Stefan Kost [Fri, 27 Jun 2008 07:42:07 +0000 (07:42 +0000)]
tests/examples/seek/seek.c: Include config.h and check if we have X. Fixes: #540334.
Original commit message from CVS:
* tests/examples/seek/seek.c:
Include config.h and check if we have X. Fixes: #540334.
Sam Morris [Thu, 26 Jun 2008 06:03:38 +0000 (06:03 +0000)]
gst-libs/gst/interfaces/mixertrack.c: API: Add "index" property to GstMixerTrack to differantiate between multiple mi...
Original commit message from CVS:
Patch by: Sam Morris <sam at robots dot org to uk>
* gst-libs/gst/interfaces/mixertrack.c:
(gst_mixer_track_class_init), (gst_mixer_track_get_property),
(gst_mixer_track_set_property):
API: Add "index" property to GstMixerTrack to differantiate between
multiple mixer tracks with the same label.
* ext/alsa/gstalsamixeroptions.c: (gst_alsa_mixer_options_new):
* ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_new):
Set the "index" property of GstMixerTrack to the index given by ALSA.
Fixes bug #528299.
Stefan Kost [Wed, 25 Jun 2008 13:15:50 +0000 (13:15 +0000)]
tests/examples/seek/: Remove libgstvideo usage. Use gtk_get_option_group instead of gtk_init().
Original commit message from CVS:
* tests/examples/seek/Makefile.am:
* tests/examples/seek/seek.c:
Remove libgstvideo usage. Use gtk_get_option_group instead of
gtk_init().
Stefan Kost [Tue, 24 Jun 2008 16:27:35 +0000 (16:27 +0000)]
tests/check/Makefile.am: Name the test registry format neutral.
Original commit message from CVS:
* tests/check/Makefile.am:
Name the test registry format neutral.
Stefan Kost [Tue, 24 Jun 2008 16:22:45 +0000 (16:22 +0000)]
gst/playback/gstqueue2.c: Do not double notify. Remove the unsued return value.
Original commit message from CVS:
* gst/playback/gstqueue2.c:
Do not double notify. Remove the unsued return value.
Stefan Kost [Tue, 24 Jun 2008 16:15:26 +0000 (16:15 +0000)]
ext/alsa/gstalsamixer.c: Also consider "speaker" as a name for master volume. If that doesn't help look for the first...
Original commit message from CVS:
* ext/alsa/gstalsamixer.c:
Also consider "speaker" as a name for master volume. If that doesn't
help look for the first non-mono volume control that also has a
playback switch.
Stefan Kost [Tue, 24 Jun 2008 16:10:50 +0000 (16:10 +0000)]
ChangeLog: Forgot to save the ChangeLog :/
Original commit message from CVS:
* ChangeLog:
Forgot to save the ChangeLog :/
Stefan Kost [Tue, 24 Jun 2008 16:05:06 +0000 (16:05 +0000)]
tests/examples/seek/: Embedd the xwindow.
Original commit message from CVS:
* tests/examples/seek/Makefile.am:
* tests/examples/seek/seek.c:
Embedd the xwindow.
Jan Schmidt [Tue, 24 Jun 2008 01:14:40 +0000 (01:14 +0000)]
sys/ximage/ximagesink.h: When the caps change, make sure to re-draw borders in force-aspect-ratio=true mode.
Original commit message from CVS:
* sys/ximage/ximagesink.c (gst_ximagesink_ximage_put),
(gst_ximagesink_setcaps):
* sys/ximage/ximagesink.h:
When the caps change, make sure to re-draw borders in
force-aspect-ratio=true mode.
* sys/xvimage/xvimagesink.c (gst_xvimagesink_xvimage_put):
Don't clear the border_draw flag until we actually draw the border.
* tests/check/Makefile.am:
Ignore alsasink/src during the states test too, so it doesn't fail
when running without access to the sound device.
Stefan Kost [Sun, 22 Jun 2008 18:35:27 +0000 (18:35 +0000)]
tests/examples/seek/seek.c: Fix crasher when playing a parse-launch line the 2nd time.
Original commit message from CVS:
* tests/examples/seek/seek.c:
Fix crasher when playing a parse-launch line the 2nd time.
Thomas Vander Stichele [Sat, 21 Jun 2008 18:56:08 +0000 (18:56 +0000)]
tests/check/pipelines/oggmux.c: Properly ifdef tests to fix compilation.
Original commit message from CVS:
* tests/check/pipelines/oggmux.c:
Properly ifdef tests to fix compilation.
Thomas Vander Stichele [Sat, 21 Jun 2008 10:25:59 +0000 (10:25 +0000)]
break long lines
Original commit message from CVS:
break long lines
Michael Smith [Fri, 20 Jun 2008 18:24:24 +0000 (18:24 +0000)]
gst/playback/: Add get-video-pad, get-audio-pad, get-text-pad action signals to playbin2. This allows the user to get...
Original commit message from CVS:
* gst/playback/gstplay-marshal.list:
* gst/playback/gstplaybin2.c:
Add get-video-pad, get-audio-pad, get-text-pad action signals to
playbin2. This allows the user to get to the selector's sinkpads, and
thus inspect a range of things - caps, tags, etc.
Michael Smith [Fri, 20 Jun 2008 17:27:03 +0000 (17:27 +0000)]
gst/playback/gstplaybin2.c: Use a different constant for the convert-frame signal id.
Original commit message from CVS:
* gst/playback/gstplaybin2.c:
Use a different constant for the convert-frame signal id.
Fixes #537009.
Michael Smith [Fri, 20 Jun 2008 17:18:55 +0000 (17:18 +0000)]
gst/playback/: Fix a whole bunch of typos in comments and log statements.
Original commit message from CVS:
* gst/playback/gstplaybin2.c:
* gst/playback/gstplaysink.c:
Fix a whole bunch of typos in comments and log statements.
Michael Smith [Fri, 20 Jun 2008 17:02:48 +0000 (17:02 +0000)]
sys/xvimage/xvimagesink.c: Don't set colour balance values on the Xv port if the user hasn't changed them (via proper...
Original commit message from CVS:
* sys/xvimage/xvimagesink.c:
Don't set colour balance values on the Xv port if the user hasn't
changed them (via properties or the interface). Avoids accumulating
rounding errors for the common case.
Partial fix for bug #537889.
Michael Smith [Fri, 20 Jun 2008 16:56:18 +0000 (16:56 +0000)]
gst/playback/gstdecodebin2.c: Ensure decodebin2 emits 'drained' signal once, and only once, when all pads are drained.
Original commit message from CVS:
* gst/playback/gstdecodebin2.c:
Ensure decodebin2 emits 'drained' signal once, and only once, when all
pads are drained.
Thomas Vander Stichele [Fri, 20 Jun 2008 16:12:50 +0000 (16:12 +0000)]
apparently it's an error to specify nc -l -p 3000 - though the short usage does not make it very clear that you can d...
Original commit message from CVS:
apparently it's an error to specify nc -l -p 3000 - though the short usage
does not make it very clear that you can drop the host arg with -l
Wim Taymans [Fri, 20 Jun 2008 09:25:44 +0000 (09:25 +0000)]
ext/vorbis/vorbisenc.c: Report the encoder latency. Fixes #538232.
Original commit message from CVS:
* ext/vorbis/vorbisenc.c: (gst_vorbis_enc_get_latency),
(gst_vorbis_enc_src_query), (gst_vorbis_enc_chain):
Report the encoder latency. Fixes #538232.
