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SoX(1)									SoX(1)

NAME
       sox - Sound eXchange : universal sound sample translator

SYNOPSIS
       sox infile1 [ infile2 ... ] outfile

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

       soxmix infile1 infile2 [ infile3 ... ] outfile

       soxmix [ general options ] [ format options ] infile1
	   [ format options ] infile2
	   [ [ format options ] infile3 ... ]
	   [ format options ] outfile
	   [ effect [ effect options ] ... ]

       General options:
	   [ -h ] [ -p ] [ -q ] [ -S ] [ -V ]

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

       Effects:
	   avg [ -l | -r | -f | -b | -1 | -2 | -3 | -4 | 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
	   mask
	   mcompand "attack1,decay1[,attack2,decay2...]
		    in-dB1,out-dB1[,in-dB2,out-dB2...]
		    [ gain [ initial-volume [ delay ] ] ]" xover_freq
	   noiseprof [profile-file]
	   noisered profile-file [threshold]
	   pan direction
	   phaser gain-in gain-out delay decay speed < -s | -t >
	   pick [ -1 | -2 | -3 | -4 | -l | -r | -f | -b ]
	   pitch shift [ width interpole fade ]
	   polyphase [ -w < nut / ham > ]
		     [	-width < long / short / # > ]
		     [ -cutoff # ]
	   rate
	   repeat count
	   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
	   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 popular 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.

       If more than one input file is specified	 then  they  are  concatenated
       into  the  output  file.	  In  this case, it has a restriction that all
       input files must be of the same data type and sample rates.

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

       There  are two types of audio file formats that SoX can work with.  The
       first are self-describing file formats.	These contain  a  header  that
       completely describe the characteristics 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 characteristics:

       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 popular 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 produce a single
       sound file.

	       soxmix music.wav voice.wav mixed.wav

       Filenames:

       SoX can be used as a part of pipe operations by using the special file‐
       names  of  "-".	 If specified as an input name, it will read data from
       stdin.  If specified as an output name, it will send data to stdout.

       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 chan‐
		 nels and a different rate than the  input  file.   Currently,
		 this  defaults to using the rate effect instead of the resam‐
		 ple effect for sample rate changes.

       -q	 Run in quite  mode  when  SoX	wouldn't  otherwise  do	 that.
		 Inverse of -S option.

       -S	 Print	status while processing audio data.  Tells how much of
		 audio data has been processed in terms of audio running  time
		 instead of samples.

       -V	 Print	a description of processing phases.  Useful for figur‐
		 ing out exactly how SoX

       is mangling your sound samples.

       Format options:

       Format options effect the input or output file  that  they  immediately
       precede.

       Self  describing	 input	files  can  obtain  all the format information
       directly from the header and so don't generally	need  format  options.
       Headerless input files lack this information and so format options must
       be used to inform SoX of the file's data type, sample rate, and	number
       of channels.

       By  default, SoX attempts to write audio data using the same data type,
       sample rate, and channel count as the input data.  If  the  user	 wants
       the  output file to be of a different format then format options can be
       used to specify the differences.

       If an output file format doesn't support the  same  data	 type,	sample
       rate,  or  channel  count  as the input file format, then SoX will auto
       select the closest values it does support so that  the  user  does  not
       have to specify these format change options manually.

       -t filetype
		 gives	the  type  of the sound sample file.  Useful when file
		 extension is not standard or can not be determeind by looking
		 at the header of the file.

       -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 format
		 options.
		 If the input and output files have  different	rates  then  a
		 sample	 rate change effect must be ran.  Since SoX has multi‐
		 ple rate changing effects, the user can specify which to  use
		 as  an	 effect.  If no rate change effect is specified then a
		 default one will be chosen.

       -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.
		 As with other format options, the volume option  effects  the
		 file its specified with.  This is useful whe processing muti‐
		 ple input files as the volume adjustment can be specified for
		 each input file or just once to adjust the output file.  This
		 can be compared to an audio mixer were you  can  control  the
		 volume	 of  each  input  as  well  as a master volume (output
		 side).
		 soxmix defaults the value of the -v  option  for  each	 input
		 file  to  1/input_file_count.	 This means if your mixing two
		 input	files  together	 then  each  input  file's  volume  is
		 adjusted  by  0.5.  This is done to prevent clipping of audio
		 data during the mixing operation.  Users will most likely not
		 be happy with this large of a volume adjustment and can spec‐
		 ify the -v option to override this default value.
		 Note: For the non-mixing case, see the stat effect for infor‐
		 mation	 on  finding the maximum volume adjustment that can be
		 done with this	 option	 without  causing  audio  data	to  be
		 clipped.

