Provided by: sox_14.3.0-1build1_i386 bug

NAME

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

SYNOPSIS

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

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

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

DESCRIPTION

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

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

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

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

                                 *        *        *

       The overall SoX processing chain can be summarised as follows:

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

       Note however, that on the  SoX  command  line,  the  positions  of  the
       Output(s)  and  the  Effects  are  swapped w.r.t. the logical flow just
       shown.  Note also that whilst options pertaining to  files  are  placed
       before  their  respective  file name, the opposite is true for effects.
       To show how this works in practice, here is a selection of examples  of
       how SoX might be used.  The simple

          sox recital.au recital.wav

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

          sox recital.au -b 16 recital.wav channels 1 rate 16k fade 3 norm

       performs the same format translation, but  also  applies  four  effects
       (down-mix  to  one channel, sample rate change, fade-in, nomalize), and
       stores the result at a bit-depth of 16.

          sox -r 16k -e signed -b 8 -c 1 voice-memo.raw voice-memo.wav

       converts ‘raw’ (a.k.a. ‘headerless’) audio to  a  self-describing  file
       format,

          sox slow.aiff fixed.aiff speed 1.027

       adjusts audio speed,

          sox short.wav long.wav longer.wav

       concatenates two audio files, and

          sox -m music.mp3 voice.wav mixed.flac

       mixes together two audio files.

          play "The Moonbeams/Greatest/*.ogg" bass +3

       plays  a  collection  of  audio  files  whilst applying a bass boosting
       effect,

          play -n -c1 synth sin %-12 sin %-9 sin %-5 sin %-2 fade h 0.1 1 0.1

       plays a synthesised ‘A minor seventh’ chord with a pipe-organ sound,

          rec -c 2 radio.aiff trim 0 30:00

       records half an hour of stereo audio, and

          rec -M take1.aiff take1-dub.aiff

       records a new track in a multi-track recording.  Finally,

          rec -r 44100 -b 16 -s -p silence 1 0.50 0.1% 1 10:00 0.1% | \
            sox -p song.ogg silence 1 0.50 0.1% 1 2.0 0.1% : \
            newfile : restart

       records a stream of audio such as LP/cassette and splits in to multiple
       audio  files  at  points  with 2 seconds of silence.  Also, it does not
       start recording until it detects audio is playing and  stops  after  it
       sees 10 minutes of silence.

       N.B.   The  above  is  just an overview of SoX’s capabilities; detailed
       explanations of how to  use  all  SoX  parameters,  file  formats,  and
       effects  can  be  found  below  in this manual, in soxformat(7), and in
       soxi(1).

   File Format Types
       There are two types of audio file format that SoX can work with: ‘self-
       describing’  -  these  (e.g.  WAV,  FLAC) have a header that completely
       describes the signal and encoding attributes of  the  audio  data  that
       follows,  and ‘raw’ (or ‘headerless’) audio - the audio characteristics
       of these must, when reading a raw file,  be  described  using  the  SoX
       command  line,  and,  when writing a raw file, be set using the 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’).  For
              example, 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, i.e.
              the stored audio data takes up less space  (i.e.  disk-space  or
              transmission  band-width)  than  the other format parameters and
              the number of samples would imply.  Commonly-used encoding types
              include  floating-point,  μ-law, ADPCM, signed-integer PCM, MP3,
              and FLAC.

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

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

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

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

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

       1.  Command-line format options.

       2.  The filename extension.

       3.  The input file format characteristics, or the closest to them  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

       and

          rec new-file.wav

       These two commands are functionally equivalent to

          sox existing-file.wav -d

       and

          sox -d new-file.wav

       Of course, further options and effects  (as  described  below)  can  be
       added to the commands in either form.

                                 *        *        *

       Some  systems  provide  more  than  one  type of (SoX-compatible) audio
       driver, e.g. ALSA & OSS, or SUNAU & AO.  Systems  can  also  have  more
       than  one  audio  device (a.k.a. ‘sound card’).  If more than one audio
       driver has been built-in to SoX, and the default selected by  SoX  when
       recording  or  playing  is  not  the  one  that  is  wanted,  then  the
       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

       or

          set AUDIODEV=hw:soundwave,1,2
          play ...
          sox ... -t alsa

       Note that the way of setting environment variables varies  from  system
       to system - for some specific examples, see ‘SOX_OPTS’ below.

       When  playing  a  file  with a sample rate that is not supported by the
       audio output device, SoX will automatically invoke the rate  effect  to
       perform  the  necessary sample rate conversion.  For compatibility with
       old hardware, here, the default rate quality level  is  set  to  ‘low’;
       however,  this  can be changed if desired, by explicitly specifying the
       rate effect with a different quality level, e.g.

          play ... rate -m

       or by using the --play-rate-arg option (see below).

                                 *        *        *

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

       To help with setting a suitable recording level, SoX includes  a  peak-
       level  meter  which can be invoked (before making the actual recording)
       as follows:

          rec -n

       The recording level should be adjusted (using the system-provided mixer
       program, not SoX) so that the meter is at most occasionally full scale,
       and never ‘in the red’ (an exclamation mark is  shown).   See  also  -S
       below.

