sox
(1)
Name
sox - lation
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
Sound eXchange SoX(1)
NAME
SoX - Sound eXchange, the Swiss Army knife of audio manipu-
lation
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 multi-
ple 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 com-
mand. 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 inter-
ested 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 process-
ing. If you need an interactive, graphical audio editor,
use audacity(1).
* * *
The overall SoX processing chain can be summarised as fol-
lows:
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
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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,
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rec -r 44100 -b 16 -s -p silence 1 0.50 0.1% 1 10:00 0.1% | \
sox -p song.ogg silence 1 0.50 0.1% 1 2.0 0.1% : \
newfile : restart
records a stream of audio such as LP/cassette and splits in
to multiple audio files at points with 2 seconds of silence.
Also, it does not start recording until it detects audio is
playing and stops after it sees 10 minutes of silence.
N.B. The above is just an overview of SoX's capabilities;
detailed explanations of how to use all SoX parameters, file
formats, and effects can be found below in this manual, in
soxformat(4), 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 `header-
less' formats do not contain this information, so the audio
characteristics of these must be described on the SoX com-
mand line or inferred from those of the input file.
The following four characteristics are used to describe the
format of audio data such that it can be processed with SoX:
sample rate
The sample rate in samples per second (`Hertz' or
`Hz'). 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 profes-
sional 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 differ-
ent 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, -law, ADPCM, signed-integer PCM, MP3,
and FLAC.
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channels
The number of audio channels contained in the file.
One (`mono') and two (`stereo') are widely used. `Sur-
round sound' audio typically contains six or more chan-
nels.
The term `bit-rate' is a measure of the amount of storage
occupied by an encoded audio signal over a unit of time. It
can depend on all of the above and is typically denoted as a
number of kilo-bits per second (kbps). An A-law telephony
signal has a bit-rate of 64 kbs. MP3-encoded stereo music
typically has a bit-rate of 128-196 kbps. FLAC-encoded
stereo music typically has a bit-rate of 550-760 kbps.
Most self-describing formats also allow textual `comments'
to be 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, 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 character-
istics 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.
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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
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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 conver-
sion. 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 qual-
ity 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 sig-
nal 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 orig-
inal audio signal can be reproduced when using a lossy
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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 uncom-
pressed 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 con-
vert 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
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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 automati-
cally added. The -D option may be given to override auto-
matic 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 repre-
sentation. 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 con-
verting one format to another, and even when simply playing
the audio.
Playing an audio file often involves resampling, and pro-
cessing by analogue components can introduce a small DC off-
set and/or amplification, all of which can produce distor-
tion 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 recom-
mend 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.
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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 meth-
ods: `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 refer-
ence 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 inter-
val -1 to +1). If the number of channels in the input files
is not the same, the missing channels are considered to con-
tain all zero.
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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. `bal-
ance') 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 adjust-
ments 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 per-
formed. 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 /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 /n instead
of /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.
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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 substi-
tuted. 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 sec-
onds 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 automati-
cally 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: press-
ing 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 sup-
port 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
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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 cir-
cumstances 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 spec-
ify 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 exam-
ple, 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 for-
mat 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
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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 file-
name 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 file-
name to specify that a `null file' is to be used. Note
that here, `null file' refers to a SoX-specific mecha-
nism and is not related to any operating-system mecha-
nism 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 pro-
duce 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(4) for a list and description of the supported
file formats and audio device drivers.
OPTIONS
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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 docu-
ment.
--buffer BYTES, --input-buffer BYTES
Set the size in bytes of the buffers used for process-
ing audio (default 8192). --buffer applies to input,
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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 `Dither' 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. 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.
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-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 for-
mats.
--i, --info
Only if given as the first parameter to sox, behave as
soxi(1).
--interactive
Deprecated alias for --no-clobber.
-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 paral-
lel on hyper-threading/multi-core architectures. This
may reduce processing time, though sometimes it may be
necessary to use this option in conjuction with a
larger buffer size than is the default to gain any ben-
efit 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 ...
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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
Automatically invoke the gain effect to guard against
clipping and to normalise the audio. E.g.
sox --norm infile -b 16 outfile rate 44100 dither -s
is shorthand for
sox infile -b 16 outfile gain -h rate 44100 gain -nh dither -s
See also -V, -G, and the gain effect.
--play-rate-arg ARG
Selects a quality option to be used when the `rate'
effect is automatically invoked whilst playing audio.
This option is typically set via the SOX_OPTS environ-
ment 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
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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 pro-
cessing 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
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to determine if an error has occurred.
1 Only error messages are shown. These are gener-
ated if SoX cannot complete the requested com-
mands.
2 Warning messages are also shown. These are gener-
ated 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 pro-
cessing 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.
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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 com-
plex encodings such as MP3 or GSM. Not necessary with
encodings that have a fixed number of bits, e.g.
A/-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.
-1/-2/-3/-4/-8
The number of bytes in each encoded sample. Deprecated
aliases for -b 8, -b 16, -b 24, -b 32, -b 64 respec-
tively.
-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 cer-
tain file types. Examples:
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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 chan-
nel 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 -3
sox input.wav output.wav bass -3 channels 1
though the second form is more flexible as it allows
the effects to be ordered arbitrarily.
