ref: 2078845b803fcc613d8410c7a6ae9c66047bb1bc
parent: 3d18ea29658bfb1ac4be84d4ae0f4660199ca76d
author: robs <robs>
date: Fri Jan 5 18:52:32 EST 2007
Minor clean-ups.
--- a/sox.1
+++ b/sox.1
@@ -1,15 +1,12 @@
'\" t
'\" The line above instructs some `man' programs to invoke tbl
-.de SP
-.if t .sp .5
-.if n .sp
-..
-.ie n .ds EM " -
-.el .ds EM \(em
+.ie n .ds m " -
+.el .ds m \(em
.ds RA \(->
+.ds d \v'-.15m'.\v'+.15m'\" Decimal point set slightly raised
.TH SoX 1 "January 31, 2007" "sox" "Sound eXchange"
.SH NAME
-SoX\*(EMSound eXchange\*(EMThe Swiss Army knife of audio manipulation
+SoX\*mSound eXchange\*mThe Swiss Army knife of audio manipulation
.SH SYNOPSIS
.nf
\fBsox [\fR\fIglobal-options\fR\fB] [\fR\fIformat-options\fR\fB]\fR \fIinfile1\fR
@@ -20,13 +17,13 @@
SoX reads and writes audio files in most popular formats and can
optionally apply effects to them; it includes a basic audio synthesiser,
and, on many systems, can play and record audio files.
-.SP
+.P
SoX can also combine multiple input files (with the same sample rate and
number of channels) to form one output file using one of three methods:
`concatenate' (the default), `mix', or `merge'.
-.SP
+.P
The overall SoX processing chain can be summarised as follows:
-.SP
+.P
.ce
Input(s) \*(RA Combiner \*(RA Effects \*(RA Output
.SS File Formats
@@ -36,7 +33,7 @@
The second type is `headerless' (or `raw data'); here,
the audio data characteristics must be described using the
SoX command line.
-.SP
+.P
The following four characteristics are sufficient to describe
the format of audio data so that it can be processed with SoX:
.TP
@@ -44,7 +41,7 @@
The sample rate in samples per second (i.e. `Hertz' or `Hz'). For
example, digital telephony traditionally uses a sample rate of 8000Hz
(8kHz);
-audio Compact Discs use 44100Hz (44.1kHz).
+audio Compact Discs use 44100Hz (44\*d1kHz).
.TP
sample size
The number of bits (or bytes) used to store each sample. Most popular are
@@ -65,7 +62,7 @@
.PP
The term `bit-rate' is sometimes used as an overall measure of an audio
format and may incorporate elements of all of the above.
-.SP
+.P
Most self-describing formats also allow textual `comments' to be
embedded in the file that can be used to describe the audio in some way,
e.g. for music, the title, the author, etc.
@@ -73,10 +70,10 @@
There are several mechanisms available for SoX to use to determine or set the
format characteristics of an audio file. Depending on the circumstances,
individual characteristics may be determined or set using different mechanisms.
-.SP
+.P
To determine the format of an input file, SoX will use, in order of
precedence and as given or available:
-.SP
+.P
.TS
tab (@);
l l l.
@@ -84,10 +81,10 @@
@2.@The contents of the file header.
@3.@The file-name extension.
.TE
-.SP
+.P
To set the output file format, SoX will use, in order of
precedence and as given or available:
-.SP
+.P
.TS
tab (@);
l l lw(6i).
@@ -98,7 +95,7 @@
to them that is supported by the output file type.
T}
.TE
-.SP
+.P
For all files, SoX will exit with an error
if the file type cannot be determined; command-line format options may
need to be added or changed to resolve the problem.
@@ -111,18 +108,18 @@
many formats used in portable music players (e.g. MP3, Vorbis) where
adequate fidelity can be retained even with the large compression ratios
that are needed to make portable players practical.
-.SP
+.P
Formats that discard audio signal information are called `lossy',
and formats that do not, `lossless'. The term `quality' is used as a
measure of how closely the original audio signal can be reproduced when
using a lossy format.
-.SP
+.P
Audio file conversion with SoX is lossless where it can be, i.e. when
not using lossy compression and the number of bits used in the
destination format is not less than in the source format. E.g.
converting from an 8-bit PCM format to a 16-bit PCM format is lossless
but converting from an 8-bit PCM format to (8-bit) A-law isn't.
-.SP
+.P
.I Note:
SoX converts all audio files to an internal uncompressed
format before performing any audio processing; this means that
@@ -132,12 +129,12 @@
SoX first decompresses the input MP3 file, then applies the
.B trim
effect, and finally creates the output MP3 file by recompressing the
-audio\*(EMwith a possible reduction in fidelity above that which
+audio\*mwith a possible reduction in fidelity above that which
occurred when the input file was created.
Hence, if what is ultimately desired is lossily compressed audio, it is
highly recommended to perform all audio processing using lossless file
formats and then convert to the lossy format at the final stage.
-.SP
+.P
.I Note:
Applying multiple effects with a single SoX invocation will,
in general, produce more accurate results than the equivalent using
@@ -147,18 +144,18 @@
level (or `volume') exceeds the range of the chosen representation.
