sox
Sound eXchange, the Swiss Army knife of audio manipulation
see also :
soxi - audacity - wget
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]] ...
add an example, a script, a trick and tips
examples
source
how to remove clipping from audio using sox?
If the track was recorded and clipped when recorded, you cannot
"fix" the clip by reducing the volume. The recorded signal will
still exhibit the clip, but at a reduced volume.
source
Using sox low pass filter with jack
SoX doesn’t have a JACK driver. It wouldn’t be too difficult to
write one, but due to the architecture of SoX, the latency would
always be higher than strictly necessary.
If all you need is simple IIR filtering, just use a LADSPA host
such as JACK Rack and suitable plugins, such as those by
Matthias Nagorni or
buttlow_iir/butthigh_iir etc. from
Steve Harris’s
collection.
source
equivalent of sox for video
source
Can't run sox, No such file or directory
It looks like your shell has cached the location of
sox for some reason. You can work around the problem
by giving the full path of the command:
/usr/bin/sox message.wav message.flac rate 16k
Try hash -r to clear your shell's command cache.
Also try alias | grep sox in case you have an alias
set to a bad location.
description
Introduction
SoX reads and writes audio files in most popular formats and
can optionally apply effects to them. It can combine
multiple input sources, synthesise audio, and, on many
systems, act as a general purpose audio player or a
multi-track audio recorder. It also has limited ability to
split the input into multiple output files.
All SoX
functionality is available using just the sox
command. To simplify playing and recording audio, if SoX is
invoked as play, the output file is automatically set
to be the default sound device, and if invoked as
rec, the default sound device is used as an input
source. Additionally, the soxi(1) command provides a
convenient way to just query audio file header
information.
The heart of
SoX is a library called libSoX. Those interested in
extending SoX or using it in other programs should refer to
the libSoX manual page: libsox(3).
SoX is a
command-line audio processing tool, particularly suited to
making quick, simple edits and to batch processing. If you
need an interactive, graphical audio editor, use
audacity(1).
The overall SoX
processing chain can be summarised as follows:
Note however,
that on the SoX command line, the positions of the Output(s)
and the Effects are swapped w.r.t. the logical flow just
shown. Note also that whilst options pertaining to files are
placed before their respective file name, the opposite is
true for effects. To show how this works in practice, here
is a selection of examples of how SoX might be used. The
simple
sox recital.au recital.wav
translates an audio file in Sun
AU format to a Microsoft WAV file, whilst
sox recital.au -b 16 recital.wav channels 1 rate 16k fade 3 norm
performs the same format
translation, but also applies four effects (down-mix to one
channel, sample rate change, fade-in, nomalize), and stores
the result at a bit-depth of 16.
sox -r 16k -e signed -b 8 -c 1 voice-memo.raw voice-memo.wav
converts ’raw’
(a.k.a. ’headerless’) audio to a self-describing
file format,
sox slow.aiff fixed.aiff speed 1.027
adjusts audio speed,
sox short.wav long.wav longer.wav
concatenates two audio files,
and
sox -m music.mp3 voice.wav mixed.flac
mixes together two audio
files.
play "The Moonbeams/Greatest/*.ogg" bass +3
plays a collection of audio
files whilst applying a bass boosting effect,
play -n -c1 synth sin %-12 sin %-9 sin %-5 sin %-2 fade h 0.1 1 0.1
plays a synthesised ’A
minor seventh’ chord with a pipe-organ sound,
rec -c 2 radio.aiff trim 0 30:00
records half an hour of stereo
audio, and
play -q take1.aiff & rec -M take1.aiff take1-dub.aiff
(with POSIX shell and where
supported by hardware) records a new track in a multi-track
recording. Finally,
rec -r 44100 -b 16 -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(7), and in soxi(1).
File Format
Types
SoX can work with ’self-describing’ and
’raw’ audio files. ’self-describing’
formats (e.g. WAV, FLAC, MP3) have a header that completely
describes the signal and encoding attributes of the audio
data that follows. ’raw’ or
’headerless’ formats do not contain this
information, so the audio characteristics of these must be
described on the SoX command line or inferred from those of
the input file.
The following
four characteristics are used to describe the format of
audio data such that it can be processed with SoX:
sample rate
The sample rate in samples per
second (’Hertz’ or ’Hz’). Digital
telephony traditionally uses a sample rate of 8000 Hz
(8 kHz), though these days, 16 and even 32 kHz are
becoming more common. Audio Compact Discs use 44100 Hz
(44.1 kHz). Digital Audio Tape and many computer
systems use 48 kHz. Professional audio systems often
use 96 kHz.
sample size
The number of bits used to
store each sample. Today, 16-bit is commonly used. 8-bit was
popular in the early days of computer audio. 24-bit is used
in the professional audio arena. Other sizes are also
used.
data encoding
The way in which each audio
sample is represented (or ’encoded’). Some
encodings have variants with different byte-orderings or
bit-orderings. Some compress the audio data so that the
stored audio data takes up less space (i.e. disk space or
transmission bandwidth) than the other format parameters and
the number of samples would imply. Commonly-used encoding
types include floating-point, μ-law, ADPCM,
signed-integer PCM, MP3, and FLAC.
channels
The number of audio channels
contained in the file. One (’mono’) and two
(’stereo’) are widely used. ’Surround
sound’ audio typically contains six or more
channels.
The term
’bit-rate’ is a measure of the amount of storage
occupied by an encoded audio signal over a unit of time. It
can depend on all of the above and is typically denoted as a
number of kilo-bits per second (kbps). An A-law telephony
signal has a bit-rate of 64 kbps. MP3-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
characteristics may be determined or set using different
mechanisms.
To determine
the format of an input file, SoX will use, in order of
precedence and as given or available:
1.
Command-line format options.
2.
The contents of the file header.
3.
The filename extension.
To set the
output file format, SoX will use, in order of precedence and
as given or available:
1.
Command-line format options.
2.
The filename extension.
3.
The input file format characteristics, or the closest
that is supported by the output file type.
For all files,
SoX will exit with an error if the file type cannot be
determined. Command-line format options may need to be added
or changed to resolve the problem.
Playing
& Recording Audio
The play and rec commands are provided so that
basic playing and recording is as simple as
play existing-file.wav
and
rec new-file.wav
These two commands are
functionally equivalent to
sox existing-file.wav -d
and
sox -d new-file.wav
Of course, further options and
effects (as described below) can be added to the commands in
either form.
Some systems
provide more than one type of (SoX-compatible) audio driver,
e.g. ALSA & OSS, or SUNAU & AO. Systems can also
have more than one audio device (a.k.a. ’sound
card’). If more than one audio driver has been
built-in to SoX, and the default selected by SoX when
recording or playing is not the one that is wanted, then the
AUDIODRIVER environment variable can be used to
override the default. For example (on many systems):
set AUDIODRIVER=oss
play ...
The AUDIODEV environment
variable can be used to override the default audio device,
e.g.
set AUDIODEV=/dev/dsp2
play ...
sox ... -t oss
or
set AUDIODEV=hw:soundwave,1,2
play ...
sox ... -t alsa
Note that the way of setting
environment variables varies from system to system - for
some specific examples, see ’SOX_OPTS’
below.
When playing a
file with a sample rate that is not supported by the audio
output device, SoX will automatically invoke the rate
effect to perform the necessary sample rate conversion. For
compatibility with old hardware, the default rate
quality level is set to ’low’. This can be
changed by explicitly specifying the rate effect with
a different quality level, e.g.
play ... rate -m
or by using the
--play-rate-arg option (see
below).
On some
systems, SoX allows audio playback volume to be adjusted
whilst using play. Where supported, this is achieved
by tapping the ’v’ & ’V’ keys
during playback.
To help with
setting a suitable recording level, SoX includes a
peak-level meter which can be invoked (before making the
actual recording) as follows:
rec -n
The recording level should be
adjusted (using the system-provided mixer program, not SoX)
so that the meter is at most occasionally full scale,
and never ’in the red’ (an exclamation mark is
shown). See also -S below.
Accuracy
Many file formats that compress audio discard some of the
audio signal information whilst doing so. Converting to such
a format and then converting back again will not produce an
exact copy of the original audio. This is the case for many
formats used in telephony (e.g. A-law, GSM) where low signal
bandwidth is more important than high audio fidelity, and
for many formats used in portable music players (e.g. MP3,
Vorbis) where adequate fidelity can be retained even with
the large compression ratios that are needed to make
portable players practical.
Formats that
discard audio signal information are called
’lossy’. Formats that do not are called
’lossless’. The term ’quality’ is
used as a measure of how closely the original audio signal
can be reproduced when using a lossy format.
Audio file
conversion with SoX is lossless when it can be, i.e. when
not using lossy compression, when not reducing the sampling
rate or number of channels, and when the number of bits used
in the destination format is not less than in the source
format. E.g. converting from an 8-bit PCM format to a 16-bit
PCM format is lossless but converting from an 8-bit PCM
format to (8-bit) A-law isn’t.
N.B. SoX
converts all audio files to an internal uncompressed format
before performing any audio processing. This means that
manipulating a file that is stored in a lossy format can
cause further losses in audio fidelity. E.g. with
sox long.mp3 short.mp3 trim 10
SoX first decompresses the
input MP3 file, then applies the trim effect, and
finally creates the output MP3 file by re-compressing the
audio - with a possible reduction in fidelity above that
which occurred when the input file was created. Hence, if
what is ultimately desired is lossily compressed audio, it
is highly recommended to perform all audio processing using
lossless file formats and then convert to the lossy format
only at the final stage.
N.B.
Applying multiple effects with a single SoX invocation will,
in general, produce more accurate results than those
produced using multiple SoX invocations.
Dithering
Dithering is a technique used to maximise the dynamic range
of audio stored at a particular bit-depth. Any distortion
introduced by quantisation is decorrelated by adding a small
amount of white noise to the signal. In most cases, SoX can
determine whether the selected processing requires dither
and will add it during output formatting if appropriate.
Specifically,
by default, SoX automatically adds TPDF dither when the
output bit-depth is less than 24 and any of the following
are true:
•
bit-depth reduction has been
specified explicitly using a command-line option
•
the output file format supports only bit-depths lower
than that of the input file format
•
an effect has increased effective bit-depth within the
internal processing chain
For example,
adjusting volume with vol 0.25 requires two
additional bits in which to losslessly store its results
(since 0.25 decimal equals 0.01 binary). So if the input
file bit-depth is 16, then SoX’s internal
representation will utilise 18 bits after processing this
volume change. In order to store the output at the same
depth as the input, dithering is used to remove the
additional bits.
Use the
-V option to see what processing SoX has
automatically added. The -D option may be given
to override automatic dithering. To invoke dithering
manually (e.g. to select a noise-shaping curve), see the
dither effect.
Clipping
Clipping is distortion that occurs when an audio signal
level (or ’volume’) exceeds the range of the
chosen representation. In most cases, clipping is
undesirable and so should be corrected by adjusting the
level prior to the point (in the processing chain) at which
it occurs.
In SoX,
clipping could occur, as you might expect, when using the
vol or gain effects to increase the audio
volume. Clipping could also occur with many other effects,
when converting one format to another, and even when simply
playing the audio.
Playing an
audio file often involves resampling, and processing by
analogue components can introduce a small DC offset and/or
amplification, all of which can produce distortion if the
audio signal level was initially too close to the clipping
point.
For these
reasons, it is usual to make sure that an audio file’s
signal level has some ’headroom’, i.e. it does
not exceed a particular level below the maximum possible
level for the given representation. Some standards bodies
recommend as much as 9dB headroom, but in most cases, 3dB
(≈ 70% linear) is enough. Note that this wisdom seems
to have been lost in modern music production; in fact, many
CDs, MP3s, etc. are now mastered at levels above
0dBFS i.e. the audio is clipped as delivered.
SoX’s
stat and stats effects can assist in
determining the signal level in an audio file. The
gain or vol effect can be used to prevent
clipping, e.g.
sox dull.wav bright.wav gain -6 treble +6
guarantees that the treble
boost will not clip.
If clipping
occurs at any point during processing, SoX will display a
warning message to that effect.
See also
-G and the gain and norm
effects.
Input File
Combining
SoX’s input combiner can be configured (see OPTIONS
below) to combine multiple files using any of the following
methods: ’concatenate’, ’sequence’,
’mix’, ’mix-power’,
’merge’, or ’multiply’. The default
method is ’sequence’ for play, and
’concatenate’ for rec and sox.
For all methods
other than ’sequence’, multiple input files must
have the same sampling rate. If necessary, separate SoX
invocations can be used to make sampling rate adjustments
prior to combining.
If the
’concatenate’ combining method is selected
(usually, this will be by default) then the input files must
also have the same number of channels. The audio from each
input will be concatenated in the order given to form the
output file.
The
’sequence’ combining method is selected
automatically for play. It is similar to
’concatenate’ in that the audio from each input
file is sent serially to the output file. However, here the
output file may be closed and reopened at the corresponding
transition between input files. This may be just what is
needed when sending different types of audio to an output
device, but is not generally useful when the output is a
normal file.
