SoX - Sound eXchange, the Swiss Army knife of audio manipulation


   sox [global-options] [format-options] infile1
        [[format-options] infile2] ... [format-options] outfile
        [effect [effect-options]] ...

   play [global-options] [format-options] infile1
        [[format-options] infile2] ... [format-options]
        [effect [effect-options]] ...

   rec [global-options] [format-options] outfile
        [effect [effect-options]] ...


   SoX  reads  and  writes  audio  files  in  most popular formats and can
   optionally apply  effects  to  them.  It  can  combine  multiple  input
   sources,  synthesise  audio,  and,  on  many  systems, act as a general
   purpose audio player or a  multi-track  audio  recorder.  It  also  has
   limited ability to split the input into multiple output files.

   All  SoX  functionality  is  available  using just the sox command.  To
   simplify playing and recording audio, if SoX is invoked  as  play,  the
   output file is automatically set to be the default sound device, and if
   invoked as rec, the default sound device is used as  an  input  source.
   Additionally,  the  soxi(1)  command  provides a convenient way to just
   query audio file header information.

   The heart of SoX is a  library  called  libSoX.   Those  interested  in
   extending  SoX or using it in other programs should refer to the libSoX
   manual page: libsox(3).

   SoX is a command-line audio processing  tool,  particularly  suited  to
   making  quick,  simple  edits  and to batch processing.  If you need an
   interactive, graphical audio editor, use audacity(1).

                             *        *        *

   The overall SoX processing chain can be summarised as follows:

                  Input(s) → Combiner → Effects → Output(s)

   Note however, that on the  SoX  command  line,  the  positions  of  the
   Output(s)  and  the  Effects  are  swapped w.r.t. the logical flow just
   shown.  Note also that whilst options pertaining to  files  are  placed
   before  their  respective  file name, the opposite is true for effects.
   To show how this works in practice, here is a selection of examples  of
   how SoX might be used.  The simple
      sox recital.wav
   translates  an  audio  file  in  Sun AU format to a Microsoft WAV file,
      sox -b 16 recital.wav channels 1 rate 16k fade 3 norm
   performs the same format translation, but  also  applies  four  effects
   (down-mix  to  one channel, sample rate change, fade-in, nomalize), and
   stores the result at a bit-depth of 16.
      sox -r 16k -e signed -b 8 -c 1 voice-memo.raw voice-memo.wav
   converts `raw' (a.k.a. `headerless') audio to  a  self-describing  file
      sox slow.aiff fixed.aiff speed 1.027
   adjusts audio speed,
      sox short.wav long.wav longer.wav
   concatenates two audio files, and
      sox -m music.mp3 voice.wav mixed.flac
   mixes together two audio files.
      play "The Moonbeams/Greatest/*.ogg" bass +3
   plays  a  collection  of  audio  files  whilst applying a bass boosting
      play -n -c1 synth sin %-12 sin %-9 sin %-5 sin %-2 fade h 0.1 1 0.1
   plays a synthesised `A minor seventh' chord with a pipe-organ sound,
      rec -c 2 radio.aiff trim 0 30:00
   records half an hour of stereo audio, and
      play -q take1.aiff & rec -M take1.aiff take1-dub.aiff
   (with POSIX shell and where supported by hardware) records a new  track
   in a multi-track recording.  Finally,
      rec -r 44100 -b 16 -s -p silence 1 0.50 0.1% 1 10:00 0.1% | \
        sox -p song.ogg silence 1 0.50 0.1% 1 2.0 0.1% : \
        newfile : restart
   records a stream of audio such as LP/cassette and splits in to multiple
   audio files at points with 2 seconds of silence.   Also,  it  does  not
   start  recording  until  it detects audio is playing and stops after it
   sees 10 minutes of silence.

   N.B.  The above is just an overview  of  SoX's  capabilities;  detailed
   explanations  of  how  to  use  all  SoX  parameters, file formats, and
   effects can be found below in this  manual,  in  soxformat(7),  and  in

   File Format Types
   SoX  can  work  with  `self-describing'  and `raw' audio files.  `self-
   describing' formats (e.g. WAV, FLAC, MP3) have a header that completely
   describes  the  signal  and  encoding attributes of the audio data that
   follows. `raw' or `headerless' formats do not contain this information,
   so  the  audio  characteristics  of  these must be described on the SoX
   command line or inferred from those of the input file.

   The following four characteristics are used to describe the  format  of
   audio data such that it can be processed with SoX:

   sample rate
          The  sample  rate  in  samples  per  second  (`Hertz'  or `Hz').
          Digital telephony traditionally uses a sample  rate  of  8000 Hz
          (8 kHz), though these days, 16 and even 32 kHz are becoming more
          common. Audio Compact Discs  use  44100 Hz  (44.1 kHz).  Digital
          Audio  Tape  and  many computer systems use 48 kHz. Professional
          audio systems often use 96 kHz.

   sample size
          The number of bits used to store each sample.  Today, 16-bit  is
          commonly  used.  8-bit was popular in the early days of computer
          audio. 24-bit is used in the  professional  audio  arena.  Other
          sizes are also used.

   data encoding
          The   way   in  which  each  audio  sample  is  represented  (or
          `encoded').  Some encodings have variants with  different  byte-
          orderings  or  bit-orderings.   Some  compress the audio data so
          that the stored audio data takes up less space (i.e. disk  space
          or  transmission bandwidth) than the other format parameters and
          the number of samples would imply.  Commonly-used encoding types
          include  floating-point,  μ-law, ADPCM, signed-integer PCM, MP3,
          and FLAC.

          The number  of  audio  channels  contained  in  the  file.   One
          (`mono')  and  two (`stereo') are widely used.  `Surround sound'
          audio typically contains six or more channels.

   The term `bit-rate' is a measure of the amount of storage  occupied  by
   an  encoded  audio signal over a unit of time.  It can depend on all of
   the above and is typically denoted as a number of kilo-bits per  second
   (kbps).   An  A-law  telephony  signal  has  a  bit-rate  of  64  kbps.
   MP3-encoded stereo music typically has  a  bit-rate  of  128-196  kbps.
   FLAC-encoded stereo music typically has a bit-rate of 550-760 kbps.

   Most  self-describing  formats  also  allow  textual  `comments'  to be
   embedded in the file that can be used to describe  the  audio  in  some
   way, e.g. for music, the title, the author, etc.

   One  important  use  of  audio file comments is to convey `Replay Gain'
   information.  SoX supports applying Replay Gain  information,  but  not
   generating it.  Note that by default, SoX copies input file comments to
   output files that support comments, so output files may contain  Replay
   Gain  information if some was present in the input file.  In this case,
   if anything other than a simple format conversion  was  performed  then
   the  output  file Replay Gain information is likely to be incorrect and
   so should be recalculated using a tool that supports this (not SoX).

   The soxi(1) command can be used to display information from audio  file

   Determining & Setting The File Format
   There  are  several mechanisms available for SoX to use to determine or
   set the format characteristics of an  audio  file.   Depending  on  the
   circumstances,  individual  characteristics  may  be  determined or set
   using different mechanisms.

   To determine the format of an input file, SoX will  use,  in  order  of
   precedence and as given or available:

   1.  Command-line format options.

   2.  The contents of the file header.

   3.  The filename extension.

   To set the output file format, SoX will use, in order of precedence and
   as given or available:

   1.  Command-line format options.

   2.  The filename extension.

   3.  The input file format  characteristics,  or  the  closest  that  is
       supported by the output file type.

   For  all  files, SoX will exit with an error if the file type cannot be
   determined. Command-line format options may need to be added or changed
   to resolve the problem.

   Playing & Recording Audio
   The  play  and  rec  commands  are  provided  so that basic playing and
   recording is as simple as
      play existing-file.wav
      rec new-file.wav
   These two commands are functionally equivalent to
      sox existing-file.wav -d
      sox -d new-file.wav
   Of course, further options and effects  (as  described  below)  can  be
   added to the commands in either form.

                             *        *        *

   Some  systems  provide  more  than  one  type of (SoX-compatible) audio
   driver, e.g. ALSA & OSS, or SUNAU & AO.  Systems  can  also  have  more
   than  one  audio  device (a.k.a. `sound card').  If more than one audio
   driver has been built-in to SoX, and the default selected by  SoX  when
   recording  or  playing  is  not  the  one  that  is  wanted,  then  the
   AUDIODRIVER environment variable can be used to override  the  default.
   For example (on many systems):
      set AUDIODRIVER=oss
      play ...
   The  AUDIODEV  environment variable can be used to override the default
   audio device, e.g.
      set AUDIODEV=/dev/dsp2
      play ...
      sox ... -t oss
      set AUDIODEV=hw:soundwave,1,2
      play ...
      sox ... -t alsa
   Note that the way of setting environment variables varies  from  system
   to system - for some specific examples, see `SOX_OPTS' below.

   When  playing  a  file  with a sample rate that is not supported by the
   audio output device, SoX will automatically invoke the rate  effect  to
   perform  the  necessary sample rate conversion.  For compatibility with
   old hardware, the default rate quality level is set to `low'. This  can
   be  changed  by  explicitly specifying the rate effect with a different
   quality level, e.g.
      play ... rate -m
   or by using the --play-rate-arg option (see below).

                             *        *        *

   On some systems, SoX allows audio playback volume to be adjusted whilst
   using play.  Where supported, this is achieved by tapping the `v' & `V'
   keys during playback.

   To help with setting a suitable recording level, SoX includes  a  peak-
   level  meter  which can be invoked (before making the actual recording)
   as follows:
      rec -n
   The recording level should be adjusted (using the system-provided mixer
   program, not SoX) so that the meter is at most occasionally full scale,
   and never `in the red' (an exclamation mark is  shown).   See  also  -S

   Many  file formats that compress audio discard some of the audio signal
   information whilst doing so. Converting  to  such  a  format  and  then
   converting  back  again  will not produce an exact copy of the original
   audio.  This is the case for many formats used in  telephony  (e.g.  A-
   law,  GSM) where low signal bandwidth is more important than high audio
   fidelity, and for many formats used in  portable  music  players  (e.g.
   MP3,  Vorbis)  where  adequate  fidelity  can be retained even with the
   large compression ratios that  are  needed  to  make  portable  players

   Formats  that  discard  audio  signal  information  are called `lossy'.
   Formats that do not are called `lossless'.  The term `quality' is  used
   as a measure of how closely the original audio signal can be reproduced
   when using a lossy format.

   Audio file conversion with SoX is lossless when it can  be,  i.e.  when
   not  using  lossy  compression,  when not reducing the sampling rate or
   number of channels, and when the number of bits used in the destination
   format is not less than in the source format.  E.g.  converting from an
   8-bit PCM format to a 16-bit PCM format is lossless but converting from
   an 8-bit PCM format to (8-bit) A-law isn't.

   N.B.   SoX  converts all audio files to an internal uncompressed format
   before performing any audio processing. This means that manipulating  a
   file that is stored in a lossy format can cause further losses in audio
   fidelity.  E.g. with
      sox long.mp3 short.mp3 trim 10
   SoX first decompresses the  input  MP3  file,  then  applies  the  trim
   effect,  and  finally creates the output MP3 file by re-compressing the
   audio - with a possible reduction in fidelity above that which occurred
   when  the input file was created.  Hence, if what is ultimately desired
   is lossily compressed audio, it is highly recommended  to  perform  all
   audio  processing  using  lossless file formats and then convert to the
   lossy format only at the final stage.

   N.B.  Applying multiple effects with a single SoX invocation  will,  in
   general,  produce  more  accurate  results  than  those  produced using
   multiple SoX invocations.

   Dithering is a technique used to maximise the dynamic  range  of  audio
   stored   at  a  particular  bit-depth.  Any  distortion  introduced  by
   quantisation is decorrelated by adding a small amount of white noise to
   the  signal.   In  most  cases,  SoX can determine whether the selected
   processing requires dither and will add it during output formatting  if

   Specifically,  by  default, SoX automatically adds TPDF dither when the
   output bit-depth is less than 24 and any of the following are true:

   ·   bit-depth reduction has been specified explicitly using a  command-
       line option

   ·   the  output file format supports only bit-depths lower than that of
       the input file format

   ·   an effect has increased effective  bit-depth  within  the  internal
       processing chain

   For  example,  adjusting  volume  with vol 0.25 requires two additional
   bits in which to losslessly  store  its  results  (since  0.25  decimal
   equals  0.01 binary).  So if the input file bit-depth is 16, then SoX's
   internal representation will utilise  18  bits  after  processing  this
   volume  change.   In order to store the output at the same depth as the
   input, dithering is used to remove the additional bits.

   Use the -V option to see what processing SoX has  automatically  added.
   The  -D option may be given to override automatic dithering.  To invoke
   dithering manually (e.g. to select  a  noise-shaping  curve),  see  the
   dither effect.

   Clipping  is  distortion  that  occurs  when  an audio signal level (or
   `volume') exceeds the range of  the  chosen  representation.   In  most
   cases,  clipping is undesirable and so should be corrected by adjusting
   the level prior to the point (in the  processing  chain)  at  which  it

   In  SoX,  clipping could occur, as you might expect, when using the vol
   or gain effects to increase the audio volume. Clipping could also occur
   with  many  other  effects,  when converting one format to another, and
   even when simply playing the audio.

   Playing an audio file often  involves  resampling,  and  processing  by
   analogue   components   can   introduce   a   small  DC  offset  and/or
   amplification, all of which can produce distortion if the audio  signal
   level was initially too close to the clipping point.

   For these reasons, it is usual to make sure that an audio file's signal
   level has some `headroom', i.e. it does not exceed a  particular  level
   below  the  maximum  possible level for the given representation.  Some
   standards bodies recommend as much as 9dB headroom, but in most  cases,
   3dB (≈ 70% linear) is enough.  Note that this wisdom seems to have been
   lost in modern music production; in fact, many CDs, MP3s, etc.  are now
   mastered at levels above 0dBFS i.e. the audio is clipped as delivered.

   SoX's stat and stats effects can assist in determining the signal level
   in an audio file. The gain  or  vol  effect  can  be  used  to  prevent
   clipping, e.g.
      sox dull.wav bright.wav gain -6 treble +6
   guarantees that the treble boost will not clip.

   If  clipping  occurs at any point during processing, SoX will display a
   warning message to that effect.

   See also -G and the gain and norm effects.

   Input File Combining
   SoX's input combiner can be configured (see OPTIONS below)  to  combine
   multiple  files  using  any  of  the  following methods: `concatenate',
   `sequence', `mix', `mix-power', `merge', or  `multiply'.   The  default
   method is `sequence' for play, and `concatenate' for rec and sox.

   For  all  methods other than `sequence', multiple input files must have
   the same sampling rate. If necessary, separate SoX invocations  can  be
   used to make sampling rate adjustments prior to combining.

   If  the  `concatenate' combining method is selected (usually, this will
   be by default) then the input files must also have the same  number  of
   channels.   The audio from each input will be concatenated in the order
   given to form the output file.

   The `sequence' combining method is selected automatically for play.  It
   is  similar  to `concatenate' in that the audio from each input file is
   sent serially to the output file. However, here the output file may  be
   closed  and  reopened  at  the  corresponding  transition between input
   files. This may be just what is needed when sending different types  of
   audio  to an output device, but is not generally useful when the output
   is a normal file.

   If either the `mix' or `mix-power' combining method  is  selected  then
   two  or  more  input  files must be given and will be mixed together to
   form the output file.  The number of channels in each input  file  need
   not  be the same, but SoX will issue a warning if they are not and some
   channels in the output file will not contain  audio  from  every  input
   file.   A  mixed audio file cannot be un-mixed without reference to the
   original input files.

   If the `merge' combining method is selected  then  two  or  more  input
   files  must  be  given  and  will be merged together to form the output
   file.  The number of channels in each input file need not be the  same.
   A merged audio file comprises all of the channels from all of the input
   files. Un-merging is possible using multiple invocations  of  SoX  with
   the  remix effect.  For example, two mono files could be merged to form
   one stereo file. The first and second mono files would become the  left
   and right channels of the stereo file.

   The  `multiply'  combining  method  multiplies  the  sample  values  of
   corresponding channels (treated as numbers in the interval -1  to  +1).
   If  the  number  of  channels  in  the input files is not the same, the
   missing channels are considered to contain all zero.

