sox(1) Sound eXchange sox(1)
NAME
SoX - Sound eXchange, the Swiss Army knife of audio manipulation
SYNOPSIS
sox [global-options] [format-options] infile1
[[format-options] infile2] ... [format-options] outfile
[effect [effect-options]] ...
play [global-options] [format-options] infile1
[[format-options] infile2] ... [format-options]
[effect [effect-options]] ...
rec [global-options] [format-options] outfile
[effect [effect-options]] ...
DESCRIPTION
Introduction
SoX reads and writes audio files in most popular formats and can
optionally apply effects to them; it can combine multiple input
sources, synthesise audio, and, on many systems, act as a general pur-
pose audio player or a multi-track audio recorder. It also has limited
ability to split the input in to multiple output files.
Almost all SoX functionality is available using just the sox command,
however, to simplify playing and recording audio, if SoX is invoked as
play the output file is automatically set to be the default sound
device and if invoked as rec the default sound device is used as an
input source. Additionally, the soxi(1) command provides a convenient
way to just query audio file header information.
The heart of SoX is a library called libSoX. Those interested in
extending SoX or using it in other programs should refer to the libSoX
manual page: libsox(3).
SoX is a command-line audio processing tool, particularly suited to
making quick, simple edits and to batch processing. If you need an
interactive, graphical audio editor, use audacity(1).
* * *
The overall SoX processing chain can be summarised as follows:
Input(s) -> Combiner -> Effects -> Output(s)
To show how this works in practise, 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 -r 12k -b 8 -c 1 recital.wav vol 0.7 dither
performs the same format translation, but also changes the audio sam-
pling rate & sample size, down-mixes to mono, and applies the vol and
dither effects.
sox -r 8k -u -b 8 -c 1 voice-memo.raw voice-memo.wav
converts `raw' (a.k.a. `headerless') audio to a self-descibing file
format,
sox slow.aiff fixed.aiff speed 1.027
adjusts audio speed,
sox short.au long.au longer.au
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 q 0.1 1 0.1
plays a synthesised `A minor seventh' chord with a pipe-organ sound,
rec -c 2 test.aiff trim 0 10
records 10 seconds of stereo audio, and
rec -M take1.aiff take1-dub.aiff
records a new track in a multi-track recording.
rec -r 44100 -2 -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 does not start
recording until it detects audio is playing and stops after it sees 10
minutes of silence.
N.B. Detailed explanations of how to use all SoX parameters, file for-
mats, and effects can be found below in this manual, and in soxfor-
mat(7).
File Format Types
There are two types of audio file format that SoX can work with. The
first is `self-describing'; these formats include a header that com-
pletely describes the characteristics of the audio data that follows.
The second type is `headerless' (or `raw data'); here, the audio data
characteristics must be described using the SoX command line.
The following four characteristics are sufficient to describe the for-
mat of audio data such that it can be processed with SoX:
sample rate
The sample rate in samples per second (`Hertz' or `Hz'). For
example, digital telephony traditionally uses a sample rate of
8000 Hz (8 kHz); audio Compact Discs use 44100 Hz (44.1 kHz);
Digital Audio Tape and many computer systems use 48 kHz; profes-
sional audio systems typically use 96 or 192 kHz.
sample size
The number of bits used to store each sample. The most popular
is 16-bit (two bytes); 8-bit (one byte) was popular in the early
days of computer audio, and is still used in telephony; 24-bit
(three bytes) is used, primarily as an intermediate format, 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, i.e.
the stored audio data takes up less space (i.e. disk-space or
transmission band-width) than the other format parameters and
the number of samples would imply. Commonly-used encoding types
include floating-point, u-law, ADPCM, signed-integer PCM, 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 sometimes used as an overall measure of an audio
format and may incorporate elements of all of the above.
Most self-describing formats also allow textual `comments' to be embed-
ded in the file that can be used to describe the audio in some way,
e.g. for music, the title, the author, etc.
One important use of audio file comments is to convey `Replay Gain'
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 cir-
cumstances, individual characteristics may be determined or set using
different mechanisms.
To determine the format of an input file, SoX will use, in order of
precedence and as given or available:
1. Command-line format options.
2. The contents of the file header.
3. The filename extension.
To set the output file format, SoX will use, in order of precedence and
as given or available:
1. Command-line format options.
2. The filename extension.
3. The input file format characteristics, or the closest to
them 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.
Play, Rec, & Default Audio Devices
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
using rec or play 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 ...
For rec, play, and sox, 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:0
play ...
sox ... -t alsa
(Note that the syntax of the set command may vary from system to sys-
tem.)
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, here, the default rate quality level is set to `low';
however, this can be changed if desired, by explicitly specifing the
rate effect with a different quality level, e.g.
play ... rate -m
or by setting the environment varible PLAY_RATE_ARG to the desired
quality option, e.g.
set PLAY_RATE_ARG=-m
play ...
(Note that the syntax of the set command may vary from system to sys-
tem.)
To help with setting a suitable recording level, SoX includes a simple
VU 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).
Accuracy
Many file formats that compress audio discard some of the audio signal
information whilst doing so; converting to such a format then convert-
ing 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', and
formats that do not, `lossless'. The term `quality' is used as a mea-
sure 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 recompressing 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 multi-
ple SoX invocations; hence this is also recommended.
Clipping
Clipping is distortion that occurs when an audio signal level (or `vol-
ume') exceeds the range of the chosen representation. It is nearly
always undesirable and so should usually be corrected by adjusting the
level prior to the point at which clipping occurs.
In SoX, clipping could occur, as you might expect, when using the vol
effect to increase the audio volume, but 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 re-sampling, and processing by
analogue components that can introduce a small DC offset and/or ampli-
fication, 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 does not exceed around 70% of the maximum (linear) range avail-
able, as this will avoid the majority of clipping problems. SoX's stat
effect 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.au bright.au gain -6 treble +6
guarantees that the treble boost will not clip.
If clipping occurs at any point during processing, then SoX will dis-
play a warning message to that effect.
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', or `merge'. 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, however, 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 orignal 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.
When combining input files, SoX applies any specified effects (includ-
ing, for example, the vol volume adjustment effect) after the audio has
been combined; however, it is often useful to be able to set the volume
of (i.e. `balance') the inputs individually, before combining takes
place.
For all combining methods, input file volume adjustments can be made
manually using the -v option (below) which can be given for one or more
input files; if it is given for only some of the input files then the
others receive no volume adjustment. In some circumstances, automatic
volume adjustments may be applied (see below).
The -V option (below) can be used to show the input file volume adjust-
ments that have been selected (either manually or automatically).
There are some special considerations that need to made when mixing
input files:
Unlike the other methods, `mix' combining has the potential to cause
clipping in the combiner if no balancing is performed. So here, if
manual volume adjustments are not given, to ensure that clipping does
not occur, SoX will automatically adjust 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 through the mixed audio
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 appropriately
equal to that of one of the input signals. This is achieved by balanc-
ing using a factor of 1/\/n instead of 1/n. Note that this balancing
factor does not guarantee that no clipping will occur, however, in many
cases, the number of clips will be low and the resultant distortion
imperceptable.
Output Files
SoX's default behavior is to take one or more input files and write
them to a single output file.
