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 -e signed-integer -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  (for  cer‐
       tain input file formats only; currently, at least FLAC and Ogg Vorbis),
       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 custom‐
       ized  by  placing a %n anywhere in the filename where the number should
       be substituted.  An optional number can be placed after the % to  indi‐
       cate 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
       specification.    The  only  work-around  to  this  is  to  avoid  such
       filenames. This is generally not difficult since most  audio  filenames
       have a filename `extension', whilst effect-names do not.

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

       -      SoX can be used in  simple  pipeline  operations  by  using  the
              special  filename  `-' which, if used as an input filename, will
              cause SoX will read audio data from  `standard  input'  (stdin),
              and  which,  if used as the output filename, will cause SoX will
              send audio data to `standard output' (stdout).  Note  that  when
              using  this option for the output file, and sometimes when using
              it for an input file, the file-type (see -t below) must also  be
              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
              formatted  signals  to  its  standard output, then the following
              command makes a stereo file from two generated signals:
                 sox -M "|genw --imd -" "|genw --thd -" out.wav
              For  headerless  (raw)  audio,  -t  (and  perhaps  other  format
              options) will need to be given, preceding the input command.

       "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  conjunction
              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.

       -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.

       --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.

       position
              A  position  within the audio stream; the syntax is [=|+|-]time‐
              spec, where timespec is a time specification (see  below).   The
              optional first character indicates whether the timespec is to be
              interpreted relative to the start (=) or end (-) of audio, or to
              the  previous  position  if the effect accepts multiple position
              arguments (+).  The audio length must be known for  end-relative
              locations  to  work; some effects do accept -0 for end-of-audio,
              though, even if the length is unknown.  Which of =, +, - is  the
              default  depends  on  the  effect and is shown in its syntax as,
              e.g., position(+).

              Examples: =2:00 (two minutes into the audio stream), -100s  (one
              hundred samples before the end of audio), +0:12+10s (twelve sec‐
              onds and ten samples after the previous position), -0.5+1s  (one
              sample less than half a second before the end of audio).

       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.

       Most effects that expect an audio position or duration in a  parameter,
       i.e. a time specification, accept either of the following two forms:

       [[hours:]minutes:]seconds[.frac][t]
              A  specification  of  `1:30.5' corresponds to one minute, thirty
              and ½ seconds.  The t suffix is entirely optional (however,  see
              the  silence  effect for an exception).  Note that the component
              values do not have to be normalized; e.g.,  `1:23:45',  `83:45',
              `79:0285',  `1:0:1425',  `1::1425'  and `5025' all are legal and
              equivalent to each other.

       sampless
              Specifies the number of samples directly, as  in  `8000s'.   For
              large  sample  counts,  e notation is supported: `1.7e6s' is the
              same as `1700000s'.

       Time specifications can also be chained with + or -  into  a  new  time
       specification  where  the right part is added to or subtracted from the
       left, respectively: `3:00-200s' means two  hundred  samples  less  than
       three minutes.

       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)]   {   start-posi‐
       tion(+),cents,end-position(+) }
              Changes pitch by specified amounts  at  specified  times.   Each
              given  triple:  start-position,cents,end-position  specifies one
              bend.  cents is the number of cents (100 cents = 1 semitone)  by
              which  to bend the pitch. The other values specify the points in
              time at which to start and end bending the pitch, respectively.

              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
              Here, the first bend runs from 0.35 to 0.6, and the  second  one
              from  0.75 to 1.28 seconds.  Note that the clipping that is pro‐
              duced 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 and requires
              the input audio sample rate to be either 44.1kHz or 48kHz.  Max‐
              imum  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 {position(=)}
              Delay one or more audio channels such that  they  start  at  the
              given  position.   For  example,  delay  1.5 +1 3000s delays the
              first channel by 1.5 seconds, the second channel by 2.5  seconds
              (one  second  more than the previous channel), the third channel
              by 3000 samples, and leaves  any  other  channels  that  may  be
              present  un-delayed.   The  following (one long) command plays a
              chime sound:
                 play -n synth -j 3 sin %3 sin %-2 sin %-5 sin %-9 \
                   sin %-14 sin %-21 fade h .01 2 1.5 delay \
                   1.3 1 .76 .54 .27 remix - fade h 0 2.7 2.5 norm -1
              and this plays a guitar chord:
                 play -n synth pl G2 pl B2 pl D3 pl G3 pl D4 pl G4 \
                   delay 0 .05 .1 .15 .2 .25 remix - fade 0 4 .1 norm -1

       dither [-S|-s|-f filter] [-a] [-p precision]
              Apply dithering to the audio.   Dithering  deliberately  adds  a
              small  amount  of  noise  to the signal in order to mask audible
              quantization effects that can occur if the output sample size is
              less than 24 bits.  With no options, this effect will add 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-shaping (if available) 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-position(=) [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  the  time  given  as  fade-in-
              length.  Specify 0 if no fade-in is wanted.

