Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_add_stream),
(qtdemux_parse_trak):
Fix crash (j might be greater than n_samples, in which case we're
writing outside the allocated space for the array) and memleak.
Original commit message from CVS:
2004-03-06 Christophe Fergeau <teuf@gnome.org>
* ext/faac/gstfaac.c: (gst_faac_chain):
* ext/flac/gstflactag.c: (gst_flac_tag_chain):
* ext/libpng/gstpngenc.c: (user_write_data):
* ext/mikmod/gstmikmod.c: (gst_mikmod_loop):
* gst/ac3parse/gstac3parse.c: (gst_ac3parse_chain):
* gst/mpeg2sub/gstmpeg2subt.c: (gst_mpeg2subt_chain_subtitle):
* gst/mpegstream/gstrfc2250enc.c: (gst_rfc2250_enc_add_slice):
Fix several misuse of gst_buffer_merge (it doesn't take ownership
of any buffer), should fix some leaks. I hope I didn't unref buffers
that shouldn't be...
Original commit message from CVS:
* ext/faad/gstfaad.c: (gst_faad_init), (gst_faad_srcgetcaps),
(gst_faad_chain): Fix negotiation.
* ext/librfb/gstrfbsrc.c: (gst_rfbsrc_handle_src_event): Add
key and button events.
* gst-libs/gst/floatcast/floatcast.h: Fix a minor bug in this
dung heap of code.
* gst-libs/gst/gconf/gstreamer-gconf-uninstalled.pc.in: gstgconf
depends on gconf
* gst-libs/gst/gconf/gstreamer-gconf.pc.in: same
* gst-libs/gst/play/play.c: (gst_play_pipeline_setup),
(gst_play_video_fixate), (gst_play_audio_fixate): Add a fixate
function to encourage better negotiation, particularly between
audioconvert and osssink.
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_chain):
* gst/qtdemux/qtdemux.c: (qtdemux_parse_trak): Make some debugging
more important.
* gst/typefind/gsttypefindfunctions.c: Fix mistake in flash
typefinding.
* gst/vbidec/vbiscreen.c: Add glib header
* pkgconfig/gstreamer-play.pc.in: Depends on gst-interfaces.
Original commit message from CVS:
* gst/videodrop/gstvideodrop.c: (gst_videodrop_init),
(gst_videodrop_chain), (gst_videodrop_change_state):
* gst/videodrop/gstvideodrop.h:
Work based on timestamp of input data, not based on the expected
framerate from the input. The consequence is that this element now
not only scales framerates, but also functions as a framerate
corrector or framerate stabilizer/constantizer.
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 Benjamin Otte <otte@gnome.org>
* ext/xine/Makefile.am:
* ext/xine/gstxine.h:
* ext/xine/xine.c:
* ext/xine/xineaudiodec.c:
* ext/xine/xinecaps.c:
add first version of xine plugin wrapper. Currently only wraps the
QDM2 win32 DLL, and even that only in proof-of-concept quality.
* configure.ac:
* ext/Makefile.am:
add xine plugin wrapper, disabled by default. Use --enable-xine to
build. Note that it'll segfault on gst-register if you don't remove
the goom and tvtime post plugins from xine.
* gst/qtdemux/qtdemux.c: (gst_qtdemux_handle_sink_event),
(qtdemux_parse), (qtdemux_parse_trak), (qtdemux_audio_caps):
add extradata parsing for QDM2.
change around debugging prints.
Original commit message from CVS:
2004-02-15 Julien MOUTTE <julien@moutte.net>
* gst/switch/gstswitch.c: (gst_switch_loop): More fixes for
correct data refcounting.
Original commit message from CVS:
2004-02-15 Julien MOUTTE <julien@moutte.net>
* gst/switch/gstswitch.c: (gst_switch_change_state),
(gst_switch_class_init): Cleaning the sinkpads correctly on state
change, mostly the EOS flag.
Original commit message from CVS:
2004-02-14 Julien MOUTTE <julien@moutte.net>
* gst-libs/gst/play/play.c: (gst_play_connect_visualization): Disable
visualization until i find a way to fix switch correctly.
