Home Man
Today's Posts

Linux & Unix Commands - Search Man Pages

RedHat 9 (Linux i386) - man page for soxmix (redhat section 1)

SoX(1)				     General Commands Manual				   SoX(1)

       sox - Sound eXchange : universal sound sample translator

       sox infile outfile

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

       soxmix infile1 infile2 outfile

       soxmix [ general options ] [ format options ] infile1
	   [ format options ] infile2
	   [ 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/-f ]
	   [ -b/-w/-l ]
	   [ -c channels ] [ -x ] [ -e ]

	   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...]
		   [ gain [ initial-volume [ delay ] ] ]
	   dcshift shift [ limitergain ]
	   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
	   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 # ]
	   resample [ -qs | -q | -ql ] [ rolloff [ beta ] ]
	   reverb gain-out reverb-time delay [ delay ... ]
	   silence above_periods [ duration threshold[ d | % ]
		   [ below_periods duration
		     threshold[ d | % ]]
	   speed [ -c ] factor
	   stat [ -s n ] [ -rms ] [ -v ] [ -d ]
	   stretch [ factor [ window fade shift fading ]
	   swap [ 1 2 | 1 2 3 4 ]
	   synth [ length ] type mix [ freq [ -freq2 ]
		 [ off ] [ ph ] [ p1 ] [ p2 ] [ p3 ]
	   trim start [ length ]
	   vibro speed [ depth ]
	   vol gain [ type [ limitergain ] ]

       SoX is a command line program that can convert most popular audio files to most other pop-
       ular 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.

       soxmix  is  functionally the same as the command line program sox expect that it takes two
       files as input and mixes the audio together to produce a single file as output.	It has	a
       restriction that both input files must be of the same data type and sample rates.

       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 characteris-
       tics of the audio data that follows.

       The  second  type  are  header-less  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  con-

       Audio data can usually be totally described by four characteristics:

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

       data size The precision the data is stored in.  Most popular are  8-bit	bytes  or  16-bit

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

       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.

       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  sam-
       pling rate to 12000 hertz, and applies the mask sound effect to the audio data.

       The following will mix two sound files together to to produce a single sound file.

	       soxmix music.wav voice.wav mixed.wav

       Format options:

       Format options effect the audio samples that they immediately precede.  If they are placed
       before the input file name then they effect the input data.  If they are placed before the
       output  file name then they will effect the output data.  By taking advantage of this, you
       can override a input file's corrupted header or produce an output  file	that  is  totally
       different  style  then the input file.  It is also how SoX is informed about the format of
       raw input data.

       -t filetype
		 gives the type of the sound sample file.  Useful  when  file  extension  is  not
		 standard or for specifying 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.

		 The  sample data encoding is signed linear (2's complement), unsigned linear, u-
		 law (logarithmic), A-law (logarithmic),  ADPCM,  IMA_ADPCM,  GSM,  or	Floating-
		 U-law	(actually  shorthand for mu-law) and A-law are the U.S. and international
		 standards for logarithmic telephone sound compression.  When uncompressed  u-law
		 has  roughly the precision of 14-byte PCM audio and A-law has roughly the preci-
		 sion of 13-bit PCM audio.
		 A-law and u-law data is sometimes encoded using a reversed bit-ordering (ie. MSB
		 becomes  LSB).   Internally,  SoX understands how to work with this encoding but
		 there is currently no command line option to specify it.  If you need this  sup-
		 port  then you can use the psuedo file types of ".la" and ".lu" to inform sox of
		 the encoding.	See supported file types for more information.
		 ADPCM is a form of sound compression that has a  good	compromise  between  good
		 sound	quality  and fast encoding/decoding time.  It is used for telephone sound
		 compression and places were full fidelity is  not  as	important.   When  uncom-
		 pressed  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  different  mean-
		 ings 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 compres-
		 sion,	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.  It usually is CPU intensive to
		 work with GSM audio data.

       -b/-w/-l  The sample data size is in bytes, 16-bit words, or 32-bit long words.

       -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-for-
		 mat floating-point data is not portable.

