SoX(1)				Sound eXchange				SoX(1)



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

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

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

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

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

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

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

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

				 *	  *	   *

       The overall SoX processing chain can be summarised as follows:

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

       Note however, that on the SoX command line, the positions of  the  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.au recital.wav

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

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

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

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

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

	  sox slow.aiff fixed.aiff speed 1.027

       adjusts audio speed,

	  sox short.wav long.wav longer.wav

       concatenates two audio files, and

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

       mixes together two audio files.

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

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

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

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

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

       records half an hour of stereo audio, and

	  rec -M take1.aiff take1-dub.aiff

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

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

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

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

   File Format Types
       There are two types of audio file format that SoX can work with: ‘self-
       describing’  -  these  (e.g.  WAV,  FLAC) have a header that completely
       describes the signal and encoding attributes of	the  audio  data  that
       follows,	 and ‘raw’ (or ‘headerless’) audio - the audio characteristics
       of these must, when reading a raw file, be described using the SoX com-
       mand  line, and, when writing a raw file, be set using the command line
       (or inferred from those of the input file).

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

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

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

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

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

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

       Most self-describing formats also allow textual ‘comments’ to be 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
       headers.

   Determining & Setting The File Format
       There  are  several mechanisms available for SoX to use to determine or
       set the format characteristics of an audio file.	 Depending on the 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 to them  that
	   is supported by the output file type.

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

   Playing & Recording Audio
       The  play  and  rec  commands  are  provided  so that basic playing and
       recording is as simple as

	  play existing-file.wav

       and

	  rec new-file.wav

       These two commands are functionally equivalent to

	  sox existing-file.wav -d

       and

	  sox -d new-file.wav

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

				 *	  *	   *

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

       or

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

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

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

	  play ... rate -m

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

				 *	  *	   *

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

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

	  rec -n

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

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

       Formats that discard audio signal information are called	 ‘lossy’,  and
       formats	that do not, ‘lossless’.  The term ‘quality’ is used as a mea-
       sure 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; hence this is also recommended.

   Dithering
       Dithering  is  a	 technique used to maximise the dynamic range of audio
       stored at a particular bit-depth: any distortion introduced by  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-
       priate.

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

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

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

       SoX’s stat and stats effects can assist in determining the signal level
       in an audio file; the gain or vol effect can be used to	prevent	 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, then SoX will dis-
       play a warning message to that effect.

       See also -G and the gain and norm effects.

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

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

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

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

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

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

       When  combining input files, SoX applies any specified effects (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
       place.

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

       The -V option (below) can be used to show the input file volume 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.	 So  here,  if
       manual  volume  adjustments are not given, to ensure that clipping does
       not occur, SoX will automatically adjust the volume (amplitude) of each
       input  signal by a factor of ¹/n, where n is the number of input files.
       If this results in audio that is too quiet or otherwise unbalanced then
       the  input  file	 volumes can be set manually as described above; using
       the norm effect on the mix is another alternative.

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

       With the ‘mix-power’ combine method, the mixed volume is	 appropriately
       equal to that of one of the input signals.  This is achieved by balanc-
       ing using a factor of ¹/√n instead of ¹/n.  Note	 that  this  balancing
       factor does not guarantee that no clipping will occur, however, in many
       cases, the number of clips will be low  and  the	 resultant  distortion
       imperceptible.

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

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

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

       In multiple output mode, a unique number will automatically be appended
       to the end of all filenames.  If the filename has an extension then the
       number is inserted before the extension.	 This behaviour	 can  be  cus-
       tomized	by  placing  a	%n  anywhere  in the filename where the number
       should be substituted.  An optional number can be placed after the % to
       indicate a minimum fixed width for the number.

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

       The following is an example of splitting the first  60  seconds	of  an
       input file in to two 30 second files and ignoring the rest.