-e ENCODING, --encoding ENCODING
The audio encoding type. Sometimes needed with file-
types that support more than one encoding type. For
example, with raw, WAV, or AU (but not, for example,
with MP3 or FLAC). The available encoding types are as
follows:
signed-integer
PCM data stored as signed (`two's complement')
integers. Commonly used with a 16 or 24 -bit
encoding size. A value of 0 represents minimum
signal power.
unsigned-integer
PCM data stored as signed (`two's complement')
integers. Commonly used with an 8-bit encoding
size. A value of 0 represents maximum signal
power.
floating-point
PCM data stored as IEEE 753 single precision
(32-bit) or double precision (64-bit) floating-
point (`real') 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
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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. -law. It
has a precision equivalent to roughly 14-bit PCM
and is sometimes encoded with reversed bit-order-
ing (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 equiva-
lent 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 sup-
port 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' (`header-
less') 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
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convert raw CD digital audio (16-bit, signed-integer)
to floating-point `WAV' files (single & double preci-
sion respectively).
By default (i.e. if this option is not given), the out-
put encoding type will (providing it is supported by
the output file type) be set to the input encoding
type.
-s/-u/-f/-A/-U/-o/-i/-a/-g
Deprecated aliases for specifying the encoding types
signed-integer, unsigned-integer, floating-point, mu-
law, a-law, oki-adpcm, ima-adpcm, ms-adpcm, gsm-full-
rate respectively (see -e above).
--no-glob
Specifies that filename `globbing' (wild-card matching)
should not be performed by SoX on the following file-
name. 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' (`head-
erless') 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:
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sox input.wav -r 48k output.wav bass -3
sox input.wav output.wav bass -3 rate 48k
though the second form is more flexible as it allows
rate options to be given, and allows the effects to be
ordered 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(4) for a list of supported file types.
-L, --endian little
-B, --endian big
-x, --endian swap
These options specify whether the byte-order of the
audio data is, respectively, `little endian', `big
endian', or the opposite to that of the system on which
SoX is being used. Endianness applies only to data
encoded as floating-pont, or as signed or unsigned
integers of 16 or more bits. It is often necessary to
specify one of these options for headerless files, and
sometimes necessary for (otherwise) self-describing
files. A given endian-setting option may be ignored
for an input file whose header contains a specific
endianness identifier, or for an output file that is
actually an audio device.
N.B. Unlike other format characteristics, the endian-
ness (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
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must be used to preserve big-endianness in the output
file.
The -V option can be used to check the selected order-
ings.
-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 head-
erless) 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 com-
ment 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 com-
pressing file formats. See the description of the file
formats that use this option in soxformat(4) for more
information.
EFFECTS
In addition to converting, playing and recording audio
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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 applica-
tions 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 Effect Chains
A single effects chain is made up of one or more effects.
Audio from the input runs through the chain until either the
end of the input file is reached or an effect in the chain
requests to terminate the chain.
SoX supports running multiple effects 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 sam-
ples that are buffered by effects to the left of the termi-
nating effect will be discarded. The amount of samples dis-
carded 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.
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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:
centre[k]
See frequency.
frequency[k]
A frequency in Hz, or, if appended with `k', kHz.
gain A power gain in dB. Zero gives no gain; less than zero
gives an attenuation.
width[h|k|o|q]
Used to specify the band-width of a filter. A number
of different methods to specify the width are available
(though not all for every effect). One of the charac-
ters 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.
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 relation-
ship without changing its frequency to amplitude
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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 intro-
duces 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 shoul-
ders.
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 shoul-
ders.
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) fre-
quencies 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 equali-
sation (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
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Sound eXchange SoX(1)
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 fol-
lowing 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 tran-
sition. 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.
tion }
bend [-f frame-rate(25)] [-o over-sample(16)] {
delay,cents,dura-
Changes pitch by specified amounts at specified times.
Each given triple: delay,cents,duration specifies one
bend. delay is the amount of time after the start of
the audio stream, or the end of the previous bend, at
which to start bending the pitch; cents is the number
of cents (100 cents = 1 semitone) by which to bend the
pitch, and duration the length of time over which the
pitch will be bent.
The pitch-bending algorithm utilises the Discrete
Fourier 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
Note that the clipping that is produced in this example
is deliberate; to remove it, use gain -5 in place of
gain 1.
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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 coef-
ficients respectively.
See http://en.wikipedia.org/wiki/Digital_biquad_filter
(where a0 = 1).
channels CHANNELS
Invoke a simple algorithm to change the number of chan-
nels in the audio signal to the given number CHANNELS:
mixing if decreasing the number of channels or dupli-
cating if increasing the number of channels.
The channels effect is invoked automatically if SoX's
-c option specifies a number of channels that is dif-
ferent 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 com-
mands are equivalent:
sox input.wav -c 1 output.wav bass -3
sox input.wav output.wav bass -3 channels 1
though the second form is more flexible as it allows
the effects to be ordered arbitrarily.
See also remix for an effect that allows channels to be
mixed/selected arbitrarily.
chorus gain-in gain-out <delay decay speed depth -s|-t>
Add a chorus effect to the audio. This can make a sin-
gle 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 out-
put.
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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 param-
eters 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 compan-
der'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
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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 ini-
tially, 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
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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 enhance-
ment.
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 prob-
lem 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
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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 classi-
cal 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 cor-
rectly on any CD player), or simply play the correctly
de-emphasised audio files on the PC. For example:
sox track1.wav track1-deemph.wav deemph
and then burn track1-deemph.wav to CD, or
play track1-deemph.wav
or simply
play track1.wav deemph
The de-emphasis filter is implemented as a biquad; its
maximum deviation from the ideal response is only
0.06dB (up to 20kHz).
This effect supports the --plot global option.
See also the bass and treble shelving equalisation
effects.
delay {length}
Delay one or more audio channels. length can specify a
time or, if appended with an `s', a number of samples.
