platform/upstream/gstreamer.git
16 years agogst-libs/gst/app/gstappsrc.c (gst_app_src_set_max_bytes)
Andy Wingo [Mon, 16 Jun 2008 14:11:36 +0000 (14:11 +0000)]
gst-libs/gst/app/gstappsrc.c (gst_app_src_set_max_bytes)

Original commit message from CVS:
2008-06-16  Andy Wingo  <wingo@pobox.com>

* gst-libs/gst/app/gstappsrc.c (gst_app_src_set_max_bytes)
(gst_app_src_get_max_bytes, gst_app_src_push_buffer): Use
G_GUINT64_FORMAT. Avoid overflow in get_max_bytes().

16 years agoFinal round of doc updates.
Stefan Kost [Mon, 16 Jun 2008 07:30:32 +0000 (07:30 +0000)]
Final round of doc updates.

Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
* gst/speed/gstspeed.c:
* gst/speexresample/gstspeexresample.c:
* gst/videosignal/gstvideoanalyse.c:
* gst/videosignal/gstvideodetect.c:
* gst/videosignal/gstvideomark.c:
* sys/dvb/gstdvbsrc.c:
* sys/oss4/oss4-mixer.c:
* sys/oss4/oss4-sink.c:
* sys/oss4/oss4-source.c:
* sys/wininet/gstwininetsrc.c:
Final round of doc updates.

16 years agodocs/plugins/: docs/plugins/inspect/plugin-mythtv.xml
Stefan Kost [Fri, 13 Jun 2008 11:59:21 +0000 (11:59 +0000)]
docs/plugins/: docs/plugins/inspect/plugin-mythtv.xml

Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-bad-plugins-docs.sgml:
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
* docs/plugins/gst-plugins-bad-plugins.args:
* docs/plugins/gst-plugins-bad-plugins.hierarchy:
* docs/plugins/gst-plugins-bad-plugins.interfaces:
* docs/plugins/gst-plugins-bad-plugins.prerequisites:
* docs/plugins/gst-plugins-bad-plugins.signals:
* docs/plugins/inspect/plugin-alsaspdif.xml:
* docs/plugins/inspect/plugin-amrwb.xml:
* docs/plugins/inspect/plugin-app.xml:
* docs/plugins/inspect/plugin-bayer.xml:
* docs/plugins/inspect/plugin-bz2.xml:
* docs/plugins/inspect/plugin-cdaudio.xml:
* docs/plugins/inspect/plugin-cdxaparse.xml:
* docs/plugins/inspect/plugin-dtsdec.xml:
* docs/plugins/inspect/plugin-dvb.xml:
* docs/plugins/inspect/plugin-dvdspu.xml:
* docs/plugins/inspect/plugin-faac.xml:
* docs/plugins/inspect/plugin-faad.xml:
* docs/plugins/inspect/plugin-fbdevsink.xml:
* docs/plugins/inspect/plugin-festival.xml:
* docs/plugins/inspect/plugin-filter.xml:
* docs/plugins/inspect/plugin-flvdemux.xml:
* docs/plugins/inspect/plugin-freeze.xml:
* docs/plugins/inspect/plugin-gsm.xml:
* docs/plugins/inspect/plugin-gstinterlace.xml:
* docs/plugins/inspect/plugin-gstrtpmanager.xml:
* docs/plugins/inspect/plugin-h264parse.xml:
* docs/plugins/inspect/plugin-interleave.xml:
* docs/plugins/inspect/plugin-jack.xml:
* docs/plugins/inspect/plugin-ladspa.xml:
* docs/plugins/inspect/plugin-metadata.xml:
* docs/plugins/inspect/plugin-mms.xml:
* docs/plugins/inspect/plugin-modplug.xml:
* docs/plugins/inspect/plugin-mpeg2enc.xml:
* docs/plugins/inspect/plugin-mpeg4videoparse.xml:
* docs/plugins/inspect/plugin-mpegtsparse.xml:
* docs/plugins/inspect/plugin-mpegvideoparse.xml:
* docs/plugins/inspect/plugin-musepack.xml:
* docs/plugins/inspect/plugin-musicbrainz.xml:
* docs/plugins/inspect/plugin-mve.xml:
* docs/plugins/inspect/plugin-mythtv.xml
* docs/plugins/inspect/plugin-nas.xml:
* docs/plugins/inspect/plugin-neon.xml:
* docs/plugins/inspect/plugin-nsfdec.xml:
* docs/plugins/inspect/plugin-nuvdemux.xml:
* docs/plugins/inspect/plugin-oss4.xml
* docs/plugins/inspect/plugin-rawparse.xml:
* docs/plugins/inspect/plugin-real.xml:
* docs/plugins/inspect/plugin-replaygain.xml:
* docs/plugins/inspect/plugin-rfbsrc.xml:
* docs/plugins/inspect/plugin-sdl.xml:
* docs/plugins/inspect/plugin-sdp.xml:
* docs/plugins/inspect/plugin-selector.xml:
* docs/plugins/inspect/plugin-sndfile.xml:
* docs/plugins/inspect/plugin-soundtouch.xml:
* docs/plugins/inspect/plugin-spcdec.xml:
* docs/plugins/inspect/plugin-speed.xml:
* docs/plugins/inspect/plugin-speexresample.xml:
* docs/plugins/inspect/plugin-stereo.xml:
* docs/plugins/inspect/plugin-subenc.xml
* docs/plugins/inspect/plugin-timidity.xml:
* docs/plugins/inspect/plugin-tta.xml:
* docs/plugins/inspect/plugin-vcdsrc.xml:
* docs/plugins/inspect/plugin-videosignal.xml:
* docs/plugins/inspect/plugin-vmnc.xml:
* docs/plugins/inspect/plugin-wildmidi.xml:
* docs/plugins/inspect/plugin-x264.xml:
* docs/plugins/inspect/plugin-xvid.xml:
* docs/plugins/inspect/plugin-y4menc.xml:
* ext/amrwb/gstamrwbdec.c:
* ext/amrwb/gstamrwbenc.c:
* ext/amrwb/gstamrwbparse.c:
* ext/dc1394/gstdc1394.c:
* ext/directfb/dfbvideosink.c:
* ext/ivorbis/vorbisdec.c:
* ext/jack/gstjackaudiosink.c:
* ext/mpeg2enc/gstmpeg2enc.cc:
* ext/mplex/gstmplex.cc:
* ext/musicbrainz/gsttrm.c:
* ext/mythtv/gstmythtvsrc.c:
* ext/theora/theoradec.c:
* ext/timidity/gsttimidity.c:
* ext/timidity/gstwildmidi.c:
* gst-libs/gst/app/gstappsink.c:
* gst/deinterlace/gstdeinterlace.c:
* gst/dvdspu/gstdvdspu.c:
* gst/festival/gstfestival.c:
* gst/freeze/gstfreeze.c:
* gst/interleave/deinterleave.c:
* gst/interleave/interleave.c:
* gst/modplug/gstmodplug.cc:
* gst/nuvdemux/gstnuvdemux.c:
Add missing elements to docs. Fix doc-markup: use convinience syntax
for examples (produces valid docbook), add several refsec2 when we
have several titles. Fix some types.

16 years agoexamples/app/: Add beefed up example app from bug #413418. It now also uses appsink...
Wim Taymans [Thu, 12 Jun 2008 15:47:03 +0000 (15:47 +0000)]
examples/app/: Add beefed up example app from bug #413418. It now also uses appsink instead of fakesink for more ulti...

Original commit message from CVS:
* examples/app/.cvsignore:
* examples/app/Makefile.am:
* examples/app/appsink-src.c: (on_new_buffer_from_source),
(on_source_message), (on_sink_message), (main):
Add beefed up example app from bug #413418. It now also uses appsink
instead of fakesink for more ultimate coolness.
* gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init),
(gst_app_src_init), (gst_app_src_set_property),
(gst_app_src_get_property), (gst_app_src_unlock),
(gst_app_src_unlock_stop), (gst_app_src_create),
(gst_app_src_set_max_bytes), (gst_app_src_push_buffer),
(gst_app_src_end_of_stream):
* gst-libs/gst/app/gstappsrc.h:
Add block property to allow push based implementation to block when we
fill up the appsrc queues.
Emit the enough-data signal while releasing our lock.

16 years agoexamples/app/.cvsignore: Ignore more.
Stefan Kost [Thu, 12 Jun 2008 14:50:27 +0000 (14:50 +0000)]
examples/app/.cvsignore: Ignore more.

