There is currently no way for users to receive incoming events from
appsink while keeping them properly serialized with the buffers flow.
This can be especially useful when application is injecting custom
downstream events into the pipeline and needs to know when they reached
appsink.
Solving this by adding a new signal notifying about new incoming events
and a set of action signals and method to pull those events.
The API is actually pulling the samples and events all together as they
are actually fetched from the same queue.
Having a specific API to pull only events would have the side effect of
discarding samples (and pulling samples would discard events) making
this API not convenient for users.
Partially fix#247
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1046>
If a buffer is dropped during resyncing on a discont because either its
end offset is already before the current output offset of the
aggregator or because it fully overlaps with the part of the current
output buffer that was already filled, then don't just assume that the
next buffer is going to start at exactly the expected offset. It might
still require some more dropping of samples.
This caused the input to be mixed with an offset to its actual position
in the output stream, causing additional latency and wrong
synchronization between the different input streams.
Instead consider each buffer after a discont as a discont until the
aggregator actually resynced and starts mixing samples from the input
again.
Also update the start output offset of a new input buffer if samples
have to be dropped at the beginning. Otherwise it might be mixed too
early into the output and overwrite part of the output buffer that
already took samples from this input into account.
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/issues/912
which is a regression introduced by https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1180/
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1224>
A video decoder can now receive subframes and start decoding
instead of waiting for the full frame to be complete.
Subframe support will reduce latency as described in the
video encoder base class.
A unit test illustrating this API is available in
tests/check/libs/videodecoder.c.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/454>
While preparing a blist for pushing, some RTP header extension may
request caps change for a specific buffer in the list. When this
happens, depayloader should immediately push those buffers from the list
that precede the currently processed buffer (for which the caps change
was requested) and only then apply the new caps to the src pad.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1011>
These parameters are incorrectly regarded as mutable in G-IR making them
"incompatible" with languages that are explicit about mutability like
Rust. In order to clean up the code and expected API there, update the
signatures here, right at the source (instead of overriding them in
Gir.toml and hoping for the best).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1005>
We've been allowing only a few known chroma-site values such as
jpeg (not co-sited), mpeg2 (horizontally co-sited) and
dv (co-sited on alternate lines). That's insufficient for
representing all possible chroma-site values. By this commit,
we can represent any combination of chroma-site flags.
But, an exception here is that any combination with
GST_VIDEO_CHROMA_SITE_NONE will be considered as invalid value.
For any combination of chroma-site flags,
gst_video_chroma_to_string() method is deprecated in order to
return newly allocated string via a new gst_video_chroma_site_to_string()
method. And for consistent API naming, gst_video_chroma_from_string()
is also deprecated. Newly written code should use
gst_video_chroma_site_from_string() instead.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/927>
New signals are added for managing the internal list of rtp header
extension implementations read by a specific depayloader instance.
If the 'extmap-$NUM' field is present in the src caps, then an
extension implementation will be requested but is not required to be able
to negotiate correctly. An extension will be requested using the
'request-extension' signal if none could be found internally.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/748>
New signals are added for managing the internal list of rtp header
extension implementations read by a specific depayloader instance.
If the 'extmap-$NUM' field is present in the sink caps, then an
extension implementation will be requested but is not requited to be
able to negotiate correctly. An extension will be requested using the
'request-extension' signal if none could be found internally.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/748>
.. and make use of that API in videoaggregator.
When setting certain properties, such as cropping or the scaled
size of pads, a new converter is created by videoaggregator.
Before that patch, this implied spawning new threads, potentially
at each aggregate cycle when interpolating pad properties. This
is obviously wasteful, and re-using a task pool removes that
overhead.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/896>
In the case `videoaggregator` is set as allowing format conversions,
and as we convert only on the sinkpads, we should ensure that the
chosen format is usable by the subclass. This in turns implies
that the format is usable on the srcpad.
When doing conversion *any* format can be used on the sinkpads, and this
is the only way that we can avoid race conditions during renegotiations
so we can not change that fact, we just need to ensure that the chosen
intermediary format is usable, which was not actually ensured before
that patch.
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/issues/834
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/909>
We validate the header extensions length of an RTP buffer by comparing
it against the block size. Since we multiply the length in words by 4 to
get the length in bytes, a suitably large length could cause a wrapround
of the uint16, giving a lower length which erroneously passes the check
and allows the buffer to be mapped.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/864>
Rename remaining `gst_video_color_transfer_{encode,decode}` functions on
the `GstVideoTransferFunction` enumeration to
`gst_video_transfer_function_{encode,decode}` permitting
gobject-introspection to turn these into associated functions and place
them under the respective `<enumeration>` block in gir XML files.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/805>
Don't run the harness in live mode, or otherwise it would output frames
already in the very beginning before a buffer was provided to it due to
timeout.
Also send EOS/a second buffer before pulling a buffer as videoaggregator
has one frame of latency.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/812>
The type is called GstVideoTransferFunction so the function names should
match, otherwise gobject-introspection is keeping the functions as
global functions instead of methods on the type.
The same mistake was also made in lots of other APIs over the years, but
here we can at least fix it for 1.18 still.
Thanks to Marijn Suijten for noticing.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/807>
Call gst_aggregator_selected_samples() after filling the queues
(but before preparing frames).
Implement GstAggregator.peek_next_sample.
Add an example that demonstrates usage of the new API in combination
with the existing buffer-consumed signal.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/728>
