The goal here is to minimize the work needed to bring all images
to a common format. A better criteria than the number of pads
with a given format is the number of pixels with a given format.
https://bugzilla.gnome.org/show_bug.cgi?id=786078
Crossfading is a bit more complex than just having two pads with the
right keyframes as the blending is not exactly the same.
The difference is in the way we compute the alpha channel, in the case
of crossfading, we have to compute an additive operation between
the destination and the source (factored by the alpha property of both
the input pad alpha property and the crossfading ratio) basically so
that the crossfade result of 2 opaque frames is also fully opaque at any
time in the crossfading process, avoid bleeding through the layer
blending.
Some rationnal can be found in https://phabricator.freedesktop.org/T7773.
https://bugzilla.gnome.org/show_bug.cgi?id=784827
When the pad has received EOS, its buffer may still be mixed
any number of times, when the pad's framerate is inferior
to the output framerate.
This was introduced by my patch in
https://bugzilla.gnome.org/show_bug.cgi?id=782962, this patch
also correctly addresses the initial issue.
When caps changes while streaming, the new caps was getting processed
immediately in videoaggregator, but the next buffer in the queue that
corresponds to this new caps was not necessarily being used immediately,
which resulted sometimes in using an old buffer with new caps. Of course
there used to be a separate buffer_vinfo for mapping the buffer with its
own caps, but in compositor the GstVideoConverter was still using wrong
info and resulted in invalid reads and corrupt output.
This approach here is more safe. We delay using the new caps
until we actually select the next buffer in the queue for use.
This way we also eliminate the need for buffer_vinfo, since the
pad->info is always in sync with the format of the selected buffer.
https://bugzilla.gnome.org/show_bug.cgi?id=780682
When entering this code path, we know that:
We received EOS on this pad.
We consumed all its buffers.
In any case, we want to replace vaggpad->buffer with NULL,
otherwise we will end up mixing the same buffer twice.
https://bugzilla.gnome.org/show_bug.cgi?id=781037
If the input buffer is after the end of the output buffer, then waiting
for more data won't help. We will never get an input buffer for this point.
This fixes compositing of streams from rtspsrc.
https://bugzilla.gnome.org/show_bug.cgi?id=766422
We weren't using the result of find_best_format at all.
Also, move the find_best_format usage to the default update_caps() to make
sure that it is also overridable.
https://bugzilla.gnome.org/show_bug.cgi?id=764363
When caps are already negotiated it should be possible to
select formats other than the one that was negotiated. If downstream
allows alpha video caps and it has already negotiated to a non-alpha
format, caps queries should still return the alpha caps as a possible
format as caps renegotiation can happen.
Includes tests (for compositor) to check that caps queries done after
a caps has been negotiated returns complete results
https://bugzilla.gnome.org/show_bug.cgi?id=757610
Allows the subclass to completely override the chosen src caps.
This is needed as videoaggregator generally has no idea exactly
what operation is being performed.
- Adds a fixate_caps vfunc for fixation
- Merges gst_video_aggregator_update_converters() into
gst_videoaggregator_update_src_caps() as we need some of its info
for proper caps handling.
- Pass the downstream caps to the update_caps vfunc
https://bugzilla.gnome.org/show_bug.cgi?id=756207
We have to queue buffers based on their running time, not based on
the segment position.
Also return running time from GstAggregator::get_next_time() instead of
a segment position, as required by the API.
Also only update the segment position after we pushed a buffer, otherwise
we're going to push down a segment event with the next position already.
https://bugzilla.gnome.org/show_bug.cgi?id=753196
Only accept alpha if downstream has alpha as well. It could
theoretically accept alpha unconditionally if blending is
properly implemented for handle it but at the moment this
is a missing feature.
Improves the caps query by also comparing with the template
caps to filter by what the subclass supports.
https://bugzilla.gnome.org/show_bug.cgi?id=754465
Before aggregator based elements always started at running time 0,
now it's possible to select the first input buffer running time or
explicitly set a start-time value.
https://bugzilla.gnome.org/show_bug.cgi?id=749966
The problem here was that after removing the formats and
all the things we could convert, we then intersected these
caps with the template caps.