Wim Taymans [Fri, 20 Jun 2008 09:19:59 +0000 (09:19 +0000)]
gst/playback/gstplaybin2.c: Implement the source property, emit notify when it changes in the underlying uridecodebin.
Original commit message from CVS:
* gst/playback/gstplaybin2.c: (gst_play_bin_get_property),
(notify_source), (activate_group):
Implement the source property, emit notify when it changes in the
underlying uridecodebin.
Wim Taymans [Fri, 20 Jun 2008 09:14:26 +0000 (09:14 +0000)]
tests/examples/seek/seek.c: Free and clear the seek element list so that we don't use invalid references when seeking...
Original commit message from CVS:
* tests/examples/seek/seek.c: (stop_cb):
Free and clear the seek element list so that we don't use invalid
references when seeking after recreating a gst-launch line.
Wim Taymans [Fri, 20 Jun 2008 09:09:37 +0000 (09:09 +0000)]
gst-libs/gst/audio/gstbaseaudiosink.c: Report latency even if we are not live instead of hiding it.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_query), (gst_base_audio_sink_skew_slaving),
(gst_base_audio_sink_render):
Report latency even if we are not live instead of hiding it.
Take ts-offset and render-delay of the basesink into account when
scheduling samples.
Rework the clipping code so that we can take the various offsets into
account and still do correct clipping.
Jan Schmidt [Fri, 20 Jun 2008 08:52:21 +0000 (08:52 +0000)]
configure.ac: Bump verion back to devel -> 0.10.20.1
Original commit message from CVS:
* configure.ac:
Bump verion back to devel -> 0.10.20.1
Sebastian Dröge [Fri, 20 Jun 2008 08:47:14 +0000 (08:47 +0000)]
gst-libs/gst/tag/tags.c: Don't increase the size of non-string image buffers by one as this might in theory confuse d...
Original commit message from CVS:
* gst-libs/gst/tag/tags.c: (gst_tag_image_data_to_image_buffer):
Don't increase the size of non-string image buffers by one as this
might in theory confuse decoders. Still increase it by one for string
image buffers to append '\0'.
Antoine Tremblay [Fri, 20 Jun 2008 08:45:13 +0000 (08:45 +0000)]
gst/gdp/gstgdppay.c: Fix a buffer memleak and remove a confusing and wrong debug output.
Original commit message from CVS:
Patch by: Antoine Tremblay <hexa00 at gmail dot com>
* gst/gdp/gstgdppay.c: (gst_gdp_pay_reset):
Fix a buffer memleak and remove a confusing and wrong debug output.
Fixes bug #538663.
Wim Taymans [Thu, 19 Jun 2008 11:25:37 +0000 (11:25 +0000)]
examples/app/appsink-src.c: Don't use a buffer after unreffing it.
Original commit message from CVS:
* examples/app/appsink-src.c: (on_new_buffer_from_source):
Don't use a buffer after unreffing it.
Jan Schmidt [Wed, 18 Jun 2008 14:36:28 +0000 (14:36 +0000)]
Release 0.10.20
Original commit message from CVS:
Release 0.10.20
Jan Schmidt [Wed, 18 Jun 2008 14:32:12 +0000 (14:32 +0000)]
Update .po files
Original commit message from CVS:
Update .po files
Stefan Kost [Wed, 18 Jun 2008 06:31:11 +0000 (06:31 +0000)]
Fix gtk-doc warnings. Also don't misuse api-doc comments for normal comments.
Original commit message from CVS:
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
* examples/app/appsrc-ra.c:
* examples/app/appsrc-seekable.c:
* examples/app/appsrc-stream.c:
* examples/app/appsrc-stream2.c:
* ext/directfb/dfbvideosink.h:
* ext/metadata/gstbasemetadata.c:
* ext/metadata/gstbasemetadata.h:
* ext/metadata/metadata.c:
* ext/metadata/metadataexif.c:
* ext/theora/theoradec.h:
* gst/deinterlace2/gstdeinterlace2.h:
* gst/deinterlace2/tvtime/speedy.c:
* gst/deinterlace2/tvtime/speedy.h:
* gst/deinterlace2/tvtime/vfir.c:
Fix gtk-doc warnings. Also don't misuse api-doc comments for normal
comments.
Andy Wingo [Mon, 16 Jun 2008 14:11:36 +0000 (14:11 +0000)]
gst-libs/gst/app/gstappsrc.c (gst_app_src_set_max_bytes)
Original commit message from CVS:
2008-06-16 Andy Wingo <wingo@pobox.com>
* gst-libs/gst/app/gstappsrc.c (gst_app_src_set_max_bytes)
(gst_app_src_get_max_bytes, gst_app_src_push_buffer): Use
G_GUINT64_FORMAT. Avoid overflow in get_max_bytes().
Stefan Kost [Mon, 16 Jun 2008 07:30:32 +0000 (07:30 +0000)]
Final round of doc updates.
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
* gst/speed/gstspeed.c:
* gst/speexresample/gstspeexresample.c:
* gst/videosignal/gstvideoanalyse.c:
* gst/videosignal/gstvideodetect.c:
* gst/videosignal/gstvideomark.c:
* sys/dvb/gstdvbsrc.c:
* sys/oss4/oss4-mixer.c:
* sys/oss4/oss4-sink.c:
* sys/oss4/oss4-source.c:
* sys/wininet/gstwininetsrc.c:
Final round of doc updates.
