ref: d6059b8fd3040c6f96f1cc170eff42431a542af9
dir: /soxeffect.7/
'\" t
'\" The line above instructs most `man' programs to invoke tbl
'\"
'\" Separate paragraphs; not the same as PP which resets indent level.
.de SP
.if t .sp .5
.if n .sp
..
'\"
'\" Replacement em-dash for nroff (default is too short).
.ie n .ds m " -
.el .ds m \(em
'\"
'\" Placeholder macro for if longer nroff arrow is needed.
.ds RA \(->
'\"
'\" Decimal point set slightly raised
.if t .ds d \v'-.15m'.\v'+.15m'
.if n .ds d .
'\"
'\" Enclosure macro for examples
.de EX
.SP
.nf
.ft CW
..
.de EE
.ft R
.SP
.fi
..
.TH SoX 7 "April 17, 2007" "soxeffect" "Sound eXchange"
.SH NAME
SoX \- Sound eXchange, the Swiss Army knife of audio manipulation
.SH DESCRIPTION
.SS SOX EFFECTS
Multiple effects may be applied to the audio by specifying them
one after another at the end of the SoX command line.
.SP
.I 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.
.TP
\fBallpass\fR \fIfrequency width\fR[\fBh\fR\^|\^\fBo\fR\^|\^\fBq\fR]
Apply a two-pole all-pass filter with central frequency (in Hz)
\fIfrequency\fR, and filter-width \fIwidth\fR: in Hz (the default, or if
appended with `\fBh\fR'), in octaves (if appended with `\fBo\fR'), or as
a Q-factor (if appended with `\fBq\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 [width\fR[\fBh\fR\^|\^\fBo\fR\^|\^\fBq\fR]]
Apply a band-pass filter.
The frequency response drops logarithmically
around the
.I center
frequency.
The
.I width
in Hz (the default, or if appended with `\fBh\fR'), in octaves (if
appended with `\fBo\fR'), or as a Q-factor (if appended with `\fBq\fR'),
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 width\fR[\fBh\fR\^|\^\fBo\fR\^|\^\fBq\fR]
Apply a two-pole Butterworth band-pass or band-reject filter with
central frequency (in Hz) \fIfrequency\fR, and (3dB-point) band-width
\fIwidth\fR: in Hz (the default, or if appended with `\fBh\fR'), in
octaves (if appended with `\fBo\fR'), or as a Q-factor (if appended with
`\fBq\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 width\fR[\fBh\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 [\fIwidth\fR[\fBs\fR\^|\^\fBh\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 (Baxandall) tone-controls. This is also
known as shelving equalisation (EQ).
.SP
\fIgain\fR gives the dB 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 the filter's shelf transition is and can be expressed as:
a `slope' (the default, or if appended with `\fBs\fR'),
a Q-factor (if appended with `\fBq\fR'),
the transition width in octaves (if appended with `\fBo\fR'),
or the transition width in Hz (if appended with `\fBh\fR').
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. 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 (\fB\-s\fR) or triangular
(\fB\-t\fR). Gain-out is the volume of the output.
.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. 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 amout 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
See also
.B mcompand
for a multiple-band companding effect.
.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.
.TP
\fBdeemph\fR
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.
.SP
This effect supports the \fB\-\-plot\fR global option.
.SP
See also the \fBbass\fR and \fBtreble\fR shelving equalisation effects.
.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 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.
.TP
\fBecho \fIgain-in gain-out\fR <\fIdelay decay\fR>
Add echoing 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.
.TP
\fBechos \fIgain-in gain-out\fR <\fIdelay decay\fR>
Add a sequence of echos 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.
.TP
\fBequalizer \fIfrequency width\fR[\fBq\fR\^|\^\fBo\fR\^|\^\fBh\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,
as a Q-factor [2] (the default, or if appended with `\fBq\fR'),
in octaves (if appended with `\fBo\fR'),
or in Hz (if appended with `\fBh\fR'),
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.