If either the
’mix’ or ’mix-power’ combining
method is selected then two or more input files must be
given and will be mixed together to form the output file.
The number of channels in each input file need not be the
same, but SoX will issue a warning if they are not and some
channels in the output file will not contain audio from
every input file. A mixed audio file cannot be un-mixed
without reference to the original input files.
If the
’merge’ combining method is selected then two or
more input files must be given and will be merged together
to form the output file. The number of channels in each
input file need not be the same. A merged audio file
comprises all of the channels from all of the input files.
Un-merging is possible using multiple invocations of SoX
with the remix effect. For example, two mono files
could be merged to form one stereo file. The first and
second mono files would become the left and right channels
of the stereo file.
The
’multiply’ combining method multiplies the
sample values of corresponding channels (treated as numbers
in the interval -1 to +1). If the number of channels
in the input files is not the same, the missing channels are
considered to contain all zero.
When combining
input files, SoX applies any specified effects (including,
for example, the vol volume adjustment effect) after
the audio has been combined. However, it is often useful to
be able to set the volume of (i.e. ’balance’)
the inputs individually, before combining takes place.
For all
combining methods, input file volume adjustments can be made
manually using the -v option (below) which can
be given for one or more input files. If it is given for
only some of the input files then the others receive no
volume adjustment. In some circumstances, automatic volume
adjustments may be applied (see below).
The
-V option (below) can be used to show the input
file volume adjustments that have been selected (either
manually or automatically).
There are some
special considerations that need to made when mixing input
files:
Unlike the
other methods, ’mix’ combining has the potential
to cause clipping in the combiner if no balancing is
performed. In this case, if manual volume adjustments are
not given, SoX will try to ensure that clipping does not
occur by automatically adjusting the volume (amplitude) of
each input signal by a factor of ¹/ n ,
where n is the number of input files. If this results in
audio that is too quiet or otherwise unbalanced then the
input file volumes can be set manually as described above.
Using the norm effect on the mix is another
alternative.
If mixed audio
seems loud enough at some points but too quiet in others
then dynamic range compression should be applied to correct
this - see the compand effect.
With the
’mix-power’ combine method, the mixed volume is
approximately equal to that of one of the input signals.
This is achieved by balancing using a factor of ¹/
√n instead of ¹/ n
. Note that this balancing factor does not guarantee that
clipping will not occur, but the number of clips will
usually be low and the resultant distortion is generally
imperceptible.
Output
Files
SoX’s default behaviour is to take one or more input
files and write them to a single output file.
This behaviour
can be changed by specifying the pseudo-effect
’newfile’ within the effects list. SoX will then
enter multiple output mode.
In multiple
output mode, a new file is created when the effects prior to
the ’newfile’ indicate they are done. The
effects chain listed after ’newfile’ is then
started up and its output is saved to the new file.
In multiple
output mode, a unique number will automatically be appended
to the end of all filenames. If the filename has an
extension then the number is inserted before the extension.
This behaviour can be customized by placing a %n anywhere in
the filename where the number should be substituted. An
optional number can be placed after the % to indicate a
minimum fixed width for the number.
Multiple output
mode is not very useful unless an effect that will stop the
effects chain early is specified before the
’newfile’. If end of file is reached before the
effects chain stops itself then no new file will be created
as it would be empty.
The following
is an example of splitting the first 60 seconds of an input
file into two 30 second files and ignoring the rest.
sox song.wav ringtone%1n.wav trim 0 30 : newfile : trim 0 30
Stopping
SoX
Usually SoX will complete its processing and exit
automatically once it has read all available audio data from
the input files.
If desired, it
can be terminated earlier by sending an interrupt signal to
the process (usually by pressing the keyboard interrupt key
which is normally Ctrl-C). This is a natural requirement in
some circumstances, e.g. when using SoX to make a recording.
Note that when using SoX to play multiple files, Ctrl-C
behaves slightly differently: pressing it once causes SoX to
skip to the next file; pressing it twice in quick succession
causes SoX to exit.
Another option
to stop processing early is to use an effect that has a time
period or sample count to determine the stopping point. The
trim effect is an example of this. Once all effects chains
have stopped then SoX will also stop.
options
Global
Options
These options can be specified on the command line at any
point before the first effect name.
The
SOX_OPTS environment variable can be used to provide
alternative default values for SoX’s global options.
For example:
SOX_OPTS="--buffer 20000 --play-rate-arg -hs --temp /mnt/temp"
Note that setting SOX_OPTS can
potentially create unwanted changes in the behaviour of
scripts or other programs that invoke SoX. SOX_OPTS might
best be used for things (such as in the given example) that
reflect the environment in which SoX is being run. Enabling
options such as --no-clobber as
default might be handled better using a shell alias since a
shell alias will not affect operation in scripts etc.
One way to
ensure that a script cannot be affected by SOX_OPTS is to
clear SOX_OPTS at the start of the script, but this of
course loses the benefit of SOX_OPTS carrying some
system-wide default options. An alternative approach is to
explicitly invoke SoX with default option values, e.g.
SOX_OPTS="-V --no-clobber"
sox -V2 --clobber $input $output ...
Note that the way to set
environment variables varies from system to system. Here are
some examples:
Unix bash:
export SOX_OPTS="-V --no-clobber"
Unix csh:
setenv SOX_OPTS "-V --no-clobber"
MS-DOS/MS-Windows:
set SOX_OPTS=-V --no-clobber
MS-Windows GUI: via Control
Panel : System : Advanced : Environment Variables
Mac OS X GUI:
Refer to Apple’s Technical Q&A QA1067 document.
--buffer BYTES,
--input-buffer BYTES
Set the size in bytes of the
buffers used for processing audio (default 8192).
--buffer applies to input, effects, and
output processing; --input-buffer
applies only to input processing (for which it overrides
--buffer if both are given).
Be aware that
large values for --buffer will cause SoX
to be become slow to respond to requests to terminate or to
skip the current input file.
--clobber
Don’t prompt before
overwriting an existing file with the same name as that
given for the output file. This is the default
behaviour.
--combine
concatenate|merge|mix|mix-power|multiply|sequence
Select the input file combining
method; for some of these, short options are available:
-m selects ’mix’, -M
selects ’merge’, and -T selects
’multiply’.
See Input
File Combining above for a description of the different
combining methods.
-D,
--no-dither
Disable automatic dither - see
’Dithering’ above. An example of why this might
occasionally be useful is if a file has been converted from
16 to 24 bit with the intention of doing some processing on
it, but in fact no processing is needed after all and the
original 16 bit file has been lost, then, strictly speaking,
no dither is needed if converting the file back to 16 bit.
See also the stats effect for how to determine the
actual bit depth of the audio within a file.
--effects-file
FILENAME
Use FILENAME to obtain all
effects and their arguments. The file is parsed as if the
values were specified on the command line. A new line can be
used in place of the special : marker to separate
effect chains. For convenience, such markers at the end of
the file are normally ignored; if you want to specify an
empty last effects chain, use an explicit : by itself
on the last line of the file. This option causes any effects
specified on the command line to be discarded.
-G,
--guard
Automatically invoke the
gain effect to guard against clipping. E.g.
sox -G infile -b 16 outfile rate 44100 dither -s
is shorthand for
sox infile -b 16 outfile gain -h rate 44100 gain -rh dither -s
See also -V,
--norm, and the gain effect.
-h,
--help
Show version number and usage
information.
--help-effect
NAME
Show usage information on the
specified effect. The name all can be used to show
usage on all effects.
--help-format
NAME
Show information about the
specified file format. The name all can be used to
show information on all formats.
--i,
--info
Only if given as the first
parameter to sox, behave as soxi(1).
-m|-M
Equivalent to --combine mix and
--combine merge, respectively.
--magic
If SoX has been built with the
optional ’libmagic’ library then this option can
be given to enable its use in helping to detect audio file
types.
--multi-threaded
| --single-threaded
By default, SoX is
’single threaded’. If the
--multi-threaded option is given
however then SoX will process audio channels for most
multi-channel effects in parallel on
hyper-threading/multi-core architectures. This may reduce
processing time, though sometimes it may be necessary to use
this option in conjuction with a larger buffer size than is
the default to gain any benefit from multi-threaded
processing (e.g. 131072; see --buffer
above).
--no-clobber
Prompt before overwriting an
existing file with the same name as that given for the
output file.
N.B.
Unintentionally overwriting a file is easier than you might
think, for example, if you accidentally enter
sox file1 file2 effect1 effect2 ...
when what you really meant
was
play file1 file2 effect1 effect2 ...
then, without this option,
file2 will be overwritten. Hence, using this option is
recommended. SOX_OPTS (above), a ’shell’ alias,
script, or batch file may be an appropriate way of
permanently enabling it.
--norm[=dB-level]
Automatically invoke the
gain effect to guard against clipping and to
normalise the audio. E.g.
sox --norm infile -b 16 outfile rate 44100 dither -s
is shorthand for
sox infile -b 16 outfile gain -h rate 44100 gain -nh dither -s
Optionally, the audio can be
normalized to a given level (usually) below 0 dBFS:
sox --norm=-3 infile outfile
See also
-V, -G, and the gain effect.
--play-rate-arg
ARG
Selects a quality option to be
used when the ’rate’ effect is automatically
invoked whilst playing audio. This option is typically set
via the SOX_OPTS environment variable (see
above).
--plot
gnuplot|octave|off
If not set to off (the
default if --plot is not given), run in a
mode that can be used, in conjunction with the gnuplot
program or the GNU Octave program, to assist with the
selection and configuration of many of the transfer-function
based effects. For the first given effect that supports the
selected plotting program, SoX will output commands to plot
the effect’s transfer function, and then exit without
actually processing any audio. E.g.
sox --plot octave input-file -n highpass 1320 > highpass.plt
octave highpass.plt
-q,
--no-show-progress
Run in quiet mode when SoX
wouldn’t otherwise do so. This is the opposite of the
-S option.
-R
Run in ’repeatable’ mode. When this option
is given, where applicable, SoX will embed a fixed
time-stamp in the output file (e.g. AIFF) and will
’seed’ pseudo random number generators (e.g.
dither) with a fixed number, thus ensuring that
successive SoX invocations with the same inputs and the same
parameters yield the same output.
--replay-gain
track|album|off
Select whether or not to apply
replay-gain adjustment to input files. The default is
off for sox and rec, album for
play where (at least) the first two input files are
tagged with the same Artist and Album names, and
track for play otherwise.
-S,
--show-progress
Display input file
format/header information, and processing progress as input
file(s) percentage complete, elapsed time, and remaining
time (if known; shown in brackets), and the number of
samples written to the output file. Also shown is a
peak-level meter, and an indication if clipping has
occurred. The peak-level meter shows up to two channels and
is calibrated for digital audio as follows (right channel
shown):
A three-second
peak-held value of headroom in dBs will be shown to the
right of the meter if this is below 6dB.
This option is
enabled by default when using SoX to play or record
audio.
-T
Equivalent to --combine multiply.
--temp
DIRECTORY
Specify that any temporary
files should be created in the given DIRECTORY. This
can be useful if there are permission or free-space problems
with the default location. In this case, using
’--temp .’ (to use the
current directory) is often a good solution.
--version
Show SoX’s version number
and exit.
-V[level]
Set verbosity. This is
particularly useful for seeing how any automatic effects
have been invoked by SoX.
SoX displays
messages on the console (stderr) according to the following
verbosity levels:
0
No messages are shown at all; use the exit status to
determine if an error has occurred.
1
Only error messages are shown. These are generated if
SoX cannot complete the requested commands.
2
Warning messages are also shown. These are generated if
SoX can complete the requested commands, but not exactly
according to the requested command parameters, or if
clipping occurs.
3
Descriptions of SoX’s processing phases are also
shown. Useful for seeing exactly how SoX is processing your
audio.
4 and above
Messages to help with debugging
SoX are also shown.
By default, the
verbosity level is set to 2 (shows errors and warnings).
Each occurrence of the -V option increases the
verbosity level by 1. Alternatively, the verbosity level can
be set to an absolute number by specifying it immediately
after the -V, e.g. -V0 sets it to
0.
Input File
Options
These options apply only to input files and may precede only
input filenames on the command line.
--ignore-length
Override an (incorrect) audio
length given in an audio file’s header. If this option
is given then SoX will keep reading audio until it reaches
the end of the input file.
-v,
--volume FACTOR
Intended for use when combining
multiple input files, this option adjusts the volume of the
file that follows it on the command line by a factor of
FACTOR. This allows it to be ’balanced’
w.r.t. the other input files. This is a linear (amplitude)
adjustment, so a number less than 1 decreases the volume and
a number greater than 1 increases it. If a negative number
is given then in addition to the volume adjustment, the
audio signal will be inverted.
See also the
norm, vol, and gain effects, and see
Input File Balancing above.
Input &
Output File Format Options
These options apply to the input or output file whose name
they immediately precede on the command line and are used
mainly when working with headerless file formats or when
specifying a format for the output file that is different to
that of the input file.
-b BITS, --bits
BITS
The number of bits (a.k.a.
bit-depth or sometimes word-length) in each encoded sample.
