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SoX(1) |
Sound eXchange |
SoX(1) |
SoX - Sound eXchange, the Swiss Army knife of audio manipulation
sox [global-options] [format-options] infile1
[[format-options] infile2] ... [format-options] outfile
[effect [effect-options]] ...
play [global-options] [format-options] infile1
[[format-options] infile2] ... [format-options]
[effect [effect-options]] ...
rec [global-options] [format-options] outfile
[effect [effect-options]] ...
SoX reads and writes audio files in most popular formats and can optionally
apply effects to them. It can combine multiple input sources, synthesise
audio, and, on many systems, act as a general purpose audio player or a
multi-track audio recorder. It also has limited ability to split the input
into multiple output files.
All SoX functionality is available using just the sox
command. To simplify playing and recording audio, if SoX is invoked as
play, the output file is automatically set to be the default sound
device, and if invoked as rec, the default sound device is used as an
input source. Additionally, the soxi(1) command provides a convenient
way to just query audio file header information.
The heart of SoX is a library called libSoX. Those interested in
extending SoX or using it in other programs should refer to the libSoX
manual page: libsox(3).
SoX is a command-line audio processing tool, particularly suited
to making quick, simple edits and to batch processing. If you need an
interactive, graphical audio editor, use audacity(1).
The overall SoX processing chain can be summarised as follows:
Input(s) → Combiner → Effects → Output(s) |
Note however, that on the SoX command line, the positions of the
Output(s) and the Effects are swapped w.r.t. the logical flow just shown.
Note also that whilst options pertaining to files are placed before their
respective file name, the opposite is true for effects. To show how this
works in practice, here is a selection of examples of how SoX might be used.
The simple
sox recital.au recital.wav
translates an audio file in Sun AU format to a Microsoft WAV file, whilst
sox recital.au -b 16 recital.wav channels 1 rate 16k fade 3 norm
performs the same format translation, but also applies four effects (down-mix to
one channel, sample rate change, fade-in, nomalize), and stores the result at
a bit-depth of 16.
sox -r 16k -e signed -b 8 -c 1 voice-memo.raw voice-memo.wav
converts `raw' (a.k.a. `headerless') audio to a self-describing file format,
sox slow.aiff fixed.aiff speed 1.027
adjusts audio speed,
sox short.wav long.wav longer.wav
concatenates two audio files, and
sox -m music.mp3 voice.wav mixed.flac
mixes together two audio files.
play "The Moonbeams/Greatest/*.ogg" bass +3
plays a collection of audio files whilst applying a bass boosting effect,
play -n -c1 synth sin %-12 sin %-9 sin %-5 sin %-2 fade h 0.1 1 0.1
plays a synthesised `A minor seventh' chord with a pipe-organ sound,
rec -c 2 radio.aiff trim 0 30:00
records half an hour of stereo audio, and
play -q take1.aiff & rec -M take1.aiff take1-dub.aiff
(with POSIX shell and where supported by hardware) records a new track in a
multi-track recording. Finally,
rec -r 44100 -b 16 -e signed-integer -p \
silence 1 0.50 0.1% 1 10:00 0.1% | \
sox -p song.ogg silence 1 0.50 0.1% 1 2.0 0.1% : \
newfile : restart
records a stream of audio such as LP/cassette and splits in to multiple audio
files at points with 2 seconds of silence. Also, it does not start recording
until it detects audio is playing and stops after it sees 10 minutes of
silence.
N.B. The above is just an overview of SoX's capabilities; detailed
explanations of how to use all SoX parameters, file formats, and
effects can be found below in this manual, in soxformat(7), and in
soxi(1).
SoX can work with `self-describing' and `raw' audio files. `self-describing'
formats (e.g. WAV, FLAC, MP3) have a header that completely describes the
signal and encoding attributes of the audio data that follows. `raw' or
`headerless' formats do not contain this information, so the audio
characteristics of these must be described on the SoX command line or inferred
from those of the input file.
The following four characteristics are used to describe the format
of audio data such that it can be processed with SoX:
- sample rate
- The sample rate in samples per second (`Hertz' or `Hz'). Digital telephony
traditionally uses a sample rate of 8000 Hz (8 kHz), though
these days, 16 and even 32 kHz are becoming more common. Audio
Compact Discs use 44100 Hz (44.1 kHz). Digital Audio Tape
and many computer systems use 48 kHz. Professional audio systems
often use 96 kHz.
- sample size
- The number of bits used to store each sample. Today, 16-bit is commonly
used. 8-bit was popular in the early days of computer audio. 24-bit is
used in the professional audio arena. Other sizes are also used.
- data encoding
- The way in which each audio sample is represented (or `encoded'). Some
encodings have variants with different byte-orderings or bit-orderings.
Some compress the audio data so that the stored audio data takes up less
space (i.e. disk space or transmission bandwidth) than the other format
parameters and the number of samples would imply. Commonly-used encoding
types include floating-point, μ-law, ADPCM, signed-integer PCM,
MP3, and FLAC.
- channels
- The number of audio channels contained in the file. One (`mono') and two
(`stereo') are widely used. `Surround sound' audio typically contains six
or more channels.
The term `bit-rate' is a measure of the amount of storage occupied
by an encoded audio signal over a unit of time. It can depend on all of the
above and is typically denoted as a number of kilo-bits per second (kbps).
An A-law telephony signal has a bit-rate of 64 kbps. MP3-encoded stereo
music typically has a bit-rate of 128-196 kbps. FLAC-encoded stereo music
typically has a bit-rate of 550-760 kbps.
Most self-describing formats also allow textual `comments' to be
embedded in the file that can be used to describe the audio in some way,
e.g. for music, the title, the author, etc.
One important use of audio file comments is to convey `Replay
Gain' information. SoX supports applying Replay Gain information (for
certain input file formats only; currently, at least FLAC and Ogg Vorbis),
but not generating it. Note that by default, SoX copies input file comments
to output files that support comments, so output files may contain Replay
Gain information if some was present in the input file. In this case, if
anything other than a simple format conversion was performed then the output
file Replay Gain information is likely to be incorrect and so should be
recalculated using a tool that supports this (not SoX).
The soxi(1) command can be used to display information from
audio file headers.
There are several mechanisms available for SoX to use to determine or set the
format characteristics of an audio file. Depending on the circumstances,
individual characteristics may be determined or set using different
mechanisms.
To determine the format of an input file, SoX will use, in order
of precedence and as given or available:
- 1.
- Command-line format options.
- 2.
- The contents of the file header.
- 3.
- The filename extension.
To set the output file format, SoX will use, in order of
precedence and as given or available:
- 1.
- Command-line format options.
- 2.
- The filename extension.
- 3.
- The input file format characteristics, or the closest that is supported by
the output file type.
For all files, SoX will exit with an error if the file type cannot
be determined. Command-line format options may need to be added or changed
to resolve the problem.
The play and rec commands are provided so that basic playing and
recording is as simple as
play existing-file.wav
and
rec new-file.wav
These two commands are functionally equivalent to
sox existing-file.wav -d
and
sox -d new-file.wav
Of course, further options and effects (as described below) can be added to the
commands in either form.
Some systems provide more than one type of (SoX-compatible) audio
driver, e.g. ALSA & OSS, or SUNAU & AO. Systems can also have more
than one audio device (a.k.a. `sound card'). If more than one audio driver
has been built-in to SoX, and the default selected by SoX when recording or
playing is not the one that is wanted, then the AUDIODRIVER
environment variable can be used to override the default. For example (on
many systems):
set AUDIODRIVER=oss
play ...
The AUDIODEV environment variable can be used to override the default
audio device, e.g.
set AUDIODEV=/dev/dsp2
play ...
sox ... -t oss
or
set AUDIODEV=hw:soundwave,1,2
play ...
sox ... -t alsa
Note that the way of setting environment variables varies from system to system
- for some specific examples, see `SOX_OPTS' below.
When playing a file with a sample rate that is not supported by
the audio output device, SoX will automatically invoke the rate
effect to perform the necessary sample rate conversion. For compatibility
with old hardware, the default rate quality level is set to `low'.
This can be changed by explicitly specifying the rate effect with a
different quality level, e.g.
play ... rate -m
or by using the --play-rate-arg option (see below).
On some systems, SoX allows audio playback volume to be adjusted
whilst using play. Where supported, this is achieved by tapping the
`v' & `V' keys during playback.
To help with setting a suitable recording level, SoX includes a
peak-level meter which can be invoked (before making the actual recording)
as follows:
rec -n
The recording level should be adjusted (using the system-provided mixer program,
not SoX) so that the meter is at most occasionally full scale, and
never `in the red' (an exclamation mark is shown). See also -S below.
Many file formats that compress audio discard some of the audio signal
information whilst doing so. Converting to such a format and then converting
back again will not produce an exact copy of the original audio. This is the
case for many formats used in telephony (e.g. A-law, GSM) where low signal
bandwidth is more important than high audio fidelity, and for many formats
used in portable music players (e.g. MP3, Vorbis) where adequate fidelity can
be retained even with the large compression ratios that are needed to make
portable players practical.
Formats that discard audio signal information are called `lossy'.
Formats that do not are called `lossless'. The term `quality' is used as a
measure of how closely the original audio signal can be reproduced when
using a lossy format.
Audio file conversion with SoX is lossless when it can be, i.e.
when not using lossy compression, when not reducing the sampling rate or
number of channels, and when the number of bits used in the destination
format is not less than in the source format. E.g. converting from an 8-bit
PCM format to a 16-bit PCM format is lossless but converting from an 8-bit
PCM format to (8-bit) A-law isn't.
N.B. SoX converts all audio files to an internal
uncompressed format before performing any audio processing. This means that
manipulating a file that is stored in a lossy format can cause further
losses in audio fidelity. E.g. with
sox long.mp3 short.mp3 trim 10
SoX first decompresses the input MP3 file, then applies the trim effect,
and finally creates the output MP3 file by re-compressing the audio - with a
possible reduction in fidelity above that which occurred when the input file
was created. Hence, if what is ultimately desired is lossily compressed audio,
it is highly recommended to perform all audio processing using lossless file
formats and then convert to the lossy format only at the final stage.
N.B. Applying multiple effects with a single SoX invocation
will, in general, produce more accurate results than those produced using
multiple SoX invocations.
Dithering is a technique used to maximise the dynamic range of audio stored at a
particular bit-depth. Any distortion introduced by quantisation is
decorrelated by adding a small amount of white noise to the signal. In most
cases, SoX can determine whether the selected processing requires dither and
will add it during output formatting if appropriate.
Specifically, by default, SoX automatically adds TPDF dither when
the output bit-depth is less than 24 and any of the following are true:
- bit-depth reduction has been specified explicitly using a command-line
option
- the output file format supports only bit-depths lower than that of the
input file format
- an effect has increased effective bit-depth within the internal processing
chain
For example, adjusting volume with vol 0.25 requires two
additional bits in which to losslessly store its results (since 0.25 decimal
equals 0.01 binary). So if the input file bit-depth is 16, then SoX's
internal representation will utilise 18 bits after processing this volume
change. In order to store the output at the same depth as the input,
dithering is used to remove the additional bits.
Use the -V option to see what processing SoX has
automatically added. The -D option may be given to override automatic
dithering. To invoke dithering manually (e.g. to select a noise-shaping
curve), see the dither effect.
Clipping is distortion that occurs when an audio signal level (or `volume')
exceeds the range of the chosen representation. In most cases, clipping is
undesirable and so should be corrected by adjusting the level prior to the
point (in the processing chain) at which it occurs.
In SoX, clipping could occur, as you might expect, when using the
vol or gain effects to increase the audio volume. Clipping
could also occur with many other effects, when converting one format to
another, and even when simply playing the audio.
Playing an audio file often involves resampling, and processing by
analogue components can introduce a small DC offset and/or amplification,
all of which can produce distortion if the audio signal level was initially
too close to the clipping point.
For these reasons, it is usual to make sure that an audio file's
signal level has some `headroom', i.e. it does not exceed a particular level
below the maximum possible level for the given representation. Some
standards bodies recommend as much as 9dB headroom, but in most cases, 3dB
(≈ 70% linear) is enough. Note that this wisdom seems to have been
lost in modern music production; in fact, many CDs, MP3s, etc. are now
mastered at levels above 0dBFS i.e. the audio is clipped as
delivered.
SoX's stat and stats effects can assist in
determining the signal level in an audio file. The gain or vol
effect can be used to prevent clipping, e.g.
sox dull.wav bright.wav gain -6 treble +6
guarantees that the treble boost will not clip.
If clipping occurs at any point during processing, SoX will
display a warning message to that effect.
See also -G and the gain and norm
effects.
SoX's input combiner can be configured (see OPTIONS below) to combine multiple
files using any of the following methods: `concatenate', `sequence', `mix',
`mix-power', `merge', or `multiply'. The default method is `sequence' for
play, and `concatenate' for rec and sox.
For all methods other than `sequence', multiple input files must
have the same sampling rate. If necessary, separate SoX invocations can be
used to make sampling rate adjustments prior to combining.
If the `concatenate' combining method is selected (usually, this
will be by default) then the input files must also have the same number of
channels. The audio from each input will be concatenated in the order given
to form the output file.
The `sequence' combining method is selected automatically for
play. It is similar to `concatenate' in that the audio from each
input file is sent serially to the output file. However, here the output
file may be closed and reopened at the corresponding transition between
input files. This may be just what is needed when sending different types of
audio to an output device, but is not generally useful when the output is a
normal file.
If either the `mix' or `mix-power' combining method is selected
then two or more input files must be given and will be mixed together to
form the output file. The number of channels in each input file need not be
the same, but SoX will issue a warning if they are not and some channels in
the output file will not contain audio from every input file. A mixed audio
file cannot be un-mixed without reference to the original input files.
If the `merge' combining method is selected then two or more input
files must be given and will be merged together to form the output file. The
number of channels in each input file need not be the same. A merged audio
file comprises all of the channels from all of the input files. Un-merging
is possible using multiple invocations of SoX with the remix effect.
For example, two mono files could be merged to form one stereo file. The
first and second mono files would become the left and right channels of the
stereo file.
The `multiply' combining method multiplies the sample values of
corresponding channels (treated as numbers in the interval -1 to +1). If the
number of channels in the input files is not the same, the missing channels
are considered to contain all zero.
When combining input files, SoX applies any specified effects
(including, for example, the vol volume adjustment effect) after the
audio has been combined. However, it is often useful to be able to set the
volume of (i.e. `balance') the inputs individually, before combining takes
place.
For all combining methods, input file volume adjustments can be
made manually using the -v option (below) which can be given for one
or more input files. If it is given for only some of the input files then
the others receive no volume adjustment. In some circumstances, automatic
volume adjustments may be applied (see below).
The -V option (below) can be used to show the input file
volume adjustments that have been selected (either manually or
automatically).
