Commit graph

1355 commits

Author SHA1 Message Date
Tim-Philipp Müller
029f05635d gst/: Recognise Kate subtitle streams (#550582).
Original commit message from CVS:
* gst-libs/gst/pbutils/descriptions.c:
* gst/typefind/gsttypefindfunctions.c:
Recognise Kate subtitle streams (#550582).
2008-09-15 15:11:18 +00:00
Tim-Philipp Müller
b579580991 gst-libs/gst/audio/audio.h: Remove trailing comma from enum list, which causes problems with -pendantic (#550729).
Original commit message from CVS:
* gst-libs/gst/audio/audio.h: (GST_AUDIO_FIELD_SIGNED):
Remove trailing comma from enum list, which causes problems
with -pendantic (#550729).
2008-09-13 11:04:02 +00:00
Tim-Philipp Müller
f5a176bb6c gst-libs/gst/interfaces/propertyprobe.c: More sanity checks for our second-favourite interface.
Original commit message from CVS:
* gst-libs/gst/interfaces/propertyprobe.c:
(gst_property_probe_get_properties),
(gst_property_probe_get_property),
(gst_property_probe_probe_property),
(gst_property_probe_probe_property_name),
(gst_property_probe_needs_probe),
(gst_property_probe_needs_probe_name),
(gst_property_probe_get_values),
(gst_property_probe_get_values_name),
(gst_property_probe_probe_and_get_values),
(gst_property_probe_probe_and_get_values_name):
More sanity checks for our second-favourite interface.
2008-09-05 19:04:47 +00:00
Stefan Kost
aa7b4f5ac7 gst-libs/gst/interfaces/propertyprobe.c: Check for NULL pointer, in the hope that this fixes #532864.
Original commit message from CVS:
* gst-libs/gst/interfaces/propertyprobe.c:
Check for NULL pointer, in the hope that this fixes #532864.
2008-09-05 14:12:01 +00:00
Wim Taymans
265a494de5 gst-libs/gst/audio/gstaudiosrc.c: Disable a code path that is now called but causes a deadlock for some reason and is...
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiosrc.c: (gst_audioringbuffer_stop):
Disable a code path that is now called but causes a deadlock for some
reason and is unneeded.
2008-09-04 16:25:06 +00:00
Edward Hervey
21952ab09a gst-libs/gst/riff/riff-media.c: Use a decent caps for TrueSpeech instead of a ffmpeg-specific one.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
Use a decent caps for TrueSpeech instead of a ffmpeg-specific one.
This will also be fixed for upcoming gst-ffmpeg release so that once
this release of -base is out, it will work with the latest gst-ffmpeg
release.
2008-09-03 14:00:06 +00:00
Edward Hervey
3a0b6ebcfb gst-libs/gst/riff/riff-media.c: Add Truespeech mapping for RIFF formats (AVI/WAV).
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
(gst_riff_create_audio_template_caps):
Add Truespeech mapping for RIFF formats (AVI/WAV).
Fixes #550656
2008-09-03 13:27:20 +00:00
Stefan Kost
54acaa5706 Use new geo location tags from core. Fixes #481169
Original commit message from CVS:
* gst-libs/gst/tag/gstvorbistag.c:
* tests/check/libs/tag.c:
Use new geo location tags from core. Fixes #481169
2008-09-02 06:37:04 +00:00
Wim Taymans
da76d5e7cb gst-libs/gst/audio/gstaudiosink.c: Since we now call stop, we trigger this code path that causes a deadlock is appare...
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_stop):
Since we now call stop, we trigger this code path that causes a deadlock
is apparently not needed.
2008-08-26 17:24:31 +00:00
Wim Taymans
440432612b gst-libs/gst/audio/gstringbuffer.c: Also allow the case where the ringbuffer was paused when we try to stop it so tha...
Original commit message from CVS:
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_start),
(gst_ring_buffer_stop):
Also allow the case where the ringbuffer was paused when we try to stop
it so that the basesrc stop function is still called.
2008-08-26 15:45:36 +00:00
Sebastian Dröge
12bccedb68 gst-libs/gst/pbutils/missing-plugins.c: When cleaning up the caps fields also remove "depth" for the same reason we r...
Original commit message from CVS:
* gst-libs/gst/pbutils/missing-plugins.c: (copy_and_clean_caps):
When cleaning up the caps fields also remove "depth" for the same
reason we remove "width".
2008-08-15 07:24:38 +00:00
Tim-Philipp Müller
87384c993d gst-libs/gst/pbutils/descriptions.c: Add Lead H.264 here as well.
Original commit message from CVS:
* gst-libs/gst/pbutils/descriptions.c: (format_info_get_desc):
Add Lead H.264 here as well.
2008-08-14 17:14:53 +00:00
Julien Moutte
33690da61a gst-libs/gst/riff/riff-media.c: Add Lead H.264 variant.
Original commit message from CVS:
2008-08-14  Julien Moutte  <julien@fluendo.com>

* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
(gst_riff_create_video_template_caps): Add Lead H.264 variant.
2008-08-14 15:17:31 +00:00
Wim Taymans
510a5befc1 gst-libs/gst/audio/gstbaseaudiosrc.c: When not slaved to another clock also subtract the base_time from our internal ...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_create):
When not slaved to another clock also subtract the base_time from our
internal clock time to get the running time.
2008-08-13 09:17:38 +00:00
Stefan Kost
c5ad1c7228 gst-libs/gst/interfaces/propertyprobe.h: Remove double "interface" from doc-string.
Original commit message from CVS:
* gst-libs/gst/interfaces/propertyprobe.h:
Remove double "interface" from doc-string.
* gst-libs/gst/interfaces/xoverlay.h:
Document interface.
* gst-libs/gst/riff/riff.c:
Add basic doc blobs.
2008-08-12 06:31:49 +00:00
Stefan Kost
5d2049cdb3 gst-libs/gst/audio/Makefile.am: Don't try to build that example anymore.
Original commit message from CVS:
* gst-libs/gst/audio/Makefile.am:
Don't try to build that example anymore.
2008-08-11 15:05:35 +00:00
Stefan Kost
3511b2772b gst-libs/gst/audio/: Move audiofiltertemplate to gst-template.
Original commit message from CVS:
* gst-libs/gst/audio/.cvsignore:
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/audio/gstaudiofiltertemplate.c:
* gst-libs/gst/audio/make_filter:
Move audiofiltertemplate to gst-template.
2008-08-11 14:51:58 +00:00
Stefan Kost
01554ac056 More docs and shuffling. What can we do with the hundreds of #defines.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/audio/gstaudiosrc.h:
More docs and shuffling. What can we do with the hundreds of #defines.
2008-08-11 09:20:33 +00:00
Stefan Kost
f73aa5b817 gst-libs/gst/: Reducing number of dundocumented symbols.
Original commit message from CVS:
* gst-libs/gst/audio/audio.h:
* gst-libs/gst/audio/gstaudiofilter.h:
* gst-libs/gst/audio/gstringbuffer.h:
* gst-libs/gst/interfaces/propertyprobe.h:
* gst-libs/gst/tag/gsttagdemux.h:
Reducing number of dundocumented symbols.
2008-08-11 08:34:56 +00:00
Stefan Kost
26ad0ba982 gst-libs/gst/audio/audio.c: Fix doc comment syntax.
Original commit message from CVS:
* gst-libs/gst/audio/audio.c:
Fix doc comment syntax.
* gst-libs/gst/interfaces/propertyprobe.c:
Add more doc-comments and a FIXME: for the signal.
2008-08-11 07:16:30 +00:00
Frederic Crozat
89be246154 Make sure gettext returns translations in UTF-8 encoding rather than in the current locale encoding (#546822).
Original commit message from CVS:
Patch by: Frederic Crozat <fcrozat@mandriva.org>
* ext/alsa/gstalsaplugin.c: (plugin_init):
* ext/cdparanoia/gstcdparanoiasrc.c: (plugin_init):
* ext/gnomevfs/gstgnomevfs.c: (plugin_init):
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_plugin_init):
* gst-libs/gst/audio/gstbaseaudiosrc.c: (_do_init):
* gst-libs/gst/pbutils/pbutils.c: (gst_pb_utils_init):
* gst-libs/gst/tag/tags.c: (gst_tag_register_tags_internal):
* gst/playback/gstdecodebin.c: (plugin_init):
* gst/playback/gstdecodebin2.c: (gst_decode_bin_plugin_init):
* gst/playback/gstplayback.c: (plugin_init):
* gst/playback/gstqueue2.c: (plugin_init):
* gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_plugin_init):
* sys/v4l/gstv4l.c: (plugin_init):
Make sure gettext returns translations in UTF-8 encoding rather
than in the current locale encoding (#546822).
2008-08-07 15:58:58 +00:00
Stefan Kost
cac199ce36 gst-libs/gst/pbutils/descriptions.c: Add audio/x-qdm for qtdemux.
Original commit message from CVS:
* gst-libs/gst/pbutils/descriptions.c:
Add audio/x-qdm for qtdemux.
2008-08-06 13:12:07 +00:00
Stefan Kost
f7a085edaa Bump requirement to latest core and use new tag for riff formats.
Original commit message from CVS:
* configure.ac:
* gst-libs/gst/riff/riff-read.c:
Bump requirement to latest core and use new tag for riff formats.
Needed for #520694.
2008-08-01 11:55:07 +00:00
Stefan Kost
4f37ce04f6 gst-libs/gst/riff/riff-ids.h: Adding acid chunk for tempo and loop information.
Original commit message from CVS:
* gst-libs/gst/riff/riff-ids.h:
Adding acid chunk for tempo and loop information.
2008-07-30 09:02:31 +00:00
Damien Lespiau
d76e33616c gst-libs/gst/sdp/gstsdpmessage.c: Use GST_STR_NULL to avoid crashes with libcs that don't like NULL strings in printf...
Original commit message from CVS:
Patch by: Damien Lespiau  <damien.lespiau gmail com>
* gst-libs/gst/sdp/gstsdpmessage.c: (print_media):
Use GST_STR_NULL to avoid crashes with libcs that don't
like NULL strings in printf args (such as the win32 one).
Fixes #544306.
2008-07-23 13:17:31 +00:00
David Schleef
47eafd3466 gst-libs/gst/video/video.c: Revert ABI change.
Original commit message from CVS:
* gst-libs/gst/video/video.c: Revert ABI change.
2008-07-15 22:43:16 +00:00
Sebastian Dröge
dd7d36320e gst-libs/gst/riff/riff-media.c: Make it impossible to have NULL caps at the point where we set framerate and other th...
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
Make it impossible to have NULL caps at the point where we set
framerate and other things. Also don't return immediately for "3ivd"
video and let framerate, etc be set. Might fix bug #542508.
2008-07-15 13:05:04 +00:00
Mark Nauwelaerts
d6d5f88174 gst-libs/gst/video/video.c: Video format can also be conveniently determined from (many) non-fixed caps.
Original commit message from CVS:
* gst-libs/gst/video/video.c: (gst_video_format_parse_caps):
Video format can also be conveniently determined from (many)
non-fixed caps.
2008-07-14 17:06:26 +00:00
Damien Lespiau
c3e1de9033 gst-libs/gst/sdp/gstsdpmessage.c: Makes libgstsdp compile with mingw32 by defining the right WINVER so that getaddrin...
Original commit message from CVS:
Patch by: Damien Lespiau <damien dot lespiau at gmail dot com>
* gst-libs/gst/sdp/gstsdpmessage.c:
Makes libgstsdp compile with mingw32 by defining the right WINVER so
that getaddrinfo() can be used. Fixes #541358.
2008-07-03 09:12:49 +00:00
Sebastian Dröge
1aca2efee8 gst-libs/gst/riff/: Add support for ADPCM IMA DK3 and DK4 variant in RIFF containers.
Original commit message from CVS:
* gst-libs/gst/riff/riff-ids.h:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
(gst_riff_create_audio_template_caps):
Add support for ADPCM IMA DK3 and DK4 variant in RIFF containers.
Fixes bug #540351.