   When  combining  input  files,  SoX  applies  any   specified   effects
   (including,  for  example,  the vol volume adjustment effect) after the
   audio has been combined. However, it is often useful to be able to  set
   the   volume  of  (i.e.  `balance')  the  inputs  individually,  before
   combining takes place.

   For all combining methods, input file volume adjustments  can  be  made
   manually using the -v option (below) which can be given for one or more
   input files. If it is given for only some of the input files  then  the
   others  receive no volume adjustment.  In some circumstances, automatic
   volume adjustments may be applied (see below).

   The -V option (below) can  be  used  to  show  the  input  file  volume
   adjustments that have been selected (either manually or automatically).

   There  are  some  special  considerations that need to made when mixing
   input files:

   Unlike the other methods, `mix' combining has the  potential  to  cause
   clipping  in  the combiner if no balancing is performed.  In this case,
   if manual volume adjustments are not given, SoX will try to ensure that
   clipping   does   not  occur  by  automatically  adjusting  the  volume
   (amplitude) of each input signal by a factor of ¹/n,  where  n  is  the
   number  of  input files.  If this results in audio that is too quiet or
   otherwise unbalanced then the input file volumes can be set manually as
   described   above.  Using  the  norm  effect  on  the  mix  is  another

   If mixed audio seems loud enough at some points but too quiet in others
   then  dynamic range compression should be applied to correct this - see
   the compand effect.

   With the `mix-power' combine method, the mixed volume is  approximately
   equal  to  that  of  one  of  the  input  signals.  This is achieved by
   balancing using a factor of  ¹/√n  instead  of  ¹/n.   Note  that  this
   balancing  factor  does not guarantee that clipping will not occur, but
   the number of clips will usually be low and the resultant distortion is
   generally imperceptible.

   Output Files
   SoX's  default  behaviour  is to take one or more input files and write
   them to a single output file.

   This behaviour can be changed by specifying the pseudo-effect `newfile'
   within the effects list.  SoX will then enter multiple output mode.

   In  multiple  output mode, a new file is created when the effects prior
   to the `newfile' indicate they are  done.   The  effects  chain  listed
   after  `newfile'  is then started up and its output is saved to the new

   In multiple output mode, a unique number will automatically be appended
   to the end of all filenames.  If the filename has an extension then the
   number is  inserted  before  the  extension.   This  behaviour  can  be
   customized  by  placing  a %n anywhere in the filename where the number
   should be substituted.  An optional number can be placed after the % to
   indicate a minimum fixed width for the number.

   Multiple output mode is not very useful unless an effect that will stop
   the effects chain early is specified before the `newfile'.  If  end  of
   file  is reached before the effects chain stops itself then no new file
   will be created as it would be empty.

   The following is an example of splitting the first  60  seconds  of  an
   input file into two 30 second files and ignoring the rest.
      sox song.wav ringtone%1n.wav trim 0 30 : newfile : trim 0 30

   Stopping SoX
   Usually SoX will complete its processing and exit automatically once it
   has read all available audio data from the input files.

   If desired, it can be terminated earlier by sending an interrupt signal
   to the process (usually by pressing the keyboard interrupt key which is
   normally Ctrl-C).  This is a natural requirement in some circumstances,
   e.g.  when  using SoX to make a recording.  Note that when using SoX to
   play multiple files, Ctrl-C behaves slightly differently:  pressing  it
   once  causes  SoX  to skip to the next file; pressing it twice in quick
   succession causes SoX to exit.

   Another option to stop processing early is to use an effect that has  a
   time  period  or sample count to determine the stopping point. The trim
   effect is an example of this.  Once all  effects  chains  have  stopped
   then SoX will also stop.


   Filenames can be simple file names, absolute or relative path names, or
   URLs (input files only).  Note that URL support requires  that  wget(1)
   is available.

   Note:  Giving SoX an input or output filename that is the same as a SoX
   effect-name will not  work  since  SoX  will  treat  it  as  an  effect
   specification.    The  only  work-around  to  this  is  to  avoid  such
   filenames. This is generally not difficult since most  audio  filenames
   have a filename `extension', whilst effect-names do not.

   Special Filenames
   The following special filenames may be used in certain circumstances in
   place of a normal filename on the command line:

   -      SoX can be used in  simple  pipeline  operations  by  using  the
          special  filename  `-' which, if used as an input filename, will
          cause SoX will read audio data from  `standard  input'  (stdin),
          and  which,  if used as the output filename, will cause SoX will
          send audio data to `standard output' (stdout).  Note  that  when
          using  this option for the output file, and sometimes when using
          it for an input file, the file-type (see -t below) must also  be

   "|program [options] ..."
          This  can  be  used in place of an input filename to specify the
          the given program's standard output (stdout) be used as an input
          file.   Unlike - (above), this can be used for several inputs to
          one SoX command.  For example,  if  `genw'  generates  mono  WAV
          formatted  signals  to  its  standard output, then the following
          command makes a stereo file from two generated signals:
             sox -M "|genw --imd -" "|genw --thd -" out.wav
          For  headerless  (raw)  audio,  -t  (and  perhaps  other  format
          options) will need to be given, preceding the input command.

          Specifies  that  filename `globbing' (wild-card matching) should
          be performed by SoX instead of by  the  shell.   This  allows  a
          single  set  of  file options to be applied to a group of files.
          For example, if  the  current  directory  contains  three  `vox'
          files, file1.vox, file2.vox, and file3.vox, then
             play --rate 6k *.vox
          will be expanded by the `shell' (in most environments) to
             play --rate 6k file1.vox file2.vox file3.vox
          which will treat only the first vox file as having a sample rate
          of 6k.  With
             play --rate 6k "*.vox"
          the given sample rate option will be applied to  all  three  vox

   -p, --sox-pipe
          This  can be used in place of an output filename to specify that
          the SoX command should be used as in input pipe to  another  SoX
          command.  For example, the command:
             play "|sox -n -p synth 2" "|sox -n -p synth 2 tremolo 10" stat
          plays two `files' in succession, each with different effects.

          -p is in fact an alias for `-t sox -'.

   -d, --default-device
          This  can  be  used  in  place of an input or output filename to
          specify that the default audio device (if  one  has  been  built
          into  SoX)  is to be used.  This is akin to invoking rec or play
          (as described above).

   -n, --null
          This can be used in place of an  input  or  output  filename  to
          specify that a `null file' is to be used.  Note that here, `null
          file' refers to a SoX-specific mechanism and is not  related  to
          any operating-system mechanism with a similar name.

          Using a null file to input audio is equivalent to using a normal
          audio file that contains an infinite amount of silence,  and  as
          such  is  not  generally  useful unless used with an effect that
          specifies a finite time length (such as trim or synth).

          Using a null file to output  audio  amounts  to  discarding  the
          audio and is useful mainly with effects that produce information
          about the audio instead of affecting it (such  as  noiseprof  or

          The  sampling  rate  associated  with  a null file is by default
          48 kHz, but, as with a normal file, this can  be  overridden  if
          desired using command-line format options (see below).

   Supported File & Audio Device Types
   See  soxformat(7)  for  a  list  and  description of the supported file
   formats and audio device drivers.


   Global Options
   These options can be specified on the command line at any point  before
   the first effect name.

   The  SOX_OPTS  environment  variable can be used to provide alternative
   default values for SoX's global options.  For example:
      SOX_OPTS="--buffer 20000 --play-rate-arg -hs --temp /mnt/temp"
   Note that setting SOX_OPTS can potentially create unwanted  changes  in
   the  behaviour  of scripts or other programs that invoke SoX.  SOX_OPTS
   might best be used for things (such  as  in  the  given  example)  that
   reflect  the  environment  in which SoX is being run.  Enabling options
   such as --no-clobber as default might be handled better using  a  shell
   alias since a shell alias will not affect operation in scripts etc.

   One  way  to  ensure that a script cannot be affected by SOX_OPTS is to
   clear SOX_OPTS at the start of the script, but this of course loses the
   benefit  of  SOX_OPTS  carrying  some  system-wide default options.  An
   alternative approach is to explicitly invoke SoX  with  default  option
   values, e.g.
      SOX_OPTS="-V --no-clobber"
      sox -V2 --clobber $input $output ...
   Note  that  the  way to set environment variables varies from system to
   system. Here are some examples:

   Unix bash:
      export SOX_OPTS="-V --no-clobber"
   Unix csh:
      setenv SOX_OPTS "-V --no-clobber"
      set SOX_OPTS=-V --no-clobber
   MS-Windows GUI: via Control Panel : System  :  Advanced  :  Environment

   Mac OS X GUI: Refer to Apple's Technical Q&A QA1067 document.

   --buffer BYTES, --input-buffer BYTES
          Set  the  size in bytes of the buffers used for processing audio
          (default 8192).  --buffer applies to input, effects, and  output
          processing; --input-buffer applies only to input processing (for
          which it overrides --buffer if both are given).

          Be aware that large values for --buffer will  cause  SoX  to  be
          become  slow  to respond to requests to terminate or to skip the
          current input file.

          Don't prompt before overwriting an existing file with  the  same
          name  as  that  given  for the output file.  This is the default

   --combine concatenate|merge|mix|mix-power|multiply|sequence
          Select the input file combining method; for some of these, short
          options are available: -m selects `mix', -M selects `merge', and
          -T selects `multiply'.

          See  Input  File  Combining  above  for  a  description  of  the
          different combining methods.

   -D, --no-dither
          Disable automatic dither - see `Dithering' above.  An example of
          why this might occasionally be useful is  if  a  file  has  been
          converted  from  16  to  24 bit with the intention of doing some
          processing on it, but in fact no processing is needed after  all
          and  the  original  16  bit  file  has been lost, then, strictly
          speaking, no dither is needed if converting the file back to  16
          bit.   See also the stats effect for how to determine the actual
          bit depth of the audio within a file.

   --effects-file FILENAME
          Use FILENAME to obtain all effects  and  their  arguments.   The
          file  is  parsed  as if the values were specified on the command
          line.  A new line can be used in place of the special  :  marker
          to separate effect chains.  For convenience, such markers at the
          end of the file are normally ignored; if you want to specify  an
          empty  last  effects  chain,  use an explicit : by itself on the
          last line of the file.  This option causes any effects specified
          on the command line to be discarded.

   -G, --guard
          Automatically  invoke the gain effect to guard against clipping.
             sox -G infile -b 16 outfile rate 44100 dither -s
          is shorthand for
             sox infile -b 16 outfile gain -h rate 44100 gain -rh dither -s
          See also -V, --norm, and the gain effect.

   -h, --help
          Show version number and usage information.

   --help-effect NAME
          Show usage information on the specified effect.   The  name  all
          can be used to show usage on all effects.

   --help-format NAME
          Show  information about the specified file format.  The name all
          can be used to show information on all formats.

   --i, --info
          Only if given as the first parameter to sox, behave as soxi(1).

   -m|-M  Equivalent to --combine mix and --combine merge, respectively.

          If SoX has been built with the optional `libmagic' library  then
          this  option can be given to enable its use in helping to detect
          audio file types.

   --multi-threaded | --single-threaded
          By default, SoX is `single threaded'.  If  the  --multi-threaded
          option is given however then SoX will process audio channels for
          most multi-channel effects in parallel on hyper-threading/multi-
          core  architectures.  This  may  reduce  processing time, though
          sometimes it may be necessary to use this option  in  conjuction
          with  a  larger  buffer  size  than  is  the default to gain any
          benefit  from  multi-threaded  processing  (e.g.   131072;   see
          --buffer above).

          Prompt before overwriting an existing file with the same name as
          that given for the output file.

          N.B.  Unintentionally overwriting a  file  is  easier  than  you
          might think, for example, if you accidentally enter
             sox file1 file2 effect1 effect2 ...
          when what you really meant was
             play file1 file2 effect1 effect2 ...
          then,  without  this  option, file2 will be overwritten.  Hence,
          using this option is recommended. SOX_OPTS  (above),  a  `shell'
          alias,  script,  or  batch  file  may  be  an appropriate way of
          permanently enabling it.

          Automatically invoke the gain effect to guard  against  clipping
          and to normalise the audio. E.g.
             sox --norm infile -b 16 outfile rate 44100 dither -s
          is shorthand for
             sox infile -b 16 outfile gain -h rate 44100 gain -nh dither -s
          Optionally,  the  audio  can  be  normalized  to  a  given level
          (usually) below 0 dBFS:
             sox --norm=-3 infile outfile

          See also -V, -G, and the gain effect.

   --play-rate-arg ARG
          Selects a quality option to be used when the  `rate'  effect  is
          automatically  invoked  whilst  playing  audio.   This option is
          typically set via the SOX_OPTS environment variable (see above).

   --plot gnuplot|octave|off
          If not set to off (the default if --plot is not given), run in a
          mode  that  can be used, in conjunction with the gnuplot program
          or the GNU Octave program, to  assist  with  the  selection  and
          configuration  of  many  of the transfer-function based effects.
          For the first given effect that supports the  selected  plotting
          program,  SoX will output commands to plot the effect's transfer
          function, and then exit without actually processing  any  audio.
             sox --plot octave input-file -n highpass 1320 > highpass.plt
             octave highpass.plt

   -q, --no-show-progress
          Run  in  quiet  mode when SoX wouldn't otherwise do so.  This is
          the opposite of the -S option.

   -R     Run in `repeatable' mode.  When  this  option  is  given,  where
          applicable, SoX will embed a fixed time-stamp in the output file
          (e.g.  AIFF) and will `seed'  pseudo  random  number  generators
          (e.g.    dither)   with  a  fixed  number,  thus  ensuring  that
          successive SoX invocations with the same  inputs  and  the  same
          parameters yield the same output.

   --replay-gain track|album|off
          Select  whether  or not to apply replay-gain adjustment to input
          files.  The default is off for sox and rec, album for play where
          (at  least)  the  first two input files are tagged with the same
          Artist and Album names, and track for play otherwise.

   -S, --show-progress
          Display input file  format/header  information,  and  processing
          progress as input file(s) percentage complete, elapsed time, and
          remaining time (if known; shown in brackets), and the number  of
          samples  written to the output file.  Also shown is a peak-level
          meter, and an indication if clipping has  occurred.   The  peak-
          level  meter  shows  up  to  two  channels and is calibrated for
          digital audio as follows (right channel shown):

                        dB FSD   Display   dB FSD   Display
                         -25     -          -11     ====
                         -23     =           -9     ====-
                         -21     =-          -7     =====
                         -19     ==          -5     =====-
                         -17     ==-         -3     ======
                         -15     ===         -1     =====!
                         -13     ===-

          A three-second peak-held value of headroom in dBs will be  shown
          to the right of the meter if this is below 6dB.

          This  option  is  enabled  by  default when using SoX to play or
          record audio.

   -T     Equivalent to --combine multiply.

   --temp DIRECTORY
          Specify that any temporary files should be created in the  given
          DIRECTORY.   This can be useful if there are permission or free-
          space problems with the default location. In  this  case,  using
          `--temp  .'  (to  use  the  current  directory)  is often a good

          Show SoX's version number and exit.

          Set verbosity. This is particularly useful for  seeing  how  any
          automatic effects have been invoked by SoX.

          SoX  displays  messages on the console (stderr) according to the
          following verbosity levels:

          0      No messages are shown at all;  use  the  exit  status  to
                 determine if an error has occurred.

          1      Only  error  messages  are shown.  These are generated if
                 SoX cannot complete the requested commands.

          2      Warning messages are also shown.  These are generated  if
                 SoX  can complete the requested commands, but not exactly
                 according to the  requested  command  parameters,  or  if
                 clipping occurs.

          3      Descriptions  of  SoX's processing phases are also shown.
                 Useful for seeing exactly  how  SoX  is  processing  your

          4 and above
                 Messages to help with debugging SoX are also shown.

          By  default,  the  verbosity level is set to 2 (shows errors and
          warnings). Each  occurrence  of  the  -V  option  increases  the
          verbosity level by 1.  Alternatively, the verbosity level can be
          set to an absolute number by specifying it immediately after the
          -V, e.g.  -V0 sets it to 0.

   Input File Options
   These  options  apply  only  to  input files and may precede only input
   filenames on the command line.

          Override an (incorrect) audio length given in  an  audio  file's
          header. If this option is given then SoX will keep reading audio
          until it reaches the end of the input file.

