Commit graph

1273 commits

Author SHA1 Message Date
Andre Guedes
c427fd1aec tests: Add AVTP source tests
This patch adds test cases for the AVTP source element. For now, only
properties get() and set() are covered.
2019-07-03 09:59:35 -07:00
Andre Guedes
e0deddbcf6 tests: Add AVTP sink tests
This patch adds test cases for the AVTP sink element. For now, only
properties get() and set() are covered.
2019-07-03 09:59:35 -07:00
Andre Guedes
82b6b0faa7 tests: Add AAF depayloader tests
This patch adds test cases for the AAF depayloader element covering the
basic functionalities.
2019-07-03 09:59:35 -07:00
Andre Guedes
e09470fac8 tests: Add AAF payloader tests
This patch adds the infrastructure to test AVTP plugin elements. It also
adds a test case to check avtpaafpay element basic functionality. The
test consists in setting the element sink caps and properties, and
verifying if the output buffer is set as expected.
2019-07-03 09:59:35 -07:00
Matthew Waters
3c164f4de2 tests/vkcolorconvert: remove extra instance/device creation
It's unnecessary.
2019-06-24 16:23:29 +10:00
Matthew Waters
0cb416db11 vkbuffermemory: report requested size of the memory
Rather than using Vulkan's much larger aligned sizes. Fixes multi-planer
video with the GstVideoFrame API.
2019-06-20 01:41:56 +10:00
Matthew Waters
5363b30f6c vulkan: add a color conversion element
Currently converts between all 4-component RGBA/RGBx formats.
2019-06-20 01:41:56 +10:00
Seungha Yang
7b8d198712 tests: hls: Add a test case for EXT-X-MAP tag
https://bugzilla.gnome.org/show_bug.cgi?id=776928
2019-06-18 07:14:28 +00:00
Seungha Yang
e779160434 tests: Enable hls m3u8 unit test with meson build 2019-06-18 07:14:28 +00:00
Seungha Yang
4e6e6ad0f9 tests: insertbin: Don't use duplicated variable name
../subprojects/gst-plugins-bad/tests/check/libs/insertbin.c(46): warning C4273:
  'mutex': ...

The "mutex" was declared in gstcheck.h already
2019-06-18 06:32:14 +00:00
Tim-Philipp Müller
4ff14c0e74 Remove VDPAU plugin
It's been replaced by NVENC/NVDEC and even NVIDIA doesn't
support VDPAU any longer and hasn't for quite some time.

The plugin has been unmaintained and unsupported for a very
long time, and given the track record over the last 10 years
it seems highly unlikely anyone is going to make it work well,
not to mention adding plumbing for proper zero-copy or
gst-gl integration.

Closes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/828
2019-06-06 18:45:09 +01:00
Matthew Waters
e2eea3b3ee tests/vulkan: allow instance creation failures
Usually means that there is no vulkan implementation available.

We skip tests in that case
2019-06-04 09:03:44 +00:00
Matthew Waters
32bde87551 test/vulkan: add simple memory test 2019-06-04 09:03:44 +00:00
Marc Leeman
3ef737605a rtpmanagerbad: add RTP streaming elements
This is a re-implementation of the RTP elements that are submitted in
2013 to handle RTP streams. The elements handle a correct connection
for the bi-directional use of the RTCP sockets.

https://bugzilla.gnome.org/show_bug.cgi?id=703111

The rtpsink and rtpsrc elements add an URI interface so that streams
can be decoded with decodebin using the rtp:// interface.

The code can be used as follows

```
gst-launch-1.0 videotestsrc ! x264enc ! rtph264pay config-interval=3 ! rtpsink uri=rtp://239.1.1.1:1234

gst-launch-1.0 videotestsrc ! x264enc ! rtph264pay config-interval=1 ! rtpsink uri=rtp://239.1.2.3:5000
gst-launch-1.0 rtpsrc uri=rtp://239.1.2.3:5000?encoding-name=H264 ! rtph264depay ! avdec_h264 ! videoconvert ! xvimagesink

gst-launch-1.0 videotestsrc ! avenc_mpeg4 ! rtpmp4vpay config-interval=1 ! rtpsink uri=rtp://239.1.2.3:5000
gst-launch-1.0 rtpsrc uri=rtp://239.1.2.3:5000?encoding-name=MP4V-ES ! rtpmp4vdepay ! avdec_mpeg4 ! videoconvert ! xvimagesink
```

