This adds "id" variants to most debugging functions, and allows providing a
string identifier instead of a GObject.
This allows providing unified and clearer debug logs for all the
non-gobject-based items, and opens the way for more unified logging.
As an extension, copying the object name is avoided as much as possible, by
using it directly instead of going through another copy.
* API : gst_debug_message_get_object_id
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3483>
This regression was introduce by fix for making buffer pool thread safe. When
we renegotiate, the pool will be setup after we set the format. But the code
has been simplified to only get the pool once before, which caused a null
pointer deref.
Fixes 94ba019 ("v4l2: Fix SIGSEGV on 'change state' during 'format change'")
Related to !3481Fixes#1626
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3513>
There are cases where user might want to be in full control of the
timeline and not be limited by the checks that are being done by GES
to go from one timeline layout to another, this should be doable as
it is a valid use case.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3501>
When coloring is in use, those escape codes are going to be created many times
for almost all debug lines.
Don't create plenty of temporary allocations, and instead build the escape code
ourselves statically
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3498>
Because of the asynchronous resolving of mDNS ICE candidates it is
possible that GstWebRTCICE outlives webrtcbin. This in turn prolongs
the lifetime of the GstWebRTCNiceStream objects via refs in
nice_stream_map. Thus the GstWebRTCICETransport objects held in
GstWebRTCNiceStream may be invalid at the time they are accessed by
the _on_candidate_gathering_done() callback since GstWebRTCNiceStream
doesn't take a reference to them. Doing so would create a circular
reference, so instead this commit introduces weak references to the
transport objects and then we can check if the objects are valid before
accessing them.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3502>
And even that vaav1dec doesn't use vabasedec negotiate vmethod, it should align
with the new scheme of using base's width & height for surface size and
output_info structure for downstream display size negotiation.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3480>
This vmethod can be used by decoders with the same VA decoder reopen logic:
same profile, chroma, width and height.
Also a new public method called gst_va_base_dec_set_output_state() with the
common GStreamer code for setting the output state, which is always called by
the negotiate vmethod.
In order to do this refactoring, new variables in vabasedec have to be populated
by the decoders:
* width and height define the resolution set in VA decoder. In the case of H264
would be de coded_width and codec_height, or max_width and max_height in AV1.
* output_info is the downstream video info used for negotiation in
gst_va_base_dec_set_output_state().
* input_state, from codec parent class shall be also held by vabasedec
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3480>
There could be multi-GPU setups where the non-first has more
entrypoints than the first one, and the elements names are not
homogeneous, leading to pipeline building error.
This patch add the render node in the elements names when they belong
to the non-first device.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3491>
To fix the warning on Alderlake
vafilter gstvafilter.c:534:gst_va_filter_ensure_filters:<vafilter0>
vaQueryVideoProcFiltersCaps: list argument exceeds maximum number
Increase the number of caps to 16 as vadumpcaps does.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3473>
We want to make it so that we prefer a higher, not lower, number of
channels. Otherwise, this pipeline would convert from 2 to 1 channels:
gst-launch-1.0 audiotestsrc ! audio/x-raw,channels=2 ! opusenc ! queue ! opusdec ! queue ! opusenc ! fakesink
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3494>
