SoX(1)                                                     Sound eXchange                                                    SoX(1)

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

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

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

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

       SoX  reads  and writes audio files in most popular formats and can optionally apply effects to them. It can combine multiple
       input sources, synthesise audio, and, on many systems, act as  a  general  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 command.  To simplify playing and recording audio, if SoX is invoked
       as play, the output file is automatically set to be the default sound device, and if invoked as rec, the default  sound  de‐
       vice  is  used  as  an  input  source.  Additionally, the soxi(1) command provides a convenient way to just query audio file
       header information.

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

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

                                                           *        *        *

       The overall SoX processing chain can be summarised as follows:

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

       Note however, that on the SoX command line, the positions of the Output(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 oppo‐
       site is true for effects.  To show how this works in practice, here is a selection of examples of how  SoX  might  be  used.
       The simple
          sox recital.wav
       translates an audio file in Sun AU format to a Microsoft WAV file, whilst
          sox -b 16 recital.wav channels 1 rate 16k fade 3 norm
       performs  the  same format translation, but also applies four effects (down-mix to one channel, sample rate change, fade-in,
       nomalize), and stores the result at a bit-depth of 16.
          sox -r 16k -e signed -b 8 -c 1 voice-memo.raw voice-memo.wav
       converts `raw' (a.k.a. `headerless') audio to a self-describing file format,
          sox slow.aiff fixed.aiff speed 1.027
       adjusts audio speed,
          sox short.wav long.wav longer.wav
       concatenates two audio files, and
          sox -m music.mp3 voice.wav mixed.flac
       mixes together two audio files.
          play "The Moonbeams/Greatest/*.ogg" bass +3
       plays a collection of audio files whilst applying a bass boosting effect,
          play -n -c1 synth sin %-12 sin %-9 sin %-5 sin %-2 fade h 0.1 1 0.1
       plays a synthesised `A minor seventh' chord with a pipe-organ sound,
          rec -c 2 radio.aiff trim 0 30:00
       records half an hour of stereo audio, and
          play -q take1.aiff & rec -M take1.aiff take1-dub.aiff
       (with POSIX shell and where supported by hardware) records a new track in a multi-track recording.  Finally,
          rec -r 44100 -b 16 -e signed-integer -p \
            silence 1 0.50 0.1% 1 10:00 0.1% | \
            sox -p song.ogg silence 1 0.50 0.1% 1 2.0 0.1% : \
            newfile : restart
       records a stream of audio such as LP/cassette and splits in to multiple audio files at points with  2  seconds  of  silence.
       Also, it does not start recording until it detects audio is playing and stops after it sees 10 minutes of silence.

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

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

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

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

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

       data encoding
              The way in which each audio sample is represented (or `encoded').  Some encodings have variants with different  byte-
              orderings  or  bit-orderings.   Some  compress the audio data 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.   Com‐
              monly-used encoding types include floating-point, μ-law, ADPCM, signed-integer PCM, MP3, and FLAC.

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

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

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

       One important use of audio file comments is to convey `Replay Gain' information.  SoX supports applying Replay Gain informa‐
       tion  (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 comments to output files that support comments, so output files may contain Replay  Gain  in‐
       formation  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 incorrect and so should be  recalculated  using  a  tool
       that supports this (not SoX).

       The soxi(1) command can be used to display information from audio file headers.

   Determining & Setting The File Format
       There  are several mechanisms available for SoX to use to determine or set the format characteristics of an audio file.  De‐
       pending on the circumstances, 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 supported by the output file type.

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

   Playing & Recording Audio
       The play and rec commands are provided so that basic playing and recording is as simple as
          play existing-file.wav
          rec new-file.wav
       These two commands are functionally equivalent to
          sox existing-file.wav -d
          sox -d new-file.wav
       Of course, further options and effects (as described below) can be added to the commands in either form.

                                                           *        *        *

       Some  systems provide more than one type of (SoX-compatible) audio driver, e.g. ALSA & OSS, or SUNAU & AO.  Systems can also
       have more than one audio device (a.k.a. `sound card').  If more than one audio driver has been built-in to SoX, and the  de‐
       fault selected by SoX when recording or playing is not the one that is wanted, then the AUDIODRIVER environment variable can
       be used to override the default.  For example (on many systems):
          set AUDIODRIVER=oss
          play ...
       The AUDIODEV environment variable can be used to override the default audio device, e.g.
          set AUDIODEV=/dev/dsp2
          play ...
          sox ... -t oss
          set AUDIODEV=hw:soundwave,1,2
          play ...
          sox ... -t alsa
       Note that the way of setting environment variables varies from system to system - for some specific examples, see `SOX_OPTS'

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

                                                           *        *        *

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

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

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

       Formats that discard audio signal information are called `lossy'.  Formats 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 sam‐
       pling 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 for‐
       mat to (8-bit) A-law isn't.

       N.B.  SoX converts all audio files to an internal uncompressed format before performing any  audio  processing.  This  means
       that manipulating a file that is stored in a lossy format can cause further losses in audio fidelity.  E.g. with
          sox long.mp3 short.mp3 trim 10
       SoX  first decompresses the input MP3 file, then applies the trim effect, and finally creates the output MP3 file by re-com‐
       pressing the audio - with a possible reduction in fidelity above that which  occurred  when  the  input  file  was  created.
       Hence,  if  what is ultimately desired is lossily compressed audio, it is highly recommended to perform all audio processing
       using lossless file formats and then convert to the lossy format only at the final stage.

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

       Dithering is a technique used to maximise the dynamic range of audio stored at a particular bit-depth. Any distortion intro‐
       duced by quantisation is decorrelated by adding a small amount of white noise to the signal.  In most cases, SoX can  deter‐
       mine whether the selected processing requires dither and will add it during output formatting if appropriate.