Original commit message from CVS:
* examples/app/.cvsignore:
Ignore more.

16 years agoDo not use short_description in section docs for elements. We extract them from eleme...
Stefan Kost [Thu, 12 Jun 2008 14:49:15 +0000 (14:49 +0000)]
Do not use short_description in section docs for elements. We extract them from element details and there will be war...

Original commit message from CVS:
* ext/dc1394/gstdc1394.c:
* ext/ivorbis/vorbisdec.c:
* ext/jack/gstjackaudiosink.c:
* ext/metadata/gstmetadatademux.c:
* ext/mythtv/gstmythtvsrc.c:
* ext/theora/theoradec.c:
* gst-libs/gst/app/gstappsink.c:
* gst/bayer/gstbayer2rgb.c:
* gst/deinterlace/gstdeinterlace.c:
* gst/rawparse/gstaudioparse.c:
* gst/rawparse/gstvideoparse.c:
* gst/rtpmanager/gstrtpbin.c:
* gst/rtpmanager/gstrtpclient.c:
* gst/rtpmanager/gstrtpjitterbuffer.c:
* gst/rtpmanager/gstrtpptdemux.c:
* gst/rtpmanager/gstrtpsession.c:
* gst/rtpmanager/gstrtpssrcdemux.c:
* gst/selector/gstinputselector.c:
* gst/selector/gstoutputselector.c:
* gst/videosignal/gstvideoanalyse.c:
* gst/videosignal/gstvideodetect.c:
* gst/videosignal/gstvideomark.c:
* sys/oss4/oss4-mixer.c:
* sys/oss4/oss4-sink.c:
* sys/oss4/oss4-source.c:
Do not use short_description in section docs for elements. We extract
them from element details and there will be warnings if they differ.
Also fixing up the ChangeLog order.

16 years agoconfigure.ac: 0.10.19.3 pre-release
Jan Schmidt [Wed, 11 Jun 2008 21:17:01 +0000 (21:17 +0000)]
configure.ac: 0.10.19.3 pre-release

Original commit message from CVS:
* configure.ac:
0.10.19.3 pre-release

16 years agogst-libs/gst/rtsp/gstrtspconnection.c: Fix build on win32.
David Schleef [Wed, 11 Jun 2008 20:13:00 +0000 (20:13 +0000)]
gst-libs/gst/rtsp/gstrtspconnection.c: Fix build on win32.

Original commit message from CVS:
* gst-libs/gst/rtsp/gstrtspconnection.c:
Fix build on win32.
Patch By: David Schleef <ds@schleef.org>
Fixes: #536874

16 years agoext/gio/gstgiobasesrc.*: Try to read the requested number of bytes, even if the first...
Sebastian Dröge [Wed, 11 Jun 2008 09:35:51 +0000 (09:35 +0000)]
ext/gio/gstgiobasesrc.*: Try to read the requested number of bytes, even if the first read returns less than requeste...

Original commit message from CVS:
* ext/gio/gstgiobasesrc.c: (gst_gio_base_src_finalize),
(gst_gio_base_src_create):
* ext/gio/gstgiobasesrc.h:
Try to read the requested number of bytes, even if the first
read returns less than requested, until nothing is read anymore
or we have the requested amount of bytes. This fixes playback of
files via Samba as Samba only allows to read 64k at once.
Implement a caching algorithm that makes sure that we read at
least 4k of data every time. Some elements will try to read a few
bytes, then seek, read again a few bytes and so on and this is
painfully slow as every operation has to go over DBus if GVfs is
used as backend.
Fixes bug #536849 and #536848.
* ext/gio/gstgiosrc.c: (gst_gio_src_class_init),
(gst_gio_src_check_get_range):
Override check_get_range() to blacklist http/https URIs
and whitelist file URIs. More to be added on demand.

16 years agoexamples/app/: Added 3 more example application for using appsrc in random-access...
Wim Taymans [Fri, 6 Jun 2008 16:50:51 +0000 (16:50 +0000)]
examples/app/: Added 3 more example application for using appsrc in random-access mode, pull-mode streaming and pull ...

Original commit message from CVS:
* examples/app/Makefile.am:
* examples/app/appsrc-ra.c: (feed_data), (seek_data),
(found_source), (bus_message), (main):
* examples/app/appsrc-seekable.c: (feed_data), (seek_data),
(found_source), (bus_message), (main):
* examples/app/appsrc-stream2.c: (feed_data), (found_source),
(bus_message), (main):
Added 3 more example application for using appsrc in random-access mode,
pull-mode streaming and pull mode seekable.
* gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init),
(gst_app_src_start), (gst_app_src_do_get_size),
(gst_app_src_create):
* gst-libs/gst/app/gstappsrc.h:
Make stream-type property writable.
Unset flushing when starting so that we reuse appsrc.
Inform basesrc about the configured size.
Emit seek-data signal when we are going to a different offset in
random-access mode.

16 years agoexamples/app/appsrc-stream.c: Use deep-notify until we can depend on a playbin2 with...
Wim Taymans [Fri, 6 Jun 2008 14:19:54 +0000 (14:19 +0000)]
examples/app/appsrc-stream.c: Use deep-notify until we can depend on a playbin2 with support for the source property.

Original commit message from CVS:
* examples/app/appsrc-stream.c: (found_source), (main):
Use deep-notify until we can depend on a playbin2 with support for the
source property.

16 years agoexamples/app/: Added an example on how to use appsrc in playbin in streaming mode...
Wim Taymans [Thu, 5 Jun 2008 16:38:50 +0000 (16:38 +0000)]
examples/app/: Added an example on how to use appsrc in playbin in streaming mode from an mmapped file.

Original commit message from CVS:
* examples/app/.cvsignore:
* examples/app/Makefile.am:
* examples/app/appsrc-stream.c: (read_data), (start_feed),
(stop_feed), (found_source), (bus_message), (main):
Added an example on how to use appsrc in playbin in streaming mode from
an mmapped file.
* examples/app/appsrc_ex.c: (main):
Set pipeline to NULL to free queued buffers.
* gst-libs/gst/app/gstapp-marshal.list:
* gst-libs/gst/app/gstappsrc.c: (stream_type_get_type), (_do_init),
(gst_app_src_class_init), (gst_app_src_init),
(gst_app_src_flush_queued), (gst_app_src_dispose),
(gst_app_src_set_property), (gst_app_src_get_property),
(gst_app_src_unlock), (gst_app_src_unlock_stop),
(gst_app_src_start), (gst_app_src_stop), (gst_app_src_is_seekable),
(gst_app_src_check_get_range), (gst_app_src_do_seek),
(gst_app_src_create), (gst_app_src_set_stream_type),
(gst_app_src_get_stream_type), (gst_app_src_set_max_bytes),
(gst_app_src_get_max_bytes), (gst_app_src_push_buffer),
(gst_app_src_end_of_stream), (gst_app_src_uri_get_type),
(gst_app_src_uri_get_protocols), (gst_app_src_uri_get_uri),
(gst_app_src_uri_set_uri), (gst_app_src_uri_handler_init):
* gst-libs/gst/app/gstappsrc.h:
Measure max queue size in bytes instead.
Add support for 3 modes of operation, streaming, seekable and
random-access, making basesrc handle the scheduling modes for each.
Add appsrc:// uri handler so that automatic plugging can be done from
playbin2 or uridecodebin, for example.
Added support for custom segment formats.
Add support for push and pull based operations from the application.
Expand the methods so that errors can be detected.
Flush the queued buffers on seeks and when shutting down.
Add signals to inform the app that a seek must happen.

16 years agoconfigure.ac: 0.10.19.2 pre-release
Jan Schmidt [Thu, 5 Jun 2008 09:47:23 +0000 (09:47 +0000)]
configure.ac: 0.10.19.2 pre-release

Original commit message from CVS:
* configure.ac:
0.10.19.2 pre-release

16 years agowin32/common/: Add new API functions to the dll exports
Jan Schmidt [Wed, 4 Jun 2008 21:48:27 +0000 (21:48 +0000)]
win32/common/: Add new API functions to the dll exports

Original commit message from CVS:
* win32/common/libgstrtsp.def:
* win32/common/libgsttag.def:
Add new API functions to the dll exports

16 years agogst/playback/gstplaybasebin.c: Disconnect signals from decodebins we created before...
Michael Smith [Wed, 4 Jun 2008 17:42:38 +0000 (17:42 +0000)]
gst/playback/gstplaybasebin.c: Disconnect signals from decodebins we created before we remove it from playbin, to avo...

