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Annotated edit history of sox(1) version 3, including all changes. View license author blame.

Rev	Author	#	Line
3	JohnMcPherson	1	See also our SoxNotes for user-submitted examples/comments.
		2	----
1	perry	3	SoX
		4	!!!SoX
		5	NAME
		6	SYNOPSIS
		7	DESCRIPTION
		8	OPTIONS
		9	FILE TYPES
		10	EFFECTS
		11	BUGS
		12	FILES
		13	SEE ALSO
		14	NOTICES
		15	AUTHOR
		16	----
		17	!!NAME
		18
		19
		20	sox - Sound eXchange : universal sound sample translator
		21	!!SYNOPSIS
		22
		23
		24	__sox__ ''infile outfile''
		25
		26
		27	__sox__ [[ ''general options'' ] [[ ''format
		28	options'' ] ''infile''
		29	[[ ''format options'' ] ''outfile''
		30	[[ ''effect'' [[ ''effect options'' ] ... ]
		31
		32
		33	__soxmix__ ''infile1 infile2 outfile''
		34
		35
		36	__soxmix__ [[ ''general options'' ] [[ ''format
		37	options'' ] ''infile1''
		38	[[ ''format options'' ] ''infile2''
		39	[[ ''format options'' ] ''outfile''
		40	[[ ''effect'' [[ ''effect options'' ] ... ]
		41
		42
		43	__General options:__
		44	[[ -h ] [[ -p ] [[ -v ''volume'' ] [[ -V ]
		45
		46
		47	__Format options:__
		48	[[ -t ''filetype'' ] [[ -r ''rate'' ] [[
		49	-s/-u/-U/-A/-a/-i/-g/-f ] [[ -b/-w/-l ] [[ -c ''channels''
		50	] [[ -x ] [[ -e ]
		51
		52
		53	__Effects:
		54	avg__ [[ -l | -r | -f | -b | n,n,...,n ]__
		55	band__ [[ -n ] ''center'' [[ ''width'' ]__
		56	bandpass__ ''frequency bandwidth''__
		57	bandreject__ ''frequency bandwidth''__
		58	chorus__ ''gain-in gain out delay decay speed
		59	depth''
		60	-s | -t [[ ''delay decay speed depth'' -s | -t ]__
		61	compand__
		62	''attack1'',''decay1''[[,''attack2'',''decay2''...]''
		63	in-dB1'',''out-dB1''[[,''in-dB2'',''out-dB2''...]
		64	[[ ''gain'' [[ ''initial-volume'' [[ ''delay'' ] ]
		65	]__
		66	copy
		67	dcshift__ ''shift'' [[ ''limitergain'' ]__
		68	deemph
		69	earwax
		70	echo__ ''gain-in gain-out delay decay'' [[ ''delay
		71	decay ...'' ]__
		72	echos__ ''gain-in gain-out delay decay'' [[ ''delay
		73	decay ...'' ]__
		74	fade__ [[ ''type'' ] ''fade-in-length'' [[
		75	''stop-time'' [[ ''fade-out-length'' ] ]__
		76	filter__ [[ ''low'' ]-[[ ''high'' ] [[
		77	''window-len'' [[ ''beta'' ]]__
		78	flanger__ ''gain-in gain-out delay decay speed''
		79	''
		80	highp__ ''frequency''__
		81	highpass__ ''frequency''__
		82	lowp__ ''frequency''__
		83	lowpass__ ''frequency''__
		84	map
		85	mask
		86	pan__ ''direction''__
		87	phaser__ ''gain-in gain-out delay decay speed''
		88	''
		89	pick__ [[ ''-1'' | ''-2'' | ''-3'' | ''-4'' |
		90	''-l'' | ''-r'' ]__
		91	pitch__ ''shift'' [[ ''width interpole fade''
		92	]__
		93	polyphase__ [[ -w __nut'' / ''ham''
		94	''-width'' ''long'' / ''short'' / #
		95	''-cutoff #'' ]__
		96	rate
		97	resample__ [[ -qs | -q | -ql ] [[ ''rolloff'' [[
		98	''beta'' ] ]__
		99	reverb__ ''gain-out reverb-time delay'' [[ ''delay''
		100	... ]__
		101	reverse
		102	silence__ ''above_periods'' [[ ''duration
		103	threshold''[[ ''d'' | ''%'' ] [[ ''below_periods
		104	duration threshold''[[ ''d'' | ''%'' ]]__
		105	speed__ [[ -c ] ''factor''__
		106	split
		107	stat__ [[ -s ''n'' ] [[ -rms ] [[ -v ] [[ -d ]__
		108	stretch__ [[ ''factor'' [[ ''window fade shift
		109	fading'' ]__
		110	swap__ [[ ''1 2'' | ''1 2 3 4'' ]__
		111	synth__ [[ ''length'' ] ''type mix'' [[ ''freq'' [[
		112	''-freq2'' ] [[ ''off'' ] [[ ''ph'' ] [[ ''p1'' ] [[
		113	''p2'' ] [[ ''p3'' ]__
		114	trim__ ''start'' [[ ''length'' ]__
		115	vibro__ ''speed'' [[ ''depth'' ]__
		116	vol__ ''gain'' [[ ''type'' [[ ''limitergain'' ]
		117	]
		118	!!DESCRIPTION
		119
		120
		121	''SoX'' is a command line program that can convert most
		122	popular audio files to most other popular audio file
		123	formats. It can optionally change the audio sample data type
		124	and apply one or more sound effects to the file during this
		125	translation.
		126
		127
		128	''soxmix'' is functionally the same as the command line
		129	program ''sox'' expect that it takes two files as input
		130	and mixes the audio together to produce a single file as
		131	output. It has a restriction that both input files must be
		132	of the same data type and sample rates.
		133
		134
		135	There are two types of audio files formats that ''SoX''
		136	can work with. The first are self-describing file formats.
		137	These contain a header that completely describe the
		138	characteristics of the audio data that follows.
		139
		140
		141	The second type are header-less data, or sometimes called
		142	raw data. A user must pass enough information to ''SoX''
		143	on the command line so that it knows what type of data it
		144	contains.
		145
		146
		147	Audio data can usually be totally described by four
		148	characteristics:
		149
		150
		151	rate
		152
		153
		154	The sample rate is in samples per second. For example, CD
		155	sample rates are at 44100.
		156
		157
		158	data size
		159
		160
		161	The precision the data is stored in. Most popular are 8-bit
		162	bytes or 16-bit words.
