SoX(1)                          Sound eXchange                          SoX(1)



NAME
       SoX − Sound eXchange, the Swiss Army knife of audio manipulation

SYNOPSIS
       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]] ...

DESCRIPTION
   Introduction
       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 pur‐
       pose 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 sim‐
       plify playing and recording audio, if SoX is invoked as play, the  out‐
       put  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  Out‐
       put(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.au recital.wav

       translates  an  audio  file  in  Sun AU format to a Microsoft WAV file,
       whilst

          sox recital.au −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
       format,

          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
       effect,

          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
       soxi(1).

   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 com‐
       mand 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’).  Digi‐
              tal 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.

       channels
              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 embed‐
       ded 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
       headers.

   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 cir‐
       cumstances,  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  sup‐
           ported 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

       and

          rec new‐file.wav

       These two commands are functionally equivalent to

          sox existing‐file.wav −d

       and

          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 AUDIO‐
       DRIVER 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

       or

          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
       below.

   Accuracy
       Many  file formats that compress audio discard some of the audio signal
       information whilst doing so. Converting to such a format and then  con‐
       verting  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 prac‐
       tical.

       Formats that discard audio signal information are called ‘lossy’.  For‐
       mats 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 multi‐
       ple SoX invocations.

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

       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 vol‐
       ume  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
       Clipping is distortion that occurs when an audio signal level (or ‘vol‐
       ume’) 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  occurs.

       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 ana‐
       logue  components can introduce a small DC offset and/or amplification,
       all of which can produce distortion if the audio signal level was  ini‐
       tially 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 clip‐
       ping, 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  corre‐
       sponding  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 (includ‐
       ing, 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 adjust‐
       ments 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  (ampli‐
       tude) 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 alternative.

       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 balanc‐
       ing 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
       file.

       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  cus‐
       tomized  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
       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 specifi‐
       cation.   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  spe‐
              cial  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
              given.

       "|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 for‐
              matted  signals  to its standard output, then the following com‐
              mand 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.

       "wildcard‐filename"
              Specifies  that  filename ‘globbing’ (wild‐card matching) should
              be performed by SoX instead of by the shell.  This allows a sin‐
              gle  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
              files.

       −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
              stat).

              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 for‐
       mats and audio device drivers.

OPTIONS
   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"

       MS‐DOS/MS‐Windows:

          set SOX_OPTS=−V −−no‐clobber

       MS‐Windows GUI: via Control Panel : System  :  Advanced  :  Environment
       Variables

       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.

       −−clobber
              Don’t prompt before overwriting an existing file with  the  same
              name as that given for the output file.  This is the default be‐
              haviour.

       −−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  differ‐
              ent 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 con‐
              verted  from  16 to 24 bit with the intention of doing some pro‐
              cessing 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.
              E.g.

                 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.

       −−magic
              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  bene‐
              fit  from  multi‐threaded  processing (e.g. 131072; see −−buffer
              above).

       −−no−clobber
              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 perma‐
              nently enabling it.

       −−norm[=dB‐level]
              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  (usu‐
              ally) 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 typ‐
              ically 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 con‐
              figuration of many of the transfer‐function based effects.   For
              the  first given effect that supports the selected plotting pro‐
              gram, SoX will output commands to  plot  the  effect’s  transfer
              function,  and  then exit without actually processing any audio.
              E.g.

                 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  succes‐
              sive  SoX  invocations with the same inputs and the same parame‐
              ters 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 digi‐
              tal 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  solu‐
              tion.

       −−version
              Show SoX’s version number and exit.

       −V[level]
              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
                     audio.

              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 ver‐
              bosity 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.

       −−ignore−length
              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 ‘bal‐
              anced’ w.r.t. the other input files.  This is a  linear  (ampli‐
              tude)  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 immedi‐
       ately 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’ (‘header‐
              less’) 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’
              file.

              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.

       −1/−2/−3/−4/−8
              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 sup‐
              ported 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’
              file.

                 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  speci‐
              fying  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:

              signed‐integer
                     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.