It is nearly always undesirable and so should usually be corrected by
adjusting the volume prior to the point at which clipping occurs.
-.SP
+.P
In SoX, clipping could occur, as you might expect, when using the
.B vol
effect to increase the audio volume, but could also occur with many
other effects, when converting one format to another, and even when
simply playing the audio.
-.SP
+.P
Playing an audio file often involves re-sampling, and processing by
analogue components that can introduce a small DC offset and/or
amplification, all of which can produce distortion if the audio signal
level was initially too close to the clipping point.
-.SP
+.P
For these reasons, it is usual to make sure that an audio
file's signal level does not exceed around 70% of the maximum (linear)
range available, as this will avoid the majority of clipping problems.
@@ -167,11 +164,11 @@
effect can assist in determining the signal level in an audio file; the
.B vol
effect can be used to prevent clipping e.g.
-.SP
+.P
sox dull.au bright.au vol \-6 dB treble +6
-.SP
+.P
guarantees that the treble boost will not clip.
-.SP
+.P
If clipping occurs at any point during processing, then
SoX will display a warning message to that effect.
.SS Input File Balancing
@@ -181,7 +178,7 @@
volume adjustment effect) after the audio has been combined; however, it
is often useful to be able to set the volume of (i.e. `balance') the
inputs individually, before combining takes place.
-.SP
+.P
For all SoX combining methods (`concatenate', `mix', or `merge'), input
file volume adjustments can be made manually using the
.B \-v
@@ -189,13 +186,13 @@
given for only some of the input files then the others receive no volume
adjustment. See the next section for a description of the automatic volume
adjustments that can apply when mixing input files.
-.SP
+.P
The \fB\-V\fR option (below) can be used to show the input file volume
adjustments that have been selected (either manually or automatically).
.SS Input File Mixing
There are some special considerations that need to made when mixing
input files:
-.SP
+.P
Unlike `concatenate' and `merge', the `mix' combining method has the
potential to cause clipping in the combiner if no balancing is
performed. So for `mix', if manual volume adjustments are not given, to
@@ -204,40 +201,39 @@
where n is the number of input files. If this results in audio that is
too quiet or otherwise unbalanced then the input file volumes should be
set manually as described above.
-.SP
+.P
If mixed audio seems loud enough at some points through the audio but
too quiet in others, then dynamic-range compression should be applied to
-correct this\*(EMsee the
+correct this\*msee the
.B compand
effect.
.SS Examples
The command line syntax can seem complex, but in essence:
-.SP
+.P
sox file.au file.wav
-.SP
+.P
translates an audio file in SUN Sparc .AU format
into a Microsoft .WAV file, while
-.SP
- sox file.au \-r 12000 \-1 file.wav vol 0.5 dither
-.SP
+.P
+ sox file.au \-r 12000 \-1 file.wav vol 0\*d5 dither
+.P
performs the same format translation but also
changes the sampling rate to 12000 Hz,
the sample size to 1 byte (8 bits),
and applies the \fBvol\fR and \fBdither\fR effects
to the audio.
-.SP
+.P
sox short.au long.au longer.au
-.SP
+.P
concatenates two audio files to produce a single file, whilst
-.SP
+.P
sox \-m music.mp3 voice.wav mixed.flac
-.SP
+.P
mixes together two audio files.
-.SP
-See also the
+.P
+See the
.BR soxexam (1)
-manual page for a more detailed description of
-SoX and further examples on how to use
+manual page for further examples on how to use
SoX with various file formats and effects.
.SH OPTIONS
.SS Special File-name Options
@@ -287,7 +283,7 @@
and will be mixed together (instead of concatenated)
to form the output file.
A mixed audio file cannot be un-mixed.
-.SP
+.P
See also \fBInput File Mixing\fR above.
.TP
\fB\-M\fR, \fB\-\-merge\fR
@@ -299,7 +295,7 @@
files; a merged file could be un-merged using the
.B pick
effect.
-.SP
+.P
For example, two mono files could be merged to form one
stereo file; the first and second mono files would become
the left and right channels of the stereo file.
@@ -311,7 +307,7 @@
that supports the \fB\-o\fR option, SoX will output Octave
commands to plot the effect's transfer function, and then exit
without actually processing any audio. E.g.
-.SP
+.P
sox \-o input-file \-n highpass 1320 > plot.m
.br
octave plot.m
@@ -374,11 +370,11 @@
decreases the volume; greater than 1 increases it. If a negative number
is given, then in addition to the volume adjustment, the audio signal
will be inverted.
-.SP
+.P
See also the \fBstat\fR effect for information on how to find
the maximum volume of an audio file; this can be used to help select
suitable values for this option.
-.SP
+.P
See also \fBInput File Balancing\fR above.
.SS Input And Output File Format Options
These options apply to the input or output file whose name they
@@ -402,7 +398,7 @@
Gives the sample rate in Hz of the file. To cause the output file to have
a different sample rate than the input file, include this option as a part
of the output format options.