       -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 specified 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 out-
		 put 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 perceive 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.  Useful for figuring out  exactly  how
		 SoX is mangling your sound samples.

       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 specify-
       ing '-' as the filename.

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

       The following file formats are supported:

       .8svx	 Amiga 8SVX musical instrument description format.

       .aiff	 AIFF files used on Apple IIc/IIgs and SGI.  Note: the AIFF format supports  only
		 one  SSND chunk.  It does not support multiple sound chunks, or the 8SVX musical
		 instrument description format.  AIFF files are multimedia archives and can  have
		 multiple  audio  and  picture	chunks.  You may need a separate archiver to work
		 with them.

       .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 han-
		 dler 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 commercial packages on the Mac.

       .cdr	 CD-R
		 CD-R files are used in mastering music on Compact Disks.  The audio  data  on	a
		 CD-R disk is a raw audio file with a format of stereo 16-bit signed samples at a
		 44khz sample rate.  There is a special blocking/padding oddity at the end of the
		 audio file and is why it needs its own handler.

       .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.  Subsequent  lines  contain
		 two numeric data items: the time since the beginning of the first sample and the
		 sample value.	Values are normalized so that the maximum and  minimum	are  1.00
		 and  -1.00.  This file format can be used to create data files for external pro-
		 grams such as FFT analyzers 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
		 telecommunications (GSM).  Its good for its purpose, shrinking audio data  size,
		 but  it  will	introduce  lots of noise when a given sound sample is encoded and
		 decoded multiple times.  This format is used by some  voice  mail  applications.
		 It is rather CPU intensive.
		 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  sup-
		 ported file formats.

       .hcom	 Macintosh  HCOM  files.  These are (apparently) Mac FSSD files with some variant
		 of Huffman compression.  The Macintosh has wacky file formats	and  this  format
		 handler  apparently  doesn't handle all the ones it should.  Mac users will need
		 your usual arsenal of file converters to deal with an HCOM file  under  Unix  or

       .maud	 An Amiga format
		 An IFF-conform sound file type, registered by MS MacroSystem Computer GmbH, pub-
		 lished along with the "Toccata" sound-card on the Amiga.   Allows  8bit  linear,
		 16bit linear, A-Law, u-law in mono and stereo.

       .nul	 Null  file  handler.	This  is  a fake file hander that act as if its reading a
		 stream of 0's from a while or fake writing output to a file.  This is not a very
		 useful  file handler in most cases.  It might be useful in some scripts were you
		 do not want to read or write from a real file but would like to specify a  file-
		 name for consistency.

       .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.
		 Decoding 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 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 out-
		 put values to use the highest quality samples your sound card can handle.  Exam-
		 ple: -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.

		 SPHERE (SPeech HEader Resources) is a file  format  defined  by  NIST	(National
		 Institute  of	Standards and Technology) and is used with speech audio.  SoX can
		 read these files when they contain u-law and  PCM  data.   It	will  ignore  any
		 header  information  that  says the data is compressed using shorten compression
		 and will treat the data as either u-law 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 Soft-
		 works.  This  package	is for communication to several MIDI samplers. All sample
		 rates are supported by the package, although not all are supported by	the  sam-
		 plers themselves. Currently loop points are ignored.

		 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 out-
		 put values to use the highest quality samples your hardware can  handle.   Exam-
		 ple:  -t  sunau  -w  -s  /dev/audio or -t sunau -U -c 1 /dev/audio for older sun

       .txw	 Yamaha TX-16W sampler.
		 A file format from a Yamaha sampling keyboard which  wrote  IBM-PC  format  3.5"
		 floppies.   Handles reading of files which do not have the sample rate field set
		 to one of the expected by looking at some other bytes in the attack/loop  length
		 fields, and defaulting to 33kHz if the sample rate is still unknown.

       .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 sam-
		 ple rates for different chunks.  On input, the silence  parts	are  filled  out,
		 loops are rejected, and sample data with a new sample rate is rejected.  Silence
		 with a different sample rate is generated appropriately.  On output, silence  is
		 not  detected, nor are impossible sample rates.  Note, this version now supports
		 playing VOC files with multiple blocks and supports playing files containing  u-
		 law and A-law samples.

