This situation happens in the situation where an input stream has a framerate
exceeding the timeout latency (Ex: 1fps with a latency of 500ms) and an input
stream greater than output framerate (ex: 60fps in, 30 fps out).
The problem that would happen is that we would timeout, but then buffers from
the fast input stream would only be popped out one by one.... until a buffer
reaches the low-framerate input stream at which point they would quickly be
popped out/used. The resulting output would be "slow ... fast ... slow ... fast"
of that input fast stream.
In order to avoid this situation, whenever we detect a late buffer, check if
there's a next one and re-check with that one.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/990>
Fix the following build failure with gcc 4.8 which has been added with
d268c193ad:
../gst-libs/gst/video/gstvideoaggregator.c: In function 'gst_video_aggregator_init':
../gst-libs/gst/video/gstvideoaggregator.c:2762:3: error: 'for' loop initial declarations are only allowed in C99 mode
for (gint i = 0; i < gst_caps_get_size (src_template); i++) {
^
Signed-off-by: Fabrice Fontaine <fontaine.fabrice@gmail.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/974>
.. 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>
Take `GST_OBJECT_LOCK` when writing `vagg->info`, so that reading in
subclasses is protected against races, as documented in the struct.
/*< public >*/
/* read-only, with OBJECT_LOCK */
GstVideoInfo info;
`gst_video_aggregator_default_negotiated_src_caps` should take the
`GST_VIDEO_AGGREGATOR_LOCK` to avoid racing with
`gst_video_aggregator_reset` called by
`gst_video_aggregator_release_pad` of the last sinkpad. Otherwise it can
happen that `latency = gst_util_uint64_scale (...` gets called with a
zero framerate.
There doesn't seem to be any reason not to use the local `info` instead
of `vagg->info`, so do that.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/915>
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>
Since 23189c60f4 we started using the
useless result of `modified_caps` which, was never used since it was
introduced 7 years ago (in videomixer2). The intersection is useless and
we should just avoid doing it at all (which was always the case before)
as it can end up failing renegotiation for bad reasons.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/905>
gst_pad_get_current_caps() may be wrong when there is a renegotiation in
progress for the pad and we have not yet received or selected the buffer
with different caps yet.
Fix by storing the caps through in a similar way to the existing code
for buffer/video-info selection machinery.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/813>
This is not actually required (anymore?). Source pad caps can be
negotiated at any time regardless of any configured (or existing) sink
pads and videoaggregator comes up with some fixated caps based on the
downstream caps.
Subclasses can override this behaviour as needed by overriding
update_src_caps().
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/793>
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>
[38/1301] Generating GstVideo-1.0.gir with a custom command.
../subprojects/gst-plugins-base/gst-libs/gst/video/gstvideoaggregator.c:231: Error: GstVideo: identifier not found on the first line:
*
^
This can be used to have compositor display either the background
or a stream on a lower zorder after a live input stream freezes
for a certain amount of time, for example because of network
issues.
This simply implies not trying to "prepare" those buffers,
as mapping an empty buffer to a video frame does not make
much sense.
This also adds a simple test in compositor that performs
some trivial checking of the handling of gap events, the test
was not failing before, but an error was logged, this is
no longer the case.
Fixes#717
If there's no known value in the best caps then the functions to convert
them to strings will return NULL. Having the fields not in the caps is
not a problem, having them with a NULL value however will cause
negotiation failures.
In prepare_frame, it is not enough for the target info
(conversion_info) to not have changed to decide not to update
the converter, as the vpad info may have changed as well.
Fixes#714
The ->skip_buffer implementation in videoaggregator replicates
the behaviour of the aggregate method to determine whether a
buffer can be skipped
(https://bugzilla.gnome.org/show_bug.cgi?id=781928).
This fixes a typo that made it so the start time of the buffer
was calculated against the output segment, not the segment of
the relevant sinkpad, which caused buffers to be skipped when
for example a sinkpad had received a segment which base had
been modified by a pad offset somewhere along the way.
This simply makes the calculation of the buffer start time
identical to the calculation in aggregate()
The start_time and end_time in this context have already
been adjusted for the input's rate by converting them to running
time above. What is needed afterwards is to compare these
with the output's start/stop running time, which also takes
into account the rate, so we are comparing equal things.
Multiplying these with the output's rate here is only breaking
this logic. In most cases the input and output rate is the same,
so this multiplication effectively reverses the rate adjustment
that happened while converting to running time, which is why
we see the video playing with the original rate in tests.
Fixes#541
This moves all the conversion related code to a single place, allows
less code-duplication inside compositor and makes the glmixer code less
awkward. It's also the same pattern as used by GstAudioAggregator.
This is only used for caching reasons and should never actually be in
the public API. If this is ever a bottleneck later, caching around a
class private struct could be implemented.
The aggregated_frame is now called prepared_frame and passed to the
prepare_frame and cleanup_frame virtual methods directly. For the
currently queued buffer there is a method on the video aggregator pad
now.