Original commit message from CVS:
* gst/playback/gstplaybasebin.c:
Disconnect signals from decodebins we created before we remove it from
playbin, to avoid crashes if the decodebin is eventually disposed after
the playbin itself (possible if the app takes a reference on the
decodebin).
Fixes #536521.

16 years agogst/typefind/gsttypefindfunctions.c: Bunch of small clean-ups: use gst_type_find_sugg...
Tim-Philipp Müller [Wed, 4 Jun 2008 17:12:40 +0000 (17:12 +0000)]
gst/typefind/gsttypefindfunctions.c: Bunch of small clean-ups: use gst_type_find_suggest_simple(); don't copy caps fo...

Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (aac_type_find),
(mp3_type_find), (musepack_type_find), (MULTIPART_MAX_HEADER_SIZE),
(mpeg_sys_type_find), (mpeg_ts_type_find), (mpeg4_video_type_find),
(h264_video_type_find), (mpeg_video_stream_type_find),
(dv_type_find), (mmsh_type_find):
Bunch of small clean-ups: use gst_type_find_suggest_simple(); don't
copy caps for no good reason (this may be desirable to make it easier
to detect leaks, but then it should probably be done for all caps
in the typefinder somewhere).

16 years agotests/check/Makefile.am: Do not try to run the check tests for subparse unless it...
Peter Kjellerstedt [Wed, 4 Jun 2008 16:06:49 +0000 (16:06 +0000)]
tests/check/Makefile.am: Do not try to run the check tests for subparse unless it has been built.

Original commit message from CVS:
* tests/check/Makefile.am:
Do not try to run the check tests for subparse unless it has been
built.

16 years agotests/check/pipelines/streamheader.c: Do not try to run a test which requires vorbise...
Peter Kjellerstedt [Wed, 4 Jun 2008 16:00:26 +0000 (16:00 +0000)]
tests/check/pipelines/streamheader.c: Do not try to run a test which requires vorbisenc unless we have actually built...

Original commit message from CVS:
* tests/check/pipelines/streamheader.c: (buffer_probe_cb),
(test_multifdsink_gdp_vorbisenc), (streamheader_suite):
Do not try to run a test which requires vorbisenc unless we have
actually built it.

16 years agogst-libs/gst/rtsp/gstrtspconnection.*: Add a couple of missing argument guards.
Peter Kjellerstedt [Wed, 4 Jun 2008 11:53:53 +0000 (11:53 +0000)]
gst-libs/gst/rtsp/gstrtspconnection.*: Add a couple of missing argument guards.

Original commit message from CVS:
* gst-libs/gst/rtsp/gstrtspconnection.c:
(gst_rtsp_connection_set_auth), (gst_rtsp_connection_set_auth_param),
(gst_rtsp_connection_clear_auth_params),
(gst_rtsp_connection_set_qos_dscp), (gst_rtsp_connection_get_ip):
* gst-libs/gst/rtsp/gstrtspconnection.h:
Add a couple of missing argument guards.
Add a way of setting the DSCP for an RTSP connection.
Add an accessor method for the ip member of GstRTSPConnection as all
members are supposed to be private.

16 years agogst/tcp/gstmultifdsink.c: Fixed accidental use of IPv4 options for all IPv6 addresses.
Peter Kjellerstedt [Wed, 4 Jun 2008 11:33:23 +0000 (11:33 +0000)]
gst/tcp/gstmultifdsink.c: Fixed accidental use of IPv4 options for all IPv6 addresses.

Original commit message from CVS:
* gst/tcp/gstmultifdsink.c: (setup_dscp_client):
Fixed accidental use of IPv4 options for all IPv6 addresses.

16 years agogst-libs/gst/interfaces/mixertrack.h: Document mixer track flags.
Tim-Philipp Müller [Wed, 4 Jun 2008 10:18:42 +0000 (10:18 +0000)]
gst-libs/gst/interfaces/mixertrack.h: Document mixer track flags.

Original commit message from CVS:
* gst-libs/gst/interfaces/mixertrack.h:
Document mixer track flags.

16 years agogst/gdp/gstgdppay.c: Don't set caps on the buffers that contain a copy of the buffer...
Antoine Tremblay [Wed, 4 Jun 2008 05:58:38 +0000 (05:58 +0000)]
gst/gdp/gstgdppay.c: Don't set caps on the buffers that contain a copy of the buffer including the caps of them resul...

Original commit message from CVS:
Patch by: Antoine Tremblay <hexa00 at gmail dot com>
* gst/gdp/gstgdppay.c: (gst_gdp_pay_reset_streamheader):
Don't set caps on the buffers that contain a copy of the buffer
including the caps of them resulting in an always increasing refcount
of the caps and insanely large caps. Instead include a buffer without
caps in the new caps. Fixes bug #536475.

16 years agogst/videoscale/gstvideoscale.c: Transform a given PAR to a range on the struct with...
Sebastian Dröge [Wed, 4 Jun 2008 05:44:06 +0000 (05:44 +0000)]
gst/videoscale/gstvideoscale.c: Transform a given PAR to a range on the struct with the generic height/width instead ...

Original commit message from CVS:
* gst/videoscale/gstvideoscale.c: (gst_video_scale_transform_caps):
Transform a given PAR to a range on the struct with the generic
height/width instead of the struct with the possibly restricted
height/width.

16 years agogst/videoscale/gstvideoscale.c: Prefer the given format if it contains something...
Sebastian Dröge [Wed, 4 Jun 2008 04:24:27 +0000 (04:24 +0000)]
gst/videoscale/gstvideoscale.c: Prefer the given format if it contains something stricter than [1,MAX] for height or ...

Original commit message from CVS:
* gst/videoscale/gstvideoscale.c: (gst_video_scale_transform_caps):
Prefer the given format if it contains something stricter than [1,MAX]
for height or width and only put a structure that requires rescaling
as second. This makes it possible to use videoscale in pipelines where
the source can actually produce the wanted height/width but usually
selects a different one from the requested.

16 years agogst-libs/gst/tag/gstvorbistag.c: Retrieve COVERART tags from vorbis comments (#512333)
John Millikin [Tue, 3 Jun 2008 20:01:58 +0000 (20:01 +0000)]
gst-libs/gst/tag/gstvorbistag.c: Retrieve COVERART tags from vorbis comments (#512333)

Original commit message from CVS:
Based on patch by: John Millikin <jmillikin gmail com>
* gst-libs/gst/tag/gstvorbistag.c: (tag_matches), (gst_vorbis_tag_add),
(gst_vorbis_tag_add_coverart):
Retrieve COVERART tags from vorbis comments (#512333)

16 years agogst-libs/gst/tag/: Don't forget to add new enum value here too (should probably use...
Tim-Philipp Müller [Tue, 3 Jun 2008 19:44:48 +0000 (19:44 +0000)]
gst-libs/gst/tag/: Don't forget to add new enum value here too (should probably use glib-mkenums here...).

Original commit message from CVS:
* gst-libs/gst/tag/tag.h:
* gst-libs/gst/tag/tags.c: (register_tag_image_type_enum):
Don't forget to add new enum value here too (should probably use
glib-mkenums here...).

16 years agogst-libs/gst/tag/: API: add gst_tag_image_data_to_image_buffer()
Tim-Philipp Müller [Tue, 3 Jun 2008 19:29:06 +0000 (19:29 +0000)]
gst-libs/gst/tag/: API: add gst_tag_image_data_to_image_buffer()

Original commit message from CVS:
* gst-libs/gst/tag/gstid3tag.c: (gst_tag_list_add_id3_image):
* gst-libs/gst/tag/tag.h: (GST_TAG_IMAGE_TYPE_NONE),
* gst-libs/gst/tag/tags.c: (register_tag_image_type_enum),
(gst_tag_image_type_get_type), (gst_tag_image_type_is_valid),
(gst_tag_image_data_to_image_buffer):
Add two utility functions to avoid code duplication (#512333):
API: add gst_tag_image_data_to_image_buffer()
API: add gst_tag_list_add_id3_image()

16 years agowin32/common/libgstaudio.def: Add gst_audio_check_channel_positions() to the exported...
Sebastian Dröge [Tue, 3 Jun 2008 08:54:29 +0000 (08:54 +0000)]
win32/common/libgstaudio.def: Add gst_audio_check_channel_positions() to the exported symbols.

Original commit message from CVS:
* win32/common/libgstaudio.def:
Add gst_audio_check_channel_positions() to the exported symbols.