-.SP
+.P
If the input and output files have
different rates then a sample rate change effect must be run. Since
SoX has
@@ -413,7 +409,7 @@
Gives the type of the audio file. This is useful when the
file extension is non-standard or when the type can not be determined by
looking at the header of the file.
-.SP
+.P
The
.B \-t
option can also be used to override the type implied by an input file-name
@@ -420,7 +416,7 @@
extension, but if overriding with a type that has a header,
SoX will exit with an appropriate error message if such a header is not
actually present.
-.SP
+.P
See \fBFILE TYPES\fR below for a list of supported file types.
.PP
\fB\-L\fR, \fB\-\-endian=little\fR
@@ -452,12 +448,12 @@
The audio data encoding is signed linear (2's complement),
unsigned linear, \(*m-law (logarithmic), A-law (logarithmic),
ADPCM, IMA-ADPCM, GSM, or floating-point.
-.SP
+.P
\(*m-law (or mu-law) and A-law are the U.S. and
international standards for logarithmic telephone audio compression.
When uncompressed \(*m-law has roughly the precision of 14-bit PCM audio
and A-law has roughly the precision of 13-bit PCM audio.
-.SP
+.P
A-law and \(*m-law are sometimes encoded using reversed bit-ordering
(i.e. MSB becomes LSB). Internally, SoX understands how to work with
these encodings but there is currently no command line option to
@@ -464,7 +460,7 @@
specify them. If you need this support then you can use the pseudo
file types of `.la' and `.lu' to inform SoX of the encoding. See
supported file types for more information.
-.SP
+.P
ADPCM is a form of audio compression that has a good
compromise between good audio quality and fast encoding/decoding
time. It is used for telephone audio compression and places were
@@ -476,7 +472,7 @@
IMA ADPCM is a specific form of ADPCM compression, slightly simpler
and slightly lower fidelity than Microsoft's flavor of ADPCM.
IMA ADPCM is also called DVI ADPCM.
-.SP
+.P
GSM is currently used for the vast majority of the world's digital
wireless telephone calls. It utilises several audio
formats with different bit-rates and associated speech quality.
@@ -516,7 +512,7 @@
option (see above for details).
File types that can be determined
by a file-name extension are listed with their names preceded by a dot.
-.SP
+.P
.TP
.B .8svx
Amiga 8SVX musical instrument description format.
@@ -534,7 +530,7 @@
AIFF-C (not compressed, linear), defined in DAVIC 1.4 Part 9 Annex B.
This format is referred from ARIB STD-B24, which is specified for
Japanese data broadcasting. Any private chunks are not supported.
-.SP
+.P
Note: The infile is processed as .aiff currently.
.TP
.B alsa
@@ -561,7 +557,7 @@
can be dealt with using the
.B .ul
format (see below).
-.SP
+.P
It is possible to override AU file header information
with the
.B \-r
@@ -577,7 +573,7 @@
\&\fB.cdda\fR, \fB.cdr\fR
`Red Book' Compact Disc Digital Audio.
CDDA has two audio channels formatted as 16-bit
-signed integers at a sample rate of 44.1kHz. The number of (stereo)
+signed integers at a sample rate of 44\*d1kHz. The number of (stereo)
samples in each CDDA track is always a multiple of 588 which is why it
needs its own handler.
.TP
@@ -616,13 +612,13 @@
music. It is similar to MP3 and Ogg Vorbis, but lossless,
meaning that audio is compressed in FLAC without any loss in
quality.
-.SP
+.P
SoX can decode native FLAC files (.flac) but not Ogg FLAC files (.ogg).
[But see
.B .ogg
below for information relating to support for Ogg
Vorbis files.]
-.SP
+.P
SoX has basic support for writing FLAC files: it can encode to
native FLAC using compression levels 0 to 8. 8 is the default
compression level and gives the best (but slowest) compression;
@@ -630,7 +626,7 @@
level can be selected using the
.B \-C
option (see above) with a whole number from 0 to 8.
-.SP
+.P
Note that Replay Gain information is not used by
SoX if present in FLAC input files and is not generated by
SoX for FLAC output files, however
@@ -640,7 +636,7 @@
file. In this case the Replay Gain information in the output file is
likely to be incorrect and so should be recalculated using a tool that
supports this (not SoX).
-.SP
+.P
FLAC support in
SoX is optional and requires optional FLAC libraries. To
see if there is support for FLAC run \fBsox \-h\fR and look for
@@ -654,7 +650,7 @@
lots of noise when a given audio signal is encoded and decoded
multiple times. This format is used by some voice mail applications.
It is rather CPU intensive.
-.SP
+.P
GSM in
SoX is optional and requires access to an external GSM library. To see
if there is support for GSM run \fBsox \-h\fR
@@ -683,13 +679,13 @@
quality loss. See also
.B Ogg Vorbis
for a similar format.
-.SP
+.P
MP3 support in
SoX is optional and requires access to either or both the external
libmad and libmp3lame libraries. To
see if there is support for Mp3 run \fBsox \-h\fR
and look for it under the list of supported file formats as `mp3'.
-.SP
+.P
.TP
.B null
Null file type.
@@ -699,22 +695,22 @@
special file-name
.B \-n
in place of an input or output file-name.
