ref: a733cacd430b07b910cf1ed288df3b779277703a
parent: 0a33a06e6109daa2edc8d2188a23418f687e11d1
author: robs <robs>
date: Sat Jan 6 11:18:29 EST 2007
Ongoing clean-ups.
--- a/sox.1
+++ b/sox.1
@@ -1,5 +1,5 @@
'\" t
-'\" The line above instructs some `man' programs to invoke tbl
+'\" The line above instructs most `man' programs to invoke tbl
'\"
'\" Separate paragraphs; not the same as .PP which resets indent level.
.de SP
@@ -22,9 +22,9 @@
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
- \fB[[\fR\fIformat-options\fR\fB]\fR \fIinfile2\fR \fB...] [\fR\fIformat-options\fR\fB]\fR \fIoutfile\fR
- \fB[\fR\fIeffect\fR \fB[\fR\fIeffect-options\fR\fB] ...]\fR
+\fBsox\fR [\fIglobal-options\fR] [\fIformat-options\fR] \fIinfile1\fR
+ [ [\fIformat-options\fR] \fIinfile2\fR ] ... [\fIformat-options\fR] \fIoutfile\fR
+ [\fIeffect\fR [\fIeffect-options\fR] ] ...
.fi
.SH DESCRIPTION
SoX reads and writes audio files in most popular formats and can
@@ -127,18 +127,21 @@
measure of how closely the original audio signal can be reproduced when
using a lossy format.
.SP
-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.
+Audio file conversion with SoX is lossless when it can be, i.e. when not
+using lossy compression, when not reducing the sampling rate or number
+of channels, and when 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
-.I Note:
+.B N.B.
SoX converts all audio files to an internal uncompressed
format before performing any audio processing; this means that
manipulating a file that is stored in a lossy format can cause further
losses in audio fidelity. E.g. with
-.BR "sox long.mp3 short.mp3 trim 10" ,
+.SP
+ sox long.mp3 short.mp3 trim 10
+.SP
SoX first decompresses the input MP3 file, then applies the
.B trim
effect, and finally creates the output MP3 file by recompressing the
@@ -148,9 +151,9 @@
highly recommended to perform all audio processing using lossless file
formats and then convert to the lossy format at the final stage.
.SP
-.I Note:
+.B N.B.
Applying multiple effects with a single SoX invocation will,
-in general, produce more accurate results than the equivalent using
+in general, produce more accurate results than those produced using
multiple SoX invocations; hence this is also recommended.
.SS Clipping
Clipping is distortion that occurs when an audio signal
@@ -176,7 +179,7 @@
.B stat
effect can assist in determining the signal level in an audio file; the
.B vol
-effect can be used to prevent clipping e.g.
+effect can be used to prevent clipping, e.g.
.SP
sox dull.au bright.au vol \-6 dB treble +6
.SP
@@ -221,33 +224,46 @@
.B compand
effect.
.SS Examples
-The command line syntax can seem complex, but in essence:
+The simple command:
.SP
- sox file.au file.wav
+ sox recital.au recital.wav
.SP
translates an audio file in Sun .au format
-into a Microsoft WAV file, while
+to a Microsoft WAV file, whilst:
.SP
- sox file.au \-r 12000 \-1 file.wav vol 0\*d5 dither
+ sox recital.au \-r 12000 \-b \-c 1 recital.wav vol 0\*d7 dither
.SP
-performs the same format translation but also
+performs the same format translation, but also
changes the sampling rate to 12000Hz,
-the sample size to 1 byte (8 bits),
+the sample size to one byte (8 bits),
+the number of channels to one (mono),
and applies the \fBvol\fR and \fBdither\fR effects
to the audio.
.SP
+Further examples:
+.SP
+ sox \-r 8000 \-u \-b \-c 1 voice-memo.raw voice-memo.wav
+.SP
+adds a header to a raw audio file,
+.SP
+ sox slow.aiff fixed.aiff speed 1\*d027 rabbit -c0
+.SP
+adjusts audio speed using the most accurate
+.B rabbit
+algorithm,
+.SP
sox short.au long.au longer.au
.SP
-concatenates two audio files to produce a single file, whilst
+concatenates two audio files, and
.SP
sox \-m music.mp3 voice.wav mixed.flac
.SP
mixes together two audio files.
.SP
-See the
+More examples can found thoughout this manual and in the
+separate
.BR soxexam (1)
-manual page for further examples on how to use
-SoX with various file formats and effects.
+manual.
.SH OPTIONS
.SS Special File-name Options
Each of these options is used in certain circumstances in place of a normal
@@ -274,7 +290,7 @@
\fB\-e\fR
This is an alias of
.B \-n
-which is retained for backwards compatibility only.
+and is retained for backwards compatibility only.
.SS Global Options
These options can be specified on the command line at any point
before the first effect name.
@@ -282,11 +298,11 @@
\fB\-h\fR, \fB\-\-help\fR
Show version number and usage information.
.TP
-\fB\-\-help\-effect=\fR\fIname\fR
+\fB\-\-help\-effect=\fIname\fR
Show usage information on the specified effect. The name
\fBall\fR can be used to show usage on all effects.
.TP
-\fB\fB\-\-interactive\fR
+\fB\-\-interactive\fR
Prompt before overwriting an existing file with the same name as that
given for the output file.
.TP
@@ -337,7 +353,7 @@
.TP
\fB\-\-version\fR
Show version number and exit.
