ref: 7289e815250f95d076562fdc438d1aa9b51f694f
dir: /soxeffect.7/
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.TH SoX 7 "September 16, 2008" "soxeffect" "Sound eXchange"
.SH NAME
SoX \- Sound eXchange, the Swiss Army knife of audio manipulation
.SH DESCRIPTION
This manual describes SoX audio effects; the SoX manual set starts with
.BR sox (1).
.SP
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.
.SP
Some of the SoX effects are primarily intended to be applied to a single
instrument or `voice'. To facilitate this, the \fBremix\fR effect and
the global SoX option \fB\-M\fR can be used to isolate then recombine
tracks from a multi-track recording.
.SP
In the descriptions that follow,
brackets [ ] are used to denote parameters that are optional, braces
{ } to denote those that are both optional and repeatable,
and angle brackets < > to denote those that are repeatable but not
optional.
Where applicable, default values for optional parameters are shown in parenthesis ( ).
.SP
The following parameters are used with, and have the same meaning for,
several effects:
.TP
\fIcentre\fR[\fBk\fR]
See
.IR frequency .
.TP
\fIfrequency\fR[\fBk\fR]
A frequency in Hz, or, if appended with `k', kHz.
.TP
\fIgain\fR
A power gain in dB.
Zero gives no gain; less than zero gives an attenuation.
.TP
\fIwidth\fR[\fBh\fR\^|\^\fBk\fR\^|\^\fBo\fR\^|\^\fBq\fR]
Used to specify the band-width of a filter. A number of different
methods to specify the width are available (though not all for every effect);
one of the characters shown may be appended to select the desired method
as follows:
.TS
center box;
cI cI lI
cB c l.
\ Method Notes
h Hz \
k kHz \
o Octaves \
q Q-factor See [2]
.TE
.DT
.SP
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.
.PP
To see if SoX has support for an optional effect, enter
.B sox \-h
and look for its name under the list: `EFFECTS'.
.SS SOX EFFECTS
.TP
\fBallpass\fR \fIfrequency\fR[\fBk\fR]\fI width\fR[\fBh\fR\^|\^\fBk\fR\^|\^\fBo\fR\^|\^\fBq\fR]
Apply a two-pole all-pass filter with central frequency (in Hz)
\fIfrequency\fR, and filter-width \fIwidth\fR.
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].
.SP
This effect supports the \fB\-\-plot\fR global option.
.TP
\fBband\fR [\fB\-n\fR] \fIcenter\fR[\fBk\fR]\fR [\fIwidth\fR[\fBh\fR\^|\^\fBk\fR\^|\^\fBo\fR\^|\^\fBq\fR]]
Apply a band-pass filter.
The frequency response drops logarithmically
around the
.I center
frequency.
The
.I width
parameter gives the slope of the drop.
The frequencies at
.I center
+
.I width
and
.I center
\-
.I width
will be half of their original amplitudes.
.B band
defaults to a mode oriented to pitched audio,
i.e. voice, singing, or instrumental music.
The \fB\-n\fR (for noise) option uses the alternate mode
for un-pitched audio (e.g. percussion).
.B Warning:
\fB\-n\fR introduces a power-gain of about 11dB in the filter, so beware
of output clipping.
.B band
introduces noise in the shape of the filter,
i.e. peaking at the
.I center
frequency and settling around it.
.SP
This effect supports the \fB\-\-plot\fR global option.
.SP
See also \fBfilter\fR for a bandpass filter with steeper shoulders.
.TP
\fBbandpass\fR\^|\^\fBbandreject\fR [\fB\-c\fR] \fIfrequency\fR[\fBk\fR]\fI width\fR[\fBh\fR\^|\^\fBk\fR\^|\^\fBo\fR\^|\^\fBq\fR]
Apply a two-pole Butterworth band-pass or band-reject filter with
central frequency \fIfrequency\fR, and (3dB-point) band-width
\fIwidth\fR. The
.B \-c
option applies only to
.B 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].
.SP
These effects support the \fB\-\-plot\fR global option.
.SP
See also \fBfilter\fR for a bandpass filter with steeper shoulders.
.TP
\fBbandreject \fIfrequency\fR[\fBk\fR]\fI width\fR[\fBh\fR\^|\^\fBk\fR\^|\^\fBo\fR\^|\^\fBq\fR]
Apply a band-reject filter.
See the description of the \fBbandpass\fR effect for details.
.TP
\fBbass\fR\^|\^\fBtreble \fIgain\fR [\fIfrequency\fR[\fBk\fR]\fR [\fIwidth\fR[\fBs\fR\^|\^\fBh\fR\^|\^\fBk\fR\^|\^\fBo\fR\^|\^\fBq\fR]]]
Boost or cut the bass (lower) or treble (upper) frequencies of the audio
using a two-pole shelving filter with a response similar to that
of a standard hi-fi's tone-controls. This is also
known as shelving equalisation (EQ).
.SP
\fIgain\fR gives the gain at 0\ Hz (for \fBbass\fR), or whichever is
the lower of \(ap22\ kHz and the Nyquist frequency (for \fBtreble\fR). Its
useful range is about \-20 (for a large cut) to +20 (for a large
boost).
Beware of
.B Clipping
when using a positive \fIgain\fR.
.SP
If desired, the filter can be fine-tuned using the following
optional parameters:
.SP
\fIfrequency\fR 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 \fBbass\fR) or 3\ kHz (for
\fBtreble\fR).
.SP
\fIwidth\fR
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 `\fBs\fR') may be used.
The useful range of `slope' is
about 0\*d3, for a gentle slope, to 1 (the maximum), for a steep slope; the
default value is 0\*d5.
.SP
The filters are described in detail in [1].
.SP
These effects support the \fB\-\-plot\fR global option.
.SP
See also \fBequalizer\fR for a peaking equalisation effect.
.TP
\fBchorus \fIgain-in gain-out\fR <\fIdelay decay speed depth \fB\-s\fR\^|\^\fB\-t\fR>
Add a chorus effect to the audio. This can make a single vocal sound
like a chorus, but can also be applied to instrumentation.
.SP
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.
.SP
Each four-tuple parameter
delay/decay/speed/depth gives the delay in milliseconds
and the decay (relative to gain-in) with a modulation
speed in Hz using depth in milliseconds.
The modulation is either sinusoidal (\fB\-s\fR) or triangular
(\fB\-t\fR). Gain-out is the volume of the output.
.SP
A typical delay is around 40ms to 60ms; the modulation speed is best
near 0\*d25Hz and the modulation depth around 2ms.
For example, a single delay:
.EX
play guitar1.wav chorus 0.7 0.9 55 0.4 0.25 2 \-t
.EE
Two delays of the original samples:
.EX
play guitar1.wav chorus 0.6 0.9 50 0.4 0.25 2 \-t \\
60 0.32 0.4 1.3 \-s
.EE
A fuller sounding chorus (with three additional delays):
.EX
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
.EE
.TP
\fBcompand \fIattack1\fB,\fIdecay1\fR{\fB,\fIattack2\fB,\fIdecay2\fR}
[\fIsoft-knee-dB\fB:\fR]\fIin-dB1\fR[\fB,\fIout-dB1\fR]{\fB,\fIin-dB2\fB,\fIout-dB2\fR}
.br
[\fIgain\fR [\fIinitial-volume-dB\fR [\fIdelay\fR]]]
.SP
Compand (compress or expand) the dynamic range of the audio.
.SP
The
.I attack
and
.I decay
parameters (in seconds) determine the time over which the
instantaneous level of the input signal is averaged to determine its
volume; attacks refer to increases in volume and decays refer to
decreases.
For most situations, the attack time (response to the music getting
louder) should be shorter than the decay time because the human ear is more
sensitive to sudden loud music than sudden soft music.
Where more than one pair of attack/decay parameters are
specified, each input channel is companded separately and the number of
pairs must agree with the number of input channels.
Typical values are
.B 0\*d3,0\*d8
seconds.
.SP
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
.I out-dB1
defaults to the same value as
.IR in-dB1 ;
levels below
.I in-dB1
are not companded (but may have gain applied to them).
The point \fB0,0\fR is assumed but may be overridden (by
\fB0,\fIout-dBn\fR).