Not applicable to complex encodings such as MP3 or GSM. Not
necessary with encodings that have a fixed number of bits,
e.g. A/μ-law, ADPCM.
For an input
file, the most common use for this option is to inform SoX
of the number of bits per sample in a ’raw’
(’headerless’) audio file. For example
sox -r 16k -e signed -b 8 input.raw output.wav
converts a particular
’raw’ file to a self-describing
’WAV’ file.
For an output
file, this option can be used (perhaps along with
-e) to set the output encoding size. By default
(i.e. if this option is not given), the output encoding size
will (providing it is supported by the output file type) be
set to the input encoding size. For example
sox input.cdda -b 24 output.wav
converts raw CD digital audio
(16-bit, signed-integer) to a 24-bit (signed-integer)
’WAV’ file.
-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 respectively.
-c
CHANNELS, --channels
CHANNELS
The number of audio channels in
the audio file. This can be any number greater than
zero.
For an input
file, the most common use for this option is to inform SoX
of the number of channels in a ’raw’
(’headerless’) audio file. Occasionally, it may
be useful to use this option with a ’headered’
file, in order to override the (presumably incorrect) value
in the header - note that this is only supported with
certain file types. Examples:
sox -r 48k -e float -b 32 -c 2 input.raw output.wav
converts a particular
’raw’ file to a self-describing
’WAV’ file.
play -c 1 music.wav
interprets the file data as
belonging to a single channel regardless of what is
indicated in the file header. Note that if the file does in
fact have two channels, this will result in the file playing
at half speed.
For an output
file, this option provides a shorthand for specifying that
the channels effect should be invoked in order to
change (if necessary) the number of channels in the audio
signal to the number given. For example, the following two
commands are equivalent:
sox input.wav -c 1 output.wav bass -b 24
sox input.wav output.wav bass -b 24 channels 1
though the second form is more
flexible as it allows the effects to be ordered
arbitrarily.
-e
ENCODING, --encoding
ENCODING
The audio encoding type.
Sometimes needed with file-types that support more than one
encoding type. For example, with raw, WAV, or AU (but not,
for example, with MP3 or FLAC). The available encoding types
are as follows:
signed-integer
PCM data stored as signed
(’two’s complement’) integers. Commonly
used with a 16 or 24 -bit encoding size. A value of 0
represents minimum signal power.
unsigned-integer
PCM data stored as unsigned
integers. Commonly used with an 8-bit encoding size. A value
of 0 represents maximum signal power.
floating-point
PCM data stored as IEEE 753
single precision (32-bit) or double precision (64-bit)
floating-point (’real’) numbers. A value of 0
represents minimum signal power.
a-law
International telephony standard for logarithmic
encoding to 8 bits per sample. It has a precision equivalent
to roughly 13-bit PCM and is sometimes encoded with reversed
bit-ordering (see the -X option).
u-law, mu-law
North American telephony
standard for logarithmic encoding to 8 bits per sample.
A.k.a. μ-law. It has a precision equivalent to roughly
14-bit PCM and is sometimes encoded with reversed
bit-ordering (see the -X option).
oki-adpcm
OKI (a.k.a. VOX, Dialogic, or
Intel) 4-bit ADPCM; it has a precision equivalent to roughly
12-bit PCM. ADPCM is a form of audio compression that has a
good compromise between audio quality and encoding/decoding
speed.
ima-adpcm
IMA (a.k.a. DVI) 4-bit ADPCM;
it has a precision equivalent to roughly 13-bit PCM.
ms-adpcm
Microsoft 4-bit ADPCM; it has a
precision equivalent to roughly 14-bit PCM.
gsm-full-rate
GSM is currently used for the
vast majority of the world’s digital wireless
telephone calls. It utilises several audio formats with
different bit-rates and associated speech quality. SoX has
support for GSM’s original 13kbps ’Full
Rate’ audio format. It is usually CPU-intensive to
work with GSM audio.
Encoding names can be abbreviated where this would not
be ambiguous; e.g. ’unsigned-integer’ can be
given as ’un’, but not ’u’
(ambiguous with ’u-law’).
For an input
file, the most common use for this option is to inform SoX
of the encoding of a ’raw’
(’headerless’) audio file (see the examples in
-b and -c above).
For an output
file, this option can be used (perhaps along with
-b) to set the output encoding type For
example
sox input.cdda -e float output1.wav
sox input.cdda -b 64 -e float output2.wav
convert raw CD digital audio
(16-bit, signed-integer) to floating-point ’WAV’
files (single & double precision respectively).
By default
(i.e. if this option is not given), the output encoding type
will (providing it is supported by the output file type) be
set to the input encoding type.
-s/-u/-f/-A/-U/-o/-i/-a/-g
Deprecated aliases for
specifying the encoding types signed-integer,
unsigned-integer, floating-point,
a-law, mu-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 filename. For example, if
the current directory contains the two files
’five-seconds.wav’ and ’five*.wav’,
then
play --no-glob "five*.wav"
can be used to play just the
single file ’five*.wav’.
-r,
--rate RATE[k]
Gives the sample rate in Hz (or
kHz if appended with ’k’) of the file.
For an input
file, the most common use for this option is to inform SoX
of the sample rate of a ’raw’
(’headerless’) audio file (see the examples in
-b and -c above). Occasionally it
may be useful to use this option with a
’headered’ file, in order to override the
(presumably incorrect) value in the header - note that this
is only supported with certain file types. For example, if
audio was recorded with a sample-rate of say 48k from a
source that played back a little, say 1.5%, too slowly,
then
sox -r 48720 input.wav output.wav
effectively corrects the speed
by changing only the file header (but see also the
speed effect for the more usual solution to this
problem).
For an output
file, this option provides a shorthand for specifying that
the rate effect should be invoked in order to change
(if necessary) the sample rate of the audio signal to the
given value. For example, the following two commands are
equivalent:
sox input.wav -r 48k output.wav bass -b 24
sox input.wav output.wav bass -b 24 rate 48k
though the second form is more
flexible as it allows rate options to be given, and
allows the effects to be ordered arbitrarily.
-t,
--type FILE-TYPE
Gives the type of the audio
file. For both input and output files, this option is
commonly used to inform SoX of the type a
’headerless’ audio file (e.g. raw, mp3) where
the actual/desired type cannot be determined from a given
filename extension. For example:
another-command | sox -t mp3 - output.wav
sox input.wav -t raw output.bin
It can also be used to override
the type implied by an input filename extension, but if
overriding with a type that has a header, SoX will exit with
an appropriate error message if such a header is not
actually present.
See
soxformat(7) for a list of supported file types.
-L,
--endian little
-B, --endian big
-x, --endian swap
These options specify whether the byte-order of the
audio data is, respectively, ’little endian’,
’big endian’, or the opposite to that of the
system on which SoX is being used. Endianness applies only
to data encoded as floating-point, or as signed or unsigned
integers of 16 or more bits. It is often necessary to
specify one of these options for headerless files, and
sometimes necessary for (otherwise) self-describing files. A
given endian-setting option may be ignored for an input file
whose header contains a specific endianness identifier, or
for an output file that is actually an audio device.
N.B.
Unlike other format characteristics, the endianness (byte,
nibble, & bit ordering) of the input file is not
automatically used for the output file; so, for example,
when the following is run on a little-endian system:
sox -B audio.s16 trimmed.s16 trim 2
trimmed.s16 will be created as
little-endian;
sox -B audio.s16 -B trimmed.s16 trim 2
must be used to preserve
big-endianness in the output file.
The
-V option can be used to check the selected
orderings.
-N,
--reverse-nibbles
Specifies that the nibble
ordering (i.e. the 2 halves of a byte) of the samples should
be reversed; sometimes useful with ADPCM-based formats.
N.B. See
also N.B. in section on -x above.
-X,
--reverse-bits
Specifies that the bit ordering
of the samples should be reversed; sometimes useful with a
few (mostly headerless) formats.
N.B. See
also N.B. in section on -x above.
Output File
Format Options
These options apply only to the output file and may precede
only the output filename on the command line.
--add-comment TEXT
Append a comment in the output
file header (where applicable).
--comment
TEXT
Specify the comment text to
store in the output file header (where applicable).
SoX will
provide a default comment if this option (or
--comment-file) is not given. To
specify that no comment should be stored in the output file,
use --comment "" .
--comment-file
FILENAME
Specify a file containing the
comment text to store in the output file header (where
applicable).
-C,
--compression FACTOR
The compression factor for
variably compressing output file formats. If this option is
not given then a default compression factor will apply. The
compression factor is interpreted differently for different
compressing file formats. See the description of the file
formats that use this option in soxformat(7) for more
information.
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.
effects
In addition to converting, playing and recording audio files, SoX
can be used to invoke a number of audio ’effects’. Multiple
effects may be applied by specifying them one after another at
the end of the SoX command line, forming an ’effects chain’. Note
that applying multiple effects in real-time (i.e. when playing
audio) is likely to require a high performance computer. Stopping
other applications may alleviate performance issues should they
occur.
Some of the SoX effects are primarily intended to be applied to a
single instrument or ’voice’. To facilitate this, the
remix effect and the global SoX option -M can be
used to isolate then recombine tracks from a multi-track
recording.
Multiple Effects Chains
A single effects chain is made up of one or more effects. Audio
from the input runs through the chain until either the end of the
input file is reached or an effect in the chain requests to
terminate the chain.
SoX supports running multiple effects chains over the input
audio. In this case, when one chain indicates it is done
processing audio, the audio data is then sent through the next
effects chain. This continues until either no more effects chains
exist or the input has reached the end of the file.
An effects chain is terminated by placing a : (colon)
after an effect. Any following effects are a part of a new
effects chain.
It is important to place the effect that will stop the chain as
the first effect in the chain. This is because any samples that
are buffered by effects to the left of the terminating effect
will be discarded. The amount of samples discarded is related to
the --buffer option and it should be kept small, relative
to the sample rate, if the terminating effect cannot be first.
Further information on stopping effects can be found in the
Stopping SoX section.
There are a few pseudo-effects that aid using multiple effects
chains. These include newfile which will start writing to
a new output file before moving to the next effects chain and
restart which will move back to the first effects chain.
Pseudo-effects must be specified as the first effect in a chain
and as the only effect in a chain (they must have a :
before and after they are specified).
The following is an example of multiple effects chains. It will
split the input file into multiple files of 30 seconds in length.
Each output filename will have unique number in its name as
documented in the Output Files section.
sox infile.wav output.wav trim 0 30 : newfile : restart
Common Notation And Parameters
In the descriptions that follow, brackets [ ] are used to denote
parameters that are optional, braces { } to denote those that are
both optional and repeatable, and angle brackets < > to
denote those that are repeatable but not optional. Where
applicable, default values for optional parameters are shown in
parenthesis ( ).
The following parameters are used with, and have the same meaning
for, several effects:
center[k]
See frequency.
frequency[k]
A frequency in Hz, or, if appended with ’k’, kHz.
gain
A power gain in dB. Zero gives no gain; less than zero gives an
attenuation.
width[h|k|o|q]
Used to specify the band-width of a filter. A number of different
methods to specify the width are available (though not all for
every effect). One of the characters shown may be appended to
select the desired method as follows:
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 relationship
without changing its frequency to amplitude relationship. The
filter is described in detail in [1].
This effect supports the --plot global option.
band [-n] center[k]
[width[h|k|o|q]]
Apply a band-pass filter. The frequency response drops
logarithmically around the center frequency. The
width parameter gives the slope of the drop. The
frequencies at center + width and center -
width will be half of their original amplitudes.
band defaults to a mode oriented to pitched audio, i.e.
voice, singing, or instrumental music. The -n (for noise)
option uses the alternate mode for un-pitched audio (e.g.
percussion). Warning: -n introduces a power-gain of about
11dB in the filter, so beware of output clipping. band
introduces noise in the shape of the filter, i.e. peaking at the
center frequency and settling around it.
This effect supports the --plot global option.
See also sinc for a bandpass filter with steeper
shoulders.
bandpass|bandreject [-c]
frequency[k]
width[h|k|o|q]
Apply a two-pole Butterworth band-pass or band-reject filter with
central frequency frequency, and (3dB-point) band-width
width. The -c option applies only to
bandpass and selects a constant skirt gain (peak gain = Q)
instead of the default: constant 0dB peak gain. The filters roll
off at 6dB per octave (20dB per decade) and are described in
detail in [1].
These effects support the --plot global option.
See also sinc for a bandpass filter with steeper
shoulders.
bandreject frequency[k]
width[h|k|o|q]
Apply a band-reject filter. See the description of the
bandpass effect for details.
bass|treble gain [frequency[k]
[width[s|h|k|o|q]]]
Boost or cut the bass (lower) or treble (upper) frequencies of
the audio using a two-pole shelving filter with a response
similar to that of a standard hi-fi’s tone-controls. This is also
known as shelving equalisation (EQ).
gain gives the gain at 0 Hz (for bass), or
whichever is the lower of ∼22 kHz and the Nyquist frequency
(for treble). Its useful range is about -20 (for a large
cut) to +20 (for a large boost). Beware of Clipping when
using a positive gain.