       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 formatting information in their  headers,
		 and  so  do not need any format options specified for an input file. If any are,
		 they will override the file header, and you will be warned to this effect.   You
		 had  better  know  what you are doing! Output format options will cause a format
		 conversion, and the .wav will written appropriately.	SoX  currently	can  read
		 PCM,  ULAW,  ALAW,  MS ADPCM, and IMA (or DVI) ADPCM.	It can write all of these
		 formats including (NEW!)  the ADPCM encoding.

       .wve	 Psion 8-bit A-law
		 These are 8-bit A-law 8khz sound files used on the Psion palmtop  portable  com-

       .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, .lu, .la, .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, al, lu, la and sl correspond to
		 "unsigned byte", "signed byte", "unsigned word", "signed word", "u-law"  (byte),
		 "A-law"  (byte),  inverse  bit  order	"u-law",  inverse  bit order "A-law", and
		 "signed long".  The sample rate defaults to 8000 hz if not explicitly	set,  and
		 the  number of channels defaults to 1.  There are lots of Sparc samples floating
		 around in u-law format with no header and fixed at a sample  rate  of	8000  hz.
		 (Certain  sound management software cheerfully ignores the headers.)  Similarly,
		 most Mac sound files are in unsigned byte format with a sample rate of 11025  or
		 22050 hz.

       .auto	 This  is  a  ``meta-type'': specifying this type for an input file triggers some
		 code that tries to guess the real type by looking for magic words in the header.
		 If  the  type	can't  be  guessed, the program exits with an error message.  The
		 input must be a plain file, not a pipe.  This type  can't  be	used  for  output

       Multiple  effects may be applied to the audio data by specifying 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 automatically used when the
		 number of input channels differ from the number of output channels.  When reduc-
		 ing 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 proportion of each input channel that is to be mixed into each  out-
		 put  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  propor-
		 tions 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
		 instrumental  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  fil-
		 ter,  so  beware  of output clipping.	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 coming 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 milliseconds and the decay (relative to gain-in) with a  modulation
		 speed	in  Hz	using depth in milliseconds.  The modulation is either sinusoidal
		 (-s) or triangular (-t).  Gain-out is the volume of the output.

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


	       [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  separately  and
		 the  number  of  pairs must agree with the number of input channels.  The second
		 parameter is a list of points on the compander's transfer function specified  in
		 dB  relative to the maximum possible signal amplitude.  The input values must be
		 in a strictly increasing order but the transfer function does	not  have  to  be
		 monotonically	rising.   The special value -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 post-processing gain in dB which is applied
		 after the compression has taken place; the fourth  (optional)	parameter  is  an
		 initial volume to be assumed for each channel when the effect starts.	This per-
		 mits the user to supply a nominal level initially, so that, for example, a  very
		 large	gain is not applied to initial signal levels before the companding action
		 has begun to operate: it is quite probable that in such  an  event,  the  output
		 would be severely clipped while the compander gain properly adjusts itself.

		 The  fifth (optional) parameter is a delay in seconds.  The input signal is ana-
		 lyzed immediately to control the compander, but it is delayed before  being  fed
		 to  the  volume  adjuster.   Specifying  a  delay  approximately  equal  to  the
		 attack/decay times allows the compander to effectively operate in a "predictive"
		 rather than a reactive mode.

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

       dcshift shift [ limitergain ]
		 DC Shift the audio data, with basic linear amplitude formula.	This is most use-
		 ful  if  your audio data tends to not be centered around a value of 0.  Shifting
		 it back will allow you to get the most volume adjustments without clipping audio
		 The first option is the dcshift value.  It is a floating point number that indi-
		 cates the amount to shift.
		 An option limtergain value can be specified as well.  It  should  have  a  value
		 much less then 1.0 and is used only on peaks to prevent clipping.

