Provided by: sox_14.4.2+git20190427-3_amd64 bug


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


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

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

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


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

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

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

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

                                          *        *        *

       The overall SoX processing chain can be summarised as follows:

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

       Note however, that on the SoX command line, the positions of the Output(s) and the Effects
       are swapped w.r.t. the logical flow just shown.  Note also that whilst options  pertaining
       to  files  are placed before their respective file name, the opposite is true for effects.
       To show how this works in practice, here is a selection of examples of how  SoX  might  be
       used.  The simple
          sox recital.wav
       translates an audio file in Sun AU format to a Microsoft WAV file, whilst
          sox -b 16 recital.wav channels 1 rate 16k fade 3 norm
       performs  the  same  format  translation,  but  also applies four effects (down-mix to one
       channel, sample rate change, fade-in, nomalize), and stores the result at a  bit-depth  of
          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
       command line or inferred from those of the input file.

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

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

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

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

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

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

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

       One  important  use  of  audio  file comments is to convey `Replay Gain' information.  SoX
       supports applying Replay Gain information (for certain 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  circumstances,  individual
       characteristics may be determined or set using different mechanisms.

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

       1.  Command-line format options.

       2.  The contents of the file header.

       3.  The filename extension.

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

       1.  Command-line format options.

       2.  The filename extension.

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

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

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

                                          *        *        *

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

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

                                          *        *        *

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

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

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

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

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

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

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

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

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

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

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

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

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

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

       Clipping is distortion that occurs when an audio signal level (or  `volume')  exceeds  the
       range  of the chosen representation.  In most cases, clipping is undesirable and so should
       be corrected by adjusting the level prior to the point (in the processing chain) at  which
       it 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 analogue components can
       introduce a small DC offset and/or amplification, all of which can produce  distortion  if
       the audio signal level was initially too close to the clipping point.

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

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

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

       See also -G and the gain and norm effects.

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

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

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

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

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

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

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

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

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

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

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

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

   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  customized by placing a %n anywhere in the filename
       where the number should be substituted.  An optional number can be placed after the  %  to
       indicate a minimum fixed width for the number.

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

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

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

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

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


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

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

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

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

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


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

       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

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

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

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

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

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

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

       -G, --guard
              Automatically invoke the gain effect to guard against clipping. 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.

              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 benefit from multi-threaded
              processing (e.g. 131072; see --buffer above).

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

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

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

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

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

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

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

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

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

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

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

       -T     Equivalent to --combine multiply.

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

              Show SoX's version number and exit.

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

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

              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 verbosity level by 1.  Alternatively, the
              verbosity level can be set to an absolute number by specifying it immediately after
              the -V, e.g.  -V0 sets it to 0.

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

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

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

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

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

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

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

              For an input file, the most common use for this option is  to  inform  SoX  of  the
              number  of  channels in a `raw' (`headerless') audio file.  Occasionally, it may be
              useful to use this option  with  a  `headered'  file,  in  order  to  override  the
              (presumably  incorrect) value in the header - note that this is only supported with
              certain file types.  Examples:
                 sox -r 48k -e float -b 32 -c 2 input.raw output.wav
              converts a particular `raw' file to a self-describing `WAV' 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  specifying  that  the
              channels  effect  should be invoked in order to change (if necessary) the number of
              channels in the audio signal to the number given.  For example, the  following  two
              commands are equivalent:
                 sox input.wav -c 1 output.wav bass -b 24
                 sox input.wav      output.wav bass -b 24 channels 1
              though  the  second  form  is  more flexible as it allows the effects to be ordered

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

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

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

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

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

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

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

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

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

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

              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

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

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

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

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

       -r, --rate RATE[k]
              Gives the sample rate in Hz (or kHz if appended with `k') of the 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 example, if audio was recorded
              with a sample-rate of say 48k from a source that played back a  little,  say  1.5%,
              too slowly, 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 specifying that  the  rate
              effect  should  be invoked in order to change (if necessary) the sample rate of the
              audio signal to the given value.  For  example,  the  following  two  commands  are
                 sox input.wav -r 48k output.wav bass -b 24
                 sox input.wav        output.wav bass -b 24 rate 48k
              though  the second form is more flexible as it allows rate options to be given, and
              allows the effects to be ordered arbitrarily.