-.IP \fB\-V\fB[\fR\fIlevel\fR\fB]\fR\fP
+.IP \fB\-V\fB[\fIlevel\fB]\fP
Set verbosity.
SoX prints messages to the console (stderr) according to the following
verbosity levels:
@@ -374,10 +390,10 @@
sets it to 0.
.IP
.SS Input File Options
-These options apply to only input files and may only precede input
+These options apply only to input files and may precede only input
file-names on the command line.
.TP
-\fB\-v \fIvolume\fR, \fB\-\-volume=\fR\fIvolume\fR
+\fB\-v \fIvolume\fR, \fB\-\-volume=\fIvolume\fR
Adjust volume by a factor of \fIvolume\fR.
This is a linear (amplitude) adjustment, so a number less than 1
decreases the volume; greater than 1 increases it. If a negative number
@@ -389,13 +405,13 @@
suitable values for this option.
.SP
See also \fBInput File Balancing\fR above.
-.SS Input And Output File Format Options
+.SS Input & Output File Format Options
These options apply to the input or output file whose name they
-immediately precede on the command line; they are used mainly when
+immediately precede on the command line and are used mainly when
working with headerless file formats or when specifying a format
for the output file that is different to that of the input file.
.TP
-\fB\-c \fIchannels\fR, \fB\-\-channels=\fR\fIchannels\fR
+\fB\-c \fIchannels\fR, \fB\-\-channels=\fIchannels\fR
The number of audio channels in the audio file.
This may be 1, 2, or 4; for mono, stereo, or quad audio. To cause
the output file to have a different number of channels than the input
@@ -407,10 +423,10 @@
effect is not specified on the
command line it will be invoked internally with default parameters.
.TP
-\fB\-r \fIrate\fR, \fB\-\-rate=\fR\fIrate\fR
+\fB\-r \fIrate\fR, \fB\-\-rate=\fIrate\fR
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.
+a different sample rate than the input file, include this option with
+the output file format options.
.SP
If the input and output files have
different rates then a sample rate change effect must be run. Since
@@ -418,7 +434,7 @@
multiple rate changing effects, the user can specify which to use as an effect.
If no rate change effect is specified then a default one will be chosen.
.TP
-\fB\-t \fIfile-type\fR, \fB\-\-type=\fR\fIfile-type\fR
+\fB\-t \fIfile-type\fR, \fB\-\-type=\fIfile-type\fR
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.
@@ -452,7 +468,7 @@
is actually an audio device.
.TP
\fB\-X\fR, \fB\-\-reverse\-bits\fR
-Specifies that the bit ordering should be reversed; currently only used
+Specifies that the bit ordering should be reversed; currently used only
with the
.B cvsd
format.
@@ -500,7 +516,7 @@
\fB\-1\fR\^/\fB\-2\fR\^/\fB\-4\fR\^/\fB\-8\fR.
Abbreviations of: byte, word, long word, double long (long long) word.
.SS Output File Format Options
-These options apply to only the output file and may only precede the output
+These options apply only to the output file and may precede only the output
file-name on the command line.
.TP
\fB\-\-comment \fItext\fR
@@ -511,7 +527,7 @@
Specify a file containing the comment text to store in the output
file header (where applicable).
.TP
-\fB\-C \fIcompression-factor\fR, \fB\-\-compression=\fR\fIcompression-factor\fR
+\fB\-C \fIcompression-factor\fR, \fB\-\-compression=\fIcompression-factor\fR
The compression factor for variably compressing output file formats. If
this option is not given, then a default compression factor will apply.
The compression factor is interpreted differently for different
@@ -722,7 +738,7 @@
.SP
The number of channels and the sampling rate associated with a null file
are by default 2 and 44\*d1kHz respectively, but these can be overridden
-if necessary by using appropriate \fBFormat Options\fR.
+if necessary by using appropriate command-line format options.
.SP
One other use of the null file type is to use it in conjunction
with
@@ -882,11 +898,9 @@
difficult.
.TP
.B .raw
-Raw files (no header).
-The sample rate, size (byte, word, etc),
-and encoding (signed, unsigned, etc.)
-of the audio file must be given.
-The number of channels defaults to 1.
+Raw (headerless) audio files. The sample rate, sample size, and data
+encoding must be given using command-line format options; the number of
+channels defaults to 1.
.TP
.B .ub .sb .uw .sw .ul .al .lu .la .sl
These file-name extensions serve as shorthand for identifying the format
@@ -898,18 +912,21 @@
inverse bit order `A-law', or `signed long' respectively. Command-line
format options can also be given to modify the selected format if it
does not provide an exact match for a particular file.
+.SP
Headerless audio files on a `Sparc' computer are likely to be of format
\fBul\fR; on a `Mac' computer, they're likely to be \fBub\fR but with a
-sample rate of 11025 or 22050Hz.
+sample rate of 11025 or 22050 Hz.
.SH EFFECTS
Multiple effects may be applied to the audio by specifying them
one after another at the end of the command line.
.SP
-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.
+.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
-avg \fB[\fR\-l\fB\^|\^\fR\-r\fB\^|\^\fR\-f\fB\^|\^\fR\-b\fB\^|\^\fR\-1\fB\^|\^\fR\-2\fB\^|\^\fR\-3\fB\^|\^\fR\-4\fB\^|\^\fR\fIn\fR,\fIn\fR,\fB...\fR,\fIn\fR\fR\fB]\fR
+\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 channels by averaging the samples,
or duplicate channels to increase the number of channels.
