shithub: sox

ref: 2010240ce3c22f65d1379ec785f754253e9759b9
dir: /sox.txt/

View raw version
SoX(1)							   SoX(1)



NAME
       sox - Sound eXchange : universal sound sample translator

SYNOPSIS
       sox infile outfile

       sox [ general options ] [ format options ] infile
	   -e effect [ effect options ]

       sox [ general options ] [ format options ] infile
	   [ format options ] outfile
	   [ effect [ effect options ] ... ]

       General options:
	   [ -h ] [ -p ] [ -v volume ] [ -V ]

       Format options:
	   [ -t filetype ] [ -r rate ] [ -s/-u/-U/-A/-a/-i/-g ]
	   [ -b/-w/-l/-f/-d/-D ]
	   [ -c channels ] [ -x ] [ -e ]

       Effects:
	   avg [ -l | -r | -f | -b | n,n,...,n ]
	   band [ -n ] center [ width ]
	   bandpass frequency bandwidth
	   bandreject frequency bandwidth
	   chorus gain-in gain out delay decay speed depth
		  -s | -t [ delay decay speed depth -s | -t ]
	   compand attack1,decay1[,attack2,decay2...]
		   in-dB1,out-dB1[,in-dB2,out-dB2...]
		   [ gain [ initial-volume [ delay ] ] ]
	   copy
	   cut
	   deemph
	   earwax
	   echo gain-in gain-out delay decay [ delay decay ... ]
	   echos gain-in gain-out delay decay [ delay decay ... ]
	   fade [ type ] fade-in-length
		[ stop-time [ fade-out-length ] ]
	   filter [ low ]-[ high ] [ window-len [ beta ]]
	   flanger gain-in gain-out delay decay speed < -s | -t >
	   highp frequency
	   highpass frequency
	   lowp frequency
	   lowpass frequency
	   map
	   mask
	   pan direction
	   phaser gain-in gain-out delay decay speed < -s | -t >
	   pick [ -1 | -2 | -3 | -4 | -l | -r ]
	   pitch shift [ width interpole fade ]
	   polyphase [ -w < nut / ham > ]
		     [	-width < long / short / # > ]
		     [ -cutoff # ]
	   rate
	   resample [ -qs | -q | -ql ] [ rolloff [ beta ] ]
	   reverb gain-out reverb-time delay [ delay ... ]
	   reverse
	   speed [ -c ] factor
	   split
	   stat [ -s n ] [ -rms ] [ -v ] [ -d ]
	   stretch [ factor [ window fade shift fading ]
	   swap [ 1 2 | 1 2 3 4 ]
	   trim start [ length ]
	   vibro speed [ depth ]
	   vol gain [ type [ limitergain ] ]

DESCRIPTION
       SoX  is a command line program that can convert most popu�
       lar 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.

       There  are  two	types of audio files formats that SoX can
       work with.  The first are  self-describing  file	 formats.
       These  contain a header that completely describe the char�
       acteristics of the audio data that follows.

       The second type are headerless data, or	sometimes  called
       raw  data.   A user must pass enough information to SoX on
       the command line so that it knows what  type  of	 data  it
       contains.

       Audio  data can usually be totally described by four char�
       acteristics:

       rate	 The sample rate is in samples per  second.   For
		 example, CD sample rates are at 44100.

       data size The precision the data is stored in.  Most popu�
		 lar are 8-bit bytes or 16-bit words.

       data encoding
		 What encoding the data type uses.  Examples  are
		 u-law, ADPCM, or signed linear data.

       channels	 How  many  channels  are  contained in the audio
		 data.	Mono and Stereo are the two most  common.

       Please  refer  to  the  soxexam(1)  manual page for a long
       description with examples on how to use sox  with  various
       types of file formats.

OPTIONS
       The option syntax is a little grotty, but in essence:

	    sox file.au file.wav

       translates  a  sound  file  in SUN Sparc .AU format into a
       Microsoft .WAV file, while

	    sox -v 0.5 file.au -r 12000 file.wav mask

       does the same  format  translation  but	also  lowers  the
       amplitude  by  1/2,  changes  the  sampling  rate to 12000
       hertz, and applies the mask  sound  effect  to  the  audio
       data.

       Format options:

       Format  options effect the audio samples that they immedi�
       ately preceed.  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 out�
       put 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.

       -t filetype
		 gives the type of the sound sample file.  Useful
		 when file extension is not standard or for spec�
		 ifying the .auto file type.

       -r rate	 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.
		 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.

