1 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  ] ifile 
       [ format options  ] ofile  [ effect [ effect options ... ] ]

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

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

Effects:
	copy 
	rate 
	avg 
 	resample
 	check 
	stat 
	echo delay volume [ delay volume ... ]
	vibro speed [ depth ]
	lowp center
	highp center
	band [ -n ] center [ width ]

2 DESCRIPTION
Sox
translates sound files from one format to another,
possibly doing a sound effect.

2 OPTIONS
The option syntax is a little grotty, but in essence:
 
     $ sox file.au file.voc 
 
translates a sound sample in SUN Sparc .AU format 
into a SoundBlaster .VOC file, while
 
     $ sox -v 0.5 file.au -rate 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.

File type options:

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

  -r rate
Give sample rate in Hertz of file.

  -s/-u/-U/-A
The sample data is signed linear (2's complement),
unsigned linear, U-law (logarithmic), or A-law (logarithmic).
U-law and A-law are the U.S. and international
standards for logarithmic telephone sound compression.

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

  -x
The sample data is in XINU format; that is,
it comes from a machine with the opposite word order 
than yours and must
be swapped according to the word-size given above.
Only 16-bit and 32-bit integer data may be swapped.
Machine-format floating-point data is not portable.
IEEE floats are a fixed, portable format. ???

  -c channels
The number of sound channels in the data file.
This may be 1, 2, or 4; for mono, stereo, or quad sound data.

General options:

  -e
after the input file allows you to avoid giving
an output file and just name an effect.
This is only useful with the  stat effect.

  -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.
Useful for figuring out exactly how sox
is mangling your sound samples.

The input and output files may be standard input and output.
This is specified by '-'.
The -t type
option must be given in this case,
else sox 
will not know the format of the given file.

The
   -t,
   -r,
   -s/-u/-U/-A,
   -b/-w/-l/-f/-d/-D
and
   -x
options refer to the input data when given before the
input file name.  After, they refer to the output data.

If you don't give an output file name, sox
will just read the input file.
This is useful for validating structured file formats;
the stat effect may also be used via the -e
option.

2 FILE_TYPES
Sox
needs to know the formats of the input and output files.
File formats which have headers are checked, 
if that header doesn't seem right,
the program exits with an appropriate message.
Currently, raw (no header) binary and textual data, 
IRCAM Sound Files,
Sound Blaster, SPARC .AU (w/header), Mac HCOM,
PC/DOS .SOU, Sndtool, and Sounder, NeXT .SND,
Windows 3.1 RIFF/WAV, Turtle Beach .SMP, CD-R, 
and Apple/SGI AIFF and 8SVX formats are supported.

   .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
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).

   .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 DOS.
 

   .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.

   ".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 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.

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

   .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 generated appropriately.
On output, silence is not detected, nor are impossible sample rates.

   .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 files.
.PP

   .cdr
CD-R
 
CD-R files are used in mastering music Compact Disks.
The file format is, as you might expect, raw stereo
raw unsigned samples at 44khz.  But, there's
some blocking/padding oddity in the format, so it
needs its own handler.

   .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 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.

   .smp
Turtle Beach SampleVision files.
 
SMP files are for use with the PC-DOS package SampleVision by Turtle Beach
Softworks. This package is for communication to several MIDI samplers. All
sample rates are supported by the package, although not all are supported by
the samplers themselves. Currently loop points are ignored.
.PP

   .wav
Windows 3.1 .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 3.1.
Obviously, Windows 3.1 is of such incredible importance
to the computer industry that it just had to have its own 
sound file format.

2 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.

  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".

  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.
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:

 resample

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.

   avg
Mix 4- or 2-channel sound file into 2- or 1-channel file
by averaging the samples for different speakers.

  check
Do a format check on the input file,
and print any errors on the standard error file.
Write no output.
If you give no output file, 
you need to specify either this effect or the following:

  stat
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 output,
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.

  echo [ delay volume ...  ]
Add echoing to a sound sample.
Each delay/volume pair gives the delay in seconds
and the volume (relative to 1.0) of that echo.
If the volumes add up to more than 1.0,
the sound will melt down instead of fading away.

  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.

  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.

  highp center
Apply a high-pass filter.
The frequency response drops logarithmically with 
center frequency in the middle of the drop.
The slope of the filter is quite gentle.

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

Sox enforces certain effects.
If the two files have different sampling
rates, the requested effect must be one of
   copy,
or
   rate,
or
   resample.

If the two files have different numbers of channels,
the avg or other channel mixing
effect must be requested.

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

2 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.

2 NOTICES
The echoplex effect is:
    Copyright (C) 1989 by Jef Poskanzer.
	
    Permission to use, copy, modify, and distribute this software 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.