This effect is automatically used when the number of input
@@ -955,7 +972,7 @@
.TE
.SP
.TP
-band \fB[\fR\-n\fB]\fR \fIcenter\fR \fB[\fR\fIwidth\fR\fB]\fR
+\fBband\fR [\fB\-n\fR] \fIcenter\fR [\fIwidth\fR]
Apply a band-pass filter.
The frequency response drops logarithmically
around the
@@ -977,7 +994,7 @@
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.
+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.
@@ -991,7 +1008,7 @@
.SP
See also \fBfilter\fR for a bandpass filter with steeper shoulders.
.TP
-bandpass\fB\^|\^\fRbandreject \fIfrequency bandwidth\fR
+\fBbandpass\fR\^|\^\fBbandreject \fIfrequency bandwidth\fR
Apply a two-pole Butterworth band-pass or band-reject filter with
central frequency (in Hz) \fIfrequency\fR,
and bandwidth (in Hz, and as determined by the 3dB points)
@@ -1000,11 +1017,11 @@
.SP
These effects support the \fB\-o\fR global option (see above).
.TP
-bandreject \fIfrequency bandwidth\fR
+\fBbandreject \fIfrequency bandwidth\fR
Apply a band-reject filter.
See the description of the \fBbandpass\fR effect for details.
.TP
-bass\fB\^|\^\fRtreble \fIgain\fR \fB[\fR\fIfrequency\fR\fB] [\fR\fIslope\fR\fB]\fR
+\fBbass\fR\^|\^\fBtreble \fIgain\fR [\fIfrequency\fR] [\fIslope\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
@@ -1035,7 +1052,7 @@
.SP
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+\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
@@ -1043,10 +1060,10 @@
The modulation is either sinusoidal (\fB\-s\fR) or triangular
(\fB\-t\fR). Gain-out is the volume of the output.
.TP
-compand \fIattack1\fR,\fIdecay1\fR\fB[\fR,\fIattack2\fR,\fIdecay2\fR\fB...]\fR
-\fIin-dB1\fR,\fIout-dB1\fR\fB[\fR,\fIin-dB2\fR,\fIout-dB2\fR\fB...]\fR
+\fBcompand \fIattack1\fB,\fIdecay1\fR{\fB,\fIattack2\fB,\fIdecay2\fR}+\fIin-dB1\fB,\fIout-dB1\fR{\fB,\fIin-dB2\fB,\fIout-dB2\fR}.br
-\fB[\fR\fIgain\fR \fB[\fR\fIinitial-volume\fR \fB[\fR\fIdelay\fR\fB] ] ]\fR
+[\fIgain\fR [\fIinitial-volume\fR [\fIdelay\fR] ] ]
.SP
Compand (compress or expand) the dynamic range of the audio. The
attack and decay time specify the integration time over which the
@@ -1079,8 +1096,12 @@
Specifying a delay approximately equal to the attack/decay times
allows the compander to effectively operate in a `predictive' rather than a
reactive mode.
+.SP
+See also
+.B mcompand
+for a multiple-band companding effect.
.TP
-dcshift \fIshift\fR \fB[\fR\fIlimitergain\fR\fB]\fR
+\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
@@ -1094,7 +1115,7 @@
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
-deemph
+\fBdeemph\fR
Apply a treble attenuation shelving filter to audio in
audio-CD format. The frequency response of pre-emphasized
recordings is rectified. The filtering is defined in the
@@ -1103,7 +1124,7 @@
This effect supports the \fB\-o\fR global option (see above).
.SP
.TP
-dither \fB[\fR\fIdepth\fR\fB]\fR
+\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
@@ -1115,7 +1136,7 @@
This effect should not be followed by any other effect that
affects the audio.
.TP
-earwax
+\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
@@ -1125,7 +1146,7 @@
http://www.geocities.com/beinges
for a full explanation.
.TP
-echo \fIgain-in gain-out delay decay\fR \fB[\fR\fIdelay decay\fR \fB... ]\fR
+\fBecho \fIgain-in gain-out\fR <\fIdelay decay\fR>
Add echoing to the audio.
Each
.I "delay decay"
@@ -1133,7 +1154,7 @@
and the decay (relative to gain-in) of that echo.
Gain-out is the volume of the output.
.TP
-echos \fIgain-in gain-out delay decay\fR \fB[\fR\fIdelay decay\fR \fB... ]\fR
+\fBechos \fIgain-in gain-out\fR <\fIdelay decay\fR>
Add a sequence of echos to the audio.
Each
.I "delay decay"
@@ -1141,7 +1162,7 @@
and the decay (relative to gain-in) of that echo.
Gain-out is the volume of the output.
.TP
-equalizer \fIcentral-frequency Q gain\fR
+\fBequalizer \fIcentral-frequency Q gain\fR
Apply a two-pole peaking equalisation (EQ) filter.
This allows modification (\fIgain\fR) of the signal level at and
around (\fIQ\fR) a central frequency (\fIcentral-frequency\fR),
@@ -1162,7 +1183,7 @@
.SP
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
+\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.
@@ -1186,7 +1207,7 @@
.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
-filter \fB[\fR\fIlow\fR\fB]\fR\-\fB[\fR\fIhigh\fR\fB] [\fR\fIwindow-len\fR \fB[\fR\fIbeta\fR\fB]]\fR
+\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.