		163
		164
		165	data encoding
		166
		167
		168	What encoding the data type uses. Examples are u-law, ADPCM,
		169	or signed linear data.
		170
		171
		172	channels
		173
		174
		175	How many channels are contained in the audio data. Mono and
		176	Stereo are the two most common.
		177
		178
		179	Please refer to the __soxexam(1)__ manual page for a long
		180	description with examples on how to use SoX with various
		181	types of file formats.
		182	!!OPTIONS
		183
		184
		185	The option syntax is a little grotty, but in
		186	essence:
		187
		188
		189	sox File.au file.wav
		190
		191
		192	translates a sound file in SUN Sparc .AU format into a
		193	Microsoft .WAV file, while
		194
		195
		196	sox -v 0.5 file.au -r 12000 file.wav mask
		197
		198
		199	does the same format translation but also lowers the
		200	amplitude by 1/2, changes the sampling rate to 12000 hertz,
		201	and applies the __mask__ sound effect to the audio
		202	data.
		203
		204
		205	The following will mix two sound files together to to
		206	produce a single sound file.
		207
		208
		209	soxmix music.wav voice.wav mixed.wav
		210
		211
		212	__Format options:__
		213
		214
		215	Format options effect the audio samples that they
		216	immediately precede. If they are placed before the input
		217	file name then they effect the input data. If they are
		218	placed before the output file name then they will effect the
		219	output data. By taking advantage of this, you can override a
		220	input file's corrupted header or produce an output file that
		221	is totally different style then the input file. It is also
		222	how SoX is informed about the format of raw input
		223	data.
		224
		225
		226	__-t__ ''filetype''
		227
		228
		229	gives the type of the sound sample file. Useful when file
		230	extension is not standard or for specifying the .auto file
		231	type.
		232
		233
		234	__-r__ ''rate''
		235
		236
		237	Gives the sample rate in Hertz of the file. To cause the
		238	output file to have a different sample rate than the input
		239	file, include this option as a part of the output
		240	options.
		241	If the input and output files have different rates then a
		242	sample rate change effect must be ran. If a sample rate
		243	changing effect is not specified then a default one will
		244	internally be ran by SoX using its default
		245	parameters.
		246
		247
		248	__-s/-u/-U/-A/-a/-i/-g/-f__
		249
		250
		251	The sample data encoding is signed linear (2's complement),
		252	unsigned linear, u-law (logarithmic), A-law (logarithmic),
		253	ADPCM, IMA_ADPCM, GSM, or Floating-point.
		254	U-law (actually shorthand for mu-law) and A-law are the U.S.
		255	and international standards for logarithmic telephone sound
		256	compression. When uncompressed u-law has roughly the
		257	precision of 14-byte PCM audio and A-law has roughly the
		258	precision of 13-bit PCM audio.
		259	A-law and u-law data is sometimes encoded using a reversed
		260	bit-ordering (ie. MSB becomes LSB). Internally, SoX
		261	understands how to work with this encoding but there is
		262	currently no command line option to specify it. If you need
		263	this support then you can use the psuedo file types of
		264	ADPCM is a form of sound compression that has a good
		265	compromise between good sound quality and fast
		266	encoding/decoding time. It is used for telephone sound
		267	compression and places were full fidelity is not as
		268	important. When uncompressed it has roughly the precision of
		269	16-bit PCM audio. Popular version of ADPCM include G.726, MS
		270	ADPCM, and IMA ADPCM. The __-a__ flag has different
		271	meanings in different file handlers. In __.wav__ files it
		272	represents MS ADPCM files, in all others it means G.726
		273	ADPCM. IMA ADPCM is a specific form of ADPCM compression,
		274	slightly simpler and slightly lower fidelity than
		275	Microsoft's flavor of ADPCM. IMA ADPCM is also called DVI
		276	ADPCM.
		277	GSM is a standard used for telephone sound compression in
		278	European countries and its gaining popularity because of its
		279	quality. It usually is CPU intensive to work with GSM audio
		280	data.
		281
		282
		283	__-b/-w/-l__
		284
		285
		286	The sample data size is in bytes, 16-bit words, or 32-bit
		287	long words.
		288
		289
		290	__-x__ The sample data is in XINU format; that is, it
		291	comes from a machine with the opposite word order than yours
		292	and must be swapped according to the word-size given above.
		293	Only 16-bit and 32-bit integer data may be swapped.
		294	Machine-format floating-point data is not
		295	portable.
		296
		297
		298	__-c__ ''channels''
		299
		300
		301	The number of sound channels in the data file. This may be
		302	1, 2, or 4; for mono, stereo, or quad sound data. To cause
		303	the output file to have a different number of channels than
		304	the input file, include this option with the output file
		305	options. If the input and output file have a different
		306	number of channels then the avg effect must be used. If the
		307	avg effect is not specified on the command line it will be
		308	invoked internally with default parameters.
		309
		310
		311	__-e__ When used after the input filename (so that it
		312	applies to the output file) it allows you to avoid giving an
		313	output filename and will not produce an output file. It will
		314	apply any specified effects to the input file. This is
		315	mainly useful with the __stat__ effect but can be used
		316	with others.
		317
		318
		319	__General options:__
		320
		321
		322	__-h__ Print version number and usage
		323	information.
		324
		325
		326	__-p__ Run in preview mode and run fast. This will
		327	somewhat speed up SoX when the output format has a different
		328	number of channels and a different rate than the input file.
		329	Currently, this defaults to using the __rate__ effect
		330	instead of the __resample__ effect for sample rate
		331	changes.
		332
		333
		334	__-v__ ''volume''
		335
		336
		337	Change amplitude (floating point); less than 1.0 decreases,
		338	greater than 1.0 increases. May use a negative number to
		339	invert the phase of the audio data. It is interesting to
		340	note that we perceive volume logarithmically but this
		341	adjusts the amplitude linearly.
		342	Note: see the __stat__ effect for information on finding
		343	the maximum value that can be used with this option without
		344	causing audio data be be clipped.
		345
		346
		347	__-V__ Print a description of processing phases. Useful
		348	for figuring out exactly how ''SoX'' is mangling your
		349	sound samples.
		350	!!FILE TYPES
		351
		352
		353	''SoX'' attempts to determine the file type of input
		354	files automatically by looking at the header of the audio
		355	file. When it is unable to detect the file type or if its an
		356	output file then it uses the file extension of the file to
		357	determine what type of file format handler to use. This can
		358	be overridden by specifying the
		359	''
		360
		361
		362	The input and output files may be read from standard in and
		363	out. This is done by specifying '-' as the
		364	filename.
