SoX(7) Sound eXchange SoX(7)
NAME
SoX - Sound eXchange, the Swiss Army knife of audio manipulation
DESCRIPTION
This manual describes SoX audio effects; the SoX manual set starts with
sox(1).
In addition to converting and playing audio files, SoX can be used to
invoke a number of audio `effects'. Multiple effects may be applied by
specifying them one after another at the end of the SoX command line.
Note that applying multiple effects in real-time (i.e. when playing
audio) is likely to need a high performance computer; stopping other
applications may alleviate performance issues should they occur.
Some of the SoX effects are primarily intended to be applied to a sin-
gle instrument or `voice'. To facilitate this, the remix effect and
the global SoX option -M can be used to isolate then recombine tracks
from a multi-track recording.
In the descriptions that follow, brackets [ ] are used to denote param-
eters that are optional, braces { } to denote those that are both
optional and repeatable, and angle brackets < > to denote those that
are repeatable but not optional. Where applicable, default values for
optional parameters are shown in parenthesis ( ).
The following parameters are used with, and have the same meaning for,
several effects:
centre[k]
See frequency.
frequency[k]
A frequency in Hz, or, if appended with `k', kHz.
gain A power gain in dB. Zero gives no gain; less than zero gives an
attenuation.
width[h|k|o|q]
Used to specify the band-width of a filter. A number of differ-
ent methods to specify the width are available (though not all
for every effect); one of the characters shown may be appended
to select the desired method as follows:
+-----------------------+
| Method Notes |
|h Hz |
|k kHz |
|o Octaves |
|q Q-factor See [2] |
+-----------------------+
For each effect that uses this parameter, the default method
(i.e. if no character is appended) is the one that it listed
first in the effect's first line of description.
To see if SoX has support for an optional effect, enter sox -h and look
for its name under the list: `EFFECTS'.
SOX EFFECTS
allpass frequency[k] width[h|k|o|q]
Apply a two-pole all-pass filter with central frequency (in Hz)
frequency, and filter-width width. An all-pass filter changes
the audio's frequency to phase relationship without changing its
frequency to amplitude relationship. The filter is described in
detail in [1].
This effect supports the --plot global option.
band [-n] center[k] [width[h|k|o|q]]
Apply a band-pass filter. The frequency response drops loga-
rithmically around the center frequency. The width parameter
gives the slope of the drop. The frequencies at center + width
and center - width will be half of their original amplitudes.
band defaults to a mode oriented to pitched audio, i.e. voice,
singing, or instrumental music. The -n (for noise) option uses
the alternate mode for un-pitched audio (e.g. percussion).
Warning: -n introduces a power-gain of about 11dB in the filter,
so beware of output clipping. band introduces noise in the
shape of the filter, i.e. peaking at the center frequency and
settling around it.
This effect supports the --plot global option.
See also filter for a bandpass filter with steeper shoulders.
bandpass|bandreject [-c] frequency[k] width[h|k|o|q]
Apply a two-pole Butterworth band-pass or band-reject filter
with central frequency frequency, and (3dB-point) band-width
width. The -c option applies only to bandpass and selects a
constant skirt gain (peak gain = Q) instead of the default: con-
stant 0dB peak gain. The filters roll off at 6dB per octave
(20dB per decade) and are described in detail in [1].
These effects support the --plot global option.
See also filter for a bandpass filter with steeper shoulders.
bandreject frequency[k] width[h|k|o|q]
Apply a band-reject filter. See the description of the bandpass
effect for details.
bass|treble gain [frequency[k] [width[s|h|k|o|q]]]
Boost or cut the bass (lower) or treble (upper) frequencies of
the audio using a two-pole shelving filter with a response simi-
lar to that of a standard hi-fi's tone-controls. This is also
known as shelving equalisation (EQ).
gain gives the gain at 0 Hz (for bass), or whichever is the
lower of ~22 kHz and the Nyquist frequency (for treble). Its
useful range is about -20 (for a large cut) to +20 (for a large
boost). Beware of Clipping when using a positive gain.
If desired, the filter can be fine-tuned using the following
optional parameters:
frequency sets the filter's central frequency and so can be used
to extend or reduce the frequency range to be boosted or cut.
The default value is 100 Hz (for bass) or 3 kHz (for treble).
width determines how steep is the filter's shelf transition. In
addition to the common width specification methods described
above, `slope' (the default, or if appended with `s') may be
used. The useful range of `slope' is about 0.3, for a gentle
slope, to 1 (the maximum), for a steep slope; the default value
is 0.5.
The filters are described in detail in [1].
These effects support the --plot global option.
See also equalizer for a peaking equalisation effect.
chorus gain-in gain-out <delay decay speed depth -s|-t>
Add a chorus effect to the audio. This can make a single vocal
sound like a chorus, but can also be applied to instrumentation.
Chorus resembles an echo effect with a short delay, but whereas
with echo the delay is constant, with chorus, it is varied using
sinusoidal or triangular modulation. The modulation depth
defines the range the modulated delay is played before or after
the delay. Hence the delayed sound will sound slower or faster,
that is the delayed sound tuned around the original one, like in
a chorus where some vocals are slightly off key. See [3] for
more discussion of the chorus effect.
Each four-tuple parameter delay/decay/speed/depth gives the
delay in milliseconds and the decay (relative to gain-in) with a
modulation speed in Hz using depth in milliseconds. The modula-
tion is either sinusoidal (-s) or triangular (-t). Gain-out is
the volume of the output.
A typical delay is around 40ms to 60ms; the modulation speed is
best near 0.25Hz and the modulation depth around 2ms. For exam-
ple, a single delay:
play guitar1.wav chorus 0.7 0.9 55 0.4 0.25 2 -t
Two delays of the original samples:
play guitar1.wav chorus 0.6 0.9 50 0.4 0.25 2 -t \
60 0.32 0.4 1.3 -s
A fuller sounding chorus (with three additional delays):
play guitar1.wav chorus 0.5 0.9 50 0.4 0.25 2 -t \
60 0.32 0.4 2.3 -t 40 0.3 0.3 1.3 -s
compand attack1,decay1{,attack2,decay2}
[soft-knee-dB:]in-dB1[,out-dB1]{,in-dB2,out-dB2}
[gain [initial-volume-dB [delay]]]
Compand (compress or expand) the dynamic range of the audio.
The attack and decay parameters (in seconds) determine the time
over which the instantaneous level of the input signal is aver-
aged to determine its volume; attacks refer to increases in vol-
ume and decays refer to decreases. For most situations, the
attack time (response to the music getting louder) should be
shorter than the decay time because the human ear is more sensi-
tive to sudden loud music than sudden soft music. Where more
than one pair of attack/decay parameters are specified, each
input channel is companded separately and the number of pairs
must agree with the number of input channels. Typical values
are 0.3,0.8 seconds.
The second parameter is a list of points on the compander's
transfer function specified in dB relative to the maximum possi-
ble signal amplitude. The input values must be in a strictly
increasing order but the transfer function does not have to be
monotonically rising. If omitted, the value of out-dB1 defaults
to the same value as in-dB1; levels below in-dB1 are not com-
panded (but may have gain applied to them). The point 0,0 is
assumed but may be overridden (by 0,out-dBn). If the list is
preceded by a soft-knee-dB value, then the points at where adja-
cent line segments on the transfer function meet will be rounded
by the amount given. Typical values for the transfer function
are 6:-70,-60,-20.
