This is the first step in getting rid of the requirement to have the codec set
during object construction. The goal is to have pipelines prepared in advance so
that the codec can be plugged in once negotiation is complete.
Having the pipelines prepared in advance let's us grab allocated local ports of
udpsrc elements for RTP and RTCP instead of setting those and hoping they're not
yet in use.
They are phased out in favour of their newly introduced ui-call-* pendants.
This was done to have a better separation of concerns and allows for some
cleanup in CallsCall.
Closes#397
This function is used in the activate callback for the per protocol dial actions
to choose the correct origin to place a call from. If an origin cannot be found
it will return NULL which will lead to the fallback "app.dial" action being
invoked.
The id property will be used to keep track of which origin was used for a call,
so that we can default to reusing the same origin when placing a call from the
history.
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.
This was handled explicitly in the Call window.
By changing the logic to delay the emission of "ui-call-removed" we make sure
that the Call UI and the exported DBus object is consistent.
We also need to change the test cases to use run a GMainLoop because we now have
to wait until signal comes in.
This "ui-active" property should be set to FALSE for new incoming calls and set
to TRUE after a short delay of 20ms. This property is used to delay showing the
call display and starting to ring.
This delay gives DBus clients like
phosh-antispam (https://gitlab.com/kop316/phosh-antispam)
some time to make a decision whether a call should be hung up or not and defer
starting to ring and showing the display only afterwards.
When introducing the per protocol dial actions we started setting the widget
self->target as the action target instead of the target property of
self->record.
Bizzarely we only sometimes got greeted with a critical:
11:30:20.5184 GLib[887977]:CRITICAL: g_variant_new_string: assertion 'g_utf8_validate (string, -1, NULL)' failed
Thread 1 "gnome-calls" received signal SIGTRAP, Trace/breakpoint trap.
g_logv (log_domain=0x7ffff7ea400e "GLib", log_level=G_LOG_LEVEL_CRITICAL, format=<optimized out>, args=<optimized out>) at ../../../glib/gmessages.c:1417
1417 ../../../glib/gmessages.c: No such file or directory.
(gdb) bt
at ../../../glib/gmessages.c:2898
Fixes: 687dd7ff383476523ae0a8a38879a171cb0b9a5a
Closes#398
By connecting to the "ui-call-added" and "ui-call-removed" signals instead of
"call-add" and "call-remove".
This allows us to shave off a few lines of code.
This is part of a refactoring which should help separating concerns between
CallsCall and CallsUiCallData better.
To accomplish this we want to stop passing around CallsCall objects and instead
pass CallsUiCallData objects.
In this commit we added new signals instead of changing the existing ones
because that results in a lot of churn. It allows to do the switch per module
allowing smaller individual patches and a more manageable patchset.
