We get loads of warnings when parsing videos from users:
gsth264parser.c:1115:gst_h264_parser_parse_user_data_unregistered: No more remaining payload data to store
gsth264parse.c:646:gst_h264_parse_process_sei:<h264parse0> failed to parse one or more SEI message
Those are raised because of unregistered SEI without user data.
The spec does not explicitly state that unregistered SEI needs to have
data and I suppose the UUID by itself can carry valuable information.
FFmpeg also parses and exposes such SEI so there is no reason for us no
too as well.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7931>
It creates a new structure for passing the codec quality structure at _start(),
where it will be filled. The quality level can be set or changed according
encoder limits.
Later the quality level will be set at _update_session_parameters() and at each
frame encoding. That's why it has to be set at _start().
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8007>
The algorithm for generating the current slot index is a simple round robin,
nonetheless it's not assured that the next slot index it's not still used by a
still living encode picture.
This new way holds an array with the still living encode pictures and the next
slot index looks for a released index in the array.
Its downside is deallocating a picture need to be removed from the array, so the
helper has to be passed to the uninit() function
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8007>
In GStreamer that buffer information is decoupled, holding other structures to
describe the stream: GstCaps. So, to keep the GStreamer design this patch
removes these information from GstVulkanEncoderPicture and pass to
gst_vulkan_encoder_encode() a pointer to GstVideoInfo.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8007>
That's the number of references that gst_vulkan_encoder_encode() receives to
process, so it has to go as a parameter, because it's part of the reference
list, not of the picture.
This commit also modified unit tests accordingly.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8007>
The structure already stored the generic video capabilities and the specific
codec capabilities both for encoding an decoding. The generic decoder
capabilities weren't stored because it was only used internally in the decoder
helper object. Nonetheless, for the encoder, the elements will need the generic
encoder capabilities to configure the encoding. That's why it's required to
expose it as part of GstVulkanVideoCapabilities. And the generic decoder is
included for the sake of symmetry.
While updating the API vkvideoencodeh265 test got some code-style fixes.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8007>
The diff between compared timestamps might be outside the gint range
resulting in wrong sorting results. This patch corrects that by
comparing the timestamps and then returning -1, 0 or 1 depending on the
result.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7726>
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>
Don't reuse the same stats state structure across multiple
get-stats calls. Make each callback take a copy of the
non-changing fields it needs and use a local working copy
to avoid crashing.
Fixes problems with the unit test crashing sometimes for the
unit test introduced in MR !7338
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7387>
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>
Prevent the default webrtc test machinery from attempting to
create and set an answer when we're just testing rollback
of the offers. Add some locking / waiting to ensure the test
is complete before exiting.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7365>
Use pattern matching against expected error strings that
might include internal element names, where the names
are default assigned with incrementing integers. When running
with CK_FORK=no, there may have been previous tests that
ran in the same process and incremented the counters more
than when running in the default fork-per-test mode.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7365>
fix playback fail, when some file with length_size_minus_one == 2
According to the spec 2 cannot be a valid value, so that stream has a
bad config record. but breaking the decoding because of that, perhaps is too much.
and ffmpeg seem not check this
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7213>
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>
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>
