SoX(1)                         Sound eXchange_ng                        SoX(1)

NAME
       SoX - Sound eXchange_ng, another Swiss Army knife of audio manipulation

SYNOPSIS
       sox_ng [global‐options] [format‐options] infile1
            [[format‐options] infile2] ... [format‐options] outfile
            [effect [effect‐options]] ...

       play_ng [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 op‐
       tionally apply effects to them. It can combine multiple input  sources,
       synthesize  audio, and, on many systems, act as a general purpose audio
       player or a multi‐track audio recorder. It also has limited ability  to
       split the input into multiple output files.

       All  SoX  functionality is available using just the sox_ng command.  To
       simplify playing and recording audio, if SoX is invoked as play_ng, 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_ng(1) command provides a convenient way to just
       query audio file header information.

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

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

       The overall SoX processing chain can be summarized 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_ng recital.au recital.wav

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

          sox_ng 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, normalize),  and
       stores the result at a bit‐depth of 16.

          sox_ng -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_ng slow.aiff fixed.aiff speed 1.027

       adjusts audio speed,

          sox_ng short.wav long.wav longer.wav

       concatenates two audio files, and

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

       mixes together two audio files.

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

       plays a collection of audio files whilst applying a bass  boosting  ef‐
       fect,

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

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

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

       records half an hour of stereo audio, and

          play_ng -q take1.aiff & rec -M take1.aiff take1-dub.aiff

       (with  POSIX shell and where supported by hardware) records a new track
       in a multi‐track recording.  Finally,

          rec_ng -r 44100 -b 16 -e signed‐integer -p \
            silence 1 0.50 0.1% 1 10:00 0.1% | \
            sox_ng -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 ex‐
       planations of how to use all SoX parameters, file formats, and  effects
       can  be  found  below  in  this  manual,  in  soxformat_ng(7),  and  in
       soxi_ng(1).

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

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

       sample rate
              The sample rate in samples per second (‘Hertz’ or ‘Hz’).   Digi‐
              tal  telephony  traditionally  uses  a  sample  rate  of 8000 Hz
              (8 kHz), though these days, 16 and even 32 kHz are becoming more
              common. Audio Compact Discs use 44100 Hz (44.1 kHz). Digital Au‐
              dio Tape and many computer systems use 48 kHz. Professional  au‐
              dio 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  ‘en‐
              coded’).   Some  encodings have variants with different byte‐or‐
              derings or bit‐orderings.  Some compress the audio data so  that
              the  stored  audio  data takes up less space (i.e. disk space or
              transmission bandwidth) than the other format parameters and the
              number of samples would imply.  Commonly‐used encoding types in‐
              clude floating‐point, μ‐law, ADPCM, signed‐integer PCM, MP3, and
              FLAC.

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

       The term ‘bit‐rate’ is a measure of the amount of storage  occupied  by
       an  encoded  audio signal over a unit of time.  It can depend on all of
       the above and is typically denoted as a number of kilo‐bits per  second
       (kbps).   An  A‐law telephony signal has a bit‐rate of 64 kbps. MP3‐en‐
       coded stereo music typically has a bit‐rate of 128-196  kbps.  FLAC‐en‐
       coded 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’ in‐
       formation.  SoX supports applying Replay Gain information (for  certain
       input  file formats only; currently, at least FLAC and Ogg Vorbis), but
       not generating it.  Note that by default, SoX copies  input  file  com‐
       ments  to  output files that support comments, so output files may con‐
       tain Replay Gain information which is incorrect.  This  can  be  fixed,
       when converting input files with --replay-gain enabled, by removing all
       comments  using  --comment ""  or  removing just the REPLAYGAIN comment
       with

       soxi_ng ‐a in.au | grep ‐v REPLAYGAIN > comments
       sox_ng ‐‐replay‐gain=track in.au ‐‐comment‐file comments out.au


       The soxi_ng(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 that is  sup‐
           ported by the output file type.

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

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

          play_ng existing‐file.wav

       and

          rec_ng new‐file.wav

       These two commands are functionally equivalent to

          sox_ng existing‐file.wav -d

       and

          sox_ng -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  dri‐
       ver,  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 record‐
       ing or playing is not the one that is wanted, then the AUDIODRIVER  en‐
       vironment  variable  can  be used to override the default.  For example
       (on many systems):

          set AUDIODRIVER=oss
          play_ng ...

       The AUDIODEV environment variable can be used to override  the  default
       audio device, e.g.

          set AUDIODEV=/dev/dsp2
          play_ng ...
          sox_ng ... -t oss

       or

          set AUDIODEV=hw:soundwave,1,2
          play_ng ...
          sox_ng ... -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 au‐
       dio output device, SoX will automatically invoke  the  rate  effect  to
       perform  the  necessary sample rate conversion.  For compatibility with
       old hardware, the default rate quality level is set to ‘low’. This  can
       be  changed  by  explicitly specifying the rate effect with a different
       quality level, e.g.

          play_ng ... 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_ng -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 be‐
       low.

   Accuracy
       Many file formats that compress audio discard some of the audio  signal
       information  whilst doing so. Converting to such a format and then con‐
       verting back again will not produce an exact copy of the  original  au‐
       dio.   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  fi‐
       delity,  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’.  For‐
       mats  that do not are called ‘lossless’.  The term ‘quality’ is used as
       a measure of how closely the original audio signal  can  be  reproduced
       when using a lossy format.

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

       N.B.   SoX  converts all audio files to an internal uncompressed format
       before performing any audio processing. This means that manipulating  a
       file that is stored in a lossy format can cause further losses in audio
       fidelity.  E.g. with

          sox_ng 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 recompressing the audio with
       a possible reduction in fidelity above that which occurred when the in‐
       put  file was created.  Hence, if what is ultimately desired is lossily
       compressed audio, it is highly recommended to perform  all  audio  pro‐
       cessing  using lossless file formats and then convert to the lossy for‐
       mat only at the final stage.

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

   Dithering
       Dithering  is  a  technique used to maximize 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 utilize 18 bits after processing this vol‐
       ume change.  In order to store the output at the same depth as the  in‐
       put, 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. Clipping could also occur
       with  many  other  effects,  when converting one format to another, and
       even when simply playing the audio.

