- prevent some escape to heap
- avoid copying by using a ring buffer for receiver reports (probably
should remove this as this is for debugging only and data so far has
shown clients sending bad data and nothing more.)
Removing some logs which have not been useful in terms of insights other
than saying that there are a bunch of packets missing. Going to start
looking at gaps in terms of time if the inter-packet gap is too high.
Also, using logging these events as first 20 and then every 200.
* Guard against timestamp inversion in RED -> Opus conversion.
Seeing timestamp inversion (sequence number is +1, but timestamp is
-960, i.e. 20ms) in the RED -> Opus conversion path. Not able to spot
any bugs in code. So, logging details upon detection and also dropping
the packet. If not dropped, downstream components like Egress treat it
as a big timestamp jump (because sequence number is moving forward) and
try to adjust pts which ends up causing drops.
* do not log time reversal at the start
* typo
* Optimise some bits in rtpstats_receiver
RTPStatsReceiver.Update is one of the high CPU bits. Taking some
suggestions from Cursor. Makes the `Update` function verbose though :-(
* zap.Inline logging fields
* rename
* Reducing some info level logs.
Also, relaxing the check for runaway RTCP receiver report to allow for
rollover to catch up if it is not too far away.
* set logger
With audio simulcast codecs, it is possible that the clock rate of the
primary codec is different from the secondary codec. If a subscriber
binds to the secondary codec, the clock rate should be set correctly. Do
it at bind time.
* Refactor receiver and buffer into Base and higher layer.
To be able to share code/functionality with relay.
* WIP
* WIP
* WIP
* WIP
* WIP
* WIP
* WIP
* WIP
* clean up
* deps
* fix test
* fix test
* Add support for RTP stream restart.
When an unhandled packet is encountered, try a restart sequence.
Restart happens when 5 packets with contiguous sequence numbers and same
or increasing time stamps are received. Note that this does not work for
B-frame type of scenarios, but that is true for receive path handling
even before this. As WebRTC does not use B-frames, it is fine. But,
needs to be looked at again if B-frames are necessary.
It is controlled by a config that is disabled by default.
* clean up
* debug log
It is possible that the stream stops just after start and
restarts much later introducing a large gap in sequence number.
That could look like an unhandled case because the wrap back handler
does not have enough packets yet.
Let other checks based on time stamp gap take effect and only if that
also leaves the sequence number unhandled, drop the packet.
- New bucket API to pass in max packet size and sequence number offset
and seequence number size generic type
- Move OWD estimator to mediatransportutil.
publisher peer connection.
While cleaning up during single peer connection changes, unintentionally
removed handler.
Also, another small change to log first packet time adjustment after
increment.
* Adjust for hold time when fowarding RTCP report.
When passing through RTCP sender report, holding it for some time before
sending means the remote receiver could see varying amount of
propagation delay if the remote uses something like local_clock -
ntp_sender_report_time and adapting to it.
Ideally, SFU should just forward RTCP Sender Report, but the current pull model to
group RTCP sender reports makes it a bigger change. So, adjust it by
hold time.
Also add a initial condition for one-way-delay estimator which can init
with a smaller value of latency if the first sample to measure
one-way-delay itself experienced higher delay than the prevailing
conditions.
* variable name
* log as duration
- With probing the packet rate can get high suddenly and remote may not
have sent receiver report as it might be sending for the non-spikey
rate. That causes metadata cache overflows. So, give RTX more cahe.
- Don't need a large cache for primary as either reports come in
regularly (or they are missing for a long time and having a biger
cache is not the solution for that, so reduce primary cache size)
- Check for receiver report falling exactly back by (1 << 16). Had done
that change in the inside for loop, but missed the top level check :-(
Receiver view is used for connection quality.
Sender view is used for analytics. One thing that this introduces is
that sender view uses the packet loss information from receiver view as
true loss is available only in the RTCP Receiver Reports received from
the remote side. So, the time alignment is off, i. e. receiver report
happens periodically and it includes information till the time at which
it was sent from remote side, but sender could have sent more packets
after that time.
The split should ensure that analytics does not rely on remote side
sending proper receiver repoerts albeit at slight misalignment of loss
statistic for remotes that send RTCP RR (which should be majority of the
cases)
* Do not skip due to large RR interval.
With sequence number adjustment, it will report some packets missing
which is fine.
* do not seed if already initialized
No functional change, just logging reason was confusing.
Also, log no packets case. Seeing some instances in staging where there
are periods of no packets received. Trying to understand better.
initially.
Out-of-order reception is detected and dropped before this change. By
moving the adjustment to that point, potentially more reports can be
processed.
* Remove alloc in packet forwarding path.
Unlikely logger creation was doing allocs. Replace it with a function
like in rtpstats_receiver.go so that allocations do not happen
unnecessarily.
* variable rename
* one more place