This behavior 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 behavior can be cus-
tomized 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 in to 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
usually Ctrl-C). This is a natural requirement in some circumstances,
e.g. when using SoX to make a recording. Note that when using SoX to
play multiple files, Ctrl-C behaves slightly differently: pressing it
once causes SoX to skip to the next file; pressing it twice in quick
succession causes SoX to exit.
Another option to stop processing early is to use an effect that has a
time period or sample count to determine the stopping point. The trim
effect is an example of this. Once all effects chains have stopped
then SoX will also stop.
FILENAMES
Filenames can be simple file names, absolute or relative path names, or
URLs (input files only). Note that URL support requires that wget(1)
is available.
Note: Giving SoX an input or output filename that is the same as a SoX
effect-name will not work since SoX will treat it as an effect specifi-
cation. The only work-around to this is to avoid such filenames; how-
ever, this is generally not difficult since most audio filenames have a
filename `extension', whilst effect-names do not.
Special Filenames
The following special filenames may be used in certain circumstances in
place of a normal filename on the command line:
- SoX can be used in simple pipeline operations by using the spe-
cial filename `-' which, if used in place of an input filename,
will cause SoX will read audio data from `standard input'
(stdin), and which, if used in place of the output filename,
will cause SoX will send audio data to `standard output' (std-
out). Note that when using this option, the file-type (see -t
below) must also be given.
"|program [options] ..."
This can be used in place of an input filename to specify the
the given program's standard output (stdout) be used as an input
file. Unlike - (above), this can be used for several inputs to
one SoX command. For example, if `genw' generates mono WAV for-
matted signals to its standard output, then the following com-
mand makes a stereo file from two generated signals:
sox -M -t wav "|genw --imd -" -t wav "|genw --thd -" out.wav
If -t is not given then the signal is assumed (and checked) to
be in SoX's native .sox format (see -p below and soxformat(7)).
-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 for-
mats and audio device drivers.
OPTIONS
Global Options
These options can be specified on the command line at any point before
the first effect name.
-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.
--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.
---effects-file=FILENAME
Use FILENAME to obtain all effects and their arguments. The
file is parsed as if the values were specified on the command
line. A new line can be used in place of the special ":" marker
to separate effect chains. This option causes any effects spec-
ified on the command line to be discarded.
--interactive
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 strongly recommended; a `shell' alias,
script, or batch file may be an appropriate way of permanently
enabling it.
-m|-M|--combine concatenate|merge|mix|mix-power|sequence
Select the input file combining method; -m selects `mix', -M
selects `merge'.
See Input File Combining above for a description of the differ-
ent combining methods.
--plot gnuplot|octave|off
If not set to off (the default if --plot is not given), run in a
mode that can be used, in conjunction with the gnuplot program
or the GNU Octave program, to assist with the selection and con-
figuration of many of the transfer-function based effects. For
the first given effect that supports the selected plotting pro-
gram, SoX will output commands to plot the effect's transfer
function, and then exit without actually processing any audio.
E.g.
sox --plot octave input-file -n highpass 1320 > plot.m
octave plot.m
-q, --no-show-progress
Run in quiet mode when SoX wouldn't otherwise do so; this is the
opposite of the -S option.
--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 VU meter,
and an indication if clipping has occurred. The VU meter shows
up to two channels and is calibrated for digital audio as fol-
lows:
+----------------------------------------+
|dB FSD Display |
| >= (right channel) |
| -25 - |
| -23 = |
| -21 =- |
| -19 == |
| -17 ==- |
| -15 === |
| -13 ===- |
| -11 ==== |
| -9 ====- |
| -7 ===== |
| -5 =====- |
| -3 ====== |
| -1 =====! `In the red' |
+----------------------------------------+
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.
--version
Show SoX's version number and exit.
-V[level]
Set verbosity. 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; each occurrence of
the -V option increases the verbosity level by 1. Alterna-
tively, 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.
-v, --volume FACTOR
Adjust volume by a factor of FACTOR. This is a linear (ampli-
tude) adjustment, so a number less than 1 decreases the volume;
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 stat effect for information on how to find the max-
imum volume of an audio file; this can be used to help select
suitable values for this option.
See also Input File Balancing above.
Input & Output File Format Options
These options apply to the input or output file whose name they immedi-
ately precede on the command line and are used mainly when working with
headerless file formats or when specifying a format for the output file
that is different to that of the input file.
-b BITS, --bits BITS
The number of bits in each encoded sample. Not applicable to
complex encodings, e.g. MP3, GSM. Not necessary with encodings
that have a fixed number of bits, e.g. A/u-law, ADPCM.
-1/-2/-3/-4/-8
The number of bytes in each encoded sample. 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. To cause the output file to have a
different number of channels than the input file, include this
option with the output file options. If the input and output
file have a different number of channels then the mixer effect
must be used. If the mixer effect is not specified on the com-
mand line it will be invoked internally with default parameters.
Alternatively, some effects (e.g. synth, remix) determine what
will be the number of output channels; in this case, neither
this option nor the mixer effect is necessary.
-e ENCODING, --encoding ENCODING
The audio encoding type.
signed-integer
PCM data stored as signed (`two's complement') integers.
Commonly used with a 16 or 24 -bit encoding size. A
value of 0 represents minimum signal power.
unsigned-integer
PCM data stored as signed (`two's complement') integers.
Commonly used with an 8-bit encoding size. A value of 0
represents maximum signal power.
floating-point
PCM data stored as IEEE 753 single precision (32-bit) or
double precision (64-bit) floating-point ('real') num-
bers. A value of 0 represents minimum signal power.
a-law International telephony standard for logarithmic encoding
to 8 bits per sample. It has a precision equivalent to
roughly 13-bit PCM and is sometimes encoded with reversed
bit-ordering (see the -X option).
u-law, mu-law
North American telephony standard for logarithmic encod-
ing to 8 bits per sample. A.k.a u-law. It has a preci-
sion 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 equiva-
lent to roughly 13-bit PCM.
ms-adpcm
Microsoft 4-bit ADPCM; it has a precision equivalent to
roughly 14-bit PCM.
gsm-full-rate
GSM is currently used for the vast majority of the
world's digital wireless telephone calls. It utilises
several audio formats with different bit-rates and asso-
ciated speech quality. SoX has support for GSM's origi-
nal 13kbps `Full Rate' audio format. It is usually CPU
intensive to work with GSM audio.
Encoding names can be abbreviated where this would not be
ambiguous; e.g. 'unsigned-integer' can be given as 'un', but not
'u' (ambiguous with 'u-law'). For reasons of forward compati-
bility, using abbreviations in scripts is not recommended.
Note that explicitly specifying other encoding types (e.g. MP3,
FLAC) is not necessary since they can be inferred from the file
type or header.
-s/-u/-f/-A/-U/-o/-i/-a/-g
Aliases for specifying the encoding types signed-inte-
ger/unsigned-integer/floating-point/mu-law/a-law/oki-adpcm/ima-
adpcm/ms-adpcm/gsm-full-rate respectively.
-r, --rate RATE[k]
Gives the sample rate in Hz (or kHz if appended with `k') of the
file. To cause the output file to have a different sample rate
than the input file, include this option with the output file
format options.
