Rename the offset field in GstVideoFormatInfo to poffset to avoid confusion with
the offset of the plane in the buffer. The poffset is the offset in the plane
where the first byte of the component data can be found.
Properly implement the COMP_OFFSET calculations.
Fix YV12 and YVU9, simply use the same offsets as the regular I420 and YUV9
variants, we use the plane info to reorder components already.
Improve the unit test.
We have removed things like protocol=gdp in the tcp elements
in favour of explicit gdppay/depay elements, so there's no need
to keep a public API and library for now. We can still add it
back later. Someone needs to think hard about 0.11 and gdp
anyway one of these days.
When the payload for an Exif tag is less than or equal to 4 bytes,
the data is simply put into the offset field. Fix writing these
kinds of payloads on big endian systems (and possibly also on
little endian systems). The caller will have already formatted
the bytes in memory according to the writer's endianness, so just
write out the bytes as they are in this case. Fixes tags unit test
on big endian systems.
We used to add a trailing \n to the end of generated xmp packets.
Windows viewer was unhappy with it and we fixed it in
96d2120c2b
The problem is that this caused xmp generated before this fix
to not be recognized and parsed anymore. This patch makes it
recognize xmp with the trailing \n and without, fixing the
regression. Also adds tests for it.
Make appsink return a GstSample. Remove the pull_buffer_list method because it
is not very useful anymore.
Pass GstSample to the conversion function.
Update playbin2 and examples
Previously, the source posted a TAG message before buffers would
even be pushed towards the sink, so we'd get the TAG message before
any ASYNC_DONE message. Now the tags get sent downstream to the sink
to get posted there, and the tag event will get queued and handled
later after preroll has finished, so now we get the ASYNC_DONE
message before the TAG message.
If they go when calling snd_config_update_free_global, they're
not really bug leaks, but more like intentional ones we don't
want to get told about.
https://bugzilla.gnome.org/show_bug.cgi?id=615342
There's no code whatsoever that uses these macros. If anyone
ever feels the need to resurrect them, we should add them to
gstutils.h in core or libgstaudio or so.
This will cause not-linked errors that usually don't happen
because normal decoders/parsers will set srcpad caps before
allocating buffers from downstream.
Make sure decodebin1 doesn't try to plug the same parser twice
in a row (so we can change all parsers to accept parsed input as
well without breaking applications still using the old decodebin1
element).
Before the behaviour was undefined and implemented differently by elements,
now core checks for this (and other problems) and returns NULL and an assertion.
The unit tests are riddled with g_assert() and friends, sometimes
containing functional code like set_state() calls in them even
(looking at you, pipeline/capsfilter-renegotiation). Make sure we
don't disable assert and cast checks for the unit tests even if
this has been specified for the rest of the code base, e.g. via
--disable-glib-asserts.
Adds a tag to inform what mode was used by a camera to calculate
the picture capturing exposure
Also adds mapping to exif and tests
API: GST_TAG_CAPTURING_METERING_MODE
https://bugzilla.gnome.org/show_bug.cgi?id=631773
Adds new tag for tagging sharpness processing used
when capturing an image. Also maps it in the exif
tags.
Tests included.
API: GST_TAG_CAPTURING_SHARPNESS
https://bugzilla.gnome.org/show_bug.cgi?id=631773
Make appsrc not set caps on buffers when its own caps is NULL.
This avoids calling make_metadata_writable on all buffers and
prevents losing buffer caps in case we are not replacing it
with something meaningful.
https://bugzilla.gnome.org/show_bug.cgi?id=630353
Binding generators apparently need this as they can't really know
that the callback is guaranteed to be called exactly once and that
the user_data can be freed at the end of it.
Adds mappings from:
GST_TAG_CAPTURING_EXPOSURE_PROGRAM -> ExposureProgram
GST_TAG_CAPTURING_EXPOSURE_MODE -> ExposureMode
GST_TAG_CAPTURING_SCENE_CAPTURE_TYPE -> SceneCaptureType
GST_TAG_CAPTURING_GAIN_ADJUSTMENT -> GainControl
GST_TAG_CAPTURING_WHITE_BALANCE -> WhiteBalance
GST_TAG_CAPTURING_CONTRAST -> Constrast
GST_TAG_CAPTURING_SATURATION -> Saturation
Also renames gst_tag_image_orientation_from_exif_value and
gst_tag_image_orientation_to_exif_value to remove the 'gst'
prefix and not including in the win32 defs.
Tests included.
If we find a bag of tags of type string in the xmp packet, we
should concat them, this is not the ideal approach, but at
least works for now as we don't know what type of tag it
is (simple, structure, seq, alt or bag)
So people can check what version of the gst-plugins-base libs they're
building against or linked against.
API: GST_PLUGINS_BASE_VERSION_MAJOR
API: GST_PLUGINS_BASE_VERSION_MINOR
API: GST_PLUGINS_BASE_VERSION_MICRO
API: GST_PLUGINS_BASE_VERSION_NANO
API: GST_CHECK_PLUGINS_BASE_VERSION
API: gst_plugins_base_version()
API: gst_plugins_base_version_string()
Use new gst_element_link_pads_full() function to link elements,
and disable all checks when linking (don't try this at home).
Down to 18s from 3m20s. Scary.