		365
		366
		367	File formats which have headers are checked, if that header
		368	doesn't seem right, the program exits with an appropriate
		369	message.
		370
		371
		372	The following file formats are supported:
		373
		374
		375	__.8svx__
		376
		377
		378	Amiga 8SVX musical instrument description
		379	format.
		380
		381
		382	__.aiff__
		383
		384
		385	AIFF files used on Apple IIc/IIgs and SGI. Note: the AIFF
		386	format supports only one SSND chunk. It does not support
		387	multiple sound chunks, or the 8SVX musical instrument
		388	description format. AIFF files are multimedia archives and
		389	can have multiple audio and picture chunks. You may need a
		390	separate archiver to work with them.
		391
		392
		393	__.au__
		394
		395
		396	SUN Microsystems AU files. There are apparently many types
		397	of .au files; DEC has invented its own with a different
		398	magic number and word order. The .au handler can read these
		399	files but will not write them. Some .au files have valid AU
		400	headers and some do not. The latter are probably original
		401	SUN u-law 8000 hz samples. These can be dealt with using the
		402	__.ul__ format (see below).
		403
		404
		405	__.avr__
		406
		407
		408	Audio Visual Research
		409	The AVR format is produced by a number of commercial
		410	packages on the Mac.
		411
		412
		413	__.cdr__
		414
		415
		416	CD-R
		417	CD-R files are used in mastering music on Compact Disks. The
		418	audio data on a CD-R disk is a raw audio file with a format
		419	of stereo 16-bit signed samples at a 44khz sample rate.
		420	There is a special blocking/padding oddity at the end of the
		421	audio file and is why it needs its own handler.
		422
		423
		424	__.cvs__
		425
		426
		427	Continuously Variable Slope Delta modulation
		428	Used to compress speech audio for applications such as voice
		429	mail.
		430
		431
		432	__.dat__
		433
		434
		435	Text Data files
		436	These files contain a textual representation of the sample
		437	data. There is one line at the beginning that contains the
		438	sample rate. Subsequent lines contain two numeric data
		439	items: the time since the beginning of the first sample and
		440	the sample value. Values are normalized so that the maximum
		441	and minimum are 1.00 and -1.00. This file format can be used
		442	to create data files for external programs such as FFT
		443	analyzers or graph routines. SoX can also convert a file in
		444	this format back into one of the other file
		445	formats.
		446
		447
		448	__.gsm__
		449
		450
		451	GSM 06.10 Lossy Speech Compression
		452	A standard for compressing speech which is used in the
		453	Global Standard for Mobil telecommunications (GSM). Its good
		454	for its purpose, shrinking audio data size, but it will
		455	introduce lots of noise when a given sound sample is encoded
		456	and decoded multiple times. This format is used by some
		457	voice mail applications. It is rather CPU intensive.
		458	GSM in __SoX__ is optional and requires access to an
		459	external GSM library. To see if there is support for gsm run
		460	__sox -h__ and look for it under the list of supported
		461	file formats.
		462
		463
		464	__.hcom__
		465
		466
		467	Macintosh HCOM files. These are (apparently) Mac FSSD files
		468	with some variant of Huffman compression. The Macintosh has
		469	wacky file formats and this format handler apparently
		470	doesn't handle all the ones it should. Mac users will need
		471	your usual arsenal of file converters to deal with an HCOM
		472	file under Unix or DOS.
		473
		474
		475	__.maud__
		476
		477
		478	An Amiga format
2	perry	479	An IFF-conform sound file type, registered by MS !MacroSystem
1	perry	480	Computer GmbH, published along with the
		481
		482
		483	__.nul__
		484
		485
		486	Null file handler. This is a fake file hander that act as if
		487	its reading a stream of 0's from a while or fake writing
		488	output to a file. This is not a very useful file handler in
		489	most cases. It might be useful in some scripts were you do
		490	not want to read or write from a real file but would like to
		491	specify a filename for consistency.
		492
		493
		494	__.ogg__
		495
		496
		497	Ogg Vorbis Compressed Audio.
		498	Ogg Vorbis is a open, patent-free CODEC designed for
		499	compressing music and streaming audio. It is similar to MP3,
		500	VQF, AAC, and other lossy formats. __SoX__ can decode all
		501	types of Ogg Vorbis files, but can only encode at 128 kbps.
		502	Decoding is somewhat CPU intensive and encoding is very CPU
		503	intensive.
		504	Ogg Vorbis in __SoX__ is optional and requires access to
		505	external Ogg Vorbis libraries. To see if there is support
		506	for Ogg Vorbis run __sox -h__ and look for it under the
		507	list of supported file formats as
		508	__
		509
		510
		511	__ossdsp__
		512
		513
		514	OSS /dev/dsp device driver
		515	This is a pseudo-file type and can be optionally compiled
		516	into SoX. Run __sox -h__ to see if you have support for
		517	this file type. When this driver is used it allows you to
		518	open up the OSS /dev/dsp file and configure it to use the
		519	same data format as passed in to __SoX__. It works for
		520	both playing and recording sound samples. When playing sound
		521	files it attempts to set up the OSS driver to use the same
		522	format as the input file. It is suggested to always override
		523	the output values to use the highest quality samples your
		524	sound card can handle. Example: ''-t ossdsp -w -s
		525	/dev/dsp''
		526
		527
		528	__.sf__
		529
		530
		531	IRCAM Sound Files.
		532	Sound Files are used by academic music software such as the
2	perry	533	CSound package, and the !MixView sound sample
1	perry	534	editor.
		535
		536
		537	__.sph__
		538
		539
		540	SPHERE (SPeech HEader Resources) is a file format defined by
		541	NIST (National Institute of Standards and Technology) and is
		542	used with speech audio. SoX can read these files when they
		543	contain u-law and PCM data. It will ignore any header
		544	information that says the data is compressed using
		545	''shorten'' compression and will treat the data as either
		546	u-law or PCM. This will allow SoX and the command line
		547	''shorten'' program to be ran together using pipes to
		548	uncompress the data and then pass the result to SoX for
		549	processing.
		550
		551
		552	__.smp__
		553
		554
2	perry	555	Turtle Beach !SampleVision files.
		556	SMP files are for use with the PC-DOS package !SampleVision
1	perry	557	by Turtle Beach Softworks. This package is for communication
		558	to several MIDI samplers. All sample rates are supported by
		559	the package, although not all are supported by the samplers
		560	themselves. Currently loop points are ignored.
		561
		562
		563	__.snd__
		564
		565
		566	Under DOS this file format is the same as the __.sndt__
		567	format. Under all other platforms it is the same as the
		568	__.au__ format.