       -s/-u/-U/-A/-a/-i/-g
		 The sample data encoding is signed  linear  (2's
		 complement),  unsigned	 linear, U-law (logarith�
		 mic), A-law (logarithmic), ADPCM, IMA_ADPCM,  or
		 GSM.
		 U-law	(actually shorthand for mu-law) and A-law
		 are the U.S.  and  international  standards  for
		 logarithmic  telephone	 sound compression.  When
		 uncompressed it has  roughly  the  precision  of
		 12-byte PCM audio.
		 ADPCM	is  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 -a flag	has  dif�
		 ferent	 meanings in different file handlers.  In
		 .wav 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.
		 GSM  is a standard used for telephone sound com�
		 pression in European countries and  its  gaining
		 popularity  because  of its quality.  It usually
		 is CPU intensive to work with GSM audio data.

       -b/-w/-l/-f/-d/-D
		 The sample data size is in bytes, 16-bit  words,
		 32-bit	 longwords,  32-bit floats, 64-bit double
		 floats, or 80-bit IEEE floats.	 Floats and  dou�
		 ble floats are in native machine format.

       -x	 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.
		 IEEE floats are a fixed, portable format.

       -c channels
		 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.

       -e	 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 spec�
		 ified effects to the input file.  This is mainly
		 useful with the stat effect but can be used with
		 others.

       General options:

       -h	 Print version number and usage information.

       -p	 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 rate effect instead of the
		 resample effect for sample rate changes.

       -v volume 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
		 percieve volume logarithmically but this adjusts
		 the amplitude linearly.
		 Note: see the stat  effect  for  information  on
		 finding  the maximum value that can be used with
		 this option without causing  audio  data  be  be
		 clipped.

       -V	 Print	a description of processing phases.  Use�
		 ful for figuring out exactly how sox is mangling
		 your sound samples.

FILE TYPES
       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.

       The  input  and	output files may be read from standard in
       and out.	 This is done by specifying '-' as the	filename.

       File  formats  which  have  headers  are	 checked, if that
       header doesn't seem  right,  the	 program  exits	 with  an
       appropriate message.

       The following file formats are supported:


       .8svx	 Amiga	8SVX  musical instrument description for�
		 mat.

       .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 descrip�
		 tion format.  AIFF files are multimedia archives
		 and  can have multiple audio and picture chunks.
		 You may need a separate archiver  to  work  with
		 them.

       .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  .ul  format
		 (see below).

       .avr	 Audio Visual Research
		 The  AVR  format is produced by a number of com�
		 mercial packages on the Mac.

       .cdr	 CD-R
		 CD-R files are used in mastering music	 on  Com�
		 pact  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 han�
		 dler.

       .cvs	 Continuously Variable Slope Delta modulation
		 Used to compress speech audio	for  applications
		 such as voice mail.

       .dat	 Text Data files
		 These	files contain a textual representation of
		 the sample data.   There  is  one  line  at  the
		 beginning that contains the sample rate.  Subse�
		 quent 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 analyz�
		 ers or graph routines.	 SoX can also  convert	a
		 file  in  this format back into one of the other
		 file formats.

       .gsm	 GSM 06.10 Lossy Speech Compression
		 A standard for compressing speech which is  used
		 in  the Global Standard for Mobil telecommunica�
		 tions (GSM).  Its good for its purpose,  shrink�
		 ing  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.
		 GSM in sox is optional and requires access to an
		 external GSM library.	To see if there	 is  sup�
		 port  for  gsm	 run sox -h and look for it under
		 the list of supported file formats.

       .hcom	 Macintosh HCOM files.	 These	are  (apparently)
		 Mac FSSD files with some variant of Huffman com�
		 pression.  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.

       .maud	 An Amiga format
		 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.

       .ogg	 Ogg Vorbis Compressed Audio.
		 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.  sox can decode all types of Ogg Vorbis
		 files, but can only encode at 128 kbps.   Decod�
		 ing  is  somewhat  CPU intensive and encoding is
		 very CPU intensive.
		 Ogg Vorbis  in	 sox  is  optional  and	 requires
		 access to external Ogg Vorbis libraries.  To see
		 if there is support for Ogg Vorbis  run  sox  -h
		 and look for it under the list of supported file
		 formats as "vorbis".

       ossdsp	 OSS /dev/dsp device driver
		 This is a pseudo-file type and can be optionally
		 compiled  into	 Sox.	Run  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.	 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: -t
		 ossdsp -w -s /dev/dsp

       .sf	 IRCAM Sound Files.
		 Sound Files are used by academic music	 software
		 such  as  the	CSound	package,  and the MixView
		 sound sample editor.