There are some special considerations that need to made when
mixing input files:
Unlike the other methods, `mix' combining has the potential to
cause clipping in the combiner if no balancing is performed. In this case,
if manual volume adjustments are not given, SoX will try to ensure that
clipping does not occur by automatically adjusting the volume (amplitude) of
each input signal by a factor of ¹/n, where n is the number of input
files. If this results in audio that is too quiet or otherwise unbalanced
then the input file volumes can be set manually as described above. Using
the norm effect on the mix is another alternative.
If mixed audio seems loud enough at some points but too quiet in
others then dynamic range compression should be applied to correct this -
see the compand effect.
With the `mix-power' combine method, the mixed volume is
approximately equal to that of one of the input signals. This is achieved by
balancing using a factor of ¹/√n instead of ¹/n. Note
that this balancing factor does not guarantee that clipping will not occur,
but the number of clips will usually be low and the resultant distortion is
generally imperceptible.
SoX's default behaviour is to take one or more input files and write them to a
single output file.
This behaviour can be changed by specifying the pseudo-effect
`newfile' within the effects list. SoX will then enter multiple output
mode.
In multiple output mode, a new file is created when the effects
prior to the `newfile' indicate they are done. The effects chain listed
after `newfile' is then started up and its output is saved to the new
file.
In multiple output mode, a unique number will automatically be
appended to the end of all filenames. If the filename has an extension then
the number is inserted before the extension. This behaviour can be
customized by placing a %n anywhere in the filename where the number should
be substituted. An optional number can be placed after the % to indicate a
minimum fixed width for the number.
Multiple output mode is not very useful unless an effect that will
stop the effects chain early is specified before the `newfile'. If end of
file is reached before the effects chain stops itself then no new file will
be created as it would be empty.
The following is an example of splitting the first 60 seconds of
an input file into two 30 second files and ignoring the rest.
sox song.wav ringtone%1n.wav trim 0 30 : newfile : trim 0 30
Usually SoX will complete its processing and exit automatically once it has read
all available audio data from the input files.
If desired, it can be terminated earlier by sending an interrupt
signal to the process (usually by pressing the keyboard interrupt key which
is normally Ctrl-C). This is a natural requirement in some circumstances,
e.g. when using SoX to make a recording. Note that when using SoX to play
multiple files, Ctrl-C behaves slightly differently: pressing it once causes
SoX to skip to the next file; pressing it twice in quick succession causes
SoX to exit.
Another option to stop processing early is to use an effect that
has a time period or sample count to determine the stopping point. The trim
effect is an example of this. Once all effects chains have stopped then SoX
will also stop.
Filenames can be simple file names, absolute or relative path names, or URLs
(input files only). Note that URL support requires that wget(1) is
available.
Note: Giving SoX an input or output filename that is the same as a
SoX effect-name will not work since SoX will treat it as an effect
specification. The only work-around to this is to avoid such filenames. This
is generally not difficult since most audio filenames have a filename
`extension', whilst effect-names do not.
The following special filenames may be used in certain circumstances in place of
a normal filename on the command line:
- -
- SoX can be used in simple pipeline operations by using the special
filename `-' which, if used as an input filename, will cause SoX will read
audio data from `standard input' (stdin), and which, if used as the output
filename, will cause SoX will send audio data to `standard output'
(stdout). Note that when using this option for the output file, and
sometimes when using it for an input file, the file-type (see -t
below) must also be given.
- "|program [options] ..."
- This can be used in place of an input filename to specify the the given
program's standard output (stdout) be used as an input file. Unlike
- (above), this can be used for several inputs to one SoX command.
For example, if `genw' generates mono WAV formatted signals to its
standard output, then the following command makes a stereo file from two
generated signals:
sox -M "|genw --imd -" "|genw --thd -" out.wav
For headerless (raw) audio, -t (and perhaps other format options)
will need to be given, preceding the input command.
- "wildcard-filename"
- Specifies that filename `globbing' (wild-card matching) should be
performed by SoX instead of by the shell. This allows a single set of file
options to be applied to a group of files. For example, if the current
directory contains three `vox' files, file1.vox, file2.vox, and file3.vox,
then
play --rate 6k *.vox
will be expanded by the `shell' (in most environments) to
play --rate 6k file1.vox file2.vox file3.vox
which will treat only the first vox file as having a sample rate of 6k. With
play --rate 6k "*.vox"
the given sample rate option will be applied to all three vox files.
- -p, --sox-pipe
- This can be used in place of an output filename to specify that the SoX
command should be used as in input pipe to another SoX command. For
example, the command:
play "|sox -n -p synth 2" "|sox -n -p synth 2 tremolo 10" stat
plays two `files' in succession, each with different effects.
-p is in fact an alias for `-t sox -'.
- -d, --default-device
- This can be used in place of an input or output filename to specify that
the default audio device (if one has been built into SoX) is to be used.
This is akin to invoking rec or play (as described
above).
- -n, --null
- This can be used in place of an input or output filename to specify that a
`null file' is to be used. Note that here, `null file' refers to a
SoX-specific mechanism and is not related to any operating-system
mechanism with a similar name.
Using a null file to input audio is equivalent to using a
normal audio file that contains an infinite amount of silence, and as
such is not generally useful unless used with an effect that specifies a
finite time length (such as trim or synth).
Using a null file to output audio amounts to discarding the
audio and is useful mainly with effects that produce information about
the audio instead of affecting it (such as noiseprof or
stat).
The sampling rate associated with a null file is by default
48 kHz, but, as with a normal file, this can be overridden if
desired using command-line format options (see below).
See soxformat(7) for a list and description of the supported file formats
and audio device drivers.
These options can be specified on the command line at any point before the first
effect name.
The SOX_OPTS environment variable can be used to provide
alternative default values for SoX's global options. For example:
SOX_OPTS="--buffer 20000 --play-rate-arg -hs --temp /mnt/temp"
Note that setting SOX_OPTS can potentially create unwanted changes in the
behaviour of scripts or other programs that invoke SoX. SOX_OPTS might best be
used for things (such as in the given example) that reflect the environment in
which SoX is being run. Enabling options such as --no-clobber as
default might be handled better using a shell alias since a shell alias will
not affect operation in scripts etc.
One way to ensure that a script cannot be affected by SOX_OPTS is
to clear SOX_OPTS at the start of the script, but this of course loses the
benefit of SOX_OPTS carrying some system-wide default options. An
alternative approach is to explicitly invoke SoX with default option values,
e.g.
SOX_OPTS="-V --no-clobber"
...
sox -V2 --clobber $input $output ...
Note that the way to set environment variables varies from system to system.
Here are some examples:
Unix bash:
export SOX_OPTS="-V --no-clobber"
Unix csh:
setenv SOX_OPTS "-V --no-clobber"
MS-DOS/MS-Windows:
set SOX_OPTS=-V --no-clobber
MS-Windows GUI: via Control Panel : System : Advanced : Environment Variables
Mac OS X GUI: Refer to Apple's Technical Q&A QA1067
document.
- --buffer BYTES, --input-buffer BYTES
- Set the size in bytes of the buffers used for processing audio (default
8192). --buffer applies to input, effects, and output processing;
--input-buffer applies only to input processing (for which it
overrides --buffer if both are given).
Be aware that large values for --buffer will cause SoX
to be become slow to respond to requests to terminate or to skip the
current input file.
- --clobber
- Don't prompt before overwriting an existing file with the same name as
that given for the output file. This is the default behaviour.
- --combine
concatenate|merge|mix|mix-power|multiply|sequence
- Select the input file combining method; for some of these, short options
are available: -m selects `mix', -M selects `merge', and
-T selects `multiply'.
See Input File Combining above for a description of the
different combining methods.
- -D, --no-dither
- Disable automatic dither - see `Dithering' above. An example of why this
might occasionally be useful is if a file has been converted from 16 to 24
bit with the intention of doing some processing on it, but in fact no
processing is needed after all and the original 16 bit file has been lost,
then, strictly speaking, no dither is needed if converting the file back
to 16 bit. See also the stats effect for how to determine the
actual bit depth of the audio within a file.
- --effects-file FILENAME
- Use FILENAME to obtain all effects and their arguments. The file is parsed
as if the values were specified on the command line. A new line can be
used in place of the special : marker to separate effect chains.
For convenience, such markers at the end of the file are normally ignored;
if you want to specify an empty last effects chain, use an explicit
: by itself on the last line of the file. This option causes any
effects specified on the command line to be discarded.
- -G, --guard
- Automatically invoke the gain effect to guard against clipping.
E.g.
sox -G infile -b 16 outfile rate 44100 dither -s
is shorthand for
sox infile -b 16 outfile gain -h rate 44100 gain -rh dither -s
See also -V, --norm, and the gain effect.
- -h, --help
- Show version number and usage information.
- --help-effect NAME
- Show usage information on the specified effect. The name all can be
used to show usage on all effects.
- --help-format NAME
- Show information about the specified file format. The name all can
be used to show information on all formats.
- --i, --info
- Only if given as the first parameter to sox, behave as
soxi(1).
- -m|-M
- Equivalent to --combine mix and --combine merge,
respectively.
- --magic
- If SoX has been built with the optional `libmagic' library then this
option can be given to enable its use in helping to detect audio file
types.
- --multi-threaded | --single-threaded
- By default, SoX is `single threaded'. If the --multi-threaded
option is given however then SoX will process audio channels for most
multi-channel effects in parallel on hyper-threading/multi-core
architectures. This may reduce processing time, though sometimes it may be
necessary to use this option in conjunction with a larger buffer size than
is the default to gain any benefit from multi-threaded processing (e.g.
131072; see --buffer above).
- --no-clobber
- Prompt before overwriting an existing file with the same name as that
given for the output file.
N.B. Unintentionally overwriting a file is easier than
you might think, for example, if you accidentally enter
sox file1 file2 effect1 effect2 ...
when what you really meant was
play file1 file2 effect1 effect2 ...
then, without this option, file2 will be overwritten. Hence, using this
option is recommended. SOX_OPTS (above), a `shell' alias, script, or batch
file may be an appropriate way of permanently enabling it.
- --norm[=dB-level]
- Automatically invoke the gain effect to guard against clipping and
to normalise the audio. E.g.
sox --norm infile -b 16 outfile rate 44100 dither -s
is shorthand for
sox infile -b 16 outfile gain -h rate 44100 gain -nh dither -s
Optionally, the audio can be normalized to a given level (usually) below 0
dBFS:
sox --norm=-3 infile outfile
See also -V, -G, and the gain effect.
- --play-rate-arg ARG
- Selects a quality option to be used when the `rate' effect is
automatically invoked whilst playing audio. This option is typically set
via the SOX_OPTS environment variable (see above).
- --plot gnuplot|octave|off
- If not set to off (the default if --plot is not given), run
in a mode that can be used, in conjunction with the gnuplot program or the
GNU Octave program, to assist with the selection and configuration of many
of the transfer-function based effects. For the first given effect that
supports the selected plotting program, SoX will output commands to plot
the effect's transfer function, and then exit without actually processing
any audio. E.g.
sox --plot octave input-file -n highpass 1320 > highpass.plt
octave highpass.plt
- -q, --no-show-progress
- Run in quiet mode when SoX wouldn't otherwise do so. This is the opposite
of the -S option.
- -R
- Run in `repeatable' mode. When this option is given, where applicable, SoX
will embed a fixed time-stamp in the output file (e.g. AIFF) and
will `seed' pseudo random number generators (e.g. dither) with a
fixed number, thus ensuring that successive SoX invocations with the same
inputs and the same parameters yield the same output.
- --replay-gain track|album|off
- Select whether or not to apply replay-gain adjustment to input files. The
default is off for sox and rec, album for
play where (at least) the first two input files are tagged with the
same Artist and Album names, and track for play
otherwise.
- -S, --show-progress
- Display input file format/header information, and processing progress as
input file(s) percentage complete, elapsed time, and remaining time (if
known; shown in brackets), and the number of samples written to the output
file. Also shown is a peak-level meter, and an indication if clipping has
occurred. The peak-level meter shows up to two channels and is calibrated
for digital audio as follows (right channel shown):
dB FSD |
Display |
dB FSD |
Display |
-25 |
- |
-11 |
==== |
-23 |
= |
-9 |
====- |
-21 |
=- |
-7 |
===== |
-19 |
== |
-5 |
=====- |
-17 |
==- |
-3 |
====== |
-15 |
=== |
-1 |
=====! |
-13 |
===- |
A three-second peak-held value of headroom in dBs will be
shown to the right of the meter if this is below 6dB.
This option is enabled by default when using SoX to play or
record audio.
- -T
- Equivalent to --combine multiply.
- --temp DIRECTORY
- Specify that any temporary files should be created in the given
DIRECTORY. This can be useful if there are permission or free-space
problems with the default location. In this case, using `--temp .'
(to use the current directory) is often a good solution.
- --version
- Show SoX's version number and exit.
- -V[level]
- Set verbosity. This is particularly useful for seeing how any automatic
effects have been invoked by SoX.
SoX displays messages on the console (stderr) according to the
following verbosity levels:
- 0
- No messages are shown at all; use the exit status to determine if an error
has occurred.
- 1
- Only error messages are shown. These are generated if SoX cannot complete
the requested commands.
- 2
- Warning messages are also shown. These are generated if SoX can complete
the requested commands, but not exactly according to the requested command
parameters, or if clipping occurs.
- 3
- Descriptions of SoX's processing phases are also shown. Useful for seeing
exactly how SoX is processing your audio.
- 4 and above
- Messages to help with debugging SoX are also shown.
- By default, the verbosity level is set to 2 (shows errors and warnings).
Each occurrence of the -V option increases the verbosity level by
1. Alternatively, the verbosity level can be set to an absolute number by
specifying it immediately after the -V, e.g. -V0 sets it to
0.
These options apply only to input files and may precede only input filenames on
the command line.
- --ignore-length
- Override an (incorrect) audio length given in an audio file's header. If
this option is given then SoX will keep reading audio until it reaches the
end of the input file.
- -v, --volume FACTOR
- Intended for use when combining multiple input files, this option adjusts
the volume of the file that follows it on the command line by a factor of
FACTOR. This allows it to be `balanced' w.r.t. the other input
files. This is a linear (amplitude) adjustment, so a number less than 1
decreases the volume and a number greater than 1 increases it. If a
negative number is given then in addition to the volume adjustment, the
audio signal will be inverted.
See also the norm, vol, and gain effects,
and see Input File Balancing above.
These options apply to the input or output file whose name they immediately
precede on the command line and are used mainly when working with headerless
file formats or when specifying a format for the output file that is different
to that of the input file.
- -b BITS, --bits BITS
- The number of bits (a.k.a. bit-depth or sometimes word-length) in each
encoded sample. Not applicable to complex encodings such as MP3 or GSM.
Not necessary with encodings that have a fixed number of bits, e.g.
A/μ-law, ADPCM.
