


SoX(1)                             SoX(1)


NAME
       sox - Sound eXchange : universal sound sample translator

SYNOPSIS
       sox infile outfile
       sox infile outfile [ effect [ effect options ... ] ]
       sox infile -e effect [ effect options ... ]
       sox  [ general options  ] [ format options  ] ifile [ for-
       mat options  ] ofile [ effect [ effect options ... ] ]

       General options: [ -e ] [ -h ] [ -p ] [ -v volume ] [ -V ]

       Format	options:   [   -t  filetype  ]	[  -r  rate  ]	[
       -s/-u/-U/-A/-a/-g ] [ -b/-w/-l/-f/-d/-D ] [ -c channels	]
       [ -x ]

       Effects:
	    avg [ -l | -r ]
	    band [ -n ] center [ width ]
	    check
	    chorus  gain-in  gain  out	delay  decay  speed depth
		 -s | -t [ delay decay speed depth -s | -fI-t ]
	    copy
	    cut
	    deemph
	    echo gain-in gain-out delay decay [ delay decay  ...]
	    echos gain-in gain-out delay decay [ delay decay ...]
	    flanger gain-in gain-out delay decay speed -s | -fI-t
	    highp center
	    lowp center
	    map
	    mask
	    phaser gain-in gain-out delay decay speed -s | -t
	    pick
	    polyphase [ -w < num / ham > ]
		      [	 -width <  long	 / short  / # > ]
		      [ -cutoff #  ]
	    rate
	    resample
	    reverb gain-out reverb-time delay [ delay ... ]
	    reverse
	    split
	    stat [ debug | -v ]
	    swap [ 1 2 3 4 ]
	    vibro speed [ depth ]

DESCRIPTION
       Sox  translates	sound  files  from one format to another,
       possibly doing a sound effect.

OPTIONS
       The option syntax is a little grotty, but in essence:
	    sox file.au file.voc
       translates a sound sample in SUN Sparc .AU format  into	a



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SoX(1)							   SoX(1)


       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.

       File type options:

       -t filetype
		 gives the type of the sound sample file.

       -r rate	 Give sample rate in Hertz of file.

       -s/-u/-U/-A/-a/-g
		 The  sample  data  is signed linear (2's comple-
		 ment), unsigned linear, U-law (logarithmic),  A-
		 law  (logarithmic), ADPCM, or GSM.  U-law and A-
		 law are the U.S. and international standards for
		 logarithmic  telephone sound compression.  ADPCM
		 is form of sound compression  that  has  a  good
		 compromise  between  good sound quality and fast
		 encoding/decoding time.  GSM is a standard  used
		 for  telephone	 sound	compression  in	 European
		 countries and its gaining popularity because  of
		 its quality.

       -b/-w/-l/-f/-d/-D
		 The  sample  data  is	in  bytes,  16-bit words,
		 32-bit longwords, 32-bit floats,  64-bit  double
		 floats,  or 80-bit IEEE floats.  Floats and dou-
		 ble floats are in native machine format.

       -x	 The sample data is in XINU format; that  is,  it
		 comes	from  a	 machine  with	the opposite word
		 order than yours and must be  swapped	according
		 to  the  word-size given above.  Only 16-bit and
		 32-bit integer data may  be  swapped.	 Machine-
		 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.

       General options:

       -e	 after the input file allows you to avoid  giving
		 an output file and just name an effect.  This is
		 mainly useful with the stat effect  but  can  be
		 used with others.

       -h	 Print version number and usage information.




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SoX(1)							   SoX(1)


       -p	 Run  in  preview  mode	 and run fast.	This will
		 somewhat speed up sox when the output format has
		 a  different  number of channels and a different
		 rate then the input file.  The	 order	that  the
		 effects  are run in will be arranged for maximum
		 speed and not quality.

       -v volume Change amplitude (floating point); less than 1.0
		 decreases, greater than 1.0 increases.	 Note: we
		 perceive volume logarithmically,  not	linearly.
		 Note: see the stat effect.

       -V	 Print	a description of processing phases.  Use-
		 ful for figuring out exactly how sox is mangling
		 your sound samples.

       The  input and output files may be standard input and out-
       put.  This is specified by '-'.	The -t type  option  must
       be  given  in this case, else sox will not know the format
       of   the	  given	  file.	   The	 -t,   -r,   -s/-u/-U/-A,
       -b/-w/-l/-f/-d/-D  and  -x options refer to the input data
       when given before the input file name.  After, they  refer
       to the output data.