Stefan Kost [Fri, 13 Jun 2008 11:59:21 +0000 (11:59 +0000)]
docs/plugins/: docs/plugins/inspect/plugin-mythtv.xml
Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-bad-plugins-docs.sgml:
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
* docs/plugins/gst-plugins-bad-plugins.args:
* docs/plugins/gst-plugins-bad-plugins.hierarchy:
* docs/plugins/gst-plugins-bad-plugins.interfaces:
* docs/plugins/gst-plugins-bad-plugins.prerequisites:
* docs/plugins/gst-plugins-bad-plugins.signals:
* docs/plugins/inspect/plugin-alsaspdif.xml:
* docs/plugins/inspect/plugin-amrwb.xml:
* docs/plugins/inspect/plugin-app.xml:
* docs/plugins/inspect/plugin-bayer.xml:
* docs/plugins/inspect/plugin-bz2.xml:
* docs/plugins/inspect/plugin-cdaudio.xml:
* docs/plugins/inspect/plugin-cdxaparse.xml:
* docs/plugins/inspect/plugin-dtsdec.xml:
* docs/plugins/inspect/plugin-dvb.xml:
* docs/plugins/inspect/plugin-dvdspu.xml:
* docs/plugins/inspect/plugin-faac.xml:
* docs/plugins/inspect/plugin-faad.xml:
* docs/plugins/inspect/plugin-fbdevsink.xml:
* docs/plugins/inspect/plugin-festival.xml:
* docs/plugins/inspect/plugin-filter.xml:
* docs/plugins/inspect/plugin-flvdemux.xml:
* docs/plugins/inspect/plugin-freeze.xml:
* docs/plugins/inspect/plugin-gsm.xml:
* docs/plugins/inspect/plugin-gstinterlace.xml:
* docs/plugins/inspect/plugin-gstrtpmanager.xml:
* docs/plugins/inspect/plugin-h264parse.xml:
* docs/plugins/inspect/plugin-interleave.xml:
* docs/plugins/inspect/plugin-jack.xml:
* docs/plugins/inspect/plugin-ladspa.xml:
* docs/plugins/inspect/plugin-metadata.xml:
* docs/plugins/inspect/plugin-mms.xml:
* docs/plugins/inspect/plugin-modplug.xml:
* docs/plugins/inspect/plugin-mpeg2enc.xml:
* docs/plugins/inspect/plugin-mpeg4videoparse.xml:
* docs/plugins/inspect/plugin-mpegtsparse.xml:
* docs/plugins/inspect/plugin-mpegvideoparse.xml:
* docs/plugins/inspect/plugin-musepack.xml:
* docs/plugins/inspect/plugin-musicbrainz.xml:
* docs/plugins/inspect/plugin-mve.xml:
* docs/plugins/inspect/plugin-mythtv.xml
* docs/plugins/inspect/plugin-nas.xml:
* docs/plugins/inspect/plugin-neon.xml:
* docs/plugins/inspect/plugin-nsfdec.xml:
* docs/plugins/inspect/plugin-nuvdemux.xml:
* docs/plugins/inspect/plugin-oss4.xml
* docs/plugins/inspect/plugin-rawparse.xml:
* docs/plugins/inspect/plugin-real.xml:
* docs/plugins/inspect/plugin-replaygain.xml:
* docs/plugins/inspect/plugin-rfbsrc.xml:
* docs/plugins/inspect/plugin-sdl.xml:
* docs/plugins/inspect/plugin-sdp.xml:
* docs/plugins/inspect/plugin-selector.xml:
* docs/plugins/inspect/plugin-sndfile.xml:
* docs/plugins/inspect/plugin-soundtouch.xml:
* docs/plugins/inspect/plugin-spcdec.xml:
* docs/plugins/inspect/plugin-speed.xml:
* docs/plugins/inspect/plugin-speexresample.xml:
* docs/plugins/inspect/plugin-stereo.xml:
* docs/plugins/inspect/plugin-subenc.xml
* docs/plugins/inspect/plugin-timidity.xml:
* docs/plugins/inspect/plugin-tta.xml:
* docs/plugins/inspect/plugin-vcdsrc.xml:
* docs/plugins/inspect/plugin-videosignal.xml:
* docs/plugins/inspect/plugin-vmnc.xml:
* docs/plugins/inspect/plugin-wildmidi.xml:
* docs/plugins/inspect/plugin-x264.xml:
* docs/plugins/inspect/plugin-xvid.xml:
* docs/plugins/inspect/plugin-y4menc.xml:
* ext/amrwb/gstamrwbdec.c:
* ext/amrwb/gstamrwbenc.c:
* ext/amrwb/gstamrwbparse.c:
* ext/dc1394/gstdc1394.c:
* ext/directfb/dfbvideosink.c:
* ext/ivorbis/vorbisdec.c:
* ext/jack/gstjackaudiosink.c:
* ext/mpeg2enc/gstmpeg2enc.cc:
* ext/mplex/gstmplex.cc:
* ext/musicbrainz/gsttrm.c:
* ext/mythtv/gstmythtvsrc.c:
* ext/theora/theoradec.c:
* ext/timidity/gsttimidity.c:
* ext/timidity/gstwildmidi.c:
* gst-libs/gst/app/gstappsink.c:
* gst/deinterlace/gstdeinterlace.c:
* gst/dvdspu/gstdvdspu.c:
* gst/festival/gstfestival.c:
* gst/freeze/gstfreeze.c:
* gst/interleave/deinterleave.c:
* gst/interleave/interleave.c:
* gst/modplug/gstmodplug.cc:
* gst/nuvdemux/gstnuvdemux.c:
Add missing elements to docs. Fix doc-markup: use convinience syntax
for examples (produces valid docbook), add several refsec2 when we
have several titles. Fix some types.
Wim Taymans [Thu, 12 Jun 2008 15:47:03 +0000 (15:47 +0000)]
examples/app/: Add beefed up example app from bug #413418. It now also uses appsink instead of fakesink for more ulti...
Original commit message from CVS:
* examples/app/.cvsignore:
* examples/app/Makefile.am:
* examples/app/appsink-src.c: (on_new_buffer_from_source),
(on_source_message), (on_sink_message), (main):
Add beefed up example app from bug #413418. It now also uses appsink
instead of fakesink for more ultimate coolness.
* gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init),
(gst_app_src_init), (gst_app_src_set_property),
(gst_app_src_get_property), (gst_app_src_unlock),
(gst_app_src_unlock_stop), (gst_app_src_create),
(gst_app_src_set_max_bytes), (gst_app_src_push_buffer),
(gst_app_src_end_of_stream):
* gst-libs/gst/app/gstappsrc.h:
Add block property to allow push based implementation to block when we
fill up the appsrc queues.
Emit the enough-data signal while releasing our lock.
Stefan Kost [Thu, 12 Jun 2008 14:50:27 +0000 (14:50 +0000)]
examples/app/.cvsignore: Ignore more.
Original commit message from CVS:
* examples/app/.cvsignore:
Ignore more.
Stefan Kost [Thu, 12 Jun 2008 14:49:15 +0000 (14:49 +0000)]
Do not use short_description in section docs for elements. We extract them from element details and there will be war...
Original commit message from CVS:
* ext/dc1394/gstdc1394.c:
* ext/ivorbis/vorbisdec.c:
* ext/jack/gstjackaudiosink.c:
* ext/metadata/gstmetadatademux.c:
* ext/mythtv/gstmythtvsrc.c:
* ext/theora/theoradec.c:
* gst-libs/gst/app/gstappsink.c:
* gst/bayer/gstbayer2rgb.c:
* gst/deinterlace/gstdeinterlace.c:
* gst/rawparse/gstaudioparse.c:
* gst/rawparse/gstvideoparse.c:
* gst/rtpmanager/gstrtpbin.c:
* gst/rtpmanager/gstrtpclient.c:
* gst/rtpmanager/gstrtpjitterbuffer.c:
* gst/rtpmanager/gstrtpptdemux.c:
* gst/rtpmanager/gstrtpsession.c:
* gst/rtpmanager/gstrtpssrcdemux.c:
* gst/selector/gstinputselector.c:
* gst/selector/gstoutputselector.c:
* gst/videosignal/gstvideoanalyse.c:
* gst/videosignal/gstvideodetect.c:
* gst/videosignal/gstvideomark.c:
* sys/oss4/oss4-mixer.c:
* sys/oss4/oss4-sink.c:
* sys/oss4/oss4-source.c:
Do not use short_description in section docs for elements. We extract
them from element details and there will be warnings if they differ.
Also fixing up the ChangeLog order.
Jan Schmidt [Wed, 11 Jun 2008 21:17:01 +0000 (21:17 +0000)]
configure.ac: 0.10.19.3 pre-release
Original commit message from CVS:
* configure.ac:
0.10.19.3 pre-release
David Schleef [Wed, 11 Jun 2008 20:13:00 +0000 (20:13 +0000)]
gst-libs/gst/rtsp/gstrtspconnection.c: Fix build on win32.
Original commit message from CVS:
* gst-libs/gst/rtsp/gstrtspconnection.c:
Fix build on win32.
Patch By: David Schleef <ds@schleef.org>
Fixes: #536874
Sebastian Dröge [Wed, 11 Jun 2008 09:35:51 +0000 (09:35 +0000)]
ext/gio/gstgiobasesrc.*: Try to read the requested number of bytes, even if the first read returns less than requeste...