If the list is preceded by a
.I soft-knee-dB
value, then the points at where adjacent line segments on the
transfer function meet will be rounded by the amount given.
Typical values for the transfer function are
.BR 6:\-70,\-60,\-20 .
.SP
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.
.SP
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
.B \-90
dB.
.SP
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
.B 0\*d2
seconds.
.SP
This effect supports the \fB\-\-plot\fR global option (for the transfer function).
.SP
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:
.EX
sox asz.au asz-car.au compand 0.3,1 6:-70,-60,-20 -5 -90 0.2
.EE
The transfer function (`6:\-70,...') says that very soft sounds (below
\-70dB) will remain unchanged. This will stop the compander from
boosting the volume on `silent' passages such as between movements.
However, sounds in the range \-60dB to 0dB (maximum
volume) will be boosted so that the 60dB dynamic range of the
original music will be compressed 3-to-1 into a 20dB range, which is
wide enough to enjoy the music but narrow enough to get around the
road noise. The `6:' selects 6dB soft-knee companding.
The \-5 (dB) output gain is needed to avoid clipping (the number is
inexact, and was derived by experimentation).
The \-90 (dB) for the initial volume will work fine for a clip that starts
with near silence, and the delay of 0\*d2 (seconds) has the effect of causing
the compander to react a bit more quickly to sudden volume changes.
.SP
See also
.B mcompand
for a multiple-band companding effect.
.TP
\fBcontrast [\fIenhancement-amount (75)\fR]
Comparable with compression, this effect modifies an audio signal to
make it sound louder.
.I enhancement-amount
controls the amount of the enhancement and is a number in the range 0\-100.
Note that
.I enhancement-amount
= 0 still gives a significant contrast enhancement.
.B contrast
is often used in conjunction with the
.B norm
effect as follows:
.EX
sox infile outfile norm -i contrast
.EE
.TP
\fBdcshift \fIshift\fR [\fIlimitergain\fR]
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.
.SP
The first option is the \fIdcshift\fR value. It is a floating point number that
indicates the amount to shift.
.SP
An optional
.I limitergain
can be specified as well. It should have a value much less than 1
(e.g. 0\*d05 or 0\*d02) and is used only on peaks to prevent clipping.
.SP
An alternative approach to removing a DC offset (albeit with a short delay)
is to use the
.B highpass
filter effect at a frequency of say 10Hz, as illustrated in the following
example:
.EX
sox -n out.au synth 5 sin %0 50 highpass 10
.EE
.TP
\fBdeemph\fR
Apply ISO 908 de-emphasis (a treble attenuation shelving filter) to
44\*d1kHz (Compact Disc) audio.
.SP
Pre-emphasis was applied in the mastering of some CDs issued in the early
1980s. These included many classical music albums, as well as now
sought-after issues of albums by The Beatles, Pink Floyd and others.
Pre-emphasis should be removed at playback time by a de-emphasis
filter in the playback device. However, not all modern CD players have
this filter, and very few PC CD drives have it; playing pre-emphasised
audio without the correct de-emphasis filter results in audio that sounds harsh
and is far from what its creators intended.
.SP
With the
.B deemph
effect, it is possible to apply the necessary de-emphasis to audio that
has been extracted from a pre-emphasised CD, and then either burn the
de-emphasised audio to a new CD (which will then play correctly on any
CD player), or simply play the correctly de-emphasised audio files on the
PC. For example:
.EX
sox track1.wav track1-deemph.wav deemph
.EE
and then burn track1-deemph.wav to CD, or
.EX
play track1-deemph.wav
.EE
or simply
.EX
play track1.wav deemph
.EE
The de-emphasis filter is implemented as a biquad; its maximum deviation
from the ideal response is only 0\*d06dB (up to 20kHz).
.SP
This effect supports the \fB\-\-plot\fR global option.
.SP
See also the \fBbass\fR and \fBtreble\fR shelving equalisation effects.
.TP
\fBdelay\fR {\fIlength\fR}
Delay one or more audio channels.
.I length
can specify a time or, if appended with an `s', a number of samples.
For example,
.B delay 1\*d5 0 0\*d5
delays the first channel by 1\*d5 seconds, the third channel by 0\*d5
seconds, and leaves the second channel (and any other channels that may be
present) un-delayed.
The following (one long) command plays a chime sound:
.EX
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
.EE
.TP
\fBdither\fR [\fIdepth\fR]
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 \(12 bit;
the optional \fIdepth\fR parameter is a (linear or voltage)
multiplier of this amount.
.SP
This effect should not be followed by any other effect that
affects the audio.
.TP
\fBearwax\fR
Makes audio easier to listen to on headphones.
Adds `cues' to 44\*d1kHz stereo (i.e. audio CD format) audio so that
when listened to on headphones the stereo image is
moved from inside
your head (standard for headphones) to outside and in front of the
listener (standard for speakers). See
http://www.geocities.com/beinges
for a full explanation.
.TP
\fBecho \fIgain-in gain-out\fR <\fIdelay decay\fR>
Add echoing to the audio.
Echoes are reflected sound and can occur naturally amongst mountains
(and sometimes large buildings) when talking or shouting; digital echo
effects emulate this behaviour and are often used to help fill
out the sound of a single instrument or vocal. The time difference
between the original signal and the reflection is the `delay' (time),
and the loudness of the relected signal is the `decay'. Multiple echoes
can have different delays and decays.
.SP
Each given
.I "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:
.EX
play lead.aiff echo 0.8 0.88 60 0.4
.EE
If the delay is very short, then it sound like a (metallic) robot playing
music:
.EX
play lead.aiff echo 0.8 0.88 6 0.4
.EE
A longer delay will sound like an open air concert in the mountains:
.EX
play lead.aiff echo 0.8 0.9 1000 0.3
.EE
One mountain more, and:
.EX
play lead.aiff echo 0.8 0.9 1000 0.3 1800 0.25
.EE
.TP
\fBechos \fIgain-in gain-out\fR <\fIdelay decay\fR>
Add a sequence of echoes to the audio.
Each
.I "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.
.SP
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.
.SP
The sample will be bounced twice in symmetric echos:
.EX
play lead.aiff echos 0.8 0.7 700 0.25 700 0.3
.EE
The sample will be bounced twice in asymmetric echos:
.EX
play lead.aiff echos 0.8 0.7 700 0.25 900 0.3
.EE
The sample will sound as if played in a garage:
.EX
play lead.aiff echos 0.8 0.7 40 0.25 63 0.3
.EE
.TP
\fBequalizer \fIfrequency\fR[\fBk\fR]\fI width\fR[\fBq\fR\^|\^\fBo\fR\^|\^\fBh\fR\^|\^\fBk\fR] \fIgain\fR
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.
.SP
\fIfrequency\fR gives the filter's central frequency in Hz,
\fIwidth\fR, the band-width,
and \fIgain\fR the required gain
or attenuation in dB.
Beware of
.B Clipping
when using a positive \fIgain\fR.
.SP
In order to produce complex equalisation curves, this effect
can be given several times, each with a different central frequency.
.SP
The filter is described in detail in [1].
.SP
This effect supports the \fB\-\-plot\fR global option.
.SP
See also \fBbass\fR and \fBtreble\fR for shelving equalisation effects.
.TP
\fBfade\fR [\fItype\fR] \fIfade-in-length\fR [\fIstop-time\fR [\fIfade-out-length\fR]]
Add a fade effect to the beginning, end, or both of the audio.
.SP
For fade-ins, this starts from the first sample and ramps the volume of the audio from 0 to full volume over \fIfade-in-length\fR seconds. Specify 0 seconds if no fade-in is wanted.
.SP
For fade-outs, the audio will be truncated at
.I stop-time
and
the volume will be ramped from full volume down to 0 starting at
\fIfade-out-length\fR seconds before the \fIstop-time\fR. If
.I fade-out-length
is not specified, it defaults to the same value as
\fIfade-in-length\fR.
No fade-out is performed if
.I stop-time
is not specified.
If the file length can be determined from the input file header and length-changing effects are not in effect, then \fB0\fR may be specified for
.I stop-time
to indicate the usual case of a fade-out that ends at the end of the input
audio stream.
.SP
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').