Hence if a subclass offered permissive sink templates
(eg all the possible formats videoconvert handles), but only
one output format, then at negotiation time getcaps returned
caps with the format restricted to that format, even though
we do handle conversion.
https://bugzilla.gnome.org/show_bug.cgi?id=751255
When set, it causes videoaggregator to repeatedly aggregate the last buffer on
an EOS pad instead of skipping it and outputting silence. This is useful, for
instance, while playing back files seamless one after the other, to avoid
videoaggregator ever outputting silence (the checkerboard pattern).
It is to be noted that if all the pads on videoaggregator have this property set
on them, the mixer will never forward EOS downstream for obvious reasons. Hence,
at least one pad with 'ignore-eos' set to FALSE must send EOS to the mixer
before it will be forwarded downstream.
https://bugzilla.gnome.org/show_bug.cgi?id=748946
Add preserve_update_caps_result boolean on the class to allow
sub-classes to disable videoaggregator removing sizes and framerate
from the update_caps() return result.
A return value of GST_FLOW_OK with a NULL buffer from get_output_buffer()
means the sub-class doesn't want to produce an output buffer, so
skip it.
If gst_videoaggregator_do_aggregate() generates an error, make sure
to propagate it - don't just ignore and discard the error by
over-writing it with the gst_pad_push() result.
This would've also triggered if for some reason the segment was updated
in such a way that PTS went backwards, but the running time increased. Like
what happens when non-flushing seeks are done.
We're doing a proper buffer-from-the-past check a few lines below based on the
running time, which is the only time we should care about here.
And keep on querying upstream until we get a reply.
Also, the _get_latency_unlocked() method required being calld
with a private lock, so removed the _unlocked() variant from the API.
And it now returns GST_CLOCK_TIME_NONE when the element is not live as
we think that 0 upstream latency is possible.
https://bugzilla.gnome.org/show_bug.cgi?id=745768
In case the original caps were missing some optional fields like
interlace-mode. We assume default values for those everywhere,
but they can still cause negotiation to fail if a downstream element
expects the field to be there and at a specific value.
If the src framerate and videoaggreator's output framerate were
different, then we were taking every single buffer that had duration=-1
as it came in regardless of the buffer's start time. This caused the src
to possibly run at a different speed to the output frames.
https://bugzilla.gnome.org/show_bug.cgi?id=744096
This can happen if this is a live pipeline and no source produced any buffer
and sent no caps until an output buffer should've been produced according to the
latency.
This fix is similar in spirit to commit be7034d1 by Sebastian for audiomixer.
Unset out buffer in clip function when we unref the buffer to be
clipped, otherwise aggregator will continue to use the already-
freed buffer. Fixes crash when buffers without timestamps are
being fed to aggregator. Partly because aggregator ignores the
error flow return.
https://bugzilla.gnome.org/show_bug.cgi?id=743334
When this is TRUE, we really have to produce output. This happens
in live mixing mode when we have to output something for the current
time, no matter if we have enough input or not.
This removes the uses of GAsyncQueue and replaces it with explicit
GMutex, GCond and wakeup count which is used for the non-live case.
For live pipelines, the aggregator waits on the clock until either
data arrives on all sink pads or the expected output buffer time
arrives plus the timeout/latency at which time, the subclass
produces a buffer.
https://bugzilla.gnome.org/show_bug.cgi?id=741146
gst_video_info_set_format() will reset the complete video-info, but
we want to keep values like the PAR, colorimetry and chroma site.
Otherwise we risk setting different values on the srcpad caps than
what is actually inside the buffers.
Otherwise we might negotiate from the sinkpad streaming threads at
the same time as on the srcpad streaming thread, and then all kinds
of crazy bugs happen that don't make any sense at all.
This gives more flexibility to the subclasses and permits to remove the
GstVideoAggregatorClass->disable_frame_conversion ugly API.
WARNING: This breaks the API as it removes the disable_frame_conversion
field
API:
+ GstVideoAggregatorClass->find_best_format
+ GstVideoAggregatorPadClass->set_format
+ GstVideoAggregatorPadClass->prepare_frame
+ GstVideoAggregatorPadClass->clean_frame
- GstVideoAggregatorClass->disable_frame_conversion
https://bugzilla.gnome.org/show_bug.cgi?id=740768
With the current code, we will end up setting the preferred downstream
format as the srcpad format, and it might not be accepted by the next
sinkpad to be added. We should instead let the next sinkpad reconfigure
everything.
The aggregator.segment is not to be initialized by the subclasses but
by the aggregator itself. Moreover, initializing it on start would make
us loose the information coming from the initial seek.