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

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

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

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

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

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

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

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

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

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

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

       "wildcard-filename"
	      Specifies	 that  filename ‘globbing’ (wild-card matching) should
	      be performed by SoX instead of the shell.	 This allows a	single
	      set  of  file  options  to  be applied to a group of files.  For
	      example, if the current directory contains  three	 ‘vox’	files:
	      file1.vox, file2.vox, and file3.vox, then

		 play --rate 6k *.vox

	      will be expanded by the ‘shell’ (in most environments) to

		 play --rate 6k file1.vox file2.vox file3.vox

	      which will treat only the first vox file as having a sample rate
	      of 6k; but with

		 play --rate 6k "*.vox"

	      the given sample rate option will be applied to  all  three  vox
	      files.

       -p, --sox-pipe
	      This  can be used in place of an output filename to specify that
	      the SoX command should be used as in input pipe to  another  SoX
	      command.	For example, the command:

		 play "|sox -n -p synth 2" "|sox -n -p synth 2 tremolo 10" stat

	      plays two ‘files’ in succession, each with different effects.

	      -p is in fact an alias for ‘-t sox -’.

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

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

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

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

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

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

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

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

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

       Note that setting SOX_OPTS can potentially create unwanted  changes  in
       the  behaviour  of  scripts  or	other  programs	 that  invoke SoX.  So
       SOX_OPTS might best be used for things (such as in the  given  example)
       that  reflect  the  environment	in  which  SoX is being run.  Enabling
       options such as --no-clobber as default might be handled better using a
       shell  alias  since  a shell alias will not affect operation in scripts
       etc.

       One way to ensure that a script can not be affected by SOX_OPTS, is  to
       clear SOX_OPTS at the start of the script (but this of course loses the
       benefit of SOX_OPTS carrying some  system-wide  default	options).   An
       alternative  approach  is  to explicitly invoke SoX with default option
       values, e.g.

	  SOX_OPTS="-V --no-clobber"
	  ...
	  sox -V2 --clobber $input $output ...

       Note that the way of setting environment variables varies  from	system
       to system - here are some examples:

       Unix bash:

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

       Unix csh:

	  setenv SOX_OPTS "-V --no-clobber"

       MS-DOS/MS-Windows:

	  set SOX_OPTS=-V --no-clobber

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

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

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

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

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

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

       --effects-file FILENAME
	      Use FILENAME to obtain all effects  and  their  arguments.   The
	      file  is	parsed	as if the values were specified on the command
	      line.  A new line can be used in place of the special ":" marker
	      to separate effect chains.  This option causes any effects spec-
	      ified on the command line to be discarded.

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

		 sox -G infile -b 16 outfile rate 44100 dither -s

	      is shorthand for

		 sox infile -b 16 outfile gain -h rate 44100 gain -rh dither -s

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

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

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

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

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

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

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

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

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

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

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

		 sox file1 file2 effect1 effect2 ...

	      when what you really meant was

		 play file1 file2 effect1 effect2 ...

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

       --norm Automatically  invoke the gain effect to guard against clipping,
	      and to normalise the audio. E.g.

		 sox --norm infile -b 16 outfile rate 44100 dither -s

	      is shorthand for

		 sox infile -b 16 outfile gain -h rate 44100 gain -nh dither -s

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

       --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.
	      E.g.

		 sox --plot octave input-file -n highpass 1320 > highpass.plt
		 octave highpass.plt


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

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

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

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

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

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

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

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


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

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

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

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

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

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

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

       -v, --volume FACTOR
	      Intended for use	when  combining	 multiple  input  files,  this
	      option  adjusts  the  volume  of the file that follows it on the
	      command line by a factor of FACTOR, thus allowing	 it  to	 ‘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;
	      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’
	      file.

	      For  an output file, this option can be used (perhaps along with
	      -e) to set the output encoding size.  By default (i.e.  if  this
	      option  is  not given), the output encoding size will (providing
	      it is supported by the output file type) be  set	to  the	 input
	      encoding size.  For example

		 sox input.cdda -b 24 output.wav

	      converts	raw  CD	 digital  audio	 (16-bit, signed-integer) to a
	      24-bit (signed-integer) ‘WAV’ file.