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

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

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

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

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

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

       Clipping is distortion that occurs when an audio signal level (or `volume') 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 process‐
       ing 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 analogue components can introduce a small DC offset
       and/or amplification, all of which can produce distortion if the audio signal level was initially too close to the  clipping

       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 mu‐
       sic production; in fact, many CDs, MP3s, etc.  are now mastered at levels above 0dBFS i.e. the audio is  clipped  as  deliv‐

       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 clipping, e.g.
          sox dull.wav bright.wav gain -6 treble +6
       guarantees that the treble boost will not clip.

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

       See also -G and the gain and norm effects.

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

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

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

       The `sequence' combining method is selected automatically for play.  It is similar to `concatenate' in that the  audio  from
       each  input file is sent serially to the output file. However, here the output file may be closed and reopened at the corre‐
       sponding 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 corresponding 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

       When combining input files, SoX applies any specified effects (including, for example, the vol volume adjustment effect) af‐
       ter the audio has been combined. However, it is often useful to be able to set the volume of (i.e. `balance') the inputs in‐
       dividually, 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 adjustments that have been selected (either manually or  au‐

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

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

       If  mixed audio seems loud enough at some points 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 balancing using a factor of ¹/√n instead of ¹/n.  Note that this balancing factor does not guarantee that clip‐
       ping 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 multi‐
       ple output mode.

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

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

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

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

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

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

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

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

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

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

       -      SoX can be used in simple pipeline operations by using the special filename `-' which, if used as an input  filename,
              will  cause  SoX  will read audio data from `standard input' (stdin), and which, if used as the output filename, will
              cause SoX will send audio data to `standard output' (stdout).  Note that when using this option for the output  file,
              and sometimes when using it for an input file, the file-type (see -t below) must also be 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 formatted signals to its standard output, then the following command makes a stereo file from two
              generated signals:
                 sox -M "|genw --imd -" "|genw --thd -" out.wav
              For headerless (raw) audio, -t (and perhaps other format options) will need to be given, preceding the input command.

              Specifies that filename `globbing' (wild-card matching) should be performed by SoX instead of by the shell.  This al‐
              lows  a single set of file options to be applied to a group of files.  For example, if the current directory contains
              three `vox' files, file1.vox, file2.vox, and file3.vox, then
                 play --rate 6k *.vox
              will be expanded by the `shell' (in most environments) to
                 play --rate 6k file1.vox file2.vox file3.vox
              which will treat only the first vox file as having a sample rate of 6k.  With
                 play --rate 6k "*.vox"
              the given sample rate option will be applied to all three vox files.

       -p, --sox-pipe
              This can be used in place of an output filename to specify that the SoX command should be used as in  input  pipe  to
              another SoX command.  For example, the command:
                 play "|sox -n -p synth 2" "|sox -n -p synth 2 tremolo 10" stat
              plays two `files' in succession, each with different effects.

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

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

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

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

              Using a null file to output audio amounts to discarding the audio and is useful mainly with effects that produce  in‐
              formation 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 overrid‐
              den if desired using command-line format options (see below).

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

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

       The SOX_OPTS environment variable can be used to provide alternative default values for SoX's global options.  For example:
          SOX_OPTS="--buffer 20000 --play-rate-arg -hs --temp /mnt/temp"
       Note that setting SOX_OPTS can potentially create unwanted changes in the behaviour of scripts or other programs that invoke
       SoX.  SOX_OPTS might best be used for things (such as in the given example) that reflect the environment in which SoX is be‐
       ing run.  Enabling options such as --no-clobber as default might be handled better using a shell alias since a  shell  alias
       will not affect operation in scripts etc.

       One  way to ensure that a script cannot be affected by SOX_OPTS is to clear SOX_OPTS at the start of the script, but this of
       course loses the benefit of SOX_OPTS carrying some system-wide default options.  An alternative approach  is  to  explicitly
       invoke SoX with default option values, e.g.
          SOX_OPTS="-V --no-clobber"
          sox -V2 --clobber $input $output ...
       Note that the way to set environment variables varies from system to system. Here are some examples:

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

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

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

       --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 different 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 converted from 16 to 24 bit with the intention of doing some processing 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 convert‐
              ing 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 -G infile -b 16 outfile rate 44100 dither -s
              is shorthand for
                 sox infile -b 16 outfile gain -h rate 44100 gain -rh dither -s
              See also -V, --norm, and the gain effect.

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

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

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

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

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

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

       --multi-threaded | --single-threaded
              By  default,  SoX  is `single threaded'.  If the --multi-threaded option is given however then SoX will process audio
              channels for most multi-channel effects in parallel on hyper-threading/multi-core architectures. This may reduce pro‐
              cessing  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 benefit from multi-threaded processing (e.g. 131072; see --buffer above).

              Prompt before overwriting an existing file with the same name as that given for the output file.

              N.B.  Unintentionally overwriting a file is easier than you might think, for example, if you accidentally enter
                 sox file1 file2 effect1 effect2 ...
              when what you really meant was
                 play file1 file2 effect1 effect2 ...
              then, without this option, file2 will be overwritten.  Hence, using this option is recommended. SOX_OPTS  (above),  a
              `shell' alias, script, or batch file may be an appropriate way of permanently enabling it.

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

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

       --play-rate-arg ARG
              Selects  a  quality option to be used when the `rate' effect is automatically invoked whilst playing audio.  This op‐
              tion is typically 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  gnu‐
              plot  program or the GNU Octave program, to assist with the selection and configuration of many of the transfer-func‐
              tion based effects.  For the first given effect that supports the selected plotting program, SoX will output commands
              to plot the effect's transfer function, and then exit without actually processing any audio.  E.g.
                 sox --plot octave input-file -n highpass 1320 > highpass.plt
                 octave highpass.plt

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

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

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

       -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 chan‐
              nels and is calibrated for digital 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  permis‐
              sion  or free-space problems with the default location. In this case, using `--temp .' (to use the current directory)
              is often a good solution.