For an input file, the most common use for this option is to
inform SoX of the number of bits per sample in a `raw' (`headerless')
audio file. For example
sox -r 16k -e signed -b 8 input.raw output.wav
converts a particular `raw' file to a self-describing `WAV' file.
For an output file, this option can be used (perhaps along
with -e) to set the output encoding size. By default (i.e. if
this option is not given), the output encoding size will (providing it
is supported by the output file type) be set to the input encoding size.
For example
sox input.cdda -b 24 output.wav
converts raw CD digital audio (16-bit, signed-integer) to a 24-bit
(signed-integer) `WAV' file.
- -c CHANNELS, --channels CHANNELS
- The number of audio channels in the audio file. This can be any number
greater than zero.
For an input file, the most common use for this option is to
inform SoX of the number of channels in a `raw' (`headerless') audio
file. Occasionally, it may be useful to use this option with a
`headered' file, in order to override the (presumably incorrect) value
in the header - note that this is only supported with certain file
types. Examples:
sox -r 48k -e float -b 32 -c 2 input.raw output.wav
converts a particular `raw' file to a self-describing `WAV' file.
play -c 1 music.wav
interprets the file data as belonging to a single channel regardless of what
is indicated in the file header. Note that if the file does in fact have
two channels, this will result in the file playing at half speed.
For an output file, this option provides a shorthand for
specifying that the channels effect should be invoked in order to
change (if necessary) the number of channels in the audio signal to the
number given. For example, the following two commands are
equivalent:
sox input.wav -c 1 output.wav bass -b 24
sox input.wav output.wav bass -b 24 channels 1
though the second form is more flexible as it allows the effects to be
ordered arbitrarily.
- -e ENCODING, --encoding ENCODING
- The audio encoding type. Sometimes needed with file-types that support
more than one encoding type. For example, with raw, WAV, or AU (but not,
for example, with MP3 or FLAC). The available encoding types are as
follows:
- signed-integer
- PCM data stored as signed (`two's complement') integers. Commonly used
with a 16 or 24 -bit encoding size. A value of 0 represents minimum signal
power.
- unsigned-integer
- PCM data stored as unsigned integers. Commonly used with an 8-bit encoding
size. A value of 0 represents maximum signal power.
- floating-point
- PCM data stored as IEEE 753 single precision (32-bit) or double precision
(64-bit) floating-point (`real') numbers. A value of 0 represents minimum
signal power.
- a-law
- International telephony standard for logarithmic encoding to 8 bits per
sample. It has a precision equivalent to roughly 13-bit PCM and is
sometimes encoded with reversed bit-ordering (see the -X
option).
- u-law, mu-law
- North American telephony standard for logarithmic encoding to 8 bits per
sample. A.k.a. μ-law. It has a precision equivalent to roughly
14-bit PCM and is sometimes encoded with reversed bit-ordering (see the
-X option).
- oki-adpcm
- OKI (a.k.a. VOX, Dialogic, or Intel) 4-bit ADPCM; it has a precision
equivalent to roughly 12-bit PCM. ADPCM is a form of audio compression
that has a good compromise between audio quality and encoding/decoding
speed.
- ima-adpcm
- IMA (a.k.a. DVI) 4-bit ADPCM; it has a precision equivalent to roughly
13-bit PCM.
- ms-adpcm
- Microsoft 4-bit ADPCM; it has a precision equivalent to roughly 14-bit
PCM.
- gsm-full-rate
- GSM is currently used for the vast majority of the world's digital
wireless telephone calls. It utilises several audio formats with different
bit-rates and associated speech quality. SoX has support for GSM's
original 13kbps `Full Rate' audio format. It is usually CPU-intensive to
work with GSM audio.
-
- Encoding names can be abbreviated where this would not be ambiguous; e.g.
`unsigned-integer' can be given as `un', but not `u' (ambiguous with
`u-law').
For an input file, the most common use for this option is to
inform SoX of the encoding of a `raw' (`headerless') audio file (see the
examples in -b and -c above).
For an output file, this option can be used (perhaps along
with -b) to set the output encoding type For example
sox input.cdda -e float output1.wav
sox input.cdda -b 64 -e float output2.wav
convert raw CD digital audio (16-bit, signed-integer) to floating-point
`WAV' files (single & double precision respectively).
By default (i.e. if this option is not given), the output
encoding type will (providing it is supported by the output file type)
be set to the input encoding type.
- --no-glob
- Specifies that filename `globbing' (wild-card matching) should not be
performed by SoX on the following filename. For example, if the current
directory contains the two files `five-seconds.wav' and `five*.wav', then
play --no-glob "five*.wav"
can be used to play just the single file `five*.wav'.
- -r, --rate RATE[k]
- Gives the sample rate in Hz (or kHz if appended with `k') of the file.
For an input file, the most common use for this option is to
inform SoX of the sample rate of a `raw' (`headerless') audio file (see
the examples in -b and -c above). Occasionally it may be
useful to use this option with a `headered' file, in order to override
the (presumably incorrect) value in the header - note that this is only
supported with certain file types. For example, if audio was recorded
with a sample-rate of say 48k from a source that played back a little,
say 1.5%, too slowly, then
sox -r 48720 input.wav output.wav
effectively corrects the speed by changing only the file header (but see
also the speed effect for the more usual solution to this problem).
For an output file, this option provides a shorthand for
specifying that the rate effect should be invoked in order to
change (if necessary) the sample rate of the audio signal to the given
value. For example, the following two commands are equivalent:
sox input.wav -r 48k output.wav bass -b 24
sox input.wav output.wav bass -b 24 rate 48k
though the second form is more flexible as it allows rate options to
be given, and allows the effects to be ordered arbitrarily.
- -t, --type FILE-TYPE
- Gives the type of the audio file. For both input and output files, this
option is commonly used to inform SoX of the type a `headerless' audio
file (e.g. raw, mp3) where the actual/desired type cannot be determined
from a given filename extension. For example:
another-command | sox -t mp3 - output.wav
sox input.wav -t raw output.bin
It can also be used to override the type implied by an input filename
extension, but if overriding with a type that has a header, SoX will exit
with an appropriate error message if such a header is not actually
present.
See soxformat(7) for a list of supported file
types.
-L, --endian little
-B, --endian big
-x, --endian swap
-
- These options specify whether the byte-order of the audio data is,
respectively, `little endian', `big endian', or the opposite to that of
the system on which SoX is being used. Endianness applies only to data
encoded as floating-point, or as signed or unsigned integers of 16 or more
bits. It is often necessary to specify one of these options for headerless
files, and sometimes necessary for (otherwise) self-describing files. A
given endian-setting option may be ignored for an input file whose header
contains a specific endianness identifier, or for an output file that is
actually an audio device.
N.B. Unlike other format characteristics, the
endianness (byte, nibble, & bit ordering) of the input file is not
automatically used for the output file; so, for example, when the
following is run on a little-endian system:
sox -B audio.s16 trimmed.s16 trim 2
trimmed.s16 will be created as little-endian;
sox -B audio.s16 -B trimmed.s16 trim 2
must be used to preserve big-endianness in the output file.
The -V option can be used to check the selected
orderings.
- -N, --reverse-nibbles
- Specifies that the nibble ordering (i.e. the 2 halves of a byte) of the
samples should be reversed; sometimes useful with ADPCM-based formats.
N.B. See also N.B. in section on -x above.
- -X, --reverse-bits
- Specifies that the bit ordering of the samples should be reversed;
sometimes useful with a few (mostly headerless) formats.
N.B. See also N.B. in section on -x above.
These options apply only to the output file and may precede only the output
filename on the command line.
- --add-comment TEXT
- Append a comment in the output file header (where applicable).
- --comment TEXT
- Specify the comment text to store in the output file header (where
applicable).
SoX will provide a default comment if this option (or
--comment-file) is not given. To specify that no comment should
be stored in the output file, use --comment "" .
- --comment-file FILENAME
- Specify a file containing the comment text to store in the output file
header (where applicable).
- -C, --compression FACTOR
- The compression factor for variably compressing output file formats. If
this option is not given then a default compression factor will apply. The
compression factor is interpreted differently for different compressing
file formats. See the description of the file formats that use this option
in soxformat(7) for more information.
In addition to converting, playing and recording audio files, SoX can be used to
invoke a number of audio `effects'. Multiple effects may be applied by
specifying them one after another at the end of the SoX command line, forming
an `effects chain'. Note that applying multiple effects in real-time (i.e.
when playing audio) is likely to require a high performance computer. Stopping
other applications may alleviate performance issues should they occur.
Some of the SoX effects are primarily intended to be applied to a
single instrument or `voice'. To facilitate this, the remix effect
and the global SoX option -M can be used to isolate then recombine
tracks from a multi-track recording.
A single effects chain is made up of one or more effects. Audio from the input
runs through the chain until either the end of the input file is reached or an
effect in the chain requests to terminate the chain.
SoX supports running multiple effects chains over the input audio.
In this case, when one chain indicates it is done processing audio, the
audio data is then sent through the next effects chain. This continues until
either no more effects chains exist or the input has reached the end of the
file.
An effects chain is terminated by placing a : (colon) after
an effect. Any following effects are a part of a new effects chain.
It is important to place the effect that will stop the chain as
the first effect in the chain. This is because any samples that are buffered
by effects to the left of the terminating effect will be discarded. The
amount of samples discarded is related to the --buffer option and it
should be kept small, relative to the sample rate, if the terminating effect
cannot be first. Further information on stopping effects can be found in the
Stopping SoX section.
There are a few pseudo-effects that aid using multiple effects
chains. These include newfile which will start writing to a new
output file before moving to the next effects chain and restart which
will move back to the first effects chain. Pseudo-effects must be specified
as the first effect in a chain and as the only effect in a chain (they must
have a : before and after they are specified).
The following is an example of multiple effects chains. It will
split the input file into multiple files of 30 seconds in length. Each
output filename will have unique number in its name as documented in the
Output Files section.
sox infile.wav output.wav trim 0 30 : newfile : restart
In the descriptions that follow, brackets [ ] are used to denote parameters that
are optional, braces { } to denote those that are both optional and
repeatable, and angle brackets < > to denote those that are repeatable
but not optional. Where applicable, default values for optional parameters are
shown in parenthesis ( ).
The following parameters are used with, and have the same meaning
for, several effects:
- center[k]
- See frequency.
- frequency[k]
- A frequency in Hz, or, if appended with `k', kHz.
- gain
- A power gain in dB. Zero gives no gain; less than zero gives an
attenuation.
- position
- A position within the audio stream; the syntax is
[=|+|-]timespec, where timespec is a
time specification (see below). The optional first character indicates
whether the timespec is to be interpreted relative to the start
(=) or end (-) of audio, or to the previous position
if the effect accepts multiple position arguments (+). The audio
length must be known for end-relative locations to work; some effects do
accept -0 for end-of-audio, though, even if the length is unknown.
Which of =, +, - is the default depends on the effect
and is shown in its syntax as, e.g., position(+).
Examples: =2:00 (two minutes into the audio stream),
-100s (one hundred samples before the end of audio),
+0:12+10s (twelve seconds and ten samples after the previous
position), -0.5+1s (one sample less than half a second before the
end of audio).
- width[h|k|o|q]
- Used to specify the band-width of a filter. A number of different methods
to specify the width are available (though not all for every effect). One
of the characters shown may be appended to select the desired method as
follows:
|
Method |
Notes |
h |
Hz |
|
k |
kHz |
|
o |
Octaves |
|
q |
Q-factor |
See [2] |
For each effect that uses this parameter, the default method
(i.e. if no character is appended) is the one that it listed first in
the first line of the effect's description.
Most effects that expect an audio position or duration in a
parameter, i.e. a time specification, accept either of the following
two forms:
- [[hours:]minutes:]seconds[.frac][t]
- A specification of `1:30.5' corresponds to one minute, thirty and ½
seconds. The t suffix is entirely optional (however, see the
silence effect for an exception). Note that the component values do
not have to be normalized; e.g., `1:23:45', `83:45', `79:0285',
`1:0:1425', `1::1425' and `5025' all are legal and equivalent to each
other.
- sampless
- Specifies the number of samples directly, as in `8000s'. For large sample
counts, e notation is supported: `1.7e6s' is the same as
`1700000s'.
Time specifications can also be chained with + or -
into a new time specification where the right part is added to or subtracted
from the left, respectively: `3:00-200s' means two hundred samples less than
three minutes.
To see if SoX has support for an optional effect, enter sox
-h and look for its name under the list: `EFFECTS'.
Note: a categorised list of the effects can be found in the accompanying
`README' file.
- allpass frequency[k]
width[h|k|o|q]
- Apply a two-pole all-pass filter with central frequency (in Hz)
frequency, and filter-width width. An all-pass filter
changes the audio's frequency to phase relationship without changing its
frequency to amplitude relationship. The filter is described in detail in
[1].
This effect supports the --plot global option.
- band [-n] center[k]
[width[h|k|o|q]]
- Apply a band-pass filter. The frequency response drops logarithmically
around the center frequency. The width parameter gives the
slope of the drop. The frequencies at center + width and
center - width will be half of their original amplitudes.
band defaults to a mode oriented to pitched audio, i.e. voice,
singing, or instrumental music. The -n (for noise) option uses the
alternate mode for un-pitched audio (e.g. percussion). Warning:
-n introduces a power-gain of about 11dB in the filter, so beware
of output clipping. band introduces noise in the shape of the
filter, i.e. peaking at the center frequency and settling around
it.
This effect supports the --plot global option.
See also sinc for a bandpass filter with steeper
shoulders.
- bandpass|bandreject [-c]
frequency[k]
width[h|k|o|q]
- Apply a two-pole Butterworth band-pass or band-reject filter with central
frequency frequency, and (3dB-point) band-width width. The
-c option applies only to bandpass and selects a constant
skirt gain (peak gain = Q) instead of the default: constant 0dB peak gain.
The filters roll off at 6dB per octave (20dB per decade) and are described
in detail in [1].
These effects support the --plot global option.
See also sinc for a bandpass filter with steeper
shoulders.
- bandreject frequency[k]
width[h|k|o|q]
- Apply a band-reject filter. See the description of the bandpass
effect for details.
- bass|treble gain [frequency[k]
[width[s|h|k|o|q]]]
- Boost or cut the bass (lower) or treble (upper) frequencies of the audio
using a two-pole shelving filter with a response similar to that of a
standard hi-fi's tone-controls. This is also known as shelving
equalisation (EQ).
gain gives the gain at 0 Hz (for bass),
or whichever is the lower of ∼22 kHz and the Nyquist
frequency (for treble). Its useful range is about -20 (for a
large cut) to +20 (for a large boost). Beware of Clipping when
using a positive gain.