		569
		570
		571	__.sndt__
		572
		573
2	perry	574	!SoundTool files.
1	perry	575	This is an older DOS file format.
		576
		577
		578	__sunau__
		579
		580
		581	Sun /dev/audio device driver
		582	This is a pseudo-file type and can be optionally compiled
		583	into SoX. Run __sox -h__ to see if you have support for
		584	this file type. When this driver is used it allows you to
		585	open up a Sun /dev/audio file and configure it to use the
		586	same data type as passed in to __SoX.__ It works for both
		587	playing and recording sound samples. When playing sound
		588	files it attempts to set up the audio driver to use the same
		589	format as the input file. It is suggested to always override
		590	the output values to use the highest quality samples your
		591	hardware can handle. Example: ''-t sunau -w -s
		592	/dev/audio'' or ''-t sunau -U -c 1 /dev/audio'' for
		593	older sun equipment.
		594
		595
		596	__.txw__
		597
		598
		599	Yamaha TX-16W sampler.
		600	A file format from a Yamaha sampling keyboard which wrote
		601	IBM-PC format 3.5
		602
		603
		604	__.vms__
		605
		606
		607	More info to come.
		608	Used to compress speech audio for applications such as voice
		609	mail.
		610
		611
		612	__.voc__
		613
		614
		615	Sound Blaster VOC files.
		616	VOC files are multi-part and contain silence parts, looping,
		617	and different sample rates for different chunks. On input,
		618	the silence parts are filled out, loops are rejected, and
		619	sample data with a new sample rate is rejected. Silence with
		620	a different sample rate is generated appropriately. On
		621	output, silence is not detected, nor are impossible sample
		622	rates. Note, this version now supports playing VOC files
		623	with multiple blocks and supports playing files containing
		624	u-law and A-law samples.
		625
		626
		627	__vorbis__
		628
		629
		630	See __.ogg__ format.
		631
		632
		633	__.wav__
		634
		635
		636	Microsoft .WAV RIFF files.
		637	These appear to be very similar to IFF files, but not the
		638	same. They are the native sound file format of Windows.
		639	(Obviously, Windows was of such incredible importance to the
		640	computer industry that it just had to have its own sound
		641	file format.) Normally __.wav__ files have all formatting
		642	information in their headers, and so do not need any format
		643	options specified for an input file. If any are, they will
		644	override the file header, and you will be warned to this
		645	effect. You had better know what you are doing! Output
		646	format options will cause a format conversion, and the
		647	__.wav__ will written appropriately. SoX currently can
		648	read PCM, ULAW, ALAW, MS ADPCM, and IMA (or DVI) ADPCM. It
		649	can write all of these formats including __(NEW!)__ the
		650	ADPCM encoding.
		651
		652
		653	__.wve__
		654
		655
		656	Psion 8-bit A-law
		657	These are 8-bit A-law 8khz sound files used on the Psion
		658	palmtop portable computer.
		659
		660
		661	__.raw__
		662
		663
		664	Raw files (no header).
		665	The sample rate, size (byte, word, etc), and encoding
		666	(signed, unsigned, etc.) of the sample file must be given.
		667	The number of channels defaults to 1.
		668
		669
		670	__.ub, .sb, .uw, .sw, .ul, .al, .lu, .la,
		671	.sl__
		672
		673
		674	These are several suffices which serve as a shorthand for
		675	raw files with a given size and encoding. Thus, __ub, sb,
		676	uw, sw, ul, al, lu, la__ and __sl__ correspond to
		677	__
		678
		679
		680	__.auto__
		681
		682
		683	This is a ``meta-type'': specifying this type for an input
		684	file triggers some code that tries to guess the real type by
		685	looking for magic words in the header. If the type can't be
		686	guessed, the program exits with an error message. The input
		687	must be a plain file, not a pipe. This type can't be used
		688	for output files.
		689	!!EFFECTS
		690
		691
		692	Multiple effects may be applied to the audio data by
		693	specifying them one after another at the end of the command
		694	line.
		695
		696
		697	avg [[ ''-l'' | ''-r'' | ''-f'' | ''-b'' |
		698	''n,n,...,n'' ]
		699
		700
		701	Reduce the number of channels by averaging the samples, or
		702	duplicate channels to increase the number of channels. This
		703	effect is automatically used when the number of input
		704	channels differ from the number of output channels. When
		705	reducing the number of channels it is possible to manually
		706	specify the avg effect and use the ''-l'', ''-r'',
		707	''-f'', or ''-b'' options to select only the left,
		708	right, front, or back channel(s) for the output instead of
		709	averaging the channels. The ''-f'' and ''-b'' options
		710	maintain left/right stereo separation; use the avg effect
		711	twice to select a single channel.
		712
		713
		714	The avg effect can also be invoked with up to 16
		715	double-precision numbers, which specify the proportion of
		716	each input channel that is to be mixed into each output
		717	channel. In two-channel mode, 4 numbers are given: l-
		718
		719
		720	It is also possible to use the 16 numbers to expand or
		721	reduce the channel count; just specify 0 for unused
		722	channels. Finally, if fewer than 4 numbers are given,
		723	certain special abbreviations may be invoked; see the source
		724	code for details.
		725
		726
		727	band __[[__ ''-n'' __]__ ''center'' __[[__
		728	''width'' __]__
		729
		730
		731	Apply a band-pass filter. The frequency response drops
		732	logarithmically around the ''center'' frequency. The
		733	''width'' gives the slope of the drop. The frequencies at
		734	''center + width'' and ''center - width'' will be half
		735	of their original amplitudes. __Band__ defaults to a mode
		736	oriented to pitched signals, i.e. voice, singing, or
		737	instrumental music. The ''-n'' (for noise) option uses
		738	the alternate mode for un-pitched signals. __Warning:__
		739	''-n'' introduces a power-gain of about 11dB in the
		740	filter, so beware of output clipping. __Band__ introduces
		741	noise in the shape of the filter, i.e. peaking at the
		742	''center'' frequency and settling around it. See
		743	__filter__ for a bandpass effect with steeper
		744	shoulders.
		745
		746
		747	bandpass ''frequency bandwidth''
		748
		749
		750	Butterworth bandpass filter. Description coming
		751	soon!
		752
		753
		754	bandreject ''frequency bandwidth''
		755
		756
		757	Butterworth bandreject filter. Description coming
		758	soon!
		759
		760
		761	chorus ''gain-in gain-out delay decay speed
		762	depth''
		763
		764
		765	-s | ''-t [[ delay decay speed depth -s'' | ''-t ...''