.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 a linear slope.
.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 cutoff, smaller windows a more gradual cutoff.
.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 \fBresample\fR effect.
.SP
.TP
\fBflanger\fR [\fIdelay depth regen width speed shape phase interp\fR]
Apply a flanging effect to the audio.
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
.SP
See [3] for a detailed description of flanging.
.TP
\fBhighpass\fR\^|\^\fBlowpass\fR [\fB-1\fR|\fB-2\fR] \fIfrequency\fR [\fRwidth\fR[\fBq\fR\^|\^\fBo\fR\^|\^\fBh\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 and is the filter-width: as a
Q-factor (the default, or if appended with `\fBq\fR'), in octaves (if
appended with `\fBo\fR'), or in Hz (if appended with `\fBh\fR');
the default Q is 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
\fBkey \fR[\fB\-q\fR] \fIshift\fR [\fIsegment\fR [\fIsearch\fR [\fIoverlap\fR]]]
Change the audio key (i.e. pitch but not tempo) using a WSOLA algorithm.
.SP
.I shift
gives the key shift as positive or negative `cents' (i.e. 100ths of a
semitone). See the
.B tempo
effect for a description of the other parameters.
.SP
See also
.B pitch
for a similar effect.
.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.
.TP
\fBlowpass\fR [\fB-1\fR|\fB-2\fR] \fIfrequency\fR [\fRwidth\fR[\fBq\fR\^|\^\fBo\fR\^|\^\fBh\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 \(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
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
.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
\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
\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
\fBphaser \fIgain-in gain-out delay decay speed\fR [\fB\-s\fR\^|\^\fB\-t\fR]
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 (\fB\-s\fR) or triangular
(\fB\-t\fR). The decay should be less than 0\*d5 to avoid
feedback. Gain-out is the volume of the output.
.TP
\fBpolyphase\fR [\fB\-w nut\fR\^|\^\fBham\fR] [\fB\-width \fIn\fR] [\fB\-cutoff \fIc\fR]
Change the sampling rate using `polyphase interpolation', a DSP algorithm.
This method is relatively slow and memory intensive.
.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\-cutoff\fR value (\fIc\fR) specifies the filter cutoff 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
.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 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 \fB\-c0\fR, 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.
.SP
See also
.B polyphase
and
.B resample
for other sample-rate changing effects, and see
\fBresample\fR for more discussion of resampling.
.TP
\fBrepeat \fIcount\fR
Repeat the entire audio \fIcount\fR 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.
.TP
\fBresample\fR [\fB\-qs\fR\^|\^\fB\-q\fR\^|\^\fB\-ql\fR] [\fIrolloff\fR [\fIbeta\fR]]
Change the sampling rate using simulated
analog filtration. Other rate changing effects available are
\fBpolyphase\fR and \fBrabbit\fR. 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.
.SP
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\*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 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 \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 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 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.
.SP
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\*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.
.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.
Default values are shown in parenthesis.
.TP
\fBreverse\fR
Reverse the audio completely.
Requires disk space to store the data to be reversed.
.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).
.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 the \fBresample\fR
effect with its default parameters. For higher quality
resampling, in addition to the \fBspeed\fR effect, specify
either the \fBresample\fR or the \fBrabbit\fR effect with
appropriate parameters.
.TP
\fBstat\fR [\fB\-s \fIn\fR] [\fB\-rms\fR] [\fB\-freq\fR] [\fB\-v\fR] [\fB\-d\fR]
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.
.SP
The `Volume Adjustment:' field in the statistics
gives you the parameter to the
.B \-v
.I number
which will make the audio as loud as possible without clipping.
Note: See the discussion on
.B Clipping
above for reasons why it is rarely a good idea to actually do this.
.SP
The option
.B \-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.
.SP
The
.B \-s
option is used to scale the input data by a given factor. The default value
of
.I 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.