If desired, the filter can be fine-tuned using the following
optional parameters:
frequency sets the filter’s central frequency and so can
be used to extend or reduce the frequency range to be boosted or
cut. The default value is 100 Hz (for bass) or
3 kHz (for treble).
width determines how steep is the filter’s shelf
transition. In addition to the common width specification methods
described above, ’slope’ (the default, or if appended with
’s’) may be used. The useful range of ’slope’ is about
0.3, for a gentle slope, to 1 (the maximum), for a steep slope;
the default value is 0.5.
The filters are described in detail in [1].
These effects support the --plot global option.
See also equalizer for a peaking equalisation effect.
bend [-f frame-rate(25)] [-o
over-sample(16)] {
delay,cents,duration }
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.
See also pitch.
biquad b0 b1 b2 a0 a1 a2
Apply a biquad IIR filter with the given coefficients. Where b*
and a* are the numerator and denominator coefficients
respectively.
See http://en.wikipedia.org/wiki/Digital_biquad_filter (where a0
= 1).
This effect supports the --plot global option.
channels CHANNELS
Invoke a simple algorithm to change the number of channels in the
audio signal to the given number CHANNELS: mixing if
decreasing the number of channels or duplicating if increasing
the number of channels.
The channels effect is invoked automatically if SoX’s
-c option specifies a number of channels that is different
to that of the input file(s). Alternatively, if this effect is
given explicitly, then SoX’s -c option need not be given.
For example, the following two commands are equivalent:
sox input.wav -c 1 output.wav bass -b 24
sox input.wav output.wav bass -b 24 channels 1
though the second form is more flexible as it allows the effects
to be ordered arbitrarily.
See also remix for an effect that allows channels to be
mixed/selected arbitrarily.
chorus gain-in gain-out <delay decay speed
depth -s|-t>
Add a chorus effect to the audio. This can make a single vocal
sound like a chorus, but can also be applied to instrumentation.
Chorus resembles an echo effect with a short delay, but whereas
with echo the delay is constant, with chorus, it is varied using
sinusoidal or triangular modulation. The modulation depth defines
the range the modulated delay is played before or after the
delay. Hence the delayed sound will sound slower or faster, that
is the delayed sound tuned around the original one, like in a
chorus where some vocals are slightly off key. See [3] for more
discussion of the chorus effect.
Each four-tuple parameter delay/decay/speed/depth gives the delay
in milliseconds and the decay (relative to gain-in) with a
modulation speed in Hz using depth in milliseconds. The
modulation is either sinusoidal (-s) or triangular
(-t). Gain-out is the volume of the output.
A typical delay is around 40ms to 60ms; the modulation speed is
best near 0.25Hz and the modulation depth around 2ms. For
example, a single delay:
play guitar1.wav chorus 0.7 0.9 55 0.4 0.25 2 -t
Two delays of the original samples:
play guitar1.wav chorus 0.6 0.9 50 0.4 0.25 2 -t \
60 0.32 0.4 1.3 -s
A fuller sounding chorus (with three additional delays):
play guitar1.wav chorus 0.5 0.9 50 0.4 0.25 2 -t \
60 0.32 0.4 2.3 -t 40 0.3 0.3 1.3 -s
compand
attack1,decay1{,attack2,decay2}
[soft-knee-dB:]in-dB1[,out-dB1]{,in-dB2,out-dB2}
[gain [initial-volume-dB [delay]]]
Compand (compress or expand) the dynamic range of the audio.
The attack and decay parameters (in seconds)
determine the time over which the instantaneous level of the
input signal is averaged to determine its volume; attacks refer
to increases in volume and decays refer to decreases. For most
situations, the attack time (response to the music getting
louder) should be shorter than the decay time because the human
ear is more sensitive to sudden loud music than sudden soft
music. Where more than one pair of attack/decay parameters are
specified, each input channel is companded separately and the
number of pairs must agree with the number of input channels.
Typical values are 0.3,0.8 seconds.
The second parameter is a list of points on the compander’s
transfer function specified in dB relative to the maximum
possible signal amplitude. The input values must be in a strictly
increasing order but the transfer function does not have to be
monotonically rising. If omitted, the value of out-dB1
defaults to the same value as in-dB1; levels below
in-dB1 are not companded (but may have gain applied to
them). The point 0,0 is assumed but may be overridden (by
0,out-dBn). If the list is preceded by a
soft-knee-dB value, then the points at where adjacent line
segments on the transfer function meet will be rounded by the
amount given. Typical values for the transfer function are
6:-70,-60,-20.
The third (optional) parameter is an additional gain in dB to be
applied at all points on the transfer function and allows easy
adjustment of the overall gain.
The fourth (optional) parameter is an initial level to be assumed
for each channel when companding starts. This permits the user to
supply a nominal level initially, so that, for example, a very
large gain is not applied to initial signal levels before the
companding action has begun to operate: it is quite probable that
in such an event, the output would be severely clipped while the
compander gain properly adjusts itself. A typical value (for
audio which is initially quiet) is -90 dB.
The fifth (optional) parameter is a delay in seconds. The input
signal is analysed immediately to control the compander, but it
is delayed before being fed to the volume adjuster. Specifying a
delay approximately equal to the attack/decay times allows the
compander to effectively operate in a ’predictive’ rather than a
reactive mode. A typical value is 0.2 seconds.
The following example might be used to make a piece of music with
both quiet and loud passages suitable for listening to in a noisy
environment such as a moving vehicle:
sox asz.wav asz-car.wav compand 0.3,1 6:-70,-60,-20 -5 -90 0.2
The transfer function (’6:-70,...’) says that very soft sounds
(below -70dB) will remain unchanged. This will stop the compander
from boosting the volume on ’silent’ passages such as between
movements. However, sounds in the range -60dB to 0dB (maximum
volume) will be boosted so that the 60dB dynamic range of the
original music will be compressed 3-to-1 into a 20dB range, which
is wide enough to enjoy the music but narrow enough to get around
the road noise. The ’6:’ selects 6dB soft-knee companding. The -5
(dB) output gain is needed to avoid clipping (the number is
inexact, and was derived by experimentation). The -90 (dB) for
the initial volume will work fine for a clip that starts with
near silence, and the delay of 0.2 (seconds) has the effect of
causing the compander to react a bit more quickly to sudden
volume changes.
In the next example, compand is being used as a noise-gate for
when the noise is at a lower level than the signal:
play infile compand .1,.2 -inf,-50.1,-inf,-50,-50 0 -90 .1
Here is another noise-gate, this time for when the noise is at a
higher level than the signal (making it, in some ways, similar to
squelch):
play infile compand .1,.1 -45.1,-45,-inf,0,-inf 45 -90 .1
This effect supports the --plot global option (for the
transfer function).
See also mcompand for a multiple-band companding effect.
contrast [enhancement-amount(75)]
Comparable with compression, this effect modifies an audio signal
to make it sound louder. enhancement-amount controls the
amount of the enhancement and is a number in the range 0-100.
Note that enhancement-amount = 0 still gives a significant
contrast enhancement.
See also the compand and mcompand effects.
dcshift shift [limitergain]
Apply a DC shift to the audio. This can be useful to remove a DC
offset (caused perhaps by a hardware problem in the recording
chain) from the audio. The effect of a DC offset is reduced
headroom and hence volume. The stat or stats effect
can be used to determine if a signal has a DC offset.
The given dcshift value is a floating point number in the
range of ±2 that indicates the amount to shift the audio (which
is in the range of ±1).
An optional limitergain can be specified as well. It
should have a value much less than 1 (e.g. 0.05 or 0.02) and is
used only on peaks to prevent clipping.
An alternative approach to removing a DC offset (albeit with a
short delay) is to use the highpass filter effect at a
frequency of say 10Hz, as illustrated in the following example:
sox -n dc.wav synth 5 sin %0 50
sox dc.wav fixed.wav highpass 10
deemph
Apply Compact Disc (IEC 60908) de-emphasis (a treble attenuation
shelving filter).
Pre-emphasis was applied in the mastering of some CDs issued in
the early 1980s. These included many classical music albums, as
well as now sought-after issues of albums by The Beatles, Pink
Floyd and others. Pre-emphasis should be removed at playback time
by a de-emphasis filter in the playback device. However, not all
modern CD players have this filter, and very few PC CD drives
have it; playing pre-emphasised audio without the correct
de-emphasis filter results in audio that sounds harsh and is far
from what its creators intended.
With the deemph effect, it is possible to apply the
necessary de-emphasis to audio that has been extracted from a
pre-emphasised CD, and then either burn the de-emphasised audio
to a new CD (which will then play correctly on any CD player), or
simply play the correctly de-emphasised audio files on the PC.
For example:
sox track1.wav track1-deemph.wav deemph
and then burn track1-deemph.wav to CD, or
play track1-deemph.wav
or simply
play track1.wav deemph
The de-emphasis filter is implemented as a biquad; 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 seconds, and leaves the second
channel (and any other channels that may be present) un-delayed.
The following (one long) command plays a chime sound:
play -n synth -j 3 sin %3 sin %-2 sin %-5 sin %-9 \
sin %-14 sin %-21 fade h .01 2 1.5 delay \
1.3 1 .76 .54 .27 remix - fade h 0 2.7 2.5 norm -1
and this plays a guitar chord:
play -n synth pl G2 pl B2 pl D3 pl G3 pl D4 pl G4 \
delay 0 .05 .1 .15 .2 .25 remix - fade 0 4 .1 norm -1
dither [-S|-s|-f filter]
[-a] [-p precision]
Apply dithering to the audio. Dithering deliberately adds a small
amount of noise to the signal in order to mask audible
quantization effects that can occur if the output sample size is
less than 24 bits. With no options, this effect will add
triangular (TPDF) white noise. Noise-shaping (only for certain
sample rates) can be selected with -s. With the -f
option, it is possible to select a particular noise-shaping
filter from the following list: lipshitz, f-weighted,
modified-e-weighted, improved-e-weighted, gesemann, shibata,
low-shibata, high-shibata. Note that most filter types are
available only with 44100Hz sample rate. The filter types are
distinguished by the following properties: audibility of noise,
level of (inaudible, but in some circumstances, otherwise
problematic) shaped high frequency noise, and processing
speed.
See http://sox.sourceforge.net/SoX/NoiseShaping for graphs of the
different noise-shaping curves.
The -S option selects a slightly ’sloped’ TPDF, biased
towards higher frequencies. It can be used at any sampling rate
but below ≈22k, plain TPDF is probably better, and above ≈ 37k,
noise-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 portions. However, auto dithering is not
fool-proof, so the fades should be carefully checked for any
noise modulation; if this occurs, then either re-dither the whole
file, or use trim, fade, and concatencate.
The -p option allows overriding the target precision.
If the SoX global option -R option is not given, then the
pseudo-random number generator used to generate the white noise
will be ’reseeded’, i.e. the generated noise will be different
between invocations.
This effect should not be followed by any other effect that
affects the audio.
See also the ’Dithering’ section above.
downsample [factor(2)]
Downsample the signal by an integer factor: Only the first out of
each factor samples is retained, the others are discarded.
No decimation filter is applied. If the input is not a properly
bandlimited baseband signal, aliasing will occur. This may be
desirable, e.g., for frequency translation.
For a general resampling effect with anti-aliasing, see
rate. See also upsample.
earwax
Makes audio easier to listen to on headphones. Adds ’cues’ to
44.1kHz stereo (i.e. audio CD format) audio so that when listened
to on headphones the stereo image is moved from inside your head
(standard for headphones) to outside and in front of the listener
(standard for speakers).
echo gain-in gain-out <delay decay>
Add echoing to the audio. Echoes are reflected sound and can
occur naturally amongst mountains (and sometimes large buildings)
when talking or shouting; digital echo effects emulate this
behaviour and are often used to help fill out the sound of a
single instrument or vocal. The time difference between the
original signal and the reflection is the ’delay’ (time), and the
loudness of the reflected signal is the ’decay’. Multiple echoes
can have different delays and decays.
Each given delay decay pair gives the delay in
milliseconds and the decay (relative to gain-in) of that echo.
Gain-out is the volume of the output. For example: This will make
it sound as if there are twice as many instruments as are
actually playing:
play lead.aiff echo 0.8 0.88 60 0.4
If the delay is very short, then it sound like a (metallic) robot
playing music:
play lead.aiff echo 0.8 0.88 6 0.4
A longer delay will sound like an open air concert in the
mountains:
play lead.aiff echo 0.8 0.9 1000 0.3
One mountain more, and:
play lead.aiff echo 0.8 0.9 1000 0.3 1800 0.25
echos gain-in gain-out <delay decay>
Add a sequence of echoes to the audio. Each delay decay
pair gives the delay in milliseconds and the decay (relative to
gain-in) of that echo. Gain-out is the volume of the output.
Like the echo effect, echos stand for ’ECHO in Sequel’, that is
the first echos takes the input, the second the input and the
first echos, the third the input and the first and the second
echos, ... and so on. Care should be taken using many echos; a
single echos has the same effect as a single echo.