16 years agoAPI: Make gst_audio_check_channel_positions() public.
Sebastian Dröge [Tue, 3 Jun 2008 08:48:32 +0000 (08:48 +0000)]
API: Make gst_audio_check_channel_positions() public.

Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/audio/multichannel.c:
(gst_audio_check_channel_positions):
* gst-libs/gst/audio/multichannel.h:
API: Make gst_audio_check_channel_positions() public.
* tests/check/libs/audio.c: (GST_START_TEST):
Add some simple checks for gst_audio_check_channel_positions().

16 years agosys/v4l/v4l_calls.c: minrange and maxrange are scaled according to the frequency...
Tim-Philipp Müller [Mon, 2 Jun 2008 20:09:14 +0000 (20:09 +0000)]
sys/v4l/v4l_calls.c: minrange and maxrange are scaled according to the frequency multiplier.

Original commit message from CVS:
* sys/v4l/v4l_calls.c: (gst_v4l_get_chan_names):
minrange and maxrange are scaled according to the frequency
multiplier.

16 years agoext/pango/: Use gstvideo functions to calculate strides and plane offsets. Fixes...
Tim-Philipp Müller [Mon, 2 Jun 2008 18:37:02 +0000 (18:37 +0000)]
ext/pango/: Use gstvideo functions to calculate strides and plane offsets. Fixes rendering issue ('ghost' images of t...

Original commit message from CVS:
* ext/pango/Makefile.am:
* ext/pango/gsttextoverlay.c: (gst_text_overlay_shade_y),
(gst_text_overlay_blit_yuv420), (gst_text_overlay_push_frame):
Use gstvideo functions to calculate strides and plane offsets. Fixes
rendering issue ('ghost' images of the text on the chroma planes)
with widths or heights that are not multiples of 8 (#506659 and
probably also #485729).
* tests/icles/test-textoverlay.c: (show_text), (test_textoverlay),
(main):
Test with odd height/width too.

16 years agogst/adder/gstadder.c: When using gst_element_iterate_pads() one has to unref every...
Sebastian Dröge [Mon, 2 Jun 2008 12:20:35 +0000 (12:20 +0000)]
gst/adder/gstadder.c: When using gst_element_iterate_pads() one has to unref every pad after usage.

Original commit message from CVS:
* gst/adder/gstadder.c: (gst_adder_query_duration),
(gst_adder_query_latency):
When using gst_element_iterate_pads() one has to unref every pad
after usage.

16 years agogst-libs/gst/audio/gstbaseaudiosrc.c: Add a gtk-doc chunk for the new properties...
Mark Nauwelaerts [Sat, 31 May 2008 19:57:57 +0000 (19:57 +0000)]
gst-libs/gst/audio/gstbaseaudiosrc.c: Add a gtk-doc chunk for the new properties to have a Since: indication.

Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_class_init):
Add a gtk-doc chunk for the new properties to have a Since: indication.

16 years agoChangeLog surgery, mark API change
Mark Nauwelaerts [Sat, 31 May 2008 19:50:59 +0000 (19:50 +0000)]
ChangeLog surgery, mark API change

Original commit message from CVS:
ChangeLog surgery, mark API change

16 years agogst-libs/gst/audio/gstbaseaudiosrc.c: Provide readable actual-buffer-time and actual...
Mark Nauwelaerts [Sat, 31 May 2008 18:10:47 +0000 (18:10 +0000)]
gst-libs/gst/audio/gstbaseaudiosrc.c: Provide readable actual-buffer-time and actual-latency-time properties that ref...

Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_class_init), (gst_base_audio_src_dispose),
(gst_base_audio_src_get_property), (gst_base_audio_src_setcaps),
(gst_base_audio_src_change_state):
Provide readable actual-buffer-time and actual-latency-time properties
that reflect the configured ringbuffer values. Fixes #524724.

16 years agogst-libs/gst/rtp/gstbasertppayload.c: Simply converting the running time into an...
Wim Taymans [Fri, 30 May 2008 15:29:20 +0000 (15:29 +0000)]
gst-libs/gst/rtp/gstbasertppayload.c: Simply converting the running time into an RTP timestamp by scaling it based on...

Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push),
(gst_basertppayload_change_state):
Simply converting the running time into an RTP timestamp by scaling it
based on the clock-rate is good enough for making an RTP timestamp. This
has the added benefit that we can later on expose a property with the
RTP timestamp of running time 0, as is needed for RTSP servers to
generate the response of the PLAY request.

16 years agogst/audioconvert/gstaudioconvert.c: Allow up to 11 positioned channels now that audio...
Sebastian Dröge [Fri, 30 May 2008 08:42:17 +0000 (08:42 +0000)]
gst/audioconvert/gstaudioconvert.c: Allow up to 11 positioned channels now that audioconvert can handle this but add ...

Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
(structure_has_fixed_channel_positions),
(gst_audio_convert_transform_caps):
Allow up to 11 positioned channels now that audioconvert can handle
this but add no default positions for > 8 channels.
* tests/check/elements/audioconvert.c: (GST_START_TEST):
Add some unit tests for the above change: Test conversion of
11 positioned channels to stereo and the other way around, test
conversion of 15 unpositioned channels in different ways.

16 years agowin32/common/libgstaudio.def: Add gst_audio_clock_reset to the list of exported symbols.
Sebastian Dröge [Thu, 29 May 2008 19:45:40 +0000 (19:45 +0000)]
win32/common/libgstaudio.def: Add gst_audio_clock_reset to the list of exported symbols.

Original commit message from CVS:
* win32/common/libgstaudio.def:
Add gst_audio_clock_reset to the list of exported symbols.

16 years agotests/check/elements/vorbisdec.c: Remove wrong_channels_identification_header unit...
Sebastian Dröge [Thu, 29 May 2008 19:37:47 +0000 (19:37 +0000)]
tests/check/elements/vorbisdec.c: Remove wrong_channels_identification_header unit test as we now support 7 (and more...

Original commit message from CVS:
* tests/check/elements/vorbisdec.c: (vorbisdec_suite):
Remove wrong_channels_identification_header unit test as we now
support 7 (and more channels).

16 years agogst/audioconvert/gstchannelmix.c: If mixing left or right to center (or the other...
Sebastian Dröge [Thu, 29 May 2008 12:17:16 +0000 (12:17 +0000)]
gst/audioconvert/gstchannelmix.c: If mixing left or right to center (or the other way around) only take the complete ...

Original commit message from CVS:
* gst/audioconvert/gstchannelmix.c:
(gst_channel_mix_fill_one_other):
If mixing left or right to center (or the other way around) only take
the complete value if we don't already have the original position in
the source.

16 years agogst-libs/gst/audio/multichannel.c: Allow rear center together with rear left/right...
Sebastian Dröge [Thu, 29 May 2008 11:34:09 +0000 (11:34 +0000)]
gst-libs/gst/audio/multichannel.c: Allow rear center together with rear left/right and other previously conflicting c...

Original commit message from CVS:
* gst-libs/gst/audio/multichannel.c:
(gst_audio_check_channel_positions),
(gst_audio_set_structure_channel_positions_list),
(gst_audio_fixate_channel_positions):
Allow rear center together with rear left/right and other previously
conflicting channel positions. The reason why they weren't allowed
was the channel mixing implementation in audioconvert.
Also take this into account when fixing channel layouts.
Allow setting channel positions for 1/2 channels when using
gst_audio_set_structure_channel_position().
* gst/audioconvert/gstchannelmix.c:
(gst_channel_mix_fill_compatible), (gst_channel_mix_detect_pos),
(gst_channel_mix_fill_one_other), (gst_channel_mix_fill_others),
(gst_channel_mix_fill_special), (gst_channel_mix_fill_matrix):
Major rewrite of the channel mixing.
We now allow previously conflicting channel positions to appear
together (rear center and rear left/right for example).
Fixes bug #533817.
Rework the way channels are mixed together to take more possible
channel positions into account, properly mix from/to side channels
and don't assume that either center, left&right or nothing of a
specific position is available anymore.
* tests/check/elements/audioconvert.c: (GST_START_TEST):
Adjust unit tests with non-standard 1/2 channel layouts to the more
correct new behaviour.
Add a unit test for 5.1->Stereo downmixing.

16 years agoext/vorbis/: Add sane defaults for the 7 and 8 channel layouts as those are undefined...
Sebastian Dröge [Thu, 29 May 2008 07:02:50 +0000 (07:02 +0000)]
ext/vorbis/: Add sane defaults for the 7 and 8 channel layouts as those are undefined in the Vorbis spec. Use NONE ch...