   -v, --volume FACTOR
          Intended for use  when  combining  multiple  input  files,  this
          option  adjusts  the  volume  of the file that follows it on the
          command line by a  factor  of  FACTOR.  This  allows  it  to  be
          `balanced'  w.r.t.  the  other  input  files.   This is a linear
          (amplitude) adjustment, so a number less than  1  decreases  the
          volume  and a number greater than 1 increases it.  If a negative
          number is given then in addition to the volume  adjustment,  the
          audio signal will be inverted.

          See  also  the  norm,  vol, and gain effects, and see Input File
          Balancing above.

   Input & Output File Format Options
   These options apply to  the  input  or  output  file  whose  name  they
   immediately  precede  on  the  command  line  and  are used mainly when
   working with headerless file formats or when specifying  a  format  for
   the output file that is different to that of the input file.

   -b BITS, --bits BITS
          The  number  of bits (a.k.a. bit-depth or sometimes word-length)
          in each encoded sample.  Not  applicable  to  complex  encodings
          such  as  MP3  or GSM.  Not necessary with encodings that have a
          fixed number of bits, e.g.  A/μ-law, ADPCM.

          For an input file, the most common use for  this  option  is  to
          inform  SoX  of  the  number  of  bits  per  sample  in  a `raw'
          (`headerless') audio file.  For example
             sox -r 16k -e signed -b 8 input.raw output.wav
          converts a particular `raw'  file  to  a  self-describing  `WAV'

          For  an output file, this option can be used (perhaps along with
          -e) to set the output encoding size.  By default (i.e.  if  this
          option  is  not given), the output encoding size will (providing
          it is supported by the output file type) be  set  to  the  input
          encoding size.  For example
             sox input.cdda -b 24 output.wav
          converts  raw  CD  digital  audio  (16-bit, signed-integer) to a
          24-bit (signed-integer) `WAV' file.

          The number of bytes in each encoded sample.  Deprecated  aliases
          for -b 8, -b 16, -b 24, -b 32, -b 64 respectively.

   -c CHANNELS, --channels CHANNELS
          The  number of audio channels in the audio file. This can be any
          number greater than zero.

          For an input file, the most common use for  this  option  is  to
          inform  SoX  of the number of channels in a `raw' (`headerless')
          audio file.  Occasionally, it may be useful to use  this  option
          with  a  `headered'  file,  in order to override the (presumably
          incorrect) value  in  the  header  -  note  that  this  is  only
          supported with certain file types.  Examples:
             sox -r 48k -e float -b 32 -c 2 input.raw output.wav
          converts  a  particular  `raw'  file  to a self-describing `WAV'
             play -c 1 music.wav
          interprets the file  data  as  belonging  to  a  single  channel
          regardless  of  what is indicated in the file header.  Note that
          if the file does in fact have two channels, this will result  in
          the file playing at half speed.

          For  an  output  file,  this  option  provides  a  shorthand for
          specifying that the channels effect should be invoked  in  order
          to  change  (if  necessary)  the number of channels in the audio
          signal to the number given.   For  example,  the  following  two
          commands are equivalent:
             sox input.wav -c 1 output.wav bass -b 24
             sox input.wav      output.wav bass -b 24 channels 1
          though the second form is more flexible as it allows the effects
          to be ordered arbitrarily.

   -e ENCODING, --encoding ENCODING
          The audio encoding type.  Sometimes needed with file-types  that
          support more than one encoding type. For example, with raw, WAV,
          or AU (but not, for example, with MP3 or FLAC).   The  available
          encoding types are as follows:

                 PCM  data stored as signed (`two's complement') integers.
                 Commonly used with a 16 or  24  -bit  encoding  size.   A
                 value of 0 represents minimum signal power.

                 PCM data stored as unsigned integers.  Commonly used with
                 an 8-bit encoding size.  A value of 0 represents  maximum
                 signal power.

                 PCM  data stored as IEEE 753 single precision (32-bit) or
                 double   precision   (64-bit)   floating-point   (`real')
                 numbers.  A value of 0 represents minimum signal power.

          a-law  International telephony standard for logarithmic encoding
                 to 8 bits per sample.  It has a precision  equivalent  to
                 roughly 13-bit PCM and is sometimes encoded with reversed
                 bit-ordering (see the -X option).

          u-law, mu-law
                 North  American  telephony   standard   for   logarithmic
                 encoding  to  8 bits per sample.  A.k.a. μ-law.  It has a
                 precision  equivalent  to  roughly  14-bit  PCM  and   is
                 sometimes  encoded with reversed bit-ordering (see the -X

                 OKI (a.k.a. VOX, Dialogic, or Intel) 4-bit ADPCM; it  has
                 a precision equivalent to roughly 12-bit PCM.  ADPCM is a
                 form of audio compression  that  has  a  good  compromise
                 between audio quality and encoding/decoding speed.

                 IMA   (a.k.a.  DVI)  4-bit  ADPCM;  it  has  a  precision
                 equivalent to roughly 13-bit PCM.

                 Microsoft 4-bit ADPCM; it has a precision  equivalent  to
                 roughly 14-bit PCM.

                 GSM  is  currently  used  for  the  vast  majority of the
                 world's digital wireless telephone  calls.   It  utilises
                 several   audio  formats  with  different  bit-rates  and
                 associated speech quality.  SoX  has  support  for  GSM's
                 original  13kbps `Full Rate' audio format.  It is usually
                 CPU-intensive to work with GSM audio.

          Encoding names can  be  abbreviated  where  this  would  not  be
          ambiguous; e.g. `unsigned-integer' can be given as `un', but not
          `u' (ambiguous with `u-law').

          For an input file, the most common use for  this  option  is  to
          inform  SoX of the encoding of a `raw' (`headerless') audio file
          (see the examples in -b and -c above).

          For an output file, this option can be used (perhaps along  with
          -b) to set the output encoding type  For example
             sox input.cdda -e float output1.wav

             sox input.cdda -b 64 -e float output2.wav
          convert   raw  CD  digital  audio  (16-bit,  signed-integer)  to
          floating-point  `WAV'   files   (single   &   double   precision

          By  default  (i.e.  if  this  option  is  not given), the output
          encoding type will (providing it is supported by the output file
          type) be set to the input encoding type.

          Deprecated  aliases  for  specifying  the encoding types signed-
          integer, unsigned-integer, floating-point, a-law,  mu-law,  oki-
          adpcm,  ima-adpcm,  ms-adpcm, gsm-full-rate respectively (see -e

          Specifies that filename `globbing' (wild-card  matching)  should
          not be performed by SoX on the following filename.  For example,
          if  the  current  directory  contains  the  two   files   `five-
          seconds.wav' and `five*.wav', then
             play --no-glob "five*.wav"
          can be used to play just the single file `five*.wav'.

   -r, --rate RATE[k]
          Gives the sample rate in Hz (or kHz if appended with `k') of the

          For an input file, the most common use for  this  option  is  to
          inform  SoX  of  the sample rate of a `raw' (`headerless') audio
          file (see the examples in -b and -c above).  Occasionally it may
          be useful to use this option with a `headered' file, in order to
          override the (presumably incorrect) value in the header  -  note
          that  this  is  only  supported  with  certain  file types.  For
          example, if audio was recorded with a  sample-rate  of  say  48k
          from  a  source that played back a little, say 1.5%, too slowly,
             sox -r 48720 input.wav output.wav
          effectively corrects the speed by changing only the file  header
          (but  see  also  the speed effect for the more usual solution to
          this problem).

          For an  output  file,  this  option  provides  a  shorthand  for
          specifying  that  the  rate effect should be invoked in order to
          change (if necessary) the sample rate of the audio signal to the
          given  value.   For  example,  the  following  two  commands are
             sox input.wav -r 48k output.wav bass -b 24
             sox input.wav        output.wav bass -b 24 rate 48k
          though the second form  is  more  flexible  as  it  allows  rate
          options  to  be  given,  and  allows  the  effects to be ordered

   -t, --type FILE-TYPE
          Gives the type of the audio file.  For  both  input  and  output
          files,  this option is commonly used to inform SoX of the type a
          `headerless' audio file (e.g. raw, mp3) where the actual/desired
          type  cannot be determined from a given filename extension.  For
             another-command | sox -t mp3 - output.wav

             sox input.wav -t raw output.bin
          It can also be used to override the type  implied  by  an  input
          filename  extension,  but  if  overriding with a type that has a
          header, SoX will exit with an appropriate error message if  such
          a header is not actually present.

          See soxformat(7) for a list of supported file types.

   -L, --endian little
   -B, --endian big
   -x, --endian swap
          These  options  specify whether the byte-order of the audio data
          is, respectively, `little endian', `big endian', or the opposite
          to  that  of  the system on which SoX is being used.  Endianness
          applies only to data encoded as floating-point, or as signed  or
          unsigned  integers of 16 or more bits.  It is often necessary to
          specify one of these options for headerless files, and sometimes
          necessary   for  (otherwise)  self-describing  files.   A  given
          endian-setting option may be ignored for  an  input  file  whose
          header  contains  a  specific  endianness  identifier, or for an
          output file that is actually an audio device.

          N.B.  Unlike other format characteristics, the endianness (byte,
          nibble,  &  bit ordering) of the input file is not automatically
          used for the output file; so, for example, when the following is
          run on a little-endian system:
             sox -B audio.s16 trimmed.s16 trim 2
          trimmed.s16 will be created as little-endian;
             sox -B audio.s16 -B trimmed.s16 trim 2
          must be used to preserve big-endianness in the output file.

          The -V option can be used to check the selected orderings.

   -N, --reverse-nibbles
          Specifies that the nibble ordering (i.e. the 2 halves of a byte)
          of the samples should be reversed; sometimes useful with  ADPCM-
          based formats.

          N.B.  See also N.B. in section on -x above.

   -X, --reverse-bits
          Specifies  that  the  bit  ordering  of  the  samples  should be
          reversed;  sometimes  useful  with  a  few  (mostly  headerless)

          N.B.  See also N.B. in section on -x above.

   Output File Format Options
   These  options  apply  only to the output file and may precede only the
   output filename on the command line.

   --add-comment TEXT
          Append a comment in the output file header (where applicable).

   --comment TEXT
          Specify the comment text to store  in  the  output  file  header
          (where applicable).

          SoX   will   provide  a  default  comment  if  this  option  (or
          --comment-file) is not given. To specify that no comment  should
          be stored in the output file, use --comment "" .

   --comment-file FILENAME
          Specify  a  file  containing  the  comment  text to store in the
          output file header (where applicable).

   -C, --compression FACTOR
          The compression factor  for  variably  compressing  output  file
          formats.  If this option is not given then a default compression
          factor  will  apply.   The  compression  factor  is  interpreted
          differently  for  different  compressing  file formats.  See the
          description  of  the  file  formats  that  use  this  option  in
          soxformat(7) for more information.


   In  addition  to converting, playing and recording audio files, SoX can
   be used to invoke a number of audio `effects'.  Multiple effects may be
   applied  by  specifying  them  one  after another at the end of the SoX
   command line, forming an `effects chain'.  Note that applying  multiple
   effects  in  real-time (i.e. when playing audio) is likely to require a
   high performance computer. Stopping other  applications  may  alleviate
   performance issues should they occur.

   Some  of  the  SoX  effects  are  primarily intended to be applied to a
   single instrument or `voice'.  To facilitate this, the remix effect and
   the  global  SoX option -M can be used to isolate then recombine tracks
   from a multi-track recording.

   Multiple Effects Chains
   A single effects chain is made up of one or more effects.   Audio  from
   the input runs through the chain until either the end of the input file
   is reached or an effect in the chain requests to terminate the chain.

   SoX supports running multiple effects chains over the input audio.   In
   this  case,  when  one chain indicates it is done processing audio, the
   audio data is then sent through the next effects chain.  This continues
   until  either no more effects chains exist or the input has reached the
   end of the file.

   An effects chain is terminated by placing a : (colon) after an  effect.
   Any following effects are a part of a new effects chain.

   It  is  important  to  place the effect that will stop the chain as the
   first effect in the chain.   This  is  because  any  samples  that  are
   buffered  by  effects  to  the  left  of the terminating effect will be
   discarded.  The amount of samples discarded is related to the  --buffer
   option and it should be kept small, relative to the sample rate, if the
   terminating effect cannot be first.  Further  information  on  stopping
   effects can be found in the Stopping SoX section.

   There  are a few pseudo-effects that aid using multiple effects chains.
   These include newfile which will start writing to  a  new  output  file
   before  moving  to  the  next effects chain and restart which will move
   back to the first effects chain.  Pseudo-effects must be  specified  as
   the  first  effect  in  a chain and as the only effect in a chain (they
   must have a : before and after they are specified).

   The following is an example of multiple effects chains.  It will  split
   the  input  file  into  multiple  files  of 30 seconds in length.  Each
   output filename will have unique number in its name  as  documented  in
   the Output Files section.
      sox infile.wav output.wav trim 0 30 : newfile : restart

   Common Notation And Parameters
   In  the  descriptions  that  follow,  brackets  [  ] are used to denote
   parameters that are optional, braces { } to denote those that are  both
   optional  and  repeatable,  and angle brackets < > to denote those that
   are repeatable but not optional.  Where applicable, default values  for
   optional parameters are shown in parenthesis ( ).

   The  following parameters are used with, and have the same meaning for,
   several effects:

          See frequency.

          A frequency in Hz, or, if appended with `k', kHz.

   gain   A power gain in dB.  Zero gives no gain; less than zero gives an

          Used  to  specify  the  band-width  of  a  filter.   A number of
          different methods to specify the width are available (though not
          all  for  every  effect).   One  of  the characters shown may be
          appended to select the desired method as follows:

                                    Method    Notes
                               h      Hz
                               k     kHz
                               o   Octaves
                               q   Q-factor   See [2]

          For each effect that uses this  parameter,  the  default  method
          (i.e.  if  no  character  is appended) is the one that it listed
          first in the first line of the effect's description.

   To see if SoX has support for an optional effect, enter sox -h and look
   for its name under the list: `EFFECTS'.

   Supported Effects
   Note:   a  categorised  list  of  the  effects  can  be  found  in  the
   accompanying `README' file.

   allpass frequency[k] width[h|k|o|q]
          Apply a two-pole all-pass filter with central frequency (in  Hz)
          frequency,  and  filter-width width.  An all-pass filter changes
          the audio's frequency to phase relationship without changing its
          frequency to amplitude relationship.  The filter is described in
          detail in [1].

          This effect supports the --plot global option.

   band [-n] center[k] [width[h|k|o|q]]
          Apply  a  band-pass  filter.   The  frequency   response   drops
          logarithmically   around   the   center  frequency.   The  width
          parameter 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  audio,
          i.e.  voice, singing, or instrumental music.  The -n (for noise)
          option uses  the  alternate  mode  for  un-pitched  audio  (e.g.
          percussion).   Warning: -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.

          This effect supports the --plot global option.

          See also sinc for a bandpass filter with steeper shoulders.

   bandpass|bandreject [-c] frequency[k] width[h|k|o|q]
          Apply a two-pole Butterworth  band-pass  or  band-reject  filter
          with  central  frequency  frequency,  and (3dB-point) band-width
          width.  The -c option applies only to  bandpass  and  selects  a
          constant  skirt  gain  (peak  gain  = Q) instead of the default:
          constant 0dB peak gain.  The filters roll off at 6dB per  octave
          (20dB per decade) and are described in detail in [1].

          These effects support the --plot global option.

          See also sinc for a bandpass filter with steeper shoulders.

   bandreject frequency[k] width[h|k|o|q]
          Apply a band-reject filter.  See the description of the bandpass
          effect for details.

   bass|treble gain [frequency[k] [width[s|h|k|o|q]]]
          Boost or cut the bass (lower) or treble (upper)  frequencies  of
          the  audio  using  a  two-pole  shelving  filter with a response
          similar to that of a standard hi-fi's  tone-controls.   This  is
          also known as shelving equalisation (EQ).

          gain  gives  the  gain  at  0 Hz (for bass), or whichever is the
          lower of ∼22 kHz and the Nyquist frequency  (for  treble).   Its
          useful  range is about -20 (for a large cut) to +20 (for a large
          boost).  Beware of Clipping when using a positive gain.

