Original commit message from CVS:
2005-08-23 Andy Wingo <wingo@pobox.com>
* sys/oss/gstossmixer.h:
* sys/oss/gstossmixer.c: Refactored to be more like alsamixer.
* sys/oss/gstossmixertrack.h:
* sys/oss/gstossmixertrack.c: Split out from gstossmixer.[ch],
like gstalsamixer.
* sys/oss/gstosssrc.c:
* sys/oss/gstosssink.c: Where before we used a gstosselement
object as a helper library, now just call functions from
gstosshelper.
* sys/oss/gstosshelper.h:
* sys/oss/gstosshelper.c: Made a real library. Removed
propertyprobe for now, should add it back later.
* sys/oss/gstosselement.h:
* sys/oss/gstosselement.c: Removed, we don't have a shared base
class.
* sys/oss/gstosshelper.c (gst_oss_helper_probe_caps): Search
higher-to-lower, makes 16 bit appear earlier in the caps, which
makes it preferred.
Original commit message from CVS:
2005-08-08 Andy Wingo <wingo@pobox.com>
* sys/oss/gstosssink.c (gst_oss_sink_open, gst_oss_sink_close)
(gst_oss_sink_prepare, gst_oss_sink_unprepare): Update for newer
audiosink api.
* ext/raw1394/gstdv1394src.c (gst_dv1394src_get_property)
(gst_dv1394src_set_property): Style. All about the style.
* ext/esd/esdsink.c (gst_esdsink_getcaps): Return specific caps
only if in READY or higher (i.e., if _open() has been called.)
(gst_esdsink_open, gst_esdsink_close, gst_esdsink_prepare)
(gst_esdsink_unprepare): Update for audiosink changes.
(gst_esdsink_change_state): Die!
Original commit message from CVS:
* ext/gdk_pixbuf/pixbufscale.c: (gst_pixbufscale_getcaps):
Correct caps negotiation
* gst/volume/gstvolume.c: (volume_chain_float),
(volume_chain_int16):
Modify debug output to be little more informative
* sys/ximage/ximagesink.c: (gst_ximagesink_check_xshm_calls):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_check_xshm_calls),
(gst_xvimagesink_xvimage_destroy):
Add XSync calls after detaching from the shared memory segment to
avoid a crash.
Original commit message from CVS:
* sys/oss/gstosssink.c:
* sys/oss/gstosssrc.c:
advertise correct template caps - we indeed do non-native endianness
and 8bit audio has no endianness
* sys/ximage/ximagesink.c: (gst_ximagesink_getcaps):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_getcaps):
avoid (wrong) duplications in getcaps function and return
template caps
Original commit message from CVS:
* configure.ac: OSS portability
* ext/arts/gst_arts.c: idem
* sys/oss/gstosselement.c: idem
* sys/oss/gstossmixer.c: idem
* sys/oss/gstosssink.c: idem
* sys/oss/gstosssrc.c: idem
* sys/oss/oss_probe.c: idem
- check for soundcard.h in different places for some BSD
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_init),
(gst_qtdemux_handle_src_query), (gst_qtdemux_handle_src_event),
(gst_qtdemux_handle_sink_event), (gst_qtdemux_change_state),
(gst_qtdemux_loop_header), (qtdemux_dump_mvhd),
(qtdemux_parse_trak):
* gst/qtdemux/qtdemux.h:
Bitch. Also known as seeking, querying & co.
* sys/oss/gstosssink.c: (gst_osssink_init), (gst_osssink_chain),
(gst_osssink_change_state):
* sys/oss/gstosssink.h:
Resyncing is for weenies, this hack is no longer needed and was
broken anyway (since it - unintendedly - always leaves resync to
TRUE).
Original commit message from CVS:
fourth batch :
remove ',' at end of enums as they could confuse older gcc, foreign compilers (forte) and gtk-doc
(in gst-plugins/sys/ this time)
Original commit message from CVS:
* sys/oss/gstosssink.c: (gst_osssink_init), (gst_osssink_get_time),
(gst_osssink_chain), (gst_osssink_change_state):
Latest fixes for A/V sync, audio playback and such. This is about
all... MPEG playback issues are mostly related to the async build-
up of MPEG files, I cannot fix that. Use basicgthread to solve it.
Original commit message from CVS:
* ext/divx/gstdivxdec.c:
Downgrade priority. We prefer ffdec_mpeg4.
* ext/faad/gstfaad.c: (gst_faad_srcgetcaps), (gst_faad_srcconnect),
(gst_faad_chain), (gst_faad_change_state):
Fix capsnego. Doesn't work for some sounds because we don't have
a 5:1 to stereo element.
* ext/xvid/gstxvid.c: (plugin_init):
Add priority.
* sys/oss/gstosssink.c: (gst_osssink_init), (gst_osssink_chain),
(gst_osssink_change_state):
Add discont handling.
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.