       -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 output file
              that is actually an audio device.

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

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

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

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

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

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

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

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

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

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

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

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


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

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

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

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

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

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

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

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

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

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

              See frequency.

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

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

              A position within the audio stream; the syntax is [=|+|-]timespec,  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  seconds  and  ten  samples  after  the
              previous  position),  -0.5+1s (one sample less than half a second before the end of

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

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

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

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

              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.

              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 accompanying `README' file.

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

              This effect supports the --plot global option.

       band [-n] center[k] [width[h|k|o|q]]
              Apply a band-pass filter.  The frequency response drops logarithmically around  the
              center  frequency.   The  width  parameter  gives  the  slope  of  the  drop.   The
              frequencies at center + width and center - width will be  half  of  their  original
              amplitudes.   band  defaults  to  a  mode  oriented  to  pitched audio, i.e. voice,
              singing, or instrumental music.  The -n (for noise) option uses the alternate  mode
              for  un-pitched  audio  (e.g.  percussion).  Warning: -n introduces a power-gain of
              about 11dB in the filter, so beware of output clipping.  band introduces  noise  in
              the  shape  of the filter, i.e. peaking at the center frequency and settling around

              This effect supports the --plot global option.

              See also sinc for a bandpass filter with steeper shoulders.

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

              These effects support the --plot global option.

              See also sinc for a bandpass filter with steeper shoulders.

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

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

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

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

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

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

              The filters are described in detail in [1].

              These effects support the --plot global option.

              See also equalizer for a peaking equalisation effect.

       bend [-f frame-rate(25)] [-o over-sample(16)] { start-position(+),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 Transform (DFT) at a
              particular frame rate and over-sampling rate.  The -f and -o parameters may be used
              to adjust these parameters and thus control the smoothness of the changes in 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 produced in this example is deliberate; to
              remove it, use gain -5 in place of gain 1.

              See also pitch.

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

              See (where a0 = 1).

              This effect supports the --plot global option.

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

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

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

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

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

              Each  four-tuple  parameter delay/decay/speed/depth gives the delay in milliseconds
              and the decay (relative to gain-in) with a modulation speed in Hz  using  depth  in
              milliseconds.   The modulation is either sinusoidal (-s) or triangular (-t).  Gain-
              out is the volume of the output.

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

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

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

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

              The second parameter is a list of  points  on  the  compander's  transfer  function
              specified  in  dB  relative  to  the  maximum possible signal amplitude.  The input
              values must be in a strictly increasing order but the transfer  function  does  not
              have  to be monotonically rising.  If omitted, the value of out-dB1 defaults to the
              same value as in-dB1; levels below in-dB1 are not  companded  (but  may  have  gain
              applied  to  them).  The point 0,0 is assumed but may be overridden (by 0,out-dBn).
              If the list is preceded by a soft-knee-dB value, then the points at where  adjacent
              line  segments  on  the transfer function meet will be rounded by the amount given.
              Typical values for the transfer function are 6:-70,-60,-20.

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

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

              The fifth (optional) parameter is a delay in seconds.  The input signal is analysed
              immediately to control the compander, but it is delayed before  being  fed  to  the
              volume  adjuster.  Specifying a delay approximately equal to the attack/decay times
              allows the compander to  effectively  operate  in  a  `predictive'  rather  than  a
              reactive mode.  A typical value is 0.2 seconds.

                                              *        *        *

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

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

              See also mcompand for a multiple-band companding effect.

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

              See also the compand and mcompand effects.

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

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

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

                                              *        *        *

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

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

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

              With  the deemph effect, it is possible to apply the necessary de-emphasis to audio
              that has been extracted from a pre-emphasised CD, and  then  either  burn  the  de-
              emphasised  audio to a new CD (which will then play correctly on any CD player), or
              simply play the correctly de-emphasised audio files on the PC.  For example:
                 sox track1.wav track1-deemph.wav deemph
              and then burn track1-deemph.wav to CD, or
                 play track1-deemph.wav
              or simply
                 play track1.wav deemph
              The de-emphasis filter is implemented as a biquad  and  requires  the  input  audio
              sample  rate  to  be  either  44.1kHz  or  48kHz.  Maximum deviation from the ideal
              response is only 0.06dB (up to 20kHz).