   Accuracy
       Many  file formats that compress audio discard some of the audio signal
       information  whilst  doing  so;  converting  to  such  a  format   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’, and
       formats that do not, ‘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; hence this is also recommended.

   Dithering
       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
       appropriate.

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

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

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

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

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

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

   Clipping
       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, but 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  that  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) will probably suffice.  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,  then  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’,  or  ‘merge’.   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, however, 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.

       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.   So  here,  if
       manual  volume  adjustments are not given, to ensure that clipping does
       not occur, SoX will automatically adjust 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 through the mixed audio
       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  appropriately
       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  no  clipping will occur,
       however, in many cases, the  number  of  clips  will  be  low  and  the
       resultant distortion imperceptible.

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

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

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

       In multiple output mode, a unique number will automatically be appended
       to the end of all filenames.  If the filename has an extension then the
       number is  inserted  before  the  extension.   This  behaviour  can  be
       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 in to two 30 second files and ignoring the rest.

          sox song.wav ringtone%1n.wav trim 0 30 : newfile : trim 0 30

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

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

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

FILENAMES

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

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

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

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

       "|program [options] ..."
              This  can  be  used in place of an input filename to specify the
              the given program’s standard output (stdout) be used as an input
              file.   Unlike - (above), this can be used for several inputs to
              one SoX command.  For example,  if  ‘genw’  generates  mono  WAV
              formatted  signals  to  its  standard output, then the following
              command makes a stereo file from two generated signals:

                 sox -M "|genw --imd -" "|genw --thd -" out.wav

              For  headerless  (raw)  audio,  -t  (and  perhaps  other  format
              options) will need to be given, preceding the input command.

       "wildcard-filename"
              Specifies  that  filename ‘globbing’ (wild-card matching) should
              be performed by SoX instead of 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; but 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 drivers.

OPTIONS

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

       The  SOX_OPTS  environment  variable can be used to provide alternative
       default values for SoX’s global options.  For example:

          SOX_OPTS="--buffer 20000 --play-rate-arg -hs --temp /mnt/temp"

       Note that setting SOX_OPTS can potentially create unwanted  changes  in
       the  behaviour  of  scripts  or  other  programs  that  invoke SoX.  So
       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 can not 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 of setting environment variables varies  from  system
       to system - here are some examples:

       Unix bash:

          export SOX_OPTS="-V --no-clobber"

       Unix csh:

          setenv SOX_OPTS "-V --no-clobber"

       MS-DOS/MS-Windows:

          set SOX_OPTS=-V --no-clobber

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

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

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

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

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

       -D, --no-dither
              Disable  automatic  dither  - see ‘Dither’ 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.   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).

       --interactive
              Deprecated alias for --no-clobber.

       -m|-M|--combine concatenate|merge|mix|mix-power|sequence
              Select the input file combining method;  -m  selects  ‘mix’,  -M
              selects ‘merge’.

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

       --magic
              If SoX has been built with the optional ‘libmagic’ library, then
              this  option can be given to enable its use in helping to detect
              audio file types.

       --no-clobber
              Prompt before overwriting an existing file with the same name as
              that given for the output file.

              N.B.   Unintentionally  overwriting  a  file  is easier than you
              might think, for example, if you accidentally enter

                 sox file1 file2 effect1 effect2 ...

              when what you really meant was

                 play file1 file2 effect1 effect2 ...

              then, without this option, file2 will  be  overwritten.   Hence,
              using  this  option  is recommended; SOX_OPTS (above), a ‘shell’
              alias, script, or batch  file  may  be  an  appropriate  way  of
              permanently enabling it.

       --norm 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

              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.

       --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.

       --single-threaded
              By default, SoX processes audio channels for most  multi-channel
              effects in parallel on hyper-threading/multi-core architectures.
              In case this should ever cause a  problem,  parallel  processing
              can be disabled by giving this option.

       --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 which case, using
              ‘--temp .’ (to use  the  current  directory)  is  often  a  good
              solution.

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

       -V[level]
              Set  verbosity  - this is particularly useful for seeing how any
              automatic effects have been invoked by SoX.

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

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

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

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

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

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

              By default, the verbosity level is set to 2  (shows  errors  and
              warnings);  each  occurrence  of  the  -V  option  increases the
              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.

       --ignore-length
              Override  an  (incorrect)  audio length given in an audio file’s
              header; if this option is given,  then  SoX  will  keep  reading
              audio until it reaches the end of the input file.

       -v, --volume FACTOR
              Intended  for  use  when  combining  multiple  input files, this
              option adjusts the volume of the file that  follows  it  on  the
              command  line  by  a  factor  of  FACTOR,  thus  allowing  it to
              ‘balanced’ w.r.t. the other  input  files.   This  is  a  linear
              (amplitude)  adjustment,  so  a number less than 1 decreases the
              volume; 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.

       -1/-2/-3/-4/-8
              The  number of bytes in each encoded sample.  Deprecated aliases
              for -b 8, -b 16, -b 24, -b 32, -b 64 respectively.