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

       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 explanation.

       echo gain-in gain-out delay decay [ delay decay ... ]
		 Add echoing to a sound sample.  Each delay/decay part gives the  delay  in  mil-
		 liseconds  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-
		 All times can be specified in either periods of time or sample counts.  To spec-
		 ify time periods use the format hh:mm:ss.frac format.	To specify  using  sample
		 counts,  specify  the	number of samples and append the letter 's' to the sample
		 count (for example 8000s).
		 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 log-
		 arithmic, 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  fil-
		 ter  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 Nuttall window by specifying anything <= 2.0 here.  For  more  dis-
		 cussion 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 milliseconds and the decay (relative to gain-in) with a modulation	speed  in
		 Hz.  The modulation is either sinodial (-s) or triangular (-t).  Gain-out is the
		 volume of the output.

       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 coming 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 fil-
		 ter 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 miscellaneous 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 contains 2 chan-
		 nels, then it will create the missing channel itself.	The direction is a  value
		 from  -1.0 to 1.0.  -1.0 represents far left and 1.0 represents far right.  Num-
		 bers in between will start the pan effect without totally  muting  the  opposite

       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 milliseconds and the decay (relative to gain-in) with a modulation	speed  in
		 Hz.   The  modulation	is  either  sinodial  (-s) or triangular (-t).	The decay
		 should be less than 0.5 to avoid feedback.  Gain-out is the volume of	the  out-

       pick [ -1 | -2 | -3 | -4 | -l | -r ]
		 Select  the left or right channel of a stereo sample, or one of four channels in
		 a quadraphonic 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 contain 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, neg-
		 ative 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.  inter-
		 pole 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 interpola-
		 tion, a DSP 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) window.  Default is nut.

		 -width long / short / # : specify the (approximate) width of the  filter.   long
		 is  1024  samples;  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 fre-
		 quency 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	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  Common  Multiple of the two sampling rates.  This is the default
		 effect if the two files have different sampling rates and  the  preview  options
		 was  specified.  This is fast but noisy: the spectrum of the original sound will
		 be shifted upwards and duplicated faintly when up-translating by a multiple.

		 Lerp-ing is acceptable for cheap 8-bit sound hardware, but for CD-quality  sound
		 you should instead use either 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-dresden.de/~wilde/resample/resample.html

       resample [ -qs | -q | -ql ] [ rolloff [ beta ] ]
		 Translate  input sampling rate to output sampling rate via simulated analog fil-
		 tration.  This method is slower than rate, but gives much better results.

		 By default, linear interpolation is used, with a window width about  45  samples
		 at  the  lower  of  the  two rate.  This gives an accuracy of about 16 bits, but
		 insufficient stopband rejection in the  case  that  you  want	to  have  rolloff
		 greater than about 0.80 of the Nyquist frequency.

		 The  -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 converter needs at least 2 samples to detect 1 cycle at the Nyquist fre-
		 quency.  Frequencies higher then the Nyquist will actually appear as lower  fre-
		 quencies  to  the  A/D converter and is called aliasing.  Normally, A/D converts
		 run the signal through a highpass filter first to avoid these problems.

		 Similar problems will happen in software when reducing the  sample  rate  of  an
		 audio	file (frequencies above the new Nyquist frequency can be aliased to lower
		 frequencies).	Therefore, a good  resample  effect  will  remove  all	frequency
		 information above the new Nyquist frequency.

		 The  rolloff  refers  to how close to the Nyquist frequency this cutoff is, with
		 closer being better.  When increasing the sample rate of an audio file you would
		 not expect to have any frequencies exist that are past the original Nyquist fre-
		 quency.  Because of resampling properties, it is common to  have  alaising  data
		 created  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 compar-
		 ing 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 preci-

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

			    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 reverbera-
		 tion.	Gain-out is the volume of the output.