       If  you don't give an output file name, sox will just read
       the input file.	This is useful for validating  structured
       file  formats; the stat effect may also be used via the -e
       option.

FILE TYPES
       Sox needs to know the formats  of  the  input  and  output
       files.	File  formats  which have headers are checked, if
       that header doesn't seem right, the program exits with  an
       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,   Cre-
       ative  Labs  VOC,  Psion	 .WVE, and Microsoft RIFF/WAV are
       supported.


       .8svx	 Amiga 8SVX musical instrument	description  for-
		 mat.

       .aiff	 AIFF  files  used  on	Apple  IIc/IIgs	 and SGI.
		 Note: the AIFF format	supports  only	one  SSND
		 chunk.	  It  does  not	 support  multiple  sound
		 chunks, or the 8SVX musical instrument	 descrip-
		 tion format.  AIFF files are multimedia archives
		 and and can  have  multiple  audio  and  picture
		 chunks.   You	may  need  a separate archiver to
		 work with them.



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SoX(1)							   SoX(1)


       .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).

       .cdr	 CD-R
		 CD-R files are used in mastering  music  Compact
		 Disks.	 The file format is, as you might expect,
		 raw stereo raw unsigned samples at 44khz.   But,
		 there's some blocking/padding oddity in the for-
		 mat, so it needs its own handler.

       .cvs	 Continuously Variable Slope Delta modulation
		 Used to compress speech audio	for  applications
		 such as voice mail.

       .dat	 Text Data files
		 These	files contain a textual representation of
		 the sample data.   There  is  one  line  at  the
		 beginning that contains the sample rate.  Subse-
		 quent lines contain two numeric data items:  the
		 time  since  the beginning of the sample and the
		 sample value.	Values are normalized so that the
		 maximum  and  minimum	are 1.00 and -1.00.  This
		 file format can be used to create data files for
		 external programs such as FFT analyzers or graph
		 routines.  SoX can also convert a file	 in  this
		 format	 back into one of the other file formats.

       .gsm	 GSM 06.10 Lossy Speech Compression
		 A standard for compressing speech which is  used
		 in  the Global Standard for Mobil telecommunica-
		 tions (GSM).  Its good for its purpose,  shrink-
		 ing  audio data size, but it will introduce lots
		 of noise when a given sound  sample  is  encoded
		 and decoded multiple times.  This format is used
		 by some voice mail applications.  It  is  rather
		 CPU  intensive.   GSM	in  sox	 is  optional and
		 requires access to an external GSM library.   To
		 see  if  there is support for gsm run sox -h and
		 look for it under the	list  of  supported  file
		 formats.

       .hcom	 Macintosh  HCOM  files.   These are (apparently)
		 Mac FSSD files with some variant of Huffman com-
		 pression.   The Macintosh has wacky file formats
		 and this format handler apparently doesn't  han-
		 dle all the ones it should.  Mac users will need
		 your usual arsenal of file  converters	 to  deal



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SoX(1)							   SoX(1)


		 with an HCOM file under Unix or DOS.

       .maud	 An Amiga format
		 An IFF-conform sound file type, registered by MS
		 MacroSystem Computer GmbH, published along  with
		 the  "Toccata"	 sound-card on the Amiga.  Allows
		 8bit linear, 16bit linear, A-Law, u-law in  mono
		 and stereo.

       ossdsp	 OSS /dev/dsp device driver
		 This is a psuedo-file type and can be optionally
		 compiled into Sox.  Run sox -h	 to  see  if  you
		 have  support	for  this  file	 type.	When this
		 driver is used it allows you to open up the  OSS
		 /dev/dsp  file	 and configure it to use the same
		 data type as passed in to  Sox.   It  works  for
		 both  playing and recording sound samples.  When
		 playing sound files it attempts to  set  up  the
		 OSS  driver  to use the same format as the input
		 file.	It is suggested to  always  override  the
		 output values to use the highest quality samples
		 your sound card can handle.  Example: -t  ossdsp
		 -w -s /dev/dsp

       .sf	 IRCAM Sound Files.
		 SoundFiles  are  used by academic music software
		 such as the  CSound  package,	and  the  MixView
		 sound sample editor.

