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


           SoX  reads  and  writes  audio  files  in  most popular formats and can
           optionally apply  effects  to  them.  It  can  combine  multiple  input
           sources,  synthesise audio, and, on many systems, act as a general pur-
           pose audio player or a multi-track audio recorder. It also has  limited
           ability to split the input into multiple output files.
           All SoX functionality is available using just the sox command.  To sim-
           plify playing and recording audio, if SoX is invoked as play, the  out-
           put  file  is  automatically set to be the default sound device, and if
           invoked as rec, the default sound device is used as  an  input  source.
           Additionally,  the  soxi(1)  command  provides a convenient way to just
           query audio file header information.
           The heart of SoX is a  library  called  libSoX.   Those  interested  in
           extending  SoX or using it in other programs should refer to the libSoX
           manual page: libsox(3).
           SoX is a command-line audio processing  tool,  particularly  suited  to
           making  quick,  simple  edits  and to batch processing.  If you need an
           interactive, graphical audio editor, use audacity(1).
                                     *        *        *
           The overall SoX processing chain can be summarised as follows:
                          Input(s) -> Combiner -> Effects -> Output(s)
           Note however, that on the SoX command line, the positions of  the  Out-
           put(s)  and the Effects are swapped w.r.t. the logical flow just shown.
           Note also that whilst options pertaining to  files  are  placed  before
           their  respective file name, the opposite is true for effects.  To show
           how this works in practice, here is a selection of examples of how  SoX
           might be used.  The simple
              sox recital.wav
           translates  an  audio  file  in  Sun AU format to a Microsoft WAV file,
           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
              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
       File Format Types
           SoX  can  work  with  'self-describing'  and 'raw' audio files.  'self-
           describing' formats (e.g. WAV, FLAC, MP3) have a header that completely
           describes  the  signal  and  encoding attributes of the audio data that
           follows. 'raw' or 'headerless' formats do not contain this information,
           so the audio characteristics of these must be described on the SoX com-
           mand line or inferred from those of the input file.
           The following four characteristics are used to describe the  format  of
           audio data such that it can be processed with SoX:
           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,  ?-law, ADPCM, signed-integer PCM, MP3,
                  and FLAC.
                  The number  of  audio  channels  contained  in  the  file.   One
                  ('mono')  and  two ('stereo') are widely used.  'Surround sound'
                  audio typically contains six or more channels.
           The term 'bit-rate' is a measure of the amount of storage  occupied  by
           an  encoded  audio signal over a unit of time.  It can depend on all of
           the above and is typically denoted as a number of kilo-bits per  second
           (kbps).    An  A-law  telephony  signal  has  a  bit-rate  of  64  kbs.
           MP3-encoded stereo music typically has  a  bit-rate  of  128-196  kbps.
           FLAC-encoded stereo music typically has a bit-rate of 550-760 kbps.
           Most self-describing formats also allow textual 'comments' to be embed-
           ded in the file that can be used to describe the  audio  in  some  way,
           e.g. for music, the title, the author, etc.
           One  important  use  of  audio file comments is to convey 'Replay Gain'
           information.  SoX supports applying Replay Gain  information,  but  not
           generating it.  Note that by default, SoX copies input file comments to
           output files that support comments, so output files may contain  Replay
           Gain  information if some was present in the input file.  In this case,
           if anything other than a simple format conversion  was  performed  then
           the  output  file Replay Gain information is likely to be incorrect and
           so should be recalculated using a tool that supports this (not SoX).
           The soxi(1) command can be used to display information from audio  file
       Determining & Setting The File Format
           There  are  several mechanisms available for SoX to use to determine or
           set the format characteristics of an audio file.  Depending on the cir-
           cumstances,  individual  characteristics may be determined or set using
           different mechanisms.
           To determine the format of an input file, SoX will  use,  in  order  of
           precedence and as given or available:
           1.  Command-line format options.
           2.  The contents of the file header.
           to resolve the problem.
       Playing & Recording Audio
           The  play  and  rec  commands  are  provided  so that basic playing and
           recording is as simple as
              play existing-file.wav
              rec new-file.wav
           These two commands are functionally equivalent to
              sox existing-file.wav -d
              sox -d new-file.wav
           Of course, further options and effects  (as  described  below)  can  be
           added to the commands in either form.
                                     *        *        *
           Some  systems  provide  more  than  one  type of (SoX-compatible) audio
           driver, e.g. ALSA & OSS, or SUNAU & AO.  Systems  can  also  have  more
           than  one  audio  device (a.k.a. 'sound card').  If more than one audio
           driver has been built-in to SoX, and the default selected by  SoX  when
           recording  or  playing  is  not the one that is wanted, then the AUDIO-
           DRIVER environment variable can be used to override the  default.   For
           example (on many systems):
              set AUDIODRIVER=oss
              play ...
           The  AUDIODEV  environment variable can be used to override the default
           audio device, e.g.
              set AUDIODEV=/dev/dsp2
              play ...
              sox ... -t oss
              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.
           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
           Many  file formats that compress audio discard some of the audio signal
           information whilst doing so. Converting to such a format and then  con-
           verting  back  again  will  not  produce  an exact copy of the original
           audio.  This is the case for many formats used in telephony  (e.g.   A-
           law,  GSM) where low signal bandwidth is more important than high audio
           fidelity, and for many formats used in  portable  music  players  (e.g.
           MP3,  Vorbis)  where  adequate  fidelity  can be retained even with the
           large compression ratios that are needed to make portable players prac-
           Formats that discard audio signal information are called 'lossy'.  For-
           mats that do not are called 'lossless'.  The term 'quality' is used  as
           a  measure  of  how closely the original audio signal can be reproduced
           when using a lossy format.
           Audio file conversion with SoX is lossless when it can  be,  i.e.  when
           not  using  lossy  compression,  when not reducing the sampling rate or
           number of channels, and when the number of bits used in the destination
           format is not less than in the source format.  E.g.  converting from an
           8-bit PCM format to a 16-bit PCM format is lossless but converting from
           an 8-bit PCM format to (8-bit) A-law isn't.
           N.B.   SoX  converts all audio files to an internal uncompressed format
           before performing any audio processing. This means that manipulating  a
           file that is stored in a lossy format can cause further losses in audio
           fidelity.  E.g. with
              sox long.mp3 short.mp3 trim 10
           SoX first decompresses the  input  MP3  file,  then  applies  the  trim
           effect,  and  finally creates the output MP3 file by re-compressing the
           audio - with a possible reduction in fidelity above that which occurred
           when  the input file was created.  Hence, if what is ultimately desired
           is lossily compressed audio, it is highly recommended  to  perform  all
           audio  processing  using  lossless file formats and then convert to the
           lossy format only at the final stage.
           ?   bit-depth reduction has been specified explicitly using a  command-
               line option
           ?   the  output file format supports only bit-depths lower than that of
               the input file format
           ?   an effect has increased effective  bit-depth  within  the  internal
               processing chain
           For  example,  adjusting  volume  with vol 0.25 requires two additional
           bits in which to losslessly  store  its  results  (since  0.25  decimal
           equals  0.01 binary).  So if the input file bit-depth is 16, then SoX's
           internal representation will utilise 18 bits after processing this vol-
           ume  change.   In  order  to  store the output at the same depth as the
           input, dithering is used to remove the additional bits.
           Use the -V option to see what processing SoX has  automatically  added.
           The  -D option may be given to override automatic dithering.  To invoke
           dithering manually (e.g. to select  a  noise-shaping  curve),  see  the
           dither effect.
           Clipping is distortion that occurs when an audio signal level (or 'vol-
           ume') exceeds the range of the chosen representation.  In  most  cases,
           clipping  is  undesirable  and  so should be corrected by adjusting the
           level prior to the point (in the processing chain) at which it  occurs.
           In  SoX,  clipping could occur, as you might expect, when using the vol
           or gain effects to increase the audio volume. Clipping could also occur
           with  many  other  effects,  when converting one format to another, and
           even when simply playing the audio.
           Playing an audio file often involves resampling, and processing by ana-
           logue  components can introduce a small DC offset and/or amplification,
           all of which can produce distortion if the audio signal level was  ini-
           tially too close to the clipping point.
           For these reasons, it is usual to make sure that an audio file's signal
           level has some 'headroom', i.e. it does not exceed a  particular  level
           below  the  maximum  possible level for the given representation.  Some
           standards bodies recommend as much as 9dB headroom, but in most  cases,
           3dB (? 70% linear) is enough.  Note that this wisdom seems to have been
           lost in modern music production; in fact, many CDs, MP3s, etc.  are now
           mastered  at levels above 0dBFS i.e. the audio is clipped as delivered.
           SoX's stat and stats effects can assist in determining the signal level
           in  an  audio file. The gain or vol effect can be used to prevent clip-
           ping, e.g.
              sox dull.wav bright.wav gain -6 treble +6
           the  same  sampling rate. If necessary, separate SoX invocations can be
           used to make sampling rate adjustments prior to combining.
           If the 'concatenate' combining method is selected (usually,  this  will
           be  by  default) then the input files must also have the same number of
           channels.  The audio from each input will be concatenated in the  order
           given to form the output file.
           The 'sequence' combining method is selected automatically for play.  It
           is similar to 'concatenate' in that the audio from each input  file  is
           sent  serially to the output file. However, here the output file may be
           closed and reopened  at  the  corresponding  transition  between  input
           files.  This may be just what is needed when sending different types of
           audio to an output device, but is not generally useful when the  output
           is a normal file.
           If  either  the  'mix' or 'mix-power' combining method is selected then
           two or more input files must be given and will  be  mixed  together  to
           form  the  output file.  The number of channels in each input file need
           not be the same, but SoX will issue a warning if they are not and  some
           channels  in  the  output  file will not contain audio from every input
           file.  A mixed audio file cannot be un-mixed without reference  to  the
           original input files.
           If  the  'merge'  combining  method  is selected then two or more input
           files must be given and will be merged  together  to  form  the  output
           file.   The number of channels in each input file need not be the same.
           A merged audio file comprises all of the channels from all of the input
           files.  Un-merging  is  possible using multiple invocations of SoX with
           the remix effect.  For example, two mono files could be merged to  form
           one  stereo file. The first and second mono files would become the left
           and right channels of the stereo file.
           The 'multiply' combining method multiplies the sample values of  corre-
           sponding  channels  (treated  as numbers in the interval -1 to +1).  If
           the number of channels in the input files is not the same, the  missing
           channels are considered to contain all zero.
           When  combining input files, SoX applies any specified effects (includ-
           ing, for example, the vol volume adjustment effect) after the audio has
           been combined. However, it is often useful to be able to set the volume
           of (i.e. 'balance') the inputs  individually,  before  combining  takes
           For  all  combining  methods, input file volume adjustments can be made
           manually using the -v option (below) which can be given for one or more
           input  files.  If it is given for only some of the input files then the
           others receive no volume adjustment.  In some circumstances,  automatic
           volume adjustments may be applied (see below).
           The -V option (below) can be used to show the input file volume adjust-
           ments that have been selected (either manually or automatically).
           then  dynamic range compression should be applied to correct this - see
           the compand effect.
           With the 'mix-power' combine method, the mixed volume is  approximately
           equal to that of one of the input signals.  This is achieved by balanc-
           ing using a factor of ?/?n instead of ?/n.  Note  that  this  balancing
           factor  does not guarantee that clipping will not occur, but the number
           of clips will usually be low and the resultant distortion is  generally
       Output Files
           SoX's  default  behaviour  is to take one or more input files and write
           them to a single output file.
           This behaviour can be changed by specifying the pseudo-effect 'newfile'
           within the effects list.  SoX will then enter multiple output mode.
           In  multiple  output mode, a new file is created when the effects prior
           to the 'newfile' indicate they are  done.   The  effects  chain  listed
           after  'newfile'  is then started up and its output is saved to the new
           In multiple output mode, a unique number will automatically be appended
           to the end of all filenames.  If the filename has an extension then the
           number is inserted before the extension.  This behaviour  can  be  cus-
           tomized  by  placing  a  %n  anywhere  in the filename where the number
           should be substituted.  An optional number can be placed after the % to
           indicate a minimum fixed width for the number.