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

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

       SoX’s stat and stats effects can assist in determining the signal level
       in  an  audio file. The gain or vol effect can be used to prevent clip‐
       ping, e.g.

          sox_ng dull.wav bright.wav gain -6 treble +6

       guarantees that the treble boost will not clip.

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

       See also -G and the gain and norm effects.

   Input File Combining
       SoX’s  input  combiner can be configured (see OPTIONS below) to combine
       multiple files using any of the following methods: ‘concatenate’,  ‘se‐
       quence’,  ‘mix’,  ‘mix‐power’,  ‘merge’,  or  ‘multiply’.   The default
       method is ‘sequence’ for play_ng,  and  ‘concatenate’  for  rec_ng  and
       sox_ng.

       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_ng.
       It  is  similar to ‘concatenate’ in that the audio from each input file
       is sent serially to the output file. However, here the output file  may
       be  closed  and  reopened at the corresponding transition between input
       files. This may be just what is needed when sending different types  of
       audio  to an output device, but is not generally useful when the output
       is a normal file.

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

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

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

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

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

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

       With  the ‘mix‐power’ combine method, the mixed volume is approximately
       equal to that of one of the input signals.  This is achieved by balanc‐
       ing using a factor of ¹/√n instead of ¹/n.  Note  that  this  balancing
       factor  does not guarantee that clipping will not occur, but the number
       of clips will usually be low and the resultant distortion is  generally
       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 af‐
       ter ‘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 in‐
       put file into two 30 second files and ignoring the rest.

          sox_ng 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. 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_ng -M "|genw --imd -" "|genw --thd -" out.wav

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

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

                 play_ng --rate 6k *.vox

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

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

              which will treat only the first vox file as having a sample rate
              of 6k.  With

                 play_ng --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_ng "|sox_ng -n -p synth 2" "|sox_ng -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_ng  or
              play_ng (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 au‐
              dio 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_ng(7) for a list and description of  the  supported  file
       formats and audio device drivers.

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

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

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

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

       One  way  to  ensure that a script cannot be affected by SOX_OPTS is to
       clear SOX_OPTS at the start of the script, but this of course loses the
       benefit of SOX_OPTS carrying some system‐wide default options.  An  al‐
       ternative approach is to explicitly invoke SoX with default option val‐
       ues, e.g.

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

       Note  that  the  way to set environment variables varies from system to
       system. Here are some examples:

       Unix bash:

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

       Unix csh:

          setenv SOX_OPTS "-V --no‐clobber"

       MS‐DOS/MS‐Windows:

          set SOX_OPTS=-V --no‐clobber

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

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

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

              Be aware that large values for --buffer will cause SoX to be be‐
              come  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 be‐
              haviour.

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

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

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

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

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

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

              is shorthand for

                 sox_ng 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_ng, behave as
              soxi_ng(1).

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

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

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

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

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

                 sox_ng file1 file2 effect1 effect2 ...

              when what you really meant was

                 play_ng 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[=dB‐level]
              Automatically invoke the gain effect to guard  against  clipping
              and to normalize the audio. E.g.

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

              is shorthand for

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

              Optionally,  the  audio can be normalized to a given level (usu‐
              ally) below 0 dBFS:

                 sox_ng --norm=-3 infile outfile


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

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

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

                 sox_ng --plot octave input‐file -n highpass 1320 > highpass.plt
                 octave highpass.plt


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

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

       --replay-gain track|album|off
              Select whether or not to apply replay gain adjustment  to  input
              files.   The  default  is  off  for sox_ng and rec_ng, album for
              play_ng where (at least) the first two input  files  are  tagged
              with the same Artist and Album names, and track for play_ng oth‐
              erwise.

       -S, --show-progress
              Display  input  file  format/header  information, and processing
              progress as input file(s) percentage complete, elapsed time, and
              remaining time (if known; shown in brackets), and the number  of
              samples  written to the output file.  Also shown is a peak‐level
              meter, and an indication if clipping has  occurred.   The  peak‐
              level meter shows up to two channels and is calibrated for digi‐
              tal audio as follows (right channel shown):
                            dB FSD   Display   dB FSD   Display
                             -25     -          -11     ====
                             -23     =           -9     ====-
                             -21     =-          -7     =====
                             -19     ==          -5     =====-
                             -17     ==-         -3     ======
                             -15     ===         -1     =====!
                             -13     ===-

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

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

       -T     Equivalent to --combine multiply.

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

              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 op‐
              tion adjusts the volume of the file that follows it on the  com‐
              mand line by a factor of FACTOR. This allows it to be ‘balanced’
              w.r.t.  the other input files.  This is a linear (amplitude) ad‐
              justment, so a number less than 1 decreases  the  volume  and  a
              number  greater  than  1  increases it.  If a negative number is
              given then in addition to the volume adjustment, the audio  sig‐
              nal 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 in‐
              form  SoX  of the number of bits per sample in a ‘raw’ (‘header‐
              less’) audio file.  For example

                 sox_ng -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 en‐
              coding size.  For example

                 sox_ng input.cdda -b 24 output.wav

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

       -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 in‐
              form SoX of the number of channels in a ‘raw’ (‘headerless’) au‐
              dio file.  Occasionally, it may be useful  to  use  this  option
              with a ‘headered’ file, in order to override the (presumably in‐
              correct)  value in the header ‐ note that this is only supported
              with certain file types.  Examples:

                 sox_ng -r 48k -e float -b 32 -c 2 input.raw output.wav

              converts a particular ‘raw’  file  to  a  self‐describing  ‘WAV’
              file.

                 play_ng -c 1 music.wav

              interprets  the  file  data as belonging to a single channel re‐
              gardless 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_ng input.wav -c 1 output.wav bass -b 24
                 sox_ng input.wav      output.wav bass -b 24 channels 1

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

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

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

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

              floating‐point
                     PCM  data stored as IEEE 753 single precision (32‐bit) or
                     double precision (64‐bit)  floating‐point  (‘real’)  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 be‐
                     tween 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 utilizes
                     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 am‐
              biguous; 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 in‐
              form  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_ng input.cdda -e float output1.wav

                 sox_ng 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.

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

                 sox_ng -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_ng input.wav -r 48k output.wav bass -b 24
                 sox_ng input.wav        output.wav bass -b 24 rate 48k

              though  the  second  form is more flexible as it allows rate op‐
              tions 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_ng -t mp3 - output.wav

                 sox_ng 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_ng(7) for a list of supported file types.