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

              For  an  input  file,  the most common use for this option is to
              inform SoX of the number of channels in a  ‘raw’  (‘headerless’)
              audio  file.   Occasionally, it may be useful to use this option
              with a ‘headered’ file, in order  to  override  the  (presumably
              incorrect)  value  in  the  header  -  note  that  this  is only
              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 -3
                 sox input.wav      output.wav bass -3 channels 1

              though the second form is more flexible as it allows the effects
              to be ordered arbitrarily.

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

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

              unsigned-integer
                     PCM data stored as signed (‘two’s complement’)  integers.
                     Commonly  used with an 8-bit encoding size.  A value of 0
                     represents maximum signal power.

              floating-point
                     PCM data stored as IEEE 753 single precision (32-bit)  or
                     double   precision   (64-bit)   floating-point   (‘real’)
                     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-adpcm
                     OKI  (a.k.a. VOX, Dialogic, or Intel) 4-bit ADPCM; it has
                     a precision equivalent to roughly 12-bit PCM.  ADPCM is a
                     form  of  audio  compression  that  has a good compromise
                     between audio quality and encoding/decoding speed.

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

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

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

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

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

              For  an output file, this option can be used (perhaps along with
              -b) to set the output encoding type  For example

                 sox input.cdda -e float output1.wav

                 sox input.cdda -b 64 -e float output2.wav

              convert  raw  CD  digital  audio  (16-bit,  signed-integer)   to
              floating-point   ‘WAV’   files   (single   &   double  precision
              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 type.

       -s/-u/-f/-A/-U/-o/-i/-a/-g
              Deprecated aliases for specifying  the  encoding  types  signed-
              integer,  unsigned-integer,  floating-point, mu-law, a-law, oki-
              adpcm, ima-adpcm, ms-adpcm, gsm-full-rate respectively  (see  -e
              above).

       --no-glob
              Specifies  that  filename ‘globbing’ (wild-card matching) should
              not be performed by SoX on the following filename.  For example,
              if   the   current  directory  contains  the  two  files  ‘five-
              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
              equivalent:

                 sox input.wav -r 48k output.wav bass -3
                 sox input.wav        output.wav bass -3 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-pont, 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 information.

EFFECTS

       In addition to converting, playing and recording audio files,  SoX  can
       be used to invoke a number of audio ‘effects’.  Multiple effects may be
       applied by specifying them one after another at  the  end  of  the  SoX
       command  line, forming an ‘effects chain’.  Note that applying multiple
       effects in real-time (i.e. when playing audio) is likely to need 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 Effect 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 chain 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 reach end of
       file.

       A  effects  chain is terminated by placing a : (colon) after an effect.
       Any following effects are apart 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 keep small, relative to the sample rate, if the
       terminating  effect  can not 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
       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:

       centre[k]
              See frequency.

       frequency[k]
              A frequency in Hz, or, if appended with ‘k’, kHz.

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

       width[h|k|o|q]
              Used to specify  the  band-width  of  a  filter.   A  number  of
              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 effect’s first line of description.

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

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

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

              This effect supports the --plot global option.

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

              This effect supports the --plot global option.

              See also sinc for a bandpass filter with steeper shoulders.

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

              These effects support the --plot global option.

              See also sinc for a bandpass filter with steeper shoulders.

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

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

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

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

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

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

              The filters are described in detail in [1].

              These effects support the --plot global option.

              See also equalizer for a peaking equalisation effect.

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

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

              For  example,  an  initial  tone  is  generated, then bent three
              times, yielding four different notes in total:

                 play -n synth 2.5 sin 667 gain 1 \
                   bend .35,180,.25  .15,740,.53  0,-520,.3

              Note that the clipping that  is  produced  in  this  example  is
              deliberate; to remove it, use gain -5 in place of gain 1.

       biquad b0 b1 b2 a0 a1 a2
              Apply a biquad IIR filter with the given coefficients.

              See http://en.wikipedia.org/wiki/Digital_biquad_filter (where a0
              = 1).

       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 -3
                 sox input.wav      output.wav bass -3 channels 1

              though the second form is more flexible as it allows the effects
              to be ordered arbitrarily.

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

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

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

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

              A typical delay is around 40ms to 60ms; the modulation speed  is
              best  near  0.25Hz  and  the  modulation  depth around 2ms.  For
              example, a single delay:

                 play guitar1.wav chorus 0.7 0.9 55 0.4 0.25 2 -t

              Two delays of the original samples:

                 play guitar1.wav chorus 0.6 0.9 50 0.4 0.25 2 -t \
                    60 0.32 0.4 1.3 -s

              A fuller sounding chorus (with three additional delays):

                 play guitar1.wav chorus 0.5 0.9 50 0.4 0.25 2 -t \
                    60 0.32 0.4 2.3 -t 40 0.3 0.3 1.3 -s

       compand attack1,decay1{,attack2,decay2}
              [soft-knee-dB:]in-dB1[,out-dB1]{,in-dB2,out-dB2}
              [gain [initial-volume-dB [delay]]]

