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.PP
\fB\\$1\fR
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.TH SoX 1 "December 11, 2001"
.SH NAME
sox \- Sound eXchange : universal sound sample translator
.SH SYNOPSIS
.P
\fBsox\fR \fIinfile outfile\fR
.P
\fBsox\fR [ \fIgeneral options\fR ] [ \fIformat options\fR ] \fIinfile\fR
.br
[ \fIformat options\fR ] \fIoutfile\fR
.br
[ \fIeffect\fR [ \fIeffect options\fR ] ... ]
.P
\fBsoxmix\fR \fIinfile1 infile2 outfile\fR
.P
\fBsoxmix\fR [ \fIgeneral options\fR ] [ \fIformat options\fR ] \fIinfile1\fR
.br
[ \fIformat options\fR ] \fIinfile2\fR
.br
[ \fIformat options\fR ] \fIoutfile\fR
.br
[ \fIeffect\fR [ \fIeffect options\fR ] ... ]
.P
.B General options:
.br
[ -h ] [ -p ] [ -v \fIvolume\fR ] [ -V ]
.P
.B Format options:
.br
[ -t \fIfiletype\fR ] [ -r \fIrate\fR ] [ -s/-u/-U/-A/-a/-i/-g/-f ]
[ -b/-w/-l/-d ]
[ -c \fIchannels\fR ] [ -x ] [ -e ]
.P
.B Effects:
.br
\fBavg\fR [ -l | -r | -f | -b | -1 | -2 | -3 | -4 | n,n,...,n ]
.br
\fBband\fR [ -n ] \fIcenter\fR [ \fIwidth\fR ]
.br
\fBbandpass\fR \fIfrequency bandwidth\fR
.br
\fBbandreject\fR \fIfrequency bandwidth\fR
.br
\fBchorus\fR \fIgain-in gain out delay decay speed depth\fR
.br
-s | -t [ \fIdelay decay speed depth\fR -s | -t ]
.br
\fBcompand\fR \fIattack1\fR,\fIdecay1\fR[,\fIattack2\fR,\fIdecay2\fR...]
.br
\fIin-dB1\fR,\fIout-dB1\fR[,\fIin-dB2\fR,\fIout-dB2\fR...]
.br
[ \fIgain\fR [ \fIinitial-volume\fR [ \fIdelay\fR ] ] ]
.br
\fBcopy\fR
.br
\fBdcshift\fR \fIshift\fR [ \fIlimitergain\fR ]
.br
\fBdeemph\fR
.br
\fBearwax\fR
.br
\fBecho\fR \fIgain-in gain-out delay decay\fR [ \fIdelay decay ...\fR ]
.br
\fBechos\fR \fIgain-in gain-out delay decay\fR [ \fIdelay decay ...\fR ]
.br
\fBfade\fR [ \fItype\fR ] \fIfade-in-length\fR
[ \fIstop-time\fR [ \fIfade-out-length\fR ] ]
.br
\fBfilter\fR [ \fIlow\fR ]-[ \fIhigh\fR ] [ \fIwindow-len\fR [ \fIbeta\fR ]]
.br
\fBflanger\fR \fIgain-in gain-out delay decay speed\fR < -s | -t >
.br
\fBhighp\fR \fIfrequency\fR
.br
\fBhighpass\fR \fIfrequency\fR
.br
\fBlowp\fR \fIfrequency\fR
.br
\fBlowpass\fR \fIfrequency\fR
.br
\fBmap\fR
.br
\fBmask\fR
.br
\fBpan\fR \fIdirection\fR
.br
\fBphaser\fR \fIgain-in gain-out delay decay speed\fR < -s | -t >
.br
\fBpick\fR [ \fI-1\fR | \fI-2\fR | \fI-3\fR | \fI-4\fR | \fI-l\fR | \fI-r\fR | \fI-f\fR | \fI-b\fR ]
.br
\fBpitch\fR \fIshift\fR [ \fIwidth interpole fade\fR ]
.br
\fBpolyphase\fR [ -w < \fInut\fR / \fIham\fR > ]
[ \fI -width\fR < \fIlong\fR / \fIshort\fR / # > ]
[ \fI-cutoff #\fR ]
.br
\fBrate\fR
.br
\fBrepeat\fR \fIcount\fR
.br
\fBresample\fR [ -qs | -q | -ql ] [ \fIrolloff\fR [ \fIbeta\fR ] ]
.br
\fBreverb\fR \fIgain-out reverb-time delay\fR [ \fIdelay\fR ... ]
.br
\fBreverse\fR
.br
\fBsilence\fR \fIabove_periods\fR [ \fIduration threshold\fR[ \fId\fR | \fI%\fR ]
[ \fIbelow_periods duration
threshold\fR[ \fId\fR | \fI%\fR ]]
.br
\fBspeed\fR [ -c ] \fIfactor\fR
.br
\fBstat\fR [ -s \fIn\fR ] [ -rms ] [ -v ] [ -d ]
.br
\fBstretch\fR [ \fIfactor\fR [ \fIwindow fade shift fading\fR ]
.br
\fBswap\fR [ \fI1 2\fR | \fI1 2 3 4\fR ]
.br
\fBsynth\fR [ \fIlength\fR ] \fItype mix\fR [ \fIfreq\fR [ \fI-freq2\fR ]
[ \fIoff\fR ] [ \fIph\fR ] [ \fIp1\fR ] [ \fIp2\fR ] [ \fIp3\fR ]
.br
\fBtrim\fR \fIstart\fR [ \fIlength\fR ]
.br
\fBvibro\fR \fIspeed\fR [ \fIdepth\fR ]
.br
\fBvol\fR \fIgain\fR [ \fItype\fR [ \fIlimitergain\fR ] ]
.SH DESCRIPTION
.I SoX
is a command line program that can convert most popular audio files
to most other popular audio file formats. It can optionally change
the audio sample data type and apply one or more
sound effects to the file during this translation.