           Multiple output mode is not very useful unless an effect that will stop
           the effects chain early is specified before the 'newfile'.  If  end  of
           file  is reached before the effects chain stops itself then no new file
           will be created as it would be empty.
           The following is an example of splitting the first  60  seconds  of  an
           input file into two 30 second files and ignoring the rest.
              sox song.wav ringtone%1n.wav trim 0 30 : newfile : trim 0 30
       Stopping SoX
           Usually SoX will complete its processing and exit automatically once it
           has read all available audio data from the input files.
           If desired, it can be terminated earlier by sending an interrupt signal
           to the process (usually by pressing the keyboard interrupt key which is
           normally Ctrl-C).  This is a natural requirement in some circumstances,
           e.g.  when  using SoX to make a recording.  Note that when using SoX to
           play multiple files, Ctrl-C behaves slightly differently:  pressing  it
           once  causes  SoX  to skip to the next file; pressing it twice in quick
           succession causes SoX to exit.
           Another option to stop processing early is to use an effect that has  a
       Special Filenames
           The following special filenames may be used in certain circumstances in
           place of a normal filename on the command line:
           -      SoX can be used in simple pipeline operations by using the  spe-
                  cial  filename  '-'  which,  if  used as an input filename, will
                  cause SoX will read audio data from  'standard  input'  (stdin),
                  and  which,  if used as the output filename, will cause SoX will
                  send audio data to 'standard output' (stdout).  Note  that  when
                  using  this option for the output file, and sometimes when using
                  it for an input file, the file-type (see -t below) must also  be
           "|program [options] ..."
                  This  can  be  used in place of an input filename to specify the
                  the given program's standard output (stdout) be used as an input
                  file.   Unlike - (above), this can be used for several inputs to
                  one SoX command.  For example, if 'genw' generates mono WAV for-
                  matted  signals  to its standard output, then the following com-
                  mand makes a stereo file from two generated signals:
                     sox -M "|genw --imd -" "|genw --thd -" out.wav
                  For  headerless  (raw)  audio,  -t  (and  perhaps  other  format
                  options) will need to be given, preceding the input command.
                  Specifies  that  filename 'globbing' (wild-card matching) should
                  be performed by SoX instead of by the shell.  This allows a sin-
                  gle  set of file options to be applied to a group of files.  For
                  example, if the current directory contains  three  'vox'  files,
                  file1.vox, file2.vox, and file3.vox, then
                     play --rate 6k *.vox
                  will be expanded by the 'shell' (in most environments) to
                     play --rate 6k file1.vox file2.vox file3.vox
                  which will treat only the first vox file as having a sample rate
                  of 6k.  With
                     play --rate 6k "*.vox"
                  the given sample rate option will be applied to  all  three  vox
           -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:
                  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
                  The  sampling  rate  associated  with  a null file is by default
                  48 kHz, but, as with a normal file, this can  be  overridden  if
                  desired using command-line format options (see below).
       Supported File & Audio Device Types
           See  soxformat(7) for a list and description of the supported file for-
           mats and audio device drivers.


       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:
           Mac OS X GUI: Refer to Apple's Technical Q&A QA1067 document.
           --buffer BYTES, --input-buffer BYTES
                  Set  the  size in bytes of the buffers used for processing audio
                  (default 8192).  --buffer applies to input, effects, and  output
                  processing; --input-buffer applies only to input processing (for
                  which it overrides --buffer if both are given).
                  Be aware that large values for --buffer will  cause  SoX  to  be
                  become  slow  to respond to requests to terminate or to skip the
                  current input file.
                  Don't prompt before overwriting an existing file with  the  same
                  name  as  that  given  for the output file.  This is the default
           --combine concatenate|merge|mix|mix-power|multiply|sequence
                  Select the input file combining method; for some of these, short
                  options are available: -m selects 'mix', -M selects 'merge', and
                  -T selects 'multiply'.
                  See Input File Combining above for a description of the  differ-
                  ent combining methods.
           -D, --no-dither
                  Disable  automatic  dither  - see 'Dither' above.  An example of
                  why this might occasionally be useful is if a file has been con-
                  verted  from  16 to 24 bit with the intention of doing some pro-
                  cessing on it, but in fact no processing is needed after all and
                  the original 16 bit file has been lost, then, strictly speaking,
                  no dither is needed if converting the file back to 16 bit.   See
                  also  the stats effect for how to determine the actual bit depth
                  of the audio within a file.
           --effects-file FILENAME
                  Use FILENAME to obtain all effects  and  their  arguments.   The
                  file  is  parsed  as if the values were specified on the command
                  line.  A new line can be used in place of the special ":" marker
                  to separate effect chains.  This option causes any effects spec-
                  ified on the command line to be discarded.
           -G, --guard
                  Automatically invoke the gain effect to guard against  clipping.
                     sox -G infile -b 16 outfile rate 44100 dither -s
                  is shorthand for
           --i, --info
                  Only  if given as the first parameter to sox, behave as soxi(1).
                  Deprecated alias for --no-clobber.
           -m|-M  Equivalent to --combine mix and --combine merge, respectively.
                  If SoX has been built with the optional 'libmagic' library  then
                  this  option can be given to enable its use in helping to detect
                  audio file types.
           --multi-threaded | --single-threaded
                  By default, SoX is 'single threaded'.  If  the  --multi-threaded
                  option is given however then SoX will process audio channels for
                  most multi-channel effects in parallel on hyper-threading/multi-
                  core  architectures.  This  may  reduce  processing time, though
                  sometimes it may be necessary to use this option  in  conjuction
                  with  a larger buffer size than is the default to gain any bene-
                  fit from multi-threaded processing (e.g.  131072;  see  --buffer
                  Prompt before overwriting an existing file with the same name as
                  that given for the output file.
                  N.B.  Unintentionally overwriting a  file  is  easier  than  you
                  might think, for example, if you accidentally enter
                     sox file1 file2 effect1 effect2 ...
                  when what you really meant was
                     play file1 file2 effect1 effect2 ...
                  then,  without  this  option, file2 will be overwritten.  Hence,
                  using this option is recommended. SOX_OPTS  (above),  a  'shell'
                  alias, script, or batch file may be an appropriate way of perma-
                  nently enabling it.
           --norm Automatically invoke the gain effect to guard  against  clipping
                  and to normalise the audio. E.g.
                     sox --norm infile -b 16 outfile rate 44100 dither -s
                  is shorthand for
                     sox infile -b 16 outfile gain -h rate 44100 gain -nh dither -s
                  See also -V, -G, and the gain effect.
                     sox --plot octave input-file -n highpass 1320 > highpass.plt
                     octave highpass.plt
           -q, --no-show-progress
                  Run in quiet mode when SoX wouldn't otherwise do  so.   This  is
                  the opposite of the -S option.
           -R     Run  in  'repeatable'  mode.   When  this option is given, where
                  applicable, SoX will embed a fixed time-stamp in the output file
                  (e.g.   AIFF)  and  will  'seed' pseudo random number generators
                  (e.g.  dither) with a fixed number, thus ensuring  that  succes-
                  sive  SoX  invocations with the same inputs and the same parame-
                  ters yield the same output.
           --replay-gain track|album|off
                  Select whether or not to apply replay-gain adjustment  to  input
                  files.  The default is off for sox and rec, album for play where
                  (at least) the first two input files are tagged  with  the  same
                  Artist and Album names, and track for play otherwise.
           -S, --show-progress
                  Display  input  file  format/header  information, and processing
                  progress as input file(s) percentage complete, elapsed time, and
                  remaining  time (if known; shown in brackets), and the number of
                  samples written to the output file.  Also shown is a  peak-level
                  meter,  and  an  indication if clipping has occurred.  The peak-
                  level meter shows up to two channels and is calibrated for digi-
                  tal audio as follows (right channel shown):
                                dB FSD   Display   dB FSD   Display
                                 -25     -          -11     ====
                                 -23     =           -9     ====-
                                 -21     =-          -7     =====
                                 -19     ==          -5     =====-
                                 -17     ==-         -3     ======
                                 -15     ===         -1     =====!
                                 -13     ===-
                  A  three-second peak-held value of headroom in dBs will be shown
                  to the right of the meter if this is below 6dB.
                  This option is enabled by default when  using  SoX  to  play  or
                  record audio.
           -T     Equivalent to --combine multiply.
           --temp DIRECTORY
                  Specify  that any temporary files should be created in the given
                  DIRECTORY.  This can be useful if there are permission or  free-
                  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
                  4 and above
                         Messages to help with debugging SoX are also shown.
                  By default, the verbosity level is set to 2  (shows  errors  and
                  warnings).  Each  occurrence of the -V option increases the ver-
                  bosity level by 1.  Alternatively, the verbosity  level  can  be
                  set to an absolute number by specifying it immediately after the
                  -V, e.g.  -V0 sets it to 0.
       Input File Options
           These options apply only to input files  and  may  precede  only  input
           filenames on the command line.
                  Override  an  (incorrect)  audio length given in an audio file's
                  header. If this option is given then SoX will keep reading audio
                  until it reaches the end of the input file.
           -v, --volume FACTOR
                  Intended  for  use  when  combining  multiple  input files, this
                  option adjusts the volume of the file that  follows  it  on  the
                  command  line  by a factor of FACTOR. This allows it to be 'bal-
                  anced' w.r.t. the other input files.  This is a  linear  (ampli-
                  tude)  adjustment,  so a number less than 1 decreases the volume
                  and a number greater than 1 increases it.  If a negative  number
                  is  given  then  in addition to the volume adjustment, the audio
                  signal will be inverted.
                  See also the norm, vol, and gain effects,  and  see  Input  File
                  Balancing above.
       Input & Output File Format Options
           These options apply to the input or output file whose name they immedi-
           ately precede on the command line and are used mainly when working with
           headerless file formats or when specifying a format for the output file
                  converts  a  particular  'raw'  file  to a self-describing 'WAV'
                  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.
                  The  number of bytes in each encoded sample.  Deprecated aliases
                  for -b 8, -b 16, -b 24, -b 32, -b 64 respectively.
           -c CHANNELS, --channels CHANNELS
                  The number of audio channels in the audio file. This can be  any
                  number greater than zero.
                  For  an  input  file,  the most common use for this option is to
                  inform SoX of the number of channels in a  'raw'  ('headerless')
                  audio  file.   Occasionally, it may be useful to use this option
                  with a 'headered' file, in order  to  override  the  (presumably
                  incorrect)  value  in  the  header - note that this is only sup-
                  ported with certain file types.  Examples:
                     sox -r 48k -e float -b 32 -c 2 input.raw output.wav
                  converts a particular 'raw'  file  to  a  self-describing  'WAV'
                     play -c 1 music.wav
                  interprets  the  file  data  as  belonging  to  a single channel
                  regardless of what is indicated in the file header.   Note  that
                  if  the file does in fact have two channels, this will result in
                  the file playing at half speed.
                  For an output file, this option provides a shorthand for  speci-
                  fying  that  the  channels  effect should be invoked in order to
                  change (if necessary) the number of channels in the audio signal
                  to  the  number  given.  For example, the following two commands
                  are equivalent:
                     sox input.wav -c 1 output.wav bass -3
                     sox input.wav      output.wav bass -3 channels 1
                  though the second form is more flexible as it allows the effects
                  to be ordered arbitrarily.
                         Commonly  used with an 8-bit encoding size.  A value of 0
                         represents maximum signal power.
