Adds a mapping to the _ELEVATION tag, this is a different
mapping as it has to be mapped into exif:GPSAltitude and
exif:GPSAltitudeRef at the same time. So we needed to refactor
a little more to be able to deserialize it properly.
Now, when parsing a xmp buffer into a taglist all tags are
added to a list before being parsed so that when one of the
altitude tags are found the deserialization function can search
for its complementary tag to do the correct parsing
Fixes#613690
Fix up videorate test for latest videotestsrc changes: just check for
the important bits in the negotiated caps, not for exact equality with
our filter caps. Also don't leak the videorate element in the test.
XMP metadata can be embedded in many media container formats. Implement own
parser and formatter that can be used to convert between an xpacket and a
GstTagList. Add unit tests.
The only difference between the two unit tests right now is,
that the decodebin2 test resets the element to READY before trying
to reuse it instead of NULL. decodebin2 guarantees to be reusable
without going back to NULL.
Add some utility functions for language tags and ISO-639
codes. These are useful for both GUIs and elements. The
iso-codes package is used for language name translations
if available.
API: gst_tag_get_language_codes()
API: gst_tag_get_language_name()
API: gst_tag_get_language_code()
API: gst_tag_get_language_code_iso_639_1()
API: gst_tag_get_language_code_iso_639_2B()
API: gst_tag_get_language_code_iso_639_2T()
Add a new video event to mark the start or end of a still-frame
sequence, and a parser function to identify and extract info from
such events.
API: gst_video_event_new_still_frame()
API: gst_video_event_parse_still_frame()
Fixes: #601942
Rhythmbox uses cdda:// URIs of the form cdda://track#device, which
worked before the fix for bug #321532.
Also adds a check for negative track numbers and some unit tests for URI
parsing.
Fixes bug #595454.
Revert previous 'fix' for bug #588717 and fix it properly, whilst
maintaining the streamheader field on the output caps. Also make
sure we don't leak header buffers we couldn't push when downstream
is unlinked. Add unit test for the presence of the streamheader
field on the output caps and for the issue from bug #588717.
There are flac-in-ogg files without the usual flac packet framing
and these files just have a 4-byte fLaC ID packet as first packet.
We need to recognise the type just from these four bytes if we
want oggdemux to recognise these streams correctly.
This tests seeking on an adder that has a normal and a live source connected.
Wheter the current behavior is the desired one needs to be discussed still
(see #586033)
Be even less restrictive in what we accept for .srt timestamps when
typefinding and parsing subrip subtitles and add a unit test for
the 'new' format. Fixes#585197.
Don't use REPLACE_ALL merge mode when that's not really what we want,
as now that REPLACE_ALL actually does what it's supposed to do in
core, we drop tags we wanted to keep, such as the various disc id
tags. Add unit test for this as well. Fixes#579463.
Add support for a set of standard commands that can be queried and executed to
support applications like DVD. Add query construction and parsing functions.
Add new messages that can be sent on the bus to provide notifications related
to commands, multiangle changes, and button highlight activity.
Add some helper functions to parse the existing GstNavigation events that
elements might receive.
Document it all and add unit tests.
If one side has a preference for a particular sample rate or set of sample rates, we
should honour this in the caps we advertise and transform to and from, so that elements
actually know about the other side's sample rate preference and can negotiate to it
if supported. Also add unit test for this.
Add some padding to the callbacks structure just to be safe.
Remove the now invisible marshaller methods from the docs.
Fix a comment in the unit test.
Based on pacth by Martin Samuelsson <martin dot samuelsson at axis dot com>
Fixes#571299.
Add gst_app_sink_set_callbacks() to install a set of callbacks. This is a more
performant alternative to connecting to the signals.
Add a unit test for appsink.
Clean up some of the appsink docs.
API: GstAppSink::gst_app_sink_set_callbacks()
It doesn't make sense to ask installers for the same codec or element twice, so filter out duplicate requests before calling the external helper script and make the unit test check this works right. Fixes#567636.
Original commit message from CVS:
* gst-libs/gst/cdda/gstcddabasesrc.c:
(gst_cdda_base_src_handle_event):
Remove erroneous gst_buffer_ref().
* tests/check/libs/rtp.c: (GST_START_TEST):
Don't forget to unref the buffer once you're done with it.