.P
.I soxmix
is functionally the same as the command line program
.I sox
expect that it takes two files as input and mixes the audio together
to produce a single file as output. It has a restriction that both
input files must be of the same data type and sample rates.
.P
There are two types of audio files formats that
.I SoX
can work with. The first are self-describing file formats. These
contain a header that completely describe the characteristics of
the audio data that follows.
.P
The second type are header-less data, or sometimes called raw data. A
user must pass enough information to
.I SoX
on the command line so that it knows what type of data it contains.
.P
Audio data can usually be totally described by four characteristics:
.TP 10
rate
The sample rate is in samples per second. For example, CD sample rates are at 44100.
.TP 10
data size
The precision the data is stored in. Most popular are 8-bit bytes or 16-bit
words.
.TP 10
data encoding
What encoding the data type uses. Examples are u-law, ADPCM, or signed linear data.
.TP 10
channels
How many channels are contained in the audio data. Mono and Stereo are the two most common.
.P
Please refer to the
.B soxexam(1)
manual page for a long description with examples on how to use SoX with
various types of file formats.
.SH OPTIONS
The option syntax is a little grotty, but in essence:
.P
.br
sox File.au file.wav
.P
.br
translates a sound file in SUN Sparc .AU format
into a Microsoft .WAV file, while
.P
.br
sox -v 0.5 file.au -r 12000 file.wav mask
.P
.br
does the same format translation but also
lowers the amplitude by 1/2, changes
the sampling rate to 12000 hertz, and applies the \fBmask\fR sound effect
to the audio data.
.P
The following will mix two sound files together to to produce a single sound
file.
.P
.br
soxmix music.wav voice.wav mixed.wav
.PP
\fBFormat options:\fR
.PP
Format options effect the audio samples that they immediately precede. If
they are placed before the input file name then they effect the input
data. If they are placed before the output file name then they will
effect the output data. By taking advantage of this, you can override
a input file's corrupted header or produce an output file that is totally
different style then the input file. It is also how SoX is informed about
the format of raw input data.
.TP 10
\fB-t \fIfiletype\fR
gives the type of the sound sample file. Useful when file extension is
not standard or for specifying the .auto file type.
.TP 10
\fB-r \fIrate\fR
Gives the sample rate in Hertz 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 options.
.br
If the input and output files have
different rates then a sample rate change effect must be ran. If a
sample rate changing effect is not specified then a default one will internally
be ran by SoX using its default parameters.
.TP 10
\fB-s/-u/-U/-A/-a/-i/-g/-f\fR
The sample data encoding is signed linear (2's complement),
unsigned linear, u-law (logarithmic), A-law (logarithmic),
ADPCM, IMA_ADPCM, GSM, or Floating-point.
.br
U-law (actually shorthand for mu-law) and A-law are the U.S. and
international standards for logarithmic telephone sound compression.
When uncompressed u-law has roughly the precision of 14-bit PCM audio
and A-law has roughly the precision of 13-bit PCM audio.
.br
A-law and u-law data is sometimes encoded using a reversed bit-ordering
(ie. MSB becomes LSB). Internally, SoX understands how to work with
this encoding but there is currently no command line option to
specify it. If you need this support then you can use the psuedo
file types of ".la" and ".lu" to inform sox of the encoding. See
supported file types for more information.
.br
ADPCM is a form of sound compression that has a good
compromise between good sound quality and fast encoding/decoding
time. It is used for telephone sound compression and places were
full fidelity is not as important. When uncompressed it has roughly
the precision of 16-bit PCM audio. Popular version of ADPCM include
G.726, MS ADPCM, and IMA ADPCM. The \fB-a\fR flag has different meanings
in different file handlers. In \fB.wav\fR files it represents MS ADPCM
files, in all others it means G.726 ADPCM.
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.
.br
GSM is a standard used for telephone sound compression in
European countries and its gaining popularity because of its
quality. It usually is CPU intensive to work with GSM audio data.
.TP 10
\fB-b/-w/-l/-d\fR
The sample data size is in bytes, 16-bit words, 32-bit long words,
or 64-bit double long (long long) words.
.TP 10
\fB-x\fR
The sample data is in XINU format; that is,
it comes from a machine with the opposite word order
than yours and must
be swapped according to the word-size given above.
Only 16-bit and 32-bit integer data may be swapped.
Machine-format floating-point data is not portable.
.TP 10
\fB-c \fIchannels\fR
The number of sound channels in the data file.
This may be 1, 2, or 4; for mono, stereo, or quad sound data. To cause
the output file to have a different number of channels than the input
file, include this option with the output file options.
If the input and output file have a different number of channels then the
avg effect must be used. If the avg effect is not specified on the
command line it will be invoked internally with default parameters.
.TP 10
\fB-e\fR
When used after the input filename (so that it applies to the output file)
it allows you to avoid giving an output filename and will not
produce an output file. It will apply any specified effects
to the input file. This is mainly useful with the \fBstat\fR effect
but can be used with others.
.PP
\fBGeneral options:\fR
.TP 10
\fB-h\fR
Print version number and usage information.
.TP 10
\fB-p\fR
Run in preview mode and run fast. This will somewhat speed up
SoX when the output format has a different number of channels and
a different rate than the input file. Currently, this defaults to
using the \fBrate\fR effect instead of the \fBresample\fR effect for sample
rate changes.
.TP 10
\fB-v \fIvolume\fR
Change amplitude (floating point);
less than 1.0 decreases, greater than 1.0 increases. May use a negative
number to invert the phase of the audio data. It is interesting to note
that we perceive volume
logarithmically but this adjusts the amplitude linearly.
.br
Note: see the \fBstat\fR effect for information on finding the maximum
value that can be used with this option without causing audio data be
be clipped.
.TP 10
\fB-V\fR
Print a description of processing phases.
