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>
Instead of dragging the last destination pipeline stage as current barrier
source pipeline stage (which isn't a valid semantic) this patch adds a parameter
to gst_vulkan_operation_add_frame_barrier() to set the source pipeline stage to
define the barrier.
The previous logic brought problems particularly with queue transfers, when the
new queue doesn't support the stage set during a previous operation in a
different queue.
Now the operation API is closer to Vulkan semantics.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7165>
since the encoded output is changing based on version
it does not make sense to check the output bitstream with a fixed
bytearray since the version in the target might vary. So sticking
to checking the number of output buffers and encoded frame size
similar to the other tests
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7141>
From the spec (chapter 34, v1.3.283):
````
UNORM: the components are unsigned normalized values in the range [0, 1]
SRGB: the R, G and B components are unsigned normalized value that represent
values using sRGB nonlinear encoding, while the A component (if one
exists) is a regular unsigned normalized value
```
The difference is the storage encoding, the first one is aimed for image
transfers, while the second is for shaders, mostly in the swapchain stage in the
pipeline, and it's done automatically if needed [1].
As far as I have checked, other frameworks (FFmpeg, GTK+), when import or export
images from/to Vulkan, use exclusively UNORM formats, while SRGB formats are
ignored.
My conclusion is that Vulkan formats are related on how bits are stored in
memory rather their transfer functions (colorimetry).
This patch does two interrelated changes:
1. It swaps certain color format maps to try first, in both
gst_vulkan_format_from_video_info() and gst_vulkan_format_from_video_info_2(),
the UNORM formats, when comparing its usage, and later check for SRGB.
2. It removes the code that check for colorimetry in
gst_vulkan_format_from_video_info_2(), since it not storage related.
1. https://community.khronos.org/t/noob-difference-between-unorm-and-srgb/106132/7
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6797>
