Fixes the deprecation warning from meson:
DEPRECATION: target sip links against shared module sip, which is incorrect.
This will be an error in the future, so please use shared_library() for sip instead.
If shared_module() was used for sip because it has references to undefined symbols,
use shared_libary() with `override_options: ['b_lundef=false']` instead.
This gives us a better separation of concerns and it will make it a bit easier
to move the sip independent media pieces out of the plugin in the mid to long
term.
This makes running tests harder as we cannot call gst_init() after gst_deinit()
has been called.
This is what the API reference has to say about it at
https://gstreamer.freedesktop.org/documentation/gstreamer/gst.html?gi-language=c#gst_deinit
It is normally not needed to call this function in a normal application as the
resources will automatically be freed when the program terminates. This function
is therefore mostly used by testsuites and other memory profiling tools.
It isn't needed in the implementation either. It was only useful because it
included system headers like sys/types.h and sys/socket.h which we should now
include directly.
This will make it easier to move the media manager into the core sources.
Previously cui_call_get_display_name() would return the ID of the caller if no
contact was found. As this has changed recently the test for got_contact broke.
Shuffles some code around so that the property bindings are at the end.
This allows us to return early if there is a NULL contact (as is the case
for anonymous callers).
This makes sure that a NULL id will return a NULL contact and additionally
asserts that even a NULL contact will give us sensible primary/secondary display
information.
This allows to get rid of any special casing that the users of the
calls_best_match_get_name() and calls_best_match_get_id() had to do previously.
We also allow passing in NULL for *_get_primary_info ()
and *_get_secondary_info () for the anonymous caller case.
The designs for the call details show information on the type of the call:
https://gitlab.gnome.org/Teams/Design/app-mockups/blob/master/calls/calls.png
So f.e. "Cellular", "Matrix WebRTC Video Call".
These properties can potentially also be useful in choosing the mechanism to
use for the audio controls from the call display.
While not every class or interface is properly documented and annotated yet,
adding them to the docs should still be valuable.
Additionally it is a good reminder for ourselves to add some proper description
because things like missing a annotated class description really stand out
currently.
