Original commit message from CVS:
* gst-libs/gst/riff/riff-read.c: (gst_riff_read_use_event):
Don't bail on unknown events.
* gst/audioscale/gstaudioscale.c: (gst_audioscale_chain):
Don't crash on events before negotiation.
* gst/avi/gstavidemux.c: (gst_avi_demux_add_stream):
Send tags on pads, too.
* gst/playback/gststreamselector.c:
(gst_stream_selector_request_new_pad):
Forward events on first pad if no input was selected yet.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_stream_index),
(gst_avi_demux_stream_scan):
Add some more debug. Fix logic error when setting movi offset
while reading index.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_stream_index),
(gst_avi_demux_stream_scan), (gst_avi_demux_handle_seek),
(gst_avi_demux_process_next_entry):
Add some debugging. Better detection of broken indexes and the
accompanying index recovery. No infinite loops on state changes
when we're still in our loopfunction.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_stream_index):
Work for truncated (unfinished download etc.) files. Fixes#160514.
Original commit message from CVS:
* ext/a52dec/gsta52dec.c: (gst_a52dec_init),
(gst_a52dec_handle_event), (gst_a52dec_update_streaminfo),
(gst_a52dec_handle_frame), (gst_a52dec_chain),
(gst_a52dec_change_state), (plugin_init):
* ext/a52dec/gsta52dec.h:
Do something useful with timestamps. Make chain-based (since
there's really no reason to be loopbased).
* gst/avi/gstavidemux.c: (gst_avi_demux_process_next_entry):
Update current_byte/frame correctly.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_stream_index),
(gst_avi_demux_massage_index):
Fix quite humiliating bug in omitting 0-sized index chunks but
forgetting to count them for timestamps.
Original commit message from CVS:
* gst/avi/gstavimux.c: (gst_avimux_audsinkconnect),
(gst_avimux_stop_file):
First calculate the rate, and only then use it. Hdr.rate is a
multiple and not a derivative of hdr.scale. Scale is not the
same as blockalign but is solely related to rate.
Original commit message from CVS:
* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnomevfssrc_get),
(gst_gnomevfssrc_srcpad_query), (gst_gnomevfssrc_srcpad_event):
Some debug.
* gst/avi/gstavidemux.c: (gst_avi_demux_reset),
(gst_avi_demux_handle_src_event), (gst_avi_demux_read_superindex),
(gst_avi_demux_read_subindexes), (gst_avi_demux_add_stream),
(gst_avi_demux_stream_index), (gst_avi_demux_skip),
(gst_avi_demux_sync), (gst_avi_demux_stream_scan),
(gst_avi_demux_massage_index), (gst_avi_demux_stream_header):
* gst/avi/gstavidemux.h:
Support for openDML-2.0 indx/ix## chunks. Support for broken index
recovery (where, if part of the index is broken, we will still read
the rest of the index and recover the broken part by stream
scanning). More broken media support. EOS workarounds. General AVI
braindamage headache recovery. Aspirin included.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_massage_index):
OK, so the original code was too strict. It makes random AVI files
hang for seconds upon opening, which is unacceptable and is far
beyond the original goal of getting multiple chunks for one-chunk
sounc stream files. So now do just that.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_event):
Fix seeking in some files. All this code is no longer needed (and
actually breaks stuff) because we now synchronize the full index
right when reading the header.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_stream_index),
(gst_avi_demux_stream_scan), (sort), (gst_avi_demux_massage_index),
(gst_avi_demux_stream_header), (gst_avi_demux_stream_data):
Improve allocation, cutting and sorting of the index. How takes a
few seconds instead of minutes.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_video_caps_with_data),
(gst_riff_create_video_template_caps):
Add wing commander format mimetype/fourccs.
* gst/avi/gstavidemux.c: (gst_avi_demux_massage_index):
Don't crash if some value is 0.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_video_caps_with_data),
(gst_riff_create_video_template_caps):
Add DIB fourcc (raw, palettized 8-bit RGB).