          If desired, the filter can be  fine-tuned  using  the  following
          optional parameters:

          frequency sets the filter's central frequency and so can be used
          to extend or reduce the frequency range to be  boosted  or  cut.
          The default value is 100 Hz (for bass) or 3 kHz (for treble).

          width determines how steep is the filter's shelf transition.  In
          addition to the common  width  specification  methods  described
          above,  `slope'  (the  default,  or if appended with `s') may be
          used.  The useful range of `slope' is about 0.3,  for  a  gentle
          slope,  to 1 (the maximum), for a steep slope; the default value
          is 0.5.

          The filters are described in detail in [1].

          These effects support the --plot global option.

          See also equalizer for a peaking equalisation effect.

   bend [-f frame-rate(25)] [-o over-sample(16)] { delay,cents,duration }
          Changes pitch by specified amounts  at  specified  times.   Each
          given triple: delay,cents,duration specifies one bend.  delay is
          the amount of time after the start of the audio stream,  or  the
          end  of  the previous bend, at which to start bending the pitch;
          cents is the number of cents (100 cents = 1 semitone)  by  which
          to  bend  the  pitch, and duration the length of time over which
          the pitch will be bent.

          The  pitch-bending  algorithm  utilises  the  Discrete   Fourier
          Transform  (DFT)  at  a  particular frame rate and over-sampling
          rate.  The -f and -o parameters may  be  used  to  adjust  these
          parameters  and  thus  control  the smoothness of the changes in

          For example, an initial  tone  is  generated,  then  bent  three
          times, yielding four different notes in total:
             play -n synth 2.5 sin 667 gain 1 \
               bend .35,180,.25  .15,740,.53  0,-520,.3
          Note  that  the  clipping  that  is  produced in this example is
          deliberate; to remove it, use gain -5 in place of gain 1.

          See also pitch.

   biquad b0 b1 b2 a0 a1 a2
          Apply a biquad IIR filter with the given coefficients. Where  b*
          and   a*   are   the   numerator  and  denominator  coefficients

          See (where a0
          = 1).

          This effect supports the --plot global option.

   channels CHANNELS
          Invoke  a  simple  algorithm to change the number of channels in
          the audio  signal  to  the  given  number  CHANNELS:  mixing  if
          decreasing  the  number of channels or duplicating if increasing
          the number of channels.

          The channels effect is invoked automatically if SoX's -c  option
          specifies  a number of channels that is different to that of the
          input  file(s).   Alternatively,  if  this   effect   is   given
          explicitly,  then  SoX's  -c  option  need  not  be  given.  For
          example, the following two commands are equivalent:
             sox input.wav -c 1 output.wav bass -b 24
             sox input.wav      output.wav bass -b 24 channels 1
          though the second form is more flexible as it allows the effects
          to be ordered arbitrarily.

          See  also  remix  for  an  effect  that  allows  channels  to be
          mixed/selected arbitrarily.

   chorus gain-in gain-out <delay decay speed depth -s|-t>
          Add a chorus effect to the audio.  This can make a single  vocal
          sound like a chorus, but can also be applied to instrumentation.

          Chorus  resembles an echo effect with a short delay, but whereas
          with echo the delay is constant, with chorus, it is varied using
          sinusoidal  or  triangular  modulation.   The  modulation  depth
          defines the range the modulated delay is played before or  after
          the  delay. Hence the delayed sound will sound slower or faster,
          that is the delayed sound tuned around the original one, like in
          a  chorus  where  some vocals are slightly off key.  See [3] for
          more discussion of the chorus effect.

          Each  four-tuple  parameter  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 sinusoidal (-s) or triangular (-t).  Gain-
          out is the volume of the output.

          A typical delay is around 40ms to 60ms; the modulation speed  is
          best  near  0.25Hz  and  the  modulation  depth around 2ms.  For
          example, a single delay:
             play guitar1.wav chorus 0.7 0.9 55 0.4 0.25 2 -t
          Two delays of the original samples:
             play guitar1.wav chorus 0.6 0.9 50 0.4 0.25 2 -t \
                60 0.32 0.4 1.3 -s
          A fuller sounding chorus (with three additional delays):
             play guitar1.wav chorus 0.5 0.9 50 0.4 0.25 2 -t \
                60 0.32 0.4 2.3 -t 40 0.3 0.3 1.3 -s

   compand attack1,decay1{,attack2,decay2}
          [gain [initial-volume-dB [delay]]]

          Compand (compress or expand) the dynamic range of the audio.

          The attack and decay parameters (in seconds) determine the  time
          over  which  the  instantaneous  level  of  the  input signal is
          averaged to determine its volume; attacks refer to increases  in
          volume  and decays refer to decreases.  For most situations, the
          attack time (response to the music  getting  louder)  should  be
          shorter  than  the  decay  time  because  the  human ear is more
          sensitive to sudden loud music than sudden  soft  music.   Where
          more  than  one  pair  of attack/decay parameters are specified,
          each input channel is companded separately  and  the  number  of
          pairs  must  agree  with  the number of input channels.  Typical
          values are 0.3,0.8 seconds.

          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.  If omitted, the value of  out-
          dB1  defaults  to  the same value as in-dB1; levels below in-dB1
          are not companded (but may have  gain  applied  to  them).   The
          point  0,0  is assumed but may be overridden (by 0,out-dBn).  If
          the list is preceded by a soft-knee-dB value, then the points at
          where  adjacent line segments on the transfer function meet will
          be rounded by the amount given.  Typical values for the transfer
          function are 6:-70,-60,-20.

          The third (optional) parameter is an additional gain in dB to be
          applied at all points on the transfer function and  allows  easy
          adjustment of the overall gain.

          The  fourth  (optional)  parameter  is  an  initial  level to be
          assumed for each channel when companding starts.   This  permits
          the  user  to  supply  a  nominal  level initially, so that, for
          example, a very large gain is  not  applied  to  initial  signal
          levels  before the companding action has begun to operate: it is
          quite probable that in  such  an  event,  the  output  would  be
          severely  clipped  while  the  compander  gain  properly adjusts
          itself.  A typical value (for audio which is initially quiet) is
          -90 dB.

          The fifth (optional) parameter is a delay in seconds.  The input
          signal is analysed immediately 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.  A typical value is 0.2 seconds.

                                *        *        *

          The following example might be used to make  a  piece  of  music
          with both quiet and loud passages suitable for listening to in a
          noisy environment such as a moving vehicle:
             sox asz.wav asz-car.wav compand 0.3,1 6:-70,-60,-20 -5 -90 0.2
          The transfer function (`6:-70,...') says that very  soft  sounds
          (below  -70dB)  will  remain  unchanged.   This  will  stop  the
          compander from boosting the volume on `silent' passages such  as
          between  movements.   However,  sounds in the range -60dB to 0dB
          (maximum volume) will be boosted so that the 60dB dynamic  range
          of  the  original  music  will  be compressed 3-to-1 into a 20dB
          range, which is wide enough to enjoy the music but narrow enough
          to  get  around  the road noise.  The `6:' selects 6dB soft-knee
          companding.  The -5 (dB) output gain is needed to avoid clipping
          (the  number  is  inexact,  and was derived by experimentation).
          The -90 (dB) for the initial volume will work fine  for  a  clip
          that  starts  with  near silence, and the delay of 0.2 (seconds)
          has the effect of causing the compander  to  react  a  bit  more
          quickly to sudden volume changes.

          In  the  next example, compand is being used as a noise-gate for
          when the noise is at a lower level than the signal:
             play infile compand .1,.2 -inf,-50.1,-inf,-50,-50 0 -90 .1
          Here is another noise-gate, this time for when the noise is at a
          higher  level  than the signal (making it, in some ways, similar
          to squelch):
             play infile compand .1,.1 -45.1,-45,-inf,0,-inf 45 -90 .1
          This effect supports the --plot global option (for the  transfer

          See also mcompand for a multiple-band companding effect.

   contrast [enhancement-amount(75)]
          Comparable  with  compression,  this  effect  modifies  an audio
          signal to make it sound louder.  enhancement-amount controls the
          amount  of  the  enhancement and is a number in the range 0-100.
          Note that enhancement-amount  =  0  still  gives  a  significant
          contrast enhancement.

          See also the compand and mcompand effects.

   dcshift shift [limitergain]
          Apply  a  DC shift to the audio.  This can be useful to remove a
          DC offset (caused perhaps by a hardware problem in the recording
          chain)  from  the  audio.   The effect of a DC offset is reduced
          headroom and hence volume.  The stat or stats effect can be used
          to determine if a signal has a DC offset.

          The  given dcshift value is a floating point number in the range
          of ±2 that indicates the amount to shift the audio (which is  in
          the range of ±1).

          An  optional  limitergain  can  be specified as well.  It should
          have a value much less than 1 (e.g. 0.05 or 0.02)  and  is  used
          only on peaks to prevent clipping.

                                *        *        *

          An  alternative  approach to removing a DC offset (albeit with a
          short delay) is to use the highpass filter effect at a frequency
          of say 10Hz, as illustrated in the following example:
             sox -n dc.wav synth 5 sin %0 50
             sox dc.wav fixed.wav highpass 10

   deemph Apply Compact Disc (IEC 60908) de-emphasis (a treble attenuation
          shelving filter).

          Pre-emphasis was applied in the mastering of some CDs issued  in
          the early 1980s.  These included many classical music albums, as
          well as now sought-after issues of albums by The  Beatles,  Pink
          Floyd  and  others.   Pre-emphasis should be removed at playback
          time by a de-emphasis filter in the playback  device.   However,
          not  all  modern CD players have this filter, and very few PC CD
          drives have it; playing pre-emphasised audio without the correct
          de-emphasis filter results in audio that sounds harsh and is far
          from what its creators intended.

          With the deemph effect, it is possible to  apply  the  necessary
          de-emphasis  to  audio  that  has  been  extracted  from  a pre-
          emphasised CD, and then either burn the de-emphasised audio to a
          new  CD  (which  will  then play correctly on any CD player), or
          simply play the correctly de-emphasised audio files on  the  PC.
          For example:
             sox track1.wav track1-deemph.wav deemph
          and then burn track1-deemph.wav to CD, or
             play track1-deemph.wav
          or simply
             play track1.wav deemph
          The  de-emphasis  filter is implemented as a biquad; its maximum
          deviation from the ideal response is only 0.06dB (up to 20kHz).

          This effect supports the --plot global option.

          See also the bass and treble shelving equalisation effects.

   delay {length}
          Delay one or more audio channels.  length can specify a time or,
          if  appended  with  an `s', a number of samples.  Do not specify
          both time and samples delays in the same command.  For  example,
          delay  1.5  0  0.5  delays the first channel by 1.5 seconds, the
          third channel by 0.5 seconds, and leaves the second channel (and
          any  other  channels  that  may  be  present)  un-delayed.   The
          following (one long) command plays a chime sound:
             play -n synth -j 3 sin %3 sin %-2 sin %-5 sin %-9 \
               sin %-14 sin %-21 fade h .01 2 1.5 delay \
               1.3 1 .76 .54 .27 remix - fade h 0 2.7 2.5 norm -1
          and this plays a guitar chord:
             play -n synth pl G2 pl B2 pl D3 pl G3 pl D4 pl G4 \
               delay 0 .05 .1 .15 .2 .25 remix - fade 0 4 .1 norm -1

   dither [-S|-s|-f filter] [-a] [-p precision]
          Apply dithering to the audio.   Dithering  deliberately  adds  a
          small  amount  of  noise  to the signal in order to mask audible
          quantization effects that can occur if the output sample size is
          less  than  24  bits.   With  no  options,  this effect will add
          triangular (TPDF) white noise.  Noise-shaping (only for  certain
          sample  rates)  can be selected with -s.  With the -f option, it
          is possible to select a particular noise-shaping filter from the
          following   list:   lipshitz,  f-weighted,  modified-e-weighted,
          improved-e-weighted,  gesemann,  shibata,   low-shibata,   high-
          shibata.   Note  that  most filter types are available only with
          44100Hz sample rate.  The filter types are distinguished by  the
          following  properties: audibility of noise, level of (inaudible,
          but in some circumstances, otherwise  problematic)  shaped  high
          frequency noise, and processing speed.
          See  for  graphs of
          the different noise-shaping curves.

          The -S option selects a slightly `sloped' TPDF,  biased  towards
          higher  frequencies.   It  can  be used at any sampling rate but
          below ≈22k, plain TPDF is probably  better,  and  above  ≈  37k,
          noise-shaped is probably better.

          The  -a option enables a mode where dithering (and noise-shaping
          if applicable) are automatically enabled only when needed.   The
          most  likely  use for this is when applying fade in or out to an
          already dithered file, so that the redithering applies  only  to
          the  faded portions.  However, auto dithering is not fool-proof,
          so  the  fades  should  be  carefully  checked  for  any   noise
          modulation;  if  this  occurs,  then  either re-dither the whole
          file, or use trim, fade, and concatencate.

          The -p option allows overriding the target precision.

          If the SoX global option  -R  option  is  not  given,  then  the
          pseudo-random  number generator used to generate the white noise
          will be `reseeded', i.e. the generated noise will  be  different
          between invocations.

          This  effect  should  not  be  followed by any other effect that
          affects the audio.

          See also the `Dithering' section above.

   downsample [factor(2)]
          Downsample the signal by an integer factor: Only the  first  out
          of each factor samples is retained, the others are discarded.

          No decimation filter is applied.  If the input is not a properly
          bandlimited baseband signal, aliasing will occur.  This  may  be
          desirable, e.g., for frequency translation.

          For  a  general  resampling effect with anti-aliasing, see rate.
          See also upsample.

   earwax Makes audio easier to listen to on headphones.  Adds  `cues'  to
          44.1kHz  stereo  (i.e.  audio  CD  format)  audio  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).

   echo gain-in gain-out <delay decay>
          Add echoing to the audio.  Echoes are reflected  sound  and  can
          occur   naturally   amongst   mountains   (and  sometimes  large
          buildings)  when  talking  or  shouting;  digital  echo  effects
          emulate  this  behaviour and are often used to help fill out the
          sound of a single instrument  or  vocal.   The  time  difference
          between  the  original  signal and the reflection is the `delay'
          (time), and the loudness of the reflected signal is the `decay'.
          Multiple echoes can have different delays and decays.

          Each  given delay decay pair gives the delay in milliseconds and
          the decay (relative to gain-in) of that echo.  Gain-out  is  the
          volume  of  the output.  For example: This will make it sound as
          if there are twice as many instruments as are actually playing:
             play lead.aiff echo 0.8 0.88 60 0.4
          If the delay is very short, then  it  sound  like  a  (metallic)
          robot playing music:
             play lead.aiff echo 0.8 0.88 6 0.4
          A  longer  delay  will  sound  like  an  open air concert in the
             play lead.aiff echo 0.8 0.9 1000 0.3
          One mountain more, and:
             play lead.aiff echo 0.8 0.9 1000 0.3 1800 0.25

   echos gain-in gain-out <delay decay>
          Add a sequence of echoes to the audio.  Each  delay  decay  pair
          gives the delay in milliseconds and the decay (relative to gain-
          in) of that echo.  Gain-out is the volume of the output.

          Like the echo effect, echos stand for `ECHO in Sequel', that  is
          the  first  echos  takes the input, the second the input and the
          first echos, the third the input and the first  and  the  second
          echos,  ... and so on.  Care should be taken using many echos; a
          single echos has the same effect as a single echo.

          The sample will be bounced twice in symmetric echos:
             play lead.aiff echos 0.8 0.7 700 0.25 700 0.3
          The sample will be bounced twice in asymmetric echos:
             play lead.aiff echos 0.8 0.7 700 0.25 900 0.3
          The sample will sound as if played in a garage:
             play lead.aiff echos 0.8 0.7 40 0.25 63 0.3

   equalizer frequency[k] width[q|o|h|k] gain
          Apply a two-pole peaking equalisation (EQ)  filter.   With  this
          filter,  the signal-level at and around a selected frequency can
          be increased or decreased, whilst (unlike  band-pass  and  band-
          reject filters) that at all other frequencies is unchanged.

          frequency gives the filter's central frequency in Hz, width, the
          band-width, and gain the required gain  or  attenuation  in  dB.
          Beware of Clipping when using a positive gain.

