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SoX(1)				Sound eXchange				SoX(1)

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

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

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

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

       SoX  reads  and	writes	audio  files  in  most popular formats and can
       optionally apply	 effects  to  them.  It	 can  combine  multiple	 input
       sources,	 synthesise audio, and, on many systems, act as a general 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,
	  sox -b 16 recital.wav channels 1 rate 16k fade 3 norm
       performs the same format translation, but  also	applies	 four  effects
       (down-mix  to  one channel, sample rate change, fade-in, nomalize), and
       stores the result at a bit-depth of 16.
	  sox -r 16k -e signed -b 8 -c 1 voice-memo.raw voice-memo.wav
       converts `raw' (a.k.a. `headerless') audio to  a	 self-describing  file
	  sox slow.aiff fixed.aiff speed 1.027
       adjusts audio speed,
	  sox short.wav long.wav longer.wav
       concatenates two audio files, and
	  sox -m music.mp3 voice.wav mixed.flac
       mixes together two audio files.
	  play "The Moonbeams/Greatest/*.ogg" bass +3
       plays  a	 collection  of	 audio	files  whilst applying a bass boosting
	  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:

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

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

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

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

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

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

       One  important  use  of	audio file comments is to convey `Replay Gain'
       information.  SoX supports applying Replay Gain	information,  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.

       3.  The filename extension.

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

       1.  Command-line format options.

       2.  The filename extension.

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

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

   Playing & Recording Audio
       The  play  and  rec  commands  are  provided  so that basic playing and
       recording is as simple as
	  play existing-file.wav
	  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.

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

				 *	  *	   *

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

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

       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.

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

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

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

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

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

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

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

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

       Clipping is distortion that occurs when an audio signal level (or `vol‐
       ume') exceeds the range of the chosen representation.  In  most	cases,
       clipping	 is  undesirable  and  so should be corrected by adjusting the
       level prior to the point (in the processing chain) at which it occurs.

       In SoX, clipping could occur, as you might expect, when using  the  vol
       or gain effects to increase the audio volume. Clipping could also occur
       with many other effects, when converting one  format  to	 another,  and
       even when simply playing the audio.

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

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

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

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

       See also -G and the gain and norm effects.

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

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

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

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

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

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

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

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

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

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

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

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

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

       With the `mix-power' combine method, the mixed volume is	 approximately
       equal to that of one of the input signals.  This is achieved by balanc‐
       ing using a factor of ¹/√n instead of ¹/n.  Note	 that  this  balancing
       factor  does not guarantee that clipping will not occur, but the number
       of clips will usually be low and the resultant distortion is  generally

   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 custom‐
       ized  by	 placing a %n anywhere in the filename where the number should
       be substituted.	An optional number can be placed after the % to	 indi‐
       cate a minimum fixed width for the number.

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

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

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

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

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

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

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

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

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

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

	      Specifies	 that  filename `globbing' (wild-card matching) should
	      be performed by SoX instead of by the shell.  This allows a 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:
		 play "|sox -n -p synth 2" "|sox -n -p synth 2 tremolo 10" stat
	      plays two `files' in succession, each with different effects.

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

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

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

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

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

	      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:

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

       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 be‐

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

	      See Input File Combining above for a description of the  differ‐
	      ent combining methods.

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

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

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

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

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

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

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

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

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

       --multi-threaded | --single-threaded
	      By default, SoX is `single threaded'.  If	 the  --multi-threaded
	      option is given however then SoX will process audio channels for
	      most multi-channel effects in parallel on hyper-threading/multi-
	      core  architectures.  This  may  reduce  processing time, though
	      sometimes it may be necessary to use this option	in  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.

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

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

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

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

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

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

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

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

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

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

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

       -T     Equivalent to --combine multiply.

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

	      Show SoX's version number and exit.

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

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

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

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

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

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

	      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
       that is different to that of the input file.