rtpmanagerbad: add pkg-config
rtpmanagerbad: Rtp should be uppercase
rtpmanagerbad: add G_OS_WIN32 for shielding unix headers
rtpmanagerbad: remove Since from documentation
rtpmanagerbad: rename lib name from nrtp to rtpmanagerbad
rtpmanagerbad: sync meson.build with other modules
rtpmanagerbad: add Makefile.am
rtpmanagerbad: use GstElement to count pads
rtpmanagerbad: use gst_bin_set_suppressed_flags
rtpmanagerbad: check element creation
rtpmanagerbad: post message when trying to access missing rtpbin
rtpmanagerbad: return FALSE with g_return tests
rtpmanagerbad: use gsocket multicast check
rtpmanagerbad: use gst_caps_new_empty_simple iso gst_caps_from_string
rtpmanagerbad: sync with gstrtppayloads.h
rtpmanagerbad: correct media type X-GST
rtpmanagerbad: test if a compatible pad was found
rtpmanagerbad: remove evil copy of GstRTPPayloadInfo
rtpmanagerbad: add gio_dep to meson
rtpmanagerbad: revert to old glib boilerplate

GStreamer 1.16 does not yet support the newer GLib templates, so revert.

rtpmanagerbad: return GST_STATE_CHANGE_NO_PREROLL for live sources

for live sources, NO_PREROLL should be returned for PLAYING->PAUSED and
READY->PAUSED transitions.

rtpmanagerbad: use GstElement pad counting
rtpmanagerbad: just use template name to request pad
rtpmanagerbad: remove commented code
rtpmanagerbad: use funnel to send multiple streams on one socket
rtpmanagerbad: avoid beaches

beaches should only be used during the summer, so rewrite the code to
return explicitly and avoid beaches during the winter.

rtpmanagerbad: add copyright to test code
rtpmanagerbad: g_free is NULL safe
rtpmanagerbad: do not trace rtpbin
rtpmanagerbad: return NULL explitly
rtpmanagerbad: warn when data port is not even

According to RFC 3550, RTP data should be sent on even ports, while RTCP
is sent on the following odd port.

rtpmanagerbad: document port allocation in rtpsink/src
rtpmanagerbad: improve uri description
rtpmanagerbad: add comment re-use socket
rtpmanagerbad: rename gst_object_set_properties_from_uri_query
rtpmanagerbad: loan prop/val setter from rist
rtpmanagerbad: rtpsrc: fix unitialised pointer
rtpmanagerbad: fix silly typo
rtpmanagerbad: test for empty key/value
rtpmanagerbad: rtpsrc: deprecate ssrc collision to INFO
rtpmanagerbad: sync debug with rist
rtpmanagerbad: small strings allocated on stack
rtpmanagerbad: correct rename
rtpmanagerbad: add locking on prop setters/getters

Locking is added because the URI allows to access the properties too.

rtpmanagerbad: allow for RTCP through NAT
rtpmanagerbad: move gio to header file
rtpmanagerbad: free small strings too
rtpmanagerbad: ttl_mc for ttl on dynudpsink
rtpmanagerbad: add comments on the URI registered
rtpmanagerbad: correct macro after file rename
rtpmanagerbad: code style
rtpmanagerbad: handle wrong URIs in setter
rtpmanagerbad: nit URI notation correction

In an URI, the first key/value pair should not have an ampersand, the
parser did not die though.
2019-06-03 20:08:23 +00:00
Alex Ashley
015566daec tests/dash_mpd: take account of Period start in expected timestamps
The start of each segment is relative to the Period start, minus
the presentation time offset.

As specified in section 5.3.9.6 of the MPEG DASH specification:
	The value of the @t attribute minus the value of the
	@presentationTimeOffset specifies the MPD start time of
	the first Segment in the series.

Several tests use a Period@start value of 10 seconds, which either
needs to be taken into account when calculating expected timestamps
or have that attribute removed.