The third (optional) parameter is an additional gain in dB to be
applied at all points on the transfer function and allows easy
adjustment of the overall gain.
The fourth (optional) parameter is an initial level to be
assumed for each channel when companding starts. This permits
the user to supply a nominal level initially, so that, for exam-
ple, a very large gain is not applied to initial signal levels
before the companding action has begun to operate: it is quite
probable that in such an event, the output would be severely
clipped while the compander gain properly adjusts itself. A
typical value (for audio which is initially quiet) is -90 dB.
The fifth (optional) parameter is a delay in seconds. The input
signal is analysed immediately to control the compander, but it
is delayed before being fed to the volume adjuster. Specifying
a delay approximately equal to the attack/decay times allows the
compander to effectively operate in a `predictive' rather than a
reactive mode. A typical value is 0.2 seconds.
This effect supports the --plot global option (for the transfer
function).
The following example might be used to make a piece of music
with both quiet and loud passages suitable for listening to in a
noisy environment such as a moving vehicle:
sox asz.au asz-car.au compand 0.3,1 6:-70,-60,-20 -5 -90 0.2
The transfer function (`6:-70,...') says that very soft sounds
(below -70dB) will remain unchanged. This will stop the compan-
der from boosting the volume on `silent' passages such as
between movements. However, sounds in the range -60dB to 0dB
(maximum volume) will be boosted so that the 60dB dynamic range
of the original music will be compressed 3-to-1 into a 20dB
range, which is wide enough to enjoy the music but narrow enough
to get around the road noise. The `6:' selects 6dB soft-knee
companding. The -5 (dB) output gain is needed to avoid clipping
(the number is inexact, and was derived by experimentation).
The -90 (dB) for the initial volume will work fine for a clip
that starts with near silence, and the delay of 0.2 (seconds)
has the effect of causing the compander to react a bit more
quickly to sudden volume changes.
See also mcompand for a multiple-band companding effect.
contrast [enhancement-amount (75)]
Comparable with compression, this effect modifies an audio sig-
nal to make it sound louder. enhancement-amount controls the
amount of the enhancement and is a number in the range 0-100.
Note that enhancement-amount = 0 still gives a significant con-
trast enhancement. contrast is often used in conjunction with
the norm effect as follows:
sox infile outfile norm -i contrast
dcshift shift [limitergain]
DC Shift the audio, with basic linear amplitude formula. This
is most useful if your audio tends to not be centered around a
value of 0. Shifting it back will allow you to get the most
volume adjustments without clipping.
The first option is the dcshift value. It is a floating point
number that indicates the amount to shift.
An optional limitergain can be specified as well. It should
have a value much less than 1 (e.g. 0.05 or 0.02) and is used
only on peaks to prevent clipping.
An alternative approach to removing a DC offset (albeit with a
short delay) is to use the highpass filter effect at a frequency
of say 10Hz; i.e.
sox -n out.au synth 5 sin %0 50 highpass 10
deemph Apply a treble attenuation shelving filter to audio in audio-CD
format. The frequency response of pre-emphasized recordings is
rectified. The filter is defined in the standard document ISO
908.
This effect supports the --plot global option.
See also the bass and treble shelving equalisation effects.
delay {length}
Delay one or more audio channels. length can specify a time or,
if appended with an `s', a number of samples. For example,
delay 1.5 0 0.5 delays the first channel by 1.5 seconds, the
third channel by 0.5 seconds, and leaves the second channel (and
any other channels that may be present) un-delayed. The follow-
ing (one long) command plays a chime sound:
play -n synth sin %-21.5 sin %-14.5 sin %-9.5 sin %-5.5 \
sin %-2.5 sin %2.5 gain -5.4 fade h 0.008 2 1.5 \
delay 0 .27 .54 .76 1.01 1.3 remix - fade h 0.1 2.72 2.5
dither [depth]
Apply dithering to the audio. Dithering deliberately adds digi-
tal white noise to the signal in order to mask audible quantiza-
tion effects that can occur if the output sample size is less
than 24 bits. By default, the amount of noise added is 1/2 bit;
the optional depth parameter is a (linear or voltage) multiplier
of this amount.
This effect should not be followed by any other effect that
affects the audio.
earwax Makes audio easier to listen to on headphones. Adds `cues' to
44.1kHz stereo (i.e. audio CD format) audio so that when lis-
tened to on headphones the stereo image is moved from inside
your head (standard for headphones) to outside and in front of
the listener (standard for speakers). See http://www.geoci-
ties.com/beinges for a full explanation.
echo gain-in gain-out <delay decay>
Add echoing to the audio. Echoes are reflected sound and can
occur naturally amongst mountains (and sometimes large build-
ings) when talking or shouting; digital echo effects emulate
this behaviour and are often used to help fill out the sound of
a single instrument or vocal. The time difference between the
original signal and the reflection is the `delay' (time), and
the loudness of the relected signal is the `decay'. Multiple
echoes can have different delays and decays.
Each given delay decay pair gives the delay in milliseconds and
the decay (relative to gain-in) of that echo. Gain-out is the
volume of the output. For example: This will make it sound as
if there are twice as many instruments as are actually playing:
play lead.aiff echo 0.8 0.88 60 0.4
If the delay is very short, then it sound like a (metallic) ro-
bot playing music:
play lead.aiff echo 0.8 0.88 6 0.4
A longer delay will sound like an open air concert in the moun-
tains:
play lead.aiff echo 0.8 0.9 1000 0.3
One mountain more, and:
play lead.aiff echo 0.8 0.9 1000 0.3 1800 0.25
echos gain-in gain-out <delay decay>
Add a sequence of echoes to the audio. Each delay decay pair
gives the delay in milliseconds and the decay (relative to gain-
in) of that echo. Gain-out is the volume of the output.
Like the echo effect, echos stand for `ECHO in Sequel', that is
the first echos takes the input, the second the input and the
first echos, the third the input and the first and the second
echos, ... and so on. Care should be taken using many echos; a
single echos has the same effect as a single echo.
The sample will be bounced twice in symmetric echos:
play lead.aiff echos 0.8 0.7 700 0.25 700 0.3
The sample will be bounced twice in asymmetric echos:
play lead.aiff echos 0.8 0.7 700 0.25 900 0.3
The sample will sound as if played in a garage:
play lead.aiff echos 0.8 0.7 40 0.25 63 0.3
equalizer frequency[k] width[q|o|h|k] gain
Apply a two-pole peaking equalisation (EQ) filter. With this
filter, the signal-level at and around a selected frequency can
be increased or decreased, whilst (unlike band-pass and band-
reject filters) that at all other frequencies is unchanged.
frequency gives the filter's central frequency in Hz, width, the
band-width, and gain the required gain or attenuation in dB.
Beware of Clipping when using a positive gain.
In order to produce complex equalisation curves, this effect can
be given several times, each with a different central frequency.
The filter is described in detail in [1].
This effect supports the --plot global option.
See also bass and treble for shelving equalisation effects.
fade [type] fade-in-length [stop-time [fade-out-length]]
Add a fade effect to the beginning, end, or both of the audio.
For fade-ins, this starts from the first sample and ramps the
volume of the audio from 0 to full volume over fade-in-length
seconds. Specify 0 seconds if no fade-in is wanted.