* gst-libs/gst/riff/riff-read.c: (gst_riff_peek_head): Fix a bug when
EOS arrives.
* gst/switch/gstswitch.c: (gst_switch_release_pad),
(gst_switch_request_new_pad), (gst_switch_poll_sinkpads),
(gst_switch_loop), (gst_switch_dispose), (gst_switch_class_init):
Reworked switch to get a more correct behaviour with events and refing
of data stored in sinkpads.
* gst/switch/gstswitch.h: Adding an eos flag for every sinkpad so that
we don't pull from a pad in EOS.
Original commit message from CVS:
2004-02-03 Benjamin Otte <in7y118@public.uni-hamburg.de>
* gst/qtdemux/qtdemux.c: (gst_qtdemux_add_stream):
set explicit caps before adding the element, so the autopluggers can
plug correctly.
* gst/typefind/gsttypefindfunctions.c: (mp3_type_find),
(mpeg2_sys_type_find), (mpeg1_sys_type_find),
(mpeg_video_type_find), (mpeg_video_stream_type_find),
(dv_type_find):
fix memleaks in typefind functions. gst_type_find_suggest takes a const
argument.
Original commit message from CVS:
2004-01-29 Ronald Bultje <rbultje@ronald.bitfreak.net>
* gst/mpeg1videoparse/gstmp1videoparse.c:
(gst_mp1videoparse_real_chain):
Committed wrong version last week... Grr... Didn't notice until now.
Original commit message from CVS:
2004-01-26 Jeremy Simon <jesimon@libertysurf.fr>
* ext/ffmpeg/gstffmpegcodecmap.c: (gst_ffmpeg_codecid_to_caps),
(gst_ffmpeg_caps_to_extradata), (gst_ffmpeg_caps_to_pixfmt):
* gst/qtdemux/qtdemux.c: (plugin_init), (qtdemux_parse_trak),
(qtdemux_video_caps):
* gst/qtdemux/qtdemux.h:
Add SVQ3 specific flags to qtdemux and ffmpeg
Original commit message from CVS:
2004-01-25 Julien MOUTTE <julien@moutte.net>
* gst-libs/gst/play/gstplay.c: (gst_play_pipeline_setup),
(gst_play_identity_handoff), (gst_play_set_location),
(gst_play_set_visualization), (gst_play_connect_visualization): Another
try in visualization implementation. Still have an issue with switch
blocking when pulling from video_queue and only audio comes out of
spider.
* gst/switch/gstswitch.c: (gst_switch_release_pad),
(gst_switch_poll_sinkpads), (gst_switch_class_init): Implementing pad
release method. And check if the pad is usable before pulling.
Original commit message from CVS:
2004-01-25 Ronald Bultje <rbultje@ronald.bitfreak.net>
* gst-libs/gst/riff/riff-read.c: (gst_riff_read_info):
Additional pad usability check.
* gst/mpeg1videoparse/gstmp1videoparse.c: (gst_mp1videoparse_init),
(mp1videoparse_find_next_gop), (gst_mp1videoparse_time_code),
(gst_mp1videoparse_real_chain):
Fix MPEG video stream parsing. The original plugin had several
issues, including not timestamping streams where the source was
not timestamped (this happens with PTS values in mpeg system
streams, but MPEG video is also a valid stream on its own so
that needs timestamps too). We use the display time code for that
for now. Also, if one incoming buffer contains multiple valid
frames, we push them all on correctly now, including proper EOS
handling. Lastly, several potential segfaults were fixed, and we
properly sync on new sequence/gop headers to include them in next,
not previous frames (since they're header for the next frame, not
the previous). Also see #119206.
* gst/mpegaudioparse/gstmpegaudioparse.c: (gst_mp3parse_chain),
(bpf_from_header):
Move caps setting so we only do it after finding several valid
MPEG-1 fraes sequentially, not right after the first one (which
might be coincidental).