The sample will be bounced twice in symmetric echos:
play lead.aiff echos 0.8 0.7 700 0.25 700 0.3
The sample will be bounced twice in asymmetric echos:
play lead.aiff echos 0.8 0.7 700 0.25 900 0.3
The sample will sound as if played in a garage:
play lead.aiff echos 0.8 0.7 40 0.25 63 0.3
equalizer frequency[k]
width[q|o|h|k] gain
Apply a two-pole peaking equalisation (EQ) filter. With this
filter, the signal-level at and around a selected frequency can
be increased or decreased, whilst (unlike band-pass and
band-reject filters) that at all other frequencies is unchanged.
frequency gives the filter’s central frequency in Hz,
width, the band-width, and gain the required gain
or attenuation in dB. Beware of Clipping when using a
positive gain.
In order to produce complex equalisation curves, this effect can
be given several times, each with a different central frequency.
The filter is described in detail in [1].
This effect supports the --plot global option.
See also bass and treble for shelving equalisation
effects.
fade [type] fade-in-length [stop-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.
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
...
This effect supports the --plot global option.
flanger [delay depth regen width speed shape phase
interp]
Apply a flanging effect to the audio. See [3] for a detailed
description of flanging.
All parameters are optional (right to left).
gain [-e|-B|-b|-r] [-n]
[-l|-h] [gain-dB]
Apply amplification or attenuation to the audio signal, or, in
some cases, to some of its channels. Note that use of any of
-e, -B, -b, -r, or -n requires
temporary file space to store the audio to be processed, so may
be unsuitable for use with ’streamed’ audio.
Without other options, gain-dB is used to adjust the
signal power level by the given number of dB: positive amplifies
(beware of Clipping), negative attenuates. With other options,
the gain-dB amplification or attenuation is (logically)
applied after the processing due to those options.
Given the -e option, the levels of the audio channels of a
multi-channel file are ’equalised’, i.e. gain is applied to all
channels other than that with the highest peak level, such that
all channels attain the same peak level (but, without also giving
-n, the audio is not ’normalised’).
The -B (balance) option is similar to -e, but with
-B, the RMS level is used instead of the peak level.
-B might be used to correct stereo imbalance caused by an
imperfect record turntable cartridge. Note that unlike -e,
-B might cause some clipping.
-b is similar to -B but has clipping protection,
i.e. if necessary to prevent clipping whilst balancing,
attenuation is applied to all channels. Note, however, that in
conjunction with -n, -B and -b are
synonymous.
The -r option is used in conjunction with a prior
invocation of gain with the -h option - see below
for details.
The -n option normalises the audio to 0dB FSD; it is often
used in conjunction with a negative gain-dB to the effect
that the audio is normalised to a given level below 0dB. For
example,
sox infile outfile gain -n
normalises to 0dB, and
sox infile outfile gain -n -3
normalises to -3dB.
The -l option invokes a simple limiter, e.g.
sox infile outfile gain -l 6
will apply 6dB of gain but never clip. Note that limiting more
than a few dBs more than occasionally (in a piece of audio) is
not recommended as it can cause audible distortion. See the
compand effect for a more capable limiter.
The -h option is used to apply gain to provide head-room
for subsequent processing. For example, with
sox infile outfile gain -h bass +6
6dB of attenuation will be applied prior to the bass boosting
effect thus ensuring that it will not clip. Of course, with bass,
it is obvious how much headroom will be needed, but with other
effects (e.g. rate, dither) it is not always as clear. Another
advantage of using gain -h rather than an explicit
attenuation, is that if the headroom is not used by subsequent
effects, it can be reclaimed with gain -r, for example:
sox infile outfile gain -h bass +6 rate 44100 gain -r
The above effects chain guarantees never to clip nor amplify; it
attenuates if necessary to prevent clipping, but by only as much
as is needed to do so.
Output formatting (dithering and bit-depth reduction) also
requires headroom (which cannot be ’reclaimed’), e.g.
sox infile outfile gain -h bass +6 rate 44100 gain -rh dither
Here, the second gain invocation, reclaims as much of the
headroom as it can from the preceding effects, but retains as
much headroom as is needed for subsequent processing. The SoX
global option -G can be given to automatically invoke
gain -h and gain -r.
See also the norm and vol effects.
highpass|lowpass [-1|-2]
frequency[k]
[width[q|o|h|k]]
Apply a high-pass or low-pass filter with 3dB point
frequency. The filter can be either single-pole (with
-1), or double-pole (the default, or with -2).
width applies only to double-pole filters; the default is
Q = 0.707 and gives a Butterworth response. The filters roll off
at 6dB per pole per octave (20dB per pole per decade). The
double-pole filters are described in detail in [1].
These effects support the --plot global option.
See also sinc for filters with a steeper roll-off.
hilbert [-n taps]
Apply an odd-tap Hilbert transform filter, phase-shifting the
signal by 90 degrees.
This is used in many matrix coding schemes and for analytic
signal generation. The process is often written as a
multiplication by i (or j), the imaginary unit.
An odd-tap Hilbert transform filter has a bandpass
characteristic, attenuating the lowest and highest frequencies.
Its bandwidth can be controlled by the number of filter taps,
which can be specified with -n. By default, the number of
taps is chosen for a cutoff frequency of about 75 Hz.
This effect supports the --plot global option.
ladspa module [plugin] [argument...]
Apply a LADSPA [5] (Linux Audio Developer’s Simple Plugin API)
plugin. Despite the name, LADSPA is not Linux-specific, and a
wide range of effects is available as LADSPA plugins, such as cmt
[6] (the Computer Music Toolkit) and Steve Harris’s plugin
collection [7]. The first argument is the plugin module, the
second the name of the plugin (a module can contain more than one
plugin) and any other arguments are for the control ports of the
plugin. Missing arguments are supplied by default values if
possible. 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
provides equalisation for the human auditory system. See
http://en.wikipedia.org/wiki/Loudness for a detailed description
of loudness. The gain is adjusted by the given gain
parameter (usually negative) and the signal equalised according
to ISO 226 w.r.t. a reference level of 65dB, though an
alternative reference level may be given if the original
audio has been equalised for some other optimal level. A default
gain of -10dB is used if a gain value is not given.
See also the gain effect.
lowpass [-1|-2] frequency[k]
[width[q|o|h|k]]
Apply a low-pass filter. See the description of the
highpass effect for details.
mcompand
"attack1,decay1{,attack2,decay2}
[soft-knee-dB:]in-dB1[,out-dB1]{,in-dB2,out-dB2}
[gain [initial-volume-dB [delay]]]"
{crossover-freq[k] "attack1,..."}
The multi-band compander is similar to the single-band compander
but the audio is first divided into bands using Linkwitz-Riley
cross-over filters and a separately specifiable compander run on
each band. See the compand effect for the definition of
its parameters. Compand parameters are specified between double
quotes and the crossover frequency for that band is given by
crossover-freq; these can be repeated to create multiple
bands.
For example, the following (one long) command shows how
multi-band companding is typically used in FM radio:
play track1.wav gain -3 sinc 8000- 29 100 mcompand \
"0.005,0.1 -47,-40,-34,-34,-17,-33" 100 \
"0.003,0.05 -47,-40,-34,-34,-17,-33" 400 \
"0.000625,0.0125 -47,-40,-34,-34,-15,-33" 1600 \
"0.0001,0.025 -47,-40,-34,-34,-31,-31,-0,-30" 6400 \
"0,0.025 -38,-31,-28,-28,-0,-25" \
gain 15 highpass 22 highpass 22 sinc -n 255 -b 16 -17500 \
gain 9 lowpass -1 17801
The audio file is played with a simulated FM radio sound (or
broadcast signal condition if the lowpass filter at the end is
skipped). Note that the pipeline is set up with US-style 75us
pre-emphasis.
See also compand for a single-band companding effect.
noiseprof [profile-file]
Calculate a profile of the audio for use in noise reduction. See
the description of the noisered effect for details.
noisered [profile-file [amount]]
Reduce noise in the audio signal by profiling and filtering. This
effect is moderately effective at removing consistent background
noise such as hiss or hum. To use it, first run SoX with the
noiseprof effect on a section of audio that ideally would
contain silence but in fact contains noise - such sections are
typically found at the beginning or the end of a recording.
noiseprof will write out a noise profile to
profile-file, or to stdout if no profile-file or if
’-’ is given. E.g.
sox speech.wav -n trim 0 1.5 noiseprof speech.noise-profile
To actually remove the noise, run SoX again, this time with the
noisered effect; noisered will reduce noise
according to a noise profile (which was generated by
noiseprof), from profile-file, or from stdin if no
profile-file or if ’-’ is given. E.g.
sox speech.wav cleaned.wav noisered speech.noise-profile 0.3
How much noise should be removed is specified by
amount-a number between 0 and 1 with a default of
0.5. Higher numbers will remove more noise but present a greater
likelihood of removing wanted components of the audio signal.
Before replacing an original recording with a noise-reduced
version, experiment with different amount values to find
the optimal one for your audio; use headphones to check that you
are happy with the results, paying particular attention to
quieter sections of the audio.
On most systems, the two stages - profiling and reduction - can
be combined using a pipe, e.g.
sox noisy.wav -n trim 0 1 noiseprof | play noisy.wav noisered
norm [dB-level]
Normalise the audio. norm is just an alias for gain
-n; see the gain effect for details.
oops
Out Of Phase Stereo effect. Mixes stereo to twin-mono where each
mono channel contains the difference between the left and right
stereo channels. This is sometimes known as the ’karaoke’ effect
as it often has the effect of removing most or all of the vocals
from a recording. It is equivalent to remix 1,2i 1,2i.
overdrive [gain(20) [colour(20)]]
Non linear distortion. The colour parameter controls the
amount of even harmonic content in the over-driven output.
pad { length[@position] }
Pad the audio with silence, at the beginning, the end, or any
specified points through the audio. 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 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 padding at each end of the
audio, whilst pad 4000s@3:00 inserts 4000 samples of
silence 3 minutes into the audio. If silence is wanted only at
the end of the audio, specify either the end position or specify
a zero-length pad at the start.
See also delay for an effect that can add silence at the
beginning of the audio on a channel-by-channel basis.
phaser gain-in gain-out delay decay speed
[-s|-t]
Add a phasing effect to the audio. See [3] for a detailed
description of phasing.
delay/decay/speed gives the delay in milliseconds and the decay
(relative to gain-in) with a modulation speed in Hz. The
modulation is either sinusoidal (-s) - preferable for
multiple instruments, or triangular (-t) - gives single
instruments a sharper phasing effect. The decay should be less
than 0.5 to avoid feedback, and usually no less than 0.1.
Gain-out is the volume of the output.
For example:
play snare.flac phaser 0.8 0.74 3 0.4 0.5 -t
Gentler:
play snare.flac phaser 0.9 0.85 4 0.23 1.3 -s
A popular sound:
play snare.flac phaser 0.89 0.85 1 0.24 2 -t
More severe:
play snare.flac phaser 0.6 0.66 3 0.6 2 -t
pitch [-q] shift [segment
[search [overlap]]]
Change the audio pitch (but not tempo).
shift gives the pitch shift as positive or negative
’cents’ (i.e. 100ths of a semitone). See the tempo effect
for a description of the other parameters.
See also the bend, speed, and tempo effects.
rate [-q|-l|-m|-h|-v]
[override-options] RATE[k]
Change the audio sampling rate (i.e. resample the audio) to any
given RATE (even non-integer if this is supported by the
output file format) using a quality level defined as follows:
where Band-width is the percentage of the audio frequency
band that is preserved and Rej dB is the level of noise
rejection. Increasing levels of resampling quality come at the
expense of increasing amounts of time to process the audio. If no
quality option is given, the quality level used is ’high’ (but
see ’Playing & Recording Audio’ above regarding playback).
The ’quick’ algorithm uses cubic interpolation; all others use
band-limited interpolation. By default, all algorithms have a
’linear’ phase response; for ’medium’, ’high’ and ’very high’,
the phase response is configurable (see below).
The rate effect is invoked automatically if SoX’s
-r option specifies a rate that is different to that of
the input file(s). Alternatively, if this effect is given
explicitly, then SoX’s -r option need not be given. For
example, the following two commands are equivalent:
sox input.wav -r 48k output.wav bass -b 24
sox input.wav output.wav bass -b 24 rate 48k
though the second command is more flexible as it allows
rate options to be given, and allows the effects to be
ordered arbitrarily.
Warning: technically detailed discussion follows.
The simple quality selection described above provides settings
that satisfy the needs of the vast majority of resampling tasks.
Occasionally, however, it may be desirable to fine-tune the
resampler’s filter response; this can be achieved using
override options, as detailed in the following table:
N.B. Override options cannot be used with the ’quick’ or ’low’
quality algorithms.
All resamplers use filters that can sometimes create ’echo’
(a.k.a. ’ringing’) artefacts with transient signals such as those
that occur with ’finger snaps’ or other highly percussive sounds.
Such artefacts are much more noticeable to the human ear if they
occur before the transient (’pre-echo’) than if they occur after
it (’post-echo’). Note that frequency of any such artefacts is
related to the smaller of the original and new sampling rates but
that if this is at least 44.1kHz, then the artefacts will lie
outside the range of human hearing.