For fade-outs, the audio will be truncated at stop-time and the
volume will be ramped from full volume down to 0 starting at
fade-out-length seconds before the stop-time. If fade-out-
length is not specified, it defaults to the same value as fade-
in-length. No fade-out is performed if stop-time is not speci-
fied. If the file length can be determined from the input file
header and length-changing effects are not in effect, then 0 may
be specified for stop-time to indicate the usual case of a fade-
out that ends at the end of the input audio stream.
All times can be specified in either periods of time or sample
counts. To specify time periods use the format hh:mm:ss.frac
format. To specify using sample counts, specify the number of
samples and append the letter `s' to the sample count (for exam-
ple `8000s').
An optional type can be specified to change the type of enve-
lope. Choices are q for quarter of a sine wave, h for half a
sine wave, t for linear slope, l for logarithmic, and p for
inverted parabola. The default is logarithmic.
filter [low]-[high] [window-len [beta]]
Apply a sinc-windowed lowpass, highpass, or bandpass filter of
given window length to the signal. low refers to the frequency
of the lower 6dB corner of the filter. high refers to the fre-
quency of the upper 6dB corner of the filter.
A low-pass filter is obtained by leaving low unspecified, or 0.
A high-pass filter is obtained by leaving high unspecified, or
0, or greater than or equal to the Nyquist frequency.
The window-len, if unspecified, defaults to 128. Longer windows
give a sharper cut-off, smaller windows a more gradual cut-off.
The beta parameter determines the type of filter window used.
Any value greater than 2 is the beta for a Kaiser window. Beta
<= 2 selects a Nuttall window. If unspecified, the default is a
Kaiser window with beta 16.
In the case of Kaiser window (beta > 2), lower betas produce a
somewhat faster transition from pass-band to stop-band, at the
cost of noticeable artifacts. A beta of 16 is the default, beta
less than 10 is not recommended. If you want a sharper cut-off,
don't use low beta's, use a longer sample window. A Nuttall win-
dow is selected by specifying any `beta' <= 2, and the Nuttall
window has somewhat steeper cut-off than the default Kaiser win-
dow. You will probably not need to use the beta parameter at
all, unless you are just curious about comparing the effects of
Nuttall vs. Kaiser windows.
flanger [delay depth regen width speed shape phase interp]
Apply a flanging effect to the audio. See [3] for a detailed
description of flanging.
All parameters are optional (right to left).
+-----------------------------------------------------------------+
| Range Default Description |
|delay 0 - 10 0 Base delay in milliseconds. |
|depth 0 - 10 2 Added swept delay in milliseconds. |
|regen -95 - 95 0 Percentage regeneration (delayed |
| signal feedback). |
|width 0 - 100 71 Percentage of delayed signal mixed |
| with original. |
|speed 0.1 - 10 0.5 Sweeps per second (Hz). |
|shape sin Swept wave shape: sine|triangle. |
|phase 0 - 100 25 Swept wave percentage phase-shift |
| for multi-channel (e.g. stereo) |
| flange; 0 = 100 = same phase on |
| each channel. |
|interp lin Digital delay-line interpolation: |
| linear|quadratic. |
+-----------------------------------------------------------------+
gain dB-gain
Apply an amplification or an attenuation to the audio signal.
This is an alias for the vol effect - handy for those who prefer
to work in dBs by default.
highpass|lowpass [-1|-2] frequency[k] [width[q|o|h|k]]
Apply a high-pass or low-pass filter with 3dB point frequency.
The filter can be either single-pole (with -1), or double-pole
(the default, or with -2). width applies only to double-pole
filters; the default is Q = 0.707 and gives a Butterworth
response. The filters roll off at 6dB per pole per octave (20dB
per pole per decade). The double-pole filters are described in
detail in [1].
These effects support the --plot global option.
See also filter for filters with a steeper roll-off.
key [-q] shift [segment [search [overlap]]]
Change the audio key (i.e. pitch but not tempo) using a WSOLA
algorithm.
shift gives the key 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 pitch for a similar effect.
ladspa module [plugin] [argument...]
Apply a LADSPA [5] (Linux Audio Developer's Simple Plugin API)
plugin. Despite the name, LADSPA is not Linux-specific, and a
wide range of effects is available as LADSPA plugins, such as
cmt [6] (the Computer Music Toolkit) and Steve Harris's plugin
collection [7]. The first argument is the plugin module, the
second the name of the plugin (a module can contain more than
one plugin) and any other arguments are for the control ports of
the plugin. Missing arguments are supplied by default values if
possible. Only plugins with at most one audio input and one
audio output port can be used. If found, the environment vari-
ble LADSPA_PATH will be used as search path for plugins.
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]]]" {xover-freq[k]
"attack1,..."}
The multi-band compander is similar to the single-band compander
but the audio is first divided into bands using Butterworth
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
xover-freq; these can be repeated to create multiple bands.
For example, the following (one long) command shows how multi-
band companding is typically used in FM radio:
play track1.wav gain -3 filter 8000- 32 100 mcompand \
"0.005,0.1 -47,-40,-34,-34,-17,-33" 100 \
"0.003,0.05 -47,-40,-34,-34,-17,-33" 400 \
"0.000625,0.0125 -47,-40,-34,-34,-15,-33" 1600 \
"0.0001,0.025 -47,-40,-34,-34,-31,-31,-0,-30" 6400 \
"0,0.025 -38,-31,-28,-28,-0,-25" \
gain 15 highpass 22 highpass 22 filter -17500 256 \
gain 9 lowpass -1 17801
The audio file is played with a simulated FM radio sound (or
broadcast signal condition if the lowpass filter at the end is
skipped). Note that the pipeline is set up with US-style 75us
preemphasis.
See also compand for a single-band companding effect.
mixer [ -l|-r|-f|-b|-1|-2|-3|-4|n{,n} ]
Reduce the number of audio channels by mixing or selecting chan-
nels, or increase the number of channels by duplicating chan-
nels. Note: this effect operates on the audio channels within
the SoX effects processing chain; it should not be confused with
the -m global option (where multiple files are mix-combined
before entering the effects chain).
This effect is automatically used when the number of input chan-
nels differ from the number of output channels. When reducing
the number of channels it is possible to manually specify the
mixer effect and use the -l, -r, -f, -b, -1, -2, -3, -4, options
to select only the left, right, front, back channel(s) or spe-
cific channel for the output instead of averaging the channels.
The -l, and -r options will do averaging in quad-channel files
so select the exact channel to prevent this.
The mixer effect can also be invoked with up to 16 numbers, sep-
arated by commas, which specify the proportion (0 = 0% and 1 =
100%) of each input channel that is to be mixed into each output
channel. In two-channel mode, 4 numbers are given: l -> l, l ->
r, r -> l, and r -> r, respectively. In four-channel mode, the
first 4 numbers give the proportions for the left-front output
channel, as follows: lf -> lf, rf -> lf, lb -> lf, and rb -> rf.
The next 4 give the right-front output in the same order, then
left-back and right-back.
It is also possible to use the 16 numbers to expand or reduce
the channel count; just specify 0 for unused channels.
Finally, certain reduced combination of numbers can be specified
for certain input/output channel combinations.
+----------------------------------------------------------+
|In Ch Out Ch Num Mappings |
| 2 1 2 l -> l, r -> l |
| 2 2 1 adjust balance |
| 4 1 4 lf -> l, rf -> l, lb -> l, rb -> l |
| 4 2 2 lf -> l&rf -> r, lb -> l&rb -> r |
| 4 4 1 adjust balance |
| 4 4 2 front balance, back balance |
+----------------------------------------------------------+
See also remix for a mixing effect that handles any number of
channels.
noiseprof [profile-file]
Calculate a profile of the audio for use in noise reduction.