* gst/typefind/gsttypefindfunctions.c: (mpeg1_sys_type_find),
(mpeg_video_type_find), (mpeg_video_stream_type_find),
(plugin_init):
Add unsynced MPEG video stream typefinding, and change some
probability values so we detect streams rightly. The idea is as
follows: I can have an unsynced system stream which contains
video. In the current code, I would randomly get a type for either
system or video stream type found, because the probabilities are
being calculated rather randomly. I now use fixed values, so we
always prefer system stream if that was found (and that is how it
should be). If no system stream was found, we can still identity
the stream as video-only.
Original commit message from CVS:
2004-01-20 Julien MOUTTE <julien@moutte.net>
* gst/switch/gstswitch.c: (gst_switch_request_new_pad),
(gst_switch_init): Fixed switch element : proxying link and setting
caps from src to sink on request.
Original commit message from CVS:
* gst-libs/gst/audio/Makefile.am:
Add gstaudiofiltertemplate.c and building of gstaudiofilterexample.c
from the template.
* gst-libs/gst/audio/gstaudiofilter.c:
* gst-libs/gst/audio/gstaudiofilter.h:
Add bytes_per_sample and size and n_samples calculation.
* gst-libs/gst/audio/gstaudiofilterexample.c:
Remove, now autogenerated.
* gst-libs/gst/audio/gstaudiofiltertemplate.c:
Moved from gstaudiofilterexample, object name changed, code added
so that it actually works.
* gst-libs/gst/audio/make_filter:
Script to build an audiofilter subclass from the template.
* gst/colorspace/Makefile.am:
* gst/colorspace/yuv2yuv.c:
Remove file, since it's GPL, and we don't use it.
Original commit message from CVS:
* ext/esd/esdsink.c: (gst_esdsink_class_init): Remove property
that handles osssink fallback.
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_init),
(gst_audio_convert_getcaps):
* gst/qtdemux/qtdemux.c: (qtdemux_audio_caps):
Add audio/x-qdm2 for QDM2 audio.
* gst/sine/gstsinesrc.c: (gst_sinesrc_get):
* gst/sine/gstsinesrc.h: Add example of how to implement tags.
* gst/videoscale/gstvideoscale.c: (gst_videoscale_getcaps):
Decrease minimum size to 16x16.
* gst/wavparse/gstwavparse.c:
Convert disabled pad template caps to new caps.
* sys/ximage/ximagesink.c: (gst_ximagesink_xcontext_get):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xcontext_get),
(gst_xvimagesink_chain): Throw element error when display cannot
be opened. Increase minimum framerate to 1.0. Check the data
free function on a buffer to make sure it is the type we expect
before manipulating it.
Original commit message from CVS:
2004-01-15 Ronald Bultje <rbultje@ronald.bitfreak.net>
* gst/colorspace/gstcolorspace.c:
* gst/colorspace/yuv2yuv.c: (gst_colorspace_yuy2_to_i420),
(gst_colorspace_i420_to_yv12):
Fix compiling... Didn't test if it actually works.
Original commit message from CVS:
* configure.ac:
* gst/colorspace/Makefile.am:
* gst/colorspace/gstcolorspace.c:
* gst/colorspace/gstcolorspace.h:
* gst/colorspace/yuv2rgb.c:
* gst/colorspace/yuv2rgb.h:
Duplicate the ext/hermes colorspace plugin, and remove Hermes
code and GPL code. Fix for new caps negotiation. Rewrite
much of the format handling code, and some of the conversion
code. Basically, rewrote almost everything. This element
handles I420, YV12 to RGB conversions.
* ext/hermes/Makefile.am:
* ext/hermes/gsthermescolorspace.c:
Rename colorspace to hermescolorspace. Fix negotiation issues.
Remove non-Hermes related code. This element handles lots of
RGB to RGB conversions, but no YUV.
* ext/hermes/gstcolorspace.c:
* ext/hermes/gstcolorspace.h:
* ext/hermes/rgb2yuv.c:
* ext/hermes/yuv2rgb.c:
* ext/hermes/yuv2rgb.h:
* ext/hermes/yuv2rgb_mmx16.s:
* ext/hermes/yuv2yuv.c:
* ext/hermes/yuv2yuv.h:
Remove old code.