@@ -1204,7 +1225,7 @@
For more discussion of beta, look under the \fBresample\fR effect.
.SP
.TP
-flanger \fB[\fR\fIdelay depth regen width speed shape phase interp\fR\fB]\fR
+\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
@@ -1223,7 +1244,7 @@
Percentage of delayed signal mixed with original.
T}
speed 0\*d1 \- 10 0\*d5 Sweeps per second (Hz).
-shape \ sin Swept wave shape: sine\^|\^triangle.
+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;
@@ -1231,12 +1252,12 @@
T}
interp \ lin T{.na
-Digital delay-line interpolation: linear\^|\^quadratic.
+Digital delay-line interpolation: \fBlinear\fR\^|\^\fBquadratic\fR.
T}
.TE
.SP
.TP
-highp\fB\^|\^\fRlowp \fIfrequency\fR
+\fBhighp\fR\^|\^\fBlowp \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).
@@ -1245,7 +1266,7 @@
.SP
See also \fBfilter\fR for filters with a sharper cutoff.
.TP
-highpass\fB\^|\^\fRlowpass \fIfrequency\fR
+\fBhighpass\fR\^|\^\fBlowpass \fIfrequency\fR
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).
@@ -1252,31 +1273,31 @@
.SP
These effects support the \fB\-o\fR global option (see above).
.TP
-lowp \fIfrequency\fR
+\fBlowp \fIfrequency\fR
Apply a low-pass filter.
See the description of the \fBhighp\fR effect for details.
.TP
-lowpass \fIfrequency\fR
+\fBlowpass \fIfrequency\fR
Apply a low-pass filter.
See the description of the \fBhighpass\fR effect for details.
.TP
-mask \fB[\fR\fIdepth\fR\fB]\fR
+\fBmask\fR [\fIdepth\fR]
This effect is just a deprecated alias for the \fBdither\fR effect, left for historical reasons.
.TP
-mcompand "\fIattack1,decay1\fR\fB[\fR,\fIattack2,decay2\fR\fB...]\fR
-\fIin-dB1,out-dB1\fR\fB[\fR,\fIin-dB2,out-dB2\fR\fB...]\fR
+\fBmcompand "\fIattack1\fB,\fIdecay1\fR{\fB,\fIattack2\fB,\fIdecay2\fR}+\fIin-dB1\fB,\fIout-dB1\fR{\fB,\fIin-dB2\fB,\fIout-dB2\fR}.br
-\fB[\fR\fIgain\fR \fB[\fR\fIinitial-volume\fR \fB[\fR\fIdelay\fR\fB] ] ]\fR" \fIxover-freq\fR
+[\fIgain\fR [\fIinitial-volume\fR [\fIdelay\fR] ] ]\fB" \fIxover-freq\fR
.SP
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.
.TP
-noiseprof \fB[\fR\fIprofile-file\fR\fB]\fR
+\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
-noisered \fIprofile-file\fR \fB[\fR\fIthreshold\fR\fB]\fR
+\fBnoisered \fIprofile-file\fR [\fIthreshold\fR]
Noise reduction filter with profiling. This filter is moderately effective at
removing consistent background noise such as hiss or hum. To use it, first run
the \fBnoiseprof\fR effect on a section of audio that ideally would
@@ -1295,7 +1316,7 @@
Experiment with different threshold values to find the optimal one for your
audio.
.TP
-pad \fB{\fR \fIlength\fR\fB[\fR@\fIposition\fR\fB] }\fR+\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
@@ -1320,7 +1341,7 @@
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
-pan \fIdirection\fR
+\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
@@ -1333,8 +1354,8 @@
far left and 1 represents far right. Numbers in between will start the
pan effect without totally muting the opposite channel.
.TP
-phaser \fIgain-in gain-out delay decay speed\fR \fB[\fR\-s\fB\^|\^\fR\-t\fB]\fR
-Add a phasing effect to the audio. Each triple
+\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.
@@ -1342,13 +1363,12 @@
(\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
+\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 avg
-effect
-which is retained for backwards compatibility only.
+effect and is retained for backwards compatibility only.
.TP
-pitch \fIshift\fR \fB[\fR\fIwidth interpolate fade\fR\fB]\fR
+\fBpitch \fIshift\fR [\fIwidth interpolate fade\fR]
Change the pitch of file without affecting its duration by cross-fading
shifted samples.
.I shift
@@ -1363,7 +1383,7 @@
option, can be \fBcos\fR, \fBhamming\fR, \fBlinear\fR or
\fBtrapezoid\fR; the default is \fBcos\fR.
.TP
-polyphase \fB[\fR\-w nut\fB\^|\^\fRham\fB] [\fR\-width long\fB\^|\^\fRshort\fB\^|\^\fR\fIn\fR\fB] [\fR\-cutoff \fIc\fR\fB]\fR
+\fBpolyphase\fR [\fB\-w nut\fR\^|\^\fBham\fR] [\fB\-width long\fR\^|\^\fBshort\fR\^|\^\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
@@ -1396,7 +1416,7 @@
.B resample
for other sample-rate changing effects.
.TP
-rabbit \fB[\fR\-c0\fB\^|\^\fR\-c1\fB\^|\^\fR\-c2\fB\^|\^\fR\-c3\fB\^|\^\fR\-c4\fB]\fR
+\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
@@ -1413,17 +1433,17 @@
for other sample-rate changing effects, and see
\fBresample\fR for more discussion of resampling.