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

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

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

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

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

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

                                    *        *        *

              The following example might be used to make  a  piece  of  music
              with both quiet and loud passages suitable for listening to in a
              noisy environment such as a moving vehicle:

                 sox asz.wav asz-car.wav compand 0.3,1 6:-70,-60,-20 -5 -90 0.2

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

              In  the  next example, compand is being used as a noise-gate for
              when the noise is at a lower level than the signal:

                 play infile compand .1,.2 -inf,-50.1,-inf,-50,-50 0 -90 .1

              Here is another noise-gate, this time for when the noise is at a
              higher  level  than the signal (making it, in some ways, similar
              to squelch):

                 play infile compand .1,.1 -45.1,-45,-inf,0,-inf 45 -90 .1

              This effect supports the --plot global option (for the  transfer
              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 ISO 908 de-emphasis (a treble attenuation shelving filter)
              to 44.1kHz (Compact Disc) audio.

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

              With the deemph effect, it is possible to  apply  the  necessary
              de-emphasis  to  audio  that  has  been  extracted  from  a pre-
              emphasised CD, and then either burn the de-emphasised audio to a
              new  CD  (which  will  then play correctly on any CD player), or
              simply play the correctly de-emphasised audio files on  the  PC.
              For example:

                 sox track1.wav track1-deemph.wav deemph

              and then burn track1-deemph.wav to CD, or

                 play track1-deemph.wav

              or simply

                 play track1.wav deemph

              The  de-emphasis  filter is implemented as a biquad; its maximum
              deviation from the ideal response is only 0.06dB (up to  20kHz).

              This effect supports the --plot global option.

              See also the bass and treble shelving equalisation effects.

       delay {length}
              Delay one or more audio channels.  length can specify a time or,
              if appended with an ‘s’, a number of samples.   Do  not  specify
              both  time and samples delays in the same command.  For example,
              delay 1.5 0 0.5 delays the first channel  by  1.5  seconds,  the
              third channel by 0.5 seconds, and leaves the second channel (and
              any  other  channels  that  may  be  present)  un-delayed.   The
              following (one long) command plays a chime sound:

                 play -n synth -j 3 sin %3 sin %-2 sin %-5 sin %-9 \
                   sin %-14 sin %-21 fade h .01 2 1.5 delay \
                   1.3 1 .76 .54 .27 remix - fade h 0 2.7 2.5 norm -1

              and this plays a guitar chord:

                 play -n synth pl G2 pl B2 pl D3 pl G3 pl D4 pl G4 \
                   delay 0 .05 .1 .15 .2 .25 remix - fade 0 4 .1 norm -1

       dither [-a] [-S|-s|-f filter]
              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 http://sox.sourceforge.net/SoX/NoiseShaping  for  graphs  of
              the different noise-shaping curves.

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

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

              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 ‘Dither’ section above.

       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).      See
              http://www.geocities.com/beinges for a full explanation.

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

              Each  given delay decay pair gives the delay in milliseconds and
              the decay (relative to gain-in) of that echo.  Gain-out  is  the
              volume  of  the output.  For example: This will make it sound as
              if there are twice as many instruments as are actually playing:

                 play lead.aiff echo 0.8 0.88 60 0.4

              If the delay is very short, then  it  sound  like  a  (metallic)
              robot playing music:

                 play lead.aiff echo 0.8 0.88 6 0.4

              A  longer  delay  will  sound  like  an  open air concert in the
              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 unchanged.

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

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

              The filter is described in detail in [1].

              This effect supports the --plot global option.

              See also bass and treble for shelving equalisation effects.

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

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

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

              For  fade-outs, the audio will be truncated at stop-time and the
              signal level will be ramped from full volume down to 0  starting
              at  fade-out-length  seconds before the stop-time.  If fade-out-
              length is not specified, it defaults to the same value as  fade-
              in-length.   No  fade-out  is  performed  if  stop-time  is  not
              specified.  If the file length can be determined from the  input
              file  header and length-changing effects are not in effect, then
              0 may be specified for stop-time to indicate the usual case of a
              fade-out that ends at the end of the input audio stream.

              All  times  can be specified in either periods of time or sample
              counts.  To specify time periods use  the  format  hh:mm:ss.frac
              format.   To  specify using sample counts, specify the number of
              samples and append the letter  ‘s’  to  the  sample  count  (for
              example ‘8000s’).

              See also the splice effect.

       fir [coefs-file|coefs]
              Use   SoX’s   FFT  convolution  engine  with  given  FIR  filter
              coefficients.  If a  single  argument  is  given  then  this  is
              treated as the name of a file containing the filter coefficients
              (white-space separated; may contain ‘#’ comments).  If the given
              filename   is  ‘-’,  or  if  no  argument  is  given,  then  the
              coefficients  are  read  from  the  ‘standard  input’   (stdin);
              otherwise,  coefficients  may  be  given  on  the  command line.
              Examples:

                 sox infile outfile fir 0.0195 -0.082 0.234 0.891 -0.145 0.043

                 sox infile outfile fir coefs.txt

              with coefs.txt containing

                 # HP filter
                 # freq=10000
                   1.2311233052619888e-01
                  -4.4777096106211783e-01
                   5.1031563346705155e-01
                  -6.6502926320995331e-02
                 ...