Do not specify both time and samples delays in the same
command. For example, delay 1.5 0 0.5 delays the first
channel by 1.5 seconds, the third channel by 0.5 sec-
onds, and leaves the second channel (and any other
channels that may be present) un-delayed. The follow-
ing (one long) command plays a chime sound:
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play -n synth -j 3 sin %3 sin %-2 sin %-5 sin %-9 \
sin %-14 sin %-21 fade h .01 2 1.5 delay \
1.3 1 .76 .54 .27 remix - fade h 0 2.7 2.5 norm -1
and this plays a guitar chord:
play -n synth pl G2 pl B2 pl D3 pl G3 pl D4 pl G4 \
delay 0 .05 .1 .15 .2 .25 remix - fade 0 4 .1 norm -1
dither [-a] [-S|-s|-f filter]
Apply dithering to the audio. Dithering deliberately
adds a small amount of noise to the signal in order to
mask audible quantization effects that can occur if the
output sample size is less than 24 bits. With no
options, this effect will add 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, modi-
fied-e-weighted, improved-e-weighted, gesemann, shi-
bata, low-shibata, high-shibata. Note that most filter
types are available only with 44100Hz sample rate. The
filter types are distinguished by the following proper-
ties: 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 sam-
pling rate but below 22k, plain TPDF is probably bet-
ter, and above 37k, noise-shaped is probably better.
The -a option enables a mode where dithering (and
noise-shaping if applicable) are automatically enabled
only when needed. The most likely use for this is when
applying fade in or out to an already dithered file, so
that the redithering applies only to the faded por-
tions. 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.
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.
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See also the `Dither' section above.
earwax
Makes audio easier to listen to on headphones. Adds
`cues' to 44.1kHz stereo (i.e. audio CD format) audio
so that when listened to on headphones the stereo image
is moved from inside your head (standard for head-
phones) to outside and in front of the listener (stan-
dard for speakers). See http://www.geoci-
ties.com/beinges for a full explanation.
echo gain-in gain-out <delay decay>
Add echoing to the audio. Echoes are reflected sound
and can occur naturally amongst mountains (and some-
times large buildings) when talking or shouting; digi-
tal 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 mil-
liseconds and the decay (relative to gain-in) of that
echo. Gain-out is the volume of the output. For exam-
ple: 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',
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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 differ-
ent central frequency.
The filter is described in detail in [1].
This effect supports the --plot global option.
See also bass and treble for shelving equalisation
effects.
fade [type] fade-in-length [stop-time [fade-out-length]]
Apply a fade effect to the beginning, end, or both of
the audio.
An optional type can be specified to select the shape
of the fade curve: q for quarter of a sine wave, h for
half a sine wave, t for linear (`triangular') slope, l
for logarithmic, and p for inverted parabola. The
default is logarithmic.
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A fade-in starts from the first sample and ramps the
signal level from 0 to full volume over fade-in-length
seconds. Specify 0 seconds if no fade-in is wanted.
For fade-outs, the audio will be truncated at stop-time
and the signal level will be ramped from full volume
down to 0 starting at fade-out-length seconds before
the stop-time. If fade-out-length is not specified, it
defaults to the same value as fade-in-length. No fade-
out is performed if stop-time is not specified. If the
file length can be determined from the input file
header and length-changing effects are not in effect,
then 0 may be specified for stop-time to indicate the
usual case of a fade-out that ends at the end of the
input audio stream.
All times can be specified in either periods of time or
sample counts. To specify time periods use the format
hh:mm:ss.frac format. To specify using sample counts,
specify the number of samples and append the letter `s'
to the sample count (for example `8000s').
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
...
flanger [delay depth regen width speed shape phase interp]
Apply a flanging effect to the audio. See [3] for a
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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 atten-
uation 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 high-
est 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
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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 lim-
iting 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 clip-
ping, 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
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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.
ladspa module [plugin] [argument...]
Apply a LADSPA [5] (Linux Audio Developer's Simple
Plugin API) plugin. Despite the name, LADSPA is not
Linux-specific, and a wide range of effects is avail-
able 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 sec-
ond 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 sup-
plied by default values if possible. Only plugins with
at most one audio input and one audio output port can
be used. If found, the environment variable
LADSPA_PATH will be used as search path for plugins.
loudness [gain [reference]]
Loudness control - similar to the gain effect, but pro-
vides 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]]
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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 sepa-
rately 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 multi-
ple bands.
For example, the following (one long) command shows how
multi-band companding is typically used in FM radio:
play track1.wav gain -3 sinc 8000- 29 100 mcompand \
"0.005,0.1 -47,-40,-34,-34,-17,-33" 100 \
"0.003,0.05 -47,-40,-34,-34,-17,-33" 400 \
"0.000625,0.0125 -47,-40,-34,-34,-15,-33" 1600 \
"0.0001,0.025 -47,-40,-34,-34,-31,-31,-0,-30" 6400 \
"0,0.025 -38,-31,-28,-28,-0,-25" \
gain 15 highpass 22 highpass 22 sinc -n 255 -b 16 -17500 \
gain 9 lowpass -1 17801
The audio file is played with a simulated FM radio
sound (or broadcast signal condition if the lowpass
filter at the end is skipped). Note that the pipeline
is set up with US-style 75us pre-emphasis.
See also compand for a single-band companding effect.
mixer [ -l|-r|-f|-b|-1|-2|-3|-4|n{,n} ]
Reduce the number of audio channels by mixing or
selecting channels, or increase the number of channels
by duplicating channels. Note: this effect operates on
the audio channels within the SoX effects processing
chain; it should not be confused with the -m global
option (where multiple files are mix-combined before
entering the effects chain).