.TP
-rate
+\fBrate\fR
Does the same as \fBresample\fR with no parameters; it exists for
backwards compatibility.
.TP
-repeat \fIcount\fR
+\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 orignal audio and the
repeated audio.
.TP
-resample \fB[\fR\-qs\fB\^|\^\fR\-q\fB\^|\^\fR\-ql\fB] [\fR\fIrolloff\fR \fB[\fR\fIbeta\fR\fB] ]\fR
+\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
@@ -1509,10 +1529,10 @@
Default parameters are, as indicated above, Kaiser window of length 45,
roll-off 0\*d80, beta 16, linear interpolation.
.SP
-\fBNOTE:\fR \fB\-qs\fR is only slightly slower, but more accurate for
+Note: \fB\-qs\fR is only slightly slower, but more accurate for
16-bit or higher precision.
.SP
-\fBNOTE:\fR In many cases of up-sampling, no interpolation is needed,
+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
.SP
@@ -1521,7 +1541,7 @@
and
output-rate \(di gcd(input-rate, output-rate) \(<= 511
.TP
-reverb \fIgain-out reverb-time delay\fR \fB[\fR\fIdelay\fR \fB... ]\fR
+\fBreverb \fIgain-out reverb-time\fR <\fIdelay\fR>
Add reverberation to the audio. Each
.I delay
is given
@@ -1536,11 +1556,11 @@
.I gain-out
is the volume of the output.
.TP
-reverse
+\fBreverse\fR
Reverse the audio completely.
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
+\fBsilence \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
@@ -1600,7 +1620,7 @@
.B %
to indicate a percentage of maximum value of the sample value (\fB0%\fR specifies pure digital silence).
.TP
-speed \fIfactor\fR\fB[\fRc\fB]\fR
+\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
@@ -1615,7 +1635,7 @@
either the \fBresample\fR or the \fBrabbit\fR effect with
appropriate parameters.
.TP
-stat \fB[\fR\-s \fIn\fR\fB] [\fR\-rms\fB] [\fR\-freq\fB] [\fR\-v\fB] [\fR\-d\fB]\fR
+\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.
@@ -1660,7 +1680,7 @@
This is mainly only of use in tracking down endian problems that
creep in to SoX on cross-platform versions.
.TP
-stretch \fIfactor\fR \fB[\fR\fIwindow fade shift fading\fR\fB]\fR
+\fBstretch \fIfactor\fR [\fIwindow fade shift fading\fR]
Time stretch the audio by the given factor. Changes duration without affecting the pitch.
.I factor
of stretching: >1 lengthen, <1 shorten duration.
@@ -1669,25 +1689,25 @@
.I fade
option, can be `lin'.
.I shift
-ratio, in \fB[\fR0 1\fB]\fR. Default depends on stretch factor. 1
+ratio, in [0 1]. Default depends on stretch factor. 1
to shorten, 0\*d8 to lengthen. The
.I fading
-ratio, in \fB[\fR0 0\*d5\fB]\fR. The amount of a fade's default depends on
+ratio, in [0 0\*d5]. The amount of a fade's default depends on
.I factor
and \fIshift\fR.
.TP
-swap \fB[\fR\fI1 2\fR \fB|\fR \fI1 2 3 4\fR\fB]\fR
+\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,
+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
-synth \fB[\fR\fIlen\fR\fB] {[\fR\fItype\fR\fB] [\fR\fIcombine\fR\fB] [\fR\fIfreq\fR\fB[\fR\fI\-freq2\fR\fB]] [\fR\fIoff\fR\fB] [\fR\fIph\fR\fB] [\fR\fIp1\fR\fB] [\fR\fIp2\fR\fB] [\fR\fIp3\fR\fB]}\fR+\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'.
@@ -1716,7 +1736,7 @@
sox \-r 8000 \-c 1 \-n output.au synth 3 sine 300\-3300
.SP
Multiple channels can be synthesised by specifying the set of
-parameters shown between braces (\fB{}\fR) multiple times;+parameters shown between braces multiple times;
the following puts the swept tone in the left channel and adds `brown'
noise in the right:
.SP
@@ -1756,7 +1776,7 @@
`s' appended to it.
.SP
\fItype\fR is one of sine, square, triangle, sawtooth, trapezium, exp,
-\fB[\fRwhite\fB]\fRnoise, pinknoise, brownnoise; default=sine
+[white]noise, pinknoise, brownnoise; default=sine
.SP
\fIcombine\fR is one of create, mix, amod (amplitude modulation), fmod
(frequency modulation); default=create
@@ -1774,17 +1794,17 @@
`rising' (triangle, exp, trapezium); default=50 (square, triangle, exp),
default=10 (trapezium).
.SP
-\fIp2\fR trapezium: the percentage through each cycle at which `falling'
+\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'
+\fIp3\fR (trapezium): the percentage through each cycle at which `falling'
ends; default=60.
.TP
-treble \fIgain\fR \fB[\fR\fIfrequency\fR\fB] [\fR\fIslope\fR\fB]\fR
+\fBtreble \fIgain\fR [\fIfrequency\fR] [\fIslope\fR]
Apply a treble tone control effect.
See the description of the \fBbass\fR effect for details.