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

              floating‐point
                     PCM data stored as IEEE 753 single precision (32‐bit)  or
                     double  precision  (64‐bit)  floating‐point (‘real’) num‐
                     bers.  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 encod‐
                     ing to 8 bits per sample.  A.k.a. μ‐law.  It has a preci‐
                     sion  equivalent  to  roughly 14‐bit PCM and is sometimes
                     encoded with reversed bit‐ordering (see the −X option).

              oki‐adpcm
                     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‐adpcm
                     IMA  (a.k.a. DVI) 4‐bit ADPCM; it has a precision equiva‐
                     lent to roughly 13‐bit PCM.

              ms‐adpcm
                     Microsoft 4‐bit ADPCM; it has a precision  equivalent  to
                     roughly 14‐bit PCM.

              gsm‐full‐rate
                     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 asso‐
                     ciated speech quality.  SoX has support for GSM’s  origi‐
                     nal  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 float‐
              ing‐point ‘WAV’ files (single & double precision  respectively).

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

       −s/−u/−f/−A/−U/−o/−i/−a/−g
              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
              above).

       −−no−glob
              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‐sec‐
              onds.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
              file.

              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 exam‐
              ple, if audio was recorded with a sample‐rate of say 48k from  a
              source that played back a little, say 1.5%, too slowly, then

                 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 speci‐
              fying 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 equivalent:

                 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 arbi‐
              trarily.

       −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
              example:

                 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 out‐
              put 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)  for‐
              mats.

              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 −−com‐
              ment−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 out‐
              put file header (where applicable).

       −C, −−compression FACTOR
              The compression factor for variably compressing output file for‐
              mats.   If  this  option is not given then a default compression
              factor will apply.  The compression factor is  interpreted  dif‐
              ferently  for  different  compressing  file  formats.   See  the
              description of the file formats that use this option in  soxfor‐
              mat(7) for more information.

EFFECTS
       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 com‐
       mand 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 sin‐
       gle 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 dis‐
       carded.  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 out‐
       put 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 param‐
       eters 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:

       center[k]
              See frequency.

       frequency[k]
              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
              attenuation.

       width[h|k|o|q]
              Used to specify the band‐width of a filter.  A number of differ‐
              ent 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 accompany‐
       ing ‘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  loga‐
              rithmically  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: con‐
              stant 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 simi‐
              lar 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 Trans‐
              form  (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 pitch.

              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  respec‐
              tively.

              See http://en.wikipedia.org/wiki/Digital_biquad_filter (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 explic‐
              itly, 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 modula‐
              tion  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 exam‐
              ple, 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}
              [soft‐knee‐dB:]in‐dB1[,out‐dB1]{,in‐dB2,out‐dB2}
              [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 aver‐
              aged to determine its volume; attacks refer to increases in vol‐
              ume  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 sensi‐
              tive 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 possi‐
              ble 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 com‐
              panded (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 adja‐
              cent 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 exam‐
              ple, 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 compan‐
              der  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
              function).

              See also mcompand for a multiple‐band companding effect.

       contrast [enhancement‐amount(75)]
              Comparable with compression, this effect modifies an audio  sig‐
              nal  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 con‐
              trast 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‐empha‐
              sised  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 follow‐
              ing (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 trian‐
              gular (TPDF) white noise.  Noise‐shaping (only for certain  sam‐
              ple  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‐shi‐
              bata.   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  http://sox.sourceforge.net/SoX/NoiseShaping  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  modula‐
              tion;  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 lis‐
              tened 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 build‐
              ings) 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)  ro‐
              bot 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 moun‐
              tains:

                 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.   Spec‐
              ify 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 speci‐
              fied.  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 exam‐
              ple ‘8000s’).

              See also the splice effect.

       fir [coefs‐file|coefs]
              Use  SoX’s  FFT convolution engine with given FIR filter coeffi‐
              cients.  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  file‐
              name  is  ‘−’, or if no argument is given, then the coefficients
              are read from the ‘standard input’ (stdin);  otherwise,  coeffi‐
              cients may be given on the command line.  Examples:

                 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
                   1.2311233052619888e−01
                  −4.4777096106211783e−01
                   5.1031563346705155e−01
                  −6.6502926320995331e−02
                 ...


              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:
                                            linear|quadratic.

       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 giv‐
              ing −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 neces‐
              sary  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 head‐
              room 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 sig‐
              nal  generation.   The process is often written as a multiplica‐
              tion by i (or j), the imaginary unit.