		766	]
		767
		768
		769	Add a chorus to a sound sample. Each quadtuple
		770	delay/decay/speed/depth gives the delay in milliseconds and
		771	the decay (relative to gain-in) with a modulation speed in
		772	Hz using depth in milliseconds. The modulation is either
		773	sinusoidal (-s) or triangular (-t). Gain-out is the volume
		774	of the output.
		775
		776
		777	compand
		778	''attack1,decay1''[[,''attack2,decay2''...]
		779
		780
		781	''in-dB1,out-dB1''[[,''in-dB2,out-dB2''...]
		782
		783
		784	[[''gain'' [[''initial-volume'' [[''delay'' ] ]
		785	]
		786
		787
		788	Compand (compress or expand) the dynamic range of a sample.
		789	The attack and decay time specify the integration time over
		790	which the absolute value of the input signal is integrated
		791	to determine its volume; attacks refer to increases in
		792	volume and decays refer to decreases. Where more than one
		793	pair of attack/decay parameters are specified, each channel
		794	is treated separately and the number of pairs must agree
		795	with the number of input channels. The second parameter is a
		796	list of points on the compander's transfer function
		797	specified in dB relative to the maximum possible signal
		798	amplitude. The input values must be in a strictly increasing
		799	order but the transfer function does not have to be
		800	monotonically rising. The special value ''-inf'' may be
		801	used to indicate that the input volume should be associated
		802	output volume. The points ''-inf,-inf'' and ''0,0''
		803	are assumed; the latter may be overridden, but the former
		804	may not.
		805
		806
		807	The third (optional) parameter is a post-processing gain in
		808	dB which is applied after the compression has taken place;
		809	the fourth (optional) parameter is an initial volume to be
		810	assumed for each channel when the effect starts. This
		811	permits the user to supply a nominal level initially, so
		812	that, for example, a very large gain is not applied to
		813	initial signal levels before the companding action has begun
		814	to operate: it is quite probable that in such an event, the
		815	output would be severely clipped while the compander gain
		816	properly adjusts itself.
		817
		818
		819	The fifth (optional) parameter is a delay in seconds. The
		820	input signal is analyzed immediately to control the
		821	compander, but it is delayed before being fed to the volume
		822	adjuster. Specifying a delay approximately equal to the
		823	attack/decay times allows the compander to effectively
		824	operate in a
		825
		826
		827	copy
		828
		829
		830	Copy the input file to the output file. This is the default
		831	effect if both files have the same sampling
		832	rate.
		833
		834
		835	dcshift ''shift'' [[ ''limitergain'' ]
		836
		837
		838	DC Shift the audio data, with basic linear amplitude
		839	formula. This is most useful if your audio data tends to not
		840	be centered around a value of 0. Shifting it back will allow
		841	you to get the most volume adjustments without clipping
		842	audio data.
		843	The first option is the ''dcshift'' value. It is a
		844	floating point number that indicates the amount to
		845	shift.
		846	An option limtergain value can be specified as well. It
		847	should have a value much less then 1.0 and is used only on
		848	peaks to prevent clipping.
		849
		850
		851	deemph
		852
		853
		854	Apply a treble attenuation shelving filter to samples in
		855	audio cd format. The frequency response of pre-emphasized
		856	recordings is rectified. The filtering is defined in the
		857	standard document ISO 908.
		858
		859
		860	earwax
		861
		862
		863	Makes sound easier to listen to on headphones. Adds
		864	audio-cues to samples in audio cd format so that when
		865	listened to on headphones the stereo image is moved from
		866	inside your head (standard for headphones) to outside and in
		867	front of the listener (standard for speakers). See
		868	www.geocities.com/beinges for a full
		869	explanation.
		870
		871
		872	echo ''gain-in gain-out delay decay'' [[ ''delay decay
		873	...'' ]
		874
		875
		876	Add echoing to a sound sample. Each delay/decay part gives
		877	the delay in milliseconds and the decay (relative to
		878	gain-in) of that echo. Gain-out is the volume of the
		879	output.
		880
		881
		882	echos ''gain-in gain-out delay decay'' [[ ''delay decay
		883	...'' ]
		884
		885
		886	Add a sequence of echos to a sound sample. Each delay/decay
		887	part gives the delay in milliseconds and the decay (relative
		888	to gain-in) of that echo. Gain-out is the volume of the
		889	output.
		890
		891
		892	fade [[ ''type'' ] ''fade-in-length''
		893
		894
		895	[[ ''stop-time'' [[ ''fade-out-length'' ] ]
		896
		897
		898	Add a fade effect to the beginning, end, or both of the
		899	audio data.
		900
		901
		902	For fade-ins, this starts from the first sample and ramps
		903	the volume of the audio from 0 to full volume over
		904	''fade-in-length'' seconds. Specify 0 seconds if no
		905	fade-in is wanted.
		906
		907
		908	For fade-outs, the audio data will be truncated at the
		909	stop-time and the volume will be ramped from full volume
		910	down to 0 starting at ''fade-out-length'' seconds before
		911	the ''stop-time''. No fade-out is performed if these
		912	options are not specified.
		913	All times can be specified in either periods of time or
		914	sample counts. To specify time periods use the format
		915	hh:mm:ss.frac format. To specify using sample counts,
		916	specify the number of samples and append the letter 's' to
		917	the sample count (for example 8000s).
		918	An optional ''type'' can be specified to change the type
		919	of envelope. Choices are q for quarter of a sinewave, h for
		920	half a sinewave, t for linear slope, l for logarithmic, and
		921	p for inverted parabola. The default is a linear
		922	slope.
		923
		924
		925	filter [[ ''low'' ]-[[ ''high'' ] [[ ''window-len'' [[
		926	''beta'' ] ]
		927
		928
		929	Apply a Sinc-windowed lowpass, highpass, or bandpass filter
		930	of given window length to the signal. ''low'' refers to
		931	the frequency of the lower 6dB corner of the filter.
		932	''high'' refers to the frequency of the upper 6dB corner
		933	of the filter.
		934
		935
		936	A lowpass filter is obtained by leaving ''low''
		937	unspecified, or 0. A highpass filter is obtained by leaving
		938	''high'' unspecified, or 0, or greater than or equal to
		939	the Nyquist frequency.
		940
		941
		942	The ''window-len'', if unspecified, defaults to 128.
		943	Longer windows give a sharper cutoff, smaller windows a more
		944	gradual cutoff.