                         PCM data stored as IEEE 753 single precision (32-bit)  or
                         double  precision  (64-bit)  floating-point ('real') num-
                         bers.  A value of 0 represents minimum signal power.
                  a-law  International telephony standard for logarithmic encoding
                         to  8  bits per sample.  It has a precision equivalent to
                         roughly 13-bit PCM and is sometimes encoded with reversed
                         bit-ordering (see the -X option).
                  u-law, mu-law
                         North  American telephony standard for logarithmic encod-
                         ing to 8 bits per sample.  A.k.a. ?-law.  It has a preci-
                         sion  equivalent  to  roughly 14-bit PCM and is sometimes
                         encoded with reversed bit-ordering (see the -X option).
                         OKI (a.k.a. VOX, Dialogic, or Intel) 4-bit ADPCM; it  has
                         a precision equivalent to roughly 12-bit PCM.  ADPCM is a
                         form of audio compression  that  has  a  good  compromise
                         between audio quality and encoding/decoding speed.
                         IMA  (a.k.a. DVI) 4-bit ADPCM; it has a precision equiva-
                         lent to roughly 13-bit PCM.
                         Microsoft 4-bit ADPCM; it has a precision  equivalent  to
                         roughly 14-bit PCM.
                         GSM  is  currently  used  for  the  vast  majority of the
                         world's digital wireless telephone  calls.   It  utilises
                         several  audio formats with different bit-rates and asso-
                         ciated speech quality.  SoX has support for GSM's  origi-
                         nal  13kbps 'Full Rate' audio format.  It is usually CPU-
                         intensive to work with GSM audio.
                  Encoding names can  be  abbreviated  where  this  would  not  be
                  ambiguous; e.g. 'unsigned-integer' can be given as 'un', but not
                  'u' (ambiguous with 'u-law').
                  For an input file, the most common use for  this  option  is  to
                  inform  SoX of the encoding of a 'raw' ('headerless') audio file
                  (see the examples in -b and -c above).
                  For an output file, this option can be used (perhaps along  with
                  -b) to set the output encoding type  For example
                  adpcm,  ima-adpcm,  ms-adpcm, gsm-full-rate respectively (see -e
                  Specifies that filename 'globbing' (wild-card  matching)  should
                  not be performed by SoX on the following filename.  For example,
                  if the current  directory  contains  the  two  files  'five-sec-
                  onds.wav' and 'five*.wav', then
                     play --no-glob "five*.wav"
                  can be used to play just the single file 'five*.wav'.
           -r, --rate RATE[k]
                  Gives the sample rate in Hz (or kHz if appended with 'k') of the
                  For an input file, the most common use for  this  option  is  to
                  inform  SoX  of  the sample rate of a 'raw' ('headerless') audio
                  file (see the examples in -b and -c above).  Occasionally it may
                  be useful to use this option with a 'headered' file, in order to
                  override the (presumably incorrect) value in the header  -  note
                  that  this is only supported with certain file types.  For exam-
                  ple, if audio was recorded with a sample-rate of say 48k from  a
                  source that played back a little, say 1.5%, too slowly, then
                     sox -r 48720 input.wav output.wav
                  effectively  corrects the speed by changing only the file header
                  (but see also the speed effect for the more  usual  solution  to
                  this problem).
                  For  an output file, this option provides a shorthand for speci-
                  fying that the rate effect should be invoked in order to  change
                  (if  necessary) the sample rate of the audio signal to the given
                  value.  For example, the following two commands are equivalent:
                     sox input.wav -r 48k output.wav bass -3
                     sox input.wav        output.wav bass -3 rate 48k
                  though the second form  is  more  flexible  as  it  allows  rate
                  options  to be given, and allows the effects to be ordered arbi-
           -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
                     another-command | sox -t mp3 - output.wav
                  is, respectively, 'little endian', 'big endian', or the opposite
                  to  that  of  the system on which SoX is being used.  Endianness
                  applies only to data encoded as floating-pont, or as  signed  or
                  unsigned  integers of 16 or more bits.  It is often necessary to
                  specify one of these options for headerless files, and sometimes
                  necessary   for  (otherwise)  self-describing  files.   A  given
                  endian-setting option may be ignored for  an  input  file  whose
                  header contains a specific endianness identifier, or for an out-
                  put file that is actually an audio device.
                  N.B.  Unlike other format characteristics, the endianness (byte,
                  nibble,  &  bit ordering) of the input file is not automatically
                  used for the output file; so, for example, when the following is
                  run on a little-endian system:
                     sox -B audio.s16 trimmed.s16 trim 2
                  trimmed.s16 will be created as little-endian;
                     sox -B audio.s16 -B trimmed.s16 trim 2
                  must be used to preserve big-endianness in the output file.
                  The -V option can be used to check the selected orderings.
           -N, --reverse-nibbles
                  Specifies that the nibble ordering (i.e. the 2 halves of a byte)
                  of the samples should be reversed; sometimes useful with  ADPCM-
                  based formats.
                  N.B.  See also N.B. in section on -x above.
           -X, --reverse-bits
                  Specifies  that  the  bit  ordering  of  the  samples  should be
                  reversed; sometimes useful with a few (mostly  headerless)  for-
                  N.B.  See also N.B. in section on -x above.
       Output File Format Options
           These  options  apply  only to the output file and may precede only the
           output filename on the command line.
           --add-comment TEXT
                  Append a comment in the output file header (where applicable).
           --comment TEXT
                  Specify the comment text to store  in  the  output  file  header
                  (where applicable).
                  SoX  will  provide  a  default comment if this option (or --com-
                  ment-file) is not given. To specify that no  comment  should  be


           In  addition  to converting, playing and recording audio files, SoX can
           be used to invoke a number of audio 'effects'.  Multiple effects may be
           applied by specifying them one after another at the end of the SoX com-
           mand line, forming an 'effects chain'.   Note  that  applying  multiple
           effects  in  real-time (i.e. when playing audio) is likely to require a
           high performance computer. Stopping other  applications  may  alleviate
           performance issues should they occur.
           Some  of the SoX effects are primarily intended to be applied to a sin-
           gle instrument or 'voice'.  To facilitate this, the  remix  effect  and
           the  global  SoX option -M can be used to isolate then recombine tracks
           from a multi-track recording.
       Multiple Effect Chains
           A single effects chain is made up of one or more effects.   Audio  from
           the input runs through the chain until either the end of the input file
           is reached or an effect in the chain requests to terminate the chain.
           SoX supports running multiple effects chains over the input audio.   In
           this  case,  when  one chain indicates it is done processing audio, the
           audio data is then sent through the next effects chain.  This continues
           until  either no more effects chains exist or the input has reached the
           end of the file.
           An effects chain is terminated by placing a : (colon) after an  effect.
           Any following effects are a part of a new effects chain.
           It  is  important  to  place the effect that will stop the chain as the
           first effect in the chain.   This  is  because  any  samples  that  are
           buffered  by effects to the left of the terminating effect will be dis-
           carded.  The amount of samples discarded is  related  to  the  --buffer
           option and it should be kept small, relative to the sample rate, if the
           terminating effect cannot be first.  Further  information  on  stopping
           effects can be found in the Stopping SoX section.
           There  are a few pseudo-effects that aid using multiple effects chains.
           These include newfile which will start writing to  a  new  output  file
           before  moving  to  the  next effects chain and restart which will move
           back to the first effects chain.  Pseudo-effects must be  specified  as
           the  first  effect  in  a chain and as the only effect in a chain (they
           must have a : before and after they are specified).
           The following is an example of multiple effects chains.  It will  split
           the  input file into multiple files of 30 seconds in length.  Each out-
           put filename will have unique number in its name as documented  in  the
           Output Files section.
              sox infile.wav output.wav trim 0 30 : newfile : restart
       Common Notation And Parameters
           gain   A power gain in dB.  Zero gives no gain; less than zero gives an
                  Used to specify the band-width of a filter.  A number of differ-
                  ent methods to specify the width are available (though  not  all
                  for  every effect).  One of the characters shown may be appended
                  to select the desired method as follows:
                                            Method    Notes
                                       h      Hz
                                       k     kHz
                                       o   Octaves
                                       q   Q-factor   See [2]
                  For each effect that uses this  parameter,  the  default  method
                  (i.e.  if  no  character  is appended) is the one that it listed
                  first in the first line of the effect's description.
           To see if SoX has support for an optional effect, enter sox -h and look
           for its name under the list: 'EFFECTS'.
       Supported Effects
           Note:  a categorised list of the effects can be found in the accompany-
           ing 'README' file.
           allpass frequency[k] width[h|k|o|q]
                  Apply a two-pole all-pass filter with central frequency (in  Hz)
                  frequency,  and  filter-width width.  An all-pass filter changes
                  the audio's frequency to phase relationship without changing its
                  frequency to amplitude relationship.  The filter is described in
                  detail in [1].
                  This effect supports the --plot global option.
           band [-n] center[k] [width[h|k|o|q]]
                  Apply a band-pass filter.  The frequency  response  drops  loga-
                  rithmically  around  the  center frequency.  The width parameter
                  gives the slope of the drop.  The frequencies at center +  width
                  and  center  -  width will be half of their original amplitudes.
                  band defaults to a mode oriented to pitched audio,  i.e.  voice,
                  singing,  or instrumental music.  The -n (for noise) option uses
                  the alternate  mode  for  un-pitched  audio  (e.g.  percussion).
                  Warning: -n introduces a power-gain of about 11dB in the filter,
                  so beware of output clipping.   band  introduces  noise  in  the
                  shape  of  the  filter, i.e. peaking at the center frequency and
                  settling around it.
                  This effect supports the --plot global option.
                  See also sinc for a bandpass filter with steeper shoulders.
                  Apply a band-reject filter.  See the description of the bandpass
                  effect for details.
           bass|treble gain [frequency[k] [width[s|h|k|o|q]]]
                  Boost or cut the bass (lower) or treble (upper)  frequencies  of
                  the audio using a two-pole shelving filter with a response simi-
                  lar to that of a standard hi-fi's tone-controls.  This  is  also
                  known as shelving equalisation (EQ).
                  gain  gives  the  gain  at  0 Hz (for bass), or whichever is the
                  lower of ~22 kHz and the Nyquist frequency  (for  treble).   Its
                  useful  range is about -20 (for a large cut) to +20 (for a large
                  boost).  Beware of Clipping when using a positive gain.
                  If desired, the filter can be  fine-tuned  using  the  following
                  optional parameters:
                  frequency sets the filter's central frequency and so can be used
                  to extend or reduce the frequency range to be  boosted  or  cut.
                  The default value is 100 Hz (for bass) or 3 kHz (for treble).
                  width determines how steep is the filter's shelf transition.  In
                  addition to the common  width  specification  methods  described
                  above,  'slope'  (the  default,  or if appended with 's') may be
                  used.  The useful range of 'slope' is about 0.3,  for  a  gentle
                  slope,  to 1 (the maximum), for a steep slope; the default value
                  is 0.5.
                  The filters are described in detail in [1].
                  These effects support the --plot global option.
                  See also equalizer for a peaking equalisation effect.
           bend [-f frame-rate(25)] [-o over-sample(16)] { delay,cents,duration }
                  Changes pitch by specified amounts  at  specified  times.   Each
                  given triple: delay,cents,duration specifies one bend.  delay is
                  the amount of time after the start of the audio stream,  or  the
                  end  of  the previous bend, at which to start bending the pitch;
                  cents is the number of cents (100 cents = 1 semitone)  by  which
                  to  bend  the  pitch, and duration the length of time over which
                  the pitch will be bent.
                  The pitch-bending algorithm utilises the Discrete Fourier Trans-
                  form  (DFT)  at  a particular frame rate and over-sampling rate.
                  The -f and -o parameters may be used to adjust these  parameters
                  and thus control the smoothness of the changes in pitch.
                  For  example,  an  initial  tone  is  generated, then bent three
                  times, yielding four different notes in total:
                     play -n synth 2.5 sin 667 gain 1 \
                  Invoke a simple algorithm to change the number  of  channels  in
                  the  audio  signal  to  the  given  number  CHANNELS:  mixing if
                  decreasing the number of channels or duplicating  if  increasing
                  the number of channels.