          In order to produce complex equalisation curves, this effect can
          be given several times, each with a different central frequency.

          The filter is described in detail in [1].

          This effect supports the --plot global option.

          See also bass and treble for shelving equalisation effects.

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

          An optional type can be specified to select  the  shape  of  the
          fade  curve:  q  for  quarter  of a sine wave, h for half a sine
          wave, t for linear (`triangular') slope, l for logarithmic,  and
          p for inverted parabola.  The default is logarithmic.

          A  fade-in  starts  from  the  first sample and ramps the signal
          level  from  0  to  full  volume  over  fade-in-length  seconds.
          Specify 0 seconds if no fade-in is wanted.

          For  fade-outs, the audio will be truncated at stop-time and the
          signal level 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  stop-time  is  not
          specified.  If the file length can be determined from the  input
          file  header and length-changing effects are not in effect, then
          0 may be specified for stop-time to indicate the usual case of a
          fade-out that ends at the end of the input audio stream.

          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
          samples and append the letter  `s'  to  the  sample  count  (for
          example `8000s').

          See also the splice effect.

   fir [coefs-file|coefs]
          Use   SoX's   FFT  convolution  engine  with  given  FIR  filter
          coefficients.  If a  single  argument  is  given  then  this  is
          treated as the name of a file containing the filter coefficients
          (white-space separated; may contain `#' comments).  If the given
          filename   is  `-',  or  if  no  argument  is  given,  then  the
          coefficients  are  read  from  the  `standard  input'   (stdin);
          otherwise,  coefficients  may  be  given  on  the  command line.
             sox infile outfile fir 0.0195 -0.082 0.234 0.891 -0.145 0.043
             sox infile outfile fir coefs.txt
          with coefs.txt containing
             # HP filter
             # freq=10000

          This effect supports the --plot global option.

   flanger [delay depth regen width speed shape phase interp]
          Apply a flanging effect to the audio.  See [3]  for  a  detailed
          description of flanging.

          All parameters are optional (right to left).

                    Range     Default   Description
          delay     0 - 30       0      Base delay in milliseconds.
          depth     0 - 10       2      Added swept delay in milliseconds.
          regen    -95 - 95      0      Percentage regeneration (delayed
                                        signal feedback).
          width    0 - 100      71      Percentage of delayed signal mixed
                                        with original.
          speed    0.1 - 10     0.5     Sweeps per second (Hz).
          shape                 sin     Swept wave shape: sine|triangle.
          phase    0 - 100      25      Swept wave percentage phase-shift
                                        for multi-channel (e.g. stereo)
                                        flange; 0 = 100 = same phase on
                                        each channel.
          interp                lin     Digital delay-line interpolation:

   gain [-e|-B|-b|-r] [-n] [-l|-h] [gain-dB]
          Apply  amplification  or attenuation to the audio signal, or, in
          some cases, to some of its channels.  Note that use  of  any  of
          -e, -B, -b, -r, or -n requires temporary file space to store the
          audio to be  processed,  so  may  be  unsuitable  for  use  with
          `streamed' audio.

          Without  other  options,  gain-dB  is  used to adjust the signal
          power level by  the  given  number  of  dB:  positive  amplifies
          (beware  of Clipping), negative attenuates.  With other options,
          the gain-dB amplification or attenuation is (logically)  applied
          after the processing due to those options.

          Given  the  -e  option,  the  levels  of the audio channels of a
          multi-channel file are `equalised', i.e.  gain is applied to all
          channels  other than that with the highest peak level, such that
          all channels attain the  same  peak  level  (but,  without  also
          giving -n, the audio is not `normalised').

          The  -B  (balance) option is similar to -e, but with -B, the RMS
          level is used instead of the peak level.  -B might  be  used  to
          correct stereo imbalance caused by an imperfect record turntable
          cartridge.   Note that unlike -e, -B might cause some clipping.

          -b is similar to  -B  but  has  clipping  protection,  i.e.   if
          necessary  to  prevent clipping whilst balancing, attenuation is
          applied to all channels.  Note,  however,  that  in  conjunction
          with -n, -B and -b are synonymous.

          The  -r option is used in conjunction with a prior invocation of
          gain with the -h option - see below for details.

          The -n option normalises the audio to 0dB FSD; it is often  used
          in  conjunction  with  a negative gain-dB to the effect that the
          audio is normalised to a given level below 0dB.  For example,
             sox infile outfile gain -n
          normalises to 0dB, and
             sox infile outfile gain -n -3
          normalises to -3dB.

          The -l option invokes a simple limiter, e.g.
             sox infile outfile gain -l 6
          will apply 6dB of gain but never clip.  Note that limiting  more
          than  a  few dBs more than occasionally (in a piece of audio) is
          not recommended as it can cause  audible  distortion.   See  the
          compand effect for a more capable limiter.

          The  -h  option  is  used to apply gain to provide head-room for
          subsequent processing.  For example, with
             sox infile outfile gain -h bass +6
          6dB of attenuation will be applied prior to  the  bass  boosting
          effect  thus  ensuring  that  it will not clip.  Of course, with
          bass, it is obvious how much headroom will be needed,  but  with
          other  effects  (e.g.   rate, dither) it is not always as clear.
          Another advantage of using  gain  -h  rather  than  an  explicit
          attenuation,  is  that if the headroom is not used by subsequent
          effects, it can be reclaimed with gain -r, for example:
             sox infile outfile gain -h bass +6 rate 44100 gain -r
          The above effects chain guarantees never to clip nor amplify; it
          attenuates if necessary to prevent clipping, but by only as much
          as is needed to do so.

          Output  formatting  (dithering  and  bit-depth  reduction)  also
          requires headroom (which cannot be `reclaimed'), e.g.
             sox infile outfile gain -h bass +6 rate 44100 gain -rh dither
          Here,  the  second  gain  invocation,  reclaims  as  much of the
          headroom as it can from the preceding effects,  but  retains  as
          much  headroom  as is needed for subsequent processing.  The SoX
          global option -G can be given to automatically  invoke  gain  -h
          and gain -r.

          See also the norm and vol effects.

   highpass|lowpass [-1|-2] frequency[k] [width[q|o|h|k]]
          Apply  a  high-pass or low-pass filter with 3dB point frequency.
          The filter can be either single-pole (with -1),  or  double-pole
          (the  default,  or  with -2).  width applies only to double-pole
          filters; the default is  Q  =  0.707  and  gives  a  Butterworth
          response.  The filters roll off at 6dB per pole per octave (20dB
          per pole per decade).  The double-pole filters are described  in
          detail in [1].

          These effects support the --plot global option.

          See also sinc for filters with a steeper roll-off.

   hilbert [-n taps]
          Apply  an  odd-tap  Hilbert transform filter, phase-shifting the
          signal by 90 degrees.

          This is used in many matrix  coding  schemes  and  for  analytic
          signal   generation.    The   process  is  often  written  as  a
          multiplication by i (or j), the imaginary unit.

          An   odd-tap   Hilbert   transform   filter   has   a   bandpass
          characteristic,  attenuating the lowest and highest frequencies.
          Its bandwidth can be controlled by the number  of  filter  taps,
          which  can be specified with -n.  By default, the number of taps
          is chosen for a cutoff frequency of about 75 Hz.

          This effect supports the --plot global option.

   ladspa module [plugin] [argument...]
          Apply a LADSPA [5] (Linux Audio Developer's Simple  Plugin  API)
          plugin.   Despite  the name, LADSPA is not Linux-specific, and a
          wide range of effects is available as LADSPA  plugins,  such  as
          cmt  [6]  (the Computer Music Toolkit) and Steve Harris's plugin
          collection [7]. The first argument is  the  plugin  module,  the
          second  the  name  of the plugin (a module can contain more than
          one plugin) and any other arguments are for the control ports of
          the  plugin. Missing arguments are supplied by default values if
          possible. Only plugins with at most  one  audio  input  and  one
          audio  output  port  can  be  used.   If  found, the environment
          variable LADSPA_PATH will be used as search path for plugins.

   loudness [gain [reference]]
          Loudness control - similar to  the  gain  effect,  but  provides
          equalisation    for    the    human    auditory   system.    See
 for a detailed description
          of  loudness.   The gain is adjusted by the given gain parameter
          (usually negative) and the signal equalised according to ISO 226
          w.r.t.   a  reference  level  of  65dB,  though  an  alternative
          reference level may be given if  the  original  audio  has  been
          equalised for some other optimal level.  A default gain of -10dB
          is used if a gain value is not given.

          See also the gain effect.

   lowpass [-1|-2] frequency[k] [width[q|o|h|k]]
          Apply a low-pass filter.  See the description  of  the  highpass
          effect for details.

   mcompand "attack1,decay1{,attack2,decay2}
          [gain     [initial-volume-dB    [delay]]]"    {crossover-freq[k]

          The multi-band compander is similar to the single-band compander
          but  the  audio is first divided into bands using Linkwitz-Riley
          cross-over filters and a separately specifiable compander run on
          each  band.   See  the  compand effect for the definition of its
          parameters.  Compand parameters  are  specified  between  double
          quotes  and  the  crossover  frequency for that band is given by
          crossover-freq; these can be repeated to create multiple bands.

          For example, the following (one long) command shows  how  multi-
          band companding is typically used in FM radio:
             play track1.wav gain -3 sinc 8000- 29 100 mcompand \
               "0.005,0.1 -47,-40,-34,-34,-17,-33" 100 \
               "0.003,0.05 -47,-40,-34,-34,-17,-33" 400 \
               "0.000625,0.0125 -47,-40,-34,-34,-15,-33" 1600 \
               "0.0001,0.025 -47,-40,-34,-34,-31,-31,-0,-30" 6400 \
               "0,0.025 -38,-31,-28,-28,-0,-25" \
               gain 15 highpass 22 highpass 22 sinc -n 255 -b 16 -17500 \
               gain 9 lowpass -1 17801
          The  audio  file  is  played with a simulated FM radio sound (or
          broadcast signal condition if the lowpass filter at the  end  is
          skipped).   Note  that the pipeline is set up with US-style 75us

          See also compand for a single-band companding effect.

   noiseprof [profile-file]
          Calculate a profile of the audio for  use  in  noise  reduction.
          See the description of the noisered effect for details.

   noisered [profile-file [amount]]
          Reduce  noise  in  the  audio signal by profiling and filtering.
          This effect  is  moderately  effective  at  removing  consistent
          background  noise such as hiss or hum.  To use it, first run SoX
          with the noiseprof effect on a section  of  audio  that  ideally
          would contain silence but in fact contains noise - such sections
          are typically found at the beginning or the end of a  recording.
          noiseprof  will write out a noise profile to profile-file, or to
          stdout if no profile-file or if `-' is given.  E.g.
             sox speech.wav -n trim 0 1.5 noiseprof speech.noise-profile
          To actually remove the noise, run SoX again, this time with  the
          noisered effect; noisered will reduce noise according to a noise
          profile (which was generated by noiseprof),  from  profile-file,
          or from stdin if no profile-file or if `-' is given.  E.g.
             sox speech.wav cleaned.wav noisered speech.noise-profile 0.3
          How much noise should be removed is specified by amount-a number
          between 0 and 1 with a default  of  0.5.   Higher  numbers  will
          remove  more  noise but present a greater likelihood of removing
          wanted components of the  audio  signal.   Before  replacing  an
          original recording with a noise-reduced version, experiment with
          different amount values to find the optimal one for your  audio;
          use  headphones  to  check  that you are happy with the results,
          paying particular attention to quieter sections of the audio.

          On most systems, the two stages - profiling and reduction -  can
          be combined using a pipe, e.g.
             sox noisy.wav -n trim 0 1 noiseprof | play noisy.wav noisered

   norm [dB-level]
          Normalise the audio.  norm is just an alias for gain -n; see the
          gain effect for details.

   oops   Out Of Phase Stereo effect.  Mixes  stereo  to  twin-mono  where
          each  mono  channel contains the difference between the left and
          right stereo channels.  This is sometimes known as the `karaoke'
          effect as it often has the effect of removing most or all of the
          vocals from a recording.  It is equivalent to remix 1,2i 1,2i.

   overdrive [gain(20) [colour(20)]]
          Non linear distortion.  The colour parameter controls the amount
          of even harmonic content in the over-driven output.

   pad { length[@position] }
          Pad  the  audio  with silence, at the beginning, the end, or any
          specified points through the audio.  Both  length  and  position
          can  specify  a  time  or,  if appended with an `s', a number of
          samples.  length is the amount of silence to insert and position
          the  position  in  the input audio stream at which to insert it.
          Any number of lengths and positions may be  specified,  provided
          that  a  specified  position  is not less that the previous one.
          position is optional for the first and  last  lengths  specified
          and  if  omitted  correspond to the beginning and the end of the
          audio respectively.  For example, pad 1.5 1.5 adds  1.5  seconds
          of  silence  padding  at  each  end  of  the  audio,  whilst pad
          4000s@3:00 inserts 4000 samples of silence 3  minutes  into  the
          audio.   If  silence  is  wanted  only  at the end of the audio,
          specify either the end position or specify a zero-length pad  at
          the start.

          See  also  delay  for  an  effect  that  can  add silence at the
          beginning of the audio on a channel-by-channel basis.

   phaser gain-in gain-out delay decay speed [-s|-t]
          Add a phasing effect to the  audio.   See  [3]  for  a  detailed
          description of phasing.

          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 sinusoidal (-s)  - preferable for multiple
          instruments, or triangular (-t)  - gives  single  instruments  a
          sharper  phasing  effect.   The decay should be less than 0.5 to
          avoid feedback, and usually no less than 0.1.  Gain-out  is  the
          volume of the output.

          For example:
             play snare.flac phaser 0.8 0.74 3 0.4 0.5 -t
             play snare.flac phaser 0.9 0.85 4 0.23 1.3 -s
          A popular sound:
             play snare.flac phaser 0.89 0.85 1 0.24 2 -t
          More severe:
             play snare.flac phaser 0.6 0.66 3 0.6 2 -t

   pitch [-q] shift [segment [search [overlap]]]
          Change the audio pitch (but not tempo).

          shift  gives  the  pitch  shift  as positive or negative `cents'
          (i.e. 100ths of  a  semitone).   See  the  tempo  effect  for  a
          description of the other parameters.

          See also the bend, speed, and tempo effects.

   rate [-q|-l|-m|-h|-v] [override-options] RATE[k]
          Change  the audio sampling rate (i.e. resample the audio) to any
          given RATE (even non-integer if this is supported by the  output
          file format) using a quality level defined as follows:

                       Quality   Band-   Rej dB   Typical Use
                 -q     quick     n/a    ≈30 @    playback on
                                          Fs/4    ancient hardware
                 -l      low      80%     100     playback on old
                 -m    medium     95%     100     audio playback
                 -h     high      95%     125     16-bit mastering
                                                  (use with dither)
                 -v   very high   95%     175     24-bit mastering

          where Band-width is the percentage of the audio  frequency  band
          that  is  preserved  and Rej dB is the level of noise rejection.
          Increasing levels of resampling quality come at the  expense  of
          increasing  amounts of time to process the audio.  If no quality
          option is given, the quality  level  used  is  `high'  (but  see
          `Playing & Recording Audio' above regarding playback).

          The  `quick'  algorithm uses cubic interpolation; all others use
          band-limited interpolation.  By default, all algorithms  have  a
          `linear'  phase  response; for `medium', `high' and `very high',
          the phase response is configurable (see below).

          The rate effect is invoked  automatically  if  SoX's  -r  option
          specifies a rate that is different to that of the input file(s).
          Alternatively, if this effect is given explicitly, then SoX's -r
          option  need  not  be  given.   For  example,  the following two
          commands are equivalent:
             sox input.wav -r 48k output.wav bass -b 24
             sox input.wav        output.wav bass -b 24 rate 48k
          though the second command is more flexible  as  it  allows  rate
          options  to  be  given,  and  allows  the  effects to be ordered

                                *        *        *

          Warning: technically detailed discussion follows.

          The simple quality selection described above  provides  settings
          that satisfy the needs of the vast majority of resampling tasks.
          Occasionally, however, it may  be  desirable  to  fine-tune  the
          resampler's   filter   response;  this  can  be  achieved  using
          override options, as detailed in the following table:

          -M/-I/-L     Phase response = minimum/intermediate/linear
          -s           Steep filter (band-width = 99%)
          -a           Allow aliasing/imaging above the pass-band
          -b 74-99.7   Any band-width %
          -p 0-100     Any phase response (0 = minimum, 25 = intermediate,
                       50 = linear, 100 = maximum)

          N.B.   Override options cannot be used with the `quick' or `low'
          quality algorithms.

