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, <mu>-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 if some was present in the input file.  In
       this  case,  if anything other than a simple format conversion was per-
       formed then the output file Replay Gain information is likely to be in-
       correct  and  so should be recalculated using a tool that supports this
       (not SoX).

       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
       driver, e.g. ALSA & OSS, or SUNAU & AO.  Systems  can  also  have  more
       than  one  audio  device (a.k.a. `sound card').  If more than one audio
       driver has been built-in to SoX, and the default selected by  SoX  when
       recording  or  playing  is  not the one that is wanted, then the AUDIO-
       DRIVER environment variable can be used to override the  default.   For
       example (on many systems):

          set AUDIODRIVER=oss
          play_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 ef-
       fect, and finally creates the output MP3 file by re-compressing the au-
       dio  -  with a possible reduction in fidelity above that which occurred
       when the input file was created.  Hence, if what is ultimately  desired
       is  lossily  compressed  audio, it is highly recommended to perform all
       audio processing using lossless file formats and then  convert  to  the
       lossy format only at the final stage.

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

   Dithering
       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:

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

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

       o   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 de-
       livered.

       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 other-
       wise unbalanced then the input file volumes can be set manually as  de-
       scribed 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/<sqrt>n instead of ^1/n.  Note that this bal-
       ancing 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 custom-
       ized by placing a %n anywhere in the filename where the  number  should
       be  substituted.  An optional number can be placed after the % to indi-
       cate a minimum fixed width for the number.

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

       The following is an example of splitting the first 60 seconds of an 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/<mu>-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. <mu>-law.  It has a
                     precision equivalent to roughly 14-bit PCM and  is  some-
                     times  encoded with reversed bit-ordering (see the -X op-
                     tion).

              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 param-
       eters  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  1/2  seconds.   The t suffix is entirely optional (however,
              see the silence effect for an exception).  Note that the  compo-
              nent  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 le-
              gal 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.

              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


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

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

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

                                    *        *        *

              An alternative approach to removing a DC offset (albeit  with  a
              short delay) is to use the highpass filter effect at a frequency
              of say 10Hz, as illustrated in the following example:

                 sox_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
              bandlimited baseband signal, aliasing will occur.  This  may  be
              desirable, e.g., for frequency translation.

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

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

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

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

                 play_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".

              If  found,  the environment variable LADSPA_PATH will be used as
              search path for plugins.

       loudness [gain [reference]]
              Loudness control - similar to  the  gain  effect,  but  provides
              equalisation    for    the    human    auditory   system.    See
              http://en.wikipedia.org/wiki/Loudness for a detailed description
              of  loudness.   The gain is adjusted by the given gain parameter
              (usually negative) and the signal 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


       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.

              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-numbers and
              hyphen-delimited channel-number ranges; alternatively, 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 input 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 numbers to
              the left and right of the hyphen are optional and default  to  1
              and to the number of input channels respectively. Thus

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

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

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

                 sox_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 adjust-
              ments, 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  input.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/<sqrt>n (`power') scaling; this  gives  a  louder
              mix but one that might occasionally clip.

                                    *        *        *

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

              #!/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:  input-01.wav,  in-
              put-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  repetition.   Re-
              quires  temporary  file space to store the audio to be repeated.
              Note that repeating once yields two copies: the  original  audio
              and the repeated audio.

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

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

              This  effect  increases the volume of the audio and continues to
              reverberate after the input finishes so, to prevent 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' signal, thus
              allowing it to be processed further, independently 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-periods  will  be 1 but it can be increased to higher
              values to trim all audio up to a specific count  of  non-silence
              periods.  For  example,  if you had an audio file with two songs
              that each contained 2 seconds of silence before  the  song,  you
              could specify an above-period of 2 to strip out both silence pe-
              riods 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  in-
              creasing  the duration, burst of noise can be treated as silence
              and trimmed off.

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

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

              Unfortunately, you must know the length of the  silence  at  the
              end  of  your  audio file to trim off silence reliably.  A work-
              around is to use the silence effect in combination with the  re-
              verse  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-pe-
              riods  that  is negative.  This value is then treated as a posi-
              tive value and is also used to indicate that the  effect  should
              restart  processing as specified by the above-periods, making it
              suitable for removing periods of silence in the  middle  of  the
              audio.

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

              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 digital silence).

              The following example shows how this effect can be used to start
              a recording that does not contain the delay at the  start  which
              usually  occurs  between  `pressing  the  record button' and the
              start of the performance:

                 rec_ng parameters filename other-effects silence 1 5 2%


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

                 sinc 3k
                 sinc -4k
                 sinc 3k-4k
                 sinc 4k-3k

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

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

              The default transition band-width of 5% of the total band can be
              overridden with -t (and tbw in Hertz); alternatively, the number
              of filter taps can be given directly with -n and is  limited  to
              the range of 11-32767.

