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gstreamer/ext/audiofile
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Wim Taymans d18d4fd5a1 - Changed plugins for new APIs 
Original commit message from CVS:
- Changed plugins for new APIs
- modularized audiofile.
- added seeking, query and convert functions for mad, mpeg2dec,
avidemux, mpegdemux, mpegparse
- sync updates to oss. removed the ossclock for now
2002-05-26 21:59:21 +00:00
..
gstaf.c - Changed plugins for new APIs 2002-05-26 21:59:21 +00:00
gstafparse.c - Changed plugins for new APIs 2002-05-26 21:59:21 +00:00
gstafparse.h - Changed plugins for new APIs 2002-05-26 21:59:21 +00:00
gstafsink.c - Changed plugins for new APIs 2002-05-26 21:59:21 +00:00
gstafsink.h - Changed plugins for new APIs 2002-05-26 21:59:21 +00:00
gstafsrc.c - Changed plugins for new APIs 2002-05-26 21:59:21 +00:00
gstafsrc.h - Changed plugins for new APIs 2002-05-26 21:59:21 +00:00
gstaftypes.c use correct mime type for the types supported by audiofile 2002-04-26 12:42:32 +00:00
Makefile.am - Changed plugins for new APIs 2002-05-26 21:59:21 +00:00
README fixed some GST_LIBS stuff added audiofile added gst-libs/audio building 2001-12-21 11:46:15 +00:00

README 

This plugin wraps the SGI Audiofile 
(http://oss.sgi.com/projects/audiofile/) library into a src and sink
element.

You can read from and write to the supported formats (WAVE, AIFF, AIFFC,
NEXTSND).

What is supported :
* all the file formats
* integer sample data, both 2's complement and unsigned
* 8 or 16 bit width & depth (haven't tested others)
* sample rate
* some sort of endianness control

What isn't supported yet :
* float data

What you can do :
* src element only accepts location argument
* sink element accepts location, endianness and type
	- location : file on the system to output
	- endianness : at this time endianness is still a bit shady
	  	you can either set 1234 or 4321;
		setting it to 4321 will byteswap the buffer data
		you might want to keep it at 1234 for now
	- type : one of the file types

Use gstreamer-inspect on afsink and afsrc to see all of the supported
options.

Examples :

* tools/gstreamer-launch afsrc location=/opt/media/wav/dark-480-16-m.wav ! afsink type=2 location=/opt/media/wav/dark-480-16-m.aiff

Future plans :

* add float support
* wrap up afsink and afsrc with pipe and fork to act like data convertors,
  allowing arbitrary choice of sink and src element