Adds mappings for:
GST_TAG_GEO_LOCATION_CAPTURE_DIRECTION
GST_TAG_GEO_LOCATION_MOVEMENT_DIRECTION
GST_TAG_GEO_LOCATION_MOVEMENT_SPEED
GST_TAG_GEO_LOCATION_ELEVATION
Does some refactoring in the code to reduce number of parameters
passed to functions
Tests included.
Adds GST_TAG_GEO_LOCATION_MOVEMENT_SPEED,
GST_TAG_GEO_LOCATION_MOVEMENT_DIRECTION and
GST_TAG_GEO_LOCATION_CAPTURE_DIRECTION to xmp
mappings.
Tests included.
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).
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.
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.
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.
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.
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.
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.
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.
Original commit message from CVS:
Patch by: Sjoerd Simons <sjoerd at luon dot net>
* gst/audioresample/gstaudioresample.c: (gst_audioresample_init):
Let audioresample use the buffer allocation of basetransform instead
of it's own stuff.
* tests/check/elements/audioresample.c: (alloc_only_48000),
(GST_START_TEST), (audioresample_suite):
Add unit test for the recent basetransform bugfix, where upstream
changes caps to something that can't be passed through anymore.
Original commit message from CVS:
Based on a patch by: Tim-Philipp Müller <tim.muller at collabora co uk>
* gst/audioconvert/audioconvert.c: (audio_convert_prepare_context):
* gst/audioconvert/audioconvert.h:
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_parse_caps),
(structure_has_fixed_channel_positions),
(gst_audio_convert_transform_caps):
* gst/audioconvert/gstchannelmix.c: (gst_channel_mix_fill_matrix):
Add support for more than 8 channels and NONE channel layouts. For
more than 8 channels no channel conversion is supported yet, only
format conversions are supported. Fixes bug #398033.
* tests/check/elements/audioconvert.c: (verify_convert),
(GST_START_TEST), (audioconvert_suite):
Add some unit tests by Tim for checking the NONE channel layouts
and more than 8 channels and add some more unit tests for channel
conversions.
Original commit message from CVS:
* gst/subparse/samiparse.c: (handle_start_sync),
(end_sami_element), (characters_sami):
Remove trailing, leading and double whitespaces.
Correctly timestamp buffers and output the last buffer too.
* tests/check/elements/subparse.c: (GST_START_TEST),
(subparse_suite):
Add a simple unit test for SAMI parsing.
Original commit message from CVS:
* tests/check/elements/subparse.c: (do_test),
(test_tmplayer_style3b), (subparse_suite):
Add unit test for the tmplayer variant from bug #530962.
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.
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.
Original commit message from CVS:
* tests/check/libs/video.c: (paintinfo), (paintinfo_struct),
(fourcc_list_struct), (fourcc_list), (fourcc_get_size),
(paint_setup_I420), (paint_setup_YV12), (paint_setup_AYUV),
(paint_setup_YUY2), (paint_setup_UYVY), (paint_setup_YVYU),
(paint_setup_IYU2), (paint_setup_Y41B), (paint_setup_Y42B),
(paint_setup_Y800), (paint_setup_YVU9), (paint_setup_YUV9),
(gst_video_format_is_packed), (video_format_is_packed):
Add unit test that makes sure that the strides, offsets and
sizes returned for the various YUV formats by the new video API
match the old reference implementation in videotestsrc.
Original commit message from CVS:
Based on patch by: Tomasz Sałaciński <tsalacinski gmail com>
* gst/subparse/gstsubparse.c: (parse_mdvdsub):
* tests/check/elements/subparse.c: (test_microdvd_with_italics),
(subparse_suite):
Forward slashes at the beginning and end of a line also signify
italics (Fixes: #518162).
Original commit message from CVS:
* tests/check/gst-plugins-base.supp:
Add a suppression for a cached value in GIO that wasn't moved
while moving gio from -bad to -base.
Original commit message from CVS:
* ext/gio/gstgiostreamsink.c:
* ext/gio/gstgiostreamsrc.c:
Add documentation and example code for giostreamsink/giostreamsrc.
* tests/check/pipelines/gio.c: (GST_START_TEST):
Ask the GMemoryOutputStream for the data instead of assuming that
the pointer to the data stayed the same. It could've been realloc'ed.
Original commit message from CVS:
* tests/check/Makefile.am:
Revert previous change of the test environment's GST_PLUGIN_PATH.
The problem is not with the plugins, but with element factories
and only occurs if elements are split out from existing plugins
or if plugins change name (see #512740).
Original commit message from CVS:
* tests/check/Makefile.am:
Fix the tests environment's GST_PLUGIN_PATH: we want the directory
with the core's plugins first and our local build directories last,
since we might be building against an installed core, and that
core's plugin directory may contain older or other versions of
our own -base plugins, but we really do want to test our local
ones (if there are multiple plugins or element factories with the
same name, those inspected last will trump those read in earlier).
Fixes#512740 for the most part.
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()
Original commit message from CVS:
* configure.ac:
* tests/check/pipelines/gio.c: (GST_START_TEST):
Update for API changes in GIO and require GIO 2.15.2 for this.
Original commit message from CVS:
* tests/check/Makefile.am:
Fix CFLAGS to also pull in the gstcheck cflags from AM_CFLAGS.
* tests/check/elements/audiorate.c: (do_perfect_stream_test):
* tests/check/elements/playbin.c:
* tests/check/libs/mixer.c: (test_element_interface_supported),
(gst_implements_interface_init):
* tests/check/libs/rtp.c: (GST_START_TEST):
Fix various assignment type mismatches.