       .sph
		 SPHERE (SPeech HEader Resources) is a file  for�
		 mat defined by NIST (National Institute of Stan�
		 dards and Technology) and is  used  with  speech
		 audio.	  SoX can read these files when they con�
		 tain ulaw and PCM  data.   It	will  ignore  any
		 header	 information  that  says the data is com�
		 pressed using shorten compression and will treat
		 the data as either ulaw or PCM.  This will allow
		 SoX and the command line shorten program  to  be
		 ran  together using pipes to uncompress the data
		 and then pass the result to SoX for  processing.

       .smp	 Turtle Beach SampleVision files.
		 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.

       .snd
		 Under	DOS  this  file format is the same as the
		 .sndt format.	Under all other platforms  it  is
		 the same as the .au format.

       .sndt	 SoundTool files.
		 This is an older DOS file format.

       sunau	 Sun /dev/audio device driver
		 This is a pseudo-file type and can be optionally
		 compiled into Sox.  Run 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  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: -t sunau  -w
		 -s /dev/audio or -t sunau -U -c 1 /dev/audio for
		 older sun equipment.

       .txw	 Yamaha TX-16W sampler.
		 A file format from a  Yamaha  sampling	 keyboard
		 which	wrote  IBM-PC format 3.5" floppies.  Han�
		 dles reading of files which do not have the sam�
		 ple  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.

       .vms	 More info to come.
		 Used to compress speech audio	for  applications
		 such as voice mail.

       .voc	 Sound Blaster VOC files.
		 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  gener�
		 ated  appropriately.	On output, silence is not
		 detected, nor are impossible sample rates.

       vorbis	 See .ogg format.

       .wav	 Microsoft .WAV RIFF files.
		 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 .wav files have all for�
		 matting 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 con�
		 version, and the  .wav	 will  written	appropri�
		 ately.	  Sox currently can read PCM, ULAW, ALAW,
		 MS ADPCM, and IMA (or DVI) ADPCM.  It can  write
		 all of these formats including (NEW!)	the ADPCM
		 encoding.

       .wve	 Psion 8-bit alaw
		 These are 8-bit a-law 8khz sound files	 used  on
		 the Psion palmtop portable computer.

       .raw	 Raw files (no header).
		 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.

       .ub, .sb, .uw, .sw, .ul, .al, .sl
		 These are several  suffices  which  serve  as	a
		 shorthand  for	 raw  files with a given size and
		 encoding.  Thus, ub, sb, uw, sw, ul and sl  cor�
		 respond   to  "unsigned  byte",  "signed  byte",
		 "unsigned word", "signed word",  "ulaw"  (byte),
		 "alaw"	 (byte),  and  "signed long".  The sample
		 rate defaults to 8000 hz if not explicitly  set,
		 and  the number of channels (as always) 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 head�
		 ers.)	Similarly, most Mac sound  files  are  in
		 unsigned byte format with a sample rate of 11025
		 or 22050 hz.

       .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  mes�
		 sage.	 The  input  must  be a plain file, not a
		 pipe.	This type can't be used for output files.

EFFECTS
       Multiple effects may be applied to the audio data by spec�
       ifying them one after another at the end	 of  the  command
       line.

       avg [ -l | -r | -f | -b | n,n,...,n ]
		 Reduce	 the  number of channels by averaging the
		 samples, or duplicate channels to  increase  the
		 number	 of  channels.	 This effect is automati�
		 cally 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
		 -l, -r, -f, or -b options  to	select	only  the
		 left,	right,	front, or back channel(s) for the
		 output instead of averaging the  channels.   The
		 -f  and  -b  options  maintain left/right stereo
		 separation; use the avg effect twice to select a
		 single channel.

		 The avg effect can also be invoked with up to 16
		 double-precision numbers, which specify the pro�
		 portion  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, if fewer than 4
		 numbers are given, certain special abbreviations
		 may be invoked; see the source code for details.

       band [ -n ] center [ width ]
		 Apply	a  band-pass   filter.	  The	frequency
		 response drops logarithmically around the center
		 frequency.  The width gives  the  slope  of  the
		 drop.	 The  frequencies  at  center + width and
		 center - width will be half  of  their	 original
		 amplitudes.  Band defaults to a mode oriented to
		 pitched signals, i.e. voice, singing, or instru�
		 mental	 music.	  The  -n (for noise) option uses
		 the  alternate	 mode  for  un-pitched	 signals.
		 Warning:  -n  introduces  a  power-gain of about
		 11dB in the filter, so beware	of  output  clip�
		 ping.	Band introduces noise in the shape of the
		 filter, i.e. peaking at the center frequency and
		 settling  around  it.	See filter for a bandpass
		 effect with steeper shoulders.