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

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

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

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

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

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

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

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

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

		     OKI  (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
		 sox input.cdda -e float output1.wav

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

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

	      Deprecated  aliases  for	specifying  the encoding types signed-
	      integer, unsigned-integer, floating-point, a-law,	 mu-law,  oki-
	      adpcm,  ima-adpcm,  ms-adpcm, gsm-full-rate respectively (see -e

	      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 -b 24
		 sox input.wav	      output.wav bass -b 24 rate 48k
	      though the second form  is  more	flexible  as  it  allows  rate
	      options  to be given, and allows the effects to be ordered arbi‐

       -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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

	      See frequency.

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

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

	      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.

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

	      These effects support the --plot global option.

	      See also sinc for a bandpass filter with steeper shoulders.

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

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

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

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

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

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

	      The filters are described in detail in [1].

	      These effects support the --plot global option.

	      See also equalizer for a peaking equalisation effect.

       bend [-f frame-rate(25)] [-o over-sample(16)] { 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 \
		   bend .35,180,.25  .15,740,.53  0,-520,.3
	      Note that the clipping that  is  produced	 in  this  example  is
	      deliberate; to remove it, use gain -5 in place of gain 1.

	      See also pitch.

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

	      See (where a0
	      = 1).

	      This effect supports the --plot global option.

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

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

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

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

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

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

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

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

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

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

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

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

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

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

				    *	     *	      *

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

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

	      See also mcompand for a multiple-band companding effect.

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

	      See also the compand and mcompand effects.

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

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

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

				    *	     *	      *

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

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

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

	      With the deemph effect, it is possible to	 apply	the  necessary
	      de-emphasis  to  audio that has been extracted from a pre-empha‐
	      sised CD, and then either burn the de-emphasised audio to a  new
	      CD  (which will then play correctly on any CD player), or simply
	      play the correctly de-emphasised audio files  on	the  PC.   For
		 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 [-S|-s|-f filter] [-a] [-p precision]
	      Apply dithering to the audio.   Dithering	 deliberately  adds  a
	      small  amount  of	 noise	to the signal in order to mask audible
	      quantization effects that can occur if the output sample size is
	      less than 24 bits.  With no options, this effect will add trian‐
	      gular (TPDF) white noise.	 Noise-shaping (only for certain  sam‐
	      ple  rates)  can be selected with -s.  With the -f option, it is
	      possible to select a particular noise-shaping  filter  from  the
	      following	  list:	  lipshitz,  f-weighted,  modified-e-weighted,
	      improved-e-weighted, gesemann, shibata,  low-shibata,  high-shi‐
	      bata.   Note  that  most	filter	types  are available only with
	      44100Hz sample rate.  The filter types are distinguished by  the
	      following	 properties: audibility of noise, level of (inaudible,
	      but in some circumstances, otherwise  problematic)  shaped  high
	      frequency noise, and processing speed.
	      See	for  graphs of
	      the different noise-shaping curves.

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

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

	      The -p option allows overriding the target precision.

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

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

	      See also the `Dithering' section above.

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

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

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

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

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

	      Each  given delay decay pair gives the delay in milliseconds and
	      the decay (relative to gain-in) of that echo.  Gain-out  is  the
	      volume  of  the output.  For example: This will make it sound as
	      if there are twice as many instruments as are actually playing:
		 play lead.aiff echo 0.8 0.88 60 0.4
	      If the delay is very short, then it sound like a (metallic)  ro‐
	      bot playing music:
		 play lead.aiff echo 0.8 0.88 6 0.4
	      A	 longer delay will sound like an open air concert in the moun‐
		 play lead.aiff echo 0.8 0.9 1000 0.3
	      One mountain more, and:
		 play lead.aiff echo 0.8 0.9 1000 0.3 1800 0.25

       echos gain-in gain-out <delay decay>
	      Add a sequence of echoes to the audio.  Each  delay  decay  pair
	      gives the delay in milliseconds and the decay (relative to gain-
	      in) of that echo.	 Gain-out is the volume of the output.

	      Like the echo effect, echos stand for `ECHO in Sequel', that  is
	      the  first  echos	 takes the input, the second the input and the
	      first echos, the third the input and the first  and  the	second
	      echos,  ... and so on.  Care should be taken using many echos; a
	      single echos has the same effect as a single echo.