                  The  channels effect is invoked automatically if SoX's -c option
                  specifies a number of channels that is different to that of  the
                  input  file(s).   Alternatively, if this effect is given explic-
                  itly, then SoX's -c option need not be given.  For example,  the
                  following two commands are equivalent:
                     sox input.wav -c 1 output.wav bass -3
                     sox input.wav      output.wav bass -3 channels 1
                  though the second form is more flexible as it allows the effects
                  to be ordered arbitrarily.
                  See also  remix  for  an  effect  that  allows  channels  to  be
                  mixed/selected arbitrarily.
           chorus gain-in gain-out <delay decay speed depth -s|-t>
                  Add  a chorus effect to the audio.  This can make a single vocal
                  sound like a chorus, but can also be applied to instrumentation.
                  Chorus  resembles an echo effect with a short delay, but whereas
                  with echo the delay is constant, with chorus, it is varied using
                  sinusoidal  or  triangular  modulation.   The  modulation  depth
                  defines the range the modulated delay is played before or  after
                  the  delay. Hence the delayed sound will sound slower or faster,
                  that is the delayed sound tuned around the original one, like in
                  a  chorus  where  some vocals are slightly off key.  See [3] for
                  more discussion of the chorus effect.
                  Each  four-tuple  parameter  delay/decay/speed/depth  gives  the
                  delay in milliseconds and the decay (relative to gain-in) with a
                  modulation speed in Hz using depth in milliseconds.  The modula-
                  tion  is either sinusoidal (-s) or triangular (-t).  Gain-out is
                  the volume of the output.
                  A typical delay is around 40ms to 60ms; the modulation speed  is
                  best near 0.25Hz and the modulation depth around 2ms.  For exam-
                  ple, a single delay:
                     play guitar1.wav chorus 0.7 0.9 55 0.4 0.25 2 -t
                  Two delays of the original samples:
                     play guitar1.wav chorus 0.6 0.9 50 0.4 0.25 2 -t \
                        60 0.32 0.4 1.3 -s
                  A fuller sounding chorus (with three additional delays):
                  attack time (response to the music  getting  louder)  should  be
                  shorter than the decay time because the human ear is more sensi-
                  tive to sudden loud music than sudden soft  music.   Where  more
                  than  one  pair  of  attack/decay parameters are specified, each
                  input channel is companded separately and the  number  of  pairs
                  must  agree  with  the number of input channels.  Typical values
                  are 0.3,0.8 seconds.
                  The second parameter is a list  of  points  on  the  compander's
                  transfer function specified in dB relative to the maximum possi-
                  ble signal amplitude.  The input values must be  in  a  strictly
                  increasing  order  but the transfer function does not have to be
                  monotonically rising.  If omitted, the value of out-dB1 defaults
                  to  the  same  value as in-dB1; levels below in-dB1 are not com-
                  panded (but may have gain applied to them).  The  point  0,0  is
                  assumed  but  may  be overridden (by 0,out-dBn).  If the list is
                  preceded by a soft-knee-dB value, then the points at where adja-
                  cent line segments on the transfer function meet will be rounded
                  by the amount given.  Typical values for the  transfer  function
                  are 6:-70,-60,-20.
                  The third (optional) parameter is an additional gain in dB to be
                  applied at all points on the transfer function and  allows  easy
                  adjustment of the overall gain.
                  The  fourth  (optional)  parameter  is  an  initial  level to be
                  assumed for each channel when companding starts.   This  permits
                  the user to supply a nominal level initially, so that, for exam-
                  ple, a very large gain is not applied to initial  signal  levels
                  before  the  companding action has begun to operate: it is quite
                  probable that in such an event, the  output  would  be  severely
                  clipped  while  the  compander  gain properly adjusts itself.  A
                  typical value (for audio which is initially quiet) is -90 dB.
                  The fifth (optional) parameter is a delay in seconds.  The input
                  signal  is analysed immediately to control the compander, but it
                  is delayed before being fed to the volume adjuster.   Specifying
                  a delay approximately equal to the attack/decay times allows the
                  compander to effectively operate in a 'predictive' rather than a
                  reactive mode.  A typical value is 0.2 seconds.
                                        *        *        *
                  The  following  example  might  be used to make a piece of music
                  with both quiet and loud passages suitable for listening to in a
                  noisy environment such as a moving vehicle:
                     sox asz.wav asz-car.wav compand 0.3,1 6:-70,-60,-20 -5 -90 0.2
                  The  transfer  function ('6:-70,...') says that very soft sounds
                  (below -70dB) will remain unchanged.  This will stop the compan-
                  der  from  boosting  the  volume  on  'silent'  passages such as
                     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
                  See also mcompand for a multiple-band companding effect.
           contrast [enhancement-amount(75)]
                  Comparable with compression, this effect modifies an audio  sig-
                  nal  to  make  it sound louder.  enhancement-amount controls the
                  amount of the enhancement and is a number in  the  range  0-100.
                  Note  that enhancement-amount = 0 still gives a significant con-
                  trast enhancement.
                  See also the compand and mcompand effects.
           dcshift shift [limitergain]
                  Apply a DC shift to the audio.  This can be useful to  remove  a
                  DC offset (caused perhaps by a hardware problem in the recording
                  chain) from the audio.  The effect of a  DC  offset  is  reduced
                  headroom and hence volume.  The stat or stats effect can be used
                  to determine if a signal has a DC offset.
                  The given dcshift value is a floating point number in the  range
                  of  ?2 that indicates the amount to shift the audio (which is in
                  the range of ?1).
                  An optional limitergain can be specified  as  well.   It  should
                  have  a  value  much less than 1 (e.g. 0.05 or 0.02) and is used
                  only on peaks to prevent clipping.
                                        *        *        *
                  An alternative approach to removing a DC offset (albeit  with  a
                  short delay) is to use the highpass filter effect at a frequency
                  of say 10Hz, as illustrated in the following example:
                     sox -n dc.wav synth 5 sin %0 50
                     sox dc.wav fixed.wav highpass 10
           deemph Apply Compact Disc (IEC 60908) de-emphasis (a treble attenuation
                  shelving filter).
                  Pre-emphasis  was applied in the mastering of some CDs issued in
                     sox track1.wav track1-deemph.wav deemph
                  and then burn track1-deemph.wav to CD, or
                     play track1-deemph.wav
                  or simply
                     play track1.wav deemph
                  The de-emphasis filter is implemented as a biquad;  its  maximum
                  deviation  from the ideal response is only 0.06dB (up to 20kHz).
                  This effect supports the --plot global option.
                  See also the bass and treble shelving equalisation effects.
           delay {length}
                  Delay one or more audio channels.  length can specify a time or,
                  if  appended  with  an 's', a number of samples.  Do not specify
                  both time and samples delays in the same command.  For  example,
                  delay  1.5  0  0.5  delays the first channel by 1.5 seconds, the
                  third channel by 0.5 seconds, and leaves the second channel (and
                  any other channels that may be present) un-delayed.  The follow-
                  ing (one long) command plays a chime sound:
                     play -n synth -j 3 sin %3 sin %-2 sin %-5 sin %-9 \
                       sin %-14 sin %-21 fade h .01 2 1.5 delay \
                       1.3 1 .76 .54 .27 remix - fade h 0 2.7 2.5 norm -1
                  and this plays a guitar chord:
                     play -n synth pl G2 pl B2 pl D3 pl G3 pl D4 pl G4 \
                       delay 0 .05 .1 .15 .2 .25 remix - fade 0 4 .1 norm -1
           dither [-a] [-S|-s|-f filter]
                  Apply dithering to the audio.   Dithering  deliberately  adds  a
                  small  amount  of  noise  to the signal in order to mask audible
                  quantization effects that can occur if the output sample size is
                  less than 24 bits.  With no options, this effect will add trian-
                  gular (TPDF) white noise.  Noise-shaping (only for certain  sam-
                  ple  rates)  can be selected with -s.  With the -f option, it is
                  possible to select a particular noise-shaping  filter  from  the
                  following   list:   lipshitz,  f-weighted,  modified-e-weighted,
                  improved-e-weighted, gesemann, shibata,  low-shibata,  high-shi-
                  bata.   Note  that  most  filter  types  are available only with
                  44100Hz sample rate.  The filter types are distinguished by  the
                  following  properties: audibility of noise, level of (inaudible,
                  but in some circumstances, otherwise  problematic)  shaped  high
                  so the fades should be carefully checked for any  noise  modula-
                  tion;  if  this occurs, then either re-dither the whole file, or
                  use trim, fade, and concatencate.
                  If the SoX global option  -R  option  is  not  given,  then  the
                  pseudo-random  number generator used to generate the white noise
                  will be 'reseeded', i.e. the generated noise will  be  different
                  between invocations.
                  This  effect  should  not  be  followed by any other effect that
                  affects the audio.
                  See also the 'Dither' section above.
           earwax Makes audio easier to listen to on headphones.  Adds  'cues'  to
                  44.1kHz  stereo  (i.e.  audio CD format) audio so that when lis-
                  tened to on headphones the stereo image  is  moved  from  inside
                  your  head  (standard for headphones) to outside and in front of
                  the listener (standard  for  speakers).   See  http://www.geoci-
         for a full explanation.
           echo gain-in gain-out <delay decay>
                  Add  echoing  to  the audio.  Echoes are reflected sound and can
                  occur naturally amongst mountains (and  sometimes  large  build-
                  ings)  when  talking  or  shouting; digital echo effects emulate
                  this behaviour and are often used to help fill out the sound  of
                  a  single  instrument or vocal.  The time difference between the
                  original signal and the reflection is the  'delay'  (time),  and
                  the  loudness  of the reflected signal is the 'decay'.  Multiple
                  echoes can have different delays and decays.
                  Each given delay decay pair gives the delay in milliseconds  and
                  the  decay  (relative to gain-in) of that echo.  Gain-out is the
                  volume of the output.  For example: This will make it  sound  as
                  if there are twice as many instruments as are actually playing:
                     play lead.aiff echo 0.8 0.88 60 0.4
                  If  the  delay  is  very  short, then it sound like a (metallic)
                  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  moun-
                     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
                     play lead.aiff echos 0.8 0.7 700 0.25 700 0.3
                  The sample will be bounced twice in asymmetric echos:
                     play lead.aiff echos 0.8 0.7 700 0.25 900 0.3
                  The sample will sound as if played in a garage:
                     play lead.aiff echos 0.8 0.7 40 0.25 63 0.3
           equalizer frequency[k] width[q|o|h|k] gain
                  Apply  a  two-pole  peaking equalisation (EQ) filter.  With this
                  filter, the signal-level at and around a selected frequency  can
                  be  increased  or  decreased, whilst (unlike band-pass and band-
                  reject filters) that at all other frequencies is unchanged.
                  frequency gives the filter's central frequency in Hz, width, the
                  band-width,  and  gain  the  required gain or attenuation in dB.
                  Beware of Clipping when using a positive gain.
                  In order to produce complex equalisation curves, this effect can
                  be given several times, each with a different central frequency.
                  The filter is described in detail in [1].
                  This effect supports the --plot global option.
                  See also bass and treble for shelving equalisation effects.
           fade [type] fade-in-length [stop-time [fade-out-length]]
                  Apply a fade effect to the beginning, end, or both of the audio.
                  An  optional  type  can  be specified to select the shape of the
                  fade curve: q for quarter of a sine wave,  h  for  half  a  sine
                  wave,  t for linear ('triangular') slope, l for logarithmic, and
                  p for inverted parabola.  The default is logarithmic.
                  A fade-in starts from the first  sample  and  ramps  the  signal
                  level  from 0 to full volume over fade-in-length seconds.  Spec-
                  ify 0 seconds if no fade-in is wanted.