		945
		946
		947	The ''beta'', if unspecified, defaults to 16. This
		948	selects a Kaiser window. You can select a Nuttall window by
		949	specifying anything
		950	''resample__ effect.
		951
		952
		953	flanger ''gain-in gain-out delay decay speed''
		954	''
		955
		956
		957	Add a flanger to a sound sample. Each triple
		958	delay/decay/speed gives the delay in milliseconds and the
		959	decay (relative to gain-in) with a modulation speed in Hz.
		960	The modulation is either sinodial (-s) or triangular (-t).
		961	Gain-out is the volume of the output.
		962
		963
		964	highp ''frequency''
		965
		966
		967	Apply a single pole recursive high-pass filter. The
		968	frequency response drops logarithmically with I frequency in
		969	the middle of the drop. The slope of the filter is quite
		970	gentle. See __filter__ for a highpass effect with sharper
		971	cutoff.
		972
		973
		974	highpass ''frequency''
		975
		976
		977	Butterworth highpass filter. Description coming
		978	soon!
		979
		980
		981	lowp ''frequency''
		982
		983
		984	Apply a single pole recursive low-pass filter. The frequency
		985	response drops logarithmically with ''frequency'' in the
		986	middle of the drop. The slope of the filter is quite gentle.
		987	See __filter__ for a lowpass effect with sharper
		988	cutoff.
		989
		990
		991	lowpass ''frequency''
		992
		993
		994	Butterworth lowpass filter. Description coming
		995	soon!
		996
		997
		998	map
		999
		1000
		1001	Display a list of loops in a sample, and miscellaneous loop
		1002	info.
		1003
		1004
		1005	mask
		1006
		1007
		1008	Add
		1009
		1010
		1011	pan ''direction''
		1012
		1013
		1014	Pan the sound of an audio file from one channel to another.
		1015	This is done by changing the volume of the input channels so
		1016	that it fades out on one channel and fades-in on another. If
		1017	the number of input channels is different then the number of
		1018	output channels then this effect tries to intelligently
		1019	handle this. For instance, if the input contains 1 channel
		1020	and the output contains 2 channels, then it will create the
		1021	missing channel itself. The ''direction'' is a value from
		1022	-1.0 to 1.0. -1.0 represents far left and 1.0 represents far
		1023	right. Numbers in between will start the pan effect without
		1024	totally muting the opposite channel.
		1025
		1026
		1027	phaser ''gain-in gain-out delay decay speed''
		1028	''
		1029
		1030
		1031	Add a phaser to a sound sample. Each triple
		1032	delay/decay/speed gives the delay in milliseconds and the
		1033	decay (relative to gain-in) with a modulation speed in Hz.
		1034	The modulation is either sinodial (-s) or triangular (-t).
		1035	The decay should be less than 0.5 to avoid feedback.
		1036	Gain-out is the volume of the output.
		1037
		1038
		1039	pick [[ ''-1'' | ''-2'' | ''-3'' | ''-4'' |
		1040	''-l'' | ''-r'' ]
		1041
		1042
		1043	Select the left or right channel of a stereo sample, or one
		1044	of four channels in a quadraphonic sample. The ''-l'' and
		1045	''-r'' options represent either the left or right
		1046	channel. It is required that you use the __-c 1__ command
		1047	line option in order to force the output file to contain
		1048	only 1 channel.
		1049
		1050
		1051	pitch ''shift [[ width interpole fade ]''
		1052
		1053
		1054	Change the pitch of file without affecting its duration by
		1055	cross-fading shifted samples. ''shift'' is given in
		1056	cents. Use a positive value to shift to treble, negative
		1057	value to shift to bass. Default shift is 0. ''width'' of
		1058	window is in ms. Default width is 20ms. Try 30ms to lower
		1059	pitch, and 10ms to raise pitch. ''interpole'' option, can
		1060	be
		1061	''fade'' option, can be
		1062	''
		1063
		1064
		1065	polyphase [[ ''-w'' ''nut'' / ''ham''
		1066	''
		1067
		1068
		1069	[[ ''-width'' ''long'' / ''short'' / ''#''
		1070	''
		1071
		1072
		1073	[[ ''-cutoff #'' ]
		1074
		1075
		1076	Translate input sampling rate to output sampling rate via
		1077	polyphase interpolation, a DSP algorithm. This method is
		1078	slow and uses lots of RAM, but gives much better results
		1079	than __rate.__
		1080
		1081
		1082	-w
		1083	nut.''
		1084
		1085
		1086	-width long / short / # : specify the (approximate) width of
		1087	the filter. ''long'' is 1024 samples; ''short'' is 128
		1088	samples. Alternatively, an exact number can be used. Default
		1089	is ''long.'' The ''short'' option is __not__
		1090	recommended, as it produces poor quality
		1091	results.
		1092
		1093
		1094	-cutoff # : specify the filter cutoff frequency in terms of
		1095	fraction of frequency bandwidth, also know as the Nyquist
		1096	frequency. Please see the ''resample'' effect for further
		1097	information on Nyquist frequency. If upsampling, then this
		1098	is the fraction of the original signal that should go
		1099	through. If downsampling, this is the fraction of the signal
		1100	left after downsampling. Default is 0.95. Remember that this
		1101	is a float.
		1102
		1103
		1104	rate
		1105
		1106
		1107	Translate input sampling rate to output sampling rate via
		1108	linear interpolation to the Least Common Multiple of the two
		1109	sampling rates. This is the default effect if the two files
		1110	have different sampling rates and the preview options was
		1111	specified. This is fast but noisy: the spectrum of the
		1112	original sound will be shifted upwards and duplicated
		1113	faintly when up-translating by a multiple.
		1114
		1115
		1116	Lerp-ing is acceptable for cheap 8-bit sound hardware, but
		1117	for CD-quality sound you should instead use either
		1118	__resample__ or __polyphase.__ If you are wondering
		1119	which rate changing effects to use, you will want to read a
		1120	detailed analysis of all of them at
		1121	http://eakaw2.et.tu-dresden.de/~wilde/resample/resample.html
		1122
		1123
		1124	resample [[ ''-qs'' __|__ ''-q'' __|__ ''-ql''
		1125	__] [[__ ''rolloff'' __[[__ ''beta'' __]
		1126	]__
		1127
		1128
		1129	Translate input sampling rate to output sampling rate via
		1130	simulated analog filtration. This method is slower than
		1131	__rate,__ but gives much better results.
		1132
		1133
		1134	By default, linear interpolation is used, with a window
		1135	width about 45 samples at the lower of the two rate. This
		1136	gives an accuracy of about 16 bits, but insufficient
		1137	stopband rejection in the case that you want to have rolloff
		1138	greater than about 0.80 of the Nyquist
		1139	frequency.