	      The sample will be bounced twice in symmetric echos:
		 play lead.aiff echos 0.8 0.7 700 0.25 700 0.3
	      The sample will be bounced twice in asymmetric echos:
		 play lead.aiff echos 0.8 0.7 700 0.25 900 0.3
	      The sample will sound as if played in a garage:
		 play lead.aiff echos 0.8 0.7 40 0.25 63 0.3

       equalizer frequency[k] width[q|o|h|k] gain
	      Apply a two-pole peaking equalisation (EQ)  filter.   With  this
	      filter,  the signal-level at and around a selected frequency can
	      be increased or decreased, whilst (unlike	 band-pass  and	 band-
	      reject filters) that at all other frequencies is unchanged.

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

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

	      The filter is described in detail in [1].

	      This effect supports the --plot global option.

	      See also bass and treble for shelving equalisation effects.

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

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

	      A	 fade-in  starts  from	the  first sample and ramps the signal
	      level from 0 to full volume over fade-in-length seconds.	 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.

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

	      See also the splice effect.

       fir [coefs-file|coefs]
	      Use  SoX's  FFT convolution engine with given FIR filter coeffi‐
	      cients.  If a single argument is given then this is  treated  as
	      the  name	 of  a file containing the filter coefficients (white-
	      space separated; may contain `#' comments).  If the given	 file‐
	      name  is	`-', or if no argument is given, then the coefficients
	      are read from the `standard input' (stdin);  otherwise,  coeffi‐
	      cients may be given on the command line.	Examples:
		 sox infile outfile fir 0.0195 -0.082 0.234 0.891 -0.145 0.043
		 sox infile outfile fir coefs.txt
	      with coefs.txt containing
		 # HP filter
		 # freq=10000

	      This effect supports the --plot global option.

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

	      All parameters are optional (right to left).

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

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

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

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

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

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

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

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

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

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

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

	      See also the norm and vol effects.

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

	      These effects support the --plot global option.

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

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

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

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

	      This effect supports the --plot global option.

       ladspa 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:
		 play track1.wav gain -3 sinc 8000- 29 100 mcompand \
		   "0.005,0.1 -47,-40,-34,-34,-17,-33" 100 \
		   "0.003,0.05 -47,-40,-34,-34,-17,-33" 400 \
		   "0.000625,0.0125 -47,-40,-34,-34,-15,-33" 1600 \
		   "0.0001,0.025 -47,-40,-34,-34,-31,-31,-0,-30" 6400 \
		   "0,0.025 -38,-31,-28,-28,-0,-25" \
		   gain 15 highpass 22 highpass 22 sinc -n 255 -b 16 -17500 \
		   gain 9 lowpass -1 17801
	      The audio file is played with a simulated	 FM  radio  sound  (or
	      broadcast	 signal	 condition if the lowpass filter at the end is
	      skipped).	 Note that the pipeline is set up with	US-style  75us

	      See also compand for a single-band companding effect.

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

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

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

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

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

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

       pad { length[@position] }
	      Pad the audio with silence, at the beginning, the	 end,  or  any
	      specified	 points	 through  the audio.  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

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

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

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

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

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

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

	      See also the bend, speed, and tempo effects.

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

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

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

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

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

				    *	     *	      *

	      Warning: technically detailed discussion follows.

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

	      -M/-I/-L	   Phase response = minimum/intermediate/linear
	      -s	   Steep filter (band-width = 99%)
	      -a	   Allow aliasing/imaging above the pass-band
	      -b 74-99.7   Any band-width %

	      -p 0-100	   Any phase response (0 = minimum, 25 = intermediate,
			   50 = linear, 100 = maximum)

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

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

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

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

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

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

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

		 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 and speed effects use the rate effect at their core.

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

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

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

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

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

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

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

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

	      The -m (manual) option disables  all  automatic  volume  adjust‐
	      ments, so
		 sox input.wav output.wav remix -m 1,2 3,4v0.8
	      results in channel level multipliers of 1,1 1,0.8.