See the description of the noisered effect for details.
noisered [profile-file [amount]]
Reduce noise in the audio signal by profiling and filtering.
This effect is moderately effective at removing consistent back-
ground noise such as hiss or hum. To use it, first run SoX with
the noiseprof effect on a section of audio that ideally would
contain silence but in fact contains noise - such sections are
typically found at the beginning or the end of a recording.
noiseprof will write out a noise profile to profile-file, or to
stdout if no profile-file or if `-' is given. E.g.
sox speech.au -n trim 0 1.5 noiseprof speech.noise-profile
To actually remove the noise, run SoX again, this time with the
noisered effect; noisered will reduce noise according to a noise
profile (which was generated by noiseprof), from profile-file,
or from stdin if no profile-file or if `-' is given. E.g.
sox speech.au cleaned.au noisered speech.noise-profile 0.3
How much noise should be removed is specified by amount-a number
between 0 and 1 with a default of 0.5. Higher numbers will
remove more noise but present a greater likelihood of removing
wanted components of the audio signal. Before replacing an
original recording with a noise-reduced version, experiment with
different amount values to find the optimal one for your audio;
use headphones to check that you are happy with the results,
paying particular attention to quieter sections of the audio.
On most systems, the two stages - profiling and reduction - can
be combined using a pipe, e.g.
sox noisy.au -n trim 0 1 noiseprof | play noisy.au noisered
norm [-i] [level]
Normalise audio to 0dB FSD or to a given level relative to 0dB.
Requires temporary file space to store the audio to be nor-
malised.
To create a normalised copy of an audio file,
sox infile outfile norm
can be used, though note that if `infile' has a simple encoding
(e.g. PCM), then
sox infile outfile vol `sox infile -n stat -v 2>&1`
(on systems that support this construct) might be quicker to
execute (though perhaps not to type!) as it doesn't require a
temporary file.
For a more complex example, suppose that `effect1' performs some
unknown or unpredictable attenuation and that `effect2' requires
up to 10dB of headroom, then
sox infile outfile effect1 norm -10 effect2 norm
gives both effect2 and the output file the highest possible sig-
nal levels.
Normally, audio is normalised based on the level of the channel
with the highest peak level, which means that whilst all chan-
nels are adjusted, only one channel attains the normalised
level. If the -i option is given, then each channel is treated
individually and will attain the normalised level.
In most cases, norm -3 should be the maximum level used at the
output file (to leave headroom for playback-resampling, etc.).
See also the discussions of clipping and Replay Gain in sox(1).
oops Out Of Phase Stereo effect. Mixes stereo to twin-mono where
each mono channel contains the difference between the left and
right stereo channels. This is sometimes known as the `karaoke'
effect as it often has the effect of removing most or all of the
vocals from a recording.
pad { length[@position] }
Pad the audio with silence, at the beginning, the end, or any
specified points through the audio. Both length and position
can specify a time or, if appended with an `s', a number of sam-
ples. length is the amount of silence to insert and position
the position in the input audio stream at which to insert it.
Any number of lengths and positions may be specified, provided
that a specified position is not less that the previous one.
position is optional for the first and last lengths specified
and if omitted correspond to the beginning and the end of the
audio respectively. For example, pad 1.5 1.5 adds 1.5 seconds
of silence padding at each end of the audio, whilst pad
4000s@3:00 inserts 4000 samples of silence 3 minutes into the
audio. If silence is wanted only at the end of the audio, spec-
ify either the end position or specify a zero-length pad at the
start.
pan direction
Pan the audio from one channel to another. This is done by
changing the volume of the input channels so that it fades out
on one channel and fades-in on another. If the number of input
channels is different then the number of output channels then
this effect tries to intelligently handle this. For instance,
if the input contains 1 channel and the output contains 2 chan-
nels, then it will create the missing channel itself. The
direction is a value from -1 to 1. -1 represents far left and 1
represents far right. Numbers in between will start the pan
effect without totally muting the opposite channel.
phaser gain-in gain-out delay decay speed [-s|-t]
Add a phasing effect to the audio. See [3] for a detailed
description of phasing.
delay/decay/speed gives the delay in milliseconds and the decay
(relative to gain-in) with a modulation speed in Hz. The modu-
lation is either sinusoidal (-s) - preferable for multiple
instruments, or triangular (-t) - gives single instruments a
sharper phasing effect. The decay should be less than 0.5 to
avoid feedback, and usually no less than 0.1. Gain-out is the
volume of the output.
For example:
play snare.flac phaser 0.8 0.74 3 0.4 0.5 -t
Gentler:
play snare.flac phaser 0.9 0.85 4 0.23 1.3 -s
A popular sound:
play snare.flac phaser 0.89 0.85 1 0.24 2 -t
More severe:
play snare.flac phaser 0.6 0.66 3 0.6 2 -t
rate [-q|-l|-m|-h|-v] [RATE[k]]
Change the audio sampling rate (i.e. resample the audio) using a
quality level as follows:
+------------------------------------------------------------+
| Quality BW % Rej dB Typical Use |
|-q quick & dirty n/a ~=30 @ Fs/4 playback on |
| ancient hardware |
|-l low 80 100 playback on old |
| hardware |
|-m medium 99 100 audio playback |
|-h high 99 125 16-bit mastering |
| (use with dither) |
|-v very high 99 175 24-bit mastering |
+------------------------------------------------------------+
where BW % is the percentage of the audio band that is preserved
(based on the 3dB point) during sample rate conversion, and Rej
dB is the level of noise rejection. The default quality level
is `high' (-h). The -q algorithm uses cubic interpolation; the
others use linear-phase bandwidth-limited interpolation.
This effect is invoked automatically if SoX's -r option speci-
fies a rate that is different to that of the input file(s).
Alternatively, this effect may be invoked with the output rate
parameter RATE and SoX's -r option need not be given. For exam-
ple, the following two commands are equivalent:
sox input.au -r 48k output.au bass -3
sox input.au output.au bass -3 rate 48k
though the second command is more flexible as it allows a rate
quality option to be given, and it allows the effects to be
ordered arbitrarily.
See also resample, polyphase and rabbit for other sample-rate
changing effects.
remix [-a|-m|-p] <out-spec>
out-spec = in-spec{,in-spec} | 0
in-spec = [in-chan][-[in-chan2]][vol-spec]
vol-spec = p|i|v[volume]
Select and mix input audio channels into output audio channels.
Each output channel is specified, in turn, by a given out-spec:
a list of contributing input channels and volume specifications.
Note that this effect operates on the audio channels within the
SoX effects processing chain; it should not be confused with the
-m global option (where multiple files are mix-combined before
entering the effects chain).
An out-spec contains comma-separated input channel-numbers and
hyphen-delimited channel-number ranges; alternatively, 0 may be
given to create a silent output channel. For example,
sox input.au output.au remix 6 7 8 0
creates an output file with four channels, where channels 1, 2,
and 3 are copies of channels 6, 7, and 8 in the input file, and
channel 4 is silent. Whereas
sox input.au output.au remix 1-3,7 3
creates a stereo output file where the left channel is a mix-
down of input channels 1, 2, 3, and 7, and the right channel is
a copy of input channel 3.
