Original commit message from CVS:
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_init),
(gst_asf_demux_loop), (gst_asf_demux_process_file),
(gst_asf_demux_process_data), (gst_asf_demux_handle_data),
(gst_asf_demux_process_object), (gst_asf_demux_get_stream),
(gst_asf_demux_process_chunk), (gst_asf_demux_handle_sink_event),
(gst_asf_demux_handle_src_event), (gst_asf_demux_handle_src_query),
(gst_asf_demux_change_state):
* gst/asfdemux/gstasfdemux.h:
You know Chimaira? "I - HATE - EVERYTHING". Yeah, that's what this
feels like. I think we should set a new requirement for demuxers
from now on to implement sane loop functions, data loops, query
and seek functions before first commit into CVS. And this commit
fixes all of the above.
Original commit message from CVS:
* gst/asfdemux/gstasfdemux.c:
- fix a mem leak and always propagate tags
- add WMV3 to known video codecs (but no decoder yet)
- replace "surplus data" at end of audio header for what
it is : codec specific data
- fix a typo
Original commit message from CVS:
* gst/asfdemux/gstasfdemux.c :
- freeing the event also frees its associated taglist,
that's unfortunate as we need taglist after that. fixed
Original commit message from CVS:
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_process_comment):
don't write to memory we might not write to - g_convert does that
for us anyway
(gst_asf_demux_audio_caps):
conmment out gst_util_dump_mem
Original commit message from CVS:
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_process_comment):
Fix some odd cases and fix BE metadata parsing of unicode16 text.
Original commit message from CVS:
* ext/jpeg/gstjpegenc.c: (gst_jpegenc_get_type),
(gst_jpegenc_chain):
fix DURATION on outgoing buffers
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_handle_sink_event):
debug using time formats
* sys/ximage/ximagesink.c: (gst_ximagesink_xcontext_get):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support),
(gst_xvimagesink_sink_link):
windows with width/height 0 generate X errors, so don't allow them
Original commit message from CVS:
* gst/asfdemux/Makefile.am:
* gst/asfdemux/asfheaders.c:
* gst/asfdemux/asfheaders.h:
* gst/asfdemux/gstasf.c: (plugin_init):
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_get_type),
(gst_asf_demux_base_init), (gst_asf_demux_process_comment),
(gst_asf_demux_setup_pad):
* gst/asfdemux/gstasfdemux.h:
* gst/asfdemux/gstasfmux.c:
* gst/asfdemux/gstasfmux.h:
Add tagging support to demuxer, split out registration in its own
file instead of in demux (hacky), and prevent having some tables
in our memory multiple times (in asfheaders.h).
Original commit message from CVS:
* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnomevfssrc_dispose):
actually free the URI string
* ext/mad/gstid3tag.c: (gst_id3_tag_src_event):
compute offset correctly when passing discont events
* ext/mad/gstid3tag.c: (gst_id3_tag_handle_event):
don't leak discont events
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_video_caps):
add some missing breaks so caps aren't copied randomly
* gst/mpegstream/gstmpegdemux.c: (gst_mpeg_demux_get_video_stream):
if we realloc memory, we better use it
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.
Original commit message from CVS:
* gst/mpegstream/gstrfc2250enc.c: (gst_rfc2250_enc_add_slice):
Fix code that ignores return value of gst_buffer_merge().
(bug #114560)
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_descramble_segment):
* gst/mpegstream/gstrfc2250enc.c: (gst_rfc2250_enc_add_slice): same
* testsuite/gst-lint: Check for above.
Original commit message from CVS:
2004-02-27 Benjamin Otte <otte@gnome.org>
* gst-libs/gst/audio/audio.h:
add macro to make sure header isn't included twice
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_process_chunk):
don't use gst_buffer_free
* gst/playondemand/filter.func:
don't usae gst_data_free. Free data only once.
Original commit message from CVS:
2004-02-20 Andy Wingo <wingo@pobox.com>
* gst/intfloat/, gst/oneton: Removed, replaced by audioconvert and
interleave respectively.
* gst/interleave/deinterleave.c: New plugin: deinterleave
(replaces on oneton).
* gst/interleave/interleave.c: New plugin: interleave.
* gst/interleave/plugin.h: Support file.
* gst/interleave/plugin.c: Support file.
* configure.ac: Remove intfloat and oneton, add interleave.
* ext/sndfile/gstsf.c: Handle events better.
* gst/audioconvert/gstaudioconvert.c: Change to support int2float
and float2int operation. int2float has scheduling problems as
noted in in2float_chain.
Original commit message from CVS:
2004-01-27 Benjamin Otte <in7y118@public.uni-hamburg.de>
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_handle_sink_event):
stop processing after EOS
Original commit message from CVS:
2004-01-27 Benjamin Otte <in7y118@public.uni-hamburg.de>
* gst/asfdemux/asfheaders.h:
* gst/asfdemux/gstasfdemux.c:
* gst/asfdemux/gstasfmux.c: (gst_asfmux_put_guid),
(gst_asfmux_put_string), (gst_asfmux_put_wav_header),
(gst_asfmux_put_vid_header), (gst_asfmux_put_bmp_header):
lot's of fixes to make data extraction simpler and get the code
architecture and compiler independant. Add debugging category
* gst/goom/gstgoom.c: (gst_goom_change_state):
reset channel count on PAUSED=>READY, not READY=>PAUSED
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:
fixes for ASF:
- merge asfdemux and asfmux into one plugin
- make gstasf a plugin and not a lib (it accidently was one before)
Original commit message from CVS:
Sorry Dave... Add mpegversion=1 to mp3 caps everywhere so that the autoplugger uses mad and not faad for mp3 decoding. This should fix mp3 playback.
Original commit message from CVS:
Riff, EBML, fourcc etc. work. Not fully finished, but better than
what we used to have and definately worth a first broad testing.
I've revived rifflib. Rifflib used to be a bytestream-for-riff, which
just dup'ed bytestream. I've rewritten rifflib to be a modern riff-
chunk parser that uses bytestream fully, plus adds some extra functions
so that riff file parsing becomes extremely easy. It also contains some
small usability functions for strh/strf and metadata parsing. Note that
it doesn't use the new tagging yet, that's a TODO.
Avidemux has been rewritten to use this. I think we all agreed that
avidemux was pretty much a big mess, which is because it used all
sort of bytestream magic all around the place. It was just ugly.
This is a lot nicer, very complete and safe. I think this is far more
robust than what the old avidemux could ever have been. Of course, it
might contain bugs, please let me know.
EBML writing has also been implemented. This is useful for matroska.
I'm intending to modify avidemux (with a riffwriter) similarly. Maybe
I'll change wavparse/-enc too to use rifflib.
Lastly, several plugins have been modified to use rifflib's fourcc
parsing instead of their own. this puts fourcc parsing in one central
place, which should make it a lot simpler to add new fourccs. We might
want to move this to its own lib instead of rifflib.
Enjoy!