.SP
An optional \fItype\fR can be specified to change the type of envelope. Choices are \fBq\fR for quarter of a sine wave, \fBh\fR for half a sine wave, \fBt\fR for linear slope, \fBl\fR for logarithmic, and \fBp\fR for inverted parabola. The default is logarithmic.
.TP
\fBfilter\fR [\fIlow\fR]\fB\-\fR[\fIhigh\fR] [\fIwindow-len\fR [\fIbeta\fR]]
Apply a sinc-windowed lowpass, highpass, or bandpass filter of given
window length to the signal.
\fIlow\fR refers to the frequency of the lower 6dB corner of the filter.
\fIhigh\fR refers to the frequency of the upper 6dB corner of the filter.
.SP
A low-pass filter is obtained by leaving \fIlow\fR unspecified, or 0.
A high-pass filter is obtained by leaving \fIhigh\fR unspecified, or 0,
or greater than or equal to the Nyquist frequency.
.SP
The \fIwindow-len\fR, if unspecified, defaults to 128.
Longer windows give a sharper cut-off, smaller windows a more gradual cut-off.
.SP
The \fIbeta\fR 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.
.SP
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 window is selected by
specifying any `beta' \(<= 2, and the Nuttall window has somewhat
steeper cut-off than the default Kaiser window. You will probably not
need to use the beta parameter at all, unless you are just curious
about comparing the effects of Nuttall vs. Kaiser windows.
.TP
\fBflanger\fR [\fIdelay depth regen width speed shape phase interp\fR]
Apply a flanging effect to the audio.
See [3] for a detailed description of flanging.
.SP
All parameters are optional (right to left).
.TS
center box;
cB cB cB lB
cI c c l.
\ Range Default Description
delay 0 \- 10 0 Base delay in milliseconds.
depth 0 \- 10 2 Added swept delay in milliseconds.
regen \-95 \- 95 0 T{
.na
Percentage regeneration (delayed signal feedback).
T}
width 0 \- 100 71 T{
.na
Percentage of delayed signal mixed with original.
T}
speed 0\*d1 \- 10 0\*d5 Sweeps per second (Hz).
shape \ sin Swept wave shape: \fBsine\fR\^|\^\fBtriangle\fR.
phase 0 \- 100 25 T{
.na
Swept wave percentage phase-shift for multi-channel (e.g. stereo) flange;
0 = 100 = same phase on each channel.
T}
interp \ lin T{
.na
Digital delay-line interpolation: \fBlinear\fR\^|\^\fBquadratic\fR.
T}
.TE
.DT
.TP
\fBgain \fIdB-gain\fR
Apply an amplification or an attenuation to the audio signal.
This is an alias for the
.B vol
effect\*mhandy for those who prefer to work in dBs by default.
.TP
\fBhighpass\fR\^|\^\fBlowpass\fR [\fB\-1\fR|\fB\-2\fR] \fIfrequency\fR[\fBk\fR]\fR [\fRwidth\fR[\fBq\fR\^|\^\fBo\fR\^|\^\fBh\fR\^|\^\fBk\fR]]
Apply a high-pass or low-pass filter with 3dB point \fIfrequency\fR.
The filter can be either single-pole (with
.BR \-1 ),
or double-pole (the default, or with
.BR \-2 ).
.I width
applies only to double-pole filters;
the default is Q = 0\*d707 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].
.SP
These effects support the \fB\-\-plot\fR global option.
.SP
See also \fBfilter\fR for filters with a steeper roll-off.
.TP
\fBladspa\fR \fBmodule\fR [\fBplugin\fR] [\fBargument\fR...]
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 varible
LADSPA_PATH will be used as search path for plugins.
.TP
\fBlowpass\fR [\fB\-1\fR|\fB\-2\fR] \fIfrequency\fR[\fBk\fR]\fR [\fRwidth\fR[\fBq\fR\^|\^\fBo\fR\^|\^\fBh\fR\^|\^\fBk\fR]]
Apply a low-pass filter.
See the description of the \fBhighpass\fR effect for details.
.TP
\fBmcompand\fR \(dq\fIattack1\fB,\fIdecay1\fR{\fB,\fIattack2\fB,\fIdecay2\fR}
[\fIsoft-knee-dB\fB:\fR]\fIin-dB1\fR[\fB,\fIout-dB1\fR]{\fB,\fIin-dB2\fB,\fIout-dB2\fR}
.br
[\fIgain\fR [\fIinitial-volume-dB\fR [\fIdelay\fR]]]\(dq {\fIxover-freq\fR[\fBk\fR] \(dqattack1,...\(dq}
.SP
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
\fBcompand\fR effect for the definition of its parameters. Compand
parameters are specified between double quotes and the crossover
frequency for that band is given by \fIxover-freq\fR; these can be
repeated to create multiple bands.
.SP
For example, the following (one long) command shows how multi-band
companding is typically used in FM radio:
.EX
play track1.wav gain -3 filter 8000- 32 100 mcompand \\
\(dq0.005,0.1 -47,-40,-34,-34,-17,-33\(dq 100 \\
\(dq0.003,0.05 -47,-40,-34,-34,-17,-33\(dq 400 \\
\(dq0.000625,0.0125 -47,-40,-34,-34,-15,-33\(dq 1600 \\
\(dq0.0001,0.025 -47,-40,-34,-34,-31,-31,-0,-30\(dq 6400 \\
\(dq0,0.025 -38,-31,-28,-28,-0,-25\(dq \\
gain 15 highpass 22 highpass 22 filter -17500 256 \\
gain 9 lowpass -1 17801
.EE
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.
.SP
See also
.B compand
for a single-band companding effect.
.TP
\fBmixer\fR [ \fB\-l\fR\^|\^\fB\-r\fR\^|\^\fB\-f\fR\^|\^\fB\-b\fR\^|\^\fB\-1\fR\^|\^\fB\-2\fR\^|\^\fB\-3\fR\^|\^\fB\-4\fR\^|\^\fIn\fR{\fB,\fIn\fR} ]
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
.I channels
within the SoX effects processing chain; it should not be confused with the
.B \-m
global option (where multiple
.I files
are mix-combined before entering the effects chain).
.SP
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
.B mixer
effect and use the \fB\-l\fR, \fB\-r\fR, \fB\-f\fR, \fB\-b\fR,
\fB\-1\fR, \fB\-2\fR, \fB\-3\fR, \fB\-4\fR, options to select only
the left, right, front, back channel(s) or specific channel
for the output instead of averaging the channels.
The \fB\-l\fR, and \fB\-r\fR options will do averaging
in quad-channel files so select the exact channel to prevent this.
.SP
The
.B 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 \*(RA l, l \*(RA r, r \*(RA l, and r \*(RA r,
respectively.
In four-channel mode, the first 4 numbers give the proportions for the
left-front output channel, as follows: lf \*(RA lf, rf \*(RA lf, lb \*(RA lf, and
rb \*(RA rf.
The next 4 give the right-front output in the same order, then
left-back and right-back.
.SP
It is also possible to use the 16 numbers to expand or reduce the
channel count; just specify 0 for unused channels.
.SP
Finally, certain reduced combination of numbers can be specified
for certain input/output channel combinations.
.TS
center box ;
cB cB cB lB
c c c l .
In Ch Out Ch Num Mappings
2 1 2 l \*(RA l, r \*(RA l
2 2 1 adjust balance
4 1 4 lf \*(RA l, rf \*(RA l, lb \*(RA l, rb \*(RA l
4 2 2 lf \*(RA l&rf \*(RA r, lb \*(RA l&rb \*(RA r
4 4 1 adjust balance
4 4 2 front balance, back balance
.TE
.DT
.SP
See also
.B remix
for a mixing effect that handles any number of channels.
.TP
\fBnoiseprof\fR [\fIprofile-file\fR]
Calculate a profile of the audio for use in noise reduction. See the
description of the \fBnoisered\fR effect for details.
.TP
\fBnoisered\fR [\fIprofile-file\fR [\fIamount\fR]]
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 \fBnoiseprof\fR
effect on a section of audio that ideally would contain silence but in
fact contains noise\*msuch sections are typically found at the beginning
or the end of a recording. \fBnoiseprof\fR will write out a noise
profile to \fIprofile-file\fR, or to stdout if no \fIprofile-file\fR or
if `\-' is given. E.g.