If desired, the filter can be fine-tuned using the following
optional parameters:
frequency sets the filter's central frequency and so
can be used to extend or reduce the frequency range to be boosted or
cut. The default value is 100 Hz (for bass) or
3 kHz (for treble).
width determines how steep is the filter's shelf
transition. In addition to the common width specification methods
described above, `slope' (the default, or if appended with `s')
may be used. The useful range of `slope' is about 0.3, for a gentle
slope, to 1 (the maximum), for a steep slope; the default value is
0.5.
The filters are described in detail in [1].
These effects support the --plot global option.
See also equalizer for a peaking equalisation
effect.
- bend [-f frame-rate(25)] [-o
over-sample(16)] {
start-position(+),cents,end-position(+)
}
- Changes pitch by specified amounts at specified times. Each given triple:
start-position,cents,end-position
specifies one bend. cents is the number of cents (100 cents = 1
semitone) by which to bend the pitch. The other values specify the points
in time at which to start and end bending the pitch, respectively.
The pitch-bending algorithm utilises the Discrete Fourier
Transform (DFT) at a particular frame rate and over-sampling rate. The
-f and -o parameters may be used to adjust these
parameters and thus control the smoothness of the changes in pitch.
For example, an initial tone is generated, then bent three
times, yielding four different notes in total:
play -n synth 2.5 sin 667 gain 1 \
bend .35,180,.25 .15,740,.53 0,-520,.3
Here, the first bend runs from 0.35 to 0.6, and the second one from 0.75 to
1.28 seconds. Note that the clipping that is produced in this example is
deliberate; to remove it, use gain -5 in place of
gain 1.
See also pitch.
- biquad b0 b1 b2 a0 a1 a2
- Apply a biquad IIR filter with the given coefficients. Where b* and a* are
the numerator and denominator coefficients respectively.
See http://en.wikipedia.org/wiki/Digital_biquad_filter (where
a0 = 1).
This effect supports the --plot global option.
- channels CHANNELS
- Invoke a simple algorithm to change the number of channels in the audio
signal to the given number CHANNELS: mixing if decreasing the
number of channels or duplicating if increasing the number of channels.
The channels effect is invoked automatically if SoX's
-c option specifies a number of channels that is different to
that of the input file(s). Alternatively, if this effect is given
explicitly, then SoX's -c option need not be given. For example,
the following two commands are equivalent:
sox input.wav -c 1 output.wav bass -b 24
sox input.wav output.wav bass -b 24 channels 1
though the second form is more flexible as it allows the effects to be
ordered arbitrarily.
See also remix for an effect that allows channels to be
mixed/selected arbitrarily.
- chorus gain-in gain-out <delay decay speed depth
-s|-t>
- Add a chorus effect to the audio. This can make a single vocal sound like
a chorus, but can also be applied to instrumentation.
Chorus resembles an echo effect with a short delay, but
whereas with echo the delay is constant, with chorus, it is varied using
sinusoidal or triangular modulation. The modulation depth defines the
range the modulated delay is played before or after the delay. Hence the
delayed sound will sound slower or faster, that is the delayed sound
tuned around the original one, like in a chorus where some vocals are
slightly off key. See [3] for more discussion of the chorus effect.
Each four-tuple parameter delay/decay/speed/depth gives the
delay in milliseconds and the decay (relative to gain-in) with a
modulation speed in Hz using depth in milliseconds. The modulation is
either sinusoidal (-s) or triangular (-t). Gain-out is the
volume of the output.
A typical delay is around 40ms to 60ms; the modulation speed
is best near 0.25Hz and the modulation depth around 2ms. For example, a
single delay:
play guitar1.wav chorus 0.7 0.9 55 0.4 0.25 2 -t
Two delays of the original samples:
play guitar1.wav chorus 0.6 0.9 50 0.4 0.25 2 -t \
60 0.32 0.4 1.3 -s
A fuller sounding chorus (with three additional delays):
play guitar1.wav chorus 0.5 0.9 50 0.4 0.25 2 -t \
60 0.32 0.4 2.3 -t 40 0.3 0.3 1.3 -s
- compand
attack1,decay1{,attack2,decay2}
- [soft-knee-dB:]in-dB1[,out-dB1]{,in-dB2,out-dB2}
[gain [initial-volume-dB [delay]]]
Compand (compress or expand) the dynamic range of the
audio.
The attack and decay parameters (in seconds)
determine the time over which the instantaneous level of the input
signal is averaged to determine its volume; attacks refer to increases
in volume and decays refer to decreases. For most situations, the attack
time (response to the music getting louder) should be shorter than the
decay time because the human ear is more sensitive to sudden loud music
than sudden soft music. Where more than one pair of attack/decay
parameters are specified, each input channel is companded separately and
the number of pairs must agree with the number of input channels.
Typical values are 0.3,0.8 seconds.
The second parameter is a list of points on the compander's
transfer function specified in dB relative to the maximum possible
signal amplitude. The input values must be in a strictly increasing
order but the transfer function does not have to be monotonically
rising. If omitted, the value of out-dB1 defaults to the same
value as in-dB1; levels below in-dB1 are not companded
(but may have gain applied to them). The point 0,0 is assumed but
may be overridden (by 0,out-dBn). If the list is preceded
by a soft-knee-dB value, then the points at where adjacent line
segments on the transfer function meet will be rounded by the amount
given. Typical values for the transfer function are
6:-70,-60,-20.
The third (optional) parameter is an additional gain in dB to
be applied at all points on the transfer function and allows easy
adjustment of the overall gain.
The fourth (optional) parameter is an initial level to be
assumed for each channel when companding starts. This permits the user
to supply a nominal level initially, so that, for example, a very large
gain is not applied to initial signal levels before the companding
action has begun to operate: it is quite probable that in such an event,
the output would be severely clipped while the compander gain properly
adjusts itself. A typical value (for audio which is initially quiet) is
-90 dB.
The fifth (optional) parameter is a delay in seconds. The
input signal is analysed immediately to control the compander, but it is
delayed before being fed to the volume adjuster. Specifying a delay
approximately equal to the attack/decay times allows the compander to
effectively operate in a `predictive' rather than a reactive mode. A
typical value is 0.2 seconds.
The following example might be used to make a piece of music
with both quiet and loud passages suitable for listening to in a noisy
environment such as a moving vehicle:
sox asz.wav asz-car.wav compand 0.3,1 6:-70,-60,-20 -5 -90 0.2
The transfer function (`6:-70,...') says that very soft sounds (below -70dB)
will remain unchanged. This will stop the compander from boosting the
volume on `silent' passages such as between movements. However, sounds in
the range -60dB to 0dB (maximum volume) will be boosted so that the 60dB
dynamic range of the original music will be compressed 3-to-1 into a 20dB
range, which is wide enough to enjoy the music but narrow enough to get
around the road noise. The `6:' selects 6dB soft-knee companding. The -5
(dB) output gain is needed to avoid clipping (the number is inexact, and
was derived by experimentation). The -90 (dB) for the initial volume will
work fine for a clip that starts with near silence, and the delay of 0.2
(seconds) has the effect of causing the compander to react a bit more
quickly to sudden volume changes.
In the next example, compand is being used as a noise-gate for
when the noise is at a lower level than the signal:
play infile compand .1,.2 -inf,-50.1,-inf,-50,-50 0 -90 .1
Here is another noise-gate, this time for when the noise is at a higher
level than the signal (making it, in some ways, similar to squelch):
play infile compand .1,.1 -45.1,-45,-inf,0,-inf 45 -90 .1
This effect supports the --plot global option (for the transfer
function).
See also mcompand for a multiple-band companding
effect.
- contrast [enhancement-amount(75)]
- Comparable with compression, this effect modifies an audio signal to make
it sound louder. enhancement-amount controls the amount of the
enhancement and is a number in the range 0-100. Note that
enhancement-amount = 0 still gives a significant contrast
enhancement.
See also the compand and mcompand effects.
- dcshift shift [limitergain]
- Apply a DC shift to the audio. This can be useful to remove a DC offset
(caused perhaps by a hardware problem in the recording chain) from the
audio. The effect of a DC offset is reduced headroom and hence volume. The
stat or stats effect can be used to determine if a signal
has a DC offset.
The given dcshift value is a floating point number in
the range of ±2 that indicates the amount to shift the audio
(which is in the range of ±1).
An optional limitergain can be specified as well. It
should have a value much less than 1 (e.g. 0.05 or 0.02) and is used
only on peaks to prevent clipping.
An alternative approach to removing a DC offset (albeit with a
short delay) is to use the highpass filter effect at a frequency
of say 10Hz, as illustrated in the following example:
sox -n dc.wav synth 5 sin %0 50
sox dc.wav fixed.wav highpass 10
- deemph
- Apply Compact Disc (IEC 60908) de-emphasis (a treble attenuation shelving
filter).
Pre-emphasis was applied in the mastering of some CDs issued
in the early 1980s. These included many classical music albums, as well
as now sought-after issues of albums by The Beatles, Pink Floyd and
others. Pre-emphasis should be removed at playback time by a de-emphasis
filter in the playback device. However, not all modern CD players have
this filter, and very few PC CD drives have it; playing pre-emphasised
audio without the correct de-emphasis filter results in audio that
sounds harsh and is far from what its creators intended.
With the deemph effect, it is possible to apply the
necessary de-emphasis to audio that has been extracted from a
pre-emphasised CD, and then either burn the de-emphasised audio to a new
CD (which will then play correctly on any CD player), or simply play the
correctly de-emphasised audio files on the PC. For example:
sox track1.wav track1-deemph.wav deemph
and then burn track1-deemph.wav to CD, or
play track1-deemph.wav
or simply
play track1.wav deemph
The de-emphasis filter is implemented as a biquad and requires the input
audio sample rate to be either 44.1kHz or 48kHz. Maximum deviation from
the ideal response is only 0.06dB (up to 20kHz).
This effect supports the --plot global option.
See also the bass and treble shelving
equalisation effects.
- delay {position(=)}
- Delay one or more audio channels such that they start at the given
position. For example, delay 1.5 +1 3000s delays the first
channel by 1.5 seconds, the second channel by 2.5 seconds (one second more
than the previous channel), the third channel by 3000 samples, and leaves
any other channels that may be present un-delayed. The following (one
long) command plays a chime sound:
play -n synth -j 3 sin %3 sin %-2 sin %-5 sin %-9 \
sin %-14 sin %-21 fade h .01 2 1.5 delay \
1.3 1 .76 .54 .27 remix - fade h 0 2.7 2.5 norm -1
and this plays a guitar chord:
play -n synth pl G2 pl B2 pl D3 pl G3 pl D4 pl G4 \
delay 0 .05 .1 .15 .2 .25 remix - fade 0 4 .1 norm -1
- dither [-S|-s|-f filter] [-a]
[-p precision]
- Apply dithering to the audio. Dithering deliberately adds a small amount
of noise to the signal in order to mask audible quantization effects that
can occur if the output sample size is less than 24 bits. With no options,
this effect will add triangular (TPDF) white noise. Noise-shaping (only
for certain sample rates) can be selected with -s. With the
-f option, it is possible to select a particular noise-shaping
filter from the following list: lipshitz, f-weighted, modified-e-weighted,
improved-e-weighted, gesemann, shibata, low-shibata, high-shibata. Note
that most filter types are available only with 44100Hz sample rate. The
filter types are distinguished by the following properties: audibility of
noise, level of (inaudible, but in some circumstances, otherwise
problematic) shaped high frequency noise, and processing speed.
See http://sox.sourceforge.net/SoX/NoiseShaping for graphs of the different
noise-shaping curves.
The -S option selects a slightly `sloped' TPDF, biased
towards higher frequencies. It can be used at any sampling rate but
below ≈22k, plain TPDF is probably better, and above ≈
37k, noise-shaping (if available) is probably better.
The -a option enables a mode where dithering (and
noise-shaping if applicable) are automatically enabled only when needed.
The most likely use for this is when applying fade in or out to an
already dithered file, so that the redithering applies only to the faded
portions. However, auto dithering is not fool-proof, so the fades should
be carefully checked for any noise modulation; if this occurs, then
either re-dither the whole file, or use trim, fade, and
concatencate.
The -p option allows overriding the target
precision.
If the SoX global option -R option is not given, then
the pseudo-random number generator used to generate the white noise will
be `reseeded', i.e. the generated noise will be different between
invocations.
This effect should not be followed by any other effect that
affects the audio.
See also the `Dithering' section above.
- downsample [factor(2)]
- Downsample the signal by an integer factor: Only the first out of each
factor samples is retained, the others are discarded.
No decimation filter is applied. If the input is not a
properly bandlimited baseband signal, aliasing will occur. This may be
desirable, e.g., for frequency translation.
For a general resampling effect with anti-aliasing, see
rate. See also upsample.
- earwax
- Makes audio easier to listen to on headphones. Adds `cues' to 44.1kHz
stereo (i.e. audio CD format) audio so that when listened to on headphones
the stereo image is moved from inside your head (standard for headphones)
to outside and in front of the listener (standard for speakers).
- echo gain-in gain-out <delay decay>
- Add echoing to the audio. Echoes are reflected sound and can occur
naturally amongst mountains (and sometimes large buildings) when talking
or shouting; digital echo effects emulate this behaviour and are often
used to help fill out the sound of a single instrument or vocal. The time
difference between the original signal and the reflection is the `delay'
(time), and the loudness of the reflected signal is the `decay'. Multiple
echoes can have different delays and decays.
Each given delay decay pair gives the delay in
milliseconds and the decay (relative to gain-in) of that echo. Gain-out
is the volume of the output. For example: This will make it sound as if
there are twice as many instruments as are actually playing:
play lead.aiff echo 0.8 0.88 60 0.4
If the delay is very short, then it sound like a (metallic) robot playing
music:
play lead.aiff echo 0.8 0.88 6 0.4
A longer delay will sound like an open air concert in the mountains:
play lead.aiff echo 0.8 0.9 1000 0.3
One mountain more, and:
play lead.aiff echo 0.8 0.9 1000 0.3 1800 0.25
- echos gain-in gain-out <delay decay>
- Add a sequence of echoes to the audio. Each delay decay pair gives
the delay in milliseconds and the decay (relative to gain-in) of that
echo. Gain-out is the volume of the output.
Like the echo effect, echos stand for `ECHO in Sequel', that
is the first echos takes the input, the second the input and the first
echos, the third the input and the first and the second echos, ... and
so on. Care should be taken using many echos; a single echos has the
same effect as a single echo.
The sample will be bounced twice in symmetric echos:
play lead.aiff echos 0.8 0.7 700 0.25 700 0.3
The sample will be bounced twice in asymmetric echos:
play lead.aiff echos 0.8 0.7 700 0.25 900 0.3
The sample will sound as if played in a garage:
play lead.aiff echos 0.8 0.7 40 0.25 63 0.3
- equalizer frequency[k]
width[q|o|h|k] gain
- Apply a two-pole peaking equalisation (EQ) filter. With this filter, the
signal-level at and around a selected frequency can be increased or
decreased, whilst (unlike band-pass and band-reject filters) that at all
other frequencies is unchanged.
frequency gives the filter's central frequency in Hz,
width, the band-width, and gain the required gain or
attenuation in dB. Beware of Clipping when using a positive
gain.
