This allows the reception of streams that don't exactly match
the codec preferences. In particular, the ssrc in the codec preferences
is local sender SSRC, the other side is expected to send a different SSRC.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2615>
This new signal allows data-channel consumers to configure signal handlers on a
newly created data-channel, before any data or state change has been notified.
The webrtcin unit-tests were refactored to make use of this new signal.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2427>
In preparation for the new element `GstGtkWaylandSink`, move reusable
parts out of `GstWaylandSink` into the already exisiting but very
barebone library.
Notable changes include:
- the `GstWaylandVideo` interface was dropped
- support for `wl-shell` was dropped
- lots of renaming in order to match established naming patterns
- lots of code modernisations, reducing boilerplate
- members were made private wherever possible
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2479>
1) check for right macro name when checking for NICE_VERSION_CHECK
2) if libnice version is 0.1.18.1 this should not satisfy
a NICE_VERSION_CHECK(0,1,19).
Fixes build with libnice 0.1.18.1 subproject checkout.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2499>
WHen bundling, if multiple medias are used with the same media payload, then
each of the fec/rtx/red additions would add a distinct payload. This could
very easily overflow the available payload space.
Instead, track the relationship between the media payload value and
the relevant fec/rtx/red payload values and reuse them whenever
necessary, even when bundling.
e.g.
...
a=group:BUNDLE video0 video1
m=video 9 UDP/SAVPF 96 97
a=mid:video0
a=rtpmap:96 VP8/90000
a=rtpmap:97 rtx/90000
a=fmtp:97 apt=96
...
m=video 9 UDP/SAVPF 96 97
a=mid:video1
a=rtpmap:96 VP8/90000
a=rtpmap:97 rtx/90000
a=fmtp:97 apt=96
...
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2474>
Since both g_value_set_object() and g_weak_ref_get() takes a reference
there will be two new references to the GstWebRTCICE object when there
should be only one. g_value_take_object() has the same functionality as
g_value_set_object() but does not take a reference.
Without this change, the GstWebRTCICE object will be leaked.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2333>
GLib guarantees libintl is always present, using proxy-libintl as
last resort. There is no need to mock gettex API any more.
This fix static build on Windows because G_INTL_STATIC_COMPILATION must
be defined before including libintl.h, and glib does it for us as part
as including glib.h.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2028>
They are part of gst_dep already and we have to make sure to always have
gst_dep. The order in dependencies matters, because it is also the order
in which Meson will set -I args. We want gstreamer's config.h to take
precedence over glib's private config.h when it's a subproject.
While at it, remove useless fallback args for gmodule/gio dependencies,
only gstreamer core needs it.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2031>
g_signal_disconnect*() doesn't stop any existing callbacks from running
which means that if the notify::state callback is in progress in one
thread and the data channel object is finalize()ed in another thread,
then there could be a use-after-free trying lock the data channel
object.
We can't reasonably use a GWeakRef as we don't have a 'parent' object to
free the GWeakRef after the data channel is finalized. This is also
complicated by the fact that the application can hold a reference to the
data channel object that would live beyond the lifetime of webrtcbin
itself.
We solve this by implementing a ghetto weak-ref solution internally with
a list of outstanding data channels.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1664>
Input (sink pads) is the already-ssrc-muxed stream with the relevant rtp
sdes header extensions already applied:
- mid
- stream-id
- repaired-stream-id
Output (src pads) have the pads separated into individual ssrc's as
that's what rtpbin gives us.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1664>
Each rtpbin exposed recv_src pad is now exposed as webrtcbin src_%u pad
now with no meaining applied to the value of %u. Previously this used
to mean the mline in the SDP. If this is is still required, then the
transceiver can be retrieved from the pad and the "mlineindex" property
from the transciever. The "mid" is also retrievable from the
transceiver.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1664>