              Show SoX's version number and exit.

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

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

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

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

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

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

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

              By default, the verbosity level is set to 2 (shows errors and warnings). Each occurrence of the -V  option  increases
              the verbosity level by 1.  Alternatively, the verbosity level can be set to an absolute number by specifying it imme‐
              diately after the -V, e.g.  -V0 sets it to 0.

   Input File Options
       These options apply only to input files and may precede only input filenames on the command line.

              Override an (incorrect) audio length given in an audio file's header. If this option is  given  then  SoX  will  keep
              reading audio until it reaches the end of the input file.

       -v, --volume FACTOR
              Intended  for  use when combining multiple input files, this option adjusts the volume of the file that follows it on
              the command line by a factor of FACTOR. This allows it to be `balanced' w.r.t. the other input files.  This is a lin‐
              ear  (amplitude)  adjustment,  so a number less than 1 decreases the volume and a number greater than 1 increases it.
              If a negative number is given then in addition to the volume adjustment, the audio signal will be inverted.

              See also the norm, vol, and gain effects, and see Input File Balancing above.

   Input & Output File Format Options
       These options apply to the input or output file whose name they immediately 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 en‐
              codings such as MP3 or GSM.  Not necessary with encodings that have a fixed number of bits, e.g.  A/μ-law, ADPCM.

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

              For an output file, this option can be used (perhaps along with -e) to set the  output  encoding  size.   By  default
              (i.e. if this option is not given), the output encoding size will (providing it is supported by the output file type)
              be set to the input encoding size.  For example
                 sox input.cdda -b 24 output.wav
              converts raw CD digital audio (16-bit, signed-integer) to a 24-bit (signed-integer) `WAV' file.

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

              For an input file, the most common use for this option is to inform SoX of the number of channels in a `raw'  (`head‐
              erless')  audio file.  Occasionally, it may be useful to use this option with a `headered' file, in order to override
              the (presumably incorrect) value in the header - note that this is only supported with certain file types.  Examples:
                 sox -r 48k -e float -b 32 -c 2 input.raw output.wav
              converts a particular `raw' file to a self-describing `WAV' file.
                 play -c 1 music.wav
              interprets the file data as belonging to a single channel regardless of what is indicated in the file  header.   Note
              that if the file does in fact have two channels, this will result in the file playing at half speed.

              For an output file, this option provides a shorthand for specifying that the channels effect should be invoked in or‐
              der to change (if necessary) the number of channels in the audio signal to the number given.  For example,  the  fol‐
              lowing two commands are equivalent:
                 sox input.wav -c 1 output.wav bass -b 24
                 sox input.wav      output.wav bass -b 24 channels 1
              though the second form is more flexible as it allows the effects to be ordered arbitrarily.

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

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

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

                     PCM data stored as IEEE 753 single precision (32-bit) or double  precision  (64-bit)  floating-point  (`real')
                     numbers.  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 encoding to 8 bits per sample.  A.k.a. μ-law.  It has a pre‐
                     cision  equivalent  to  roughly 14-bit PCM and is sometimes encoded with reversed bit-ordering (see the -X op‐

                     OKI (a.k.a. VOX, Dialogic, or Intel) 4-bit ADPCM; it has a precision equivalent to roughly 12-bit PCM.   ADPCM
                     is a form of audio compression that has a good compromise between audio quality and encoding/decoding speed.

                     IMA (a.k.a. DVI) 4-bit ADPCM; it has a precision equivalent to roughly 13-bit PCM.

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

                     GSM is currently used for the vast majority of the world's digital wireless telephone calls.  It utilises sev‐
                     eral audio formats with different bit-rates and associated speech quality.  SoX has support for GSM's original
                     13kbps `Full Rate' audio format.  It is usually CPU-intensive to work with GSM audio.

              Encoding  names  can  be abbreviated where this would not be ambiguous; e.g. `unsigned-integer' can be given as `un',
              but not `u' (ambiguous with `u-law').

              For an input file, the most common use for this option is to inform SoX of the encoding of a `raw' (`headerless') au‐
              dio file (see the examples in -b and -c above).

              For an output file, this option can be used (perhaps along with -b) to set the output encoding type  For example
                 sox input.cdda -e float output1.wav

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

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

              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-seconds.wav' and `five*.wav', then
                 play --no-glob "five*.wav"
              can be used to play just the single file `five*.wav'.

       -r, --rate RATE[k]
              Gives the sample rate in Hz (or kHz if appended with `k') of the file.

              For an input file, the most common use for this option is to inform SoX of the sample rate of a `raw'  (`headerless')
              audio file (see the examples in -b and -c above).  Occasionally it may be useful to use this option with a `headered'
              file, in order to override the (presumably incorrect) value in the header - note that this  is  only  supported  with
              certain  file types.  For example, if audio was recorded with a sample-rate of say 48k from a source that played back
              a little, say 1.5%, too slowly, then
                 sox -r 48720 input.wav output.wav
              effectively corrects the speed by changing only the file header (but see also the speed effect for the more usual so‐
              lution to this problem).

