/WAV are supported.
.PP
.TP 10
.B .8svx
Amiga 8SVX musical instrument description format.
.TP 10
.B .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
and can have multiple audio and picture chunks.
You may need a separate archiver to work with them.
.TP 10
.B .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 
.B .ul
format (see below).
.TP 10
.B .cdr
CD-R
.br
CD-R files are used in mastering music Compact Disks.
The file format is, as you might expect, raw stereo
raw unsigned samples at 44khz.  But, there's
some blocking/padding oddity in the format, so it
needs its own handler.
.TP 10
.B .cvs
Continuously Variable Slope Delta modulation
.br
Used to compress speech audio for applications such as voice mail.
.TP 10
.B .dat      
Text Data files
.br
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 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.
.TP 10
.B .gsm
GSM 06.10 Lossy Speech Compression
.br
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
.B sox
is optional and requires access to an external GSM library.  To see
if there is support for gsm run
.I sox -h
and look for it under the list of supported file formats.
.TP 10
.B .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.
.TP 10
.B .maud
An Amiga format
.br
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.
.TP 10
.B ossdsp
OSS /dev/dsp device driver
.br
This is a psuedo-file type and can be optionally compiled into Sox.  Run
.B 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
.B 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:
.I -t ossdsp -w -s /dev/dsp
.TP 10
.B .sf
IRCAM Sound Files.
.br
SoundFiles are used by academic music software 
such as the CSound package, and the MixView sound sample editor.
.TP 10
.B .smp
Turtle Beach SampleVision files.
.br
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.
.TP 10
.B sunau
Sun /dev/audio device driver
.br
This is a psuedo-file type and can be optionally compiled into Sox.  Run
.B 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
.B 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:
.I -t sunau -w -s /dev/audio
or
.I -t sunau -U -c 1 /dev/audio
for older sun equipment.
.TP 10
.B .txw
Yamaha TX-16W sampler.
.br
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.
.TP 10
.B .vms
More info to come.
.br
Used to compress speech audio for applications such as voice mail.
.TP 10
.B .voc
Sound Blaster VOC files.
.br
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.
.TP 10
.B .wav
Microsoft .WAV RIFF files.
.br
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 \fB.wav\fR 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 format
options will cause a format conversion, and the \fB.wav\fR
will written appropriately.  Note that it is possible to
write data of a type that cannot be specified by
the \fB.wav\fR header, and you will be warned that
you a writing a bad file !
Sox currently can read PCM, ULAW, ALAW, MS ADPCM, and IMA (or DVI) ADPCM.
It can output all of these formats except the ADPCM styles.
.TP 10
.B .wve
Psion 8-bit alaw
.br
These are 8-bit a-law 8khz sound files used on the
Psion palmtop portable computer.
.TP 10
.B .raw
Raw files (no header).
.br
The sample rate, size (byte, word, etc), 
and style (signed, unsigned, etc.)
of the sample file must be given.
The number of channels defaults to 1.
.TP 10
.B ".ub, .sb, .uw, .sw, .ul"
These are several suffices which serve as
a shorthand for raw files with a given size and style.
Thus, \fBub, sb, uw, sw,\fR and \fBul\fR
correspond to "unsigned byte", "signed byte",
"unsigned word", "signed word", and "ulaw" (byte).
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.
.TP 10
.B .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.
.SH EFFECTS
Only one effect from the palette may be applied to a sound sample.
To do multiple effects you'll need to run 
.I sox 
in a pipeline.
.TP 10
avg [ \fI-l\fR | \fI-r\fR ]
Reduce the number of channels by averaging the samples,
or duplicate channels to increase the number of channels.
Valid combinations are 1 - 2, 1 - 4, 2 - 4, 4 - 2, 4 - 1,
2 - 1. The \fI-l\fR or \fI-r\fR option is not really averaging but
either duplicates or leaves just the left or right channel, depending
on if your increasing or decreasing the number of output channels.
.TP 10
band \fB[ \fI-n \fB] \fIcenter \fB[ \fIwidth\fB ]
Apply a band-pass filter.
The frequency response drops logarithmically
around the
.I center
frequency.
The
.I width
gives the slope of the drop.
The frequencies at 
.I "center + width"
and
.I "center - width"
will be half of their original amplitudes.
.B Band
defaults to a mode oriented to pitched signals,
i.e. voice, singing, or instrumental music.
The 
.I -n
(for noise) option uses the alternate mode
for un-pitched signals.
.B Band
introduces noise in the shape of the filter,
i.e. peaking at the 
.I center
frequency and settling around it.
.TP
chorus \fIgain-in gain-out delay decay speed deptch 
.TP 10
       -s \fR| \fI-t [ \fIdelay decay speed depth -s \fR| \fI-t ... \fR]
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.
.TP 10
copy
Copy the input file to the output file.
This is the default effect if both files have the same 
sampling rate.
.TP 10
cut \fIloopnumber
Extract loop #N from a sample.
.TP 10
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.
.TP 10
echo \fIgain-in gain-out delay decay \fR[ \fIdelay decay ... \fR]
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.
.TP 10
echos \fIgain-in gain-out delay decay \fR[ \fIdelay decay ... \fR]
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.
.TP 10
flanger \fIgain-in gain-out delay decay speed -s \fR| \fI-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.
.TP 10
highp \fIcenter
Apply a high-pass filter.
The frequency response drops logarithmically with 
.I center
frequency in the middle of the drop.
The slope of the filter is quite gentle.
.TP 10
lowp \fIcenter
Apply a low-pass filter.
The frequency response drops logarithmically with 
.I center
frequency in the middle of the drop.
The slope of the filter is quite gentle.
.TP 10
map 
Display a list of loops in a sample,
and miscellaneous loop info.
.TP 10
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.
.TP 10
phaser \fIgain-in gain-out delay decay speed -s \fR| \fI-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.
.TP 10
pick
Select the left or right channel of a stereo sample,
or one of four channels in a quadrophonic sample.
.TP
polyphase [ \fI-w \fR< \fInum\fR / \fIham\fR > ] 
.TP
          [ \fI -width \fR< \fI long \fR / \fIshort \fR / \fI# \fR> ] 
.TP 10
          [ \fI-cutoff # \fR ]
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 then 
.B rate.
.br
-w < nut / ham > : select either a Nuttal (~90 dB stopband) or Hamming
(~43 dB stopband) window.
.B Warning:
Nuttall windows require 2x length than Hamming windows.  Default is
.I nut.
.br
-width long / short / # : specify the width of the filter.
.I long
is 1024 samples;
.I short
is 128 samples.  Alternatively, an exact number can be used.  Default is
.I long.
.br
-cutoff # : specify the filter cutoff frequency in terms of fraction of
bandwidth.  If upsampling, then this is the fraction of the orignal signal
that should go through.  If downsampling, this is the fraction of the
signal left after downsampling.  Default is 0.95.  Remember that
this is a float.