Useful for figuring out exactly how
.I SoX
is mangling your sound samples.
.SH FILE TYPES
.I SoX
attempts to determine the file type of input files automatically by looking
at the header of the audio file. When it is unable to detect the file
type or if its an output file
then it uses the file extension of the file to determine what type of file
format handler to use. This can be overridden by specifying the
"-t" option on the command line.
.P
The input and output files may be read from standard in and out. This
is done by specifying '-' as the filename.
.P
File formats which have headers are checked,
if that header doesn't seem right,
the program exits with an appropriate message.
.P
The following file formats are supported:
.PP
.TP 10
.B .8svx
Amiga 8SVX musical instrument description format.
.TP 10
.B .aiff
AIFF files used on Apple IIc/IIgs and SGI.
Note: the AIFF format supports only one SSND chunk.
It does not support multiple sound chunks,
or the 8SVX musical instrument description format.
AIFF files are multimedia archives and
can have multiple audio and picture chunks.
You may need a separate archiver to work with them.
.TP 10
.B .au
SUN Microsystems AU files.
There are apparently many types of .au files;
DEC has invented its own with a different magic number
and word order.
The .au handler can read these files but will not write them.
Some .au files have valid AU headers and some do not.
The latter are probably original SUN u-law 8000 hz samples.
These can be dealt with using the
.B .ul
format (see below).
.TP 10
.B .avr
Audio Visual Research
.br
The AVR format is produced by a number of commercial packages
on the Mac.
.TP 10
.B .cdr
CD-R
.br
CD-R files are used in mastering music on Compact Disks.
The audio data on a CD-R disk is a raw audio file
with a format of stereo 16-bit signed samples at a 44khz sample
rate. There is a special blocking/padding oddity at the end
of the audio file and is why it needs its own handler.
.TP 10
.B .cvs
Continuously Variable Slope Delta modulation
.br
Used to compress speech audio for applications such as voice mail.
.TP 10
.B .dat
Text Data files
.br
These files contain a textual representation of the
sample data. There is one line at the beginning
that contains the sample rate. Subsequent lines
contain two numeric data items: the time since
the beginning of the first sample and the sample value.
Values are normalized so that the maximum and minimum
are 1.00 and -1.00. This file format can be used to
create data files for external programs such as
FFT analyzers or graph routines. SoX can also convert
a file in this format back into one of the other file
formats.
.TP 10
.B .gsm
GSM 06.10 Lossy Speech Compression
.br
A standard for compressing speech which is used in the
Global Standard for Mobil telecommunications (GSM). Its good
for its purpose, shrinking audio data size, but it will introduce
lots of noise when a given sound sample is encoded and decoded
multiple times. This format is used by some voice mail applications.
It is rather CPU intensive.
.br
GSM in
.B SoX
is optional and requires access to an external GSM library. To see
if there is support for gsm run \fBsox -h\fR
and look for it under the list of supported file formats.
.TP 10
.B .hcom
Macintosh HCOM files.
These are (apparently) Mac FSSD files with some variant
of Huffman compression.
The Macintosh has wacky file formats and this format
handler apparently doesn't handle all the ones it should.
Mac users will need your usual arsenal of file converters
to deal with an HCOM file under Unix or DOS.
.TP 10
.B .maud
An Amiga format
.br
An IFF-conform sound file type, registered by
MS MacroSystem Computer GmbH, published along
with the "Toccata" sound-card on the Amiga.
Allows 8bit linear, 16bit linear, A-Law, u-law
in mono and stereo.
.TP 10
.B .mp3
MP3 Compressed Audio
.br
MP3 audio files come from the MPEG standards for audio and video compression. They are a lossy compression format that achieves good compression rates with a minimum amount of quality loss. Also see Ogg Vorbis for a similar format.
MP3 support in
.B 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".
.TP 10
.B .nul
Null file handler. This is a fake file hander that act as if its reading
a stream of 0's from a while or fake writing output to a file. This
is not a very useful file handler in most cases. It might be useful in
some scripts were you do not want to read or write from a real file
but would like to specify a filename for consistency.
.TP 10
.B .ogg
Ogg Vorbis Compressed Audio.
.br
Ogg Vorbis is a open, patent-free CODEC designed for compressing music
and streaming audio. It is similar to MP3, VQF, AAC, and other lossy
formats.
.B SoX
can decode all types of Ogg Vorbis files, but can only encode at 128 kbps.
Decoding is somewhat CPU intensive and encoding is very CPU intensive.
.br
Ogg Vorbis in
.B SoX
is optional and requires access to external Ogg Vorbis libraries. To
see if there is support for Ogg Vorbis run \fBsox -h\fR
and look for it under the list of supported file formats as "vorbis".
.TP 10
.B ossdsp
OSS /dev/dsp device driver
.br
This is a pseudo-file type and can be optionally compiled into SoX. Run
.B sox -h
to see if you have support for this file type. When this driver is used
it allows you to open up the OSS /dev/dsp file and configure it to
use the same data format as passed in to \fBSoX\fR.
It works for both playing and recording sound samples. When playing sound
files it attempts to set up the OSS driver to use the same format as the
input file. It is suggested to always override the output values to use
the highest quality samples your sound card can handle. Example:
.I -t ossdsp -w -s /dev/dsp
.TP 10
.B .prc
Psion record.app
.br
Used in some Psion devices for System alarms. This format is newer then
the .wve format that is used in some Psion devices.
.TP 10
.B .sf
IRCAM Sound Files.
.br
Sound Files are used by academic music software
such as the CSound package, and the MixView sound sample editor.
.TP 10
.B .sph
.br
SPHERE (SPeech HEader Resources) is a file format defined by NIST
(National Institute of Standards and Technology) and is used with
speech audio. SoX can read these files when they contain
u-law and PCM data. It will ignore any header information that
says the data is compressed using \fIshorten\fR compression and
will treat the data as either u-law or PCM. This will allow SoX
and the command line \fIshorten\fR program to be ran together using
pipes to uncompress the data and then pass the result to SoX for processing.