-.SP
+.P
Using this file type to input audio is equivalent to
using a normal audio file that contains an infinite amount
of silence, and as such is not generally useful unless used
with an effect that specifies a finite time length
(such as \fBtrim\fR or \fBsynth\fR).
-.SP
+.P
Using this type to output audio amounts to discarding the audio
and is useful mainly with effects that produce information about the
audio instead of affecting it
(such as \fBnoiseprof\fR or \fBstat\fR).
-.SP
+.P
The number of channels and the sampling rate associated with a null file
-are by default 2 and 44.1kHz respectively, but these can be overridden
+are by default 2 and 44\*d1kHz respectively, but these can be overridden
if necessary by using appropriate \fBFormat Options\fR.
-.SP
+.P
One other use of the null file type is to use it in conjunction
with
.B \-V
@@ -732,7 +728,7 @@
quality loss. See also
.B MP3
for a similar format.
-.SP
+.P
SoX can decode all types of Ogg Vorbis files, and can encode at different
compression levels/qualities given as a number from \-1 (highest
compression/lowest quality) to 10 (lowest compression, highest quality).
@@ -740,10 +736,10 @@
of approx. 112kbps), but this can be changed using the
.B \-C
option (see above) with a number from \-1 to 10; fractional numbers (e.g.
-3.6) are also allowed.
-.SP
+3\*d6) are also allowed.
+.P
Decoding is somewhat CPU intensive and encoding is very CPU intensive.
-.SP
+.P
Ogg Vorbis in
SoX is optional and requires access to external Ogg Vorbis libraries. To
see if there is support for Ogg Vorbis run \fBsox \-h\fR
@@ -813,7 +809,7 @@
.B .txw
Yamaha TX-16W sampler.
A file format from a Yamaha sampling keyboard which wrote IBM-PC
-format 3.5" floppies. Handles reading of files which do not have
+format 3\*d5" floppies. Handles reading of files which do not have
the sample rate field set to one of the expected by looking at some
other bytes in the attack/loop length fields, and defaulting to
33kHz if the sample rate is still unknown.
@@ -846,7 +842,7 @@
.B .wav
Microsoft .WAV RIFF files.
This is the native audio file format of Windows, and widely used for uncompressed audio.
-.SP
+.P
Normally \fB.wav\fR files have all formatting information
in their headers, and so do not need any format options
specified for an input file. If any are, they will
@@ -854,7 +850,7 @@
You had better know what you are doing! Output format
options will cause a format conversion, and the \fB.wav\fR
will written appropriately.
-.SP
+.P
SoX currently can read PCM, ULAW, ALAW, MS ADPCM, and IMA (or DVI) ADPCM.
It can write all of these formats including the ADPCM encoding.
Big endian versions of RIFF files, called RIFX, can also be read
@@ -894,7 +890,7 @@
.SH EFFECTS
Multiple effects may be applied to the audio by specifying them
one after another at the end of the command line.
-.SP
+.P
Optionality is denoted by brackets \fB[ ]\fR;
multiplicity is denoted by braces \fB{ }\fR or an ellipsis \fB...\fR;alternatives are indicated with a vertical bar \fB|\fR.
@@ -912,7 +908,7 @@
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
+.P
The
.B avg
effect can also be invoked with up to 16
@@ -925,10 +921,10 @@
rb \*(RA rf.
The next 4 give the right-front output in the same order, then
left-back and right-back.
-.SP
+.P
It is also possible to use the 16 numbers to expand or reduce the
channel count; just specify 0 for unused channels.
-.SP
+.P
Finally, certain reduced combination of numbers can be specified
for certain input/output channel combinations.
.TS
@@ -943,7 +939,7 @@
4 4 1 adjust balance
4 4 2 front balance, back balance
.TE
-.SP
+.P
.TP
band \fB[\fR\-n\fB]\fR \fIcenter\fR \fB[\fR\fIwidth\fR\fB]\fR
Apply a band-pass filter.
@@ -976,9 +972,9 @@
i.e. peaking at the
.I center
frequency and settling around it.
-.SP
+.P
This effect supports the \fB\-o\fR global option (see above).
-.SP
+.P
See also \fBfilter\fR for a bandpass filter with steeper shoulders.
.TP
bandpass\fB\^|\^\fRbandreject \fIfrequency bandwidth\fR
@@ -987,7 +983,7 @@
and bandwidth (in Hz, and as determined by the 3dB points)
\fIbandwidth\fR.
The filter rolls off at 6dB per octave (20dB per decade).
-.SP
+.P
These effects support the \fB\-o\fR global option (see above).
.TP
bandreject \fIfrequency bandwidth\fR
@@ -999,30 +995,30 @@
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 or EQ.
-.SP
+.P
\fIgain\fR gives the dB gain at 0Hz (for \fBbass\fR), or whichever is
-the lower of ~22kHz and the Nyquist frequency (for \fBtreble\fR). Its
+the lower of \(ap22kHz 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
+.P
If desired, the filter can be fine-tuned using the following
optional parameters (in either order):
-.SP
+.P
\fIfrequency\fR sets the filter's center frequency and so can be
used to extend or reduce the frequency range to be boosted or
cut. The default value is 100Hz (for \fBbass\fR) or 3kHz (for
\fBtreble\fR).
