Original commit message from CVS:
* configure.ac: look for musepack headers as musepack/*.h
(fixes#159847)
* ext/musepack/gstmusepackdec.h: use <musepack/*.h>
* ext/musepack/gstmusepackreader.h: same
Original commit message from CVS:
Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
* ext/musicbrainz/gsttrm.c: (gst_musicbrainz_class_init),
(gst_musicbrainz_init), (gst_musicbrainz_chain),
(gst_musicbrainz_set_property), (gst_musicbrainz_get_property):
* ext/musicbrainz/gsttrm.h:
Add support for using a proxy server when getting a trm id from
the MusicBrainz database (#149613).
Original commit message from CVS:
* configure.ac: add audioresample and cairo plugins. Remove
HAVE_MMX stuff, because it's not used.
* ext/Makefile.am: same
* ext/audioresample/Makefile.am: You are not ready for an
audio resampling element based on audioresample.
* ext/audioresample/gstaudioresample.c:
* ext/audioresample/gstaudioresample.h:
* ext/cairo/Makefile.am: You are not ready for overlay elements
based on cairo. Don't look too closely, these elements kinda
suck right now.
* ext/cairo/gstcairo.c: new
* ext/cairo/gsttextoverlay.c: new
* ext/cairo/gsttextoverlay.h: new
* ext/cairo/gsttimeoverlay.c: new
* ext/cairo/gsttimeoverlay.h: new
* gst-libs/gst/media-info/media-info-priv.h: fix compile
problem with compilers that don't support variadic macros.
Original commit message from CVS:
* ext/faad/gstfaad.c: (gst_faad_chanpos_from_gst),
(gst_faad_chanpos_to_gst), (gst_faad_chain):
Set DURATION even if source buffer didn't. Also use increasing
timestamps.
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_audio_caps_with_data):
Block_align can have larger values than 8192.
Original commit message from CVS:
* configure.ac:
* ext/libvisual/visual.c: (gst_visual_get_type),
(libvisual_log_handler), (gst_visual_getcaps),
(gst_visual_srclink), (gst_visual_change_state), (make_valid_name),
(plugin_init):
Update libvisual to 0.1.7. Link in the debug handling to gstreamer
* ext/smoothwave/Makefile.am:
* ext/smoothwave/demo-osssrc.c: (main):
* ext/smoothwave/gstsmoothwave.c: (gst_smoothwave_class_init),
(gst_smoothwave_init), (gst_smoothwave_dispose), (gst_sw_sinklink),
(gst_sw_srclink), (gst_smoothwave_chain), (gst_sw_change_state),
(plugin_init):
* ext/smoothwave/gstsmoothwave.h:
Make gstsmoothwave a working element in the 20th century.
* gst/chart/gstchart.c: (gst_chart_init), (gst_chart_srcconnect):
Fix incorrect link function
Original commit message from CVS:
* ext/alsa/gstalsasink.c: (gst_alsa_sink_get_time):
This seems to be antique leftover. It needs to pass error
checking.
* ext/sdl/sdlvideosink.c: (gst_sdlvideosink_init),
(gst_sdlvideosink_deinitsdl), (gst_sdlvideosink_initsdl),
(gst_sdlvideosink_destroy), (gst_sdlvideosink_create),
(gst_sdlvideosink_sinkconnect), (gst_sdlvideosink_chain):
Fix GstXOverlay implementation (#151059).
Original commit message from CVS:
* ext/alsa/gstalsa.c: (gst_alsa_get_caps_internal):
buffer-frames property was missing
* ext/arts/gst_arts.c:
rate missing from sinkcaps
* ext/audiofile/gstafparse.c:
* ext/audiofile/gstafsink.c:
* ext/audiofile/gstafsrc.c:
* ext/swfdec/gstswfdec.c:
int audio doesn't know buffer-frames
* ext/cdparanoia/gstcdparanoia.c:
int audio doesn't know chunksize either
* ext/nas/nassink.c:
it's endianness, not endianess
* gst-libs/gst/audio/audio.h:
make float standard pad template caps really describe float
* gst/law/mulaw.c: (linear_factory):
signed only, please
* gst/mpegstream/gstdvddemux.c:
widths of 20 are not valid
Original commit message from CVS:
* ext/cdaudio/gstcdaudio.c: (_do_init), (gst_cdaudio_base_init),
(gst_cdaudio_get_event_masks), (gst_cdaudio_send_event),
(gst_cdaudio_query), (plugin_init), (cdaudio_uri_get_type),
(cdaudio_uri_get_protocols), (cdaudio_uri_get_uri),
(cdaudio_uri_set_uri), (cdaudio_uri_handler_init):
Added uri handler for cd://
Port to new API.