When reducing the number of channels it is possible to
use the -l, -r, -f, -b, -1, -2, -3, -4, options to
select only the left, right, front, back channel(s) or
specific channel for the output instead of averaging
the channels. The -l, and -r options will do averaging
in quad-channel files so select the exact channel to
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prevent this.
The mixer effect can also be invoked with up to 16 num-
bers, separated by commas, which specify the proportion
(0 = 0% and 1 = 100%) of each input channel that is to
be mixed into each output channel. In two-channel
mode, 4 numbers are given: l -> l, l -> r, r -> l, and
r -> r, respectively. In four-channel mode, the first
4 numbers give the proportions for the left-front out-
put channel, as follows: lf -> lf, rf -> lf, lb -> lf,
and rb -> rf. The next 4 give the right-front output
in the same order, then left-back and right-back.
It is also possible to use the 16 numbers to expand or
reduce the channel count; just specify 0 for unused
channels.
Finally, certain reduced combination of numbers can be
specified for certain input/output channel combina-
tions.
In Ch Out Ch Num Mappings
2 1 2 l -> l, r -> l
2 2 1 adjust balance
4 1 4 lf -> l, rf -> l, lb -> l, rb -> l
4 2 2 lf -> l&rf -> r, lb -> l&rb -> r
4 4 1 adjust balance
4 4 2 front balance, back balance
See also remix for a mixing effect that handles any
number of channels.
noiseprof [profile-file]
Calculate a profile of the audio for use in noise
reduction. See the description of the noisered effect
for details.
noisered [profile-file [amount]]
Reduce noise in the audio signal by profiling and fil-
tering. This effect is moderately effective at remov-
ing 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
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according to a noise profile (which was generated by
noiseprof), from profile-file, or from stdin if no pro-
file-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 sec-
tions of the audio.
On most systems, the two stages - profiling and reduc-
tion - can be combined using a pipe, e.g.
sox noisy.wav -n trim 0 1 noiseprof | play noisy.wav noisered
norm [dB-level]
Normalise the audio. norm is just an alias for gain
-n; see the gain effect for details.
Note that norm's -i and -b options are deprecated (hav-
ing been superseded by gain -en and gain -B respec-
tively) and will be removed in a future release.
oops Out Of Phase Stereo effect. Mixes stereo to twin-mono
where each mono channel contains the difference between
the left and right stereo channels. This is sometimes
known as the `karaoke' effect as it often has the
effect of removing most or all of the vocals from a
recording.
overdrive [gain(20) [colour(20)]]
Non linear distortion. The colour parameter controls
the amount of even harmonic content in the over-driven
output.
pad { length[@position] }
Pad the audio with silence, at the beginning, the end,
or any specified points through the audio. Both length
and position can specify a time or, if appended with an
`s', a number of samples. 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
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one. position is optional for the first and last
lengths specified and if omitted correspond to the
beginning and the end of the audio respectively. For
example, pad 1.5 1.5 adds 1.5 seconds of silence pad-
ding 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 speed and tempo effects.
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rate [-q|-l|-m|-h|-v] [override-options] RATE[k]
Change the audio sampling rate (i.e. resample the
audio) to any given RATE (even non-integer if this is
supported by the output file format) using a quality
level defined as follows:
Quality Band- Rej dB Typical Use
width
-q quick n/a ~=30 @ playback on
Fs/4 ancient hardware
-l low 80% 100 playback on old
hardware
-m medium 95% 100 audio playback
-h high 95% 125 16-bit mastering
(use with dither)
-v very high 95% 175 24-bit mastering
where Band-width is the percentage of the audio fre-
quency band that is preserved and Rej dB is the level
of noise rejection. Increasing levels of resampling
quality come at the expense of increasing amounts of
time to process the audio. If no quality option is
given, the quality level used is `high'.
The `quick' algorithm uses cubic interpolation; all
others use band-limited interpolation. By default, all
algorithms have a `linear' phase response; for
`medium', `high' and `very high', the phase response is
configurable (see below).
The rate effect is invoked automatically if SoX's -r
option specifies a rate that is different to that of
the input file(s). Alternatively, if this effect is
given explicitly, then SoX's -r option need not be
given. For example, the following two commands are
equivalent:
sox input.wav -r 48k output.wav bass -3
sox input.wav output.wav bass -3 rate 48k
though the second command is more flexible as it allows
rate options to be given, and allows the effects to be
ordered 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
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detailed in the following table:
-M/-I/-L Phase response = minimum/intermediate/linear
-s Steep filter (band-width = 99%)
-a Allow aliasing/imaging above the pass-band
-b 74-99.7 Any band-width %
-p 0-100 Any phase response (0 = minimum, 25 = intermediate,
50 = linear, 100 = maximum)
N.B. Override options can not be used with the `quick'
or `low' quality algorithms.
All resamplers use filters that can sometimes create
`echo' (a.k.a. `ringing') artefacts with transient
signals such as those that occur with `finger snaps' or
other highly percussive sounds. Such artefacts are
much more noticeable to the human ear if they occur
before the transient (`pre-echo') than if they occur
after it (`post-echo'). Note that frequency of any
such artefacts is related to the smaller of the origi-
nal 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-
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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 dis-
torted; however, since this is above the pass-band
(i.e. above the highest frequency of interest/audibil-
ity), this may not be a problem. The benefits of
allowing aliasing/imaging are reduced processing time,
and reduced (by almost half) transient echo artefacts.
Note that if this option is given, then the minimum
band-width allowable with -b increases to 85%.
Examples:
sox input.wav -b 16 output.wav rate -s -a 44100 dither -s
default (high) quality resampling; overrides: steep
filter, allow aliasing; to 44.1kHz sample rate; noise-
shaped dither to 16-bit WAV file.
sox input.wav -b 24 output.aiff rate -v -I -b 90 48k
very high quality resampling; overrides: intermediate
phase, band-width 90%; to 48k sample rate; store output
to 24-bit AIFF file.
