Original commit message from CVS:
* sys/oss/gstosssink.c: (gst_osssink_chain):
And another caller that couldn't handle delay < 0 (unsigned
integer overflow). Video now continues playing on an audio
buffer underrun, and the clock continues working. Audio still
stalls.
Original commit message from CVS:
* sys/oss/gstosssink.c: (gst_osssink_get_delay),
(gst_osssink_get_time):
get_delay() may return values lower than 0. In those cases, we
should not actually cast to *unsigned* int64, that will break
stuff horribly. In my case, it screwed up A/V sync in movies
in totem rather badly.
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:
2004-01-27 Benjamin Otte <in7y118@public.uni-hamburg.de>
* sys/oss/gstosssink.c: (gst_osssink_sink_query):
use gst_element_get_time to get correct time
Original commit message from CVS:
2004-01-15 Benjamin Otte <in7y118@public.uni-hamburg.de>
* ext/alsa/gstalsasink.c: (gst_alsa_sink_check_event):
Don't update the time of the clock
(gst_alsa_sink_loop):
sync to the clock given to alsasink, not the own clock
* sys/oss/gstosssink.c: (gst_osssink_chain):
sync to the clock
(gst_osssink_change_state):
activate the clock
* sys/ximage/ximagesink.c: (gst_ximagesink_chain):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_chain):
remove bogus code that made DISCONT events unhandled
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_video_caps):
explicitly case to double in _set_simple. (fixes 2nd warning in bug
#131502)
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_read_object_header),
(gst_asf_demux_handle_sink_event), (gst_asf_demux_audio_caps),
(gst_asf_demux_add_audio_stream), (gst_asf_demux_video_caps):
convert g_warning because of wrong asf data to GST_WARNINGs (fixes
2nd warning in bug #131502)
Original commit message from CVS:
2004-01-07 Benjamin Otte <in7y118@public.uni-hamburg.de>
* sys/oss/gstosssink.c: (gst_osssink_sink_fixate):
Fix for bug shown by poisoning
Original commit message from CVS:
Fix some clocking issue in OSS. The issue is that if we seek forward (note: specifically forward-only), then we call handle_discont() before re-setting the clock to active. However, gstclock.c tells us that handle_discont only succeeds if allow_discont=TRUE, which is set in... set_active(TRUE). So, we first need to re-activate the clock and *then* call handle_discont(). More importantly, though, we should **NEVER EVER EVER EVER EVER** **NEVER EVER EVER EVER EVER** call clock_wait() after a forward discont without first having called handle_discont(). I don't know who added that code, but it's beyond fundamentally broken. clock_wait() **WAITS** until we're at the new given buftime, so if we do that on a forward-seek buffer, we... yes! we wait the amount of time that we seeked forward. Anyway, Apparently this code has been in here for quite a long time so I don't get how this can ever have worked...
Original commit message from CVS:
first bunch of conversions to new plugin_init. Includes libs/gst, gst/id3, sys/oss, ext/gnomevfs, gst/typefind and ext/mad.
You guessed it, everything Rhythmbox needs ;)
fixed BMP typefind and made gnomevfs one plugin instead of two while doing this
Original commit message from CVS:
Interface implementation example: OSS mixer. Also osscommon->osselement so it can be loaded without being a source/sink (for a stand-alone mixer)
Original commit message from CVS:
* actually recurse into sndfile if we are able
* big ladspa cleanups, mainly to comply with the buffer-frames caps property, but also general
cleanups
- the samplerate prop is gone, if you want to set it explicitly (as in for get-based plugins)
you need to use a filtered connection, just like with buffer-frames
* big float2int and int2float changes for buffer-frames compatibility - I think it's quite a bit
simpler
* make the ossclock general, add it to gstaudio, and use it in sndfile as well
i need to update mimetypes, but that's coming soon. there are some other plugins that don't
support buffer-frames, i guess i need to get around to fixing them as well.
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:
implemented wait_async and unschedule ossclock, and support it in osssink -- really should make this a general clock, ill need it in gstsf
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:
- UNITS -> DEFAULT
- added chunk_size option to osssink, buffers will be written to the
devive in chunks of this size, this can increase the accuracy of the
clock on some devices.
Original commit message from CVS:
another batch of connect->link fixes
please let me know about issues
and please refrain of making them yourself, so that I don't spend double
the time resolving conflicts