.EX
sox speech.au -n trim 0 1.5 noiseprof speech.noise-profile
.EE
To actually remove the noise, run SoX again, this time with the \fBnoisered\fR
effect;
.B noisered
will reduce noise according to a noise profile (which was generated by
.BR noiseprof ),
from
.IR profile-file ,
or from stdin if no \fIprofile-file\fR or if `\-' is given. E.g.
.EX
sox speech.au cleaned.au noisered speech.noise-profile 0.3
.EE
How much noise should be removed is specified by
.IR amount \*ma
number between 0 and 1 with a default of 0\*d5. 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
.I 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.
.SP
On most systems, the two stages\*mprofiling and reduction\*mcan be combined
using a pipe, e.g.
.EX
sox noisy.au -n trim 0 1 noiseprof | play noisy.au noisered
.EE
.TP
\fBnorm\fR [\fB\-i\fR\^|\^\fB\-b\fR] [\fIlevel\fR]
Normalise audio to 0dB FSD, to a given level relative to 0dB, or normalise
the balance of multi-channel audio.
Requires temporary file space to store the audio to be normalised.
.SP
To create a normalised copy of an audio file,
.EX
sox infile outfile norm
.EE
can be used, though note that if `infile' has a simple encoding (e.g.
PCM), then
.EX
sox infile outfile vol \`sox infile -n stat -v 2>&1\`
.EE
(on systems that support this construct) might be quicker to execute
(though perhaps not to type!) as it doesn't require a temporary file.
.SP
For a more complex example, suppose that `effect1' performs some unknown
or unpredictable attenuation and that `effect2' requires up to 10dB of
headroom, then
.EX
sox infile outfile effect1 norm -10 effect2 norm
.EE
gives both effect2 and the output file the highest possible signal
levels.
.SP
Normally, audio is normalised based on the level of the channel with
the highest peak level, which means that whilst all channels are adjusted,
only one channel attains
the normalised level. If the
.B \-i
option is given, then each channel is treated individually and
will attain the normalised level.
.SP
If the
.B \-b
option is given (with a multi-channel audio file), then the audio
channels will be balanced; i.e. the RMS level of each channel will be
normalised to that of the channel with the highest RMS level. This can
be used, for example, to correct stereo imbalance. Note that
.B \-b
can cause clipping.
.SP
In most cases,
.B 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
.BR sox (1).
.TP
\fBoops\fR
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.
.TP
\fBpad\fR { \fIlength\fR[\fB@\fIposition\fR] }
Pad the audio with silence, at the beginning, the end, or any
specified points through the audio.
Both
.I length
and
.I position
can specify a time or, if appended with an `s', a number of samples.
.I length
is the amount of silence to insert and
.I 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.
.I 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,
.B pad 1\*d5 1\*d5
adds 1\*d5 seconds of silence padding at each end of the audio, whilst
.B 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.
.TP
\fBphaser \fIgain-in gain-out delay decay speed\fR [\fB\-s\fR\^|\^\fB\-t\fR]
Add a phasing effect to the audio.
See [3] for a detailed description of phasing.
.SP
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 (\fB\-s\fR) \*mpreferable for multiple
instruments, or triangular
(\fB\-t\fR) \*mgives single instruments a sharper phasing effect.
The decay should be less than 0\*d5 to avoid
feedback, and usually no less than 0\*d1. Gain-out is the volume of the output.
.SP
For example:
.EX
play snare.flac phaser 0.8 0.74 3 0.4 0.5 -t
.EE
Gentler:
.EX
play snare.flac phaser 0.9 0.85 4 0.23 1.3 -s
.EE
A popular sound:
.EX
play snare.flac phaser 0.89 0.85 1 0.24 2 -t
.EE
More severe:
.EX
play snare.flac phaser 0.6 0.66 3 0.6 2 -t
.EE
.TP
\fBpitch \fR[\fB\-q\fR] \fIshift\fR [\fIsegment\fR [\fIsearch\fR [\fIoverlap\fR]]]
Change the audio pitch (but not tempo).
.SP
.I shift
gives the pitch shift as positive or negative `cents' (i.e. 100ths of a
semitone). See the
.B tempo
effect for a description of the other parameters.
.TP
\fBrate\fR [\fB\-q\fR\^|\^\fB\-l\fR\^|\^\fB\-m\fR\^|\^\fB\-h\fR\^|\^\fB\-v\fR] [override-options] \fIRATE\fR[\fBk\fR]
Change the audio sampling rate (i.e. resample the audio) to any given
.I RATE
(even non-integer if this is supported by the output file format)
using a quality level defined as follows:
.TS
center box;
cI cI2w9 cI cI2w6 cIw6 lIw17
cB c c c c l.
\ Quality T{
\ Phase Response
T} T{
Band-width
T} Rej dB T{
.na
Typical Use
T}
\-q T{
.na
quick
T} linear n/a T{
.na
\(~=30 @ \ Fs/4
T} T{
.na
playback on ancient hardware
T}
\-l low linear 80% 100 T{
.na
playback on old hardware
T}
\-m medium intermediate 95% 100 T{
.na
audio playback
T}
\-h high intermediate 95% 125 T{
.na
16-bit mastering (use with dither)
T}
\-v T{
.na
very high
T} intermediate 95% 175 24-bit mastering
.TE
.DT
.SP
where
.I Band-width
is the percentage of the audio frequency band that is preserved and
.I Rej dB
is the level of noise rejection. Increasing levels of resampling
quality come at the expense of increasing amounts of time to process the
audio. If no quality option is given, the quality level used is `high'.
.SP
The `quick' algorithm uses cubic interpolation; all others use
band-width limited interpolation. The `quick' and `low' quality
algorithms have a `linear' phase response; for `medium', `high' and
`very high', the phase response is configurable (see below), but
defaults to `intermediate'.
.SP
The
.B rate
effect is invoked automatically if SoX's \fB\-r\fR option specifies a
rate that is different to that of the input file(s). Alternatively, if
this effect is given explicitly, then SoX's
.B \-r
option need not be given. For example, the following two commands are
equivalent:
.EX
.ne 2
sox input.au -r 48k output.au bass -3
sox input.au output.au bass -3 rate 48k
.EE
though the second command is more flexible as it allows
.B rate
options to be given, and allows the effects to be ordered arbitrarily.
.TS
center;
c8 c8 c.
* * *
.TE
.DT
.SP
The simple quality selection described above provides settings that
satisfy the needs of the vast majority of resampling tasks.
Occasionally, however, it may be desirable to fine-tune the resampler's
filter response; this can be achieved using
.IR override\ options ,
as detailed in the following table:
.TS
center box;
lB lw52.
\-M/\-I/\-L Phase response = minimum/intermediate/linear
\-p\ 0\-100 T{
.na
Any phase response (0 = minimum, 25 = intermediate, 50 = linear, 100 = maximum)
T}
\-s Steep filter (band-width = 99%)
\-b\ 74\-99.7 Any band-width %
\-a Allow aliasing above the pass-band
.TE
.DT
.SP
N.B. Override options can not be used with the `quick' or `low'
quality algorithms.
.SP
All resamplers use filters that can sometimes create `echo' (a.k.a.
`ringing') artefacts with transient signals such as those that occur
with `finger snaps' or other highly percussive sounds. Such artefacts are
much more noticable to the human ear if they occur before the transient
(`pre-echo') than if they occur after it (`post-echo'). The phase
response setting controls the distribution of any transient echo between
`pre' and `post': with minimum phase, there is no pre-echo but the
longest post-echo; with linear phase, pre and post echo are in equal
amounts (in signal terms, but not audibility terms); the intermediate
phase setting attempts to find the best compromise by selecting a small
length (and level) of pre-echo and a medium lengthed post-echo.
.SP
Minimum, intermediate, or linear phase response is selected using the
.BR \-M ,
.BR \-I ,
or
.B \-L
option; a custom phase response can be created with the
.B \-p
option. Note that phase responses between `linear' and `maximum'
(greater than 50) are rarely useful.
.SP
A resampler's band-width setting determines how much of the frequency
content of the original signal (w.r.t. the orignal sample rate when
up-sampling, or the new sample rate when down-sampling) is preserved
during conversion. The term `pass-band' is used to refer to all frequencies
up to the band-width point (e.g. for 44\*d1kHz sampling rate, and a
resampling band-width of 95%, the pass-band represents frequencies from
0Hz (D.C.) to circa 21kHz). Increasing the resampler's band-width
results in a slower conversion and can increase transient echo
artefacts (and vice versa).