                  For fade-outs, the audio will be truncated at stop-time and  the
                  signal  level will be ramped from full volume down to 0 starting
                  at fade-out-length seconds before the stop-time.   If  fade-out-
                  length  is not specified, it defaults to the same value as fade-
                  in-length.  No fade-out is performed if stop-time is not  speci-
                  fied.   If the file length can be determined from the input file
                  header and length-changing effects are not in effect, then 0 may
                  be specified for stop-time to indicate the usual case of a fade-
                  out that ends at the end of the input audio stream.
                  are  read  from the 'standard input' (stdin); otherwise, coeffi-
                  cients may be given on the command line.  Examples:
                     sox infile outfile fir 0.0195 -0.082 0.234 0.891 -0.145 0.043
                     sox infile outfile fir coefs.txt
                  with coefs.txt containing
                     # HP filter
                     # freq=10000
           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:
           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
                  sary  to  prevent  clipping  whilst  balancing,  attenuation  is
                  applied  to  all  channels.   Note, however, that in conjunction
                  with -n, -B and -b are synonymous.
                  The -r option is used in conjunction with a prior invocation  of
                  gain with the -h option - see below for details.
                  The  -n option normalises the audio to 0dB FSD; it is often used
                  in conjunction with a negative gain-dB to the  effect  that  the
                  audio is normalised to a given level below 0dB.  For example,
                     sox infile outfile gain -n
                  normalises to 0dB, and
                     sox infile outfile gain -n -3
                  normalises to -3dB.
                  The -l option invokes a simple limiter, e.g.
                     sox infile outfile gain -l 6
                  will  apply 6dB of gain but never clip.  Note that limiting more
                  than a few dBs more than occasionally (in a piece of  audio)  is
                  not  recommended  as  it  can cause audible distortion.  See the
                  compand effect for a more capable limiter.
                  The -h option is used to apply gain  to  provide  head-room  for
                  subsequent processing.  For example, with
                     sox infile outfile gain -h bass +6
                  6dB  of  attenuation  will be applied prior to the bass boosting
                  effect thus ensuring that it will not  clip.   Of  course,  with
                  bass,  it  is obvious how much headroom will be needed, but with
                  other effects (e.g.  rate, dither) it is not  always  as  clear.
                  Another  advantage  of  using  gain  -h  rather than an explicit
                  attenuation, is that if the headroom is not used  by  subsequent
                  effects, it can be reclaimed with gain -r, for example:
                     sox infile outfile gain -h bass +6 rate 44100 gain -r
                  The above effects chain guarantees never to clip nor amplify; it
                  attenuates if necessary to prevent clipping, but by only as much
                  as is needed to do so.
                  Output  formatting  (dithering  and  bit-depth  reduction)  also
                  requires headroom (which cannot be 'reclaimed'), e.g.
                     sox infile outfile gain -h bass +6 rate 44100 gain -rh dither
                  per  pole per decade).  The double-pole filters are described in
                  detail in [1].
                  These effects support the --plot global option.
                  See also sinc for filters with a steeper roll-off.
           ladspa module [plugin] [argument...]
                  Apply a LADSPA [5] (Linux Audio Developer's Simple  Plugin  API)
                  plugin.   Despite  the name, LADSPA is not Linux-specific, and a
                  wide range of effects is available as LADSPA  plugins,  such  as
                  cmt  [6]  (the Computer Music Toolkit) and Steve Harris's plugin
                  collection [7]. The first argument is  the  plugin  module,  the
                  second  the  name  of the plugin (a module can contain more than
                  one plugin) and any other arguments are for the control ports of
                  the  plugin. Missing arguments are supplied by default values if
                  possible. Only plugins with at most  one  audio  input  and  one
                  audio  output port can be used.  If found, the environment vari-
                  able LADSPA_PATH will be used as search path for plugins.
           loudness [gain [reference]]
                  Loudness control - similar to  the  gain  effect,  but  provides
                  equalisation    for    the    human    auditory   system.    See
         for a detailed description
                  of  loudness.   The gain is adjusted by the given gain parameter
                  (usually negative) and the signal equalised according to ISO 226
                  w.r.t.  a  reference level of 65dB, though an alternative refer-
                  ence level may be given if the original audio has been equalised
                  for  some  other optimal level.  A default gain of -10dB is used
                  if a gain value is not given.
                  See also the gain effect.
           lowpass [-1|-2] frequency[k] [width[q|o|h|k]]
                  Apply a low-pass filter.  See the description  of  the  highpass
                  effect for details.
           mcompand "attack1,decay1{,attack2,decay2}
                  [gain     [initial-volume-dB    [delay]]]"    {crossover-freq[k]
                  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:
                  See also compand for a single-band companding effect.
           mixer [ -l|-r|-f|-b|-1|-2|-3|-4|n{,n} ]
                  Reduce the number of audio channels by mixing or selecting chan-
                  nels, or increase the number of channels  by  duplicating  chan-
                  nels.   Note:  this effect operates on the audio channels within
                  the SoX effects processing chain; it should not be confused with
                  the  -m  global  option  (where  multiple files are mix-combined
                  before entering the effects chain).
                  When reducing the number of channels it is possible to  use  the
                  -l, -r, -f, -b, -1, -2, -3, -4, options to select only the left,
                  right, front, back channel(s) or specific channel for the output
                  instead  of averaging the channels.  The -l, and -r options will
                  do averaging in quad-channel files so select the  exact  channel
                  to prevent this.
                  The mixer effect can also be invoked with up to 16 numbers, sep-
                  arated by commas, which specify the proportion (0 = 0% and  1  =
                  100%) of each input channel that is to be mixed into each output
                  channel.  In two-channel mode, 4 numbers are given: l -> l,  l  ->
                  r,  r  ->  l, and r -> r, respectively.  In four-channel mode, the
                  first 4 numbers give the proportions for the  left-front  output
                  channel,  as  follows:  lf  -> lf, rf -> lf, lb -> lf, and rb -> rf.
                  The next 4 give the right-front output in the same  order,  then
                  left-back and right-back.
                  It  is  also  possible to use the 16 numbers to expand or reduce
                  the channel count; just specify 0 for unused channels.
                  Finally, certain reduced combination of numbers can be specified
                  for certain input/output channel combinations.
                       In Ch   Out Ch   Num   Mappings
                         2       1       2    l -> l, r -> l
                         2       2       1    adjust balance
                         4       1       4    lf -> l, rf -> l, lb -> l, rb -> l
                         4       2       2    lf -> l&rf -> r, lb -> l&rb -> r
                         4       4       1    adjust balance
                         4       4       2    front balance, back balance
                  See  also  remix  for a mixing effect that handles any number of
           noiseprof [profile-file]
                  Calculate a profile of the audio for  use  in  noise  reduction.
                  See the description of the noisered effect for details.
           noisered [profile-file [amount]]
                  Reduce  noise  in  the  audio signal by profiling and filtering.
                  This effect is moderately effective at removing consistent back-
                     sox speech.wav cleaned.wav noisered speech.noise-profile 0.3
                  How much noise should be removed is specified by amount-a number
                  between 0 and 1 with a default  of  0.5.   Higher  numbers  will
                  remove  more  noise but present a greater likelihood of removing
                  wanted components of the  audio  signal.   Before  replacing  an
                  original recording with a noise-reduced version, experiment with
                  different amount values to find the optimal one for your  audio;
                  use  headphones  to  check  that you are happy with the results,
                  paying particular attention to quieter sections of the audio.
                  On most systems, the two stages - profiling and reduction -  can
                  be combined using a pipe, e.g.
                     sox noisy.wav -n trim 0 1 noiseprof | play noisy.wav noisered
           norm [dB-level]
                  Normalise the audio.  norm is just an alias for gain -n; see the
                  gain effect for details.
                  Note that norm's -i and -b options are deprecated  (having  been
                  superseded  by  gain  -en  and gain -B respectively) and will be
                  removed in a future release.
           oops   Out Of Phase Stereo effect.  Mixes  stereo  to  twin-mono  where
                  each  mono  channel contains the difference between the left and
                  right stereo channels.  This is sometimes known as the 'karaoke'
                  effect as it often has the effect of removing most or all of the
                  vocals from a recording.
           overdrive [gain(20) [colour(20)]]
                  Non linear distortion.  The colour parameter controls the amount
                  of even harmonic content in the over-driven output.
           pad { length[@position] }
                  Pad  the  audio  with silence, at the beginning, the end, or any
                  specified points through the audio.  Both  length  and  position
                  can specify a time or, if appended with an 's', a number of sam-
                  ples.  length is the amount of silence to  insert  and  position
                  the  position  in  the input audio stream at which to insert it.
                  Any number of lengths and positions may be  specified,  provided
                  that  a  specified  position  is not less that the previous one.
                  position is optional for the first and  last  lengths  specified
                  and  if  omitted  correspond to the beginning and the end of the
                  audio respectively.  For example, pad 1.5 1.5 adds  1.5  seconds
                  of  silence  padding  at  each  end  of  the  audio,  whilst pad
                  4000s@3:00 inserts 4000 samples of silence 3  minutes  into  the
                  audio.  If silence is wanted only at the end of the audio, spec-
                  ify either the end position or specify a zero-length pad at  the
                  For example:
                     play snare.flac phaser 0.8 0.74 3 0.4 0.5 -t
                     play snare.flac phaser 0.9 0.85 4 0.23 1.3 -s
                  A popular sound:
                     play snare.flac phaser 0.89 0.85 1 0.24 2 -t
                  More severe:
                     play snare.flac phaser 0.6 0.66 3 0.6 2 -t
           pitch [-q] shift [segment [search [overlap]]]
                  Change the audio pitch (but not tempo).
                  shift  gives  the  pitch  shift  as positive or negative 'cents'
                  (i.e. 100ths of  a  semitone).   See  the  tempo  effect  for  a
                  description of the other parameters.
                  See also the speed and tempo effects.
           rate [-q|-l|-m|-h|-v] [override-options] RATE[k]
                  Change  the audio sampling rate (i.e. resample the audio) to any
                  given RATE (even non-integer if this is supported by the  output
                  file format) using a quality level defined as follows:
                               Quality   Band-  Rej dB   Typical Use
                         -q     quick     n/a   ?30 @    playback on
                                                 Fs/4    ancient hardware
                         -l      low      80%    100     playback on old
                         -m    medium     95%    100     audio playback
                         -h     high      95%    125     16-bit mastering
                                                         (use with dither)
                         -v   very high   95%    175     24-bit mastering
                  where Band-width is the percentage of the audio  frequency  band
                  that  is  preserved  and Rej dB is the level of noise rejection.
                  Increasing levels of resampling quality come at the  expense  of
                  increasing  amounts of time to process the audio.  If no quality
                  option is given, the quality level used is 'high'.
                  The 'quick' algorithm uses cubic interpolation; all  others  use
                  band-limited  interpolation.   By default, all algorithms have a
                                        *        *        *
                  Warning: technically detailed discussion follows.
                  The  simple  quality selection described above provides settings
                  that satisfy the needs of the vast majority of resampling tasks.
                  Occasionally,  however,  it  may  be  desirable to fine-tune the
                  resampler's filter response; this can be  achieved  using  over-
                  ride options, as detailed in the following table:
                  -M/-I/-L     Phase response = minimum/intermediate/linear
                  -s           Steep filter (band-width = 99%)
                  -a           Allow aliasing/imaging above the pass-band
                  -b 74-99.7   Any band-width %
                  -p 0-100     Any phase response (0 = minimum, 25 = intermediate,
                               50 = linear, 100 = maximum)
                  N.B.  Override options can not be used with the 'quick' or 'low'
                  quality algorithms.
                  All  resamplers  use  filters  that  can sometimes create 'echo'
                  (a.k.a.  'ringing') artefacts with  transient  signals  such  as
                  those  that occur with 'finger snaps' or other highly percussive
                  sounds.  Such artefacts are much more noticeable  to  the  human
                  ear if they occur before the transient ('pre-echo') than if they
                  occur after it ('post-echo').  Note that frequency of  any  such
                  artefacts is related to the smaller of the original and new sam-
                  pling rates but that if this is at least 44.1kHz, then the arte-
                  facts will lie outside the range of human hearing.