In order to produce complex equalisation curves, this effect
can be given several times, each with a different central frequency.
The filter is described in detail in [1].
This effect supports the --plot global option.
See also bass and treble for shelving
equalisation effects.
- fade [type] fade-in-length [stop-position(=)
[fade-out-length]]
- Apply a fade effect to the beginning, end, or both of the audio.
An optional type can be specified to select the shape
of the fade curve: q for quarter of a sine wave, h for
half a sine wave, t for linear (`triangular') slope, l for
logarithmic, and p for inverted parabola. The default is
logarithmic.
A fade-in starts from the first sample and ramps the signal
level from 0 to full volume over the time given as
fade-in-length. Specify 0 if no fade-in is wanted.
For fade-outs, the audio will be truncated at
stop-position and the signal level will be ramped from full
volume down to 0 over an interval of fade-out-length before the
stop-position. If fade-out-length is not specified, it
defaults to the same value as fade-in-length. No fade-out is
performed if stop-position is not specified. If the audio length
can be determined from the input file header and any previous effects,
then -0 (or, for historical reasons, 0) may be specified
for stop-position to indicate the usual case of a fade-out that
ends at the end of the input audio stream.
Any time specification may be used for fade-in-length
and fade-out-length.
See also the splice effect.
- fir [coefs-file|coefs]
- Use SoX's FFT convolution engine with given FIR filter coefficients. If a
single argument is given then this is treated as the name of a file
containing the filter coefficients (white-space separated; may contain `#'
comments). If the given filename is `-', or if no argument is given, then
the coefficients are read from the `standard input' (stdin); otherwise,
coefficients may be given on the command line. Examples:
sox infile outfile fir 0.0195 -0.082 0.234 0.891 -0.145 0.043
sox infile outfile fir coefs.txt
with coefs.txt containing
# HP filter
# freq=10000
1.2311233052619888e-01
-4.4777096106211783e-01
5.1031563346705155e-01
-6.6502926320995331e-02
...
This effect supports the --plot global option.
- flanger [delay depth regen width speed shape phase
interp]
- Apply a flanging effect to the audio. See [3] for a detailed description
of flanging.
All parameters are optional (right to left).
|
Range |
Default |
Description |
delay |
0 - 30 |
0 |
Base delay in milliseconds. |
depth |
0 - 10 |
2 |
Added swept delay in milliseconds. |
regen |
-95 - 95 |
0 |
Percentage regeneration (delayed signal feedback). |
width |
0 - 100 |
71 |
Percentage of delayed signal mixed with original. |
speed |
0.1 - 10 |
0.5 |
Sweeps per second (Hz). |
shape |
|
sin |
Swept wave shape: sine|triangle. |
phase |
0 - 100 |
25 |
Swept wave percentage phase-shift for multi-channel (e.g. stereo)
flange; 0 = 100 = same phase on each channel. |
interp |
|
lin |
Digital delay-line interpolation:
linear|quadratic. |
- gain [-e|-B|-b|-r] [-n]
[-l|-h] [gain-dB]
- Apply amplification or attenuation to the audio signal, or, in some cases,
to some of its channels. Note that use of any of -e, -B,
-b, -r, or -n requires temporary file space to store
the audio to be processed, so may be unsuitable for use with `streamed'
audio.
Without other options, gain-dB is used to adjust the
signal power level by the given number of dB: positive amplifies (beware
of Clipping), negative attenuates. With other options, the
gain-dB amplification or attenuation is (logically) applied after
the processing due to those options.
Given the -e option, the levels of the audio channels
of a multi-channel file are `equalised', i.e. gain is applied to all
channels other than that with the highest peak level, such that all
channels attain the same peak level (but, without also giving -n,
the audio is not `normalised').
The -B (balance) option is similar to -e, but
with -B, the RMS level is used instead of the peak level.
-B might be used to correct stereo imbalance caused by an
imperfect record turntable cartridge. Note that unlike -e,
-B might cause some clipping.
-b is similar to -B but has clipping protection,
i.e. if necessary to prevent clipping whilst balancing, attenuation is
applied to all channels. Note, however, that in conjunction with
-n, -B and -b are synonymous.
The -r option is used in conjunction with a prior
invocation of gain with the -h option - see below for
details.
The -n option normalises the audio to 0dB FSD; it is
often used in conjunction with a negative gain-dB to the effect
that the audio is normalised to a given level below 0dB. For
example,
sox infile outfile gain -n
normalises to 0dB, and
sox infile outfile gain -n -3
normalises to -3dB.
The -l option invokes a simple limiter, e.g.
sox infile outfile gain -l 6
will apply 6dB of gain but never clip. Note that limiting more than a few
dBs more than occasionally (in a piece of audio) is not recommended as it
can cause audible distortion. See the compand effect for a more
capable limiter.
The -h option is used to apply gain to provide
head-room for subsequent processing. For example, with
sox infile outfile gain -h bass +6
6dB of attenuation will be applied prior to the bass boosting effect thus
ensuring that it will not clip. Of course, with bass, it is obvious how
much headroom will be needed, but with other effects (e.g. rate, dither)
it is not always as clear. Another advantage of using gain -h
rather than an explicit attenuation, is that if the headroom is not used
by subsequent effects, it can be reclaimed with gain -r, for
example:
sox infile outfile gain -h bass +6 rate 44100 gain -r
The above effects chain guarantees never to clip nor amplify; it attenuates
if necessary to prevent clipping, but by only as much as is needed to do
so.
Output formatting (dithering and bit-depth reduction) also
requires headroom (which cannot be `reclaimed'), e.g.
sox infile outfile gain -h bass +6 rate 44100 gain -rh dither
Here, the second gain invocation, reclaims as much of the headroom as
it can from the preceding effects, but retains as much headroom as is
needed for subsequent processing. The SoX global option -G can be
given to automatically invoke gain -h and gain -r.
See also the norm and vol effects.
- highpass|lowpass [-1|-2]
frequency[k] [width[q|o|h|k]]
- Apply a high-pass or low-pass filter with 3dB point frequency. The
filter can be either single-pole (with -1), or double-pole (the
default, or with -2). width applies only to double-pole
filters; the default is Q = 0.707 and gives a Butterworth response. The
filters roll off at 6dB per pole per octave (20dB per pole per decade).
The double-pole filters are described in detail in [1].
These effects support the --plot global option.
See also sinc for filters with a steeper roll-off.
- hilbert [-n taps]
- Apply an odd-tap Hilbert transform filter, phase-shifting the signal by 90
degrees.
This is used in many matrix coding schemes and for analytic
signal generation. The process is often written as a multiplication by
i (or j), the imaginary unit.
An odd-tap Hilbert transform filter has a bandpass
characteristic, attenuating the lowest and highest frequencies. Its
bandwidth can be controlled by the number of filter taps, which can be
specified with -n. By default, the number of taps is chosen for a
cutoff frequency of about 75 Hz.
This effect supports the --plot global option.
- ladspa [-l|-r] module [plugin]
[argument ...]
- Apply a LADSPA [5] (Linux Audio Developer's Simple Plugin API) plugin.
Despite the name, LADSPA is not Linux-specific, and a wide range of
effects is available as LADSPA plugins, such as cmt [6] (the Computer
Music Toolkit) and Steve Harris's plugin collection [7]. The first
argument is the plugin module, the second the name of the plugin (a module
can contain more than one plugin), and any other arguments are for the
control ports of the plugin. Missing arguments are supplied by default
values if possible.
Normally, the number of input ports of the plugin must match
the number of input channels, and the number of output ports determines
the output channel count. However, the -r (replicate) option
allows cloning a mono plugin to handle multi-channel input.
Some plugins introduce latency which SoX may optionally
compensate for. The -l (latency compensation) option
automatically compensates for latency as reported by the plugin via an
output control port named "latency".
If found, the environment variable LADSPA_PATH will be used as
search path for plugins.
- loudness [gain [reference]]
- Loudness control - similar to the gain effect, but provides
equalisation for the human auditory system. See
http://en.wikipedia.org/wiki/Loudness for a detailed description of
loudness. The gain is adjusted by the given gain parameter (usually
negative) and the signal equalised according to ISO 226 w.r.t. a reference
level of 65dB, though an alternative reference level may be given
if the original audio has been equalised for some other optimal level. A
default gain of -10dB is used if a gain value is not given.
See also the gain effect.
- lowpass [-1|-2] frequency[k]
[width[q|o|h|k]]
- Apply a low-pass filter. See the description of the highpass effect
for details.
- mcompand
"attack1,decay1{,attack2,decay2}
- [soft-knee-dB:]in-dB1[,out-dB1]{,in-dB2,out-dB2}
[gain [initial-volume-dB [delay]]]"
{crossover-freq[k] "attack1,..."}
The multi-band compander is similar to the single-band
compander but the audio is first divided into bands using Linkwitz-Riley
cross-over filters and a separately specifiable compander run on each
band. See the compand effect for the definition of its
parameters. Compand parameters are specified between double quotes and
the crossover frequency for that band is given by crossover-freq;
these can be repeated to create multiple bands.
For example, the following (one long) command shows how
multi-band companding is typically used in FM radio:
play track1.wav gain -3 sinc 8000- 29 100 mcompand \
"0.005,0.1 -47,-40,-34,-34,-17,-33" 100 \
"0.003,0.05 -47,-40,-34,-34,-17,-33" 400 \
"0.000625,0.0125 -47,-40,-34,-34,-15,-33" 1600 \
"0.0001,0.025 -47,-40,-34,-34,-31,-31,-0,-30" 6400 \
"0,0.025 -38,-31,-28,-28,-0,-25" \
gain 15 highpass 22 highpass 22 sinc -n 255 -b 16 -17500 \
gain 9 lowpass -1 17801
The audio file is played with a simulated FM radio sound (or broadcast
signal condition if the lowpass filter at the end is skipped). Note that
the pipeline is set up with US-style 75us pre-emphasis.
See also compand for a single-band companding
effect.
- noiseprof [profile-file]
- Calculate a profile of the audio for use in noise reduction. See the
description of the noisered effect for details.
- noisered [profile-file [amount]]
- Reduce noise in the audio signal by profiling and filtering. This effect
is moderately effective at removing consistent background noise such as
hiss or hum. To use it, first run SoX with the noiseprof effect on
a section of audio that ideally would contain silence but in fact contains
noise - such sections are typically found at the beginning or the end of a
recording. noiseprof will write out a noise profile to
profile-file, or to stdout if no profile-file or if `-' is
given. E.g.
sox speech.wav -n trim 0 1.5 noiseprof speech.noise-profile
To actually remove the noise, run SoX again, this time with the
noisered effect; noisered will reduce noise according to a
noise profile (which was generated by noiseprof), from
profile-file, or from stdin if no profile-file or if `-' is
given. E.g.
sox speech.wav cleaned.wav noisered speech.noise-profile 0.3
How much noise should be removed is specified by amount-a
number between 0 and 1 with a default of 0.5. Higher numbers will remove
more noise but present a greater likelihood of removing wanted components
of the audio signal. Before replacing an original recording with a
noise-reduced version, experiment with different amount values to
find the optimal one for your audio; use headphones to check that you are
happy with the results, paying particular attention to quieter sections of
the audio.
On most systems, the two stages - profiling and reduction -
can be combined using a pipe, e.g.
sox noisy.wav -n trim 0 1 noiseprof | play noisy.wav noisered
- norm [dB-level]
- Normalise the audio. norm is just an alias for gain -n; see
the gain effect for details.
- oops
- Out Of Phase Stereo effect. Mixes stereo to twin-mono where each mono
channel contains the difference between the left and right stereo
channels. This is sometimes known as the `karaoke' effect as it often has
the effect of removing most or all of the vocals from a recording. It is
equivalent to remix 1,2i 1,2i.
- overdrive [gain(20) [colour(20)]]
- Non linear distortion. The colour parameter controls the amount of
even harmonic content in the over-driven output.
- pad { length[@position(=)] }
- Pad the audio with silence, at the beginning, the end, or any specified
points through the audio. length is the amount of silence to insert
and position the position in the input audio stream at which to
insert it. Any number of lengths and positions may be specified, provided
that a specified position is not less that the previous one, and any time
specification may be used for them. position is optional for the
first and last lengths specified and if omitted correspond to the
beginning and the end of the audio respectively. For example, pad 1.5
1.5 adds 1.5 seconds of silence padding at each end of the audio,
whilst pad 4000s@3:00 inserts 4000 samples of silence 3 minutes
into the audio. If silence is wanted only at the end of the audio, specify
either the end position or specify a zero-length pad at the start.
See also delay for an effect that can add silence at
the beginning of the audio on a channel-by-channel basis.
- phaser gain-in gain-out delay decay speed
[-s|-t]
- Add a phasing effect to the audio. See [3] for a detailed description of
phasing.
delay/decay/speed gives the delay in milliseconds and the
decay (relative to gain-in) with a modulation speed in Hz. The
modulation is either sinusoidal (-s) - preferable for multiple
instruments, or triangular (-t) - gives single instruments a
sharper phasing effect. The decay should be less than 0.5 to avoid
feedback, and usually no less than 0.1. Gain-out is the volume of the
output.
For example:
play snare.flac phaser 0.8 0.74 3 0.4 0.5 -t
Gentler:
play snare.flac phaser 0.9 0.85 4 0.23 1.3 -s
A popular sound:
play snare.flac phaser 0.89 0.85 1 0.24 2 -t
More severe:
play snare.flac phaser 0.6 0.66 3 0.6 2 -t
- pitch [-q] shift [segment [search
[overlap]]]
- Change the audio pitch (but not tempo).
shift gives the pitch shift as positive or negative
`cents' (i.e. 100ths of a semitone). See the tempo effect for a
description of the other parameters.
See also the bend, speed, and tempo
effects.
- rate [-q|-l|-m|-h|-v]
[override-options] RATE[k]
- Change the audio sampling rate (i.e. resample the audio) to any given
RATE (even non-integer if this is supported by the output file
format) using a quality level defined as follows:
|
Quality |
Band-width |
Rej dB |
Typical Use |
-q |
quick |
n/a |
≈30 @ Fs/4 |
playback on ancient hardware |
-l |
low |
80% |
100 |
playback on old hardware |
-m |
medium |
95% |
100 |
audio playback |
-h |
high |
95% |
125 |
16-bit mastering (use with dither) |
-v |
very high |
95% |
175 |
24-bit mastering |
where Band-width is the percentage of the audio
frequency band that is preserved and Rej dB is the level of noise
rejection. Increasing levels of resampling quality come at the expense
of increasing amounts of time to process the audio. If no quality option
is given, the quality level used is `high' (but see `Playing &
Recording Audio' above regarding playback).