.SP
The
.B \-s
`steep filter' option changes resampling band-width from the default 95%
(based on the 3dB point), to 99%. The
.B \-b
option allows the band-width to be set to any value in the range
74\-99\*d7 %, but note that band-width values greater than 99% are not
recommended for normal use as they can cause excessive transient echo.
.SP
If the
.B \-a
option is given, then aliasing above the pass-band is allowed. For
example, with 44\*d1kHz sampling rate, and a
resampling band-width of 95%, this means that frequency content above
21kHz can be distorted; however, since this is above the pass-band (i.e.
above the highest frequency of interest/audibility), this may not be a
problem. The benefits of allowing aliasing are reduced processing time,
and reduced (by almost half) transient echo artefacts.
Note that if this option is given, then
the minimum band-width allowable with
.B \-b
increases to 85%.
.SP
Examples:
.EX
sox input.wav -2 output.wav rate -s -a 44100 dither
.EE
default (high) quality resampling; overrides: steep filter, allow
aliasing; to 44\*d1kHz sample rate; dither output to 2-byte (16-bit) WAV
file.
.EX
sox input.wav -3 output.aiff rate -v -L -b 90 48k
.EE
very high quality resampling; overrides: linear phase, band-width 90%;
to 48k sample rate; store output to 3-byte (24-bit) AIFF file.
.TS
center;
c8 c8 c.
* * *
.TE
.DT
.SP
The
.BR key ,
.B speed
and
.B tempo
effects all use the
.B rate
effect at their core.
.SP
See also
.BR resample ,
.B polyphase
and
.B rabbit
for other sample-rate changing effects.
.TP
\fBremix\fR [\fB\-a\fR\^|\^\fB\-m\fR\^|\^\fB\-p\fR] <\fIout-spec\fR>
\fIout-spec\fR = \fIin-spec\fR{\fB,\fIin-spec\fR} | \fB0\fR
.br
\fIin-spec\fR = [\fIin-chan\fR]\^[\fB\-\fR[\fIin-chan2\fR]]\^[\fIvol-spec\fR]
.br
\fIvol-spec\fR = \fBp\fR\^|\^\fBi\fR\^|\^\fBv\^\fR[\fIvolume\fR]
.br
.SP
Select and mix input audio channels into output audio channels. Each output
channel is specified, in turn, by a given \fIout-spec\fR: a list of
contributing input channels and volume specifications.
.SP
Note that this effect operates on the audio
.I channels
within the SoX effects processing chain; it should not be confused with the
.B \-m
global option (where multiple
.I files
are mix-combined before entering the effects chain).
.SP
An
.I out-spec
contains comma-separated input channel-numbers and hyphen-delimited
channel-number ranges; alternatively,
.B 0
may be given to create a silent output channel. For example,
.EX
sox input.au output.au remix 6 7 8 0
.EE
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
.EX
sox input.au output.au remix 1-3,7 3
.EE
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.
.SP
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
.EX
sox input.au output.au remix -
.EE
performs a mix-down of all input channels to mono.
.SP
By default, where an output channel is mixed from multiple (n) input
channels, each input channel will be scaled by a factor of \(S1/\s-2n\s+2.
Custom mixing volumes can be set by following a given input channel or range
of input channels with a \fIvol-spec\fR (volume specification).
This is one of the letters \fBp\fR, \fBi\fR, or \fBv\fR,
followed by a volume number, the meaning of which depends on the given
letter and is defined as follows:
.TS
center;
lI lI lI
c l l.
Letter Volume number Notes
p power adjust in dB 0 = no change
i power adjust in dB T{
.na
As `p', but invert the audio
T}
v voltage multiplier T{
.na
1 = no change, 0\*d5 \(~= 6dB attenuation, 2 \(~= 6dB gain, \-1 = invert
T}
.TE
.DT
.SP
If an
.I out-spec
includes at least one
.I vol-spec
then, by default, \(S1/\s-2n\s+2 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,
.EX
sox input.au output.au remix 1,2 3,4v0.8
.EE
results in channel level multipliers of 0\*d5,0\*d5 1,0\*d8, whereas
.EX
sox input.au output.au remix -a 1,2 3,4v0.8
.EE
results in channel level multipliers of 0\*d5,0\*d5 0\*d5,0\*d8.
.SP
The \-m (manual) option disables all automatic volume adjustments, so
.EX
sox input.au output.au remix -m 1,2 3,4v0.8
.EE
results in channel level multipliers of 1,1 1,0\*d8.
.SP
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
.BR i .
For example, if
.I input.au
is stereo, then
.EX
sox input.au output.au remix 1,2i
.EE
is a mono equivalent of the
.B oops
effect.
.SP
If the \fB\-p\fR option is given, then any automatic \(S1/\s-2n\s+2 scaling
is replaced by \(S1/\s-2\(srn\s+2 (`power') scaling; this gives a louder mix
but one that might occasionally clip.
.TS
center;
c8 c8 c.
* * *
.TE
.DT
.SP
One typical use of the
.B remix
effect is to split an audio file into a set of files, each containing
one of the constituent channels (in order to perform subsequent
processing on individual audio channels). Where more than a few
channels are involved, a script such as the following is useful:
.EX
#!/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
.EE
If a file
.I input.au
containing six audio channels were given, the script would produce six
output files:
.IR input-01.au ,
\fIinput-02.au\fR, ...,
.IR input-06.au .
.SP
See also
.B mixer
and
.B swap
for similar effects.
.TP
\fBrepeat \fIcount\fR
Repeat the entire audio \fIcount\fR 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.
.TP
\fBreverb\fR [\fB\-w\fR|\fB\-\-wet-only\fR] [\fIreverberance\fR (50%) [\fIHF-damping\fR (50%)
[\fIroom-scale\fR (100%) [\fIstereo-depth\fR (100%)
.br
[\fIpre-delay\fR (0ms) [\fIwet-gain\fR (0dB)]]]]]]
.SP
Add reverberation to the audio using the `freeverb' algorithm. A
reverberation effect is sometimes desirable for concert halls that are too
small or contain so many people that the hall's natural reverberance is
diminished. Applying a small amount of stereo reverb to a (dry) mono signal
will usually make it sound more natural. See [3] for a detailed description
of reverberation.
.SP
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:
.EX
play dry.au gain -3 pad 0 3 reverb
.EE
.TP
\fBreverse\fR
Reverse the audio completely.
Requires temporary file space to store the audio to be reversed.
.TP
\fBriaa\fR
Apply RIAA vinyl playback equalisation.
The sampling rate must be one of: 44\*d1, 48, 88\*d2, 96 kHz.
.SP
This effect supports the \fB\-\-plot\fR global option.
.TP
\fBsilence \fR[\fB\-l\fR] \fIabove-periods\fR [\fIduration
threshold\fR[\fBd\fR\^|\^\fB%\fR] [\fIbelow-periods duration
threshold\fR[\fBd\fR\^|\^\fB%\fR]]
.SP
Removes silence from the beginning, middle, or end of the audio.
Silence is anything below a specified threshold.
.SP
The \fIabove-periods\fR 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 \fIabove-periods\fR, it trims audio up until it finds
non-silence. Normally, when trimming silence from beginning of audio
the \fIabove-periods\fR 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 \fIabove-period\fR of 2 to strip out both silence periods and the
first song.
.SP
When \fIabove-periods\fR is non-zero, you must also specify a
\fIduration\fR and \fIthreshold\fR. \fIDuration\fR 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.
.SP
\fIThreshold\fR 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.
.SP
When optionally trimming silence from the end of the audio, you specify
a \fIbelow-periods\fR count. In this case, \fIbelow-period\fR 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.
.SP
For \fIbelow-periods\fR, \fIduration\fR 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.
.SP
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 \fBsilence\fR effect in combination with the \fBreverse\fR effect.
By first reversing the audio, you can use the \fIabove-periods\fR
to reliably trim all audio from what looks like the front of the file.
Then reverse the file again to get back to normal.
.SP
To remove silence from the middle of a file, specify a
\fIbelow-periods\fR 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
\fIabove-periods\fR, making it suitable for removing periods of
silence in the middle of the audio.