       -L, --endian little
       -B, --endian big
       -x, --endian swap
              These  options  specify whether the byte order of the audio data
              is, respectively, ‘little endian’, ‘big endian’ or the  opposite
              to  that  of  the system on which SoX is being used.  Endianness
              applies only to data encoded as floating‐point, or as signed  or
              unsigned  integers of 16 or more bits.  It is often necessary to
              specify one of these options for headerless files, and sometimes
              necessary for (otherwise) self‐describing files.   A  given  en‐
              dian‐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_ng -B audio.s16 trimmed.s16 trim 2

              trimmed.s16 will be created as little‐endian;

                 sox_ng -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  re‐
              versed; sometimes useful with a few (mostly headerless) formats.

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

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

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

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

              SoX  will  provide  a  default comment if this option (or --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  de‐
              scription  of  the  file formats that use this option in soxfor‐
              mat_ng(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 ef‐
       fects  in  real‐time  (i.e.  when playing audio) is likely to require a
       high performance computer. Stopping other  applications  may  alleviate
       performance issues should they occur.

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

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

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

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

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

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

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

          sox_ng infile.wav output.wav trim 0 30 : newfile : restart


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

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

       center[k]
              See frequency.

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

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

       position
              A position within the audio stream; the syntax  is  [=|+|-]time‐
              spec,  where  timespec is a time specification (see below).  The
              optional first character indicates whether the timespec is to be
              interpreted relative to the start (=) or end (-) of audio, or to
              the previous position if the effect  accepts  multiple  position
              arguments  (+).  The audio length must be known for end‐relative
              locations to work; some effects do accept -0  for  end‐of‐audio,
              though,  even if the length is unknown.  Which of =, +, - is the
              default depends on the effect and is shown  in  its  syntax  as,
              e.g., position(+).

              Examples:  =2:00 (two minutes into the audio stream), -100s (one
              hundred samples before the end of audio), +0:12+10s (twelve sec‐
              onds and ten samples after the previous position), -0.5+1s  (one
              sample less than half a second before the end of audio).

       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 first line of the effect’s description.

       Most  effects that expect an audio position or duration in a parameter,
       i.e. a time specification, accept either of the following two forms:

       [[hours:]minutes:]seconds[.frac][t]
              A specification of ‘1:30.5’ corresponds to  one  minute,  thirty
              and  ½ seconds.  The t suffix is entirely optional (however, see
              the silence effect for an exception).  Note that  the  component
              values  do  not have to be normalized; e.g., ‘1:23:45’, ‘83:45’,
              ‘79:0285’, ‘1:0:1425’, ‘1::1425’ and ‘5025’ all  are  legal  and
              equivalent to each other.

       sampless
              Specifies  the  number  of samples directly, as in ‘8000s’.  For
              large sample counts, e notation is supported:  ‘1.7e6s’  is  the
              same as ‘1700000s’.

       Time  specifications  can  also  be chained with + or - into a new time
       specification where the right part is added to or subtracted  from  the
       left,  respectively:  ‘3:00-200s’  means  two hundred samples less than
       three minutes.

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

   Supported Effects
       Note:  a categorized 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 op‐
              tional 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)] { start‐posi‐
       tion(+),cents,end‐position(+) }
              Changes  pitch  by  specified  amounts at specified times.  Each
              given triple:  start‐position,cents,end‐position  specifies  one
              bend.   cents is the number of cents (100 cents = 1 semitone) by
              which to bend the pitch. The other values specify the points  in
              time at which to start and end bending the pitch, respectively.

              The pitch‐bending algorithm utilizes 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_ng -n synth 2.5 sin 667 gain 1 \
                   bend .35,180,.25  .15,740,.53  0,-520,.3

              Here, the first bend runs from 0.35 to 0.6, and the  second  one
              from  0.75 to 1.28 seconds.  Note that the clipping that is pro‐
              duced in this example is deliberate; to remove it,  use  gain -5
              in place of gain 1.

              See also pitch.

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

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

              This effect supports the --plot global option.

       channels CHANNELS
              Invoke  a  simple  algorithm to change the number of channels in
              the audio signal to the given number  CHANNELS:  mixing  if  de‐
              creasing 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_ng input.wav -c 1 output.wav bass -b 24
                 sox_ng input.wav      output.wav bass -b 24 channels 1

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

              For  example,  when  making a stereo file quadraphonic, the left
              and right channels are copied into the third and fourth and when
              mixing a four‐channel file down to stereo, the left  channel  is
              the  mix  of the first and third and the right of the second and
              fourth.

              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 de‐
              fines 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 de‐
              lay 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_ng guitar1.wav chorus 0.7 0.9 55 0.4 0.25 2 -t

              Two delays of the original samples:

                 play_ng 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_ng 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

              flanger  can do everything that chorus can do except for gain‐in
              and gain‐out but works internally in floating point and interpo‐
              lates between samples so is slower but gives much less noisy re‐
              sults.

       compand attack1,decay1{,attack,decay}
              [soft‐knee‐dB:]in‐dB1[,out‐dB1]{,in‐dB,out‐dB}
              [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 at‐
              tack 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 in‐
              put 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 as‐
              sumed for each channel when companding starts.  This permits the
              user to supply a nominal level initially, so that, for  example,
              a very large gain is not applied to initial signal levels before
              the companding action has begun to operate: it is quite probable
              that  in  such  an  event,  the output would be severely clipped
              while the compander gain properly  adjusts  itself.   A  typical
              value (for audio which is initially quiet) is -90 dB.

