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       sox infile outfile

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

       soxmix infile1 infile2 outfile

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

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

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

           avg [ -l | -r | -f | -b | n,n,...,n ]
           band [ -n ] center [ width ]
           bandpass frequency bandwidth
           bandreject frequency bandwidth
           chorus gain-in gain out delay decay speed depth
                  -s | -t [ delay decay speed depth -s | -t ]
           compand attack1,decay1[,attack2,decay2...]
                   [ gain [ initial-volume [ delay ] ] ]
           dcshift shift [ limitergain ]
           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
           pan direction
           phaser gain-in gain-out delay decay speed < -s | -t >
           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 ] ]


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

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

       There are two types of audio files formats  that  SoX  can
       work  with.   The  first are self-describing file formats.
       These contain a header that completely describe the  char­
       acteristics of the audio data that follows.

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

       Audio data can usually be totally described by four  char­

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

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

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

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

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

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

               soxmix music.wav voice.wav mixed.wav

       Format options:

       Format  options effect the audio samples that they immedi­
       ately precede.  If they are placed before the  input  file
       name  then they effect the input data.  If they are placed
       before the output file name then they will effect the out­
       put data.  By taking advantage of this, you can override a
       input file's corrupted header or produce  an  output  file
       that  is  totally different style then the input file.  It
       is also how SoX is informed about the format of raw  input

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

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

                 The sample data encoding is signed  linear  (2's
                 complement),  unsigned  linear, u-law (logarith­
                 mic),  A-law  (logarithmic),  ADPCM,  IMA_ADPCM,
                 GSM, or Floating-point.
                 U-law  (actually shorthand for mu-law) and A-law
                 are the U.S.  and  international  standards  for
                 logarithmic  telephone  sound compression.  When
                 uncompressed u-law has roughly the precision  of
                 14-byte PCM audio and A-law has roughly the pre­
                 cision of 13-bit PCM audio.
                 A-law and u-law data is sometimes encoded  using
                 a  reversed  bit-ordering (ie. MSB becomes LSB).
                 Internally, SoX understands  how  to  work  with
                 this  encoding but there is currently no command
                 line option to specify it.   If  you  need  this
                 support  then  you can use the psuedo file types
                 of ".la" and ".lu" to inform sox of  the  encod­
                 ing.  See supported file types for more informa­
                 Microsoft's  flavor of ADPCM.  IMA ADPCM is also
                 called DVI ADPCM.
                 GSM is a standard used for telephone sound  com­
                 pression  in  European countries and its gaining
                 popularity because of its quality.   It  usually
                 is CPU intensive to work with GSM audio data.

                 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 number  of  channels  then  the  avg
                 effect  must  be used.  If the avg effect is not
                 specified on the command line it will be invoked
                 internally with default parameters.

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

       General options:

       -h        Print version number and usage information.

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

       -v volume Change amplitude (floating point); less than 1.0
                 decreases,  greater than 1.0 increases.  May use

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

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

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

       The following file formats are supported:

       .8svx     Amiga 8SVX musical instrument  description  for­

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

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

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

       .cdr      CD-R
                 CD-R  files  are used in mastering music on Com­
                 pact Disks.  The audio data on a CD-R disk is  a
                 raw  audio  file  with a format of stereo 16-bit
                 signed samples at a 44khz sample rate.  There is
                 a  special blocking/padding oddity at the end of
                 the audio file and is why it needs its own  han­
                 files for external programs such as FFT  analyz­
                 ers  or  graph routines.  SoX can also convert a
                 file in this format back into one of  the  other
                 file formats.

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

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

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

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

       .nul      Null file handler.  This is a fake  file  hander
                 that  act as if its reading a stream of 0's from
                 a while or fake writing output to a file.   This
                 is not a very useful file handler in most cases.
                 It might be useful in some scripts were  you  do
                 if  there  is  support for Ogg Vorbis run sox -h
                 and look for it under the list of supported file
                 formats as "vorbis".

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

       .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.