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

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

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

		 play -c 1 music.wav

	      interprets the file  data	 as  belonging	to  a  single  channel
	      regardless  of  what is indicated in the file header.  Note that
	      if the file does in fact have two channels, this will result  in
	      the file playing at half speed.

	      For  an output file, this option provides a shorthand for speci-
	      fying that the channels effect should be	invoked	 in  order  to
	      change (if necessary) the number of channels in the audio signal
	      to the number given.  For example, the  following	 two  commands
	      are equivalent:

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

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

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

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

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

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

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

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

	      gsm-full-rate
		     GSM is currently  used  for  the  vast  majority  of  the
		     world’s  digital  wireless	 telephone calls.  It utilises
		     several audio formats with different bit-rates and	 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.

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

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

	      For  an  input  file,  the most common use for this option is to
	      inform SoX of the sample rate of a  ‘raw’	 (‘headerless’)	 audio
	      file  (see  the  examples in -b and -c above).  Occasionally, it
	      may be useful to use this option	with  a	 ‘headered’  file,  in
	      order to override the (presumably incorrect) value in the header
	      - note that this is only supported with certain file types.  For
	      example,	if  audio  was	recorded with a sample-rate of say 48k
	      from a source that played back a little, say 1.5%,  too  slowly,
	      then

		 sox -r 48720 input.wav output.wav

	      effectively  corrects the speed by changing only the file header
	      (but see also the speed effect for the more  usual  solution  to
	      this problem).

	      For  an output file, this option provides a shorthand for speci-
	      fying that the rate effect should be invoked in order to	change
	      (if  necessary) the sample rate of the audio signal to the given
	      value.  For example, the following two commands are equivalent:

		 sox input.wav -r 48k output.wav bass -3
		 sox input.wav	      output.wav bass -3 rate 48k

	      though the second form  is  more	flexible  as  it  allows  rate
	      options  to be given, and allows the effects to be ordered arbi-
	      trarily.

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

		 another-command | sox -t mp3 - output.wav

		 sox input.wav -t raw output.bin

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

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

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

	      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.

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

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

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

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

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

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

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

       The following is an example of multiple effects chains.	It will	 split
       the  input file into multiple files of 30 seconds in length.  Each out-
       put filename will have unique number in its name as documented in  Out-
       put 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:

       centre[k]
	      See frequency.

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

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

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

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

   Supported Effects
       Note:  a categorised list of the effects can be found in the 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.

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

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

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

	      The  channels effect is invoked automatically if SoX’s -c option
	      specifies a number of channels that is different to that of  the
	      input  file(s).	Alternatively, if this effect is given explic-
	      itly, then SoX’s -c option need not be given.  For example,  the
	      following two commands are equivalent:

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

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

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

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

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

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

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

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

	      Two delays of the original samples:

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

	      A fuller sounding chorus (with three additional delays):

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


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

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

	      The attack and decay parameters (in seconds) determine the  time
	      over  which the instantaneous level of the input signal is 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
	      function).

	      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 ISO 908 de-emphasis (a treble attenuation shelving filter)
	      to 44.1kHz (Compact Disc) audio.

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

	      With  the	 deemph	 effect, it is possible to apply the necessary
	      de-emphasis to audio that has been extracted from	 a  pre-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
	      example:

		 sox track1.wav track1-deemph.wav deemph

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

		 play track1-deemph.wav

	      or simply

		 play track1.wav deemph

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

	      This effect supports the --plot global option.

	      See also the bass and treble shelving equalisation effects.

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

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

	      and this plays a guitar chord:

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


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

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

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

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

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

	      See also the ‘Dither’ section above.

       earwax Makes audio easier to listen to on headphones.  Adds  ‘cues’  to
	      44.1kHz  stereo  (i.e.  audio CD format) audio so that when lis-
	      tened to on headphones the stereo image  is  moved  from	inside
	      your  head  (standard for headphones) to outside and in front of
	      the listener (standard  for  speakers).	See  http://www.geoci-
	      ties.com/beinges for a full explanation.