	      The  volume number is optional and omitting it corresponds to no
	      volume change; however, the only case in which this is useful is
	      in  conjunction  with  i.	  For example, if input.wav is stereo,
		 sox input.wav output.wav remix 1,2i
	      is a mono equivalent of the oops effect.

	      If the -p option is given, then any  automatic  ¹/n  scaling  is
	      replaced	by ¹/√n (`power') scaling; this gives a louder mix but
	      one that might occasionally clip.

				    *	     *	      *

	      One use of the remix effect is to split an audio file into a set
	      of  files,  each	containing one of the constituent channels (in
	      order to perform subsequent processing on individual audio chan‐
	      nels).   Where  more  than a few channels are involved, a script
	      such as the following (Bourne shell script) is useful:
	      chans=`soxi -c "$1"`
	      while [ $chans -ge 1 ]; do
		 chans0=`printf %02i $chans`   # 2 digits hence up to 99 chans
		 out=`echo "$1"|sed "s/\(.*\)\.\(.*\)/\1-$chans0.\2/"`
		 sox "$1" "$out" remix $chans
		 chans=`expr $chans - 1`
	      If a file input.wav containing six audio	channels  were	given,
	      the   script  would  produce  six	 output	 files:	 input-01.wav,
	      input-02.wav, ..., input-06.wav.

	      See also the swap effect.

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

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

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

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

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

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

	      This effect supports the --plot global option.

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

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

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

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

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

	      When  optionally trimming silence from the end of the audio, you
	      specify a below-periods count.  In this case, below-period means
	      to  remove  all audio after silence is detected.	Normally, this
	      will be a value 1 of but it can be increased to skip over	 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
	      value of the sample value (0% specifies pure digital silence).

	      The following example shows how this effect can be used to start
	      a	 recording  that does not contain the delay at the start which
	      usually occurs between `pressing	the  record  button'  and  the
	      start of the performance:
		 rec parameters filename other-effects silence 1 5 2%

       sinc [-a att|-b beta] [-p phase|-M|-I|-L] [-t tbw|-n taps] [freqHP]
       [-freqLP [-t tbw|-n taps]]
	      Apply a sinc kaiser-windowed low-pass, high-pass, band-pass,  or
	      band-reject filter to the signal.	 The freqHP and freqLP parame‐
	      ters give the frequencies of the 6dB points of a	high-pass  and
	      low-pass	filter	that may be invoked individually, or together.
	      If both are given, then freqHP less than freqLP creates a	 band-
	      pass  filter,  freqHP  greater than freqLP creates a band-reject
	      filter.  For example, the invocations
		 sinc 3k
		 sinc -4k
		 sinc 3k-4k
		 sinc 4k-3k
	      create a high-pass, low-pass, band-pass, and band-reject	filter

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

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

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

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

	      This effect supports the --plot global option.

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

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

	      For example, if `my.wav' is a stereo file, then with
		 sox my.wav -n spectrogram
	      a spectrogram of the entire file will be	created	 in  the  file
	      `spectrogram.png'.   More	 often	though,	 analysis of a smaller
	      portion of the audio is required; e.g. with
		 sox my.wav -n remix 2 trim 20 30 spectrogram
	      the spectrogram shows information only from the  second  (right)
	      channel,	and  of	 thirty	 seconds of audio starting from twenty
	      seconds in.  To analyse a small portion of the frequency domain,
	      the rate effect may be used, e.g.
		 sox my.wav -n rate 6k spectrogram
	      allows  detailed	analysis  of  frequencies up to 3kHz (half the
	      sampling rate) i.e. where the human auditory system is most sen‐
	      sitive.  With
		 sox my.wav -n trim 0 10 spectrogram -x 600 -y 200 -z 100
	      the given options control the size of the spectrogram's X, Y & Z
	      axes (in this case, the spectrogram area of the  produced	 image
	      will  be	600 by 200 pixels in size and the Z-axis range will be
	      100 dB).	Note that the produced	image  includes	 axes  legends
	      etc.  and so will be a little larger than the specified spectro‐
	      gram size.  In this example:
		 sox -n -n synth 6 tri 10k:14k spectrogram -z 100 -w kaiser
	      an analysis `window' with high dynamic range is selected to best
	      display  the spectrogram of a swept triangular wave.  For a smi‐
	      lar example, append the following to the `chime' command in  the
	      description of the delay effect (above):
		 rate 2k spectrogram -X 200 -Z -10 -w kaiser
	      Options  are  also  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 200000.  See also -X and -d.