              For  an  output file, this option provides a shorthand for specifying 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 com‐
              mands are equivalent:
                 sox input.wav -r 48k output.wav bass -b 24
                 sox input.wav        output.wav bass -b 24 rate 48k
              though  the  second form is more flexible as it allows rate options to be given, and allows the effects to be ordered

       -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 file‐
              name extension.  For example:
                 another-command | sox -t mp3 - output.wav

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

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

       -L, --endian little
       -B, --endian big
       -x, --endian swap
              These  options  specify  whether the byte-order of the audio data is, respectively, `little endian', `big endian', or
              the opposite to that of the system on which SoX is being used.  Endianness applies only to data encoded as  floating-
              point,  or  as signed or unsigned integers of 16 or more bits.  It is often necessary to specify one of these options
              for headerless files, and sometimes necessary for (otherwise) self-describing files.  A given  endian-setting  option
              may  be  ignored for an input file whose header contains a specific endianness identifier, or for an output 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 au‐
              tomatically used for the output file; so, for example, when the following is run on a little-endian system:
                 sox -B audio.s16 trimmed.s16 trim 2
              trimmed.s16 will be created as little-endian;
                 sox -B audio.s16 -B trimmed.s16 trim 2
              must be used to preserve big-endianness in the output file.

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

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

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

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

              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 --comment-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 output file header (where applicable).

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

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

       Some  of  the  SoX  effects are primarily intended to be applied to a single 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 pro‐
       cessing 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

       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 discarded.  The amount of samples discarded is
       related to the --buffer option and it should be kept small, relative to the sample rate, if the terminating effect cannot be
       first.  Further information on stopping effects can be found in the Stopping SoX section.

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

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

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

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

              See frequency.

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

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

              A  position  within  the audio stream; the syntax is [=|+|-]timespec, where timespec is a time specification (see be‐
              low).  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.,

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

              Used to specify the band-width of a filter.  A number of different 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  fol‐
       lowing two forms:

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

              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  sub‐
       tracted 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 -h and look for its name under the list: `EFFECTS'.

   Supported Effects
       Note: a categorised list of the effects can be found in the accompanying `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 fil‐
              ter 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 logarithmically around the center frequency.  The width pa‐
              rameter 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 fil‐
              ter, 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: constant 0dB peak gain.  The filters roll off at 6dB per octave (20dB per decade) and are described  in  de‐
              tail 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 re‐
              sponse similar 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

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

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

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

              The filters are described in detail in [1].

              These effects support the --plot global option.

              See also equalizer for a peaking equalisation effect.

       bend [-f frame-rate(25)] [-o over-sample(16)] { start-position(+),cents,end-position(+) }
              Changes  pitch  by specified amounts at specified times.  Each given triple: start-position,cents,end-position speci‐
              fies 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  utilises the Discrete Fourier Transform (DFT) at a particular frame rate and over-sam‐
              pling rate.  The -f and -o parameters may be used to adjust these parameters and thus control the smoothness  of  the
              changes in pitch.

              For example, an initial tone is generated, then bent three times, yielding four different notes in total:
                 play -n synth 2.5 sin 667 gain 1 \
                   bend .35,180,.25  .15,740,.53  0,-520,.3
              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 produced in this example is deliberate; to remove it, use gain -5 in place of gain 1.

              See also pitch.

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

              See (where a0 = 1).

              This effect supports the --plot global option.

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

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

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

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

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

              Each four-tuple parameter delay/decay/speed/depth gives the delay in milliseconds and the decay (relative to gain-in)
              with a modulation speed in Hz using depth in milliseconds.  The modulation 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 example, a single delay:
                 play guitar1.wav chorus 0.7 0.9 55 0.4 0.25 2 -t
              Two delays of the original samples:
                 play guitar1.wav chorus 0.6 0.9 50 0.4 0.25 2 -t \
                    60 0.32 0.4 1.3 -s
              A fuller sounding chorus (with three additional delays):
                 play guitar1.wav chorus 0.5 0.9 50 0.4 0.25 2 -t \
                    60 0.32 0.4 2.3 -t 40 0.3 0.3 1.3 -s

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

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

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

              The second parameter is a list of points on the compander's transfer function specified in dB relative to the maximum
              possible 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 companded (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 adjacent line segments on
              the transfer function meet will be rounded by the amount  given.   Typical  values  for  the  transfer  function  are

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

              The fourth (optional) parameter is an initial level to be assumed for each channel when companding starts.  This per‐
              mits  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 out‐
              put  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  com‐
              pander, but it is delayed before being fed to the volume adjuster.  Specifying a delay approximately equal to the at‐
              tack/decay times allows the compander to effectively operate in a `predictive' rather than a reactive mode.  A  typi‐
              cal value is 0.2 seconds.

                                                               *        *        *

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

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

              See also mcompand for a multiple-band companding effect.

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

              See also the compand and mcompand effects.

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

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

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

                                                               *        *        *

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

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

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

              With  the  deemph  effect,  it is possible to apply the necessary de-emphasis to audio that has been extracted from a
              pre-emphasised CD, and then either burn the de-emphasised audio to a new CD (which will then play correctly on any CD
              player), or simply play the correctly de-emphasised audio files on the PC.  For example:
                 sox track1.wav track1-deemph.wav deemph
              and then burn track1-deemph.wav to CD, or
                 play track1-deemph.wav
              or simply
                 play track1.wav deemph
              The  de-emphasis  filter  is implemented as a biquad and requires the input audio sample rate to be either 44.1kHz or
              48kHz.  Maximum deviation from the ideal response is only 0.06dB (up to 20kHz).

              This effect supports the --plot global option.