* * *
The pitch, speed and tempo effects all use the rate
effect at their core.
remix [-a|-m|-p] <out-spec>
out-spec = in-spec{,in-spec} | 0
in-spec = [in-chan][-[in-chan2]][vol-spec]
vol-spec = p|i|v[volume]
Select and mix input audio channels into output audio
channels. Each output channel is specified, in turn,
by a given out-spec: a list of contributing input chan-
nels 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).
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An out-spec contains comma-separated input channel-num-
bers and hyphen-delimited channel-number ranges; alter-
natively, 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 chan-
nels 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 mul-
tiple (n) input channels, each input channel will be
scaled by a factor of /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, /n scaling is not applied to any other chan-
nels 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,
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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 corre-
sponds 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 /n scal-
ing is replaced by /n (`power') scaling; this gives a
louder mix but one that might occasionally clip.
* * *
One use of the remix effect is to split an audio file
into a set of files, each containing one of the con-
stituent channels (in order to perform subsequent pro-
cessing on individual audio channels). Where more than
a few channels are involved, a script such as the fol-
lowing (Bourne shell script) is useful:
#!/bin/sh
chans=`soxi -c "$1"`
while [ $chans -ge 1 ]; do
chans0=`printf %02i $chans` # 2 digits hence up to 99 chans
out=`echo "$1"|sed "s/\(.*\)\.\(.*\)/\1-$chans0.\2/"`
sox "$1" "$out" remix $chans
chans=`expr $chans - 1`
done
If a file input.wav containing six audio channels were
given, the script would produce six output files:
input-01.wav, input-02.wav, ..., input-06.wav.
See also mixer and swap for similar effects.
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repeat count
Repeat the entire audio count times. Requires tempo-
rary file space to store the audio to be repeated.
Note that repeating once yields two copies: the origi-
nal 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 desir-
able 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 rever-
beration.
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, inde-
pendently 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
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from the beginning. When specifying an non-zero above-
periods, it trims audio up until it finds non-silence.
Normally, when trimming silence from beginning of audio
the above-periods will be 1 but it can be increased to
higher values to trim all audio up to a specific count
of non-silence periods. For example, if you had an
audio file with two songs that each contained 2 seconds
of silence before the song, you could specify an above-
period of 2 to strip out both silence periods and the
first song.
When above-periods is non-zero, you must also specify a
duration and threshold. Duration indications the amount
of time that non-silence must be detected before it
stops trimming audio. By increasing the duration, burst
of noise can be treated as silence and trimmed off.
Threshold is used to indicate what sample value you
should treat as silence. For digital audio, a value of
0 may be fine but for audio recorded from analog, you
may wish to increase the value to account for back-
ground 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 sec-
ond 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 reli-
ably. A work around is to use the silence effect in
combination with the reverse effect. By first revers-
ing the audio, you can use the above-periods to reli-
ably trim all audio from what looks like the front of
the file. Then reverse the file again to get back to
normal.
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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 indi-
cate the effect should restart processing as specified
by the above-periods, making it suitable for removing
periods of silence in the middle of the audio.
The option -l indicates that below-periods duration
length of audio should be left intact at the beginning
of each period of silence. For example, if you want to
remove long pauses between words but do not want to
remove the pauses completely.
The period counts are in units of samples. Duration
counts may be in the format of hh:mm:ss.frac, or the
exact count of samples. Threshold numbers may be suf-
fixed 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%
qHP][-freqLP [-t tbw|-n taps]]
sinc [-a att|-b beta] [-p phase|-M|-I|-L] [-t tbw|-n taps]
[fre-
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 < freqLP creates a band-pass filter, freqHP
> freqLP creates a band-reject filter.
The default stop-band attenuation of 120dB can be over-
ridden 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.
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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, fre-
quency 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 start-
ing 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:
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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 fol-
lowing to the `chime' command in the description of the
delay effect (above):
rate 2k spectrogram -X 200 -Z -10 -w kaiser
Options are also avaliable to control the appearance
(colour-set, brightness, contrast, etc.) and filename
of the 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 spec-
trogram from its default value of 800 pixels to a
given number between 100 and 5000. See also -X
and -d.
-X num
X-axis pixels/second; the default is auto-calcu-
lated to fit the given or known audio duration to
the X-axis size, or 100 otherwise. If given in
conjunction with -d, this option affects the width
of the spectrogram; otherwise, it affects the
duration of the spectrogram. num can be from 1
(low time resolution) to 5000 (high time resolu-
tion) and need not be an integer. SoX may make a
slight adjustment to the given number for process-
ing quantisation reasons; if so, SoX will report
the actual number used (viewable when the SoX
global option -V is in effect). See also -x and
-d.
-y num
Sets the Y-axis size in pixels (per channel); this
is the number of frequency `bins' used in the
Fourier analysis that produces the spectrogram.
N.B. it can be slow to produce the spectrogram if
this number is not one more than a power of two
(e.g. 129). By default the Y-axis size is chosen
automatically (depending on the number of chan-
nels). See -Y for alternative way of setting
spectrogram height.
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-Y num
Sets the target total height of the spectro-
gram(s). The default value is 550 pixels. Using
this option (and by default), SoX will choose a
height for individual spectrogram channels that is
one more than a power of two, so the actual total
height may fall short of the given number. How-
ever, there is also a minimum height per channel
so if there are many channels, the number may be
exceeded. See -y for alternative way of setting
spectrogram height.
