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
These are outside the expected range of sequence numbers and should be
clipped, especially for RTSP they might belong to packets from before a seek
or a previous stream in general.
We (currently?) can't really handle gaps between RTP packets if they're not
properly timestamped. The current code would go into calculations with
GST_CLOCK_TIME_NONE and then cause assertions everywhere. It's probably
better to error out cleanly instead.
We used to setup an iterator with 1 GValue set with a NULL object
pointer which is not the normal way to do that. Instead we should make
sure that the first call to gst_iterator_next returns GST_ITERATOR_DONE.
When the RTT and jitter are very low (such as on a local network), the
calculated retransmission timeout is very small. Set some sensible lower
boundary to the timeout by adding a new property. We use the packet
spacing as a lower boundary by default.
The jitterbuffer shouldn't force clock-rate on its sink pad, this will cause a negotiation issue since rtpssrcdemux doesn't have the clock-rate and doesn't add it to the caps. The documentation states that the clock-rate can either be specified through the caps or through the request-pt-map signal, so we must remove clock-rate from the pad templates and we must accept the GST_EVENT_CAPS if the caps don't have the clock-rate.
https://bugzilla.gnome.org/show_bug.cgi?id=734322
Implement 3 different cases for handling the SR:
1) we don't have enough timing information to handle the SR packet and
we need to wait a little for more RTP packets. In that case we keep
the SR packet around and retry when we get an RTP packet in the
chain function.
2) the SR packet has a too old timestamp and should be discarded. It is
labeled invalid and the last_sr is cleared.
3) the SR packet is ok and there is enough timing information, proceed
with processing the SR packet.
Before this patch, case 2) and 1) were handled in the same way,
resulting that SR packets with too old timestamps were checked over and
over again for each RTP packet.
They are very confusing for people, and more often than not
also just not very accurate. Seeing 'last reviewed: 2005' in
your docs is not very confidence-inspiring. Let's just remove
those comments.
Make a new method to disable the jitterbuffer buffering.
Rework the update_estimated_eos() method. Calculate how much time
there is left to play. If we have less than the delay of the
jitterbuffer, we disabled buffering because we might never be able to
fill the complete jitterbuffer again.
If we receive an EOS event, disable buffering. We will drain the
buffer and eventually push the EOS event out.
When we reach the estimated NPT timeout and we didn't receive an EOS
event, make one and queue it so that it can be pushed.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728017
Rework the logic to make buffering messages a little, make sure we
don't make the same message multiple times.
Consider the buffer full when EOS was received.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728017
When we are buffering, we can't block and wait for the serialized query
to complete because the jitterbuffer will not try to forward the query
while buffering. Instead, just refuse the query.
gstrtpjitterbuffer.c: In function 'gst_rtp_jitter_buffer_loop':
gstrtpjitterbuffer.c:2978:3: error: 'result' may be used uninitialized in this function
while (result == GST_FLOW_OK);
^
Several conditional statements perform comparison on RTP sequence
numbers without taking the sequence number rollover into account.
Instead, use the gst_rtp_buffer_compare_seqnum function to perform the
comparison.
https://bugzilla.gnome.org/show_bug.cgi?id=725159
If the expected packet (do_next_seqnum is TRUE) is the one we requested
for retranmission earlier, do the logic to update the retransmission
statistics as well before setting up the timers for the next expected
packet.
Also reset the retransmission counter if the timer is reused for another
seqnum.
Use the round-trip-time and average jitter to dynamically calculate the
retransmission interval and expected packet arrival time.
Based on patches from Torrie Fischer <torrie.fischer@collabora.co.uk>
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711412
Don't use the current time calculated from the tmieout loop for when we
last scheduled the NACK because it might be unscheduled because of a max
packet misorder and then we don't accurately calculate the current time.
Instead, take the current element running time using the clock.
Don't reset the expected output seqnum when clearing the pt map because this
could stall the jitterbuffer forever.