              This effect supports the --plot global option.

              See also the bass and treble shelving equalisation effects.

       delay {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
              triangular  (TPDF)  white noise.  Noise-shaping (only for certain sample rates) can
              be selected with -s.  With the -f option, it is possible  to  select  a  particular
              noise-shaping  filter  from  the  following list: lipshitz, f-weighted, modified-e-
              weighted, improved-e-weighted, gesemann, shibata, low-shibata, high-shibata.   Note
              that  most  filter  types  are available only with 44100Hz sample rate.  The filter
              types are distinguished by the following properties: audibility of noise, level  of
              (inaudible, but in some circumstances, otherwise problematic) shaped high frequency
              noise, and processing speed.
              See for graphs of the different  noise-
              shaping curves.

              The  -S option selects a slightly `sloped' TPDF, biased towards higher frequencies.
              It can be used at any sampling rate but below ≈22k, plain TPDF is probably  better,
              and above ≈ 37k, noise-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 modulation; if this occurs,
              then either re-dither the whole file, or use trim, fade, and concatencate.

              The -p option allows overriding the target precision.

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

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

              See also the `Dithering' section above.

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

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

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

       earwax Makes audio easier to listen to on headphones.  Adds `cues' to 44.1kHz stereo (i.e.
              audio CD format) audio so that when listened to on headphones the stereo  image  is
              moved  from  inside  your head (standard for headphones) to outside and in front of
              the listener (standard for speakers).

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

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

              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 coefficients.  If a single
              argument is given then this is treated as the name of a file containing the  filter
              coefficients  (white-space  separated;  may  contain  `#'  comments).  If the given
              filename is `-', or if no argument is given, then the coefficients  are  read  from
              the  `standard  input' (stdin); otherwise, coefficients may be given on the command
              line.  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

              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

              All parameters are optional (right to left).

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

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

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

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

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

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

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

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

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

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

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

              See also the norm and vol effects.

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

              These effects support the --plot global option.

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

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

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

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

              This effect supports the --plot global option.

       ladspa [-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  determines  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  compensate  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

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

              See also the gain effect.

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

       mcompand "attack1,decay1{,attack2,decay2}
              [gain [initial-volume-dB [delay]]]" {crossover-freq[k] "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 background noise such as hiss or hum.
              To use it, first run SoX with the noiseprof effect  on  a  section  of  audio  that
              ideally  would  contain  silence  but  in  fact  contains noise - such sections are
              typically found at the beginning or the end of a recording.  noiseprof  will  write
              out  a  noise profile to profile-file, or to stdout if no profile-file or if `-' is
              given.  E.g.
                 sox speech.wav -n trim 0 1.5 noiseprof speech.noise-profile
              To actually remove the noise, run SoX again, this time with  the  noisered  effect;
              noisered  will  reduce  noise  according to a noise profile (which was generated by
              noiseprof), from profile-file, or from stdin if no profile-file or if `-' is given.
                 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

              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

       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 positions 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 beginning of the audio on
              a channel-by-channel basis.

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

              delay/decay/speed gives the delay in milliseconds and the decay (relative to  gain-
              in)  with  a  modulation  speed in Hz.  The modulation is either sinusoidal (-s)  -
              preferable for multiple instruments, or triangular (-t)  - gives single instruments
              a sharper phasing effect.  The decay should be less than 0.5 to avoid feedback, and
              usually no less than 0.1.  Gain-out is the volume of the output.

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

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

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

              See also the bend, speed, and tempo effects.

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

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

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

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

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

                                              *        *        *

              Warning: technically detailed discussion follows.

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

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

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

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

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

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

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

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

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

                 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

                            Letter   Volume number        Notes
                              p      power adjust in dB   0 = no change
                              i      power adjust in dB   As `p', but invert the
                              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 adjustments, so
                 sox input.wav output.wav remix -m 1,2 3,4v0.8
              results in channel level multipliers of 1,1 1,0.8.