       -s/-u/-U/-A/-a/-i/-g/-f
		 The  sample  data encoding is signed linear (2's complement),
		 unsigned linear, u-law	 (logarithmic),	 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 preci‐
		 sion of 14-bit PCM audio and A-law has roughly the  precision
		 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	under‐
		 stands	 how to work with this encoding but there is currently
		 no command line option to specify it.	If you need this  sup‐
		 port  then  you  can  use  the psuedo file types of ".la" and
		 ".lu" to inform sox of	 the  encoding.	  See  supported  file
		 types for more information.
		 ADPCM	is a form of sound compression that has a good compro‐
		 mise 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 ver‐
		 sion of ADPCM include G.726, MS ADPCM, and IMA ADPCM.	The -a
		 flag  has  different 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  compression  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/-d
		 The  sample  data size is in bytes, 16-bit words, 32-bit long
		 words, or 64-bit double long (long 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  num‐
		 ber of channels then the avg effect must be used.  If the avg
		 effect is not specified  on  the  command  line  it  will  be
		 invoked internally with default parameters.

       -e	 When  specified  after	 the  last  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 specified effects  to  the  input  file.   This  is
		 mainly useful with the stat effect but can be used.

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 format.

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

       .alsa	 ALSA /dev/snd/pcmCxDxp 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  ALSA  /dev/snd/pcmCxDxp 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 ALSA 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: sox infile -t alsa -w -s
		 /dev/snd/pcmC0D0p

       .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 head‐
		 ers 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 commercial packages
		 on the Mac.

       .cdr	 CD-R
		 CD-R files are used in mastering music on Compact 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 handler.

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

       .dat	 Text Data files
		 These files contain a textual representation  of  the	sample
		 data.	 There	is one line at the beginning that contains the
		 sample rate.	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  ana‐
		 lyzers	 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  telecommunications (GSM).  Its good for
		 its purpose, shrinking audio data size, but it will introduce
		 lots  of  noise  when	a  given  sound	 sample is encoded and
		 decoded multiple times.  This format is used  by  some	 voice
		 mail applications.  It is rather CPU intensive.
		 GSM in SoX is optional and requires access to an external GSM
		 library.  To see if there is support for gsm run sox  -h  and
		 look for it under the list of supported file formats.

       .hcom	 Macintosh  HCOM files.	 These are (apparently) Mac FSSD files
		 with some variant of Huffman compression.  The Macintosh  has
		 wacky file formats and this format handler apparently doesn't
		 handle all the ones it should.	  Mac  users  will  need  your
		 usual	arsenal	 of  file converters to deal with an HCOM file
		 under Unix or DOS.

       .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	 compression  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 support 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 out‐
		 put 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 consistency.

       .ogg	 Ogg Vorbis Compressed Audio.
		 Ogg  Vorbis  is  a  open, patent-free CODEC designed for com‐
		 pressing 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 inten‐
		 sive.
		 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  out‐
		 put values to use the highest quality samples your sound card
		 can handle.  Example: sox infile -t ossdsp -w -s /dev/dsp

       .prc	 Psion record.app
		 Used in some Psion devices for System alarms.	This format is
		 newer	then  the  .wve	 format	 that  is  used	 in some Psion
		 devices.

       .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 format defined by
		 NIST (National Institute of Standards and Technology) and  is
		 used  with  speech audio.  SoX can read these files when they
		 contain u-law and PCM data.  It will ignore any header infor‐
		 mation	 that  says  the data is compressed using shorten com‐
		 pression 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 processing.

       .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 them‐
		 selves. 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 out‐
		 put values to use the highest quality samples	your  hardware
		 can handle.  Example: sox infile -t sunau -w -s /dev/audio or
		 sox infile -t sunau -U -c 1 /dev/audio for older  sun	equip‐
		 ment.

       .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 generated appropriately.  On out‐
		 put, 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.