              See also the bass and treble shelving equalisation effects.

       delay {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 -n synth -j 3 sin %3 sin %-2 sin %-5 sin %-9 \
                   sin %-14 sin %-21 fade h .01 2 1.5 delay \
                   1.3 1 .76 .54 .27 remix - fade h 0 2.7 2.5 norm -1
              and this plays a guitar chord:
                 play -n synth pl G2 pl B2 pl D3 pl G3 pl D4 pl G4 \
                   delay 0 .05 .1 .15 .2 .25 remix - fade 0 4 .1 norm -1

       dither [-S|-s|-f filter] [-a] [-p precision]
              Apply dithering to the audio.  Dithering deliberately adds a small amount of noise to the signal in order to mask au‐
              dible quantization effects that can occur if the output sample size is less than 24 bits.  With no options, this  ef‐
              fect  will add triangular (TPDF) white noise.  Noise-shaping (only for certain sample rates) can be selected with -s.
              With the -f option, it is possible to select a particular noise-shaping filter from the following list: lipshitz,  f-
              weighted,  modified-e-weighted,  improved-e-weighted,  gesemann,  shibata, low-shibata, high-shibata.  Note that most
              filter types are available only with 44100Hz sample rate.  The filter types are distinguished by the following  prop‐
              erties:  audibility of noise, level of (inaudible, but in some circumstances, otherwise problematic) shaped high fre‐
              quency noise, and processing speed.
              See for graphs of the different noise-shaping curves.

              The -S option selects a slightly `sloped' TPDF, biased towards higher frequencies.  It can be used  at  any  sampling
              rate but below ≈22k, plain TPDF is probably better, and above ≈ 37k, noise-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  red‐
              ithering applies only to the faded portions.  However, auto dithering is not fool-proof, so the fades should be care‐
              fully checked for any noise modulation; if this occurs, then either re-dither the whole file, or use trim, fade,  and

              The -p option allows overriding the target precision.

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

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

              See also the `Dithering' section above.

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

              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  listened  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
              buildings)  when talking or shouting; digital echo effects emulate this behaviour and are often used to help fill out
              the sound of a single instrument or vocal.  The time difference between the original signal and the reflection is the
              `delay'  (time),  and the loudness of the reflected signal is the `decay'.  Multiple echoes can have different delays
              and decays.

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

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

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

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

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

              frequency  gives  the filter's central frequency in Hz, width, the band-width, and gain the required gain or attenua‐
              tion 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  cen‐
              tral 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  loga‐

              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 indicate the usual case of a fade-out that ends at the end of the  input  audio

              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 coefficients.  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  filename is `-', or if no argument is given, then the coefficients are read from the `standard input'
              (stdin); otherwise, coefficients may be given on the command line.  Examples:
                 sox infile outfile fir 0.0195 -0.082 0.234 0.891 -0.145 0.043
                 sox infile outfile fir coefs.txt
              with coefs.txt containing
                 # HP filter
                 # freq=10000

              This effect supports the --plot global option.

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

              All parameters are optional (right to left).

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

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

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

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

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

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

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

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

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

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

              Output formatting (dithering and bit-depth reduction) also requires headroom (which cannot be `reclaimed'), e.g.
                 sox infile outfile gain -h bass +6 rate 44100 gain -rh dither
              Here,  the second gain invocation, reclaims as much of the headroom 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  in‐
              voke gain -h and gain -r.

              See also the norm and vol effects.

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

              These effects support the --plot global option.

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

       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 signal generation.  The process is  often  written  as  a
              multiplication by i (or j), the imaginary unit.

              An  odd-tap  Hilbert  transform filter has a bandpass characteristic, attenuating the lowest and highest frequencies.
              Its bandwidth 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-spe‐
              cific, 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 determines the output channel count.  However, the -r (replicate) option allows cloning a mono plugin to handle
              multi-channel input.

              Some  plugins  introduce latency which SoX may optionally compensate for.  The -l (latency compensation) option auto‐
              matically 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
     for a detailed description of loudness.  The gain is adjusted by the given gain
              parameter (usually negative) and the signal equalised according to ISO 226 w.r.t. a reference level of  65dB,  though
              an alternative reference level may be given if the original audio has been equalised for some other optimal level.  A
              default gain of -10dB is used if a gain value is not given.

              See also the gain effect.

       lowpass [-1|-2] frequency[k] [width[q|o|h|k]]
              Apply a low-pass filter.  See the description of the highpass effect for details.

       mcompand "attack1,decay1{,attack2,decay2}
              [gain [initial-volume-dB [delay]]]" {crossover-freq[k] "attack1,..."}

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

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

              See also compand for a single-band companding effect.

       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 consis‐
              tent  background noise such as hiss or hum.  To use it, first run SoX with the noiseprof effect on a section of audio
              that ideally would contain silence but in fact contains noise - such sections are typically found at the beginning or
              the end of a recording.  noiseprof will write out a noise profile to profile-file, or to stdout if no profile-file or
              if `-' is given.  E.g.
                 sox speech.wav -n trim 0 1.5 noiseprof speech.noise-profile
              To actually remove the noise, run SoX again, this time with the noisered effect; noisered will reduce noise according
              to  a noise profile (which was generated by noiseprof), from profile-file, or from stdin if no profile-file or if `-'
              is given.  E.g.
                 sox speech.wav cleaned.wav noisered speech.noise-profile 0.3
              How much noise should be removed is specified by amount-a number between 0 and 1 with a default of 0.5.  Higher  num‐
              bers  will remove more noise but present a greater likelihood of removing wanted components of the audio signal.  Be‐
              fore 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 atten‐
              tion to quieter sections of the audio.

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

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

       oops   Out Of Phase Stereo effect.  Mixes stereo to twin-mono where each mono channel contains the  difference  between  the
              left and right stereo channels.  This is sometimes known as the `karaoke' effect as it often has the effect of remov‐
              ing 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 silence to insert and position the position in the input audio stream at which to insert it.  Any number of
              lengths and positions may be specified, provided that a specified position is not less that the previous one, and any
              time  specification may be used for them.  position is optional for the first and last lengths specified and if omit‐
              ted correspond to the beginning and the end of the audio respectively.  For example, pad 1.5 1.5 adds 1.5 seconds  of
              silence  padding  at  each end of the audio, whilst pad 4000s@3:00 inserts 4000 samples of silence 3 minutes into the
              audio.  If silence is wanted only at the end of the audio, 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 beginning of the audio on a channel-by-channel basis.