If the input and output files have different rates then a sample
rate change effect must be run. Since SoX has multiple rate
changing effects, the user can specify which to use as an
effect. If no rate change effect is specified then the rate
effect will be chosen by default.
-t, --type file-type
Gives the type of the audio file. This is useful when the file
extension is non-standard or when the type can not be determined
by looking at the header of the file.
The -t option 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 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 (oth-
erwise) self-describing files. A given endian-setting option
may be ignored for an input file whose header contains a spe-
cific endianness identifier, or for an output file that is actu-
ally an audio device.
N.B. Unlike normal format characteristics, the endianness
(byte, nibble, & bit ordering) of the input file is not automat-
ically used for the output file; so, for example, when the fol-
lowing is run on a little-endian system:
sox -B audio.s2 trimmed.s2 trim 2
trimmed.s2 will be created as little-endian;
sox -B audio.s2 -B trimmed.s2 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) for-
mats.
N.B. See also N.B. in section on -x above.
Output File Format Options
These options apply only to the output file and may precede only the
output filename on the command line.
--add-comment TEXT
Append a comment in the output file header (where applicable).
--comment TEXT
Specify the comment text to store in the output file header
(where applicable).
SoX will provide a default comment if this option (or --com-
ment-file) is not given; to specify that no comment should be
stored in the output file, use --comment "" .
--comment-file FILENAME
Specify a file containing the comment text to store in the out-
put file header (where applicable).
-C, --compression FACTOR
The compression factor for variably compressing output file for-
mats. If this option is not given, then a default compression
factor will apply. The compression factor is interpreted dif-
ferently for different compressing file formats. See the
description of the file formats that use this option in soxfor-
mat(7) for more information.
EFFECTS
In addition to converting and playing 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 need a high performance
computer; stopping other applications may alleviate performance issues
should they occur.
Some of the SoX effects are primarily intended to be applied to a sin-
gle instrument or `voice'. To facilitate this, the remix effect and
the global SoX option -M can be used to isolate then recombine tracks
from a multi-track recording.
Multiple Effect Chains
A single effects chain is made up of one or more effects. Audio from
the input in ran through the chain until either the input file reaches
end of file or an effects in the chain requests to terminate the chain.
SoX supports running multiple effects chain 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 reach end of
file.
A effects chain is terminated by placing a : (colon) after an effect.
Any following effects are apart of a new effects chain.
It is important to place the effect that will stop the chain as the
first effect in the chain. This is because any samples that are
buffered by effects to the left of the terminating effect will be dis-
carded. The amount of samples discarded is related to the --buffer
option and it should be keep small, relative to the sample rate, if the
terminating effect can not 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 out-
put filename will have unique number in its name as documented in Out-
put 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 param-
eters that are optional, braces { } to denote those that are both
optional and repeatable, and angle brackets < > to denote those that
are repeatable but not optional. Where applicable, default values for
optional parameters are shown in parenthesis ( ).
The following parameters are used with, and have the same meaning for,
several effects:
centre[k]
See frequency.
frequency[k]
A frequency in Hz, or, if appended with `k', kHz.
gain A power gain in dB. Zero gives no gain; less than zero gives an
attenuation.
width[h|k|o|q]
Used to specify the band-width of a filter. A number of differ-
ent methods to specify the width are available (though not all
for every effect); one of the characters shown may be appended
to select the desired method as follows:
+-----------------------+
| Method Notes |
|h Hz |
|k kHz |
|o Octaves |
|q Q-factor See [2] |
+-----------------------+
For each effect that uses this parameter, the default method
(i.e. if no character is appended) is the one that it listed
first in the effect's first line of 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
allpass frequency[k] width[h|k|o|q]
Apply a two-pole all-pass filter with central frequency (in Hz)
frequency, and filter-width width. An all-pass filter changes
the audio's frequency to phase relationship without changing its
frequency to amplitude relationship. The filter is described in
detail in [1].
This effect supports the --plot global option.
band [-n] center[k] [width[h|k|o|q]]
Apply a band-pass filter. The frequency response drops loga-
rithmically around the center frequency. The width parameter
gives the slope of the drop. The frequencies at center + width
and center - width will be half of their original amplitudes.
band defaults to a mode oriented to pitched audio, i.e. voice,
singing, or instrumental music. The -n (for noise) option uses
the alternate mode for un-pitched audio (e.g. percussion).
Warning: -n introduces a power-gain of about 11dB in the filter,
so beware of output clipping. band introduces noise in the
shape of the filter, i.e. peaking at the center frequency and
settling around it.
This effect supports the --plot global option.
See also filter for a bandpass filter with steeper shoulders.
bandpass|bandreject [-c] frequency[k] width[h|k|o|q]
Apply a two-pole Butterworth band-pass or band-reject filter
with central frequency frequency, and (3dB-point) band-width
width. The -c option applies only to bandpass and selects a
constant skirt gain (peak gain = Q) instead of the default: con-
stant 0dB peak gain. The filters roll off at 6dB per octave
(20dB per decade) and are described in detail in [1].
These effects support the --plot global option.
See also filter for a bandpass filter with steeper shoulders.
bandreject frequency[k] width[h|k|o|q]
Apply a band-reject filter. See the description of the bandpass
effect for details.
bass|treble gain [frequency[k] [width[s|h|k|o|q]]]
Boost or cut the bass (lower) or treble (upper) frequencies of
the audio using a two-pole shelving filter with a response simi-
lar to that of a standard hi-fi's tone-controls. This is also
known as shelving equalisation (EQ).
gain gives the gain at 0 Hz (for bass), or whichever is the
lower of ~22 kHz and the Nyquist frequency (for treble). Its
useful range is about -20 (for a large cut) to +20 (for a large
boost). Beware of Clipping when using a positive gain.
If desired, the filter can be fine-tuned using the following
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 Trans-
form (DFT) at a particular frame rate and over-sampling rate.
The -f and -o parameters may be used to adjust these parameters
and thus control the smoothness of the changes in pitch.
For example, an initial tone is generated, then bent three
times, yeilding 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.
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 modula-
tion is either sinusoidal (-s) or triangular (-t). Gain-out is
the volume of the output.
A typical delay is around 40ms to 60ms; the modulation speed is
best near 0.25Hz and the modulation depth around 2ms. For exam-
ple, a single delay:
play 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 aver-
aged to determine its volume; attacks refer to increases in vol-
ume 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 sensi-
tive 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 possi-
ble signal amplitude. The input values must be in a strictly
increasing order but the transfer function does not have to be
monotonically rising. If omitted, the value of out-dB1 defaults
to the same value as in-dB1; levels below in-dB1 are not com-
panded (but may have gain applied to them). The point 0,0 is
assumed but may be overridden (by 0,out-dBn). If the list is
preceded by a soft-knee-dB value, then the points at where adja-
cent line segments on the transfer function meet will be rounded
by the amount given. Typical values for the transfer function
are 6:-70,-60,-20.
The third (optional) parameter is an additional gain in dB to be
applied at all points on the transfer function and allows easy
adjustment of the overall gain.