-.SP
+.P
\fIslope\fR is a number between 0 and 1 that determines how
steep the filter's shelf transition is. Its useful range is
-about 0.3 (for a gentle slope) to 1 (for a steep slope). The
-default value is 0.5.
-.SP
+about 0\*d3 (for a gentle slope) to 1 (for a steep slope). The
+default value is 0\*d5.
+.P
These effects support the \fB\-o\fR global option (see above).
-.SP
+.P
See also \fBequalizer\fR for a peaking equalisation effect.
.TP
chorus \fIgain-in gain-out\fR \fB{\fR \fIdelay decay speed depth\fR \-s\fB\^|\^\fR\-t \fB}\fR@@ -1037,7 +1033,7 @@
\fIin-dB1\fR,\fIout-dB1\fR\fB[\fR,\fIin-dB2\fR,\fIout-dB2\fR\fB...]\fR
.br
\fB[\fR\fIgain\fR \fB[\fR\fIinitial-volume\fR \fB[\fR\fIdelay\fR\fB] ] ]\fR
-.SP
+.P
Compand (compress or expand) the dynamic range of the audio. The
attack and decay time specify the integration time over which the
absolute value of the input signal is integrated to determine its
@@ -1052,7 +1048,7 @@
be used to indicate that the input volume should be associated output
volume. The points \fB\-inf,\-inf\fR and \fB0,0\fR are assumed; the
latter may be overridden, but the former may not.
-.SP
+.P
The third
(optional) parameter is a post-processing gain in dB which is applied
after the compression has taken place; the fourth (optional) parameter
@@ -1062,7 +1058,7 @@
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.
-.SP
+.P
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.
@@ -1075,14 +1071,14 @@
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
+.P
The first option is the \fIdcshift\fR value. It is a floating point number that
indicates the amount to shift.
-.SP
+.P
An optional
.I limitergain
can be specified as well. It should have a value much less than 1
-(e.g. 0.05 or 0.02) and is used only on peaks to prevent clipping.
+(e.g. 0\*d05 or 0\*d02) and is used only on peaks to prevent clipping.
.TP
deemph
Apply a treble attenuation shelving filter to audio in
@@ -1089,9 +1085,9 @@
audio-CD format. The frequency response of pre-emphasized
recordings is rectified. The filtering is defined in the
standard document ISO 908.
-.SP
+.P
This effect supports the \fB\-o\fR global option (see above).
-.SP
+.P
.TP
dither \fB[\fR\fIdepth\fR\fB]\fR
Apply dithering to the audio.
@@ -1098,10 +1094,10 @@
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 1/2 bit;
+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
+.P
This effect should not be followed by any other effect that
affects the audio.
.TP
@@ -1137,7 +1133,7 @@
around (\fIQ\fR) a central frequency (\fIcentral-frequency\fR),
leaving all other frequencies untouched (unlike
bandpass/bandreject filters).
-.SP
+.P
\fIcentral-frequency\fR is the filter's central frequency in Hz, \fIQ\fR
its `Q-factor' (see http://en.wikipedia.org/wiki/Q_factor), and
\fIgain\fR is the gain or attenuation in dB.
@@ -1144,19 +1140,19 @@
Beware of
.B Clipping
when using a positive \fIgain\fR.
-.SP
+.P
In order to produce complex equalisation curves, this effect
can be given several times, each with a different central frequency.
-.SP
+.P
This effect supports the \fB\-o\fR global option (see above).
-.SP
+.P
See also \fBbass\fR and \fBtreble\fR for shelving equalisation effects.
.TP
fade \fB[\fR\fItype\fR\fB]\fR \fIfade-in-length\fR \fB[\fR\fIstop-time\fR \fB[\fR\fIfade-out-length\fR\fB] ]\fR
Add a fade effect to the beginning, end, or both of the audio.
-.SP
+.P
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
+.P
For fade-outs, the audio will be truncated at
.I stop-time
and
@@ -1168,12 +1164,12 @@
No fade-out is performed if
.I stop-time
is not specified.
-.SP
+.P
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
+.P
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
filter \fB[\fR\fIlow\fR\fB]\fR\-\fB[\fR\fIhigh\fR\fB] [\fR\fIwindow-len\fR \fB[\fR\fIbeta\fR\fB]]\fR
@@ -1181,18 +1177,18 @@
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
+.P
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
+.P
The \fIwindow-len\fR, if unspecified, defaults to 128.
Longer windows give a sharper cutoff, smaller windows a more gradual cutoff.
-.SP
+.P
The \fIbeta\fR, if unspecified, defaults to 16. This selects a Kaiser window.
-You can select a Nuttall window by specifying anything <= 2 here.
+You can select a Nuttall window by specifying anything \(<= 2 here.
For more discussion of beta, look under the \fBresample\fR effect.
-.SP
+.P
.TP
flanger \fB[\fR\fIdelay depth regen width speed shape phase interp\fR\fB]\fR
Apply a flanging effect to the audio.
