SoX(1) Sound eXchange SoX(1)

NAME

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

SYNOPSIS

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

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

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

DESCRIPTION

Introduction

SoX reads and writes audio files in most popular formats and can
optionally apply effects to them. It can combine multiple input
sources, synthesise audio, and, on many systems, act as a general
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 device is used as an input
source. Additionally, the soxi(1) command provides a convenient way
to just query audio file header information.

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

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

* * *

The overall SoX processing chain can be summarised as follows:

Input(s) -> Combiner -> Effects -> Output(s)

Note however, that on the SoX command line, the positions of the
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 opposite is true for effects.
To show how this works in practice, here is a selection of examples
of how SoX might be used. The simple
sox recital.au recital.wav
translates an audio file in Sun AU format to a Microsoft WAV file,
whilst
sox recital.au -b 16 recital.wav channels 1 rate 16k fade 3 norm
performs the same format translation, but also applies four effects
(down-mix to one channel, sample rate change, fade-in, nomalize), and
stores the result at a bit-depth of 16.
sox -r 16k -e signed -b 8 -c 1 voice-memo.raw voice-memo.wav
converts `raw' (a.k.a. `headerless') audio to a self-describing file
format,
sox slow.aiff fixed.aiff speed 1.027
adjusts audio speed,
sox short.wav long.wav longer.wav
concatenates two audio files, and
sox -m music.mp3 voice.wav mixed.flac
mixes together two audio files.
play "The Moonbeams/Greatest/*.ogg" bass +3
plays a collection of audio files whilst applying a bass boosting
effect,
play -n -c1 synth sin %-12 sin %-9 sin %-5 sin %-2 fade h 0.1 1 0.1
plays a synthesised `A minor seventh' chord with a pipe-organ sound,
rec -c 2 radio.aiff trim 0 30:00
records half an hour of stereo audio, and
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 formats, 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. Commonly-
used encoding types include floating-point, <mu>-law, ADPCM,
signed-integer PCM, MP3, and FLAC.

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

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

Most self-describing formats also allow textual `comments' to be
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 information (for
certain input file formats only; currently, at least FLAC and Ogg
Vorbis), but not generating it. Note that by default, SoX copies
input file comments to output files that support comments, so output
files may contain Replay Gain information if some was present in the
input file. In this case, if anything other than a simple format
conversion was performed then the output file Replay Gain information
is likely to be incorrect and so should be recalculated using a tool
that supports this (not SoX).

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

Determining & Setting The File Format
There are several mechanisms available for SoX to use to determine or
set the format characteristics of an audio file. Depending on the
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
and
rec new-file.wav
These two commands are functionally equivalent to
sox existing-file.wav -d
and
sox -d new-file.wav
Of course, further options and effects (as described below) can be
added to the commands in either form.

* * *

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

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

* * *

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

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

Accuracy

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 formats used in telephony
(e.g. A-law, GSM) where low signal bandwidth is more important than
high audio fidelity, and for many formats used in portable music
players (e.g. MP3, Vorbis) where adequate fidelity can be retained
even with the large compression ratios that are needed to make
portable players practical.

Formats that discard audio signal information are called `lossy'.
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 sampling rate or
number of channels, and when the number of bits used in the
destination format is not less than in the source format. E.g.
converting from an 8-bit PCM format to a 16-bit PCM format is
lossless but converting from an 8-bit PCM format to (8-bit) A-law
isn't.

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

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

Dithering

Dithering is a technique used to maximise the dynamic range of audio
stored at a particular bit-depth. Any distortion introduced by
quantisation is decorrelated by adding a small amount of white noise
to the signal. In most cases, SoX can determine 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 following are true:

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

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

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

For example, adjusting volume with vol 0.25 requires two additional
bits in which to losslessly store its results (since 0.25 decimal
equals 0.01 binary). So if the input file bit-depth is 16, then
SoX's internal representation will 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

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 processing chain) at
which it occurs.

