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sfconvert(1)							  sfconvert(1)

NAME    [Toc]    [Back]

     sfconvert - convert soundfiles

SYNOPSIS    [Toc]    [Back]

     sfconvert infile outfile [	options	... ] [	output keywords	... ]

     where options are:

	-inputraw [ input keywords ... ] end

     The full option string need not be	specified; -i and -o will work.

DESCRIPTION    [Toc]    [Back]

     sfconvert converts	a soundfile to another soundfile of a different	file
     format or data format.

     If	invoked	with no	arguments, sfconvert displays help.

     The default conversion is to the same format.  So,	the following

	sfconvert in.snd out.snd

     Is	basically like "cp in.snd out.snd," except that	it will	not copy
     comments or undocumented chunks found in the input	file to	the output

     If	invoked	with a pair of filenames and some output keywords, sfconvert
     will attempt to open 'infile', convert it to a soundfile described	by the
     keywords, and write the result in 'outfile' .

     For example, the following	converts a (presumably stereo) input file to a
     mono output file, keeping all other parameters the	same:

	sfconvert in.snd out.snd channels 1

     And this converts an input	file (of any allowed format) to	a 24-bit, 2's
     complement, 2-channel integer aiff	file.

	sfconvert in.snd out.aif format	aiff int 24 2 chan 2

     See sfkeywords(1) for a complete description of the keywords used to
     describe soundfile	formats.

     The sfconvert program now performs	high-quality sampling rate conversion
     between all of the	standard sampling rates.  See sfkeywords(1) for	more
     information about sampling	rate conversion.

									Page 1

sfconvert(1)							  sfconvert(1)

RAW OUTPUT    [Toc]    [Back]

     The -outputraw (-o) option	will cause sfconvert to	write its output in a
     file of raw (headerless) data.  If	you supply this	option and you also
     supply a "format" keyword (which normally indicates what type of header
     format to use), the "format" keyword will be ignored.

     If	you specify other keywords, the	input file's data will be converted
     appropriately before it is	written	to the raw output file.

RAW INPUT    [Toc]    [Back]

     sfconvert will also change	a file of raw data into	a valid	soundfile of
     any format	(such as aiff or wave).

     In	this case the data format of the raw input data	must be	specified,
     using the appropriate keywords with the -inputraw (-i) option.

     The input keywords	following -i must be terminated	with 'end' in order to
     distinguish them from the output keywords.

     For example, a raw	data file consisting of	8-bit unsigned mono integer
     samples could be converted	to a playable aiff file	with the following

      sfconvert	in.raw out.aif -i int 8	u chan 1 end format aiff
				  ^^^^^^^^^^^^^^     ^^^^^^^^^^^
				  input	format	    output format

     The keywords are underlined and labeled as	to which soundfile they

     See sfkeywords(1) for a complete description of the keywords used to
     describe soundfile	formats.

     If	the keywords are not specified or not fully specified, the default raw
     input soundfile format is:

     Format: 1 channel 8-bit integer (2's complement) file.
     Sampling Rate: 8.000 kHz.
     Data begins at offset 0 (0	hex).

     So, because a sampling rate was not given,	the above example would
     produce an	8 kHz soundfile.

LIMITATIONS    [Toc]    [Back]

     sfconvert will convert almost any popular soundfile format	to a playable
     or	interchangeable	format.	 However, it does have certain limitations.
     sfconvert will:

     - only read and write mulaw files with 8-bit encoding.

     - read and	write only mono	and stereo files (although sfinfo will tell

									Page 2

sfconvert(1)							  sfconvert(1)

     you if a file has more than 2 channels).  4-channel support (and probably
     more) is coming in	a future release.

     - read and	write only single precision IEEE floating point	files (double
     is	not supported yet; this	is coming in a future release).

     - only read and write integer files of 1-32 bits.

     - perform endian conversions only for integer data.

     - not manipulate MIDI files or MIDI chunks	in soundfiles.

     sfconvert uses a high-quality linear phase	sampling rate conversion
     algorithm for conversions between the standard rates displayed on the
     Audio Control Panel.  For conversions to or from a	nonstandard rate,
     sfconvert uses a lower-quality algorithm, and issues a warning to this
     effect.  For these	lower-quality conversions, some	loss of	quality	is
     likely, and audible artifacts may occur in	the output sound, especially
     on	conversions from a higher to a lower sampling rate.  This lower
     quality algorithm,	which was present in earlier releases, uses thirdorder
 polynomial interpolation and	does no	pre- or	post-filtering.	 A
     high-quality algorithm capable of conversion between arbitrary pairs of
     sampling rates is under development.

     Note also that no dithering is done on conversions	from integers of
     higher resolution to lower	resolution.  This will be amended in a future

AUTHOR    [Toc]    [Back]

     Silicon Graphics Inc.; Apple Computer, Inc. for AIFF code.

SEE ALSO    [Toc]    [Back]

     intro(3a) for more	about the audio	library.
     sfplay(1),	sfinfo(1), sfkeywords(1),
     dmconvert(1), dminfo(1), mediaconvert(1)

									PPPPaaaaggggeeee 3333
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