       .vox	 A  headerless	file of Dialogic/OKI ADPCM audio data commonly
		 comes with the extension .vox.	 This ADPCM  data  has	12-bit
		 precision packed into only 4-bits.

       .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 formatting information in their
		 headers, and so do not need any format options specified  for
		 an  input  file.  If  any  are,  they	will override the file
		 header, and you will be warned to this effect.	 You had  bet‐
		 ter know what you are doing! Output format options will cause
		 a format conversion, and the .wav will written appropriately.
		 SoX currently can read PCM, ULAW, ALAW, MS ADPCM, and IMA (or
		 DVI)  ADPCM.  It can write all of these formats including the
		 ADPCM encoding.  Big endian versions of  RIFF	files,	called
		 RIFX,	can  also  be read and written.	 To write a RIFX file,
		 use the -x option with the output file options.

       .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'' and is the default file type  if  the
		 user  does  not specify one. This file type attempts to guess
		 the real type by looking for magic words in  the  header.  If
		 the  type  can't  be guessed, the program exits with an error
		 message.  The input must be a plain file, not a  pipe.	  This
		 type can't be used for output files.

EFFECTS
       Multiple	 effects  may  be applied to the audio data by specifying them
       one after another at the end of the command line.

       avg [ -l | -r | -f | -b | -1 | -2 | -3 | -4 | n,n,...,n ]
		 Reduce the number of channels by averaging  the  samples,  or
		 duplicate  channels to increase the number of channels.  This
		 effect is automatically used when the number of  input	 chan‐
		 nels  differ from the number of output channels.  When reduc‐
		 ing the number of channels it is possible to manually specify
		 the  avg  effect  and use the -l, -r, -f, -b, -1, -2, -3, -4,
		 options to select only the left,  right,  front,  back	 chan‐
		 nel(s)	 or specific channel for the output instead of averag‐
		 ing the channels.  The -l, and -r options will	 do  averaging
		 in  quad-channel files so select the exact channel to prevent
		 this.

		 The avg effect can also be invoked with up to 16  double-pre‐
		 cision	 numbers,  seperated by commas, which specify the pro‐
		 portion (0.0 = 0% and 1.0 = 100%) 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  fol‐
		 lows:	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, certain reduced combination of numbers can be speci‐
		 fied for certain input/output channel combinations.

		 In Ch	Out Ch Num Mappings
		 _____	______ ___ _____________________________
		   2	  1	2   l->l, r->l
		   2	  2	1   adjust balance
		   4	  1	4   lf->l, rf->l, lb->l, rb-l
		   4	  2	2   lf->l&rf->r, lb->l&rb->r
		   4	  4	1   adjust balance
		   4	  4	2   front balance, back balance

       band [ -n ] center [ width ]
		 Apply a band-pass filter.  The frequency response drops loga‐
		 rithmically 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 instrumental music.	 The  -n  (for	noise)
		 option uses the alternate mode for un-pitched signals.	 Warn‐
		 ing: -n introduces a power-gain of about 11dB in the  filter,
		 so  beware  of output clipping.  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 coming 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 milliseconds 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  separately  and  the number of pairs must agree with
		 the number of input channels.	The second parameter is a list
		 of  points  on the compander's transfer function specified in
		 dB relative to the maximum possible  signal  amplitude.   The
		 input	values	must be in a strictly increasing order but the
		 transfer 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-processing  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 per‐
		 mits the user to supply a nominal level initially,  so	 that,
		 for example, a very large gain is not applied to initial sig‐
		 nal levels before the companding action has begun to operate:
		 it  is quite probable that in such an event, the output would
		 be severely clipped while the compander gain properly adjusts
		 itself.

		 The  fifth  (optional)	 parameter is a delay in seconds.  The
		 input signal is analyzed immediately to control  the  compan‐
		 der,  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 oper‐
		 ate 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  cen‐
		 tered	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 rectified.  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 lis‐
		 tener (standard for speakers). See
		 www.geocities.com/beinges for a full explanation.

       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 out‐
		 put.