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

Playing an audio file often involves resampling, and processing by
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 point.

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

SoX's stat and stats effects can assist in determining the signal
level in an audio file. The gain or vol effect can be used to prevent
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 `concatenate' for rec and sox.

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

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

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

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

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

The `multiply' combining method multiplies the sample values of
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 zero.

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

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

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

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

Unlike the other methods, `mix' combining has the potential to cause
clipping in the combiner if no balancing is performed. In this case,
if manual volume adjustments are not given, SoX will try to ensure
that clipping does not occur by automatically adjusting the volume
(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/<sqrt>n instead of ^1/n.
Note that this balancing factor does not guarantee that clipping will
not occur, but the number of clips will usually be low and the
resultant distortion is generally imperceptible.

Output Files

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

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

In multiple output mode, a new file is created when the effects prior
to the `newfile' indicate they are done. The effects chain listed
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 empty.

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

Stopping SoX

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

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

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

FILENAMES

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.

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

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

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

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

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

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

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

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

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

OPTIONS

Global Options

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

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

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

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

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

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

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

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

--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 converting the file back to
16 bit. See also the stats effect for how to determine the
actual bit depth of the audio within a file.

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

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

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

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

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

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

--norm[=dB-level]
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 option 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 gnuplot
program or the GNU Octave program, to assist with the
selection and configuration of many of the transfer-function
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 output
file (e.g. AIFF) and will `seed' pseudo random number
generators (e.g. dither) with a fixed number, thus ensuring
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 otherwise.

-S, --show-progress
Display input file format/header information, and processing
progress as input file(s) percentage complete, elapsed time,
and remaining time (if known; shown in brackets), and the
number of samples written to the output file. Also shown is a
peak-level meter, and an indication if clipping has occurred.
The peak-level meter shows up to two channels and is
calibrated for 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 permission
or free-space problems with the default location. In this
case, using `--temp .' (to use the current directory) is often
a good solution.

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

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

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

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

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

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

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

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

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

Input File Options

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

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

-v, --volume FACTOR
Intended for use when combining multiple input files, this
option adjusts the volume of the file that follows it on the
command line by a factor of FACTOR. This allows it to be
`balanced' w.r.t. the other input files. This is a linear
(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 encodings
such as MP3 or GSM. Not necessary with encodings that have a
fixed number of bits, e.g. A/<mu>-law, ADPCM.

For an input file, the most common use for this option is to
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' (`headerless')
audio file. Occasionally, it may be useful to use this option
with a `headered' file, in order to override the (presumably
incorrect) value in the header - note that this is only
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 order
to change (if necessary) the number of channels in the audio
signal to the number given. For example, the following two
commands are equivalent:
sox input.wav -c 1 output.wav bass -b 24
sox input.wav output.wav bass -b 24 channels 1
though the second form is more flexible as it allows the
effects to be ordered arbitrarily.

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

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

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

floating-point
PCM data stored as IEEE 753 single precision (32-bit)
or double precision (64-bit) floating-point (`real')
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. <mu>-law. It
has a precision equivalent to roughly 14-bit PCM and is
sometimes encoded with reversed bit-ordering (see the
-X option).

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

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

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

gsm-full-rate
GSM is currently used for the vast majority of the
world's digital wireless telephone calls. It utilises
several audio formats with different bit-rates and
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') audio
file (see the examples in -b and -c above).

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

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

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.

--no-glob
Specifies that filename `globbing' (wild-card matching) should
not be performed by SoX on the following filename. For
example, if the current directory contains the two files
`five-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
solution 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 commands are
equivalent:
sox input.wav -r 48k output.wav bass -b 24
sox input.wav output.wav bass -b 24 rate 48k
though the second form is more flexible as it allows rate
options to be given, and allows the effects to be ordered
arbitrarily.