       .smp	 Turtle Beach SampleVision files.
		 SMP  files  are  for use with the PC-DOS package
		 SampleVision by  Turtle  Beach	 Softworks.  This
		 package  is  for  communication  to several MIDI
		 samplers. All sample rates are supported by  the
		 package,  although  not all are supported by the
		 samplers themselves. Currently loop  points  are
		 ignored.

       sunau	 Sun /dev/audio device driver
		 This is a psuedo-file type and can be optionally
		 compiled into Sox.  Run sox -h	 to  see  if  you
		 have  support	for  this  file	 type.	When this
		 driver is used it allows you to open  up  a  Sun
		 /dev/audio file and configure it to use the same
		 data type as passed in to  Sox.   It  works  for
		 both  playing and recording sound samples.  When
		 playing sound files it attempts to  set  up  the
		 audio driver to use the same format as the input
		 file.	It is suggested to  always  override  the
		 output values to use the highest quality samples
		 your hardware can handle.  Example: -t sunau  -w
		 -s /dev/audio or -t sunau -U -c 1 /dev/audio for
		 older sun equipment.




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SoX(1)							   SoX(1)


       .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
		 effect.  You had better know what you are doing!
		 Output	 format	 options will cause a format con-
		 version, and the  .wav	 will  written	appropri-
		 ately.	  Note	that it is possible to write data
		 of a type that cannot be specified by	the  .wav
		 header,  and you will be warned that you a writ-
		 ing a bad file !  Sox currently  can  read  PCM,
		 ULAW,	ALAW,  MS  ADPCM, and IMA (or DVI) ADPCM.
		 It can output all of these  formats  except  the
		 ADPCM styles.

       .wve	 Psion 8-bit alaw
		 These	are  8-bit a-law 8khz sound files used on
		 the Psion palmtop portable computer.

       .raw	 Raw files (no header).
		 The sample rate, size	(byte,	word,  etc),  and
		 style	(signed,  unsigned,  etc.)  of the sample
		 file must be  given.	The  number  of	 channels
		 defaults to 1.



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SoX(1)							   SoX(1)


       .ub, .sb, .uw, .sw, .ul
		 These	are  several  suffices	which  serve as a
		 shorthand for raw files with a	 given	size  and
		 style.	  Thus, ub, sb, uw, sw, and ul correspond
		 to "unsigned  byte",  "signed	byte",	"unsigned
		 word",	 "signed  word",  and "ulaw" (byte).  The
		 sample rate defaults to 8000 hz if  not  explic-
		 itly set, and the number of channels (as always)
		 defaults to 1.	 There are lots of Sparc  samples
		 floating  around  in u-law format with no header
		 and fixed at a sample rate of 8000 hz.	 (Certain
		 sound management software cheerfully ignores the
		 headers.)  Similarly, most Mac sound  files  are
		 in  unsigned  byte  format with a sample rate of
		 11025 or 22050 hz.

       .auto	 This is a ``meta-type'':  specifying  this  type
		 for  an input file triggers some code that tries
		 to guess the real  type  by  looking  for  magic
		 words	in  the	 header.   If  the  type can't be
		 guessed, the program exits with  an  error  mes-
		 sage.	 The  input  must  be a plain file, not a
		 pipe.	This type can't be used for output files.

EFFECTS
       Only one effect from the palette may be applied to a sound
       sample.	To do multiple effects you'll need to run sox  in
       a pipeline.

       avg [ -l | -r ]
		 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 -l  or
		 -r  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.

       band [ -n ] center [ width ]
		 Apply	a  band-pass   filter.	  The	frequency
		 response drops logarithmically around the center
		 frequency.  The width gives  the  slope  of  the
		 drop.	 The  frequencies  at  center + width and
		 center - width will be half  of  their	 original
		 amplitudes.  Band defaults to a mode oriented to
		 pitched signals, i.e. voice, singing, or instru-
		 mental	 music.	  The  -n (for noise) option uses
		 the alternate mode for un-pitched signals.  Band
		 introduces  noise  in	the  shape of the filter,
		 i.e. peaking at the center  frequency	and  set-
		 tling around it.