		1140
		1141
		1142	The ''-q*'' options will change the default values for
		1143	rolloff and beta as well as use quadratic interpolation of
		1144	filter coefficients, resulting in about 24 bits precision.
		1145	The ''-qs'', ''-q'', or ''-ql'' options specify
		1146	increased accuracy at the cost of lower execution speed. It
		1147	is optional to specify rolloff and beta parameters when
		1148	using the ''-q*'' options.
		1149
		1150
		1151	Following is a table of the reasonable defaults which are
		1152	built-in to SoX:
		1153
		1154
		1155	__Option Window rolloff beta interpolation
		1156	------ ------ ------- ---- -------------__
		1157	(none) 45 0.80 16 linear
		1158	-qs 45 0.80 16 quadratic
		1159	-q 75 0.875 16 quadratic
		1160	-ql 149 0.94 16 quadratic__
		1161	------ ------ ------- ---- -------------__
		1162
		1163
		1164	''-qs'', ''-q'', or ''-ql'' use window lengths of
		1165	45, 75, or 149 samples, respectively, at the lower
		1166	sample-rate of the two files. This means progressively
		1167	sharper stop-band rejection, at proportionally slower
		1168	execution times.
		1169
		1170
		1171	''rolloff'' refers to the cut-off frequency of the low
		1172	pass filter and is given in terms of the Nyquist frequency
		1173	for the lower sample rate. rolloff therefore should be
		1174	something between 0.0 and 1.0, in practice 0.8-0.95. The
		1175	defaults are indicated above.
		1176
		1177
		1178	The ''Nyquist frequency'' is equal to (sample rate / 2).
		1179	Logically, this is because the A/D converter needs at least
		1180	2 samples to detect 1 cycle at the Nyquist frequency.
		1181	Frequencies higher then the Nyquist will actually appear as
		1182	lower frequencies to the A/D converter and is called
		1183	aliasing. Normally, A/D converts run the signal through a
		1184	highpass filter first to avoid these problems.
		1185
		1186
		1187	Similar problems will happen in software when reducing the
		1188	sample rate of an audio file (frequencies above the new
		1189	Nyquist frequency can be aliased to lower frequencies).
		1190	Therefore, a good resample effect will remove all frequency
		1191	information above the new Nyquist frequency.
		1192
		1193
		1194	The ''rolloff'' refers to how close to the Nyquist
		1195	frequency this cutoff is, with closer being better. When
		1196	increasing the sample rate of an audio file you would not
		1197	expect to have any frequencies exist that are past the
		1198	original Nyquist frequency. Because of resampling
		1199	properties, it is common to have alaising data created that
		1200	is above the old Nyquist frequency. In that case the
		1201	''rolloff'' refers to how close to the original Nyquist
		1202	frequency to use a highpass filter to remove this false
		1203	data, with closer also being better.
		1204
		1205
		1206	The ''beta'' parameter determines the type of filter
		1207	window used. Any value greater than 2.0 is the beta for a
		1208	Kaiser window. Beta
		1209	''
		1210
		1211
		1212	In the case of Kaiser window (beta
		1213
		1214
		1215	This is the default effect if the two files have different
		1216	sampling rates. Default parameters are, as indicated above,
		1217	Kaiser window of length 45, rolloff 0.80, beta 16, linear
		1218	interpolation.
		1219
		1220
		1221	__NOTE:__ ''-qs'' is only slightly slower, but more
		1222	accurate for 16-bit or higher precision.
		1223
		1224
		1225	__NOTE:__ In many cases of up-sampling, no interpolation
		1226	is needed, as exact filter coefficients can be computed in a
		1227	reasonable amount of space. To be precise, this is done
		1228	when
		1229
		1230
		1231	input_rate
		1232	output_rate/gcd(input_rate,output_rate)
		1233
		1234
		1235	reverb ''gain-out delay'' [[ ''delay ...''
		1236	]
		1237
		1238
		1239	Add reverberation to a sound sample. Each delay is given in
		1240	milliseconds and its feedback is depending on the
		1241	reverb-time in milliseconds. Each delay should be in the
		1242	range of half to quarter of reverb-time to get a realistic
		1243	reverberation. Gain-out is the volume of the
		1244	output.
		1245
		1246
		1247	reverse
		1248
		1249
		1250	Reverse the sound sample completely. Included for finding
		1251	Satanic subliminals.
		1252
		1253
		1254	__silence__ ''above_periods'' [[ ''duration
		1255	threshold''[[ ''d'' | ''%'' ]
		1256
		1257
		1258	[[ ''below_periods duration''
		1259
		1260
		1261	threshold[[ ''d'' | ''%'' ]]
		1262
		1263
		1264	Removes silence from the beginning or end of a sound file.
		1265	Silence is anything below a specified threshold.
		1266	When trimming silence from the beginning of a sound file,
		1267	you specify a duration of audio that is above a given
		1268	silence threshold before audio data is processed. You can
		1269	also specify the count of periods of none silence you want
		1270	to detect before processing audio data. Specify a period of
		1271	0 if you do not want to trim data from the front of the
		1272	sound file.
		1273	When optionally trimming silence form the end of a sound
		1274	file, you specify the duration of audio that must be below a
		1275	given threshold before stopping to process audio data. A
		1276	count of periods that occur below the threshold may also be
		1277	specified. If this options are not specified then data is
		1278	not trimmed from the end of the audio file.
		1279	Duration counts may be in the format of time, hh:mm:ss.frac,
		1280	or in the exact count of samples.
		1281	Threshold may be suffixed with d, or % to indicated the
		1282	value is in decibels or a percentage of max value of the
		1283	sample value. A value of '0%' will look for total
		1284	silence.
		1285
		1286
		1287	speed [[ -c ] ''factor''
		1288
		1289
		1290	Speed up or down the sound, as a magnetic tape with a speed
		1291	control. It affects both pitch and time. A factor of 1.0
		1292	means no change, and is the default. 2.0 doubles speed, thus
		1293	time length is cut by a half and pitch is one octave higher.
		1294	0.5 halves speed thus time length doubles and pitch is one
		1295	octave lower. If the optional -c parameter is used then the
		1296	factor is specified in
		1297
		1298
		1299	split
		1300
		1301
		1302	Turn a mono sample into a stereo sample by copying the input
		1303	channel to the left and right channels.