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

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

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

-L, --endian little
-B, --endian big
-x, --endian swap
These options specify whether the byte-order of the audio data
is, respectively, `little endian', `big endian', or the
opposite to that of the system on which SoX is being used.
Endianness applies only to data encoded as floating-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
automatically used for the output file; so, for example, when
the following is run on a little-endian system:
sox -B audio.s16 trimmed.s16 trim 2
trimmed.s16 will be created as little-endian;
sox -B audio.s16 -B trimmed.s16 trim 2
must be used to preserve big-endianness in the output file.

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

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

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

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

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

Output File Format Options

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

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

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

SoX will provide a default comment if this option (or
--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
compression factor will apply. The compression factor is
interpreted differently for different compressing file
formats. See the description of the file formats that use
this option in soxformat(7) for more information.

EFFECTS

In addition to converting, playing and recording audio files, SoX can
be used to invoke a number of audio `effects'. Multiple effects may
be applied by specifying them one after another at the end of the SoX
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 processing audio,
the audio data is then sent through the next effects chain. This
continues until either no more effects chains exist or the input has
reached the end of the file.

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

It is important to place the effect that will stop the chain as the
first effect in the chain. This is because any samples that are
buffered by effects to the left of the terminating effect will be
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:

center[k]
See frequency.

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

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

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

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

width[h|k|o|q]
Used to specify the band-width of a filter. A number of
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 following
two forms:

[[hours:]minutes:]seconds[.frac][t]
A specification of `1:30.5' corresponds to one minute, thirty
and 1/2 seconds. The t suffix is entirely optional (however,
see the silence effect for an exception). Note that the
component values do not have to be normalized; e.g.,
`1:23:45', `83:45', `79:0285', `1:0:1425', `1::1425' and
`5025' all are legal and equivalent to each other.

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

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

To see if SoX has support for an optional effect, enter sox -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 filter
changes the audio's frequency to phase relationship without
changing its frequency to amplitude relationship. The filter
is described in detail in [1].

This effect supports the --plot global option.

band [-n] center[k] [width[h|k|o|q]]
Apply a band-pass filter. The frequency response drops
logarithmically around the center frequency. The width
parameter gives the slope of the drop. The frequencies at
center + width and center - width will be half of their
original amplitudes. band defaults to a mode oriented to
pitched audio, i.e. voice, singing, or instrumental music.
The -n (for noise) option uses the alternate mode for un-
pitched audio (e.g. percussion). Warning: -n introduces a
power-gain of about 11dB in the filter, so beware of output
clipping. band introduces noise in the shape of the filter,
i.e. peaking at the center frequency and settling around it.

This effect supports the --plot global option.

See also sinc for a bandpass filter with steeper shoulders.

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

These effects support the --plot global option.

See also sinc for a bandpass filter with steeper shoulders.

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

bass|treble gain [frequency[k] [width[s|h|k|o|q]]]
Boost or cut the bass (lower) or treble (upper) frequencies of
the audio using a two-pole shelving filter with a response
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 gain.

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

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

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

The filters are described in detail in [1].

These effects support the --plot global option.

See also equalizer for a peaking equalisation effect.

bend [-f frame-rate(25)] [-o over-sample(16)] { start-
position(+),cents,end-position(+) }
Changes pitch by specified amounts at specified times. Each
given triple: start-position,cents,end-position specifies one
bend. cents is the number of cents (100 cents = 1 semitone)
by which to bend the pitch. The other values specify the
points in time at which to start and end bending the pitch,
respectively.

The pitch-bending algorithm utilises the Discrete Fourier
Transform (DFT) at a particular frame rate and over-sampling
rate. The -f and -o parameters may be used to adjust these
parameters and thus control the smoothness of the changes in
pitch.

For example, an initial tone is generated, then bent three
times, yielding four different notes in total:
play -n synth 2.5 sin 667 gain 1 \
bend .35,180,.25 .15,740,.53 0,-520,.3
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
respectively.

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

This effect supports the --plot global option.

channels CHANNELS
Invoke a simple algorithm to change the number of channels in
the audio signal to the given number CHANNELS: mixing if
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
instrumentation.

