0a350c610d broke the build by only
building enum types with meson. It also removed gstsrt.c from the list
of sources, causing the plugin to fail to load.
squash! srt: Fix autotools build
gstsrtobject.c: In function ‘gst_srt_object_close’:
gstsrtobject.c:1036:7: error: function called through a non-compatible type [-Werror]
(GDestroyNotify) g_closure_unref);
/usr/include/glib-2.0/glib/gmem.h:121:8: note: in definition of macro ‘g_clear_pointer’
(destroy) (_ptr); \
^~~~~~~
gstsrtobject.c:1038:7: error: function called through a non-compatible type [-Werror]
(GDestroyNotify) g_closure_unref);
/usr/include/glib-2.0/glib/gmem.h:121:8: note: in definition of macro ‘g_clear_pointer’
(destroy) (_ptr); \
^~~~~~~
Arch Linux
gcc 8.2.1 20181127
glib 2.58.2
We have srt{client,server}{src,sink} elements in accordance to the
norm of the connection oriented protocols. However, SRT connection
mode can be changed by uri parameters so it requires an integrated
element to handle the parameters.
fix: #740
As a side-effect we can now actually store the line offset in the
line21dec element, and have to perform fewer transformations in the
decklink elements (which were also buggy as they assumed a single byte
triplet per meta).
When waylandsink is used on some other thread than the main wayland
client thread, the waylandsink implementation is vulnerable to a
condition related to registry and surface events which handled in
seperated event queue.
The race that may happen is that after a proxy is created, but
before the queue is set, events meant to be emitted via the yet to
set queue may already have been queued on the wrong queue.
Wayland 1.11 introduced new API that allows creating a proxy
wrappper which can help to avoid this race condition.
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.
The wpe element is used to produce a video texture representing a web page
rendered off-screen by WPE. This element can be used to overlay HTML on top of
another video stream for instance.
The latter is going away in libfdk-aac 2.0.0. Instead, MPEG-style output
is always non-interleaved and WAV-style output is always interleaved.
Earlier libfdk-aac also defaults interleaving accordingly.
Since our reordering looks at the associated PCE indices instead of the
actual channel order, we're agnostic to the mapping.
For https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/825
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
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
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
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.
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
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'
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
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!
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.
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