              For  fade-outs, the audio will be truncated at stop-position and
              the signal level will be ramped from full volume down to 0  over
              an  interval  of  fade-out-length  before the stop-position.  If
              fade-out-length is not specified, it defaults to the same  value
              as fade-in-length.  No fade-out is performed if stop-position is
              not specified.  If the audio length can be determined  from  the
              input  file  header  and  any previous effects, then -0 (or, for
              historical reasons, 0) may be  specified  for  stop-position  to
              indicate  the  usual  case of a fade-out that ends at the end of
              the input audio stream.

              Any time specification may be used for fade-in-length and  fade-
              out-length.

              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 [-l|-r] 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.

              Normally, the number of input ports of the plugin must match the
              number of input channels, and the number of output ports  deter‐
              mines  the  output  channel  count.  However, the -r (replicate)
              option allows cloning a  mono  plugin  to  handle  multi-channel
              input.

              Some  plugins introduce latency which SoX may optionally compen‐
              sate for.  The -l (latency  compensation)  option  automatically
              compensates  for latency as reported by the plugin via an output
              control port named "latency".

              If found, the environment variable LADSPA_PATH will be  used  as
              search path for plugins.

       loudness [gain [reference]]
              Loudness  control  -  similar  to  the gain effect, but provides
              equalisation   for   the    human    auditory    system.     See
              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.  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 posi‐
              tions may be specified, provided that a  specified  position  is
              not  less  that the previous one, and any time specification may
              be used for them.  position is optional for the first  and  last
              lengths specified and if omitted correspond to the beginning and
              the end of the audio respectively.  For  example,  pad  1.5  1.5
              adds  1.5  seconds  of silence padding at each end of the audio,
              whilst pad 4000s@3:00 inserts 4000 samples of silence 3  minutes
              into  the  audio.   If  silence is wanted only at the end of the
              audio, specify either the end position or specify a  zero-length
              pad at the start.

              See  also delay for an effect that can add silence at the 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.   The  special  value  -  requests  infinite  repetition.
              Requires temporary file space to store the audio to be repeated.
              Note that repeating once yields two copies: the  original  audio
              and the repeated audio.

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

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

              duration is a time specification with  the  peculiarity  that  a
              bare number is interpreted as a sample count, not as a number of
              seconds.  For specifying seconds, either use the t suffix (as in
              `2t') or specify minutes, too (as in `0:02').

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

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

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

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

              -h     Selects  a  high-colour  palette - less visually pleasing
                     than the default colour palette, but it may make it  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 file
                     Name of the spectrogram output PNG file,  default  `spec‐
                     trogram.png'.   If  `-' is given, the spectrogram will be
                     sent to standard output (stdout).

              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 (a time specification) 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 position(=)
                     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).  Any time specification may be  used  for  these
              parameters.   SoX should then be invoked with the two audio sec‐
              tions as input files and the splice effect given with the  posi‐
              tion  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
              # No chained time specifications allowed for the parameters
              # (i.e. such that contain +/-).
              e=0.005                      # Using default excess
              l=$e                         # and leeway.
              sox "$1" piece.wav trim $2-$e-$l =$3+$e
              sox "$1" part1.wav trim 0 $4+$e
              sox "$1" part2.wav trim $4+$3-$2-$e-$l
              sox part1.wav piece.wav part2.wav "$5" \
                 splice $4+$e +$3-$2+$e+$l+$e
              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.  If the input  is  not  stereo,  pairs  of
              channels  are  swapped,  and  a possible odd last channel passed
              through.  E.g., for seven channels, the output order will be  2,
              1, 4, 3, 6, 5, 7.

              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 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  (any  time  specifica‐
              tion);  a value of 0 indicated to use the input length, which is
              also the default.

              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.  Using a  value  of  0  for  the
              first  position  parameter  allows copying from the beginning of
              the 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
              and
                 play infile trim 12:34 2:26 -2:00
              will both 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 (imaging) 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-imaging,  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.

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                            December 31, 2014                        SoX(1)