          All resamplers use filters  that  can  sometimes  create  `echo'
          (a.k.a.   `ringing')  artefacts  with  transient signals such as
          those that occur with `finger snaps' or other highly  percussive
          sounds.   Such  artefacts  are much more noticeable to the human
          ear if they occur before the transient (`pre-echo') than if they
          occur  after  it (`post-echo').  Note that frequency of any such
          artefacts is related to the smaller  of  the  original  and  new
          sampling  rates  but  that if this is at least 44.1kHz, then the
          artefacts will lie outside the range of human hearing.

          A phase response setting may be used to control the distribution
          of  any  transient  echo  between `pre' and `post': with minimum
          phase, there is no pre-echo  but  the  longest  post-echo;  with
          linear  phase, pre and post echo are in equal amounts (in signal
          terms, but not audibility terms); the intermediate phase setting
          attempts to find the best compromise by selecting a small length
          (and level) of pre-echo and a medium lengthed post-echo.

          Minimum, intermediate, or  linear  phase  response  is  selected
          using  the  -M, -I, or -L option; a custom phase response can be
          created with the -p option.  Note that phase  responses  between
          `linear' and `maximum' (greater than 50) are rarely useful.

          A  resampler's  band-width  setting  determines  how much of the
          frequency content of the original signal  (w.r.t.  the  original
          sample  rate when up-sampling, or the new sample rate when down-
          sampling) is preserved during conversion.  The term  `pass-band'
          is  used  to refer to all frequencies up to the band-width point
          (e.g. for 44.1kHz sampling rate, and a resampling band-width  of
          95%,  the  pass-band  represents  frequencies from 0Hz (D.C.) to
          circa 21kHz).  Increasing the resampler's band-width results  in
          a  slower  conversion  and can increase transient echo artefacts
          (and vice versa).

          The -s `steep filter' option changes resampling band-width  from
          the default 95% (based on the 3dB point), to 99%.  The -b option
          allows the band-width to be  set  to  any  value  in  the  range
          74-99.7  %, but note that band-width values greater than 99% are
          not recommended for normal  use  as  they  can  cause  excessive
          transient echo.

          If the -a option is given, then aliasing/imaging above the pass-
          band is allowed.  For example, with 44.1kHz sampling rate, and a
          resampling  band-width of 95%, this means that frequency content
          above 21kHz can be distorted; however, since this is  above  the
          pass-band    (i.e.     above    the    highest    frequency   of
          interest/audibility), this may not be a problem.   The  benefits
          of  allowing  aliasing/imaging  are reduced processing time, and
          reduced (by almost half) transient echo artefacts.  Note that if
          this option is given, then the minimum band-width allowable with
          -b increases to 85%.

             sox input.wav -b 16 output.wav rate -s -a 44100 dither -s
          default (high)  quality  resampling;  overrides:  steep  filter,
          allow  aliasing;  to 44.1kHz sample rate; noise-shaped dither to
          16-bit WAV file.
             sox input.wav -b 24 output.aiff rate -v -I -b 90 48k
          very high quality  resampling;  overrides:  intermediate  phase,
          band-width  90%; to 48k sample rate; store output to 24-bit AIFF

                                *        *        *

          The pitch and speed effects use the rate effect at their core.

   remix [-a|-m|-p] <out-spec>
          out-spec  = in-spec{,in-spec} | 0
          in-spec   = [in-chan][-[in-chan2]][vol-spec]
          vol-spec  = p|i|v[volume]

          Select and mix input audio channels into output audio  channels.
          Each  output channel is specified, in turn, by a given out-spec:
          a list of contributing input channels and volume specifications.

          Note that this effect operates on the audio channels within  the
          SoX effects processing chain; it should not be confused with the
          -m global option (where multiple files are  mix-combined  before
          entering the effects chain).

          An  out-spec  contains comma-separated input channel-numbers and
          hyphen-delimited channel-number ranges; alternatively, 0 may  be
          given to create a silent output channel.  For example,
             sox input.wav output.wav remix 6 7 8 0
          creates  an output file with four channels, where channels 1, 2,
          and 3 are copies of channels 6, 7, and 8 in the input file,  and
          channel 4 is silent.  Whereas
             sox input.wav output.wav remix 1-3,7 3
          creates  a  (somewhat bizarre) stereo output file where the left
          channel is a mix-down of input channels 1, 2, 3, and 7, and  the
          right channel is a copy of input channel 3.

          Where  a  range of channels is specified, the channel numbers to
          the left and right of the hyphen are optional and default  to  1
          and to the number of input channels respectively. Thus
             sox input.wav output.wav remix -
          performs a mix-down of all input channels to mono.

          By  default,  where an output channel is mixed from multiple (n)
          input channels, each input channel will be scaled by a factor of
          ¹/n.   Custom  mixing  volumes  can  be set by following a given
          input channel or range of input channels with a vol-spec (volume
          specification).  This is one of the letters p, i, or v, followed
          by a volume number, the meaning of which depends  on  the  given
          letter and is defined as follows:

                  Letter   Volume number        Notes
                    p      power adjust in dB   0 = no change
                    i      power adjust in dB   As `p', but invert
                                                the audio
                    v      voltage multiplier   1 = no change, 0.5
                                                ≈ 6dB attenuation,
                                                2 ≈ 6dB gain, -1 =

          If  an out-spec includes at least one vol-spec then, by default,
          ¹/n scaling is not applied to any other  channels  in  the  same
          out-spec (though may be in other out-specs).  The -a (automatic)
          option however, can be given to retain the automatic scaling  in
          this case.  For example,
             sox input.wav output.wav remix 1,2 3,4v0.8
          results in channel level multipliers of 0.5,0.5 1,0.8, whereas
             sox input.wav output.wav remix -a 1,2 3,4v0.8
          results in channel level multipliers of 0.5,0.5 0.5,0.8.

          The   -m   (manual)   option   disables   all  automatic  volume
          adjustments, so
             sox input.wav output.wav remix -m 1,2 3,4v0.8
          results in channel level multipliers of 1,1 1,0.8.

          The volume number is optional and omitting it corresponds to  no
          volume change; however, the only case in which this is useful is
          in conjunction with i.  For example,  if  input.wav  is  stereo,
             sox input.wav output.wav remix 1,2i
          is a mono equivalent of the oops effect.

          If  the  -p  option  is given, then any automatic ¹/n scaling is
          replaced by ¹/√n (`power') scaling; this gives a louder mix  but
          one that might occasionally clip.

                                *        *        *

          One use of the remix effect is to split an audio file into a set
          of files, each containing one of the  constituent  channels  (in
          order  to  perform  subsequent  processing  on  individual audio
          channels).  Where more than  a  few  channels  are  involved,  a
          script such as the following (Bourne shell script) is useful:
          chans=`soxi -c "$1"`
          while [ $chans -ge 1 ]; do
             chans0=`printf %02i $chans`   # 2 digits hence up to 99 chans
             out=`echo "$1"|sed "s/\(.*\)\.\(.*\)/\1-$chans0.\2/"`
             sox "$1" "$out" remix $chans
             chans=`expr $chans - 1`
          If  a  file  input.wav containing six audio channels were given,
          the  script  would  produce  six  output  files:   input-01.wav,
          input-02.wav, ..., input-06.wav.

          See also the swap effect.

   repeat [count (1)]
          Repeat  the  entire  audio  count times, or once if count is not
          given.  Requires temporary file space to store the audio  to  be
          repeated.   Note  that  repeating  once  yields  two copies: the
          original audio and the repeated audio.

   reverb [-w|--wet-only] [reverberance (50%) [HF-damping (50%)
          [room-scale (100%) [stereo-depth (100%)
          [pre-delay (0ms) [wet-gain (0dB)]]]]]]

          Add reverberation to the audio using the  `freeverb'  algorithm.
          A  reverberation effect is sometimes desirable for concert halls
          that are too small or contain so many  people  that  the  hall's
          natural  reverberance is diminished.  Applying a small amount of
          stereo reverb to a (dry) mono signal will usually make it  sound
          more   natural.    See   [3]   for  a  detailed  description  of

          Note that this effect increases both the volume and  the  length
          of the audio, so to prevent clipping in these domains, a typical
          invocation might be:
             play dry.wav gain -3 pad 0 3 reverb
          The -w option can be given to select only the `wet' signal, thus
          allowing  it to be processed further, independently of the `dry'
          signal.  E.g.
             play -m voice.wav "|sox voice.wav -p reverse reverb -w reverse"
          for a reverse reverb effect.

          Reverse the audio completely.  Requires temporary file space  to
          store the audio to be reversed.

   riaa   Apply  RIAA vinyl playback equalisation.  The sampling rate must
          be one of: 44.1, 48, 88.2, 96 kHz.

          This effect supports the --plot global option.

   silence [-l] above-periods [duration threshold[d|%]
          [below-periods duration threshold[d|%]]

          Removes silence from the beginning, middle, or end of the audio.
          `Silence' is determined by a specified threshold.

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

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

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

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

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

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

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

          The option -l indicates that below-periods  duration  length  of
          audio  should  be left intact at the beginning of each period of
          silence.  For example, if you want to remove long pauses between
          words but do not want to remove the pauses completely.

          The  period  counts are in units of samples. Duration counts may
          be in the  format  of  hh:mm:ss.frac,  or  the  exact  count  of
          samples.   Threshold  numbers may be suffixed with d to indicate
          the value is in decibels, or  %  to  indicate  a  percentage  of
          maximum  value  of  the  sample value (0% specifies pure digital

          The following example shows how this effect can be used to start
          a  recording  that does not contain the delay at the start which
          usually occurs between `pressing  the  record  button'  and  the
          start of the performance:
             rec parameters filename other-effects silence 1 5 2%

   sinc [-a att|-b beta] [-p phase|-M|-I|-L] [-t tbw|-n taps] [freqHP]
   [-freqLP [-t tbw|-n taps]]
          Apply a sinc kaiser-windowed low-pass, high-pass, band-pass,  or
          band-reject  filter  to  the  signal.   The  freqHP  and  freqLP
          parameters give the frequencies of the 6dB points of a high-pass
          and  low-pass  filter  that  may  be  invoked  individually,  or
          together.  If both are  given,  then  freqHP  less  than  freqLP
          creates a band-pass filter, freqHP greater than freqLP creates a
          band-reject filter.  For example, the invocations
             sinc 3k
             sinc -4k
             sinc 3k-4k
             sinc 4k-3k
          create a high-pass, low-pass, band-pass, and band-reject  filter

          The  default  stop-band  attenuation  of 120dB can be overridden
          with -a; alternatively, the kaiser-window `beta'  parameter  can
          be given directly with -b.

          The default transition band-width of 5% of the total band can be
          overridden with -t (and tbw in Hertz); alternatively, the number
          of filter taps can be given directly with -n.

          If  both  freqHP  and  freqLP  are given, then a -t or -n option
          given  to  the  left  of  the  frequencies   applies   to   both
          frequencies;  one  of  these  options  given to the right of the
          frequencies applies only to freqLP.

          The -p, -M, -I,  and  -L  options  control  the  filter's  phase
          response; see the rate effect for details.

          This effect supports the --plot global option.

   spectrogram [options]
          Create   a  spectrogram  of  the  audio;  the  audio  is  passed
          unmodified through the SoX processing  chain.   This  effect  is
          optional  -  type  sox  --help  and  check the list of supported
          effects to see if it has been included.

          The spectrogram is rendered in a Portable Network Graphic  (PNG)
          file, and shows time in the X-axis, frequency in the Y-axis, and
          audio  signal  magnitude  in  the  Z-axis.   Z-axis  values  are
          represented  by  the colour (or optionally the intensity) of the
          pixels in the X-Y plane.  If the audio signal contains  multiple
          channels  then  these are shown from top to bottom starting from
          channel 1 (which is the left channel for stereo audio).

          For example, if `my.wav' is a stereo file, then with
             sox my.wav -n spectrogram
          a spectrogram of the entire file will be  created  in  the  file
          `spectrogram.png'.   More  often  though,  analysis of a smaller
          portion of the audio is required; e.g. with
             sox my.wav -n remix 2 trim 20 30 spectrogram
          the spectrogram shows information only from the  second  (right)
          channel,  and  of  thirty  seconds of audio starting from twenty
          seconds in.  To analyse a small portion of the frequency domain,
          the rate effect may be used, e.g.
             sox my.wav -n rate 6k spectrogram
          allows  detailed  analysis  of  frequencies up to 3kHz (half the
          sampling rate) i.e. where the  human  auditory  system  is  most
          sensitive.  With
             sox my.wav -n trim 0 10 spectrogram -x 600 -y 200 -z 100
          the given options control the size of the spectrogram's X, Y & Z
          axes (in this case, the spectrogram area of the  produced  image
          will  be  600 by 200 pixels in size and the Z-axis range will be
          100 dB).  Note that the produced  image  includes  axes  legends
          etc.  and  so  will  be  a  little  larger  than  the  specified
          spectrogram size.  In this example:
             sox -n -n synth 6 tri 10k:14k spectrogram -z 100 -w kaiser
          an analysis `window' with high dynamic range is selected to best
          display  the  spectrogram  of  a  swept  triangular wave.  For a
          smilar example, append the following to the `chime'  command  in
          the description of the delay effect (above):
             rate 2k spectrogram -X 200 -Z -10 -w kaiser
          Options  are  also  avaliable to control the appearance (colour-
          set,  brightness,  contrast,   etc.)   and   filename   of   the
          spectrogram; e.g. with
             sox my.wav -n spectrogram -m -l -o print.png
          a  spectrogram  is created suitable for printing on a `black and
          white' printer.


          -x num Change the (maximum) width (X-axis)  of  the  spectrogram
                 from  its  default  value of 800 pixels to a given number
                 between 100 and 200000.  See also -X and -d.

          -X num X-axis pixels/second; the default is  auto-calculated  to
                 fit the given or known audio duration to the X-axis size,
                 or 100 otherwise.  If given in conjunction with -d,  this
                 option  affects  the width of the spectrogram; otherwise,
                 it affects the duration of the spectrogram.  num  can  be
                 from   1   (low  time  resolution)  to  5000  (high  time
                 resolution) and need not be an integer.  SoX may  make  a
                 slight  adjustment  to  the  given  number for processing
                 quantisation reasons; if so, SoX will report  the  actual
                 number used (viewable when the SoX global option -V is in
                 effect).  See also -x and -d.

          -y num Sets the Y-axis size in pixels (per channel); this is the
                 number  of  frequency `bins' used in the Fourier analysis
                 that produces the spectrogram.  N.B. it can  be  slow  to
                 produce  the  spectrogram  if this number is not one more
                 than a power of two (e.g. 129).  By  default  the  Y-axis
                 size  is chosen automatically (depending on the number of
                 channels).   See  -Y  for  alternative  way  of   setting
                 spectrogram height.

          -Y num Sets  the target total height of the spectrogram(s).  The
                 default value is 550 pixels.  Using this option  (and  by
                 default),   SoX  will  choose  a  height  for  individual
                 spectrogram channels that is one more  than  a  power  of
                 two,  so  the  actual  total height may fall short of the
                 given number.  However, there is also  a  minimum  height
                 per channel so if there are many channels, the number may
                 be exceeded.  See  -y  for  alternative  way  of  setting
                 spectrogram height.

          -z num Z-axis  (colour) range in dB, default 120.  This sets the
                 dynamic-range of  the  spectrogram  to  be  -num dBFS  to
                 0 dBFS.   Num  may  range  from  20  to  180.  Decreasing
                 dynamic-range effectively increases the `contrast' of the
                 spectrogram display, and vice versa.

          -Z num Sets  the  upper limit of the Z-axis in dBFS.  A negative
                 num  effectively  increases  the  `brightness'   of   the
                 spectrogram display, and vice versa.

