Original commit message from CVS:
* gst/asfdemux/gstasfdemux.c: (_read_var_length), (_read_guid),
(gst_asf_demux_process_segment), (gst_asf_demux_handle_data),
(gst_asf_demux_process_chunk), (gst_asf_demux_handle_sink_event):
Prevent infinite loops. More correct error reporting.
* gst/auparse/gstauparse.c: (gst_auparse_chain):
Error out if negotiation fails.
* gst/playback/gstplaybasebin.c: (setup_source),
(gst_play_base_bin_change_state), (gst_play_base_bin_error),
(gst_play_base_bin_found_tag):
Error/tag forwarding. Pre-roll fixes for source errors on state
changes (e.g. "file does not exist") to prevent hangs.
Original commit message from CVS:
* ext/mad/gstmad.c: (gst_mad_check_caps_reset),
(gst_mad_change_state):
Allow for mp3 rate/channels changes. However, only very
conservatively. Reason that we *have* to enable this is smiply
because the mad find_sync() function is not good enough, it will
regularly sync on random data as valid frames and therefore make
us provide random caps as *final* caps of the stream. The best fix
I could think of is to simply require several of the same stream
changes in a row before we change caps.
The actual testcase that works now is #
* ext/ogg/Makefile.am:
* ext/ogg/gstogg.c: (plugin_init):
* ext/ogg/gstogmparse.c:
OGM support (video only for now; I need an audio sample file).
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_base_init),
(gst_asf_demux_process_stream), (gst_asf_demux_video_caps),
(gst_asf_demux_add_video_stream):
WMV extradata.
* gst/playback/gstplaybasebin.c: (unknown_type):
Don't error out on single unknown-types after all. It's wrong.
If we found type of video and audio but not of a subtitle stream,
it will still error out (which is unwanted). Will find a better fix
later on.
* gst/typefind/gsttypefindfunctions.c: (ogmvideo_type_find),
(ogmaudio_type_find), (plugin_init):
OGM support.
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.
