Provided by:
sox_14.0.1-2build2_i386 
NAME
SoX - Sound eXchange, the Swiss Army knife of audio manipulation
DESCRIPTION
SOX EFFECTS
Multiple effects may be applied to the audio by specifying them one
after another at the end of the SoX command line.
Note: 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.
allpass frequency width[h|o|q]
Apply a two-pole all-pass filter with central frequency (in Hz)
frequency, and filter-width width: in Hz (the default, or if
appended with ‘h’), in octaves (if appended with ‘o’), or as a
Q-factor (if appended with ‘q’). 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 [width[h|o|q]]
Apply a band-pass filter. The frequency response drops
logarithmically around the center frequency. The width in Hz
(the default, or if appended with ‘h’), in octaves (if appended
with ‘o’), or as a Q-factor (if appended with ‘q’), 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 width[h|o|q]
Apply a two-pole Butterworth band-pass or band-reject filter
with central frequency (in Hz) frequency, and (3dB-point) band-
width width: in Hz (the default, or if appended with ‘h’), in
octaves (if appended with ‘o’), or as a Q-factor (if appended
with ‘q’). 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 filter for a bandpass filter with steeper shoulders.
bandreject frequency width[h|o|q]
Apply a band-reject filter. See the description of the bandpass
effect for details.
bass|treble gain [frequency [width[s|h|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 (Baxandall) tone-controls.
This is also known as shelving equalisation (EQ).
gain gives the dB 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 the filter’s shelf transition is and
can be expressed as: a ‘slope’ (the default, or if appended with
‘s’), a Q-factor (if appended with ‘q’), the transition width in
octaves (if appended with ‘o’), or the transition width in Hz
(if appended with ‘h’). 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. Each four-tuple
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.
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. 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 amout 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.
This effect supports the --plot global option (for the transfer
function).
See also mcompand for a multiple-band companding effect.
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.
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.
dither [depth]
Apply dithering to the audio. Dithering deliberately adds
digital white noise to the signal in order to mask audible
quantization effects that can occur if the output sample size is
less than 24 bits. By default, the amount of noise added is ½
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
audio in audio-CD format 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). See http://www.geocities.com/beinges for a full
explanation.
echo gain-in gain-out <delay decay>
Add echoing 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.
echos gain-in gain-out <delay decay>
Add a sequence of echos 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.
equalizer frequency width[q|o|h] 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, as a Q-factor [2] (the default, or if appended with
‘q’), in octaves (if appended with ‘o’), or in Hz (if appended
with ‘h’), 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
specified.
All times can be specified in either periods of time or sample
counts. To specify time periods use the format hh:mm:ss.frac
format. To specify using sample counts, specify the number of
samples and append the letter ‘s’ to the sample count (for
example ‘8000s’).
An optional type can be specified to change the type of
envelope. 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 a linear slope.
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
frequency 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 cutoff, smaller windows a more gradual cutoff.
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 resample
effect.
flanger [delay depth regen width speed shape phase interp]
Apply a flanging effect to the audio. 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. |
+-----------------------------------------------------------------+
See [3] for a detailed description of flanging.
highpass|lowpass [-1|-2] frequency [width[q|o|h]]
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 and is the filter-width: as a Q-factor (the default, or
if appended with ‘q’), in octaves (if appended with ‘o’), or in
Hz (if appended with ‘h’); the default Q is 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 enviornment
varible LADSPA_PATH will be used as search path for plugins.
lowpass [-1|-2] frequency [width[q|o|h]]
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 "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.
See also compand for a single-band companding effect.
mixer [ -l|-r|-f|-b|-1|-2|-3|-4|n{,n} ]
Reduce the number of audio channels by mixing or selecting
channels, or increase the number of channels by duplicating
channels. Note: this effect operates on the audio channels
within the SoX effects processing chain; it should not be
confused with the -m global option (where multiple files are
mix-combined before entering the effects chain).
This effect is automatically used when the number of input
channels 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 specific channel for the output instead of
averaging the channels. The -l, and -r options will do
averaging in quad-channel files so select the exact channel to
prevent this.
The mixer effect can also be invoked with up to 16 numbers,
separated by commas, which specify the proportion (0 = 0% and 1
= 100%) of each input channel that is to be mixed into each
output channel. In two-channel mode, 4 numbers are given: l →
l, l → r, r → l, and r → r, respectively. In four-channel mode,
the first 4 numbers give the proportions for the left-front
output channel, as follows: lf → lf, rf → lf, lb → lf, and rb →
rf. The next 4 give the right-front output in the same order,
then left-back and right-back.