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

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

                 sox_ng 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 be‐
              tween 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_ng 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_ng 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 [limiter-gain]
              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  limiter-gain  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_ng -n dc.wav synth 5 sin %0 50
                 sox_ng dc.wav fixed.wav highpass 10


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

              Pre‐emphasis was applied in the mastering of some CDs issued  in
              the early 1980s.  These included many classical music albums, as
              well  as  now sought‐after issues of albums by The Beatles, Pink
              Floyd and others.  Pre‐emphasis should be  removed  at  playback
              time  by  a de‐emphasis filter in the playback device.  However,
              not all modern CD players have this filter, and very few  PC  CD
              drives have it; playing pre‐emphasized 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‐
              sized  CD, and then either burn the de‐emphasized audio to a new
              CD (which will then play correctly on any CD player), or  simply
              play the correctly de‐emphasized audio files on the PC.  For ex‐
              ample:

                 sox_ng track1.wav track1-deemph.wav deemph

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

                 play_ng track1-deemph.wav

              or simply

                 play_ng track1.wav deemph

              The  de‐emphasis  filter is implemented as a biquad and requires
              the input audio sample rate to be either 44.1kHz or 48kHz.  Max‐
              imum 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 {position(=)}
              Delay  one  or  more  audio channels such that they start at the
              given position.  For example, delay  1.5  +1  3000s  delays  the
              first  channel by 1.5 seconds, the second channel by 2.5 seconds
              (one second more than the previous channel), the  third  channel
              by  3000  samples,  and  leaves  any  other channels that may be
              present un‐delayed.  The following (one long)  command  plays  a
              chime sound:

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


       dither [-S|-s|-f filter] [-a] [-p precision]
              Apply  dithering  to  the  audio.  Dithering deliberately adds a
              small amount of noise to the signal in  order  to  mask  audible
              quantization effects that can occur if the output sample size is
              less than 24 bits.  With no options, this effect will add trian‐
              gular  (TPDF) white noise.  Noise‐shaping (only for certain sam‐
              ple rates) can be selected with -s.  With the -f option,  it  is
              possible  to  select  a particular noise‐shaping filter from the
              following list: lipshitz, f‐weighted,  modified‐e‐weighted,  im‐
              proved‐e‐weighted, gesemann, shibata, low‐shibata, high‐shibata.
              Note that most filter types are available only with 44100Hz sam‐
              ple  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.

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

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

              The -p option allows overriding the target precision.

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

              This effect should not be followed by any other effect that  af‐
              fects the audio.

              See also the ‘Dithering’ section above.

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

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

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

       earwax This effect takes a 44.1kHz stereo signal and  adds  audio  cues
              that,  when listened to on headphones, move the sound stage from
              inside your head to outside and in front of you, as if listening
              to loudspeakers.

       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_ng lead.aiff echo 0.8 0.88 60 0.4

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

                 play_ng 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_ng lead.aiff echo 0.8 0.9 1000 0.3

              One mountain more, and:

                 play_ng 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_ng lead.aiff echos 0.8 0.7 700 0.25 700 0.3

              The sample will be bounced twice in asymmetric echos:

                 play_ng lead.aiff echos 0.8 0.7 700 0.25 900 0.3

              The sample will sound as if played in a garage:

                 play_ng 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‐re‐
              ject 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‐position(=) [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  the  time  given  as  fade‐in‐
              length.  Specify 0 if no fade‐in is wanted.

              For  fade‐outs, the audio will be truncated at stop‐position and
              the signal level will be ramped from full volume down to 0  over
              an  interval  of  fade‐out‐length  before the stop‐position.  If
              fade‐out‐length is not specified, it defaults to the same  value
              as fade‐in‐length.  No fade‐out is performed if stop‐position is
              not  specified.   If the audio length can be determined from the
              input file header and any previous effects,  then  -0  (or,  for
              historical reasons, 0) may be specified for stop‐position to in‐
              dicate  the usual case of a fade‐out that ends at the end of the
              input audio stream.

              Any time specification may be used for fade‐in‐length and  fade‐
              out‐length.

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


                 sox_ng infile outfile fir coefs.txt

              with coefs.txt containing

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


              This effect supports the --plot global option.

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

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

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

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

              Given the -e option, the levels  of  the  audio  channels  of  a
              multi‐channel file are ‘equalized’, 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 ‘normalized’).

              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 ap‐
              plied 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 normalizes the audio to 0dB FSD; it is often  used
              in  conjunction  with  a negative gain‐dB to the effect that the
              audio is normalized to a given level below 0dB.  For example,

                 sox_ng infile outfile gain -n

              normalizes to 0dB, and

                 sox_ng infile outfile gain -n -3

              normalizes to -3dB.

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

                 sox_ng 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_ng 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  at‐
              tenuation, is that if the headroom is not used by subsequent ef‐
              fects, it can be reclaimed with gain -r, for example:

                 sox_ng 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 re‐
              quires headroom (which cannot be ‘reclaimed’), e.g.

                 sox_ng 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  re‐
              sponse.   The  filters roll off at 6dB per pole per octave (20dB
              per pole per decade).  The double‐pole filters are described  in
              detail in [1].

              These effects support the --plot global option.

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

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

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

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

              This effect supports the --plot global option.

       ladspa [‐l|‐r] 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.

              Normally, the number of input ports of the plugin must match the
              number  of input channels, and the number of output ports deter‐
              mines the output channel count.  However, the -r (replicate) op‐
              tion allows cloning a mono plugin to handle multi‐channel input.

              Some plugins introduce latency which SoX may optionally  compen‐
              sate  for.   The  -l (latency compensation) option automatically
              compensates for latency as reported by the plugin via an  output
              control port named "latency".

              The  environment  variable LADSPA_PATH is a colon‐separated list
              of directories in which to search for LADSPA plugins.   The  de‐
              fault  depends  on  how SoX was built but on Unix it defaults to
              /usr/lib/ladspa   and   on   MacOS/X   to   /Library/Audio/Plug‐
              Ins/LADSPA.   Windows  doesn’t  have  a "usual place" for LADSPA
              plugins, but  Ardour  puts  them  in  \f(CWC:\Program  Files\Ar‐
              dour6\lib\ardour6\ladspa\fR or similar.

       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 equalized 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 equalized
              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 "compand‐args" {frequency "compand‐args"}
              +.SP +The quoted compand‐args are as for the compand effect:
              attack1,decay1{,attack,decay}        [soft‐knee‐dB:]in‐dB1[,out‐
              dB1]{,in‐dB,out‐dB}
              [gain [initial‐volume‐dB [delay]]]

              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 pa‐
              rameters.   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.