The fourth (optional) parameter is an initial level to be
assumed for each channel when companding starts. This permits
the user to supply a nominal level initially, so that, for exam-
ple, 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.au asz-car.au compand 0.3,1 6:-70,-60,-20 -5 -90 0.2
The transfer function (`6:-70,...') says that very soft sounds
(below -70dB) will remain unchanged. This will stop the compan-
der from boosting the volume on `silent' passages such as
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.
This effect supports the --plot global option (for the transfer
function).
See also mcompand for a multiple-band companding effect.
contrast [enhancement-amount (75)]
Comparable with compression, this effect modifies an audio sig-
nal to make it sound louder. enhancement-amount controls the
amount of the enhancement and is a number in the range 0-100.
Note that enhancement-amount = 0 still gives a significant con-
trast enhancement. contrast is often used in conjunction with
the norm effect as follows:
sox infile outfile norm -i contrast
dcshift shift [limitergain]
DC Shift the audio, with basic linear amplitude formula. This
is most useful if your audio tends to not be centered around a
value of 0. Shifting it back will allow you to get the most
volume adjustments without clipping.
The first option is the dcshift value. It is a floating point
number that indicates the amount to shift.
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 out.au synth 5 sin %0 50 highpass 10
deemph Apply ISO 908 de-emphasis (a treble attenuation shelving filter)
to 44.1kHz (Compact Disc) audio.
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-empha-
sised 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 follow-
ing (one long) command plays a chime sound:
play -n synth sin %-21.5 sin %-14.5 sin %-9.5 sin %-5.5 \
sin %-2.5 sin %2.5 gain -5.4 fade h 0.008 2 1.5 \
delay 0 .27 .54 .76 1.01 1.3 remix - fade h 0.1 2.72 2.5
dither [-r|-t] [-s|-f filter] [depth]
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. The default (or with the -t option) is Tri-
angular (TPDF) white noise. The -r option can be used to select
Rectangular Probability Density Function (RPDF) white noise.
Noise-shaping (only for certain sample rates) can be selected
with -s. With the -f option, it is possible to select a partic-
ular 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 distiguished by the following properties: audibility of
noise, level of (inaudible, but in some circumstances, otherwise
problematic) shaped high frequency noise, and processing speed.
By default, the amount of noise added is +-1/2 bit for RPDF, +-1
bit for TPDF; the optional depth parameter (0.5 to 1) is a (lin-
ear or voltage) multiplier of this amount. Reducing this value
reduces the audibility of the added white noise, but correspond-
ingly creates residual quantization noise, so it should not nor-
mally be changed.
This effect should not be followed by any other effect that
affects the audio.
earwax Makes audio easier to listen to on headphones. Adds `cues' to
44.1kHz stereo (i.e. audio CD format) audio so that when lis-
tened to on headphones the stereo image is moved from inside
your head (standard for headphones) to outside and in front of
the listener (standard for speakers). See http://www.geoci-
ties.com/beinges for a full explanation.
echo gain-in gain-out <delay decay>
Add echoing to the audio. Echoes are reflected sound and can
occur naturally amongst mountains (and sometimes large build-
ings) when talking or shouting; digital echo effects emulate
this behaviour and are often used to help fill out the sound of
a single instrument or vocal. The time difference between the
original signal and the reflection is the `delay' (time), and
the loudness of the relected 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) ro-
bot 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 moun-
tains:
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]]
Add a fade effect to the beginning, end, or both of the audio.
For fade-ins, this starts from the first sample and ramps the
volume of the audio 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
volume 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 speci-
fied. 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 exam-
ple `8000s').
An optional type can be specified to change the type of enve-
lope. Choices are q for quarter of a sine wave, h for half a
sine wave, t for linear slope, l for logarithmic, and p for
inverted parabola. The default is logarithmic.
filter [low]-[high] [window-len [beta]]
Apply a sinc-windowed lowpass, highpass, or bandpass filter of
given window length to the signal. low refers to the frequency
of the lower 6dB corner of the filter. high refers to the fre-
quency of the upper 6dB corner of the filter.
A low-pass filter is obtained by leaving low unspecified, or 0.
A high-pass filter is obtained by leaving high unspecified, or
0, or greater than or equal to the Nyquist frequency.
The window-len, if unspecified, defaults to 128. Longer windows
give a sharper cut-off, smaller windows a more gradual cut-off.
The beta parameter determines the type of filter window used.
Any value greater than 2 is the beta for a Kaiser window. Beta
<= 2 selects a Blackman-Nuttall window. If unspecified, the
default is a Kaiser window with beta 16.
In the case of Kaiser window (beta > 2), lower betas produce a
somewhat faster transition from pass-band to stop-band, at the
cost of noticeable artifacts. A beta of 16 is the default, beta
less than 10 is not recommended. If you want a sharper cut-off,
don't use low beta's, use a longer sample window. A Blackman-
Nuttall window is selected by specifying any `beta' <= 2, and
the Blackman-Nuttall window has somewhat steeper cut-off than
the default Kaiser window. You will probably not need to use the
beta parameter at all, unless you are just curious about compar-
ing the effects of Blackman-Nuttall vs. Kaiser windows.
This effect supports the --plot global option.
flanger [delay depth regen width speed shape phase interp]
Apply a flanging effect to the audio. See [3] for a detailed
description of flanging.
All parameters are optional (right to left).
+-----------------------------------------------------------------+
| Range Default Description |
|delay 0 - 10 0 Base delay in milliseconds. |
|depth 0 - 10 2 Added swept delay in milliseconds. |
|regen -95 - 95 0 Percentage regeneration (delayed |
| signal feedback). |
|width 0 - 100 71 Percentage of delayed signal mixed |
| with original. |
|speed 0.1 - 10 0.5 Sweeps per second (Hz). |
|shape sin Swept wave shape: sine|triangle. |
|phase 0 - 100 25 Swept wave percentage phase-shift |
| for multi-channel (e.g. stereo) |
| flange; 0 = 100 = same phase on |
| each channel. |
|interp lin Digital delay-line interpolation: |
| linear|quadratic. |
+-----------------------------------------------------------------+
gain dB-gain
Apply an amplification or an attenuation to the audio signal.
The signal level is adjusted by the given number of dB - posi-
tive amplifies (beware of Clipping), negative attenuates.
See also the vol effect.
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 filter for filters with a steeper roll-off.
ladspa module [plugin] [argument...]
Apply a LADSPA [5] (Linux Audio Developer's Simple Plugin API)
plugin. Despite the name, LADSPA is not Linux-specific, and a
wide range of effects is 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 vari-
ble 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 refer-
ence 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 filter 8000- 32 100 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 filter -17500 256 \
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
preemphasis.
See also compand for a single-band companding effect.
mixer [ -l|-r|-f|-b|-1|-2|-3|-4|n{,n} ]
Reduce the number of audio channels by mixing or selecting chan-
nels, or increase the number of channels by duplicating chan-
nels. 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).
This effect is automatically used when the number of input chan-
nels differ from the number of output channels. When reducing
the number of channels it is possible to manually specify the
mixer effect and use the -l, -r, -f, -b, -1, -2, -3, -4, options
to select only the left, right, front, back channel(s) or spe-
cific 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, sep-
arated 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.