It is also possible to use the 16 numbers to expand or reduce
the channel count; just specify 0 for unused channels.
Finally, certain reduced combination of numbers can be specified
for certain input/output channel combinations.
+------------------------------------------------------+
|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 |
+------------------------------------------------------+
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.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
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
samples. length is the amount of silence to insert and position
the position in the input audio stream at which to insert it.
Any number of lengths and positions may be specified, provided
that a specified position is not less that the previous one.
position is optional for the first and last lengths specified
and if omitted correspond to the beginning and the end of the
audio respectively. For example: pad 1.5 1.5 adds 1.5 seconds
of silence padding at each end of the audio, whilst pad
4000s@3:00 inserts 4000 samples of silence 3 minutes into the
audio. If silence is wanted only at the end of the audio,
specify either the end position or specify a zero-length pad at
the start.
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
channels, 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. 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) or triangular (-t). The decay should be less than 0.5 to
avoid feedback. Gain-out is the volume of the output.
polyphase [-w nut|ham] [-width n] [-cutoff c]
Change the sampling rate using ‘polyphase interpolation’, a DSP
algorithm. This method is relatively slow and memory intensive.
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 -cutoff value (c) specifies the filter cutoff frequency in
terms of fraction of frequency bandwidth, also know as the
Nyquist frequency. See the resample effect for further
information 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 rabbit and resample for other sample-rate changing
effects.
rabbit [-c0|-c1|-c2|-c3|-c4]
Change the sampling rate using libsamplerate, also known as
‘Secret Rabbit Code’. This effect is optional and must have
been selected at compile time of SoX. See http://www.mega-
nerd.com/SRC for details of the algorithms. Algorithms 0
through 2 are progressively faster and lower quality versions of
the sinc algorithm; the default is -c0, which is probably the
best quality algorithm for general use currently available in
SoX. Algorithm 3 is zero-order hold, and 4 is linear
interpolation.
See also polyphase and resample for other sample-rate changing
effects, and see resample for more discussion of resampling.
repeat count
Repeat the entire audio count times. Requires disk space to
store the data to be repeated. Note that repeating once yields
two copies: the original audio and the repeated audio.
resample [-qs|-q|-ql] [rolloff [beta]]
Change the sampling rate using simulated analog filtration.
Other rate changing effects available are polyphase and rabbit.
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.
By default, linear interpolation 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 insufficient 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
coefficients, resulting in about 24 bits precision. The -qs,
-q, or -ql options specify increased accuracy at the cost of
lower execution 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
proportionally 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.
Logically, this is because the A/D converter needs at least 2
samples to detect 1 cycle at the Nyquist frequency. Frequencies
higher then the Nyquist will actually appear as lower
frequencies 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
sample 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
cutoff is, with closer being better. When increasing the sample
rate of an audio file you would not expect to have any
frequencies 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
frequency to use a highpass filter to remove these artifacts,
with closer also being better.
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 cutoff,
don’t use low beta’s, use a longer sample window. A Nuttall
window is selected by specifying any ‘beta’ ≤ 2, and the Nuttall
window has somewhat steeper cutoff than the default Kaiser
window. You will probably not need to use the beta parameter at
all, unless you are just curious about comparing the effects of
Nuttall vs. Kaiser windows.
This is the default effect if the two files have different
sampling rates. 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.
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.
Default values are shown in parenthesis.
Note that reverb 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 vol -3dB pad 0 3 reverb
reverse
Reverse the audio completely. Requires disk space to store the
data 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
periods of silence that are wanted. For example, if you have a
song with 2 seconds of silence in the middle and 2 second at the
end, you could set below-period to a value of 2 to skip over the
silence in the middle of the audio.
For below-periods, duration specifies a period of silence that
must exist before audio is not copied any more. By specifying a
higher duration, silence that is wanted can be left in the
audio. For example, if you have a song with an expected 1
second of silence in the middle and 2 seconds of silence at the
end, a duration of 2 seconds could be used to skip over the
middle silence.
Unfortunately, you must know the length of the silence at the
end of your audio file to trim off silence reliably. A work
around is to use the silence effect in combination with the
reverse effect. By first reversing the audio, you can use the
above-periods to reliably trim all audio from what looks like
the front of the file. Then reverse the file again to get back
to normal.
To remove silence from the middle of a file, specify a below-
periods that is negative. This value is then treated as a
positive value and is also used to indicate the effect should
restart processing as specified by the above-periods, making it
suitable for removing periods of silence in the middle of the
audio.
