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gstreamer/ext/audiofile
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Ronald S. Bultje 1b63854c1f Fix for instantiate-test (see core). Also remove dead code from jpegenc (which still needs fixing, but that's lower o... 
Original commit message from CVS:
2004-01-07  Ronald Bultje  <rbultje@ronald.bitfreak.net>

* ext/audiofile/gstafsink.c: (gst_afsink_init), (gst_afsink_chain),
(gst_afsink_handle_event):
* ext/jpeg/gstjpegenc.c: (gst_jpegenc_init):
* gst/avi/gstavimux.c: (gst_avimux_request_new_pad):
* sys/dxr3/dxr3audiosink.c: (dxr3audiosink_init):
* sys/dxr3/dxr3spusink.c: (dxr3spusink_init):
* sys/dxr3/dxr3videosink.c: (dxr3videosink_init):
Fix for instantiate-test (see core). Also remove dead code from
jpegenc (which still needs fixing, but that's lower on my TODO
list...).
* sys/v4l2/gstv4l2src.c: (gst_v4l2src_getcaps):
Never return NULL as caps.
2004-01-07 13:18:08 +00:00
..
gstaf.c remove copyright field from plugins 2003-12-04 10:37:38 +00:00
gstafparse.c Convert elements to use gst_pad_use_explicit_caps() where appropriate. 2004-01-02 07:09:23 +00:00
gstafparse.h Remove all config.h includes from header files, add it to each source file and remove duplicate config.h includes fro... 2003-11-07 12:47:02 +00:00
gstafsink.c Fix for instantiate-test (see core). Also remove dead code from jpegenc (which still needs fixing, but that's lower o... 2004-01-07 13:18:08 +00:00
gstafsink.h Remove all config.h includes from header files, add it to each source file and remove duplicate config.h includes fro... 2003-11-07 12:47:02 +00:00
gstafsrc.c Convert elements to use gst_pad_use_explicit_caps() where appropriate. 2004-01-02 07:09:23 +00:00
gstafsrc.h Remove all config.h includes from header files, add it to each source file and remove duplicate config.h includes fro... 2003-11-07 12:47:02 +00:00
Makefile.am remove audiofile typefinding because it is buggy and we support all of its formats anyway. 2003-11-03 19:18:36 +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