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

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

              The  -r option controls whether the filter should round the num-
              ber 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  unmodi-
              fied  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 repre-
              sented by the colour (or optionally the intensity) of the pixels
              in  the  X-Y plane.  If the audio signal contains multiple chan-
              nels then these are shown from top to bottom starting from chan-
              nel 1 (which is the left channel for stereo audio).

              For example, if `my.wav' is a stereo file, then with

                 sox_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 sen-
              sitive.  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 image includes axes legends
              etc. and so will be a little larger than the specified  spectro-
              gram size.  In this example:

                 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 simi-
              lar  example, append the following to the `chime' command in the
              description of the delay effect (above):

                 rate 2k spectrogram -X 200 -Z -10 -w kaiser

              Options are also available to control  the  appearance  (colour-
              set,  brightness,  contrast,  etc.) and filename of the spectro-
              gram; 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 spectrogram
                     from its default value of 800 pixels to  a  given  number
                     between 100 and 200000.  See also -X and -d.

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

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

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

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

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

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

              -w name
                     Window:  Hann  (default), Hamming, Bartlett, Rectangular,
                     Kaiser or Dolph.  The spectrogram is produced  using  the
                     Discrete  Fourier  Transform (DFT) algorithm.  A signifi-
                     cant parameter to this algorithm is the choice of `window
                     function'.   By  default,  SoX uses the Hann window which
                     has good all-round frequency-resolution and dynamic-range
                     properties.   For  better frequency resolution (but lower
                     dynamic-range), select a Hamming window; for  higher  dy-
                     namic-range  (but  poorer frequency-resolution), 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 window shape.  A
                     positive  number (up to ten) increases its dynamic range,
                     a negative number decreases it.

              -s     Allow slack overlapping of DFT  windows.   This  can,  in
                     some cases, increase image sharpness and give greater ad-
                     herence to the -x value, but at the expense of  a  little
                     spectral loss.

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

              -h     Selects  a  high-colour  palette - less visually pleasing
                     than the default colour palette, but it may make it  eas-
                     ier to differentiate different levels.  If this option is
                     used in conjunction with -m, the result will be a  hybrid
                     monochrome/colour palette.

              -p num Permute  the  colours in a colour or hybrid palette.  The
                     num parameter, from 1 (the default)  to  6,  selects  the
                     permutation.

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

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

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

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

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

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

              -o file
                     Name of the spectrogram output PNG file,  default  `spec-
                     trogram.png'.   If  `-' is given, the spectrogram 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 specification) 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 au-
                     dio, 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, receives the  entire
                     audio stream).

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

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

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

              See also the bend, pitch, and tempo effects.

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

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

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

              To perform a splice, first use the trim effect to select the au-
              dio sections to be joined together.  As when performing  a  tape
              splice,  the  end  of  the  section to be spliced onto should be
              trimmed with a small excess (default 0.005 seconds) of audio af-
              ter 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  parame-
              ters.  SoX should then be invoked with the two audio sections as
              input files and the splice effect given  with  the  position  at
              which  to perform the splice - this is length of the first audio
              section (including the excess).

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

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

              where * indicates the joining points.

              For example, a long song begins with two verses which start  (as
              determined  e.g.  by  using  the  play_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 con-
              catenating the audio, i.e. by appending splice 1 to the command.
              (Clicks at the beginning and end of the audio can be removed  by
              preceding the splice effect with fade q .01 2 .01).

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

              #!/bin/sh
              # Audio Copy and Paste Over
              # acpo infile copy-start copy-stop paste-over-start outfile
              # 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, excess would typi-
              cally  be an number of seconds, the -q option would typically be
              given (to select an `equal power' cross-fade), and leeway should
              be  zero (which is the default if -q is given).  For example, if
              f1.wav and f2.wav are audio files to be cross-faded, then

                 sox_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 x 3 = 6 seconds (Note: the  $(...)  notation  is
              POSIX shell).

