Original commit message from CVS:
* ext/alsa/gstalsa.c: (gst_alsa_class_probe_devices):
* sys/oss/gstosselement.c: (gst_osselement_class_probe_devices):
Don't block during probing...
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:
2004-02-29 Christophe Fergeau <teuf@gnome.org>
* sys/oss/gstosselement.c: (gst_osselement_probe),
(device_combination_append), (gst_osselement_class_probe_devices):
* sys/oss/gstosselement.h:
Reworked enumeration of oss dsps and mixers so that gst-mixer works
on my system using alsa oss emulation, fixes bug #135597
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:
2004-01-22 Ronald Bultje <rbultje@ronald.bitfreak.net>
* sys/oss/gstosselement.c: (gst_osselement_class_probe_devices):
Fix the ossmixer case where we shouldn't open /dev/dsp* because
it might block operations (which is bad for a mixer).
Original commit message from CVS:
2004-01-15 Julien MOUTTE <julien@moutte.net>
* ext/alsa/gstalsamixer.c: (gst_alsa_mixer_interface_init): Setting
mixer interface type to HARDWARE.
* gst-libs/gst/mixer/mixer.c: (gst_mixer_class_init): Adding a default
type to SOFTWARE.
* gst-libs/gst/mixer/mixer.h: Adding mixer interface type and macro.
* gst-libs/gst/mixer/mixertrack.h: Adding mixertrack flag SOFTWARE.
* gst/volume/gstvolume.c: (gst_volume_interface_supported),
(gst_volume_interface_init), (gst_volume_list_tracks),
(gst_volume_set_volume), (gst_volume_get_volume),
(gst_volume_set_mute), (gst_volume_mixer_init),
(gst_volume_dispose), (gst_volume_get_type), (volume_class_init),
(volume_init): Implementing mixer interface.
* gst/volume/gstvolume.h: Adding tracklist for mixer interface.
* sys/oss/gstosselement.c: (gst_osselement_get_type),
(gst_osselement_change_state): Removing some trailing commas in
structures.
* sys/oss/gstossmixer.c: (gst_ossmixer_interface_init): Setting mixer
interface type to HARDWARE.
* sys/v4l/gstv4lcolorbalance.c:
(gst_v4l_color_balance_interface_init): Setting colorbalance interface
type to HARDWARE.
* sys/v4l2/gstv4l2colorbalance.c:
(gst_v4l2_color_balance_interface_init): Setting colorbalance
interface type to HARDWARE.
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_chain): use exactly the
same code than ximagesink for event handling.
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:
2004-01-06 Ronald Bultje <rbultje@ronald.bitfreak.net>
* ext/shout/gstshout.c: (gst_icecastsend_base_init),
(gst_icecastsend_init):
fix for new caps system.
* gst-libs/gst/mixer/mixertrack.h:
* sys/oss/gstossmixer.c: (gst_ossmixer_build_list):
Add 'master track' flag (for tools like ACME that only want to
change the main volume).
Original commit message from CVS:
2003-12-21 Ronald Bultje <rbultje@ronald.bitfreak.net>
* configure.ac:
Improve mpeg2enc detection. This is for distributions that do
ship mjpegtools, but without mpeg2enc. Also does object check
for might there ever be ABI incompatibility.
* ext/mpeg2enc/gstmpeg2enc.cc:
Add Andrew as second maintainer (he's helping me), and also add
an error if no caps was set. This happens if I pull before capsnego
and that's something I should solve sometime else.
* gst/matroska/matroska-demux.c:
(gst_matroska_demux_parse_blockgroup):
Fix time parsing.
* gst/matroska/matroska-mux.c: (gst_matroska_mux_audio_pad_link),
(gst_matroska_mux_track_header):
Add caps to templates.
* gst/mpegaudioparse/gstmpegaudioparse.c: (mp3_sink_factory):
Add mpegversion=1 to prevent confusion with MPEG/AAC.
* gst/mpegstream/gstmpegdemux.c:
Remove layer since it causes warnings about unfixed caps.
* gst/videotestsrc/gstvideotestsrc.c: (gst_videotestsrc_get):
Fix obvious typo (we error out if caps were set, we should of
course error out if *no* caps were set).
* sys/oss/gstosselement.c: (gst_osselement_convert):
Fix format conversion, we confused bits/bytes.
* sys/oss/gstosselement.h:
Improve documentation for 'bps'.
* sys/v4l/TODO:
Remove stuff about plugins that need removing - this was done
ages ago.
* sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_init),
(gst_v4lmjpegsrc_src_convert), (gst_v4lmjpegsrc_src_query):
* sys/v4l/gstv4lsrc.c: (gst_v4lsrc_init), (gst_v4lsrc_src_convert),
(gst_v4lsrc_src_query):
* sys/v4l2/gstv4l2src.c: (gst_v4l2src_init),
(gst_v4l2src_src_convert), (gst_v4l2src_src_query):
Add get_query_types(), get_formats() and query() functions.
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:
Remove ossgst... It's a crude hack (beyond ugly), it's broken and it failed to load during the last few months. If anyone wants to revive it, have fun finding it back in the CVS history
Original commit message from CVS:
Fix device probing from multiple childs. It's done once in the parent class only now, but the childs do get the correct values. Also fixes an incorrect succesful state change if we opened a v4l device that doesn't have the capabilities that are needed by the plugin.
Original commit message from CVS:
Don't change mixer if there's nothing to change. This caused a bug if the soundcard only supports one input and I call this function with rec=TRUE twice.
Original commit message from CVS:
Remove all config.h includes from header files, add it to each source file and remove duplicate config.h includes from several source files
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:
Rename osselement to ossmixer and only open audio device if we have at least one pad. This makes ossmixer *only* open the mixer, which means we can open multiple mixer sessions.
Original commit message from CVS:
Make GstMixerTrack a GObject. I also want to make it emit several signals, starting work is in here but it's not fully implemented yet. for OSS, this will cause issues, but for ALSA, this is all automated.
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:
* implement clocking
* set clock counter back to zero on ready->paused
* move open/close to ready/null instead of paused/ready.
* add random typos
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
Original commit message from CVS:
Make the OSSSrc set the correct rate/number of channels as set in the GstCaps.
Handle state changes correctly according to docs/random/wtay/states
