SoX(1)							   SoX(1)



NAME
       sox - Sound eXchange : universal sound sample translator

SYNOPSIS
       sox infile outfile

       sox [ general options ] [ format options ] infile
	   [ format options ] outfile
	   [ effect [ effect options ] ... ]

       soxmix infile1 infile2 outfile

       soxmix [ general options ] [ format options ] infile1
	   [ format options ] infile2
	   [ format options ] outfile
	   [ effect [ effect options ] ... ]


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

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

       Effects:
	   avg [ -l | -r | -f | -b | n,n,...,n ]
	   band [ -n ] center [ width ]
	   bandpass frequency bandwidth
	   bandreject frequency bandwidth
	   chorus gain-in gain out delay decay speed depth
		  -s | -t [ delay decay speed depth -s | -t ]
	   compand attack1,decay1[,attack2,decay2...]
		   in-dB1,out-dB1[,in-dB2,out-dB2...]
		   [ gain [ initial-volume [ delay ] ] ]
	   copy
	   dcshift shift [ limitergain ]
	   deemph
	   earwax
	   echo gain-in gain-out delay decay [ delay decay ... ]
	   echos gain-in gain-out delay decay [ delay decay ... ]
	   fade [ type ] fade-in-length
		[ stop-time [ fade-out-length ] ]
	   filter [ low ]-[ high ] [ window-len [ beta ]]
	   flanger gain-in gain-out delay decay speed < -s | -t >
	   highp frequency
	   highpass frequency
	   lowp frequency
	   lowpass frequency
	   map
	   mask
	   pan direction
	   phaser gain-in gain-out delay decay speed < -s | -t >
	   pick [ -1 | -2 | -3 | -4 | -l | -r ]
	   pitch shift [ width interpole fade ]
	   polyphase [ -w < nut / ham > ]
		     [	-width < long / short / # > ]
		     [ -cutoff # ]
	   rate
	   resample [ -qs | -q | -ql ] [ rolloff [ beta ] ]
	   reverb gain-out reverb-time delay [ delay ... ]
	   reverse
	   silence above_periods [ duration threshold[ d | % ]
		   [ below_periods duration
		     threshold[ d | % ]]
	   speed [ -c ] factor
	   split
	   stat [ -s n ] [ -rms ] [ -v ] [ -d ]
	   stretch [ factor [ window fade shift fading ]
	   swap [ 1 2 | 1 2 3 4 ]
	   synth [ length ] type mix [ freq [ -freq2 ]
		 [ off ] [ ph ] [ p1 ] [ p2 ] [ p3 ]
	   trim start [ length ]
	   vibro speed [ depth ]
	   vol gain [ type [ limitergain ] ]

DESCRIPTION
       SoX is a command line program that can convert most  popu­
       lar  audio files to most other popular audio file formats.
       It can optionally change the audio sample  data	type  and
       apply  one  or  more sound effects to the file during this
       translation.

       soxmix is functionally the same as the command  line  pro­
       gram sox expect that it takes two files as input and mixes
       the audio together to produce a single file as output.  It
       has  a  restriction  that  both input files must be of the
       same data type and sample rates.

       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 header-less data, or sometimes called
       raw data.  A user must pass enough information to  SoX  on
       the  command  line  so  that it knows what type of data it
       contains.

       Audio data can usually be totally described by four  char­
       acteristics:

       rate	 The  sample  rate is in samples per second.  For
		 example, CD sample rates are at 44100.

       data size The precision the data is stored in.  Most popu­
		 lar are 8-bit bytes or 16-bit words.

       data encoding
		 What  encoding the data type uses.  Examples are
		 u-law, ADPCM, or signed linear data.

       channels	 How many channels are	contained  in  the  audio
		 data.	 Mono and Stereo are the two most common.

       Please refer to the soxexam(1)  manual  page  for  a  long
       description  with  examples on how to use SoX with various
       types of file formats.

OPTIONS
       The option syntax is a little grotty, but in essence:

	    sox File.au file.wav

       translates a sound file in SUN Sparc  .AU  format  into	a
       Microsoft .WAV file, while

	    sox -v 0.5 file.au -r 12000 file.wav mask

       does  the  same	format	translation  but  also lowers the
       amplitude by 1/2,  changes  the	sampling  rate	to  12000
       hertz,  and  applies  the  mask	sound effect to the audio
       data.

       The following will mix two sound files together to to pro­
       duce a single sound file.

	       soxmix music.wav voice.wav mixed.wav

       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.  It
       is also how SoX is informed about the format of raw  input
       data.

       -t filetype
		 gives the type of the sound sample file.  Useful
		 when file extension is not standard or for spec­
		 ifying the .auto file type.