Original commit message from CVS:
* ext/theora/theoradec.c: (gst_theora_dec_class_init),
(_theora_granule_frame), (_theora_granule_start_time),
(theora_dec_sink_convert), (theora_dec_decode_buffer):
Adapt for post-alpha meaning of granulepos, when we
have a newer version of libtheora.
* ext/theora/theoraenc.c: (gst_theora_enc_class_init),
(theora_enc_get_ogg_packet_end_time), (theora_enc_sink_event),
(theora_enc_is_discontinuous), (theora_enc_chain):
Likewise.
* tests/check/Makefile.am:
Link libtheora into theoraenc test so we can check which version of
libtheora we're testing against.
* tests/check/pipelines/theoraenc.c: (check_libtheora),
(check_buffer_granulepos),
(check_buffer_granulepos_from_starttime), (GST_START_TEST),
(theoraenc_suite):
Adapt tests to check the values that are now defined for theora; make
the tests backwards-adapt the passed values if we're running against an
old libtheora.
Fixes#497964
Original commit message from CVS:
* configure.ac:
* ext/gio/gstgiobasesrc.c: (gst_gio_base_src_get_size):
* tests/check/pipelines/gio.c: (free_input), (GST_START_TEST):
Update to GMemoryInputStream API changes in GLib SVN and require
gio-2.0 >= 2.15.1 for this. Fixes bug #507584.
We can also report the duration for every GSeekable, not only
GFileInputStream and GMemoryInputStream.
Original commit message from CVS:
* tests/check/pipelines/theoraenc.c: (check_buffer_is_header),
(check_buffer_timestamp), (check_buffer_duration):
Turn these functions into macros so we can see right away
where the failure occured.
Original commit message from CVS:
* tests/check/Makefile.am:
* tests/check/pipelines/theoraenc.c:
Reenable theoraenc test, which fails on the buildbot but
not locally.
Original commit message from CVS:
* tests/check/pipelines/theoraenc.c:
Disable theoraenc test long enough to get the buildbot to
compile a recent -base.
Original commit message from CVS:
* gst/videoscale/gstvideoscale.c: (gst_video_scale_set_property),
(gst_video_scale_get_property), (gst_video_scale_transform_caps),
(gst_video_scale_transform):
Don't claim to be able to handle/transform caps that can't really
be handled by the currently selected scaling method (here: RGB or
packed YUV with 4-tap method). Also add locking to method property.
* tests/check/pipelines/simple-launch-lines.c: (setup_pipeline),
(test_basetransform_based):
Some test pipelines for the above (not entirely valgrind clean yet
apparently).
Original commit message from CVS:
* tests/check/elements/playbin.c: (test_sink_usage_video_only_stream),
(test_suburi_error_unknowntype), (test_suburi_error_invalidfile),
(test_suburi_error_wrongproto), (test_missing_primary_decoder):
* tests/check/libs/cddabasesrc.c: (GST_START_TEST), (GST_START_TEST),
(cddabasesrc_suite):
Don't use GST_PLUGIN_DEFINE_STATIC, it's not portable and will be
deprecated in the future (see #498924).
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.
Original commit message from CVS:
Patch by: Robin Stocker <robin dot stocker at gmx dot ch>
* gst/subparse/gstsubparse.c: (gst_sub_parse_data_format_autodetect):
Some .srt files start with chunk number 0 and not chunk number 1,
recognise and accept those as well (fixes#502497).
* tests/check/elements/subparse.c: (srt_input), (srt_input0),
(test_src):
Add unit test for the above.
Original commit message from CVS:
* configure.ac:
* tests/check/pipelines/gio.c: (GST_START_TEST):
Require GIO >= 0.1.2 and adjust unit test for an API change.
Original commit message from CVS:
* ext/gio/gstgio.h:
Add macro to check if a stream supports seeking.
* ext/gio/Makefile.am:
* ext/gio/gstgiobasesink.c: (gst_gio_base_sink_base_init),
(gst_gio_base_sink_class_init), (gst_gio_base_sink_init),
(gst_gio_base_sink_finalize), (gst_gio_base_sink_start),
(gst_gio_base_sink_stop), (gst_gio_base_sink_unlock),
(gst_gio_base_sink_unlock_stop), (gst_gio_base_sink_event),
(gst_gio_base_sink_render), (gst_gio_base_sink_query),
(gst_gio_base_sink_set_stream):
* ext/gio/gstgiobasesink.h:
* ext/gio/gstgiobasesrc.c: (gst_gio_base_src_base_init),
(gst_gio_base_src_class_init), (gst_gio_base_src_init),
(gst_gio_base_src_finalize), (gst_gio_base_src_start),
(gst_gio_base_src_stop), (gst_gio_base_src_get_size),
(gst_gio_base_src_is_seekable), (gst_gio_base_src_unlock),
(gst_gio_base_src_unlock_stop), (gst_gio_base_src_check_get_range),
(gst_gio_base_src_create), (gst_gio_base_src_set_stream):
* ext/gio/gstgiobasesrc.h:
Refactor common GIO functions to GstGioBaseSink and GstGioBaseSrc
base classes that only require a GInputStream or GOutputStream to
work.