       bandpass frequency bandwidth
		 Butterworth bandpass filter. Description  coming
		 soon!

       bandreject frequency bandwidth
		 Butterworth bandreject filter.	 Description com�
		 ing soon!

       chorus gain-in gain-out delay decay speed depth

	      -s | -t [ delay decay speed depth -s | -t ... ]
		 Add a chorus to a sound sample.  Each	quadtuple
		 delay/decay/speed/depth  gives the delay in mil�
		 liseconds and the decay  (relative  to	 gain-in)
		 with  a  modulation  speed  in Hz using depth in
		 milliseconds.	The modulation is either sinodial
		 (-s) or triangular (-t).  Gain-out is the volume
		 of the output.

       compand attack1,decay1[,attack2,decay2...]

	       in-dB1,out-dB1[,in-dB2,out-dB2...]

	       [gain [initial-volume [delay ] ] ]
		 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 sepa�
		 rately and the number of pairs must  agree  with
		 the number of input channels.	The second param�
		 eter 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 -inf
		 may be used to indicate that  the  input  volume
		 should	 be associated output volume.  The points
		 -inf,-inf and 0,0 are assumed; the latter may be
		 overridden, but the former may not.

		 The third (optional) parameter is a postprocess�
		 ing gain in dB which is applied after	the  com�
		 pression  has taken place; the fourth (optional)
		 parameter is an initial volume to be assumed for
		 each  channel when the effect starts.	This per�
		 mits the user to supply  a  nominal  level  ini�
		 tially,  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 com�
		 pander gain properly adjusts itself.

		 The fifth (optional) parameter	 is  a	delay  in
		 seconds.   The	 input signal is analyzed immedi�
		 ately	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.

       copy	 Copy the input file to the output file.  This is
		 the default effect if both files have	the  same
		 sampling rate.

       cut loopnumber
		 Extract loop #N from a sample.

       deemph	 Apply	a  treble  attenuation shelving filter to
		 samples  in  audio  cd	 format.   The	frequency
		 response  of pre-emphasized recordings is recti�
		 fied.	The filtering is defined in the	 standard
		 document ISO 908.

       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
		 www.geocities.com/beinges for	a  full	 explana�
		 tion.

       echo gain-in gain-out delay decay [ delay decay ... ]
		 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.

       echos gain-in gain-out delay decay [ delay decay ... ]
		 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.

       fade [ type ] fade-in-length

	    [ stop-time [ fade-out-length ] ]
		 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 fade-in-length 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 fade-out-
		 length seconds before the stop-time.	No  fade-
		 out is performed if these options are not speci�
		 fied.	All times can be  specified  in	 seconds,
		 mm:ss.frac, or hh:mm:ss.frac format.

		 An  optional type 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.

       filter [ low ]-[ high ] [ window-len [ beta ] ]
		 Apply	a  Sinc-windowed  lowpass,  highpass,  or
		 bandpass filter of given window  length  to  the
		 signal.   low	refers	to  the	 frequency of the
		 lower 6dB corner of the filter.  high refers  to
		 the  frequency	 of  the  upper 6dB corner of the
		 filter.

		 A lowpass filter  is  obtained	 by  leaving  low
		 unspecified,	or   0.	  A  highpass  filter  is
		 obtained by leaving high unspecified, or  0,  or
		 greater  than or equal to the Nyquist frequency.

		 The window-len, if unspecified, defaults to 128.
		 Longer	 windows  give	a sharper cutoff, smaller
		 windows a more gradual cutoff.

		 The beta, if unspecified, defaults to 16.   This
		 selects  a Kaiser window.  You can select a Nut�
		 tall window by specifying anything <= 2.0  here.
		 For  more  discussion	of  beta,  look under the
		 resample effect.


       flanger gain-in gain-out delay decay speed < -s | -t >
		 Add a flanger to a sound  sample.   Each  triple
		 delay/decay/speed  gives  the delay in millisec�
		 onds 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.

       highp frequency
		 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 filter
		 for a highpass effect with sharper cutoff.

       highpass frequency
		 Butterworth  highpass	filter.	 Description com�
		 ming soon!

       lowp frequency
		 Apply a single pole recursive	low-pass  filter.
		 The  frequency	 response  drops  logarithmically
		 with frequency in the middle of the  drop.   The
		 slope of the filter is quite gentle.  See filter
		 for a lowpass effect with sharper cutoff.

       lowpass frequency
		 Butterworth lowpass filter.  Description  coming
		 soon!

       map	 Display a list of loops in a sample, and miscel�
		 laneous loop info.

