* do not NACK on fec recovered packet
* refactor common code into unlockAndInvokeFECRecovery
* add FEC stats to RTPStats
* add log for when fecdecoder unavailable after creation
* rename markAsFECRepair to setFECRepairBufferLimit for clarity
* refactor packet generation for tests into util
* fix test race in CI
* move the unlocks back inline for readability
* update RTPstats
* Fix FEC write return after merging RTX feedback changes
* Use shared RTP packet helper in merged FEC NACK test
* agent: release JobTerminate handler when job ends before registration
JobRequest registered the job's JobTerminate handler after AssignJob
returned, so an ended UpdateJobStatus or worker disconnect handled in
between deregistered nothing and the handler leaked. AssignJob also sent
the assignment before recording the job as running, so an immediate
ended update was dropped.
Record the job before sending the assignment, and after registering the
handler recheck under h.mu that the job is still running and its worker
still registered, deregistering otherwise.
Fixes#4901
* replace utils.CloneProto with proto.CloneOf
* agent: wait on worker readiness and poll for released handlers in test
* sfu: add opt-in verbatim abs-capture-time forwarding
Downtrack ACT forwarding gates on an RTCP sender report and
unconditionally rewrites the capture timestamp into SFU clock domain,
which suits cross-stream A/V sync but breaks consumers that need the
original publisher capture time (e.g. end-to-end latency measurement).
Add ForwardAbsCaptureTimeVerbatim (RTC config: forward_abs_capture_time_verbatim),
off by default, which skips both the gate and the rewrite.
* fix: use DisableSenderReportPassThrough setting to also gate abs-capture-time forwarding
- 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.
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
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
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.
* Log more details of RTP stats snap shots.
Seeing cases of loss more than 100%. Logging snap shots to understand it
better.
* log message
* use delta to update packets lost from RR
* remove cast
* Make lite version of RTPStatsReceiver called RTPStatsReceiverLite.
Refactor around that.
Will probably make some more flavors to have lighter versions still.
* update deps
* use MarshalLogArray
* use util
* Splitting out rtp stats stuff into its own package.
Going to be making some lighter versions of these.
Will be cleaner to have all of these grouped together.
So, as a first step, just making a package for it.
* tests
David Chen