The buffer is not for padding packets. So, calculate
adjusted sequence numbers before comparing against size.
Also, it is possible that invalidated slot is accessed
due to not being able to exclude padding range. This was
causing time stamp reset to 0. Will remove the error log
after this goes out and the condition does not show up
for a few days.
Seeing a time stamp jump that I am not able to explain.
Basically, it looks like the time stamp doubles at some
point. There is no code which doubles the timestamp.
Can understand an erroneous roll over/wrap around, but
doubling is very strange.
So, logging only audio packets. Will disable as soon
as I have some smaples from canary.
Seeing some unexplained jumps in sender report time stamp
in canary. Wonder if the calculated clock rate is way off
during some interval. Logging clock deviations to understand
better.
* Error log of padding updating highest time to get backtrace.
* Do not update highest time on padding packet.
Padding packets use time stamp of last packet sent.
Padding packets could be sent when probing much after last packet
was sent. Updating highest time on that screws up sender report
calculations. We have ways of making sure sender reports do not
get too out-of-whack, but it logs during that repair.
That repair should be unnecessary unless the source is behaving weird
(things like publisher sending all packets at the same time, publisher
sample rate is incorrect, etc.)
* Log highest time update on padding packet.
Seeing a strange case of what looks like highest time getting
updated on a padding packet. Can't see how it happens in code.
So, logging to check. Will be removing log after checking.
* log sequence number also
* Use 32-bit time stamp to get reference time stamp on a switch.
With relay and dyncast and migration, it is possible that different
layers of a simulcast get out of sync in terms of extended type,
i. e. layer 0 could keep running and its timestamp could have
wrapped around and bumped the extended timestamp. But, another layer
could start and stop.
One possible solution is sending the extended timestamp across relay.
But, that breaks down during migration if publisher has started afresh.
Subscriber could still be using extended range.
So, use 32-bit timestamp to infer reference timestamp and patch it with
expected extended time stamp to derive the extended reference.
* use calculated value
* make it test friendly
Seeing a large positive gap which I am not able to explain.
Wondering if at some other time, a large negative is happening
and the large positive is just a correction.
* Log potential sequence number de-sync in receicer reports.
Seeing some cases of a roll over being missed. That ends up
as largish range to search in an interval and reports missing packets
in the packet metadata cache.
Logging some details.
* just log in one place
Ideally, can remove the nil return when there are too many packets
as we have more information with extended sequence numbers, but
logging duration first to understand what is happening better.
When converting from RED -> Opus, if there is a loss, SFU recovers
that loss if it can using a subsequent redundant packet. That path
was not setting the extended sequence number properly.
Also, ensuring use of monotonic clock for first packet time adjustment
also.
* Log resync at Infow.
Seeing potentially large sequence number jumps on a resync.
And it seems to happen on a lot of subscribe/unsubscribe.
Logging at Infow to understand better.
Probably need to find a way to avoid resync. But, logging for now to
check if I can catch one.
* Remove resync and log large sequence number jumps
* Reduce packet meta data cache - part 1
Packet meta data cache takes a good amount of space.
That cache is 8K entries deep and each entry is 8 bytes.
So, that takes 64KB per RTP stream.
It is mostly needed for down stream to line up with receiver reports.
So, removing cache from up stream (RTPStatsReceiver) as part 1.
Will look at optimising the down stream in part 2.
* Remove caching from RTPStatsReceiver
* clean up a bit more
* maintain history and fix test
Streaming could start after 16-bits has rolled over. So, have to add
that base back to what is received in receiver report.
Otherwise, it looks like there are not packets received in window
leading to poor quality.
* Split RTPStats into receiver and sender.
For receiver, short types are input and need to calculate extended type.
For sender (subscriber), it can operate only in extended type.
This makes the subscriber side a little simpler and should make it more
efficient as it can do simple comparisons in extended type space.
There was also an issue with subscriber using shorter type and
calculating extended type. When subscriber starts after the publisher
has already rolled over in sequence number OR timestamp, when
subsequent publisher side sender reports are used to adjust subscriber
time stamps, they were out of whack. Using extended type on subscriber
does not face that.
* fix test
* extended types from sequencer
* log
