Provided by: sox_14.4.1-3ubuntu1_amd64 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 into multiple output files.

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

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

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

                                          *        *        *

       The overall SoX processing chain can be summarised as follows:

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

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

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

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

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

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

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

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

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

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

       Most self-describing formats also allow textual `comments' to be 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 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, the default rate quality level is set to
       `low'. This can be changed by explicitly specifying  the  rate  effect  with  a  different
       quality level, e.g.
          play ... rate -m
       or by using the --play-rate-arg option (see below).

                                          *        *        *

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

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

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

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

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

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

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

   Dithering
       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.  Clipping  could  also  occur  with many other effects, when
       converting one format to another, and even when simply playing the audio.

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

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

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

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

       See also -G and the gain and norm effects.

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

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

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

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

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

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

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

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

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

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

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

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

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

       With the `mix-power' combine method, the mixed volume is approximately equal  to  that  of
       one of the input signals.  This is achieved by balancing using a factor of ¹/√n instead of
       ¹/n.  Note that this balancing factor does not guarantee that clipping will not occur, but
       the  number  of  clips  will  usually  be  low  and  the resultant distortion is generally
       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 into two 30
       second files and ignoring the rest.
          sox song.wav ringtone%1n.wav trim 0 30 : newfile : trim 0 30

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

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

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

FILENAMES

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

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

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

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

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

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

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

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

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

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

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

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

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

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

OPTIONS

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

                                      dB FSD   Display   dB FSD   Display
                                       -25     -          -11     ====
                                       -23     =           -9     ====-
                                       -21     =-          -7     =====
                                       -19     ==          -5     =====-
                                       -17     ==-         -3     ======

                                       -15     ===         -1     =====!
                                       -13     ===-

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

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

       -T     Equivalent to --combine multiply.

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

       --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. This
              allows it to be `balanced'  w.r.t.  the  other  input  files.   This  is  a  linear
              (amplitude)  adjustment,  so a number less than 1 decreases the volume and a number
              greater than 1 increases it.  If a negative number is given then in addition to the
              volume adjustment, the audio signal will be inverted.

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

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

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

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

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

       -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 -b 24
                 sox input.wav      output.wav bass -b 24 channels 1
              though  the  second  form  is  more flexible as it allows the effects to be ordered
              arbitrarily.

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

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

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

              floating-point
                     PCM  data  stored  as IEEE 753 single precision (32-bit) or double precision
                     (64-bit) floating-point (`real') 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, a-law, mu-law, oki-adpcm, ima-adpcm,  ms-adpcm,  gsm-full-
              rate respectively (see -e above).

       --no-glob
              Specifies  that filename `globbing' (wild-card matching) should not be performed by
              SoX on the following filename.  For example, if the current directory contains  the
              two files `five-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 -b 24
                 sox input.wav        output.wav bass -b 24 rate 48k
              though  the second form is more flexible as it allows rate options to be given, and
              allows the effects to be ordered arbitrarily.

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

       -C, --compression FACTOR
              The  compression  factor  for  variably  compressing  output file formats.  If this
              option is not given then a default compression factor will apply.  The  compression
              factor  is interpreted differently for different compressing file formats.  See the
              description of the file formats that use  this  option  in  soxformat(7)  for  more
              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  require  a
       high  performance  computer.  Stopping other applications may alleviate performance issues
       should they occur.

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

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

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

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

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

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

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

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

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

       center[k]
              See frequency.

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

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

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

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

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

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

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

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

              This effect supports the --plot global option.

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

              This effect supports the --plot global option.

              See also sinc for a bandpass filter with steeper shoulders.

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

              These effects support the --plot global option.

              See also sinc for a bandpass filter with steeper shoulders.

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

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

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

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

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

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

              The filters are described in detail in [1].

              These effects support the --plot global option.

              See also equalizer for a peaking equalisation effect.

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

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

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

              See also pitch.

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

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

              This effect supports the --plot global option.

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

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

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

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

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

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

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

       compand attack1,decay1{,attack2,decay2}
              [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 Compact Disc (IEC 60908) de-emphasis (a treble attenuation shelving filter).

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

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

              This effect supports the --plot global option.

              See also the bass and treble shelving equalisation effects.

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

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

              The -p option allows overriding the target precision.

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

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

              See also the `Dithering' section above.

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

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

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

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

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

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

              This effect supports the --plot global option.

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

              All parameters are optional (right to left).

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

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

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

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

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

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

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

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

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

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

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

              See also the norm and vol effects.

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

              These effects support the --plot global option.

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

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

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

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

              This effect supports the --plot global option.

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

       loudness [gain [reference]]
              Loudness control - similar to the gain effect, but provides  equalisation  for  the
              human  auditory  system.   See 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.

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

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

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

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

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

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

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

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

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

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

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

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

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

              See also the bend, speed, and tempo effects.

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

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

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

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

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

                                              *        *        *

              Warning: technically detailed discussion follows.

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

              The -m (manual) option disables all automatic volume 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 the swap effect.

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

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

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

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

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

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

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

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

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

              To  remove  silence  from  the  middle  of  a file, specify a below-periods that is
              negative.  This value is then treated as a positive  value  and  is  also  used  to
              indicate  the  effect  should restart processing as specified by the above-periods,
              making it suitable for removing periods of silence in the middle of the 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 less than freqLP creates a band-pass
              filter, freqHP greater than freqLP creates a band-reject filter.  For example,  the
              invocations
                 sinc 3k
                 sinc -4k
                 sinc 3k-4k
                 sinc 4k-3k
              create a high-pass, low-pass, band-pass, and band-reject filter respectively.

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

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

              If both freqHP and freqLP are given, then a -t or -n option given to  the  left  of
              the  frequencies  applies  to  both  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 200000.  See also -X
                     and -d.

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

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

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

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

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

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

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

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

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

              See also the bend, pitch, and tempo effects.

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

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

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

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

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

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

              where * indicates the joining points.

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

              Provided  your  arithmetic is good enough, multiple splices can be performed with a
              single splice invocation.  For example:
              #!/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] [-m|-s|-l] factor [segment [search [overlap]]]
              Change the audio playback speed but not its  pitch.  This  effect  uses  the  WSOLA
              algorithm. The audio is chopped up into segments which are then shifted in the time
              domain and overlapped (cross-faded)  at  points  where  their  waveforms  are  most
              similar as determined by measurement of `least squares'.

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

              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 preserve before the  trigger  point  and
                     any found quieter/shorter bursts.

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

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

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

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

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

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

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

              -M num Time constant used to smooth spectral measurements.

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

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

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

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

              See also the silence effect.

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

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

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

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

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

              Beware of Clipping when the increasing the volume.

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

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

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

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

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

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

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

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

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

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

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

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 Developer's Simple Plugin API, http://www.ladspa.org

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

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

LICENSE

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

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

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