* ext/gio/gstgiosink.c: (gst_gio_sink_base_init),
(gst_gio_sink_class_init), (gst_gio_sink_init),
(gst_gio_sink_finalize), (gst_gio_sink_start):
* ext/gio/gstgiosink.h:
* ext/gio/gstgiosrc.c: (gst_gio_src_base_init),
(gst_gio_src_class_init), (gst_gio_src_init),
(gst_gio_src_finalize), (gst_gio_src_start):
* ext/gio/gstgiosrc.h:
Use the newly created base classes here.
* ext/gio/gstgio.c: (plugin_init):
* ext/gio/gstgiostreamsink.c: (gst_gio_stream_sink_base_init),
(gst_gio_stream_sink_class_init), (gst_gio_stream_sink_init),
(gst_gio_stream_sink_finalize), (gst_gio_stream_sink_set_property),
(gst_gio_stream_sink_get_property):
* ext/gio/gstgiostreamsink.h:
* ext/gio/gstgiostreamsrc.c: (gst_gio_stream_src_base_init),
(gst_gio_stream_src_class_init), (gst_gio_stream_src_init),
(gst_gio_stream_src_finalize), (gst_gio_stream_src_set_property),
(gst_gio_stream_src_get_property):
* ext/gio/gstgiostreamsrc.h:
Implement GstGioStreamSink and GstGioStreamSrc that have a property
to set the GInputStream/GOutputStream that should be used.
* tests/check/Makefile.am:
* tests/check/pipelines/.cvsignore:
* tests/check/pipelines/gio.c: (message_handler), (GST_START_TEST),
(gio_testsuite), (main):
Add unit test for giostreamsrc and giostreamsink.
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.
Original commit message from CVS:
Patch by: Thijs Vermeir <thijsvermeir@gmail.com>
* tests/check/elements/audioconvert.c: (verify_convert):
Add check to make sure that the out caps have a channel layout
set on them where they should have one.
Original commit message from CVS:
* tests/check/elements/decodebin.c: (test_text_plain_streams):
Make sure the pipeline really operates in push mode as it should
in this case.
Original commit message from CVS:
* tests/check/libs/audio.c: (init_value_to_channel_layout),
(test_channel_layout_value_intersect), (audio_suite):
Add simple unit test to make sure GstValue intersection
of channel layouts works the way I think it does.
Original commit message from CVS:
* tests/check/libs/tag.c: (test_musicbrainz_tag_registration):
GST_TAG_MUSICBRAINZ_SORTNAME is deprecated and we compile with
-DGST_DISABLE_DEPRECATED, so use new GST_TAG_ARTIST_SORTNAME
instead.
Original commit message from CVS:
* tests/check/elements/libvisual.c:
Test all libvisual plugins, not just the first one; this reproduces
bug #450336 quite easily. Looks like a problem with the 'jess'
visualisation.
Original commit message from CVS:
* tests/check/Makefile.am:
* tests/check/elements/.cvsignore:
* tests/check/elements/libvisual.c:
Add basic libvisual test case in an attempt to reproduce bug #450336.
Doesn't reproduce that bug, but some other crasher instead (invalid
free), at least with make elements/libvisual.forever and the bumscope
plugin on x86-64/gutsy. Leaving test disabled for now.
Original commit message from CVS:
* gst/volume/gstvolume.c: (volume_choose_func):
* tests/check/elements/volume.c: (GST_START_TEST):
Revert the latest change: floating point samples are allowed to
have any value, not only values in the range [-1,1]. Thanks to Andy
Wingo for noticing.
Also fix processing of int32 samples with volumes > 4 by making the
unity value smaller which prevents overflows.
Original commit message from CVS:
* gst-libs/gst/rtp/gstrtpbuffer.c:
* tests/check/libs/rtp.c:
Fix gst_rtp_buffer_set_csrc() and enable csrc-related unit test checks.
Original commit message from CVS:
Based on patch by: Haakon Sporsheim <haakon.sporsheim at tandberg com>
* gst-libs/gst/rtp/gstrtpbuffer.c:
Fix up GstRTPHeader helper struct so that compilers will not under
any circumstances add padding in between our fields, as currently
happens with MSVC on win32, because that would lead to us sending
out RTP payloads with broken RTP headers (#471194).
Fix assertion guards for gst_rtp_buffer_get_csrc() and _set_csrc().
* tests/check/Makefile.am:
* tests/check/libs/.cvsignore:
* tests/check/libs/rtp.c:
Add some simple unit tests for GstRTPBuffer. Some are disabled
because the code tested still needs fixing (set_csrc() does not work).
Original commit message from CVS:
* gst/volume/gstvolume.c: (volume_choose_func),
(volume_process_double), (volume_process_double_clamp),
(volume_process_float_clamp):
Correctly clamp float/double samples in the [-1.0,1.0] range to
prevent weird effects.
* tests/check/elements/volume.c: (GST_START_TEST), (volume_suite):
Add unit tests for all samples types that had none before.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/pbutils/install-plugins.c:
* gst-libs/gst/pbutils/install-plugins.h:
* tests/check/libs/pbutils.c:
API: also add gst_install_plugins_supported() while we're at it
(see #470456).
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/pbutils/missing-plugins.c:
* gst-libs/gst/pbutils/missing-plugins.h:
* tests/check/libs/pbutils.c:
API: add gst_missing_*_installer_detail_new() convenience API so
that applications that know exactly what they're missing can request
installer detail strings for those items directly instead of having
to first create a dummy missing-plugin message and then get the
installer detail string from that. Fixes#470456.