Original commit message from CVS:
* ext/gio/gstgiobasesrc.c: (gst_gio_base_src_finalize),
(gst_gio_base_src_create):
* ext/gio/gstgiobasesrc.h:
Try to read the requested number of bytes, even if the first
read returns less than requested, until nothing is read anymore
or we have the requested amount of bytes. This fixes playback of
files via Samba as Samba only allows to read 64k at once.
Implement a caching algorithm that makes sure that we read at
least 4k of data every time. Some elements will try to read a few
bytes, then seek, read again a few bytes and so on and this is
painfully slow as every operation has to go over DBus if GVfs is
used as backend.
Fixes bug #536849 and #536848.
* ext/gio/gstgiosrc.c: (gst_gio_src_class_init),
(gst_gio_src_check_get_range):
Override check_get_range() to blacklist http/https URIs
and whitelist file URIs. More to be added on demand.
Wim Taymans [Fri, 6 Jun 2008 16:50:51 +0000 (16:50 +0000)]
examples/app/: Added 3 more example application for using appsrc in random-access mode, pull-mode streaming and pull ...
Original commit message from CVS:
* examples/app/Makefile.am:
* examples/app/appsrc-ra.c: (feed_data), (seek_data),
(found_source), (bus_message), (main):
* examples/app/appsrc-seekable.c: (feed_data), (seek_data),
(found_source), (bus_message), (main):
* examples/app/appsrc-stream2.c: (feed_data), (found_source),
(bus_message), (main):
Added 3 more example application for using appsrc in random-access mode,
pull-mode streaming and pull mode seekable.
* gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init),
(gst_app_src_start), (gst_app_src_do_get_size),
(gst_app_src_create):
* gst-libs/gst/app/gstappsrc.h:
Make stream-type property writable.
Unset flushing when starting so that we reuse appsrc.
Inform basesrc about the configured size.
Emit seek-data signal when we are going to a different offset in
random-access mode.
Wim Taymans [Fri, 6 Jun 2008 14:19:54 +0000 (14:19 +0000)]
examples/app/appsrc-stream.c: Use deep-notify until we can depend on a playbin2 with support for the source property.
Original commit message from CVS:
* examples/app/appsrc-stream.c: (found_source), (main):
Use deep-notify until we can depend on a playbin2 with support for the
source property.
Wim Taymans [Thu, 5 Jun 2008 16:38:50 +0000 (16:38 +0000)]
examples/app/: Added an example on how to use appsrc in playbin in streaming mode from an mmapped file.
Original commit message from CVS:
* examples/app/.cvsignore:
* examples/app/Makefile.am:
* examples/app/appsrc-stream.c: (read_data), (start_feed),
(stop_feed), (found_source), (bus_message), (main):
Added an example on how to use appsrc in playbin in streaming mode from
an mmapped file.
* examples/app/appsrc_ex.c: (main):
Set pipeline to NULL to free queued buffers.
* gst-libs/gst/app/gstapp-marshal.list:
* gst-libs/gst/app/gstappsrc.c: (stream_type_get_type), (_do_init),
(gst_app_src_class_init), (gst_app_src_init),
(gst_app_src_flush_queued), (gst_app_src_dispose),
(gst_app_src_set_property), (gst_app_src_get_property),
(gst_app_src_unlock), (gst_app_src_unlock_stop),
(gst_app_src_start), (gst_app_src_stop), (gst_app_src_is_seekable),
(gst_app_src_check_get_range), (gst_app_src_do_seek),
(gst_app_src_create), (gst_app_src_set_stream_type),
(gst_app_src_get_stream_type), (gst_app_src_set_max_bytes),
(gst_app_src_get_max_bytes), (gst_app_src_push_buffer),
(gst_app_src_end_of_stream), (gst_app_src_uri_get_type),
(gst_app_src_uri_get_protocols), (gst_app_src_uri_get_uri),
(gst_app_src_uri_set_uri), (gst_app_src_uri_handler_init):
* gst-libs/gst/app/gstappsrc.h:
Measure max queue size in bytes instead.
Add support for 3 modes of operation, streaming, seekable and
random-access, making basesrc handle the scheduling modes for each.
Add appsrc:// uri handler so that automatic plugging can be done from
playbin2 or uridecodebin, for example.
Added support for custom segment formats.
Add support for push and pull based operations from the application.
Expand the methods so that errors can be detected.
Flush the queued buffers on seeks and when shutting down.
Add signals to inform the app that a seek must happen.
Jan Schmidt [Thu, 5 Jun 2008 09:47:23 +0000 (09:47 +0000)]
configure.ac: 0.10.19.2 pre-release
Original commit message from CVS:
* configure.ac:
0.10.19.2 pre-release
Jan Schmidt [Wed, 4 Jun 2008 21:48:27 +0000 (21:48 +0000)]
win32/common/: Add new API functions to the dll exports
Original commit message from CVS:
* win32/common/libgstrtsp.def:
* win32/common/libgsttag.def:
Add new API functions to the dll exports
Michael Smith [Wed, 4 Jun 2008 17:42:38 +0000 (17:42 +0000)]
gst/playback/gstplaybasebin.c: Disconnect signals from decodebins we created before we remove it from playbin, to avo...
Original commit message from CVS:
* gst/playback/gstplaybasebin.c:
Disconnect signals from decodebins we created before we remove it from
playbin, to avoid crashes if the decodebin is eventually disposed after
the playbin itself (possible if the app takes a reference on the
decodebin).
Fixes #536521.
Tim-Philipp Müller [Wed, 4 Jun 2008 17:12:40 +0000 (17:12 +0000)]
gst/typefind/gsttypefindfunctions.c: Bunch of small clean-ups: use gst_type_find_suggest_simple(); don't copy caps fo...
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (aac_type_find),
(mp3_type_find), (musepack_type_find), (MULTIPART_MAX_HEADER_SIZE),
(mpeg_sys_type_find), (mpeg_ts_type_find), (mpeg4_video_type_find),
(h264_video_type_find), (mpeg_video_stream_type_find),
(dv_type_find), (mmsh_type_find):
Bunch of small clean-ups: use gst_type_find_suggest_simple(); don't
copy caps for no good reason (this may be desirable to make it easier
to detect leaks, but then it should probably be done for all caps
in the typefinder somewhere).
Peter Kjellerstedt [Wed, 4 Jun 2008 16:06:49 +0000 (16:06 +0000)]
tests/check/Makefile.am: Do not try to run the check tests for subparse unless it has been built.
Original commit message from CVS:
* tests/check/Makefile.am:
Do not try to run the check tests for subparse unless it has been
built.
Peter Kjellerstedt [Wed, 4 Jun 2008 16:00:26 +0000 (16:00 +0000)]
tests/check/pipelines/streamheader.c: Do not try to run a test which requires vorbisenc unless we have actually built...
Original commit message from CVS:
* tests/check/pipelines/streamheader.c: (buffer_probe_cb),
(test_multifdsink_gdp_vorbisenc), (streamheader_suite):
Do not try to run a test which requires vorbisenc unless we have
actually built it.
Peter Kjellerstedt [Wed, 4 Jun 2008 11:53:53 +0000 (11:53 +0000)]
gst-libs/gst/rtsp/gstrtspconnection.*: Add a couple of missing argument guards.
Original commit message from CVS:
* gst-libs/gst/rtsp/gstrtspconnection.c:
(gst_rtsp_connection_set_auth), (gst_rtsp_connection_set_auth_param),
(gst_rtsp_connection_clear_auth_params),
(gst_rtsp_connection_set_qos_dscp), (gst_rtsp_connection_get_ip):
* gst-libs/gst/rtsp/gstrtspconnection.h:
Add a couple of missing argument guards.