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

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

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

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

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

              See also the bend, speed, and tempo effects.

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

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

              where Band-width is the percentage of the audio frequency band that is preserved and Rej dB is the level of noise re‐
              jection.  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  algo‐
              rithms  have a `linear' phase response; for `medium', `high' and `very high', the phase response is configurable (see

              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 commands are equivalent:
                 sox input.wav -r 48k output.wav bass -b 24
                 sox input.wav        output.wav bass -b 24 rate 48k
              though the second command is more flexible as it allows rate options to be given, and allows the effects  to  be  or‐
              dered arbitrarily.

                                                               *        *        *

              Warning: technically detailed discussion follows.

              The  simple quality selection described above provides settings that satisfy the needs of the vast majority of resam‐
              pling tasks.  Occasionally, however, it may be desirable to fine-tune the resampler's filter response;  this  can  be
              achieved using override 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 sampling rates but that if
              this is at least 44.1kHz, then the artefacts 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 linear 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 compro‐
              mise by selecting a small length (and level) of pre-echo and a medium lengthed post-echo.

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

              A  resampler's  band-width  setting  determines  how much of the frequency content of the original signal (w.r.t. the
              original sample rate when up-sampling, or the new sample rate when down-sampling)  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 transient 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/audibility), this may not be a prob‐
              lem.  The benefits of allowing aliasing/imaging are reduced processing time, and reduced (by almost  half)  transient
              echo artefacts.  Note that if this option is given, then the minimum band-width allowable with -b increases to 85%.

                 sox input.wav -b 16 output.wav rate -s -a 44100 dither -s
              default  (high)  quality  resampling;  overrides:  steep filter, allow aliasing; to 44.1kHz sample rate; noise-shaped
              dither to 16-bit WAV file.
                 sox input.wav -b 24 output.aiff rate -v -I -b 90 48k
              very high quality resampling; overrides: intermediate phase, band-width 90%; to 48k  sample  rate;  store  output  to
              24-bit AIFF 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  con‐
              fused with the -m global option (where multiple files are mix-combined before entering the effects chain).

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

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

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

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

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

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

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

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

                                                               *        *        *

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

              See also the swap effect.

       repeat [count(1)|-]
              Repeat the entire audio count times, or once if count is not given.  The special value -  requests  infinite  repeti‐
              tion.   Requires temporary file space to store the audio to be repeated.  Note that repeating once yields two copies:
              the original audio and the repeated audio.

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

              Add reverberation to the audio using the `freeverb' 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.  Ap‐
              plying a small amount of stereo reverb to a (dry) mono signal will usually make it sound more natural.  See [3] for a
              detailed description of reverberation.

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

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

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

              This effect supports the --plot global option.

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

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

              The above-periods value is used to indicate if audio should be trimmed at the beginning of the audio. A value of zero
              indicates no silence should be trimmed from the beginning. When specifying 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  exam‐
              ple,  if  you  had  an  audio file with two songs that each contained 2 seconds of silence before the song, you could
              specify an above-period of 2 to strip out both silence periods and the first song.

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

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

              When  optionally trimming silence from the end of the 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  in‐
              creased  to  skip over periods of silence that are wanted.  For example, if you have a song with 2 seconds of silence
              in the middle and 2 second at the end, you could set below-period to a value of 2 to skip over  the  silence  in  the
              middle of the audio.

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

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

              To remove silence from the middle of a file, specify a below-periods that is negative.  This value is then treated as
              a positive value and is also used to indicate that the effect should restart processing as specified by the above-pe‐
              riods, 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

              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

              threshold numbers may be suffixed with d to indicate the value is in decibels, or % to indicate a percentage of maxi‐
              mum value of the sample value (0% specifies pure digital silence).

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

       sinc [-a att|-b beta] [-p phase|-M|-I|-L] [-t tbw|-n taps] [freqHP][-freqLP [-t tbw|-n taps]]
              Apply  a  sinc  kaiser-windowed  low-pass, high-pass, band-pass, or band-reject filter to the signal.  The freqHP and
              freqLP parameters give the frequencies of the 6dB points of a high-pass and low-pass filter that may be invoked indi‐
              vidually,  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' parame‐
              ter can be given directly with -b.

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

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

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

              This effect supports the --plot global option.

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

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

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


              -x num Change the (maximum) width (X-axis) of the spectrogram from its default value of 800 pixels to a given  number
                     between 100 and 200000.  See also -X and -d.

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

              -y num Sets the Y-axis size in pixels (per channel); this is the number of  frequency  `bins'  used  in  the  Fourier
                     analysis  that produces the spectrogram.  N.B. it can be slow to produce the spectrogram if this number is not
                     one more than a power of two (e.g. 129).  By default the Y-axis size is chosen automatically (depending on the
                     number of channels).  See -Y for alternative way of setting spectrogram 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 spectrogram channels that is one more  than  a  power  of
                     two,  so  the actual total height may fall short of the given number.  However, there is also a minimum height
                     per channel so if there are many channels, the number may be exceeded.  See -y for alternative way of  setting
                     spectrogram height.

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

              -q num Sets the Z-axis quantisation, i.e. the number of different colours (or intensities) in which to render  Z-axis
                     values.   A small number (e.g. 4) will give a `poster'-like effect making it easier to discern magnitude bands
                     of similar level.  Small numbers also usually result in small PNG files.  The number given specifies the  num‐
                     ber 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 significant 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 dy‐
                     namic-range properties.  For better frequency resolution (but lower dynamic-range), select a  Hamming  window;
                     for  higher dynamic-range (but poorer frequency-resolution), select a Dolph window.  Kaiser, Bartlett and Rec‐
                     tangular 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
                     adherence to the -x value, but at the expense of a little spectral loss.