* gst-libs/gst/riff/riff-read.c:
(gst_riff_read_strf_vids_with_data):
Oops, fix strf_data reading bug.
* gst/avi/gstavidemux.c: (gst_avi_demux_add_stream):
Use a non-NULL tag.
* gst/qtdemux/qtdemux.c: (qtdemux_parse_trak):
Time for hacks. Sorry Dave. At least one quicktime movie (a
trailer) that I've encountered contains multiple video tracks.
One of those is the actual video track, the other are one-frame
tracks (images). Unfortunately, the number of frames according
to the trak header is 1 for each, so that doesn't help. So
instead, I look at the duration and discard tracks with a
duration shorter than 20% of the length of the stream. Better
than nothing.
Original commit message from CVS:
* ext/flac/gstflacdec.c: (gst_flacdec_src_query):
Only return true if we actually filled something in. Prevents
player applications from showing a random length for flac files.
* gst-libs/gst/riff/riff-read.c: (gst_riff_read_class_init),
(gst_riff_read_use_event), (gst_riff_read_handle_event),
(gst_riff_read_seek), (gst_riff_read_skip), (gst_riff_read_strh),
(gst_riff_read_strf_vids_with_data),
(gst_riff_read_strf_auds_with_data), (gst_riff_read_strf_iavs):
OK, ok, so I implemented event handling. Apparently it's normal
that we receive random events at random points without asking
for it.
* gst/avi/gstavidemux.c: (gst_avi_demux_reset),
(gst_avi_demux_src_convert), (gst_avi_demux_handle_src_query),
(gst_avi_demux_handle_src_event), (gst_avi_demux_stream_index),
(gst_avi_demux_sync), (gst_avi_demux_stream_scan),
(gst_avi_demux_massage_index), (gst_avi_demux_stream_header),
(gst_avi_demux_handle_seek), (gst_avi_demux_process_next_entry),
(gst_avi_demux_stream_data), (gst_avi_demux_loop):
* gst/avi/gstavidemux.h:
Implement non-lineair chunk handling and subchunk processing.
The first solves playback of AVI files where the audio and video
data of individual buffers that we read are not synchronized.
This should not happen according to the wonderful AVI specs, but
of course it does happen in reality. It is also a prerequisite for
the second. Subchunk processing allows us to cut chunks in small
pieces and process each of these pieces separately. This is
required because I've seen several AVI files with incredibly large
audio chunks, even some files with only one audio chunk for the
whole file. This allows for proper playback including seeking.
This patch is supposed to fix all AVI A/V sync issues.
* gst/flx/gstflxdec.c: (gst_flxdec_class_init),
(flx_decode_chunks), (flx_decode_color), (gst_flxdec_loop):
Work.
* gst/modplug/gstmodplug.cc:
Proper return value setting for the query() function.
* gst/playback/gstplaybasebin.c: (setup_source):
Being in non-playing state (after, e.g., EOS) is not necessarily
a bad thing. Allow for that. This fixes playback of short files.
They don't actually playback fully now, because the clock already
runs. This means that small files (<500kB) with a small length
(<2sec) will still not or barely play. Other files, such as mod
or flx, will work correctly, however.
Original commit message from CVS:
* ext/dirac/Makefile.am:
* ext/dirac/gstdirac.cc:
* ext/dirac/gstdiracdec.cc:
* ext/dirac/gstdiracdec.h:
Do something. Don't actually know if this works because I don't
have a demuxer yet.
* ext/gsm/gstgsmdec.c: (gst_gsmdec_getcaps):
Add channels=1 to caps returned from _getcaps().
* ext/ogg/gstogmparse.c: (gst_ogm_audio_parse_get_type),
(gst_ogm_video_parse_get_type), (gst_ogm_audio_parse_base_init),
(gst_ogm_video_parse_base_init), (gst_ogm_parse_init),
(gst_ogm_audio_parse_init), (gst_ogm_video_parse_init),
(gst_ogm_parse_sink_convert), (gst_ogm_parse_chain),
(gst_ogm_parse_change_state):
Separate between audio/video so ogmaudioparse actually uses the
audio pad templates. Both audio and video work now, including
autoplugging. Also use sometimes-srcpad hack.