              An odd‐tap Hilbert transform filter has a bandpass  characteris‐
              tic,  attenuating the lowest and highest frequencies.  Its band‐
              width 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 vari‐
              able 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
              http://en.wikipedia.org/wiki/Loudness 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 refer‐
              ence 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}
              [soft‐knee‐dB:]in‐dB1[,out‐dB1]{,in‐dB2,out‐dB2}
              [gain     [initial‐volume‐dB    [delay]]]"    {crossover‐freq[k]
              "attack1,..."}

              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
              pre‐emphasis.

              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 back‐
              ground 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 sam‐
              ples.   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, spec‐
              ify  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  begin‐
              ning 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 modu‐
              lation 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

              Gentler:

                 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
                                     width
                     −q     quick     n/a   ≈30 @    playback on
                                             Fs/4    ancient hardware
                     −l      low      80%    100     playback on old
                                                     hardware
                     −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  com‐
              mands 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  arbi‐
              trarily.

                                    *        *        *

              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  over‐
              ride 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 sam‐
              pling rates but that if this is at least 44.1kHz, then the arte‐
              facts 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 lin‐
              ear 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 fre‐
              quency  content of the original signal (w.r.t. the original sam‐
              ple rate when up‐sampling, or the new sample rate when down‐sam‐
              pling)  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 tran‐
              sient 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/audi‐
              bility),  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%.

              Examples:

                 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
              file.

                                    *        *        *

              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 =
                                                    invert

              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 adjust‐
              ments, 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,
              then

                 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 chan‐
              nels).  Where more than a few channels are  involved,  a  script
              such as the following (Bourne shell script) is useful:

              #!/bin/sh
              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`
              done

              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 orig‐
              inal 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 reverbera‐
              tion.

              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
              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 peri‐
              ods 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  sec‐
              ond  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 mid‐
              dle 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 posi‐
              tive  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
              audio.

              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  sam‐
              ples.   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 silence).

              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 parame‐
              ters 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
              respectively.

              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  frequen‐
              cies; 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 unmodi‐
              fied 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  repre‐
              sented by the colour (or optionally the intensity) of the pixels
              in the X‐Y plane.  If the audio signal contains  multiple  chan‐
              nels then these are shown from top to bottom starting from chan‐
              nel 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 sen‐
              sitive.  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 spectro‐
              gram 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 smi‐
              lar 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  spectro‐
              gram; 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.

              Options:

              −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 resolu‐
                     tion) and need not be an integer.  SoX may make a  slight
                     adjustment  to  the given number for processing quantisa‐
                     tion 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  spec‐
                     trogram 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 spec‐
                     trogram channels that is one more than a power of two, so
                     the  actual total height may fall short of the given num‐
                     ber.  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 spectro‐
                     gram 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  spec‐
                     trogram display, and vice versa.

              −q num Sets  the Z‐axis quantisation, i.e. the number of differ‐
                     ent 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  magni‐
                     tude  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 values.

              −w name
                     Window: Hann (default), Hamming, Bartlett, Rectangular or
                     Kaiser.   The  spectrogram is produced using the Discrete
                     Fourier Transform (DFT) algorithm.  A significant parame‐
                     ter 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  win‐
                     dow.   Bartlett  and  Rectangular windows are also avail‐
                     able.

              −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  nega‐
                     tive 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  eas‐
                     ier 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
                     permutation.

              −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  some‐
                     times useful in helping to discern artefacts at the spec‐
                     trogram edges.

              −r     Raw spectrogram: suppress the display of  axes  and  leg‐
                     ends.

              −A     Selects  an  alternative, fixed colour‐set.  This is pro‐
                     vided 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  spectro‐
                     gram.

              −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  ‘spec‐
                     trogram.png’.

              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
                     resolution.