       -r rate	 Gives	the sample rate in Hertz of the file.  To
		 cause the output file to have a different sample
		 rate than the input file, include this option as
		 a part of the output options.
		 If the input and  output  files  have	different
		 rates	then  a sample rate change effect must be
		 ran.  If a sample rate changing  effect  is  not
		 specified  then a default one will internally be
		 ran by SoX using its default parameters.

       -s/-u/-U/-A/-a/-i/-g/-f
		 The sample data encoding is signed  linear  (2's
		 complement),  unsigned	 linear, u-law (logarith­
		 mic),	A-law  (logarithmic),  ADPCM,  IMA_ADPCM,
		 GSM, or Floating-point.
		 U-law	(actually shorthand for mu-law) and A-law
		 are the U.S.  and  international  standards  for
		 logarithmic  telephone	 sound compression.  When
		 uncompressed u-law has roughly the precision  of
		 14-byte PCM audio and A-law has roughly the pre­
		 cision of 13-bit PCM audio.
		 A-law and u-law data is sometimes encoded  using
		 a  reversed  bit-ordering (ie. MSB becomes LSB).
		 Internally, SoX understands  how  to  work  with
		 this  encoding but there is currently no command
		 line option to specify it.   If  you  need  this
		 support  then	you can use the psuedo file types
		 of ".la" and ".lu" to inform sox of  the  encod­
		 ing.  See supported file types for more informa­
		 tion.
		 ADPCM is a form of sound compression that has	a
		 good  compromise  between good sound quality and
		 fast encoding/decoding time.	It  is	used  for
		 telephone sound compression and places were full
		 fidelity is not as important.	When uncompressed
		 it  has  roughly  the	precision  of  16-bit PCM
		 audio.	 Popular version of ADPCM include  G.726,
		 MS  ADPCM,  and IMA ADPCM.  The -a flag has dif­
		 ferent meanings in different file handlers.   In
		 .wav  files it represents MS ADPCM files, in all
		 others it means G.726 ADPCM.	IMA  ADPCM  is	a
		 specific  form	 of  ADPCM  compression, slightly
		 simpler  and  slightly	  lower	  fidelity   than
		 Microsoft's  flavor of ADPCM.	IMA ADPCM is also
		 called DVI ADPCM.
		 GSM is a standard used for telephone sound  com­
		 pression  in  European countries and its gaining
		 popularity because of its quality.   It  usually
		 is CPU intensive to work with GSM audio data.

       -b/-w/-l	 The  sample data size is in bytes, 16-bit words,
		 or 32-bit long words.

       -x	 The sample data is in XINU format; that  is,  it
		 comes	from  a	 machine  with	the opposite word
		 order than yours and must be  swapped	according
		 to  the  word-size given above.  Only 16-bit and
		 32-bit integer data may  be  swapped.	 Machine-
		 format floating-point data is not portable.

       -c channels
		 The  number  of sound channels in the data file.
		 This may be 1, 2, or 4;  for  mono,  stereo,  or
		 quad  sound  data.   To cause the output file to
		 have a different number  of  channels	than  the
		 input	file, include this option with the output
		 file options.	If the input and output file have
		 a  different  number  of  channels  then the avg
		 effect must be used.  If the avg effect  is  not
		 specified on the command line it will be invoked
		 internally with default parameters.

       -e	 When used after the input filename (so	 that  it
		 applies  to  the  output  file) it allows you to
		 avoid giving an output	 filename  and	will  not
		 produce an output file.  It will apply any spec­
		 ified effects to the input file.  This is mainly
		 useful with the stat effect but can be used with
		 others.

       General options:

       -h	 Print version number and usage information.

       -p	 Run in preview mode and  run  fast.   This  will
		 somewhat speed up SoX when the output format has
		 a different number of channels and  a	different
		 rate  than  the  input	 file.	 Currently,  this
		 defaults to using the rate effect instead of the
		 resample effect for sample rate changes.

       -v volume Change amplitude (floating point); less than 1.0
		 decreases, greater than 1.0 increases.	 May  use
		 a  negative  number  to  invert the phase of the
		 audio data.  It is interesting to note	 that  we
		 perceive volume logarithmically but this adjusts
		 the amplitude linearly.
		 Note: see the stat  effect  for  information  on
		 finding  the maximum value that can be used with
		 this option without causing  audio  data  be  be
		 clipped.

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

FILE TYPES
       SoX  attempts  to  determine  the file type of input files
       automatically by looking at the header of the audio  file.
       When  it	 is  unable  to detect the file type or if its an
       output file then it uses the file extension of the file to
       determine  what	type of file format handler to use.  This
       can be overridden by specifying the  "-t"  option  on  the
       command line.

       The  input  and	output files may be read from standard in
       and out.	 This is done by specifying '-' as the	filename.

       File  formats  which  have  headers  are	 checked, if that
       header doesn't seem  right,  the	 program  exits	 with  an
       appropriate message.