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

              All parameters are optional (right to left).

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

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

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

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

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

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

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

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

                 sox infile outfile gain -n

              normalises to 0dB, and

                 sox infile outfile gain -n -3

              normalises to -3dB.

              The -l option invokes a simple limiter, e.g.

                 sox infile outfile gain -l 6

              will  apply 6dB of gain but never clip.  Note that limiting more
              than a few dBs more than occasionally (in a piece of  audio)  is
              not  recommended  as  it  can cause audible distortion.  See the
              compand effect for a more capable limiter.

              The -h option is used to apply gain  to  provide  head-room  for
              subsequent processing.  For example, with

                 sox infile outfile gain -h bass +6

              6dB  of  attenuation  will be applied prior to the bass boosting
              effect thus ensuring that it will not  clip.   Of  course,  with
              bass,  it  is obvious how much headroom will be needed, but with
              other effects (e.g.  rate, dither) it is not  always  as  clear.
              Another  advantage  of  using  gain  -h  rather than an explicit
              attenuation, is that if the headroom is not used  by  subsequent
              effects, it can be reclaimed with gain -r, for example:

                 sox infile outfile gain -h bass +6 rate 44100 gain -r

              The above effects chain guarantees never to clip nor amplify; it
              attenuates if necessary to prevent clipping, but by only as much
              as is needed to do so.

              Output  formatting  (dithering  and  bit-depth  reduction)  also
              requires headroom (which cannot be ‘reclaimed’), e.g.

                 sox infile outfile gain -h bass +6 rate 44100 gain -rh dither

              Here, the second  gain  invocation,  reclaims  as  much  of  the
              headroom  as  it  can from the preceding effects, but retains as
              much headroom as is needed for subsequent processing.   The  SoX
              global  option  -G  can be given to automatically invoke gain -h
              and gain -r.

              See also the norm and vol effects.

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

              These effects support the --plot global option.

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

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

       loudness [gain [reference]]
              Loudness control - similar to  the  gain  effect,  but  provides
              equalisation    for    the    human    auditory   system.    See
              http://en.wikipedia.org/wiki/Loudness for a detailed description
              of  loudness.   The gain is adjusted by the given gain parameter
              (usually negative) and the signal equalised according to ISO 226
              w.r.t.   a  reference  level  of  65dB,  though  an  alternative
              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}
              [soft-knee-dB:]in-dB1[,out-dB1]{,in-dB2,out-dB2}
              [gain     [initial-volume-dB    [delay]]]"    {crossover-freq[k]
              "attack1,..."}

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

              For  example,  the following (one long) command shows how multi-
              band companding is typically used in FM radio:

                 play track1.wav gain -3 sinc 8000- 29 100 mcompand \
                   "0.005,0.1 -47,-40,-34,-34,-17,-33" 100 \
                   "0.003,0.05 -47,-40,-34,-34,-17,-33" 400 \
                   "0.000625,0.0125 -47,-40,-34,-34,-15,-33" 1600 \
                   "0.0001,0.025 -47,-40,-34,-34,-31,-31,-0,-30" 6400 \
                   "0,0.025 -38,-31,-28,-28,-0,-25" \
                   gain 15 highpass 22 highpass 22 sinc -n 255 -b 16 -17500 \
                   gain 9 lowpass -1 17801

              The audio file is played with a simulated  FM  radio  sound  (or
              broadcast  signal  condition if the lowpass filter at the end is
              skipped).  Note that the pipeline is set up with  US-style  75us
              pre-emphasis.

              See also compand for a single-band companding effect.

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

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

              The  mixer  effect  can  also  be invoked with up to 16 numbers,
              separated by commas, which specify the proportion (0 = 0% and  1
              =  100%)  of  each  input  channel that is to be mixed into each
              output channel.  In two-channel mode, 4 numbers are given:  l  →
              l, l → r, r → l, and r → r, respectively.  In four-channel mode,
              the first 4 numbers give  the  proportions  for  the  left-front
              output  channel, as follows: lf → lf, rf → lf, lb → lf, and rb →
              rf.  The next 4 give the right-front output in the  same  order,
              then left-back and right-back.

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

              Finally, certain reduced combination of numbers can be specified
              for certain input/output channel combinations.

                   In Ch   Out Ch   Num   Mappings
                     2       1       2    l → l, r → l
                     2       2       1    adjust balance
                     4       1       4    lf → l, rf → l, lb → l, rb → l
                     4       2       2    lf → l&rf → r, lb → l&rb → r
                     4       4       1    adjust balance
                     4       4       2    front balance, back balance

              See  also  remix  for a mixing effect that handles any number of
              channels.