-z num
Z-axis (colour) range in dB, default 120. This
sets the dynamic-range of the spectrogram to be
-num dBFS to 0 dBFS. Num may range from 20 to
180. Decreasing dynamic-range effectively
increases the `contrast' of the spectrogram dis-
play, and vice versa.
-Z num
Sets the upper limit of the Z-axis in dBFS. A
negative num effectively increases the `bright-
ness' of the spectrogram display, and vice versa.
-q num
Sets the Z-axis quantisation, i.e. the number of
different colours (or intensities) in which to
render Z-axis values. A small number (e.g. 4)
will give a `poster'-like effect making it easier
to discern magnitude bands of similar level.
Small numbers also usually result in small PNG
files. The number given specifies the number of
colours to use inside the Z-axis range; two
colours are reserved to represent out-of-range
values.
-w name
Window: Hann (default), Hamming, Bartlett, Rectan-
gular or Kaiser. The spectrogram is produced
using the Discrete Fourier Transform (DFT) algo-
rithm. 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-resolu-
tion), select a Kaiser window. Bartlett and Rect-
angular windows are also available.
-W num
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Window adjustment parameter. This can be used to
make small adjustments to the Kaiser window shape.
A positive number (up to ten) increases its
dynamic range, a 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 lev-
els. If this option is used in conjunction with
-m, the result will be a hybrid monochrome/colour
palette.
-p num
Permute the colours in a colour or hybrid palette.
The num parameter, from 1 (the default) to 6,
selects the permutation.
-l Creates a `printer friendly' spectrogram with a
light background (the default has a dark back-
ground).
-a Suppress the display of the axis lines. This is
sometimes useful in helping to discern artefacts
at the spectrogram edges.
-r Raw spectrogram: suppress the display of axes and
legends.
-A Selects an alternative, fixed colour-set. This is
provided only for compatibility with spectrograms
produced by another package. It should not nor-
mally be used as it has some problems, not least,
a lack of differentiation at the bottom end which
results in masking of low-level artefacts.
-t text
Set the image title - text to display above the
spectrogram.
-c text
Set (or clear) the image comment - text to display
below and to the left of the spectrogram.
-o text
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Name of the spectrogram output PNG file, default
`spectrogram.png'.
Advanced Options:
In order to process a smaller section of audio without
affecting other effects or the output signal (unlike
when the trim effect is used), the following options
may be used.
-d duration
This option sets the X-axis resolution such that
audio with the given duration ([[HH:]MM:]SS) fits
the selected (or default) X-axis width. For exam-
ple,
sox input.mp3 output.wav -n spectrogram -d 1:00 stats
creates a spectrogram showing the first minute of
the audio, whilst
the stats effect is applied to the entire audio
signal.
See also -X for an alternative way of setting the
X-axis resolution.
-S time
Start the spectrogram at the given point in the
audio stream. For example
sox input.aiff output.wav spectrogram -S 1:00
creates a spectrogram showing all but the first
minute of the audio (the output file however,
receives the entire audio stream).
For the ability to perform off-line processing of spec-
tral data, see the stat effect.
speed factor[c]
Adjust the audio speed (pitch and tempo together).
factor is either the ratio of the new speed to the old
speed: greater than 1 speeds up, less than 1 slows
down, or, if appended with the letter `c', the number
of cents (i.e. 100ths of a semitone) by which the pitch
(and tempo) should be adjusted: greater than 0
increases, less than 0 decreases.
By default, the speed change is performed by resampling
with the rate effect using its default quality/speed.
For higher quality or higher speed resampling, in addi-
tion to the speed effect, specify the rate effect with
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the desired quality option.
See also the pitch and tempo effects.
splice [-h|-t|-q] { position[,excess[,leeway]] }
Splice together audio sections. This effect provides
two things over simple audio concatenation: a (usually
short) cross-fade is applied at the join, and a wave
similarity comparison is made to help determine the
best place at which to make the join.
One of the options -h, -t, or -q may be given to select
the fade envelope as triangular (a.k.a. linear) (the
default), half-cosine wave, or quarter-cosine wave
respectively.
Type Audio Fade level Transitions
t correlated constant gain abrupt
h correlated constant gain smooth
q uncorrelated constant power smooth
To perform a splice, first use the trim effect to
select the audio sections to be joined together. As
when performing a tape splice, the end of the section
to be spliced onto should be trimmed with a small
excess (default 0.005 seconds) of audio after the ideal
joining point. The beginning of the audio section to
splice on should be trimmed with the same excess
(before the ideal joining point), plus an additional
leeway (default 0.005 seconds). SoX should then be
invoked with the two audio sections as input files and
the splice effect given with the position at which to
perform the splice - this is length of the first audio
section (including the excess).
For example, a long song begins with two verses which
start (as determined e.g. by using the play command
with the trim (start) effect) at times 0:30.125 and
1:03.432. The following commands cut out the first
verse:
sox too-long.wav part1.wav trim 0 30.130
(5 ms excess, after the first verse starts)
sox too-long.wav part2.wav trim 1:03.422
(5 ms excess plus 5 ms leeway, before the second verse
starts)
sox part1.wav part2.wav just-right.wav splice 30.130
For another example, the SoX command
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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 pre-
ceding the splice effect with fade q .01 2 .01).