.TP
-trim \fIstart\fR \fB[\fR\fIlength\fR\fB]\fR
+\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.
@@ -1802,7 +1822,7 @@
it. A value of 8000s will wait until 8000 samples are read before
starting to process audio.
.TP
-vibro \fIspeed\fR \fB[\fR\fIdepth\fR\fB]\fR
+\fBvibro \fIspeed\fR [\fIdepth\fR]
Apply low frequency sinusoidal amplitude modulation to the audio.
Otherwise known as `tremolo', in the guitar world
this effect is often referred to as `vibrato' (which in fact
@@ -1813,7 +1833,7 @@
.I depth
(0 to 1, default 0\*d5).
.TP
-vol \fIgain\fR \fB[\fR\fItype\fR \fB[\fR\fIlimitergain\fR\fB] ]\fR
+\fBvol \fIgain\fR [\fItype\fR [\fIlimitergain\fR] ]
Apply an amplification or an attenuation to the audio signal.
Unlike the
.B \-v
--- a/soxexam.1
+++ b/soxexam.1
@@ -1,69 +1,55 @@
-.ie n .ds EM " -
-.el .ds EM \(em
-.ds d \v'-.15m'.\v'+.15m'\" Decimal point set slightly raised
+'\" 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
+.ds d \v'-.15m'.\v'+.15m'
+'\"
.TH SoX 1 "January 31, 2007" "soxexam" "Sound eXchange"
.SH NAME
-soxexam\*(EMSoX Examples
-.SH CONVERSIONS
-To convert from unsigned bytes to signed words:
-.P
- sox filename.ub newfile.sw
-.P
-To convert from Apple's AIFF format to Microsoft's WAV format:
-.P
- sox filename.aiff filename.wav
-.P
-To convert from mono raw 8000 Hz 8-bit unsigned PCM data to a WAV file:
-.P
- sox \-r 8000 \-u \-b \-c 1 filename.raw filename.wav
-.P
-SoX may even be used to convert sample rates. Downconverting will
-reduce the bandwidth of the audio, and reduce storage space on
-your disk. All such conversions are lossy and will introduce some noise.
-You should really pass your sample through a low pass filter
-prior to downconverting as this will prevent alias signals (which
-would sound like additional noise). For example to convert from a
-sample recorded at 11025 Hz to a \(*m-law file at 8000 Hz sample rate:
-.P
- sox input.wav \-t au \-r 8000 \-U \-b \-c 1 output.au
-.P
-To add a low-pass filter (note use of stdout for output of
-the first stage and stdin for input on the second stage):
-.P
- sox input.wav \-t raw \-s \-w \-c 1 \- lowpass 3700 |
-.br
- sox \-t raw \-r 11025 \-s \-w \-c 1 \- \-t au \-r 8000 \-U \-b \-c 1 output.au
-.P
-If you hear some clicks and pops when converting to \(*m-law or A-law,
-reduce the output level slightly, for example this will decrease
-it by 20%:
-.P
- sox input.wav \-t au \-r 8000 \-U \-b \-c 1 \-v 0\*d8 outputfile.au
-.P
-The following example applies various effects to an 8000 Hz ADPCM input
-file and then end up with the final file as 44100 Hz ADPCM.
-.P
- sox firstfile.wav \-r 44100 \-s \-w secondfile.wav
-.br
- sox secondfile.wav thirdfile.wav swap
-.br
- sox thirdfile.wav \-a \-b finalfile.wav mask
-.SH EFFECTS
+soxexam\*mSoX Examples
+.SH SYNOPSIS
+.nf
+\fBsox\fR [\fIglobal-options\fR] [\fIformat-options\fR] \fIinfile1\fR
+ [ [\fIformat-options\fR] \fIinfile2\fR ] ... [\fIformat-options\fR] \fIoutfile\fR
+ [\fIeffect\fR [\fIeffect-options\fR] ] ...
+.SP
+\fBplay\fR [\fIgeneral-options\fR] [\fIformat-options\fR] \fIinfile1\fR
+ [ [\fIformat-options\fR] \fIinfile2\fR ] ...
+ [\fIeffect\fR [\fIeffect-options\fR] ] ...
+.SP
+\fBrec\fR [\fIgeneral-options\fR] [\fIformat-options\fR] \fIoutfile\fR
+ [\fIeffect\fR [\fIeffect-options\fR] ] ...
+.fi
+.SH DESCRIPTION
+.SS Introduction
The core problem is that you need some experience in using effects
in order to say `that any old sound file sounds with effects
absolutely hip'. There isn't any rule-based system which tell you
the correct setting of all the parameters for every effect.
But after some time you will become an expert in using effects.
-.P
+.SP
Here are some examples which can be used with any music sample.
(For a sample where only a single instrument is playing, extreme
parameter setting may make well-known `typically' or `classical'
sounds. Likewise, for drums, vocals or guitars.)
-.P
+.SP
Single effects will be explained and some given parameter settings
that can be used to understand the theory by listening to the sound file
with the added effect.
-.P
+.SP
Using multiple effects in parallel or in series can result either
in a very nice sound or (mostly) in a dramatic overloading in
variations of sounds such that your ear may follow the sound but
@@ -72,36 +58,36 @@
composition of effects in the examples because too many combinations
are possible and you really need a very fast machine and a lot of
memory to play them in real-time.