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

Each four-tuple parameter delay/decay/speed/depth gives the
delay in milliseconds and the decay (relative to gain-in) with
a modulation speed in Hz using depth in milliseconds. The
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}
[soft-knee-dB:]in-dB1[,out-dB1]{,in-dB2,out-dB2}
[gain [initial-volume-dB [delay]]]

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

The attack and decay parameters (in seconds) determine the
time over which the instantaneous level of the input signal is
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/decay parameters are specified,
each input channel is companded separately and the number of
pairs must agree with the number of input channels. Typical
values are 0.3,0.8 seconds.

The second parameter is a list of points on the compander's
transfer function specified in dB relative to the maximum
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 6:-70,-60,-20.

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

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

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

* * *

The following example might be used to make a piece of music
with both quiet and loud passages suitable for listening to in
a noisy environment such as a moving vehicle:
sox 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 inexact, and was derived by
experimentation). The -90 (dB) for the initial volume will
work fine for a clip that starts with near silence, and the
delay of 0.2 (seconds) has the effect of causing the compander
to react a bit more quickly to sudden volume changes.

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

See also mcompand for a multiple-band companding effect.

contrast [enhancement-amount(75)]
Comparable with compression, this effect modifies an audio
signal to make it sound louder. enhancement-amount controls
the amount of the enhancement and is a number in the range
0-100. Note that enhancement-amount = 0 still gives a
significant 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 audible
quantization effects that can occur if the output sample size
is less than 24 bits. With no options, this effect 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 properties: audibility of
noise, level of (inaudible, but in some circumstances,
otherwise problematic) shaped high frequency noise, and
processing speed.
See http://sox.sourceforge.net/SoX/NoiseShaping for graphs of
the different noise-shaping curves.

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

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

The -p option allows overriding the target precision.

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

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

See also the `Dithering' section above.

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

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

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

earwax Makes audio easier to listen to on headphones. Adds `cues' to
44.1kHz stereo (i.e. audio CD format) audio so that when
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 using many
echos; a single echos has the same effect as a single echo.

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

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

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

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

The filter is described in detail in [1].

This effect supports the --plot global option.

See also bass and treble for shelving equalisation effects.

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

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

A fade-in starts from the first sample and ramps the signal
level from 0 to full volume over the time given as fade-in-
length. Specify 0 if no fade-in is wanted.

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

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

See also the splice effect.

fir [coefs-file|coefs]
Use SoX's FFT convolution engine with given FIR filter
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
1.2311233052619888e-01
-4.4777096106211783e-01
5.1031563346705155e-01
-6.6502926320995331e-02
...

This effect supports the --plot global option.

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

All parameters are optional (right to left).

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

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

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

Given the -e option, the levels of the audio channels of a
multi-channel file are `equalised', i.e. gain is applied to
all channels other than that with the highest peak level, such
that all channels attain the same peak level (but, without
also 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 always as clear.
Another advantage of using gain -h rather than an explicit
attenuation, is that if the headroom is not used by subsequent
effects, it can be reclaimed with gain -r, for example:
sox infile outfile gain -h bass +6 rate 44100 gain -r
The above effects chain guarantees never to clip nor amplify;
it attenuates if necessary to prevent clipping, but by only as
much as is needed to do so.

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

See also the norm and vol effects.

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

These effects support the --plot global option.

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

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-specific, and a
wide range of effects is available as LADSPA plugins, such as
cmt [6] (the Computer Music Toolkit) and Steve Harris's plugin
collection [7]. The first argument is the plugin module, the
second the name of the plugin (a module can contain more than
one plugin), and any other arguments are for the control ports
of the plugin. Missing arguments are supplied by default
values if possible.

Normally, the number of input ports of the plugin must match
the number of input channels, and the number of output ports
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
automatically compensates for latency as reported by the
plugin via an output control port named "latency".

