With this patch, configure time is identical no matter whether doc is
enabled or not.
The configuration files also now contain explicitly-listed sources with
no wildcards.
For the four libraries where hotdoc needs to use clang to generate the
documentation (as opposed to the rest of the libraries where hotdoc uses
the gir), the script will call pkg-config to determine the appropriate
C flags.
This means a side effect of this patch is that pkg-config files are now
generated for the gstadaptivedemux and gstopencv libraries.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8312>
People might have GST_TRACERS=leaks set in their environment
by default, which will now trigger criticals during the build
when calling g-ir-scanner, because we unset GST_PLUGIN_SYSTEM_PATH
so that the scanner doesn't load any plugins.
Fixes#4093
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8121>
The gst_dep.get_variable('libexecdir') may fail in some scenarios
(e.g. building a module alone inside an uninstalled devenv) and
it shouldn't really be reached in the first place if docs are
disabled via options.
Also to avoid confusing meson messages when cross-compiling or
doing a static build.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7818>
- Align `glib_debug`, `glib_assert` and `glib_checks` options with GLib,
otherwise glib subproject won't inherit their value. Previous names
and values are preserved using Meson's deprecation mechanism.
- Add `extra-checks` and `benchmarks` options in the main project so it
can be inherited in GStreamer subprojects.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1165>
Since commit 4d86f994, when setting an RTSP media both shared and
reusable, streaming cannot be restarted after the first time all the
clients disconnect. That happens because the sockets (unlike
addresses) of GstRTSPStream are not cleared in
gst_rtsp_stream_leave_bin, and on restart sockets and addresses are
not allocated in gst_rtsp_stream_allocate_udp_sockets, and then the
check in create_sender_part fails. Fix this by clearing sockets in
gst_rtsp_stream_leave_bin.
Fixesgstreamer/gst-rtsp-server#113
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6325>
While the suspend modes NONE and PAUSED provided a low startup latency
for connecting clients they did not ensure that streams started on
fresh data.
With this property we can maintain the low startup latency of those
suspend modes while also ensuring that a stream starts on a key unit.
Furthermore, by modifying the value of a new property,
ensure-keyunit-on-start-timeout, it is possible to accept a keyunit of
a certain age but discard it if too much time has passed and instead
force a new keyunit.
Fixes#2443
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4334>
Allow a project to use gstreamer-full as a static library
and link to create a binary without dependencies.
Introduce the option 'gst-full-target-type' to
select the build type, dynamic(default) or static.
In gstreamer-full/static build configuration gstreamer (gst.c)
needs the symbol gst_init_static_plugins which is defined
in gstreamer-full.
All the tests and examples are linking with gstreamer but the
symbol gst_init_static_plugins is only defined in the gstreamer-full
library. gstreamer-full can not be built first as it needs to know what plugins
will be built.
One option would be to build all the examples and tests after
gstreamer-full as the tools.
Disable tools build in subprojects too as it will be built at the end of
build process.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4128>
Previously it was possible that a shared media was just in the process
of being unprepared because the last client disappeared, while another
client retrieved it from the cache and then tried to use it. Unless the
media was reusable this would've then failed unnecessarily.
To avoid this it is necessary to lock the media directly in
gst_rtsp_media_factory_construct() and return a locked media. After
locking the cached media it is necessary to check if the media was ever
unprepared or is actually reusable and based on that either reuse it or
create a new media.
This minimally changes the gst_rtsp_media_factory_construct() API to
always return a locked media, and adds a new
gst_rtsp_media_can_be_shared() function to check if a media can actually
be shared in practice.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4606>