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

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

		 play lead.aiff echo 0.8 0.88 60 0.4

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

		 play lead.aiff echo 0.8 0.88 6 0.4

	      A longer delay will sound like an open air concert in the	 moun-
	      tains:

		 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
		   1.2311233052619888e-01
		  -4.4777096106211783e-01
		   5.1031563346705155e-01
		  -6.6502926320995331e-02
		 ...


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

	      All parameters are optional (right to left).

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

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

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

	      Given  the  -e  option,  the  levels  of the audio channels of a
	      multi-channel file are ‘equalised’, i.e.	gain is applied to all
	      channels	other than that with the highest peak level, such that
	      all channels attain the same peak level (but, without also  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
	      -r.

	      See also the norm and vol effects.

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

	      These effects support the --plot global option.

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

       ladspa module [plugin] [argument...]
	      Apply a LADSPA [5] (Linux Audio Developer’s Simple  Plugin  API)
	      plugin.	Despite	 the name, LADSPA is not Linux-specific, and a
	      wide range of effects is available as LADSPA  plugins,  such  as
	      cmt  [6]	(the Computer Music Toolkit) and Steve Harris’s plugin
	      collection [7]. The first argument is  the  plugin  module,  the
	      second  the  name	 of the plugin (a module can contain more than
	      one plugin) and any other arguments are for the control ports of
	      the  plugin. Missing arguments are supplied by default values if
	      possible. Only plugins with at most  one	audio  input  and  one
	      audio  output port can be used.  If found, the environment 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
	      http://en.wikipedia.org/wiki/Loudness for a detailed description
	      of  loudness.   The gain is adjusted by the given gain parameter
	      (usually negative) and the signal equalised according to ISO 226
	      w.r.t.  a	 reference level of 65dB, though an alternative 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}
	      [soft-knee-dB:]in-dB1[,out-dB1]{,in-dB2,out-dB2}
	      [gain	[initial-volume-dB    [delay]]]"    {crossover-freq[k]
	      "attack1,..."}

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

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

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

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

	      See also compand for a single-band companding effect.

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

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

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

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

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

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

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

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

       noisered [profile-file [amount]]
	      Reduce  noise  in	 the  audio signal by profiling and filtering.
	      This effect is moderately effective at removing consistent 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.

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

       oops   Out Of Phase Stereo effect.  Mixes  stereo  to  twin-mono	 where
	      each  mono  channel contains the difference between the left and
	      right stereo channels.  This is sometimes known as the ‘karaoke’
	      effect as it often has the effect of removing most or all of the
	      vocals from a recording.

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

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

	      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

	      Gentler:

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

	      A popular sound:

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

	      More severe:

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


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

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

	      See also the speed and tempo effects.

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

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

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

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

	      The  rate	 effect	 is  invoked  automatically if SoX’s -r option
	      specifies a rate that is different to that of the input file(s).
	      Alternatively, if this effect is given explicitly, then SoX’s -r
	      option need not be given.	 For example, the following  two  com-
	      mands are equivalent:

		 sox input.wav -r 48k output.wav bass -3
		 sox input.wav	      output.wav bass -3 rate 48k

	      though  the  second  command  is more flexible as it allows rate
	      options to be given, and allows the effects to be ordered	 arbi-
	      trarily.

				    *	     *	      *

	      Warning: technically detailed discussion follows.

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

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

	      N.B.  Override options can not be used with the ‘quick’ or ‘low’
	      quality algorithms.

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

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

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

	      A resampler’s band-width setting determines how much of the fre-
	      quency  content of the original signal (w.r.t. the original sam-
	      ple rate when up-sampling, or the new sample rate when down-sam-
	      pling)  is preserved during conversion.  The term ‘pass-band’ is
	      used to refer to all frequencies	up  to	the  band-width	 point
	      (e.g.  for 44.1kHz sampling rate, and a resampling band-width of
	      95%, the pass-band represents frequencies	 from  0Hz  (D.C.)  to
	      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%.