@@ -1212,7 +1208,7 @@
.na
Percentage of delayed signal mixed with original.
T}
-speed 0.1 \- 10 0.5 Sweeps per second (Hz).
+speed 0\*d1 \- 10 0\*d5 Sweeps per second (Hz).
shape \ sin Swept wave shape: sine\^|\^triangle.
phase 0 \- 100 25 T{.na
@@ -1224,15 +1220,15 @@
Digital delay-line interpolation: linear\^|\^quadratic.
T}
.TE
-.SP
+.P
.TP
highp\fB\^|\^\fRlowp \fIfrequency\fR
Apply a single-pole recursive high-pass or low-pass filter with
3dB point \fIfrequency\fR.
The filters roll off at 6dB per octave (20dB per decade).
-.SP
+.P
These effects support the \fB\-o\fR global option (see above).
-.SP
+.P
See also \fBfilter\fR for filters with a sharper cutoff.
.TP
highpass\fB\^|\^\fRlowpass \fIfrequency\fR
@@ -1239,7 +1235,7 @@
Apply a two-pole Butterworth high-pass or low-pass filter with
3dB point \fIfrequency\fR.
The filters roll off at 12dB per octave (40dB per decade).
-.SP
+.P
These effects support the \fB\-o\fR global option (see above).
.TP
lowp \fIfrequency\fR
@@ -1258,7 +1254,7 @@
\fIin-dB1,out-dB1\fR\fB[\fR,\fIin-dB2,out-dB2\fR\fB...]\fR
.br
\fB[\fR\fIgain\fR \fB[\fR\fIinitial-volume\fR \fB[\fR\fIdelay\fR\fB] ] ]\fR" \fIxover-freq\fR
-.SP
+.P
Multi-band compander is similar to the single band compander but
the audio is first divided up into bands and then the compander
is run on each band. See the \fBcompand\fR effect for the definition of its options. Compand options are specified between double quotes and the crossover frequency for that band is specified separately with \fIxover-fre\fR. This can be repeated multiple times to create multiple bands.
@@ -1276,12 +1272,12 @@
to \fIprofile-file\fR, or to stdout if no \fIprofile-file\fR is specified.
If there is audio output on stdout then the profile will instead be directed to
stderr.
-.SP
+.P
To actually remove the noise, run
SoX again with the \fInoisered\fR filter. The
filter needs one parameter, \fIprofile-file\fR, which contains the noise profile
from \fBnoiseprof\fR. \fIthreshold\fR specifies how much noise should be removed, and
-may be between 0 and 1 with a default of 0.5. Higher values will remove more
+may be between 0 and 1 with a default of 0\*d5. Higher values will remove more
noise but present a greater likelyhood of distorting the desired audio signal.
Experiment with different threshold values to find the optimal one for your
audio.
@@ -1304,8 +1300,8 @@
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 1.5 1.5
-adds 1.5 seconds of silence padding at each end of the audio, whilst
+.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
@@ -1330,7 +1326,7 @@
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 0.5 to avoid
+(\fB\-t\fR). The decay should be less than 0\*d5 to avoid
feedback. Gain-out is the volume of the output.
.TP
pick \fB[\fR\-1\fB\^|\^\fR\-2\fB\^|\^\fR\-3\fB\^|\^\fR\-4\fB\^|\^\fR\-l\fB\^|\^\fR\-r\fB\^|\^\fR\-f\fB\^|\^\fR\-b\fB]\fR
@@ -1357,11 +1353,11 @@
polyphase \fB[\fR\-w nut\fB\^|\^\fRham\fB] [\fR\-width long\fB\^|\^\fRshort\fB\^|\^\fR\fIn\fR\fB] [\fR\-cutoff \fIc\fR\fB]\fR
Change the sampling rate using `polyphase interpolation', a DSP algorithm.
This method is relatively slow and memory intensive.
-.SP
+.P
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 Nutall.
-.SP
+.P
The \fB\-width\fR parameter specifies the (approximate) width of the filter.
.B long
is 1024 samples;
@@ -1372,7 +1368,7 @@
The
.B short
option is not recommended, as it produces poor quality results.
-.SP
+.P
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
@@ -1379,8 +1375,8 @@
further information on Nyquist frequency. If up-sampling, then this is the
fraction of the original signal
that should go through. If down-sampling, this is the fraction of the
-signal left after down-sampling. The default is 0.95.
-.SP
+signal left after down-sampling. The default is 0\*d95.
+.P
See also
.B rabbit
and
@@ -1396,7 +1392,7 @@
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
+.P
See also
.B polyphase
and
@@ -1421,13 +1417,13 @@
\fBresample\fR and \fBpolyphase\fR at
http://leute.server.de/wilde/resample.html; see \fBrabbit\fR for a
pointer to its own documentation.
-.SP
+.P
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 0.80 of
+in the case that you want to have roll-off greater than about 0\*d8 of
the Nyquist frequency.
-.SP
+.P
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.