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

loudness [gain [reference]]
Loudness control - similar to the gain effect, but provides
equalisation for the human auditory system. See
http://en.wikipedia.org/wiki/Loudness for a detailed
description of loudness. The gain is adjusted by the given
gain parameter (usually negative) and the signal 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}
[soft-knee-dB:]in-dB1[,out-dB1]{,in-dB2,out-dB2}
[gain [initial-volume-dB [delay]]]" {crossover-freq[k]
"attack1,..."}

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

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

See also compand for a single-band companding effect.

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

noisered [profile-file [amount]]
Reduce noise in the audio signal by profiling and filtering.
This effect is moderately effective at removing consistent
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 numbers
will remove more noise but present a greater likelihood of
removing wanted components of the audio signal. Before
replacing an original recording with a noise-reduced version,
experiment with different amount values to find the optimal
one for your audio; use headphones to check that you are happy
with the results, paying particular attention to quieter
sections of the audio.

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

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

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

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

pad { length[@position(=)] }
Pad the audio with silence, at the beginning, the end, or any
specified points through the audio. 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 omitted correspond to the
beginning and the end of the audio respectively. For example,
pad 1.5 1.5 adds 1.5 seconds of silence padding at each end of
the audio, whilst pad 4000s@3:00 inserts 4000 samples of
silence 3 minutes into the audio. If silence is wanted only
at the end of the audio, 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
Gentler:
play snare.flac phaser 0.9 0.85 4 0.23 1.3 -s
A popular sound:
play snare.flac phaser 0.89 0.85 1 0.24 2 -t
More severe:
play snare.flac phaser 0.6 0.66 3 0.6 2 -t

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

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

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

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

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

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

* * *

Warning: technically detailed discussion follows.

The simple quality selection described above provides settings
that satisfy the needs of the vast majority of resampling
tasks. Occasionally, however, it may be desirable to fine-
tune the resampler's filter response; this can be achieved
using 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
compromise by selecting a small length (and level) of pre-echo
and a medium lengthed post-echo.

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

A resampler's band-width setting determines how much of the
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 problem. The
benefits of allowing aliasing/imaging are reduced processing
time, and reduced (by almost half) transient echo artefacts.
Note that if this option is given, then the minimum band-width
allowable with -b increases to 85%.

Examples:
sox input.wav -b 16 output.wav rate -s -a 44100 dither -s
default (high) quality resampling; overrides: steep filter,
allow aliasing; to 44.1kHz sample rate; noise-shaped dither to
16-bit WAV file.
sox input.wav -b 24 output.aiff rate -v -I -b 90 48k
very high quality resampling; overrides: intermediate phase,
band-width 90%; to 48k sample rate; store output to 24-bit
AIFF file.

* * *

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

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

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

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

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

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

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

Letter Volume number Notes
p power adjust in dB 0 = no change
i power adjust in dB As `p', but invert
the audio
v voltage multiplier 1 = no change, 0.5
~= 6dB attenuation,
2 ~= 6dB gain, -1 =
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/<sqrt>n (`power') scaling; this gives a louder
mix but one that might occasionally clip.

* * *

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

See also the swap effect.

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

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

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

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

This effect supports the --plot global option.

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

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

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

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

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

To remove silence from the middle of a file, specify a below-
periods that is negative. This value is then treated as a
positive value and is also used to indicate that the effect
should restart processing as specified by the above-periods,
making it suitable for removing periods of silence in the
middle of the audio.

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

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

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

The following example shows how this effect can be used to
start a recording that does not contain the delay at the start
which usually occurs between `pressing the record button' and
the start of the performance:
rec 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 individually, or
together. If both are given, then freqHP less than freqLP
creates a band-pass filter, freqHP greater than freqLP creates
a band-reject filter. For example, the invocations
sinc 3k
sinc -4k
sinc 3k-4k
sinc 4k-3k
create a high-pass, low-pass, band-pass, and band-reject
filter respectively.