.SP
The option
.B \-l
indicates that \fIbelow-periods\fR \fIduration\fR 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.
.SP
The \fIperiod\fR counts are in units of samples. \fIDuration\fR counts
may be in the format of hh:mm:ss.frac, or the exact count of samples.
\fIThreshold\fR numbers may be suffixed with
.B d
to indicate the value is in decibels, or
.B %
to indicate a percentage of maximum value of the sample value
(\fB0%\fR specifies pure digital silence).
.SP
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:
.EX
rec \fIparameters filename other-effects\fR silence 1 5 2%
.EE
.TP
\fBspeed \fIfactor\fR[\fBc\fR]
Adjust the audio speed (pitch and tempo together). \fIfactor\fR
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.
.SP
By default, the speed change is performed by resampling with the \fBrate\fR
effect using its default quality/speed. For higher quality or higher speed
resampling, in addition to the \fBspeed\fR effect, specify
the \fBrate\fR effect with the desired quality option.
.TP
\fBspectrogram \fR[options]
Create a spectrogram of the audio. This effect is optional; type \fBsox
\-\-help\fR and check the list of supported effects to see if it has
been included.
.SP
The spectrogram is rendered in a Portable Network Graphic (PNG) file,
and shows time in the X-axis, frequency in the Y-axis, and audio signal
magnitude in the Z-axis. Z-axis values are represented by the colour
(or intensity) of the pixels in the X-Y plane.
.SP
This effect supports only one channel; for multi-channel input files,
use either SoX's
.B \-c 1
option with the output file (to obtain a spectrogram on the mix-down),
or the
.B remix
.I n
effect to select a particular channel. Be aware though, that both of
these methods affect the audio in the effects chain.
.RS
.IP \fB\-x\ \fInum\fR
X-axis pixels/second, default 100. This controls the width of the
spectrogram;
.I 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
.B \-\-verbose
is in effect).
.SP
The maximum width of the spectrogram is 999 pixels; if the audio length
and the given
.B \-x
number are such that this would be exceeded, then the spectrogram (and
the effects chain) will be truncated. To move the spectrogram to a
point later in the audio stream, first invoke the
.B trim
effect; e.g.
.EX
sox audio.ogg -n trim 1:00 spectrogram
.EE
starts the spectrogram at 1 minute through the audio.
.IP \fB\-y\ \fInum\fR
Y-axis resolution (1 \- 4), default 2.
This controls the height of the spectrogram;
.I num
can be from 1 (low frequency 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 manipulation package
(such as
.BR ImageMagick (1))
can be used to re-size the image.
.SP
To increase the frequency resolution without increasing the height of
the spectrogram, the
.B rate
effect may be invoked to reduce the sampling rate of the signal before
invoking
.BR spectrogram ;
e.g.
.EX
sox audio.ogg -r 4k -n rate spectrogram
.EE
allows detailed analysis of frequencies up to 2kHz (half the sampling
rate).
.IP \fB\-z\ \fInum\fR
Z-axis (colour) range in dB, default 120. This sets the dynamic-range
of the spectrogram to be \-\fInum\fR\ dBFS to 0\ dBFS.
.I Num
may range from 20 to 180. Decreasing dynamic-range effectively
increases the `contrast' of the spectrogram display, and vice versa.
.IP \fB\-Z\ \fInum\fR
Sets the upper limit of the Z-axis in dBFS.
A negative
.I num
effectively increases the `brightness' of the spectrogram display,
and vice versa.
.IP \fB\-q\ \fInum\fR
Sets the Z-axis quantisation, i.e. the number of different colours (or
intensities) in which to render Z-axis
values. A small number (e.g. 4) will give a `poster'-like effect making
it easier to discern magnitude bands of similar level. Smaller numbers
also usually
result in smaller PNG files. The number given specifies the number of
colours to use inside the Z-axis range; two colours are reserved to
represent out-of-range values.
.IP \fB\-w\ \fIname\fR
Window: Hann (default), Hamming, Bartlett, Rectangular or Kaiser. The
spectrogram is produced using the Discrete Fourier Transform (DFT)
algorithm. A significant parameter to this algorithm is the choice of
`window function'. By default, SoX uses the Hann window which has good
all-round frequency-resolution and dynamic-range properties. For better
frequency resolution (but lower dynamic-range), select a Hamming window;
for higher dynamic-range (but poorer frequency-resolution), select a
Kaiser window. Bartlett and Rectangular windows are also available.
Selecting a window other than Hann will usually require
a corresponding
.B \-z
setting.
.IP \fB\-s\fR
Allow slack overlapping of DFT windows.
This can, in some cases, increase image sharpness and give greater adherence
to the
.B \-x
value, but at the expense of a little spectral loss.
.IP \fB\-m\fR
Creates a monochrome spectrogram (the default is colour).
.IP \fB\-h\fR
Selects a high-colour palette\*mless visually pleasing than the default
colour palette, but it may make it easier to differentiate different levels.
If this option is used in conjunction with
.BR \-m ,
the result will be a hybrid monochrome/colour palette.
.IP \fB\-p\ \fInum\fR
Permute the colours in a colour or hybrid palette.
The
.I num
parameter (from 1 to 6) selects the permutation.
.IP \fB\-l\fR
Creates a `printer friendly' spectrogram with a light background (the
default has a dark background).
.IP \fB\-a\fR
Suppress the display of the axis lines. This is sometimes useful in
helping to discern artefacts at the spectrogram edges.
.IP \fB\-t\ \fItext\fR
Set the image title\*mtext to display above the spectrogram.
.IP \fB\-c\ \fItext\fR
Set the image comment\*mtext to display below and to the left of the
spectrogram.
.IP \fB\-o\ \fItext\fR
Name of the spectrogram output PNG file, default `spectrogram.png'.
.RE
.TP
\
For example, let's see what the spectrogram of a swept triangular wave looks
like:
.EX
sox -n -n synth 6 tri 10k:14k spectrogram -z 100 -w k
.EE
For the ability to perform off-line processing of spectral data, see the
.B stat
effect.
.TP
\fBsplice \fR { \fIposition\fR[\fB,\fIexcess\fR[\fB,\fIleeway\fR]] }
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.
.SP
To perform a splice, first use the
.B 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
.I excess
(default 0\*d005 seconds) of audio after the ideal joining point. The
beginning of the audio section to splice on should be trimmed with the
same
.IR excess
(before the ideal joining point), plus an additional
.I leeway
(default 0\*d005 seconds). SoX should then be invoked with the two
audio sections as input files and the
.B splice
effect given with the position at which to perform the splice\*mthis is
length of the first audio section (including the excess).
.SP
For example, a long song begins with two verses which start (as
determined e.g. by using the
.B play
command with the
.B trim
(\fIstart\fR) effect) at times 0:30\*d125 and 1:03\*d432.
The following commands cut out the first verse:
.EX
sox too-long.au part1.au trim 0 30.130
.EE
(5 ms excess, after the first verse starts)
.EX
sox long.au part2.au trim 1:03.422
.EE
(5 ms excess plus 5 ms leeway, before the second verse starts)
.EX
sox part1.au part2.au just-right.au splice 30.130
.EE
Provided your arithmetic is good enough, multiple splices can be
performed with a single
.B splice
invocation. For example:
.EX
#!/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
.EE
In the above Bourne shell script,
two splices are used to `copy and paste' audio.
.TS
center;
c8 c8 c.
* * *
.TE
.DT
.SP
It is also possible to use this effect to perform general cross-fades, e.g. to
join two songs.
In this case,
.I excess
would typically be an number of seconds, and
.I leeway
should be set to zero.
.TP
\fBstat\fR [\fB\-s \fIscale\fR] [\fB\-rms\fR] [\fB\-freq\fR] [\fB\-v\fR] [\fB\-d\fR]
Display time and frequency domain statistical information about the audio.
Audio is passed unmodified through the SoX processing chain.
.SP
The information is output to the `standard error' (stderr) stream and is
calculated, where
.I n
is the duration of the audio in samples,
.I c
is the number of audio channels,
.I r
is the audio sample rate, and
.I x\s-2\dk\u\s0
represents the PCM value (in the range \-1 to +1 by default) of each successive
sample in the audio,
as follows:
.TS
center;
lI l l.