		1304
		1305
		1306	stat [[ ''-s n'' __] [[__''-rms'' __] [[__
		1307	''-v'' __] [[__ ''-d'' __]__
		1308
		1309
		1310	Do a statistical check on the input file, and print results
		1311	on the standard error file. Audio data is passed unmodified
		1312	from input to output file unless used along with the
		1313	__-e__ option.
		1314
		1315
		1316	The
		1317	-v__ ''number'' which
		1318	will make the sample as loud as possible without
		1319	clipping.
		1320
		1321
		1322	The option __-v__ will print out the
		1323	__
		1324
		1325
		1326	The __-s n__ option is used to scale the input data by a
		1327	given factor. The default value of n is the max value of a
		1328	signed long variable (0x7fffffff). Internal effects always
		1329	work with signed long PCM data and so the value should
		1330	relate to this fact.
		1331
		1332
		1333	The __-rms__ option will convert all output average
		1334	values to ''root mean square'' format.
		1335
		1336
		1337	There is also an optional parameter __-d__ that will
		1338	print out a hex dump of the sound file from the internal
		1339	buffer that is in 32-bit signed PCM data. This is mainly
		1340	only of use in tracking down endian problems that creep in
		1341	to SoX on cross-platform versions.
		1342
		1343
		1344	stretch ''factor [[window fade shift
		1345	fading]''
		1346
		1347
		1348	Time stretch file by a given factor. Change duration without
		1349	affecting the pitch. ''factor'' of stretching:
		1350	''window'' size is in
		1351	ms. Default is 20ms. The ''fade'' option, can be
		1352	''shift'' ratio, in [[0.0 1.0]. Default
		1353	depends on stretch factor. 1.0 to shorten, 0.8 to lengthen.
		1354	The ''fading'' ratio, in [[0.0 0.5]. The amount of a
		1355	fade's default depends on factor and shift.
		1356
		1357
		1358	swap [[ ''1 2'' __|__ ''1 2 3 4''
		1359	__]__
		1360
		1361
		1362	Swap channels in multi-channel sound files. Optionally, you
		1363	may specify the channel order you would like the output in.
		1364	This defaults to output channel 2 and then 1 for stereo and
		1365	2, 1, 4, 3 for quad-channels. An interesting feature is that
		1366	you may duplicate a given channel by overwriting another.
		1367	This is done by repeating an output channel on the command
		1368	line. For example, swap 2 2 will overwrite channel 1 with
		1369	channel 2's data; creating a stereo file with both channels
		1370	containing the same audio data.
		1371
		1372
		1373	synth [[ ''length'' ] ''type mix'' [[ ''freq'' [[
		1374	''-freq2'' ]
		1375
		1376
		1377	[[ ''off'' ] [[ ''ph'' ] [[ ''p1'' ] [[ ''p2'' ] [[
		1378	''p3'' ]
		1379
		1380
		1381	The synth effect will generate various types of audio data.
		1382	Although this effect is used to generate audio data, an
		1383	input file must be specified. The length of the input audio
		1384	file determines the length of the output audio file.
		1385
		1386
		1387	trim ''start'' [[ ''length'' ]
		1388
		1389
		1390	Trim can trim off unwanted audio data from the beginning and
		1391	end of the audio file. Audio samples are not sent to the
		1392	output stream until the ''start'' location is
		1393	reached.
		1394	The optional ''length'' parameter tells the number of
		1395	samples to output after the ''start'' sample and is used
		1396	to trim off the back side of the audio data. Using a value
		1397	of 0 for the ''start'' parameter will allow trimming off
		1398	the back side only.
		1399	Both options can be specified using either an amount of time
		1400	and an exact count of samples. The format for specifying
		1401	lengths in time is hh:mm:ss.frac. A start value of 1:30.5
		1402	will not start until 1 minute, thirty and 1/2 seconds into
		1403	the audio data. The format for specifying sample counts is
		1404	the number of samples with the letter 's' appended to it. A
		1405	value of 8000s will wait until 8000 samples are read before
		1406	starting to process audio data.
		1407
		1408
		1409	vibro ''speed'' __[[__ ''depth''
		1410	__]__
		1411
		1412
		1413	Add the world-famous Fender Vibro-Champ sound effect to a
		1414	sound sample by using a sine wave as the volume knob.
		1415	__Speed__ gives the Hertz value of the wave. This must be
		1416	under 30. __Depth__ gives the amount the volume is cut
		1417	into by the sine wave, ranging 0.0 to 1.0 and defaulting to
		1418	0.5.
		1419
		1420
		1421	vol ''gain'' [[ ''type'' __[[__ ''limitergain'' ]
		1422	]
		1423
		1424
		1425	The vol effect is much like the command line option -v. It
		1426	allows you to adjust the volume of an input file and allows
		1427	you to specify the adjustment in relation to amplitude,
		1428	power, or dB. If ''type'' is not specified then it
		1429	defaults to ''amplitude''.
		1430	When type is ''amplitude'' then a linear change of the
		1431	amplitude is performed based on the gain. Therefore, a value
		1432	of 1.0 will keep the volume the same, 0.0 to
		1433	''
		1434	When type is ''power'' then a value of 1.0 also means no
		1435	change in volume.
		1436	When type is ''dB'' the amplitude is changed
		1437	logarithmically. 0.0 is constant while +6 doubles the
		1438	amplitude.
		1439	An optional ''limitergain'' value can be specified and
		1440	should be a value much less then 1.0 (ie 0.05 or 0.02) and
		1441	is used only on peaks to prevent clipping. Not specifying
		1442	this parameter will cause no limiter to be used. In verbose
		1443	mode, this effect will display the percentage of audio data
		1444	that needed to be limited.
		1445	!!BUGS
		1446
		1447
		1448	The syntax is horrific. Thats the breaks when trying to
		1449	handle all things from the command line.
		1450
		1451
		1452	Please report any bugs found in this version of SoX to Chris
		1453	Bagwell (cbagwell@sprynet.com)
		1454	!!FILES
		1455	!!SEE ALSO
		1456
		1457
		1458	play(1), rec(1),
		1459	__soxexam(1)__
		1460	!!NOTICES
		1461
		1462
		1463	The version of SoX that accompanies this manual page is
		1464	support by Chris Bagwell (cbagwell@users.sourceforge.net).
		1465	Please refer any questions regarding it to this address. You
		1466	may obtain the latest version at the the web site
		1467	http://sox.sourceforge.net/
		1468	!!AUTHOR
		1469
		1470
		1471	Chris Bagwell (cbagwell@users.sourceforge.net).
		1472
		1473
		1474	Updates by Anonymous
		1475	----