2008-06-30 09:20:59 +00:00
Sam Morris
752cf09704 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.
2008-06-26 06:03:38 +00:00
Wim Taymans
d2f328f55b 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.
2008-06-20 09:09:37 +00:00
Sebastian Dröge
31f3f65d53 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'.
2008-06-20 08:47:14 +00:00
Andy Wingo
7ba9110637 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().
2008-06-16 14:11:36 +00:00
Stefan Kost
4ad8ad3db5 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.
2008-06-13 11:59:21 +00:00
Wim Taymans
c30d479783 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.
2008-06-12 15:47:03 +00:00
Stefan Kost
e54b324d7d 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.
2008-06-12 14:49:15 +00:00
David Schleef
526b2e63a2 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
2008-06-11 20:13:00 +00:00
Wim Taymans
593d4b1af3 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.
2008-06-06 16:50:51 +00:00
Wim Taymans
20d64607d4 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.
2008-06-05 16:38:50 +00:00
Peter Kjellerstedt
26cd5ea1c8 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.
2008-06-04 11:53:53 +00:00
Tim-Philipp Müller
44e087f8c9 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.
2008-06-04 10:18:42 +00:00
John Millikin
f934d1c233 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)
2008-06-03 20:01:58 +00:00
Tim-Philipp Müller
8b491df810 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...).
2008-06-03 19:44:48 +00:00
Tim-Philipp Müller
cd9bb9a674 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()
2008-06-03 19:29:06 +00:00
Sebastian Dröge
0de81029c8 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().
2008-06-03 08:48:32 +00:00
Mark Nauwelaerts
9fa61c528d 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.
2008-05-31 19:57:57 +00:00
Mark Nauwelaerts
c660bbd6dd 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.
2008-05-31 18:10:47 +00:00
Wim Taymans
11309247f3 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.
2008-05-30 15:29:20 +00:00
Sebastian Dröge
45ef6b5e13 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.
2008-05-29 11:34:09 +00:00
Wim Taymans
2855fb48ad 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.
2008-05-28 15:48:33 +00:00
Wim Taymans
35e4b75b86 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.
2008-05-27 16:20:17 +00:00
Wim Taymans
13d7048f69 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.
2008-05-26 17:18:52 +00:00
Wim Taymans
79a725148d 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.
2008-05-23 14:14:28 +00:00
Felipe Contreras
b5f896dad6 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
2008-05-21 17:01:16 +00:00
Stefan Kost
eda6d89b8c 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 ';'.
2008-05-21 14:35:41 +00:00
Sebastian Dröge
d03bbd1e3e gst-libs/gst/audio/multichannel.c: Allow non-standard 2 channel layouts.
Original commit message from CVS:
* gst-libs/gst/audio/multichannel.c:
(gst_audio_check_channel_positions):
Allow non-standard 2 channel layouts.
* tests/check/elements/audioconvert.c: (GST_START_TEST):
Add some tests for converting and remapping non-standard 1 and 2
channel layouts.
2008-05-21 07:39:56 +00:00
Sebastian Dröge
3ee2676c2e gst-libs/gst/rtsp/gstrtsptransport.h: Make the GstRTSPTransport struct members public as there are no setters/getters...
Original commit message from CVS:
* gst-libs/gst/rtsp/gstrtsptransport.h:
Make the GstRTSPTransport struct members public as there are no
setters/getters and it's supposed to be changed directly.
Fixes bug #533087.
2008-05-21 06:39:20 +00:00
Wim Taymans
f36d9d6b08 gst-libs/gst/audio/gstbaseaudiosink.c: We can only use our optimal calibration if we prerolled before the latency exp...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_sync_latency):
We can only use our optimal calibration if we prerolled before the
latency expired.
2008-05-20 16:26:53 +00:00
Wim Taymans
d8dc371c0d ext/gnomevfs/gstgnomevfssrc.*: Set the ICY caps on the srcpad from where they get picked up by the base class now and...
Original commit message from CVS:
* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_init),
(gst_gnome_vfs_src_finalize),
(gst_gnome_vfs_src_received_headers_callback),
(gst_gnome_vfs_src_create), (gst_gnome_vfs_src_stop):
* ext/gnomevfs/gstgnomevfssrc.h:
Set the ICY caps on the srcpad from where they get picked up by the base
class now and set on the outgoing buffers.
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_create):
* sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_buffer_new):
BaseSrc now sets the caps on outgoing buffers automatically.
2008-05-20 11:13:27 +00:00
Wim Taymans
95d162fb71 gst-libs/gst/audio/gstbaseaudiosink.c: Change the way in which the ringbuffer is started when dealing with a slaved c...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_resample_slaving),
(gst_base_audio_sink_skew_slaving),
(gst_base_audio_sink_sync_latency), (gst_base_audio_sink_render),
(gst_base_audio_sink_async_play),
(gst_base_audio_sink_change_state):
Change the way in which the ringbuffer is started when dealing with a
slaved clock and latency. We now sync to the clock until we reach
upstream latency before starting the ringbuffer. This has the effect
that we can accurately align the master and slave clocks and let the
rate correction code take care of the initial drift or rounding errors
instead of leaving them uncorrected with the old approach.
2008-05-20 11:09:06 +00:00
Wim Taymans
86ab51207b gst-libs/gst/rtp/gstbasertpdepayload.c: Validate the RTP packet before further processing it. It's just too dangerous...
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_chain):
Validate the RTP packet before further processing it. It's just too
dangerous to accept random packets and people are not forced to use a
jitterbuffer or session manager to filter out the bad packets.
* gst-libs/gst/rtp/gstrtpbuffer.c:
(gst_rtp_buffer_set_extension_data),
(gst_rtp_buffer_get_payload_subbuffer):
Small cleanups.
When setting extension data in a buffer that is too small, we fail and
we should not set the extension bit.
Change GST_WARNINGS into g_warning because they really are
programming errors.
* tests/check/libs/rtp.c: (GST_START_TEST):
Catch the g_warnings now in the unit tests and that fact that failing to
set extension data left the extension bit untouched.
2008-05-14 20:28:02 +00:00
Bernard B
d06df554a9 gst-libs/gst/rtp/gstrtpbuffer.c: Fix seqnum compare function for bordercase values and fix the docs again. Fixes #533...
Original commit message from CVS:
Patch by: Bernard B <b-gnome at largestprime dot net>
* gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_compare_seqnum):
Fix seqnum compare function for bordercase values and fix the docs
again. Fixes #533075.
* tests/check/libs/rtp.c: (GST_START_TEST), (rtp_suite):
Add a testcase for seqnum compare function.
2008-05-14 13:43:12 +00:00
Wim Taymans
0c9b13988c gst-libs/gst/audio/gstbaseaudiosink.c: Revert previous patch that attempted to more accurately calculate the initial ...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render), (gst_base_audio_sink_async_play):
Revert previous patch that attempted to more accurately calculate the
initial offset between master and slave clock. The best thing we can do
in general is take the time of both clocks as the diff since we don't
know when the actual preroll happened.
2008-05-12 08:45:11 +00:00
Tim-Philipp Müller
1482332184 gst-libs/gst/pbutils/install-plugins.c: Fix docs: type and missing word.
Original commit message from CVS:
* gst-libs/gst/pbutils/install-plugins.c:
Fix docs: type and missing word.
2008-05-11 19:52:59 +00:00
Jan Schmidt
f11cf32c3f Document the GstTuner and GstColorBalance interfaces, and some other random API functions that needed it. 70% symbol ...
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/interfaces/colorbalance.c:
* gst-libs/gst/interfaces/colorbalance.h:
* gst-libs/gst/interfaces/colorbalancechannel.c:
* gst-libs/gst/interfaces/colorbalancechannel.h:
* gst-libs/gst/interfaces/tuner.c:
* gst-libs/gst/interfaces/tunerchannel.c:
* gst-libs/gst/interfaces/tunerchannel.h:
* gst-libs/gst/interfaces/tunernorm.c:
* gst-libs/gst/interfaces/tunernorm.h:
* gst-libs/gst/video/video.c:
* gst-libs/gst/video/video.h:
Document the GstTuner and GstColorBalance interfaces, and some
other random API functions that needed it. 70% symbol coverage, woo.
2008-05-09 21:42:26 +00:00
Wim Taymans
fc523e047c gst-libs/gst/audio/gstaudiosink.c: Choose to allocate one less segment but require one additional segment as latency.
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_acquire):
Choose to allocate one less segment but require one additional segment
as latency.
* gst-libs/gst/audio/gstaudiosrc.c: (gst_audioringbuffer_acquire):
No need to increment the number of segments in the source.
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_get_time), (clock_convert_external),
(gst_base_audio_sink_resample_slaving),
(gst_base_audio_sink_skew_slaving),
(gst_base_audio_sink_none_slaving), (gst_base_audio_sink_render),
(gst_base_audio_sink_async_play):
Remove adding latency when returning the internal time while subtracting
it again when we use the value a little later.
When calculating the end timestamp, we are making a rounding error
with the current algorithm. Ensure that we don't accumulate these
rounding errors when aligning samples by not resampling at all if we
don't need to. Fixes #419351.
Make the initial calibration of the clock slaving a little more
predictable and accurate. Also handle the case where we don't do
clock slaving.
2008-05-09 16:38:10 +00:00
Wouter Cloetens
a8a2b9c717 gst-libs/gst/rtsp/: Add Digest authorization support for RTSP connections. See #532065.
Original commit message from CVS:
Patch by: Wouter Cloetens <zombie at e2big dot org>
* gst-libs/gst/rtsp/Makefile.am:
* gst-libs/gst/rtsp/gstrtspconnection.c:
(gst_rtsp_connection_create), (md5_digest_to_hex_string),
(auth_digest_compute_hex_urp), (auth_digest_compute_response),
(add_auth_header), (gst_rtsp_connection_free),
(gst_rtsp_connection_set_auth), (str_case_hash), (str_case_equal),
(gst_rtsp_connection_set_auth_param),
(gst_rtsp_connection_clear_auth_params):
* gst-libs/gst/rtsp/gstrtspconnection.h:
Add Digest authorization support for RTSP connections. See #532065.
* gst-libs/gst/rtsp/md5.c:
* gst-libs/gst/rtsp/md5.h:
Yeap, another md5 implementation until we can depend on a glib that has
support for it.
2008-05-08 14:46:27 +00:00
Wim Taymans
09f7dee84d gst-libs/gst/audio/gstbaseaudiosink.c: Report the latency with the new seglatency parameter.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_query):
Report the latency with the new seglatency parameter.
* gst-libs/gst/audio/gstringbuffer.c:
(gst_ring_buffer_debug_spec_buff), (gst_ring_buffer_parse_caps),
(gst_ring_buffer_acquire):
* gst-libs/gst/audio/gstringbuffer.h:
Add new field to the ringbufferspec to specify the expected latency
between the underlying device read/write pointer, this is needed
when writing sinks that sit a little closer to the hardware.
Add some more docs for other fields.
2008-05-07 15:47:03 +00:00
Wim Taymans
8c3b00be4d gst-libs/gst/app/: Add marshal.list, make it compile and add to cvsignore.
Original commit message from CVS:
* gst-libs/gst/app/.cvsignore:
* gst-libs/gst/app/Makefile.am:
* gst-libs/gst/app/gstapp-marshal.list:
Add marshal.list, make it compile and add to cvsignore.
* gst-libs/gst/app/gstappsink.c: (gst_app_sink_dispose),
(gst_app_sink_stop):
Small cleanups.
* 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_start), (gst_app_src_stop),
(gst_app_src_create), (gst_app_src_set_caps),
(gst_app_src_get_caps), (gst_app_src_set_size),
(gst_app_src_get_size), (gst_app_src_set_seekable),
(gst_app_src_get_seekable), (gst_app_src_set_max_buffers),
(gst_app_src_get_max_buffers), (gst_app_src_push_buffer),
(gst_app_src_end_of_stream):
* gst-libs/gst/app/gstappsrc.h:
Beat appsrc in shape, add signals and actions.
Add some docs.
Add properties for caps, size, seekability and max-buffers.
Fix unlock/stop code.
