Sending an EOS event is actually really bad because rtpbin doesn't
handle that very well. It was only being used as a way to notify
webrtcbin to check if re-negotiation is needed.
We don't need that anymore, since changing the direction is enough to
notify webrtcbin to check for re-negotiation.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5045>
When a video track is muted, hide the video element to differentiate
it from a track that is stuck because we stopped receiving RTP data.
Show it again when it is unmuted.
When a video track is removed, remove the video element. It will be
re-added on renegotiation.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5045>
The current way of dma caps uses the drm-format to replace the orginal
format field. The absence of format field means it can accept all formats.
It causes problems when clipping with other old DMA or video/x-raw(ANY)
caps, the result will contain both format field and drm-format field,
which is not valid DMA caps.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4981>
This GST_VIDEO_FORMAT_DMA_DRM is introduced for DMABuf kind feature
usage. It represent the DMA DRM kind memory. And like the ENCODED
format, it should not be interpreted and mapped as normal video format.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4981>
Setting the input field on the empty slot prevents future linking of it and will
result in flow errors later on.
This was observed in WebKit's MediaStream source element, when it changes the
caps on one of its associated streams, from an encoded format to a raw video
format. The associated stream-id on the sticky stream-start event doesn´t
change, but the element creates a new GstStream with a different ID and sets it
on the stream-start event. Stream parsing is disabled in urisourcebin, so
decodebin3 handles the parsing. Without this patch we would end-up with unlinked
pads in decodebin3 after switching to the raw video format.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5048>
The drop-frame rules are specified in “SMPTE ST 12-3:2016” and are
consistent with the traditional ones:
“
To minimize fractional time deviation from real time, the first two
super-frame numbers (00 and 01) shall be omitted from the count at the
start of each minute except minutes 00, 10, 20, 30, 40, and 50. Thus the
first eight frame numbers (0 through 7) are omitted from the count at
the start of each minute except minutes 00, 10, 20, 30, 40, and 50.
”
Where “super-frame” is a group of 4 frames for 120 FPS.
Fixes#2797
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5028>
The current implementation copies metas without checking if the buffer
is writable.
The operation that needs to be done, replacing the input buffer and
copying the metas, is only part of that process. We create a new function
that does both.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4912>
This check fixes a critical warning that can happen when a pointer motion
happens and the video doesn't have its width/height information available.
GStreamer-Video-CRITICAL **: gst_video_center_rect: assertion 'src->h != 0' failed
#0 g_logv (log_domain=0x7ffff705e176 "GStreamer-Video", log_level=G_LOG_LEVEL_CRITICAL, format=<optimized out>, args=<optimized out>) at ../../../../Projects/jhbuild/glib/glib/gmessages.c:1422
#1 0x00007ffff7e1a81d in g_log (log_domain=<optimized out>, log_level=log_level@entry=G_LOG_LEVEL_CRITICAL, format=format@entry=0x7ffff7e77a9d "%s: assertion '%s' failed") at ../../../../Projects/jhbuild/glib/glib/gmessages.c:1460
#2 0x00007ffff7e1b749 in g_return_if_fail_warning (log_domain=<optimized out>, pretty_function=<optimized out>, expression=<optimized out>) at ../../../../Projects/jhbuild/glib/glib/gmessages.c:2930
#3 0x00007ffff701d90b in gst_video_sink_center_rect (src=..., dst=..., result=result@entry=0x7fffffffc6d0, scaling=scaling@entry=1) at ../../../../Projects/jhbuild/gstreamer/subprojects/gst-plugins-base/gst-libs/gst/video/gstvideosink.c:105
#4 0x00007fffe5652dbb in _fit_stream_to_allocated_size (result=0x7fffffffc6d0, allocation=0x7fffffffc6c0, base_widget=0x9396f0) at ../../../../Projects/jhbuild/gstreamer/subprojects/gst-plugins-good/ext/gtk/gtkgstbasewidget.c:326