Samples read \fIn\fR\^\(mu\^\fIc\fR \
Length (seconds) \fIn\fR\^\(di\^\fIr\fR
Scaled by \ See \-s below.
Maximum amplitude max(\fIx\s-2\dk\u\s0\fR) T{
The maximum sample value in the audio; usually this will be a positive number.
T}
Minimum amplitude min(\fIx\s-2\dk\u\s0\fR) T{
The minimum sample value in the audio; usually this will be a negative number.
T}
Midline amplitude \(12\^min(\fIx\s-2\dk\u\s0\fR)\^+\^\(12\^max(\fIx\s-2\dk\u\s0\fR)
Mean norm \(S1/\s-2n\s+2\^\(*S\^\^\(br\^\fIx\s-2\dk\u\s0\fR\^\(br\^ T{
The average of the absolute value of each sample in the audio.
T}
Mean amplitude \(S1/\s-2n\s+2\^\(*S\^\fIx\s-2\dk\u\s0\fR T{
The average of each sample in the audio. If this figure is non-zero, then it indicates the
presence of a D.C. offset (which could be removed using the
.B dcshift
effect).
T}
RMS amplitude \(sr(\(S1/\s-2n\s+2\^\(*S\^\fIx\s-2\dk\u\s0\fR\(S2) T{
The level of a D.C. signal that would have the same power
as the audio's average power.
T}
Maximum delta max(\^\(br\^\fIx\s-2\dk\u\s0\fR\^\-\^\fIx\s-2\dk\-1\u\s0\fR\^\(br\^)
Minimum delta min(\^\(br\^\fIx\s-2\dk\u\s0\fR\^\-\^\fIx\s-2\dk\-1\u\s0\fR\^\(br\^)
Mean delta \(S1/\s-2n\-1\s+2\^\(*S\^\^\(br\^\fIx\s-2\dk\u\s0\fR\^\-\^\fIx\s-2\dk\-1\u\s0\fR\^\(br\^
RMS delta \(sr(\(S1/\s-2n\-1\s+2\^\(*S\^(\fIx\s-2\dk\u\s0\fR\^\-\^\fIx\s-2\dk\-1\u\s0\fR)\(S2)
Rough frequency \ In Hz.
Volume Adjustment \ T{
The parameter to the
.B vol
effect which would make the audio as loud as possible without clipping.
Note: See the discussion on
.B Clipping
in
.BR sox (1)
for reasons why it is rarely a good idea actually to do this.
T}
.TE
.DT
.SP
The
.B \-s
option can be used to scale the input data by a given factor.
The default value of
.I scale
is 2147483647 (i.e. the maximum value of a 32-bit signed integer).
Internal effects
always work with signed long PCM data and so the value should relate to this
fact.
.SP
The
.B \-rms
option will convert all output average values to `root mean square'
format.
.SP
The
.B \-v
option displays only the `Volume Adjustment' value.
.SP
The
.B \-freq
option calculates the input's power spectrum (4096 point DFT) instead of the
statistics listed above.
.SP
The
.B \-d
option
displays a hex dump of the 32-bit signed PCM data
audio in SoX's internal buffer.
This is mainly used to help track down endian problems that
sometimes occur in cross-platform versions of SoX.
.TP
\fBswap\fR [\fI1 2\fR | \fI1 2 3 4\fR]
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,
.B swap 2 2
will overwrite channel 1 with channel 2; creating a stereo
file with both channels containing the same audio.
.SP
See also the
.B remix
effect.
.TP
\fBsynth\fR [\fIlen\fR] {[\fItype\fR] [\fIcombine\fR] [[\fB%\fR]\fIfreq\fR[\fBk\fR][\fB:\fR\^|\^\fB+\fR\^|\^\fB/\fR\^|\^\fB\-\fR[\fB%\fR]\fIfreq2\fR[\fBk\fR]]] [\fIoff\fR] [\fIph\fR] [\fIp1\fR] [\fIp2\fR] [\fIp3\fR]}
This effect can be used to generate fixed or swept frequency audio tones
with various wave shapes, or to generate wide-band noise of various
`colours'.
Multiple synth effects can be cascaded to produce more complex
waveforms; at each stage it is possible to choose whether the generated
waveform will be mixed with, or modulated onto
the output from the previous stage.
Audio for each channel in a multi-channel audio file can be synthesised
independently.
.SP
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 \fB\-n\fR) is often given instead (and the
length specified as a parameter to \fBsynth\fR or by another given
effect that can has an associated length).
.SP
For example, the following produces a 3 second, 48kHz,
audio file containing a sine-wave swept from 300 to 3300\ Hz:
.EX
sox -n output.au synth 3 sine 300-3300
.EE
and this produces an 8\ kHz version:
.EX
sox -r 8000 -n output.au synth 3 sine 300-3300
.EE
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:
.EX
sox -n output.au synth 3 sine 300-3300 brownnoise
.EE
The following example shows how two synth effects can be cascaded
to create a more complex waveform:
.EX
sox -n output.au synth 0\*d5 sine 200-500 \(rs
synth 0\*d5 sine fmod 700-100
.EE
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:
.EX
play -n synth sine %-17
.EE
.B 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 follow this effect with the \fBgain\fR
effect to prevent this from happening. (See also
.B Clipping
in
.BR sox (1).)
.SP
A detailed description of each
.B synth
parameter follows:
.SP
\fIlen\fR is the length of audio to synthesise expressed as a time
or as a number of samples;
0=inputlength, default=0.
.SP
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.
.SP
\fItype\fR is one of sine, square, triangle, sawtooth, trapezium, exp,
[white]noise, pinknoise, brownnoise; default=sine
.SP
\fIcombine\fR is one of create, mix, amod (amplitude modulation), fmod
(frequency modulation); default=create
.SP
\fIfreq\fR/\fIfreq2\fR 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
.I freq2
is given, then
.I 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:
.RS
.IP \fB:\fR
Linear: the tone will change by a fixed number of hertz per second.
.IP \fB+\fR
Square: a second-order function is used to change the tone.
.IP \fB/\fR
Exponential: the tone will change by a fixed number of semitones per second.
.IP \fB\-\fR
Exponential: as `/', but initial phase always zero, and stepped (less
smooth) frequency changes.
.RE
.TP
\
Not used for noise.
.SP
\fIoff\fR is the bias (DC-offset) of the signal in percent; default=0.
.SP
\fIph\fR is the phase shift in percentage of 1 cycle; default=0. Not
used for noise.
.SP
\fIp1\fR is the percentage of each cycle that is `on' (square), or
`rising' (triangle, exp, trapezium); default=50 (square, triangle, exp),
default=10 (trapezium).
.SP
\fIp2\fR (trapezium): the percentage through each cycle at which `falling'
begins; default=50. exp: the amplitude in percent; default=100.
.SP
\fIp3\fR (trapezium): the percentage through each cycle at which `falling'
ends; default=60.
.TP
\fBtempo \fR[\fB\-q\fR] \fIfactor\fR [\fIsegment\fR [\fIsearch\fR [\fIoverlap\fR]]]
Change the audio tempo (but not its pitch).
The audio is chopped up into segments which are then shifted in the time
domain and overlapped (cross-faded) at points where their waveforms are
most similar (as determined by measurement of `least squares').
.SP
By default, linear searches are used to find the best overlapping
points; if the optional
.B \-q
parameter is given, tree searches are used instead, giving a quicker,
but possibly lower quality, result.
.SP
.I 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%.
.SP
The optional
.I 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.
.SP
The optional
.I 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.
.SP
The optional
.I overlap
parameter gives the segment overlap length in milliseconds (default 12).
.SP
See also
.B speed
for an effect that changes tempo and pitch together, and
.B pitch
for an effect that changes pitch without changing tempo.
.TP
\fBtreble \fIgain\fR [\fIfrequency\fR[\fBk\fR]\fR [\fIwidth\fR[\fBs\fR\^|\^\fBh\fR\^|\^\fBk\fR\^|\^\fBo\fR\^|\^\fBq\fR]]]
Apply a treble tone-control effect.
See the description of the \fBbass\fR effect for details.
.TP
\fBtremolo \fIspeed\fR [\fIdepth\fR]
Apply a tremolo (low frequency amplitude modulation) effect to the audio.