Where a range of channels is specified, the channel numbers to
the left and right of the hyphen are optional and default to 1
and to the number of input channels respectively. Thus
sox input.au output.au remix -
performs a mix-down of all input channels to mono.
By default, where an output channel is mixed from multiple (n)
input channels, each input channel will be scaled by a factor of
1/n. Custom mixing volumes can be set by following a given
input channel or range of input channels with a vol-spec (volume
specification). This is one of the letters p, i, or v, followed
by a volume number, the meaning of which depends on the given
letter and is defined as follows:
Letter Volume number Notes
p power adjust in dB 0 = no change
i power adjust in dB As `p', but invert
the audio
v voltage multiplier 1 = no change, 0.5
~= 6dB attenuation,
2 ~= 6dB gain, -1 =
invert
If an out-spec includes at least one vol-spec then, by default,
1/n scaling is not applied to any other channels in the same
out-spec (though may be in other out-specs). The -a (automatic)
option however, can be given to retain the automatic scaling in
this case. For example,
sox input.au output.au remix 1,2 3,4v0.8
results in channel level multipliers of 0.5,0.5 1,0.8, whereas
sox input.au output.au remix -a 1,2 3,4v0.8
results in channel level multipliers of 0.5,0.5 0.5,0.8.
The -m (manual) option disables all automatic volume adjust-
ments, so
sox input.au output.au remix -m 1,2 3,4v0.8
results in channel level multipliers of 1,1 1,0.8.
The volume number is optional and omitting it corresponds to no
volume change; however, the only case in which this is useful is
in conjunction with i. For example, if input.au is stereo, then
sox input.au output.au remix 1,2i
is a mono equivalent of the oops effect.
If the -p option is given, then any automatic 1/n scaling is
replaced by 1/\/n (`power') scaling; this gives a louder mix but
one that might occasionally clip.
* * *
One typical 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 individ-
ual audio channels). Where more than a few channels are
involved, a script such as the following is useful:
#!/bin/sh # This is a Bourne shell script
chans=`soxi -c "$1"`
while [ $chans -ge 1 ]; do
chans0=`printf %02i $chans` # 2 digits hence up to 99 chans
out=`echo "$1"|sed "s/\(.*\)\.\(.*\)/\1-$chans0.\2/"`
sox "$1" "$out" remix $chans
chans=`expr $chans - 1`
done
If a file input.au containing six audio channels were given, the
script would produce six output files: input-01.au, input-02.au,
..., input-06.au.
See also mixer and swap for similar effects.
repeat count
Repeat the entire audio count times. Requires temporary file
space to store the audio to be repeated. Note that repeating
once yields two copies: the original audio and the repeated
audio.
reverb [-w|--wet-only] [reverberance (50%) [HF-damping (50%)
[room-scale (100%) [stereo-depth (100%)
[pre-delay (0ms) [wet-gain (0dB)]]]]]]
Add reverberation to the audio using the `freeverb' algorithm.
A reverberation effect is sometimes desirable for concert halls
that are too small or contain so many people that the hall's
natural reverberance is diminished. Applying a small amount of
stereo reverb to a (dry) mono signal will usually make it sound
more natural. See [3] for a detailed description of reverbera-
tion.
Note that this effect increases both the volume and the length
of the audio, so to prevent clipping in these domains, a typical
invocation might be:
play dry.au gain -3 pad 0 3 reverb
reverse
Reverse the audio completely. Requires temporary file space to
store the audio to be reversed.
silence [-l] above-periods [duration
threshold[d|%] [below-periods duration threshold[d|%]]
Removes silence from the beginning, middle, or end of the audio.
Silence is anything below a specified threshold.
The above-periods value is used to indicate if audio should be
trimmed at the beginning of the audio. A value of zero indicates
no silence should be trimmed from the beginning. When specifying
an non-zero above-periods, it trims audio up until it finds non-
silence. Normally, when trimming silence from beginning of audio
the above-periods will be 1 but it can be increased to higher
values to trim all audio up to a specific count of non-silence
periods. For example, if you had an audio file with two songs
that each contained 2 seconds of silence before the song, you
could specify an above-period of 2 to strip out both silence
periods and the first song.
When above-periods is non-zero, you must also specify a duration
and threshold. Duration indications the amount of time that non-
silence must be detected before it stops trimming audio. By
increasing the duration, burst of noise can be treated as
silence and trimmed off.
Threshold is used to indicate what sample value you should treat
as silence. For digital audio, a value of 0 may be fine but for
audio recorded from analog, you may wish to increase the value
to account for background noise.
When optionally trimming silence from the end of the audio, you
specify a below-periods count. In this case, below-period means
to remove all audio after silence is detected. Normally, this
will be a value 1 of but it can be increased to skip over peri-
ods of silence that are wanted. For example, if you have a song
with 2 seconds of silence in the middle and 2 second at the end,
you could set below-period to a value of 2 to skip over the
silence in the middle of the audio.
For below-periods, duration specifies a period of silence that
must exist before audio is not copied any more. By specifying a
higher duration, silence that is wanted can be left in the
audio. For example, if you have a song with an expected 1 sec-
ond of silence in the middle and 2 seconds of silence at the
end, a duration of 2 seconds could be used to skip over the mid-
dle silence.
Unfortunately, you must know the length of the silence at the
end of your audio file to trim off silence reliably. A work
around is to use the silence effect in combination with the
reverse effect. By first reversing the audio, you can use the
above-periods to reliably trim all audio from what looks like
the front of the file. Then reverse the file again to get back
to normal.
To remove silence from the middle of a file, specify a below-
periods that is negative. This value is then treated as a posi-
tive value and is also used to indicate the effect should
restart processing as specified by the above-periods, making it
suitable for removing periods of silence in the middle of the
audio.
The option -l indicates that below-periods duration length of
audio should be left intact at the beginning of each period of
silence. For example, if you want to remove long pauses between
words but do not want to remove the pauses completely.
The period counts are in units of samples. Duration counts may
be in the format of hh:mm:ss.frac, or the exact count of sam-
ples. Threshold numbers may be suffixed with d to indicate the
value is in decibels, or % to indicate a percentage of maximum
value of the sample value (0% specifies pure digital silence).
The following example shows how this effect can be used to start
a recording that does not contain the delay at the start which
usually occurs between `pressing the record button' and the
start of the performance:
rec parameters filename other-effects silence 1 5 2%
speed factor[c]
Adjust the audio speed (pitch and tempo together). factor is
either the ratio of the new speed to the old speed: greater than
1 speeds up, less than 1 slows down, or, if appended with the
letter `c', the number of cents (i.e. 100ths of a semitone) by
which the pitch (and tempo) should be adjusted: greater than 0
increases, less than 0 decreases.
By default, the speed change is performed by resampling with the
rate effect using its default quality/speed. For higher quality
or higher speed resampling, in addition to the speed effect,
specify the rate effect with the desired quality option.
spectrogram [options]
Create a spectrogram of the audio. This effect is optional;
type sox --help and check the list of supported effects to see
if it has been included.
The spectrogram is rendered in a Portable Network Graphic (PNG)
file, and shows time in the X-axis, frequency in the Y-axis, and
audio signal magnitude in the Z-axis. Z-axis values are repre-
sented by the colour (or intensity) of the pixels in the X-Y
plane.