Original commit message from CVS:
* ext/vorbis/vorbisdec.c: (vorbis_handle_identification_packet):
* ext/vorbis/vorbisenc.c: (gst_vorbis_enc_generate_sink_caps):
Add sane defaults for the 7 and 8 channel layouts as those are
undefined in the Vorbis spec. Use NONE channel layouts when decoding
more than 8 channels instead of erroring out. Fixes bug #535356.

16 years agoAdd theoraparse to the docs and fix some docs.
Wim Taymans [Wed, 28 May 2008 16:10:20 +0000 (16:10 +0000)]
Add theoraparse to the docs and fix some docs.

Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
* docs/plugins/gst-plugins-base-plugins-sections.txt:
* ext/theora/theoraparse.c:
Add theoraparse to the docs and fix some docs.

16 years agogst-libs/gst/cdda/gstcddabasesrc.c: Fix EOS condition and track addition check, the...
Wim Taymans [Wed, 28 May 2008 15:48:33 +0000 (15:48 +0000)]
gst-libs/gst/cdda/gstcddabasesrc.c: Fix EOS condition and track addition check, the track.end sector is included in t...

Original commit message from CVS:
* gst-libs/gst/cdda/gstcddabasesrc.c:
(gst_cdda_base_src_add_track), (gst_cdda_base_src_create):
Fix EOS condition and track addition check, the track.end sector is
included in the track. Fixes #533265.

16 years agogst/videorate/gstvideorate.*: React (more) to NEWSEGMENT
Mark Nauwelaerts [Wed, 28 May 2008 14:49:24 +0000 (14:49 +0000)]
gst/videorate/gstvideorate.*: React (more) to NEWSEGMENT

Original commit message from CVS:
Patch by: Mark Nauwelaerts <manauw at skynet be>
* gst/videorate/gstvideorate.c: (gst_video_rate_reset),
(gst_video_rate_flush_prev), (gst_video_rate_event),
(gst_video_rate_chain):
* gst/videorate/gstvideorate.h:
React (more) to NEWSEGMENT
Small adjustment in timestamp calculation to prevent mismatches
Fixes #435633.

16 years agotests/examples/seek/seek.c: Initialise error to NULL as we should.
Tim-Philipp Müller [Wed, 28 May 2008 11:31:44 +0000 (11:31 +0000)]
tests/examples/seek/seek.c: Initialise error to NULL as we should.

Original commit message from CVS:
* tests/examples/seek/seek.c: (make_parselaunch_pipeline):
Initialise error to NULL as we should.

16 years agogst/adder/gstadder.c: Implement latency query.
Sebastian Dröge [Wed, 28 May 2008 08:14:47 +0000 (08:14 +0000)]
gst/adder/gstadder.c: Implement latency query.

Original commit message from CVS:
* gst/adder/gstadder.c: (gst_adder_query_duration),
(gst_adder_query_latency), (gst_adder_query):
Implement latency query.

16 years agogst/adder/gstadder.c: Correctly resync the iterator if gst_iterator_next() returns
Sebastian Dröge [Tue, 27 May 2008 18:10:00 +0000 (18:10 +0000)]
gst/adder/gstadder.c: Correctly resync the iterator if gst_iterator_next() returns

Original commit message from CVS:
* gst/adder/gstadder.c: (gst_adder_query_duration):
Correctly resync the iterator if gst_iterator_next() returns
GST_ITERATOR_RESYNC.

16 years agowin32/vs6/libgstpbutils.dsp: Add pbutils-enumtypes.c to sources (#518037).
Tim-Philipp Müller [Tue, 27 May 2008 17:14:07 +0000 (17:14 +0000)]
win32/vs6/libgstpbutils.dsp: Add pbutils-enumtypes.c to sources (#518037).

Original commit message from CVS:
* win32/vs6/libgstpbutils.dsp:
Add pbutils-enumtypes.c to sources (#518037).

16 years agogst-libs/gst/audio/gstaudioclock.*: Add method to inform the clock that the time...
Wim Taymans [Tue, 27 May 2008 16:20:17 +0000 (16:20 +0000)]
gst-libs/gst/audio/gstaudioclock.*: Add method to inform the clock that the time starts from 0 again. We use this inf...

Original commit message from CVS:
* gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_init),
(gst_audio_clock_reset), (gst_audio_clock_get_internal_time):
* gst-libs/gst/audio/gstaudioclock.h:
Add method to inform the clock that the time starts from 0 again. We use
this info to calculate a clock offset so that the time we report in
internal_time is monotonically increasing, as required by the clock base
class. Fixes #521761.
API: GstAudioClock::gst_audio_clock_reset()
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_skew_slaving),
(gst_base_audio_sink_change_state):
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_create), (gst_base_audio_src_change_state):
Reset reported time when we (re)create the ringbuffer.

16 years agoext/alsa/gstalsamixertrack.c: Make sure playback volumes aren't accidentally overwrit...
Tim-Philipp Müller [Tue, 27 May 2008 16:11:32 +0000 (16:11 +0000)]
ext/alsa/gstalsamixertrack.c: Make sure playback volumes aren't accidentally overwritten by capture volumes if an als...

Original commit message from CVS:
* ext/alsa/gstalsamixertrack.c:
(gst_alsa_mixer_track_update_alsa_capabilities):
Make sure playback volumes aren't accidentally overwritten by
capture volumes if an alsa mixer track has both playback and
capture capabilities: we create two GstMixerTracks in that
case, so make sure we query only the alsa capabilities that
refer to the type of GstMixerTrack we created from the dual
capability alsa element. Should fix issues with Audigy2 sound
cards (#518082).

16 years agotests/check/pipelines/oggmux.c: Don't use deprecated function.
Tim-Philipp Müller [Tue, 27 May 2008 10:57:56 +0000 (10:57 +0000)]
tests/check/pipelines/oggmux.c: Don't use deprecated function.

Original commit message from CVS:
* tests/check/pipelines/oggmux.c: (test_pipeline):
Don't use deprecated function.

16 years agogst/playback/gstdecodebin2.c: Check for NULL cases and log them, creating ghostpads...
Wim Taymans [Tue, 27 May 2008 10:35:55 +0000 (10:35 +0000)]
gst/playback/gstdecodebin2.c: Check for NULL cases and log them, creating ghostpads can, for example, fail when the p...

Original commit message from CVS:
* gst/playback/gstdecodebin2.c:
(gst_decode_group_control_source_pad), (gst_decode_group_expose):
Check for NULL cases and log them, creating ghostpads can, for example,
fail when the pad returns wrong caps.
* gst/playback/gstplaybin2.c: (perform_eos):
When pushing out the EOS event, collect the return value and warn when
something failed.

16 years agogst-libs/gst/riff/riff-media.c: Add support for DVCPRO.
Wim Taymans [Mon, 26 May 2008 17:18:52 +0000 (17:18 +0000)]
gst-libs/gst/riff/riff-media.c: Add support for DVCPRO.

Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
(gst_riff_create_video_template_caps):
Add support for DVCPRO.

16 years agogst/videoscale/gstvideoscale.c: Change default scaling method from nearest-neighbour...
Tim-Philipp Müller [Mon, 26 May 2008 10:29:20 +0000 (10:29 +0000)]
gst/videoscale/gstvideoscale.c: Change default scaling method from nearest-neighbour to bilinear.

Original commit message from CVS:
* gst/videoscale/gstvideoscale.c: (DEFAULT_PROP_METHOD):
Change default scaling method from nearest-neighbour to bilinear.

16 years agotests/check/libs/video.c: More checks.
Tim-Philipp Müller [Mon, 26 May 2008 10:26:00 +0000 (10:26 +0000)]
tests/check/libs/video.c: More checks.

Original commit message from CVS:
* tests/check/libs/video.c:
More checks.

16 years agoLimit duration to a maximum of five seconds for tmplayer format where we can guess...
Tim-Philipp Müller [Sun, 25 May 2008 20:51:35 +0000 (20:51 +0000)]
Limit duration to a maximum of five seconds for tmplayer format where we can guess the duration only from the timesta...

Original commit message from CVS:
* gst/subparse/gstsubparse.c: (parser_state_init),
(gst_sub_parse_format_autodetect), (handle_buffer):
* gst/subparse/gstsubparse.h:
* tests/check/elements/subparse.c: (test_tmplayer_style3b):
Limit duration to a maximum of five seconds for tmplayer format where
we can guess the duration only from the timestamp of the next line of
text. We don't want to show a text for eternities just because nothing
else is being said for a while.

