* Make data message naming a bit more consistent.
OnDataPacket and OnDataMessage half-way in the chain made it confusing
(for me at least). Use same name throughout.
API still uses SendDataPacket, but that is not harder to read.
* test
* Use published track for model access in data down track.
No need to pass in the model.
* implementation type assertion
* remove redundant log key
* one more type
* Close interface method
* clean up redundant log key
* remove unused method
* some interface changes to support data tracks in replay
* correct mock signature
Published counter was bumped up only when not migrating in, but it was
decremented when a migrating participant leaves without expectation to
resume. That could have resulted in negative counts.
Always change counters irrespective of migration or expected to resume
on leave. Control events send based on migration/resume.
* Some logging changes.
Trying to chase a case of large sequence number gap on subscriber side
where packets are sent after a long time.
* return values instead of logging
* 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
* Avoid duplicate track add to room track manager.
Don't have proof that this happens, but in the leak chase, this is
another component at room level and holds references to tracks. Guessing
this is not cleaning tracks till room is closed.
* add a report
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.
Seeing a bunch of objects in ReadStreamSRTP.write which does not make
a lot of sense as the function does not allocate anything
(8fe528a0c4/stream_srtp.go (L68-L77))
RTP buffer was marked noinline in an easrlier PR.
Marking RTCP buffer write also as noinline to check if heap reporting
changes.
* switch participant callbacks to room to listener interface
* mage generate
* clean up
* clear listener
* clean up
* use interface in up data track manager
* tweaks
* Paul feedback - should reduce the diff as this keeps the room handlers as is except making methods for a couple of anonymous handlers
* clean up
When a participant is closing, RTCP readers should be cleaned up from
factory even if the participant is expected to resume. The resumed
participant will be a new participant session and peer connection(s) and
everything will be set up again.