Add a way of setting the DSCP for an RTSP connection.
Add an accessor method for the ip member of GstRTSPConnection as all
members are supposed to be private.
Peter Kjellerstedt [Wed, 4 Jun 2008 11:33:23 +0000 (11:33 +0000)]
gst/tcp/gstmultifdsink.c: Fixed accidental use of IPv4 options for all IPv6 addresses.
Original commit message from CVS:
* gst/tcp/gstmultifdsink.c: (setup_dscp_client):
Fixed accidental use of IPv4 options for all IPv6 addresses.
Tim-Philipp Müller [Wed, 4 Jun 2008 10:18:42 +0000 (10:18 +0000)]
gst-libs/gst/interfaces/mixertrack.h: Document mixer track flags.
Original commit message from CVS:
* gst-libs/gst/interfaces/mixertrack.h:
Document mixer track flags.
Antoine Tremblay [Wed, 4 Jun 2008 05:58:38 +0000 (05:58 +0000)]
gst/gdp/gstgdppay.c: Don't set caps on the buffers that contain a copy of the buffer including the caps of them resul...
Original commit message from CVS:
Patch by: Antoine Tremblay <hexa00 at gmail dot com>
* gst/gdp/gstgdppay.c: (gst_gdp_pay_reset_streamheader):
Don't set caps on the buffers that contain a copy of the buffer
including the caps of them resulting in an always increasing refcount
of the caps and insanely large caps. Instead include a buffer without
caps in the new caps. Fixes bug #536475.
Sebastian Dröge [Wed, 4 Jun 2008 05:44:06 +0000 (05:44 +0000)]
gst/videoscale/gstvideoscale.c: Transform a given PAR to a range on the struct with the generic height/width instead ...
Original commit message from CVS:
* gst/videoscale/gstvideoscale.c: (gst_video_scale_transform_caps):
Transform a given PAR to a range on the struct with the generic
height/width instead of the struct with the possibly restricted
height/width.
Sebastian Dröge [Wed, 4 Jun 2008 04:24:27 +0000 (04:24 +0000)]
gst/videoscale/gstvideoscale.c: Prefer the given format if it contains something stricter than [1,MAX] for height or ...
Original commit message from CVS:
* gst/videoscale/gstvideoscale.c: (gst_video_scale_transform_caps):
Prefer the given format if it contains something stricter than [1,MAX]
for height or width and only put a structure that requires rescaling
as second. This makes it possible to use videoscale in pipelines where
the source can actually produce the wanted height/width but usually
selects a different one from the requested.
John Millikin [Tue, 3 Jun 2008 20:01:58 +0000 (20:01 +0000)]
gst-libs/gst/tag/gstvorbistag.c: Retrieve COVERART tags from vorbis comments (#512333)
Original commit message from CVS:
Based on patch by: John Millikin <jmillikin gmail com>
* gst-libs/gst/tag/gstvorbistag.c: (tag_matches), (gst_vorbis_tag_add),
(gst_vorbis_tag_add_coverart):
Retrieve COVERART tags from vorbis comments (#512333)
Tim-Philipp Müller [Tue, 3 Jun 2008 19:44:48 +0000 (19:44 +0000)]
gst-libs/gst/tag/: Don't forget to add new enum value here too (should probably use glib-mkenums here...).
Original commit message from CVS:
* gst-libs/gst/tag/tag.h:
* gst-libs/gst/tag/tags.c: (register_tag_image_type_enum):
Don't forget to add new enum value here too (should probably use
glib-mkenums here...).
Tim-Philipp Müller [Tue, 3 Jun 2008 19:29:06 +0000 (19:29 +0000)]
gst-libs/gst/tag/: API: add gst_tag_image_data_to_image_buffer()
Original commit message from CVS:
* gst-libs/gst/tag/gstid3tag.c: (gst_tag_list_add_id3_image):
* gst-libs/gst/tag/tag.h: (GST_TAG_IMAGE_TYPE_NONE),
* gst-libs/gst/tag/tags.c: (register_tag_image_type_enum),
(gst_tag_image_type_get_type), (gst_tag_image_type_is_valid),
(gst_tag_image_data_to_image_buffer):
Add two utility functions to avoid code duplication (#512333):
API: add gst_tag_image_data_to_image_buffer()
API: add gst_tag_list_add_id3_image()
Sebastian Dröge [Tue, 3 Jun 2008 08:54:29 +0000 (08:54 +0000)]
win32/common/libgstaudio.def: Add gst_audio_check_channel_positions() to the exported symbols.
Original commit message from CVS:
* win32/common/libgstaudio.def:
Add gst_audio_check_channel_positions() to the exported symbols.
Sebastian Dröge [Tue, 3 Jun 2008 08:48:32 +0000 (08:48 +0000)]
API: Make gst_audio_check_channel_positions() public.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/audio/multichannel.c:
(gst_audio_check_channel_positions):
* gst-libs/gst/audio/multichannel.h:
API: Make gst_audio_check_channel_positions() public.
* tests/check/libs/audio.c: (GST_START_TEST):
Add some simple checks for gst_audio_check_channel_positions().
Tim-Philipp Müller [Mon, 2 Jun 2008 20:09:14 +0000 (20:09 +0000)]
sys/v4l/v4l_calls.c: minrange and maxrange are scaled according to the frequency multiplier.
Original commit message from CVS:
* sys/v4l/v4l_calls.c: (gst_v4l_get_chan_names):
minrange and maxrange are scaled according to the frequency
multiplier.
Tim-Philipp Müller [Mon, 2 Jun 2008 18:37:02 +0000 (18:37 +0000)]
ext/pango/: Use gstvideo functions to calculate strides and plane offsets. Fixes rendering issue ('ghost' images of t...
Original commit message from CVS:
* ext/pango/Makefile.am:
* ext/pango/gsttextoverlay.c: (gst_text_overlay_shade_y),
(gst_text_overlay_blit_yuv420), (gst_text_overlay_push_frame):
Use gstvideo functions to calculate strides and plane offsets. Fixes
rendering issue ('ghost' images of the text on the chroma planes)
with widths or heights that are not multiples of 8 (#506659 and
probably also #485729).
* tests/icles/test-textoverlay.c: (show_text), (test_textoverlay),
(main):
Test with odd height/width too.
Sebastian Dröge [Mon, 2 Jun 2008 12:20:35 +0000 (12:20 +0000)]
gst/adder/gstadder.c: When using gst_element_iterate_pads() one has to unref every pad after usage.
Original commit message from CVS:
* gst/adder/gstadder.c: (gst_adder_query_duration),
(gst_adder_query_latency):
When using gst_element_iterate_pads() one has to unref every pad
after usage.
Mark Nauwelaerts [Sat, 31 May 2008 19:57:57 +0000 (19:57 +0000)]
gst-libs/gst/audio/gstbaseaudiosrc.c: Add a gtk-doc chunk for the new properties to have a Since: indication.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_class_init):
Add a gtk-doc chunk for the new properties to have a Since: indication.
Mark Nauwelaerts [Sat, 31 May 2008 19:50:59 +0000 (19:50 +0000)]
ChangeLog surgery, mark API change
Original commit message from CVS:
ChangeLog surgery, mark API change
Mark Nauwelaerts [Sat, 31 May 2008 18:10:47 +0000 (18:10 +0000)]
gst-libs/gst/audio/gstbaseaudiosrc.c: Provide readable actual-buffer-time and actual-latency-time properties that ref...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_class_init), (gst_base_audio_src_dispose),
(gst_base_audio_src_get_property), (gst_base_audio_src_setcaps),
(gst_base_audio_src_change_state):
Provide readable actual-buffer-time and actual-latency-time properties
that reflect the configured ringbuffer values. Fixes #524724.