16 years agogst-libs/gst/rtp/gstbasertpdepayload.c: Check sequence numbers, mark input buffers...
Wim Taymans [Fri, 23 May 2008 14:14:28 +0000 (14:14 +0000)]
gst-libs/gst/rtp/gstbasertpdepayload.c: Check sequence numbers, mark input buffers with a discont flag for the subcla...

Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_chain),
(gst_base_rtp_depayload_handle_sink_event),
(gst_base_rtp_depayload_push_full),
(gst_base_rtp_depayload_change_state):
Check sequence numbers, mark input buffers with a discont flag for the
subclass when we detected a gap, drop duplicate buffers. We do this
because one can use the element without a jitterbuffer in front and we
don't want to feed the subclasses invalid or reordered data.
Do an error when the subclass did not provide a process function instead
of crashing.
Some other small cleanups.

16 years agogst/videotestsrc/videotestsrc.c: May just as well use the precalculated uvstride...
Tim-Philipp Müller [Thu, 22 May 2008 22:35:40 +0000 (22:35 +0000)]
gst/videotestsrc/videotestsrc.c: May just as well use the precalculated uvstride here.

Original commit message from CVS:
* gst/videotestsrc/videotestsrc.c: (paint_hline_NV12_NV21):
May just as well use the precalculated uvstride here.

16 years agoAdd some documentation comments, and some new headers to be scanned.
Jan Schmidt [Thu, 22 May 2008 22:09:16 +0000 (22:09 +0000)]
Add some documentation comments, and some new headers to be scanned.

Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-base-plugins-overrides.txt:
* docs/plugins/gst-plugins-base-plugins-sections.txt:
* docs/plugins/gst-plugins-base-plugins.args:
* docs/plugins/gst-plugins-base-plugins.hierarchy:
* docs/plugins/gst-plugins-base-plugins.interfaces:
* docs/plugins/gst-plugins-base-plugins.prerequisites:
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-decodebin.xml:
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
* docs/plugins/inspect/plugin-gdp.xml:
* docs/plugins/inspect/plugin-gio.xml:
* docs/plugins/inspect/plugin-gnomevfs.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playback.xml:
* docs/plugins/inspect/plugin-queue2.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-uridecodebin.xml:
* docs/plugins/inspect/plugin-video4linux.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
* ext/cdparanoia/gstcdparanoiasrc.c:
* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggdemux.h:
* ext/ogg/gstoggmux.c:
* ext/ogg/gstoggmux.h:
* gst/audioconvert/audioconvert.c:
* gst/audioconvert/audioconvert.h:
* gst/audioconvert/gstaudioconvert.h:
* gst/gdp/gstgdpdepay.h:
* gst/gdp/gstgdppay.h:
* gst/playback/gstdecodebin.c:
* gst/playback/gstdecodebin2.c:
* gst/playback/gstplaybin.c:
* gst/playback/gstplaybin2.c:
* gst/playback/gsturidecodebin.c:
* gst/tcp/gstmultifdsink.c:
* gst/tcp/gstmultifdsink.h:
* gst/tcp/gsttcp.h:
Add some documentation comments, and some new headers to be scanned.
Rename some internal enum declarations (audioconvert's DitherType and
NoiseShapingType, GstUnitType from the TCP elements) to match the
documented GObject type names so that the docs pick them up.
Name the playbin2 docs markups properly so they get picked up. They'll
need renaming back when/if playbin2 becomes playbin.
100% symbol coverage for the plugin docs, booya.

16 years agogst/videotestsrc/videotestsrc.c: Fix generation of NV12/NV21 frames. Fixes bug #532454.
Thijs Vermeir [Thu, 22 May 2008 18:30:15 +0000 (18:30 +0000)]
gst/videotestsrc/videotestsrc.c: Fix generation of NV12/NV21 frames. Fixes bug #532454.

Original commit message from CVS:
Patch by: Thijs Vermeir <thijsvermeir@gmail.com>
* gst/videotestsrc/videotestsrc.c: (paint_hline_NV12_NV21):
Fix generation of NV12/NV21 frames. Fixes bug #532454.

16 years agogst/playback/gstdecodebin.c: Lock the fakesink before setting the state to NULL and...
Sjoerd Simons [Thu, 22 May 2008 11:59:33 +0000 (11:59 +0000)]
gst/playback/gstdecodebin.c: Lock the fakesink before setting the state to NULL and removing it from the bin so that ...

Original commit message from CVS:
Patch by: Sjoerd Simons <sjoerd at luon dot net>
* gst/playback/gstdecodebin.c: (remove_fakesink):
Lock the fakesink before setting the state to NULL and removing it from
the bin so that a concurrent state change cannot interfere.
Fixes #534331.

16 years agodocs/Makefile.am: Fix installing plugin documentation when gtk-doc is disabled.
Felipe Contreras [Wed, 21 May 2008 17:09:42 +0000 (17:09 +0000)]
docs/Makefile.am: Fix installing plugin documentation when gtk-doc is disabled.

Original commit message from CVS:
* docs/Makefile.am:
Fix installing plugin documentation when gtk-doc is disabled.

16 years agogst-libs/gst/rtsp/Makefile.am: Distribute, don't install md5.h
Felipe Contreras [Wed, 21 May 2008 17:01:16 +0000 (17:01 +0000)]
gst-libs/gst/rtsp/Makefile.am: Distribute, don't install md5.h

Original commit message from CVS:
* gst-libs/gst/rtsp/Makefile.am:
Distribute, don't install md5.h

16 years agogst/tcp/gstmultifdsink.c: Use IPPROTO_IP instead of SOL_IP, works on more platforms.
Julien Moutte [Wed, 21 May 2008 16:47:58 +0000 (16:47 +0000)]
gst/tcp/gstmultifdsink.c: Use IPPROTO_IP instead of SOL_IP, works on more platforms.

Original commit message from CVS:
2008-05-21  Julien Moutte  <julien@fluendo.com>

* gst/tcp/gstmultifdsink.c: (setup_dscp_client): Use IPPROTO_IP
instead of SOL_IP, works on more platforms.
* gst/typefind/gsttypefindfunctions.c: (aac_type_find): Fix printf
arguments.

16 years agoSome debug and comment fixes.
Wim Taymans [Wed, 21 May 2008 16:44:15 +0000 (16:44 +0000)]
Some debug and comment fixes.

Original commit message from CVS:
* ext/vorbis/vorbisdec.c:
* gst/videoscale/gstvideoscale.c: (gst_video_scale_transform):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_show_frame):
Some debug and comment fixes.
* tests/examples/dynamic/addstream.c: (main):
Fix , to ;

16 years agoDon't use bad gst_element_get_pad().
Wim Taymans [Wed, 21 May 2008 16:36:50 +0000 (16:36 +0000)]
Don't use bad gst_element_get_pad().

Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_typefind):
* gst/playback/decodetest.c: (new_decoded_pad_cb):
* gst/playback/gstdecodebin.c: (gst_decode_bin_init),
(try_to_link_1), (elem_is_dynamic), (close_link), (type_found),
(cleanup_decodebin):
* gst/playback/gstdecodebin2.c: (gst_decode_bin_init),
(connect_element), (gst_decode_group_control_demuxer_pad):
* gst/playback/gstplaybasebin.c: (queue_remove_probe),
(queue_out_of_data), (gen_preroll_element), (preroll_unlinked),
(mute_group_type):
* gst/playback/gstplaybin.c: (gst_play_bin_vis_blocked),
(gst_play_bin_set_property), (handoff), (gen_video_element),
(gen_text_element), (gen_audio_element), (gen_vis_element),
(remove_sinks), (add_sink), (setup_sinks):
* gst/playback/gstplaybin2.c: (pad_added_cb), (no_more_pads_cb):
* gst/playback/gstplaysink.c: (gst_play_sink_get_video_sink),
(gst_play_sink_get_audio_sink), (gst_play_sink_vis_unblocked),
(gst_play_sink_vis_blocked), (gst_play_sink_set_vis_plugin),
(gst_play_sink_get_vis_plugin), (gst_play_sink_set_mute),
(gen_video_chain), (gen_text_chain), (gen_audio_chain),
(gen_vis_chain), (gst_play_sink_reconfigure),
(gst_play_sink_set_font_desc), (gst_play_sink_get_font_desc),
(gst_play_sink_request_pad):
* gst/playback/gsturidecodebin.c: (type_found), (setup_source):
* gst/playback/test.c: (gen_video_element), (gen_audio_element),
(cb_newpad):
* gst/playback/test6.c: (new_decoded_pad_cb):
* tests/check/elements/audioconvert.c: (GST_START_TEST):
* tests/check/elements/audiorate.c: (test_injector_chain),
(do_perfect_stream_test):
* tests/check/elements/ffmpegcolorspace.c: (GST_START_TEST):
* tests/check/elements/gdpdepay.c: (GST_START_TEST):
* tests/check/elements/gnomevfssink.c:
* tests/check/elements/textoverlay.c:
(notgst_check_setup_src_pad2), (notgst_check_teardown_src_pad2):
* tests/check/elements/videotestsrc.c: (GST_START_TEST):
* tests/check/libs/cddabasesrc.c: (GST_START_TEST):
* tests/check/pipelines/oggmux.c: (test_pipeline):
* tests/check/pipelines/streamheader.c: (GST_START_TEST):
* tests/check/pipelines/theoraenc.c: (GST_START_TEST):
* tests/check/pipelines/vorbisenc.c: (GST_START_TEST):
* tests/examples/seek/scrubby.c: (make_wav_pipeline):
* tests/examples/seek/seek.c: (make_mod_pipeline),
(make_dv_pipeline), (make_wav_pipeline), (make_flac_pipeline),
(make_sid_pipeline), (make_parse_pipeline), (make_vorbis_pipeline),
(make_theora_pipeline), (make_vorbis_theora_pipeline),
(make_avi_msmpeg4v3_mp3_pipeline), (make_mp3_pipeline),
(make_avi_pipeline), (make_mpeg_pipeline), (make_mpegnt_pipeline),
(update_fill), (msg_buffering):
Don't use bad gst_element_get_pad().

16 years agogst-libs/gst/riff/riff-media.c: Fix wrong method name in docs. Fix calculation of...
Stefan Kost [Wed, 21 May 2008 14:35:41 +0000 (14:35 +0000)]
gst-libs/gst/riff/riff-media.c: Fix wrong method name in docs. Fix calculation of strf fields for broken mulaw/alaw.

Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c:
Fix wrong method name in docs. Fix calculation of strf fields for
broken mulaw/alaw.
* gst-libs/gst/riff/riff-read.c:
Whitespace fix and removing double ';'.

16 years agodocs/design/part-playbin2.txt: Add some leftover doc.
Wim Taymans [Wed, 21 May 2008 11:52:30 +0000 (11:52 +0000)]
docs/design/part-playbin2.txt: Add some leftover doc.

Original commit message from CVS:
* docs/design/part-playbin2.txt:
Add some leftover doc.

16 years agogst/audioconvert/gstchannelmix.c: Fix copy & paste error in last commit.
Sebastian Dröge [Wed, 21 May 2008 11:36:37 +0000 (11:36 +0000)]
gst/audioconvert/gstchannelmix.c: Fix copy & paste error in last commit.

Original commit message from CVS:
* gst/audioconvert/gstchannelmix.c: (gst_channel_mix_fill_others):
Fix copy & paste error in last commit.

16 years agogst/audioconvert/gstchannelmix.c: Add support for mixing GST_AUDIO_CHANNEL_POSITION_S...
Sebastian Dröge [Wed, 21 May 2008 11:30:58 +0000 (11:30 +0000)]
gst/audioconvert/gstchannelmix.c: Add support for mixing GST_AUDIO_CHANNEL_POSITION_SIDE_* from/to other channel posi...

Original commit message from CVS:
* gst/audioconvert/gstchannelmix.c: (gst_channel_mix_fill_others):
Add support for mixing GST_AUDIO_CHANNEL_POSITION_SIDE_* from/to
other channel positions when source has SIDE channels and dest doesn't
or the other way around.

16 years agogst/tcp/gstmultifdsink.*: Add support for DSCP QOS. Fixes #469933.
Henrik Eriksson [Wed, 21 May 2008 11:29:25 +0000 (11:29 +0000)]
gst/tcp/gstmultifdsink.*: Add support for DSCP QOS. Fixes #469933.

Original commit message from CVS:
Patch by: Henrik Eriksson <henriken at axis dot com>
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init),
(gst_multi_fd_sink_init), (setup_dscp_client), (setup_dscp),
(gst_multi_fd_sink_add_full), (gst_multi_fd_sink_set_property),
(gst_multi_fd_sink_get_property):
* gst/tcp/gstmultifdsink.h:
Add support for DSCP QOS. Fixes #469933.

16 years agotests/check/elements/audioconvert.c: Add another test that checks if conversion betwe...
Sebastian Dröge [Wed, 21 May 2008 07:46:02 +0000 (07:46 +0000)]
tests/check/elements/audioconvert.c: Add another test that checks if conversion between standard 1 and 2 channel layo...

Original commit message from CVS:
* tests/check/elements/audioconvert.c: (GST_START_TEST):
Add another test that checks if conversion between standard 1 and 2
channel layouts with and without positions set is working.

16 years agogst-libs/gst/audio/multichannel.c: Allow non-standard 2 channel layouts.
Sebastian Dröge [Wed, 21 May 2008 07:39:56 +0000 (07:39 +0000)]
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.

16 years agogst/audioconvert/gstchannelmix.c: Prevent division by zero if the channel mix matrix...
Sebastian Dröge [Wed, 21 May 2008 07:28:04 +0000 (07:28 +0000)]
gst/audioconvert/gstchannelmix.c: Prevent division by zero if the channel mix matrix contains only zeroes.

Original commit message from CVS:
* gst/audioconvert/gstchannelmix.c:
(gst_channel_mix_fill_normalize):
Prevent division by zero if the channel mix matrix contains only
zeroes.

16 years agogst/gdp/gstgdppay.c: Close a buffer memory leak. Fixes bug #534071.
Antoine Tremblay [Wed, 21 May 2008 06:45:22 +0000 (06:45 +0000)]
gst/gdp/gstgdppay.c: Close a buffer memory leak. Fixes bug #534071.

Original commit message from CVS:
Patch by: Antoine Tremblay <hexa00 at gmail dot com>
* gst/gdp/gstgdppay.c: (gst_gdp_pay_chain):
Close a buffer memory leak. Fixes bug #534071.

16 years agogst-libs/gst/rtsp/gstrtsptransport.h: Make the GstRTSPTransport struct members public...
Sebastian Dröge [Wed, 21 May 2008 06:39:20 +0000 (06:39 +0000)]
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.

16 years agogst/adder/gstadder.c: Adder also doesn't support audio/x-raw-int with width!=depth...
Sebastian Dröge [Wed, 21 May 2008 05:48:05 +0000 (05:48 +0000)]
gst/adder/gstadder.c: Adder also doesn't support audio/x-raw-int with width!=depth so don't claim this on the pad tem...

Original commit message from CVS:
* gst/adder/gstadder.c:
Adder also doesn't support audio/x-raw-int with width!=depth so don't
claim this on the pad template caps.

16 years agogst-libs/gst/audio/gstbaseaudiosink.c: We can only use our optimal calibration if...
Wim Taymans [Tue, 20 May 2008 16:26:53 +0000 (16:26 +0000)]
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.

16 years agoconfigure.ac: Require core CVS for GstBaseSrc buffer caps setting magic.
Tim-Philipp Müller [Tue, 20 May 2008 14:35:42 +0000 (14:35 +0000)]
configure.ac: Require core CVS for GstBaseSrc buffer caps setting magic.

Original commit message from CVS:
* configure.ac:
Require core CVS for GstBaseSrc buffer caps setting magic.

16 years agogst/audioconvert/gstaudioconvert.c: Fix logic in last commit.
Sebastian Dröge [Tue, 20 May 2008 12:26:32 +0000 (12:26 +0000)]
gst/audioconvert/gstaudioconvert.c: Fix logic in last commit.

Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_fixate_channels):
Fix logic in last commit.

16 years agogst/audioconvert/gstaudioconvert.c: Passthrough the channel positions if the number...
Sebastian Dröge [Tue, 20 May 2008 12:15:34 +0000 (12:15 +0000)]
gst/audioconvert/gstaudioconvert.c: Passthrough the channel positions if the number of output channels is the same as...

Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_fixate_channels):
Passthrough the channel positions if the number of output channels is
the same as the number of input channels, the input had a channel
layout and downstream requests no special one. We did this already for
> 2 channels but now it's also done for 1 channel. Fixes bug #533617.