              The  following  examples approximate Dolby A compression and de‐
              compression, as used for tape noise  reduction  in  professional
              recording studios:

              # Dolby A compressor
              sox_ng in.au dolbyA.au mcompand     ".1,.1 4:‐56,‐46,‐36,‐26,‐26,‐20,‐17,‐15,‐9,‐9" 80     ".1,.1 4:‐56,‐46,‐36,‐26,‐26,‐20,‐17,‐15,‐9,‐9" 3k     ".1,.1 4:‐56,‐46,‐36,‐26,‐26,‐20,‐17,‐15,‐9,‐9" 9k     ".1,.1 4:‐56,‐42,‐36,‐23,‐26,‐18,‐17,‐14,‐9,‐9"

              # Dolby A decompressor
              sox_ng dolbyA.au out.au mcompand     ".1,.1 4:‐46,‐56,‐26,‐36,‐20,‐26,‐15,‐17,‐9,‐9" 80     ".1,.1 4:‐46,‐56,‐26,‐36,‐20,‐26,‐15,‐17,‐9,‐9" 3k     ".1,.1 4:‐46,‐56,‐26,‐36,‐20,‐26,‐15,‐17,‐9,‐9" 9k     ".1,.1 4:‐42,‐56,‐23,‐36,‐18,‐26,‐14,‐17,‐9,‐9"

              Real  Dolby A probably compands each channel separately but that
              is left as an exercise to interested readers.

              See also compand for a single‐band companding effect.

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

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

                 sox_ng 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_ng 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  re‐
              move  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_ng noisy.wav -n trim 0 1 noiseprof | play_ng noisy.wav noisered


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

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

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

       pad { length[@position(=)] }
              Pad the audio with silence, at the beginning, the  end,  or  any
              specified points through the audio.  length is the amount of si‐
              lence  to  insert  and  position the position in the input audio
              stream at which to insert it.  Any number of lengths  and  posi‐
              tions  may  be  specified, provided that a specified position is
              not less that the previous one, and any time  specification  may
              be  used  for them.  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 au‐
              dio,  specify  either  the end position or specify a zero‐length
              pad at the start.

              See also delay for an effect that can add silence at the  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 de‐
              scription 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  in‐
              struments,  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_ng snare.flac phaser 0.8 0.74 3 0.4 0.5 -t

              Gentler:

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

              A popular sound:

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

              More severe:

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

              flanger can do everything that phaser can do except for  gain‐in
              and gain‐out but works internally in floating point and interpo‐
              lates between samples so is slower but gives much less noisy re‐
              sults.

       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  de‐
              scription of the other parameters.

              See also the bend, speed, and tempo effects.

       rate [-q|-l|-m|-h|-v] [override‐options] RATE[k]
              Change  the audio sampling rate (i.e. resample the audio) to any
              given RATE (even non‐integer if this is supported by the  output
              file format) using a quality level defined as follows:
                           Quality   Band‐   Rej dB   Typical Use
                                     width
                     -q     quick     n/a    ≈30 @    playback on an‐
                                              Fs/4    cient hardware
                     -l      low      80%     100     playback on old
                                                      hardware
                     -m    medium     95%     100     audio playback
                     -h     high      95%     125     16‐bit mastering
                                                      (use with dither)
                     -v   very high   95%     175     24‐bit mastering

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

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

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

                 sox_ng input.wav -r 48k output.wav bass -b 24
                 sox_ng input.wav        output.wav bass -b 24 rate 48k

              though the second command is more flexible as it allows rate op‐
              tions  to  be  given, and allows the effects to be ordered arbi‐
              trarily.
                                    *        *        *

              Warning: technically detailed discussion follows.

              A user notes that resampling tracks and then concatenating  them
              is  more likely to create clicks at the joints than joining them
              first and resampling the result, due to edge effects.

              Override Options
              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 re‐
              sampler’s filter response; this  can  be  achieved  using  over‐
              ride options, as detailed in the following table:
              -M/-I/-L     Phase response = minimum/intermediate/linear
              -s           Steep filter (band‐width = 99%)
              -a           Allow aliasing/imaging above the pass‐band
              -b 74-99.7   Any band‐width %
              -p 0-100     Any phase response (0 = minimum, 25 = intermediate,
                           50 = linear, 100 = maximum)

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

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

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

              Minimum,  intermediate, or linear phase response is selected us‐
              ing 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 in‐
              creases to 85%.

              Examples:

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

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

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

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

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

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

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

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

                     An  out‐spec  contains comma‐separated input channel‐num‐
                     bers and hyphen‐delimited channel‐number ranges; alterna‐
                     tively, 0 may be given to create a silent output channel.
                     For example,

                        sox_ng 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 in‐
                     put file, and channel 4 is silent.  Whereas

                        sox_ng 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  num‐
                     bers to the left and right of the hyphen are optional and
                     default  to 1 and to the number of input channels respec‐
                     tively. Thus

                        sox_ng input.wav output.wav remix -

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

                     By default, where an output channel is mixed from  multi‐
                     ple (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  chan‐
                     nels 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_ng 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_ng 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  ad‐
                     justments, so

                        sox_ng 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 in‐
                     put.wav is stereo, then

                        sox_ng 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  con‐
                     stituent  channels  (in  order to perform subsequent pro‐
                     cessing 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_ng -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_ng "$1" "$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: in‐
                     put‐01.wav, input‐02.wav, ..., input‐06.wav.

                     See also the swap effect.

              repeat [count(1)|-]
                     Repeat the entire audio count times, or once if count  is
                     not given.  The special value - requests infinite repeti‐
                     tion.   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’ algo‐
                     rithm.  A reverberation effect is sometimes desirable for
                     concert  halls that are too small or contain so many peo‐
                     ple that the hall’s natural reverberance  is  diminished.
                     Applying  a small amount of stereo reverb to a (dry) mono
                     signal will usually make it sound more natural.  See  [3]
                     for a detailed description of reverberation.

                     This effect increases the volume of the audio and contin‐
                     ues  to  reverberate after the input finishes so, to pre‐
                     vent clipping and keep the  the  final  reverberation,  a
                     typical invocation might be:

                        play_ng dry.wav gain -3 pad 0 1 reverb

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

                        play_ng -m voice.wav "|sox_ng voice.wav -p reverse reverb -w reverse"

                     for a reverse reverb effect.

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

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

                     This effect supports the --plot global option.