The `quick' algorithm uses cubic interpolation; all others use
band-limited interpolation. By default, all algorithms have a `linear'
phase response; for `medium', `high' and `very high', the phase response
is configurable (see below).
The rate effect is invoked automatically if SoX's
-r option specifies a rate that is different to that of the input
file(s). Alternatively, if this effect is given explicitly, then SoX's
-r option need not be given. For example, the following two
commands are equivalent:
sox input.wav -r 48k output.wav bass -b 24
sox input.wav output.wav bass -b 24 rate 48k
though the second command is more flexible as it allows rate options
to be given, and allows the effects to be ordered arbitrarily.
Warning: technically detailed discussion follows.
The simple quality selection described above provides settings
that satisfy the needs of the vast majority of resampling tasks.
Occasionally, however, it may be desirable to fine-tune the resampler's
filter response; this can be achieved using
override options, as detailed in the following table:
-M/-I/-L |
Phase response = minimum/intermediate/linear |
-s |
Steep filter (band-width = 99%) |
-a |
Allow aliasing/imaging above the pass-band |
-b 74-99.7 |
Any band-width % |
-p 0-100 |
Any phase response (0 = minimum, 25 = intermediate, 50 = linear, 100
= maximum) |
N.B. Override options cannot be used with the `quick' or `low'
quality algorithms.
All resamplers use filters that can sometimes create `echo'
(a.k.a. `ringing') artefacts with transient signals such as those that
occur with `finger snaps' or other highly percussive sounds. Such
artefacts are much more noticeable to the human ear if they occur before
the transient (`pre-echo') than if they occur after it (`post-echo').
Note that frequency of any such artefacts is related to the smaller of
the original and new sampling rates but that if this is at least
44.1kHz, then the artefacts will lie outside the range of human
hearing.
A phase response setting may be used to control the
distribution of any transient echo between `pre' and `post': with
minimum phase, there is no pre-echo but the longest post-echo; with
linear phase, pre and post echo are in equal amounts (in signal terms,
but not audibility terms); the intermediate phase setting attempts to
find the best compromise by selecting a small length (and level) of
pre-echo and a medium lengthed post-echo.
Minimum, intermediate, or linear phase response is selected
using the -M, -I, or -L option; a custom phase
response can be created with the -p option. Note that phase
responses between `linear' and `maximum' (greater than 50) are rarely
useful.
A resampler's band-width setting determines how much of the
frequency content of the original signal (w.r.t. the original sample
rate when up-sampling, or the new sample rate when down-sampling) is
preserved during conversion. The term `pass-band' is used to refer to
all frequencies up to the band-width point (e.g. for 44.1kHz sampling
rate, and a resampling band-width of 95%, the pass-band represents
frequencies from 0Hz (D.C.) to circa 21kHz). Increasing the resampler's
band-width results in a slower conversion and can increase transient
echo artefacts (and vice versa).
The -s `steep filter' option changes resampling
band-width from the default 95% (based on the 3dB point), to 99%. The
-b option allows the band-width to be set to any value in the
range 74-99.7 %, but note that band-width values greater than 99% are
not recommended for normal use as they can cause excessive transient
echo.
If the -a option is given, then aliasing/imaging above
the pass-band is allowed. For example, with 44.1kHz sampling rate, and a
resampling band-width of 95%, this means that frequency content above
21kHz can be distorted; however, since this is above the pass-band (i.e.
above the highest frequency of interest/audibility), this may not be a
problem. The benefits of allowing aliasing/imaging are reduced
processing time, and reduced (by almost half) transient echo artefacts.
Note that if this option is given, then the minimum band-width allowable
with -b increases to 85%.
Examples:
sox input.wav -b 16 output.wav rate -s -a 44100 dither -s
default (high) quality resampling; overrides: steep filter, allow aliasing;
to 44.1kHz sample rate; noise-shaped dither to 16-bit WAV file.
sox input.wav -b 24 output.aiff rate -v -I -b 90 48k
very high quality resampling; overrides: intermediate phase, band-width 90%;
to 48k sample rate; store output to 24-bit AIFF file.
The pitch and speed effects use the rate
effect at their core.
- remix [-a|-m|-p] <out-spec>
- out-spec = in-spec{,in-spec} | 0
in-spec =
[in-chan][-[in-chan2]][vol-spec]
vol-spec = p|i|v[volume]
Select and mix input audio channels into output audio
channels. Each output channel is specified, in turn, by a given
out-spec: a list of contributing input channels and volume
specifications.
Note that this effect operates on the audio channels
within the SoX effects processing chain; it should not be confused with
the -m global option (where multiple files are
mix-combined before entering the effects chain).
An out-spec contains comma-separated input
channel-numbers and hyphen-delimited channel-number ranges;
alternatively, 0 may be given to create a silent output channel.
For example,
sox input.wav output.wav remix 6 7 8 0
creates an output file with four channels, where channels 1, 2, and 3 are
copies of channels 6, 7, and 8 in the input file, and channel 4 is silent.
Whereas
sox input.wav output.wav remix 1-3,7 3
creates a (somewhat bizarre) stereo output file where the left channel is a
mix-down of input channels 1, 2, 3, and 7, and the right channel is a copy
of input channel 3.
Where a range of channels is specified, the channel numbers to
the left and right of the hyphen are optional and default to 1 and to
the number of input channels respectively. Thus
sox input.wav output.wav remix -
performs a mix-down of all input channels to mono.
By default, where an output channel is mixed from multiple (n)
input channels, each input channel will be scaled by a factor of
¹/n. Custom mixing volumes can be set by following a given input
channel or range of input channels with a vol-spec (volume
specification). This is one of the letters p, i, or
v, followed by a volume number, the meaning of which depends on
the given letter and is defined as follows:
Letter |
Volume number |
Notes |
p |
power adjust in dB |
0 = no change |
i |
power adjust in dB |
As `p', but invert the audio |
v |
voltage multiplier |
1 = no change, 0.5 ≈ 6dB attenuation, 2 ≈ 6dB gain, -1
= invert |
If an out-spec includes at least one vol-spec
then, by default, ¹/n scaling is not applied to any other
channels in the same out-spec (though may be in other out-specs). The -a
(automatic) option however, can be given to retain the automatic scaling
in this case. For example,
sox input.wav output.wav remix 1,2 3,4v0.8
results in channel level multipliers of 0.5,0.5 1,0.8, whereas
sox input.wav output.wav remix -a 1,2 3,4v0.8
results in channel level multipliers of 0.5,0.5 0.5,0.8.
The -m (manual) option disables all automatic volume
adjustments, so
sox input.wav output.wav remix -m 1,2 3,4v0.8
results in channel level multipliers of 1,1 1,0.8.
The volume number is optional and omitting it corresponds to
no volume change; however, the only case in which this is useful is in
conjunction with i. For example, if input.wav is stereo,
then
sox input.wav output.wav remix 1,2i
is a mono equivalent of the oops effect.
If the -p option is given, then any automatic
¹/n scaling is replaced by ¹/√n (`power') scaling;
this gives a louder mix but one that might occasionally clip.
One use of the remix effect is to split an audio file
into a set of files, each containing one of the constituent channels (in
order to perform subsequent processing on individual audio channels).
Where more than a few channels are involved, a script such as the
following (Bourne shell script) is useful:
#!/bin/sh
chans=`soxi -c "$1"`
while [ $chans -ge 1 ]; do
chans0=`printf %02i $chans` # 2 digits hence up to 99 chans
out=`echo "$1"|sed "s/\(.*\)\.\(.*\)/\1-$chans0.\2/"`
sox "$1" "$out" remix $chans
chans=`expr $chans - 1`
done
If a file input.wav containing six audio channels were given, the
script would produce six output files: input-01.wav,
input-02.wav, ..., input-06.wav.
See also the swap effect.
- repeat [count(1)|-]
- Repeat the entire audio count times, or once if count is not
given. The special value - requests infinite repetition. Requires
temporary file space to store the audio to be repeated. Note that
repeating once yields two copies: the original audio and the repeated
audio.
- reverb [-w|--wet-only] [reverberance (50%)
[HF-damping (50%)
- [room-scale (100%) [stereo-depth (100%)
[pre-delay (0ms) [wet-gain (0dB)]]]]]]
Add reverberation to the audio using the `freeverb' algorithm.
A reverberation effect is sometimes desirable for concert halls that are
too small or contain so many people that the hall's natural reverberance
is diminished. Applying a small amount of stereo reverb to a (dry) mono
signal will usually make it sound more natural. See [3] for a detailed
description of reverberation.
Note that this effect increases both the volume and the length
of the audio, so to prevent clipping in these domains, a typical
invocation might be:
play dry.wav gain -3 pad 0 3 reverb
The -w option can be given to select only the `wet' signal, thus
allowing it to be processed further, independently of the `dry' signal.
E.g.
play -m voice.wav "|sox voice.wav -p reverse reverb -w reverse"
for a reverse reverb effect.
- reverse
- Reverse the audio completely. Requires temporary file space to store the
audio to be reversed.
- riaa
- Apply RIAA vinyl playback equalisation. The sampling rate must be one of:
44.1, 48, 88.2, 96 kHz.
This effect supports the --plot global option.
- silence [-l] above-periods [duration
threshold[d|%]
- [below-periods duration threshold[d|%]]
Removes silence from the beginning, middle, or end of the
audio. `Silence' is determined by a specified threshold.
The above-periods value is used to indicate if audio
should be trimmed at the beginning of the audio. A value of zero
indicates no silence should be trimmed from the beginning. When
specifying a non-zero above-periods, it trims audio up until it
finds non-silence. Normally, when trimming silence from beginning of
audio the above-periods will be 1 but it can be increased to
higher values to trim all audio up to a specific count of non-silence
periods. For example, if you had an audio file with two songs that each
contained 2 seconds of silence before the song, you could specify an
above-period of 2 to strip out both silence periods and the first
song.
When above-periods is non-zero, you must also specify a
duration and threshold. duration indicates the
amount of time that non-silence must be detected before it stops
trimming audio. By increasing the duration, burst of noise can be
treated as silence and trimmed off.
threshold is used to indicate what sample value you
should treat as silence. For digital audio, a value of 0 may be fine but
for audio recorded from analog, you may wish to increase the value to
account for background noise.
When optionally trimming silence from the end of the audio,
you specify a below-periods count. In this case,
below-period means to remove all audio after silence is detected.
Normally, this will be a value 1 of but it can be increased to skip over
periods of silence that are wanted. For example, if you have a song with
2 seconds of silence in the middle and 2 second at the end, you could
set below-period to a value of 2 to skip over the silence in the middle
of the audio.
For below-periods, duration specifies a period
of silence that must exist before audio is not copied any more. By
specifying a higher duration, silence that is wanted can be left in the
audio. For example, if you have a song with an expected 1 second of
silence in the middle and 2 seconds of silence at the end, a duration of
2 seconds could be used to skip over the middle silence.
Unfortunately, you must know the length of the silence at the
end of your audio file to trim off silence reliably. A workaround is to
use the silence effect in combination with the reverse
effect. By first reversing the audio, you can use the
above-periods to reliably trim all audio from what looks like the
front of the file. Then reverse the file again to get back to
normal.
To remove silence from the middle of a file, specify a
below-periods that is negative. This value is then treated as a
positive value and is also used to indicate that the effect should
restart processing as specified by the above-periods, making it
suitable for removing periods of silence in the middle of the audio.
The option -l indicates that below-periods
duration length of audio should be left intact at the beginning
of each period of silence. For example, if you want to remove long
pauses between words but do not want to remove the pauses
completely.
duration is a time specification with the peculiarity
that a bare number is interpreted as a sample count, not as a number of
seconds. For specifying seconds, either use the t suffix (as in
`2t') or specify minutes, too (as in `0:02').
threshold numbers may be suffixed with d to
indicate the value is in decibels, or % to indicate a percentage
of maximum value of the sample value (0% specifies pure digital
silence).
The following example shows how this effect can be used to
start a recording that does not contain the delay at the start which
usually occurs between `pressing the record button' and the start of the
performance:
rec parameters filename other-effects silence 1 5 2%
- sinc [-a att|-b beta] [-p
phase|-M|-I|-L] [-t tbw|-n
taps] [freqHP][-freqLP [-t tbw|-n
taps]]
- Apply a sinc kaiser-windowed low-pass, high-pass, band-pass, or
band-reject filter to the signal. The freqHP and freqLP
parameters give the frequencies of the 6dB points of a high-pass and
low-pass filter that may be invoked individually, or together. If both are
given, then freqHP less than freqLP creates a band-pass
filter, freqHP greater than freqLP creates a band-reject
filter. For example, the invocations
sinc 3k
sinc -4k
sinc 3k-4k
sinc 4k-3k
create a high-pass, low-pass, band-pass, and band-reject filter
respectively.
The default stop-band attenuation of 120dB can be overridden
with -a; alternatively, the kaiser-window `beta' parameter can be
given directly with -b.
The default transition band-width of 5% of the total band can
be overridden with -t (and tbw in Hertz); alternatively,
the number of filter taps can be given directly with -n.
If both freqHP and freqLP are given, then a
-t or -n option given to the left of the frequencies
applies to both frequencies; one of these options given to the right of
the frequencies applies only to freqLP.
The -p, -M, -I, and -L options
control the filter's phase response; see the rate effect for
details.
This effect supports the --plot global option.
- spectrogram [options]
- Create a spectrogram of the audio; the audio is passed unmodified through
the SoX processing chain. This effect is optional - type sox --help
and check the list of supported effects to see if it has been included.
The spectrogram is rendered in a Portable Network Graphic
(PNG) file, and shows time in the X-axis, frequency in the Y-axis, and
audio signal magnitude in the Z-axis. Z-axis values are represented by
the colour (or optionally the intensity) of the pixels in the X-Y plane.
If the audio signal contains multiple channels then these are shown from
top to bottom starting from channel 1 (which is the left channel for
stereo audio).
For example, if `my.wav' is a stereo file, then with
sox my.wav -n spectrogram
a spectrogram of the entire file will be created in the file
`spectrogram.png'. More often though, analysis of a smaller portion of the
audio is required; e.g. with
sox my.wav -n remix 2 trim 20 30 spectrogram
the spectrogram shows information only from the second (right) channel, and
of thirty seconds of audio starting from twenty seconds in. To analyse a
small portion of the frequency domain, the rate effect may be used,
e.g.
sox my.wav -n rate 6k spectrogram
allows detailed analysis of frequencies up to 3kHz (half the sampling rate)
i.e. where the human auditory system is most sensitive. With
sox my.wav -n trim 0 10 spectrogram -x 600 -y 200 -z 100
the given options control the size of the spectrogram's X, Y & Z axes
(in this case, the spectrogram area of the produced image will be 600 by
200 pixels in size and the Z-axis range will be 100 dB). Note that the
produced image includes axes legends etc. and so will be a little larger
than the specified spectrogram size. In this example:
sox -n -n synth 6 tri 10k:14k spectrogram -z 100 -w kaiser
an analysis `window' with high dynamic range is selected to best display the
spectrogram of a swept triangular wave. For a smilar example, append the
following to the `chime' command in the description of the delay
effect (above):
rate 2k spectrogram -X 200 -Z -10 -w kaiser
Options are also available to control the appearance (colour-set,
brightness, contrast, etc.) and filename of the spectrogram; e.g. with
sox my.wav -n spectrogram -m -l -o print.png
a spectrogram is created suitable for printing on a `black and white'
printer.