16 years agoext/gnomevfs/gstgnomevfssrc.*: Set the ICY caps on the srcpad from where they get...
Wim Taymans [Tue, 20 May 2008 11:13:27 +0000 (11:13 +0000)]
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.

16 years agogst-libs/gst/audio/gstbaseaudiosink.c: Change the way in which the ringbuffer is...
Wim Taymans [Tue, 20 May 2008 11:09:06 +0000 (11:09 +0000)]
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.

16 years agogst/audioconvert/gstaudioconvert.c: Correctly set the default channel positions when...
Sebastian Dröge [Tue, 20 May 2008 08:12:19 +0000 (08:12 +0000)]
gst/audioconvert/gstaudioconvert.c: Correctly set the default channel positions when converting to 8 channels.

Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_fixate_channels):
Correctly set the default channel positions when converting to 8
channels.

16 years agoconfigure.ac: Error out if we don't have the required version of core.
Tim-Philipp Müller [Mon, 19 May 2008 16:13:25 +0000 (16:13 +0000)]
configure.ac: Error out if we don't have the required version of core.

Original commit message from CVS:
* configure.ac:
Error out if we don't have the required version of core.

16 years agogst/typefind/gsttypefindfunctions.c: Use data scan helper in aac typefinder and stop...
Tim-Philipp Müller [Mon, 19 May 2008 15:59:40 +0000 (15:59 +0000)]
gst/typefind/gsttypefindfunctions.c: Use data scan helper in aac typefinder and stop scanning for headers when we've ...

Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (aac_type_find):
Use data scan helper in aac typefinder and stop scanning
for headers when we've found a type. Also fix potential invalid
memory access when calculating the frame length.

16 years agogst/typefind/gsttypefindfunctions.c: Don't modify scan context when we return FALSE...
Tim-Philipp Müller [Mon, 19 May 2008 14:09:08 +0000 (14:09 +0000)]
gst/typefind/gsttypefindfunctions.c: Don't modify scan context when we return FALSE in ensure_data, so it's possible ...

Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (data_scan_ctx_ensure_data),
(mpeg_sys_is_valid_pack):
Don't modify scan context when we return FALSE in ensure_data, so
it's possible to continue scanning, and we don't end up with a NULL
data pointer and a positive size, which might bite us the next time
we're called. Small constification.

16 years agogst/adder/gstadder.c: Adder doesn't support 24 bit samples so don't claim it supports...
Sebastian Dröge [Fri, 16 May 2008 21:12:02 +0000 (21:12 +0000)]
gst/adder/gstadder.c: Adder doesn't support 24 bit samples so don't claim it supports them in the pad template caps.

Original commit message from CVS:
* gst/adder/gstadder.c:
Adder doesn't support 24 bit samples so don't claim it supports them
in the pad template caps.

16 years agogst-libs/gst/rtp/gstbasertpdepayload.c: Validate the RTP packet before further proces...
Wim Taymans [Wed, 14 May 2008 20:28:02 +0000 (20:28 +0000)]
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.

16 years agogst/audioresample/gstaudioresample.c: Revert previous change which made basetransform...
Tim-Philipp Müller [Wed, 14 May 2008 13:57:41 +0000 (13:57 +0000)]
gst/audioresample/gstaudioresample.c: Revert previous change which made basetransform handle buffer_alloc and which b...

Original commit message from CVS:
* gst/audioresample/gstaudioresample.c: (gst_audioresample_init):
Revert previous change which made basetransform handle buffer_alloc
and which breaks things badly in the non-passthrough case since it
returned buffers with a different (ie. sometimes smaller) size than
the size requested.

16 years agogst-libs/gst/rtp/gstrtpbuffer.c: Fix seqnum compare function for bordercase values...
Bernard B [Wed, 14 May 2008 13:43:12 +0000 (13:43 +0000)]
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.

16 years agogst/adder/gstadder.c: Correctly declare the supported endianness on the pad templates...
Sebastian Dröge [Wed, 14 May 2008 10:58:52 +0000 (10:58 +0000)]
gst/adder/gstadder.c: Correctly declare the supported endianness on the pad templates and check for correct endiannes...

Original commit message from CVS:
* gst/adder/gstadder.c: (gst_adder_setcaps),
(gst_adder_class_init):
Correctly declare the supported endianness on the pad templates
and check for correct endianness in the set caps function. Adder
only supports native endianness.
Also use gst_element_class_set_details_simple().

16 years agosys/xvimage/xvimagesink.c: Better debug logging in port value handling. Merging separ...
Stefan Kost [Wed, 14 May 2008 09:12:10 +0000 (09:12 +0000)]
sys/xvimage/xvimagesink.c: Better debug logging in port value handling. Merging separate port value loops into one.

Original commit message from CVS:
* sys/xvimage/xvimagesink.c:
Better debug logging in port value handling. Merging separate port
value loops into one.

16 years agogst/tcp/: Fix regression in clientsrc because we did not add the fd to the poll set...
Hannes Bistry [Tue, 13 May 2008 16:02:19 +0000 (16:02 +0000)]
gst/tcp/: Fix regression in clientsrc because we did not add the fd to the poll set anymore. Fixes #532364.

Original commit message from CVS:
Patch by: Hannes Bistry <hannesb at gmx dot de>
* gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_start):
* gst/tcp/gsttcpserversink.c:
(gst_tcp_server_sink_handle_server_read),
(gst_tcp_server_sink_handle_wait), (gst_tcp_server_sink_init_send):
Fix regression in clientsrc because we did not add the fd to the poll
set anymore. Fixes #532364.
Do some cleanups here and there.

16 years agogst/playback/: Use correct marshallers. GstCaps are a boxed type and no GObject subclass.
Sebastian Dröge [Tue, 13 May 2008 13:04:24 +0000 (13:04 +0000)]
gst/playback/: Use correct marshallers. GstCaps are a boxed type and no GObject subclass.

Original commit message from CVS:
* gst/playback/gstdecodebin.c: (gst_decode_bin_class_init):
* gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init):
* gst/playback/gstplay-marshal.list:
* gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init):
Use correct marshallers. GstCaps are a boxed type and no GObject
subclass.

16 years agowin32/common/libgstrtsp.def: Add gst_rtsp_connection_(set|clear)_auth_param() to...
Sebastian Dröge [Tue, 13 May 2008 11:37:15 +0000 (11:37 +0000)]
win32/common/libgstrtsp.def: Add gst_rtsp_connection_(set|clear)_auth_param() to the exported symbols.

Original commit message from CVS:
* win32/common/libgstrtsp.def:
Add gst_rtsp_connection_(set|clear)_auth_param() to the exported
symbols.

16 years agotests/check/elements/audioresample.c: Add unit test for the latest basetransform...
Sjoerd Simons [Tue, 13 May 2008 10:59:49 +0000 (10:59 +0000)]
tests/check/elements/audioresample.c: Add unit test for the latest basetransform negotiation changes.

Original commit message from CVS:
Patch by: Sjoerd Simons <sjoerd at luon dot net>
* tests/check/elements/audioresample.c:
(live_switch_alloc_only_48000), (live_switch_get_sink_caps),
(live_switch_push), (GST_START_TEST):
Add unit test for the latest basetransform negotiation changes.
See bug #526768.

16 years agogst/ffmpegcolorspace/imgconvert.c: Fix nv12<->nv21 conversion if stride is larger...
Sebastian Dröge [Tue, 13 May 2008 09:14:44 +0000 (09:14 +0000)]
gst/ffmpegcolorspace/imgconvert.c: Fix nv12<->nv21 conversion if stride is larger than width.

Original commit message from CVS:
* gst/ffmpegcolorspace/imgconvert.c: (nv12_to_nv21):
Fix nv12<->nv21 conversion if stride is larger than width.

16 years agoext/ogg/gstoggdemux.*: Parse presentation time from skeleton streams and use it as...
j^ [Tue, 13 May 2008 07:28:21 +0000 (07:28 +0000)]
ext/ogg/gstoggdemux.*: Parse presentation time from skeleton streams and use it as offset for the timestamps. Fixes b...

Original commit message from CVS:
Patch by: j^ <j at oil21 dot org>
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_parse_skeleton_fishead),
(gst_ogg_pad_parse_skeleton_fisbone):
* ext/ogg/gstoggdemux.h:
Parse presentation time from skeleton streams and use it as offset
for the timestamps. Fixes bug #530068.