A phase response setting may be used to control the distribution
of any transient echo between ’pre’ and ’post’: with minimum
phase, there is no pre-echo but the longest post-echo; with
linear phase, pre and post echo are in equal amounts (in signal
terms, but not audibility terms); the intermediate phase setting
attempts to find the best compromise by selecting a small length
(and level) of pre-echo and a medium lengthed post-echo.
Minimum, intermediate, or linear phase response is selected using
the -M, -I, or -L option; a custom phase
response can be created with the -p option. Note that
phase responses between ’linear’ and ’maximum’ (greater than 50)
are rarely useful.
A resampler’s band-width setting determines how much of the
frequency content of the original signal (w.r.t. the original
sample rate when up-sampling, or the new sample rate when
down-sampling) is preserved during conversion. The term
’pass-band’ is used to refer to all frequencies up to the
band-width point (e.g. for 44.1kHz sampling rate, and a
resampling band-width of 95%, the pass-band represents
frequencies from 0Hz (D.C.) to circa 21kHz). Increasing the
resampler’s band-width results in a slower conversion and can
increase transient echo artefacts (and vice versa).
The -s ’steep filter’ option changes resampling band-width
from the default 95% (based on the 3dB point), to 99%. The
-b option allows the band-width to be set to any value in
the range 74-99.7 %, but note that band-width values greater than
99% are not recommended for normal use as they can cause
excessive transient echo.
If the -a option is given, then aliasing/imaging above the
pass-band is allowed. For example, with 44.1kHz sampling rate,
and a resampling band-width of 95%, this means that frequency
content above 21kHz can be distorted; however, since this is
above the pass-band (i.e. above the highest frequency of
interest/audibility), this may not be a problem. The benefits of
allowing aliasing/imaging are reduced processing time, and
reduced (by almost half) transient echo artefacts. Note that if
this option is given, then the minimum band-width allowable with
-b increases to 85%.
Examples:
sox input.wav -b 16 output.wav rate -s -a 44100 dither -s
default (high) quality resampling; overrides: steep filter, allow
aliasing; to 44.1kHz sample rate; noise-shaped dither to 16-bit
WAV file.
sox input.wav -b 24 output.aiff rate -v -I -b 90 48k
very high quality resampling; overrides: intermediate phase,
band-width 90%; to 48k sample rate; store output to 24-bit AIFF
file.
The pitch and speed effects use the rate
effect at their core.
remix [-a|-m|-p]
<out-spec>
out-spec = in-spec{,in-spec} |
0
in-spec =
[in-chan][-[in-chan2]][vol-spec]
vol-spec = p|i|v[volume]
Select and mix input audio channels into output audio channels.
Each output channel is specified, in turn, by a given
out-spec: a list of contributing input channels and volume
specifications.
Note that this effect operates on the audio channels
within the SoX effects processing chain; it should not be
confused with the -m global option (where multiple
files are mix-combined before entering the effects chain).
An out-spec contains comma-separated input channel-numbers
and hyphen-delimited channel-number ranges; alternatively,
0 may be given to create a silent output channel. For
example,
sox input.wav output.wav remix 6 7 8 0
creates an output file with four channels, where channels 1, 2,
and 3 are copies of channels 6, 7, and 8 in the input file, and
channel 4 is silent. Whereas
sox input.wav output.wav remix 1-3,7 3
creates a (somewhat bizarre) stereo output file where the left
channel is a mix-down of input channels 1, 2, 3, and 7, and the
right channel is a copy of input channel 3.
Where a range of channels is specified, the channel numbers to
the left and right of the hyphen are optional and default to 1
and to the number of input channels respectively. Thus
sox input.wav output.wav remix -
performs a mix-down of all input channels to mono.
By default, where an output channel is mixed from multiple (n)
input channels, each input channel will be scaled by a factor of
¹/ n . Custom mixing volumes can be set by
following a given input channel or range of input channels with a
vol-spec (volume specification). This is one of the
letters p, i, or v, followed by a volume
number, the meaning of which depends on the given letter and is
defined as follows:
If an out-spec includes at least one vol-spec then,
by default, ¹/ n scaling is not applied to any
other channels in the same out-spec (though may be in other
out-specs). The -a (automatic) option however, can be given to
retain the automatic scaling in this case. For example,
sox input.wav output.wav remix 1,2 3,4v0.8
results in channel level multipliers of 0.5,0.5 1,0.8, whereas
sox input.wav output.wav remix -a 1,2 3,4v0.8
results in channel level multipliers of 0.5,0.5 0.5,0.8.
The -m (manual) option disables all automatic volume adjustments,
so
sox input.wav output.wav remix -m 1,2 3,4v0.8
results in channel level multipliers of 1,1 1,0.8.
The volume number is optional and omitting it corresponds to no
volume change; however, the only case in which this is useful is
in conjunction with i. For example, if input.wav is
stereo, then
sox input.wav output.wav remix 1,2i
is a mono equivalent of the oops effect.
If the -p option is given, then any automatic ¹/
n scaling is replaced by ¹/ √n
(’power’) scaling; this gives a louder mix but one that might
occasionally clip.
One use of the remix effect is to split an audio file into
a set of files, each containing one of the constituent channels
(in order to perform subsequent processing on individual audio
channels). Where more than a few channels are involved, a script
such as the following (Bourne shell script) is useful:
#!/bin/sh
chans=`soxi -c "$1"`
while [ $chans -ge 1 ]; do
chans0=`printf %02i $chans` # 2 digits hence up to 99 chans
out=`echo "$1"|sed "s/\(.*\)\.\(.*\)/\1-$chans0.\2/"`
sox "$1" "$out" remix $chans
chans=`expr $chans - 1`
done
If a file input.wav containing six audio channels were
given, the script would produce six output files:
input-01.wav, input-02.wav, ...,
input-06.wav.
See also the swap effect.
repeat [count (1)]
Repeat the entire audio count times, or once if
count is not given. Requires temporary file space to store
the audio to be repeated. Note that repeating once yields two
copies: the original audio and the repeated audio.
reverb [-w|--wet-only] [reverberance
(50%) [HF-damping (50%)
[room-scale (100%) [stereo-depth (100%)
[pre-delay (0ms) [wet-gain (0dB)]]]]]]
Add reverberation to the audio using the ’freeverb’ algorithm. A
reverberation effect is sometimes desirable for concert halls
that are too small or contain so many people that the hall’s
natural reverberance is diminished. Applying a small amount of
stereo reverb to a (dry) mono signal will usually make it sound
more natural. See [3] for a detailed description of
reverberation.
Note that this effect increases both the volume and the length of
the audio, so to prevent clipping in these domains, a typical
invocation might be:
play dry.wav gain -3 pad 0 3 reverb
The -w option can be given to select only the ’wet’
signal, thus allowing it to be processed further, independently
of the ’dry’ signal. E.g.
play -m voice.wav "|sox voice.wav -p reverse reverb -w reverse"
for a reverse reverb effect.
reverse
Reverse the audio completely. Requires temporary file space to
store the audio to be reversed.
riaa
Apply RIAA vinyl playback equalisation. The sampling rate must be
one of: 44.1, 48, 88.2, 96 kHz.
This effect supports the --plot global option.
silence [-l] above-periods [duration
threshold[d|%]
[below-periods duration threshold[d|%]]
Removes silence from the beginning, middle, or end of the audio.
’Silence’ is determined by a specified threshold.
The above-periods value is used to indicate if audio
should be trimmed at the beginning of the audio. A value of zero
indicates no silence should be trimmed from the beginning. When
specifying 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 background noise.
When optionally trimming silence from the end of the audio, you
specify a below-periods count. In this case,
below-period means to remove all audio after silence is
detected. Normally, this will be a value 1 of but it can be
increased to skip over periods of silence that are wanted. For
example, if you have a song with 2 seconds of silence in the
middle and 2 second at the end, you could set below-period to a
value of 2 to skip over the silence in the middle of the audio.
For below-periods, duration specifies a period of
silence that must exist before audio is not copied any more. By
specifying a higher duration, silence that is wanted can be left
in the audio. For example, if you have a song with an expected 1
second of silence in the middle and 2 seconds of silence at the
end, a duration of 2 seconds could be used to skip over the
middle silence.
Unfortunately, you must know the length of the silence at the end
of your audio file to trim off silence reliably. A work around is
to use the silence effect in combination with the
reverse effect. By first reversing the audio, you can use
the above-periods to reliably trim all audio from what
looks like the front of the file. Then reverse the file again to
get back to normal.
To remove silence from the middle of a file, specify a
below-periods that is negative. This value is then treated
as a positive value and is also used to indicate 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 suffixed with
d to indicate the value is in decibels, or % to
indicate a percentage of maximum value of the sample value
(0% specifies pure digital silence).
The following example shows how this effect can be used to start
a recording that does not contain the delay at the start which
usually occurs between ’pressing the record button’ and the start
of the performance:
rec parameters filename other-effects silence 1 5 2%
sinc [-a att|-b beta]
[-p phase|-M|-I|-L] [-t
tbw|-n taps] [freqHP]
[-freqLP [-t tbw|-n taps]]
Apply a sinc kaiser-windowed low-pass, high-pass, band-pass, or
band-reject filter to the signal. The freqHP and
freqLP parameters give the frequencies of the 6dB points
of a high-pass and low-pass filter that may be invoked
individually, or together. If both are given, then freqHP
less than freqLP creates a band-pass filter, freqHP
greater than freqLP creates a band-reject filter. For
example, the invocations
sinc 3k
sinc -4k
sinc 3k-4k
sinc 4k-3k
create a high-pass, low-pass, band-pass, and band-reject filter
respectively.
The default stop-band attenuation of 120dB can be overridden with
-a; alternatively, the kaiser-window ’beta’ parameter can
be given directly with -b.
The default transition band-width of 5% of the total band can be
overridden with -t (and tbw in Hertz);
alternatively, the number of filter taps can be given directly
with -n.
If both freqHP and freqLP are given, then a
-t or -n option given to the left of the
frequencies applies to both frequencies; one of these options
given to the right of the frequencies applies only to
freqLP.
The -p, -M, -I, and -L options
control the filter’s phase response; see the rate effect
for details.
This effect supports the --plot global option.
spectrogram [options]
Create a spectrogram of the audio; the audio is passed unmodified
through the SoX processing chain. This effect is optional - type
sox --help and check the list of supported effects to see
if it has been included.
The spectrogram is rendered in a Portable Network Graphic (PNG)
file, and shows time in the X-axis, frequency in the Y-axis, and
audio signal magnitude in the Z-axis. Z-axis values are
represented by the colour (or optionally the intensity) of the
pixels in the X-Y plane. If the audio signal contains multiple
channels then these are shown from top to bottom starting from
channel 1 (which is the left channel for stereo audio).
For example, if ’my.wav’ is a stereo file, then with
sox my.wav -n spectrogram
a spectrogram of the entire file will be created in the file
’spectrogram.png’. More often though, analysis of a smaller
portion of the audio is required; e.g. with
sox my.wav -n remix 2 trim 20 30 spectrogram
the spectrogram shows information only from the second (right)
channel, and of thirty seconds of audio starting from twenty
seconds in. To analyse a small portion of the frequency domain,
the rate effect may be used, e.g.
sox my.wav -n rate 6k spectrogram
allows detailed analysis of frequencies up to 3kHz (half the
sampling rate) i.e. where the human auditory system is most
sensitive. With
sox my.wav -n trim 0 10 spectrogram -x 600 -y 200 -z 100
the given options control the size of the spectrogram’s X, Y & Z
axes (in this case, the spectrogram area of the produced image
will be 600 by 200 pixels in size and the Z-axis range will be
100 dB). Note that the produced image includes axes legends etc.
and so will be a little larger than the specified spectrogram
size. In this example:
sox -n -n synth 6 tri 10k:14k spectrogram -z 100 -w kaiser
an analysis ’window’ with high dynamic range is selected to best
display the spectrogram of a swept triangular wave. For a smilar
example, append the following to the ’chime’ command in the
description of the delay effect (above):
rate 2k spectrogram -X 200 -Z -10 -w kaiser
Options are also 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 spectrogram from its
default value of 800 pixels to a given number between 100 and
200000. See also -X and -d.
-X num
X-axis pixels/second; the default is auto-calculated to fit the
given or known audio duration to the X-axis size, or 100
otherwise. If given in conjunction with -d, this option
affects the width of the spectrogram; otherwise, it affects the
duration of the spectrogram. num can be from 1 (low time
resolution) to 5000 (high time resolution) and need not be an
integer. SoX may make a slight adjustment to the given number for
processing quantisation reasons; if so, SoX will report the
actual number used (viewable when the SoX global option -V
is in effect). See also -x and -d.
-y num
Sets the Y-axis size in pixels (per channel); this is the number
of frequency ’bins’ used in the Fourier analysis that produces
the spectrogram. N.B. it can be slow to produce the spectrogram
if this number is not one more than a power of two (e.g. 129). By
default the Y-axis size is chosen automatically (depending on the
number of channels). See -Y for alternative way of setting
spectrogram height.