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

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

                     The  above‐periods  value  is  used  to indicate if audio
                     should be trimmed at the beginning of the audio. A  value
                     of  zero  indicates no silence should be trimmed from the
                     beginning. When specifying a non‐zero  above‐periods,  it
                     trims audio up until it finds non‐silence. Normally, when
                     trimming  silence from beginning of audio the above‐peri‐
                     ods will be 1 but it can be increased to higher values to
                     trim all audio up to a specific count of non‐silence  pe‐
                     riods.  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 indicates 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 au‐
                     dio, you specify a below‐periods count.   In  this  case,
                     below‐period  means  to remove all audio after silence is
                     detected.  Normally, this will be a value 1 of but it can
                     be increased to skip over periods  of  silence  that  are
                     wanted.   For  example, if you have a song with 2 seconds
                     of silence in the middle and 2 second  at  the  end,  you
                     could  set  below‐period to a value of 2 to skip over the
                     silence in the middle of the audio.

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

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

                     To  remove  silence  from the middle of a file, specify a
                     below‐periods that  is  negative.   This  value  is  then
                     treated  as a positive value and is also used to indicate
                     that the effect should restart processing as specified by
                     the above‐periods, making it suitable for removing  peri‐
                     ods 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  re‐
                     move  long pauses between words but do not want to remove
                     the pauses completely.

                     duration is a time  specification  with  the  peculiarity
                     that  a bare number is interpreted as a sample count, not
                     as a number of seconds.  For specifying  seconds,  either
                     use the t suffix (as in ‘2t’) or specify minutes, too (as
                     in ‘0:02’).

                     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 dig‐
                     ital 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_ng 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]] [-r]]
                     Apply a sinc kaiser‐windowed low‐pass,  high‐pass,  band‐
                     pass,  or  band‐reject  filter to the signal.  The freqHP
                     and freqLP parameters give the  frequencies  of  the  6dB
                     points of a high‐pass and low‐pass filter that may be in‐
                     voked individually, or together.  If both are given, then
                     freqHP  less than freqLP creates a band‐pass filter, fre‐
                     qHP greater than freqLP  creates  a  band‐reject  filter.
                     For example, the invocations

                        sinc 3k
                        sinc ‐4k
                        sinc 3k‐4k
                        sinc 4k‐3k

                     create  a high‐pass, low‐pass, band‐pass, and band‐reject
                     filter respectively.

                     The default stop band attenuation of 120dB can  be  over‐
                     ridden with -a; alternatively, the kaiser window’s ‘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); alterna‐
                     tively, the number of filter taps can be  given  directly
                     with -n and is limited to the range of 11-32767.

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

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

                     The  -r  option  controls whether the filter should round
                     the number of taps to  the  closest  integer  instead  of
                     truncating it..

                     This effect supports the --plot global option.

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

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

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

                        sox_ng 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_ng 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_ng my.wav -n rate 6k spectrogram

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

                        sox_ng 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 im‐
                     age  includes  axes  legends etc. and so will be a little
                     larger than the specified spectrogram size.  In this  ex‐
                     ample:

                        sox_ng -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 similar 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  available  to  control  the  appearance
                     (colour‐set,  brightness, contrast, etc.) and filename of
                     the spectrogram; e.g. with

                        sox_ng 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 spec‐
                            trogram from its default value of 800 pixels to  a
                            given  number between 100 and 200000.  See also -X
                            and -d.

                     -X num X‐axis pixels/second; the default  is  auto‐calcu‐
                            lated  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 du‐
                            ration of the spectrogram.  num can be from 1 (low
                            time resolution) to 5000  (high  time  resolution)
                            and need not be an integer.  SoX may make a slight
                            adjustment  to  the  given  number  for processing
                            quantisation reasons; if so, SoX will  report  the
                            actual  number  used (viewable when the SoX global
                            option -V is in effect).  See also -x and -d.

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

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

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

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

                     -w name
                            Window: Hann (default), Hamming, Bartlett, Rectan‐
                            gular, Kaiser or Dolph.  The spectrogram  is  pro‐
                            duced  using  the Discrete Fourier Transform (DFT)
                            algorithm.  A significant parameter to this  algo‐
                            rithm  is the choice of ‘window function’.  By de‐
                            fault, 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‐resolu‐
                            tion),  select  a  Dolph window.  Kaiser, Bartlett
                            and Rectangular windows are also available.

                     -W num Window adjustment parameter.  This can be used  to
                            make small adjustments to the Kaiser or Dolph win‐
                            dow  shape.   A  positive  number  (up to ten) in‐
                            creases its dynamic range, a negative  number  de‐
                            creases 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  ex‐
                            pense of a little spectral loss.

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

                     -h     Selects a  high‐colour  palette  ‐  less  visually
                            pleasing  than  the default colour palette, but it
                            may make it easier to differentiate different lev‐
                            els.  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,  se‐
                            lects the permutation.

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

                     -a     Suppress  the  display of the axis lines.  This is
                            sometimes useful in helping to  discern  artefacts
                            at the spectrogram edges.

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

                     -A     Selects an alternative, fixed color set.  This  is
                            provided  only for compatibility with spectrograms
                            produced by another package.  It should  not  nor‐
                            mally  be used as it has some problems, not least,
                            a lack of differentiation at the bottom end  which
                            results in masking of low‐level artefacts.

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

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

                     -o file
                            Name of the spectrogram output PNG  file,  default
                            ‘spectrogram.png’.   If ‘‐’ is given, the spectro‐
                            gram will be sent to standard output (stdout).

                     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 (a time specifica‐
                            tion) fits the selected (or default) X‐axis width.
                            For example,

                               sox_ng 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 position(=)
                            Start  the  spectrogram  at the given point in the
                            audio stream.  For example

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

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

                     For  the  ability to perform off‐line processing of spec‐
                     tral data, see the stat effect.

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

                     Technically,  the  speed  effect  only changes the sample
                     rate information,  leaving  the  samples  themselves  un‐
                     touched.  The rate effect is invoked automatically to re‐
                     sample to the output sample rate, using its default qual‐
                     ity/speed.   For  higher  quality  or higher speed resam‐
                     pling, in addition to the speed effect, specify the  rate
                     effect with the desired quality option.

                     See also the bend, pitch, and tempo effects.