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

       noisered [profile-file [amount]]
              Reduce  noise  in  the  audio signal by profiling and filtering.
              This effect  is  moderately  effective  at  removing  consistent
              background  noise such as hiss or hum.  To use it, first run SoX
              with the noiseprof effect on a section  of  audio  that  ideally
              would contain silence but in fact contains noise - such sections
              are typically found at the beginning or the end of a  recording.
              noiseprof  will write out a noise profile to profile-file, or to
              stdout if no profile-file or if ‘-’ is given.  E.g.

                 sox speech.wav -n trim 0 1.5 noiseprof speech.noise-profile

              To actually remove the noise, run SoX again, this time with  the
              noisered effect; noisered will reduce noise according to a noise
              profile (which was generated by noiseprof),  from  profile-file,
              or from stdin if no profile-file or if ‘-’ is given.  E.g.

                 sox speech.wav cleaned.wav noisered speech.noise-profile 0.3

              How much noise should be removed is specified by amount-a number
              between 0 and 1 with a default  of  0.5.   Higher  numbers  will
              remove  more  noise but present a greater likelihood of removing
              wanted components of the  audio  signal.   Before  replacing  an
              original recording with a noise-reduced version, experiment with
              different amount values to find the optimal one for your  audio;
              use  headphones  to  check  that you are happy with the results,
              paying particular attention to quieter sections of the audio.

              On most systems, the two stages - profiling and reduction -  can
              be combined using a pipe, e.g.

                 sox noisy.wav -n trim 0 1 noiseprof | play noisy.wav noisered

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

              Note that norm’s -i and -b options are deprecated  (having  been
              superseded  by  gain  -en  and gain -B respectively) and will be
              removed in a future release.

       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.

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

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

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

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

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

              For example:

                 play snare.flac phaser 0.8 0.74 3 0.4 0.5 -t

              Gentler:

                 play snare.flac phaser 0.9 0.85 4 0.23 1.3 -s

              A popular sound:

                 play snare.flac phaser 0.89 0.85 1 0.24 2 -t

              More severe:

                 play snare.flac phaser 0.6 0.66 3 0.6 2 -t

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

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

              See also the speed and tempo effects.

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

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

              where Band-width is the percentage of the audio  frequency  band
              that  is  preserved  and Rej dB is the level of noise rejection.
              Increasing levels of resampling quality come at the  expense  of
              increasing  amounts of time to process the audio.  If no quality
              option is given, the quality level used is ‘high’.

              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 -3
                 sox input.wav        output.wav bass -3 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 can not be used with the ‘quick’ or ‘low’
              quality algorithms.

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

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

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

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

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

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

              Examples:

                 sox input.wav -b 16 output.wav rate -s -a 44100 dither -s

              default  (high)  quality  resampling;  overrides:  steep filter,
              allow aliasing; to 44.1kHz sample rate; noise-shaped  dither  to
              16-bit WAV file.

                 sox input.wav -b 24 output.aiff rate -v -I -b 90 48k

              very  high  quality  resampling;  overrides: intermediate phase,
              band-width 90%; to 48k sample rate; store output to 24-bit  AIFF
              file.

                                    *        *        *

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

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

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

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

              An out-spec contains comma-separated input  channel-numbers  and
              hyphen-delimited  channel-number ranges; alternatively, 0 may be
              given to create a silent output channel.  For example,

                 sox input.wav output.wav remix 6 7 8 0

              creates an output file with four channels, where channels 1,  2,
              and  3 are copies of channels 6, 7, and 8 in the input file, and
              channel 4 is silent.  Whereas

                 sox input.wav output.wav remix 1-3,7 3

              creates a (somewhat bizarre) stereo output file where  the  left
              channel  is a mix-down of input channels 1, 2, 3, and 7, and the
              right channel is a copy of input channel 3.

              Where a range of channels is specified, the channel  numbers  to
              the  left  and right of the hyphen are optional and default to 1
              and to the number of input channels respectively. Thus

                 sox input.wav output.wav remix -

              performs a mix-down of all input channels to mono.

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

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

              If an out-spec includes at least one vol-spec then, by  default,
              ¹/n  scaling  is  not  applied to any other channels in the same
              out-spec (though may be in other out-specs).  The -a (automatic)
              option  however, can be given to retain the automatic scaling in
              this case.  For example,

                 sox input.wav output.wav remix 1,2 3,4v0.8

              results in channel level multipliers of 0.5,0.5 1,0.8, whereas

                 sox input.wav output.wav remix -a 1,2 3,4v0.8

              results in channel level multipliers of 0.5,0.5 0.5,0.8.

              The  -m  (manual)   option   disables   all   automatic   volume
              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:

              #!/bin/sh
              chans=`soxi -c "$1"`
              while [ $chans -ge 1 ]; do
                 chans0=`printf %02i $chans`   # 2 digits hence up to 99 chans
                 out=`echo "$1"|sed "s/\(.*\)\.\(.*\)/\1-$chans0.\2/"`
                 sox "$1" "$out" remix $chans
                 chans=`expr $chans - 1`
              done

              If a file input.wav containing six audio  channels  were  given,
              the   script  would  produce  six  output  files:  input-01.wav,
              input-02.wav, ..., input-06.wav.

              See also mixer and swap for similar effects.

       repeat count
              Repeat the entire audio count times.   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
              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 anything below a specified threshold.