Wim Taymans [Fri, 30 May 2008 15:29:20 +0000 (15:29 +0000)]
gst-libs/gst/rtp/gstbasertppayload.c: Simply converting the running time into an RTP timestamp by scaling it based on...
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push),
(gst_basertppayload_change_state):
Simply converting the running time into an RTP timestamp by scaling it
based on the clock-rate is good enough for making an RTP timestamp. This
has the added benefit that we can later on expose a property with the
RTP timestamp of running time 0, as is needed for RTSP servers to
generate the response of the PLAY request.
Sebastian Dröge [Fri, 30 May 2008 08:42:17 +0000 (08:42 +0000)]
gst/audioconvert/gstaudioconvert.c: Allow up to 11 positioned channels now that audioconvert can handle this but add ...
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
(structure_has_fixed_channel_positions),
(gst_audio_convert_transform_caps):
Allow up to 11 positioned channels now that audioconvert can handle
this but add no default positions for > 8 channels.
* tests/check/elements/audioconvert.c: (GST_START_TEST):
Add some unit tests for the above change: Test conversion of
11 positioned channels to stereo and the other way around, test
conversion of 15 unpositioned channels in different ways.
Sebastian Dröge [Thu, 29 May 2008 19:45:40 +0000 (19:45 +0000)]
win32/common/libgstaudio.def: Add gst_audio_clock_reset to the list of exported symbols.
Original commit message from CVS:
* win32/common/libgstaudio.def:
Add gst_audio_clock_reset to the list of exported symbols.
Sebastian Dröge [Thu, 29 May 2008 19:37:47 +0000 (19:37 +0000)]
tests/check/elements/vorbisdec.c: Remove wrong_channels_identification_header unit test as we now support 7 (and more...
Original commit message from CVS:
* tests/check/elements/vorbisdec.c: (vorbisdec_suite):
Remove wrong_channels_identification_header unit test as we now
support 7 (and more channels).
Sebastian Dröge [Thu, 29 May 2008 12:17:16 +0000 (12:17 +0000)]
gst/audioconvert/gstchannelmix.c: If mixing left or right to center (or the other way around) only take the complete ...
Original commit message from CVS:
* gst/audioconvert/gstchannelmix.c:
(gst_channel_mix_fill_one_other):
If mixing left or right to center (or the other way around) only take
the complete value if we don't already have the original position in
the source.
Sebastian Dröge [Thu, 29 May 2008 11:34:09 +0000 (11:34 +0000)]
gst-libs/gst/audio/multichannel.c: Allow rear center together with rear left/right and other previously conflicting c...
Original commit message from CVS:
* gst-libs/gst/audio/multichannel.c:
(gst_audio_check_channel_positions),
(gst_audio_set_structure_channel_positions_list),
(gst_audio_fixate_channel_positions):
Allow rear center together with rear left/right and other previously
conflicting channel positions. The reason why they weren't allowed
was the channel mixing implementation in audioconvert.
Also take this into account when fixing channel layouts.
Allow setting channel positions for 1/2 channels when using
gst_audio_set_structure_channel_position().
* gst/audioconvert/gstchannelmix.c:
(gst_channel_mix_fill_compatible), (gst_channel_mix_detect_pos),
(gst_channel_mix_fill_one_other), (gst_channel_mix_fill_others),
(gst_channel_mix_fill_special), (gst_channel_mix_fill_matrix):
Major rewrite of the channel mixing.
We now allow previously conflicting channel positions to appear
together (rear center and rear left/right for example).
Fixes bug #533817.
Rework the way channels are mixed together to take more possible
channel positions into account, properly mix from/to side channels
and don't assume that either center, left&right or nothing of a
specific position is available anymore.
* tests/check/elements/audioconvert.c: (GST_START_TEST):
Adjust unit tests with non-standard 1/2 channel layouts to the more
correct new behaviour.
Add a unit test for 5.1->Stereo downmixing.
Sebastian Dröge [Thu, 29 May 2008 07:02:50 +0000 (07:02 +0000)]
ext/vorbis/: Add sane defaults for the 7 and 8 channel layouts as those are undefined in the Vorbis spec. Use NONE ch...
Original commit message from CVS:
* ext/vorbis/vorbisdec.c: (vorbis_handle_identification_packet):
* ext/vorbis/vorbisenc.c: (gst_vorbis_enc_generate_sink_caps):
Add sane defaults for the 7 and 8 channel layouts as those are
undefined in the Vorbis spec. Use NONE channel layouts when decoding
more than 8 channels instead of erroring out. Fixes bug #535356.
Wim Taymans [Wed, 28 May 2008 16:10:20 +0000 (16:10 +0000)]
Add theoraparse to the docs and fix some docs.
Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
* docs/plugins/gst-plugins-base-plugins-sections.txt:
* ext/theora/theoraparse.c:
Add theoraparse to the docs and fix some docs.
Wim Taymans [Wed, 28 May 2008 15:48:33 +0000 (15:48 +0000)]
gst-libs/gst/cdda/gstcddabasesrc.c: Fix EOS condition and track addition check, the track.end sector is included in t...
Original commit message from CVS:
* gst-libs/gst/cdda/gstcddabasesrc.c:
(gst_cdda_base_src_add_track), (gst_cdda_base_src_create):
Fix EOS condition and track addition check, the track.end sector is
included in the track. Fixes #533265.
Mark Nauwelaerts [Wed, 28 May 2008 14:49:24 +0000 (14:49 +0000)]
gst/videorate/gstvideorate.*: React (more) to NEWSEGMENT
Original commit message from CVS:
Patch by: Mark Nauwelaerts <manauw at skynet be>
* gst/videorate/gstvideorate.c: (gst_video_rate_reset),
(gst_video_rate_flush_prev), (gst_video_rate_event),
(gst_video_rate_chain):
* gst/videorate/gstvideorate.h:
React (more) to NEWSEGMENT
Small adjustment in timestamp calculation to prevent mismatches
Fixes #435633.
Tim-Philipp Müller [Wed, 28 May 2008 11:31:44 +0000 (11:31 +0000)]
tests/examples/seek/seek.c: Initialise error to NULL as we should.
Original commit message from CVS:
* tests/examples/seek/seek.c: (make_parselaunch_pipeline):
Initialise error to NULL as we should.
Sebastian Dröge [Wed, 28 May 2008 08:14:47 +0000 (08:14 +0000)]
gst/adder/gstadder.c: Implement latency query.
Original commit message from CVS:
* gst/adder/gstadder.c: (gst_adder_query_duration),
(gst_adder_query_latency), (gst_adder_query):
Implement latency query.
Sebastian Dröge [Tue, 27 May 2008 18:10:00 +0000 (18:10 +0000)]
gst/adder/gstadder.c: Correctly resync the iterator if gst_iterator_next() returns
Original commit message from CVS:
* gst/adder/gstadder.c: (gst_adder_query_duration):
Correctly resync the iterator if gst_iterator_next() returns
GST_ITERATOR_RESYNC.
Tim-Philipp Müller [Tue, 27 May 2008 17:14:07 +0000 (17:14 +0000)]
win32/vs6/libgstpbutils.dsp: Add pbutils-enumtypes.c to sources (#518037).
