This will end up as a "received" packet, due to the code in
source_push_rtp, which will think this is a packet being received.
Instead drop the packet and hope that either:
1. Something upstream responds to the GstRTPCollision event and changes
SSRC used for sending.
2. That the application responds to the "on-ssrc-collision" signal, and
forces the sender (payloader) to change its SSRC.
3. That the BYE sent to the existing user of this SSRC will respond to
the BYE, and that we timeout this source, so we can continue sending
using the chosen SSRC.
The test reproduces a scenario where we previously would have sent
on a non-internal source.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/817>
In baseparse we set the fixed caps flag on all src pads, therefore the
source pad caps query in get_allowed_caps will return the current caps.
Current caps won't necessarily intersect with the new caps (e.g. sample
rate change). Replace get_allowed_caps with peer_query_caps.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/816>
Add a new state for ending the overall stream, and use it to decide
whether to pass the final EOS message up the bus instead of dropping
it. Fixes a small race that makes the testsuite sometimes not generate
the last fragment(s) sometimes because the wrong EOS gets
allowed through too early.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/798>
Using the element state lock to avoid splitmuxsink shutting
down while doing element manipulations can lead to a deadlock on
shutdown if a fragment switch happens at exactly the wrong moment.
Use a private mutex and a shutdown boolean instead.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/798>
If a pad gets into the check_completed_gop method and then
the underlying conditions change on the reference context,
things could get stuck in a busy loop when the context should
instead jump back out and wait for more data.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/798>
Make sure that any late gst_element_call_async() callbacks
know that the elements is shutting down and bail out instead
of operating on the element we're trying to stop.
Fixes a spurious test failure in elements_splitmuxsrc
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/798>
- Fix start and end of picture to support multiple layers. Start of
picture is the first packet of the base layer, while end of picture
is when the marker bit is set (last packet of the enhancement
layers).
- All "layers" (aka "frames") of a picture are pushed downstream in a
single buffer when picture is complete.
- Forgive SID=0 for enhancement layers (invalid, but Chrome and
Firefox sends it)
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/773>
This is ad adaptation of a Pexip patch for dealing with spurious
GstRTPPacketLost events caused by lost ulpfec packets: as FEC packets
under that scheme are spliced in the same sequence domain as the media
packets, it is not generally possible to determine whether a lost packet
was a FEC packet or a media packet.
When upstreaming pexip's ulpfec patches, we decided to drop all lost
events at the base depayloader level, and where the original patch
from pexip was making use of picture ids and marker bits to determine
whether a packet should be forwarded, this patch makes use of those
to determine whether they should be dropped instead (by removing their
might-have-been-fec field).
Spurious lost events coming out of the depayloader can cause the
decoder to stop decoding until the next keyframe and / or request a new
keyframe, and while this is not desirable it makes sense to forward
that information when we have other means to determine whether a lost
packet was indeed a FEC packet, as is the case with VP8 / VP9 payloads
when they carry a picture id.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/769>
The AVC codec_data has a flaw that it can only accomodate
31 SPS headers, even though H.264 can have 32, and 255 PPS,
when there can be 256 in H.264. When streaming RTP some
clients like to cycle through SPS/PPS ids when changing
configuration and can eventually accumulate a full set.
In that case, we have no choice but to discard one (oldest)
entry, or else the count written into the codec_data is wrong
and downstream decoding failures ensue.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/775>
Due to us not properly acknowleding the time when the last RTX was sent
when scheduling a new one, it can easily happen that due to the packet
you are requesting have a PTS that is slightly old (but not too old when
adding the latency of the jitterbuffer), both its calculated second and
third (etc.) timeout could already have passed. This would lead to a burst
of RTX requests, which acts completely against its purpose, potentially
spending a lot more bandwidth than needed.
This has been properly reproduced in the test:
test_rtx_not_bursting_requests
The good news is that slightly re-thinking the logic concerning
re-requesting RTX, made it a lot simpler to understand, and allows us
to remove two members of the RtpTimer which no longer serves any purpose
due to the refactoring. If desirable the whole "delay" concept can actually
be removed completely from the timers, and simply just added to the timeout
by the caller of the API. But that can be a change for a another time.
The only external change (other than the improved behavior around bursting
RTX) is that the "delay" field now stricly represents the delay between
the PTS of the RTX-requested packet and the time it is requested on,
whereas before this calculation was more about the theoretical calculated
delay. This is visible in three other RTX-tests where the delay had
to be adjusted slightly. I am confident however that this change is
correct.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/789>
The crash is caused by a race condition where the render thread
calls a method on the QtGLVideoItem instance that was
previously destroyed by the main thread.
Also, less frequently, QtGLVideoItem::onSceneGraphInitialized
is called when QQuickItem::window is null, also causing a crash.
Fixes#798
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/777>
This action signal will delegate to clear-ssrc onto the rtpssrcdemux element
associated with the session. This allow rtpbin users to clear pads and
elements for a specific ssrc that is known to no longer be in use. This
happens when a pad is reused in rtpsrc or ristsrc.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/736>