	      Examples:

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

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

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

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

				    *	     *	      *

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

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

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

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

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

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

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

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

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

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

		 sox input.wav output.wav remix -

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

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

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

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

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

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

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

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

	      The -m (manual) option disables  all  automatic  volume  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,
	      then

		 sox input.wav output.wav remix 1,2i

	      is a mono equivalent of the oops effect.

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

				    *	     *	      *

	      One use of the remix effect is to split an audio file into a set
	      of  files,  each	containing one of the constituent channels (in
	      order to	perform	 subsequent  processing	 on  individual	 audio
	      channels).   Where  more	than  a	 few  channels are involved, a
	      script such as the following (Bourne shell script) is useful:

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

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

	      See also mixer and swap for similar effects.

       repeat count
	      Repeat the entire audio count times.   Requires  temporary  file
	      space  to	 store	the audio to be repeated.  Note that repeating
	      once yields two copies: the  original  audio  and	 the  repeated
	      audio.

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

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

	      Note  that  this effect increases both the volume and the length
	      of the audio, so to prevent clipping in these domains, a typical
	      invocation might be:

		 play dry.wav gain -3 pad 0 3 reverb

	      The -w option can be given to select only the ‘wet’ signal, thus
	      allowing it to be processed further, independently of the	 ‘dry’
	      signal.  E.g.

		 play -m voice.wav "|sox voice.wav -p reverse reverb -w reverse"

	      for a reverse reverb effect.

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

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

	      This effect supports the --plot global option.

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

	      Removes silence from the beginning, middle, or end of the audio.
	      Silence is anything below a specified threshold.

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

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

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

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

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

	      The period counts are in units of samples. Duration  counts  may
	      be  in  the  format of hh:mm:ss.frac, or the exact count of 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] [fre-
       qHP][-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 < freqLP creates a band-pass fil-
	      ter, freqHP > freqLP creates a band-reject filter.

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

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

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

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

	      This effect supports the --plot global option.

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

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

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

		 sox my.wav -n spectrogram

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

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

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

		 sox 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.

	      Options:

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

	      -X num X-axis  pixels/second;  the default is auto-calculated to
		     fit the given or known audio duration to the X-axis size,
		     or	 100 otherwise.	 If given in conjunction with -d, this
		     option affects the width of the  spectrogram;  otherwise,
		     it	 affects  the duration of the spectrogram.  num can be
		     from 1 (low time resolution) to 5000 (high	 time  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-
		     able.

	      -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
		     permutation.

	      -l     Creates a ‘printer friendly’  spectrogram	with  a	 light
		     background (the default has a dark background).

	      -a     Suppress  the  display  of the axis lines.	 This is some-
		     times useful in helping to discern artefacts at the spec-
		     trogram edges.

	      -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
		     artefacts.

	      -t text
		     Set the image title - text to display above the  spectro-
		     gram.

	      -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-
		     trogram.png’.

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

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

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

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

		     the stats effect is applied to the entire audio signal.

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

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

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

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

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

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

	      By default, the speed change is performed by resampling with the
	      rate effect using its default quality/speed.  For higher quality
	      or higher speed resampling, in addition  to  the	speed  effect,
	      specify the rate effect with the desired quality option.

	      See also the pitch and tempo effects.

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

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

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

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

	      For example, a long song begins with two verses which start  (as
	      determined  e.g. by using the play command with the trim (start)
	      effect) at times 0:30.125 and 1:03.432.  The following  commands
	      cut out the first verse:

		 sox too-long.wav part1.wav trim 0 30.130

	      (5 ms excess, after the first verse starts)

		 sox too-long.wav part2.wav trim 1:03.422

	      (5 ms excess plus 5 ms leeway, before the second verse starts)

		 sox part1.wav part2.wav just-right.wav splice 30.130

	      For another example, the SoX command

		 play "|sox -n -p synth 1 sin %1" "|sox -n -p synth 1 sin %3"

	      generates and plays two notes, but there is a nasty click at the
	      transition; the click can be removed by splicing instead of con-
	      catenating the audio, i.e. by appending splice 1 to the command.
	      (Clicks at the beginning and end of the audio can be removed  by
	      preceding the splice effect with fade q .01 2 .01).

