sox.txt

linux下录音程序

		 matting  information in their headers, and so do
		 not need any format  options  specified  for  an
		 input	file.  If  any are, they will overide 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.	 Note that it is possible to  write  data
		 of  a	type that cannot be specified by the .wav
		 header, and you will be warned that you a  writ-
		 ing  a	 bad  file !  Sox currently can read PCM,
		 ULAW, ALAW, MS ADPCM, and IMA	(or  DVI)  ADPCM.
		 It  can  output  all of these formats except the
		 ADPCM styles.

       .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
		 style (signed, unsigned, etc.)	  of  the  sample
		 file  must  be	 given.	  The  number of channels
		 defaults to 1.





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SOX(1)							   SOX(1)


       .ub, .sb, .uw, .sw, .ul
		 These are several  suffices  which  serve  as	a
		 shorthand  for	 raw  files with a given size and
		 style.	 Thus, ub, sb, uw, sw, and ul  correspond
		 to  "unsigned	byte",	"signed	 byte", "unsigned
		 word", "signed word", and  "ulaw"  (byte).   The
		 sample	 rate  defaults to 8000 hz if not explic-
		 itly 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.  Valid combinations are 1	-
		 2,  1 - 4, 2 - 4, 4 - 2, 4 - 1, 2 - 1. The -l or
		 -r option averages from just left or right chan-
		 nels/duplicates  to just the left or right chan-
		 nels.

       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.  Band
		 introduces  noise  in	the  shape of the filter,
		 i.e. peaking at the center  frequency	and  set-
		 tling around it.

       chorus gain-in gain-out delay decay speed deptch




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SOX(1)							   SOX(1)


	      -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.

       copy	 Copy the input file to the output file.  This is
		 the default effect if both files have	the  same
		 sampling rate, or the rates are "close".

       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.

       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.

       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.




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SOX(1)							   SOX(1)


       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.

       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.

       polyphase [ -w < num / 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 then rate.
		 -w < nut / ham > : select either a  Nuttal  (~90
		 dB  stopband)	or Hamming (~43 dB stopband) win-
		 dow.  Warning: Nuttall windows require 2x length
		 than Hamming windows.	Default is nut.
		 -width	 long  / short / # : specify the width of
		 the filter.  long is 1024 samples; short is  128
		 samples.   Alternatively, an exact number can be
		 used.	Default is long.
		 -cutoff # : specify the filter cutoff	frequency
		 in  terms  of	fraction of bandwidth.	If upsam-
		 pling, then this is the fraction of the  orignal
		 signal that should go through.	 If downsampling,
		 this is the fraction of the  signal  left  after
		 downsampling.	 Default  is 0.95.  Remember that
		 this is a float.


       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



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SOX(1)							   SOX(1)


		 different sampling  rates.   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  won-
		 dering	 which	of Sox's rate changing effects to
		 ues, you will want to read a  detailed	 analysis
		 of all of them at http://usa.ece.cmu.edu/Sox/

       resample [ rolloff [ beta ] ]
		 Translate input sampling rate to output sampling
		 rate  via  simulated  analog  filtration.   This
		 method	 is  slow and uses lots of RAM, but gives
		 much better results then rate (This has  empiri-
		 cally	been  shown  to	 be  false.  The resample
		 algorthym needs to be updated from its	 original
		 source).

       reverb gain-out delay [ delay ... ]
		 Add  reverbation  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-
		 bation.  Gain-out is the volume of the output.

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

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

       stat [ debug | -v ]
		 Do  a	statistical  check on the input file, and
		 print results on the standard error file.   stat
		 may  copy  the file untouched from input to out-
		 put, if you select an output file.  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.
		 There is an  optional	parameter  -v  that  will
		 print out the "Volume Adjustment:" field's value
		 and return.  This could be of use in scripts  to
		 auto  convert	the  volume.  There is an also an
		 optional parameter debug  that	 will  place  sox
		 into  debug mode and 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.




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SOX(1)							   SOX(1)


       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.

       Sox  enforces certain effects.  If the two files have dif-
       ferent sampling rates, the requested effect must be one of
       copy,  or rate, If the two files have different numbers of
       channels, the avg effect must be requested.

BUGS
       The syntax is horrific.	It's very tempting to  include	a
       default	system	that allows an effect name as the program
       name and just pipes a sound sample from standard input  to
       standard	 output,  but the problem of inputting the sample
       rates makes this unworkable.

       Please report any bugs found in this  version  of  sox  to
       Chris Bagwell ([e-mail])

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

NOTICES
       The   echoplex  effect  is:  Copyright  (C)  1989  by  Jef
       Poskanzer.

       Permission to use, copy, modify, and distribute this soft-
       ware and its documentation for any purpose and without fee
       is hereby  granted,  provided  that  the	 above	copyright
       notice  appear  in all copies and that both that copyright
       notice and this permission  notice  appear  in  supporting
       documentation.	This software is provided "as is" without
       express or implied warranty.

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













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