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))