Original commit message from CVS:
Patch by: Olivier Crete <tester at tester ca>
* gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_packet_remove):
* gst-libs/gst/rtp/gstrtcpbuffer.h:
Implement gst_rtcp_packet_remove(). Fixes#563174.
* tests/check/libs/rtp.c: (GST_START_TEST), (rtp_suite):
Add unit test for some RTCP functions.
Original commit message from CVS:
* tests/check/elements/speexresample.c: (test_pipeline):
Make unit test again faster to prevent timeouts with valgrind.
Original commit message from CVS:
* tests/check/elements/speexresample.c: (GST_START_TEST):
Make the unit test a bit faster to prevent timeouts, especially
with valgrind.
Original commit message from CVS:
* tests/check/elements/speexresample.c: (element_message_cb),
(eos_message_cb), (test_pipeline), (GST_START_TEST),
(speexresample_suite):
Add pipeline unit tests for testing all supported formats with
up/downsampling and different in/outrates.
* gst/speexresample/gstspeexresample.c:
(gst_speex_resample_push_drain), (gst_speex_resample_process):
* gst/speexresample/speex_resampler_wrapper.h:
Fix bugs identified by the testsuite.
Original commit message from CVS:
* gst/speexresample/gstspeexresample.c: (gst_speex_resample_start),
(gst_speex_resample_get_unit_size),
(gst_speex_resample_push_drain), (gst_speex_resample_event),
(gst_speex_resample_check_discont), (gst_speex_resample_process),
(gst_speex_resample_transform):
* gst/speexresample/gstspeexresample.h:
Rewrite timestamp tracking to make it more robust and guarantee
a continous stream.
* tests/check/Makefile.am:
* tests/check/elements/speexresample.c: (setup_speexresample),
(cleanup_speexresample), (fail_unless_perfect_stream),
(test_perfect_stream_instance), (GST_START_TEST),
(test_discont_stream_instance), (live_switch_alloc_only_48000),
(live_switch_get_sink_caps), (live_switch_push),
(speexresample_suite):
Add unit tests for speexresample based on the audioresample unit tests.
Original commit message from CVS:
* gst/subparse/gstsubparse.c:
(gst_sub_parse_data_format_autodetect), (handle_buffer),
(gst_sub_parse_change_state):
* gst/subparse/gstsubparse.h:
* tests/check/elements/subparse.c: (GST_START_TEST):
Add support for subtitle files with UTF-8 BOM at the beginning
by simple stripping it from the first line before passing it
to any parsing code. Fixes bug #555257 and playback of files
created by Gnome Subtitles.
Original commit message from CVS:
Based on a patch by: xavierb at gmail dot com
* gst/subparse/gstsubparse.c:
(gst_sub_parse_data_format_autodetect):
* tests/check/elements/subparse.c: (GST_START_TEST):
Make the detection of the used subtitle a bit less strict
for srt subtitles. Fixes bug #555607.
Original commit message from CVS:
* configure.ac:
* gst/subparse/Makefile.am:
* gst/subparse/gstsubparse.c:
* gst/subparse/samiparse.c:
* tests/check/elements/subparse.c:
Rework last change, so that we build subparse, but just disable the
sami parse functionality, if we're configured to not use xml. In the
tests only the sami test is disabled now.
Original commit message from CVS:
* tests/check/elements/audioresample.c: (setup_audioresample),
(fail_unless_perfect_stream), (test_perfect_stream_instance),
(test_discont_stream_instance):
Now that GstBaseTransform is 'fixed' ... remove cruft from tests.
Add debugging for coherence.
Original commit message from CVS:
* tests/check/elements/gdpdepay.c: (gdpdepay_suite):
* tests/check/pipelines/streamheader.c: (streamheader_suite):
Enable unit tests on PPC again as the bugs are now fixed.
Original commit message from CVS:
* sys/ximage/ximagesink.c (gst_ximagesink_ximage_put),
(gst_ximagesink_setcaps):
* sys/ximage/ximagesink.h:
When the caps change, make sure to re-draw borders in
force-aspect-ratio=true mode.
* sys/xvimage/xvimagesink.c (gst_xvimagesink_xvimage_put):
Don't clear the border_draw flag until we actually draw the border.
* tests/check/Makefile.am:
Ignore alsasink/src during the states test too, so it doesn't fail
when running without access to the sound device.
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.
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().
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.
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).