@@ -1434,10 +1430,10 @@
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
+.P
Following is a table of the reasonable defaults which are built-in to
SoX:
-.SP
+.P
.TS
center box;
cB cB cB cB cB
@@ -1444,35 +1440,35 @@
c c n c c
cB c n c c.
Option Window Roll-off Beta Interpolation
-(none) 45 0.80 16 linear
-\-qs 45 0.80 16 quadratic
-\-q 75 0.875 16 quadratic
-\-ql 149 0.94 16 quadratic
+(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
-.SP
+.P
\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
+.P
\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 0.8\-0.95. The defaults are
+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 (sample rate / 2). Logically,
+.P
+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 highpass filter first to avoid these problems.
-.SP
+a lowpass filter first to avoid these problems.
+.P
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
+.P
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
@@ -1481,36 +1477,36 @@
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
+.P
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.
+the beta for a Kaiser window. Beta \(<= 2 selects a Nuttall window.
If unspecified, the default is a Kaiser window with beta 16.
-.SP
+.P
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
+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
+.P
This is the default effect if the two files have different sampling rates.
Default parameters are, as indicated above, Kaiser window of length 45,
-roll-off 0.80, beta 16, linear interpolation.
-.SP
+roll-off 0\*d80, beta 16, linear interpolation.
+.P
\fBNOTE:\fR \fB\-qs\fR is only slightly slower, but more accurate for
16-bit or higher precision.
-.SP
+.P
\fBNOTE:\fR 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
-.SP
+.P
.ce 3
input-rate < output-rate
and
-output-rate / gcd(input-rate, output-rate) <= 511
+output-rate \(di gcd(input-rate, output-rate) \(<= 511
.TP
reverb \fIgain-out reverb-time delay\fR \fB[\fR\fIdelay\fR \fB... ]\fR
Add reverberation to the audio. Each
@@ -1532,9 +1528,9 @@
Requires disk space to store the data to be reversed.
.TP
silence \fIabove-periods\fR \fB[\fR\fIduration threshold\fR\fB[\fRd\fB\^|\^\fR%\fB] [\fR\fIbelow-periods duration threshold\fR\fB[\fRd\fB\^|\^\fR%\fB] ]\fR
-.SP
+.P
Removes silence from the beginning, middle, or end of the audio. Silence is anything below a specified threshold.
-.SP
+.P
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
@@ -1545,16 +1541,16 @@
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
+.P
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
+.P
\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
+.P
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.
@@ -1563,7 +1559,7 @@
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
+.P
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.
@@ -1570,7 +1566,7 @@
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
+.P
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.
@@ -1577,7 +1573,7 @@
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
+.P
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
@@ -1584,7 +1580,7 @@
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
+.P
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
@@ -1599,7 +1595,7 @@
`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
+.P
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
@@ -1610,7 +1606,7 @@
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
+.P
The `Volume Adjustment:' field in the statistics
gives you the parameter to the
.B \-v
@@ -1619,13 +1615,13 @@
Note: See the discussion on
.B Clipping
above for reasons why it is rarely a good idea to actually do this.
-.SP
+.P
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
+.P
The
.B \-s
option is used to scale the input data by a given factor. The default value
@@ -1634,16 +1630,16 @@
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
+.P
The
.B \-rms
option will convert all output average values to `root mean square'
format.
-.SP
+.P
The
.B \-freq
option calculates the input's power spectrum and prints it to standard error.
-.SP
+.P
There is also an optional parameter
.B \-d
that will print out a hex dump of the
@@ -1650,7 +1646,6 @@
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.
-.SP
.TP
stretch \fIfactor\fR \fB[\fR\fIwindow fade shift fading\fR\fB]\fR
Time stretch the audio by the given factor. Changes duration without affecting the pitch.
@@ -1662,9 +1657,9 @@
option, can be `lin'.
.I shift
ratio, in \fB[\fR0 1\fB]\fR. Default depends on stretch factor. 1
-to shorten, 0.8 to lengthen. The
+to shorten, 0\*d8 to lengthen. The
.I fading
-ratio, in \fB[\fR0 0.5\fB]\fR. The amount of a fade's default depends on
+ratio, in \fB[\fR0 0\*d5\fB]\fR. The amount of a fade's default depends on
.I factor
and \fIshift\fR.
.TP
@@ -1689,7 +1684,7 @@
the output from the previous stage.
Audio for each channel in a multi-channel audio file can be synthesised
independently.
-.SP
+.P
Though this effect is used to generate audio, an input file must
still be specified. This can be used to set the synthesised audio
length, the number of channels, and the sampling rate, however since the
@@ -1697,78 +1692,78 @@
.B null
file `\fB\-n\fR' is usually used instead (and the length specified
as a parameter to \fBsynth\fR).
-.SP
-For example, the following produces a 3 second, 44.1kHz,
+.P
+For example, the following produces a 3 second, 44\*d1kHz,
stereo audio file containing a sine-wave swept from 300 to 3300 Hz.