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

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

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

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

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

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

              -A     Selects an alternative, fixed colour-set.  This is provided 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 differentia‐
                     tion at the bottom end which results in masking of low-level artefacts.

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

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

              -o file
                     Name of the spectrogram output PNG file, default `spectrogram.png'.  If `-' is given, the 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 input.mp3 output.wav -n spectrogram -d 1:00 stats
                     creates a spectrogram showing the first minute of the audio, whilst
                     the stats effect is applied to the entire audio signal.

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

              -S position(=)
                     Start the spectrogram at the given point in the audio stream.  For example
                        sox 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  de‐

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

              See also the bend, pitch, and tempo effects.

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

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

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

              To  perform a splice, first use the trim effect to select the audio sections to be joined together.  As when perform‐
              ing a tape splice, the end of the section to be spliced onto should be trimmed with a  small  excess  (default  0.005
              seconds)  of  audio after the ideal joining point.  The beginning of the audio section to splice on should be trimmed
              with the same excess (before the ideal joining point), plus an additional leeway (default 0.005 seconds).   Any  time
              specification  may  be  used  for  these parameters.  SoX should then be invoked with the two 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  au‐
              dio section (including the excess).

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

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

              where * indicates the joining points.

              For example, a long song begins with two verses which start (as determined e.g. by using the play  command  with  the
              trim (start) effect) at times 0:30.125 and 1:03.432.  The following commands cut out the first verse:
                 sox too-long.wav part1.wav trim 0 30.130
              (5 ms excess, after the first verse starts)
                 sox too-long.wav part2.wav trim 1:03.422
              (5 ms excess plus 5 ms leeway, before the second verse starts)
                 sox part1.wav part2.wav just-right.wav splice 30.130
              For another example, the SoX command
                 play "|sox -n -p synth 1 sin %1" "|sox -n -p synth 1 sin %3"
              generates and plays two notes, but there is a nasty click at the transition; the click can be removed by splicing in‐
              stead of concatenating the audio, i.e. by appending splice 1 to the command. (Clicks at the beginning and end of  the
              audio can be removed by preceding the splice effect with fade q .01 2 .01).

              Provided  your arithmetic is good enough, multiple splices can be performed with a single splice invocation.  For ex‐
              # 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 "$1" piece.wav trim $2-$e-$l =$3+$e
              sox "$1" part1.wav trim 0 $4+$e
              sox "$1" part2.wav trim $4+$3-$2-$e-$l
              sox 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  typically  be  an number of seconds, the -q option would typically be given (to select an `equal power' cross-
              fade), and leeway should be zero (which is the default if -q is given).  For example, if f1.wav and f2.wav are  audio
              files to be cross-faded, then
                 sox f1.wav f2.wav out.wav splice -q $(soxi -D f1.wav),3
              cross-fades  the files where the point of equal loudness is 3 seconds before the end of f1.wav, i.e. the total length
              of the cross-fade is 2 × 3 = 6 seconds (Note: the $(...) notation is POSIX shell).

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

                                   Samples read        n×c
                                   Length (seconds)    n÷r
                                   Scaled by                                 See -s below.
                                   Maximum amplitude   max(xk)               The  maximum  sample value in the
                                                                             audio; usually  this  will  be  a
                                                                             positive number.
                                   Minimum amplitude   min(xk)               The  minimum  sample value in the
                                                                             audio; usually  this  will  be  a
                                                                             negative number.
                                   Midline amplitude   ½min(xk)+½max(xk)
                                   Mean norm           ¹/nΣ│xk│              The average of the absolute value
                                                                             of each sample in the audio.
                                   Mean amplitude      ¹/nΣxk                The average of each sample in the
                                                                             audio.   If  this  figure is non-
                                                                             zero, then it indicates the pres‐
                                                                             ence  of  a  D.C.  offset  (which
                                                                             could be removed  using  the  dc‐
                                                                             shift effect).
                                   RMS amplitude       √(¹/nΣxk²)            The  level  of a D.C. signal that
                                                                             would have the same power as  the
                                                                             audio's average power.
                                   Maximum delta       max(│xk-xk-1│)
                                   Minimum delta       min(│xk-xk-1│)
                                   Mean delta          ¹/n-1Σ│xk-xk-1│
                                   RMS delta           √(¹/n-1Σ(xk-xk-1)²)
                                   Rough frequency                           In Hz.
                                   Volume Adjustment                         The  parameter  to the vol effect
                                                                             which would  make  the  audio  as
                                                                             loud  as  possible  without clip‐
                                                                             ping.  Note: See  the  discussion
                                                                             on Clipping above for reasons why
                                                                             it is rarely a good idea actually
                                                                             to do this.

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

              The  -s option can be used to scale the input data by a given factor.  The default value of scale is 2147483647 (i.e.
              the maximum value of a 32-bit signed integer).  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-platform versions of SoX.

              See also the stats effect.

       stats [-b bits|-x bits|-s scale] [-w window-time]
              Display time domain statistical information about the audio channels; audio is passed unmodified through the SoX pro‐
              cessing chain.  Statistics are calculated and displayed for each audio channel and, where applicable, an overall fig‐
              ure 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 hexadecimal.  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 measured in dBFS.  RMS Pk dB  and  RMS Tr dB  are  peak  and
              trough values for RMS level measured over a short window (default 50ms).

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

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

              The  right-hand  Bit-depth  figure  is the standard 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 significant bits that are fixed at zero (or one
              for negative numbers) subtracted from the right-hand figure (the number subtracted 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 samples is equal to the sample-rate multiplied by Length.
              Scale Max is the scaling applied to the first three measurements; specifically, it is the maximum  value  that  could
              apply to Max level.  Window s is the length of the window used for the peak and trough RMS measurements.