Original commit message from CVS:
* gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
Debug output fixes.
* tests/check/elements/audiorate.c: (do_perfect_stream_test),
(GST_START_TEST):
Change the number of buffers used; 500 is too many and leads to
timeouts.
Original commit message from CVS:
* gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
If we have a large (> 1 second) discontinuity, push a series of
smaller buffers rather than a single very large buffer. Avoids
unreasonably large single buffer allocations when encountering a
large gap.
* tests/check/elements/audiorate.c: (GST_START_TEST),
(audiorate_suite):
Add a test for this.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/audio/audio.c: (gst_audio_buffer_clip):
* gst-libs/gst/audio/audio.h:
* tests/check/libs/audio.c: (GST_START_TEST), (audio_suite):
API: Add buffer clipping function for raw audio buffers. Fixes#456656.
Also add deprecation guards for gst_audio_structure_set_int() to the
header.
Original commit message from CVS:
* gst/audioconvert/Makefile.am:
* gst/audioconvert/audioconvert.c: (audio_convert_get_func_index),
(check_default), (audio_convert_prepare_context),
(audio_convert_clean_context), (audio_convert_convert):
* gst/audioconvert/audioconvert.h:
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_dithering_get_type),
(gst_audio_convert_ns_get_type), (gst_audio_convert_class_init),
(gst_audio_convert_init), (gst_audio_convert_set_caps),
(gst_audio_convert_set_property), (gst_audio_convert_get_property):
* gst/audioconvert/gstaudioconvert.h:
* gst/audioconvert/gstaudioquantize.c:
(gst_audio_quantize_setup_noise_shaping),
(gst_audio_quantize_free_noise_shaping),
(gst_audio_quantize_setup_dither),
(gst_audio_quantize_free_dither),
(gst_audio_quantize_setup_quantize_func),
(gst_audio_quantize_setup), (gst_audio_quantize_free):
* gst/audioconvert/gstaudioquantize.h:
Implement dithering and noise shaping in audioconvert. By default now
TPDF dithering (and no noise shaping) will be used when converting
from a higher bit depth to 20 bit depth or smaller, otherwise
everything will be as it is now.
For the last audioconvert in a pipeline it would make sense to
use some kind of noise shaping, enabling it by default for all
conversions would give undesired results though. Fixes#360246.
* tests/check/elements/audioconvert.c: (setup_audioconvert),
(GST_START_TEST):
Adjust unit test for the new audioconvert.
Original commit message from CVS:
* gst-libs/gst/tag/tags.c: (gst_tag_freeform_string_to_utf8):
Don't pass trailing zeroes in fixed-size string arrays in ID3v1 tags
to utf8-validate; fixes recognition of ID3v1 tags in UTF-8 encoding
(#451707); also, output some debugging info when dealing with
freeform strings.
* tests/check/libs/tag.c: (GST_START_TEST), (tag_suite):
Add unit test for the above.
Original commit message from CVS:
* ext/vorbis/vorbisdec.c: (vorbis_dec_decode_buffer):
Skip empty buffers, but not empty header buffers. That way the original
vorbisdec unit test still passes (#451145); also, take into account
that those empty packets might carry a granulepos.
* tests/check/Makefile.am:
* tests/check/elements/vorbisdec.c:
(_create_codebook_header_buffer), (_create_audio_buffer),
(GST_START_TEST), (vorbisdec_suite):
Add unit test that sends an empty packet.
Original commit message from CVS:
* tests/check/pipelines/oggmux.c: (validate_ogg_page), (is_video),
(eos_buffer_probe):
Add a test that ensures we set DELTA_UNIT on all non-header,
non-video buffers, if we have a video stream.
* ext/ogg/gstoggmux.c: (gst_ogg_mux_queue_pads),
(gst_ogg_mux_process_best_pad):
Move setting delta_pad to earlier, where we inspect all pads, so
that leading audio pages don't get DELTA_UNIT unset if they come
before the first DELTA_UNIT from video pages. Fixes the newly-added
test. Fixes#385527.
Original commit message from CVS:
* tests/check/pipelines/streamheader.c: (streamheader_suite):
Disable test_multifdsink_gdp_vorbisenc() on ppc64 since it
fails on the p5-ppc64 build bot and the failure looks like it is due
to the same issue as #348114, ie. a compiler bug.
Original commit message from CVS:
Based on a patch by Sven Arvidsson <sa at whiz dot se>:
* gst/subparse/gstsubparse.c: (parse_subrip),
(subviewer_unescape_newlines), (parse_subviewer),
(gst_sub_parse_data_format_autodetect),
(gst_sub_parse_format_autodetect), (gst_subparse_type_find):
* gst/subparse/gstsubparse.h:
Add support for SubViewer version 1 and 2 subtitles. Fixes#394061.
* tests/check/elements/subparse.c: (GST_START_TEST),
(subparse_suite):
Add a unit test for both SubViewer formats.
Original commit message from CVS:
* tests/check/elements/playbin.c: (test_suburi_error_unknowntype):
Use /dev/zero instead of /dev/urandom to produce an invalid subtitle
file. Avoids flukes where the input gets typefound to some valid but
useless type.