              The volume number is optional and omitting it  corresponds  to  no  volume  change;
              however,  the  only  case  in  which  this is useful is in conjunction with i.  For
              example, if input.wav is stereo, 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  channels).   Where  more than a few channels are
              involved, a script such as the following (Bourne shell script) is useful:
              chans=`soxi -c "$1"`
              while [ $chans -ge 1 ]; do
                 chans0=`printf %02i $chans`   # 2 digits hence up to 99 chans
                 out=`echo "$1"|sed "s/\(.*\)\.\(.*\)/\1-$chans0.\2/"`
                 sox "$1" "$out" remix $chans
                 chans=`expr $chans - 1`
              If a file input.wav containing six audio channels  were  given,  the  script  would
              produce six output files: input-01.wav, input-02.wav, ..., input-06.wav.

              See also the swap effect.

       repeat [count(1)|-]
              Repeat  the  entire  audio count times, or once if count is not given.  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 reverberation.

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

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

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

              This effect supports the --plot global option.

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

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

              The  above-periods  value  is  used  to  indicate if audio should be trimmed at the
              beginning of the audio. A value of zero indicates no silence should be trimmed from
              the beginning. When specifying 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

              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  periods  of  silence that are wanted.  For example, if you have a song with 2
              seconds of silence in the middle and 2 second at the  end,  you  could  set  below-
              period to a value of 2 to skip over the silence in the middle of the audio.

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

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

              To  remove  silence  from  the  middle  of  a file, specify a below-periods that is
              negative.  This value is then treated as a positive  value  and  is  also  used  to
              indicate  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

              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
              Apply a sinc kaiser-windowed low-pass, high-pass, band-pass, or band-reject  filter
              to  the  signal.   The freqHP and freqLP parameters give the frequencies of the 6dB
              points of a high-pass and low-pass filter that  may  be  invoked  individually,  or
              together.   If  both  are  given,  then freqHP less than freqLP creates a band-pass
              filter, freqHP greater than freqLP creates a band-reject filter.  For example,  the
                 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  frequencies; one of these options given to the
              right of the frequencies applies only to freqLP.

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

              This effect supports the --plot global option.

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

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

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


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

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

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

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

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

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

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

              -w name
                     Window: Hann (default), Hamming, Bartlett,  Rectangular,  Kaiser  or  Dolph.
                     The  spectrogram  is  produced  using  the  Discrete Fourier Transform (DFT)
                     algorithm.  A significant parameter to  this  algorithm  is  the  choice  of
                     `window function'.  By default, SoX uses the Hann window which has good all-
                     round  frequency-resolution  and  dynamic-range  properties.    For   better
                     frequency resolution (but lower dynamic-range), select a Hamming window; for
                     higher dynamic-range  (but  poorer  frequency-resolution),  select  a  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 easier to differentiate different levels.
                     If  this  option is used in conjunction with -m, the result will be a hybrid
                     monochrome/colour palette.

              -p num Permute the colours in a colour or hybrid palette.  The num parameter,  from
                     1 (the default) to 6, selects the 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 sometimes useful in helping
                     to discern artefacts at the spectrogram edges.

              -r     Raw spectrogram: suppress the display of axes and legends.

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

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

              -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 `spectrogram.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

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

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

              See also the bend, pitch, and tempo effects.

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

              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

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

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

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

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

              where * indicates the joining points.

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

              Provided your arithmetic is good enough, multiple splices can be performed  with  a
              single splice invocation.  For example:
              # Audio Copy and Paste Over
              # acpo infile copy-start copy-stop paste-over-start outfile
              # 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 typically be an number of  seconds,  the  -q
              option would typically be given (to select an `equal power' cross-fade), and leeway
              should be zero (which is the default if -q is given).  For example, if  f1.wav  and
              f2.wav are audio files to be cross-faded, 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 processing chain.

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

                      Samples read        n×c
                      Length (seconds)    n÷r
                      Scaled by                                 See -s below.
                      Maximum amplitude   max(xk)               The maximum sample value
                                                                in  the  audio;  usually
                                                                this  will be a positive
                      Minimum amplitude   min(xk)               The minimum sample value
                                                                in  the  audio;  usually
                                                                this will be a  negative
                      Midline amplitude   ½min(xk)+½max(xk)
                      Mean norm           ¹/nΣ│xk│              The   average   of   the
                                                                absolute value  of  each
                                                                sample in the 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-1RMS delta           √(¹/n-1Σ(xk-xk-1)²)
                      Rough frequency                           In Hz.
                      Volume Adjustment                         The parameter to the vol
                                                                effect which would  make
                                                                the  audio  as  loud  as
                                                                possible         without
                                                                clipping.  Note: See the
                                                                discussion  on  Clipping
                                                                above for reasons why it
                                                                is rarely  a  good  idea
                                                                actually to do 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 maximum value of a 32-bit  signed  integer).
              Internal  effects  always  work  with  signed long PCM data and so the value should
              relate to this fact.