Options:
- -x num
- Change the (maximum) width (X-axis) of the spectrogram from its default
value of 800 pixels to a given number between 100 and 200000. See also
-X and -d.
- -X num
- X-axis pixels/second; the default is auto-calculated to fit the given or
known audio duration to the X-axis size, or 100 otherwise. If given in
conjunction with -d, this option affects the width of the
spectrogram; otherwise, it affects the duration of the spectrogram.
num can be from 1 (low time resolution) to 5000 (high time
resolution) and need not be an integer. SoX may make a slight adjustment
to the given number for processing quantisation reasons; if so, SoX will
report the actual number used (viewable when the SoX global option
-V is in effect). See also -x and -d.
- -y num
- Sets the Y-axis size in pixels (per channel); this is the number of
frequency `bins' used in the Fourier analysis that produces the
spectrogram. N.B. it can be slow to produce the spectrogram if this number
is not one more than a power of two (e.g. 129). By default the Y-axis size
is chosen automatically (depending on the number of channels). See
-Y for alternative way of setting spectrogram height.
- -Y num
- Sets the target total height of the spectrogram(s). The default value is
550 pixels. Using this option (and by default), SoX will choose a height
for individual spectrogram channels that is one more than a power of two,
so the actual total height may fall short of the given number. However,
there is also a minimum height per channel so if there are many channels,
the number may be exceeded. See -y for alternative way of setting
spectrogram height.
- -z num
- Z-axis (colour) range in dB, default 120. This sets the dynamic-range of
the spectrogram to be -num dBFS to 0 dBFS. Num
may range from 20 to 180. Decreasing dynamic-range effectively increases
the `contrast' of the spectrogram display, and vice versa.
- -Z num
- Sets the upper limit of the Z-axis in dBFS. A negative num
effectively increases the `brightness' of the spectrogram display, and
vice versa.
- -q num
- Sets the Z-axis quantisation, i.e. the number of different colours (or
intensities) in which to render Z-axis values. A small number (e.g. 4)
will give a `poster'-like effect making it easier to discern magnitude
bands of similar level. Small numbers also usually result in small PNG
files. The number given specifies the number of colours to use inside the
Z-axis range; two colours are reserved to represent out-of-range
values.
- -w name
- Window: Hann (default), Hamming, Bartlett, Rectangular, Kaiser or Dolph.
The spectrogram is produced using the Discrete Fourier Transform (DFT)
algorithm. A significant parameter to this algorithm is the choice of
`window function'. By default, SoX uses the Hann window which has good
all-round frequency-resolution and dynamic-range properties. For better
frequency resolution (but lower dynamic-range), select a Hamming window;
for higher dynamic-range (but poorer frequency-resolution), select a Dolph
window. Kaiser, Bartlett and Rectangular windows are also available.
- -W num
- Window adjustment parameter. This can be used to make small adjustments to
the Kaiser or Dolph window shape. A positive number (up to ten) increases
its dynamic range, a negative number decreases it.
- -s
- Allow slack overlapping of DFT windows. This can, in some cases, increase
image sharpness and give greater adherence to the -x value, but at
the expense of a little spectral loss.
- -m
- Creates a monochrome spectrogram (the default is colour).
- -h
- Selects a high-colour palette - less visually pleasing than the default
colour palette, but it may make it easier to differentiate different
levels. If this option is used in conjunction with -m, the result
will be a hybrid monochrome/colour palette.
- -p num
- Permute the colours in a colour or hybrid palette. The num
parameter, from 1 (the default) to 6, selects the permutation.
- -l
- Creates a `printer friendly' spectrogram with a light background (the
default has a dark background).
- -a
- Suppress the display of the axis lines. This is sometimes useful in
helping to discern artefacts at the spectrogram edges.
- -r
- Raw spectrogram: suppress the display of axes and legends.
- -A
- Selects an alternative, fixed colour-set. This is provided only for
compatibility with spectrograms produced by another package. It should not
normally be used as it has some problems, not least, a lack of
differentiation at the bottom end which results in masking of low-level
artefacts.
- -t text
- Set the image title - text to display above the spectrogram.
- -c text
- Set (or clear) the image comment - text to display below and to the left
of the spectrogram.
- -o file
- Name of the spectrogram output PNG file, default `spectrogram.png'. If `-'
is given, the spectrogram will be sent to standard output (stdout).
-
- Advanced Options:
In order to process a smaller section of audio without affecting other
effects or the output signal (unlike when the trim effect is used),
the following options may be used.
- -d duration
- This option sets the X-axis resolution such that audio with the given
duration (a time specification) fits the selected (or default)
X-axis width. For example,
sox input.mp3 output.wav -n spectrogram -d 1:00 stats
creates a spectrogram showing the first minute of the audio, whilst the
stats effect is applied to the entire audio signal.
See also -X for an alternative way of setting the
X-axis resolution.
- -S position(=)
- Start the spectrogram at the given point in the audio stream. For example
sox input.aiff output.wav spectrogram -S 1:00
creates a spectrogram showing all but the first minute of the audio (the
output file, however, receives the entire audio stream).
-
- For the ability to perform off-line processing of spectral data, see the
stat effect.
- speed factor[c]
- Adjust the audio speed (pitch and tempo together). factor is either
the ratio of the new speed to the old speed: greater than 1 speeds up,
less than 1 slows down, or, if appended with the letter `c', the number of
cents (i.e. 100ths of a semitone) by which the pitch (and tempo) should be
adjusted: greater than 0 increases, less than 0 decreases.
Technically, the speed effect only changes the sample rate
information, leaving the samples themselves untouched. The rate
effect is invoked automatically to resample to the output sample rate,
using its default quality/speed. For higher quality or higher speed
resampling, in addition to the speed effect, specify the
rate effect with the desired quality option.
See also the bend, pitch, and tempo
effects.
- splice [-h|-t|-q] {
position(=)[, excess[,leeway]] }
- Splice together audio sections. This effect provides two things over
simple audio concatenation: a (usually short) cross-fade is applied at the
join, and a wave similarity comparison is made to help determine the best
place at which to make the join.
One of the options -h, -t, or -q may be
given to select the fade envelope as half-cosine wave (the default),
triangular (a.k.a. linear), or quarter-cosine wave respectively.
Type |
Audio |
Fade level |
Transitions |
t |
correlated |
constant gain |
abrupt |
h |
correlated |
constant gain |
smooth |
q |
uncorrelated |
constant power |
smooth |
To perform a splice, first use the trim effect to
select the audio sections to be joined together. As when performing a
tape splice, the end of the section to be spliced onto should be trimmed
with a small excess (default 0.005 seconds) of audio after the
ideal joining point. The beginning of the audio section to splice on
should be trimmed with the same excess (before the ideal joining
point), plus an additional leeway (default 0.005 seconds). Any
time specification may be used for these parameters. SoX should then be
invoked with the two audio sections as input files and the splice
effect given with the position at which to perform the splice - this is
length of the first audio section (including the excess).
The following diagram uses the tape analogy to illustrate the
splice operation. The effect simulates the diagonal cuts and joins the
two pieces:
length1 excess
-----------><--->
_________ : : _________________
\ : : :\ `
\ : : : \ `
\: : : \ `
* : : * - - *
\ : : :\ `
\ : : : \ `
_______________\: : : \_____`____
: : : :
<---> <----->
excess leeway
where * indicates the joining points.
For example, a long song begins with two verses which start
(as determined e.g. by using the play command with the
trim (start) effect) at times 0:30.125 and 1:03.432. The
following commands cut out the first verse:
sox too-long.wav part1.wav trim 0 30.130
(5 ms excess, after the first verse starts)
sox too-long.wav part2.wav trim 1:03.422
(5 ms excess plus 5 ms leeway, before the second verse starts)
sox part1.wav part2.wav just-right.wav splice 30.130
For another example, the SoX command
play "|sox -n -p synth 1 sin %1" "|sox -n -p synth 1 sin %3"
generates and plays two notes, but there is a nasty click at the transition;
the click can be removed by splicing instead of concatenating the audio,
i.e. by appending splice 1 to the command. (Clicks at the beginning
and end of the audio can be removed by preceding the splice effect
with fade q .01 2 .01).
Provided your arithmetic is good enough, multiple splices can
be performed with a single splice invocation. For example:
#!/bin/sh
# Audio Copy and Paste Over
# acpo infile copy-start copy-stop paste-over-start outfile
# No chained time specifications allowed for the parameters
# (i.e. such that contain +/-).
e=0.005 # Using default excess
l=$e # and leeway.
sox "$1" piece.wav trim $2-$e-$l =$3+$e
sox "$1" part1.wav trim 0 $4+$e
sox "$1" part2.wav trim $4+$3-$2-$e-$l
sox part1.wav piece.wav part2.wav "$5" \
splice $4+$e +$3-$2+$e+$l+$e
In the above Bourne shell script, two splices are used to `copy and paste'
audio.
It is also possible to use this effect to perform general
cross-fades, e.g. to join two songs. In this case, excess would
typically be an number of seconds, the -q option would typically
be given (to select an `equal power' cross-fade), and leeway
should be zero (which is the default if -q is given). For
example, if f1.wav and f2.wav are audio files to be cross-faded,
then
sox f1.wav f2.wav out.wav splice -q $(soxi -D f1.wav),3
cross-fades the files where the point of equal loudness is 3 seconds before
the end of f1.wav, i.e. the total length of the cross-fade is 2 × 3
= 6 seconds (Note: the $(...) notation is POSIX shell).
- stat [-s scale] [-rms] [-freq]
[-v] [-d]
- Display time and frequency domain statistical information about the audio.
Audio is passed unmodified through the SoX processing chain.
The information is output to the `standard error' (stderr)
stream and is calculated, where n is the duration of the audio in
samples, c is the number of audio channels, r is the audio
sample rate, and xk represents the PCM value (in the range -1 to
+1 by default) of each successive sample in the audio, as follows:
Samples read |
n×c |
|
Length (seconds) |
n÷r |
Scaled by |
|
See -s below. |
Maximum amplitude |
max(xk) |
The maximum sample value in the audio; usually this will be a
positive number. |
Minimum amplitude |
min(xk) |
The minimum sample value in the audio; usually this will be a
negative number. |
Midline amplitude |
½min(xk)+½max(xk) |
Mean norm |
¹/nΣ│xk│ |
The average of the absolute value of each sample in the audio. |
Mean amplitude |
¹/nΣxk |
The average of each sample in the audio. If this figure is non-zero,
then it indicates the presence of a D.C. offset (which could be
removed using the dcshift effect). |
RMS amplitude |
√(¹/nΣxk²) |
The level of a D.C. signal that would have the same power as the
audio's average power. |
Maximum delta |
max(│xk-xk-1│) |
Minimum delta |
min(│xk-xk-1│) |
Mean delta |
¹/n-1Σ│xk-xk-1│ |
RMS delta |
√(¹/n-1Σ(xk-xk-1)²) |
Rough frequency |
|
In Hz. |
Volume Adjustment |
|
The parameter to the vol effect which would make the audio as loud
as possible without clipping. Note: See the discussion on Clipping
above for reasons why it is rarely a good idea actually to do
this. |
Note that the delta measurements are not applicable for
multi-channel audio.
The -s option can be used to scale the input data by a
given factor. The default value of scale is 2147483647 (i.e. the
maximum value of a 32-bit signed integer). Internal effects always work
with signed long PCM data and so the value should relate to this
fact.
The -rms option will convert all output average values
to `root mean square' format.
The -v option displays only the `Volume Adjustment'
value.
The -freq option calculates the input's power spectrum
(4096 point DFT) instead of the statistics listed above. This should
only be used with a single channel audio file.
The -d option displays a hex dump of the 32-bit signed
PCM data audio in SoX's internal buffer. This is mainly used to help
track down endian problems that sometimes occur in cross-platform
versions of SoX.
See also the stats effect.
- stats [-b bits|-x bits|-s
scale] [-w window-time]
- Display time domain statistical information about the audio channels;
audio is passed unmodified through the SoX processing chain. Statistics
are calculated and displayed for each audio channel and, where applicable,
an overall figure is also given.
For example, for a typical well-mastered stereo music
file:
Overall Left Right |
DC offset 0.000803 -0.000391 0.000803 |
Min level -0.750977 -0.750977 -0.653412 |
Max level 0.708801 0.708801 0.653534 |
Pk lev dB -2.49 -2.49 -3.69 |
RMS lev dB -19.41 -19.13 -19.71 |
RMS Pk dB -13.82 -13.82 -14.38 |
RMS Tr dB -85.25 -85.25 -82.66 |
Crest factor - 6.79 6.32 |
Flat factor 0.00 0.00 0.00 |
Pk count 2 2 2 |
Bit-depth 16/16 16/16 16/16 |
Num samples 7.72M |
Length s 174.973 |
Scale max 1.000000 |
Window s 0.050 |
DC offset, Min level, and
Max level are shown, by default, in the range ±1.
If the -b (bits) options is given, then these three measurements
will be scaled to a signed integer with the given number of bits; for
example, for 16 bits, the scale would be -32768 to +32767. The -x
option behaves the same way as -b except that the signed integer
values are displayed in hexadecimal. The -s option scales the
three measurements by a given floating-point number.
Pk lev dB and
RMS lev dB are standard peak and RMS level measured
in dBFS. RMS Pk dB and
RMS Tr dB are peak and trough values for RMS level
measured over a short window (default 50ms).
Crest factor is the standard ratio of peak to
RMS level (note: not in dB).
Flat factor is a measure of the flatness (i.e.
consecutive samples with the same value) of the signal at its peak
levels (i.e. either Min level, or
Max level). Pk count is the number of
occasions (not the number of samples) that the signal attained either
Min level, or Max level.
The right-hand Bit-depth figure is the standard
definition of bit-depth i.e. bits less significant than the given number
are fixed at zero. The left-hand figure is the number of most
significant bits that are fixed at zero (or one for negative numbers)
subtracted from the right-hand figure (the number subtracted is directly
related to Pk lev dB).
For multi-channel audio, an overall figure for each of the
above measurements is given and derived from the channel figures as
follows: DC offset: maximum magnitude;
Max level, Pk lev dB,
RMS Pk dB, Bit-depth: maximum;
Min level, RMS Tr dB: minimum;
RMS lev dB, Flat factor,
Pk count: average; Crest factor: not
applicable.