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

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

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

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

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 intensity) of
the pixels in the X-Y plane. If the audio signal contains
multiple channels then these are shown from top to bottom
starting from channel 1 (which is the left channel for stereo
audio).

For example, if `my.wav' is a stereo file, then with
sox 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 produced
image will be 600 by 200 pixels in size and the Z-axis range
will be 100 dB). Note that the produced image includes axes
legends etc. and so will be a little larger than the specified
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.

Options:

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

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

-y num Sets the Y-axis size in pixels (per channel); this is
the number of frequency `bins' used in the Fourier
analysis that produces the spectrogram. N.B. it can be
slow to produce the spectrogram if this number is not
one more than a power of two (e.g. 129). By default
the Y-axis size is chosen automatically (depending on
the number of 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 number of colours to use inside the Z-
axis range; two colours are reserved to represent out-
of-range values.

-w name
Window: Hann (default), Hamming, Bartlett, Rectangular,
Kaiser or Dolph. The spectrogram is produced using the
Discrete Fourier Transform (DFT) algorithm. A
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 dynamic-range properties. For better frequency
resolution (but lower dynamic-range), select a Hamming
window; for higher dynamic-range (but poorer frequency-
resolution), select a Dolph window. Kaiser, Bartlett
and Rectangular windows are also available.

-W num Window adjustment parameter. This can be used to make
small adjustments to the Kaiser or Dolph window shape.
A positive number (up to ten) increases its dynamic
range, a negative number decreases it.

-s Allow slack overlapping of DFT windows. This can, in
some cases, increase image sharpness and give greater
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
permutation.

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

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

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

-A Selects an alternative, fixed 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
differentiation at the bottom end which results in
masking of low-level artefacts.

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

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

-o file
Name of the spectrogram output PNG file, default
`spectrogram.png'. If `-' is given, the 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 decreases.

Technically, the speed effect only changes the sample rate
information, leaving the samples themselves untouched. The
rate effect is invoked automatically to resample to the output
sample rate, using its default quality/speed. For higher
quality or higher speed resampling, in addition to the speed
effect, 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), triangular
(a.k.a. linear), or quarter-cosine wave respectively.

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

To perform a splice, first use the trim effect to select the
audio sections to be joined together. As when performing a
tape splice, the end of the section to be spliced onto should
be trimmed with a small excess (default 0.005 seconds) of
audio after the ideal joining point. The beginning of the
audio 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 audio section (including the excess).

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

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

where * indicates the joining points.

For example, a long song begins with two verses which start
(as determined e.g. by using the play command with the trim
(start) effect) at times 0:30.125 and 1:03.432. The following
commands cut out the first verse:
sox too-long.wav part1.wav trim 0 30.130
(5 ms excess, after the first verse starts)
sox too-long.wav part2.wav trim 1:03.422
(5 ms excess plus 5 ms leeway, before the second verse starts)
sox part1.wav part2.wav just-right.wav splice 30.130
For another example, the SoX command
play "|sox -n -p synth 1 sin %1" "|sox -n -p synth 1 sin %3"
generates and plays two notes, but there is a nasty click at
the transition; the click can be removed by splicing instead
of 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 example:
#!/bin/sh
# Audio Copy and Paste Over
# acpo infile copy-start copy-stop paste-over-start outfile
# No chained time specifications allowed for the parameters
# (i.e. such that contain +/-).
e=0.005 # Using default excess
l=$e # and leeway.
sox "$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 x 3 = 6 seconds (Note: the $(...) notation is
POSIX shell).