2008-05-07 10:38:23 +00:00
Wim Taymans
1275eda108 gst-libs/gst/app/gstappsink.*: Start some docs.
Original commit message from CVS:
* gst-libs/gst/app/gstappsink.c: (gst_app_sink_class_init),
(gst_app_sink_init), (gst_app_sink_set_property),
(gst_app_sink_get_property), (gst_app_sink_unlock_start),
(gst_app_sink_unlock_stop), (gst_app_sink_flush_unlocked),
(gst_app_sink_start), (gst_app_sink_stop), (gst_app_sink_event),
(gst_app_sink_preroll), (gst_app_sink_render),
(gst_app_sink_set_caps), (gst_app_sink_set_drop),
(gst_app_sink_get_drop):
* gst-libs/gst/app/gstappsink.h:
Start some docs.
Add property to drop buffers when the queue is filled
Fix unlocking and flushing when the queues are filled.
2008-05-05 10:27:45 +00:00
Edward Hervey
b98072f957 gst-libs/gst/pbutils/descriptions.c: It's SorensOn and not SorensEn.
Original commit message from CVS:
* gst-libs/gst/pbutils/descriptions.c: (formats):
It's SorensOn and not SorensEn.
2008-05-05 07:41:03 +00:00
Tim-Philipp Müller
6451cbf5b7 gst-libs/gst/pbutils/descriptions.c: Fix description of video/x-flash-video.
Original commit message from CVS:
* gst-libs/gst/pbutils/descriptions.c: (formats):
Fix description of video/x-flash-video.
2008-05-04 15:23:36 +00:00
Sebastian Dröge
83f0729394 Remove some unused code.
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func):
* gst-libs/gst/audio/gstaudiosrc.c: (audioringbuffer_thread_func):
* gst/tcp/gsttcp.c: (gst_tcp_socket_write):
* sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_get_fps_list):
Remove some unused code.
* gst/audioconvert/gstaudioquantize.c:
(gst_audio_quantize_free_noise_shaping):
Don't return before freeing the noise shaping history.
2008-05-04 15:02:20 +00:00
Wim Taymans
c6389eec57 gst-libs/gst/rtp/gstbasertppayload.c: Rename the setcaps/getcaps function internally to make it clear that they are c...
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_init),
(gst_basertppayload_sink_setcaps),
(gst_basertppayload_sink_getcaps):
Rename the setcaps/getcaps function internally to make it clear that
they are called for the sink pad.
2008-05-02 12:13:08 +00:00
Wim Taymans
f0f6476aff gst-libs/gst/rtp/gstbasertpdepayload.*: Catch packet-lost events from the jitterbuffer and convert them into a vmetho...
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_class_init),
(gst_base_rtp_depayload_handle_sink_event), (create_segment_event),
(gst_base_rtp_depayload_packet_lost),
(gst_base_rtp_depayload_set_gst_timestamp):
* gst-libs/gst/rtp/gstbasertpdepayload.h:
Catch packet-lost events from the jitterbuffer and convert them into a
vmethod call (lost-packet) so that depayloaders can do something smart.
Also add a default packet-lost function that sends out a segment update
to the decoders.
2008-05-02 12:11:07 +00:00
Wim Taymans
7916e386ca gst-libs/gst/audio/gstbaseaudiosink.h: Clarify some docs.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.h:
Clarify some docs.
* gst-libs/gst/audio/gstbaseaudiosrc.c: (slave_method_get_type),
(gst_base_audio_src_class_init), (gst_base_audio_src_init),
(gst_base_audio_src_set_slave_method),
(gst_base_audio_src_get_slave_method),
(gst_base_audio_src_set_property),
(gst_base_audio_src_get_property), (gst_base_audio_src_create):
* gst-libs/gst/audio/gstbaseaudiosrc.h:
Add property and methods for selecting the clock slave method in the
source, like in the sink.
We only implement "none" and "re-timestamp" for now.
API: gst_base_audio_src_set_slave_method()
API: gst_base_audio_src_get_slave_method()
2008-04-28 08:51:38 +00:00
Wim Taymans
0c5d01b59f gst-libs/gst/app/gstappsink.*: Add more docs.
Original commit message from CVS:
* gst-libs/gst/app/gstappsink.c: (gst_app_sink_class_init),
(gst_app_sink_init), (gst_app_sink_set_property),
(gst_app_sink_get_property), (gst_app_sink_event),
(gst_app_sink_preroll), (gst_app_sink_render),
(gst_app_sink_set_emit_signals), (gst_app_sink_get_emit_signals),
(gst_app_sink_set_max_buffers), (gst_app_sink_get_max_buffers),
(gst_app_sink_pull_buffer):
* gst-libs/gst/app/gstappsink.h:
Add more docs.
Add signals for when preroll and render buffers are available.
Add property to control signal emission.
Add property to control the max queue size.
2008-04-25 18:18:47 +00:00
Wim Taymans
5b8afead80 gst-libs/gst/rtp/gstrtpbuffer.c: Fix the docs about the seqnum compare function, it returns a difference.
Original commit message from CVS:
* gst-libs/gst/rtp/gstrtpbuffer.c:
Fix the docs about the seqnum compare function, it returns a difference.
2008-04-25 07:37:09 +00:00
Sebastian Dröge
b8d03daeab gst-libs/gst/rtp/gstrtppayloads.c: Use g_ascii_strcasecmp() instead of the deprecated g_strcasecmp().
Original commit message from CVS:
* gst-libs/gst/rtp/gstrtppayloads.c:
(gst_rtp_payload_info_for_name):
Use g_ascii_strcasecmp() instead of the deprecated g_strcasecmp().
2008-04-19 16:33:24 +00:00
Sebastian Dröge
66bbadadd0 gst-libs/gst/audio/gstringbuffer.c: Use g_atomic_int_set() instead of gst_atomic_int_set().
Original commit message from CVS:
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_may_start):
Use g_atomic_int_set() instead of gst_atomic_int_set().
2008-04-17 07:33:46 +00:00
Tim-Philipp Müller
96a3780816 gst-libs/gst/video/video.c: Add guards to these functions to ensure sane input values.
Original commit message from CVS:
* gst-libs/gst/video/video.c: (gst_video_format_new_caps),
(gst_video_format_to_fourcc), (gst_video_format_get_row_stride),
(gst_video_format_get_pixel_stride),
(gst_video_format_get_component_width),
(gst_video_format_get_component_height),
(gst_video_format_get_component_offset), (gst_video_format_get_size),
(gst_video_format_convert):
Add guards to these functions to ensure sane input values.
* tests/check/libs/video.c:
Fix unit test not to create caps with width=0 and height=0.
2008-04-11 17:13:52 +00:00
Tim-Philipp Müller
7a29d716bd gst/: Work around missing bits of thread-safety on older GLibs some more to avoid assertions when starting up multipl...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_class_init):
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_class_init):
* gst/playback/gstplayback.c: (plugin_init):
* gst/volume/gstvolume.c: (plugin_init):
Work around missing bits of thread-safety on older GLibs some
more to avoid assertions when starting up multiple playbin
objects concurrently (see #512382).
2008-04-06 20:16:27 +00:00
Tim-Philipp Müller
c1a32ecf46 gst-libs/gst/pbutils/missing-plugins.c: Remove some more fields.
Original commit message from CVS:
* gst-libs/gst/pbutils/missing-plugins.c: (copy_and_clean_caps):
Remove some more fields.
2008-04-06 17:19:39 +00:00
Wim Taymans
ce67ac6373 gst-libs/gst/audio/gstbaseaudiosink.c: Guard against over and underflows because of clock slaving.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_skew_slaving), (gst_base_audio_sink_render):
Guard against over and underflows because of clock slaving.
When we are using our own clock, still compensate for any calibrations
that we might have done to our clock.
2008-04-03 10:37:03 +00:00
Sebastian Dröge
49467c4a63 gst-libs/gst/cdda/: Use GLib's base64 implementation instead of our own.
Original commit message from CVS:
* gst-libs/gst/cdda/Makefile.am:
* gst-libs/gst/cdda/base64.c:
* gst-libs/gst/cdda/base64.h:
* gst-libs/gst/cdda/gstcddabasesrc.c:
(gst_cddabasesrc_calculate_musicbrainz_discid):
Use GLib's base64 implementation instead of our own.
2008-04-03 06:39:27 +00:00
Wim Taymans
4b7e9b70f7 gst-libs/gst/riff/riff-media.c: ms-gsm can have arbitrarty sample rates. See #481354.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
ms-gsm can have arbitrarty sample rates. See #481354.
2008-03-31 16:08:45 +00:00
Wim Taymans
9ac953ef0f gst-libs/gst/riff/riff-media.c: MP4S is generic MPEG-4, not a microsoft variant.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
MP4S is generic MPEG-4, not a microsoft variant.
2008-03-28 16:22:35 +00:00
Wim Taymans
fc97139103 gst-libs/gst/rtsp/gstrtspconnection.c: Don't error when poll_wait returns EAGAIN.
Original commit message from CVS:
* gst-libs/gst/rtsp/gstrtspconnection.c:
(gst_rtsp_connection_connect), (gst_rtsp_connection_write),
(gst_rtsp_connection_read_internal), (gst_rtsp_connection_poll):
Don't error when poll_wait returns EAGAIN.
2008-03-24 16:40:08 +00:00
Wim Taymans
877a45b791 gst-libs/gst/audio/gstaudiosink.c: Small debug improvement.
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_release):
Small debug improvement.
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
Fix bug in determining the sample start/stop position, we want to base
this decision on the fact that we are going forwards or backwards, not
slower or faster. This fixes some ugly resync warnings when playing at
very slow speeds.
2008-03-24 11:24:22 +00:00
Sebastian Dröge
49deb0c05d Use G_PARAM_STATIC_STRINGS everywhere for GParamSpecs that use static strings (i.e. all). This gives us less memory u...
Original commit message from CVS:
* configure.ac:
* ext/alsa/gstalsamixerelement.c:
(gst_alsa_mixer_element_class_init):
* ext/alsa/gstalsasink.c: (gst_alsasink_class_init):
* ext/alsa/gstalsasrc.c: (gst_alsasrc_class_init):
* ext/cdparanoia/gstcdparanoiasrc.c:
(gst_cd_paranoia_src_class_init):
* ext/gio/gstgiosink.c: (gst_gio_sink_class_init):
* ext/gio/gstgiosrc.c: (gst_gio_src_class_init):
* ext/gio/gstgiostreamsink.c: (gst_gio_stream_sink_class_init):
* ext/gio/gstgiostreamsrc.c: (gst_gio_stream_src_class_init):
* ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_class_init):
* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_class_init):
* ext/ogg/gstoggmux.c: (gst_ogg_mux_class_init):
* ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init):
* ext/pango/gsttextrender.c: (gst_text_render_class_init):
* ext/theora/theoradec.c: (gst_theora_dec_class_init):
* ext/theora/theoraenc.c: (gst_theora_enc_class_init):
* ext/theora/theoraparse.c: (gst_theora_parse_class_init):
* ext/vorbis/vorbisenc.c: (gst_vorbis_enc_class_init):
* gst-libs/gst/audio/gstaudiofiltertemplate.c:
(gst_audio_filter_template_class_init):
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_class_init):
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_class_init):
* gst-libs/gst/cdda/gstcddabasesrc.c:
(gst_cdda_base_src_class_init):
* gst-libs/gst/interfaces/mixertrack.c:
(gst_mixer_track_class_init):
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_class_init):
* gst-libs/gst/rtp/gstbasertppayload.c:
(gst_basertppayload_class_init):
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_class_init):
* gst/audiorate/gstaudiorate.c: (gst_audio_rate_class_init):
* gst/audioresample/gstaudioresample.c:
(gst_audioresample_class_init):
* gst/audiotestsrc/gstaudiotestsrc.c:
(gst_audio_test_src_class_init):
* gst/gdp/gstgdppay.c: (gst_gdp_pay_class_init):
* gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init):
* gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
(preroll_unlinked):
* gst/playback/gstplaybin.c: (gst_play_bin_class_init):
* gst/playback/gstplaybin2.c: (gst_play_bin_class_init):
* gst/playback/gstplaysink.c: (gst_play_sink_class_init):
* gst/playback/gstqueue2.c: (gst_queue_class_init):
* gst/playback/gststreaminfo.c: (gst_stream_info_class_init):
* gst/playback/gststreamselector.c: (gst_selector_pad_class_init),
(gst_stream_selector_class_init):
* gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init):
* gst/subparse/gstsubparse.c: (gst_sub_parse_class_init):
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
* gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_class_init):
* gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_class_init):
* gst/tcp/gsttcpserversink.c: (gst_tcp_server_sink_class_init):
* gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_class_init):
* gst/videorate/gstvideorate.c: (gst_video_rate_class_init):
* gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init):
* gst/videotestsrc/gstvideotestsrc.c:
(gst_video_test_src_class_init):
* gst/volume/gstvolume.c: (gst_volume_class_init):
* sys/v4l/gstv4lelement.c: (gst_v4lelement_class_init):
* sys/v4l/gstv4lmjpegsink.c: (gst_v4lmjpegsink_class_init):
* sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_class_init):
* sys/v4l/gstv4lsrc.c: (gst_v4lsrc_class_init):
* sys/ximage/ximagesink.c: (gst_ximagesink_class_init):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_class_init):
Use G_PARAM_STATIC_STRINGS everywhere for GParamSpecs that use
static strings (i.e. all). This gives us less memory usage,
fewer allocations and thus less memory defragmentation. Depend
on core CVS for this. Fixes bug #523806.
