When that property is left to its default, the width and height
property considers frames from input pads with width or height <= 0
should be left unscaled in that dimension.
Setting this property to FALSE changes that behaviour to < 0, as when
animating these properties, 0 should be a valid end value (eg. shrinking
an input stream until it disappears).
The default value of the width and height properties is set to -1, so that
the default behaviour stays consistent whether that new property is set
or not.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/923>
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>
Increases the throughput of compositing by using more CPU cycles across
multiple threads. Simple cases (the output contains one pixel from at
most one input) can have up to a 70% increase in throughput. Not so
simple cases are limited by the region with the most number of
composite operations.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/755>
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>
When a pad has alpha != 1.0 it means that the resulting frames will
contain alpha and thus can't fully obscure with a lower zorder.
Also simplifies the other checks as blending with an OVER or on a
transparent is not a no-op as previously assumed.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/764>
It was not working properly and the implementation of the smartencoder
element was weird. This introduce a number of changes (which are all
in one single commit because they basically all work together and lead
to basically reimplementing the element):
* Make smartencoder a bin so that the reencoding chain of elements are
inside of it instead of not having any parent. Those elements were not
be visible when dumping the pipeline which was very confusing.
* Make encodebin create the right encoder with a capsfilter (and parser)
to properly enforce the format specified by the user, and so that the
encoder properties specified in the encoding profile are respected.
* Use `decodebin` to do the decoding instead of selecting a decoder
ourself and not plug any parser etc...
* Ensure that negotiated format in the sinkpad of smart encoder is fixed
through time when the user requested a non dynamic output
* Add a parser at the beginning of the smart encoder
* Handle errors when reencoding
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/751>
When using tile format, the stride has a different meaning. It used
the MSB and LSB 16bits to encode respectively the width and height in
number of tiles.
This issue was introduce with commit e5b70d384c which was fixing
missing size recalculation when strides and offset is updated.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/753>
Whenever a new collection is calculated, the internal `select_streams_seqnum`
variable is reset. This ensures that we reliably know whether a select-streams
event has been received for that new collection.
Use that to decide whether we should add previously un-selected streams or new
streams in the current selection
Fixes#784
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/733>
For example, BT709, BT601, and BT2020_10 all have theoretically
different transfer functions, but the same function in practice. In
these cases, we should use the fast path for negotiating. Also,
BT2020_12 is essentially the same as the other three, just with one more
decimal point, so it gives the same result for fewer bits. This is now
also aliased to the former three.
Also make videoconvert do passthrough if the caps have equivalent
transfer functions but are otherwise matching.
As of the previous commit, we write the correct transfer function for
BT601, instead of the (functionally identical but different ISO code)
transfer function for BT709. Files created using GStreamer prior to that
commit write the wrong transfer function for BT601 and are, strictly
speaking, 2:4:5:4 instead. However, this commit takes care of
negotiation, so that conversions from/to the same transfer function are
done using the fast path.
Fixes#783
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/724>
It is possible for subtitle files to have a string length less than 30.
WebVTT for example may contain only the 'WEBVTT' string in the file
without any cues.
As an example in hls streams, since WEBVTT files can be segmented
like video/audio, some subtitle segments may only contain just the
header string.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/708>
When linking source pads to decodebin, make sure we use the *specified* new
source pad and not some random one.
This avoids ending up with source pads being unlinked.
Main cause of random timeouts with rtsp change_state_intensive validate tests
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/687>
Otherwise there is a mismatch between the QoS values and what upstream
would expect, leading to too much buffer dropping in video decoders in
case rate < 1.0 or not enough buffer dropping in case rate > 1.0
Adding validate tests with and without decoders.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/679>
We need to take into account the base_ts to compute next_ts and it needs
to be updated on rate change.
This introduces `pending_rate` so that change rate is properly handled
in the streaming thread in a safe way.
Added tests
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/679>