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

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

         Samples read        nxc
         Length (seconds)    n/r
         Scaled by                                              See -s below.
         Maximum amplitude   max(xk)                            The maximum  sample
                                                                value in the audio;
                                                                usually  this  will
                                                                be  a positive num-
                                                                ber.
         Minimum amplitude   min(xk)                            The minimum  sample
                                                                value in the audio;
                                                                usually  this  will
                                                                be  a negative num-
                                                                ber.
         Midline amplitude   1/2min(xk)+1/2max(xk)
         Mean norm           ^1/n<Sigma>|xk|                    The average of  the
                                                                absolute  value  of
                                                                each sample in  the
                                                                audio.
         Mean amplitude      ^1/n<Sigma>xk                      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       <sqrt>(^1/n<Sigma>xk^2)            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-1<Sigma>|xk-xk-1|
         RMS delta           <sqrt>(^1/n-1<Sigma>(xk-xk-1)^2)
         Rough frequency                                        In Hz.
         Volume Adjustment                                      The  parameter   to
                                                                the    vol   effect
                                                                which  would   make
                                                                the  audio  as loud
                                                                as possible without
                                                                clipping.     Note:
                                                                See the  discussion
                                                                on  Clipping  above
                                                                for reasons why  it
                                                                is  rarely  a  good
                                                                idea actually to do
                                                                this.

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

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

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

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

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

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

              See also the stats effect.

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

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

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

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

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

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

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

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

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

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

              See also the stat effect.

       swap   Swap stereo channels.  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 order will be  2,
              1, 4, 3, 6, 5, 7.

              See  also  remix for an effect that allows arbitrary channel se-
              lection and ordering (and mixing).

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

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

              See also the tempo effect.

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

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

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

                 sox_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  cas-
              caded to create a more complex waveform:

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

              Frequencies can also be given in `scientific' note notation, or,
              by prefixing a `%' character, as a number of semitones  relative
              to  `middle  A'  (440 Hz).   For example, the following could be
              used to help tune a guitar's low `E' string:

                 play_ng -n synth 4 pluck %-29

              or with a (Bourne shell) loop, the whole guitar:

                 for n in E2 A2 D3 G3 B3 E4; do
                   play_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 clipping  when  using
              the  audio subsequently, so in many cases, you will want to fol-
              low this effect with the gain effect to prevent this  from  hap-
              pening.  (See  also Clipping above.)  Note that, by default, the
              synth effect incorporates the functionality of gain -h (see  the
              gain effect for details); synth's -n option may be given to dis-
              able this behaviour.

              A detailed description of each synth parameter follows:

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

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

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

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

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

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

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

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

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

              Not used for noise.

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

              See  also  speed  for an effect that changes tempo and pitch to-
              gether, pitch and bend for effects that change pitch  only,  and
              stretch for an effect that changes tempo using a different algo-
              rithm.

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

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

       trim {position(+)}
              Cuts  portions out of the audio.  Any number of positions may be
              given; audio is not sent to the output until the first  position
              is reached.  The effect then alternates between copying and dis-
              carding audio at each position.  Using a  value  of  0  for  the
              first  position  parameter  allows 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-value
              samples  are  inserted between each pair of input samples.  As a
              result, the original spectrum is replicated into  the  new  fre-
              quency  space (imaging) and attenuated.  This attenuation can be
              compensated for by adding vol factor after any further  process-
              ing.   The upsample effect is typically used in combination with
              filtering effects.

              For a general resampling effect  with  anti-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 measurement to detect voice, so may
              be  fooled  by  other  things, especially music.  The effect can
              trim only from the front of the audio, so in order to trim  from
              the back, the reverse effect must also be used.  E.g.

                 play_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 recom-
              mended, but remember that neither reverse nor norm  is  suitable
              for use with streamed audio.

              Options:
              Default values are shown in parenthesis.

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

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

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

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

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

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

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

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

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

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

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

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

              -M num Time constant used to smooth spectral measurements.

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

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

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

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

              See also the silence effect.

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

              The amount to change the volume is given by gain which is inter-
              preted,  according to the given type, as follows: if type is am-
              plitude (or is omitted), then gain is an amplitude (i.e. voltage
              or  linear) ratio, if power, then a power (i.e. wattage or volt-
              age-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 in-
              creases it; a negative gain inverts the audio signal in addition
              to adjusting its volume.

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

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

              Beware of Clipping when the increasing the volume.

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

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

              See also gain for a volume-changing effect with different  capa-
              bilities,  and  compand  for  a dynamic-range compression/expan-
              sion/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.ar-
              chive.org/web/20100210031754/http://musicdsp.org/files/Audio-EQ-
              Cookbook.txt

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

       [3]    Scott     Lehman,     Effects     Explained,     https://web.ar-
              chive.org/web/20100102125223/http://harmony-central.com/Ef-
              fects/effects-explained.html

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

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

       [6]    Richard   Furse,   Computer   Music   Toolkit,    http://web.ar-
              chive.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)
