When a packet arrives that has already been considered lost as part of a
large gap the "lost timer" for this will be cancelled. If the remaining
packets of this large gap never arrives, there will be missing entries
in the queue and the loop function will keep waiting for these packets
to arrive and never push another packet, effectively stalling the
pipeline.
The proposed fix conciders parts of a large gap definitely lost (since
they are calculated from latency) and ignores the late arrivals.
In practice the issue is rare since large gaps are scheduled immediately,
and for the stall to happen the late arrival needs to be processed
before this times out.
https://bugzilla.gnome.org/show_bug.cgi?id=765933
When downstream blocks, "lost" timers are created to notify the
outgoing thread that packets are lost.
The problem is that for high packet-rate streams, we might end up with
a big list of lost timeouts (had a use-case with ~1000...).
The problem isn't so much the amount of lost timeouts to handle, but
rather the way they were handled. All timers would first be iterated,
then the one selected would be handled ... to re-iterate the list again.
All of this is being done while the jbuf lock is taken, which in some use-cases
would return in holding that lock for 10s... blocking any buffers from
being accepted in input... which would then arrive late ... which would
create plenty of lost timers ... which would cause the same issue.
In order to avoid that situation, handle the lost timers immediately when
iterating the list of pending timers. This modifies the complexity from
a quadratic to a linear complexity.
https://bugzilla.gnome.org/show_bug.cgi?id=762988
We would queue 5 consective packets before considering a reset and a proper
discont here. Instead of expecting the next output packet to have the current
seqnum (i.e. the fifth), expect it to have the first seqnum. Otherwise we're
going to drop all queued up packets.
No need to use G_GINT64_FORMAT for potentially negative values of
GstClockTimeDiff. Since 1.6 these can be handled with GST_STIME_ARGS.
Plus it creates more readable values in the logs.
https://bugzilla.gnome.org/show_bug.cgi?id=757480
Estimating it from the RTP time will give us the PTS, so in cases of PTS!=DTS
we would produce wrong DTS. As now the estimated DTS is based on the clock,
don't store it in the jitterbuffer items as it would otherwise be used in the
skew calculations and would influence the results. We only really need the DTS
for timer calculations.
https://bugzilla.gnome.org/show_bug.cgi?id=749536
The amount of time that is completely expired and not worth waiting for,
is the duration of the packets in the gap (gap * duration) - the
latency (size) of the jitterbuffer (priv->latency_ns). This is the duration
that we make a "multi-lost" packet for.
The "late" concept made some sense in 0.10 as it reflected that a buffer
coming in had not been waited for at all, but had a timestamp that was
outside the jitterbuffer to wait for. With the rewrite of the waiting
(timeout) mechanism in 1.0, this no longer makes any sense, and the
variable no longer reflects anything meaningful (num > 0 is useless,
the duration is what matters)
Fixed up the tests that had been slightly modified in 1.0 to allow faulty
behavior to sneak in, and port some of them to use GstHarness.
https://bugzilla.gnome.org/show_bug.cgi?id=738363
This reverts commit 05bd708fc5.
The reverted patch is wrong and introduces a regression because there
may still be time to receive some of the packets included in the gap
if they are reordered.
If we have a clock, update "now" now with the very latest running time we have.
If timers are unscheduled below we otherwise wouldn't update now (it's only updated
when timers expire), and also for the very first loop iteration now would otherwise
always be 0.
Also the time is used for the timeout functions, e.g. to calculate any times
for the next timeouts and we would otherwise pass too old times there.
https://bugzilla.gnome.org/show_bug.cgi?id=751636
This reverts commit 0c21cd7177.
If we have multiple immediate timers, we want to first handle the one with the
lowest sequence number... which would be broken now.
Instead of this we should just use a GSequence for the timers, and have them
sorted first by timestamp, and for equal timestamps by sequence number. Then
we would always only have to take the very first timer from the list and never
have to look at any others.
It might just be a late retransmission or spurious packet from elsewhere, but
resetting everything would mean that we will cause a noticeable hickup. Let's
get some confidence first that the sequence numbers changed for whatever
reason.
https://bugzilla.gnome.org/show_bug.cgi?id=747922
We now take the maximum of 2*jitter and 0.5*packet_spacing for the extra
delay. If jitter is very low, this should prevent unnecessary retransmission
requests to some degree.
https://bugzilla.gnome.org/show_bug.cgi?id=748041
Right above we consider lost_packet packets, each of them having duration,
as lost and triggered their timers immediately. Below we use expected_dts
to schedule retransmission or schedule lost timers for the packets that
come after expected_dts.
As we just triggered lost_packets packets as lost, there's no point in
scheduling new timers for them and we can just skip over all lost packets.
https://bugzilla.gnome.org/show_bug.cgi?id=739868
Resetting the jitterbuffer drops all packets and other things, and will cause
a discontinuity in the packets received by the depayloaders. They should now
also flush anything they had pending as the new data will start at a different
position.
https://bugzilla.gnome.org/show_bug.cgi?id=739868