#5 gtk_gst_base_widget_display_size_to_stream_size (base_widget=base_widget@entry=0x9396f0, x=1207.7109375, y=811.84765625, stream_x=stream_x@entry=0x7fffffffc720, stream_y=stream_y@entry=0x7fffffffc728) at ../../../../Projects/jhbuild/gstreamer/subprojects/gst-plugins-good/ext/gtk/gtkgstbasewidget.c:344
#6 0x00007fffe5651a4b in gst_gtk_base_sink_navigation_send_event (navigation=0x5ff990, event=0x178a730) at ../../../../Projects/jhbuild/gstreamer/subprojects/gst-plugins-good/ext/gtk/gstgtkbasesink.c:340
#7 0x00007fffe5652432 in gtk_gst_base_widget_motion_event (widget=<optimized out>, event=event@entry=0x1f14b60) at ../../../../Projects/jhbuild/gstreamer/subprojects/gst-plugins-good/ext/gtk/gtkgstbasewidget.c:404
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5049>
The libpsl subproject wasn't building successfully and CI didn't
notice because:
1. The plugin wasn't explicitly enabled
2. Even when the plugin is explicitly enabled, the dep is not required
at build time when not building a static plugin
So fix all of these issues.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5038>
The videoencoder base class uses getcaps() to ask a subclass for the caps in its
sink_query_default() implementation.
Replace the custom handling of the QUERY_CAPS in the v4l2videoenc with an
implementation of getcaps() that returns the caps that are supported by the
v4l2videoenc to return these caps in the query.
This getcaps() implementation also calls the provided proxy_getcaps(), which
sends a caps query to downstream. This fixes the v4l2videoenc element to respect
limits of downstream elements in a sink query.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5034>
Given the amount of complains about artifacts when negotiating dmabuf
given incompatible drm-formats, and that there's no enough bandwidth
for a proper and quick fix in gstreamer-vaapi, this patch disables,
from decoders and postprocessor, the DMABuf caps feature.
For those who needs DMABuf can use the va elements in -bad, increasing
their ranking for autoplugging by using the environment variable
GST_PLUGIN_FEATURE_RANK=vah264dec:MAX, for example.
This can be considered a first step to the deprecation of
gstreamer-vaapi in favor of the va plugin in -bad.
Fixes: #1137
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5010>
The videoencoder base class always uses the negotiated allocator for allocating
coded buffers and ignores the negotiated buffer pool. Therefore, the
v4l2videoenc always has to copy buffers from the pool into the allocated
output buffers.
This breaks downstream elements that want to import the CAPTURE buffers of the
v4l2videoenc, since the v4l2videoenc copies the exported CAPTURE buffers and
sends the copies downstream.
Always use the CAPTURE buffer pool for acquiring CAPTURE buffers instead of
allocating the buffers in the base class.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4230>
It's possible and normal to tear down a harness while the pipeline is
running. At the same time, it's desired for the
`gst_harness_pad_link_tear_down()` function to be synchronous.
This has created the conflict where the main thread may request a
harness to be torn down while it's in use or about to be used by a pad
in the streaming thread.
The previous implementation of `gst_harness_pad_link_tear_down()` tried
to handle this by taking the stream lock of the harnessed pad and
resetting all the pad functions while holding it. That approach was
however insufficient to handle the case where a non-serialized event
or query is being handled or about to be handled in a different thread.
This edge case was one race condition behind the flakes in the flvmux
check tests -- the rest being covered by https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2803.
This patch fixes the problem by adding an intermediate ref-counted
object, GstHarnessLink, which replaces the usage of the HARNESS_KEY
association. GstHarnessLink allows the pad functions such as event,
query and chain to borrow a reference to GstHarness and more
importantly, to lock the GstHarnessLink during their usage to block
(delay) its destruction until no users are left, and guarantee that any
future user will not receive an invalid GstHarness handle past its
destruction.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5017>