* gst-libs/gst/riff/riff-read.c: (gst_riff_read_seek):
Handle events better. Don't hang on infinite loops.
* gst/avi/gstavidemux.c: (gst_avi_demux_class_init),
(gst_avi_demux_init), (gst_avi_demux_reset),
(gst_avi_demux_src_convert), (gst_avi_demux_handle_src_query),
(gst_avi_demux_stream_header), (gst_avi_demux_stream_data),
(gst_avi_demux_change_state):
* gst/avi/gstavidemux.h:
Improve A/V sync. Still not perfect.
* gst/matroska/ebml-read.c: (gst_ebml_read_seek),
(gst_ebml_read_skip):
Handle events better.
* gst/qtdemux/qtdemux.c: (gst_qtdemux_handle_sink_event),
(gst_qtdemux_loop_header), (qtdemux_parse_trak),
(qtdemux_audio_caps):
Add IMA4. Improve event handling. Save offset after a seek when
the headers are at the end of the file so that we don't end up in
an infinite loop.
* gst/typefind/gsttypefindfunctions.c: (qt_type_find):
Add low-priority typefind support for files with no length.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_query),
(gst_avi_demux_stream_data):
Just hardcode for raw audio then. AVI audio sucks.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_query),
(gst_avi_demux_add_stream), (gst_avi_demux_stream_data):
* gst/avi/gstavidemux.h:
Fix for compressed audio (mp3) timestamp generation. How did this
ever work?
Original commit message from CVS:
* gst-libs/gst/riff/riff-read.c:
(gst_riff_read_strf_vids_with_data),
(gst_riff_read_strf_auds_with_data):
* gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_query),
(gst_avi_demux_add_stream), (gst_avi_demux_stream_header):
Make sure we don't create 0 sized subbufers in riff-read.
Signal the no more pads signal after reading the avi header.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_query):
Fix potential division by zero error and hopefully get
the position query right to get correct timestamps on avi
audio.
Original commit message from CVS:
reviewed by: <delete if not using a buddy>
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_video_caps_with_data),
(gst_riff_create_video_caps), (gst_riff_create_audio_caps),
(gst_riff_create_video_template_caps),
(gst_riff_create_audio_template_caps):
* gst-libs/gst/riff/riff-media.h:
* gst-libs/gst/riff/riff-read.c:
(gst_riff_read_strf_vids_with_data), (gst_riff_read_strf_vids):
* gst-libs/gst/riff/riff-read.h:
* gst/avi/gstavidemux.c: (gst_avi_demux_add_stream):
Add MS RLE support. I added some functions to read out strf chunks
into strf chunks and the data behind it. This is usually color
palettes (as in RLE, but also in 8-bit RGB). Also use those during
caps creation. Lastly, add ADPCM (similar to wavparse - which
should eventually be rifflib based).
* gst/matroska/matroska-demux.c: (gst_matroska_demux_class_init),
(gst_matroska_demux_init), (gst_matroska_demux_reset):
* gst/matroska/matroska-demux.h:
Remove placeholders for some prehistoric tagging system. Didn't add
support for any tag system really anyway.
* gst/qtdemux/qtdemux.c:
Add support for audio/x-m4a (MPEG-4) through spider.
* gst/wavparse/gstwavparse.c: (gst_wavparse_parse_fmt),
(gst_wavparse_loop):
ADPCM support (#135862). Increase max. buffer size because we
cannot split buffers for ADPCM (screws references) and I've seen
files with 2048 byte chunks. 4096 seems safe for now.
Original commit message from CVS:
* common/m4/gst-feature.m4: Call -config scripts with
--plugin-libs if it is supported.
* gst/avi/gstavimux.c: (gst_avimux_vidsinkconnect): sequences of
JPEG images are image/jpeg.