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SoX(1)							   SoX(1)


       chorus gain-in gain-out delay decay speed deptch

	      -s | -t [ delay decay speed depth -s | -t ... ]
		 Add  a chorus to a sound sample.  Each quadtuple
		 delay/decay/speed/depth gives the delay in  mil-
		 liseconds  and	 the  decay (relative to gain-in)
		 with a modulation speed in  Hz	 using	depth  in
		 milliseconds.	The modulation is either sinodial
		 (-s) or triangular (-t).  Gain-out is the volume
		 of the output.

       copy	 Copy the input file to the output file.  This is
		 the default effect if both files have	the  same
		 sampling rate.

       cut loopnumber
		 Extract loop #N from a sample.

       deemph	 Apply	a  treble  attenuation shelving filter to
		 samples  in  audio  cd	 format.   The	frequency
		 response  of pre-emphasized recordings is recti-
		 fied.	The filtering is defined in the	 standard
		 document ISO 908.

       echo gain-in gain-out delay decay [ delay decay ... ]
		 Add echoing to a sound sample.	 Each delay/decay
		 part gives the delay  in  milliseconds	 and  the
		 decay (relative to gain-in) of that echo.  Gain-
		 out is the volume of the output.

       echos gain-in gain-out delay decay [ delay decay ... ]
		 Add a sequence of echos to a sound sample.  Each
		 delay/decay part gives the delay in milliseconds
		 and the decay	(relative  to  gain-in)	 of  that
		 echo.	Gain-out is the volume of the output.

       flanger gain-in gain-out delay decay speed -s | -t
		 Add  a	 flanger  to a sound sample.  Each triple
		 delay/decay/speed gives the delay  in	millisec-
		 onds  and the decay (relative to gain-in) with a
		 modulation  speed  in	Hz.   The  modulation  is
		 either	 sinodial (-s) or triangular (-t).  Gain-
		 out is the volume of the output.

       highp center
		 Apply	a  high-pass   filter.	  The	frequency
		 response  drops logarithmically with center fre-
		 quency in the middle of the drop.  The slope  of
		 the filter is quite gentle.

       lowp center
		 Apply a low-pass filter.  The frequency response
		 drops logarithmically with center  frequency  in
		 the middle of the drop.  The slope of the filter



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SoX(1)							   SoX(1)


		 is quite gentle.

       map	 Display a list of loops in a sample, and miscel-
		 laneous loop info.

       mask	 Add  "masking	noise"	to  signal.   This effect
		 deliberately adds white  noise	 to  a	sound  in
		 order	to  mask quantization effects, created by
		 the process of playing a  sound  digitally.   It
		 tends	to  mask buzzing voices, for example.  It
		 adds 1/2 bit of noise to the sound file  at  the
		 output bit depth.

       phaser gain-in gain-out delay decay speed -s | -t
		 Add  a	 phaser	 to  a sound sample.  Each triple
		 delay/decay/speed gives the delay  in	millisec-
		 onds  and the decay (relative to gain-in) with a
		 modulation  speed  in	Hz.   The  modulation  is
		 either	 sinodial  (-s)	 or triangular (-t).  The
		 decay should be less than 0.5 to avoid feedback.
		 Gain-out is the volume of the output.

       pick	 Select	 the  left  or	right channel of a stereo
		 sample, or one of four	 channels  in  a  quadro-
		 phonic sample.

       polyphase [ -w < num / ham > ]

		 [  -width <  long  / short  / # > ]

		 [ -cutoff #  ]
		 Translate input sampling rate to output sampling
		 rate via polyphase interpolation,  a  DSP  algo-
		 rithm.	  This	method	is  slow and uses lots of
		 RAM, but gives much better results then rate.
		 -w < nut / ham > : select either a  Nuttal  (~90
		 dB  stopband)	or Hamming (~43 dB stopband) win-
		 dow.  Warning: Nuttall windows require 2x length
		 than Hamming windows.	Default is nut.
		 -width	 long  / short / # : specify the width of
		 the filter.  long is 1024 samples; short is  128
		 samples.   Alternatively, an exact number can be
		 used.	Default is long.
		 -cutoff # : specify the filter cutoff	frequency
		 in  terms  of	fraction of bandwidth.	If upsam-
		 pling, then this is the fraction of the  orignal
		 signal that should go through.	 If downsampling,
		 this is the fraction of the  signal  left  after
		 downsampling.	 Default  is 0.95.  Remember that
		 this is a float.


       rate	 Translate input sampling rate to output sampling
		 rate  via  linear  interpolation  to  the  Least



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SoX(1)							   SoX(1)


