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)

AUDIORECORD(1)						    BSD General Commands Manual 					    AUDIORECORD(1)

NAME

     audiorecord -- record audio files

SYNOPSIS

     audiorecord [-afhqV] [-B buffersize] [-b balance] [-c channels] [-d device] [-e encoding] [-F format] [-i info] [-m monvol] [-P precision]
		 [-p port] [-s rate] [-t time] [-v volume] file

DESCRIPTION

     The audiorecord program copies the audio device to the named audiofile or, if the file name is -, to the standard output.

     The output file will contain either a Sun/NeXT audio header, a RIFF/WAVE audio header or no header at all.  Sun output files using a linear
     PCM encoding are written with big-endian signed samples, possibly after converting these from little-endian or unsigned samples.  RIFF/WAVE
     files are written in little-endian, signed samples, also converting if necessary.	The default output is Sun/NeXT format, but if the output
     file file ends with a .wav file extension it will be written as RIFF/WAVE.

OPTIONS

     The following options are available:

     -a 	    Append to the specified file, rather than overwriting.

     -B buffersize  Set the audio device read buffer size to buffersize.  The default value is the record.buffer_size of the audio device.

     -b balance     Set the balance to balance.  This value must be between 0 and 63.

     -c channels    Set number of channels to channels.

     -d device	    Set the audio device to be device.	The default is /dev/sound.

     -e encoding    Set encoding to either ``alaw'', ``ulaw'', or ``linear'', or any other value reported by audioctl encodings.  The default
		    encoding is ``ulaw''.  If the output format is ``sun'', the file will contain slinear_be samples, if it is ``wav'', then slin-
		    ear_le, independent of the argument to -e.	Setting the argument to -e still may be important since it is used in an ioctl(2)
		    call to the kernel to choose the kind of data provided.

     -F format	    Set the output header format to format.  Currently supported formats are ``sun'', ``wav'', and ``none'' for Sun/NeXT audio,
		    WAV, and no header, respectively.

     -f 	    Force.  Normally when appending to audiofiles using the -a option, the sample rates must match.  The -f option will allow a
		    discrepancy to be ignored.

     -h 	    Print a help message.

     -i info	    If supported by the -F format, add the string info to the output header.

     -m monvol	    Set the monitor volume.

     -P precision   Set the precision.	This value is the number of bits per sample, and is normally either ``8'' or ``16'', though the values
		    ``4'', ``24'', and ``32'' are also valid.

     -p port	    Set the input port to port.  The valid values of port are ``cd'', ``internal-cd'', ``mic'', and ``line''.

     -q 	    Be quiet.

     -s rate	    Set the sampling rate.  This value is per-second.  Typical values are 8000, 44100, and 48000, which are the telephone, CD
		    Audio, and DAT Audio default sampling rates.

     -t time	    Sets the maximum amount of time to record.	Format is [hh:]mm:ss[.dddddd].

     -V 	    Be verbose.

     -v volume	    Set the volume (gain) to volume.  This value must be between 0 and 255.

ENVIRONMENT

     AUDIOCTLDEVICE  the audio control device to be used.

     AUDIODEVICE     the audio device to be used.

SEE ALSO

     audioctl(1), audioplay(1), aria(4), audio(4), audioamd(4), auich(4), autri(4), auvia(4), clcs(4), clct(4), cmpci(4), eap(4), emuxki(4),
     esm(4), eso(4), ess(4), fms(4), gus(4), guspnp(4), neo(4), sb(4), sv(4), wss(4), yds(4), ym(4)

HISTORY

     The audiorecord program was first seen in SunOS 5.  It was first made available in NetBSD 1.4.  RIFF/WAVE support, and support for converting
     signed/unsigned and big/little-endian samples was first made available in NetBSD 1.6.

AUTHORS

     The audiorecord program was written by Matthew R. Green <mrg@eterna.com.au>.

BUGS

     WAV big-endian samples are converted to little-endian, rather than a RIFX header being written.

BSD
								 December 30, 2010							       BSD