Original commit message from CVS:
* win32/vs6/libgstpbutils.dsp:
Add pbutils-enumtypes.c to sources (#518037).
Wim Taymans [Tue, 27 May 2008 16:20:17 +0000 (16:20 +0000)]
gst-libs/gst/audio/gstaudioclock.*: Add method to inform the clock that the time starts from 0 again. We use this inf...
Original commit message from CVS:
* gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_init),
(gst_audio_clock_reset), (gst_audio_clock_get_internal_time):
* gst-libs/gst/audio/gstaudioclock.h:
Add method to inform the clock that the time starts from 0 again. We use
this info to calculate a clock offset so that the time we report in
internal_time is monotonically increasing, as required by the clock base
class. Fixes #521761.
API: GstAudioClock::gst_audio_clock_reset()
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_skew_slaving),
(gst_base_audio_sink_change_state):
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_create), (gst_base_audio_src_change_state):
Reset reported time when we (re)create the ringbuffer.
Tim-Philipp Müller [Tue, 27 May 2008 16:11:32 +0000 (16:11 +0000)]
ext/alsa/gstalsamixertrack.c: Make sure playback volumes aren't accidentally overwritten by capture volumes if an als...
Original commit message from CVS:
* ext/alsa/gstalsamixertrack.c:
(gst_alsa_mixer_track_update_alsa_capabilities):
Make sure playback volumes aren't accidentally overwritten by
capture volumes if an alsa mixer track has both playback and
capture capabilities: we create two GstMixerTracks in that
case, so make sure we query only the alsa capabilities that
refer to the type of GstMixerTrack we created from the dual
capability alsa element. Should fix issues with Audigy2 sound
cards (#518082).
Tim-Philipp Müller [Tue, 27 May 2008 10:57:56 +0000 (10:57 +0000)]
tests/check/pipelines/oggmux.c: Don't use deprecated function.
Original commit message from CVS:
* tests/check/pipelines/oggmux.c: (test_pipeline):
Don't use deprecated function.
Wim Taymans [Tue, 27 May 2008 10:35:55 +0000 (10:35 +0000)]
gst/playback/gstdecodebin2.c: Check for NULL cases and log them, creating ghostpads can, for example, fail when the p...
Original commit message from CVS:
* gst/playback/gstdecodebin2.c:
(gst_decode_group_control_source_pad), (gst_decode_group_expose):
Check for NULL cases and log them, creating ghostpads can, for example,
fail when the pad returns wrong caps.
* gst/playback/gstplaybin2.c: (perform_eos):
When pushing out the EOS event, collect the return value and warn when
something failed.
Wim Taymans [Mon, 26 May 2008 17:18:52 +0000 (17:18 +0000)]
gst-libs/gst/riff/riff-media.c: Add support for DVCPRO.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
(gst_riff_create_video_template_caps):
Add support for DVCPRO.
Tim-Philipp Müller [Mon, 26 May 2008 10:29:20 +0000 (10:29 +0000)]
gst/videoscale/gstvideoscale.c: Change default scaling method from nearest-neighbour to bilinear.
Original commit message from CVS:
* gst/videoscale/gstvideoscale.c: (DEFAULT_PROP_METHOD):
Change default scaling method from nearest-neighbour to bilinear.
Tim-Philipp Müller [Mon, 26 May 2008 10:26:00 +0000 (10:26 +0000)]
tests/check/libs/video.c: More checks.
Original commit message from CVS:
* tests/check/libs/video.c:
More checks.
Tim-Philipp Müller [Sun, 25 May 2008 20:51:35 +0000 (20:51 +0000)]
Limit duration to a maximum of five seconds for tmplayer format where we can guess the duration only from the timesta...
Original commit message from CVS:
* gst/subparse/gstsubparse.c: (parser_state_init),
(gst_sub_parse_format_autodetect), (handle_buffer):
* gst/subparse/gstsubparse.h:
* tests/check/elements/subparse.c: (test_tmplayer_style3b):
Limit duration to a maximum of five seconds for tmplayer format where
we can guess the duration only from the timestamp of the next line of
text. We don't want to show a text for eternities just because nothing
else is being said for a while.
Wim Taymans [Fri, 23 May 2008 14:14:28 +0000 (14:14 +0000)]
gst-libs/gst/rtp/gstbasertpdepayload.c: Check sequence numbers, mark input buffers with a discont flag for the subcla...
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_chain),
(gst_base_rtp_depayload_handle_sink_event),
(gst_base_rtp_depayload_push_full),
(gst_base_rtp_depayload_change_state):
Check sequence numbers, mark input buffers with a discont flag for the
subclass when we detected a gap, drop duplicate buffers. We do this
because one can use the element without a jitterbuffer in front and we
don't want to feed the subclasses invalid or reordered data.
Do an error when the subclass did not provide a process function instead
of crashing.
Some other small cleanups.
Tim-Philipp Müller [Thu, 22 May 2008 22:35:40 +0000 (22:35 +0000)]
gst/videotestsrc/videotestsrc.c: May just as well use the precalculated uvstride here.
Original commit message from CVS:
* gst/videotestsrc/videotestsrc.c: (paint_hline_NV12_NV21):
May just as well use the precalculated uvstride here.
Jan Schmidt [Thu, 22 May 2008 22:09:16 +0000 (22:09 +0000)]
Add some documentation comments, and some new headers to be scanned.
Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-base-plugins-overrides.txt:
* docs/plugins/gst-plugins-base-plugins-sections.txt:
* docs/plugins/gst-plugins-base-plugins.args:
* docs/plugins/gst-plugins-base-plugins.hierarchy:
* docs/plugins/gst-plugins-base-plugins.interfaces:
* docs/plugins/gst-plugins-base-plugins.prerequisites:
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-decodebin.xml:
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
* docs/plugins/inspect/plugin-gdp.xml:
* docs/plugins/inspect/plugin-gio.xml:
* docs/plugins/inspect/plugin-gnomevfs.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playback.xml:
* docs/plugins/inspect/plugin-queue2.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-uridecodebin.xml:
* docs/plugins/inspect/plugin-video4linux.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
* ext/cdparanoia/gstcdparanoiasrc.c:
* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggdemux.h:
* ext/ogg/gstoggmux.c:
* ext/ogg/gstoggmux.h:
* gst/audioconvert/audioconvert.c:
* gst/audioconvert/audioconvert.h:
* gst/audioconvert/gstaudioconvert.h:
* gst/gdp/gstgdpdepay.h:
* gst/gdp/gstgdppay.h:
* gst/playback/gstdecodebin.c:
* gst/playback/gstdecodebin2.c:
* gst/playback/gstplaybin.c:
* gst/playback/gstplaybin2.c:
* gst/playback/gsturidecodebin.c:
* gst/tcp/gstmultifdsink.c:
* gst/tcp/gstmultifdsink.h:
* gst/tcp/gsttcp.h:
Add some documentation comments, and some new headers to be scanned.
Rename some internal enum declarations (audioconvert's DitherType and
NoiseShapingType, GstUnitType from the TCP elements) to match the
documented GObject type names so that the docs pick them up.
Name the playbin2 docs markups properly so they get picked up. They'll
need renaming back when/if playbin2 becomes playbin.
100% symbol coverage for the plugin docs, booya.
Thijs Vermeir [Thu, 22 May 2008 18:30:15 +0000 (18:30 +0000)]
gst/videotestsrc/videotestsrc.c: Fix generation of NV12/NV21 frames. Fixes bug #532454.