This commit uses a mix of updating the timestamps and removing the
start attribute, so that both the case of its presence and absence
is tested.
2019-06-01 21:25:33 +00:00
Alex Ashley
a11f7ed924 dashdemux: include both Period start and presentationTimeOffset in segment start
The start of each segment is relative to the Period start, minus
the presentation time offset.

As specified in section 5.3.9.6 of the MPEG DASH specification:
    The value of the @t attribute minus the value of the
    @presentationTimeOffset specifies the MPD start time of
    the first Segment in the series.

dashdemux was not taking account of presentationTimeOffset and in
some methods was not taking into account the Period start time.
This commit modifies the segment->start value to always be
relative to the MPD start time (zero for VOD,
availabilityStartTime for live streams). This makes all uses of
the segment list consistent.

Fixes #841
2019-06-01 21:25:33 +00:00
Matthew Waters
62cc5e51d1 tests/webrtc: wait until the SDP has been set before continuing
If we renegotiate, then it is currently possible for an added stream to
be added to webrtcbin before the SDP is complete.  This causes an
internal inconsistency as there is a 'pending sink transceiver' without
a corresponding media section in the sdp.  It also does not have an
associated transport stream and will fail in _connect_input_stream().
2019-05-30 21:33:09 +10:00
Matthew Waters
979daea7f2 tests/webrtc: fix racy test with a prenegotiated data channel
If both data channels become ready simultaneously, then the two integer
read-add-update cycles can execute concurrently and only ever increment
once instead of the required twice.  Use an atomic add instead.
2019-05-30 21:33:09 +10:00
Matthew Waters
177aa22bcd webrtc: Initial support for stream addition/removal
Limitations:
- No transport changes at all (ICE, DTLS)
- Codec changes are untested and probably don't work
- Stream removal doesn't remove transports (i.e. non-bundled transports
  will stay around until webrtcbin is shutdown)
- Unified Plan SDP only. No Plan-B support.
2019-05-30 21:33:09 +10:00
Matthew Waters
015cb75f66 tests/webrtc: a couple of debug/error string fixes 2019-05-30 21:33:09 +10:00
Matthew Waters
be35735989 tests/webrtc: rewrite bundle checks for separate validate_sdp passes
Improves reusability
2019-05-30 21:33:09 +10:00
Matthew Waters
2bb1fde47c tests/webrtc: add helper for getting the offer/answer element 2019-05-30 21:33:09 +10:00
Matthew Waters
b48e2947bf tests/webrtc: only check audio/video for direction attributes 2019-05-30 21:33:09 +10:00
Matthew Waters
bd92b2f7c4 webrtc: fix answer creation with multiple streams and similar caps 2019-05-30 21:26:46 +10:00
Matthew Waters
ebb9c3c298 tests/webrtc: factor out sdp validation into a single function 2019-05-30 21:26:46 +10:00
Matthew Waters
eb79f95bf8 tests/webrtc: validate number of sdp media using validate_sdp 2019-05-30 21:26:46 +10:00
Matthew Waters
7e1cdbfd4d tests/webrtc: allow multiple validation functions 2019-05-30 21:26:46 +10:00
Matthew Waters
120a40cf25 tests/webrtc: test that duplicate negotiations succeed 2019-05-30 21:26:46 +10:00
Mathieu Duponchelle
a1cadd11b8 mpegtsmux: aggregator port 2019-05-19 19:40:48 +00:00
Nirbheek Chauhan
8388a031ab meson: Remove redundant env set operation
This is immediately overriden 4 lines later, and starting with Meso
0.50.1 this will lead to a noisy warning.
2019-04-13 11:34:22 +05:30
Tim-Philipp Müller
a94f4064cb tests: h264parse: add minimal unit test for closed caption SEI parsing 2019-04-08 19:21:34 +01:00
Seungha Yang
231c76b0ce tests: Add nvenc unit test 2019-03-10 13:58:38 +09:00
Tim-Philipp Müller
6dad8b7c78 closedcaption: fix leak in line21 decoder
Destroy old decoder instance when shutting down the
element, and also in case the input format changes.