-.P
+.SP
However, real-time playing of sounds will greatly speed up learning
and/or tuning the parameter settings for your sounds in order to
get that `perfect' effect.
-.P
+.SP
Basically, we will use the `play' front-end of SoX since it is easier
to listen sounds coming out of the speaker or earphone instead
of looking at cryptic data in sound files.
-.P
+.SP
For easy listening of file.xxx (`xxx' is any sound format):
-.P
+.SP
play file.xxx effect-name effect-parameters
-.P
+.SP
Or more SoX-like (for `dsp' output on a UNIX/Linux computer):
-.P
+.SP
sox file.xxx \-t ossdsp \-w \-s /dev/dsp effect-name effect-parameters
-.P
+.SP
or (for `au' output):
-.P
+.SP
sox file.xxx \-t sunau \-w \-s /dev/audio effect-name effect-parameters
-.P
+.SP
And for date freaks:
-.P
+.SP
sox file.xxx file.yyy effect-name effect-parameters
-.P
+.SP
Additional options can be used. However, in this case, for real-time
playing you'll need a very fast machine.
-.P
+.SP
Notes:
-.P
+.SP
I played all examples in real-time on a Pentium 100 with 32 MB and
Linux 2.0.30 using a self-recorded sample ( 3:15 min long in `wav'
format with 44\*d1 kHz sample rate and stereo 16 bit ).
@@ -113,37 +99,37 @@
in the sample is suitable. Likewise, the combination vocal, drums, bass
and guitar.)
.SS Echo
-.P
+.SP
An echo effect can be naturally found in the mountains, standing somewhere
on a mountain and shouting a single word will result in one or more repetitions
of the word (if not, turn a bit around and try again, or climb to the next
mountain).
-.P
+.SP
However, the time difference between shouting and repeating is the delay
(time), its loudness is the decay. Multiple echos can have different delays and
decays.
-.P
+.SP
It is very popular to use echos to play an instrument with itself together,
like some guitar players (Brain May from Queen) or vocalists are doing.
For music samples of more than one instrument, echo can be used to add a
second sample shortly after the original one.
-.P
+.SP
This will sound as if you are doubling the number of instruments playing
in the same sample:
-.P
+.SP
play file.xxx echo 0\*d8 0\*d88 60 0\*d4
-.P
+.SP
If the delay is very short, then it sound like a (metallic) robot playing
music:
-.P
+.SP
play file.xxx echo 0\*d8 0\*d88 6 0\*d4
-.P
+.SP
Longer delay will sound like an open air concert in the mountains:
-.P
+.SP
play file.xxx echo 0\*d8 0\*d9 1000 0\*d3
-.P
+.SP
One mountain more, and:
-.P
+.SP
play file.xxx echo 0\*d8 0\*d9 1000 0\*d3 1800 0\*d25
.SS Echos
Like the echo effect, echos stand for `ECHO in Sequel', that is the first echos
@@ -151,23 +137,23 @@
and the first and the second echos, ... and so on.
Care should be taken using many echos (see introduction); a single echos
has the same effect as a single echo.
-.P
+.SP
The sample will be bounced twice in symmetric echos:
-.P
+.SP
play file.xxx echos 0\*d8 0\*d7 700 0\*d25 700 0\*d3
-.P
+.SP
The sample will be bounced twice in asymmetric echos:
-.P
+.SP
play file.xxx echos 0\*d8 0\*d7 700 0\*d25 900 0\*d3
-.P
+.SP
The sample will sound as if played in a garage:
-.P
+.SP
play file.xxx echos 0\*d8 0\*d7 40 0\*d25 63 0\*d3
.SS Chorus
The chorus effect has its name because it will often be used to make a single
vocal sound like a chorus. But it can be applied to other instrument samples
too.
-.P
+.SP
It works like the echo effect with a short delay, but the delay isn't constant.
The delay is varied using a sinusoidal or triangular modulation. The modulation
depth defines the range the modulated delay is played before or after the
@@ -174,35 +160,35 @@
delay. Hence the delayed sound will sound slower or faster, that is the delayed
sound tuned around the original one, like in a chorus where some vocals are
a bit out of tune.
-.P
+.SP
The typical delay is around 40ms to 60ms, the speed of the modulation is best
near 0\*d25Hz and the modulation depth around 2ms.
-.P
+.SP
A single delay will make the sample more overloaded:
-.P
+.SP
play file.xxx chorus 0\*d7 0\*d9 55 0\*d4 0\*d25 2 \-t
-.P
+.SP
Two delays of the original samples sound like this:
-.P
+.SP
play file.xxx chorus 0\*d6 0\*d9 50 0\*d4 0\*d25 2 \-t 60 0\*d32 0\*d4 1\*d3 \-s
-.P
+.SP
A big chorus of the sample is (three additional samples):
-.P
+.SP
play file.xxx chorus 0\*d5 0\*d9 50 0\*d4 0\*d25 2 \-t 60 0\*d32 0\*d4 2\*d3 \-t 40 0\*d3 0\*d3 1\*d3 \-s
.SS Flanger
The flanger effect is like the chorus effect, but the delay varies between
0ms and maximal 5ms. It sound like wind blowing, sometimes faster or slower
including changes of the speed.
-.P
+.SP
The flanger effect is widely used in funk and soul music, where the guitar
sound varies frequently slow or a bit faster.