              See also the stat effect.

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

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

       stretch factor [window fade shift fading]
              Change the audio duration (but not its pitch).  This effect is broadly equivalent to the tempo  effect  with  (factor
              inverted  and)  search set to zero, so in general, its results are comparatively poor; it is retained as it can some‐
              times 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
              This  effect  can  be  used to generate fixed or swept frequency audio tones with various wave shapes, or to generate
              wide-band noise of various `colours'.  Multiple synth effects can be cascaded to produce more complex  waveforms;  at
              each  stage  it is possible to choose whether the generated waveform will be mixed with, or modulated onto the output
              from the previous stage.  Audio for each channel in a multi-channel audio file can be synthesised independently.

              Though this effect is used to generate audio, an input file must still be given, the characteristics of which will be
              used  to  set  the  synthesised audio length, the number of channels, and the sampling rate; however, since the input
              file's audio is not normally needed, a `null file' (with the special name -n) is often given instead (and the  length
              specified as a parameter to synth or by another given effect that has an associated length).

              For example, the following produces a 3 second, 48kHz, audio file containing a sine-wave swept from 300 to 3300 Hz:
                 sox -n output.wav synth 3 sine 300-3300
              and this produces an 8 kHz version:
                 sox -r 8000 -n output.wav synth 3 sine 300-3300
              Multiple  channels  can  be  synthesised by specifying the set of parameters shown between braces multiple times; the
              following puts the swept tone in the left channel and adds `brown' noise in the right:
                 sox -n output.wav synth 3 sine 300-3300 brownnoise
              The following example shows how two synth effects can be cascaded to create a more complex waveform:
                 play -n synth 0.5 sine 200-500 synth 0.5 sine fmod 700-100
              Frequencies can also be given in `scientific' note notation, or, by prefixing a `%' character, as a number  of  semi‐
              tones  relative  to  `middle  A'  (440 Hz).  For example, the following could be used to help tune a guitar's low `E'
                 play -n synth 4 pluck %-29
              or with a (Bourne shell) loop, the whole guitar:
                 for n in E2 A2 D3 G3 B3 E4; do
                   play -n synth 4 pluck $n repeat 2; done
              See the delay effect (above) and the reference to `SoX scripting examples' (below) for more synth examples.

              N.B.  This effect generates audio at maximum volume (0dBFS), which means that there is a high chance of clipping when
              using  the  audio subsequently, so in many cases, you will want to follow this effect with the gain effect to prevent
              this from happening. (See also Clipping above.)  Note that, by default, the synth effect incorporates the functional‐
              ity of gain -h (see the gain effect for details); synth's -n option may be given to disable this behaviour.

              A detailed description of each synth parameter follows:

              len  is  the  length of audio to synthesise (any time specification); 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; default=sine.

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

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

              If freq2 is given, then len must also have been given and the generated tone will be swept between the given frequen‐
              cies.  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,
              triangle, exp), default=10 (trapezium), or sustain (pluck); default=40.

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

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

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

              By default, linear searches are used to find the best overlapping 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 override these default values.

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

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

              The  optional  search parameter gives the audio length in milliseconds over which the algorithm will search for over‐
              lapping 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  de‐
              fault 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 in‐
              crease  quality.  A  practical  maximum for overlap is the value of search, with overlap typically being (at least) a
              little smaller then search.

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

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

       tremolo speed [depth]
              Apply  a  tremolo  (low frequency amplitude modulation) 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 discarding 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 infile outfile trim 0 10
              will copy the first ten seconds, while
                 play infile trim 12:34 =15:00 -2:00
                 play 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  sec‐
              onds 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 sam‐
              ples.  As a result, the original spectrum is replicated into the new frequency space (imaging) and attenuated.   This
              attenuation  can  be compensated for by adding vol factor after any further processing.  The upsample effect is typi‐
              cally 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  resolu‐
              tion  i.e.  16-bit, 44-48kHz) recordings of speech.  The algorithm currently uses a simple cepstral power measurement
              to detect voice, so may be fooled by other things, especially music.  The effect can trim only from the front of  the
              audio, so in order to trim from the back, the reverse effect must also be used.  E.g.
                 play speech.wav norm vad
              to trim from the front,
                 play speech.wav norm reverse vad reverse
              to trim from the back, and
                 play speech.wav norm vad reverse vad reverse
              to  trim  from  both  ends.  The use of the norm effect is recommended, but remember that neither reverse nor norm is
              suitable for use with streamed audio.

              Default values are shown in parenthesis.

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

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

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

              -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 estimation/reduction in order to detect the start of the wanted
                     audio.  This option sets the time for the initial noise estimate.

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

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

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

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

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

              -M num Time constant used to smooth spectral measurements.

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

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

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

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

              See also the silence effect.

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

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

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

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

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

              Beware of Clipping when the increasing the volume.

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

              An optional 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 capabilities, and compand for a dynamic-range compres‐
              sion/expansion/limiting effect.

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

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

       soxi(1), soxformat(7), libsox(3)
       audacity(1), gnuplot(1), octave(1), wget(1)
       The SoX web site at
       SoX scripting examples at

       [1]    R.  Bristow-Johnson,  Cookbook  formulae for audio EQ biquad filter coefficients,

       [2]    Wikipedia, Q-factor,

       [3]    Scott Lehman, Effects Explained,

       [4]    Wikipedia, Decibel,

       [5]    Richard Furse, Linux Audio Developer's Simple Plugin API,

       [6]    Richard Furse, Computer Music Toolkit,

       [7]    Steve Harris, LADSPA plugins,

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

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

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

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

sox                                                      December 31, 2014                                                   SoX(1)