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sfconvert(1) [debian man page]

SFCONVERT(1)							 Debian GNU/Linux						      SFCONVERT(1)

NAME

       sfconvert - convert between various audio formats

SYNOPSIS

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

DESCRIPTION

       The  sfconvert  tool can be used to convert audio files from one audio format to another.  The files' audio formats have to be supported by
       libaudiofile.

OPTIONS

       The following keywords specify the format of the output sound file:

       byteorder e
	      Desired endianness of output sample data. e may be one of big or little.

       channels n
	      Number of output channels. n is 1 for mono, and 2 for stereo files.

       format f
	      Audio format of output file. f has to be one of the currently supported formats: aiff (Audio Interchange File Format), aifc  (AIFF-C
	      File  Format),  next (NeXT/Sun Format), wave (MS RIFF WAVE Format), bicsf (Berkeley/IRCAM/CARL Sound File Format), avr (Audio Visual
	      Research File Format), iff (Amiga IFF/8SVX Sound File Format), or nist (NIST SPHERE File Format).

       integer n s
	      Produce integer samples. n specifies the width of individual samples in bits, s yields the encoding and may be one  of  2scomp  (2's
	      complement signed data), or unsigned (unsigned data). The integer and float options (see below) are mutually exclusive.

       float m
	      Produce  floating point samples with a maximum amplitude of m (usually 1.0). This options may not be used together with option inte-
	      ger.

SEE ALSO

       sfinfo(1).

AUTHOR

       sfconvert was written by Michael Pruett <michael@68k.org>.

       This manual page was written by Daniel Kobras <kobras@debian.org> for the Debian GNU/Linux system (but may be used by others).  It is based
       on the sfconvert plain text documentation as distributed with audiofile.

Debian Project							    March 2001							      SFCONVERT(1)

Check Out this Related Man Page

AFGETFRAMESIZE(3)														 AFGETFRAMESIZE(3)

NAME

       afGetFrameSize - calculate the frame size in bytes for an audio track

SYNOPSIS

       #include <audiofile.h>

       float afGetFrameSize (AFfilehandle file, int track, int expand3to4);

PARAMETERS

       file is a valid AFfilehandle.

       track is an integer which refers to a specific audio track in the file. At present no supported audio file format allows for more than one
       audio track within a file, so track should always be AF_DEFAULT_TRACK.

       expand3to4 is a boolean-valued integer indicating whether frame size calculation will treat 24-bit data as having a size of 3 bytes or 4
       bytes.

DESCRIPTION

       afGetFrameSize returns the number of bytes in a frame in a given audio track.

       A sample frame consists of one or more samples. For a monaural track, a sample frame will always contain one sample. For a stereophonic
       track, a sample frame will always contain two samples, one for the left channel and one for the right channel.

       A non-zero value of expand3to4 should be used when calculating the frame size for storage in memory (since 24-bit audio data is presented
       in memory as a 4-byte sign-extended integer), while a value of zero should be used for calculating storage on disk where no padding is
       added. The parameter expand3to4 is ignored unless the specified audio track contains 24-bit audio data.

AUTHOR

       Michael Pruett <michael@68k.org>

Audio File Library 0.3.6					    03/06/2013							 AFGETFRAMESIZE(3)
Man Page

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