Fix harness/pipeline leak in unit test.
2019-03-08 01:18:39 +00:00
Tim-Philipp Müller
2ab238a4d0 tests: fix bus leak in hls_demux testSeek unit test
We add the signal watch in testSeekPreTestCallback so
remove it in testSeekPostTestCallback and not deep inside
some if clause in some other callback somewhere.
2019-03-08 01:18:39 +00:00
Tim-Philipp Müller
5b68e12d01 .gitignore more things 2019-03-08 01:17:40 +00:00
Mathieu Duponchelle
55abe39d76 line21: fix tests build with autotools 2019-03-06 15:17:24 +01:00
Mathieu Duponchelle
156865541f closedcaption: add line21 encoder
This element acts as a counterpart of line21encoder.

Also adds a simple test validating each element using the
other.
2019-03-06 11:29:20 +01:00
Alex Ashley
d2d912f34a tests: curlhttpsrc: add unit tests
Based upon the souphttpsrc tests, add unit tests for the curlhttpsrc
element. The souphttpsrc tests are able to use an HTTP server that
is provided as part of the soup library. This does not exist in the
curl library, therefore these tests provide a very simple HTTP server
using the GIO library.

These curlhttpsrc tests contain one new test that does not come from
the souphttpsrc tests. The test_multiple_http_requests test tries to
reproduce the way in which GstAdaptiveDemux makes use of URI source
elements. GstAdaptiveDemux creates a bin with the httpsrc element
and a queue element and sets the locked state of that bin to TRUE,
so that it does not follow the state transitions of its parent. It
then moves this bin to the PLAYING state to start each download and
back to READY when the download completes.
2019-02-19 17:55:12 +00:00
Bastien Nocera
2d5ed45913 vcdsrc: Remove unusable VCD source
The VCD source was ported in 2014 (commit 89eb1e9), but the necessary
"cdxaparse" plugin, which is used to "Parse a .dat file (VCD) into
raw mpeg1" was never ported.

This means that the probable main user for the feature, totem, hasn't
actually been able to play back VCDs, since 2012, when it switched to
using GStreamer 1.0.

Note that even if cdxaparse was finally ported, a lot of work would
still be necessary before it is considered usable. Notably, it is
missing disc image support [1] and some VCDs just cannot be opened for
reading [2].

[1]: https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/898
[2]: https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/899
2019-02-18 15:39:43 +01:00
Nirbheek Chauhan
572a283b4a meson: orc-test is not required
This is especially never available on iOS.
2019-01-31 15:29:24 +05:30
Seungha Yang
a9c61e467a meson: Add support orc fallback
Allow fallback to orc subproject if any, and add missing orc version check.
Additionally 'dependencies' keyword is removed from find_library,
because it's invalid keyword for find_library.
2019-01-30 19:55:05 +09:00
Tim-Philipp Müller
7ea8ec8929 meson: enable tests for orc code 2019-01-16 01:03:40 +00:00
Seungha Yang
82b921e1a4 tests: Enable more tests on Windows 2018-12-30 22:34:43 +09:00
Seungha Yang
33f867236b tests: Remove compositor test
It's moved to -base and no more part of -bad
2018-12-30 22:34:43 +09:00
Seungha Yang
8cd89b2cdd tests: Remove pointless unistd include 2018-12-30 22:34:43 +09:00
Tim-Philipp Müller
e42efbccb1 Remove compositor plugin which was moved to -base
https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/issues/138
2018-12-27 15:31:58 +01:00
Sebastian Dröge
fcceb37bf2 ccconverter: Add unit test 2018-12-19 00:20:26 +02:00
Seungha Yang
fa3ae13250 tests: Re-enable isoff test on Windows
... and drop libxml2 dependency since it never be used by
both test and gstisoff.
2018-12-18 16:48:17 +00:00
Seungha Yang
b606ff16a9 tests: Fix build with MSVC 2018-12-18 16:48:17 +00:00
Seungha Yang
1b0b2c37cc meson: Prefer to use join_paths() over '/'
... to avoid mixing '/' and '\' in a path string on Windows.
2018-12-18 16:48:17 +00:00
Seungha Yang
f74ad82ddc tests: Enable unit test on Windows
Allow run some unit tests on Windows.
* Add dependency explicitly for some test cases, otherwise plugins couldn't be
  loaded on uninstalled environment of Windows.
* Add missing GST_PLUGIN_LOADING_WHITELIST on meson build.
2018-12-18 16:48:17 +00:00
Tim-Philipp Müller
fa9697002a tests: mpegvideoparse: add unit test for CEA-708 closed captions extraction 2018-12-11 13:56:06 +00:00
Sebastian Dröge
fed05fcdf2 mpegtsmux: Change unit test to not generate zero-sized buffers ever 2018-12-10 14:20:14 +00:00
Sebastian Dröge
98b2c705c4 ccextractor: Copy over timecode meta from the input buffers to the outgoing caption buffers
Formats like SCC and MCC work based on timecodes so ideally we pass
through the timecodes when writing them.
2018-12-06 16:06:05 +00:00
Sebastian Dröge
f0571a94fb ccextractor: Include framerate in the closedcaption caps
It depends on the framerate how many cc_data byte pairs are allowed per
frame, and the framerate is also needed for converting into the CDP or
MCC format as the framerate is part of the header metadata.
2018-12-06 16:06:05 +00:00
Jordan Petridis
1f562870ee Run gst-indent through the files
This is required before we enabled an indent test in the CI.