The tremolo frequency in Hz is given by
.IR speed ,
and the depth as a percentage by
.I depth
(default 40).
.SP
Note: This effect is a special case of the
.B synth
effect.
.TP
\fBtrim \fIstart\fR [\fIlength\fR]
Trim can trim off unwanted audio from the beginning and end of the
audio. Audio is not sent to the output stream until
the \fIstart\fR location is reached.
.SP
The optional \fIlength\fR parameter tells the number of samples to output
after the \fIstart\fR sample and is used to trim off the back side of the
audio. Using a value of 0 for the \fIstart\fR parameter will allow
trimming off the back side only.
.SP
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\*d5 will not start until 1 minute,
thirty and \(12 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.
.TP
\fBvol \fIgain\fR [\fItype\fR [\fIlimitergain\fR]]
Apply an amplification or an attenuation to the audio signal.
Unlike the
.B \-v
option (which is used for balancing multiple input files as they enter the
SoX effects processing chain),
.B vol
is an effect like any other so can be applied anywhere, and several times
if necessary, during the processing chain.
.SP
The amount to change the volume is given by
.I gain
which is interpreted, according to the given \fItype\fR, as follows: if
.I type
is \fBamplitude\fR (or is omitted), then
.I gain
is an amplitude (i.e. voltage or linear) ratio,
if \fBpower\fR, then a power (i.e. wattage or voltage-squared) ratio,
and if \fBdB\fR, then a power change in dB.
.SP
When
.I type
is \fBamplitude\fR or \fBpower\fR, a
.I gain
of 1 leaves the volume unchanged,
less than 1 decreases it,
and greater than 1 increases it;
a negative
.I gain
inverts the audio signal in addition to adjusting its volume.
.SP
When
.I type
is \fBdB\fR, a
.I gain
of 0 leaves the volume unchanged,
less than 0 decreases it,
and greater than 0 increases it.
.SP
See [4]
for a detailed discussion on electrical (and hence audio signal)
voltage and power ratios.
.SP
Beware of
.B Clipping
when the increasing the volume.
.SP
The
.I gain
and the
.I type
parameters can be concatenated if desired, e.g.
.BR "vol 10dB" .
.SP
An optional \fIlimitergain\fR value can be specified and should be a
value much less
than 1 (e.g. 0\*d05 or 0\*d02) 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.
.SP
See also
.B compand
for a dynamic-range compression/expansion/limiting effect.
.SS 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.
.TP
\fBkey \fR[\fB\-q\fR] \fIshift\fR [\fIsegment\fR [\fIsearch\fR [\fIoverlap\fR]]]
Change the audio key (i.e. pitch but not tempo).
This is just an alias for the
.B pitch
effect.
.TP
\fBpan \fIdirection\fR
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
.I 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.
.TP
\fBpolyphase\fR [\fB\-w nut\fR\^|\^\fBham\fR] [\fB\-width \fIn\fR] [\fB\-cut-off \fIc\fR]
Change the sampling rate using `polyphase interpolation', a DSP algorithm.
\fBpolyphase\fR copes with only certain rational fraction resampling ratios,
and, compared with the \fBrate\fR effect, is generally slow, memory intensive,
and has poorer stop-band rejection.
.SP
If the \fB\-w\fR parameter is \fBnut\fR, then a Nuttall (~90 dB
stop-band) window will be used; \fBham\fR selects a Hamming (~43
dB stop-band) window. The default is Nuttall.
.SP
The \fB\-width\fR parameter specifies the (approximate) width of the filter. The default is 1024 samples, which produces reasonable results.
.SP
The \fB\-cut-off\fR value (\fIc\fR) specifies the filter cut-off frequency in terms of fraction of
frequency bandwidth, also know as the Nyquist frequency. See
the \fBresample\fR 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\*d95.
.SP
See also
.BR rate ,
.B rabbit
and
.B resample
for other sample-rate changing effects.
.TP
\fBrabbit\fR [\fB\-c0\fR\^|\^\fB\-c1\fR\^|\^\fB\-c2\fR\^|\^\fB\-c3\fR\^|\^\fB\-c4\fR]
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.
Compared with the \fBrate\fR effect, \fBrabbit\fR is very slow.
.SP
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 \fB\-c0\fR.
Algorithm 3 is zero-order hold, and 4 is linear interpolation.
.SP
See also
.BR rate ,
.B polyphase
and
.B resample
for other sample-rate changing effects, and see
\fBresample\fR for more discussion of resampling.
.TP
\fBresample\fR [\fB\-qs\fR\^|\^\fB\-q\fR\^|\^\fB\-ql\fR] [\fIrolloff\fR [\fIbeta\fR]]
Change the sampling rate using simulated analog filtration.
Compared with the \fBrate\fR effect, \fBresample\fR is slow, and has poorer
stop-band rejection.
Only the low quality option works with all resampling ratios.
.SP
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 insufficient stop-band rejection
in the case that you want to have roll-off greater than about 0\*d8 of
the Nyquist frequency.
.SP
The \fB\-q*\fR 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 \fB\-qs\fR, \fB\-q\fR, or \fB\-ql\fR options specify increased accuracy
at the cost of lower execution speed. It is optional to specify
roll-off and beta parameters when using the \fB\-q*\fR options.
.SP
Following is a table of the reasonable defaults which are built-in to
SoX:
.SP
.TS
center box;
cB cB cB cB cB
c c n c c
cB c n c c.
Option Window Roll-off Beta Interpolation
(none) 45 0\*d80 16 linear
\-qs 45 0\*d80 16 quadratic
\-q 75 0\*d875 16 quadratic
\-ql 149 0\*d94 16 quadratic
.TE
.DT
.SP
\fB\-qs\fR, \fB\-q\fR, or \fB\-ql\fR 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.
.SP
\fIrolloff\fR 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\*d8\-0\*d95. The defaults are
indicated above.
.SP
The \fINyquist frequency\fR 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.
.SP
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.
.SP
The \fIrolloff\fR refers to how close to the Nyquist frequency this cut-off
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 \fIrolloff\fR refers to how close
to the original Nyquist frequency to use a highpass filter to remove
these artifacts, with closer also being better.
.SP
The \fIbeta\fR, 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 \fBfilter\fR effect.
.SP
Default parameters are, as indicated above, Kaiser window of length 45,
roll-off 0\*d80, beta 16, linear interpolation.
.SP
Note: \fB\-qs\fR is only slightly slower, but more accurate for
16-bit or higher precision.
.SP
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 \(di gcd(input-rate, output-rate) \(<= 511.
.SP
See also
.BR rate ,
.B polyphase
and
.B rabbit
for other sample-rate changing effects.
There is a detailed analysis of
\fBresample\fR and \fBpolyphase\fR at
http://leute.server.de/wilde/resample.html; see \fBrabbit\fR for a
pointer to its own documentation.
.SH SEE ALSO
.BR sox (1),
.BR soxi (1),
.BR soxformat (7),
.BR libsox (3),
.SP
The SoX web page at http://sox.sourceforge.net
.br
SoX scripting examples at http://sox.sourceforge.net/Docs/Scripts
.SS References
.TP
[1]
R. Bristow-Johnson,
.IR "Cookbook formulae for audio EQ biquad filter coefficients" ,
http://musicdsp.org/files/Audio-EQ-Cookbook.txt
.TP
[2]
Wikipedia,
.IR "Q-factor" ,
http://en.wikipedia.org/wiki/Q_factor
.TP
[3]
Scott Lehman,
.IR "Effects Explained" ,
http://harmony-central.com/Effects/effects-explained.html
.TP
[4]
Wikipedia,
.IR "Decibel" ,
http://en.wikipedia.org/wiki/Decibel
.TP
[5]
Richard Furse,
.IR "Linux Audio Developer's Simple Plugin API" ,
http://www.ladspa.org
.TP
[6]
Richard Furse,
.IR "Computer Music Toolkit" ,
http://www.ladspa.org/cmt
.TP
[7]
Steve Harris,
.IR "LADSPA plugins" ,
http://plugin.org.uk
.SH AUTHORS
Chris Bagwell (cbagwell@users.sourceforge.net).
Other authors and contributors are listed in the AUTHORS file that
is distributed with the source code.