* gst/debug/Makefile.am:
* gst/debug/negotiation.c: (gst_negotiation_class_init),
(gst_negotiation_getcaps), (gst_negotiation_pad_link),
(gst_negotiation_update_caps), (gst_negotiation_get_property),
(gst_negotiation_plugin_init): Add a property that acts like
filter caps.
* testsuite/gst-lint: Move license checking to be a standard
test.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
(gst_riff_create_video_template_caps):
Add MS video v1.
* gst/avi/gstavidemux.c: (gst_avi_demux_stream_index),
(gst_avi_demux_stream_data):
Add support for "rec-list" chunks.
Original commit message from CVS:
* gst/ac3parse/gstac3parse.c: update to checklist 5
* gst/adder/gstadder.c: rewrite negotiation. update to checklist 5
* gst/audioconvert/gstaudioconvert.c: update to checklist 5
* gst/audioscale/gstaudioscale.c: same
* gst/auparse/gstauparse.c: same
* gst/avi/gstavidemux.c: same
Original commit message from CVS:
2004-01-25 Ronald Bultje <rbultje@ronald.bitfreak.net>
* gst/avi/gstavidemux.c: (gst_avi_demux_stream_avih),
(gst_avi_demux_stream_odml), (gst_avi_demux_stream_index):
The index reading was broken. The rest worked fine, but the whole
goal of my rewrite was to make avidemux readable, and this was
not at all readable. Please use typed variables.
Original commit message from CVS:
2004-01-23 Benjamin Otte <in7y118@public.uni-hamburg.de>
* gst/avi/gstavidemux.c: (gst_avi_demux_stream_avih),
(gst_avi_demux_stream_odml), (gst_avi_demux_stream_index):
don't write to buffer. Extract data without the need of
__attribute__ ((packed))
Original commit message from CVS:
2004-01-07 Ronald Bultje <rbultje@ronald.bitfreak.net>
* ext/audiofile/gstafsink.c: (gst_afsink_init), (gst_afsink_chain),
(gst_afsink_handle_event):
* ext/jpeg/gstjpegenc.c: (gst_jpegenc_init):
* gst/avi/gstavimux.c: (gst_avimux_request_new_pad):
* sys/dxr3/dxr3audiosink.c: (dxr3audiosink_init):
* sys/dxr3/dxr3spusink.c: (dxr3spusink_init):
* sys/dxr3/dxr3videosink.c: (dxr3videosink_init):
Fix for instantiate-test (see core). Also remove dead code from
jpegenc (which still needs fixing, but that's lower on my TODO
list...).
* sys/v4l2/gstv4l2src.c: (gst_v4l2src_getcaps):
Never return NULL as caps.
Original commit message from CVS:
2004-01-01 Ronald Bultje <rbultje@ronald.bitfreak.net>
* configure.ac:
Fix configure check for mpeg2enc. We need 1.6.1.93 instead of
1.6.1.92, since the pkg-config file of 1.6.1.92 is borked and
it therefore uses the wrong include paths. Too bad... Note
that 1.6.1.93 is not release yet. ;).
Also add a check for mplex, which is now using the lib'ified
mplex from mjpegtools, too.
* ext/ffmpeg/gstffmpegcodecmap.c:
Add codec_tag for 3ivx/xvid. For xvid, this should fix playback
issues. I don't think ffmpeg handles 3ivx correctly, so this
probably won't work. But it won't hurt either.
* ext/ffmpeg/gstffmpegdec.c: (gst_ffmpegdec_connect),
(gst_ffmpegdec_chain):
* ext/ffmpeg/gstffmpegenc.c: (gst_ffmpegenc_connect),
(gst_ffmpegenc_chain_audio):
Fix memleak in audio encoding. Close codec if open fails, this
calls the cleanup routines so we can re-use the context.
* ext/mpeg2enc/gstmpeg2enc.cc:
Fix pad template names/types, fix memory issue with getcaps().