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

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

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

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

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

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

	      -w name
		     Window: Hann (default), Hamming, Bartlett, Rectangular 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‐

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

	      -t text
		     Set  the image title - text to display above the spectro‐

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

	      -o text
		     Name  of  the spectrogram output PNG file, default `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.

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

	      See also the bend, pitch, and tempo effects.

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

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

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

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

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

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

	      where * indicates the joining points.

	      For  example, a long song begins with two verses which start (as
	      determined e.g. by using the play command with the trim  (start)
	      effect)  at times 0:30.125 and 1:03.432.	The following commands
	      cut out the first verse:
		 sox too-long.wav part1.wav trim 0 30.130
	      (5 ms excess, after the first verse starts)
		 sox too-long.wav part2.wav trim 1:03.422
	      (5 ms excess plus 5 ms leeway, before the second verse starts)
		 sox part1.wav part2.wav just-right.wav splice 30.130
	      For another example, the SoX command
		 play "|sox -n -p synth 1 sin %1" "|sox -n -p synth 1 sin %3"
	      generates and plays two notes, but there is a nasty click at the
	      transition; the click can be removed by splicing instead of 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 \
		 `expr $3 - $2 + $e + $l + $e`s
	      sox "$1" part1.wav trim 0 `expr $4 + $e`s
	      sox "$1" part2.wav trim `expr $4 + $3 - $2 - $e - $l`s
	      sox part1.wav piece.wav part2.wav "$5" splice \
		 `expr $4 + $e`s \
		 `expr $4 + $e + $3 - $2 + $e + $l + $e`s
	      In  the above Bourne shell script, two splices are used to `copy
	      and paste' audio.

				    *	     *	      *

	      It is also possible to use this effect to perform general cross-
	      fades, e.g. to join two songs.  In this case, excess would typi‐
	      cally be an number of seconds, the -q option would typically  be
	      given (to select an `equal power' cross-fade), and leeway should
	      be zero (which is the default if -q is given).  For example,  if
	      f1.wav and f2.wav are audio files to be cross-faded, then
		 sox f1.wav f2.wav out.wav splice -q $(soxi -D f1.wav),3
	      cross-fades  the	files  where  the point of equal loudness is 3
	      seconds before the end of f1.wav, i.e. the total length  of  the
	      cross-fade  is  2	 × 3 = 6 seconds (Note: the $(...) notation is
	      POSIX shell).

       stat [-s scale] [-rms] [-freq] [-v] [-d]
	      Display time and frequency domain statistical information	 about
	      the  audio.  Audio is passed unmodified through the SoX process‐
	      ing chain.

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

	       Samples read	   n×c
	       Length (seconds)	   n÷r
	       Scaled by				 See -s below.
	       Maximum amplitude   max(xk)		 The  maximum sample
							 value in the audio;
							 usually  this	will
							 be a positive	num‐
	       Minimum amplitude   min(xk)		 The  minimum sample
							 value in the audio;
							 usually  this	will
							 be a negative	num‐
	       Midline amplitude   ½min(xk)max(xk)
	       Mean norm	   ¹/nΣ│xk│		 The  average of the
							 absolute  value  of
							 each  sample in the
	       Mean amplitude	   ¹/nΣxk		 The average of each
							 sample	   in	 the
							 audio.	   If	this
							 figure is non-zero,
							 then  it  indicates
							 the  presence	of a
							 D.C. offset  (which
							 could	 be  removed
							 using	the  dcshift
	       RMS amplitude	   √(¹/nΣxk²)		 The level of a D.C.
							 signal	 that  would
							 have the same power
							 as   the    audio's
							 average power.
	       Maximum delta	   max(│xk-xk-1│)
	       Minimum delta	   min(│xk-xk-1│)