The option -l indicates that below-periods duration length of
audio should be left intact at the beginning of each period of
silence. For example, if you want to remove long pauses between
words but do not want to remove the pauses completely.
The period counts are in units of samples. Duration counts may
be in the format of hh:mm:ss.frac, or the exact count of
samples. Threshold numbers may be suffixed with d to indicate
the value is in decibels, or % to indicate a percentage of
maximum value of the sample value (0% specifies pure digital
silence).
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 the resample effect
with its default parameters. For higher quality resampling, in
addition to the speed effect, specify either the resample or the
rabbit effect with appropriate parameters.
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
above 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 max value of a signed long
variable (0x7fffffff). 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
example, swap 2 2 will overwrite channel 1 with channel 2;
creating a stereo file with both channels containing the same
audio.
synth [len] {[type] [combine] [freq[-freq2]] [off] [ph] [p1] [p2] [p3]}
This effect can be used to generate fixed or linearly swept
frequency audio tones with various wave shapes, or to generate
wide-band noise of various ‘colours’. Multiple synth effects
can be cascaded to produce more complex waveforms; at each stage
it is possible to choose whether the generated waveform will be
mixed with, or modulated onto the output from the previous
stage. Audio for each channel in a multi-channel audio file can
be synthesised independently.
Though this effect is used to generate audio, an input file must
still be given, the characteristics of which will be used to set
the synthesised audio length, the number of channels, and the
sampling rate; however, since the input file’s audio is not
normally needed, a ‘null file’ (with the special name -n) is
often given instead (and the length specified as a parameter to
synth or by another given effect that can has an associated
length).
For example, the following produces a 3 second, 44.1 kHz, audio
file containing a sine-wave swept linearly 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
cascaded 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, which means
that there is a high chance of clipping when using the audio
subsequently, so in most cases, you will want to follow this
effect with the vol effect to prevent this from happening. (See
also Clipping above.)
A detailed description of each synth parameter follows:
len is the length of audio to synthesise expressed as a time or
as a number of samples; 0=inputlength, default=0.
The format for specifying lengths in time is hh:mm:ss.frac. The
format for specifying sample counts is the number of samples
with the letter ‘s’ appended to it.
type is one of sine, square, triangle, sawtooth, trapezium, exp,
[white]noise, pinknoise, brownnoise; default=sine
combine is one of create, mix, amod (amplitude modulation), fmod
(frequency modulation); default=create
freq/freq2 are the frequencies at the beginning/end of synthesis
in Hz or, if preceded with ‘%’, semitones relative to A
(440 Hz); for both, default=%0. If freq2 is given, then len
must also have been given. 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).
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’
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, giving a quicker, but possibly lower
quality, result.
factor gives the ratio of new tempo to the old tempo.
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
milliseconds (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.
treble gain [frequency [width[s|h|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 ½ seconds into the audio. The format
for specifying sample counts is the number of samples with the
letter ‘s’ appended to it. A value of 8000s 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
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 compand for a dynamic-range
compression/expansion/limiting effect.
Deprecated Effects
The following effects have been renamed or have their functionality
included in another effect. They continue to work in this version of
SoX but may be removed in future.
avg [ -l|-r|-f|-b|-1|-2|-3|-4|n{,n} ]
Reduce the number of audio channels by mixing or selecting
channels, or duplicate channels to increase the number of
channels. This effect is just an alias of the mixer effect and
is retained for backwards compatibility only.
highp frequency
Apply a high-pass filter. This effect is just an alias for the
highpass effect used with its -1 option; it is retained for
backwards compatibility only.
lowp frequency
Apply a low-pass filter. This effect is just an alias for the
lowpass effect used with its -1 option; it is retained for
backwards compatibility only.
mask [depth]
This effect is just a deprecated alias for the dither effect,
left for historical reasons.
pick [ -l|-r|-f|-b|-1|-2|-3|-4|n{,n} ]
Pick a subset of channels to be copied into the output file.
This effect is just an alias of the mixer effect and is retained
for backwards compatibility only.
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 comparitively 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.
rate Does the same as resample with no parameters; it exists for
backwards compatibility.
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 comparitively 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.
vibro speed [depth]
This is a deprecated alias for the tremolo effect. It differs
in that the depth parameter ranges from 0 to 1 and defaults to
0.5.
SEE ALSO
sox(1), soxformat(7), libsox(3), soxexam(7), wget(1)
The SoX web page at http://sox.sourceforge.net
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, Flanging, http://harmony-
central.com/Effects/Articles/Flanging
[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
contributors are listed in the AUTHORS file that is distributed with
the source code.