+----------------------------------------------------------+
|In Ch Out Ch Num Mappings |
| 2 1 2 l -> l, r -> l |
| 2 2 1 adjust balance |
| 4 1 4 lf -> l, rf -> l, lb -> l, rb -> l |
| 4 2 2 lf -> l&rf -> r, lb -> l&rb -> r |
| 4 4 1 adjust balance |
| 4 4 2 front balance, back balance |
+----------------------------------------------------------+
See also remix for a mixing effect that handles any number of
channels.
noiseprof [profile-file]
Calculate a profile of the audio for use in noise reduction.
See the description of the noisered effect for details.
noisered [profile-file [amount]]
Reduce noise in the audio signal by profiling and filtering.
This effect is moderately effective at removing consistent back-
ground noise such as hiss or hum. To use it, first run SoX with
the noiseprof effect on a section of audio that ideally would
contain silence but in fact contains noise - such sections are
typically found at the beginning or the end of a recording.
noiseprof will write out a noise profile to profile-file, or to
stdout if no profile-file or if `-' is given. E.g.
sox speech.au -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.au cleaned.au 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.au -n trim 0 1 noiseprof | play noisy.au noisered
norm [-i|-b] [level]
Normalise audio to 0dB FSD, to a given level relative to 0dB, or
normalise the balance of multi-channel audio. Requires tempo-
rary file space to store the audio to be normalised.
To create a normalised copy of an audio file,
sox infile outfile norm
can be used, though note that if `infile' has a simple encoding
(e.g. PCM), then
sox infile outfile vol `sox infile -n stat -v 2>&1`
(on systems that support this construct) might be quicker to
execute (though perhaps not to type!) as it doesn't require a
temporary file.
For a more complex example, suppose that `effect1' performs some
unknown or unpredictable attenuation and that `effect2' requires
up to 10dB of headroom, then
sox infile outfile effect1 norm -10 effect2 norm
gives both effect2 and the output file the highest possible sig-
nal levels.
Normally, audio is normalised based on the level of the channel
with the highest peak level, which means that whilst all chan-
nels are adjusted, only one channel attains the normalised
level. If the -i option is given, then each channel is treated
individually and will attain the normalised level.
If the -b option is given (with a multi-channel audio file),
then the audio channels will be balanced; i.e. the RMS level of
each channel will be normalised to that of the channel with the
highest RMS level. This can be used, for example, to correct
stereo imbalance. Note that -b can cause clipping.
In most cases, norm -3 should be the maximum level used at the
output file (to leave headroom for playback-resampling, etc.).
See also the discussions of Clipping and Replay Gain above.
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.
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 sam-
ples. 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, spec-
ify either the end position or specify a zero-length pad at the
start.
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 modu-
lation 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.
rate [-q|-l|-m|-h|-v] [override-options] RATE[k]
Change the audio sampling rate (i.e. resample the audio) to any
given RATE (even non-integer if this is supported by the output
file format) using a quality level defined as follows:
+--------------------------------------------------+
| Quality Band- Rej dB Typical Use |
| width |
|-q quick n/a ~=30 @ playback on |
| Fs/4 ancient hardware |
|-l low 80% 100 playback on old |
| hardware |
|-m medium 95% 100 audio playback |
|-h high 95% 125 16-bit mastering |
| (use with dither) |
|-v very high 95% 175 24-bit mastering |
+--------------------------------------------------+
where Band-width is the percentage of the audio 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'.
The `quick' algorithm uses cubic interpolation; all others use
band-limited interpolation. By default, all algorithms have a
`linear' phase response; for `medium', `high' and `very high',
the phase response is configurable (see below).
The rate effect is invoked automatically if SoX's -r option
specifies a rate that is different to that of the input file(s).
Alternatively, if this effect is given explicitly, then SoX's -r
option need not be given. For example, the following two com-
mands are equivalent:
sox input.au -r 48k output.au bass -3
sox input.au output.au bass -3 rate 48k
though the second command is more flexible as it allows rate
options to be given, and allows the effects to be ordered arbi-
trarily.
* * *
Warning: technically detailed discussion follows.
The simple quality selection described above provides settings
that satisfy the needs of the vast majority of resampling tasks.
Occasionally, however, it may be desirable to fine-tune the
resampler's filter response; this can be achieved using over-
ride options, as detailed in the following table:
+-----------------------------------------------------------------+
|-M/-I/-L Phase response = minimum/intermediate/linear |
|-s Steep filter (band-width = 99%) |
|-a Allow aliasing above the pass-band |
|-b 74-99.7 Any band-width % |
|-p 0-100 Any phase response (0 = minimum, 25 = intermediate, |
| 50 = linear, 100 = maximum) |
+-----------------------------------------------------------------+
N.B. Override options can not be used with the `quick' or `low'
quality algorithms.
All resamplers use filters that can sometimes create `echo'
(a.k.a. `ringing') artefacts with transient signals such as
those that occur with `finger snaps' or other highly percussive
sounds. Such artefacts are much more noticable to the human ear
if they occur before the transient (`pre-echo') than if they
occur after it (`post-echo'). Note that frequency of any such
artefacts is related to the smaller of the original and new sam-
pling rates but that if this is at least 44.1kHz, then the arte-
facts will lie outside the range of human hearing.
A phase response setting may be used to control the distribution
of any transient echo between `pre' and `post': with minimum
phase, there is no pre-echo but the longest post-echo; with lin-
ear phase, pre and post echo are in equal amounts (in signal
terms, but not audibility terms); the intermediate phase setting
attempts to find the best compromise by selecting a small length
(and level) of pre-echo and a medium lengthed post-echo.
Minimum, intermediate, or linear phase response is selected
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 fre-
quency content of the original signal (w.r.t. the orignal sample
rate when up-sampling, or the new sample rate when down-sam-
pling) is preserved during conversion. The term `pass-band' is
used to refer to all frequencies up to the band-width point
(e.g. for 44.1kHz sampling rate, and a resampling band-width of
95%, the pass-band represents frequencies from 0Hz (D.C.) to
circa 21kHz). Increasing the resampler's band-width results in
a slower conversion and can increase transient echo artefacts
(and vice versa).
The -s `steep filter' option changes resampling band-width from
the default 95% (based on the 3dB point), to 99%. The -b option
allows the band-width to be set to any value in the range
74-99.7 %, but note that band-width values greater than 99% are
not recommended for normal use as they can cause excessive tran-
sient echo.
If the -a option is given, then aliasing above the pass-band is
allowed. For example, with 44.1kHz sampling rate, and a resam-
pling 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
are reduced processing time, and reduced (by almost half) tran-
sient 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
default (high) quality resampling; overrides: steep filter,
allow aliasing; to 44.1kHz sample rate; dither output to 16-bit
WAV file.
sox input.wav -b 24 output.aiff rate -v -L -b 90 48k
very high quality resampling; overrides: linear phase, band-
width 90%; to 48k sample rate; store output to 24-bit AIFF file.
* * *
The pitch, speed and tempo effects all use the rate effect at
their core.
See also resample, polyphase and rabbit for other sample-rate
changing effects.
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.au output.au 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.au output.au 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.au output.au remix -
performs a mix-down of all input channels to mono.