	       Mean delta	   ¹/n-1Σ│xk-xk-1│
	       RMS delta	   √(¹/n-1Σ(xk-xk-1)²)
	       Rough frequency				 In Hz.
	       Volume Adjustment			 The   parameter  to
							 the   vol    effect
							 which	 would	make
							 the audio  as	loud
							 as possible without
							 clipping.     Note:
							 See  the discussion
							 on  Clipping  above
							 for  reasons why it
							 is  rarely  a	good
							 idea actually to do

	      Note that the delta measurements are not applicable  for	multi-
	      channel audio.

	      The  -s  option  can  be used to scale the input data by a given
	      factor.  The default value of scale is 2147483647 (i.e. the max‐
	      imum value of a 32-bit signed integer).  Internal effects always
	      work with signed long PCM data and so the value should relate to
	      this fact.

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

	      The -v option displays only the `Volume Adjustment' value.

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

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

	      See also the stats effect.

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

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

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

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

	      Pk lev dB	 and  RMS lev dB  are standard peak and RMS level mea‐
	      sured in dBFS.  RMS Pk dB and RMS Tr dB are peak and trough val‐
	      ues for RMS level measured over a short window (default 50ms).

	      Crest factor  is	the standard ratio of peak to RMS level (note:
	      not in dB).

	      Flat factor is a measure of the flatness (i.e. consecutive  sam‐
	      ples with the same value) of the signal at its peak levels (i.e.
	      either Min level, or Max level).	 Pk count  is  the  number  of
	      occasions	 (not  the number of samples) that the signal attained
	      either Min level, or Max level.

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

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

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

	      See also the stat effect.

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

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

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

	      See also the tempo effect.

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

	      Though this effect is used to generate audio, an input file must
	      still be given, the characteristics of which will be used to set
	      the synthesised audio length, the number of  channels,  and  the
	      sampling rate; however, since the input file's audio is not nor‐
	      mally needed, a `null file' (with the special name -n) is	 often
	      given  instead (and the length specified as a parameter to synth
	      or by another given effect that 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‐
	      able this behaviour.

	      A detailed description of each synth parameter follows:

	      len is the length of audio to synthesise expressed as a time  or
	      as a number of samples; 0=inputlength, default=0.

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

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

	      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
	      uses  the WSOLA algorithm. The audio is chopped up into segments
	      which are then shifted in the time domain and overlapped (cross-
	      faded)  at  points  where	 their	waveforms  are most similar as
	      determined by measurement of `least squares'.

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

       vad [options]
	      Voice  Activity  Detector.   Attempts  to trim silence and quiet
	      background sounds from the ends of (fairly high resolution  i.e.
	      16-bit, 44-48kHz) recordings of speech.  The algorithm currently
	      uses a simple cepstral power measurement to detect voice, so may
	      be  fooled  by  other  things, especially music.	The effect can
	      trim only from the front of the audio, so in order to trim  from
	      the back, the reverse effect must also be used.  E.g.
		 play speech.wav norm vad
	      to trim from the front,
		 play speech.wav norm reverse vad reverse
	      to trim from the back, and
		 play speech.wav norm vad reverse vad reverse
	      to  trim	from  both ends.  The use of the norm effect is 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.

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

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

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

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

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

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

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

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

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

	      -M num Time constant used to smooth spectral measurements.

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

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

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

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

	      See also the silence effect.

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

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

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

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

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

	      Beware of Clipping when the increasing the volume.

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

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

	      See also gain for a volume-changing effect with different	 capa‐
	      bilities,	 and  compand  for  a dynamic-range compression/expan‐
	      sion/limiting effect.

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

       mixer [ -l|-r|-f|-b|-1|-2|-3|-4|n{,n} ]
	      Reduce the number of audio channels by mixing or selecting 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

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

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

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

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

       This  program  is  distributed  in the hope that it will be useful, but
       WITHOUT ANY  WARRANTY;  without	even  the  implied  warranty  of  MER‐
       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 1, 2013				SoX(1)

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