	      Provided your arithmetic is good enough, multiple splices can be
	      performed with a single splice invocation.  For example:

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

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

				    *	     *	      *

	      It is also possible to use this effect to perform general cross-
	      fades, e.g. to join two songs.  In this case, excess would 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-
	      lows:

	       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-
							 ber.
	       Minimum amplitude   min(xk)		 The minimum  sample
							 value in the audio;
							 usually  this	will
							 be  a negative num-
							 ber.
	       Midline amplitude   ½min(xk)+½max(xk)
	       Mean norm	   ¹/nΣ│xk│		 The average of	 the
							 absolute  value  of
							 each sample in	 the
							 audio.






	       Mean amplitude	   ¹/nΣxk		 The average of each
							 sample	   in	 the
							 audio.	   If	this
							 figure is non-zero,
							 then  it  indicates
							 the presence  of  a
							 D.C.  offset (which
							 could	be   removed
							 using	the  dcshift
							 effect).
	       RMS amplitude	   √(¹/nΣxk²)		 The level of a D.C.
							 signal	 that  would
							 have the same power
							 as    the   audio’s
							 average power.
	       Maximum delta	   max(│xk-xk-1│)
	       Minimum delta	   min(│xk-xk-1│)
	       Mean delta	   ¹/n-1Σ│xk-xk-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
							 this.

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

	      The -s option can be used to scale the input  data  by  a	 given
	      factor.  The default value of scale is 2147483647 (i.e. the 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
	      applicable.

	      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;
	      default=sine.

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

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

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

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

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

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

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

	      Not used for noise.

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

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

	      p1 is the percentage of each cycle that  is  ‘on’	 (square),  or
	      ‘rising’	(triangle, exp, trapezium); default=50 (square, trian-
	      gle,  exp),  default=10	(trapezium),   or   sustain   (pluck);
	      default=40.

	      p2  (trapezium):	the  percentage	 through  each	cycle at which
	      ‘falling’ begins; default=50. exp: the amplitude in multiples of
	      2dB; default=50, or tone-1 (pluck); default=20.

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

       tempo [-q] factor [segment [search [overlap]]]
	      Change the audio tempo  (but  not	 its  pitch).	The  audio  is
	      chopped  up  into	 segments  which  are then shifted in the time
	      domain and overlapped (cross-faded) at points where their	 wave-
	      forms  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, giving a quicker, but possibly	 lower
	      quality, result.

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

	      The optional segment parameter selects the  algorithm’s  segment
	      size  in milliseconds.  The default value is 82 and is typically
	      suited to making small changes to the tempo of music; for larger
	      changes  (e.g.  a	 factor of 2), 50 ms may give a better result.
	      When changing the tempo of speech,  a  segment  size  of	around
	      30 ms often works well.

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

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

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

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

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

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

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

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

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

		 play speech.wav norm vad

	      to trim from the front,

		 play speech.wav norm reverse vad reverse

	      to trim from the back, and

		 play speech.wav norm vad reverse vad reverse

	      to trim from both ends.  The use of the norm  effect  is	recom-
	      mended,  but  remember that neither reverse nor norm is suitable
	      for use with streamed audio.

	      Options:
	      Default values are shown in parenthesis.

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

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

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

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

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

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

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

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

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

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

	      -M num Time constant used to smooth spectral measurements.

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

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

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

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

	      See also the silence effect.

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

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

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

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

	      See [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.

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

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

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

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

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

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

	      This effect supports the --plot global option.

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

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

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

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

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

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

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

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

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

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

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

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

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

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

LICENSE
       Copyright 1998-2009 Chris Bagwell and SoX Contributors.
       Copyright 1991 Lance Norskog and Sundry Contributors.

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

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

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



sox				 June 14, 2009				SoX(1)