* ext/mpeg2enc/gstmpeg2encoder.cc:
* ext/mpeg2enc/gstmpeg2encoder.hh:
Fix compile issue with new caps system (const thingy).
* ext/mpeg2enc/gstmpeg2encpicturereader.cc:
* ext/mpeg2enc/gstmpeg2encpicturereader.hh:
We read a first frame right on initing, so that we have a caps
when we init the output. This caps is cached in padprivate and
read as first frame.
* ext/mplex/Makefile.am:
* ext/mplex/gstmplex.cc:
* ext/mplex/gstmplex.h:
* ext/mplex/gstmplex.hh:
* ext/mplex/gstmplexibitstream.cc:
* ext/mplex/gstmplexibitstream.hh:
* ext/mplex/gstmplexjob.cc:
* ext/mplex/gstmplexjob.hh:
* ext/mplex/gstmplexoutputstream.cc:
* ext/mplex/gstmplexoutputstream.hh:
We wrap mjpegtools mplex. So I rewrote the plugin. The old plugin
had issues, didn't do capsnego, supported only a subset of the
mplex features and required a mplex fork in our local CVS. Plus
that it worked agaist a very old mplex version. Rewriting was
faster than updating it.
* gst-libs/ext/Makefile.am:
* gst-libs/ext/mplex/INSTRUCT:
* gst-libs/ext/mplex/Makefile.am:
* gst-libs/ext/mplex/README:
* gst-libs/ext/mplex/TODO:
* gst-libs/ext/mplex/ac3strm_in.cc:
* gst-libs/ext/mplex/audiostrm.hh:
* gst-libs/ext/mplex/audiostrm_out.cc:
* gst-libs/ext/mplex/aunit.hh:
* gst-libs/ext/mplex/bits.cc:
* gst-libs/ext/mplex/bits.hh:
* gst-libs/ext/mplex/buffer.cc:
* gst-libs/ext/mplex/buffer.hh:
* gst-libs/ext/mplex/fastintfns.h:
* gst-libs/ext/mplex/format_codes.h:
* gst-libs/ext/mplex/inputstrm.cc:
* gst-libs/ext/mplex/inputstrm.hh:
* gst-libs/ext/mplex/lpcmstrm_in.cc:
* gst-libs/ext/mplex/mjpeg_logging.cc:
* gst-libs/ext/mplex/mjpeg_logging.h:
* gst-libs/ext/mplex/mjpeg_types.h:
* gst-libs/ext/mplex/mpastrm_in.cc:
* gst-libs/ext/mplex/mpegconsts.cc:
* gst-libs/ext/mplex/mpegconsts.h:
* gst-libs/ext/mplex/mplexconsts.hh:
* gst-libs/ext/mplex/multplex.cc:
* gst-libs/ext/mplex/outputstream.hh:
* gst-libs/ext/mplex/padstrm.cc:
* gst-libs/ext/mplex/padstrm.hh:
* gst-libs/ext/mplex/stillsstream.cc:
* gst-libs/ext/mplex/stillsstream.hh:
* gst-libs/ext/mplex/systems.cc:
* gst-libs/ext/mplex/systems.hh:
* gst-libs/ext/mplex/vector.cc:
* gst-libs/ext/mplex/vector.hh:
* gst-libs/ext/mplex/videostrm.hh:
* gst-libs/ext/mplex/videostrm_in.cc:
* gst-libs/ext/mplex/videostrm_out.cc:
* gst-libs/ext/mplex/yuv4mpeg.cc:
* gst-libs/ext/mplex/yuv4mpeg.h:
* gst-libs/ext/mplex/yuv4mpeg_intern.h:
* gst-libs/ext/mplex/yuv4mpeg_ratio.cc:
We don't fork mjpegtools' mplex in our CVS anymore.
* gst/avi/gstavidemux.c: (gst_avi_demux_src_getcaps),
(gst_avi_demux_add_stream):
* gst/avi/gstavidemux.h:
Add getcaps() function for proper caps nego. This makes some
parts of AVI playback/reading work.