       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.

       pan direction
		 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 con�
		 tains 2 channels, then it will create the  miss�
		 ing  channel  itself.	 The 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.

       phaser gain-in gain-out delay decay speed < -s | -t >
		 Add  a	 phaser	 to  a sound sample.  Each triple
		 delay/decay/speed gives the delay  in	millisec�
		 onds  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.

       pick [ -1 | -2 | -3 | -4 | -l | -r ]
		 Select	 the  left  or	right channel of a stereo
		 sample, or one of four	 channels  in  a  quadro�
		 phonic	 sample.  The -l and -r options represent
		 either	 the  left  or	right  channel.	  It   is
		 required  that	 you  use  the	-c 1 command line
		 option in order to force the output file to con�
		 tain only 1 channel.

       pitch shift [ width interpole fade ]
		 Change	 the  pitch of file without affecting its
		 duration by cross-fading shifted samples.  shift
		 is given in cents. Use a positive value to shift
		 to treble, negative  value  to	 shift	to  bass.
		 Default  shift	 is 0.	width of window is in ms.
		 Default width is 20ms. Try 30ms to lower  pitch,
		 and  10ms to raise pitch.  interpole option, can
		 be "cubic" or "linear". Default is "cubic".  The
		 fade  option,	can be "cos", "hamming", "linear"
		 or "trapezoid".  Default is "cos".

       polyphase [ -w < nut / ham > ]

		 [  -width <  long  / short  / # > ]

		 [ -cutoff #  ]
		 Translate input sampling rate to output sampling
		 rate  via  polyphase  interpolation, a DSP algo�
		 rithm.	 This method is slow  and  uses	 lots  of
		 RAM, but gives much better results than rate.

		 -w  <	nut / ham > : select either a Nuttal (~90
		 dB stopband) or Hamming (~43 dB  stopband)  win�
		 dow.  Default is nut.

		 -width	 long / short / # : specify the (approxi�
		 mate) width of the filter.  long  is  1024  sam�
		 ples;	short  is 128 samples.	Alternatively, an
		 exact number can be used.  Default is long.  The
		 short	option is not recommended, as it produces
		 poor quality results.

		 -cutoff # : specify the filter cutoff	frequency
		 in  terms  of	fraction  of frequency bandwidth,
		 also know as the Nyquist frequency.  Please  see
		 the  resample	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  frac�
		 tion  of  the	signal	left  after downsampling.
		 Default is 0.95.  Remember that this is a float.


       rate	 Translate input sampling rate to output sampling
		 rate via linear interpolation to the Least  Com�
		 mon Multiple of the two sampling rates.  This is
		 the default effect if the two files have differ�
		 ent  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  resample or 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-dres�
		 den.de/~wilde/resample/resample.html

       resample [ -qs | -q | -ql ] [ rolloff [ beta ] ]
		 Translate input sampling rate to output sampling
		 rate  via  simulated  analog  filtration.   This
		 method	 is slower than rate, but gives much bet�
		 ter 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  -q*  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 -qs, -q, or -ql
		 options  specify  increased accuracy at the cost
		 of lower execution speed.   It	 is  optional  to
		 specify  rolloff  and beta parameters when using
		 the -q* options.

		 Following is a table of the reasonable	 defaults
		 which are built-in to sox:

		    Option  Window rolloff beta interpolation
		    ------  ------ ------- ---- -------------
		    (none)    45    0.80    16	   linear
		      -qs     45    0.80    16	  quadratic
		      -q      75    0.875   16	  quadratic
		      -ql    149    0.94    16	  quadratic
		    ------  ------ ------- ---- -------------

		 -qs, -q, or -ql 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.

		 rolloff  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 Nyquist frequency is equal to (sample rate /
		 2).   Logically,  this	 is  because the A/D con�
		 verter 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 (fre�
		 quencies 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  rolloff  refers to how close to the Nyquist
		 frequency this cutoff is, with closer being bet�
		 ter.	When  increasing  the  sample  rate of an
		 audio file you would not expect to have any fre�
		 quencies   exist  that	 are  past  the	 original
		 Nyquist frequency.  Because of resampling  prop�
		 erties,  it is common to have alaising data cre�
		 ated that is above the	 old  Nyquist  frequency.
		 In  that case the rolloff 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 beta 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.