                 SPHERE  (SPeech HEader Resources) is a file for­
                 mat defined by NIST (National Institute of Stan­
                 dards  and  Technology)  and is used with speech
                 audio.  SoX can read these files when they  con­
                 tain  u-law  and  PCM  data.  It will ignore any
                 header information that says the  data  is  com­
                 pressed using shorten compression and will treat
                 the data as either  u-law  or  PCM.   This  will
                 allow  SoX  and the command line shorten program
                 to be ran together using pipes to uncompress the
                 data  and  then  pass the result to SoX for pro­

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

                 data  type  as  passed  in to SoX.  It works for
                 both playing and recording sound samples.   When
                 playing  sound  files  it attempts to set up the
                 audio driver to use the same format as the input
                 file.   It  is  suggested to always override the
                 output values to use the highest quality samples
                 your  hardware can handle.  Example: -t sunau -w
                 -s /dev/audio or -t sunau -U -c 1 /dev/audio for
                 older sun equipment.

       .txw      Yamaha TX-16W sampler.
                 A  file  format  from a Yamaha sampling keyboard
                 which wrote IBM-PC format 3.5"  floppies.   Han­
                 dles reading of files which do not have the sam­
                 ple rate field set to one  of  the  expected  by
                 looking  at  some other bytes in the attack/loop
                 length fields, and defaulting to  33kHz  if  the
                 sample rate is still unknown.

       .vms      More info to come.
                 Used  to  compress speech audio for applications
                 such as voice mail.

       .voc      Sound Blaster VOC files.
                 VOC files are  multi-part  and  contain  silence
                 parts,  looping,  and different sample rates for
                 different chunks.  On input, the  silence  parts
                 are  filled  out, loops are rejected, and sample
                 data  with  a  new  sample  rate  is   rejected.
                 Silence  with  a different sample rate is gener­
                 ated appropriately.  On output, silence  is  not
                 detected,   nor  are  impossible  sample  rates.
                 Note, this  version  now  supports  playing  VOC
                 files  with multiple blocks and supports playing
                 files containing u-law and A-law samples.

       vorbis    See .ogg format.

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

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

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


       Multiple effects may be applied to the audio data by spec­
       ifying  them  one  after another at the end of the command

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

                 The avg effect can also be invoked with up to 16
                 double-precision numbers, which specify the pro­
                 portion of each input  channel  that  is  to  be
                 mixed  into each output channel.  In two-channel
                 mode, 4 numbers are given: l->l, l->r, r->l, and
                 Apply   a   band-pass   filter.   The  frequency
                 response drops logarithmically around the center
                 frequency.   The  width  gives  the slope of the
                 drop.  The frequencies at  center  +  width  and
                 center  -  width  will be half of their original
                 amplitudes.  Band defaults to a mode oriented to
                 pitched signals, i.e. voice, singing, or instru­
                 mental music.  The -n (for  noise)  option  uses
                 the   alternate  mode  for  un-pitched  signals.
                 Warning: -n introduces  a  power-gain  of  about
                 11dB  in  the  filter, so beware of output clip­
                 ping.  Band introduces noise in the shape of the
                 filter, i.e. peaking at the center frequency and
                 settling around it.  See filter for  a  bandpass
                 effect with steeper shoulders.

       bandpass frequency bandwidth
                 Butterworth  bandpass filter. Description coming

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

       chorus gain-in gain-out delay decay speed depth

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

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


               [gain [initial-volume [delay ] ] ]
                 Compand  (compress  or expand) the dynamic range
                 of a sample.  The attack and decay time  specify
                 the  integration  time  over  which the absolute
                 value of  the  input  signal  is  integrated  to
                 determine its volume; attacks refer to increases
                 in volume and decays refer to decreases.   Where
                 more  than  one  pair of attack/decay parameters
                 are specified, each  channel  is  treated  sepa­
                 rately  and  the number of pairs must agree with
                 the number of input channels.  The second param­
                 eter  is  a  list  of  points on the compander's
                 transfer function specified in  dB  relative  to
                 This  permits the user to supply a nominal level
                 initially, so that, for example,  a  very  large
                 gain  is  not  applied  to initial signal levels
                 before the companding action has begun to  oper­
                 ate: it is quite probable that in such an event,
                 the output would be severely clipped  while  the
                 compander gain properly adjusts itself.

                 The  fifth  (optional)  parameter  is a delay in
                 seconds.  The input signal is  analyzed  immedi­
                 ately  to  control  the  compander,  but  it  is
                 delayed before being fed to the volume adjuster.
                 Specifying  a  delay  approximately equal to the
                 attack/decay  times  allows  the  compander   to
                 effectively  operate  in  a  "predictive" rather
                 than a reactive mode.