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

       silence above_periods [ duration threshold[ d | % ]

	       [ below_periods duration

		 threshold[ d | % ]]
		 Removes silence from the beginning or end of a sound file.  Silence is  anything
		 below a specified threshold.
		 When trimming silence from the beginning of a sound file, you specify a duration
		 of audio that is above a given silence threshold before audio data is processed.
		 You  can  also  specify  the count of periods of none silence you want to detect
		 before processing audio data.	Specify a period of 0 if you do not want to  trim
		 data from the front of the sound file.
		 When  optionally  trimming silence form the end of a sound file, you specify the
		 duration of audio that must be  below	a  given  threshold  before  stopping  to
		 process  audio data.  A count of periods that occur below the threshold may also
		 be specified.	If this options are not specified then data is not  trimmed  from
		 the end of the audio file.
		 Duration  counts  may	be  in the format of time, hh:mm:ss.frac, or in the exact
		 count of samples.
		 Threshold may be suffixed with d, or % to indicated the value is in decibels  or
		 a  percentage	of  max value of the sample value.  A value of '0%' will look for
		 total silence.

       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 doubles 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 copying 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 Adjustment:"	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  for-

		 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 duration.  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

       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.

       synth [ length ] type mix [ freq [ -freq2 ]

	     [ off ] [ ph ] [ p1 ] [ p2 ] [ p3 ]
		 The  synth  effect  will  generate  various  types of audio data.  Although this
		 effect is used to generate audio data, an input file  must  be  specified.   The
		 length of the input audio file determines the length of the output audio file.
		 <length> length in sec or hh:mm:ss.frac, 0=inputlength, default=0
		 <type> is sine, square, triangle, sawtooth, trapetz, exp, whitenoise, pinknoise,
		 brownnoise, default=sine
		 <mix> is create, mix, amod, default=create
		 <freq> frequency at beginning in Hz, not used	for noise..
		 <freq2> frequency at end in Hz, not used for noise..  <freq/2> can be	given  as
		 %%n, where 'n' is the number of half notes in respect to A (440Hz)
		 <off> Bias (DC-offset)  of signal in percent, default=0
		 <ph> phase shift 0..100 shift phase 0..2*Pi, not used for noise..
		 <p1> square: Ton/Toff, triangle+trapetz: rising slope time (0..100)
		 <p2> trapetz: ON time (0..100)
		 <p3> trapetz: falling slope position (0..100)

       trim start [ length ]
		 Trim  can  trim  off unwanted audio data from the beginning and end of the audio
		 file.	Audio samples are not sent to the output stream until the start  location
		 is reached.
		 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 parameter will allow trimming off the back side only.
		 Both  options can be specified using either an amount of time and an exact count
		 of samples.  The format for specifying lengths  in  time  is  hh:mm:ss.frac.	A
		 start value of 1:30.5 will not start until 1 minute, thirty and 1/2 seconds into
		 the audio data.  The format for specifying sample counts is the number  of  sam-
		 ples  with the letter 's' appended to it.  A value of 8000s will wait until 8000
		 samples are read before starting to process 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 [ 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 ampli-
		 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  volume
		 to  increase.	 Beware of clipping audio data when the gain is greater then 1.0.
		 A negative value performs the same adjustment while also changing the phase.
		 When type is power then a value of 1.0 also means no change in volume.
		 When type is dB the amplitude is changed logarithmically.  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 prevent  clipping.   Not
		 specifying  this  parameter  will cause no limiter to be used.  In verbose mode,
		 this effect will display the percentage of audio data that needed to be limited.

       The syntax is horrific.	Thats the breaks when trying to handle all things from	the  com-
       mand line.

       Please	report	 any  bugs  found  in  this  version  of  SoX  to  Chris  Bagwell  (cbag-

       play(1), rec(1), soxexam(1)

       The version of SoX that accompanies this manual page is support by  Chris  Bagwell  (cbag-
       well@users.sourceforge.net).   Please  refer  any  questions regarding it to this address.
       You may obtain the latest version at the the web site http://sox.sourceforge.net/

       Chris Bagwell (cbagwell@users.sourceforge.net).

       Updates by Anonymous

					December 11, 2001				   SoX(1)

All times are GMT -4. The time now is 12:32 PM.

Unix & Linux Forums Content Copyrightę1993-2018. All Rights Reserved.
Show Password