              −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,  spec‐
              ify 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  sec‐
              tion 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 con‐
              catenating 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:

              #!/bin/sh
              # 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 typi‐
              cally  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, then

                 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  process‐
              ing 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 fol‐
              lows:

               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 num‐
                                                         ber.
               Minimum amplitude   min(xk)               The minimum  sample
                                                         value in the audio;
                                                         usually  this  will
                                                         be  a negative num‐
                                                         ber.
               Midline amplitude   ½min(xk)+½max(xk)
               Mean norm           ¹/nΣ│xk│              The average of  the
                                                         absolute  value  of
                                                         each sample in  the
                                                         audio.
               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
                                                         effect).
               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−1│
               RMS 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
                                                         this.

              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 max‐
              imum 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‐plat‐
              form 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 hexadeci‐
              mal.  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 mea‐
              sured in dBFS.  RMS Pk dB and RMS Tr dB are peak and trough val‐
              ues for RMS level measured over a short window (default 50ms).

              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  sam‐
              ples 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  sig‐
              nificant  bits  that are fixed at zero (or one for negative num‐
              bers) subtracted from the right‐hand  figure  (the  number  sub‐
              tracted 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
              applicable.

              Length s is the duration in seconds of the audio,  and  Num sam‐
              ples   is   equal  to  the  sample‐rate  multiplied  by  Length.
              Scale Max is the scaling applied to  the  first  three  measure‐
              ments; 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 compara‐
              tively  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  cas‐
              caded  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 syn‐
              thesised 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 nor‐
              mally  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 length).

              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 cas‐
              caded 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  fol‐
              low  this  effect with the gain effect to prevent this from hap‐
              pening. (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 dis‐
              able 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;
              default=sine.

              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
                     tone.

              /      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, trian‐
              gle,  exp),  default=10   (trapezium),   or   sustain   (pluck);
              default=40.

              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  overlap‐
              ping  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  over‐
              ride 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 mil‐
              liseconds 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 over‐
              lap 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 algo‐
              rithm.

       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 dis‐
              carding 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  fre‐
              quency space (aliasing) and attenuated.  This attenuation can be
              compensated for by adding vol factor after any further  process‐
              ing.   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 recom‐
              mended, but remember that neither reverse nor norm  is  suitable
              for use with streamed audio.

              Options:
              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
                     audio.

              −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 qui‐
                     eter/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 estima‐
                     tion/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  algo‐
                     rithm (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 inter‐
              preted,  according  to  the  given  type, as follows: if type is
              amplitude (or is omitted), then gain is an amplitude (i.e. volt‐
              age  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  addi‐
              tion 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  capa‐
              bilities,  and  compand  for  a dynamic‐range compression/expan‐
              sion/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 chan‐
              nels,  or  increase  the number of channels by duplicating chan‐
              nels.  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, sep‐
              arated  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.

DIAGNOSTICS
       Exit  status  is  0 for no error, 1 if there is a problem with the com‐
       mand‐line parameters, or 2 if an error occurs during file processing.

BUGS
       Please report any bugs found in this version of SoX to the mailing list
       (sox‐users@lists.sourceforge.net).

SEE ALSO
       soxi(1), soxformat(7), libsox(3)
       audacity(1), gnuplot(1), octave(1), wget(1)
       The SoX web site at http://sox.sourceforge.net
       SoX scripting examples at http://sox.sourceforge.net/Docs/Scripts

   References
       [1]    R. Bristow‐Johnson, Cookbook formulae for audio EQ biquad filter
              coefficients, http://musicdsp.org/files/Audio‐EQ‐Cookbook.txt

       [2]    Wikipedia, Q‐factor, http://en.wikipedia.org/wiki/Q_factor

       [3]    Scott    Lehman,    Effects    Explained,    http://harmony‐cen‐
              tral.com/Effects/effects‐explained.html

       [4]    Wikipedia, Decibel, http://en.wikipedia.org/wiki/Decibel

       [5]    Richard  Furse,  Linux  Audio  Developer’s  Simple  Plugin  API,
              http://www.ladspa.org

       [6]    Richard Furse, Computer Music Toolkit, http://www.ladspa.org/cmt

       [7]    Steve Harris, LADSPA plugins, http://plugin.org.uk

LICENSE
       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 MER‐
       CHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU  General
       Public License for more details.

AUTHORS
       Chris Bagwell (cbagwell@users.sourceforge.net).  Other authors and con‐
       tributors are listed in the ChangeLog file that is distributed with the
       source code.



sox                            February 1, 2013                         SoX(1)