Original commit message from CVS:
2003-12-22 Benjamin Otte <in7y118@public.uni-hamburg.de>
* gst/qtdemux/qtdemux.c: (plugin_init):
qtdemux requires bytestream
Original commit message from CVS:
2003-12-21 Ronald Bultje <rbultje@ronald.bitfreak.net>
* configure.ac:
Improve mpeg2enc detection. This is for distributions that do
ship mjpegtools, but without mpeg2enc. Also does object check
for might there ever be ABI incompatibility.
* ext/mpeg2enc/gstmpeg2enc.cc:
Add Andrew as second maintainer (he's helping me), and also add
an error if no caps was set. This happens if I pull before capsnego
and that's something I should solve sometime else.
* gst/matroska/matroska-demux.c:
(gst_matroska_demux_parse_blockgroup):
Fix time parsing.
* gst/matroska/matroska-mux.c: (gst_matroska_mux_audio_pad_link),
(gst_matroska_mux_track_header):
Add caps to templates.
* gst/mpegaudioparse/gstmpegaudioparse.c: (mp3_sink_factory):
Add mpegversion=1 to prevent confusion with MPEG/AAC.
* gst/mpegstream/gstmpegdemux.c:
Remove layer since it causes warnings about unfixed caps.
* gst/videotestsrc/gstvideotestsrc.c: (gst_videotestsrc_get):
Fix obvious typo (we error out if caps were set, we should of
course error out if *no* caps were set).
* sys/oss/gstosselement.c: (gst_osselement_convert):
Fix format conversion, we confused bits/bytes.
* sys/oss/gstosselement.h:
Improve documentation for 'bps'.
* sys/v4l/TODO:
Remove stuff about plugins that need removing - this was done
ages ago.
* sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_init),
(gst_v4lmjpegsrc_src_convert), (gst_v4lmjpegsrc_src_query):
* sys/v4l/gstv4lsrc.c: (gst_v4lsrc_init), (gst_v4lsrc_src_convert),
(gst_v4lsrc_src_query):
* sys/v4l2/gstv4l2src.c: (gst_v4l2src_init),
(gst_v4l2src_src_convert), (gst_v4l2src_src_query):
Add get_query_types(), get_formats() and query() functions.
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:
Adding a new plugin: switch.
It takes N input and only has 1 output. You can "switch" the forwarded input through properties ("nb_sources", "active_source") and i will probably add tuner interface support soon.
It should be able to handle any kind of data passing through it.
It is still a work in progress don't consider it usable for production yet.
Original commit message from CVS:
tagging stuff and build fixes. In detail:
- make gdk-pixbuf loader work when distchecking
- fix invalid syntax in ffmpeg Makefile. wildcards for EXTRA_DIST are not allowed. This broke builds where distdir != srcdir
- fix ffmpeg cvs grabbing when srcdir != distdir
- new id3tag plugin for id3 tag reading/writing (uses mad's libid3tag)
- mad and libid3tag require mad/libid3tag v0.15. Fixed configure to require that
- added ogg demuxer in ext/ogg. The demuxer does not handle events yet. Especially getting seeking right will require some effort or code copying from libvorbis.
- added raw vorbis detection to typefinding. oggdemux requires a typefind function to detect its contents.
- tags plugin in gst/tags. Provides API in <gst/tags/gsttagediting.h>. API includes tag matching GStreamer <=> ID3 and GStreamer <=> vorbis and writing/reading vorbiscomments or ID3v1 tags. Also included is a simple vorbiscomment reader/writer. Writing will not really work though until someone writes oggmux.
- various build fixes. Mostly missing (DIST)CLEANFILES.
- vorbisenc handles tag writing.
Now it's YOUR turn to fix and write more plugins that handle writing/reading of tags. :)
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:
Woah, I'm f***ing annoyed that someonme never tests his changes and figures out that every freakin' format is identified as text/plain
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
