Original commit message from CVS:
* ext/speex/gstspeexdec.c: (gst_speexdec_base_init),
(gst_speexdec_init):
* ext/speex/gstspeexenc.c: (gst_speexenc_base_init),
(gst_speexenc_init): Create the pad template correctly (from
the static pad template, not a NULL pointer.)
Original commit message from CVS:
* gst/debug/Makefile.am:
* gst/debug/breakmydata.c:
add element that quasi-randomly changes bytes in the stream.
Intended use is robustness checking of demuxers and decoders in
media tests.
Original commit message from CVS:
* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnomevfssrc_set_property):
don't g_return_if_fail if element is PLAYING, fail silently as every
other element.
* gst/effectv/gstquark.c: (gst_quarktv_chain):
only fix needed for cast lvalue issues in gst-plugins
* gst/volenv/gstvolenv.c: (gst_volenv_init):
add proxy_getcaps
Original commit message from CVS:
2004-03-23 Jeremy Simon <jesimon@libertysurf.fr>
* gst/typefind/gsttypefindfunctions.c: (ape_type_find),
(plugin_init):
Add a monkeysaudio typefind function
Original commit message from CVS:
* gst-libs/gst/play/play.c (gst_play_audio_fixate)
(gst_play_video_fixate): Check so the structure has the field
before trying to fixate them, this makes it possible to have
fakesinks for video and audio output without printing errors on
the output console.
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_link),
(_fixate_caps_to_int), (gst_audio_convert_fixate):
add a fixation function that pretty much does the right thing (fixes
#137556)
Original commit message from CVS:
reviewed by: Benjamin Otte <otte@gnome.org>
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_push):
terminate gst_event_new_discontinuous correctly (fixes parts of
#137711)
Original commit message from CVS:
2004-03-15 Julien MOUTTE <julien@moutte.net>
* gst-libs/gst/play/play.c: (gst_play_get_framerate),
(gst_play_get_sink_element): First draft of gst_play_get_framerate.
* gst-libs/gst/play/play.h:
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:
* gst/modplug/gstmodplug.cc:
handle events - don't do crap when a discont arrives that's not
necessary
This allows correct loading and playback of mods in Rhythmbox
Original commit message from CVS:
* configure.ac:
* gst-libs/gst/gconf/Makefile.am:
* pkgconfig/Makefile.am:
move gstreamer-gconf pkgconfig files to pkgconfig/ dir. Make sure
they get rebuilt properly
* configure.ac:
when checking for vorbis, try pkgconfig first.
* gst/modplug/gstmodplug.cc:
add fixate function
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (qtdemux_parse_trak):
Fix for obvious mistake, where we first shift the offset and then
read a samplesize element assuming the old offset. Note that this
part still has something weird, i.e. my movies containing those
don't actually play well, but at least there's something that looks
like sound now.
Original commit message from CVS:
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_video_caps),
(gst_asf_demux_setup_pad):
Use 25fps as our "fake" fps value (marked for fixage in 0.9.x)
instead of 0. Reason is simple: some elements have a fps range
of 1-max instead of 0-max. So now ASF video actually works.