2008-03-22 15:00:53 +00:00
Wim Taymans
6d1fdf60a3 gst-libs/gst/sdp/gstsdpmessage.c: Use GST_STR_NULL when trying to print strings that could be NULL because this might...
Original commit message from CVS:
* gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_message_dump):
Use GST_STR_NULL when trying to print strings that could be NULL because
this might crash on some platforms. See #520808.
2008-03-18 11:20:05 +00:00
Ole André Vadla Ravnås
aca58fb8ac gst-libs/gst/rtsp/gstrtspconnection.c: Generic Windows fixes that makes libgstrtsp work on Windows when coupled with ...
Original commit message from CVS:
Patch by: Ole André Vadla Ravnås  <ole.andre.ravnas@tandberg.com>
* gst-libs/gst/rtsp/gstrtspconnection.c:
(gst_rtsp_connection_connect), (gst_rtsp_connection_write),
(read_line), (gst_rtsp_connection_read_internal):
Generic Windows fixes that makes libgstrtsp work on Windows when
coupled with the new GstPoll API. See #520808.
2008-03-18 11:10:12 +00:00
Tim-Philipp Müller
a0de863603 gst-libs/gst/video/video.c: Fix gst_video_format_parse_caps() for RGB caps with alpha channel (#522635).
Original commit message from CVS:
* gst-libs/gst/video/video.c: (gst_video_format_parse_caps),
(gst_video_format_from_rgba32_masks):
Fix gst_video_format_parse_caps() for RGB caps with alpha channel
(#522635).
* tests/check/libs/video.c: (test_parse_caps_rgb), (video_suite):
Add unit test for the RGB caps parsing and creation, checking for
internal consistency of the new API and consistency of the API with
the old GST_VIDEO_CAPS_* defines.
2008-03-17 10:32:28 +00:00
Michael Smith
15e209d20e gst-libs/gst/audio/gstringbuffer.*: Rename recently added buffer types to make more sense.
Original commit message from CVS:
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_parse_caps):
* gst-libs/gst/audio/gstringbuffer.h:
Rename recently added buffer types to make more sense.
* ext/alsa/gstalsasink.c: (alsasink_parse_spec),
(gst_alsasink_write):
Adapt for above API changes.
Fixes bug #520523.
2008-03-12 12:39:13 +00:00
Wim Taymans
579949e2c5 gst-libs/gst/audio/gstbaseaudiosrc.c: Fix duration when no clock was provided. Fixes #520300.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_create):
Fix duration when no clock was provided. Fixes #520300.
2008-03-10 17:19:56 +00:00
Olivier Crete
cf273d8add Add trivial function to compare GstNetAddress. See #520626.
Original commit message from CVS:
Patch by: Olivier Crete  <tester at tester ca>
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/netbuffer/gstnetbuffer.c: (gst_netaddress_equal):
* gst-libs/gst/netbuffer/gstnetbuffer.h:
Add trivial function to compare GstNetAddress. See #520626.
API: GstNetBuffer::gst_netaddress_equal
2008-03-07 18:17:44 +00:00
Wim Taymans
8a822e70be gst/: Remove GstPollMode from gstpoll constructor.
Original commit message from CVS:
* gst-libs/gst/rtsp/gstrtspconnection.c:
(gst_rtsp_connection_create):
* gst/tcp/gstmultifdsink.c: (gst_fdset_mode_get_type),
(gst_multi_fd_sink_class_init), (gst_multi_fd_sink_start):
* gst/tcp/gstmultifdsink.h:
* gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_start):
* gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_start):
Remove GstPollMode from gstpoll constructor.
2008-03-07 15:48:51 +00:00
Mersad Jelacic
ed814cbaed gst-libs/gst/rtp/gstbasertpaudiopayload.*: API: add gst_base_rtp_audio_payload_set_samplebits_options() to make it po...
Original commit message from CVS:
Patch by: Mersad Jelacic  <mersad at axis dot com>
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
API: add gst_base_rtp_audio_payload_set_samplebits_options() to make it
possible to specify the sample size in bits. (#509637)
2008-03-03 16:11:50 +00:00
Tim-Philipp Müller
5a3d087279 API: add gst_mixer_options_list_changed(), gst_mixer_mixer_changed() and gst_mixer_message_parse_options_list_changed...
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/interfaces/mixer.c: (gst_mixer_option_changed),
(gst_mixer_options_list_changed), (gst_mixer_mixer_changed),
(gst_mixer_message_get_type),
(gst_mixer_message_parse_option_changed),
(gst_mixer_message_parse_options_list_changed):
* gst-libs/gst/interfaces/mixer.h: (GstMixerType),
(GST_MIXER_MESSAGE_OPTION_CHANGED),
(GST_MIXER_MESSAGE_OPTIONS_LIST_CHANGED),
(GST_MIXER_MESSAGE_MIXER_CHANGED):
API: add gst_mixer_options_list_changed(), gst_mixer_mixer_changed()
and gst_mixer_message_parse_options_list_changed(). Fixes #519916.
2008-03-03 13:56:38 +00:00
Tim-Philipp Müller
225d9b5d08 gst-libs/gst/interfaces/mixeroptions.*: API: add GstMixerOptions::get_values vfunc (#519906)
Original commit message from CVS:
* gst-libs/gst/interfaces/mixeroptions.c: (gst_mixer_options_init),
(gst_mixer_options_get_values):
* gst-libs/gst/interfaces/mixeroptions.h:
(GST_MIXER_OPTIONS_GET_CLASS), (GstMixerOptionsClass),
(_GstMixerOptions), (_GstMixerOptionsClass):
API: add GstMixerOptions::get_values vfunc (#519906)
2008-03-03 13:50:18 +00:00
Sebastian Dröge
ec7afb6f84 Correct all relevant warnings found by the sparse semantic code analyzer. This include marking several symbols static...
Original commit message from CVS:
* ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_get_type):
* ext/alsa/gstalsasink.c: (set_hwparams):
* ext/alsa/gstalsasrc.c: (set_hwparams):
* ext/gio/gstgio.c: (gst_gio_uri_handler_get_uri):
* ext/ogg/gstoggmux.h:
* ext/ogg/gstogmparse.c:
* gst-libs/gst/audio/audio.c:
* gst-libs/gst/fft/kiss_fft_f64.c: (kiss_fft_f64_alloc):
* gst-libs/gst/pbutils/missing-plugins.c:
(gst_missing_uri_sink_message_new),
(gst_missing_element_message_new),
(gst_missing_decoder_message_new),
(gst_missing_encoder_message_new):
* gst-libs/gst/rtp/gstbasertppayload.c:
* gst-libs/gst/rtp/gstrtcpbuffer.c:
(gst_rtcp_packet_bye_get_reason):
* gst/audioconvert/gstaudioconvert.c:
* gst/audioresample/gstaudioresample.c:
* gst/ffmpegcolorspace/imgconvert.c:
* gst/playback/test.c: (gen_video_element), (gen_audio_element):
* gst/typefind/gsttypefindfunctions.c:
* gst/videoscale/vs_4tap.c:
* gst/videoscale/vs_4tap.h:
* sys/v4l/gstv4lelement.c:
* sys/v4l/gstv4lsrc.c: (gst_v4lsrc_get_any_caps):
* sys/v4l/v4l_calls.c:
* sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_capture_init),
(gst_v4lsrc_try_capture):
* sys/ximage/ximagesink.c: (gst_ximagesink_check_xshm_calls),
(gst_ximagesink_ximage_new):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_check_xshm_calls),
(gst_xvimagesink_xvimage_new):
* tests/check/elements/audioconvert.c:
* tests/check/elements/audioresample.c:
(fail_unless_perfect_stream):
* tests/check/elements/audiotestsrc.c: (setup_audiotestsrc):
* tests/check/elements/decodebin.c:
* tests/check/elements/gdpdepay.c: (setup_gdpdepay),
(setup_gdpdepay_streamheader):
* tests/check/elements/gdppay.c: (setup_gdppay), (GST_START_TEST),
(setup_gdppay_streamheader):
* tests/check/elements/gnomevfssink.c: (setup_gnomevfssink):
* tests/check/elements/multifdsink.c: (setup_multifdsink):
* tests/check/elements/textoverlay.c:
* tests/check/elements/videorate.c: (setup_videorate):
* tests/check/elements/videotestsrc.c: (setup_videotestsrc):
* tests/check/elements/volume.c: (setup_volume):
* tests/check/elements/vorbisdec.c: (setup_vorbisdec):
* tests/check/elements/vorbistag.c:
* tests/check/generic/clock-selection.c:
* tests/check/generic/states.c: (setup), (teardown):
* tests/check/libs/cddabasesrc.c:
* tests/check/libs/video.c:
* tests/check/pipelines/gio.c:
* tests/check/pipelines/oggmux.c:
* tests/check/pipelines/simple-launch-lines.c:
(simple_launch_lines_suite):
* tests/check/pipelines/streamheader.c:
* tests/check/pipelines/theoraenc.c:
* tests/check/pipelines/vorbisdec.c:
* tests/check/pipelines/vorbisenc.c:
* tests/examples/seek/scrubby.c:
* tests/examples/seek/seek.c: (query_positions_elems),
(query_positions_pads):
* tests/icles/stress-xoverlay.c: (myclock):
Correct all relevant warnings found by the sparse semantic code
analyzer. This include marking several symbols static, using
NULL instead of 0 for pointers and using "foo (void)" instead
of "foo ()" for declarations.
* win32/common/libgstrtp.def:
Add gst_rtp_buffer_set_extension_data to the symbol definition file.
2008-03-03 06:04:31 +00:00
Tim-Philipp Müller
d85688b8a2 gst-libs/gst/rtsp/gstrtspconnection.c: Don't include unix headers unconditionally (fixes #518037).
Original commit message from CVS:
* gst-libs/gst/rtsp/gstrtspconnection.c:
Don't include unix headers unconditionally (fixes #518037).
2008-03-02 18:32:36 +00:00
Tim-Philipp Müller
843366350e gst-libs/gst/video/video.*: API: add GST_VIDEO_FORMAT_Y41B and GST_VIDEO_FORMAT_Y42B.
Original commit message from CVS:
* gst-libs/gst/video/video.c: (gst_video_calculate_display_ratio),
(gst_video_format_from_fourcc), (gst_video_format_to_fourcc),
(gst_video_format_is_rgb), (gst_video_format_is_yuv),
(gst_video_format_has_alpha), (gst_video_format_get_row_stride),
(gst_video_format_get_pixel_stride),
(gst_video_format_get_component_width),
(gst_video_format_get_component_height),
(gst_video_format_get_component_offset), (gst_video_format_get_size):
* gst-libs/gst/video/video.h: (GST_VIDEO_FORMAT_Y41B),
(GST_VIDEO_FORMAT_Y42B):
API: add GST_VIDEO_FORMAT_Y41B and GST_VIDEO_FORMAT_Y42B.
