Commit graph

6260 commits

Author SHA1 Message Date
Nicola Murino
24d5c9b200 opencv: define CASCADE_DO_CANNY_PRUNING for opencv < 4 2018-11-26 13:29:03 +00:00
Nicola Murino
b0773c6ea7 opencv: make compatible with opencv 4
Closed #826
2018-11-26 13:29:03 +00:00
Nicola Murino
890dbb560f opencv: fix indentation 2018-11-26 13:29:03 +00:00
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
Matthew Waters
14ee6f9d35 webrtc: fix typo in RTCRemoteOutboundRTPStreamStats 2018-11-26 16:21:58 +11:00
Matthew Waters
a42fdbb012 webrtc: add a few comments on bundle and src pad exposure 2018-11-26 16:21:19 +11:00
Matthew Waters
6f91a191de webrtcbin: factor out dtls fingerprint setting 2018-11-26 16:20:02 +11:00
Matthew Waters
3a2566c61f webrtc: remove extra 'pad' from log line 2018-11-26 16:12:03 +11:00
Matthew Waters
5ecca0bb22 webrtc: move some functions to the appropriate files 2018-11-26 16:07:57 +11:00
Harshad Khedkar
9ad618e487 Webrtcbin : Need to use 'host' from gst_uri_get_host s libnice agent expects it
Currently master code of gst1-plugins-bad use plain-string host name while passing it to
libnice agent: nice_agent_set_relay_info() in gstwebrtcice.c while adding turn_server(_add_turn_server).

It is observered that if we don't convert the host parameter by using gst_uri_get_host, it fails in libnice agent(0.1.14-1).

Code does, actually, set the host correctly but while passing params to nice_agent_set_relay_info, it uses incorrect one.

https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/823
2018-11-22 18:47:13 +05:30
Rosen Penev
29d48ce8a1 dtls: Fix compilation without deprecated APIs on OpenSSL 1.1.x 2018-11-14 14:55:37 -08:00
Justin Kim
e76c61d74c cameracalibrate: Fix build error on Mac OSX
It fails to build only on Mac OSX with the following error.

In file included from ../subprojects/gst-plugins-bad/ext/opencv/gstopencv.cpp:45:
../subprojects/gst-plugins-bad/ext/opencv/gstcameracalibrate.h:96:38: error: a space is required between consecutive right angle brackets (use '> >')
  std::vector<std::vector<cv::Point2f>> imagePoints;
                                     ^~
                                     > >
1 error generated.

Fix: #817
2018-11-13 18:23:19 +09:00
Matthew Waters
c78f81f410 build: link against the gnustl dependency on android for c++ plugins 2018-11-12 21:37:07 +11:00
Olivier Crête
75d4ede815 srtpenc: Add support for adding a MKI
https://bugzilla.gnome.org/show_bug.cgi?id=797305
2018-11-09 15:08:05 -05:00
Olivier Crête
46b5756d30 srtpdec: Add support for MKI
Add support for MKIs which allow multiple keys to be used with a single SRTP stream.

https://bugzilla.gnome.org/show_bug.cgi?id=797305
2018-11-09 15:06:27 -05:00
Jan Alexander Steffens (heftig)
84d3f58c4a
dtls: Link against winsock2 on Windows
Should fix the Windows build after
b1509b1047.
2018-11-06 18:48:44 +01:00
Jan Alexander Steffens (heftig)
b1509b1047 dtlsconnection: Print out errno info for syscall errors
As suggested in [the SSL_get_error manpage][1]. Upgrade the message to a
warning if the errno isn't 0 (success). The latter apparently means the
transport encountered an EOF (shutdown) without the shut down handshake
on the (D)TLS level. This happens quite often for otherwise normal DTLS
connections.

[1]: https://www.openssl.org/docs/man1.1.1/man3/SSL_get_error.html
2018-11-06 16:23:50 +00:00
Andreas Frisch
51f0307900 dtls: Properly display all errors/warnings from ERR queue
Print out all errors from the OpenSSL error queue instead of just
looking at the topmost error. Using the callback interface also removes
the need for formatting using a buffer on the stack.
2018-11-06 16:23:50 +00:00
Jan Alexander Steffens (heftig)
7bb6ce352b dtlsconnection: Replace if-else chain with switch
A more idiomatic construct for testing a bunch of integers.
2018-11-06 16:23:50 +00:00
Jan Alexander Steffens (heftig)
49cc7b8091 dtlsconnection: Clear error queue before SSL_do_handshake
As documented on [the SSL_get_error manpage][1] we need to empty the
error queue before making any call that we check with SSL_get_error.