Original commit message from CVS:
* tests/check/Makefile.am:
* tests/check/elements/.cvsignore:
* tests/check/elements/gnomevfssink.c: (setup_gnomevfssink),
(cleanup_gnomevfssink), (GST_START_TEST), (gnomevfssink_suite):
Add unit test for gnomevfssink seeking and position reporting for
file:// URIs.
Original commit message from CVS:
* tests/check/Makefile.am:
* tests/check/elements/volume.c: (GST_START_TEST), (volume_suite):
Minimal check for volume's GstController usability; also another
test for #422295.
Original commit message from CVS:
* tests/check/elements/videorate.c: (GST_START_TEST):
Set buffer timestamp to a valid value in order to test the buffer
really does stay in videorate.
Original commit message from CVS:
Patch by: Dan Williams <dcbw redhat com>
* gst/videorate/gstvideorate.c: (gst_video_rate_chain):
Don't leak incoming buffer if gst_pad_push() returns a
non-OK flow. Fixes#432755.
* tests/check/elements/videorate.c: (GST_START_TEST),
(videorate_suite):
Unit test for the above by Yours Truly.
Original commit message from CVS:
* gst/subparse/gstsubparse.c:
* gst/subparse/samiparse.c:
Use GST_DISABLE_XML here
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_put),
(gst_xvimagesink_xwindow_new), (gst_xvimagesink_get_xv_support),
(gst_xvimagesink_buffer_alloc),
(gst_xvimagesink_navigation_send_event):
* sys/xvimage/xvimagesink.h:
Include stdlib.h when using atoi.
* tests/check/elements/playbin.c: (playbin_suite):
Use GST_DISABLE_REGISTRY here
Original commit message from CVS:
* gst/gdp/gstgdppay.c (gst_gdp_pay_chain):
* tests/check/pipelines/streamheader.c (tag_event_probe_cb,
GST_START_TEST, buffer_probe_cb, GST_START_TEST):
Fix a bug where serialized IN_CAPS buffers needed to be set IN_CAPS.
Original commit message from CVS:
* tests/check/pipelines/streamheader.c (tag_event_probe_cb,
GST_START_TEST, n_in_caps, buffer_probe_cb, GST_START_TEST,
streamheader_suite):
Add another test set up for failure
Original commit message from CVS:
* tests/check/Makefile.am:
* tests/check/pipelines/streamheader.c (n_tags, tag_event_probe_cb,
GST_START_TEST, streamheader_suite, main):
Add a test for the streamheader bug Wim fixed.
Original commit message from CVS:
* gst/videorate/gstvideorate.c: (gst_video_rate_setcaps),
(gst_video_rate_chain):
Add some debug.
* tests/check/elements/videorate.c: (GST_START_TEST),
(videorate_suite):
Added check for videorate changing caps handling. Closes#421834.
Original commit message from CVS:
* tests/check/elements/playbin.c:
(test_sink_usage_video_only_stream), (playbin_suite):
Add small test for stream-info-value-array code paths.
Original commit message from CVS:
Patch by: René Stadler <mail at renestadler dot de>
with some minor changes
* gst-libs/gst/floatcast/floatcast.h:
Use more efficient float endianness conversion functions that don't
involve 2 function calls per value.
* gst/audioconvert/audioconvert.c: (audio_convert_get_func_index),
(check_default), (audio_convert_prepare_context):
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_parse_caps), (make_lossless_changes):
Support non-native endianness floats as input and output.
Fixes#339838.
* tests/check/elements/audioconvert.c: (verify_convert),
(GST_START_TEST):
Add unit tests for the non-native endianness float conversions.
Original commit message from CVS:
2007-03-29 Andy Wingo <wingo@pobox.com>
* gst/videorate/gstvideorate.c (gst_video_rate_flush_prev): Make
perfect offsets also, not just timestamps.
* tests/check/elements/videorate.c (test_more): Test that given
any incoming offsets, that videorate produces perfect offsets.
Original commit message from CVS:
* gst/audioconvert/audioconvert.c:
Add docs to the integer pack functions and implement proper
rounding. Before we had rounding towards negative infinity, i.e.
always the smaller number was taken. Now we use natural rounding,
i.e. rounding to the nearest integer and to the one with the largest
absolute value for X.5. The old rounding introduced some minor
distortions. Fixes#420079
* tests/check/elements/audioconvert.c: (GST_START_TEST):
Fix one unit test that assumed the old rounding and added unit tests
for checking signed/unsigned int16 <-> signed/unsigned int16 with
depth 8, one for signed int16 <-> unsigned int16 and one for the new
rounding from signed int32 to signed/unsigned int16.
Original commit message from CVS:
* gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
* tests/check/libs/tag.c: (GST_START_TEST):
Make sure we parse floating-point numbers in vorbis comments
correctly with either '.' or ',' as separator, no matter what
the current locale is. Add unit test for this too.
Original commit message from CVS:
Patch by: René Stadler <mail at renestadler de>
* gst-libs/gst/tag/gstvorbistag.c: (gst_tag_to_vorbis_comments):
When writing out floating-point numbers to vorbis comment tags, always
use the same character as separator no matter what the current locale is
(fixes#423051).
* tests/check/libs/tag.c: (GST_START_TEST):
Add unit tests for replaygain tags in vorbis comments (closes#423055).