       The following file formats are supported:


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

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

       .au	 SUN Microsystems AU files.  There are apparently
		 many types of .au files; DEC  has  invented  its
		 own  with  a  different  magic	 number	 and word
		 order.	 The .au handler can read these files but
		 will  not write them.	Some .au files have valid
		 AU headers and some  do  not.	 The  latter  are
		 probably  original  SUN  u-law	 8000 hz samples.
		 These can be dealt with  using	 the  .ul  format
		 (see below).

       .avr	 Audio Visual Research
		 The  AVR  format is produced by a number of com­
		 mercial packages on the Mac.

       .cdr	 CD-R
		 CD-R files are used in mastering music	 on  Com­
		 pact  Disks.  The audio data on a CD-R disk is a
		 raw audio file with a format  of  stereo  16-bit
		 signed samples at a 44khz sample rate.	 There is
		 a special blocking/padding oddity at the end  of
		 the  audio file and is why it needs its own han­
		 dler.

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

       .dat	 Text Data files
		 These	files contain a textual representation of
		 the sample data.   There  is  one  line  at  the
		 beginning   that   contains   the  sample  rate.
		 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 analyz­
		 ers or graph routines.	 SoX can also  convert	a
		 file  in  this format back into one of the other
		 file formats.

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

       .hcom	 Macintosh HCOM files.	 These	are  (apparently)
		 Mac FSSD files with some variant of Huffman 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.

       .nul	 Null file handler.  This is a fake  file  hander
		 that  act as if its reading a stream of 0's from
		 a while or fake writing output to a file.   This
		 is not a very useful file handler in most cases.
		 It might be useful in some scripts were  you  do
		 not  want  to read or write from a real file but
		 would like to specify	a  filename  for  consis­
		 tency.

       .ogg	 Ogg Vorbis Compressed Audio.
		 Ogg Vorbis is a open, patent-free CODEC designed
		 for compressing music and streaming  audio.   It
		 is  similar  to  MP3,	VQF, AAC, and other lossy
		 formats.  SoX can decode all types of Ogg Vorbis
		 files,	 but can only encode at 128 kbps.  Decod­
		 ing is somewhat CPU intensive	and  encoding  is
		 very CPU intensive.
		 Ogg  Vorbis  in  SoX  is  optional  and requires
		 access to external Ogg Vorbis libraries.  To see
		 if  there  is	support for Ogg Vorbis run sox -h
		 and look for it under the list of supported file
		 formats as "vorbis".

       ossdsp	 OSS /dev/dsp device driver
		 This is a pseudo-file type and can be optionally
		 compiled into SoX.  Run sox -h	 to  see  if  you
		 have  support	for  this  file	 type.	When this
		 driver is used it allows you to open up the  OSS
		 /dev/dsp  file	 and configure it to use the same
		 data format as passed in to 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.

       .sph
		 SPHERE	 (SPeech HEader Resources) is a file for­
		 mat defined by NIST (National Institute of Stan­
		 dards	and  Technology)  and is used with speech
		 audio.	 SoX can read these files when they  con­
		 tain  u-law  and  PCM	data.  It will ignore any
		 header information that says the  data	 is  com­
		 pressed using shorten compression and will treat
		 the data as either  u-law  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 pro­
		 cessing.

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

       .snd
		 Under DOS this file format is the  same  as  the
		 .sndt	format.	  Under all other platforms it is
		 the same as the .au format.

       .sndt	 SoundTool files.
		 This is an older DOS file format.

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

       .txw	 Yamaha TX-16W sampler.
		 A  file  format  from a Yamaha sampling keyboard
		 which	wrote  IBM-PC	format	 3.5"	floppies.
		 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 gener­
		 ated appropriately.  On output, silence  is  not
		 detected,   nor  are  impossible  sample  rates.
		 Note, this  version  now  supports  playing  VOC
		 files	with multiple blocks and supports playing
		 files containing u-law and A-law samples.

       vorbis	 See .ogg format.

       .wav	 Microsoft .WAV RIFF files.
		 These appear to be very similar  to  IFF  files,
		 but  not  the	same.	They are the native sound
		 file format of Windows.  (Obviously, Windows was
		 of  such  incredible  importance to the computer
		 industry that it just had to have its own  sound
		 file format.)	Normally .wav files have all for­
		 matting information in their headers, and so  do
		 not  need  any	 format	 options specified for an
		 input file. If any are, they will  override  the
		 file  header,	and  you  will	be warned to this
		 effect.  You had better know what you are doing!
		 Output	 format	 options will cause a format con­
		 version, and the  .wav	 will  written	appropri­
		 ately.	  SoX currently can read PCM, ULAW, ALAW,
		 MS ADPCM, and IMA (or DVI) ADPCM.  It can  write
		 all of these formats including (NEW!)	the ADPCM
		 encoding.