By default, where an output channel is mixed from multiple (n)
input channels, each input channel will be scaled by a factor of
1/n. Custom mixing volumes can be set by following a given
input channel or range of input channels with a vol-spec (volume
specification). This is one of the letters p, i, or v, followed
by a volume number, the meaning of which depends on the given
letter and is defined as follows:
Letter Volume number Notes
p power adjust in dB 0 = no change
i power adjust in dB As `p', but invert
the audio
v voltage multiplier 1 = no change, 0.5
~= 6dB attenuation,
2 ~= 6dB gain, -1 =
invert
If an out-spec includes at least one vol-spec then, by default,
1/n scaling is not applied to any other channels in the same
out-spec (though may be in other out-specs). The -a (automatic)
option however, can be given to retain the automatic scaling in
this case. For example,
sox input.au output.au remix 1,2 3,4v0.8
results in channel level multipliers of 0.5,0.5 1,0.8, whereas
sox input.au output.au remix -a 1,2 3,4v0.8
results in channel level multipliers of 0.5,0.5 0.5,0.8.
The -m (manual) option disables all automatic volume adjust-
ments, so
sox input.au output.au 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.au is stereo, then
sox input.au output.au 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 chan-
nels). Where more than a few channels are involved, a script
such as the following (Bourne shell script) is useful:
#!/bin/sh
chans=`soxi -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.au containing six audio channels were given, the
script would produce six output files: input-01.au, input-02.au,
..., input-06.au.
See also mixer and swap for similar effects.
repeat count
Repeat the entire audio count times. 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 reverbera-
tion.
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.au gain -3 pad 0 3 reverb
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 anything below 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 peri-
ods of silence that are wanted. For example, if you have a song
with 2 seconds of silence in the middle and 2 second at the end,
you could set below-period to a value of 2 to skip over the
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 sec-
ond 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 mid-
dle 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 posi-
tive 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 sam-
ples. 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%
speed factor[c]
Adjust the audio speed (pitch and tempo together). factor is
either the ratio of the new speed to the old speed: greater than
1 speeds up, less than 1 slows down, or, if appended with the
letter `c', the number of cents (i.e. 100ths of a semitone) by
which the pitch (and tempo) should be adjusted: greater than 0
increases, less than 0 decreases.
By default, the speed change is performed by resampling with the
rate effect using its default quality/speed. For higher quality
or higher speed resampling, in addition to the speed effect,
specify the rate effect with the desired quality option.
spectrogram [options]
Create a spectrogram of the audio. 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 repre-
sented by the colour (or intensity) of the pixels in the X-Y
plane.
This effect supports only one channel; for multi-channel input
files, use either SoX's -c 1 option with the output file (to
obtain a spectrogram on the mix-down), or the remix n effect to
select a particular channel. Be aware though, that both of
these methods affect the audio in the effects chain.
-x num X-axis pixels/second, default 100. This controls the
width 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
--verbose is in effect).
The maximum width of the spectrogram is 999 pixels; if
the audio length and the given -x number are such that
this would be exceeded, then the spectrogram (and the
effects chain) will be truncated. To move the spectro-
gram to a point later in the audio stream, first invoke
the trim effect; e.g.
sox audio.ogg -n trim 1:00 spectrogram
starts the spectrogram at 1 minute through the audio.
-y num Y-axis resolution (1 - 4), default 2. This controls the
height of the spectrogram; num can be from 1 (low fre-
quency resolution) to 4 (high frequency resolution). For
values greater than 2, the resulting image may be too
tall to display on the screen; if so, a graphic manipula-
tion package (such as ImageMagick(1)) can be used to re-
size the image.
To increase the frequency resolution without increasing
the height of the spectrogram, the rate effect may be
invoked to reduce the sampling rate of the signal before
invoking spectrogram; e.g.
sox audio.ogg -r 4k -n rate spectrogram
allows detailed analysis of frequencies up to 2kHz (half
the sampling rate).
-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 spec-
trogram display, and vice versa.
-q num Sets the Z-axis quantisation, i.e. the number of differ-
ent colours (or intensities) in which to render Z-axis
values. A small number (e.g. 4) will give a
`poster'-like effect making it easier to discern magni-
tude bands of similar level. Small numbers also usually
result in small PNG files. The number given specifies
the number of colours to use inside the Z-axis range; two
colours are reserved to represent out-of-range values.
-w name
Window: Hann (default), Hamming, Bartlett, Rectangular or
Kaiser. The spectrogram is produced using the Discrete
Fourier Transform (DFT) algorithm. A significant parame-
ter 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 win-
dow. Bartlett and Rectangular windows are also avail-
able. Selecting a window other than Hann will usually
require a corresponding -z setting.
-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 eas-
ier to differentiate different levels. If this option is
used in conjunction with -m, the result will be a hybrid
monochrome/colour palette.
-p num Permute the colours in a colour or hybrid palette. The
num parameter (from 1 to 6) selects the permutation.
-l Creates a `printer friendly' spectrogram with a light
background (the default has a dark background).
-a Suppress the display of the axis lines. This is some-
times useful in helping to discern artefacts at the spec-
trogram edges.
-t text
Set the image title - text to display above the spectro-
gram.
-c text
Set the image comment - text to display below and to the
left of the spectrogram.
-o text
Name of the spectrogram output PNG file, default `spec-
trogram.png'.
For example, let's see what the spectrogram of a swept triangu-
lar wave looks like:
sox -n -n synth 6 tri 10k:14k spectrogram -z 100 -w k
Append the following to the `chime' example in the delay effect
to see its spectrogram:
rate 2k spectrogram -x 200 -Z -15 -w k
For the ability to perform off-line processing of spectral data,
see the stat effect.
splice { 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.
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 sec-
tion to splice on should be trimmed with the same excess (before
the ideal joining point), plus an additional leeway (default
0.005 seconds). SoX should then be invoked with the two audio
sections as input files and the splice effect given with the
position at which to perform the splice - this is length of the
first audio section (including the excess).
For example, a long song begins with two verses which start (as
determined e.g. by using the play command with the trim (start)
effect) at times 0:30.125 and 1:03.432. The following commands
cut out the first verse:
sox too-long.au part1.au trim 0 30.130
(5 ms excess, after the first verse starts)
sox long.au part2.au trim 1:03.422
(5 ms excess plus 5 ms leeway, before the second verse starts)
sox part1.au part2.au just-right.au splice 30.130
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.au trim `expr $2 - $e - $l`s \
`expr $3 - $2 + $e + $l + $e`s
sox "$1" part1.au trim 0 `expr $4 + $e`s
sox "$1" part2.au trim `expr $4 + $3 - $2 - $e - $l`s
sox part1.au piece.au part2.au "$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.
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 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).
* * *
It is also possible to use this effect to perform general cross-
fades, e.g. to join two songs. In this case, excess would typi-
cally be an number of seconds, and leeway should be set to zero.
stat [-s scale] [-rms] [-freq] [-v] [-d]
Display time and frequency domain statistical information about
the audio. Audio is passed unmodified through the SoX process-
ing chain.
The information is output to the `standard error' (stderr)
stream and is calculated, where n is the duration of the audio
in samples, c is the number of audio channels, r is the audio
sample rate, and xk represents the PCM value (in the range -1 to
+1 by default) of each successive sample in the audio, as fol-
lows:
Samples read nxc
Length (seconds) n-:-r
Scaled by See -s below.