This effect supports only one channel; for multi-channel input
files, use either SoX's -c 1 option with the output file (to
obtain a spectrogram on the mix-down), or the remix n effect to
select a particular channel. Be aware though, that both of
these methods affect the audio in the effects chain.
-x num X-axis pixels/second, default 100. This controls the
width of the spectrogram; num can be from 1 (low time
resolution) to 5000 (high time resolution) and need not
be an integer. SoX may make a slight adjustment to the
given number for processing quantisation reasons; if so,
SoX will report the actual number used (viewable when
--verbose is in effect).
The maximum width of the spectrogram is 999 pixels; if
the audio length and the given -x number are such that
this would be exceeded, then the spectrogram (and the
effects chain) will be truncated. To move the spectro-
gram to a point later in the audio stream, first invoke
the trim effect; e.g.
sox audio.ogg -n trim 1:00 spectrogram
starts the spectrogram at 1 minute through the audio.
-y num Y-axis resolution (1 - 4), default 2. This controls the
height of the spectrogram; num can be from 1 (low fre-
quency resolution) to 4 (high frequency resolution). For
values greater than 2, the resulting image may be too
tall to display on the screen; if so, a graphic manipula-
tion package (such as ImageMagick(1)) can be used to re-
size the image.
To increase the frequency resolution without increasing
the height of the spectrogram, the rate effect may be
invoked to reduce the sampling rate of the signal before
invoking spectrogram; e.g.
sox audio.ogg -r 4k -n rate spectrogram
allows detailed analysis of frequencies up to 2kHz (half
the sampling rate).
-z num Z-axis (colour) range in dB, default 120. This sets the
dynamic-range of the spectrogram to be -num dBFS to
0 dBFS. Num may range from 20 to 180. Decreasing
dynamic-range effectively increases the `contrast' of the
spectrogram display, and vice versa.
-Z num Sets the upper limit of the Z-axis in dBFS. A negative
num effectively increases the `brightness' of the spec-
trogram display, and vice versa.
-q num Sets the Z-axis quantisation, i.e. the number of differ-
ent colours (or intensities) in which to render Z-axis
values. A small number (e.g. 4) will give a
`poster'-like effect making it easier to discern magni-
tude bands of similar level. Smaller numbers also usu-
ally result in smaller PNG files. The number given spec-
ifies the number of colours to use inside the Z-axis
range; two colours are reserved to represent out-of-range
values.
-w name
Window: Hann (default), Hamming, Bartlett, Rectangular or
Kaiser. The spectrogram is produced using the Discrete
Fourier Transform (DFT) algorithm. A significant parame-
ter to this algorithm is the choice of `window function'.
By default, SoX uses the Hann window which has good all-
round frequency-resolution and dynamic-range properties.
For better frequency resolution (but lower dynamic-
range), select a Hamming window; for higher dynamic-range
(but poorer frequency-resolution), select a Kaiser win-
dow. Bartlett and Rectangular windows are also avail-
able. Selecting a window other than Hann will usually
require a corresponding -z setting.
-s Allow slack overlapping of DFT windows. This can, in
some cases, increase image sharpness and give greater
adherence to the -x value, but at the expense of a little
spectral loss.
-m Creates a monochrome spectrogram (the default is colour).
-h Selects a high-colour palette - less visually pleasing
than the default colour palette, but it may make it eas-
ier to differentiate different levels. If this option is
used in conjunction with -m, the result will be a hybrid
monochrome/colour palette.
-p num Permute the colours in a colour or hybrid palette. The
num parameter (from 1 to 6) selects the permutation.
-l Creates a `printer friendly' spectrogram with a light
background (the default has a dark background).
-a Suppress the display of the axis lines. This is some-
times useful in helping to discern artefacts at the spec-
trogram edges.
-t text
Set the image title - text to display above the spectro-
gram.
-c text
Set the image comment - text to display below and to the
left of the spectrogram.
-o text
Name of the spectrogram output PNG file, default `spec-
trogram.png'.
For example, let's see what the spectrogram of a swept triangu-
lar wave looks like:
sox -n -n synth 6 tri 10k:14k spectrogram -z 100 -w k
For the ability to perform off-line processing of spectral data,
see the stat effect.
splice { position[,excess[,leeway]] }
Splice together audio sections. This effect provides two things
over simple audio concatenation: a (usually short) cross-fade is
applied at the join, and a wave similarity comparison is made to
help determine the best place at which to make the join.
To perform a splice, first use the trim effect to select the
audio sections to be joined together. As when performing a tape
splice, the end of the section to be spliced onto should be
trimmed with a small excess (default 0.005 seconds) of audio
after the ideal joining point. The beginning of the audio sec-
tion to splice on should be trimmed with the same excess (before
the ideal joining point), plus an additional leeway (default
0.005 seconds). SoX should then be invoked with the two audio
sections as input files and the splice effect given with the
position at which to perform the splice - this is length of the
first audio section (including the excess).
For example, a long song begins with two verses which start (as
determined e.g. by using the play command with the trim (start)
effect) at times 0:30.125 and 1:03.432. The following commands
cut out the first verse:
sox too-long.au part1.au trim 0 30.130
(5 ms excess, after the first verse starts)
sox long.au part2.au trim 1:03.422
(5 ms excess plus 5 ms leeway, before the second verse starts)
sox part1.au part2.au just-right.au splice 30.130
Provided your arithmetic is good enough, multiple splices can be
performed with a single splice invocation. For example:
#!/bin/sh
# Audio Copy and Paste Over
# acpo infile copy-start copy-stop paste-over-start outfile
# All times measured in samples.
rate=`soxi -r "$1"`
e=`expr $rate '*' 5 / 1000` # Using default excess
l=$e # and leeway.
sox "$1" piece.au trim `expr $2 - $e - $l`s \
`expr $3 - $2 + $e + $l + $e`s
sox "$1" part1.au trim 0 `expr $4 + $e`s
sox "$1" part2.au trim `expr $4 + $3 - $2 - $e - $l`s
sox part1.au piece.au part2.au "$5" splice \
`expr $4 + $e`s \
`expr $4 + $e + $3 - $2 + $e + $l + $e`s
In the above Bourne shell script, two splices are used to `copy
and paste' audio.
* * *
It is also possible to use this effect to perform general cross-
fades, e.g. to join two songs. In this case, excess would typi-
cally be an number of seconds, and leeway should be set to zero.
stat [-s n] [-rms] [-freq] [-v] [-d]
Do a statistical check on the input file, and print results on
the standard error file. Audio is passed unmodified through the
SoX processing chain.
The `Volume Adjustment:' field in the statistics gives you the
parameter to the -v number which will make the audio as loud as
possible without clipping. Note: See the discussion on Clipping
in sox(1) for reasons why it is rarely a good idea to actually
do this.
The option -v will print out the `Volume Adjustment:' field's
value only and return. This could be of use in scripts to auto
convert the volume.
The -s option is used to scale the input data by a given factor.
The default value of n is the maximum value of a signed long
integer (7fffffff in hexadecimal). 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 -freq option calculates the input's power spectrum and
prints it to standard error.
There is also an optional parameter -d that will print out a hex
dump of the audio from the internal buffer that is in 32-bit
signed PCM data. This is mainly only of use in tracking down
endian problems that creep in to SoX on cross-platform versions.
swap [1 2 | 1 2 3 4]
Swap channels in multi-channel audio files. Optionally, you may
specify the channel order you would like the output in. This
defaults to output channel 2 and then 1 for stereo and 2, 1, 4,
3 for quad-channels. An interesting feature is that you may
duplicate a given channel by overwriting another. This is done
by repeating an output channel on the command-line. For exam-
ple, swap 2 2 will overwrite channel 1 with channel 2; creating
a stereo file with both channels containing the same audio.