https://gitlab.freedesktop.org/gstreamer/gstreamer-project/issues/33
2018-11-28 14:18:26 +00:00
Matthew Waters
65c9c4b406 tests/webrtc: fix tests for no libnice
webrtcbin will fail the state change to READY when libnice elements are
not available.
2018-11-26 17:13:08 +11:00
Matthew Waters
57a006d8a5 tests/webrtc: use the existing functions in the plugin
Instead of redefining our own, use the function implementations in
webrtcsdp.c and utils.c
2018-11-26 17:13:08 +11:00
Olivier Crête
f401f4ca32 srtp: Add unit tests for MKI support
Both as a srtpenc ! srtpdec pipeline and as a test that uses
fake packets for multiple mixed MKIs.

https://bugzilla.gnome.org/show_bug.cgi?id=797305
2018-11-09 15:08:05 -05:00
Seungha Yang
4e5894fa3d tests: h265parser: Add test parsing nonzero vps_max_layer_id in VPS 2018-11-07 16:13:44 +09:00
Sebastian Dröge
c651741ef0 cccombiner: Add unit test 2018-11-06 16:21:42 +00:00
Nicolas Dufresne
95c99aa0a7 h265parser: Add same parsing test as for H264
This adds the same test as found in H264 test.
2018-11-06 07:26:03 +00:00
Nicolas Dufresne
1df4562077 h264parser: Add unit test parsing 5 valid bytes of a NAL
In the case 5 valid bytes of a slice NAL, the parser should return
NO_NAL_END.
2018-11-06 07:26:03 +00:00
Sebastian Dröge
fca02f3dac ccextractor: Add unit test
https://bugzilla.gnome.org/show_bug.cgi?id=797370
2018-11-02 12:22:15 +02:00
Edward Hervey
08d1e817d0 check: Disable mssdemux unit test for now
This is for the same reason as the dash tests. This should ideally
be converted to gst-validate tests. These tests randomly timeout also
due to the tests doing seeks from the streaming thread (sic).
2018-10-27 17:59:49 +02:00
Edward Hervey
2f7d87a7d1 tests: Disable most gstplayer tests for now
https://bugzilla.gnome.org/show_bug.cgi?id=787374
2018-10-27 13:37:38 +02:00
Mathieu Duponchelle
9f684a2f81 webrtcbin: implement support for group: BUNDLE 2018-10-15 14:17:35 +02:00
Josep Torra
3f83205399 tests: shm: add an end to end test for live reception
Add a unit test to exchange data between two pipelines with shmsrc
acting as a live source.

https://bugzilla.gnome.org/show_bug.cgi?id=797203
2018-10-10 13:42:05 -04:00
Matthew Waters
21bf3a35ac webrtc/datachannel: fix support for prenegotiated channels
With prenegotiated channels, the data-channel protocol is not used and
instead the channel's negotiation is intended to be performed out of band in
some application-specific manner.