Original commit message from CVS:
* ext/vorbis/vorbisdec.c (vorbis_dec_push_forward,
vorbis_handle_data_packet):
Correctly set DURATION to generate a timestamp-continuous stream.
One bug left at the end; see
ihttp://bugzilla.gnome.org/show_bug.cgi?id=423086
* tests/check/Makefile.am:
* tests/check/pipelines/vorbisenc.c (GST_START_TEST):
Add a test to check this. Without the above patch this test fails.
Original commit message from CVS:
* gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
Also accept partial dates with only year and month,
like 1999-12-00 (fixes#410396 even more).
* tests/check/libs/tag.c: (GST_START_TEST):
Add unit test for the above.
Original commit message from CVS:
* tests/check/elements/subparse.c: (GST_START_TEST),
(subparse_suite):
Add unit test for MPL2 subtitle format (#413799).
Original commit message from CVS:
* tests/check/elements/alsa.c: (GST_START_TEST):
Unref the mixer if the state change fails too (if the
alsa devices are inaccessible, for example)
Original commit message from CVS:
* tests/check/Makefile.am:
Don't test libvisual elements in the states check, because libvisual
seems to leak internally.
Re-enable the alsa and states tests now that there's new suppressions
in gst.supp.
* tests/check/elements/alsa.c: (GST_START_TEST):
Don't leak the alsamixer we instantiated.
Original commit message from CVS:
* gst-libs/gst/interfaces/mixertrack.c:
(gst_mixer_track_class_init), (gst_mixer_track_get_property),
(gst_mixer_track_set_property):
API: add "untranslated-label" property which should be set by
implementations at construct time (#414645).
* ext/alsa/gstalsamixeroptions.c: (gst_alsa_mixer_options_new):
* ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_new):
Set "untranslated-label" when constructing mixer track objects.
* tests/check/elements/alsa.c: (GST_START_TEST), (alsa_suite):
Unit test to check the above.
Original commit message from CVS:
* tests/check/generic/states.c: (GST_START_TEST):
Copy the states.c test from core again
* tests/check/Makefile.am:
ignore cdio and cdparanoiasrc
Original commit message from CVS:
* gst/audioconvert/audioconvert.c: (float), (double), (float_hq),
(double_hq), (audio_convert_get_func_index),
(audio_convert_prepare_context), (audio_convert_convert):
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_class_init), (gst_audio_convert_get_unit_size),
(gst_audio_convert_transform_caps):
* tests/check/elements/audioconvert.c: (GST_START_TEST),
(audioconvert_suite):
Don't run inplace if that overwrites source data as we go. Add more
tests. Fixes#339837 even more.
Original commit message from CVS:
Patch by: René Stadler <mail at renestadler de>
* gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
Parse date strings in vorbis comments that have an invalid (zero)
month or day (#410396).
* tests/check/libs/tag.c: (GST_START_TEST):
Test case for the above.
Original commit message from CVS:
* gst-libs/gst/utils/install-plugins.c:
* gst-libs/gst/utils/missing-plugins.c:
* tests/check/libs/utils.c: (missing_msg_check_getters):
Change GStreamer marker prefix in detail string from 'gstreamer.net'
to just 'gstreamer'. Document the caps string component of the
decoder/encoder detail a bit better, since not everyone will be
familiar with the GStreamer media type/caps system (but they better
enjoy nested itemized lists).
Original commit message from CVS:
* gst-libs/gst/netbuffer/gstnetbuffer.c:
(notgst_buffer_copy_fields_in_place), (gst_netbuffer_copy):
Fix copying of GstNetBuffer (would crash before, or at least lead to
invalid memory access, #410772), for now by copying the GstBuffer copy
code from the core over here so we can copy the GstBuffer fields on a
provided buffer instance (of type GstNetBuffer in this case). Would be
better to fix this with some support by the core though (and in the long
run change the broken GstBuffer/GstMiniObject copy semantics, #393099).
* tests/check/Makefile.am:
Enable unit test for GstNetBuffer.
Original commit message from CVS:
* gst/audioconvert/audioconvert.c: (float), (double), (float_hq),
(double_hq), (audio_convert_get_func_index),
(audio_convert_prepare_context), (audio_convert_convert):
* gst/audioconvert/audioconvert.h:
* gst/audioconvert/gstchannelmix.c: (gst_channel_mix_setup_matrix),
(gst_channel_mix_mix_int), (gst_channel_mix_mix_float):
* gst/audioconvert/gstchannelmix.h:
* tests/check/elements/audioconvert.c: (GST_START_TEST):
Add float as an intermediate format, as well as float mixing. Enable
test that was failing before. Fixes#339837
Original commit message from CVS:
* gst/subparse/gstsubparse.c: (subrip_remove_unhandled_tag),
(subrip_remove_unhandled_tags), (parse_subrip):
For SubRip (.srt) subtitles, ignore all markup tags we don't
handle (like font tags, for example).
* tests/check/elements/subparse.c:
Add test for this.
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (add_fakesink),
(gst_decode_bin_change_state):
* gst/playback/gstdecodebin2.c: (add_fakesink),
(gst_decode_bin_change_state):
Don't error out if there is no fakesink in the READY to NULL state
change, since when decodebin is re-used, we're only adding the
fakesink element in READY to PAUSED.
* tests/check/elements/decodebin.c:
(new_decoded_pad_plug_fakesink_cb), (GST_START_TEST),
(decodebin_suite):
Minimal unit test to make sure we can use the same decodebin
instance twice (at least with audiotestsrc input).
