Original commit message from CVS:
* ext/alsa/gstalsa.c: (gst_alsa_get_property),
(gst_alsa_open_audio), (gst_alsa_close_audio):
* ext/alsa/gstalsa.c:
Don't open the device if we're a mixer (= padless).
* ext/alsa/gstalsamixer.c: (gst_alsa_mixer_class_init),
(gst_alsa_mixer_init), (gst_alsa_mixer_open),
(gst_alsa_mixer_close), (gst_alsa_mixer_change_state):
Open mixer during state change rather than during object
initialization. Also, get a device name. Currently in a somewhat
hackish fashion, but I didn't really find something better.
Original commit message from CVS:
* ext/alsa/gstalsamixer.c: (gst_alsa_mixer_free_list):
* ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_class_init),
(gst_alsa_mixer_track_new):
* ext/alsa/gstalsamixertrack.h:
Fix ancient leftovers... MixerTrack is a GObject.
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:
* 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:
* ext/alsa/Makefile.am: Fix linking against libgstinterfaces.
(bug #133886) Noticed by bugs@leroutier.net (Stephane LOEUILLET)
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:
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:
Remove all usage of gst_pad_get_caps(), and replace it with
gst_pad_get_allowed_caps() or gst_pad_get_negotiated_cap().
Original commit message from CVS:
2004-01-03 Thomas Canty <tommydal@optushome.com.au>
reviewed by: Ronald Bultje <rbultje@ronald.bitfreak.net>
* ext/alsa/gstalsamixer.c: (gst_alsa_mixer_dispose):
Correct logic of dispose function (see #129306).
Original commit message from CVS:
* ext/alsa/gstalsasrc.c: (gst_alsa_src_pad_factory),
(gst_alsa_src_base_init): Remove bogus "src" request pad.
* gst/mpegstream/gstmpegparse.c: (gst_mpeg_parse_base_init),
(gst_mpeg_parse_class_init): Move pad template registration
to class_init, since the derived class (mpegdemux) doesn't
want them.
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