-Y num
Sets the target total height of the spectrogram(s). The default
value is 550 pixels. Using this option (and by default), SoX will
choose a height for individual spectrogram channels that is one
more than a power of two, so the actual total height may fall
short of the given number. However, there is also a minimum
height per channel so if there are many channels, the number may
be exceeded. See -y for alternative way of setting
spectrogram height.
-z num
Z-axis (colour) range in dB, default 120. This sets the
dynamic-range of the spectrogram to be -num dBFS to
0 dBFS. Num may range from 20 to 180. Decreasing
dynamic-range effectively increases the ’contrast’ of the
spectrogram display, and vice versa.
-Z num
Sets the upper limit of the Z-axis in dBFS. A negative num
effectively increases the ’brightness’ of the spectrogram
display, and vice versa.
-q num
Sets the Z-axis quantisation, i.e. the number of different
colours (or intensities) in which to render Z-axis values. A
small number (e.g. 4) will give a ’poster’-like effect making it
easier to discern magnitude bands of similar level. Small numbers
also usually result in small PNG files. The number given
specifies the number of colours to use inside the Z-axis range;
two colours are reserved to represent out-of-range values.
-w name
Window: Hann (default), Hamming, Bartlett, Rectangular or Kaiser.
The spectrogram is produced using the Discrete Fourier Transform
(DFT) algorithm. A significant parameter to this algorithm is the
choice of ’window function’. By default, SoX uses the Hann window
which has good all-round frequency-resolution and dynamic-range
properties. For better frequency resolution (but lower
dynamic-range), select a Hamming window; for higher dynamic-range
(but poorer frequency-resolution), select a Kaiser window.
Bartlett and Rectangular windows are also available.
-W num
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 levels. If this option is used in
conjunction with -m, the result will be a hybrid
monochrome/colour palette.
-p num
Permute the colours in a colour or hybrid palette. The num
parameter, from 1 (the default) to 6, selects the permutation.
-l
Creates a ’printer friendly’ spectrogram with a light background
(the default has a dark background).
-a
Suppress the display of the axis lines. This is sometimes useful
in helping to discern artefacts at the spectrogram edges.
-r
Raw spectrogram: suppress the display of axes and legends.
-A
Selects an alternative, fixed colour-set. This is provided only
for compatibility with spectrograms produced by another package.
It should not normally be used as it has some problems, not
least, a lack of differentiation at the bottom end which results
in masking of low-level artefacts.
-t text
Set the image title - text to display above the spectrogram.
-c text
Set (or clear) the image comment - text to display below and to
the left of the spectrogram.
-o text
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 example,
sox input.mp3 output.wav -n spectrogram -d 1:00 stats
creates a spectrogram showing the first minute of the audio,
whilst
the stats effect is applied to the entire audio signal.
See also -X for an alternative way of setting the X-axis
resolution.
-S 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 spectral data,
see the stat effect.
speed factor[c]
Adjust the audio speed (pitch and tempo together). factor
is either the ratio of the new speed to the old speed: greater
than 1 speeds up, less than 1 slows down, or, if appended with
the letter ’c’, the number of cents (i.e. 100ths of a semitone)
by which the pitch (and tempo) should be adjusted: greater than 0
increases, less than 0 decreases.
Technically, the speed effect only changes the sample rate
information, leaving the samples themselves untouched. The
rate effect is invoked automatically to resample to the
output sample rate, using its default quality/speed. For higher
quality or higher speed resampling, in addition to the
speed effect, specify the rate effect with the
desired quality option.
See also the bend, pitch, and tempo effects.
splice [-h|-t|-q] {
position[,excess[,leeway]] }
Splice together audio sections. This effect provides two things
over simple audio concatenation: a (usually short) cross-fade is
applied at the join, and a wave similarity comparison is made to
help determine the best place at which to make the join.
One of the options -h, -t, or -q may be
given to select the fade envelope as half-cosine wave (the
default), triangular (a.k.a. linear), or quarter-cosine wave
respectively.
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).
The following diagram uses the tape analogy to illustrate the
splice operation. The effect simulates the diagonal cuts and
joins the two pieces:
length1 excess
-----------><--->
_________ : : _________________
\ : : :\ ’
\ : : : \ ’
\: : : \ ’
\ : : :\ ’
\ : : : \ ’
_______________\: : : \_____’____
<---> <----->
excess leeway
where * indicates the joining points.
For example, a long song begins with two verses which start (as
determined e.g. by using the play command with the
trim (start) effect) at times 0:30.125 and
1:03.432. The following commands cut out the first verse:
sox too-long.wav part1.wav trim 0 30.130
(5 ms excess, after the first verse starts)
sox too-long.wav part2.wav trim 1:03.422
(5 ms excess plus 5 ms leeway, before the second verse starts)
sox part1.wav part2.wav just-right.wav splice 30.130
For another example, the SoX command
play "|sox -n -p synth 1 sin %1" "|sox -n -p synth 1 sin %3"
generates and plays two notes, but there is a nasty click at the
transition; the click can be removed by splicing instead of
concatenating the audio, i.e. by appending splice 1 to the
command. (Clicks at the beginning and end of the audio can be
removed by preceding the splice effect with fade q .01
2 .01).
Provided your arithmetic is good enough, multiple splices can be
performed with a single splice invocation. For example:
#!/bin/sh
# Audio Copy and Paste Over
# acpo infile copy-start copy-stop paste-over-start outfile
# 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 general
cross-fades, e.g. to join two songs. In this case, excess
would typically be an number of seconds, the -q option
would typically be given (to select an ’equal power’ cross-fade),
and leeway should be zero (which is the default if
-q is given). For example, if f1.wav and f2.wav are audio
files to be cross-faded, then
sox f1.wav f2.wav out.wav splice -q $(soxi -D f1.wav),3
cross-fades the files where the point of equal loudness is 3
seconds before the end of f1.wav, i.e. the total length of the
cross-fade is 2 × 3 = 6 seconds (Note: the $(...) notation is
POSIX shell).
stat [-s scale] [-rms] [-freq]
[-v] [-d]
Display time and frequency domain statistical information about
the audio. Audio is passed unmodified through the SoX processing
chain.
The information is output to the ’standard error’ (stderr) stream
and is calculated, where n is the duration of the audio in
samples, c is the number of audio channels, r is
the audio sample rate, and x k represents
the PCM value (in the range -1 to +1 by default) of each
successive sample in the audio, as follows:
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:
DC offset, Min level, and
Max level are shown, by default, in the range ±1. If
the -b (bits) options is given, then these three
measurements will be scaled to a signed integer with the given
number of bits; for example, for 16 bits, the scale would be
-32768 to +32767. The -x option behaves the same way as
-b except that the signed integer values are displayed in
hexadecimal. The -s option scales the three measurements
by a given floating-point number.
Pk lev dB and RMS lev dB are
standard peak and RMS level measured in dBFS.
RMS Pk dB and RMS Tr dB are
peak and trough values for RMS level measured over a short window
(default 50ms).
Crest factor is the standard ratio of peak to RMS
level (note: not in dB).
Flat factor is a measure of the flatness (i.e.
consecutive samples with the same value) of the signal at its
peak levels (i.e. either Min level, or
Max level). Pk count is the number of
occasions (not the number of samples) that the signal attained
either Min level, or Max level.
The right-hand Bit-depth figure is the standard definition
of bit-depth i.e. bits less significant than the given number are
fixed at zero. The left-hand figure is the number of most
significant bits that are fixed at zero (or one for negative
numbers) subtracted from the right-hand figure (the number
subtracted is directly related to Pk lev dB).
For multi-channel audio, an overall figure for each of the above
measurements is given and derived from the channel figures as
follows: DC offset: maximum magnitude;
Max level, Pk lev dB,
RMS Pk dB, Bit-depth: maximum;
Min level, RMS Tr dB: minimum;
RMS lev dB, Flat factor,
Pk count: average; Crest factor: not
applicable.
Length s is the duration in seconds of the audio, and
Num samples is equal to the sample-rate multiplied by
Length. Scale Max is the scaling applied to
the first three measurements; specifically, it is the maximum
value that could apply to Max level.
Window s is the length of the window used for the
peak and trough RMS measurements.
See also the stat effect.
swap
Swap stereo channels. See also remix for an effect that
allows arbitrary channel selection and ordering (and mixing).
stretch factor [window fade shift fading]
Change the audio duration (but not its pitch). This effect is
broadly equivalent to the tempo effect with (factor
inverted and) search set to zero, so in general, its
results are comparatively poor; it is retained as it can
sometimes out-perform tempo for small factors.
factor of stretching: >1 lengthen, <1 shorten
duration. window size is in ms. Default is 20ms. The
fade option, can be ’lin’. shift ratio, in [0 1].
Default depends on stretch factor. 1 to shorten, 0.8 to lengthen.
The fading ratio, in [0 0.5]. The amount of a fade’s
default depends on factor and shift.
See also the tempo effect.
synth [-j KEY] [-n] [len
[off [ph [p1 [p2 [p3]]]]]]
{[type] [combine]
[[%]freq[k][:|+|/|-[%]freq2[k]]]
[off [ph [p1 [p2 [p3]]]]]}
This effect can be used to generate fixed or swept frequency
audio tones with various wave shapes, or to generate wide-band
noise of various ’colours’. Multiple synth effects can be
cascaded to produce more complex waveforms; at each stage it is
possible to choose whether the generated waveform will be mixed
with, or modulated onto the output from the previous stage. Audio
for each channel in a multi-channel audio file can be synthesised
independently.
Though this effect is used to generate audio, an input file must
still be given, the characteristics of which will be used to set
the synthesised audio length, the number of channels, and the
sampling rate; however, since the input file’s audio is not
normally needed, a ’null file’ (with the special name -n)
is often given instead (and the length specified as a parameter
to synth or by another given effect that can has an
associated length).
For example, the following produces a 3 second, 48kHz, audio file
containing a sine-wave swept from 300 to 3300 Hz:
sox -n output.wav synth 3 sine 300-3300
and this produces an 8 kHz version:
sox -r 8000 -n output.wav synth 3 sine 300-3300
Multiple channels can be synthesised by specifying the set of
parameters shown between braces multiple times; the following
puts the swept tone in the left channel and adds ’brown’ noise in
the right:
sox -n output.wav synth 3 sine 300-3300 brownnoise
The following example shows how two synth effects can be cascaded
to create a more complex waveform:
play -n synth 0.5 sine 200-500 synth 0.5 sine fmod 700-100
Frequencies can also be given in ’scientific’ note notation, or,
by prefixing a ’%’ character, as a number of semitones relative
to ’middle A’ (440 Hz). For example, the following could be
used to help tune a guitar’s low ’E’ string:
play -n synth 4 pluck %-29
or with a (Bourne shell) loop, the whole guitar:
for n in E2 A2 D3 G3 B3 E4; do
play -n synth 4 pluck $n repeat 2; done
See the delay effect (above) and the reference to ’SoX
scripting examples’ (below) for more synth examples.
N.B. This effect generates audio at maximum volume
(0dBFS), which means that there is a high chance of clipping when
using the audio subsequently, so in many cases, you will want to
follow this effect with the gain effect to prevent this
from happening. (See also Clipping above.) Note that, by
default, the synth effect incorporates the functionality
of gain -h (see the gain effect for details);
synth’s -n option may be given to disable this
behaviour.
A detailed description of each synth parameter follows:
len is the length of audio to synthesise 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
modulation), fmod (frequency modulation); default=create.
freq/freq2 are the frequencies at the beginning/end
of synthesis in Hz or, if preceded with ’%’, semitones relative
to A (440 Hz); alternatively, ’scientific’ note notation
(e.g. E2) may be used. The default frequency is 440Hz. By
default, the tuning used with the note notations is ’equal
temperament’; the -j KEY option selects ’just
intonation’, where KEY is an integer number of semitones
relative to A (so for example, -9 or 3 selects the key of C), or
a note in scientific notation.
If freq2 is given, then len must also have been
given and the generated tone will be swept between the given
frequencies. The two given frequencies must be separated by one
of the characters ’:’, ’+’, ’/’, or ’-’. This character is used
to specify the sweep function as follows:
:
Linear: the tone will change by a fixed number of hertz per
second.
+
Square: a second-order function is used to change the tone.
/
Exponential: the tone will change by a fixed number of semitones
per second.
-
Exponential: as ’/’, but initial phase always zero, and stepped
(less smooth) frequency changes.
Not used for noise.
off is the bias (DC-offset) of the signal in percent;
default=0.
ph is the phase shift in percentage of 1 cycle; default=0.
Not used for noise.
p1 is the percentage of each cycle that is ’on’ (square),
or ’rising’ (triangle, exp, trapezium); default=50 (square,
triangle, exp), default=10 (trapezium), or sustain (pluck);
default=40.
p2 (trapezium): the percentage through each cycle at which
’falling’ begins; default=50. exp: the amplitude in multiples of
2dB; default=50, or tone-1 (pluck); default=20.
p3 (trapezium): the percentage through each cycle at which
’falling’ ends; default=60, or tone-2 (pluck); default=90.
tempo [-q] [-m|-s|-l]
factor [segment [search [overlap]]]
Change the audio playback speed but not its pitch. This effect
uses the WSOLA algorithm. The audio is chopped up into segments
which are then shifted in the time domain and overlapped
(cross-faded) at points where their waveforms are most similar as
determined by measurement of ’least squares’.