.TP 10
.B .smp
Turtle Beach SampleVision files.
.br
SMP files are for use with the PC-DOS package SampleVision by Turtle Beach
Softworks. This package is for communication to several MIDI samplers. All
sample rates are supported by the package, although not all are supported by
the samplers themselves. Currently loop points are ignored.
.TP 10
.B .snd
.br
Under DOS this file format is the same as the \fB.sndt\fR format. Under all
other platforms it is the same as the \fB.au\fR format.
.TP 10
.B .sndt
SoundTool files.
.br
This is an older DOS file format.
.TP 10
.B sunau
Sun /dev/audio device driver
.br
This is a pseudo-file type and can be optionally compiled into SoX. Run
.B sox -h
to see if you have support for this file type. When this driver is used
it allows you to open up a Sun /dev/audio file and configure it to
use the same data type as passed in to
.B SoX.
It works for both playing and recording sound samples. When playing sound
files it attempts to set up the audio driver to use the same format as the
input file. It is suggested to always override the output values to use
the highest quality samples your hardware can handle. Example:
.I -t sunau -w -s /dev/audio
or
.I -t sunau -U -c 1 /dev/audio
for older sun equipment.
.TP 10
.B .txw
Yamaha TX-16W sampler.
.br
A file format from a Yamaha sampling keyboard which wrote IBM-PC
format 3.5" 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.
.TP 10
.B .vms
More info to come.
.br
Used to compress speech audio for applications such as voice mail.
.TP 10
.B .voc
Sound Blaster VOC files.
.br
VOC files are multi-part and contain silence parts, looping, and
different sample rates for different chunks.
On input, the silence parts are filled out, loops are rejected,
and sample data with a new sample rate is rejected.
Silence with a different sample rate is generated appropriately.
On output, silence is not detected, nor are impossible sample rates.
Note, this version now supports playing VOC files with multiple
blocks and supports playing files containing u-law and A-law samples.
.TP 10
.B vorbis
See
.B .ogg
format.
.TP 10
.B vox
A headerless file of Dialogic/OKI ADPCM audio data commonly comes with the
extension .vox. This ADPCM data has 12-bit precision packed into only 4-bits.
.TP 10
.B .wav
Microsoft .WAV RIFF files.
.br
These appear to be very similar to IFF files,
but not the same.
They are the native sound file format of Windows.
(Obviously, Windows was of such incredible importance
to the computer industry that it just had to have its own
sound file format.)
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
override the file header, and you will be warned to this effect.
You had better know what you are doing! Output format
options will cause a format conversion, and the \fB.wav\fR
will written appropriately.
SoX currently can read PCM, ULAW, ALAW, MS ADPCM, and IMA (or DVI) ADPCM.
It can write all of these formats including
.B (NEW!)
the ADPCM encoding.
.TP 10
.B .wve
Psion 8-bit A-law
.br
These are 8-bit A-law 8khz sound files used on the
Psion palmtop portable computer.
.TP 10
.B .raw
Raw files (no header).
.br
The sample rate, size (byte, word, etc),
and encoding (signed, unsigned, etc.)
of the sample file must be given.
The number of channels defaults to 1.
.TP 10
.B ".ub, .sb, .uw, .sw, .ul, .al, .lu, .la, .sl"
These are several suffices which serve as
a shorthand for raw files with a given size and encoding.
Thus, \fBub, sb, uw, sw, ul, al, lu, la\fR and \fBsl\fR
correspond to "unsigned byte", "signed byte",
"unsigned word", "signed word", "u-law" (byte), "A-law" (byte),
inverse bit order "u-law", inverse bit order "A-law", and "signed long".
The sample rate defaults to 8000 hz if not explicitly set,
and the number of channels defaults to 1.
There are lots of Sparc samples floating around in u-law format
with no header and fixed at a sample rate of 8000 hz.
(Certain sound management software cheerfully ignores the headers.)
Similarly, most Mac sound files are in unsigned byte format with
a sample rate of 11025 or 22050 hz.
.TP 10
.B .auto
This is a ``meta-type'': specifying this type for an input file
triggers some code that tries to guess the real type by looking for
magic words in the header. If the type can't be guessed, the program
exits with an error message. The input must be a plain file, not a
pipe. This type can't be used for output files.
.SH EFFECTS
Multiple effects may be applied to the audio data by specifying them
one after another at the end of the command line.
.TP 10
avg [ \fI-l\fR | \fI-r\fR | \fI-f\fR | \fI-b\fR | \fI-1\fR | \fI-2\fR | \fI-3\fR | \fI-4\fR | \fIn,n,...,n\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
channels differ from the number of output channels. When reducing
the number of channels it is possible to manually specify the
avg effect and use the \fI-l\fR, \fI-r\fR, \fI-f\fR, \fI-b\fR,
\fI-1\fR, \fI-2\fR, \fI-3\fR, \fI-4\fR, options to select only
the left, right, front, back channel(s) or specific channel
for the output instead of averaging the channels.
The \fI-l\fR, and \fI-r\fR options will do averaging
in quad-channel files so select the exact channel to prevent this.
The avg effect can also be invoked with up to 16 double-precision
numbers, which specify the proportion (0.0 = 0% and 1.0 = 100%)
of each input channel that is to be mixed into each output channel.
In two-channel mode, 4 numbers are given: l->l, l->r, r->l, and r->r,
respectively.
In four-channel mode, the first 4 numbers give the proportions for the
left-front output channel, as follows: lf->lf, rf->lf, lb->lf, and
rb->rf.
The next 4 give the right-front output in the same order, then
left-back and right-back.
It is also possible to use the 16 numbers to expand or reduce the
channel count; just specify 0 for unused channels.
Finally, certain reduced combination of numbers can be specified
for certain input/output channel combinations.