2008-03-02 18:20:44 +00:00
Tim-Philipp Müller
d3273657f0 gst-libs/gst/video/video.c: YV12 is I420 with swapped components 1 and 2, so the offset of component 1 for I420 shoul...
Original commit message from CVS:
* gst-libs/gst/video/video.c: (gst_video_format_get_component_offset):
YV12 is I420 with swapped components 1 and 2, so the offset of
component 1 for I420 should be the offset for component 2 for YV12
and vice versa.
2008-03-02 18:07:10 +00:00
Julien Moutte
f0154849b0 ext/alsa/gstalsa.c: Probe for IEC958 pcm to detect if we can do SPDIF output.
Original commit message from CVS:
2008-02-29  Julien Moutte  <julien@fluendo.com>

* ext/alsa/gstalsa.c: (gst_alsa_open_iec958_pcm),
(gst_alsa_probe_supported_formats): Probe for IEC958 pcm to
detect
if we can do SPDIF output.
* ext/alsa/gstalsa.h:
* ext/alsa/gstalsasink.c: (set_hwparams), (alsasink_parse_spec),
(gst_alsasink_prepare), (gst_alsasink_close),
(gst_alsasink_write):
* ext/alsa/gstalsasink.h: Initial support for SPDIF.
* gst-libs/gst/audio/gstringbuffer.c:
(gst_ring_buffer_parse_caps):
* gst-libs/gst/audio/gstringbuffer.h: Add non linear buffer
types
to support AC3, EC3 and IEC958 buffers.
2008-02-29 18:44:36 +00:00
Tim-Philipp Müller
ca1b8db04a gst-libs/gst/interfaces/mixer.c: De-cruft and fix message type assertions (NULL is not a really valid mixer message t...
Original commit message from CVS:
* gst-libs/gst/interfaces/mixer.c: (GST_MIXER_MESSAGE_HAS_TYPE),
(gst_mixer_message_parse_mute_toggled),
(gst_mixer_message_parse_record_toggled),
(gst_mixer_message_parse_volume_changed),
(gst_mixer_message_parse_option_changed):
De-cruft and fix message type assertions (NULL is not a really
valid mixer message type string).
2008-02-29 17:59:16 +00:00
David Schleef
3fb3c580b1 gst-libs/gst/riff/riff-media.c: Add Dirac mapping
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: Add Dirac mapping
2008-02-29 00:04:57 +00:00
Wim Taymans
0667334b2d gst-libs/gst/rtsp/gstrtspconnection.*: Use GstPoll for the rtsp connection.
Original commit message from CVS:
* gst-libs/gst/rtsp/gstrtspconnection.c:
(gst_rtsp_connection_create), (gst_rtsp_connection_connect),
(gst_rtsp_connection_write), (gst_rtsp_connection_read_internal),
(gst_rtsp_connection_receive), (gst_rtsp_connection_close),
(gst_rtsp_connection_free), (gst_rtsp_connection_poll),
(gst_rtsp_connection_flush):
* gst-libs/gst/rtsp/gstrtspconnection.h:
Use GstPoll for the rtsp connection.
2008-02-28 09:50:52 +00:00
Sebastian Dröge
07cd425c25 gst-libs/gst/cdda/sha1.c: Use memcpy() instead of upcasting a byte array to long *. This fixes an unaligned memory ac...
Original commit message from CVS:
* gst-libs/gst/cdda/sha1.c: (sha_transform):
Use memcpy() instead of upcasting a byte array to long *. This
fixes an unaligned memory access, resulting in SIGBUS on IA64.
This should be ported to GCheckSum once we can use GLib 2.16.
Partially fixes bug #500833.
2008-02-27 10:42:08 +00:00
Tim-Philipp Müller
b191c8f609 gst-libs/gst/tag/gsttagdemux.c: Push tag event after the newsegment event. Log the pointer of the buffer we're actual...
Original commit message from CVS:
* gst-libs/gst/tag/gsttagdemux.c: (gst_tag_demux_chain):
Push tag event after the newsegment event. Log the pointer of
the buffer we're actually going to push rather than the buffer
we're feeding to _make_metadata_writable().
2008-02-27 10:23:27 +00:00
Tim-Philipp Müller
2c538ea740 gst-libs/gst/audio/multichannel.c: Fix confusing terminology in docs and code: structure fields are 'fields' and not ...
Original commit message from CVS:
* gst-libs/gst/audio/multichannel.c:
(GST_AUDIO_CHANNEL_POSITIONS_FIELD_NAME),
(gst_audio_get_channel_positions), (gst_audio_set_channel_positions),
(gst_audio_set_structure_channel_positions_list),
(add_list_to_struct), (gst_audio_set_caps_channel_positions_list),
(gst_audio_fixate_channel_positions):
Fix confusing terminology in docs and code: structure fields are
'fields' and not 'properties'.
2008-02-19 20:42:09 +00:00
Tim-Philipp Müller
a1e59086ba gst-libs/gst/audio/multichannel.c: Give more useful warning messages if one of the channel layout enums passed to us ...
Original commit message from CVS:
* gst-libs/gst/audio/multichannel.c:
(gst_audio_check_channel_positions), (add_list_to_struct):
Give more useful warning messages if one of the channel
layout enums passed to us is invalid and if the "channels"
field in the caps has a GType we don't expect.
2008-02-19 20:36:58 +00:00
Tim-Philipp Müller
29162d0a46 gst-libs/gst/audio/multichannel.c: Fix typo in docs blurb.
Original commit message from CVS:
* gst-libs/gst/audio/multichannel.c:
Fix typo in docs blurb.
2008-02-19 20:22:09 +00:00
Wim Taymans
d8c28a99c8 gst-libs/gst/rtp/gstbasertpaudiopayload.c: Fix potential leaks.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
(gst_base_rtp_audio_payload_handle_frame_based_buffer),
(gst_base_rtp_audio_payload_handle_sample_based_buffer):
Fix potential leaks.
* gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_chain):
Fix leak when there is no function configured.
2008-02-12 14:51:26 +00:00
Jan Schmidt
a30f5934a6 gst-libs/gst/interfaces/: Comment out a couple of other things which break the build when
Original commit message from CVS:
* gst-libs/gst/interfaces/mixeroptions.c:
* gst-libs/gst/interfaces/mixertrack.c:
Comment out a couple of other things which break the build when
GST_DISABLE_DEPRECATED isn't on but -Werror is.
2008-02-07 22:39:00 +00:00
Wim Taymans
d4bcdb7340 gst-libs/gst/rtsp/gstrtspmessage.c: Fix compiler warning.
Original commit message from CVS:
* gst-libs/gst/rtsp/gstrtspmessage.c: (gst_rtsp_message_unset):
Fix compiler warning.
2008-02-07 12:17:49 +00:00
Peter Kjellerstedt
ed373d3072 gst-libs/gst/sdp/gstsdpmessage.c: Clear the addrinfo struct using memset. Fixes #514937.
Original commit message from CVS:
Patch by: Peter Kjellerstedt  <pkj at axis com>
* gst-libs/gst/sdp/gstsdpmessage.c: (is_multicast_address):
Clear the addrinfo struct using memset. Fixes #514937.
2008-02-07 11:00:45 +00:00
Sebastian Dröge
8b970c5581 Use gmtime_r if available as gmtime is not MT-safe.
Original commit message from CVS:
* configure.ac:
* gst-libs/gst/rtsp/gstrtspconnection.c: (add_date_header):
Use gmtime_r if available as gmtime is not MT-safe.
Fixes bug #511810.
2008-02-02 07:13:15 +00:00
Sebastian Dröge
044f629784 gst-libs/gst/rtsp/gstrtspconnection.c: Cast glong to time_t as time_t might have a different type on other platforms,...
Original commit message from CVS:
* gst-libs/gst/rtsp/gstrtspconnection.c: (add_date_header):
Cast glong to time_t as time_t might have a different type on
other platforms, like FreeBSD, and we get a compiler warning
otherwise. Fixes bug #511825.
2008-02-02 06:52:41 +00:00
Thijs Vermeir
b8d39bc200 Add gst_rtp_buffer_set_extension_data()
Original commit message from CVS:
Patch by: Thijs Vermeir <thijsvermeir at gmail dot com>
* gst-libs/gst/rtp/gstrtpbuffer.c:
(gst_rtp_buffer_set_extension_data):
* gst-libs/gst/rtp/gstrtpbuffer.h:
* tests/check/libs/rtp.c: (GST_START_TEST), (rtp_suite):
Add gst_rtp_buffer_set_extension_data()
Add a unit test for this addition. Fixes #511478.
API: GstRTPBuffer:gst_rtp_buffer_set_extension_data()
2008-02-01 11:09:16 +00:00
Wim Taymans
88322ce012 gst-libs/gst/app/gstappsink.c: Really clean up the queue instead of just unreffing all buffers in it.
Original commit message from CVS:
* gst-libs/gst/app/gstappsink.c: (gst_app_sink_dispose):
Really clean up the queue instead of just unreffing all buffers
in it.
* gst-libs/gst/app/gstappsrc.c: (gst_app_src_base_init),
(gst_app_src_class_init), (gst_app_src_init),
(gst_app_src_dispose), (gst_app_src_finalize):
Fix dispose/finalize.
2008-01-31 17:18:46 +00:00
Jan Schmidt
67ab0ec286 gst-libs/gst/interfaces/: Also remove the conditional registration of the signals that disappeared with the ABI chang...
Original commit message from CVS:
* gst-libs/gst/interfaces/mixeroptions.c:
* gst-libs/gst/interfaces/mixertrack.c:
Also remove the conditional registration of the signals
that disappeared with the ABI change in 0.10.14
2008-01-30 13:45:27 +00:00
Jan Schmidt
2531fca577 gst-libs/gst/rtsp/gstrtspconnection.c: Revert patch to gstrtspconnection.c for brown paper bag release of -base. Re-o...
Original commit message from CVS:
* gst-libs/gst/rtsp/gstrtspconnection.c:
Revert patch to gstrtspconnection.c for brown paper bag
release of -base. Re-opens: #511825
2008-01-30 12:28:59 +00:00
Jan Schmidt
bb132eda11 gst-libs/gst/interfaces/: Change the way these deprecated function pointers are removed so that the compiled ABI is u...
Original commit message from CVS:
* gst-libs/gst/interfaces/mixeroptions.h:
* gst-libs/gst/interfaces/mixertrack.h:
Change the way these deprecated function pointers are removed
so that the compiled ABI is unconditionally smaller. This
sets in stone an ABI break that actually occurred when the
things were deprecated in 0.10.14, which seems to be the best
fix as the only known users are oss-mixer and sunaudio-mixer in
gst-plugins-good.
2008-01-30 12:19:02 +00:00
Sebastian Dröge
bd4bb423e9 gst-libs/gst/rtsp/gstrtspconnection.c: Cast glong to time_t as time_t might have a different type on other platforms,...
Original commit message from CVS:
* gst-libs/gst/rtsp/gstrtspconnection.c: (add_date_header):
Cast glong to time_t as time_t might have a different type on
other platforms, like FreeBSD, and we get a compiler warning
otherwise. Fixes bug #511825.
2008-01-29 09:59:03 +00:00
Sebastian Dröge
a6e4222c70 gst-libs/gst/audio/gstaudiofilter.c: Initialize the GstRingerBuffer class to get it's debug category initialized. gst...
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiofilter.c:
(gst_audio_filter_class_init):
Initialize the GstRingerBuffer class to get it's debug category
initialized. gst_ring_buffer_parse_caps() uses the ringbuffer debug
category and otherwise we get some g_critical(). Fixes bug #512334.