          -q num Sets   the   Z-axis  quantisation,  i.e.  the  number  of
                 different colours (or intensities) in which to render  Z-
                 axis  values.   A  small  number  (e.g.  4)  will  give a
                 `poster'-like  effect  making  it   easier   to   discern
                 magnitude  bands  of  similar  level.  Small numbers also
                 usually result in small  PNG  files.   The  number  given
                 specifies  the number of colours to use inside the Z-axis
                 range; two colours are reserved to represent out-of-range

          -w name
                 Window: Hann (default), Hamming, Bartlett, Rectangular or
                 Kaiser.  The spectrogram is produced using  the  Discrete
                 Fourier   Transform   (DFT)   algorithm.   A  significant
                 parameter to this algorithm  is  the  choice  of  `window
                 function'.   By  default,  SoX uses the Hann window which
                 has good all-round frequency-resolution and dynamic-range
                 properties.   For  better frequency resolution (but lower
                 dynamic-range),  select  a  Hamming  window;  for  higher
                 dynamic-range (but poorer frequency-resolution), select a
                 Kaiser window.  Bartlett and Rectangular windows are also

          -W num Window  adjustment  parameter.   This can be used to make
                 small adjustments to the Kaiser window shape.  A positive
                 number  (up  to  ten)  increases  its  dynamic  range,  a
                 negative number decreases it.

          -s     Allow slack overlapping of DFT  windows.   This  can,  in
                 some  cases,  increase  image  sharpness and give greater
                 adherence to the -x value, but at the expense of a little
                 spectral loss.

          -m     Creates a monochrome spectrogram (the default is colour).

          -h     Selects  a  high-colour  palette - less visually pleasing
                 than the default colour  palette,  but  it  may  make  it
                 easier to differentiate different levels.  If this option
                 is used in conjunction with -m,  the  result  will  be  a
                 hybrid monochrome/colour palette.

          -p num Permute  the  colours in a colour or hybrid palette.  The
                 num parameter, from 1 (the default)  to  6,  selects  the

          -l     Creates  a  `printer  friendly'  spectrogram with a light
                 background (the default has a dark background).

          -a     Suppress  the  display  of  the  axis  lines.   This   is
                 sometimes  useful  in helping to discern artefacts at the
                 spectrogram edges.

          -r     Raw  spectrogram:  suppress  the  display  of  axes   and

          -A     Selects   an  alternative,  fixed  colour-set.   This  is
                 provided  only  for   compatibility   with   spectrograms
                 produced  by  another package.  It should not normally be
                 used as it has  some  problems,  not  least,  a  lack  of
                 differentiation  at  the  bottom  end  which  results  in
                 masking of low-level artefacts.

          -t text
                 Set  the  image  title  -  text  to  display  above   the

          -c text
                 Set  (or clear) the image comment - text to display below
                 and to the left of the spectrogram.

          -o text
                 Name  of  the  spectrogram  output  PNG   file,   default

          Advanced Options:
          In order to process a smaller section of audio without affecting
          other effects or the output signal (unlike when the trim  effect
          is used), the following options may be used.

          -d duration
                 This  option  sets  the X-axis resolution such that audio
                 with the given duration ([[HH:]MM:]SS) fits the  selected
                 (or default) X-axis width.  For example,
                    sox input.mp3 output.wav -n spectrogram -d 1:00 stats
                 creates  a  spectrogram  showing  the first minute of the
                 audio, whilst
                 the stats effect is applied to the entire audio signal.

                 See also -X for an alternative way of setting the  X-axis

          -S time
                 Start  the  spectrogram  at  the given point in the audio
                 stream.  For example
                    sox input.aiff output.wav spectrogram -S 1:00
                 creates a spectrogram showing all but the first minute of
                 the  audio  (the output file however, receives the entire
                 audio stream).

          For the ability to perform off-line processing of spectral data,
          see the stat effect.

   speed factor[c]
          Adjust  the  audio  speed (pitch and tempo together).  factor is
          either the ratio of the new speed to the old speed: greater than
          1  speeds  up,  less than 1 slows down, or, if appended with the
          letter `c', the number of cents (i.e. 100ths of a  semitone)  by
          which  the  pitch (and tempo) should be adjusted: greater than 0
          increases, less than 0 decreases.

          Technically, the speed  effect  only  changes  the  sample  rate
          information, leaving the samples themselves untouched.  The rate
          effect is invoked automatically to resample to the output sample
          rate,  using  its  default quality/speed.  For higher quality or
          higher speed  resampling,  in  addition  to  the  speed  effect,
          specify the rate effect with the desired quality option.

          See also the bend, pitch, and tempo effects.

   splice  [-h|-t|-q] { position[,excess[,leeway]] }
          Splice together audio sections.  This effect provides two things
          over simple audio concatenation: a (usually short) cross-fade is
          applied at the join, and a wave similarity comparison is made to
          help determine the best place at which to make the join.

          One of the options -h, -t, or -q may be given to select the fade
          envelope  as  half-cosine wave (the default), triangular (a.k.a.
          linear), or quarter-cosine wave respectively.

                 Type   Audio          Fade level       Transitions
                  t     correlated     constant gain    abrupt
                  h     correlated     constant gain    smooth
                  q     uncorrelated   constant power   smooth

          To perform a splice, first use the trim  effect  to  select  the
          audio sections to be joined together.  As when performing a tape
          splice, the end of the section to  be  spliced  onto  should  be
          trimmed  with  a  small  excess (default 0.005 seconds) of audio
          after the ideal joining  point.   The  beginning  of  the  audio
          section  to  splice  on  should  be trimmed with the same excess
          (before the ideal joining  point),  plus  an  additional  leeway
          (default  0.005  seconds).   SoX should then be invoked with the
          two audio sections as input files and the  splice  effect  given
          with  the  position  at  which  to  perform the splice - this is
          length of the first audio section (including the excess).

          The following diagram uses the tape analogy  to  illustrate  the
          splice  operation.   The  effect simulates the diagonal cuts and
          joins the two pieces:

                length1   excess
              _________   :   :  _________________
                       \  :   : :\     `
                        \ :   : : \     `
                         \:   : :  \     `
                          *   : :   * - - *
                           \  : :   :\     `
                            \ : :   : \     `
              _______________\: :   :  \_____`____
                                :   :   :     :
                                <--->   <----->
                                excess  leeway

          where * indicates the joining points.

          For example, a long song begins with two verses which start  (as
          determined  e.g. by using the play command with the trim (start)
          effect) at times 0:30.125 and 1:03.432.  The following  commands
          cut out the first verse:
             sox too-long.wav part1.wav trim 0 30.130
          (5 ms excess, after the first verse starts)
             sox too-long.wav part2.wav trim 1:03.422
          (5 ms excess plus 5 ms leeway, before the second verse starts)
             sox part1.wav part2.wav just-right.wav splice 30.130
          For another example, the SoX command
             play "|sox -n -p synth 1 sin %1" "|sox -n -p synth 1 sin %3"
          generates and plays two notes, but there is a nasty click at the
          transition; the click can be  removed  by  splicing  instead  of
          concatenating  the  audio,  i.e.  by  appending  splice 1 to the
          command. (Clicks at the beginning and end of the  audio  can  be
          removed by preceding the splice effect with fade q .01 2 .01).

          Provided your arithmetic is good enough, multiple splices can be
          performed with a single splice invocation.  For example:
          # Audio Copy and Paste Over
          # acpo infile copy-start copy-stop paste-over-start outfile
          # All times measured in samples.
          rate=`soxi -r "$1"`
          e=`expr $rate '*' 5 / 1000`  # Using default excess
          l=$e                         # and leeway.
          sox "$1" piece.wav trim `expr $2 - $e - $l`s \
             `expr $3 - $2 + $e + $l + $e`s
          sox "$1" part1.wav trim 0 `expr $4 + $e`s
          sox "$1" part2.wav trim `expr $4 + $3 - $2 - $e - $l`s
          sox part1.wav piece.wav part2.wav "$5" splice \
             `expr $4 + $e`s \
             `expr $4 + $e + $3 - $2 + $e + $l + $e`s
          In the above Bourne shell script, two splices are used to  `copy
          and paste' audio.

                                *        *        *

          It is also possible to use this effect to perform general cross-
          fades, e.g. to join two  songs.   In  this  case,  excess  would
          typically be an number of seconds, the -q option would typically
          be given (to select an `equal  power'  cross-fade),  and  leeway
          should  be  zero  (which  is  the  default if -q is given).  For
          example, if f1.wav and f2.wav are audio files to be cross-faded,
             sox f1.wav f2.wav out.wav splice -q $(soxi -D f1.wav),3
          cross-fades  the  files  where  the point of equal loudness is 3
          seconds before the end of f1.wav, i.e. the total length  of  the
          cross-fade  is  2  × 3 = 6 seconds (Note: the $(...) notation is
          POSIX shell).

   stat [-s scale] [-rms] [-freq] [-v] [-d]
          Display time and frequency domain statistical information  about
          the   audio.    Audio  is  passed  unmodified  through  the  SoX
          processing chain.

          The information is  output  to  the  `standard  error'  (stderr)
          stream  and  is calculated, where n is the duration of the audio
          in samples, c is the number of audio channels, r  is  the  audio
          sample rate, and xk represents the PCM value (in the range -1 to
          +1 by default) of  each  successive  sample  in  the  audio,  as

           Samples read        n×c
           Length (seconds)    n÷r
           Scaled by                                 See -s below.
           Maximum amplitude   max(xk)               The  maximum sample
                                                     value in the audio;
                                                     usually  this  will
                                                     be    a    positive
           Minimum amplitude   min(xk)               The  minimum sample
                                                     value in the audio;
                                                     usually  this  will
                                                     be    a    negative
           Midline amplitude   ½min(xk)+½max(xk)
           Mean norm           ¹/nΣ│xk│              The  average of the
                                                     absolute  value  of
                                                     each  sample in the
           Mean amplitude      ¹/nΣxk                The average of each
                                                     sample    in    the
                                                     audio.    If   this
                                                     figure is non-zero,
                                                     then  it  indicates
                                                     the  presence  of a
                                                     D.C. offset  (which
                                                     could   be  removed
                                                     using  the  dcshift
           RMS amplitude       √(¹/nΣxk²)            The level of a D.C.
                                                     signal  that  would
                                                     have the same power
                                                     as   the    audio's
                                                     average power.
           Maximum delta       max(│xk-xk-1│)
           Minimum delta       min(│xk-xk-1│)
           Mean delta          ¹/n-1Σ│xk-xk-1RMS delta           √(¹/n-1Σ(xk-xk-1)²)
           Rough frequency                           In Hz.
           Volume Adjustment                         The   parameter  to
                                                     the   vol    effect
                                                     which   would  make
                                                     the audio  as  loud
                                                     as possible without
                                                     clipping.     Note:
                                                     See  the discussion
                                                     on  Clipping  above
                                                     for  reasons why it
                                                     is  rarely  a  good
                                                     idea actually to do

          Note that the delta measurements are not applicable  for  multi-
          channel audio.

          The  -s  option  can  be used to scale the input data by a given
          factor.  The default value of  scale  is  2147483647  (i.e.  the
          maximum  value  of  a  32-bit signed integer).  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.

          The -v option displays only the `Volume Adjustment' value.

          The -freq option calculates the  input's  power  spectrum  (4096
          point  DFT) instead of the statistics listed above.  This should
          only be used with a single channel audio file.

          The -d option displays a hex dump of the 32-bit signed PCM  data
          audio  in  SoX's  internal  buffer.  This is mainly used to help
          track down  endian  problems  that  sometimes  occur  in  cross-
          platform versions of SoX.

          See also the stats effect.

   stats [-b bits|-x bits|-s scale] [-w window-time]
          Display  time  domain  statistical  information  about the audio
          channels; audio is passed unmodified through the SoX  processing
          chain.   Statistics  are calculated and displayed for each audio
          channel and, where applicable, an overall figure is also given.

          For example, for a typical well-mastered stereo music file:

                                   Overall     Left      Right
                      DC offset   0.000803 -0.000391  0.000803
                      Min level  -0.750977 -0.750977 -0.653412
                      Max level   0.708801  0.708801  0.653534
                      Pk lev dB      -2.49     -2.49     -3.69
                      RMS lev dB    -19.41    -19.13    -19.71
                      RMS Pk dB     -13.82    -13.82    -14.38
                      RMS Tr dB     -85.25    -85.25    -82.66
                      Crest factor       -      6.79      6.32
                      Flat factor     0.00      0.00      0.00
                      Pk count           2         2         2
                      Bit-depth      16/16     16/16     16/16
                      Num samples    7.72M
                      Length s     174.973
                      Scale max   1.000000
                      Window s       0.050

          DC offset, Min level, and Max level are shown,  by  default,  in
          the  range  ±1.   If  the -b (bits) options is given, then these
          three measurements will be scaled to a signed integer  with  the
          given  number of bits; for example, for 16 bits, the scale would
          be -32768 to +32767.  The -x option behaves the same way  as  -b
          except   that   the  signed  integer  values  are  displayed  in
          hexadecimal.  The -s option scales the three measurements  by  a
          given floating-point number.

          Pk lev dB  and  RMS lev dB  are  standard  peak  and  RMS  level
          measured in dBFS.  RMS Pk dB and RMS Tr dB are peak  and  trough
          values  for  RMS  level  measured  over  a short window (default

          Crest factor is the standard ratio of peak to RMS  level  (note:
          not in dB).

          Flat factor  is  a  measure  of  the  flatness (i.e. consecutive
          samples with the same value) of the signal at  its  peak  levels
          (i.e.  either  Min level, or Max level).  Pk count is the number
          of occasions  (not  the  number  of  samples)  that  the  signal
          attained either Min level, or Max level.

          The  right-hand  Bit-depth  figure is the standard definition of
          bit-depth i.e. bits less significant than the given  number  are
          fixed  at  zero.   The  left-hand  figure  is the number of most
          significant bits that are fixed at zero  (or  one  for  negative
          numbers)  subtracted  from  the  right-hand  figure  (the number
          subtracted is directly related to Pk lev dB).

          For multi-channel audio, an overall figure for each of the above
          measurements  is  given  and derived from the channel figures as
          follows: DC offset:  maximum  magnitude;  Max level,  Pk lev dB,
          RMS Pk dB,  Bit-depth:  maximum;  Min level, RMS Tr dB: minimum;
          RMS lev dB, Flat factor, Pk count:  average;  Crest factor:  not

          Length s   is   the  duration  in  seconds  of  the  audio,  and
          Num samples is equal to the sample-rate  multiplied  by  Length.
          Scale Max   is   the   scaling   applied   to  the  first  three
          measurements; specifically, it is the maximum value  that  could
          apply  to  Max level.  Window s is the length of the window used
          for the peak and trough RMS measurements.

          See also the stat effect.

   swap   Swap stereo channels.  See also remix for an effect that  allows
          arbitrary channel selection and ordering (and mixing).

   stretch factor [window fade shift fading]
          Change  the  audio duration (but not its pitch).  This effect is
          broadly equivalent to the tempo  effect  with  (factor  inverted
          and)  search  set  to  zero,  so  in  general,  its  results are
          comparatively poor; it is retained  as  it  can  sometimes  out-
          perform tempo for small factors.

          factor  of stretching: >1 lengthen, <1 shorten duration.  window
          size is in ms.  Default is 20ms.  The fade option, can be `lin'.
          shift  ratio, in [0 1].  Default depends on stretch factor. 1 to
          shorten, 0.8 to lengthen.  The fading ratio, in  [0  0.5].   The
          amount of a fade's default depends on factor and shift.

          See also the tempo effect.

   synth [-j KEY] [-n] [len [off [ph [p1 [p2 [p3]]]]]] {[type] [combine]
   [[%]freq[k][:|+|/|-[%]freq2[k]]] [off [ph [p1 [p2 [p3]]]]]}
          This effect can be used to generate  fixed  or  swept  frequency
          audio  tones  with various wave shapes, or to generate wide-band
          noise of various  `colours'.   Multiple  synth  effects  can  be
          cascaded  to produce more complex waveforms; at each stage it is
          possible to choose whether the generated waveform will be  mixed
          with,  or  modulated  onto  the  output from the previous stage.
          Audio for each channel in a  multi-channel  audio  file  can  be
          synthesised independently.