.TP 10
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
.B resample
or
.B polyphase.
If you are wondering which of
.B Sox's
rate changing effects to ues, you will want to read a
detailed analysis of all of them at http://eakaw2.et.tu-dresden.de/~andreas/resample/resample.html
.TP 10
resample [ \fIrolloff\fR [ \fIbeta\fR ] ]
Translate input sampling rate to output sampling rate
via simulated analog filtration.
This method is slower than 
.B rate,
but gives much
better results.  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 therefor should
be something between 0. and 1., in practice 0.8-0.95.  beta trades stop band
rejection against transition width from passband to stop band.  Larger
beta means a slower transition and greater stopband rejection.  beta
should be at least greater than 2.  The default is rollof 0.8, beta 17.5,
which is rather conservative with respect to aliasing.  Lower beta
and higher rolloff values preserve more high frequency signal energy,
but introduce measurable artifacts.
This is the default effect 
if the two files have different sampling rates.
.TP 10
reverb \fIgain-out delay \fR[ \fIdelay ... \fR]
Add reverbation to a sound sample.  Each delay is given 
in milliseconds and its feedback is depending on the
reverb-time in milliseconds.  Each delay should be in 
the range of half to quarter of reverb-time to get
a realistic reverbation.  Gain-out is the volume of the
output.
.TP 10
reverse 
Reverse the sound sample completely.
Included for finding Satanic subliminals.
.TP 10
split
Turn a mono sample into a stereo sample by copying
the input channel to the left and right channels.
.TP 10
stat [ debug | -v ]
Do a statistical check on the input file,
and print results on the standard error file.
.B stat
may copy the file untouched from input to output,
if you select an output file.  
The "Volume Adjustment:" field in the statistics
gives you the argument to the
.B -v
.I number
which will make the sample as loud as possible without clipping. 
There is an optional parameter
.B -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
.B 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.
.TP 10
swap [ \fI1 2 3 4\fB ]
Swap channels in multi-channel sound files.  In files with more than 2
channels you may specify the order that the channels should be rearranged
in.
.TP 10
vibro \fIspeed \fB [ \fIdepth\fB ]
Add the world-famous Fender Vibro-Champ sound
effect to a sound sample by using
a sine wave as the volume knob.
.B Speed 
gives the Hertz value of the wave.
This must be under 30.
.B Depth
gives the amount the volume is cut into
by the sine wave,
ranging 0.0 to 1.0 and defaulting to 0.5.
.P
.I Sox
enforces certain effects.
If the two files have different sampling
rates, the requested effect must be one of
.B copy,
or
.B rate,
." or
." .B resample.
If the two files have different numbers of channels,
the 
.B avg
." or other channel mixing
effect must be requested.
.SH BUGS
The syntax is horrific.
It's very tempting to include a default system that allows
an effect name as the program name
and just pipes a sound sample from standard input 
to standard output, but the problem of inputting the
sample rates makes this unworkable.
.P
Please report any bugs found in this version of sox to Chris Bagwell (cbagwell@sprynet.com)
.SH FILES
.SH SEE ALSO
.BR play (1) ,
.BR rec (1)
.SH NOTICES
The echoplex effect is:
Copyright (C) 1989 by Jef Poskanzer.

Permission to use, copy, modify, and distribute this software and its
documentation for any purpose and without fee is hereby granted, provided
that the above copyright notice appear in all copies and that both that
copyright notice and this permission notice appear in supporting
documentation.  This software is provided "as is" without express or
implied warranty.

The version of Sox that accompanies this manual page is support by 
Chris Bagwell (cbagwell@sprynet.com).  Please refer any questions 
regarding it to this address.  You may obtain the latest version at the 
the web site http://home.sprynet.com/~cbagwell/sox.html