       .wve	 Psion 8-bit A-law
		 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, .lu, .la, .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, al, lu,  la
		 and  sl  correspond  to "unsigned byte", "signed
		 byte", "unsigned word", "signed  word",  "u-law"
		 (byte),  "A-law"  (byte),  inverse bit order "u-
		 law", inverse bit  order  "A-law",  and  "signed
		 long".	  The  sample rate defaults to 8000 hz if
		 not explicitly set, and the number  of	 channels
		 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
       Multiple effects may be applied to the audio data by spec­
       ifying  them  one  after another at the end of the command
       line.

       avg [ -l | -r | -f | -b | n,n,...,n ]
		 Reduce the number of channels by  averaging  the
		 samples,  or  duplicate channels to increase the
		 number of channels.  This  effect  is	automati­
		 cally	used  when  the	 number of input channels
		 differ from the number of output channels.  When
		 reducing  the	number of channels it is possible
		 to manually specify the avg effect and	 use  the
		 -l,  -r,  -f,	or  -b options to select only the
		 left, right, front, or back channel(s)	 for  the
		 output	 instead  of averaging the channels.  The
		 -f and -b  options  maintain  left/right  stereo
		 separation; use the avg effect twice to select a
		 single channel.

		 The avg effect can also be invoked with up to 16
		 double-precision numbers, which specify the pro­
		 portion of each input	channel	 that  is  to  be
		 mixed	into each output channel.  In two-channel
		 mode, 4 numbers are given: l->l, l->r, r->l, and
		 r->r,	respectively.	In four-channel mode, the
		 first 4 numbers give  the  proportions	 for  the
		 left-front  output  channel, as follows: lf->lf,
		 rf->lf, lb->lf, and rb->rf.  The next 4 give the
		 right-front output in the same order, then left-
		 back and right-back.

		 It is also possible to use  the  16  numbers  to
		 expand or reduce the channel count; just specify
		 0 for unused channels.	 Finally, if fewer than 4
		 numbers are given, certain special abbreviations
		 may be invoked; see the source code for details.

       band [ -n ] center [ width ]
		 Apply	 a   band-pass	 filter.   The	frequency
		 response drops logarithmically around the center
		 frequency.   The  width  gives	 the slope of the
		 drop.	The frequencies at  center  +  width  and
		 center	 -  width  will be half of their original
		 amplitudes.  Band defaults to a mode oriented to
		 pitched signals, i.e. voice, singing, or instru­
		 mental music.	The -n (for  noise)  option  uses
		 the   alternate  mode	for  un-pitched	 signals.
		 Warning: -n introduces	 a  power-gain	of  about
		 11dB  in  the	filter, so beware of output clip­
		 ping.	Band introduces noise in the shape of the
		 filter, i.e. peaking at the center frequency and
		 settling around it.  See filter for  a	 bandpass
		 effect with steeper shoulders.

       bandpass frequency bandwidth
		 Butterworth  bandpass filter. Description coming
		 soon!

       bandreject frequency bandwidth
		 Butterworth bandreject filter.	 Description com­
		 ing soon!

       chorus gain-in gain-out delay decay speed depth

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

       compand attack1,decay1[,attack2,decay2...]

	       in-dB1,out-dB1[,in-dB2,out-dB2...]

	       [gain [initial-volume [delay ] ] ]
		 Compand  (compress  or expand) the dynamic range
		 of a sample.  The attack and decay time  specify
		 the  integration  time	 over  which the absolute
		 value of  the	input  signal  is  integrated  to
		 determine its volume; attacks refer to increases
		 in volume and decays refer to decreases.   Where
		 more  than  one  pair of attack/decay parameters
		 are specified, each  channel  is  treated  sepa­
		 rately	 and  the number of pairs must agree with
		 the number of input channels.	The second param­
		 eter  is  a  list  of	points on the compander's
		 transfer function specified in	 dB  relative  to
		 the  maximum  possible	 signal	 amplitude.   The
		 input values must be in  a  strictly  increasing
		 order but the transfer function does not have to
		 be monotonically rising.  The special value -inf
		 may  be  used	to indicate that the input volume
		 should be associated output volume.  The  points
		 -inf,-inf and 0,0 are assumed; the latter may be
		 overridden, but the former may not.

		 The third (optional) parameter	 is  a	post-pro­
		 cessing  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  oper­
		 ate: 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  immedi­
		 ately	to  control  the  compander,  but  it  is
		 delayed before being fed to the volume adjuster.
		 Specifying  a	delay  approximately equal to the
		 attack/decay  times  allows  the  compander   to
		 effectively  operate  in  a  "predictive" rather
		 than a reactive mode.

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

       dcshift shift [ limitergain ]
		 DC Shift  the	audio  data,  with  basic  linear
		 amplitude  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.