Provided your arithmetic is good enough, multiple
splices can be performed with a single splice invoca-
tion. For example:
#!/bin/sh
# Audio Copy and Paste Over
# acpo infile copy-start copy-stop paste-over-start outfile
# All times measured in samples.
rate=`soxi -r "$1"`
e=`expr $rate '*' 5 / 1000` # Using default excess
l=$e # and leeway.
sox "$1" piece.wav trim `expr $2 - $e - $l`s \
`expr $3 - $2 + $e + $l + $e`s
sox "$1" part1.wav trim 0 `expr $4 + $e`s
sox "$1" part2.wav trim `expr $4 + $3 - $2 - $e - $l`s
sox part1.wav piece.wav part2.wav "$5" splice \
`expr $4 + $e`s \
`expr $4 + $e + $3 - $2 + $e + $l + $e`s
In the above Bourne shell script, two splices are used
to `copy and paste' audio.
* * *
It is also possible to use this effect to perform gen-
eral 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 informa-
tion about the audio. Audio is passed unmodified
through the SoX processing chain.
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The information is output to the `standard error'
(stderr) stream and is calculated, where n is the dura-
tion of the audio in samples, c is the number of audio
channels, r is the audio sample rate, and xk represents
the PCM value (in the range -1 to +1 by default) of
each successive sample in the audio, as follows:
the dcshift
effect).
the vol effect
which would make
the audio as
loud as possible
without clip-
ping. Note: See
the discussion
on Clipping
above for rea-
sons why it is
rarely a good
idea actually to
do this.
Samples read nxc
Length (seconds) nr
Scaled by See -s below.
Maximum amplitude max(xk) The maximum sam-
ple value in the
audio; usually
this will be a
positive number.
Minimum amplitude min(xk) The minimum sam-
ple value in the
audio; usually
this will be a
negative number.
Midline amplitude 1/2min(xk)+1/2max(xk)
Mean norm /n|xk| The average of
the absolute
value of each
sample in the
audio.
Mean amplitude /nxk 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
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RMS amplitude (/nxk) 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 /n-1|xk-xk-1|
RMS delta (/n-1(xk-xk-1))
Rough frequency In Hz.
Volume Adjustment The parameter to
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
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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 measure-
ments 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. consecu-
tive samples with the same value) of the signal at its
peak levels (i.e. either Min level, or Max level).
Pk count is the number of occasions (not the number of
samples) that the signal attained either Min level, or
Max level.
The right-hand Bit-depth figure is the standard defini-
tion of bit-depth i.e. bits less significant than the
given number are fixed at zero. The left-hand figure
is the number of most significant bits that are fixed
at zero (or one for negative numbers) subtracted from
the right-hand figure (the number subtracted is
directly 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 magni-
tude; Max level, Pk lev dB, RMS Pk dB, Bit-depth:
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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. 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 gen-
eral, 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.
bine] [[%]freq[k][:|+|/|-[%]freq2[k]]] [off [ph [p1 [p2 [p3]]]]]}
synth [-j KEY] [-n] [len [off [ph [p1 [p2 [p3]]]]]] {[type]
[com-
This effect can be used to generate fixed or swept fre-
quency audio tones with various wave shapes, or to gen-
erate 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,
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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 can has an asso-
ciated 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
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given to disable this behaviour.
A detailed description of each synth parameter follows:
len is the length of audio to synthesise expressed as a
time or as a number of samples; 0=inputlength,
default=0.
The format for specifying lengths in time is
hh:mm:ss.frac. The format for specifying sample counts
is the number of samples with the letter `s' appended
to it.
type is one of sine, square, triangle, sawtooth,
trapezium, exp, [white]noise, tpdfnoise pinknoise,
brownnoise, pluck; default=sine.
combine is one of create, mix, amod (amplitude modula-
tion), fmod (frequency modulation); default=create.
freq/freq2 are the frequencies at the beginning/end of
synthesis in Hz or, if preceded with `%', semitones
relative to A (440 Hz); alternatively, `scientific'
note notation (e.g. E2) may be used. The default fre-
quency 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 inte-
ger 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 sepa-
rated 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 num-
ber of semitones per second.
- Exponential: as `/', but initial phase always
zero, and stepped (less smooth) frequency changes.
Not used for noise.
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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 (trapez-
ium), 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 mea-
surement 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.
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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 seg-
ment. 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 automati-
cally adjusted based on segment.
The optional overlap parameter gives the segment over-
lap length in milliseconds. Default value is 12, but
-m, -s, or -l flags automatically adjust overlap based
on segment size. Increasing overlap increases process-
ing time and may increase quality. A practical maximum
for overlap is the value of search, with overlap typi-
cally being (at least) a little smaller then search.
See also speed for an effect that changes tempo and
pitch together, pitch for an effect that changes tempo
and pitch together, and stretch for an effect that
changes tempo using a different algorithm.
treble gain [frequency[k] [width[s|h|k|o|q]]]
Apply a treble tone-control effect. See the descrip-
tion of the bass effect for details.
tremolo speed [depth]
Apply a tremolo (low frequency amplitude modulation)
effect to the audio. The tremolo frequency in Hz is
given by speed, and the depth as a percentage by depth
(default 40).
trim start [length|=end]
Trim can trim off unwanted audio from the beginning and
end of the audio. Audio is not sent to the output
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stream until the start location is reached.
The optional length parameter gives the length of audio
to output after the start sample and is thus used to
trim off the end of the audio. Alternatively, an abso-
lute end location can be given by preceding it with an
equals sign. Using a value of 0 for the start parame-
ter will allow trimming off the end only.
Both parameters can be specified using either an amount
of time or an exact count of samples. The format for
specifying lengths in time is hh:mm:ss.frac. A start
value of 1:30.5 will not start until 1 minute, thirty
and 1/2 seconds into the audio. The format for speci-
fying sample counts is the number of samples with the
letter `s' appended to it. A value of 8000s for the
start parameter will wait until 8000 samples are read
before starting to process audio.
vad [options]
Voice Activity Detector. Attempts to trim silence and
quiet background sounds from the ends of (fairly high
resolution i.e. 16-bit, 44-48kHz) recordings of speech.
