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SoX(1)							   SoX(1)


NAME
       sox - Sound eXchange : universal sound sample translator

SYNOPSIS
       sox infile outfile
       sox infile outfile [ effect [ effect options ... ] ]
       sox infile -e effect [ effect options ... ]
       sox [ general options  ] [ format options  ] infile [ for-
       mat options  ] outfile [ effect [ effect options ... ] ]

       General options: [ -e ] [ -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 ]

       Effects:
	    avg [ -l | -r ]
	    band [ -n ] center [ width ]
	    bandpass frequency bandwidth
	    bandreject frequency bandwidth
	    check
	    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]
	    copy
	    cut
	    deemph
	    echo gain-in gain-out delay decay [ delay decay  ...]
	    echos gain-in gain-out delay decay [ delay decay ...]
	    filter [ low ]-[ high ] [ window-len [ beta ]]
	    flanger gain-in gain-out delay decay speed -s | -t
	    highp center
	    highpass frequency
	    lowp center
	    lowpass frequency
	    map
	    mask
	    pan direction
	    phaser gain-in gain-out delay decay speed -s | -t
	    pick
	    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 factor
	    split
	    stat [ -s n ] [-rms ] [ -v ] [ -d ]



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SoX(1)							   SoX(1)


	    stretch [ factor [ window fade shift fading ]
	    swap [ 1 2 | 1 2 3 4 ]
	    vibro speed [ depth ]
	    vol gain [ type ]

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 apply a sound effect 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 type What format the data is stored in.  Most popular
		 are 8-bit or 16-bit words.

       data format
		 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.voc

       translates  a  sound  file  in SUN Sparc .AU format into a
       SoundBlaster .VOC file, while

	    sox -v 0.5 file.au -r 12000 file.voc rate

       does the same  format  translation  but	also  lowers  the
       amplitude  by  1/2 and changes the sampling rate from 8000
       hertz to 12000 hertz via the rate sound effect loop.



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SoX(1)							   SoX(1)


       Format options:

       Format options effect the audio samples that they  immedi-
       ately  precede.	 If they are placed before the input file
       name then they effect the input data.  If they are  placed
       before the output file name then they will effect the 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.

       -t filetype
		 gives the type of the sound sample file.

       -r rate	 Give sample rate in Hertz of file.  To cause the
		 output file to have a different sample rate than
		 the input file, include  this	option	with  the
		 appropriate  rate  value  along  with 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 be used
		 with its default parameters.

       -s/-u/-U/-A/-a/-i/-g
		 The sample data format	 is  signed  linear  (2's
		 complement),  unsigned	 linear, U-law (logarith-
		 mic), A-law (logarithmic), ADPCM, IMA_ADPCM,  or
		 GSM.	U-law and A-law are the U.S. and interna-
		 tional standards for logarithmic telephone sound
		 compression.  ADPCM is form of sound compression
		 that has a good compromise  between  good  sound
		 quality   and	 fast	encoding/decoding   time.
		 IMA_ADPCM is also a 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	compression  in	 European
		 countries  and its gaining popularity because of
		 its quality.

       -b/-w/-l/-f/-d/-D
		 The sample data type 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.




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SoX(1)							   SoX(1)


       -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 appro-
		 priate value with the output file  options.   If
		 the  input and output file have a different num-
		 ber of channels then  the  avg	 effect	 must  be
		 used.	If the avg effect is not specified on the
		 command line it will  be  invoked  with  default
		 parameters.

       General options:

       -e	 When  used  after  the	 input	file  (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.

       -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.  The	 order	that  the
		 effects  are run in will be arranged for maximum
		 speed and not quality.

       -v volume Change amplitude (floating point); less than 1.0
		 decreases, greater than 1.0 increases.	 Note: we
		 perceive volume logarithmically,  not	linearly.
		 Note: see the stat effect.

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

FILE TYPES
       SoX  uses  the file extension of the input and output file
       to determine what type of file format 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.




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SoX(1)							   SoX(1)


       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 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 sample	 and  the
		 sample value.	Values are normalized so that the
		 maximum and minimum are 1.00  and  -1.00.   This
		 file format can be used to create data files for
		 external programs such as FFT analyzers or graph
		 routines.   SoX  can also convert a file in this
		 format back into one of the other file	 formats.



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SoX(1)							   SoX(1)


       .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 support 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 han-
		 dle 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.

       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  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
		 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.

       .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



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SoX(1)							   SoX(1)


		 package, although not all are supported  by  the
		 samplers  themselves.	Currently loop points are
		 ignored.

       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.

       .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



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SoX(1)							   SoX(1)


		 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, .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),
		 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  headers.)   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
       Only one effect from the palette may be applied to a sound
       sample.	To do multiple effects you'll need to run sox  in
       a pipeline.

       avg [ -l | -r ]
		 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



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SoX(1)							   SoX(1)


		 to  manually  specify the avg effect and use the
		 -l and -r options to select  only  the	 left  or
		 right	channel for the output instead of averag-
		 ing the two channels.

       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]
		 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.	Where more than one  pair
		 of  attack/decay  parameters are specified, each
		 channel is treated separately and the number  of
		 pairs	must  agree  with  the	number	of  input



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SoX(1)							   SoX(1)


		 channels.  The second parameter  is  a	 list  of
		 points	 on  the  compander's  transfer	 function
		 specified in dB relative to the maximum possible
		 signal amplitude.  The input values must be in a
		 strictly increasing order but the transfer func-
		 tion  does  not have to be monotonically rising.
		 The special value -inf may be used  to	 indicate
		 that  the input volume should be associated out-
		 put 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 postprocessing gain in dB which is applied
		 after	the  compression  has  taken  place;  the
		 fourth (optional) parameter is an initial volume
		 to be assumed for each channel when  the  effect
		 starts.  This permits the user to supply a nomi-
		 nal level initially, so  that,	 for  example,	a
		 very large gain is not applied to initial signal
		 levels before the companding action has begun to
		 operate:  it  is  quite probable that in such an
		 event, the  output  would  be	severely  clipped
		 while	 the   compander  gain	properly  adjusts
		 itself.

       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.

       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.

       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



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SoX(1)							   SoX(1)


		 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 center
		 Apply	a  high-pass   filter.	  The	frequency
		 response  drops logarithmically with center fre-
		 quency 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 center
		 Apply a low-pass filter.  The frequency response
		 drops logarithmically with center  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



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SoX(1)							   SoX(1)


		 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	 Select the left or right  channel  of	a  stereo
		 sample,  or  one  of  four channels in a quadro-
		 phonic sample.

       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



			  July 24, 2000			       12





SoX(1)							   SoX(1)


		 algorithm.  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



			  July 24, 2000			       13





SoX(1)							   SoX(1)


		 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



			  July 24, 2000			       14





SoX(1)							   SoX(1)


		 better.  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



			  July 24, 2000			       15





SoX(1)							   SoX(1)


		 reverberation.	  Gain-out  is	the volume of the
		 output.

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

       speed 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.

       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.






			  July 24, 2000			       16





SoX(1)							   SoX(1)


       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.

       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 ]
		 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.  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  change  loga-
		 rithmically.	0.0  is constant while +6 doubles
		 the amplitude.

       Sox enforces certain effects.  If the two files have  dif-
       ferent sampling rates, the requested effect must be one of



			  July 24, 2000			       17





SoX(1)							   SoX(1)


       polyphase, rate, or resample.  If the two files have  dif-
       ferent  numbers of channels, the avg or other channel mix-
       ing effect must be requested.

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




































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