Original commit message from CVS:
* configure.ac: update for swfdec-0.3 and liboil-0.2
* ext/swfdec/gstswfdec.c: update for swfdec-0.3
* ext/swfdec/gstswfdec.h: same
* gst/videofilter/gstvideobalance.c: update for liboil-0.2
* gst/videotestsrc/videotestsrc.c: same
Original commit message from CVS:
Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
* ext/faad/gstfaad.c: (gst_faad_init), (gst_faad_sinkconnect),
(gst_faad_chain), (gst_faad_change_state):
* ext/faad/gstfaad.h:
Allow playback of raw (unframed) MPEG AAC files (#148993).
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:
* configure.ac: remove NASM check, since we don't use it. Update
dirac check to 0.4
* ext/dirac/gstdiracdec.cc: update to current 0.4 API
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_link):
Initialized variables.
* gst/qtdemux/qtdemux.c: (gst_qtdemux_change_state),
(gst_qtdemux_loop_header), (qtdemux_parse), (qtdemux_parse_trak),
(gst_qtdemux_handle_esds), (qtdemux_audio_caps): Fix seeking, add
SVQ3 format
Original commit message from CVS:
* Trying to correct the GST_CHECK_CONFIGPROG macro. It works perfectly for
I hope i dont break anything.
* Shifting the MAS plugin back from gst-rotten.
Original commit message from CVS:
* ext/sdl/sdlvideosink.c: (gst_sdlvideosink_base_init): Only allow
sane framerates.
* sys/ximage/ximagesink.c: (gst_ximagesink_xcontext_get): same
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support): same
* testsuite/gst-lint: Test for G_GUINT64_FORMAT usage near gettext.
Original commit message from CVS:
* configure.ac: OSS portability
* ext/arts/gst_arts.c: idem
* sys/oss/gstosselement.c: idem
* sys/oss/gstossmixer.c: idem
* sys/oss/gstosssink.c: idem
* sys/oss/gstosssrc.c: idem
* sys/oss/oss_probe.c: idem
- check for soundcard.h in different places for some BSD
Original commit message from CVS:
* ext/Makefile.am:
add rules to build shout2send (was removed by accident
when this module was no more marked experimental/broken)
* ext/shout2/gstshout2.c:
* ext/shout2/gstshout2.h:
adding a "connection problem" signal to shout2send
(fixes#142954)
Original commit message from CVS:
second batch :
remove ',' at end of enums as they could confuse older gcc, foreign compilers (forte) and gtk-doc
(in gst-plugins/ext/ this time)
Original commit message from CVS:
* ext/audiofile/gstafparse.c : change class to Codec/Demuxer/Audio
* gst/auparse/gstauparse.c : idem
* gst/wavparse/gstwavparse.c : idem
Original commit message from CVS:
* ext/audiofile/gstafsrc.c: (gst_afsrc_get):
Remove old debug output
* ext/dv/gstdvdec.c: (gst_dvdec_quality_get_type),
(gst_dvdec_class_init), (gst_dvdec_loop), (gst_dvdec_change_state),
(gst_dvdec_set_property), (gst_dvdec_get_property):
Change the quality setting to an enum, so it works from gst-launch
Don't renegotiate a non-linked pad. Allows audio only decoding.
* gst/deinterlace/gstdeinterlace.c: (gst_deinterlace_getcaps),
(gst_deinterlace_link), (gst_deinterlace_init):
* gst/videodrop/gstvideodrop.c: (gst_videodrop_getcaps),
(gst_videodrop_link):
Some caps negotiation fixes
Original commit message from CVS:
* ext/tarkin/gsttarkin.c :
- Change RANK from NONE to PRIMARY (decoder)
* ext/gdk_pixbuf/gstgdkpixbuf.c :
- Change RANK from NONE to MARGINAL (decoder)
* ext/divx/gstdivxenc.c :
- Change RANK from PRIMARY to NONE (encoder/spider issue)