Original commit message from CVS:
* tests/check/pipelines/oggmux.c: (GST_START_TEST), (oggmux_suite):
Add small test to make sure request pads are cleaned up properly
even if oggmux never changes state out of NULL.
Original commit message from CVS:
* tests/check/libs/utils.c: (GST_START_TEST):
Fix unit test. Turns out things work much better when you
NULL-terminate string arrays. Should make p5 build bot happy again.
Original commit message from CVS:
* gst-libs/gst/utils/install-plugins.c:
(gst_install_plugins_spawn_child):
* tests/check/libs/utils.c:
(test_base_utils_install_plugins_do_callout):
Lowering log level to see why things fail on the p5 build bot;
fix some typos in unit test messages.
Original commit message from CVS:
* tests/check/libs/utils.c:
(test_base_utils_install_plugins_do_callout):
Don't hard-code temp directory for test helper; use GLib functions
to write out file and do error checking etc.
Original commit message from CVS:
* gst-libs/gst/utils/Makefile.am:
* gst-libs/gst/utils/base-utils.h:
* gst-libs/gst/utils/install-plugins.c:
(gst_install_plugins_context_set_xid),
(gst_install_plugins_context_new),
(gst_install_plugins_context_free),
(gst_install_plugins_get_helper),
(gst_install_plugins_spawn_child),
(gst_install_plugins_return_from_status),
(gst_install_plugins_installer_exited),
(gst_install_plugins_async), (gst_install_plugins_sync),
(gst_install_plugins_return_get_name),
(gst_install_plugins_installation_in_progress):
* gst-libs/gst/utils/install-plugins.h:
API: add API for applications to initiate installation of missing
plugins, ie. gst_install_plugins_async() primarily.
Based on libgimme-codec by Ryan Lortie.
* configure.ac:
Add --with-install-plugins-helper configure option so distros can specify
the path of the helper script or program to call when plugin installation
is requested (distros: please do any argument munging in this helper
script instead of patching GStreamer to pass arguments differently
to another program directly).
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
Build and document new API.
* tests/check/libs/utils.c: (result_cb),
(test_base_utils_install_plugins_do_callout), (GST_START_TEST),
(libgstbaseutils_suite):
Some simple checks for the new API.
Original commit message from CVS:
* tests/check/elements/audioconvert.c: (test_float_conversion):
Add small test for 32bit float <=> 64bit float conversion (works
only one way so far, 32=>64 produces structured noise).
Original commit message from CVS:
* ext/pango/gsttextoverlay.c: (gst_text_overlay_src_event),
(gst_text_overlay_text_event):
Don't unnecessarily ref (and then leak) upstream events if the text
pad is not linked. Fixes#399948.
* tests/check/gst-plugins-base.supp:
Add suppression for pango on edgy/x86 for textoverlay test.
Original commit message from CVS:
* gst/playback/gstdecodebin2.c: (gst_decode_bin_finalize):
Don't leak mutex.
* tests/check/elements/playbin.c:
(test_sink_usage_video_only_stream),
(test_suburi_error_unknowntype), (test_suburi_error_invalidfile),
(test_suburi_error_wrongproto), (test_missing_urisource_handler),
(test_missing_suburisource_handler),
(test_missing_primary_decoder), (playbin_suite):
Run all tests once with decodebin and once with decodebin2.
One test does not pass yet with decodebin2.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (mpeg4_video_type_find):
Don't go into an endless loop if the file starts with 00 00 01 2X,
like quicktime redirect files might. Fixes#396042.
* tests/check/Makefile.am:
* tests/check/gst/.cvsignore:
* tests/check/gst/typefindfunctions.c: (GST_START_TEST),
(typefindfunctions_suite):
Add unit test for the above.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (setup_subtitle),
(gen_source_element), (gst_play_base_bin_change_state):
Attempt at a better error message in case we don't have the required
URI handler installed; post missing-plugin message also when we're
missing an URI handler for the subtitle URI; clean up properly also
when an error occurs and we never made it to PAUSED state.
* tests/check/elements/playbin.c: (GST_START_TEST),
(playbin_suite):
Check that we're also getting a missing-plugin messsage for a
missing subtitle URI handler (and clean up properly).
Original commit message from CVS:
* gst/playback/Makefile.am:
* gst/playback/gstplaybasebin.c: (string_arr_has_str),
(unknown_type), (setup_subtitle), (gen_source_element):
* gst/playback/gstplaybin.c: (plugin_init):
Post missing-plugin messages on the bus for missing sources and
missing decoders/demuxers/depayloaders; fix error code used when
we're missing an URI handler source; for media types that we are not
handling on purpose at the moment, don't print "don't know how to
handle xyz" messages to the terminal or post missing-plugin
messages on the bus.
* tests/check/elements/playbin.c: (create_playbin),
(GST_START_TEST), (gst_codec_src_uri_get_type),
(gst_codec_src_uri_get_protocols), (gst_codec_src_uri_get_uri),
(gst_codec_src_uri_set_uri), (gst_codec_src_uri_handler_init),
(gst_codec_src_init_type), (gst_codec_src_base_init),
(gst_codec_src_create), (gst_codec_src_class_init),
(gst_codec_src_init), (plugin_init), (playbin_suite):
Add some tests for the missing-plugin stuff.
