The current GstVideoRegionOfInterestMeta API allows elements to detect
and name ROI but doesn't tell anything about how this information is
meant to be consumed by downstream elements.
Typically, encoders may want to tweak their encoding settings for a
given ROI to increase or decrease their quality.
Each encoder has its own set of settings so that's not something that
can be standardized.
This patch adds encoder-specific parameters to the meta which can be
used to configure the encoding of a specific ROI.
A typical use case would be: source ! roi-detector ! encoder
with a buffer probe on the encoder sink pad set by the application.
Thanks to the probe the application will be able to tell to the encoder
how this specific region should be encoded.
Users could also develop their specific roi detectors meant to be used with a
specific encoder and directly putting the encoder parameters when
detecting the ROI.
https://bugzilla.gnome.org/show_bug.cgi?id=793338
test_negotiation would occasionally time out, for unknown reasons.
Simplify the test setup and get rid of the main loop, busses, and
notify signals. With this I can no longer easily reproduce the
timeout. Fingers crossed.
Performance optimisation: Keep track whenever the streaming
thread or the application thread are waiting on the GCond for
more space or new data, and only signal on the GCond if someone
is actually waiting. Avoids unnecessary syscalls and thus
context switches.
Performance optimisation: Keep track whenever the streaming
thread or the application thread are waiting on the GCond
for more space or new data, and only signal on the GCond if
someone is actually waiting. Avoids unnecessary syscalls and
thus context switches.
These are very much artificial of course, but got to
measure something. appsink one contains lots of buffer
creation/free overhead, while appsrc one does not.
When trying to create a wayland display, it may fail because there
is not actually display to connect. It this case NULL is returned
but the created instance is not freed.
This patch unrefs the failed display.
https://bugzilla.gnome.org/show_bug.cgi?id=793483
If new headers arrive after we are initialized, we need to make
sure that they are indeed valid.
A vorbis bitstream always begins with three header packets and must
be in order.
Also some streams have unframed (invalid?) headers that might
confuse and disrupt the decoding process.
Therefore if ever we see new headers, we accumulate them and once
we get a non-header packet we check them to make sure that:
* We have at least 3 headers
* They are the expected ones (identification, comments and setup)
* They are in order
* Any other "header" is ignored
If those conditions are met, we reset and reconfigure the decoder
https://bugzilla.gnome.org/show_bug.cgi?id=784530
Buffering messages are only sent for the active group (in case there
is more than one).
If the inactive group posts buffering messages we keep the last one
around and will post it once it becomes the playing one.
In order to flush out multiqueue, we send again a STREAM_START and
then a EOS event.
The problem was that was that we might end up pushing out on the
output of multiqueue (and therefore decodebin3) a series of:
* EOS / STREAM_START / EOS
Apart from the uglyness of such output, If decodebin3 is used with
elements such as concat on their output, they might potentially
block on that second STREAM_START.
In order to make sure we don't end up in that situation we send
a custom STREAM_START event when refreshing multiqueue (which we
drop on the output) and we don't special case EOS events on streams
on which we already got EOS.
At worst we now end up sending at most two EOS on the output of
multiqueue (and decodebin3).
Similar in vein to the playbin2 architecture except that uridecodebin3
are prerolled much earlier and all streams of the same type are
fed through a 'concat' element.
This keeps the philosphy of having all elements connected as soon
as possible.
The 'about-to-finish' signal is emitted whenever one of the uridecodebin
is about to finish, allowing the users to set the next uri/suburi.
The notion of a group being active has changed. It now means that the
uridecodebin3 has been activated, but doesn't mean it is the one
currently being outputted by the sinks (i.e. curr_group and next_group).
This is done via detecting GST_MESSAGE_STREAM_START emission by playsink
and figuring out which group is really playing.
When the current group changes, a new thread is started to deactivate
the previous one and optionnaly fire 'about-to-finish'.
Apologies for the big commit, but it wasn't really possible to split it
in anything smaller.
* Switch to uridecodebin3 instead of managing urisourcebin and decodebin3
ourselves. No major architectural change with this.
* Reconfigure sinks/outputs when needed. This is possible thanks to the
various streams-related API. Instead of blocking new pads and waiting
for a (fake) no-more-pads to decide what to connect, we instead reconfigure
playsink and the combiners to whatever types are currently selected. All of
this is done in reconfigure_output().
New pads are immediately connected to (combiners and) sinks, allowing
immediate negotiation and usage.
* Since elements are always connected, the "cached-duration" feature is gone
and queries can reach the target elements.
* The auto-plugging related code is currently disabled entirely until
we get the new proper API.
* Store collections at the GstSourceGroup level and not globally
* And more comments a bit everywhere
NOTE: gapless is still not functional, but this opens the way to be able
to handle it in a streams-aware fashion (where several uridecodebin3 can
be active at the same time).
With push-based sources, urisourcebin will emit this signal when
the stream has been fully consumed.
This signal can be used to know when the source is done providing
data.