-.SP
+.P
sox \-n output.au synth 3 sine 300\-3300
-.SP
+.P
This produces an 8kHz mono version:
-.SP
+.P
sox \-r 8000 \-c 1 \-n output.au synth 3 sine 300\-3300
-.SP
+.P
Multiple channels can be synthesised by specifying the set of
parameters shown between braces (\fB{}\fR) multiple times;the following puts the swept tone in the left channel and adds `brown'
noise in the right:
-.SP
+.P
sox \-n output.au synth 3 sine 300\-3300 brownnoise
-.SP
+.P
The following example shows how two synth effects can be cascaded
to create a more complex waveform:
-.SP
- sox \-n output.au synth .5 sine 200\-500 synth .5 sine fmod 700\-100
-.SP
+.P
+ sox \-n output.au synth 0\*d5 sine 200\-500 synth 0\*d5 sine fmod 700\-100
+.P
Frequencies can also specified in terms of musical semitones relative to
`middle A' (440Hz); the following could be used to help tune
a guitar's `low E' string (on a system that supports
\fBalsa\fR):
-.SP
+.P
sox \-n \-t alsa default synth sine %\-5
-.SP
+.P
The following produces a chord with a pipe-organ sound:
-.SP
- sox \-c4 \-n \-c1 Am7.au synth sin %0 sin %3 sin %7 sin %10 avg fade q .1 1 .1
-.SP
+.P
+ sox \-c4 \-n \-c1 Am7.au synth sin %0 sin %3 sin %7 sin %10 avg fade q 0\*d1 1 0\*d1
+.P
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 select a suitable attenuation.
-.SP
+.P
A detailed description of each
.B synth
parameter follows:
-.SP
+.P
\fIlen\fR is the length of audio to synthesise expressed as a time
or as a number of samples;
0=inputlength, default=0.
-.SP
+.P
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
+.P
\fItype\fR is one of sine, square, triangle, sawtooth, trapezium, exp,
\fB[\fRwhite\fB]\fRnoise, pinknoise, brownnoise; default=sine
-.SP
+.P
\fIcombine\fR is one of create, mix, amod (amplitude modulation), fmod
(frequency modulation); default=create
-.SP
+.P
\fIfreq\fR/\fIfreq2\fR are the frequencies at the beginning/end of
synthesis in Hz or, if preceded with `%', semitones relative to A
(440Hz); for both, default=%0. Not used for noise.
-.SP
+.P
\fIoff\fR is the bias (DC-offset) of the signal in percent; default=0.
-.SP
+.P
\fIph\fR is the phase shift in percentage of 1 cycle; default=0. Not
used for noise.
-.SP
+.P
\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
+.P
\fIp2\fR trapezium: the percentage through each cycle at which `falling'
begins; default=50. exp: the amplitude in percent; default=100.
-.SP
+.P
\fIp3\fR trapezium: the percentage through each cycle at which `falling'
ends; default=60.
.TP
@@ -1780,16 +1775,16 @@
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
+.P
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
+.P
Both options can be specified using either an amount of time or an
exact count of samples. The format for specifying lengths in time is
-hh:mm:ss.frac. A start value of 1:30.5 will not start until 1 minute,
-thirty and 1/2 seconds into the audio. The format for specifying
+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.
@@ -1803,7 +1798,7 @@
.I speed
(0 to 30), and the modulation depth by
.I depth
-(0 to 1, default 0.5).
+(0 to 1, default 0\*d5).
.TP
vol \fIgain\fR \fB[\fR\fItype\fR \fB[\fR\fIlimitergain\fR\fB] ]\fR
Apply an amplification or an attenuation to the audio signal.
@@ -1814,7 +1809,7 @@
.B vol
is an effect like any other so can be applied anywhere, and several times
if necessary, during the processing chain.
-.SP
+.P
The amount to change the volume is given by
.I gain
which is interpreted, according to the given \fItype\fR, as follows: if
@@ -1824,7 +1819,7 @@
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
+.P
When
.I type
is \fBamplitude\fR or \fBpower\fR, a
@@ -1835,7 +1830,7 @@
a negative
.I gain
inverts the audio signal in addition to adjusting its volume.
-.SP
+.P
When
.I type
is \fBdB\fR, a
@@ -1843,18 +1838,18 @@
of 0 leaves the volume unchanged,
less than 0 decreases it,
and greater than 0 increases it.
-.SP
+.P
See http://en.wikipedia.org/wiki/Decibel
for a detailed discussion on electrical (and hence audio signal)
voltage and power ratios.
-.SP
+.P
Beware of
.B Clipping
when the increasing the volume.
-.SP
+.P
An optional \fIlimitergain\fR value can be specified and should be a
value much less
-than 1 (e.g. 0.05 or 0.02) and is used only on peaks to prevent clipping.
+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.
@@ -1868,17 +1863,17 @@
.BR play (1),
.BR rec (1),
.BR soxexam (1)
-.SP
+.P
The SoX web page at http://sox.sourceforge.net
.SH LICENSE
Copyright 1991 Lance Norskog and Sundry Contributors.
Copyright 1998\-2007 by Chris Bagwell and SoX Contributors.
-.SP
+.P
This program is free software; you can redistribute it and/or modify
it under the terms of the GNU General Public License as published by
the Free Software Foundation; either version 2, or (at your option)
any later version.
-.SP
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This program is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the