Maximum amplitude max(xk) The maximum sample
value in the audio;
usually this will
be a positive num-
ber.
Minimum amplitude min(xk) The minimum sample
value in the audio;
usually this will
be a negative num-
ber.
Midline amplitude 1/2_in(xk)+1/2max(xk)
Mean norm 1/n>|xk| The average of the
absolute value of
each sample in the
_ audio.
Mean amplitude 1/n>xk The average of each
sample in the
audio. If this
figure is non-zero,
then it indicates
the presence of a
D.C. offset (which
could be removed
using the dcshift
_ effect).
RMS amplitude \/(1/n>xk2) The level of a D.C.
signal that would
have the same power
as the audio's
average power.
Maximum delta max(|xk-xk-1|)
Minimum delta min(|x_k-xk-1|)
Mean delta 1/n-1>|xk_-xk-1|
RMS delta \/(1/n-1>(xk-xk-1)2)
Rough frequency In Hz.
Volume Adjustment The parameter to
the vol effect
which would make
the audio as loud
as possible without
clipping. Note:
See the discussion
on Clipping above
for reasons why it
is rarely a good
idea actually to do
this.
The -s option can be used to scale the input data by a given
factor. The default value of scale is 2147483647 (i.e. the max-
imum value of a 32-bit signed integer). Internal effects always
work with signed long PCM data and so the value should relate to
this fact.
The -rms option will convert all output average values to `root
mean square' format.
The -v option displays only the `Volume Adjustment' value.
The -freq option calculates the input's power spectrum (4096
point DFT) instead of the statistics listed above.
The -d option displays a hex dump of the 32-bit signed PCM data
audio in SoX's internal buffer. This is mainly used to help
track down endian problems that sometimes occur in cross-plat-
form versions of SoX.
swap [1 2 | 1 2 3 4]
Swap channels in multi-channel audio files. Optionally, you may
specify the channel order you would like the output in. This
defaults to output channel 2 and then 1 for stereo and 2, 1, 4,
3 for quad-channels. An interesting feature is that you may
duplicate a given channel by overwriting another. This is done
by repeating an output channel on the command-line. For exam-
ple, swap 2 2 will overwrite channel 1 with channel 2; creating
a stereo file with both channels containing the same audio.
See also the remix effect.
stretch factor [window fade shift fading]
Change the audio duration (but not its pitch). This effect is
broadly equivalent to the tempo effect with (factor inverted
and) search set to zero, so in general, its results are compara-
tively poor; it is retained as it can sometimes out-perform
tempo for small factors.
factor of stretching: >1 lengthen, <1 shorten duration. window
size is in ms. Default is 20ms. The fade option, can be `lin'.
shift ratio, in [0 1]. Default depends on stretch factor. 1 to
shorten, 0.8 to lengthen. The fading ratio, in [0 0.5]. The
amount of a fade's default depends on factor and shift.
See also the tempo effect.
synth [len] {[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 cas-
caded to produce more complex waveforms; at each stage it is
possible to choose whether the generated waveform will be mixed
with, or modulated onto the output from the previous stage.
Audio for each channel in a multi-channel audio file can be syn-
thesised 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 nor-
mally needed, a `null file' (with the special name -n) is often
given instead (and the length specified as a parameter to synth
or by another given effect that 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.au synth 3 sine 300-3300
and this produces an 8 kHz version:
sox -r 8000 -n output.au 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.au synth 3 sine 300-3300 brownnoise
The following example shows how two synth effects can be cas-
caded to create a more complex waveform:
sox -n output.au synth 0.5 sine 200-500 \
synth 0.5 sine fmod 700-100
Frequencies can also be given as a number of musical semitones
relative to `middle A' (440 Hz) by prefixing a `%' character;
for example, the following could be used to help tune a guitar's
`E' strings:
play -n synth sine %-17
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 most cases, you will want to fol-
low this effect with the gain effect to prevent this from hap-
pening. (See also Clipping above.)
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, pinknoise, brownnoise; 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); for both, default=%0. If freq2 is given, then len
must also have been given and the generated tone will be swept
between the given frequencies. The two given frequencies must
be separated by one of the characters `:', `+', `/', or `-'.
This character is used to specify the sweep function as follows:
: Linear: the tone will change by a fixed number of hertz
per second.
+ Square: a second-order function is used to change the
tone.
/ Exponential: the tone will change by a fixed number of
semitones per second.
- Exponential: as `/', but initial phase always zero, and
stepped (less smooth) frequency changes.
Not used for noise.
off is the bias (DC-offset) of the signal in percent; default=0.
ph is the phase shift in percentage of 1 cycle; default=0. Not
used for noise.
p1 is the percentage of each cycle that is `on' (square), or
`rising' (triangle, exp, trapezium); default=50 (square, trian-
gle, exp), default=10 (trapezium).
p2 (trapezium): the percentage through each cycle at which
`falling' begins; default=50. exp: the amplitude in percent;
default=100.
p3 (trapezium): the percentage through each cycle at which
`falling' ends; default=60.
tempo [-q] factor [segment [search [overlap]]]
Change the audio tempo (but not its pitch). The audio is
chopped up into segments which are then shifted in the time
domain and overlapped (cross-faded) at points where their wave-
forms are most similar (as determined by measurement of `least
squares').
By default, linear searches are used to find the best overlap-
ping points; if the optional -q parameter is given, tree
searches are used instead, giving a quicker, but possibly lower
quality, result.
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. 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), 50 ms may give a better result.
When changing the tempo of speech, a segment size of around
30 ms often works well.
The optional search parameter gives the audio length in mil-
liseconds (default 14) over which the algorithm will search for
overlapping points. Larger values use more processing time and
do not necessarily produce better results.
The optional overlap parameter gives the segment overlap length
in milliseconds (default 12).
See also speed for an effect that changes tempo and pitch
together, and pitch for an effect that changes pitch without
changing tempo.
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).
Note: This effect is a special case of the synth effect.
trim start [length]
Trim can trim off unwanted audio from the beginning and end of
the audio. Audio is not sent to the output stream until the
start location is reached.
The optional length parameter tells the number of samples to
output after the start sample and is used to trim off the back
side of the audio. Using a value of 0 for the start parameter
will allow trimming off the back side only.
Both options can be specified using either an amount of time or
an exact count of samples. The format for specifying lengths in
time is hh:mm:ss.frac. A start value of 1:30.5 will not start
until 1 minute, thirty and 1/2 seconds into the audio. The for-
mat for specifying sample counts is the number of samples with
the letter `s' appended to it. A value of 8000s will wait until
8000 samples are read before starting to process audio.
vol gain [type [limitergain]]
Apply an amplification or an attenuation to the audio signal.
Unlike the -v option (which is used for balancing multiple input
files as they enter the SoX effects processing chain), vol is an
effect like any other so can be applied anywhere, and several
times if necessary, during the processing chain.
The amount to change the volume is given by gain which is inter-
preted, according to the given type, as follows: if type is
amplitude (or is omitted), then gain is an amplitude (i.e. volt-
age 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 addi-
tion 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 compand for a dynamic-range compression/expansion/lim-
iting 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.
key [-q] shift [segment [search [overlap]]]
Change the audio key (i.e. pitch but not tempo). This is just
an alias for the pitch effect.
pan direction
Mix the audio from one channel to another. Use mixer or remix
instead of this effect.