* sys/ximage/ximagesink.c: (gst_ximagesink_sinkconnect):
Resize window on new capsnego. This is probably wrong, but
I'm still committing it because with current capsnego, the
first successfull capsnego is auto-fixated, therefore rounded
down to the lowest values in the caps. this results in a 16x16
XWindow that is not reized when real capsnego finishes.
Dave, I see more cases of this, do you know a proper solution?
* tools/gst-launch-ext.in:
Fix MPEG-4 AAC (Apple iPod/iTunes) file commandline.
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!
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:
merge TYPEFIND branch. Major changes:
- totally reworked type(find) system
- all typefind functions are in gst/typefind now
- more typefind functions then before
- some plugins might fail to compile now because I don't have them installed and they
a) require bytestream or
b) haven't had their typefind fixed.
Please fix those plugins and put the typefind functions into gst/typefind if they don't have dependencies
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
Original commit message from CVS:
Set keyframe flag only when it is a keyframe. This will break quite some formats because many encoders dont set this flag correctly, but I'll fix that as I encounter them. divx5 works correctly now. ;).
Original commit message from CVS:
Fix padding (2 bytes instead of 4), improve index offset calculation, fix some minor other issues and make avimux actually work with mp3 audio
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:
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:
Actually push the event over to the next element instead of leaving it into eternity - thanks for Joshua for detecting
Original commit message from CVS:
Fix avimux (somewhat). Note: the EOS event still doesn't get through. This is completely braindead, I have no clue why, but setting this thing to PAUSE should do the trick too. EOS fix will come later on...
Original commit message from CVS:
Updated autogen.sh/configure.ac and various Makefiles to make the
configure script set up all gcc specific compiler arguments, rather
than hardcoding them in the Makefile.am files
Original commit message from CVS:
- new avi demuxer doesn't use a recursive infinite loop anymore
- removed temporary riff.[ch] files
- removed aviparse from build
Original commit message from CVS:
Handle JUNK and dmlh tags and when we find a broken/unknown chunk, just move to the next byte and try to see if it's a known chunk
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 it loopbased and use only one video and one audio pad. framerate is also no longer a property but is requested from the source pad
Original commit message from CVS:
Separate the movi processing loop from the index/entry parsing loop
Detect when the index starts from 0 or from the movi chunck offset
Original commit message from CVS:
add ranks only for plugins who participate in autoplugging. If you have a file that used to autoplug but doesn't anymore, then let me know or add a rank to the missing element.
Original commit message from CVS:
- Changed plugins for new APIs
- modularized audiofile.
- added seeking, query and convert functions for mad, mpeg2dec,
avidemux, mpegdemux, mpegparse
- sync updates to oss. removed the ossclock for now
Original commit message from CVS:
* a hack to work around intltool's brokenness
* a current check for mpeg2dec
* details->klass reorganizations
* an element browser that uses details->klass
* separated cdxa parse out from the avi directory
Original commit message from CVS:
* removal of //-style comments
* don't link plugins to core libs -- the versioning is done internally to the plugins with the plugin_info struct,
and symbol resolution is lazy, so we can always know if a plugin can be loaded by the plugin_info data. in theory.
Original commit message from CVS:
s/@GST_PLUGIN_LDFLAGS@/$(GST_PLUGIN_LDFLAGS)/
@-substitued variables variables are defined as make variables automagically,
and this gives the user the freedom to say make GST_PLUGIN_LDFLAGS=-myflag
Original commit message from CVS:
WARNING: avimux is still broken, but less broken than it used to be... Code is under heavy development and will work sooner or later... Uploaded for generic development and testing purposes, not intended for generic use whatsoever
Original commit message from CVS:
made changes everywhere to accomodate for the headers being in
<gst/(lib)/...>
we'll need to conclude this fast because we will also need to change stuff in core real soon for the libs in order to fix everything
and I can't do it right now because I disabled all of the plugins here ;)