See also the remix effect.
synth [len] {[type] [combine] [[%]freq[k][:|+|/|-[%]freq2[k]]] [off]
[ph] [p1] [p2] [p3]}
This effect can be used to generate fixed or swept frequency
audio tones with various wave shapes, or to generate wide-band
noise of various `colours'. Multiple synth effects can be cas-
caded to produce more complex waveforms; at each stage it is
possible to choose whether the generated waveform will be mixed
with, or modulated onto the output from the previous stage.
Audio for each channel in a multi-channel audio file can be syn-
thesised independently.
Though this effect is used to generate audio, an input file must
still be given, the characteristics of which will be used to set
the synthesised audio length, the number of channels, and the
sampling rate; however, since the input file's audio is not nor-
mally needed, a `null file' (with the special name -n) is often
given instead (and the length specified as a parameter to synth
or by another given effect that can has an associated length).
For example, the following produces a 3 second, 48kHz, audio
file containing a sine-wave swept from 300 to 3300 Hz:
sox -n output.au synth 3 sine 300-3300
and this produces an 8 kHz version:
sox -r 8000 -n output.au synth 3 sine 300-3300
Multiple channels can be synthesised by specifying the set of
parameters shown between braces multiple times; the following
puts the swept tone in the left channel and adds `brown' noise
in the right:
sox -n output.au synth 3 sine 300-3300 brownnoise
The following example shows how two synth effects can be cas-
caded to create a more complex waveform:
sox -n output.au synth 0.5 sine 200-500 \
synth 0.5 sine fmod 700-100
Frequencies can also be given as a number of musical semitones
relative to `middle A' (440 Hz) by prefixing a `%' character;
for example, the following could be used to help tune a guitar's
`E' strings:
play -n synth sine %-17
N.B. This effect generates audio at maximum volume (0dBFS),
which means that there is a high chance of clipping when using
the audio subsequently, so in most cases, you will want to fol-
low this effect with the gain effect to prevent this from hap-
pening. (See also Clipping in sox(1).)
A detailed description of each synth parameter follows:
len is the length of audio to synthesise expressed as a time or
as a number of samples; 0=inputlength, default=0.
The format for specifying lengths in time is hh:mm:ss.frac. The
format for specifying sample counts is the number of samples
with the letter `s' appended to it.
type is one of sine, square, triangle, sawtooth, trapezium, exp,
[white]noise, pinknoise, brownnoise; default=sine
combine is one of create, mix, amod (amplitude modulation), fmod
(frequency modulation); default=create
freq/freq2 are the frequencies at the beginning/end of synthesis
in Hz or, if preceded with `%', semitones relative to A
(440 Hz); for both, default=%0. If freq2 is given, then len
must also have been given and the generated tone will be swept
between the given frequencies. The two given frequencies must
be separated by one of the characters `:', `+', `/', or `-'.
This character is used to specify the sweep function as follows:
: Linear: the tone will change by a fixed number of hertz
per second.
+ Square: a second-order function is used to change the
tone.
/ Exponential: the tone will change by a fixed number of
semitones per second.
- Exponential: as `/', but initial phase always zero, and
stepped (less smooth) frequency changes.
Not used for noise.
off is the bias (DC-offset) of the signal in percent; default=0.
ph is the phase shift in percentage of 1 cycle; default=0. Not
used for noise.
p1 is the percentage of each cycle that is `on' (square), or
`rising' (triangle, exp, trapezium); default=50 (square, trian-
gle, exp), default=10 (trapezium).
p2 (trapezium): the percentage through each cycle at which
`falling' begins; default=50. exp: the amplitude in percent;
default=100.
p3 (trapezium): the percentage through each cycle at which
`falling' ends; default=60.
tempo [-q] factor [segment [search [overlap]]]
Change the audio tempo (but not its pitch) using a `WSOLA' algo-
rithm. 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 overlap-
ping points; if the optional -q parameter is given, tree
searches are used instead, giving a quicker, but possibly lower
quality, result.
factor gives the ratio of new tempo to the old tempo, so e.g.
1.1 speeds up the tempo by 10%, and 0.9 slows it down by 10%.
The optional segment parameter selects the algorithm's segment
size in milliseconds. The default value is 82 and is typically
suited to making small changes to the tempo of music; for larger
changes (e.g. a factor of 2), 50 ms may give a better result.
When changing the tempo of speech, a segment size of around
30 ms often works well.
The optional search parameter gives the audio length in mil-
liseconds (default 14) over which the algorithm will search for
overlapping points. Larger values use more processing time and
do not necessarily produce better results.
The optional overlap parameter gives the segment overlap length
in milliseconds (default 12).
See also stretch for a similar effect, speed for an effect that
changes tempo and key together, and key for an effect that
changes key without changing tempo.
treble gain [frequency[k] [width[s|h|k|o|q]]]
Apply a treble tone-control effect. See the description of the
bass effect for details.
tremolo speed [depth]
Apply a tremolo (low frequency amplitude modulation) effect to
the audio. The tremolo frequency in Hz is given by speed, and
the depth as a percentage by depth (default 40).
Note: This effect is a special case of the synth effect.
trim start [length]
Trim can trim off unwanted audio from the beginning and end of
the audio. Audio is not sent to the output stream until the
start location is reached.
The optional length parameter tells the number of samples to
output after the start sample and is used to trim off the back
side of the audio. Using a value of 0 for the start parameter
will allow trimming off the back side only.
Both options can be specified using either an amount of time or
an exact count of samples. The format for specifying lengths in
time is hh:mm:ss.frac. A start value of 1:30.5 will not start
until 1 minute, thirty and 1/2 seconds into the audio. The for-
mat for specifying sample counts is the number of samples with
the letter `s' appended to it. A value of 8000s will wait until
8000 samples are read before starting to process audio.
vol gain [type [limitergain]]
Apply an amplification or an attenuation to the audio signal.
Unlike the -v option (which is used for balancing multiple input
files as they enter the SoX effects processing chain), vol is an
effect like any other so can be applied anywhere, and several
times if necessary, during the processing chain.
The amount to change the volume is given by gain which is inter-
preted, according to the given type, as follows: if type is
amplitude (or is omitted), then gain is an amplitude (i.e. volt-
age or linear) ratio, if power, then a power (i.e. wattage or
voltage-squared) ratio, and if dB, then a power change in dB.
When type is amplitude or power, a gain of 1 leaves the volume
unchanged, less than 1 decreases it, and greater than 1
increases it; a negative gain inverts the audio signal in addi-
tion to adjusting its volume.
When type is dB, a gain of 0 leaves the volume unchanged, less
than 0 decreases it, and greater than 0 increases it.
See [4] for a detailed discussion on electrical (and hence audio
signal) voltage and power ratios.
Beware of Clipping when the increasing the volume.
The gain and the type parameters can be concatenated if desired,
e.g. vol 10dB.
An optional limitergain value can be specified and should be a
value much less than 1 (e.g. 0.05 or 0.02) and is used only on
peaks to prevent clipping. Not specifying this parameter will
cause no limiter to be used. In verbose mode, this effect will
display the percentage of the audio that needed to be limited.