Add a unit test for this.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=709800
The problem here was that the jitterbuffer lock was unlocked to push
the event, but that caused another thread to remove the timer currently
being processed, probably because the amount of rtx events
(and therefore timers) was getting too high. The solution is to
unlock and push the event only after timer processing has finished.
fixes https://bugzilla.gnome.org/show_bug.cgi?id=711131
When flush-stop arrives before we process the result of the _push() in the
loop function, we might pause even though we are not flushing anymore. Fix this
race by waiting for the srcpad loop function to completely pause after doing the
flush-start.
If we were not waiting for the missing seqnum when we insert the lost packet
event in the jitterbuffer, we end up not updating the next_seqnum and wait
forever for the lost packets to arrive. Instead, keep track of the amount of
packets contained by the jitterbuffer item and update the next expected
seqnum only after pushing the buffer/event. This makes sure we correctly handle
GAPS in the sequence numbers.
Always prepare a lost event in the jitterbuffer, it is to wake up and make the
pushing thread continue. We drop the event when we are not supposed to push lost
events downstream.
Schedule the lost event by placing it inside the jitterbuffer with the seqnum
that was lost so that the pushing thread can interleave and push it properly.
Make the jitterbuffer operate on a structure containing all the packet
information. This avoids mapping the buffer multiple times just to get the RTP
information. It will also make it possible to store other miniobjects such as
events later.
Improve the order of the timeout events, if there are timers with the same
timeout, we want to trigger the lowest seqnum first. For this we need to loop
over the complete array of timers to find the best one before triggering the
timeout.
First send the lost event, then update the next_seqnum counter and then
send the signal to the pushing thread that it can retry to push a buffer. This
avoids pushing out buffers before the lost event is pushed.
There is no need to unschedule the timer in flush-start, flush-stop will remove
the timers and unschedule.
Unschedule the current timer before attempting to join the timer thread.
Keep a separate delay in the timer so that we still know the original timestamp
of the packet that this timer refers to. We can then place the correct
running-time in the Retransmission event.
Instead of pushing the lost event from the chain function, schedule a timeout
that will push the lost event from the timer thread. This avoid blocking the
upstream thread while we push and sync the event.
When we have a large number of missing packets, generate one lost event for all
the packets that have no chance of being pushed out in time.
Fix and activate unit test for large gaps.
Keep track of the current time in the timeout loop.
Loop over all timers and trigger all the expired ones, we can do this in the
same loop that selects the new best timer.
Also update the timers when retransmission is disabled. We need to
do this because when we added LOST timers when we detected missing packets and
we need to remove those timers when the packet finally arrives.
We keep the DTS and PTS in running-time inside the jitterbuffer. Make sure to
transform it back to a buffer timestamp before pushing out the buffer.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=707931
Restructure handling of incomming packet and the gap with the expected seqnum
and register all timers from the _chain function.
Convert a timer to a LOST packet timer when the max amount of retransmission
requests has been reached.
If we change the seqnum of an existing timer and we were waiting for
that timer, unschedule it. If we change the timeout of an existing timer and we
were waiting on it, only unschedule when the new time is smaller.
Make the jitterbuffer schedule the timeouts based on the DTS instead
of the PTS. This makes it all smoother with reordered frames and gives
the decoder time to reorder the frames in time.
Refactor the jitterbuffer code. Make separate function for peeking a buffer,
pushing the next buffer, waiting for timeouts and handling the timeouts.
The main loop now tries to push as many buffers as it can until it runs out of
buffers or when it detects a seqnum discont. Then it will wait for some event to
happen before attempting to push more buffers.
Make methods to register timeouts in an array. These timeouts are registered
when we detect a missing packet, sync for the first packet or when we find an
estimation for the end-of-stream.
This greatly simplifies and clarifies the code and also makes it possible to
register more complicated timeout schemes later.