In Ch Out Ch Num Mappings
.br
_____ ______ ___ _____________________________
.b4
2 1 2 l->l, r->l
.br
2 2 1 adjust balance
.br
4 1 4 lf->l, rf->l, lb->l, rb-l
.br
4 2 2 lf->l&rf->r, lb->l&rb->r
.br
4 4 1 adjust balance
.br
4 4 2 front balance, back balance
.br
.TP 10
band \fB[ \fI-n \fB] \fIcenter \fB[ \fIwidth\fB ]
Apply a band-pass filter.
The frequency response drops logarithmically
around the
.I center
frequency.
The
.I width
gives the slope of the drop.
The frequencies at
.I "center + width"
and
.I "center - width"
will be half of their original amplitudes.
.B Band
defaults to a mode oriented to pitched signals,
i.e. voice, singing, or instrumental music.
The
.I -n
(for noise) option uses the alternate mode
for un-pitched signals.
.B Warning:
.I -n
introduces a power-gain of about 11dB in the filter, so beware
of output clipping.
.B Band
introduces noise in the shape of the filter,
i.e. peaking at the
.I center
frequency and settling around it.
See \fBfilter\fR for a bandpass effect with steeper shoulders.
.TP 10
bandpass \fIfrequency bandwidth\fB
Butterworth bandpass filter. Description coming soon!
.TP 10
bandreject \fIfrequency bandwidth\fB
Butterworth bandreject filter. Description coming soon!
.TP
chorus \fIgain-in gain-out delay decay speed depth
.TP 10
-s \fR| \fI-t [ \fIdelay decay speed depth -s \fR| \fI-t ... \fR]
Add a chorus to a sound sample. Each quadtuple
delay/decay/speed/depth gives the delay in milliseconds
and the decay (relative to gain-in) with a modulation
speed in Hz using depth in milliseconds.
The modulation is either sinusoidal (-s) or triangular
(-t). Gain-out is the volume of the output.
.TP
compand \fIattack1,decay1\fR[,\fIattack2,decay2\fR...]
.TP
\fIin-dB1,out-dB1\fR[,\fIin-dB2,out-dB2\fR...]
.TP 10
[\fIgain\fR [\fIinitial-volume\fR [\fIdelay\fR ] ] ]
Compand (compress or expand) the dynamic range of a sample. The
attack and decay time specify the integration time over which the
absolute value of the input signal is integrated to determine its
volume; attacks refer to increases in volume and decays refer to
decreases. Where more than one pair of attack/decay parameters are
specified, each channel is treated separately and the number of pairs
must agree with the number of input channels. The second parameter is
a list of points on the compander's transfer function specified in dB
relative to the maximum possible signal amplitude. The input values
must be in a strictly increasing order but the transfer function does
not have to be monotonically rising. The special value \fI-inf\fR may
be used to indicate that the input volume should be associated output
volume. The points \fI-inf,-inf\fR and \fI0,0\fR are assumed; the
latter may be overridden, but the former may not.
The third
(optional) parameter is a post-processing gain in dB which is applied
after the compression has taken place; the fourth (optional) parameter
is an initial volume to be assumed for each channel when the effect
starts. This permits the user to supply a nominal level initially, so
that, for example, a very large gain is not applied to initial signal
levels before the companding action has begun to operate: it is quite
probable that in such an event, the output would be severely clipped
while the compander gain properly adjusts itself.
The fifth (optional) parameter is a delay in seconds.
The input signal is analyzed immediately to control the compander, but
it is delayed before being fed to the volume adjuster.
Specifying a delay approximately equal to the attack/decay times
allows the compander to effectively operate in a "predictive" rather than a
reactive mode.
.TP 10
copy
Copy the input file to the output file.
This is the default effect if both files have the same
sampling rate.
.TP 10
dcshift \fIshift\fR [ \fIlimitergain\fR ]
DC Shift the audio data, with basic linear amplitude formula.
This is most useful if your audio data tends to not be centered around
a value of 0. Shifting it back will allow you to get the most volume
adjustments without clipping audio data.
.br
The first option is the \fIdcshift\fR value. It is a floating point number that
indicates the amount to shift.
.br
An option limtergain value can be specified as well. It should have a value much less then 1.0 and is used only on peaks to prevent clipping.
.TP 10
deemph
Apply a treble attenuation shelving filter to samples in
audio cd format. The frequency response of pre-emphasized
recordings is rectified. The filtering is defined in the
standard document ISO 908.
.TP 10
earwax
Makes sound easier to listen to on headphones.
Adds audio-cues to samples in audio cd format so that
when listened to on headphones the stereo image is
moved from inside
your head (standard for headphones) to outside and in front of the
listener (standard for speakers). See
.br
www.geocities.com/beinges
for a full explanation.
.TP 10
echo \fIgain-in gain-out delay decay \fR[ \fIdelay decay ... \fR]
Add echoing to a sound sample.
Each delay/decay part gives the delay in milliseconds
and the decay (relative to gain-in) of that echo.
Gain-out is the volume of the output.
.TP 10
echos \fIgain-in gain-out delay decay \fR[ \fIdelay decay ... \fR]
Add a sequence of echos to a sound sample.
Each delay/decay part gives the delay in milliseconds
and the decay (relative to gain-in) of that echo.
Gain-out is the volume of the output.
.TP
fade [ \fItype\fR ] \fIfade-in-length\fR
.TP 10
[ \fIstop-time\fR [ \fIfade-out-length\fR ] ]
Add a fade effect to the beginning, end, or both of the audio data.
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.
For fade-outs, the audio data will be truncated at the stop-time and
the volume will be ramped from full volume down to 0 starting at
\fIfade-out-length\fR seconds before the \fIstop-time\fR. No fade-out
is performed if these options are not specified.
.br
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).