[1]: https://www.openssl.org/docs/man1.1.1/man3/SSL_get_error.html
2018-11-06 16:23:50 +00:00
Jan Alexander Steffens (heftig)
fa92909d0d dtlsconnection: Lower BIO error log level to DEBUG
Periodic SSL_ERROR_SYSCALL errors with errno == 0 seem to be normal
behavior for DTLS connections.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/677
2018-11-06 16:23:50 +00:00
Jan Alexander Steffens (heftig)
a9b9994738 dtlsconnection: Run gst-indent 2018-11-06 16:23:50 +00:00
Sebastian Dröge
eb92865ea1 cccombiner: Add new closed caption combiner element
This is the inverse of ccextractor: it takes a video and closed caption
stream and places the closed caption metas on the video output buffers.
2018-11-06 16:21:42 +00:00
Sebastian Dröge
7ba9493ff1 ccextractor: Handle multiple caption metas per buffer 2018-11-06 16:21:06 +00:00
Tim-Philipp Müller
247cd2113e webrtc: update default libnice options
Uses feature options now.
2018-11-02 20:16:56 +00:00
Sebastian Dröge
7cfe6b0bf3 ccextractor: Support changes of the caption type
https://bugzilla.gnome.org/show_bug.cgi?id=797370
2018-11-02 12:22:15 +02:00
Tim-Philipp Müller
a4cfb1fa14 meson: add 'curl-ssh2' option for curl libssh2 support
https://bugzilla.gnome.org/show_bug.cgi?id=797346
2018-10-28 11:53:42 +00:00
Xavier Claessens
8dfe0dd9ce meson: Use the 'curl' feature option
https://bugzilla.gnome.org/show_bug.cgi?id=797346
2018-10-28 11:51:34 +00:00
Luis de Bethencourt
16c1eee36f webrtcbin: options is a placeholder argument
Make it clear this unused argument is there for planned future use.
2018-10-26 15:15:57 +01:00
Luis de Bethencourt
83b29b813e webrtcbin: ws-semantic is not supported
Don't offer something that isn't supported.
2018-10-26 14:36:47 +01:00
Edward Hervey
4a24739a14 hls: Remove stray '\n' from debug statement 2018-10-22 09:59:24 +02:00
Olivier Crête
a10c46a93a Revert "srtpdec: also check for "roc" in caps"
This reverts commit 73ebdb888e.

This isn't needed and it breaks srtpenc ! srtpdec, specifying the
roll-over counter manually is an advanced feature.

Also revert "srtp: Add "roc" caps field to the gst-launch example"
This reverts commit 67ae35813b.

https://bugzilla.gnome.org/show_bug.cgi?id=765079
2018-10-17 18:34:31 -04:00
Matthew Waters
59f40eb1a0 sctp: link against ws2_32 on windows
ext/sctp/ext@sctp@@gstsctp@sha/sctpassociation.c.obj: In function `receive_cb':
/var/lib/jenkins/workspace/cerbero-cross-mingw32/workdir/sources/windows_x86/gst-plugins-bad-1.0-1.15.0.1/_builddir/../ext/sctp/sctpassociation.c:692: undefined reference to `_imp__ntohl@4'
2018-10-17 21:07:36 +11:00
Mathieu Duponchelle
9f684a2f81 webrtcbin: implement support for group: BUNDLE 2018-10-15 14:17:35 +02:00
Wonchul Lee
51d5db3f47 av1enc: Add to configure image formats
Expanded to support image format to YV12/I422/I444. It's related to the
color bit-depth and profile of the codec. It can make configuring
appropriate profile according to bit-depth and format.

https://bugzilla.gnome.org/show_bug.cgi?id=791674
2018-10-10 15:44:34 -04:00
Wonchul Lee
cc9d65a512 av1enc: fix compliation with removed defines
AOM_FRAME_IS_INVISIBLE is unused and remove in aom_encoder.h

https://aomedia-review.googlesource.com/c/aom/+/70421

https://bugzilla.gnome.org/show_bug.cgi?id=791674
2018-10-10 15:44:34 -04:00
Wonchul Lee
7d73307bc4 av1enc: Add configurations
Added rate control and profile related configuration. The profile
configuration has been copied from vpx.

https://bugzilla.gnome.org/show_bug.cgi?id=791674
2018-10-10 15:44:34 -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
Aleix Conchillo Flaqué
c4fe52395b webrtcbin: start and stop thread when changing state
It might be possible that if we set webrtcbin to the NULL state some
tasks (idle sources) are still executed and they might even freeze. The freeze
is caused because the webrtcbin tasks don't hold a reference to webrtcbin and
if it's last unref inside the idle source itself this will not allow the main
loop to finish because the main loop is waiting on the idle source to finish.

We now start and stop webrtcbin thread when changing states. This will allow
the idle sources to finish properly.

https://bugzilla.gnome.org/show_bug.cgi?id=797251
2018-10-08 13:46:55 +11:00
Matthew Waters
d5da9e080a glvideomixer: fix the default blend modes for unpremultipled alpha
Alpha in GStreamer is unpremultiplied
2018-10-05 01:18:59 +10:00
Matthew Waters
ea355ca0a3 glmixerbin: add gloverlaycompositor to each input stream
Flattens the overlay compositions into the stream before the mixer will
mix them.

https://bugzilla.gnome.org/show_bug.cgi?id=759867
2018-10-04 14:32:14 +10:00
Sebastian Dröge
b5ae66ea1d glstereomix: Fix build after deprecating misnamed GST_TYPE_GL_STEREO_DOWNMIX_MODE_TYPE 2018-10-03 16:00:23 +03:00
Matthew Waters
8e8eb41ddf webrtcdatachannel: take ref of data so it doesn't disappear 2018-09-26 16:01:57 +10:00
Sebastian Dröge
47b69bca12 glmixerbin: "latency" property on aggregator is uint64, not int64
Also update the property description and range with aggregator's values.
2018-09-24 23:26:52 +03:00
Matthew Waters
ce9e2a7b78 glvideomixer: fix constant alpha enum value for constant alpha 2018-09-21 19:45:13 +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
Matthew Waters
cf46d49b1e webrtcbin: functionify dependent element checks 2018-09-21 19:36:52 +10:00
Matthew Waters
f0a4713932 webrtc/stats: rename debug category not to be ice related 2018-09-21 19:36:52 +10:00
Mathieu Duponchelle
01216845c7 sctp elements: avoid assertions on shutdown 2018-09-21 19:36:52 +10:00