Commit graph

856 commits

Author SHA1 Message Date
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
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
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
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
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
b606ff16a9 tests: Fix build with MSVC 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
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
Sebastian Dröge
c651741ef0 cccombiner: Add unit test 2018-11-06 16:21:42 +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
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