-.P
+.SP
Now, let's groove the sample:
-.P
+.SP
play file.xxx flanger
-.P
+.SP
listen carefully between the difference of sinusoidal and triangular modulation:
-.P
+.SP
play file.xxx flanger triangle
.SS Reverb
The reverb effect is often used in audience hall which are to small or contain
@@ -211,7 +197,7 @@
a large hall. You can try the reverb effect in your bathroom or garage or
sport halls by shouting loud some words. You'll hear the words reflected from
the walls.
-.P
+.SP
The biggest problem in using the reverb effect is the correct setting of the
(wall) delays such that the sound is realistic and doesn't sound like music
playing in a tin can or has overloaded feedback which destroys any illusion
@@ -228,24 +214,24 @@
The walls shouldn't stand to close to each other and not in a multiple integer
distance to each other ( so avoid wall like: 200 and 202, or something
like 100 and 200 ).
-.P
+.SP
Since audience halls do have a lot of walls, we will start designing one
beginning with one wall:
-.P
+.SP
play file.xxx reverb 1 600 180
-.P
+.SP
One wall more:
-.P
+.SP
play file.xxx reverb 1 600 180 200
-.P
+.SP
Next two walls:
-.P
+.SP
play file.xxx reverb 1 600 180 200 220 240
-.P
+.SP
Now, why not a futuristic hall with six walls:
-.P
+.SP
play file.xxx reverb 1 600 180 200 220 240 280 300
-.P
+.SP
If you run out of machine power or memory, then stop as many applications
as possible (every interrupt will consume a lot of CPU time which for
bigger halls is absolutely necessary).
@@ -252,28 +238,28 @@
.SS Phaser
The phaser effect is like the flanger effect, but it uses a reverb instead of
an echo and does phase shifting. You'll hear the difference in the examples
-comparing both effects (simply change the effect name).
+comparing both effects.
The delay modulation can be sinusoidal or triangular, preferable is the
later for multiple instruments. For single instrument sounds,
the sinusoidal phaser effect will give a sharper phasing effect.
The decay shouldn't be to close to 1 which will cause dramatic feedback.
A good range is about 0\*d5 to 0\*d1 for the decay.
-.P
-We will take a parameter setting as for the flanger before (gain-out is
+.SP
+We will take a parameter setting as before (gain-out is
lower since feedback can raise the output dramatically):
-.P
+.SP
play file.xxx phaser 0\*d8 0\*d74 3 0\*d4 0\*d5 \-t
-.P
+.SP
The drunken loudspeaker system (now less alcohol):
-.P
+.SP
play file.xxx phaser 0\*d9 0\*d85 4 0\*d23 1\*d3 \-s
-.P
+.SP
A popular sound of the sample is as follows:
-.P
+.SP
play file.xxx phaser 0\*d89 0\*d85 1 0\*d24 2 \-t
-.P
+.SP
The sample sounds if ten springs are in your ears:
-.P
+.SP
play file.xxx phaser 0\*d6 0\*d66 3 0\*d6 2 \-t
.SS Compander
The compander effect allows the dynamic range of a signal to be
@@ -281,15 +267,15 @@
For most situations, the attack time (response to the music getting
louder) should be shorter than the decay time because our ears are more
sensitive to suddenly loud music than to suddenly soft music.
-.P
+.SP
For example, suppose you are listening to Strauss' `Also Sprach
Zarathustra' in a noisy environment such as a car.
If you turn up the volume enough to hear the soft passages over the
road noise, the loud sections will be too loud.
You could try this:
-.P
+.SP
play file.xxx compand 0\*d3,1 \-90,\-90,\-70,\-70,\-60,\-20,0,0 \-5 0 0\*d2
-.P
+.SP
The transfer function (`\-90,...') says that
.I very
soft sounds between \-90 and \-70 decibels (\-90 is about the limit of
@@ -309,33 +295,33 @@
.SS Changing the Rate of Playback
You can use stretch to change the rate of playback of an audio sample
while preserving the pitch. For example to play at half the speed:
-.P
+.SP
play file.wav stretch 2
-.P
+.SP
To play a file at twice the speed:
-.P
+.SP
play file.wav stretch 0\*d5
-.P
+.SP
Other related options are `speed' to change the speed of play
(and changing the pitch accordingly), and pitch, to alter the
pitch of a sample. For example to speed a sample so it plays in
half the time (for those Mickey Mouse voices):
-.P
+.SP
play file.wav speed 2
-.P
+.SP
To raise the pitch of a sample 1 while note (100 cents):
-.P
+.SP
play file.wav pitch 100
-.P
+.SP
.SS Reducing noise in a recording
-.P
+.SP
First find a period of silence in your recording, such as the beginning or
end of a piece. If the first 1\*d5 seconds of the recording are silent, do
-.P
+.SP
sox file.wav \-n trim 0 1\*d5 noiseprof /tmp/profile
-.P
+.SP
Next, use the noisered effect to actually reduce the noise:
-.P
+.SP
play file.wav noisered /tmp/profile
.SH SEE ALSO
.BR sox (1),
@@ -342,6 +328,6 @@
.BR play (1),
.BR rec (1)
.SH AUTHOR
-Juergen Mueller (jmueller@uia.ua.ac.be).
-Additional authors and contributors are listed in the AUTHORS file that
+This manual was written by Juergen Mueller (jmueller@uia.ua.ac.be).
+Other SoX authors and contributors are listed in the AUTHORS file that
is distributed with the source code.