sox



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 ] [ -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 then 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

       Format options:

       Format  options effect the audio samples that they immediately 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 output 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 specifying  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.

       -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 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 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.

       -V        Print  a description of processing phases.  Useful for figur-
                 ing 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 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 spec-
                 ified for an input file. If any are, they will  override  the
                 file  header, and you will be warned to this effect.  You had
                 better know what you are doing! Output  format  options  will
                 cause  a  format conversion, and the .wav will written appro-
                 priately.  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 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-fire-fR, or to stdout if no profile-file is
                 specified.  If there is sound output on stdout then the  pro-
                 file will next 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://eakaw2.et.tu-dresden.de/~wilde/resam-
                 ple/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 or end of a sound file.
                 Silence is anything below a specified 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 spec-
                 ify  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.  If below_periods
                 is  negative,  it  is 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 remov-
                 ing periods of silence in the middle of a sound 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 indicated 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 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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