       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.  If
		 fade-out-length is not specified, it  defaults	 to  the  same
		 value	as  fade-in-length.   No  fade-out is performed if the
		 stop-time is not specified.
		 All times can be specified in either periods of time or  sam‐
		 ple   counts.	  To  specify  time  periods  use  the	format
		 hh:mm:ss.frac format.	To specify using sample counts,	 spec‐
		 ify  the  number  of samples and append the letter 's' to the
		 sample count (for example 8000s).
		 An optional type can be specified to change the type of enve‐
		 lope.	 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 fre‐
		 quency 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 unspeci‐
		 fied, or 0, or greater than or	 equal	to  the	 Nyquist  fre‐
		 quency.

		 The window-len, if unspecified, defaults to 128.  Longer win‐
		 dows 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 Nuttall 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 milliseconds and the
		 decay (relative to gain-in) with a modulation	speed  in  Hz.
		 The  modulation  is  either sinodial (-s) or triangular (-t).
		 Gain-out is the volume of the output.

       highp frequency
		 Apply a single pole recursive	high-pass  filter.   The  fre‐
		 quency 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!

       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.

       mcompand "attack1,decay1[,attack2,decay2...]

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

		[gain [initial-volume [delay ] ] ]" xover_freq

		 Multi-band compander is similar to the single band  compander
		 but  the  audio  file is first divided up into bands and then
		 the compander is ran on each band.  See  the  compand	effect
		 for definition of its options.	 Compand options are specified
		 between double quotes and the crossover  frequency  for  that
		 band  is  specefied  seperately  with xover_fre.  This can be
		 repeated multiple times to create multiple bands.

       noiseprof [profile-file]

       noisered profile-file [threshold]
		 Noise reduction filter with profiling. This filter is	moder‐
		 ately	effective at removing consistent background noise such
		 as hiss or hum. To use it, first run the noiseprof effect  on
		 a section of silence (that is, a section which contains noth‐
		 ing but noise). The noiseprof effect will print a noise  pro‐
		 file  to  profile-file,  or  to  stdout if no profile-file is
		 specified.  If there is sound output on stdout then the  pro‐
		 file will instead be directed to stderr.

		 To actually remove the noise, run SoX again with the noisered
		 filter. The filter needs one  argument,  profile-file,	 which
		 contains  the	noise profile from noiseprof. thershold speci‐
		 fies how much noise should be removed, and may be  between  0
		 and  1	 with a default of 0.5. Higher values will remove more
		 noise but present a greater  possibility  of  distorting  the
		 desired  audio	 signal.   Experiment with different threshold
		 values to find the optimal one for your sample.

       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  han‐
		 dle  this.  For instance, if the input contains 1 channel and
		 the output contains 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 milliseconds and the
		 decay (relative to gain-in) with a modulation	speed  in  Hz.
		 The  modulation  is  either sinodial (-s) or triangular (-t).
		 The decay should be less than 0.5 to avoid  feedback.	 Gain-
		 out is the volume of the output.

       pick [ -1 | -2 | -3 | -4 | -l | -r | -f | -b ]
		 Pick  a subset of channels to be copied into the output file.
		 This effect is just an alias of the "avg" effect but is  left
		 here for historical reasons.

       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", "lin‐
		 ear" 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  algorithm.	This method is
		 slow and uses lots of RAM, but gives much better results than
		 rate.

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

		 -width long / short / # : specify the (approximate) width  of
		 the  filter.	long  is  1024	samples; 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 infor‐
		 mation on Nyquist frequency.  If upsampling, then this is the
		 fraction of the original signal that should go	 through.   If
		 downsampling,	this  is the fraction of the signal left after
		 downsampling.	Default is 0.95.   Remember  that  this	 is  a
		 float.

       rate	 Translate  input  sampling  rate  to output sampling rate via
		 linear interpolation to the Least Common Multiple of the  two
		 sampling  rates.  This is the default effect if the two files
		 have different sampling rates and  the	 preview  options  was
		 specified.  This is fast but noisy: the spectrum of the orig‐
		 inal 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://leute.server.de/wilde/resample.html

       repeat count
		 Repeats  the  audio data count times.	Requires disk space to
		 store the data to be repeated.

       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 better 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	rejec‐
		 tion  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 pro‐
		 portionally slower execution times.

		 rolloff  refers to the cut-off frequency of the low pass fil‐
		 ter 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).	 Logi‐
		 cally, this is because the A/D converter  needs  at  least  2
		 samples to detect 1 cycle at the Nyquist frequency.  Frequen‐
		 cies 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 fil‐
		 ter first to avoid these problems.

