* Add debug for receiver restart.
Have a suspicion that something is deadlocking between restart receiver
and buffer bind during replay. Adding debug to get a better picture of
state of receiver restart.
* consistent logging
* Set up audio config in audio level module when config is updated.
It is possible to get audio config after bind (bind is where the audio
level module is created) for remote tracks. So, split out setting audio
level config in audio level module and invoke it when config is updated.
* coderabbit review
* prevent divide-by-0
* not active before config
* 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.
* Do not restart receiver on codec change mid-session.
This is not supported and was an erroneous change during the
receiver_base/buffer_base + RTP stream restart consolidation. Also make
the codec munger creation more resilient.
* fix test
In some paths, it is better to preserve pre-recorded time. So, make the
base implementations preserve the RTCP Sender Report receive time.
Also, add a method to enable forwarding packet arrival time. Could be
used across relay.
* Rework receiver restart.
- Protect against concurrent restarts
- Clean up and consolidate code around restarts
- Use `RestartStream` of buffer rather than creating new buffers.
* fix test
* Minor refactor in buffer base and audio level
- Make a function for `restartStream`. Will be useful
when external signal needs to restart a stream. Also restart all the
bits (audio level, dd parser and frame rate calculator)
- make an audio level mode with RTP timestamp so that some state can be
moved out of buffer base
* clean up
* log restart
This is not all of it as it is not possible (or at least I do not know
of a way) to get all suggestions for a repo/project. Did this via loop
searching mainly and taking the modernize suggestions.
* Return extended sequence number only and not packet.
Callers need only the extended sequence number.
Extended packet could get release if the forwarder processes it before
caller accesses it causing a data race.
* grow bucket in a go routine
* 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
* Store buffer after creating it.
Also changing signature of creator function as it could call TrackInfo()
and get into a deadlock.
* fix double unlock
* add some more debug logging
* 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.
* 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
- New bucket API to pass in max packet size and sequence number offset
and seequence number size generic type
- Move OWD estimator to mediatransportutil.
* Use sync.Pool for objects in packet path.
Seeing cases of forwarding latency spikes that aling with GC.
This might be a bit overkill, but using sync.Pool for small +
short-lived objects in packet path.
Before this, all these were increasing in alloc_space heap profile
samples over time. With these, there is no increase (actually the lines
corresponding to geting from pool does not even show up in heap
accounting when doing `list` in `pprof`)
* merge
* Paul feedback
Raja Subramanian