See also compand for a dynamic-range compression/expansion/lim-
iting effect.
Deprecated Effects
The following effects have been renamed or have their functionality
included in another effect; they continue to work in this version of
SoX but may be removed in future.
pitch shift [width interpolate fade]
Change the audio pitch (but not its duration). This effect is
equivalent to the key effect with search set to zero, so its
results are comparatively poor; it is retained for backwards
compatibility only.
Change by cross-fading shifted samples. shift is given in
cents. Use a positive value to shift to treble, negative value
to shift to bass. Default shift is 0. width of window is in
ms. Default width is 20ms. Try 30ms to lower pitch, and 10ms
to raise pitch. interpolate option, can be cubic or linear.
Default is cubic. The fade option, can be cos, hamming, linear
or trapezoid; the default is cos.
polyphase [-w nut|ham] [-width n] [-cut-off c]
Change the sampling rate using `polyphase interpolation', a DSP
algorithm. polyphase copes with only certain rational fraction
resampling ratios, and, compared with the rate effect, is gener-
ally slow, memory intensive, and has poorer stop-band rejection.
If the -w parameter is nut, then a Nuttall (~90 dB stop-band)
window will be used; ham selects a Hamming (~43 dB stop-band)
window. The default is Nuttall.
The -width parameter specifies the (approximate) width of the
filter. The default is 1024 samples, which produces reasonable
results.
The -cut-off value (c) specifies the filter cut-off frequency in
terms of fraction of frequency bandwidth, also know as the
Nyquist frequency. See the resample effect for further informa-
tion on Nyquist frequency. If up-sampling, then this is the
fraction of the original signal that should go through. If
down-sampling, this is the fraction of the signal left after
down-sampling. The default is 0.95.
See also rate, rabbit and resample for other sample-rate chang-
ing effects.
rabbit [-c0|-c1|-c2|-c3|-c4]
Change the sampling rate using libsamplerate, also known as
`Secret Rabbit Code'. This effect is optional and, due to
licence issues, is not included in all versions of SoX. Com-
pared with the rate effect, rabbit is very slow.
See http://www.mega-nerd.com/SRC for details of the algorithms.
Algorithms 0 through 2 are progressively faster and lower qual-
ity versions of the sinc algorithm; the default is -c0. Algo-
rithm 3 is zero-order hold, and 4 is linear interpolation.
See also rate, polyphase and resample for other sample-rate
changing effects, and see resample for more discussion of resam-
pling.
resample [-qs|-q|-ql] [rolloff [beta]]
Change the sampling rate using simulated analog filtration.
Compared with the rate effect, resample is slow, and has poorer
stop-band rejection. Only the low quality option works with all
resampling ratios.
By default, linear interpolation of the filter coefficients is
used, with a window width about 45 samples at the lower of the
two rates. This gives an accuracy of about 16 bits, but insuf-
ficient stop-band rejection in the case that you want to have
roll-off greater than about 0.8 of the Nyquist frequency.
The -q* options will change the default values for roll-off and
beta as well as use quadratic interpolation of filter coeffi-
cients, resulting in about 24 bits precision. The -qs, -q, or
-ql options specify increased accuracy at the cost of lower exe-
cution speed. It is optional to specify roll-off and beta
parameters when using the -q* options.
Following is a table of the reasonable defaults which are built-
in to SoX:
+--------------------------------------------------+
|Option Window Roll-off Beta Interpolation |
|(none) 45 0.80 16 linear |
| -qs 45 0.80 16 quadratic |
| -q 75 0.875 16 quadratic |
| -ql 149 0.94 16 quadratic |
+--------------------------------------------------+
-qs, -q, or -ql use window lengths of 45, 75, or 149 samples,
respectively, at the lower sample-rate of the two files. This
means progressively sharper stop-band rejection, at proportion-
ally slower execution times.
rolloff refers to the cut-off frequency of the low pass filter
and is given in terms of the Nyquist frequency for the lower
sample rate. rolloff therefore should be something between 0
and 1, in practise 0.8-0.95. The defaults are indicated above.
The Nyquist frequency is equal to half the sample rate. Logi-
cally, this is because the A/D converter needs at least 2 sam-
ples to detect 1 cycle at the Nyquist frequency. Frequencies
higher then the Nyquist will actually appear as lower frequen-
cies to the A/D converter and is called aliasing. Normally, A/D
converts run the signal through a lowpass filter first to avoid
these problems.
Similar problems will happen in software when reducing the sam-
ple rate of an audio file (frequencies above the new Nyquist
frequency can be aliased to lower frequencies). Therefore, a
good resample effect will remove all frequency information above
the new Nyquist frequency.
The rolloff refers to how close to the Nyquist frequency this
cut-off is, with closer being better. When increasing the sam-
ple rate of an audio file you would not expect to have any fre-
quencies exist that are past the original Nyquist frequency.
Because of resampling properties, it is common to have aliasing
artifacts created above the old Nyquist frequency. In that case
the rolloff refers to how close to the original Nyquist fre-
quency to use a highpass filter to remove these artifacts, with
closer also being better.
The beta, if unspecified, defaults to 16. This selects a Kaiser
window. You can select a Nuttall window by specifying anything
<= 2 here. For more discussion of beta, look under the filter
effect.
Default parameters are, as indicated above, Kaiser window of
length 45, roll-off 0.80, beta 16, linear interpolation.
Note: -qs is only slightly slower, but more accurate for 16-bit
or higher precision.
Note: In many cases of up-sampling, no interpolation is needed,
as exact filter coefficients can be computed in a reasonable
amount of space. To be precise, this is done when both input-
rate < output-rate, and output-rate -:- gcd(input-rate, output-
rate) <= 511.
See also rate, polyphase and rabbit for other sample-rate chang-
ing effects. There is a detailed analysis of resample and
polyphase at http://leute.server.de/wilde/resample.html; see
rabbit for a pointer to its own documentation.
stretch factor [window fade shift fading]
Change the audio duration (but not its pitch). This effect is
equivalent to the tempo effect with (factor inverted and) search
set to zero, so its results are comparatively poor; it is
retained for backwards compatibility only.
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
sox(1), soxi(1), soxformat(7), libsox(3),
The SoX web page at http://sox.sourceforge.net
SoX scripting examples at http://sox.sourceforge.net/Docs/Scripts
References
[1] R. Bristow-Johnson, Cookbook formulae for audio EQ biquad filter
coefficients, http://musicdsp.org/files/Audio-EQ-Cookbook.txt
[2] Wikipedia, Q-factor, http://en.wikipedia.org/wiki/Q_factor
[3] Scott Lehman, Effects Explained, http://harmony-cen-
tral.com/Effects/effects-explained.html
[4] Wikipedia, Decibel, http://en.wikipedia.org/wiki/Decibel
[5] Richard Furse, Linux Audio Developer's Simple Plugin API,
http://www.ladspa.org
[6] Richard Furse, Computer Music Toolkit, http://www.ladspa.org/cmt
[7] Steve Harris, LADSPA plugins, http://plugin.org.uk
AUTHORS
Chris Bagwell (cbagwell@users.sourceforge.net). Other authors and con-
tributors are listed in the AUTHORS file that is distributed with the
source code.
soxeffect July 27, 2008 SoX(7)
soxeffect 14.1.0 - Generated Tue Aug 26 09:14:38 CDT 2008