		 Similar  problems  will  happen in software when reducing the
		 sample rate of an  audio  file	 (frequencies  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  better.	 When  increasing  the
		 sample rate of an audio file you would not expect to have any
		 frequencies exist that are past  the  original	 Nyquist  fre‐
		 quency.   Because  of	resampling properties, it is common to
		 have aliasing data created that is above the old Nyquist fre‐
		 quency.   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  pro‐
		 duce  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 cut‐
		 off  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  interpolation  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 reverbe-time 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 reverberation.
		 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, middle, or end of a sound
		 file.	Silence is anything below a specified threshold.

		 The above_periods value is used to indicate if	 sound	should
		 be  trimmed  at  the beginning of the audio file.  A value of
		 zero indicates no silence should be trimmed from  the	begin‐
		 ning.	 When  specifing  an  non-zero above_periods, it trims
		 audio up until it finds non-silence.  Normally, when trimming
		 silence  from	beginning of audio the above_periods will be 1
		 but it can be increased to higher values to trim all data  up
		 to  a specific count of non-silence periods.  For example, if
		 you had an audio file with two songs that  each  contained  2
		 seconds  of  silence  before  the  song, you could specify an
		 above_period of 2 to strip out both silence periods  and  the
		 first song.

		 When above_periods is non-zero, you must also specify a dura‐
		 tion and threshold.  Duration indications the amount of  time
		 that  non-silence  must  be detected before it stops trimming
		 data.	By increasing the duration,  burst  of	noise  can  be
		 treated as silence and trimmed off.

		 Threshold  is	used  to indicate what sample value you should
		 treat as silence.  For digital audio, a value	of  0  may  be
		 fine  but  for	 audio	recorded  from analog, you may wish to
		 increase ths value to account for background noise.

		 When optionally trimming silence from	the  end  of  a	 sound
		 file,	you  specify  a	 below_periods	count.	 In this case,
		 below_period means to remove all audio data after silence  is
		 detected.   Normally, this will be a value 1 of but it can be
		 increased to skip over periods of silence  that  are  wanted.
		 For  example, if you have a song with 2 seconds of silence in
		 the  middle  and  2  second  at  the  end,  you   could   set
		 below_period  to a value of 2 to skip over the silence in the
		 middle of the audio file.

		 For below_periods, duration specifies	a  period  of  silence
		 that must exist before data is not copied any more.  By spec‐
		 ifying a higher duration, silence that is wanted can be  left
		 in  the  audio.   For	example,  if  you  have a song with an
		 expected 1 second of silence in the middle and 2  seconds  of
		 silence  at the end, a duration of 2 seconds could be used to
		 skip over the middle silence.

		 Unfortunetly, you must know the length of the silence at  the
		 end  of your audio file to trim off silence reliably.	A work
		 around is to use the silence effect in combination  with  the
		 reverse  effect.   By	first reversing the audio, you can use
		 the above_periods to reliably trim all audio from what	 looks
		 like  the  front of the file.	Then reverse the file again to
		 get back to normal.

		 To remove silence from	 the  middle  of  a  file,  specify  a
		 below_periods	that  is negative.  This value is then treated
		 as a positive value and is also used to indicate  the	effect
		 should	 restart processing as specified by the above_periods,
		 making it suitable for removing periods  of  silence  in  the
		 middle of the sound file.

		 The  period  counts are in units of samples.  Duration counts
		 may be in the format of hh:mm:ss.frac, or the exact count  of
		 samples.   Threshold  numbers may be suffixed iwth d, or % to
		 indicate the value is in decibels or a percentage of  maximum
		 value	 of  the  sample  value	 (0%  specifies	 pure  digital
		 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 doubles and	pitch  is  one
		 octave	 lower.	 If the optional -c parameter is used then the
		 factor is specified in "cents".

       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 Adjustment:" 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 duration.	 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 specified.  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 samples 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 parameter 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	volume
		 to  increase.	Beware of clipping audio data when the gain is
		 greater then 1.0.  A negative value performs the same adjust‐
		 ment 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  logarithmically.
		 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 prevent 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@users.sourceforge.net)

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 ques‐
       tions 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)
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