The algorithm currently uses a simple cepstral power
measurement to detect voice, so may be fooled by other
things, especially music. The effect can trim only
from the front of the audio, so in order to trim from
the back, the reverse effect must also be used. E.g.
play speech.wav norm vad
to trim from the front,
play speech.wav norm reverse vad reverse
to trim from the back, and
play speech.wav norm vad reverse vad reverse
to trim from both ends. The use of the norm effect is
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 depend-
ing on the noise level, signal level and other
charactistics of the input audio.
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-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 qui-
eter/shorter bursts.
Advanced Options:
These allow fine tuning of the alogithm'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/measure-
ments.
-m num
Measurement duration; by default, twice the mea-
surement period; i.e. with overlap.
-M num
Time constant used to smooth spectral measure-
ments.
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-h num
`Brick-wall' frequency of high-pass filter applied
at the input to the detector algorithm.
-l num
`Brick-wall' frequency of low-pass filter applied
at the input to the detector algorithm.
-H num
`Brick-wall' quefrency of high-pass lifter used in
the detector algorithm.
-L num
`Brick-wall' quefrency of low-pass lifter used in
the detector algorithm.
See also the silence effect.
vol gain [type [limitergain]]
Apply an amplification or an attenuation to the audio
signal. Unlike the -v option (which is used for bal-
ancing 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 fol-
lows: 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)
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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 differ-
ent capabilities, and compand for a dynamic-range com-
pression/expansion/limiting effect.
Deprecated Effects
The following effects have been renamed or have their func-
tionality included in another effect; they continue to work
in this version of SoX but may be removed in future.
filter [low]-[high] [window-len [beta]]
Apply a sinc-windowed lowpass, highpass, or bandpass
filter of given window length to the signal. This
effect has been superseded by the sinc effect. Com-
pared with `sinc', `filter' is slower and has fewer
capabilities.
low refers to the frequency of the lower 6dB corner of
the filter. high refers to the frequency of the upper
6dB corner of the filter.
A low-pass filter is obtained by leaving low unspeci-
fied, or 0. A high-pass filter is obtained by leaving
high unspecified, or 0, or greater than or equal to the
Nyquist frequency.
The window-len, if unspecified, defaults to 128.
Longer windows give a sharper cut-off, smaller windows
a more gradual cut-off.
The beta parameter determines the type of filter window
used. Any value greater than 2 is the beta for a
Kaiser window. Beta <= 2 selects a Blackman-Nuttall
window. If unspecified, the default is a Kaiser window
with beta 16.
In the case of Kaiser window (beta > 2), lower betas
produce a somewhat faster transition from pass-band to
stop-band, at the cost of noticeable artifacts. A beta
of 16 is the default, beta less than 10 is not recom-
mended. If you want a sharper cut-off, don't use low
beta's, use a longer sample window. A Blackman-Nuttall
window is selected by specifying any `beta' <= 2, and
the Blackman-Nuttall window has somewhat steeper cut-
off than the default Kaiser window. You will probably
not need to use the beta parameter at all, unless you
are just curious about comparing the effects of Black-
man-Nuttall vs. Kaiser windows.
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This effect supports the --plot global option.
key [-q] shift [segment [search [overlap]]]
Change the audio key (i.e. pitch but not tempo). This
is just an alias for the pitch effect.
pan direction
Mix the audio from one channel to another. Use mixer
or remix instead of this effect.
The direction is a value from -1 to 1. -1 represents
far left and 1 represents far right.
polyphase [-w nut|ham] [-width n] [-cut-off c]
rabbit [-c0|-c1|-c2|-c3|-c4]
resample [-qs|-q|-ql] [rolloff [beta]]
Formerly sample-rate-changing effects in their own
right, these are now just aliases for the rate effect.
DIAGNOSTICS
Exit status is 0 for no error, 1 if there is a problem with
the 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 ([email protected]).
ATTRIBUTES
See attributes(5) for descriptions of the following
attributes:
+---------------+------------------+
|ATTRIBUTE TYPE | ATTRIBUTE VALUE |
+---------------+------------------+
|Availability | audio/sox |
+---------------+------------------+
|Stability | Uncommitted |
+---------------+------------------+
SEE ALSO
soxi(1), soxformat(4), 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.source-
forge.net/Docs/Scripts
References
[1] R. Bristow-Johnson, Cookbook formulae for audio EQ
biquad filter coefficients,
http://musicdsp.org/files/Audio-EQ-Cookbook.txt
sox Last change: February 19, 2011 73
Sound eXchange SoX(1)
[2] Wikipedia, Q-factor,
http://en.wikipedia.org/wiki/Q_factor
[3] Scott Lehman, Effects Explained, http://harmony-cen-
tral.com/Effects/effects-explained.html
[4] Wikipedia, Decibel, http://en.wikipedia.org/wiki/Deci-
bel
[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 1998-2011 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 use-
ful, but WITHOUT ANY WARRANTY; without even the implied war-
ranty of MERCHANTABILITY or FITNESS FOR A PARTICULAR PUR-
POSE. See the GNU General Public License for more details.
AUTHORS
Chris Bagwell ([email protected]). Other
authors and contributors are listed in the ChangeLog file
that is distributed with the source code.
NOTES
This software was built from source available at
https://java.net/projects/solaris-userland. The original
community source was downloaded from http://down-
loads.source-
forge.net/project/sox/sox/14.3.2/sox-14.3.2.tar.gz
Further information about this software can be found on the
open source community website at http://sox.source-
forge.net/.
sox Last change: February 19, 2011 74