                  A phase response setting may be used to control the distribution
                  of any transient echo between 'pre'  and  'post':  with  minimum
                  phase, there is no pre-echo but the longest post-echo; with lin-
                  ear phase, pre and post echo are in  equal  amounts  (in  signal
                  terms, but not audibility terms); the intermediate phase setting
                  attempts to find the best compromise by selecting a small length
                  (and level) of pre-echo and a medium lengthed post-echo.
                  Minimum,  intermediate,  or  linear  phase  response is selected
                  using the -M, -I, or -L option; a custom phase response  can  be
                  created  with  the -p option.  Note that phase responses between
                  'linear' and 'maximum' (greater than 50) are rarely useful.
                  A resampler's band-width setting determines how much of the fre-
                  quency  content of the original signal (w.r.t. the original sam-
                  ple rate when up-sampling, or the new sample rate when down-sam-
                  pling)  is preserved during conversion.  The term 'pass-band' is
                  used to refer to all frequencies  up  to  the  band-width  point
                  (e.g.  for 44.1kHz sampling rate, and a resampling band-width of
                  95%, the pass-band represents frequencies  from  0Hz  (D.C.)  to
                  above  21kHz  can be distorted; however, since this is above the
                  pass-band (i.e.  above the highest frequency  of  interest/audi-
                  bility),  this  may  not be a problem.  The benefits of allowing
                  aliasing/imaging are reduced processing time,  and  reduced  (by
                  almost half) transient echo artefacts.  Note that if this option
                  is  given,  then  the  minimum  band-width  allowable  with   -b
                  increases to 85%.
                     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
                                        *        *        *
                  The  pitch,  speed  and tempo effects all use the rate effect at
                  their core.
           remix [-a|-m|-p] <out-spec>
                  out-spec  = in-spec{,in-spec} | 0
                  in-spec   = [in-chan][-[in-chan2]][vol-spec]
                  vol-spec  = p|i|v[volume]
                  Select and mix input audio channels into output audio  channels.
                  Each  output channel is specified, in turn, by a given out-spec:
                  a list of contributing input channels and volume specifications.
                  Note  that this effect operates on the audio channels within the
                  SoX effects processing chain; it should not be confused with the
                  -m  global  option (where multiple files are mix-combined before
                  entering the effects chain).
                  An out-spec contains comma-separated input  channel-numbers  and
                  hyphen-delimited  channel-number ranges; alternatively, 0 may be
                  given to create a silent output channel.  For example,
                     sox input.wav output.wav remix 6 7 8 0
                  creates an output file with four channels, where channels 1,  2,
                  and  3 are copies of channels 6, 7, and 8 in the input file, and
                  channel 4 is silent.  Whereas
                     sox input.wav output.wav remix 1-3,7 3
                  input channels, each input channel will be scaled by a factor of
                  ?/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 =
                  If an out-spec includes at least one vol-spec then, by  default,
                  ?/n  scaling  is  not  applied to any other channels in the same
                  out-spec (though may be in other out-specs).  The -a (automatic)
                  option  however, can be given to retain the automatic scaling in
                  this case.  For example,
                     sox input.wav output.wav remix 1,2 3,4v0.8
                  results in channel level multipliers of 0.5,0.5 1,0.8, whereas
                     sox input.wav output.wav remix -a 1,2 3,4v0.8
                  results in channel level multipliers of 0.5,0.5 0.5,0.8.
                  The -m (manual) option disables  all  automatic  volume  adjust-
                  ments, so
                     sox input.wav output.wav remix -m 1,2 3,4v0.8
                  results in channel level multipliers of 1,1 1,0.8.
                  The  volume number is optional and omitting it corresponds to no
                  volume change; however, the only case in which this is useful is
                  in  conjunction  with  i.   For example, if input.wav is stereo,
                     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  ?/n  scaling  is
                  replaced  by ?/?n ('power') scaling; this gives a louder mix but
                  one that might occasionally clip.
                                        *        *        *
                  If a file input.wav containing six audio  channels  were  given,
                  the   script  would  produce  six  output  files:  input-01.wav,
                  input-02.wav, ..., input-06.wav.
                  See also mixer and swap for similar effects.
           repeat count
                  Repeat the entire audio count times.   Requires  temporary  file
                  space  to  store  the audio to be repeated.  Note that repeating
                  once yields two copies: the  original  audio  and  the  repeated
           reverb [-w|--wet-only] [reverberance (50%) [HF-damping (50%)
                  [room-scale (100%) [stereo-depth (100%)
                  [pre-delay (0ms) [wet-gain (0dB)]]]]]]
                  Add  reverberation  to the audio using the 'freeverb' algorithm.
                  A reverberation effect is sometimes desirable for concert  halls
                  that  are  too  small  or contain so many people that the hall's
                  natural reverberance is diminished.  Applying a small amount  of
                  stereo  reverb to a (dry) mono signal will usually make it sound
                  more natural.  See [3] for a detailed description of  reverbera-
                  Note  that  this effect increases both the volume and the length
                  of the audio, so to prevent clipping in these domains, a typical
                  invocation might be:
                     play dry.wav gain -3 pad 0 3 reverb
                  The -w option can be given to select only the 'wet' signal, thus
                  allowing it to be processed further, independently of the  'dry'
                  signal.  E.g.
                     play -m voice.wav "|sox voice.wav -p reverse reverb -w reverse"
                  for a reverse reverb effect.
                  Reverse  the audio completely.  Requires temporary file space to
                  store the audio to be reversed.
           riaa   Apply RIAA vinyl playback equalisation.  The sampling rate  must
                  be one of: 44.1, 48, 88.2, 96 kHz.
                  This effect supports the --plot global option.
           silence [-l] above-periods [duration threshold[d|%]
                  [below-periods duration threshold[d|%]]
                  Removes silence from the beginning, middle, or end of the audio.
                  When above-periods is non-zero, you must also specify a duration
                  and threshold. Duration indications the amount of time that non-
                  silence must be detected before  it  stops  trimming  audio.  By
                  increasing  the  duration,  burst  of  noise  can  be treated as
                  silence and trimmed off.
                  Threshold is used to indicate what sample value you should treat
                  as silence.  For digital audio, a value of 0 may be fine but for
                  audio recorded from analog, you may wish to increase  the  value
                  to account for background noise.
                  When  optionally trimming silence from the end of the audio, you
                  specify a below-periods count.  In this case, below-period means
                  to  remove  all audio after silence is detected.  Normally, this
                  will be a value 1 of but it can be increased to skip over  peri-
                  ods of silence that are wanted.  For example, if you have a song
                  with 2 seconds of silence in the middle and 2 second at the end,
                  you  could  set  below-period  to  a value of 2 to skip over the
                  silence in the middle of the audio.
                  For below-periods, duration specifies a period of  silence  that
                  must exist before audio is not copied any more.  By specifying a
                  higher duration, silence that is  wanted  can  be  left  in  the
                  audio.   For example, if you have a song with an expected 1 sec-
                  ond of silence in the middle and 2 seconds  of  silence  at  the
                  end, a duration of 2 seconds could be used to skip over the mid-
                  dle silence.
                  Unfortunately, you must know the length of the  silence  at  the
                  end  of  your  audio  file to trim off silence reliably.  A work
                  around is to use the silence  effect  in  combination  with  the
                  reverse  effect.   By first reversing the audio, you can use the
                  above-periods to reliably trim all audio from  what  looks  like
                  the  front of the file.  Then reverse the file again to get back
                  to normal.
                  To remove silence from the middle of a file,  specify  a  below-
                  periods that is negative.  This value is then treated as a posi-
                  tive value and is  also  used  to  indicate  the  effect  should
                  restart  processing as specified by the above-periods, making it
                  suitable for removing periods of silence in the  middle  of  the
                  The  option  -l  indicates that below-periods duration length of
                  audio should be left intact at the beginning of each  period  of
                  silence.  For example, if you want to remove long pauses between
                  words but do not want to remove the pauses completely.
                  The period counts are in units of samples. Duration  counts  may
                  be  in  the  format of hh:mm:ss.frac, or the exact count of sam-
                  ples.  Threshold numbers may be suffixed with d to indicate  the
                  value  is  in decibels, or % to indicate a percentage of maximum
                  ters  give  the frequencies of the 6dB points of a high-pass and
                  low-pass filter that may be invoked individually,  or  together.
                  If both are given, then freqHP < freqLP creates a band-pass fil-
                  ter, freqHP > freqLP creates a band-reject filter.
                  The default stop-band attenuation of  120dB  can  be  overridden
                  with  -a;  alternatively, the kaiser-window 'beta' parameter can
                  be given directly with -b.
                  The default transition band-width of 5% of the total band can be
                  overridden with -t (and tbw in Hertz); alternatively, the number
                  of filter taps can be given directly with -n.
                  If both freqHP and freqLP are given, then  a  -t  or  -n  option
                  given  to  the  left of the frequencies applies to both frequen-
                  cies; one of these options given to the right of the frequencies
                  applies only to freqLP.
                  The  -p,  -M,  -I,  and  -L  options  control the filter's phase
                  response; see the rate effect for details.
                  This effect supports the --plot global option.
           spectrogram [options]
                  Create a spectrogram of the audio; the audio is  passed  unmodi-
                  fied  through the SoX processing chain.  This effect is optional
                  - type sox --help and check the list of supported effects to see
                  if it has been included.
                  The  spectrogram is rendered in a Portable Network Graphic (PNG)
                  file, and shows time in the X-axis, frequency in the Y-axis, and
                  audio  signal magnitude in the Z-axis.  Z-axis values are repre-
                  sented by the colour (or optionally the intensity) of the pixels
                  in  the  X-Y plane.  If the audio signal contains multiple chan-
                  nels then these are shown from top to bottom starting from chan-
                  nel 1 (which is the left channel for stereo audio).
                  For example, if 'my.wav' is a stereo file, then with
                     sox my.wav -n spectrogram
                  a  spectrogram  of  the  entire file will be created in the file
                  'spectrogram.png'.  More often though,  analysis  of  a  smaller
                  portion of the audio is required; e.g. with
                     sox my.wav -n remix 2 trim 20 30 spectrogram
                  the  spectrogram  shows information only from the second (right)
                  channel, and of thirty seconds of  audio  starting  from  twenty
                  seconds in.  To analyse a small portion of the frequency domain,
                  the rate effect may be used, e.g.
                     sox -n -n synth 6 tri 10k:14k spectrogram -z 100 -w kaiser
                  an analysis 'window' with high dynamic range is selected to best
                  display the spectrogram of a swept triangular wave.  For a  smi-
                  lar  example, append the following to the 'chime' command in the
                  description of the delay effect (above):
                     rate 2k spectrogram -X 200 -Z -10 -w kaiser
                  Options are also avaliable to control  the  appearance  (colour-
                  set,  brightness,  contrast,  etc.) and filename of the spectro-
                  gram; e.g. with
                     sox my.wav -n spectrogram -m -l -o print.png
                  a spectrogram is created suitable for printing on a  'black  and
                  white' printer.
                  -x num Change  the  (maximum)  width (X-axis) of the spectrogram
                         from its default value of 800 pixels to  a  given  number
                         between 100 and 5000.  See also -X and -d.
                  -X num X-axis  pixels/second;  the default is auto-calculated to
                         fit the given or known audio duration to the X-axis size,
                         or  100 otherwise.  If given in conjunction with -d, this
                         option affects the width of the  spectrogram;  otherwise,
                         it  affects  the duration of the spectrogram.  num can be
                         from 1 (low time resolution) to 5000 (high  time  resolu-
                         tion)  and need not be an integer.  SoX may make a slight
                         adjustment to the given number for  processing  quantisa-
                         tion  reasons;  if  so, SoX will report the actual number
                         used (viewable when  the  SoX  global  option  -V  is  in
                         effect).  See also -x and -d.
