The mapping between an RTP session and the SDP m= line is not always the
same, especially when BUNDLEing is used.
This causes a failure in a specific case where if when bundling,
if mline 0 is a data channel, and mline 1 an audio/video section,
then retrieving the transceiver at mline 0 (rtp session used) will fail
and cause an assertion.
This fix is actually potentially a regression for cases where the remote
part does not provide the a=ssrc: media level SDP attributes as is now
becoming common, especially when simulcast is involved.
The correct fix actually requires reading out header extensions as used
with bundle for signalling in the actual data, what media and therefore
transceiver is being used.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2467>
When converting from one framerate to another, counters are
reset periodically, however when not converting they never are
and can_genearte_output ends up making overflow-prone calculations
with large values for input_frames and output_frames.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2465>
The GST_ELEMENT_ERROR will call the gst_object_get_path_string and
use gst_object_get_parent to get the full object path name, which
needs to lock the object. But we are already in a locked context and
so this will cause a deadlock, the pipeline can not exit normally.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2451>
Some manifests use the ContentProtection node to store additional information
such as the license server url. Our MPD parser used to process the
ContentProtection node, extracting Playready PSSH boxes. However for other DRM
systems, only the `value` attribute was passed down to the protection event, so
for example, Widevine data was not parsed at all and "Widevine" was passed to
the event, which is not very useful for decryptors that require a PSSH init
data.
Parsing should now be done by decryptors which will receive the entire
ContentProtection XML node as a string. This gives more "freedom" to the
decryptor which can then detect and parse custom nodes as well.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2400>
if there is no pad name, the representation id
was NULL, causing a crash when writing the mpd file.
gst-launch-1.0 videotestsrc num-buffers=900 ! video/x-raw, width=800,
height=600, framerate=30/1 ! x264enc ! video/x-h264, profile=high !
dashsink
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2064>
We can only convert from non-cdp to cdp within the confines of valid cdp
framerates. The existing caps negotiation code was allowing any
framerate to convert to a cdp output which is incorrect and would hit an
assertion later.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2372>
If they're not dropped, they can be blocked in the queue even if it is
leaky in the case where there is a buffer being pushed downstream. Since
in webrtc, it's unlikely that there will be a special allocator to
receive RTP packets, there is almost no downside to just ignoring the
queries.
Also drop queries if they get caught in the pad probe after the queue.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2363>
By using the clocksync inside the dtlssrtpenc, all streams inside a
bundled are synchronized together. This will cause problems if their
buffers are not already arriving synchronized: clocksync would wait for
a buffer on one stream and then buffers from the other stream(s) with
lower timestamps would all be sent out too late.
Placing the clocksync before the rtpbin and rtpfunnel synchronizes each
stream individually and they will be send out more smoothly as a result.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2355>
For those using context from the application which
would be the embedded video case, if the frame callback
is entering at the same time as window is finalizing,
a wayland proxy object would be destroyed twice, leading
the refcout less than zero in the second time, it can
throw an abort() in wayland.
For those top window case, which as a directly connection
to the compositor, they can stop the message queue then
the frame callback won't happen at the same time as the
window is finalizing. It doesn't think it would bother
them about this.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1883>
We were doing our initial "empty" commit on the subsurface instead of the
toplevel surface. As an incidence, we should not have received a configure
event ever, not just on mutter. This fixes the following warning when using
mutter compositor (aka gnome-shell):
waylandsink wlwindow.c:304:gst_wl_window_new_toplevel: The compositor did not send configure event.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2299>