.br
An optional \fItype\fR can be specified to change the type of envelope. Choices are q for quarter of a sinewave, h for half a sinewave, t for linear slope, l for logarithmic, and p for inverted parabola. The default is a linear slope.
.TP 10
filter [ \fIlow\fR ]-[ \fIhigh\fR ] [ \fIwindow-len\fR [ \fIbeta\fR ] ]
Apply a Sinc-windowed lowpass, highpass, or bandpass filter of given
window length to the signal.
\fIlow\fR refers to the frequency of the lower 6dB corner of the filter.
\fIhigh\fR refers to the frequency of the upper 6dB corner of the filter.
A lowpass filter is obtained by leaving \fIlow\fR unspecified, or 0.
A highpass filter is obtained by leaving \fIhigh\fR unspecified, or 0,
or greater than or equal to the Nyquist frequency.
The \fIwindow-len\fR, if unspecified, defaults to 128.
Longer windows give a sharper cutoff, smaller windows a more gradual cutoff.
The \fIbeta\fR, if unspecified, defaults to 16. This selects a Kaiser window.
You can select a Nuttall window by specifying anything <= 2.0 here.
For more discussion of beta, look under the \fBresample\fR effect.
.TP 10
flanger \fIgain-in gain-out delay decay speed\fR < -s | -t >
Add a flanger to a sound sample. Each triple
delay/decay/speed gives the delay in milliseconds
and the decay (relative to gain-in) with a modulation
speed in Hz.
The modulation is either sinodial (-s) or triangular
(-t). Gain-out is the volume of the output.
.TP 10
highp \fIfrequency\fR
Apply a single pole recursive high-pass filter.
The frequency response drops logarithmically with
I frequency
in the middle of the drop.
The slope of the filter is quite gentle.
See \fBfilter\fR for a highpass effect with sharper cutoff.
.TP 10
highpass \fIfrequency\fB
Butterworth highpass filter. Description coming soon!
.TP 10
lowp \fIfrequency\fR
Apply a single pole recursive low-pass filter.
The frequency response drops logarithmically with
.I frequency
in the middle of the drop.
The slope of the filter is quite gentle.
See \fBfilter\fR for a lowpass effect with sharper cutoff.
.TP 10
lowpass \fIfrequency\fB
Butterworth lowpass filter. Description coming soon!
.TP 10
map
Display a list of loops in a sample,
and miscellaneous loop info.
.TP 10
mask
Add "masking noise" to signal.
This effect deliberately adds white noise to a sound
in order to mask quantization effects,
created by the process of playing a sound digitally.
It tends to mask buzzing voices, for example.
It adds 1/2 bit of noise to the sound file at the
output bit depth.
.TP 10
pan \fIdirection\fB
Pan the sound of an audio file from one channel to another. This is done by
changing the volume of the input channels so that it fades out on one
channel and fades-in on another. If the number of input channels is
different then the number of output channels then this effect tries to
intelligently handle this. For instance, if the input contains 1 channel
and the output contains 2 channels, then it will create the missing channel
itself. The
.I direction
is a value from -1.0 to 1.0. -1.0 represents
far left and 1.0 represents far right. Numbers in between will start the
pan effect without totally muting the opposite channel.
.TP 10
phaser \fIgain-in gain-out delay decay speed\fR < -s | -t >
Add a phaser to a sound sample. Each triple
delay/decay/speed gives the delay in milliseconds
and the decay (relative to gain-in) with a modulation
speed in Hz.
The modulation is either sinodial (-s) or triangular
(-t). The decay should be less than 0.5 to avoid
feedback. Gain-out is the volume of the output.
.TP 10
pick [ \fI-1\fR | \fI-2\fR | \fI-3\fR | \fI-4\fR | \fI-l\fR | \fI-r\fR | \fI-f\fR | \fI-b\fR ]
Pick a subset of channels to be copied into the output file. This effect is just an alias of the "avg" effect but is left here for historical reasons.
.TP 10
pitch \fIshift [ width interpole fade ]\fB
Change the pitch of file without affecting its duration by cross-fading
shifted samples.
.I shift
is given in cents. Use a positive value to shift to treble, negative value to shift to bass.
Default shift is 0.
.I width
of window is in ms. Default width is 20ms. Try 30ms to lower pitch,
and 10ms to raise pitch.
.I interpole
option, can be "cubic" or "linear". Default is "cubic". The
.I fade
option, can be "cos", "hamming", "linear" or "trapezoid".
Default is "cos".
.TP
polyphase [ \fI-w \fR< \fInut\fR / \fIham\fR > ]
.TP
[ \fI -width \fR< \fI long \fR / \fIshort \fR / \fI# \fR> ]
.TP 10
[ \fI-cutoff # \fR ]
Translate input sampling rate to output sampling rate via polyphase
interpolation, a DSP algorithm. This method is slow and uses lots
of RAM, but gives much better results than
.B rate.
.br
-w < nut / ham > : select either a Nuttal (~90 dB stopband) or Hamming
(~43 dB stopband) window. Default is
.I nut.
.br
-width long / short / # : specify the (approximate) width of the filter.
.I long
is 1024 samples;
.I short
is 128 samples. Alternatively, an exact number can be used. Default is
.I long.
The
.I short
option is
.B not
recommended, as it produces poor quality results.
.br
-cutoff # : specify the filter cutoff frequency in terms of fraction of
frequency bandwidth, also know as the Nyquist frequency. Please see
the \fIresample\fR effect for
further information on Nyquist frequency. If upsampling, then this is the
fraction of the original signal
that should go through. If downsampling, this is the fraction of the
signal left after downsampling. Default is 0.95. Remember that
this is a float.
.TP 10
rate
Translate input sampling rate to output sampling rate
via linear interpolation to the Least Common Multiple
of the two sampling rates.
This is the default effect
if the two files have different sampling rates and the preview options
was specified.
This is fast but noisy:
the spectrum of the original sound will be shifted upwards
and duplicated faintly when up-translating by a multiple.