          Though this effect is used to generate audio, an input file must
          still be given, the characteristics of which will be used to set
          the  synthesised  audio  length, the number of channels, and the
          sampling rate; however, since the  input  file's  audio  is  not
          normally  needed,  a  `null  file' (with the special name -n) is
          often given instead (and the length specified as a parameter  to
          synth  or  by  another  given  effect that can has an associated

          For example, the following produces a  3  second,  48kHz,  audio
          file containing a sine-wave swept from 300 to 3300 Hz:
             sox -n output.wav synth 3 sine 300-3300
          and this produces an 8 kHz version:
             sox -r 8000 -n output.wav synth 3 sine 300-3300
          Multiple  channels  can  be synthesised by specifying the set of
          parameters shown between braces multiple  times;  the  following
          puts  the  swept tone in the left channel and adds `brown' noise
          in the right:
             sox -n output.wav synth 3 sine 300-3300 brownnoise
          The following  example  shows  how  two  synth  effects  can  be
          cascaded to create a more complex waveform:
             play -n synth 0.5 sine 200-500 synth 0.5 sine fmod 700-100
          Frequencies can also be given in `scientific' note notation, or,
          by prefixing a `%' character, as a number of semitones  relative
          to  `middle  A'  (440 Hz).   For example, the following could be
          used to help tune a guitar's low `E' string:
             play -n synth 4 pluck %-29
          or with a (Bourne shell) loop, the whole guitar:
             for n in E2 A2 D3 G3 B3 E4; do
               play -n synth 4 pluck $n repeat 2; done
          See the delay effect (above) and the reference to `SoX scripting
          examples' (below) for more synth examples.

          N.B.   This  effect  generates  audio at maximum volume (0dBFS),
          which means that there is a high chance of clipping  when  using
          the  audio  subsequently,  so  in  many  cases, you will want to
          follow this effect with the gain effect  to  prevent  this  from
          happening.  (See  also  Clipping above.)  Note that, by default,
          the synth effect incorporates the functionality of gain -h  (see
          the  gain effect for details); synth's -n option may be given to
          disable this behaviour.

          A detailed description of each synth parameter follows:

          len is the length of audio to synthesise expressed as a time  or
          as a number of samples; 0=inputlength, default=0.

          The format for specifying lengths in time is hh:mm:ss.frac.  The
          format for specifying sample counts is  the  number  of  samples
          with the letter `s' appended to it.

          type is one of sine, square, triangle, sawtooth, trapezium, exp,
          [white]noise,   tpdfnoise    pinknoise,    brownnoise,    pluck;

          combine is one of create, mix, amod (amplitude modulation), fmod
          (frequency modulation); default=create.

          freq/freq2 are the frequencies at the beginning/end of synthesis
          in  Hz  or,  if  preceded  with  `%',  semitones  relative  to A
          (440 Hz); alternatively, `scientific' note  notation  (e.g.  E2)
          may  be  used.  The default frequency is 440Hz.  By default, the
          tuning used with the note notations is `equal temperament';  the
          -j KEY option selects `just intonation', where KEY is an integer
          number of semitones relative to A  (so  for  example,  -9  or  3
          selects the key of C), or a note in scientific notation.

          If  freq2  is  given, then len must also have been given and the
          generated tone will be swept between the given frequencies.  The
          two given frequencies must be separated by one of the characters
          `:', `+', `/', or `-'.  This character is used  to  specify  the
          sweep function as follows:

          :      Linear:  the  tone will change by a fixed number of hertz
                 per second.

          +      Square: a second-order function is  used  to  change  the

          /      Exponential:  the  tone  will change by a fixed number of
                 semitones per second.

          -      Exponential: as `/', but initial phase always  zero,  and
                 stepped (less smooth) frequency changes.

          Not used for noise.

          off is the bias (DC-offset) of the signal in percent; default=0.

          ph  is the phase shift in percentage of 1 cycle; default=0.  Not
          used for noise.

          p1 is the percentage of each cycle that  is  `on'  (square),  or
          `rising'   (triangle,   exp,   trapezium);  default=50  (square,
          triangle, exp),  default=10  (trapezium),  or  sustain  (pluck);

          p2  (trapezium):  the  percentage  through  each  cycle at which
          `falling' begins; default=50. exp: the amplitude in multiples of
          2dB; default=50, or tone-1 (pluck); default=20.

          p3  (trapezium):  the  percentage  through  each  cycle at which
          `falling' ends; default=60, or tone-2 (pluck); default=90.

   tempo [-q] [-m|-s|-l] factor [segment [search [overlap]]]
          Change the audio playback speed but not its pitch.  This  effect
          uses  the WSOLA algorithm. The audio is chopped up into segments
          which are then shifted in the time domain and overlapped (cross-
          faded)  at  points  where  their  waveforms  are most similar as
          determined by measurement of `least squares'.

          By  default,  linear  searches  are  used  to  find   the   best
          overlapping  points. If the optional -q parameter is given, tree
          searches are used instead.  This  makes  the  effect  work  more
          quickly,  but  the result may not sound as good. However, if you
          must improve the processing speed, this  generally  reduces  the
          sound quality less than reducing the search or overlap values.

          The  -m  option  is  used to optimize default values of segment,
          search and overlap for music processing.

          The -s option is used to optimize  default  values  of  segment,
          search and overlap for speech processing.

          The  -l  option  is  used to optimize default values of segment,
          search and overlap for `linear' processing that tends  to  cause
          more  noticeable  distortion  but  may  be useful when factor is
          close to 1.

          If -m, -s, or -l is specified, the default value of segment will
          be  calculated based on factor, while default search and overlap
          values are based  on  segment.  Any  values  you  provide  still
          override these default values.

          factor  gives  the  ratio of new tempo to the old tempo, so e.g.
          1.1 speeds up the tempo by 10%, and 0.9 slows it down by 10%.

          The optional segment parameter selects the  algorithm's  segment
          size  in  milliseconds.   If  no  other flags are specified, the
          default value is 82 and is  typically  suited  to  making  small
          changes to the tempo of music. For larger changes (e.g. a factor
          of 2), 41 ms may give a better result.  The -m, -s, and -l flags
          will  cause  the  segment  default  to be automatically adjusted
          based on factor.  For example using -s (for speech) with a tempo
          of 1.25 will calculate a default segment value of 32.

          The   optional  search  parameter  gives  the  audio  length  in
          milliseconds  over  which  the   algorithm   will   search   for
          overlapping  points.   If  no  other  flags  are  specified, the
          default value is 14.68.  Larger values use more processing  time
          and  may or may not produce better results.  A practical maximum
          is half the value of segment.  Search  can  be  reduced  to  cut
          processing time at the risk of degrading output quality. The -m,
          -s,  and  -l  flags  will  cause  the  search  default   to   be
          automatically adjusted based on segment.

          The  optional overlap parameter gives the segment overlap length
          in milliseconds.  Default value is 12, but -m, -s, or  -l  flags
          automatically  adjust  overlap based on segment size. Increasing
          overlap increases processing time and may  increase  quality.  A
          practical  maximum  for  overlap  is  the  value of search, with
          overlap typically being (at least) a little smaller then search.

          See also speed for  an  effect  that  changes  tempo  and  pitch
          together, pitch and bend for effects that change pitch only, and
          stretch for an effect  that  changes  tempo  using  a  different

   treble gain [frequency[k] [width[s|h|k|o|q]]]
          Apply  a treble tone-control effect.  See the description of the
          bass effect for details.

   tremolo speed [depth]
          Apply a tremolo (low frequency amplitude modulation)  effect  to
          the  audio.   The tremolo frequency in Hz is given by speed, and
          the depth as a percentage by depth (default 40).

   trim {[=|-]position}
          Cuts portions out of the audio.  Any number of positions may  be
          given;  audio is not sent to the output until the first position
          is reached.  The effect  then  alternates  between  copying  and
          discarding audio at each position.

          If  a  position  is  preceded  by an equals or minus sign, it is
          interpreted relative to the beginning or the end of  the  audio,
          respectively.   (The audio length must be known for end-relative
          locations to work.)  Otherwise, it is considered an offset  from
          the  last  position,  or  from  the start of audio for the first
          parameter.  Using a value of 0 for the first position  parameter
          allows copying from the beginning of the audio.

          All  parameters  can be specified using either an amount of time
          or an exact count of samples.  The format for specifying lengths
          in  time  is  hh:mm:ss.frac.   A  value  of 1:30.5 for the first
          parameter will not start until 1 minute, thirty  and  ½  seconds
          into  the audio.  The format for specifying sample counts is the
          number of samples with the letter `s' appended to it.   A  value
          of  8000s  for  the first parameter will wait until 8000 samples
          are read before starting to process audio.

          For example,
             sox infile outfile trim 0 10
          will copy the first ten seconds, while
             play infile trim 12:34 =15:00 -2:00
          will play from 12 minutes 34 seconds into the  audio  up  to  15
          minutes  into  the  audio  (i.e. 2 minutes and 26 seconds long),
          then resume playing two minutes before the end of audio.

   upsample [factor]
          Upsample the signal by an integer  factor:  factor-1  zero-value
          samples  are  inserted between each pair of input samples.  As a
          result,  the  original  spectrum  is  replicated  into  the  new
          frequency space (aliasing) and attenuated.  This attenuation can
          be compensated for  by  adding  vol  factor  after  any  further
          processing.    The   upsample   effect   is  typically  used  in
          combination with filtering effects.

          For a general resampling effect with  anti-aliasing,  see  rate.
          See also downsample.

   vad [options]
          Voice  Activity  Detector.   Attempts  to trim silence and quiet
          background sounds from the ends of (fairly high resolution  i.e.
          16-bit, 44-48kHz) recordings of speech.  The algorithm currently
          uses a simple cepstral power measurement to detect voice, so may
          be  fooled  by  other  things, especially music.  The effect can
          trim only from the front of the audio, so in order to trim  from
          the back, the reverse effect must also be used.  E.g.
             play speech.wav norm vad
          to trim from the front,
             play speech.wav norm reverse vad reverse
          to trim from the back, and
             play speech.wav norm vad reverse vad reverse
          to  trim  from  both  ends.   The  use  of  the  norm  effect is
          recommended, but remember  that  neither  reverse  nor  norm  is
          suitable for use with streamed audio.

          Default values are shown in parenthesis.

          -t num (7)
                 The measurement level used to trigger activity detection.
                 This might need to be  changed  depending  on  the  noise
                 level,  signal level and other charactistics of the input

          -T num (0.25)
                 The time constant (in seconds) used to help ignore  short
                 bursts of sound.

          -s num (1)
                 The   amount   of   audio  (in  seconds)  to  search  for
                 quieter/shorter bursts of audio to include prior  to  the
                 detected trigger point.

          -g num (0.25)
                 Allowed  gap  (in seconds) between quieter/shorter bursts
                 of audio to include prior to the detected trigger point.

          -p num (0)
                 The amount of audio (in seconds) to preserve  before  the
                 trigger point and any found quieter/shorter bursts.

          Advanced Options:
          These allow fine tuning of the algorithm's internal parameters.

          -b num The    algorithm   (internally)   uses   adaptive   noise
                 estimation/reduction in order to detect the start of  the
                 wanted  audio.  This option sets the time for the initial
                 noise estimate.

          -N num Time constant used by the adaptive  noise  estimator  for
                 when the noise level is increasing.

          -n num Time  constant  used  by the adaptive noise estimator for
                 when the noise level is decreasing.

          -r num Amount  of  noise  reduction  to  use  in  the  detection
                 algorithm (e.g. 0, 0.5, ...).

          -f num Frequency of the algorithm's processing/measurements.

          -m num Measurement  duration;  by default, twice the measurement
                 period; i.e.  with overlap.

          -M num Time constant used to smooth spectral measurements.

          -h num `Brick-wall' frequency of high-pass filter applied at the
                 input to the detector algorithm.

          -l num `Brick-wall'  frequency of low-pass filter applied at the
                 input to the detector algorithm.

          -H num `Brick-wall' frequency of high-pass lifter  used  in  the
                 detector algorithm.

          -L num `Brick-wall'  frequency  of  low-pass  lifter used in the
                 detector algorithm.

          See also the silence effect.

   vol gain [type [limitergain]]
          Apply an amplification or an attenuation to  the  audio  signal.
          Unlike the -v option (which is used for balancing multiple input
          files as they enter the SoX effects processing chain), vol is an
          effect  like  any  other so can be applied anywhere, and several
          times if necessary, during the processing chain.

          The amount to change the  volume  is  given  by  gain  which  is
          interpreted, according to the given type, as follows: if type is
          amplitude (or is omitted),  then  gain  is  an  amplitude  (i.e.
          voltage  or  linear) ratio, if power, then a power (i.e. wattage
          or voltage-squared) ratio, and if dB, then a power change in dB.

          When type is amplitude or power, a gain of 1 leaves  the  volume
          unchanged,  less  than  1  decreases  it,  and  greater  than  1
          increases it; a  negative  gain  inverts  the  audio  signal  in
          addition to adjusting its volume.

          When  type  is dB, a gain of 0 leaves the volume unchanged, less
          than 0 decreases it, and greater than 0 increases it.

          See [4] for a detailed discussion on electrical (and hence audio
          signal) voltage and power ratios.

          Beware of Clipping when the increasing the volume.

          The gain and the type parameters can be concatenated if desired,
          e.g.  vol 10dB.

          An optional limitergain value can be specified and should  be  a
          value  much  less than 1 (e.g. 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 the audio that needed to be limited.

          See also  gain  for  a  volume-changing  effect  with  different
          capabilities,     and     compand     for     a    dynamic-range
          compression/expansion/limiting effect.

   Deprecated Effects
   The following effects have been renamed  or  have  their  functionality
   included  in  another  effect; they continue to work in this version of
   SoX but may be removed in future.

   mixer [ -l|-r|-f|-b|-1|-2|-3|-4|n{,n} ]
          Reduce the number of  audio  channels  by  mixing  or  selecting
          channels,  or  increase  the  number  of channels by duplicating
          channels.  Note: this effect  operates  on  the  audio  channels
          within  the  SoX  effects  processing  chain;  it  should not be
          confused with the -m global option  (where  multiple  files  are
          mix-combined before entering the effects chain).

          When  reducing  the number of channels it is possible to use the
          -l, -r, -f, -b, -1, -2, -3, -4, options to select only the left,
          right, front, back channel(s) or specific channel for the output
          instead of averaging the channels.  The -l, and -r options  will
          do  averaging  in quad-channel files so select the exact channel
          to prevent this.

          The mixer effect can also be invoked  with  up  to  16  numbers,
          separated  by commas, which specify the proportion (0 = 0% and 1
          = 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 follows: lf → lf, rf → lf, lb → lf, and rb  →
          rf.   The  next 4 give the right-front output in the same order,
          then left-back and right-back.

          It is also possible to use the 16 numbers to  expand  or  reduce
          the channel count; just specify 0 for unused channels.

          Finally, certain reduced combination of numbers can be specified
          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

          This effect has been superseded by the remix effect that handles
          any number of channels.


   Exit  status  is  0  for  no  error,  1  if there is a problem with the
   command-line  parameters,  or  2  if  an  error  occurs   during   file


   Please report any bugs found in this version of SoX to the mailing list


   soxi(1), soxformat(7), libsox(3)
   audacity(1), gnuplot(1), octave(1), wget(1)
   The SoX web site at
   SoX scripting examples at

   [1]    R. Bristow-Johnson, Cookbook formulae for audio EQ biquad filter

   [2]    Wikipedia, Q-factor,

   [3]    Scott     Lehman,     Effects     Explained,     http://harmony-

   [4]    Wikipedia, Decibel,

   [5]    Richard  Furse,  Linux  Audio  Developer's  Simple  Plugin  API,

   [6]    Richard Furse, Computer Music Toolkit,

   [7]    Steve Harris, LADSPA plugins,


   Copyright 1998-2013 Chris Bagwell and SoX Contributors.
   Copyright 1991 Lance Norskog and Sundry Contributors.

   This program is free software; you can redistribute it and/or modify it
   under the terms of the GNU General Public License as published  by  the
   Free  Software  Foundation;  either  version 2, or (at your option) any
   later version.

   This program is distributed in the hope that it  will  be  useful,  but
   WITHOUT   ANY   WARRANTY;   without   even   the  implied  warranty  of
   General Public License for more details.


   Chris  Bagwell  (   Other  authors  and
   contributors are listed in the ChangeLog file that is distributed  with
   the source code.


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