              The -rms option will convert all  output  average  values  to  `root  mean  square'

              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

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

              See also the stats effect.

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

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

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

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

              Pk lev dB  and  RMS lev dB  are  standard  peak  and  RMS  level  measured in dBFS.
              RMS Pk dB and RMS Tr dB are peak and trough values for RMS level  measured  over  a
              short window (default 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 samples with the same
              value) of the signal at its peak levels  (i.e.  either  Min level,  or  Max level).
              Pk count  is  the  number  of occasions (not the number of samples) that the signal
              attained either Min level, or Max level.

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

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

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

              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 comparatively poor; it is retained as it can sometimes out-perform
              tempo for small factors.

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

              See also the tempo effect.

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

              Though  this  effect  is used to generate audio, an input file must still be given,
              the characteristics of which will be used to set the synthesised audio length,  the
              number of channels, and the sampling rate; however, since the input file's audio is
              not normally needed, a `null file' (with  the  special  name  -n)  is  often  given
              instead  (and  the  length  specified  as  a parameter to synth or by another given
              effect that 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 cascaded to create a more
              complex waveform:
                 play -n synth 0.5 sine 200-500 synth 0.5 sine fmod 700-100
              Frequencies can also be given in `scientific' note notation, or, by prefixing a `%'
              character,  as a number of semitones relative to `middle A' (440 Hz).  For example,
              the following could be used to help tune a guitar's low `E' string:
                 play -n synth 4 pluck %-29
              or with a (Bourne shell) loop, the whole guitar:
                 for n in E2 A2 D3 G3 B3 E4; do
                   play -n synth 4 pluck $n repeat 2; done
              See the delay effect (above) and the reference to `SoX scripting examples'  (below)
              for more synth examples.

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

              A detailed description of each synth parameter follows:

              len is the length of audio to synthesise (any time specification);  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,  triangle, 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);

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

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

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

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

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

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

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

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

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

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

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

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

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

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

       trim {position(+)}
              Cuts portions out of the audio.  Any number of positions may be given; audio is not
              sent to the output until the first position is reached.  The effect then alternates
              between copying and discarding audio at each position.  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
                 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 frequency space (imaging) and attenuated.  This attenuation
              can be compensated for by adding vol factor  after  any  further  processing.   The
              upsample effect is typically used in combination with filtering effects.

              For a general resampling effect with anti-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  recommended,  but  remember
              that neither reverse nor norm is suitable for use with streamed audio.

              Default values are shown in parenthesis.

              -t num (7)
                     The  measurement  level used to trigger activity detection.  This might need
                     to be  changed  depending  on  the  noise  level,  signal  level  and  other
                     charactistics of the input 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 quieter/shorter bursts of
                     audio to include prior to the detected trigger point.

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

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

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

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

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

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

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

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

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

              -M num Time constant used to smooth spectral measurements.

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

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

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

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

              See also the silence effect.

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

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

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

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

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

              Beware of Clipping when the increasing the volume.

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

              An optional limitergain value can be specified and should be a value much less than
              1  (e.g.  0.05  or  0.02)  and  is  used  only  on  peaks to prevent clipping.  Not
              specifying this parameter will cause no limiter to be used.  In verbose mode,  this
              effect will display the percentage of the audio that needed to be limited.

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


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


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


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

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

       [2]    Wikipedia, Q-factor,

       [3]    Scott   Lehman,   Effects   Explained,

       [4]    Wikipedia, Decibel,

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

       [6]    Richard Furse, Computer Music Toolkit,

       [7]    Steve Harris, LADSPA plugins,


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

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

       This program is distributed in the hope that it will be useful, but WITHOUT ANY  WARRANTY;
       without  even the implied warranty of MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.
       See the GNU General Public License for more details.


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