                  -y num Sets the Y-axis size in pixels (per channel); this is the
                         number of frequency 'bins' used in the  Fourier  analysis
                         that  produces  the  spectrogram.  N.B. it can be slow to
                         produce the spectrogram if this number is  not  one  more
                         than  a  power  of two (e.g. 129).  By default the Y-axis
                         size is chosen automatically (depending on the number  of
                         channels).   See  -Y for alternative way of setting spec-
                         trogram height.
                  -Y num Sets the target total height of the spectrogram(s).   The
                         default  value  is 550 pixels.  Using this option (and by
                         default), SoX will choose a height for  individual  spec-
                         trogram channels that is one more than a power of two, so
                         the actual total height may fall short of the given  num-
                         ber.  However, there is also a minimum height per channel
                  -q num Sets the Z-axis quantisation, i.e. the number of  differ-
                         ent  colours  (or  intensities) in which to render Z-axis
                         values.   A  small  number   (e.g.   4)   will   give   a
                         'poster'-like  effect  making it easier to discern magni-
                         tude bands of similar level.  Small numbers also  usually
                         result  in  small  PNG files.  The number given specifies
                         the number of colours to use inside the Z-axis range; two
                         colours are reserved to represent out-of-range values.
                  -w name
                         Window: Hann (default), Hamming, Bartlett, Rectangular or
                         Kaiser.  The spectrogram is produced using  the  Discrete
                         Fourier Transform (DFT) algorithm.  A significant parame-
                         ter to this algorithm is the choice of 'window function'.
                         By  default, SoX uses the Hann window which has good all-
                         round frequency-resolution and dynamic-range  properties.
                         For  better  frequency  resolution  (but  lower  dynamic-
                         range), select a Hamming window; for higher dynamic-range
                         (but  poorer  frequency-resolution), select a Kaiser win-
                         dow.  Bartlett and Rectangular windows  are  also  avail-
                  -W num Window  adjustment  parameter.   This can be used to make
                         small adjustments to the Kaiser window shape.  A positive
                         number  (up  to ten) increases its dynamic range, a nega-
                         tive number decreases it.
                  -s     Allow slack overlapping of DFT  windows.   This  can,  in
                         some  cases,  increase  image  sharpness and give greater
                         adherence to the -x value, but at the expense of a little
                         spectral loss.
                  -m     Creates a monochrome spectrogram (the default is colour).
                  -h     Selects a high-colour palette -  less  visually  pleasing
                         than  the default colour palette, but it may make it eas-
                         ier to differentiate different levels.  If this option is
                         used  in conjunction with -m, the result will be a hybrid
                         monochrome/colour palette.
                  -p num Permute the colours in a colour or hybrid  palette.   The
                         num  parameter,  from  1  (the default) to 6, selects the
                  -l     Creates a 'printer friendly'  spectrogram  with  a  light
                         background (the default has a dark background).
                  -a     Suppress  the  display  of the axis lines.  This is some-
                         times useful in helping to discern artefacts at the spec-
                         trogram edges.
                  -r     Raw  spectrogram:  suppress  the display of axes and leg-
                         Set (or clear) the image comment - text to display  below
                         and to the left of the spectrogram.
                  -o text
                         Name  of  the spectrogram output PNG file, default 'spec-
                  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
                  -S time
                         Start the spectrogram at the given  point  in  the  audio
                         stream.  For example
                            sox input.aiff output.wav spectrogram -S 1:00
                         creates a spectrogram showing all but the first minute of
                         the audio (the output file however, receives  the  entire
                         audio stream).
                  For the ability to perform off-line processing of spectral data,
                  see the stat effect.
           speed factor[c]
                  Adjust the audio speed (pitch and tempo  together).   factor  is
                  either the ratio of the new speed to the old speed: greater than
                  1 speeds up, less than 1 slows down, or, if  appended  with  the
                  letter  'c',  the number of cents (i.e. 100ths of a semitone) by
                  which the pitch (and tempo) should be adjusted: greater  than  0
                  increases, less than 0 decreases.
                  By default, the speed change is performed by resampling with the
                  rate effect using its default quality/speed.  For higher quality
                  or  higher  speed  resampling,  in addition to the speed effect,
                  specify the rate effect with the desired quality option.
                          t     correlated     constant gain    abrupt
                          h     correlated     constant gain    smooth
                          q     uncorrelated   constant power   smooth
                  To  perform  a  splice,  first use the trim effect to select the
                  audio sections to be joined together.  As when performing a tape
                  splice,  the  end  of  the  section to be spliced onto should be
                  trimmed with a small excess (default  0.005  seconds)  of  audio
                  after  the ideal joining point.  The beginning of the audio sec-
                  tion to splice on should be trimmed with the same excess (before
                  the  ideal  joining  point),  plus an additional leeway (default
                  0.005 seconds).  SoX should then be invoked with the  two  audio
                  sections  as  input  files  and the splice effect given with the
                  position at which to perform the splice - this is length of  the
                  first audio section (including the excess).
                  For  example, a long song begins with two verses which start (as
                  determined e.g. by using the play command with the trim  (start)
                  effect)  at times 0:30.125 and 1:03.432.  The following commands
                  cut out the first verse:
                     sox too-long.wav part1.wav trim 0 30.130
                  (5 ms excess, after the first verse starts)
                     sox too-long.wav part2.wav trim 1:03.422
                  (5 ms excess plus 5 ms leeway, before the second verse starts)
                     sox part1.wav part2.wav just-right.wav splice 30.130
                  For another example, the SoX command
                     play "|sox -n -p synth 1 sin %1" "|sox -n -p synth 1 sin %3"
                  generates and plays two notes, but there is a nasty click at the
                  transition; the click can be removed by splicing instead of con-
                  catenating the audio, i.e. by appending splice 1 to the command.
                  (Clicks  at the beginning and end of the audio can be removed by
                  preceding the splice effect with fade q .01 2 .01).
                  Provided your arithmetic is good enough, multiple splices can be
                  performed with a single splice invocation.  For example:
                  # 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 \
                  cally be an number of seconds, the -q option would typically  be
                  given (to select an 'equal power' cross-fade), and leeway should
                  be zero (which is the default if -q is given).  For example,  if
                  f1.wav and f2.wav are audio files to be cross-faded, then
                     sox f1.wav f2.wav out.wav splice -q $(soxi -D f1.wav),3
                  cross-fades  the  files  where  the point of equal loudness is 3
                  seconds before the end of f1.wav, i.e. the total length  of  the
                  cross-fade  is  2  x 3 = 6 seconds (Note: the $(...) notation is
                  POSIX shell).
           stat [-s scale] [-rms] [-freq] [-v] [-d]
                  Display time and frequency domain statistical information  about
                  the  audio.  Audio is passed unmodified through the SoX process-
                  ing chain.
                  The information is  output  to  the  'standard  error'  (stderr)
                  stream  and  is calculated, where n is the duration of the audio
                  in samples, c is the number of audio channels, r  is  the  audio
                  sample rate, and xk represents the PCM value (in the range -1 to
                  +1 by default) of each successive sample in the audio,  as  fol-
                   Samples read        nxc
                   Length (seconds)    n?r
                   Scaled by                                 See -s below.
                   Maximum amplitude   max(xk)               The  maximum sample
                                                             value in the audio;
                                                             usually  this  will
                                                             be a positive  num-
                   Minimum amplitude   min(xk)               The  minimum sample
                                                             value in the audio;
                                                             usually  this  will
                                                             be a negative  num-
                   Midline amplitude    1/2 min(xk)+ 1/2 max(xk)
                   Mean norm           ?/n?|xk|              The  average of the
                                                             absolute  value  of
                                                             each  sample in the
                   Mean amplitude      ?/n?xk                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
                                                             as possible without
                                                             clipping.     Note:
                                                             See  the discussion
                                                             on  Clipping  above
                                                             for  reasons why it
                                                             is  rarely  a  good
                                                             idea actually to do
                  Note that the delta measurements are not applicable  for  multi-
                  channel audio.
                  The  -s  option  can  be used to scale the input data by a given
                  factor.  The default value of scale is 2147483647 (i.e. the max-
                  imum value of a 32-bit signed integer).  Internal effects always
                  work with signed long PCM data and so the value should relate to
                  this fact.
                  The  -rms option will convert all output average values to 'root
                  mean square' format.
                  The -v option displays only the 'Volume Adjustment' value.
                  The -freq option calculates the  input's  power  spectrum  (4096
                  point  DFT) instead of the statistics listed above.  This should
                  only be used with a single channel audio file.
                  The -d option displays a hex dump of the 32-bit signed PCM  data
                  audio  in  SoX's  internal  buffer.  This is mainly used to help
                  track down endian problems that sometimes occur  in  cross-plat-
                  form versions of SoX.
                  See also the stats effect.
           stats [-b bits|-x bits|-s scale] [-w window-time]
                  Display  time  domain  statistical  information  about the audio
                  channels; audio is passed unmodified through the SoX  processing
                  chain.   Statistics  are calculated and displayed for each audio
                  channel and, where applicable, an overall figure is also  given.
                  For example, for a typical well-mastered stereo music file:
                                           Overall     Left      Right
                              DC offset   0.000803 -0.000391  0.000803
                              Min level  -0.750977 -0.750977 -0.653412
                              Max level   0.708801  0.708801  0.653534
                              Pk lev dB      -2.49     -2.49     -3.69
                              RMS lev dB    -19.41    -19.13    -19.71
                              RMS Pk dB     -13.82    -13.82    -14.38
                              RMS Tr dB     -85.25    -85.25    -82.66
                              Crest factor       -      6.79      6.32
                              Flat factor     0.00      0.00      0.00
                  floating-point number.
                  Pk lev dB and RMS lev dB are standard peak and  RMS  level  mea-
                  sured in dBFS.  RMS Pk dB and RMS Tr dB are peak and trough val-
                  ues for RMS level measured over a short window (default 50ms).
                  Crest factor is the standard ratio of peak to RMS  level  (note:
                  not in dB).
                  Flat factor  is a measure of the flatness (i.e. consecutive sam-
                  ples with the same value) of the signal at its peak levels (i.e.
                  either  Min level,  or  Max level).   Pk count  is the number of
                  occasions (not the number of samples) that the  signal  attained
                  either Min level, or Max level.
                  The  right-hand  Bit-depth  figure is the standard definition of
                  bit-depth i.e. bits less significant than the given  number  are
                  fixed  at zero.  The left-hand figure is the number of most sig-
                  nificant bits that are fixed at zero (or one for  negative  num-
                  bers)  subtracted  from  the  right-hand figure (the number sub-
                  tracted is directly related to Pk lev dB).
                  For multi-channel audio, an overall figure for each of the above
                  measurements  is  given  and derived from the channel figures as
                  follows: DC offset:  maximum  magnitude;  Max level,  Pk lev dB,
                  RMS Pk dB,  Bit-depth:  maximum;  Min level, RMS Tr dB: minimum;
                  RMS lev dB, Flat factor, Pk count:  average;  Crest factor:  not
                  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 measure-
                  ments; specifically, it is the maximum value that could apply to
                  Max level.   Window s  is  the length of the window used for the
                  peak and trough RMS measurements.
                  See also the stat effect.
           swap   Swap stereo channels.  See also remix for an effect that  allows
                  arbitrary channel selection and ordering (and mixing).
           stretch factor [window fade shift fading]
                  Change  the  audio duration (but not its pitch).  This effect is
                  broadly equivalent to the tempo  effect  with  (factor  inverted
                  and) search set to zero, so in general, its results are compara-
                  tively poor; it is retained  as  it  can  sometimes  out-perform
                  tempo for small factors.
                  factor  of stretching: >1 lengthen, <1 shorten duration.  window
                  size is in ms.  Default is 20ms.  The fade option, can be 'lin'.
                  shift  ratio, in [0 1].  Default depends on stretch factor. 1 to
                  shorten, 0.8 to lengthen.  The fading ratio, in  [0  0.5].   The
                  Though this effect is used to generate audio, an input file must
                  still be given, the characteristics of which will be used to set
                  the synthesised audio length, the number of  channels,  and  the
                  sampling rate; however, since the input file's audio is not nor-
                  mally needed, a 'null file' (with the special name -n) is  often
                  given  instead (and the length specified as a parameter to synth
                  or by another given effect that can has an associated length).