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

       earwax	 Makes	sound  easier to listen to on headphones.
		 Adds audio-cues to samples in audio cd format so
		 that  when  listened to on headphones the stereo
		 image is moved from inside your  head	(standard
		 for  headphones)  to outside and in front of the
		 listener (standard for speakers). See
		 www.geocities.com/beinges for	a  full	 explana­
		 tion.

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

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

       fade [ type ] fade-in-length

	    [ stop-time [ fade-out-length ] ]
		 Add a fade effect to the beginning, end, or both
		 of the audio data.

		 For fade-ins, this starts from the first  sample
		 and ramps the volume of the audio from 0 to full
		 volume over fade-in-length seconds.   Specify	0
		 seconds if no fade-in is wanted.

		 For  fade-outs, the audio data will be truncated
		 at the stop-time and the volume will  be  ramped
		 from full volume down to 0 starting at fade-out-
		 length seconds before the stop-time.	No  fade-
		 out is performed if these options are not speci­
		 fied.
		 All times can be specified in either periods  of
		 time  or sample counts.  To specify time periods
		 use the format hh:mm:ss.frac format.  To specify
		 using	sample counts, specify the number of sam­
		 ples and append the letter  's'  to  the  sample
		 count (for example 8000s).
		 An  optional type can be specified to change the
		 type of envelope.  Choices are q for quarter  of
		 a  sinewave, h for half a sinewave, t for linear
		 slope, l for logarithmic,  and	 p  for	 inverted
		 parabola.  The default is a linear slope.

       filter [ low ]-[ high ] [ window-len [ beta ] ]
		 Apply	a  Sinc-windowed  lowpass,  highpass,  or
		 bandpass filter of given window  length  to  the
		 signal.   low	refers	to  the	 frequency of the
		 lower 6dB corner of the filter.  high refers  to
		 the  frequency	 of  the  upper 6dB corner of the
		 filter.

		 A lowpass filter  is  obtained	 by  leaving  low
		 unspecified,	or   0.	  A  highpass  filter  is
		 obtained by leaving high unspecified, or  0,  or
		 greater  than or equal to the Nyquist frequency.

		 The window-len, if unspecified, defaults to 128.
		 Longer	 windows  give	a sharper cutoff, smaller
		 windows a more gradual cutoff.

		 The beta, if unspecified, defaults to 16.   This
		 selects  a Kaiser window.  You can select a 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 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 coming
		 soon!

       lowp frequency
		 Apply a single pole recursive	low-pass  filter.
		 The  frequency	 response  drops  logarithmically
		 with frequency in the middle of the  drop.   The
		 slope of the filter is quite gentle.  See filter
		 for a lowpass effect with sharper cutoff.

       lowpass frequency
		 Butterworth lowpass filter.  Description  coming
		 soon!

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

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

       pan direction
		 Pan the sound of an audio file from one  channel
		 to another.  This is done by changing the volume
		 of the input channels so that it  fades  out  on
		 one  channel  and  fades-in  on another.  If the
		 number of input channels is different	then  the
		 number of output channels then this effect tries
		 to intelligently handle this.	For instance,  if
		 the input contains 1 channel and the output con­
		 tains 2 channels, then it will create the  miss­
		 ing  channel  itself.	 The direction is a value
		 from -1.0 to 1.0.  -1.0 represents far left  and
		 1.0  represents  far  right.  Numbers in between
		 will start the pan effect without totally muting
		 the opposite channel.

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

       pick [ -1 | -2 | -3 | -4 | -l | -r ]
		 Select	 the  left  or	right channel of a stereo
		 sample, or one of four	 channels  in  a  quadra­
		 phonic	 sample.  The -l and -r options represent
		 either	 the  left  or	right  channel.	  It   is
		 required  that	 you  use  the	-c 1 command line
		 option in order to force the output file to con­
		 tain only 1 channel.

       pitch shift [ width interpole fade ]
		 Change	 the  pitch of file without affecting its
		 duration by cross-fading shifted samples.  shift
		 is given in cents. Use a positive value to shift
		 to treble, negative  value  to	 shift	to  bass.
		 Default  shift	 is 0.	width of window is in ms.
		 Default width is 20ms. Try 30ms to lower  pitch,
		 and  10ms to raise pitch.  interpole option, can
		 be "cubic" or "linear". Default is "cubic".  The
		 fade  option,	can be "cos", "hamming", "linear"
		 or "trapezoid".  Default is "cos".

       polyphase [ -w < nut / ham > ]

		 [  -width <  long  / short  / # > ]

		 [ -cutoff #  ]
		 Translate input sampling rate to output sampling
		 rate  via  polyphase  interpolation, a DSP algo­
		 rithm.	 This method is slow  and  uses	 lots  of
		 RAM, but gives much better results than rate.

		 -w  <	nut / ham > : select either a Nuttal (~90
		 dB stopband) or Hamming (~43 dB  stopband)  win­
		 dow.  Default is nut.