By default, linear searches are used to find the best overlapping
points. If the optional -q parameter is given, tree
searches are used instead. This makes the effect work more
quickly, but the result may not sound as good. However, if you
must improve the processing speed, this generally reduces the
sound quality less than reducing the search or overlap values.
The -m option is used to optimize default values of
segment, search and overlap for music processing.
The -s option is used to optimize default values of
segment, search and overlap for speech processing.
The -l option is used to optimize default values of
segment, search and overlap for ’linear’ processing that tends to
cause more noticeable distortion but may be useful when factor is
close to 1.
If -m, -s, or -l is specified, the default value of segment will
be calculated based on factor, while default search and overlap
values are based on segment. Any values you provide still
override these default values.
factor gives the ratio of new tempo to the old tempo, so
e.g. 1.1 speeds up the tempo by 10%, and 0.9 slows it down by
10%.
The optional segment parameter selects the algorithm’s
segment size in milliseconds. If no other flags are specified,
the default value is 82 and is typically suited to making small
changes to the tempo of music. For larger changes (e.g. a factor
of 2), 41 ms may give a better result. The -m, -s, and -l
flags will cause the segment default to be automatically adjusted
based on factor. For example using -s (for speech) with a tempo
of 1.25 will calculate a default segment value of 32.
The optional search parameter gives the audio length in
milliseconds over which the algorithm will search for overlapping
points. If no other flags are specified, the default value is
14.68. Larger values use more processing time and may or may not
produce better results. A practical maximum is half the value of
segment. Search can be reduced to cut processing time at the risk
of degrading output quality. The -m, -s, and -l flags will cause
the search default to be automatically adjusted based on segment.
The optional overlap parameter gives the segment overlap
length in milliseconds. Default value is 12, but -m, -s, or -l
flags automatically adjust overlap based on segment size.
Increasing overlap increases processing time and may increase
quality. A practical maximum for overlap is the value of search,
with overlap typically being (at least) a little smaller then
search.
See also speed for an effect that changes tempo and pitch
together, pitch and bend for effects that change
pitch only, and stretch for an effect that changes tempo
using a different algorithm.
treble gain [frequency[k]
[width[s|h|k|o|q]]]
Apply a treble tone-control effect. See the description of the
bass effect for details.
tremolo speed [depth]
Apply a tremolo (low frequency amplitude modulation) effect to
the audio. The tremolo frequency in Hz is given by speed,
and the depth as a percentage by depth (default 40).
trim {[=|-]position}
Cuts portions out of the audio. Any number of positions
may be given; audio is not sent to the output until the first
position is reached. The effect then alternates between
copying and discarding audio at each position.
If a position is preceded by an equals or minus sign, it
is interpreted relative to the beginning or the end of the audio,
respectively. (The audio length must be known for end-relative
locations to work.) Otherwise, it is considered an offset from
the last position, or from the start of audio for the
first parameter. Using a value of 0 for the first position
parameter allows copying from the beginning of the audio.
All 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 value of 1:30.5 for the first parameter
will not start until 1 minute, thirty and ½ seconds into the
audio. The format for specifying sample counts is the number of
samples with the letter ’s’ appended to it. A value of 8000s for
the first parameter will wait until 8000 samples are read before
starting to process audio.
For example,
sox infile outfile trim 0 10
will copy the first ten seconds, while
play infile trim 12:34 =15:00 -2:00
will play from 12 minutes 34 seconds into the audio up to 15
minutes into the audio (i.e. 2 minutes and 26 seconds long), then
resume playing two minutes before the end of audio.
upsample [factor]
Upsample the signal by an integer factor: factor-1
zero-value samples are inserted between each pair of input
samples. As a result, the original spectrum is replicated into
the new frequency space (aliasing) and attenuated. This
attenuation can be compensated for by adding vol
factor after any further processing. The upsample effect
is typically used in combination with filtering effects.
For a general resampling effect with anti-aliasing, see
rate. See also downsample.
vad [options]
Voice Activity Detector. Attempts to trim silence and quiet
background sounds from the ends of (fairly high resolution i.e.
16-bit, 44-48kHz) recordings of speech. The algorithm currently
uses a simple cepstral power measurement to detect voice, so may
be fooled by other things, especially music. The effect can trim
only from the front of the audio, so in order to trim from the
back, the reverse effect must also be used. E.g.
play speech.wav norm vad
to trim from the front,
play speech.wav norm reverse vad reverse
to trim from the back, and
play speech.wav norm vad reverse vad reverse
to trim from both ends. The use of the norm effect is
recommended, but remember that neither reverse nor
norm is suitable for use with streamed audio.
Options:
Default values are shown in parenthesis.
-t num (7)
The measurement level used to trigger activity detection. This
might need to be changed depending on the noise level, signal
level and other charactistics of the input audio.
-T num (0.25)
The time constant (in seconds) used to help ignore short bursts
of sound.
-s num (1)
The amount of audio (in seconds) to search for quieter/shorter
bursts of audio to include prior to the detected trigger point.
-g num (0.25)
Allowed gap (in seconds) between quieter/shorter bursts of audio
to include prior to the detected trigger point.
-p num (0)
The amount of audio (in seconds) to preserve before the trigger
point and any found quieter/shorter bursts.
Advanced Options:
These allow fine tuning of the algorithm’s internal parameters.
-b num
The algorithm (internally) uses adaptive noise
estimation/reduction in order to detect the start of the wanted
audio. This option sets the time for the initial noise estimate.
-N num
Time constant used by the adaptive noise estimator for when the
noise level is increasing.
-n num
Time constant used by the adaptive noise estimator for when the
noise level is decreasing.
-r num
Amount of noise reduction to use in the detection algorithm (e.g.
0, 0.5, ...).
-f num
Frequency of the algorithm’s processing/measurements.
-m num
Measurement duration; by default, twice the measurement period;
i.e. with overlap.
-M num
Time constant used to smooth spectral measurements.
-h num
’Brick-wall’ frequency of high-pass filter applied at the input
to the detector algorithm.
-l num
’Brick-wall’ frequency of low-pass filter applied at the input to
the detector algorithm.
-H num
’Brick-wall’ frequency of high-pass lifter used in the detector
algorithm.
-L num
’Brick-wall’ frequency of low-pass lifter used in the detector
algorithm.
See also the silence effect.
vol gain [type [limitergain]]
Apply an amplification or an attenuation to the audio signal.
Unlike the -v option (which is used for balancing multiple
input files as they enter the SoX effects processing chain),
vol is an effect like any other so can be applied
anywhere, and several times if necessary, during the processing
chain.
The amount to change the volume is given by gain which is
interpreted, according to the given type, as follows: if
type is amplitude (or is omitted), then gain
is an amplitude (i.e. voltage or linear) ratio, if power,
then a power (i.e. wattage or voltage-squared) ratio, and if
dB, then a power change in dB.
When type is amplitude or power, a
gain of 1 leaves the volume unchanged, less than 1
decreases it, and greater than 1 increases it; a negative
gain inverts the audio signal in addition to adjusting its
volume.
When type is dB, a gain of 0 leaves the
volume unchanged, less than 0 decreases it, and greater than 0
increases it.
See [4] for a detailed discussion on electrical (and hence audio
signal) voltage and power ratios.
Beware of Clipping when the increasing the volume.
The gain and the type parameters can be
concatenated if desired, e.g. vol 10dB.
An optional limitergain value can be specified and should
be a value much less than 1 (e.g. 0.05 or 0.02) and is used only
on peaks to prevent clipping. Not specifying this parameter will
cause no limiter to be used. In verbose mode, this effect will
display the percentage of the audio that needed to be limited.
See also gain for a volume-changing effect with different
capabilities, and compand for a dynamic-range
compression/expansion/limiting effect.
Deprecated Effects
The following effects have been renamed or have their
functionality included in another effect; they continue to work
in this version of SoX but may be removed in future.
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 prevent this.
The mixer effect can also be invoked with up to 16
numbers, 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
output 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 combinations.
This effect has been superseded by the remix effect that
handles any number of channels.
filenames
Filenames can be simple file names, absolute or relative path
names, or URLs (input files only). Note that URL support requires
that wget(1) is available.
Note: Giving SoX an input or output filename that is the same as
a SoX effect-name will not work since SoX will treat it as an
effect specification. The only work-around to this is to avoid
such filenames. This is generally not difficult since most audio
filenames have a filename ’extension’, whilst effect-names do
not.
Special Filenames
The following special filenames may be used in certain
circumstances in place of a normal filename on the command line:
-
SoX can be used in simple pipeline operations by using the
special filename ’-’ which, if used as an input filename, will
cause SoX will read audio data from ’standard input’ (stdin), and
which, if used as the output filename, will cause SoX will send
audio data to ’standard output’ (stdout). Note that when using
this option for the output file, and sometimes when using it for
an input file, the file-type (see -t below) must also be
given.
"|program [options] ..."
This can be used in place of an input filename to specify the the
given program’s standard output (stdout) be used as an input
file. Unlike - (above), this can be used for several
inputs to one SoX command. For example, if ’genw’ generates mono
WAV formatted signals to its standard output, then the following
command makes a stereo file from two generated signals:
sox -M "|genw --imd -" "|genw --thd -" out.wav
For headerless (raw) audio, -t (and perhaps other format
options) will need to be given, preceding the input command.
"wildcard-filename"
Specifies that filename ’globbing’ (wild-card matching) should be
performed by SoX instead of by the shell. This allows a single
set of file options to be applied to a group of files. For
example, if the current directory contains three ’vox’ files,
file1.vox, file2.vox, and file3.vox, then
play --rate 6k *.vox
will be expanded by the ’shell’ (in most environments) to
play --rate 6k file1.vox file2.vox file3.vox
which will treat only the first vox file as having a sample rate
of 6k. With
play --rate 6k "*.vox"
the given sample rate option will be applied to all three vox
files.
-p, --sox-pipe
This can be used in place of an output filename to specify that
the SoX command should be used as in input pipe to another SoX
command. For example, the command:
play "|sox -n -p synth 2" "|sox -n -p synth 2 tremolo 10" stat
plays two ’files’ in succession, each with different effects.
-p is in fact an alias for ’-t sox -’.
-d, --default-device
This can be used in place of an input or output filename to
specify that the default audio device (if one has been built into
SoX) is to be used. This is akin to invoking rec or
play (as described above).
-n, --null
This can be used in place of an input or output filename to
specify that a ’null file’ is to be used. Note that here, ’null
file’ refers to a SoX-specific mechanism and is not related to
any operating-system mechanism with a similar name.
Using a null file to input audio is equivalent to using a normal
audio file that contains an infinite amount of silence, and as
such is not generally useful unless used with an effect that
specifies a finite time length (such as trim or
synth).
Using a null file to output audio amounts to discarding the audio
and is useful mainly with effects that produce information about
the audio instead of affecting it (such as noiseprof or
stat).
The sampling rate associated with a null file is by default
48 kHz, but, as with a normal file, this can be overridden
if desired using command-line format options (see below).
Supported File & Audio Device Types
See soxformat(7) for a list and description of the
supported file formats and audio device drivers.
license
Copyright 1998-2013 Chris Bagwell and SoX Contributors.
Copyright 1991 Lance Norskog and Sundry Contributors.
This program is free software; you can redistribute it and/or
modify it under the terms of the GNU General Public License as
published by the Free Software Foundation; either version 2, or
(at your option) any later version.
This program is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
General Public License for more details.
bugs
Please report
any bugs found in this version of SoX to the mailing list
(sox-users[:at:]lists.sourceforge[:dot:]net).
see also
soxi ,
soxformat, libsox
audacity , gnuplot, octave,
wget
The SoX web site at http://sox.sourceforge.net
SoX scripting examples at
http://sox.sourceforge.net/Docs/Scripts
References
[1]
R. Bristow-Johnson, Cookbook formulae for audio EQ
biquad filter coefficients,
http://musicdsp.org/files/Audio-EQ-Cookbook.txt
[2]
Wikipedia, Q-factor,
http://en.wikipedia.org/wiki/Q_factor
[3]
Scott Lehman, Effects Explained,
http://harmony-central.com/Effects/effects-explained.html
[4]
Wikipedia, Decibel,
http://en.wikipedia.org/wiki/Decibel
[5]
Richard Furse, Linux Audio Developer’s Simple
Plugin API, http://www.ladspa.org
[6]
Richard Furse, Computer Music Toolkit,
http://www.ladspa.org/cmt
[7]
Steve Harris, LADSPA plugins,
http://plugin.org.uk
authors
Chris Bagwell
(cbagwell[:at:]users.sourceforge[:dot:]net). Other authors and
contributors are listed in the ChangeLog file that is
distributed with the source code.