		 NOTE: -qs is  only  slightly  slower,	but  more
		 accurate for 16-bit or higher precision.

		 NOTE:	In many cases of up-sampling, no interpo�
		 lation is needed, as exact  filter  coefficients
		 can be computed in a reasonable amount of space.
		 To be precise, this is done when

			    input_rate < output_rate
				       &&
		   output_rate/gcd(input_rate,output_rate) <= 511

       reverb gain-out delay [ delay ... ]
		 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 rever�
		 beration.  Gain-out is the volume of the output.

       reverse	 Reverse  the  sound sample completely.	 Included
		 for finding Satanic subliminals.

       speed [ -c ] factor
		 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 dou�
		 bles and pitch is  one	 octave	 lower.	  If  the
		 optional -c parameter is used then the factor is
		 specified in "cents".

       split	 Turn a mono sample into a stereo sample by copy�
		 ing  the  input  channel  to  the left and right
		 channels.

       stat [ -s n ] [-rms ] [ -v ] [ -d ]
		 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 -e option.

		 The "Volume Adjustment:" field in the statistics
		 gives you the argument to the	-v  number  which
		 will make the sample as loud as possible without
		 clipping.

		 The option -v will print out the "Volume Adjust�
		 ment:"	 field's  value	 only  and  return.  This
		 could be of use in scripts to auto  convert  the
		 volume.

		 The  -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 -rms option will convert all output  average
		 values to root mean square format.

		 There is also an optional parameter -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.


       stretch factor [window fade shift fading]
		 Time stretch file  by	a  given  factor.  Change
		 duration without affecting the pitch.	factor of
		 stretching: >1.0 lengthen,  <1.0  shorten  dura�
		 tion.	 window	 size  is in ms. Default is 20ms.
		 The fade option, can be "lin".	 shift ratio,  in
		 [0.0  1.0].  Default  depends on stretch factor.
		 1.0 to shorten, 0.8  to  lengthen.   The  fading
		 ratio,	 in  [0.0  0.5].  The  amount of a fade's
		 default depends on factor and shift.

       swap [ 1 2 | 1 2 3 4 ]
		 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.

       trim start [ length ]
		 Trim can trim off unwanted audio data	from  the
		 beginning and end of the audio file.  Audio sam�
		 ples are not sent to the output stream until the
		 start	location is reached.  start is a floating
		 point number that tells the number of seconds to
		 wait  before  starting.   If you know the sample
		 number you would like to start at then the  sec�
		 onds  can be obtained by multiplying (sample # *
		 sample rate).
		 The optional length parameter tells  the  number
		 of  samples to output after the start sample and
		 is used to trim off the back side of  the  audio
		 data.	 Using a value of 0 for the start parame�
		 ter will allow trimming off the back side  only.
		 Both  start  and length can also be specified in
		 mm:ss.frac or hh:mm:ss.frac format.

       vibro speed  [ depth ]
		 Add the world-famous  Fender  Vibro-Champ  sound
		 effect to a sound sample by using a sine wave as
		 the volume knob.  Speed gives the Hertz value of
		 the  wave.   This must be under 30.  Depth gives
		 the amount the volume is cut into  by	the  sine
		 wave,	ranging 0.0 to 1.0 and defaulting to 0.5.

       vol gain [ type [ limitergain ] ]
		 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 type is not specified then  it	 defaults
		 to amplitude.
		 When  type  is 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 vol�
		 ume 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.
		 When  type  is	 power	then  a value of 1.0 also
		 means no change in volume.
		 When type is dB the amplitude is  changed  loga�
		 rithmically.	0.0  is constant while +6 doubles
		 the amplitude.
		 An optional limitergain 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  pre�
		 vent  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.

BUGS
       The syntax is horrific.	Thats the breaks when  trying  to
       handle all things from the command line.

       Please  report  any  bugs  found in this version of sox to
       Chris Bagwell (cbagwell@sprynet.com)

FILES
SEE ALSO
       play(1), rec(1), soxexam(1)

NOTICES
       The version of Sox that accompanies this	 manual	 page  is
       support	by  Chris Bagwell (cbagwell@sprynet.com).  Please
       refer any questions regarding it to this address.  You may
       obtain	the   latest   version	 at   the  the	web  site
       http://home.sprynet.com/~cbagwell/sox.html



			  July 24, 2000			   SoX(1)