		 -width	 long / short / # : specify the (approxi­
		 mate) width of the filter.  long  is  1024  sam­
		 ples;	short  is 128 samples.	Alternatively, an
		 exact number can be used.  Default is long.  The
		 short	option is not recommended, as it produces
		 poor quality results.

		 -cutoff # : specify the filter cutoff	frequency
		 in  terms  of	fraction  of frequency bandwidth,
		 also know as the Nyquist frequency.  Please  see
		 the  resample	effect for further information on
		 Nyquist frequency.  If upsampling, then this  is
		 the  fraction of the original signal that should
		 go through.  If downsampling, this is the  frac­
		 tion  of  the	signal	left  after downsampling.
		 Default is 0.95.  Remember that this is a float.


       rate	 Translate input sampling rate to output sampling
		 rate via linear interpolation to the Least  Com­
		 mon Multiple of the two sampling rates.  This is
		 the default effect if the two files have differ­
		 ent  sampling	rates and the preview options was
		 specified.  This is fast but noisy: the spectrum
		 of  the  original  sound will be shifted upwards
		 and duplicated faintly when up-translating by	a
		 multiple.

		 Lerp-ing  is  acceptable  for	cheap 8-bit sound
		 hardware, but for CD-quality  sound  you  should
		 instead  use  either  resample or polyphase.  If
		 you are wondering which rate changing effects to
		 use,  you  will want to read a detailed analysis
		 of  all  of  them  at	http://eakaw2.et.tu-dres­
		 den.de/~wilde/resample/resample.html

       resample [ -qs | -q | -ql ] [ rolloff [ beta ] ]
		 Translate input sampling rate to output sampling
		 rate  via  simulated  analog  filtration.   This
		 method	 is slower than rate, but gives much bet­
		 ter results.

		 By default, linear interpolation is used, with a
		 window	 width	about  45 samples at the lower of
		 the two rate.	This gives an accuracy	of  about
		 16  bits, but insufficient stopband rejection in
		 the case that you want to have	 rolloff  greater
		 than about 0.80 of the Nyquist frequency.

		 The  -q*  options will change the default values
		 for rolloff and beta as well  as  use	quadratic
		 interpolation	of filter coefficients, resulting
		 in about 24 bits precision.  The -qs, -q, or -ql
		 options  specify  increased accuracy at the cost
		 of lower execution speed.   It	 is  optional  to
		 specify  rolloff  and beta parameters when using
		 the -q* options.

		 Following is a table of the reasonable	 defaults
		 which are built-in to SoX:

		    Option  Window rolloff beta interpolation
		    ------  ------ ------- ---- -------------
		    (none)    45    0.80    16	   linear
		      -qs     45    0.80    16	  quadratic
		      -q      75    0.875   16	  quadratic
		      -ql    149    0.94    16	  quadratic
		    ------  ------ ------- ---- -------------

		 -qs, -q, or -ql use window lengths of 45, 75, or
		 149 samples, respectively, at the lower  sample-
		 rate of the two files.	 This means progressively
		 sharper stop-band rejection,  at  proportionally
		 slower execution times.

		 rolloff  refers  to the cut-off frequency of the
		 low pass filter and is given  in  terms  of  the
		 Nyquist  frequency  for  the  lower sample rate.
		 rolloff therefore should  be  something  between
		 0.0 and 1.0, in practice 0.8-0.95.  The defaults
		 are indicated above.

		 The Nyquist frequency is equal to (sample rate /
		 2).   Logically,  this	 is  because the A/D con­
		 verter needs at least	2  samples  to	detect	1
		 cycle	at  the	 Nyquist  frequency.  Frequencies
		 higher then the Nyquist will actually appear  as
		 lower	frequencies  to	 the A/D converter and is
		 called aliasing.  Normally, A/D converts run the
		 signal	 through a highpass filter first to avoid
		 these problems.

		 Similar problems will happen  in  software  when
		 reducing  the sample rate of an audio file (fre­
		 quencies above the new Nyquist frequency can  be
		 aliased  to  lower  frequencies).   Therefore, a
		 good resample effect will remove  all	frequency
		 information above the new Nyquist frequency.

		 The  rolloff  refers to how close to the Nyquist
		 frequency this cutoff is, with closer being bet­
		 ter.	When  increasing  the  sample  rate of an
		 audio file you would not expect to have any fre­
		 quencies   exist  that	 are  past  the	 original
		 Nyquist frequency.  Because of resampling  prop­
		 erties,  it is common to have alaising data cre­
		 ated that is above the	 old  Nyquist  frequency.
		 In  that case the rolloff refers to how close to
		 the original Nyquist frequency to use a highpass
		 filter	 to  remove  this false data, with closer
		 also being better.

		 The beta parameter determines the type of filter
		 window	 used.	Any value greater than 2.0 is the
		 beta for a Kaiser window.  Beta <= 2.0 selects a
		 Nuttall  window.  If unspecified, the default is
		 a Kaiser window with beta 16.