stat [-s scale] [-rms] [-freq] [-v] [-d]
Display time and frequency domain statistical information
about the audio. Audio is passed unmodified through the SoX
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 nxc
Length (seconds) n/r
Scaled by See -s below.
Maximum amplitude max(xk) The maximum sample
value in the audio;
usually this will
be a positive
number.
Minimum amplitude min(xk) The minimum sample
value in the audio;
usually this will
be a negative
number.
Midline amplitude 1/2min(xk)+1/2max(xk)
Mean norm ^1/n<Sigma>|xk| The average of the
absolute value of
each sample in the
audio.
Mean amplitude ^1/n<Sigma>xk The average of each
sample in the
audio. If this
figure is non-zero,
then it indicates
the presence of a
D.C. offset (which
could be removed
using the dcshift
effect).
RMS amplitude <sqrt>(^1/n<Sigma>xk^2) 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-1<Sigma>|xk-xk-1|
RMS delta <sqrt>(^1/n-1<Sigma>(xk-xk-1)^2)
Rough frequency In Hz.
Volume Adjustment The parameter to
the vol effect
which would make
the audio as loud
as possible without
clipping. Note: See
the discussion on
Clipping above for
reasons why it is
rarely a good idea
actually to do
this.

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

The -s option can be used to scale the input data by a given
factor. The default value of scale is 2147483647 (i.e. the
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
processing chain. Statistics are calculated and displayed for
each audio channel and, where applicable, an overall figure is
also given.

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

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

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

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

See also the tempo effect.

synth [-j KEY] [-n] [len [off [ph [p1 [p2 [p3]]]]]] {[type] [combine]
[[%]freq[k][:|+|/|-[%]freq2[k]]] [off [ph [p1 [p2 [p3]]]]]}
This effect can be used to generate fixed or swept frequency
audio tones with various wave shapes, or to generate wide-band
noise of various `colours'. Multiple synth effects can be
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 semitones
relative to `middle A' (440 Hz). For example, the following
could be used to help tune a guitar's low `E' string:
play -n synth 4 pluck %-29
or with a (Bourne shell) loop, the whole guitar:
for n in E2 A2 D3 G3 B3 E4; do
play -n synth 4 pluck $n repeat 2; done
See the delay effect (above) and the reference to `SoX
scripting examples' (below) for more synth examples.

N.B. This effect generates audio at maximum volume (0dBFS),
which means that there is a high chance of clipping when using
the audio subsequently, so in many cases, you will want to
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 functionality of gain -h
(see the gain effect for details); synth's -n option may be
given to disable this behaviour.

A detailed description of each synth parameter follows:

len is the length of audio to 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 frequencies.
The two given frequencies must be separated by one of the
characters `:', `+', `/', or `-'. This character is used to
specify the sweep function as follows:

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

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

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

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

Not used for noise.

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

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

p1 is the percentage of each cycle that is `on' (square), or
`rising' (triangle, exp, trapezium); default=50 (square,
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 multiples
of 2dB; default=50, or tone-1 (pluck); default=20.

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

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

By default, linear searches are used to find the best
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 specified, the
default value is 82 and is typically suited to making small
changes to the tempo of music. For larger changes (e.g. a
factor of 2), 41 ms may give a better result. The -m, -s, and
-l flags will cause the segment default to be automatically
adjusted based on factor. For example using -s (for speech)
with a tempo of 1.25 will calculate a default segment value of
32.

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

The optional overlap parameter gives the segment overlap
length in milliseconds. Default value is 12, but -m, -s, or
-l flags automatically adjust overlap based on segment size.
Increasing overlap increases processing time and may increase
quality. A practical maximum for overlap is the value of
search, with 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
and
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 seconds
long), then resume playing two minutes before the end of
audio.

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

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

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

Options:
Default values are shown in parenthesis.

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

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

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

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

-p num (0)
The amount of audio (in seconds) to preserve 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 multiple
input files as they enter the SoX effects processing chain),
vol is an effect like any other so can be applied anywhere,
and several times if necessary, during the processing chain.

The amount to change the volume is given by gain which is
interpreted, according to the given type, as follows: if type
is amplitude (or is omitted), then gain is an 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
compression/expansion/limiting effect.

DIAGNOSTICS

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
processing.

BUGS

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

SEE ALSO

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

References

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

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

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

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

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

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

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

LICENSE

Copyright 1998-2013 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.

AUTHORS

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

sox December 31, 2014 SoX(1)