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

       dcshift shift [ limitergain ]
                 DC Shift  the  audio  data,  with  basic  linear
                 amplitude  formula.  This is most useful if your
                 audio data tends to not  be  centered  around  a
                 value  of 0.  Shifting it back will allow you to
                 get the most volume adjustments without clipping
                 audio data.
                 The  first option is the dcshift value.  It is a
                 floating point number that indicates the  amount
                 to shift.
                 An  option  limtergain value can be specified as
                 well.  It should have a value much less then 1.0
                 and is used only on peaks to prevent clipping.

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

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

       echo gain-in gain-out delay decay [ delay decay ... ]
                 Add echoing to a sound sample.  Each delay/decay
                 of the audio data.

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

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

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

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

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

                 The beta, if unspecified, defaults to 16.   This
                 selects  a Kaiser window.  You can select a Nut­
                 tall window by specifying anything <= 2.0  here.
                 For  more  discussion  of  beta,  look under the
                 resample effect.

       flanger gain-in gain-out delay decay speed < -s | -t >
                 Add a flanger to a sound  sample.   Each  triple
                 delay/decay/speed  gives  the delay in millisec­

       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

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

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

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

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

       pick [ -1 | -2 | -3 | -4 | -l | -r ]
                 Select  the  left  or  right channel of a stereo
                 sample, or one of four  channels  in  a  quadra­
                 be "cubic" or "linear". Default is "cubic".  The
                 fade  option,  can be "cos", "hamming", "linear"
                 or "trapezoid".  Default is "cos".

       polyphase [ -w < nut / ham > ]

                 [  -width <  long  / short  / # > ]

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

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

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

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

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

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

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

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

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

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

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

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

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

                 The  rolloff  refers to how close to the Nyquist
                 beta for a Kaiser window.  Beta <= 2.0 selects a
                 Nuttall  window.  If unspecified, the default is
                 a Kaiser window with beta 16.

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

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

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

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

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

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

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

       silence above_periods [ duration threshold[ d | % ]

               [ below_periods duration

                 threshold[ d | % ]]
                 Removes silence from the beginning or end  of  a
                 periods  that occur below the threshold may also
                 be specified.  If this options are not specified
                 then  data  is  not  trimmed from the end of the
                 audio file.
                 Duration counts may be in the  format  of  time,
                 hh:mm:ss.frac, or in the exact count of samples.
                 Threshold may be suffixed with d, or % to  indi­
                 cated  the  value is in decibels or a percentage
                 of max value of the sample value.   A  value  of
                 '0%' will look for total silence.

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

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

       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

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

                 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.

                 [0.0 1.0]. Default depends  on  stretch  factor.
                 1.0  to  shorten,  0.8  to lengthen.  The fading
                 ratio, in [0.0 0.5].  The  amount  of  a  fade's
                 default depends on factor and shift.

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

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

             [ off ] [ ph ] [ p1 ] [ p2 ] [ p3 ]
                 The  synth effect will generate various types of
                 audio data.  Although this  effect  is  used  to
                 generate audio data, an input file must be spec­
                 ified.  The  length  of  the  input  audio  file
                 determines  the length of the output audio file.
                 <length>  length  in   sec   or   hh:mm:ss.frac,
                 0=inputlength, default=0
                 <type>  is  sine,  square,  triangle,  sawtooth,
                 trapetz, exp, whitenoise, pinknoise, brownnoise,
                 <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
                 <off>  Bias  (DC-offset)   of signal in percent,
                 <ph> phase shift 0..100 shift phase 0..2*Pi, not
                 used for noise..
                 <p1>  square: Ton/Toff, triangle+trapetz: rising
                 slope time (0..100)
                 <p2> trapetz: ON time (0..100)
                 <p3> trapetz: falling slope position (0..100)

       trim start [ length ]
                 Trim can trim off unwanted audio data  from  the
                 beginning and end of the audio file.  Audio sam­
                 ples are not sent to the output stream until the
                 start location is reached.
                 starting to process audio data.

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

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


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

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



       play(1), rec(1), soxexam(1)


       The  version  of  SoX that accompanies this manual page is

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