		 In the case of Kaiser window (beta > 2.0), lower
		 betas	produce a somewhat faster transition from
		 passband to stopband, at the cost of  noticeable
		 artifacts.   A	 beta  of 16 is the default, beta
		 less than 10 is not recommended.  If you want	a
		 sharper  cutoff,  don't  use  low  beta's, use a
		 longer	 sample	 window.   A  Nuttall  window  is
		 selected  by specifying any 'beta' <= 2, and the
		 Nuttall window has somewhat steeper cutoff  than
		 the  default  Kaiser  window.	You will probably
		 not need to  use  the	beta  parameter	 at  all,
		 unless	 you are just curious about comparing the
		 effects of Nuttall vs. Kaiser windows.

		 This is the default effect if the two files have
		 different  sampling  rates.   Default parameters
		 are, as indicated above, Kaiser window of length
		 45, rolloff 0.80, beta 16, linear interpolation.

		 NOTE: -qs is  only  slightly  slower,	but  more
		 accurate for 16-bit or higher precision.

		 NOTE:	In many cases of up-sampling, no interpo­
		 lation is needed, as exact  filter  coefficients
		 can be computed in a reasonable amount of space.
		 To be precise, this is done when

			    input_rate < output_rate
				       &&
		   output_rate/gcd(input_rate,output_rate) <= 511

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

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

       silence above_periods [ duration threshold[ d | % ]

	       [ below_periods duration

		 threshold[ d | % ]]
		 Removes silence from the beginning or end  of	a
		 sound	file.  Silence is anything below a speci­
		 fied threshold.
		 When trimming silence from the	 beginning  of	a
		 sound file, you specify a duration of audio that
		 is above a given silence threshold before  audio
		 data  is  processed.	You  can also specify the
		 count of periods of none  silence  you	 want  to
		 detect	 before processing audio data.	Specify a
		 period of 0 if you do not want to trim data from
		 the front of the sound file.
		 When optionally trimming silence form the end of
		 a sound file, you specify the duration of  audio
		 that  must  be	 below	a  given threshold before
		 stopping to process  audio  data.   A	count  of
		 periods  that occur below the threshold may also
		 be specified.	If this options are not specified
		 then  data  is	 not  trimmed from the end of the
		 audio file.
		 Duration counts may be in the	format	of  time,
		 hh:mm:ss.frac, or in the exact count of samples.
		 Threshold may be suffixed with d, or % to  indi­
		 cated	the  value is in decibels or a percentage
		 of max value of the sample value.   A	value  of
		 '0%' will look for total silence.

       speed [ -c ] factor
		 Speed	up  or down the sound, as a magnetic tape
		 with a speed control.	It affects both pitch and
		 time.	A  factor  of 1.0 means no change, and is
		 the  default.	 2.0  doubles  speed,  thus  time
		 length	 is cut by a half and pitch is one octave
		 higher.  0.5 halves speed thus time length  dou­
		 bles  and  pitch  is  one  octave lower.  If the
		 optional -c parameter is used then the factor is
		 specified in "cents".

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

       stat [ -s n ] [-rms ] [ -v ] [ -d ]
		 Do  a	statistical  check on the input file, and
		 print results on the standard error file.  Audio
		 data  is  passed unmodified from input to output
		 file unless used along with the -e option.

		 The "Volume Adjustment:" field in the statistics
		 gives	you  the  argument to the -v number which
		 will make the sample as loud as possible without
		 clipping.

		 The option -v will print out the "Volume Adjust­
		 ment:" field's	 value	only  and  return.   This
		 could	be  of use in scripts to auto convert the
		 volume.

		 The -s n option is used to scale the input  data
		 by  a	given  factor.	The default value of n is
		 the  max  value  of  a	 signed	  long	 variable
		 (0x7fffffff).	Internal effects always work with
		 signed long PCM data and  so  the  value  should
		 relate to this fact.

		 The  -rms option will convert all output average
		 values to root mean square format.

		 There is also an optional parameter -d that will
		 print	out a hex dump of the sound file from the
		 internal buffer that is  in  32-bit  signed  PCM
		 data.	 This  is  mainly only of use in tracking
		 down endian problems that creep  in  to  SoX  on
		 cross-platform versions.


       stretch factor [window fade shift fading]
		 Time  stretch	file  by  a  given factor. Change
		 duration without affecting the pitch.	factor of
		 stretching:  >1.0  lengthen,  <1.0 shorten dura­
		 tion.	window size is in ms.  Default	is  20ms.
		 The  fade option, can be "lin".  shift ratio, in
		 [0.0 1.0]. Default depends  on	 stretch  factor.
		 1.0  to  shorten,  0.8	 to lengthen.  The fading
		 ratio, in [0.0 0.5].  The  amount  of	a  fade's
		 default depends on factor and shift.