                  For example, the following produces a  3  second,  48kHz,  audio
                  file containing a sine-wave swept from 300 to 3300 Hz:
                     sox -n output.wav synth 3 sine 300-3300
                  and this produces an 8 kHz version:
                     sox -r 8000 -n output.wav synth 3 sine 300-3300
                  Multiple  channels  can  be synthesised by specifying the set of
                  parameters shown between braces multiple  times;  the  following
                  puts  the  swept tone in the left channel and adds 'brown' noise
                  in the right:
                     sox -n output.wav synth 3 sine 300-3300 brownnoise
                  The following example shows how two synth effects  can  be  cas-
                  caded to create a more complex waveform:
                     play -n synth 0.5 sine 200-500 synth 0.5 sine fmod 700-100
                  Frequencies can also be given in 'scientific' note notation, or,
                  by prefixing a '%' character, as a number of semitones  relative
                  to  'middle  A'  (440 Hz).   For example, the following could be
                  used to help tune a guitar's low 'E' string:
                     play -n synth 4 pluck %-29
                  or with a (Bourne shell) loop, the whole guitar:
                     for n in E2 A2 D3 G3 B3 E4; do
                       play -n synth 4 pluck $n repeat 2; done
                  See the delay effect (above) and the reference to 'SoX scripting
                  examples' (below) for more synth examples.
                  N.B.   This  effect  generates  audio at maximum volume (0dBFS),
                  which means that there is a high chance of clipping  when  using
                  the  audio subsequently, so in many cases, you will want to fol-
                  low this effect with the gain effect to prevent this  from  hap-
                  pening.  (See  also Clipping above.)  Note that, by default, the
                  synth effect incorporates the functionality of gain -h (see  the
                  gain effect for details); synth's -n option may be given to dis-
                  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
                  /      Exponential:  the  tone  will change by a fixed number of
                         semitones per second.
                  -      Exponential: as '/', but initial phase always  zero,  and
                         stepped (less smooth) frequency changes.
                  Not used for noise.
                  off is the bias (DC-offset) of the signal in percent; default=0.
                  ph is the phase shift in percentage of 1 cycle; default=0.   Not
                  used for noise.
                  p1  is  the  percentage  of each cycle that is 'on' (square), or
                  'rising' (triangle, exp, trapezium); default=50 (square,  trian-
                  gle,   exp),   default=10   (trapezium),   or  sustain  (pluck);
                  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
                  The  -s  option  is  used to optimize default values of segment,
                  search and overlap for speech processing.
                  The -l option is used to optimize  default  values  of  segment,
                  search  and  overlap for 'linear' processing that tends to cause
                  more noticeable distortion but may  be  useful  when  factor  is
                  close to 1.
                  If -m, -s, or -l is specified, the default value of segment will
                  be calculated based on factor, while default search and  overlap
                  values  are based on segment. Any values you provide still over-
                  ride these default values.
                  factor gives the ratio of new tempo to the old  tempo,  so  e.g.
                  1.1 speeds up the tempo by 10%, and 0.9 slows it down by 10%.
                  The  optional  segment parameter selects the algorithm's segment
                  size in milliseconds.  If no  other  flags  are  specified,  the
                  default  value  is  82  and  is typically suited to making small
                  changes to the tempo of music. For larger changes (e.g. a factor
                  of 2), 41 ms may give a better result.  The -m, -s, and -l flags
                  will cause the segment  default  to  be  automatically  adjusted
                  based on factor.  For example using -s (for speech) with a tempo
                  of 1.25 will calculate a default segment value of 32.
                  The optional search parameter gives the  audio  length  in  mil-
                  liseconds  over  which the algorithm will search for overlapping
                  points.  If no other flags are specified, the default  value  is
                  14.68.   Larger  values  use more processing time and may or may
                  not produce better results.  A practical  maximum  is  half  the
                  value  of  segment. Search can be reduced to cut processing time
                  at the risk of degrading output quality.  The  -m,  -s,  and  -l
                  flags will cause the search default to be automatically adjusted
                  based on segment.
                  The optional overlap parameter gives the segment overlap  length
                  in  milliseconds.   Default value is 12, but -m, -s, or -l flags
                  automatically adjust overlap based on segment  size.  Increasing
                  overlap  increases  processing  time and may increase quality. A
                  practical maximum for overlap is the value of search, with over-
                  lap typically being (at least) a little smaller then search.
                  See  also  speed  for  an  effect  that  changes tempo and pitch
                  together, pitch for an  effect  that  changes  tempo  and  pitch
                  together,  and  stretch for an effect that changes tempo using a
                  different algorithm.
           treble gain [frequency[k] [width[s|h|k|o|q]]]
                  Apply a treble tone-control effect.  See the description of  the
                  bass effect for details.
                  for the start parameter will allow trimming off the end only.
                  Both  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  start value of 1:30.5 will not
                  start until 1 minute, thirty and  1/2  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
                  start  parameter  will  wait  until 8000 samples are read before
                  starting to process audio.
           vad [options]
                  Voice Activity Detector.  Attempts to  trim  silence  and  quiet
                  background  sounds from the ends of (fairly high resolution i.e.
                  16-bit, 44-48kHz) recordings of speech.  The algorithm currently
                  uses a simple cepstral power measurement to detect voice, so may
                  be fooled by other things, especially  music.   The  effect  can
                  trim  only from the front of the audio, so in order to trim from
                  the back, the reverse effect must also be used.  E.g.
                     play speech.wav norm vad
                  to trim from the front,
                     play speech.wav norm reverse vad reverse
                  to trim from the back, and
                     play speech.wav norm vad reverse vad reverse
                  to trim from both ends.  The use of the norm  effect  is  recom-
                  mended,  but  remember that neither reverse nor norm is suitable
                  for use with streamed audio.
                  Default values are shown in parenthesis.
                  -t num (7)
                         The measurement level used to trigger activity detection.
                         This  might  need  to  be  changed depending on the noise
                         level, signal level and other charactistics of the  input
                  -T num (0.25)
                         The  time constant (in seconds) used to help ignore short
                         bursts of sound.
                  -s num (1)
                         The amount of audio  (in  seconds)  to  search  for  qui-
                         eter/shorter  bursts  of  audio  to  include prior to the
                         detected trigger point.
                  -N num Time constant used by the adaptive  noise  estimator  for
                         when the noise level is increasing.
                  -n num Time  constant  used  by the adaptive noise estimator for
                         when the noise level is decreasing.
                  -r num Amount of noise reduction to use in the  detection  algo-
                         rithm (e.g. 0, 0.5, ...).
                  -f num Frequency of the algorithm's processing/measurements.
                  -m num Measurement  duration;  by default, twice the measurement
                         period; i.e.  with overlap.
                  -M num Time constant used to smooth spectral measurements.
                  -h num 'Brick-wall' frequency of high-pass filter applied at the
                         input to the detector algorithm.
                  -l num 'Brick-wall'  frequency of low-pass filter applied at the
                         input to the detector algorithm.
                  -H num 'Brick-wall' quefrency of high-pass lifter  used  in  the
                         detector algorithm.
                  -L num 'Brick-wall'  quefrency  of  low-pass  lifter used in the
                         detector algorithm.
                  See also the silence effect.
           vol gain [type [limitergain]]
                  Apply an amplification or an attenuation to  the  audio  signal.
                  Unlike the -v option (which is used for balancing multiple input
                  files as they enter the SoX effects processing chain), vol is an
                  effect  like  any  other so can be applied anywhere, and several
                  times if necessary, during the processing chain.
                  The amount to change the volume is given by gain which is inter-
                  preted,  according  to  the  given  type, as follows: if type is
                  amplitude (or is omitted), then gain is an amplitude (i.e. volt-
                  age  or  linear)  ratio, if power, then a power (i.e. wattage or
                  voltage-squared) ratio, and if dB, then a power change in dB.
                  When type is amplitude or power, a gain of 1 leaves  the  volume
                  unchanged,  less  than  1  decreases  it,  and  greater  than  1
                  increases it; a negative gain inverts the audio signal in  addi-
                  tion to adjusting its volume.
                  When  type  is dB, a gain of 0 leaves the volume unchanged, less
                  than 0 decreases it, and greater than 0 increases it.
                  See also gain for a volume-changing effect with different  capa-
                  bilities,  and  compand  for  a dynamic-range compression/expan-
                  sion/limiting effect.
       Deprecated Effects
           The following effects have been renamed  or  have  their  functionality
           included  in  another  effect; they continue to work in this version of
           SoX but may be removed in future.
           filter [low]-[high] [window-len [beta]]
                  Apply a sinc-windowed lowpass, highpass, or bandpass  filter  of
                  given  window length to the signal.  This effect has been super-
                  seded by the sinc effect.  Compared  with  'sinc',  'filter'  is
                  slower and has fewer capabilities.
                  low  refers to the frequency of the lower 6dB corner of the fil-
                  ter.  high refers to the frequency of the upper  6dB  corner  of
                  the filter.
                  A  low-pass filter is obtained by leaving low unspecified, or 0.
                  A high-pass filter is obtained by leaving high  unspecified,  or
                  0, or greater than or equal to the Nyquist frequency.
                  The window-len, if unspecified, defaults to 128.  Longer windows
                  give a sharper cut-off, smaller windows a more gradual  cut-off.
                  The  beta  parameter  determines the type of filter window used.
                  Any value greater than 2 is the beta for a Kaiser window.   Beta
                  <=  2  selects  a  Blackman-Nuttall  window.  If unspecified, the
                  default is a Kaiser window with beta 16.
                  In the case of Kaiser window (beta > 2), lower betas  produce  a
                  somewhat  faster  transition from pass-band to stop-band, at the
                  cost of noticeable artifacts. A beta of 16 is the default,  beta
                  less  than 10 is not recommended. If you want a sharper cut-off,
                  don't use low beta's, use a longer sample  window.  A  Blackman-
                  Nuttall window is selected by specifying any 'beta' <= 2, and the
                  Blackman-Nuttall window has somewhat steeper  cut-off  than  the
                  default  Kaiser  window.  You  will probably not need to use the
                  beta parameter at all, unless you are just curious about compar-
                  ing the effects of Blackman-Nuttall vs. Kaiser windows.
                  This effect supports the --plot global option.
           key [-q] shift [segment [search [overlap]]]
                  Change  the  audio key (i.e. pitch but not tempo).  This is just
                  an alias for the pitch effect.
           pan direction
                  Mix the audio from one channel to another.  Use mixer  or  remix
                  instead of this effect.


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


           soxi(1), soxformat(7), libsox(3)
           audacity(1), gnuplot(1), octave(1), wget(1)
           The SoX web site at
           SoX scripting examples at
           [1]    R. Bristow-Johnson, Cookbook formulae for audio EQ biquad filter
           [2]    Wikipedia, Q-factor,
           [3]    Scott    Lehman,    Effects    Explained,    http://harmony-cen-
           [4]    Wikipedia, Decibel,
           [5]    Richard  Furse,  Linux  Audio  Developer's  Simple  Plugin  API,
           [6]    Richard Furse, Computer Music Toolkit,
           [7]    Steve Harris, LADSPA plugins,


           Copyright 1998-2011 Chris Bagwell and SoX Contributors.
           Copyright 1991 Lance Norskog and Sundry Contributors.
           This program is free software; you can redistribute it and/or modify it
           under the terms of the GNU General Public License as published  by  the
           Free  Software  Foundation;  either  version 2, or (at your option) any
           later version.
           This program is distributed in the hope that it  will  be  useful,  but
           WITHOUT  ANY  WARRANTY;  without  even  the  implied  warranty  of MER-
           Public License for more details.


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

    sox February 19, 2011 SoX(1)


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