Original commit message from CVS:
* ext/hermes/gsthermescolorspace.c: (plugin_init): decrease rank
by 2 to not interfere with other colorspaces.
* ext/pango/gsttextoverlay.c: (plugin_init): change rank to NONE
* gst/colorspace/gstcolorspace.c: (plugin_init): decrease rank by
one to not interfere with ffmpeg_colorspace.
Original commit message from CVS:
* ext/alsa/gstalsa.c: (gst_alsa_fixate): Don't fixate fields that
aren't in the caps.
* gst/sine/gstsinesrc.c: change rate caps to [1,MAX]
* gst/videocrop/gstvideocrop.c: (plugin_init): Change rank to NONE.
Original commit message from CVS:
* gst/modplug/gstmodplug.cc:
handle events - don't do crap when a discont arrives that's not
necessary
This allows correct loading and playback of mods in Rhythmbox
Original commit message from CVS:
* configure.ac:
* gst-libs/gst/gconf/Makefile.am:
* pkgconfig/Makefile.am:
move gstreamer-gconf pkgconfig files to pkgconfig/ dir. Make sure
they get rebuilt properly
* configure.ac:
when checking for vorbis, try pkgconfig first.
* gst/modplug/gstmodplug.cc:
add fixate function
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (qtdemux_parse_trak):
Fix for obvious mistake, where we first shift the offset and then
read a samplesize element assuming the old offset. Note that this
part still has something weird, i.e. my movies containing those
don't actually play well, but at least there's something that looks
like sound now.
Original commit message from CVS:
* configure.ac: the Hermes library controls hermescolorspace, not
colorspace.
* ext/mpeg2dec/gstmpeg2dec.c: (gst_mpeg2dec_base_init),
(gst_mpeg2dec_init): minor pet peeve: disable code with #ifdef,
not /* */
* ext/sdl/sdlvideosink.c: Change XID to unsigned long.
* ext/sdl/sdlvideosink.h: ditto.
* gst/colorspace/gstcolorspace.c: Fix old comments about Hermes
Original commit message from CVS:
* ext/mikmod/gstmikmod.c: (gst_mikmod_init), (gst_mikmod_loop),
(gst_mikmod_change_state):
* ext/mikmod/gstmikmod.h:
make mikmod's loop function not loop infinitely and call
gst_element_yield anymore
* gst/modplug/gstmodplug.cc:
fix pad negotiation
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.