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

                     One  of  the options -h, -t, or -q may be given to select
                     the fade envelope as half‐cosine wave (the default), tri‐
                     angular (a.k.a. linear), or quarter‐cosine  wave  respec‐
                     tively.
                        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  per‐
                     forming  a  tape  splice,  the  end  of the section to be
                     spliced onto should be trimmed with a small  excess  (de‐
                     fault  0.005  seconds)  of  audio after the ideal joining
                     point.  The beginning of the audio section to  splice  on
                     should  be trimmed with the same excess (before the ideal
                     joining point), plus an additional leeway (default  0.005
                     seconds).   Any  time specification may be used for these
                     parameters.  SoX should then be invoked with the two  au‐
                     dio  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 ex‐
                     cess).

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

                           length1   excess
                         ‐‐‐‐‐‐‐‐‐‐‐><‐‐‐>
                         _________   :   :  _________________
                                  \  :   : :\     ‘
                                   \ :   : : \     ‘
                                    \:   : :  \     ‘
                                     *   : :   * ‐ ‐ *
                                      \  : :   :\     ‘
                                       \ : :   : \     ‘
                         _______________\: :   :  \_____‘____
                                           :   :   :     :
                                           <‐‐‐>   <‐‐‐‐‐>
                                           excess  leeway

                     where * indicates the joining points.

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

                        sox_ng too‐long.wav part1.wav trim 0 30.130

                     (5 ms excess, after the first verse starts)

                        sox_ng too‐long.wav part2.wav trim 1:03.422

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

                        sox_ng part1.wav part2.wav just‐right.wav splice 30.130

                     For another example, the SoX command

                        play_ng "|sox_ng -n -p synth 1 sin %1" "|sox_ng -n -p synth 1 sin %3"

                     generates and plays two notes, but there is a nasty click
                     at the transition; the click can be removed  by  splicing
                     instead  of  concatenating  the  audio, i.e. by appending
                     splice 1 to the command. (Clicks at the beginning and end
                     of the audio can be removed by preceding the  splice  ef‐
                     fect 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
                     # No chained time specifications allowed for the parameters
                     # (i.e. such that contain +/-).
                     e=0.005                      # Using default excess
                     l=$e                         # and leeway.
                     sox_ng "$1" piece.wav trim $2-$e-$l =$3+$e
                     sox_ng "$1" part1.wav trim 0 $4+$e
                     sox_ng "$1" part2.wav trim $4+$3-$2-$e-$l
                     sox_ng part1.wav piece.wav part2.wav "$5" \
                        splice $4+$e +$3-$2+$e+$l+$e

                     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,  ex‐
                     cess  would typically be an number of seconds, the -q op‐
                     tion would typically be given (to select an ‘equal power’
                     cross‐fade), and leeway should be zero (which is the  de‐
                     fault if -q is given).  For example, if f1.wav and f2.wav
                     are audio files to be cross‐faded, then

                        sox_ng f1.wav f2.wav out.wav splice -q $(soxi_ng -D f1.wav),3

                     cross‐fades  the  files where the point of equal loudness
                     is 3 seconds before the end of  f1.wav,  i.e.  the  total
                     length  of the cross‐fade is 2 × 3 = 6 seconds (Note: the
                     $(...) notation is POSIX shell).

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

                     The   information  is  output  to  the  ‘standard  error’
                     (stderr) stream and is calculated, where n is  the  dura‐
                     tion  of  the  audio in samples, c is the number of audio
                     channels, r is the audio sample rate, and  xk  represents
                     the  PCM value (in the range -1 to +1 by default) of each
                     successive sample in the audio, as follows:
                   Samples read        n×c
                   Length (seconds)    n÷r
                   Scaled by                                 See -s below.
                   Maximum amplitude   max(xk)               The maximum  sample
                                                             value in the audio;
                                                             usually  this  will
                                                             be a positive  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 au‐
                                                             dio.  If this  fig‐
                                                             ure   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  av‐
                                                             erage 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 maximum value of a 32‐bit signed integer).  In‐
                     ternal  effects always work with signed long PCM data and
                     so the value should relate to this fact.

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

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

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

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

                     See also the stats effect.

              stats [-b bits|-x bits|-s scale] [-w window‐time]
                     Display time domain statistical information about the au‐
                     dio  channels; audio is passed unmodified through the SoX
                     processing chain.  Statistics  are  calculated  and  dis‐
                     played  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 de‐
                     fault, 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  exam‐
                     ple,  for  16  bits, the scale would be -32768 to +32767.
                     The -x option behaves the same way as -b except that  the
                     signed  integer values are displayed in hexadecimal.  The
                     -s option scales the three measurements by a given float‐
                     ing‐point number.

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

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

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

                     The  right‐hand  Bit‐depth figure is the standard defini‐
                     tion of bit‐depth i.e. bits  less  significant  than  the
                     given  number are fixed at zero.  The left‐hand figure is
                     the number of most significant bits  that  are  fixed  at
                     zero  (or  one  for negative numbers) subtracted from the
                     right‐hand figure (the number subtracted is directly  re‐
                     lated to Pk lev dB).

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

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

                     See also the stat effect.

              swap   Swap stereo channels.  If the input is not stereo,  pairs
                     of  channels are swapped, and a possible odd last channel
                     passed through.  E.g., for seven channels, the output or‐
                     der will be 2, 1, 4, 3, 6, 5, 7.

                     See also remix for an effect that allows arbitrary  chan‐
                     nel selection and ordering (and mixing).

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

                     factor of stretching: >1 lengthen, <1  shorten  duration.
                     window  is  the length of the cross‐fading window in mil‐
                     liseconds with a default of 20.

                     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  fre‐
                     quency audio tones with various wave shapes, or to gener‐
                     ate wide‐band noise of various ‘colours’.  Multiple synth
                     effects  can  be  cascaded  to produce more complex wave‐
                     forms; 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 chan‐
                     nel  in a multi‐channel audio file can be synthesized in‐
                     dependently.

                     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  synthesized  audio  length,  the
                     number of channels, and the sampling rate; however, since
                     the  input  file’s  audio is not normally needed, a ‘null
                     file’ (with the special name -n) is often  given  instead
                     (and  the  length specified as a parameter to synth or by
                     another given effect that has an associated length).

