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.

       .mp3	 MP3 Compressed Audio
		 MP3 audio files come from the MPEG standards for
		 audio	and  video compression.	 They are a lossy
		 compression format that achieves  good	 compres­
		 sion  rates  with  a  minimum	amount of quality
		 loss.	Also see Ogg Vorbis for a similar format.
		 MP3  support  in  SoX	is  optional and requires
		 access to either or both the external libmad and
		 libmp3lame  libraries.	  To see if there is sup­
		 port for Mp3 run sox -h and look  for	it  under
		 the list of supported file formats as "mp3".


       .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.  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.
		 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)