       swap [ 1 2 | 1 2 3 4 ]
		 Swap  channels	 in  multi-channel  sound  files.
		 Optionally, you may specify  the  channel  order
		 you  would like the output in.	 This defaults to
		 output channel 2 and then 1 for stereo and 2, 1,
		 4,  3 for quad-channels.  An interesting feature
		 is that you may duplicate  a  given  channel  by
		 overwriting  another.	This is done by repeating
		 an output channel  on	the  command  line.   For
		 example,  swap 2 2 will overwrite channel 1 with
		 channel 2's data; creating a  stereo  file  with
		 both channels containing the same audio data.

       synth [ length ] type mix [ freq [ -freq2 ]

	     [ off ] [ ph ] [ p1 ] [ p2 ] [ p3 ]
		 The  synth effect will generate various types of
		 audio data.  Although this  effect  is	 used  to
		 generate audio data, an input file must be spec­
		 ified.	 The  length  of  the  input  audio  file
		 determines  the length of the output audio file.
		 <length>  length  in	sec   or   hh:mm:ss.frac,
		 0=inputlength, default=0
		 <type>	 is  sine,  square,  triangle,	sawtooth,
		 trapetz, exp, whitenoise, pinknoise, brownnoise,
		 default=sine
		 <mix> is create, mix, amod, default=create
		 <freq>	 frequency  at	beginning in Hz, not used
		 for noise..
		 <freq2> frequency at end in  Hz,  not	used  for
		 noise..  <freq/2> can be given as %%n, where 'n'
		 is the number of half	notes  in  respect  to	A
		 (440Hz)
		 <off>	Bias  (DC-offset)   of signal in percent,
		 default=0
		 <ph> phase shift 0..100 shift phase 0..2*Pi, not
		 used for noise..
		 <p1>  square: Ton/Toff, triangle+trapetz: rising
		 slope time (0..100)
		 <p2> trapetz: ON time (0..100)
		 <p3> trapetz: falling slope position (0..100)

       trim start [ length ]
		 Trim can trim off unwanted audio data	from  the
		 beginning and end of the audio file.  Audio sam­
		 ples are not sent to the output stream until the
		 start location is reached.
		 The  optional	length parameter tells the number
		 of samples to output after the start sample  and
		 is  used  to trim off the back side of the audio
		 data.	Using a value of 0 for the start  parame­
		 ter  will allow trimming off the back side only.
		 Both options can be specified	using  either  an
		 amount	 of  time  and an exact count of samples.
		 The format for specifying  lengths  in	 time  is
		 hh:mm:ss.frac.	 A start value of 1:30.5 will not
		 start until 1 minute,	thirty	and  1/2  seconds
		 into  the audio data.	The format for specifying
		 sample counts is the number of samples with  the
		 letter	 's'  appended	to  it.	 A value of 8000s
		 will wait until 8000  samples	are  read  before
		 starting to process audio data.

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

       vol gain [ type [ limitergain ] ]
		 The  vol  effect  is  much like the command line
		 option -v.  It allows you to adjust  the  volume
		 of  an	 input file and allows you to specify the
		 adjustment in relation to amplitude,  power,  or
		 dB.   If  type is not specified then it defaults
		 to amplitude.
		 When type is amplitude then a linear  change  of
		 the  amplitude	 is  performed based on the gain.
		 Therefore, a value of 1.0 will keep  the  volume
		 the  same, 0.0 to < 1.0 will cause the volume to
		 decrease and values of > 1.0 will cause the vol­
		 ume  to increase.  Beware of clipping audio data
		 when the gain is greater then 1.0.   A	 negative
		 value	performs  the  same adjustment while also
		 changing the phase.
		 When type is power then  a  value  of	1.0  also
		 means no change in volume.
		 When  type  is dB the amplitude is changed loga­
		 rithmically.  0.0 is constant while  +6  doubles
		 the amplitude.
		 An  optional  limitergain value can be specified
		 and should be a value much  less  then	 1.0  (ie
		 0.05  or 0.02) and is used only on peaks to pre­
		 vent clipping.	 Not  specifying  this	parameter
		 will  cause  no  limiter to be used.  In verbose
		 mode, this effect will display the percentage of
		 audio data that needed to be limited.

BUGS
       The  syntax  is horrific.  Thats the breaks when trying to
       handle all things from the command line.

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

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

NOTICES
       The  version  of	 SoX that accompanies this manual page is
       support by Chris Bagwell (cbagwell@users.sourceforge.net).
       Please  refer  any questions regarding it to this address.
       You may obtain the latest version  at  the  the	web  site
       http://sox.sourceforge.net/

AUTHOR
       Chris Bagwell (cbagwell@users.sourceforge.net).

       Updates by Anonymous



			December 11, 2001		   SoX(1)