Lerp-ing is acceptable for cheap 8-bit sound hardware,
but for CD-quality sound you should instead use either
.B resample
or
.B polyphase.
If you are wondering which rate changing effects to use, you will want to read a
detailed analysis of all of them at http://eakaw2.et.tu-dresden.de/~wilde/resample/resample.html
.TP 10
repeat \fIcount\fR
Repeats the audio data \fIcount\fR times. Requires disk space to store the data to be repeated.
.TP 10
resample [ \fI-qs\fB | \fI-q\fB | \fI-ql\fB ] [ \fIrolloff\fB [ \fIbeta\fB ] ]\fR
Translate input sampling rate to output sampling rate
via simulated analog filtration.
This method is slower than
.B rate,
but gives much better results.
By default, linear interpolation is used,
with a window width about 45 samples at the lower of the two rate.
This gives an accuracy of about 16 bits, but insufficient stopband rejection
in the case that you want to have rolloff greater than about 0.80 of
the Nyquist frequency.
The \fI-q*\fR options will change the default values for rolloff and beta
as well as use quadratic interpolation of filter
coefficients, resulting in about 24 bits precision.
The \fI-qs\fR, \fI-q\fR, or \fI-ql\fR options specify increased accuracy
at the cost of lower execution speed. It is optional to specify
rolloff and beta parameters when using the \fI-q*\fR options.
Following is a table of the reasonable defaults which are built-in to SoX:
.br
\fBOption Window rolloff beta interpolation\fR
.br
\fB------ ------ ------- ---- -------------\fR
.br
(none) 45 0.80 16 linear
.br
-qs 45 0.80 16 quadratic
.br
-q 75 0.875 16 quadratic
.br
-ql 149 0.94 16 quadratic
.br
\fB------ ------ ------- ---- -------------\fR
\fI-qs\fR, \fI-q\fR, or \fI-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.
\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 0.0 and 1.0, in practice 0.8-0.95. The defaults are
indicated above.
The \fINyquist frequency\fR is equal to (sample rate / 2). 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.
Similar problems will happen in software when reducing the sample rate of
an audio file (frequencies above the new Nyquist frequency can be aliased
to lower frequencies). Therefore, a good resample effect
will remove all frequency information above the new Nyquist frequency.
The \fIrolloff\fR refers to how close to the Nyquist frequency this cutoff
is, with closer being better. When increasing the sample rate of an
audio file you would not expect to have any frequencies exist that are
past the original Nyquist frequency. Because of resampling properties, it
is common to have aliasing data created that is above the old
Nyquist frequency. In that case the \fIrolloff\fR refers to how close
to the original Nyquist frequency to use a highpass filter to remove
this false data, with closer also being better.
The \fIbeta\fR parameter
determines the type of filter window used. Any value greater than 2.0 is
the beta for a Kaiser window. Beta <= 2.0 selects a Nuttall window.
If unspecified, the default is a Kaiser window with beta 16.
In the case of Kaiser window (beta > 2.0), lower betas produce a somewhat
faster transition from passband to stopband, at the cost of noticeable artifacts.
A beta of 16 is the default, beta less than 10 is not recommended. If you want
a sharper cutoff, don't use low beta's, use a longer sample window.
A Nuttall window is selected by specifying any 'beta' <= 2, and the
Nuttall window has somewhat steeper cutoff than the default Kaiser window.
You will probably not need to use the beta parameter at all, unless you are
just curious about comparing the effects of Nuttall vs. Kaiser windows.
This is the default effect if the two files have different sampling rates.
Default parameters are, as indicated above, Kaiser window of length 45,
rolloff 0.80, beta 16, linear interpolation.
\fBNOTE:\fR \fI-qs\fR is only slightly slower, but more accurate for
16-bit or higher precision.
\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
.br
input_rate < output_rate
.br
&&
.br
output_rate/gcd(input_rate,output_rate) <= 511
.br
.TP 10
reverb \fIgain-out reverbe-time delay \fR[ \fIdelay ... \fR]
Add reverberation to a sound sample. Each delay is given
in milliseconds and its feedback is depending on the
reverb-time in milliseconds. Each delay should be in
the range of half to quarter of reverb-time to get
a realistic reverberation. Gain-out is the volume of the
output.
.TP 10
reverse
Reverse the sound sample completely.
Included for finding Satanic subliminals.
.TP
\fBsilence\fR \fIabove_periods\fR [ \fIduration threshold\fR[ \fId\fR | \fI%\fR ]
.TP
[ \fIbelow_periods duration
.TP 10
threshold\fR[ \fId\fR | \fI%\fR ]]
Removes silence from the beginning or end of a sound file. Silence is anything below a specified threshold.
.br
When trimming silence from the beginning of a sound file, you specify a duration of audio that is above a given silence threshold before audio data is processed. You can also specify the count of periods of none silence you want to detect before processing audio data. Specify a period of 0 if you do not want to trim data from the front of the sound file.
.br
When optionally trimming silence form the end of a sound file, you specify the duration of audio that must be below a given threshold before stopping to process audio data. A count of periods that occur below the threshold may also be specified. If this options are not specified then data is not trimmed from the end of the audio file.
.br
Duration counts may be in the format of time, hh:mm:ss.frac, or in the exact count of samples.
.br
Threshold may be suffixed with d, or % to indicated the value is in decibels or a percentage of max value of the sample value. A value of '0%' will look for total silence.
.TP 10
speed [ -c ] \fIfactor\fB
Speed up or down the sound, as a magnetic tape with a speed control.
It affects both pitch and time. A factor of 1.0 means no change,
and is the default.
2.0 doubles speed, thus time length is cut by a half and pitch
is one octave higher.
0.5 halves speed thus time length doubles and pitch is one octave lower.
If the optional -c parameter is used then the factor is specified in "cents".