                     For example, the following produces a 3‐second 48kHz  au‐
                     dio file containing a sine wave swept from 300 to 3300Hz:

                        sox_ng -n output.wav synth 3 sine 300-3300

                     and this produces an 8 kHz version:

                        sox_ng -r 8000 -n output.wav synth 3 sine 300-3300

                     Multiple  channels  can  be synthesized 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_ng -n output.wav synth 3 sine 300-3300 brownnoise

                     The  following example shows how two synth effects can be
                     cascaded to create a more complex waveform:

                        play_ng -n synth 0.5 sine 200-500 synth 0.5 sine fmod 700-100

                     The following could be used to help tune a guitar:

                        for n in E2 A2 D3 G3 B3 E4; do
                          play_ng -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 clip‐
                     ping when using the audio subsequently, so in many cases,
                     you will want to follow this effect with the gain  effect
                     to  prevent  this  from  happening.  (See  also  Clipping
                     above.)  Note that, by default, the synth effect incorpo‐
                     rates the functionality of gain -h (see the  gain  effect
                     for  details);  synth’s -n option may be given to disable
                     this behaviour.

                     A detailed description of each synth parameter follows:

                     len is the length of audio to synthesize (any time speci‐
                     fication); a value  of  0  indicated  to  use  the  input
                     length, which is also the default.

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

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

                     freq/freq2  are  the  frequencies at the beginning/end of
                     synthesis in Hz or, if preceded with ‘%’, semitones rela‐
                     tive to A (440 Hz); alternatively, ‘scientific’ note  no‐
                     tation  (e.g.  E2) may be used.  The default frequency is
                     440Hz.  By default, the tuning used with the  note  nota‐
                     tions  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  fre‐
                     quencies.  The two given frequencies must be separated by
                     one  of the characters ‘:’, ‘+’, ‘/’, or ‘-’.  This char‐
                     acter 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  num‐
                            ber 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; de‐
                     fault=0.

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

                     p1 is the percentage of each cycle that is ‘on’ (square),
                     or   ‘rising’   (triangle,  exp,  trapezium);  default=50
                     (square, triangle, exp), default=10 (trapezium), or  sus‐
                     tain (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);  de‐
                     fault=20.

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

              tempo [-q] [-m|-s|-l] factor [segment [search [overlap]]]
                     Change  the  audio playback speed but not its pitch. This
                     effect uses the WSOLA algorithm. The audio is chopped  up
                     into  segments  which are then shifted in the time domain
                     and overlapped (cross‐faded) at points where their  wave‐
                     forms  are  most  similar as determined by measurement of
                     ‘least squares’.

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

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

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

                     The -l option is used to optimize default values of  seg‐
                     ment,  search  and  overlap  for ‘linear’ processing that
                     tends to cause more noticeable distortion but may be use‐
                     ful when factor is close to 1.

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

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

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

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

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

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

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

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

              trim {position(+)}
                     Cuts  portions out of the audio.  Any number of positions
                     may be given; audio is not sent to the output  until  the
                     first  position  is  reached.  The effect then alternates
                     between copying and discarding audio  at  each  position.
                     Using  a  value of 0 for the first position parameter al‐
                     lows copying from the beginning of the audio.

                     For example,

                        sox_ng infile outfile trim 0 10

                     will copy the first ten seconds, while

                        play_ng infile trim 12:34 =15:00 ‐2:00

                     and

                        play_ng infile trim 12:34 2:26 ‐2:00

                     will both play from 12 minutes 34 seconds into the  audio
                     up  to  15  minutes into the audio (i.e. 2 minutes and 26
                     seconds long), then resume playing two minutes before the
                     end of audio.

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

                     For a general resampling effect  with  anti‐imaging,  see
                     rate.  See also downsample.

              vad [options]
                     Voice  Activity  Detector.   Attempts to trim silence and
                     quiet background sounds from the  ends  of  (fairly  high
                     resolution  i.e.  16‐bit, 44-48kHz) recordings of speech.
                     The algorithm currently uses a simple cepstral power mea‐
                     surement 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_ng speech.wav norm vad

                     to trim from the front,

                        play_ng speech.wav norm reverse vad reverse

                     to trim from the back, and

                        play_ng speech.wav norm vad reverse vad reverse

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

                     Options:
                     Default values are shown in parenthesis.

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

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

                     -s num (1)
                            The amount of audio (in  seconds)  to  search  for
                            quieter/shorter  bursts  of audio to include prior
                            to the detected trigger point.

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

                     -p num (0)
                            The amount of audio (in seconds) to  preserve  be‐
                            fore   the   trigger  point  and  any  found  qui‐
                            eter/shorter bursts.

                     Advanced Options:
                     These allow fine tuning of the algorithm’s internal para‐
                     meters.

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

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

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

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

                     -f num Frequency of the  algorithm’s  processing/measure‐
                            ments.

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

                     -M num Time constant used  to  smooth  spectral  measure‐
                            ments.

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

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

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

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

                     See also the silence effect.

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

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

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

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

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

                     Beware of Clipping when the increasing the volume.

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

                     An  optional  limiter-gain  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 spec‐
                     ifying  this  parameter will cause no limiter to be used.
                     In verbose mode, this effect will display the  percentage
                     of the audio that needed to be limited.

                     See also gain for a volume‐changing effect with different
                     capabilities,  and  compand  for a dynamic‐range compres‐
                     sion/expansion/limiting 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‐ngs@groups.io).

SEE ALSO
       soxi_ng(1), soxformat_ng(7), libsox_ng(3)
       audacity(1), gnuplot(1), octave(1), wget(1)
       The SoX web site at https://sox_ng.codeberg.page
       SoX scripting examples at
       https://codeberg.org/sox_ng/sox_ng/src/branch/main/scripts

   References
       [1]    R. Bristow‐Johnson, Cookbook formulae for audio EQ biquad filter
              coefficients,
              http://web.archive.org/web/20100210031754/http://mu‐
              sicdsp.org/files/Audio‐EQ‐Cookbook.txt

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

       [3]    Scott Lehman, Effects Explained, https://codeberg.org/sox_ng/Ef‐
              fects‐Explained

       [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://web.archive.org/web/20100106120257/http://www.ladspa.org/cmt

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

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

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

       This  program  is  distributed  in the hope that it will be useful, but
       WITHOUT ANY  WARRANTY;  without  even  the  implied  warranty  of  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_ng                         December 31, 2014                        SoX(1)