		 Common Multiple of the two sampling rates.  This
		 is the default effect if the two files have dif-
		 ferent 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-translat-
		 ing by a multiple.  Lerp-ing is  acceptable  for
		 cheap	8-bit  sound hardware, but for CD-quality
		 sound you should instead use either resample  or
		 polyphase.   If you are wondering which of Sox's
		 rate changing effects to ues, you will	 want  to
		 read  a  detailed  analysis  of  all  of them at
		 http://eakaw2.et.tu-dresden.de/~andreas/resam-
		 ple/resample.html

       resample [ rolloff [ beta ] ]
		 Translate input sampling rate to output sampling
		 rate  via  simulated  analog  filtration.   This
		 method	 is slower than rate, but gives much bet-
		 ter results.  rolloff refers to the cut-off fre-
		 quency	 of  the  low pass filter and is given in
		 terms of the Nyquist  frequency  for  the  lower
		 sample	 rate.	 rolloff therefor should be some-
		 thing between 0. and 1., in  practice	0.8-0.95.
		 beta  trades stop band rejection against transi-
		 tion width from passband to stop  band.   Larger
		 beta means a slower transition and greater stop-
		 band rejection.  beta should be at least greater
		 than  2.   The default is rollof 0.8, beta 17.5,
		 which is rather  conservative	with  respect  to
		 aliasing.   Lower beta and higher rolloff values
		 preserve more high frequency signal energy,  but
		 introduce  measurable	artifacts.   This  is the
		 default effect if the two files  have	different
		 sampling rates.

       reverb gain-out delay [ delay ... ]
		 Add  reverbation  to a sound sample.  Each delay
		 is given in milliseconds  and	its  feedback  is
		 depending  on	the  reverb-time in milliseconds.
		 Each delay should be in the  range  of	 half  to
		 quarter of reverb-time to get a realistic rever-
		 bation.  Gain-out is the volume of the output.

       reverse	 Reverse the sound sample  completely.	 Included
		 for finding Satanic subliminals.

       split	 Turn a mono sample into a stereo sample by copy-
		 ing the input channel	to  the	 left  and  right
		 channels.

       stat [ debug | -v ]
		 Do  a	statistical  check on the input file, and
		 print results on the standard error file.   stat



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SoX(1)							   SoX(1)


		 may  copy  the file untouched from input to out-
		 put, if you select an output file.  The  "Volume
		 Adjustment:"  field  in the statistics gives you
		 the argument to the -v number	which  will  make
		 the sample as loud as possible without clipping.
		 There is an  optional	parameter  -v  that  will
		 print out the "Volume Adjustment:" field's value
		 and return.  This could be of use in scripts  to
		 auto  convert	the  volume.  There is an also an
		 optional parameter debug  that	 will  place  sox
		 into  debug mode and print out a hex dump of the
		 sound file from the internal buffer that  is  in
		 32-bit	 signed PCM data.  This is mainly only of
		 use in tracking down endian problems that  creep
		 in to sox on cross-platform versions.

       swap [ 1 2 3 4 ]
		 Swap  channels in multi-channel sound files.  In
		 files with more than 2 channels you may  specify
		 the order that the channels should be rearranged
		 in.

       vibro speed  [ depth ]
		 Add the world-famous  Fender  Vibro-Champ  sound
		 effect to a sound sample by using a sine wave as
		 the volume knob.  Speed gives the Hertz value of
		 the  wave.   This must be under 30.  Depth gives
		 the amount the volume is cut into  by	the  sine
		 wave,	ranging 0.0 to 1.0 and defaulting to 0.5.

       Sox enforces certain effects.  If the two files have  dif-
       ferent sampling rates, the requested effect must be one of
       copy, or rate, If the two files have different numbers  of
       channels, the avg effect must be requested.

BUGS
       The  syntax  is horrific.  It's very tempting to include a
       default system that allows an effect name as  the  program
       name  and just pipes a sound sample from standard input to
       standard output, but the problem of inputting  the  sample
       rates makes this unworkable.

       Please  report  any  bugs  found in this version of sox to
       Chris Bagwell (cbagwell@sprynet.com)

FILES
SEE ALSO
       play(1), rec(1)

NOTICES
       The  echoplex  effect  is:  Copyright  (C)  1989	 by   Jef
       Poskanzer.

       Permission  to  use,  copy,  modify,  and  distribute this



			  June 28, 1999			       11





SoX(1)							   SoX(1)


       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













































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