Original commit message from CVS:
Patch by: Thijs Vermeir <thijsvermeir@gmail.com>
* gst/videotestsrc/videotestsrc.c: (paint_hline_NV12_NV21):
Fix generation of NV12/NV21 frames. Fixes bug #532454.
Sjoerd Simons [Thu, 22 May 2008 11:59:33 +0000 (11:59 +0000)]
gst/playback/gstdecodebin.c: Lock the fakesink before setting the state to NULL and removing it from the bin so that ...
Original commit message from CVS:
Patch by: Sjoerd Simons <sjoerd at luon dot net>
* gst/playback/gstdecodebin.c: (remove_fakesink):
Lock the fakesink before setting the state to NULL and removing it from
the bin so that a concurrent state change cannot interfere.
Fixes #534331.
Felipe Contreras [Wed, 21 May 2008 17:09:42 +0000 (17:09 +0000)]
docs/Makefile.am: Fix installing plugin documentation when gtk-doc is disabled.
Original commit message from CVS:
* docs/Makefile.am:
Fix installing plugin documentation when gtk-doc is disabled.
Felipe Contreras [Wed, 21 May 2008 17:01:16 +0000 (17:01 +0000)]
gst-libs/gst/rtsp/Makefile.am: Distribute, don't install md5.h
Original commit message from CVS:
* gst-libs/gst/rtsp/Makefile.am:
Distribute, don't install md5.h
Julien Moutte [Wed, 21 May 2008 16:47:58 +0000 (16:47 +0000)]
gst/tcp/gstmultifdsink.c: Use IPPROTO_IP instead of SOL_IP, works on more platforms.
Original commit message from CVS:
2008-05-21 Julien Moutte <julien@fluendo.com>
* gst/tcp/gstmultifdsink.c: (setup_dscp_client): Use IPPROTO_IP
instead of SOL_IP, works on more platforms.
* gst/typefind/gsttypefindfunctions.c: (aac_type_find): Fix printf
arguments.
Wim Taymans [Wed, 21 May 2008 16:44:15 +0000 (16:44 +0000)]
Some debug and comment fixes.
Original commit message from CVS:
* ext/vorbis/vorbisdec.c:
* gst/videoscale/gstvideoscale.c: (gst_video_scale_transform):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_show_frame):
Some debug and comment fixes.
* tests/examples/dynamic/addstream.c: (main):
Fix , to ;
Wim Taymans [Wed, 21 May 2008 16:36:50 +0000 (16:36 +0000)]
Don't use bad gst_element_get_pad().
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_typefind):
* gst/playback/decodetest.c: (new_decoded_pad_cb):
* gst/playback/gstdecodebin.c: (gst_decode_bin_init),
(try_to_link_1), (elem_is_dynamic), (close_link), (type_found),
(cleanup_decodebin):
* gst/playback/gstdecodebin2.c: (gst_decode_bin_init),
(connect_element), (gst_decode_group_control_demuxer_pad):
* gst/playback/gstplaybasebin.c: (queue_remove_probe),
(queue_out_of_data), (gen_preroll_element), (preroll_unlinked),
(mute_group_type):
* gst/playback/gstplaybin.c: (gst_play_bin_vis_blocked),
(gst_play_bin_set_property), (handoff), (gen_video_element),
(gen_text_element), (gen_audio_element), (gen_vis_element),
(remove_sinks), (add_sink), (setup_sinks):
* gst/playback/gstplaybin2.c: (pad_added_cb), (no_more_pads_cb):
* gst/playback/gstplaysink.c: (gst_play_sink_get_video_sink),
(gst_play_sink_get_audio_sink), (gst_play_sink_vis_unblocked),
(gst_play_sink_vis_blocked), (gst_play_sink_set_vis_plugin),
(gst_play_sink_get_vis_plugin), (gst_play_sink_set_mute),
(gen_video_chain), (gen_text_chain), (gen_audio_chain),
(gen_vis_chain), (gst_play_sink_reconfigure),
(gst_play_sink_set_font_desc), (gst_play_sink_get_font_desc),
(gst_play_sink_request_pad):
* gst/playback/gsturidecodebin.c: (type_found), (setup_source):
* gst/playback/test.c: (gen_video_element), (gen_audio_element),
(cb_newpad):
* gst/playback/test6.c: (new_decoded_pad_cb):
* tests/check/elements/audioconvert.c: (GST_START_TEST):
* tests/check/elements/audiorate.c: (test_injector_chain),
(do_perfect_stream_test):
* tests/check/elements/ffmpegcolorspace.c: (GST_START_TEST):
* tests/check/elements/gdpdepay.c: (GST_START_TEST):
* tests/check/elements/gnomevfssink.c:
* tests/check/elements/textoverlay.c:
(notgst_check_setup_src_pad2), (notgst_check_teardown_src_pad2):
* tests/check/elements/videotestsrc.c: (GST_START_TEST):
* tests/check/libs/cddabasesrc.c: (GST_START_TEST):
* tests/check/pipelines/oggmux.c: (test_pipeline):
* tests/check/pipelines/streamheader.c: (GST_START_TEST):
* tests/check/pipelines/theoraenc.c: (GST_START_TEST):
* tests/check/pipelines/vorbisenc.c: (GST_START_TEST):
* tests/examples/seek/scrubby.c: (make_wav_pipeline):
* tests/examples/seek/seek.c: (make_mod_pipeline),
(make_dv_pipeline), (make_wav_pipeline), (make_flac_pipeline),
(make_sid_pipeline), (make_parse_pipeline), (make_vorbis_pipeline),
(make_theora_pipeline), (make_vorbis_theora_pipeline),
(make_avi_msmpeg4v3_mp3_pipeline), (make_mp3_pipeline),
(make_avi_pipeline), (make_mpeg_pipeline), (make_mpegnt_pipeline),
(update_fill), (msg_buffering):
Don't use bad gst_element_get_pad().
Stefan Kost [Wed, 21 May 2008 14:35:41 +0000 (14:35 +0000)]
gst-libs/gst/riff/riff-media.c: Fix wrong method name in docs. Fix calculation of strf fields for broken mulaw/alaw.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c:
Fix wrong method name in docs. Fix calculation of strf fields for
broken mulaw/alaw.
* gst-libs/gst/riff/riff-read.c:
Whitespace fix and removing double ';'.
Wim Taymans [Wed, 21 May 2008 11:52:30 +0000 (11:52 +0000)]
docs/design/part-playbin2.txt: Add some leftover doc.
Original commit message from CVS:
* docs/design/part-playbin2.txt:
Add some leftover doc.
Sebastian Dröge [Wed, 21 May 2008 11:36:37 +0000 (11:36 +0000)]
gst/audioconvert/gstchannelmix.c: Fix copy & paste error in last commit.
Original commit message from CVS:
* gst/audioconvert/gstchannelmix.c: (gst_channel_mix_fill_others):
Fix copy & paste error in last commit.
Sebastian Dröge [Wed, 21 May 2008 11:30:58 +0000 (11:30 +0000)]
gst/audioconvert/gstchannelmix.c: Add support for mixing GST_AUDIO_CHANNEL_POSITION_SIDE_* from/to other channel posi...
Original commit message from CVS:
* gst/audioconvert/gstchannelmix.c: (gst_channel_mix_fill_others):
Add support for mixing GST_AUDIO_CHANNEL_POSITION_SIDE_* from/to
other channel positions when source has SIDE channels and dest doesn't
or the other way around.