Original commit message from CVS:
* ext/alsa/gstalsa.c: (gst_alsa_get_type), (gst_alsa_class_init),
(gst_alsa_get_property), (gst_alsa_probe_get_properties),
(gst_alsa_class_probe_devices), (gst_alsa_class_list_devices),
(gst_alsa_probe_probe_property), (gst_alsa_probe_needs_probe),
(gst_alsa_probe_get_values), (gst_alsa_probe_interface_init),
(gst_alsa_open_audio), (gst_alsa_close_audio):
* ext/alsa/gstalsa.h:
Add propertyprobe interface implementation, add some device-name
property, all this so that it looks good in gnome-volume-control.
Original commit message from CVS:
2004-02-14 Benjamin Otte <in7y118@public.uni-hamburg.de>
* ext/alsa/gstalsa.c: (gst_alsa_pcm_wait),
(gst_alsa_xrun_recovery):
* ext/alsa/gstalsa.h:
try xrun recovery when wait failed. Make xrun recovery function
return TRUE/FALSE to indicate success. (might fix#134354)
Original commit message from CVS:
2004-02-05 Benjamin Otte <in7y118@public.uni-hamburg.de>
* ext/alsa/gstalsa.c: (gst_alsa_change_state):
be sure to stop the clock when going to paused
* sys/oss/gstosssink.c: (gst_osssink_change_state):
reset number of transmitted when going to ready.
fixes#132935
2004-02-05 Charles Schmidt <cschmidt2@emich.edu>
reviewed by Benjamin Otte
* ext/mad/gstid3tag.c: (gst_mad_id3_to_tag_list):
extract track count (fixes#133410)
Original commit message from CVS:
2004-01-31 Benjamin Otte <in7y118@public.uni-hamburg.de>
* ext/alsa/gstalsa.c: (gst_alsa_change_state), (gst_alsa_start),
(gst_alsa_drain_audio), (gst_alsa_stop_audio):
really start/stop clock only on PLAYING <=> PAUSED
* ext/alsa/gstalsasink.c: (gst_alsa_sink_loop):
remove \n from debugging lines
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_chain):
make it work when seeking does not
* ext/vorbis/vorbisdec.c: (vorbis_dec_event):
reset on DISCONT
Original commit message from CVS:
2004-01-31 Benjamin Otte <in7y118@public.uni-hamburg.de>
* ext/alsa/gstalsa.c: (gst_alsa_change_state), (gst_alsa_start):
start clock on PAUSED=>PLAYING, not later
* ext/alsa/gstalsasink.c: (gst_alsa_sink_check_event):
extract correct time for different discont formats
(gst_alsa_sink_get_time):
don't segfault when no format is negotiated yet, just return 0
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_src_event),
(gst_ogg_demux_handle_event), (gst_ogg_demux_push),
(gst_ogg_pad_push):
handle flush and discont events correctly
* ext/vorbis/vorbisdec.c: (vorbis_dec_event), (vorbis_dec_chain):
handle discont events correctly
Original commit message from CVS:
2004-01-28 Benjamin Otte <in7y118@public.uni-hamburg.de>
* ext/alsa/gstalsa.c: (gst_alsa_query_func):
use gst_element_get_time to get correct time
Original commit message from CVS:
2003-12-27 Benjamin Otte <in7y118@public.uni-hamburg.de>
* ext/alsa/gstalsa.c: (gst_alsa_open_audio):
Don't send ALSA debugging to stderr.
* ext/alsa/gstalsa.h:
Use GST_WARNING instead of g_warning when ALSA functions fail.
Original commit message from CVS:
2003-12-22 Benjamin Otte <in7y118@public.uni-hamburg.de>
* ext/alsa/gstalsa.c: (gst_alsa_get_caps), (gst_alsa_link):
Fix remaining caps handling errors due to CAPS merge.
Original commit message from CVS:
New typefind system:
* bytestream is now part of the core
* all plugins have been modified to use this new typefind system
* asf typefinding added
* mpeg video stream typefiding removed because it's broken
* duplicate typefind entries removed
* extra id3 typefinding added, because we've seen 4 types of files
(riff/wav, flac, vorbis, mp3) with id3 headers and each of these needs
to work. Instead, I've added an id3 element and let it redo typefiding
after the id3 header. this needs a hack because spider only typefinds
once. We can remove this hack once spider supports multiple typefinds.
* with all this, mp3 typefinding is semi-rewritten
* id3 typefinding in flac/vorbis is removed, it's no longer needed
* fixed spider and gst-typefind to use this, too.
* Other general cleanups
Original commit message from CVS:
New mimetypes gone into effect today - this commit changes all old mimetypes over to the new mimetypes spec as described in the previous commit's document. Note: some plugins will break, some pipelines will break, expect HEAD to be broken or at least not 100% working for a few days, but don't forget to report bugs
Original commit message from CVS:
compatibility fix for new GST_DEBUG stuff.
Includes fixes for missing includes for config.h and unistd.h
I only ensured for plugins I can build that they work, so if some of them are still broken, you gotta fix them yourselves unfortunately.
Original commit message from CVS:
next big bunch of stuff:
- proper caps setting in alsasrc
- query / conversion functions
WARNING: Alsa 0.9.2 is heavily borked wrt recording - expect segfaults
Original commit message from CVS:
bugfixes:
- better error reporting
- segfault when using alsasrc without alsasink (d'oh)
- don't try to round when doing samples => time conversion
Original commit message from CVS:
total code reorganization as a start to get alsasrc working - sink and src are now really different classes, not just on paper - includes a fix that makes the testsuite work that might be an older bug
Original commit message from CVS:
fix clock - seeking, xruns etc should be handled correctly now
includes bugfix to not play the rest of the audio buffer when going PAUSED => READY
Original commit message from CVS:
fix timestamp syncing
timestamps are only guessed so add a (big) threshold before starting to drop/insert
fix some clocking madness
Original commit message from CVS:
ALSA rewrite, part 5:
- sync to timestamps (which breaks a _lot_, because most plugins send out wrong timestamps)
- clocking support (A/V sync is superb as long as you don't sync and don't get wrong timestamps)
- 1/2 of format conversion
- assorted bugfixes
I'd like to get people to check the timestamps the plugins send out.
mpegdemux seems to be pretty broken, mad works (I just patched it...), avidemux works at least sometimes.
Haven't checked more so far.