2008-01-29 09:47:12 +00:00
Thijs Vermeir
d821d8b783 gst-libs/gst/rtp/gstrtpbuffer.c: Fix typos and wrong extension check. Fixes #511274.
Original commit message from CVS:
Patch by: Thijs Vermeir <thijsvermeir at gmail dot com>
* gst-libs/gst/rtp/gstrtpbuffer.c:
(gst_rtp_buffer_get_extension_data):
Fix typos and wrong extension check. Fixes #511274.
2008-01-22 15:37:49 +00:00
Sébastien Moutte
db4d879623 gst-libs/gst/sdp/gstsdpmessage.c: Include Winsock2.h for VS6 and use a different way initialize hints structure so it...
Original commit message from CVS:
* gst-libs/gst/sdp/gstsdpmessage.c: (is_multicast_address):
Include Winsock2.h for VS6 and use a different way initialize
hints structure so it can build with VS6.
* win32/MANIFEST:
* win32/vs6/libgstsdp.dsp:
* win32/common/libgstsdp.def:
Add new files for libgstsdp.
* win32/vs6/grammar.dsp:
Copy pbutils-enumtypes* from win32/common to pbutils sources folder.
* win32/vs6/gst_plugins_base.dsw:
* win32/vs6/libgstdecodebin.dsp:
* win32/vs6/libgstdecodebin2.dsp:
* win32/vs6/libgstplaybin.dsp:
* win32/vs6/libgstvolume.dsp:
Add new dependencies to the link list.
2008-01-13 21:40:45 +00:00
Jan Schmidt
fa2f3fb98e Add test to see if hstrerror is available or if we need libresolv (Solaris) for it, then use it in libgstrtsp.
Original commit message from CVS:
* configure.ac:
* gst-libs/gst/rtsp/Makefile.am:
Add test to see if hstrerror is available or if we need libresolv
(Solaris) for it, then use it in libgstrtsp.
2008-01-12 23:08:28 +00:00
Jan Schmidt
c571cc1e63 gst-libs/gst/tag/Makefile.am: Fix include path order
Original commit message from CVS:
* gst-libs/gst/tag/Makefile.am:
Fix include path order
2008-01-12 14:54:51 +00:00
Tim-Philipp Müller
145bd67812 Ignore more and make buildbot happy
Original commit message from CVS:
Ignore more and make buildbot happy
2008-01-11 17:15:23 +00:00
Edward Hervey
8efa9193a3 gst-libs/gst/pbutils/install-plugins.*: Add GBoxed GType for GstInstallPluginsContext, this eases the wrapping for bi...
Original commit message from CVS:
* gst-libs/gst/pbutils/install-plugins.c:
(gst_install_plugins_context_copy),
(gst_install_plugins_context_get_type):
* gst-libs/gst/pbutils/install-plugins.h:
Add GBoxed GType for GstInstallPluginsContext, this eases the wrapping
for bindings.
2008-01-11 16:18:10 +00:00
Tim-Philipp Müller
3feb4bc8c5 gst-libs/gst/audio/: Ref audio clock class from a thread-safe context to make sure however unlikely that may be in pr...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_class_init):
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_class_init):
Ref audio clock class from a thread-safe context to make sure
we're not bit by GObjects lack of thread-safety here (#349410),
however unlikely that may be in practice.
2008-01-10 17:55:53 +00:00
Tim-Philipp Müller
526fd3d7da gst-libs/gst/tag/gsttagdemux.c: Post an error message if we can't pull as many bytes as we need for the tag. This mak...
Original commit message from CVS:
* gst-libs/gst/tag/gsttagdemux.c: (gst_tag_demux_pull_start_tag):
Post an error message if we can't pull as many bytes as we need
for the tag. This makes sure the user gets to see a proper error
message if a file with a partial ID3 tag is fed to decodebin, and
not a 'no ID3 tag demuxer' error, which would be confusing
(see #508138).
2008-01-08 21:10:02 +00:00
Tim-Philipp Müller
939d39cdf4 gst-libs/gst/pbutils/descriptions.c: Add description strings for ID3, APE, and ICY tags.
Original commit message from CVS:
* gst-libs/gst/pbutils/descriptions.c: (formats):
Add description strings for ID3, APE, and ICY tags.
2008-01-08 20:59:20 +00:00
Sebastian Dröge
a000758477 gst-libs/gst/audio/gstaudiofilter.c: Don't set element details for the abstract GstAudioFilter class.
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_get_type):
Don't set element details for the abstract GstAudioFilter class.
2008-01-03 07:17:05 +00:00
Sebastian Dröge
0e5857ea26 gst-libs/gst/audio/gstaudiofilter.c: Implement get_unit_size() vmethod of GstBaseTransform.
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiofilter.c:
(gst_audio_filter_class_init), (gst_audio_filter_get_unit_size):
Implement get_unit_size() vmethod of GstBaseTransform.
2008-01-02 12:09:48 +00:00
Edward Hervey
7701190c01 gst-libs/gst/pbutils/: Use glib-enum generator to have a proper enum GType for
Original commit message from CVS:
* gst-libs/gst/pbutils/Makefile.am:
* gst-libs/gst/pbutils/pbutils.h:
Use glib-enum generator to have a proper enum GType for
GST_TYPE_INSTALL_PLUGINS_RETURN so we can easily wrap it in bindings.
2008-01-01 12:53:48 +00:00
Wim Taymans
db8c746760 gst-libs/gst/rtsp/gstrtspdefs.*: Add Location header so that we can start implementing redirects.
Original commit message from CVS:
* gst-libs/gst/rtsp/gstrtspdefs.c: (rtsp_init_status):
* gst-libs/gst/rtsp/gstrtspdefs.h:
Add Location header so that we can start implementing redirects.
See #506025.
2007-12-30 19:21:16 +00:00
David Schleef
d605ca3ba2 gst-libs/gst/video/video.*: Add additional RGBA and RGB-24 video formats.
Original commit message from CVS:
* gst-libs/gst/video/video.c:
* gst-libs/gst/video/video.h:
Add additional RGBA and RGB-24 video formats.
2007-12-22 05:19:00 +00:00
Thijs Vermeir
169259294f gst-libs/gst/riff/riff-read.c: Use GST_ROUND_UP_2 macro
Original commit message from CVS:
* gst-libs/gst/riff/riff-read.c:
Use GST_ROUND_UP_2 macro
2007-12-20 19:43:25 +00:00
Wim Taymans
355e8a940d gst-libs/gst/audio/gstaudiosink.c: Improve debug output.
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func),
(gst_audioringbuffer_open_device),
(gst_audioringbuffer_close_device), (gst_audioringbuffer_acquire),
(gst_audioringbuffer_release), (gst_audioringbuffer_start),
(gst_audioringbuffer_pause), (gst_audioringbuffer_stop),
(gst_audio_sink_create_ringbuffer):
Improve debug output.
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_start),
(gst_ring_buffer_pause), (gst_ring_buffer_delay):
Prevent some functions from doing things and failing when the
ringbuffer is not yet acquired.
2007-12-18 15:56:51 +00:00
Sebastian Dröge
c25cb3fc98 gst-libs/gst/interfaces/interfaces.h: Also remove interfaces.h from CVS as it is not needed anymore.
Original commit message from CVS:
* gst-libs/gst/interfaces/interfaces.h:
Also remove interfaces.h from CVS as it is not needed anymore.
2007-12-18 15:32:49 +00:00
Sebastian Dröge
ee3edd8901 gst-libs/gst/interfaces/Makefile.am: interfaces.h is not used anymore so remove it from the build process.
Original commit message from CVS:
* gst-libs/gst/interfaces/Makefile.am:
interfaces.h is not used anymore so remove it from the build
process.
2007-12-18 15:20:12 +00:00
David Schleef
1c3e012fc3 Add new GstVideFormat enum and write a bunch of helper functions based around it.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/video/video.c:
* gst-libs/gst/video/video.h:
Add new GstVideFormat enum and write a bunch of helper functions
based around it.
2007-12-18 00:13:26 +00:00
Wim Taymans
2ea251a366 gst-libs/gst/audio/gstbaseaudiosrc.c: Add debug info.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_create), (gst_base_audio_src_change_state):
Add debug info.
When going from PLAYING to PAUSED, pause the ringbuffer before calling
the parent state change function, just like the audiosink, because the
parent waits for the element to finish its processing before completing
the state change. This makes going to PAUSED a lot snappier.
When going from READY to PAUSED, don't allow the ringbuffer to start
yet.
2007-12-17 16:44:51 +00:00
Edward Hervey
80d1fbac43 gst-libs/gst/riff/riff-media.c: Yet another fix for broken software that produce files with an empty blockalign field...
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
Yet another fix for broken software that produce files with an empty
blockalign field. Instead of completely failing, make a second attempt
at guessing the width/depth by looking at strf->size.
2007-12-17 00:01:00 +00:00
Tim-Philipp Müller
aa47d077c1 gst-libs/gst/: Turn a few g_assert_not_reached() into g_return_val_if_reached() to avoid compiler warnings (#503930).
Original commit message from CVS:
* gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_do_seek),
(gst_cdda_base_src_handle_track_seek), (gst_cdda_base_src_create):
* gst-libs/gst/pbutils/install-plugins.c:
(gst_install_plugins_spawn_child), (gst_install_plugins_supported):
* gst-libs/gst/pbutils/missing-plugins.c:
(gst_missing_plugin_message_get_installer_detail),
(gst_missing_encoder_installer_detail_new):
* gst-libs/gst/rtsp/gstrtspconnection.c: (gst_rtsp_connection_send):
* gst-libs/gst/rtsp/gstrtspmessage.c: (gst_rtsp_message_unset):
Turn a few g_assert_not_reached() into g_return_val_if_reached() to
avoid compiler warnings (#503930).
2007-12-16 23:52:58 +00:00
Edward Hervey
ea931c47d6 gst-libs/gst/riff/riff-media.c: Be apologetic of software that use the 'jpeg' instead of 'JPEG' FOURCC for jpeg video...
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
Be apologetic of software that use the 'jpeg' instead of 'JPEG' FOURCC
for jpeg video streams.
Add the 'avc1'/'AVC1' fourcc mapping for h264, same software-comment as
for the above modification.
2007-12-16 23:46:16 +00:00
Tim-Philipp Müller
d23f2ae402 gst-libs/gst/interfaces/xoverlay.c: More guards (we don't want klass to end up being NULL).
Original commit message from CVS:
* gst-libs/gst/interfaces/xoverlay.c: (gst_x_overlay_expose),
(gst_x_overlay_handle_events):
More guards (we don't want klass to end up being NULL).
2007-12-15 17:27:48 +00:00
Peter Kjellerstedt
ba4bb0fa8c gst-libs/gst/rtsp/gstrtspconnection.c: Close control sockets. Fixes #503440.
Original commit message from CVS:
Patch by: Peter Kjellerstedt  <pkj at axis com>
* gst-libs/gst/rtsp/gstrtspconnection.c:
(gst_rtsp_connection_free):
Close control sockets. Fixes #503440.
2007-12-13 15:54:00 +00:00
Tim-Philipp Müller
41e54ce132 gst-libs/gst/pbutils/descriptions.c: Add description for 'private' dts caps (who come up with that name?).
Original commit message from CVS:
* gst-libs/gst/pbutils/descriptions.c: (formats):
Add description for 'private' dts caps (who come up with that name?).
2007-12-13 11:40:10 +00:00
Tim-Philipp Müller
b9cbedfb59 Makefile.am: Add check-exports target and run it with 'make check'.
Original commit message from CVS:
* Makefile.am:
Add check-exports target and run it with 'make check'.
* configure.ac:
Be stricter about what we export in our libraries: change regexp so that
we only export _gst_foo(), but not __gst_foo().
* gst-libs/gst/cdda/base64.h: (rfc822_binary):
* gst-libs/gst/cdda/sha1.h: (sha_init), (sha_update), (sha_final):
Change internal functions to __gst_foo so they dont' get exported.
* win32/common/libgstaudio.def:
Add missing symbols.
2007-12-13 10:10:35 +00:00
Wim Taymans
74170deee7 gst-libs/gst/riff/riff-media.c: No need for floating point operations here. avoids having to link against the math li...