Comes with test!
2018-10-09 02:38:14 +11:00
Matthew Waters
7bf18ad258 webrtc: start in the closed state
This means that we will reject all operations before we've transitioned
into READY.

This also fixes the tests using the default GMainContext in the NULL
state instead of the webrtcbin internal GMainContext and thread.  Also
removes a potential ordering race where on the element transitioning to
READY, an operations could have been queued on two different threads and
removing a guarentee on operation ordering.
2018-10-08 21:56:31 +11:00
Matthew Waters
faf212cd3a tests/webrtcbin: add a couple more dependant element checks
Without the sctp elements, all the data channel tests will fail
2018-10-04 16:44:55 +10:00
Matthew Waters
07e9374eff webrtcbin: add support for data channels based on SCTP
Mostly follows the W3C specification
https://www.w3.org/TR/webrtc/#peer-to-peer-data-api

With contributions from:
Mathieu Duponchelle <mathieu@centricular.com>

https://bugzilla.gnome.org/show_bug.cgi?id=794351
2018-09-21 19:45:12 +10:00
Vivia Nikolaidou
ff952374b5 avwait: Start video and audio together if audio starts late
Also add test to meson

https://bugzilla.gnome.org/show_bug.cgi?id=796977
2018-08-17 14:57:36 +03:00
Víctor Manuel Jáquez Leal
b2d5c1ed3c check: update gitignore 2018-08-14 13:53:00 +02:00
George Kiagiadakis
8507730351 tests/check/Makefile.am: add GST_AUDIO_CFLAGS also on the planar audio adapter unit test 2018-08-03 15:24:03 +03:00
George Kiagiadakis
9cf58eb3e4 libs: audio: add new GstPlanarAudioAdapter class
This is a GstAdapter, but for planar audio buffers.

https://bugzilla.gnome.org/show_bug.cgi?id=793605
2018-08-03 13:20:02 +03:00
Nirbheek Chauhan
b55dfb5313 Add feature options for almost all plugins
The only plugins remaining are those that haven't been ported to Meson
yet, and msdk. Also, the tests are still automagic.

https://bugzilla.gnome.org/show_bug.cgi?id=795107
2018-07-27 19:04:38 +05:30
Vivia Nikolaidou
854baf4fdb avwait: Add recording property
It works like a valve in front of the actual avwait. When recording ==
TRUE, other rules are then examined. When recording == FALSE, nothing is
passing through.

https://bugzilla.gnome.org/show_bug.cgi?id=796836
2018-07-24 13:20:59 +03:00
Jan Schmidt
0af199e25e webrtc: Add some ssrc to caps in the unit test.
The unit test uses incomplete caps to test webrtcbin,
causing some weirdness generating RTX stream mappings.

https://bugzilla.gnome.org/show_bug.cgi?id=796810
2018-07-15 23:07:37 +10:00
Sebastian Dröge
be6afc6a90 tests: Make sure to provide the KEY_UNIT flag always together with the SNAP flags in seeks
SNAP flags alone make no sense and nowadays cause warnings.
2018-06-18 12:18:07 +03:00
Edward Hervey
58070fa12c gitignore: Add h265parser test 2018-06-01 08:20:21 +02:00
Mathieu Duponchelle
5c450c5992 webrtcbin: implement support for FEC and RTX
https://bugzilla.gnome.org/show_bug.cgi?id=795044
2018-05-09 14:46:14 +02:00
Nirbheek Chauhan
feac77ace3 meson: Update option names to omit disable_ and with- prefixes
Also yield common options to the outer project (gst-build in our case)
so that they don't have to be set manually.
2018-05-05 20:06:39 +05:30
Sebastian Dröge
10b7b13732 videoaggregator: Rename ignore-eos pad property to repeat-after-eos
What it does is to repeat the last frame forever after EOS, it does not
literally ignore EOS.
2018-05-04 16:46:00 +02:00
Tim-Philipp Müller
c2bcc2711a meson: fix build when msdk is not found 2018-03-09 23:59:16 +00:00
Wang,Fei
1192a598ee tests: add msdkh264enc element unit test
https://bugzilla.gnome.org/show_bug.cgi?id=793236
2018-03-09 10:48:11 -09:00
Tim-Philipp Müller
fd3bb44031 tests: player: disable broken test_play_media_info test
Until someone can be bothered to debug + fix it.