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

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

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

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

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

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

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

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

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

              The following example shows how this effect can be used to start
              a recording that does not contain the delay at the  start  which
              usually  occurs  between  ‘pressing  the  record button’ and the
              start of the performance:

                 rec parameters filename other-effects silence 1 5 2%

       sinc [-a att|-b beta] [-p phase|-M|-I|-L] [-t tbw|-n taps]
       [freqHP][-freqLP [-t tbw|-n taps]]
              Apply a sinc kaiser-windowed low-pass, high-pass, band-pass,  or
              band-reject  filter  to  the  signal.   The  freqHP  and  freqLP
              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  <  freqLP  creates  a
              band-pass  filter, freqHP > freqLP creates a band-reject filter.

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

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

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

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

              This effect supports the --plot global option.

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

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

              For example, if ‘my.wav’ is a stereo file, then with

                 sox my.wav -n spectrogram

              a  spectrogram  of  the  entire file will be created in the file
              ‘spectrogram.png’.  More often though,  analysis  of  a  smaller
              portion of the audio is required; e.g. with

                 sox my.wav -n remix 2 trim 20 30 spectrogram

              the  spectrogram  shows information only from the second (right)
              channel, and of thirty seconds of  audio  starting  from  twenty
              seconds in.  To analyse a small portion of the frequency domain,
              the rate effect may be used, e.g.

                 sox my.wav -n rate 6k spectrogram

              allows detailed analysis of frequencies up  to  3kHz  (half  the
              sampling  rate)  i.e.  where  the  human auditory system is most
              sensitive.  With

                 sox my.wav -n trim 0 10 spectrogram -x 600 -y 200 -z 100

              the given options control the size of the spectrogram’s X, Y & Z
              axes  (in  this case, the spectrogram area of the produced image
              will be 600 by 200 pixels in size and the Z-axis range  will  be
              100  dB).   Note  that  the produced image includes axes legends
              etc.  and  so  will  be  a  little  larger  than  the  specified
              spectrogram size.  In this example:

                 sox -n -n synth 6 tri 10k:14k spectrogram -z 100 -w kaiser

              an analysis ‘window’ with high dynamic range is selected to best
              display the spectrogram of  a  swept  triangular  wave.   For  a
              smilar  example,  append the following to the ‘chime’ command in
              the description of the delay effect (above):

                 rate 2k spectrogram -X 200 -Z -10 -w kaiser

              Options are also avaliable to control  the  appearance  (colour-
              set,   brightness,   contrast,   etc.)   and   filename  of  the
              spectrogram; e.g. with

                 sox my.wav -n spectrogram -m -l -o print.png

              a spectrogram is created suitable for printing on a  ‘black  and
              white’ printer.

              Options:

              -x num Change  the  (maximum)  width (X-axis) of the spectrogram
                     from its default value of 800 pixels to  a  given  number
                     between 100 and 5000.  See also -X and -d.

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

              -o text
                     Name   of   the  spectrogram  output  PNG  file,  default
                     ‘spectrogram.png’.

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

              -d duration
                     This option sets the X-axis resolution  such  that  audio
                     with  the given duration ([[HH:]MM:]SS) fits the selected
                     (or default) X-axis width.  For example,

                        sox input.mp3 output.wav -n spectrogram -d 1:00 stats

                     creates a spectrogram showing the  first  minute  of  the
                     audio, whilst

                     the stats effect is applied to the entire audio signal.

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

              -S time
                     Start the spectrogram at the given  point  in  the  audio
                     stream.  For example

                        sox input.aiff output.wav spectrogram -S 1:00

                     creates a spectrogram showing all but the first minute of
                     the audio (the output file however, receives  the  entire
                     audio stream).

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

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

              By default, the speed change is performed by resampling with the
              rate effect 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 pitch and tempo effects.

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

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

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

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

              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:

              #!/bin/sh
              # Audio Copy and Paste Over
              # acpo infile copy-start copy-stop paste-over-start outfile
              # All times measured in samples.
              rate=`soxi -r "$1"`
              e=`expr $rate ’*’ 5 / 1000`  # Using default excess
              l=$e                         # and leeway.
              sox "$1" piece.wav trim `expr $2 - $e - $l`s \
                 `expr $3 - $2 + $e + $l + $e`s
              sox "$1" part1.wav trim 0 `expr $4 + $e`s
              sox "$1" part2.wav trim `expr $4 + $3 - $2 - $e - $l`s
              sox part1.wav piece.wav part2.wav "$5" splice \
                 `expr $4 + $e`s \
                 `expr $4 + $e + $3 - $2 + $e + $l + $e`s

              In  the above Bourne shell script, two splices are used to ‘copy
              and paste’ audio.

                                    *        *        *

              It is also possible to use this effect to perform general cross-
              fades,  e.g.  to  join  two  songs.   In this case, excess would
              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
                                                         number.
               Minimum amplitude   min(xk)               The minimum  sample
                                                         value in the audio;
                                                         usually  this  will
                                                         be    a    negative
                                                         number.
               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’ format.

              The -v option displays only the ‘Volume Adjustment’ value.

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

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

              See also the stats effect.

