When a datachannel within a session is removed after proper close,
reference to the error_ignore_bin elements of the datachannel
appsrc/appsink were left in webrtcbin.
This caused the bin-objects to be left and not freed until the whole
webrtc session was terminated. Among other things that includes a thread
from the appsrc.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7675>
Check and generate remote reception statistics from the info stored on
internal sources, as they are stored there when running against newer rtpbin
since MR !7424
This fixes cases where statistics are incomplete when
peers send RR reports from a single remote ssrc, which GStreamer does
when bundling is enabled and other RTP stacks may too.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7425>
This new LCEVC encoder plugin is meant to implement all LCEVC encoder elements.
For now, it only implements the LCEVC H264 encoder (lcevch264enc) element. This
element essentially encodes raw video frames using a specific EIL plugin, and
outputs H264 frames with LCEVC data. Depending on the encoder properties, the
LCEVC data can be either part of the video stream as SEI NAL Units, or attached
to buffers as GstMeta.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7330>
This new element wraps both the base H264 decoder and lcevcdec elements into a
bin so that LCEVC decoding works with auto-plugging elements such as decodebin.
By default, the H264 decoder element with higher rank is used as base decoder,
but any particular H264 decoder can be used by manually setting the base-decoder
property.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7330>
This new LCEVC decoder plugin is meant to implement all LCEVC decoder elements.
For now, it only implements the LCEVC enhancement decoder (lcevcdec) element.
This element essentially enhances raw video frames using the LCEVC metadata
attached to input buffers into a higher resolution frame. The element is only
meant to be used after any base decoder (eg avdec_h264).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7330>
While transforming the internals of waylandsink into a library, the
context type name was accidentally changed, causing an ABI break. Change
it back to its original (as used by the libgstgl), and add support for
the misnamed version as a backward compatibility measure.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7482>
If the pending remote description has an invalid BUNDLE group _parse_bundle()
triggers early return from _create_answer_task(), before ret has been
initialized, so it needs to be checked before attempting to call
gst_sdp_message_copy().
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7423>
webrtcsrc first creates recvonly transceivers with codec-preferences
and expects that after applying a remote description, the
previously created transceivers are used rather than having new
transceivers created.
When pairing webrtcsink + webrtcsrc, the offer sdp from webrtcsink has a media
section with sendonly direction. In !7156, which was implemented following
RFC9429 Section 5.10, we only reuse a unassociated transceiver when applying a
remote description if the media is sendrecv or recvonly, and that caused creation
of new transceivers when applying a remote offer in webrtcsrc, thus losing
information from codec preferences like the RTP extension headers in the
previously created transceivers.
Since the change in !7156 broke existing code from webrtcsrc, relax the condition
for reusing unassociated transceivers and add a test to document this behavior which
wasn't covered by any tests before.
Fixes#3753.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7417>
When multiple streams are bundled on the same transport,
the statistics would end up incorrectly generated,
as each pad would regenerate stats for every ssrc on the
transport, overwriting previous iterations and assigning
bogus media kind and other values to the wrong ssrc.
Fix by making sure each pad only loops and generates
statistics for the one ssrc that pad is receiving / sending.
Add a unit test that the codec kind field in RTP statistics
are now generated correctly.
Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2555
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7338>
When checking for renegotiation against a local offer,
reverse the remote direction in the corresponding answer
to fix falsely not triggering on-negotiation needed when
switching (for example) from local sendrecv -> recvonly
against a peer that answered 'recvonly'.
In the other direction, when the local was the answerer,
renegotiation might trigger when it didn't need to -
whenever the local transceiver direction differs from
the intersected direction we chose. Instead what we want
is to check if the intersected direction we would now
choose differs from what was previously chosen.
This makes the behaviour in both cases match the
behaviour described in
https://www.w3.org/TR/webrtc/#dfn-check-if-negotiation-is-needed
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7303>
Fixes for basic rollback (from have-local-offer or have-remote-offer to
stable). Allow having no SDP attached to the webrtc session description
in that case, and avoid all the transceiver and ICE update logic
normally applied when entering the stable signalling state
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7304>
Fix an inverted condition when checking if sink pad caps match
the codec-preference of an unassociated transceiver, and
fix a condition check for transceiver media kind to
avoid matching sinkpad requests where caps aren't provided
against unassociated transceivers where the caps might
not match later.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7237>
With MR 7156, transceivers and transports are created earlier,
but for sendrecv media we could get `not-linked` errors due to
transportreceivebin not being connected to rtpbin yet when incoming
data arrives.
This condition wasn't being tested in elements_webrtcbin, but could be
reproduced in the webrtcbidirectional example. This commit now also
adds a test for this, so that this doesn't regress anymore.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7294>
According to https://w3c.github.io/webrtc-pc/#set-the-session-description
(steps in 4.6.10.), we should be creating and associating transceivers when
setting session descriptions.
Before this commit, webrtcbin deviated from the spec:
1. Transceivers from sink pads where created when the sink pad was
requested, but not associated after setting local description, only
when signaling is STABLE.
2. Transceivers from remote offers were not created after applying the
the remote description, only when the answer is created, and were then
only associated once signaling is STABLE.
This commit makes webrtcbin follow the spec more closely with regards to
timing of transceivers creation and association.
A unit test is added, checking that the transceivers are created and
associated after every session description is set.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7156>
If a downstream buffer pool is offered, vulkanupload checks its allocation
parameters to honor them. Only adds to usage the TRANSFER bits, which are
required to upload buffers.
Also, fail if the buffer pool cannot be configured with the current parameters.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7219>
None of the symbols in webrtc-audio-coding-1 are marked with
`__declspec(dllexport)`, rendering the library usable only if
it was built with GCC/Clang.
The only fix available (as the pulseaudio copy has not been updated
with Google's upstream) is to ensure the fallback builds statically.
Although this change will also affect webrtcdsp's dependency on
webrtc-audio-processing-1, it does not break its compilation.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6407>