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
No need for floating point operations here. avoids having to link
against the math library too.
2007-12-10 15:13:55 +00:00
Tim-Philipp Müller
6623ddbfbd Add one or two missing formats. Generate ADPCM description dynamically depending on layout/format.
Original commit message from CVS:
* gst-libs/gst/pbutils/descriptions.c: (formats),
(format_info_get_desc):
* tests/check/libs/pbutils.c: (GST_START_TEST), (caps_strings),
(GST_START_TEST):
Add one or two missing formats.  Generate ADPCM description
dynamically depending on layout/format.
2007-12-10 11:16:25 +00:00
Wim Taymans
75aa042450 gst-libs/gst/rtp/gstbasertppayload.c: Use runnning time as the base time instead of the timestamp.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
Use runnning time as the base time instead of the timestamp.
Spotted by Saur on IRC.
2007-12-03 13:47:00 +00:00
Edward Hervey
ab433ee7c9 gst-libs/gst/riff/riff-media.c: Add 'WVC1' codec mapping for Windows Media VC-1 video codec.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
Add 'WVC1' codec mapping for Windows Media VC-1 video codec.
2007-12-03 11:32:30 +00:00
Wim Taymans
ac1cc82165 gst-libs/gst/audio/gstbaseaudiosink.c: Our EOS time contains the base_time, _wait_eos() expects a running_time so we ...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_drain):
Our EOS time contains the base_time, _wait_eos() expects a running_time
so we have to subtract the base_time again before calling the function.
This fixes an EOS regression where the base_time was added twice and EOS
took longer and longer in certain situations.
Fixes #498767.
2007-11-21 18:02:21 +00:00
Wim Taymans
157a65b15e Expose methods for some object properties so that subclasses can more easily configure them.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/audio/gstbaseaudiosink.c: (slave_method_get_type),
(gst_base_audio_sink_set_provide_clock),
(gst_base_audio_sink_get_provide_clock),
(gst_base_audio_sink_set_slave_method),
(gst_base_audio_sink_get_slave_method),
(gst_base_audio_sink_set_property),
(gst_base_audio_sink_get_property), (gst_base_audio_sink_drain),
(gst_base_audio_sink_none_slaving),
(gst_base_audio_sink_handle_slaving):
* gst-libs/gst/audio/gstbaseaudiosink.h:
Expose methods for some object properties so that subclasses can more
easily configure them.
Added slave method none, that completely disables slaving to the
internal clock.
API: gst_base_audio_sink_set_provide_clock()
API: gst_base_audio_sink_get_provide_clock()
API: gst_base_audio_sink_set_slave_method()
API: gst_base_audio_sink_get_slave_method()
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_set_provide_clock),
(gst_base_audio_src_get_provide_clock),
(gst_base_audio_src_set_property),
(gst_base_audio_src_get_property), (gst_base_audio_src_create):
* gst-libs/gst/audio/gstbaseaudiosrc.h:
Expose methods for some object properties so that subclasses can more
easily configure them.
API: gst_base_audio_src_set_provide_clock()
API: gst_base_audio_src_get_provide_clock()
2007-11-21 13:04:17 +00:00
Joe Peterson
bd8885220e gst-libs/gst/sdp/gstsdpmessage.c: Fix compilation on FreeBSD (Gentoo). Fixes #498228.
Original commit message from CVS:
Patch by: Joe Peterson <lavajoe at gentoo dot org>
* gst-libs/gst/sdp/gstsdpmessage.c:
Fix compilation on FreeBSD (Gentoo). Fixes #498228.
2007-11-20 07:53:56 +00:00
Tommi Myöhänen
385ed6384f gst-libs/gst/rtsp/gstrtspmessage.c: Fix leaking headers. Fixes #496761.
Original commit message from CVS:
Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
* gst-libs/gst/rtsp/gstrtspmessage.c: (gst_rtsp_message_unset):
Fix leaking headers. Fixes #496761.
2007-11-16 11:22:09 +00:00
Tim-Philipp Müller
dffc29eb13 gst-libs/gst/tag/gstid3tag.c: Add mapping for audio cd discid tags, so we can extract them from tags as well (see #34...
Original commit message from CVS:
* gst-libs/gst/tag/gstid3tag.c: (user_tag_matches),
(gst_tag_from_id3_user_tag):
Add mapping for audio cd discid tags, so we can extract
them from tags as well (see #347848). Also compare identifiers
in ID3v2 TXXX frames in a case-insensitive way to increase
compatibility when reading tags (discid vs. DiscID vs. DiscId).
2007-11-16 10:14:34 +00:00
Jan Schmidt
31f97ca497 gst-libs/gst/fft/: Don't include malloc.h which doesn't exist on Mac OSX.
Original commit message from CVS:
* gst-libs/gst/fft/kiss_fft_f32.h:
* gst-libs/gst/fft/kiss_fft_f64.h:
* gst-libs/gst/fft/kiss_fft_s16.h:
* gst-libs/gst/fft/kiss_fft_s32.h:
Don't include malloc.h which doesn't exist on Mac OSX.
Instead, pull in glib.h and use g_malloc/g_free for
consistency. Fixes: #496548
2007-11-14 11:08:48 +00:00
Sebastien Moutte
13d89656c0 gst-libs/gst/rtp/: Fix some C99-isms and and a missing function that some versions of
Original commit message from CVS:
Patch by: Sebastien Moutte  <sebastien moutte net>
* gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_ntp_to_unix),
(gst_rtcp_unix_to_ntp):
* gst-libs/gst/rtp/gstrtppayloads.c: (gst_rtp_payload_info_for_name):
Fix some C99-isms and and a missing function that some versions of
MSVC don't like too much (#494346).
* win32/vs6/gst_plugins_base.dsw:
* win32/vs6/libgstaudio.dsp:
* win32/vs6/libgstrtp.dsp:
* win32/vs6/libgsttag.dsp:
Update vs6 projects files (#494346).
2007-11-06 23:35:39 +00:00
Sebastian Dröge
edb4a505d7 Remove the magnitude and phase calculation functions as these have very special use cases and can't even be used for ...
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/fft/gstfftf32.c:
* gst-libs/gst/fft/gstfftf32.h:
* gst-libs/gst/fft/gstfftf64.c:
* gst-libs/gst/fft/gstfftf64.h:
* gst-libs/gst/fft/gstffts16.c:
* gst-libs/gst/fft/gstffts16.h:
* gst-libs/gst/fft/gstffts32.c:
* gst-libs/gst/fft/gstffts32.h:
* tests/check/libs/fft.c: (GST_START_TEST):
Remove the magnitude and phase calculation functions as these have
very special use cases and can't even be used for the spectrum
element. Also adjust the docs to mention some properties of the used
FFT implemention, i.e. how the values are scaled. Fixes #492098.
2007-11-06 11:58:59 +00:00
Vincent Torri
be04e7c294 gst-libs/gst/fft/: Include our own _stdint.h instead of sys/types.h, makes MingW happy (#492306).
Original commit message from CVS:
Patch by: Vincent Torri <vtorri at univ-evry dot fr>
* gst-libs/gst/fft/kiss_fft_s16.h: (KISS_FFT_S16_MALLOC):
* gst-libs/gst/fft/kiss_fft_s32.h: (KISS_FFT_S32_MALLOC):
Include our own _stdint.h instead of sys/types.h, makes MingW happy
(#492306).
* gst-libs/gst/rtsp/gstrtspconnection.c: (gst_rtsp_connection_create):
Use _pipe directly, GLib doesn't have a pipe() macro any longer
(it disappeared in GLib 2.14.0) (#492306).
* gst-libs/gst/sdp/Makefile.am:
* gst-libs/gst/sdp/gstsdpmessage.c:
Fix includes and LIBS for win32/Mingw (#492306).
* tests/examples/dynamic/addstream.c (pause_play_stream):
Use more portable g_usleep() instead of sleep() (#492306).
2007-11-01 13:28:59 +00:00
Ole André Vadla Ravnås
05a205860d gst-libs/gst/audio/gstringbuffer.c: Return NULL instead of an enum that happens to be 0, fixes warning on MSVC (#4921...
Original commit message from CVS:
Patch by: Ole André Vadla Ravnås  <ole.andre.ravnas@tandberg.com>
* gst-libs/gst/audio/gstringbuffer.c: (build_linear_format),
(gst_ring_buffer_parse_caps):
Return NULL instead of an enum that happens to be 0, fixes warning
on MSVC (#492114).
* gst-libs/gst/audio/gstringbuffer.h:
No trailing commas in enum list (for gcc-2.9x).
* gst/videotestsrc/videotestsrc.c: (random_char):
Make information loss explicit instead of implicitly truncating to
eight bits via the return value.  Fixes runtime error on MSVC when
using the debug CRT (#492114).
* win32/common/config.h.in:
Fix a bunch of '#undef FOO bar', which MSVC doesn't like (#492114).
* win32/common/libgstinterfaces.def:
* win32/common/libgstrtp.def:
Export a few more symbols (#492114).
2007-11-01 12:51:57 +00:00
Stefan Kost
28b46c1e5d gst-libs/gst/audio/audio.*: Readd the deprecation guards, but preserve compilability.
Original commit message from CVS:
* gst-libs/gst/audio/audio.c:
* gst-libs/gst/audio/audio.h:
Readd the deprecation guards, but preserve compilability.
2007-11-01 08:06:13 +00:00
Tim-Philipp Müller
55a3eaafea gst-libs/gst/audio/audio.h: Don't guard deprecated enum with #ifndef DISABLE_DEPRECATED, or compilation with DISABLE_...
Original commit message from CVS:
* gst-libs/gst/audio/audio.h:
Don't guard deprecated enum with #ifndef DISABLE_DEPRECATED, or
compilation with DISABLE_DEPRECATED and without REMOVE_DEPRECATED
(ie. normal cvs builds) will fail.
2007-10-31 15:30:15 +00:00
Stefan Kost
e37568c196 tell gtk-doc about the deprecation guard. Apply more doc fixes.
Original commit message from CVS:
* docs/libs/Makefile.am:
* gst-libs/gst/audio/audio.c:
* gst-libs/gst/audio/audio.h:
* gst-libs/gst/interfaces/mixer.c:
tell gtk-doc about the deprecation guard. Apply more doc fixes.
2007-10-31 12:47:41 +00:00
Stefan Kost
ffa52e2eac Fix the docs according to what gtk-doc complained about.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/audio/gstaudiofilter.h:
* gst-libs/gst/interfaces/mixer.h:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
* gst-libs/gst/rtp/gstbasertpdepayload.h:
* gst-libs/gst/sdp/gstsdpmessage.c:
Fix the docs according to what gtk-doc complained about.
2007-10-30 20:32:14 +00:00
Sebastian Dröge
a81de45df3 gst-libs/gst/riff/riff-media.c: Use the ALSA channel layout as default for wav files without channel layout informati...
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c:
(gst_riff_wavext_add_channel_layout),
(gst_riff_wave_add_default_channel_layout),
(gst_riff_wavext_get_default_channel_mask),
(gst_riff_create_audio_caps):
Use the ALSA channel layout as default for wav files without channel
layout information. This fixes playback of chan-id.wav on 5.1 systems
for example. Also refactor the channel layout setting a bit and add
more default channel orders. Fixes #489010.
2007-10-28 11:53:36 +00:00
Tim-Philipp Müller
698e4050b3 gst-libs/gst/tag/: Deprecate GST_TAG_MUSICBRAINZ_SORTNAME, replaced by the newly-added
Original commit message from CVS:
* gst-libs/gst/tag/tag.h: (GST_TAG_MUSICBRAINZ_SORTNAME):
* gst-libs/gst/tag/tags.c:
Deprecate GST_TAG_MUSICBRAINZ_SORTNAME, replaced by the newly-added
GST_TAG_ARTIST_SORTNAME (in an API and ABI compatible way).
* gst-libs/gst/tag/gstid3tag.c: (tag_matches):
Map ID3v2 TSOP, TSOA and TSOT frames to new SORTNAME tags (#414539).