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

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

                                       Overall     Left      Right
                          DC offset   0.000803 -0.000391  0.000803
                          Min level  -0.750977 -0.750977 -0.653412
                          Max level   0.708801  0.708801  0.653534
                          Pk lev dB      -2.49     -2.49     -3.69
                          RMS lev dB    -19.41    -19.13    -19.71

                          RMS Pk dB     -13.82    -13.82    -14.38
                          RMS Tr dB     -85.25    -85.25    -82.66
                          Crest factor       -      6.79      6.32
                          Flat factor     0.00      0.00      0.00
                          Pk count           2         2         2
                          Bit-depth      16/16     16/16     16/16
                          Num samples    7.72M
                          Length s     174.973
                          Scale max   1.000000
                          Window s       0.050

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

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

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

              See also the stat effect.

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

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

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

              See also the tempo effect.

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

              Though this effect is used to generate audio, an input file must
              still be given, the characteristics of which will be used to set
              the synthesised audio length, the number of  channels,  and  the
              sampling  rate;  however,  since  the  input file’s audio is not
              normally needed, a ‘null file’ (with the  special  name  -n)  is
              often  given instead (and the length specified as a parameter to
              synth or by another given effect  that  can  has  an  associated
              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 expressed as a time or
              as a number of samples; 0=inputlength, default=0.

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

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

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

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

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

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

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

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

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

              Not used for noise.

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

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

              p1 is the percentage of each cycle that  is  ‘on’  (square),  or
              ‘rising’   (triangle,   exp,   trapezium);  default=50  (square,
              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); default=20.

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

       tempo [-q] factor [segment [search [overlap]]]
              Change the audio tempo  (but  not  its  pitch).   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, giving a quicker, but possibly  lower
              quality, result.

              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.  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), 50 ms may give a better result.
              When changing the tempo of speech,  a  segment  size  of  around
              30 ms often works well.

              The   optional  search  parameter  gives  the  audio  length  in
              milliseconds (default 14) over which the algorithm  will  search
              for  overlapping points.  Larger values use more processing time
              and do not necessarily produce better results.

              The optional overlap parameter gives the segment overlap  length
              in milliseconds (default 12).

              See  also  speed  for  an  effect  that  changes tempo and pitch
              together, pitch for an  effect  that  changes  tempo  and  pitch
              together,  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 details.

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

       trim start [length]
              Trim  can  trim off unwanted audio from the beginning and end of
              the audio.  Audio is not sent to the  output  stream  until  the
              start location is reached.

              The  optional  length  parameter  gives  the  length of audio to
              output after the start sample and is thus used to trim  off  the
              end  of  the  audio.  Using a value of 0 for the start parameter
              will allow trimming off the end only.

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

       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.

              Options:
              Default values are shown in parenthesis.

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

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

              -s num (1)
                     The  amount  of  audio  (in  seconds)   to   search   for
                     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  preseve  before  the
                     trigger point and any found quieter/shorter bursts.

              Advanced Options:
              These allow fine tuning of the alogithm’s internal parameters.

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

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

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

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

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

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

              -M num Time constant used to smooth spectral measurements.

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

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

              -H num ‘Brick-wall’ quefrency of high-pass lifter  used  in  the
                     detector algorithm.

              -L num ‘Brick-wall’  quefrency  of  low-pass  lifter used in the
                     detector algorithm.

              See also the silence effect.

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

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

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

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

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

              Beware of Clipping when the increasing the volume.

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

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

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

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

       filter [low]-[high] [window-len [beta]]
              Apply  a  sinc-windowed lowpass, highpass, or bandpass filter of
              given window  length  to  the  signal.   This  effect  has  been
              superseded  by  the sinc effect.  Compared with ‘sinc’, ‘filter’
              is slower and has fewer capabilities.

              low refers to the frequency of  the  lower  6dB  corner  of  the
              filter.  high refers to the frequency of the upper 6dB corner of
              the filter.

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

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

              The beta parameter determines the type of  filter  window  used.
              Any  value greater than 2 is the beta for a Kaiser window.  Beta
              ≤ 2 selects a  Blackman-Nuttall  window.   If  unspecified,  the
              default is a Kaiser window with beta 16.

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

              This effect supports the --plot global option.

       key [-q] shift [segment [search [overlap]]]
              Change the audio key (i.e. pitch but not tempo).  This  is  just
              an alias for the pitch effect.

       pan direction
              Mix  the  audio from one channel to another.  Use mixer or remix
              instead of this effect.

              The direction is a value from -1 to 1.  -1 represents  far  left
              and 1 represents far right.

       polyphase [-w nut|ham] [-width n] [-cut-off c]
       rabbit [-c0|-c1|-c2|-c3|-c4]
       resample [-qs|-q|-ql] [rolloff [beta]]
              Formerly  sample-rate-changing effects in their own right, these
              are now just aliases for the rate effect.

DIAGNOSTICS

       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.

BUGS

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

SEE ALSO

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

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

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

       [3]    Scott     Lehman,     Effects     Explained,     http://harmony-
              central.com/Effects/effects-explained.html

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

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

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

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

LICENSE

       Copyright 1998-2009 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.

AUTHORS

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