.TP 10
stat [ \fI-s n\fB ] [\fI-rms\fB ] [ \fI-v\fB ] [ \fI-d\fB ]
Do a statistical check on the input file,
and print results on the standard error file. Audio data is passed
unmodified from input to output file unless used along with the
.B -e
option.
The "Volume Adjustment:" field in the statistics
gives you the argument to the
.B -v
.I number
which will make the sample as loud as possible without clipping.
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.
The
.B -s n
option is used to scale the input data by a given factor. The default value
of n is the max value of a signed long variable (0x7fffffff). Internal effects
always work with signed long PCM data and so the value should relate to this
fact.
The
.B -rms
option will convert all output average values to \fIroot mean square\fR
format.
There is also an optional parameter
.B -d
that will print out a hex dump of the
sound file from the internal buffer that is in 32-bit signed PCM data.
This is mainly only of use in tracking down endian problems that
creep in to SoX on cross-platform versions.
.TP 10
stretch \fIfactor [window fade shift fading]\fB
Time stretch file by a given factor. Change duration without affecting the pitch.
.I factor
of stretching: >1.0 lengthen, <1.0 shorten duration.
.I window
size is in ms. Default is 20ms. The
.I fade
option, can be "lin".
.I shift
ratio, in [0.0 1.0]. Default depends on stretch factor. 1.0
to shorten, 0.8 to lengthen. The
.I fading
ratio, in [0.0 0.5]. The amount of a fade's default depends on factor
and shift.
.TP 10
swap [ \fI1 2\fB | \fI1 2 3 4\fB ]
Swap channels in multi-channel sound files. Optionally, you may
specify the channel order you would like the output in. This defaults
to output channel 2 and then 1 for stereo and 2, 1, 4, 3 for quad-channels.
An interesting
feature is that you may duplicate a given channel by overwriting another.
This is done by repeating an output channel on the command line. For example,
swap 2 2 will overwrite channel 1 with channel 2's data; creating a stereo
file with both channels containing the same audio data.
.TP
synth [ \fIlength\fR ] \fItype mix\fR [ \fIfreq\fR [ \fI-freq2\fR ]
.TP 10
[ \fIoff\fR ] [ \fIph\fR ] [ \fIp1\fR ] [ \fIp2\fR ] [ \fIp3\fR ]
The synth effect will generate various types of audio data. Although
this effect is used to generate audio data, an input file must be specified.
The length of the input audio file determines the length of the output
audio file.
.br
<length> length in sec or hh:mm:ss.frac, 0=inputlength, default=0
.br
<type> is sine, square, triangle, sawtooth, trapetz, exp,
whitenoise, pinknoise, brownnoise, default=sine
.br
<mix> is create, mix, amod, default=create
.br
<freq> frequency at beginning in Hz, not used for noise..
.br
<freq2> frequency at end in Hz, not used for noise..
<freq/2> can be given as %%n, where 'n' is the number of
half notes in respect to A (440Hz)
.br
<off> Bias (DC-offset) of signal in percent, default=0
.br
<ph> phase shift 0..100 shift phase 0..2*Pi, not used for noise..
.br
<p1> square: Ton/Toff, triangle+trapetz: rising slope time (0..100)
.br
<p2> trapetz: ON time (0..100)
.br
<p3> trapetz: falling slope position (0..100)
.TP 10
trim \fIstart\fR [ \fIlength\fR ]
Trim can trim off unwanted audio data from the beginning and end of the
audio file. Audio samples are not sent to the output stream until
the \fIstart\fR location is reached.
.br
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 data. Using a value of 0 for the \fIstart\fR parameter will allow
trimming off the back side only.
.br
Both options can be specified using either an amount of time and 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 data. 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 data.
.TP 10
vibro \fIspeed \fB [ \fIdepth\fB ]
Add the world-famous Fender Vibro-Champ sound
effect to a sound sample by using
a sine wave as the volume knob.
.B Speed
gives the Hertz value of the wave.
This must be under 30.
.B Depth
gives the amount the volume is cut into
by the sine wave,
ranging 0.0 to 1.0 and defaulting to 0.5.
.TP 10
vol \fIgain\fR [ \fItype\fB [ \fIlimitergain\fR ] ]
The vol effect is much like the command line option -v. It allows you to
adjust the volume of an input file and allows you to specify the adjustment
in relation to amplitude, power, or dB. If \fItype\fR is not specified then
it defaults to \fIamplitude\fR.
.br
When type is
.I amplitude
then a linear change of the amplitude is performed based on the gain. Therefore,
a value of 1.0 will keep the volume the same, 0.0 to < 1.0 will cause the
volume to decrease and values of > 1.0 will cause the volume to increase.
Beware of clipping audio data when the gain is greater then 1.0. A negative
value performs the same adjustment while also changing the phase.
.br
When type is
.I power
then a value of 1.0 also means no change in volume.
.br
When type is
.I dB
the amplitude is changed logarithmically.
0.0 is constant while +6 doubles the amplitude.
.br
An optional \fIlimitergain\fR value can be specified and should be a
value much less
then 1.0 (ie 0.05 or 0.02) and is used only on peaks to prevent clipping.
Not specifying this parameter will cause no limiter to be used. In verbose
mode, this effect will display the percentage of audio data that needed to be
limited.
.SH BUGS
The syntax is horrific. Thats the breaks when trying to handle all things from the command line.
.P
Please report any bugs found in this version of SoX to Chris Bagwell (cbagwell@users.sourceforge.net)
.SH FILES
.SH SEE ALSO
.BR play (1),
.BR rec (1),
.BR soxexam(1)
.SH NOTICES
The version of SoX that accompanies this manual page is support by
Chris Bagwell (cbagwell@users.sourceforge.net). Please refer any questions
regarding it to this address. You may obtain the latest version at the
the web site http://sox.sourceforge.net/
.SH AUTHOR
Chris Bagwell (cbagwell@users.sourceforge.net).
.P
Updates by Anonymous