The direction is a value from -1 to 1. -1 represents far left
and 1 represents far right.
polyphase [-w nut|ham] [-width n] [-cut-off c]
Change the sampling rate using `polyphase interpolation', a DSP
algorithm. polyphase copes with only certain rational fraction
resampling ratios, and, compared with the rate effect, is gener-
ally slow, memory intensive, and has poorer stop-band rejection.
If the -w parameter is nut, then a Blackman-Nuttall (~90 dB
stop-band) window will be used; ham selects a Hamming (~43 dB
stop-band) window. The default is Blackman-Nuttall.
The -width parameter specifies the (approximate) width of the
filter. The default is 1024 samples, which produces reasonable
results.
The -cut-off value (c) specifies the filter cut-off frequency in
terms of fraction of frequency bandwidth, also know as the
Nyquist frequency. See the resample effect for further informa-
tion on Nyquist frequency. If up-sampling, then this is the
fraction of the original signal that should go through. If
down-sampling, this is the fraction of the signal left after
down-sampling. The default is 0.95.
See also rate, rabbit and resample for other sample-rate chang-
ing effects.
rabbit [-c0|-c1|-c2|-c3|-c4]
Change the sampling rate using libsamplerate, also known as
`Secret Rabbit Code'. This effect is optional and, due to
licence issues, is not included in all versions of SoX. Com-
pared with the rate effect, rabbit is very slow.
See http://www.mega-nerd.com/SRC for details of the algorithms.
Algorithms 0 through 2 are progressively faster and lower qual-
ity versions of the sinc algorithm; the default is -c0. Algo-
rithm 3 is zero-order hold, and 4 is linear interpolation.
See also rate, polyphase and resample for other sample-rate
changing effects, and see resample for more discussion of resam-
pling.
resample [-qs|-q|-ql] [rolloff [beta]]
Change the sampling rate using simulated analog filtration.
Compared with the rate effect, resample is slow, and has poorer
stop-band rejection. Only the low quality option works with all
resampling ratios.
By default, linear interpolation of the filter coefficients is
used, with a window width about 45 samples at the lower of the
two rates. This gives an accuracy of about 16 bits, but insuf-
ficient stop-band rejection in the case that you want to have
roll-off greater than about 0.8 of the Nyquist frequency.
The -q* options will change the default values for roll-off and
beta as well as use quadratic interpolation of filter coeffi-
cients, resulting in about 24 bits precision. The -qs, -q, or
-ql options specify increased accuracy at the cost of lower exe-
cution speed. It is optional to specify roll-off and beta
parameters when using the -q* options.
Following is a table of the reasonable defaults which are built-
in to SoX:
+--------------------------------------------------+
|Option Window Roll-off Beta Interpolation |
|(none) 45 0.80 16 linear |
| -qs 45 0.80 16 quadratic |
| -q 75 0.875 16 quadratic |
| -ql 149 0.94 16 quadratic |
+--------------------------------------------------+
-qs, -q, or -ql use window lengths of 45, 75, or 149 samples,
respectively, at the lower sample-rate of the two files. This
means progressively sharper stop-band rejection, at proportion-
ally slower execution times.
rolloff refers to the cut-off frequency of the low pass filter
and is given in terms of the Nyquist frequency for the lower
sample rate. rolloff therefore should be something between 0
and 1, in practise 0.8-0.95. The defaults are indicated above.
The Nyquist frequency is equal to half the sample rate. Logi-
cally, this is because the A/D converter needs at least 2 sam-
ples to detect 1 cycle at the Nyquist frequency. Frequencies
higher then the Nyquist will actually appear as lower frequen-
cies to the A/D converter and is called aliasing. Normally, A/D
converts run the signal through a lowpass filter first to avoid
these problems.
Similar problems will happen in software when reducing the sam-
ple rate of an audio file (frequencies above the new Nyquist
frequency can be aliased to lower frequencies). Therefore, a
good resample effect will remove all frequency information above
the new Nyquist frequency.
The rolloff refers to how close to the Nyquist frequency this
cut-off is, with closer being better. When increasing the sam-
ple rate of an audio file you would not expect to have any fre-
quencies exist that are past the original Nyquist frequency.
Because of resampling properties, it is common to have aliasing
artifacts created above the old Nyquist frequency. In that case
the rolloff refers to how close to the original Nyquist fre-
quency to use a highpass filter to remove these artifacts, with
closer also being better.
The beta, if unspecified, defaults to 16. This selects a Kaiser
window. You can select a Blackman-Nuttall window by specifying
anything <= 2 here. For more discussion of beta, look under the
filter effect.
Default parameters are, as indicated above, Kaiser window of
length 45, roll-off 0.80, beta 16, linear interpolation.
Note: -qs is only slightly slower, but more accurate for 16-bit
or higher precision.
See also rate, polyphase and rabbit for other sample-rate chang-
ing effects. There is a detailed analysis of resample and
polyphase at http://leute.server.de/wilde/resample.html; see
rabbit for a pointer to its own documentation.
DIAGNOSTICS
Exit status is 0 for no error, 1 if there is a problem with the com-
mand-line parameters, or 2 if an error occurs during file processing.
BUGS
Please report any bugs found in this version of SoX to the mailing list
(sox-users@lists.sourceforge.net).
SEE ALSO
soxi(1), soxformat(7), libsox(3)
audacity(1), ImageMagick(1), gnuplot(1), octave(1), wget(1)
The SoX web site at http://sox.sourceforge.net
SoX scripting examples at http://sox.sourceforge.net/Docs/Scripts
References
[1] R. Bristow-Johnson, Cookbook formulae for audio EQ biquad filter
coefficients, http://musicdsp.org/files/Audio-EQ-Cookbook.txt
[2] Wikipedia, Q-factor, http://en.wikipedia.org/wiki/Q_factor
[3] Scott Lehman, Effects Explained, http://harmony-cen-
tral.com/Effects/effects-explained.html
[4] Wikipedia, Decibel, http://en.wikipedia.org/wiki/Decibel
[5] Richard Furse, Linux Audio Developer's Simple Plugin API,
http://www.ladspa.org
[6] Richard Furse, Computer Music Toolkit, http://www.ladspa.org/cmt
[7] Steve Harris, LADSPA plugins, http://plugin.org.uk
LICENSE
Copyright 1991 Lance Norskog and Sundry Contributors.
Copyright 1998-2008 Chris Bagwell and SoX Contributors.
This program is free software; you can redistribute it and/or modify it
under the terms of the GNU General Public License as published by the
Free Software Foundation; either version 2, or (at your option) any
later version.
This program is distributed in the hope that it will be useful, but
WITHOUT ANY WARRANTY; without even the implied warranty of MER-
CHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU General
Public License for more details.
AUTHORS
Chris Bagwell (cbagwell@users.sourceforge.net). Other authors and con-
tributors are listed in the AUTHORS file that is distributed with the
source code.
sox October 28, 2008 sox(1)
sox 14.2.0 - Generated Tue Nov 11 07:30:14 CST 2008