* gst-libs/gst/tag/gstvorbistag.c: (tag_matches),
(gst_tag_to_vorbis_comments):
Map new SORTNAME tags (these tags aren't even semi-official, so I'm
just mapping everything I found in the wild) (#414539).
2007-10-25 15:10:59 +00:00
Tim-Philipp Müller
d33e6ca818 gst-libs/gst/tag/gsttagdemux.c: Don't abort with an assertion if we receive a seek event with a start type of NONE (s...
Original commit message from CVS:
* gst-libs/gst/tag/gsttagdemux.c:
Don't abort with an assertion if we receive a seek event with
a start type of NONE (see launchpad bug #155878).
2007-10-23 14:23:14 +00:00
Wim Taymans
6a20747e83 gst-libs/gst/audio/gstbaseaudiosink.c: Also explicitly release the ringbuffer when going to NULL because it is requir...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_drain), (gst_base_audio_sink_change_state):
Also explicitly release the ringbuffer when going to NULL because it
is required in the setcaps function, before the state change to PAUSED
completes.
2007-10-16 15:33:31 +00:00
Tim-Philipp Müller
689ca135f3 Extract vorbis comment LICENSE tags correctly.
Original commit message from CVS:
* gst-libs/gst/tag/gstvorbistag.c:
* tests/check/libs/tag.c:
Extract vorbis comment LICENSE tags correctly.
2007-10-11 18:24:09 +00:00
Jason Kivlighn
054e43aa3c Map ID3v2 WCOP frame to GST_TAG_COPYRIGHT_URI (#447000).
Original commit message from CVS:
Patch by: Jason Kivlighn  <jkivlighn gmail com>
* gst-libs/gst/tag/gstid3tag.c:
* tests/check/libs/tag.c:
Map ID3v2 WCOP frame to GST_TAG_COPYRIGHT_URI (#447000).
2007-10-11 16:12:21 +00:00
Tim-Philipp Müller
8458a74526 gst-libs/gst/tag/gsttagdemux.c: Don't error out when a buggy downstream element doesn't handle the newsegment event w...
Original commit message from CVS:
* gst-libs/gst/tag/gsttagdemux.c:
Don't error out when a buggy downstream element doesn't
handle the newsegment event we send properly (especially
not without posting a meaningful error message on the
bus). See bug #471370 and launchpad bug #136264.
2007-10-10 17:01:51 +00:00
Wim Taymans
02f280a9a0 gst-libs/gst/audio/gstbaseaudiosink.c: Use new basesink method to make our EOS drain interruptable.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_drain):
Use new basesink method to make our EOS drain interruptable.
2007-10-10 15:36:56 +00:00
Jan Schmidt
c96877779a gst-libs/gst/rtp/gstrtppayloads.c: Fix silly search-replace oversight.
Original commit message from CVS:
* gst-libs/gst/rtp/gstrtppayloads.c:
Fix silly search-replace oversight.
2007-10-10 09:37:09 +00:00
Laurent Glayal
0f3c871859 gst-libs/gst/rtp/gstbasertppayload.c: Fix caps memleak. Fixes #484989.
Original commit message from CVS:
Patch by: Laurent Glayal <spglegle at yahoo dot fr>
* gst-libs/gst/rtp/gstbasertppayload.c: (copy_fixed),
(gst_basertppayload_set_outcaps):
Fix caps memleak. Fixes #484989.
2007-10-09 09:57:17 +00:00
Wim Taymans
4f707e2172 gst-libs/gst/rtp/gstbasertpdepayload.c: Fix debug output.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_setcaps), (gst_base_rtp_depayload_chain):
Fix debug output.
2007-10-08 18:04:34 +00:00
Wim Taymans
c3dda05a8b gst-libs/gst/audio/gstbaseaudiosrc.c: Also handle the case where there is no clock set on the audio source, like in t...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_create):
Also handle the case where there is no clock set on the audio source,
like in the unit tests.
2007-10-08 18:02:53 +00:00
Jan Schmidt
41ae026041 gst-libs/gst/rtp/gstrtppayloads.c: Use unsigned G_MAXUINT8 instead of -1 to initialise a guint8 to avoid compiler war...
Original commit message from CVS:
* gst-libs/gst/rtp/gstrtppayloads.c:
Use unsigned G_MAXUINT8 instead of -1 to initialise a guint8
to avoid compiler warnings
2007-10-08 17:40:17 +00:00
Tim-Philipp Müller
d3648ca534 gst-libs/gst/tag/gsttagdemux.c: Don't leak caps.
Original commit message from CVS:
* gst-libs/gst/tag/gsttagdemux.c:
Don't leak caps.
2007-10-06 16:49:55 +00:00
Tim-Philipp Müller
f277cd8801 gst-libs/gst/tag/: API: add GstTagDemux base class for simple tag demuxers.
Original commit message from CVS:
* gst-libs/gst/tag/Makefile.am:
* gst-libs/gst/tag/gsttagdemux.c:
* gst-libs/gst/tag/gsttagdemux.h:
API: add GstTagDemux base class for simple tag demuxers.
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
Add GstTagDemux to docs.
2007-10-06 15:04:53 +00:00
Sebastian Dröge
6e51063659 gst-libs/gst/rtp/gstrtpbuffer.c: Fix bug introduced with last commit which inverted the logic and caused all buffers ...
Original commit message from CVS:
* gst-libs/gst/rtp/gstrtpbuffer.c:
(gst_rtp_buffer_get_payload_subbuffer):
Fix bug introduced with last commit which inverted the logic and
caused all buffers to be dropped. Fixes #483620.
Thanks to Laurent Glayal <spglegle at yahoo dot fr> for noticing.
2007-10-05 07:49:25 +00:00
Stefan Kost
ff01787e4e gst-libs/gst/rtp/gstrtpbuffer.c: with regular return and warning.
Original commit message from CVS:
* gst-libs/gst/rtp/gstrtpbuffer.c:
Replace g_return_if_val (as it could be disabled), with regular return
and warning.
2007-10-04 06:50:53 +00:00
Wim Taymans
5ba1ed3a21 gst-libs/gst/audio/gstbaseaudiosink.c: When slaved to the clock, don't try to align a sample with the previous one wh...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_async_play):
When slaved to the clock, don't try to align a sample with the previous
one when going to PLAYING again.
2007-10-02 11:11:13 +00:00
Wim Taymans
767366a1e1 gst-libs/gst/sdp/gstsdpmessage.h: Add RFC 3556 bandwidth modifiers.
Original commit message from CVS:
* gst-libs/gst/sdp/gstsdpmessage.h:
Add RFC 3556 bandwidth modifiers.
2007-10-01 16:33:00 +00:00
Wim Taymans
d874f65f6e Update documentation.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/rtp/gstrtppayloads.c:
Update documentation.
2007-10-01 13:37:31 +00:00
Wim Taymans
7cdfb6d154 gst-libs/gst/rtp/: Added new file and header to deal with payload info.
Original commit message from CVS:
* gst-libs/gst/rtp/Makefile.am:
* gst-libs/gst/rtp/gstrtppayloads.c: (gst_rtp_payload_info_for_pt),
(gst_rtp_payload_info_for_name):
* gst-libs/gst/rtp/gstrtppayloads.h:
Added new file and header to deal with payload info.
* gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_validate_data),
(gst_rtp_buffer_default_clock_rate):
* gst-libs/gst/rtp/gstrtpbuffer.h:
Payload specific stuff is move to new headers.
Implement _default_clock rate using the new payload function.
* gst-libs/gst/sdp/gstsdpmessage.c: (is_multicast_address),
(gst_sdp_parse_line):
* gst-libs/gst/sdp/gstsdpmessage.h:
Add some more comments.
2007-10-01 13:22:14 +00:00
Tim-Philipp Müller
3263d790f9 Add mappings for the new GST_TAG_COMPOSER for vorbis comments and ID3v2 tags.
Original commit message from CVS:
* configure.ac:
* gst-libs/gst/tag/gstid3tag.c:
* gst-libs/gst/tag/gstvorbistag.c:
Add mappings for the new GST_TAG_COMPOSER for vorbis comments
and ID3v2 tags.
2007-09-25 19:06:47 +00:00
Tim-Philipp Müller
3f7d77fc29 gst-libs/gst/floatcast/floatcast.h: Don't include config.h in an installed public header, this might break compilatio...
Original commit message from CVS:
* gst-libs/gst/floatcast/floatcast.h:
Don't include config.h in an installed public header, this
might break compilation of applications that don't have such
a header and doesn't necessarily do what it's supposed to do
anyway (ie. check for the lrint/lrintf defines) (#442065).
Add docs for the various macros and document how this header
has to be used (link against libm, etc.); add a few FIXMEs;
include math.h for non-c99 code path.  Based on patch by
Jan Schmidt.
2007-09-25 11:54:09 +00:00
Wim Taymans
0bba976bd3 gst-libs/gst/app/gstappsink.*: Add properties, signals and actions to access the element even without linking to the ...
Original commit message from CVS:
* gst-libs/gst/app/gstappsink.c: (gst_app_marshal_OBJECT__VOID),
(gst_app_sink_class_init), (gst_app_sink_init),
(gst_app_sink_dispose), (gst_app_sink_finalize),
(gst_app_sink_set_property), (gst_app_sink_get_property),
(gst_app_sink_flush_unlocked), (gst_app_sink_start),
(gst_app_sink_event), (gst_app_sink_getcaps),
(gst_app_sink_set_caps), (gst_app_sink_get_caps),
(gst_app_sink_is_eos), (gst_app_sink_pull_preroll),
(gst_app_sink_pull_buffer):
* gst-libs/gst/app/gstappsink.h:
Add properties, signals and actions to access the element even without
linking to the library.
Fix some method names and signatures.
2007-09-20 13:59:50 +00:00
Wim Taymans
2f0a269ddd gst-libs/gst/rtp/gstbasertpdepayload.c: Only copy timestamp on outgoing packets if the depayloader did not set one.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_chain),
(gst_base_rtp_depayload_set_gst_timestamp):
Only copy timestamp on outgoing packets if the depayloader did not set
one.
Also copy duration on outgoing packets.
2007-09-19 16:09:57 +00:00
Wim Taymans
70ad18deee gst-libs/gst/rtp/gstbasertppayload.c: Fix compilation because of missing %d in printf.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertppayload.c: (copy_fixed),
(gst_basertppayload_set_outcaps):
Fix compilation because of missing %d in printf.
When fixating caps, fixate what we can and throw away all remaining
unfixed caps, subclasses should do something smart if they need to.
2007-09-19 15:55:08 +00:00
Jan Schmidt
d5996e9c37 Fix a bunch of compile warnings shown with Forte.
Original commit message from CVS:
* ext/pango/gsttextoverlay.c: (gst_text_overlay_init),
(gst_text_overlay_set_property):
* ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
* gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_ntp_to_unix),
(gst_rtcp_unix_to_ntp):
* gst-libs/gst/rtsp/gstrtspmessage.c: (gst_rtsp_message_get_type):
* gst/playback/gstqueue2.c:
* tests/examples/seek/seek.c: (set_scale):
Fix a bunch of compile warnings shown with Forte.
* gst/audiorate/gstaudiorate.c:
Always pull in config.h before including any system headers.
2007-09-17 17:24:55 +00:00
Wim Taymans
523fd097e6 gst-libs/gst/rtp/gstbasertpdepayload.c: Remove code to deal with RTP to GST time conversion, we now just copy the GST...
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_init), (gst_base_rtp_depayload_setcaps),
(gst_base_rtp_depayload_chain),
(gst_base_rtp_depayload_handle_sink_event),
(gst_base_rtp_depayload_push_full),
(gst_base_rtp_depayload_set_gst_timestamp),
(gst_base_rtp_depayload_change_state):
Remove code to deal with RTP to GST time conversion, we now just copy
the GST timestamp we receive to the outgoing buffers.
Handle segment and flushes correctly.
* gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
When we have no valid input timestamp, use the previous rtp timestamp on
the outgoing RTP packet instead of the RTP base time.
2007-09-16 19:31:06 +00:00