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gstreamer/ext/audiofile
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Stefan Kost bf3fd4f95d Define GstElementDetails as const and also static (when defined as global) 
Original commit message from CVS:
* ext/amrwb/gstamrwbdec.c:
* ext/amrwb/gstamrwbenc.c:
* ext/amrwb/gstamrwbparse.c:
* ext/arts/gst_arts.c:
* ext/artsd/gstartsdsink.c:
* ext/audiofile/gstafparse.c:
* ext/audiofile/gstafsink.c:
* ext/audiofile/gstafsrc.c:
* ext/audioresample/gstaudioresample.c:
* ext/bz2/gstbz2dec.c:
* ext/bz2/gstbz2enc.c:
* ext/cdaudio/gstcdaudio.c:
* ext/directfb/dfbvideosink.c:
* ext/divx/gstdivxdec.c:
* ext/divx/gstdivxenc.c:
* ext/dts/gstdtsdec.c: (gst_dtsdec_base_init):
* ext/faac/gstfaac.c: (gst_faac_base_init):
* ext/faad/gstfaad.c:
* ext/gsm/gstgsmdec.c:
* ext/gsm/gstgsmenc.c:
* ext/hermes/gsthermescolorspace.c:
* ext/ivorbis/vorbisfile.c:
* ext/lcs/gstcolorspace.c:
* ext/libfame/gstlibfame.c:
* ext/libmms/gstmms.c: (gst_mms_base_init):
* ext/musepack/gstmusepackdec.c: (gst_musepackdec_base_init):
* ext/musicbrainz/gsttrm.c: (gst_musicbrainz_base_init):
* ext/nas/nassink.c: (gst_nassink_base_init):
* ext/neon/gstneonhttpsrc.c:
* ext/sdl/sdlaudiosink.c:
* ext/sdl/sdlvideosink.c:
* ext/shout/gstshout.c:
* ext/snapshot/gstsnapshot.c:
* ext/sndfile/gstsf.c:
* ext/swfdec/gstswfdec.c:
* ext/tarkin/gsttarkindec.c:
* ext/tarkin/gsttarkinenc.c:
* ext/theora/theoradec.c:
* ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_base_init):
* ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_base_init):
* ext/xvid/gstxviddec.c:
* ext/xvid/gstxvidenc.c:
* gst/cdxaparse/gstcdxaparse.c: (gst_cdxa_parse_base_init):
* gst/cdxaparse/gstcdxastrip.c: (gst_cdxastrip_base_init):
* gst/chart/gstchart.c:
* gst/colorspace/gstcolorspace.c:
* gst/deinterlace/gstdeinterlace.c:
* gst/equalizer/gstiirequalizer.c: (gst_iir_equalizer_base_init):
* gst/festival/gstfestival.c:
* gst/filter/gstbpwsinc.c:
* gst/filter/gstiir.c:
* gst/filter/gstlpwsinc.c:
* gst/freeze/gstfreeze.c:
* gst/games/gstpuzzle.c: (gst_puzzle_base_init):
* gst/librfb/gstrfbsrc.c:
* gst/mixmatrix/mixmatrix.c:
* gst/mpeg1sys/gstmpeg1systemencode.c:
* gst/mpeg1videoparse/gstmp1videoparse.c:
* gst/mpeg2sub/gstmpeg2subt.c:
* gst/mpegaudioparse/gstmpegaudioparse.c:
* gst/multifilesink/gstmultifilesink.c:
* gst/overlay/gstoverlay.c:
* gst/passthrough/gstpassthrough.c:
* gst/playondemand/gstplayondemand.c:
* gst/qtdemux/qtdemux.c:
* gst/rtjpeg/gstrtjpegdec.c:
* gst/rtjpeg/gstrtjpegenc.c:
* gst/smooth/gstsmooth.c:
* gst/smoothwave/gstsmoothwave.c:
* gst/spectrum/gstspectrum.c:
* gst/speed/gstspeed.c:
* gst/stereo/gststereo.c:
* gst/switch/gstswitch.c:
* gst/tta/gstttadec.c: (gst_tta_dec_base_init):
* gst/tta/gstttaparse.c: (gst_tta_parse_base_init):
* gst/vbidec/gstvbidec.c:
* gst/videocrop/gstvideocrop.c:
* gst/videodrop/gstvideodrop.c:
* gst/virtualdub/gstxsharpen.c:
* gst/xingheader/gstxingmux.c: (gst_xing_mux_base_init):
* gst/y4m/gsty4mencode.c:
* sys/cdrom/gstcdplayer.c:
* sys/directdraw/gstdirectdrawsink.c:
* sys/directsound/gstdirectsoundsink.c:
* sys/glsink/glimagesink.c:
* sys/qcam/gstqcamsrc.c:
* sys/v4l2/gstv4l2src.c:
* sys/vcd/vcdsrc.c: (gst_vcdsrc_base_init):
* sys/ximagesrc/ximagesrc.c:
Define GstElementDetails as const and also static (when defined as
global)
2006-04-25 21:56:38 +00:00
..
gstaf.c rework build; add translations for v4l2 2006-04-01 10:09:11 +00:00
gstafparse.c Define GstElementDetails as const and also static (when defined as global) 2006-04-25 21:56:38 +00:00
gstafparse.h expand tabs 2005-12-06 19:55:58 +00:00
gstafsink.c Define GstElementDetails as const and also static (when defined as global) 2006-04-25 21:56:38 +00:00
gstafsink.h expand tabs 2005-12-06 19:55:58 +00:00
gstafsrc.c Define GstElementDetails as const and also static (when defined as global) 2006-04-25 21:56:38 +00:00
gstafsrc.h expand tabs 2005-12-06 19:55:58 +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