Length s is the duration in seconds of the
audio, and Num samples is equal to the sample-rate
multiplied by Length. Scale Max is the scaling
applied to the first three measurements; specifically, it is the maximum
value that could apply to Max level.
Window s is the length of the window used for the peak and
trough RMS measurements.
See also the stat effect.
- swap
- Swap stereo channels. If the input is not stereo, pairs of channels are
swapped, and a possible odd last channel passed through. E.g., for seven
channels, the output order will be 2, 1, 4, 3, 6, 5, 7.
See also remix for an effect that allows arbitrary
channel selection and ordering (and mixing).
- stretch factor [window fade shift fading]
- Change the audio duration (but not its pitch). This effect is broadly
equivalent to the tempo effect with (factor inverted and)
search set to zero, so in general, its results are comparatively
poor; it is retained as it can sometimes out-perform tempo for
small factors.
factor of stretching: >1 lengthen, <1 shorten
duration. window size is in ms. Default is 20ms. The fade
option, can be `lin'. shift ratio, in [0 1]. Default depends on
stretch factor. 1 to shorten, 0.8 to lengthen. The fading ratio,
in [0 0.5]. The amount of a fade's default depends on factor and
shift.
See also the tempo effect.
- synth [-j KEY] [-n] [len [off
[ph [p1 [p2 [p3]]]]]] {[type]
[combine]
[[%]freq[k][:|+|/|-[%]freq2[k]]]
[off [ph [p1 [p2 [p3]]]]]}
- This effect can be used to generate fixed or swept frequency audio tones
with various wave shapes, or to generate wide-band noise of various
`colours'. Multiple synth effects can be cascaded to produce more complex
waveforms; at each stage it is possible to choose whether the generated
waveform will be mixed with, or modulated onto the output from the
previous stage. Audio for each channel in a multi-channel audio file can
be synthesised independently.
Though this effect is used to generate audio, an input file
must still be given, the characteristics of which will be used to set
the synthesised audio length, the number of channels, and the sampling
rate; however, since the input file's audio is not normally needed, a
`null file' (with the special name -n) is often given instead
(and the length specified as a parameter to synth or by another
given effect that has an associated length).
For example, the following produces a 3 second, 48kHz, audio
file containing a sine-wave swept from 300 to 3300 Hz:
sox -n output.wav synth 3 sine 300-3300
and this produces an 8 kHz version:
sox -r 8000 -n output.wav synth 3 sine 300-3300
Multiple channels can be synthesised by specifying the set of parameters
shown between braces multiple times; the following puts the swept tone in
the left channel and adds `brown' noise in the right:
sox -n output.wav synth 3 sine 300-3300 brownnoise
The following example shows how two synth effects can be cascaded to create
a more complex waveform:
play -n synth 0.5 sine 200-500 synth 0.5 sine fmod 700-100
Frequencies can also be given in `scientific' note notation, or, by
prefixing a `%' character, as a number of semitones relative to `middle A'
(440 Hz). For example, the following could be used to help tune a
guitar's low `E' string:
play -n synth 4 pluck %-29
or with a (Bourne shell) loop, the whole guitar:
for n in E2 A2 D3 G3 B3 E4; do
play -n synth 4 pluck $n repeat 2; done
See the delay effect (above) and the reference to `SoX scripting
examples' (below) for more synth examples.
N.B. This effect generates audio at maximum volume
(0dBFS), which means that there is a high chance of clipping when using
the audio subsequently, so in many cases, you will want to follow this
effect with the gain effect to prevent this from happening. (See
also Clipping above.) Note that, by default, the synth
effect incorporates the functionality of gain -h (see the
gain effect for details); synth's -n option may be
given to disable this behaviour.
A detailed description of each synth parameter
follows:
len is the length of audio to synthesise (any time
specification); a value of 0 indicated to use the input length, which is
also the default.
type is one of sine, square, triangle, sawtooth,
trapezium, exp, [white]noise, tpdfnoise, pinknoise, brownnoise, pluck;
default=sine.
combine is one of create, mix, amod (amplitude
modulation), fmod (frequency modulation); default=create.
freq/freq2 are the frequencies at the
beginning/end of synthesis in Hz or, if preceded with `%', semitones
relative to A (440 Hz); alternatively, `scientific' note notation
(e.g. E2) may be used. The default frequency is 440Hz. By default, the
tuning used with the note notations is `equal temperament'; the
-j KEY option selects `just intonation', where KEY
is an integer number of semitones relative to A (so for example, -9 or 3
selects the key of C), or a note in scientific notation.
If freq2 is given, then len must also have been
given and the generated tone will be swept between the given
frequencies. The two given frequencies must be separated by one of the
characters `:', `+', `/', or `-'. This character is used to specify the
sweep function as follows:
- :
- Linear: the tone will change by a fixed number of hertz per second.
- +
- Square: a second-order function is used to change the tone.
- /
- Exponential: the tone will change by a fixed number of semitones per
second.
- -
- Exponential: as `/', but initial phase always zero, and stepped (less
smooth) frequency changes.
-
- Not used for noise.
off is the bias (DC-offset) of the signal in percent;
default=0.
ph is the phase shift in percentage of 1 cycle;
default=0. Not used for noise.
p1 is the percentage of each cycle that is `on'
(square), or `rising' (triangle, exp, trapezium); default=50 (square,
triangle, exp), default=10 (trapezium), or sustain (pluck);
default=40.
p2 (trapezium): the percentage through each cycle at
which `falling' begins; default=50. exp: the amplitude in multiples of
2dB; default=50, or tone-1 (pluck); default=20.
p3 (trapezium): the percentage through each cycle at
which `falling' ends; default=60, or tone-2 (pluck); default=90.
- tempo [-q] [-m|-s|-l] factor
[segment [search [overlap]]]
- Change the audio playback speed but not its pitch. This effect uses the
WSOLA algorithm. The audio is chopped up into segments which are then
shifted in the time domain and overlapped (cross-faded) at points where
their waveforms are most similar as determined by measurement of `least
squares'.
By default, linear searches are used to find the best
overlapping points. If the optional -q parameter is given, tree
searches are used instead. This makes the effect work more quickly, but
the result may not sound as good. However, if you must improve the
processing speed, this generally reduces the sound quality less than
reducing the search or overlap values.
The -m option is used to optimize default values of
segment, search and overlap for music processing.
The -s option is used to optimize default values of
segment, search and overlap for speech processing.
The -l option is used to optimize default values of
segment, search and overlap for `linear' processing that tends to cause
more noticeable distortion but may be useful when factor is close to
1.
If -m, -s, or -l is specified, the default value of segment
will be calculated based on factor, while default search and overlap
values are based on segment. Any values you provide still override these
default values.
factor gives the ratio of new tempo to the old tempo,
so e.g. 1.1 speeds up the tempo by 10%, and 0.9 slows it down by
10%.
The optional segment parameter selects the algorithm's
segment size in milliseconds. If no other flags are specified, the
default value is 82 and is typically suited to making small changes to
the tempo of music. For larger changes (e.g. a factor of 2),
41 ms may give a better result. The -m, -s, and -l flags will
cause the segment default to be automatically adjusted based on factor.
For example using -s (for speech) with a tempo of 1.25 will calculate a
default segment value of 32.
The optional search parameter gives the audio length in
milliseconds over which the algorithm will search for overlapping
points. If no other flags are specified, the default value is 14.68.
Larger values use more processing time and may or may not produce better
results. A practical maximum is half the value of segment. Search can be
reduced to cut processing time at the risk of degrading output quality.
The -m, -s, and -l flags will cause the search default to be
automatically adjusted based on segment.
The optional overlap parameter gives the segment
overlap length in milliseconds. Default value is 12, but -m, -s, or -l
flags automatically adjust overlap based on segment size. Increasing
overlap increases processing time and may increase quality. A practical
maximum for overlap is the value of search, with overlap typically being
(at least) a little smaller then search.
See also speed for an effect that changes tempo and
pitch together, pitch and bend for effects that change
pitch only, and stretch for an effect that changes tempo using a
different algorithm.
- treble gain [frequency[k]
[width[s|h|k|o|q]]]
- Apply a treble tone-control effect. See the description of the bass
effect for details.
- tremolo speed [depth]
- Apply a tremolo (low frequency amplitude modulation) effect to the audio.
The tremolo frequency in Hz is given by speed, and the depth as a
percentage by depth (default 40).
- trim {position(+)}
- Cuts portions out of the audio. Any number of positions may be
given; audio is not sent to the output until the first position is
reached. The effect then alternates between copying and discarding audio
at each position. Using a value of 0 for the first position
parameter allows copying from the beginning of the audio.
For example,
sox infile outfile trim 0 10
will copy the first ten seconds, while
play infile trim 12:34 =15:00 -2:00
and
play infile trim 12:34 2:26 -2:00
will both play from 12 minutes 34 seconds into the audio up to 15 minutes
into the audio (i.e. 2 minutes and 26 seconds long), then resume playing
two minutes before the end of audio.
- upsample [factor]
- Upsample the signal by an integer factor: factor-1 zero-value
samples are inserted between each pair of input samples. As a result, the
original spectrum is replicated into the new frequency space (imaging) and
attenuated. This attenuation can be compensated for by adding vol
factor after any further processing. The upsample effect is
typically used in combination with filtering effects.
For a general resampling effect with anti-imaging, see
rate. See also downsample.
- vad [options]
- Voice Activity Detector. Attempts to trim silence and quiet background
sounds from the ends of (fairly high resolution i.e. 16-bit, 44-48kHz)
recordings of speech. The algorithm currently uses a simple cepstral power
measurement to detect voice, so may be fooled by other things, especially
music. The effect can trim only from the front of the audio, so in order
to trim from the back, the reverse effect must also be used. E.g.
play speech.wav norm vad
to trim from the front,
play speech.wav norm reverse vad reverse
to trim from the back, and
play speech.wav norm vad reverse vad reverse
to trim from both ends. The use of the norm effect is recommended,
but remember that neither reverse nor norm is suitable for
use with streamed audio.
Options:
Default values are shown in parenthesis.
- -t num (7)
- The measurement level used to trigger activity detection. This might need
to be changed depending on the noise level, signal level and other
charactistics of the input audio.
- -T num (0.25)
- The time constant (in seconds) used to help ignore short bursts of
sound.
- -s num (1)
- The amount of audio (in seconds) to search for quieter/shorter bursts of
audio to include prior to the detected trigger point.
- -g num (0.25)
- Allowed gap (in seconds) between quieter/shorter bursts of audio to
include prior to the detected trigger point.
- -p num (0)
- The amount of audio (in seconds) to preserve before the trigger point and
any found quieter/shorter bursts.
-
- Advanced Options:
These allow fine tuning of the algorithm's internal parameters.
- -b num
- The algorithm (internally) uses adaptive noise estimation/reduction in
order to detect the start of the wanted audio. This option sets the time
for the initial noise estimate.
- -N num
- Time constant used by the adaptive noise estimator for when the noise
level is increasing.
- -n num
- Time constant used by the adaptive noise estimator for when the noise
level is decreasing.
- -r num
- Amount of noise reduction to use in the detection algorithm (e.g. 0, 0.5,
...).
- -f num
- Frequency of the algorithm's processing/measurements.
- -m num
- Measurement duration; by default, twice the measurement period; i.e. with
overlap.
- -M num
- Time constant used to smooth spectral measurements.
- -h num
- `Brick-wall' frequency of high-pass filter applied at the input to the
detector algorithm.
- -l num
- `Brick-wall' frequency of low-pass filter applied at the input to the
detector algorithm.
- -H num
- `Brick-wall' frequency of high-pass lifter used in the detector
algorithm.
- -L num
- `Brick-wall' frequency of low-pass lifter used in the detector
algorithm.
-
- See also the silence effect.
- vol gain [type [limitergain]]
- Apply an amplification or an attenuation to the audio signal. Unlike the
-v option (which is used for balancing multiple input files as they
enter the SoX effects processing chain), vol is an effect like any
other so can be applied anywhere, and several times if necessary, during
the processing chain.
The amount to change the volume is given by gain which
is interpreted, according to the given type, as follows: if
type is amplitude (or is omitted), then gain is an
amplitude (i.e. voltage or linear) ratio, if power, then a power
(i.e. wattage or voltage-squared) ratio, and if dB, then a power
change in dB.
When type is amplitude or power, a
gain of 1 leaves the volume unchanged, less than 1 decreases it,
and greater than 1 increases it; a negative gain inverts the
audio signal in addition to adjusting its volume.
When type is dB, a gain of 0 leaves the
volume unchanged, less than 0 decreases it, and greater than 0 increases
it.
See [4] for a detailed discussion on electrical (and hence
audio signal) voltage and power ratios.
Beware of Clipping when the increasing the volume.
The gain and the type parameters can be
concatenated if desired, e.g. vol 10dB.
An optional limitergain value can be specified and
should be a value much less than 1 (e.g. 0.05 or 0.02) and is used only
on peaks to prevent clipping. Not specifying this parameter will cause
no limiter to be used. In verbose mode, this effect will display the
percentage of the audio that needed to be limited.
See also gain for a volume-changing effect with
different capabilities, and compand for a dynamic-range
compression/expansion/limiting effect.
Exit status is 0 for no error, 1 if there is a problem with the command-line
parameters, or 2 if an error occurs during file processing.
Please report any bugs found in this version of SoX to the mailing list
(sox-users@lists.sourceforge.net).
soxi(1), soxformat(7), libsox(3)
audacity(1), gnuplot(1), octave(1), wget(1)
The SoX web site at http://sox.sourceforge.net
SoX scripting examples at http://sox.sourceforge.net/Docs/Scripts
- [1]
- R. Bristow-Johnson, Cookbook formulae for audio EQ biquad filter
coefficients, http://musicdsp.org/files/Audio-EQ-Cookbook.txt
- [2]
- Wikipedia, Q-factor, http://en.wikipedia.org/wiki/Q_factor
- [3]
- Scott Lehman, Effects Explained,
http://harmony-central.com/Effects/effects-explained.html
- [4]
- Wikipedia, Decibel, http://en.wikipedia.org/wiki/Decibel
- [5]
- Richard Furse, Linux Audio Developer's Simple Plugin API,
http://www.ladspa.org
- [6]
- Richard Furse, Computer Music Toolkit,
http://www.ladspa.org/cmt
- [7]
- Steve Harris, LADSPA plugins, http://plugin.org.uk
Copyright 1998-2013 Chris Bagwell and SoX Contributors.
Copyright 1991 Lance Norskog and Sundry Contributors.
This program is free software; you can redistribute it and/or
modify it under the terms of the GNU General Public License as published by
the Free Software Foundation; either version 2, or (at your option) any
later version.
This program is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU General
Public License for more details.
Chris Bagwell (cbagwell@users.sourceforge.net). Other authors and contributors
are listed in the ChangeLog file that is distributed with the source code.
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