.SP
The
.B \-rms
option will convert all output average values to `root mean square'
format.
.SP
The
.B \-freq
option calculates the input's power spectrum and prints it to standard error.
.SP
There is also an optional parameter
.B \-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.
.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.
.TP
\fBsynth\fR [\fIlen\fR] {[\fItype\fR] [\fIcombine\fR] [\fIfreq\fR[\fI\-freq2\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, 44\*d1\ kHz,
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, 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 \fBvol\fR
effect to prevent this from happening. (See also
.B Clipping
above.)
.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.
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) 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').
.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.
.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 stretch
for a similar effect.
.TP
\fBtreble \fIgain\fR [\fIfrequency\fR [\fIwidth\fR[\fBs\fR\^|\^\fBh\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
\fBavg\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 duplicate channels to increase the number of channels.
This effect is just an alias of the
.B mixer
effect and is retained for backwards compatibility only.
.TP
\fBhighp\fR \fIfrequency\fR
Apply a high-pass filter.
This effect is just an alias for the
.B highpass
effect used with its
.B \-1
option; it is retained for backwards compatibility only.
.TP
\fBlowp \fIfrequency\fR
Apply a low-pass filter.
This effect is just an alias for the
.B lowpass
effect used with its
.B \-1
option; it is retained for backwards compatibility only.
.TP
\fBmask\fR [\fIdepth\fR]
This effect is just a deprecated alias for the \fBdither\fR effect, left for historical reasons.
.TP
\fBpick\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} ]
Pick a subset of channels to be copied into the output file.
This effect is just an alias of the
.B mixer
effect and is retained for backwards compatibility only.
.TP
\fBpitch \fIshift\fR [\fIwidth interpolate fade\fR]
Change the audio pitch (but not its duration).
This effect is equivalent to the
.B key
effect with
.I search
set to zero, so its results are comparitively poor;
it is retained for backwards compatibility only.
.SP
Change by cross-fading shifted samples.
.I shift
is given in cents. Use a positive value to shift to treble, negative value to shift to bass.
Default shift is 0.
.I width
of window is in ms. Default width is 20ms. Try 30ms to lower pitch,
and 10ms to raise pitch.
.I interpolate
option, can be \fBcubic\fR or \fBlinear\fR. Default is \fBcubic\fR. The
.I fade
option, can be \fBcos\fR, \fBhamming\fR, \fBlinear\fR or
\fBtrapezoid\fR; the default is \fBcos\fR.
.TP
\fBrate\fR
Does the same as \fBresample\fR with no parameters; it exists for
backwards compatibility.
.TP
\fBstretch \fIfactor\fR [\fIwindow fade shift fading\fR]
Change the audio duration (but not its pitch).
This effect is equivalent to the
.B tempo
effect with (\fIfactor\fR inverted and)
.I search
set to zero, so its results are comparitively poor;
it is retained for backwards compatibility only.
.SP
.I factor
of stretching: >1 lengthen, <1 shorten duration.
.I window
size is in ms. Default is 20ms. The
.I fade
option, can be `lin'.
.I shift
ratio, in [0 1]. Default depends on stretch factor. 1
to shorten, 0\*d8 to lengthen. The
.I fading
ratio, in [0 0\*d5]. The amount of a fade's default depends on
.I factor
and \fIshift\fR.
.TP
\fBvibro \fIspeed\fR [\fIdepth\fR]
This is a deprecated alias for the
.B tremolo
effect. It differs in that the depth parameter ranges from 0 to 1 and defaults to 0\*d5.
.SH SEE ALSO
.BR sox (1),
.BR soxformat (7),
.BR libsox (3),
.BR soxexam (7),
.BR wget (1)
.SP
The SoX web page at http://sox.sourceforge.net
.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 "Flanging" ,
http://harmony-central.com/Effects/Articles/Flanging
.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.