https://bugzilla.gnome.org/show_bug.cgi?id=787372
https://bugzilla.gnome.org/show_bug.cgi?id=787374
2018-03-05 19:37:52 +00:00
Guillaume Desmottes
6dd997541c h265parser: allow partial matching on range extension profile
Best to return a valid profiles rather than no profile if bitstream uses
a not standard profile.

https://bugzilla.gnome.org/show_bug.cgi?id=793876
2018-03-05 13:22:12 -05:00
Guillaume Desmottes
9f25fcdfc9 h265parse: add support for 'Format range extensions profiles'
Those profiles have been introduced in version 2 of the HEVC spec
(A.3.5).

https://bugzilla.gnome.org/show_bug.cgi?id=793876
2018-03-05 13:22:12 -05:00
Guillaume Desmottes
d252f503fc h265parser: decouple GstH265Profile and GstH265ProfileIDC
We used to have the same enum to represent H265 profiles and idc values.
Those are no longer the same with extension profiles defined from
version 2 of the spec.
Split those enums so the semantic of each is clearer and we'll be able
to add extension profiles to GstH265Profile.

Also add gst_h265_profile_tier_level_get_profile() to retrieve the
GstH265Profile from the GstH265ProfileTierLevel. It will be used to
implement the detection of extension profiles.

https://bugzilla.gnome.org/show_bug.cgi?id=793876
2018-03-05 13:19:42 -05:00
Tim-Philipp Müller
a0cc9d1af0 Remove schroedinger plugin
No upstream (website disappeared), no maintainer, and
pretty much a fringe format anyway.

https://bugzilla.gnome.org/show_bug.cgi?id=776215
2018-03-01 15:24:39 +00:00
Tim-Philipp Müller
c180f8ffed audiomixer: remove, moved to -base
https://bugzilla.gnome.org/show_bug.cgi?id=791218
2018-02-13 00:37:35 +00:00
Matthew Waters
4ef2a20a83 dtls: add meson definition for recently added test 2018-02-08 15:29:56 +11:00
Matthew Waters
18156a76fb tests: add a test for the dtls{enc,dec} elements 2018-02-05 17:44:33 +11:00
Matthew Waters
1894293d63 webrtcbin: an element that handles the transport aspects of webrtc connections
SDP's are generated and consumed according to the W3C PeerConnection API
available from https://www.w3.org/TR/webrtc/

The SDP is either created initially from the connected
sink pads/attached transceivers as in the case of generating an offer or
intersected with the connected sink pads/attached transceivers as in
the case for creating an answer.  In both cases, the rtp payloaded streams
sent by the peer are exposed as separate src pads.

The implementation supports trickle ICE, RTCP muxing, reduced size RTCP.

With contributions from:
Nirbheek Chauhan <nirbheek@centricular.com>
Mathieu Duponchelle <mathieu@centricular.com>
Edward Hervey <edward@centricular.com>

https://bugzilla.gnome.org/show_bug.cgi?id=792523
2018-02-02 15:02:21 +11:00
Tim-Philipp Müller
92400a3b69 tests: camerabin: skip checks if we don't have encoders for the default format
https://bugzilla.gnome.org/show_bug.cgi?id=767863
2018-01-20 15:49:35 +00:00
Tim-Philipp Müller
e62eb34bfc tests: ipcpipeline: fix struct initialisation warnings with clang
Use init macros to avoid gst-indent making a mess of it.

https://bugzilla.gnome.org/show_bug.cgi?id=787980
2018-01-17 12:06:29 +00:00
Edward Hervey
5d8d1d359c check: Update gitignore 2018-01-11 11:49:06 +01:00
Mathieu Duponchelle
536cb12577 audioaggregator: implement input conversion
https://bugzilla.gnome.org/show_bug.cgi?id=786344
2017-12-19 23:39:37 +01:00
Tim-Philipp Müller
06e4403fdb gl: update plugins to use GstGL from -base 2017-12-19 12:02:31 +00:00