* Support XR request/response for rtt calculation
* Update pkg/sfu/downtrack.go
Co-authored-by: David Zhao <dz@livekit.io>
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Co-authored-by: David Zhao <dz@livekit.io>
* Buffer size config for video and audio.
There was only one buffer size in config.
In upstream, config value was used for video.
Audio used a hard coded value of 200 packets.
But, in the down stream sequencer, the config value was used for both
video and audio. So, if video was set up for high bit rate (deep
buffers), audio sequencer ended up using a lot of memory too in
sequencer.
Split config to be able to control that and also not hard code audio.
Another optimisation here would be to not instantiate sequencer unkess
NACK is negotiated.
* deprecate packet_buffer_size
* Reduce heap for dependency descriptor in forwarding path.
Marshaled dependency descriptor is held in sequencer adding heap objcts.
Store DD bytes in sequencer to avoid heap usage.
Also, accomodating over sized objects via storing in slice and using it
in case the bytes do not fit in the internal array.
NOTE: Marshal DD still does a make([]byte...), but I think it should be
on the stack given the short use of it. Have to verify.
* fix test and also add cases for oversized codec/dd bytes
* Remove some logs.
Also, changing Errorw -> Warnw in a bunch of places.
Going to move towards using `Errorw` for cases where a functionally
unexpected condition happens, i.e by design a condition should not
happen yet it triggered kind of scenarios.
* log error
Firefox on Windows 10 seems to be producing simulcast tracks with
duplicate RID. That causes a leak as only one buffer is processed.
Ignore duplicate rid.
NOTE: This is not perfect as the actual layer -> rid is indeterminable
at addition time. It would require looking at packets to determine the
video dimensions and match to rid/layer to figure out which one is
correct and which one is duplicate.
To simplify though, taking the first one and dropping later ones.
This could mean the correct resolution is not streamed, but that should
be okay. The leak is far more destructive.
Was at 20 when LOST was introduced, but was going to 20 even when under
not LOST conditions. When there are packets, want the min to be at 30.
Going down to 20 resulted in reporting LOST quality even when packets
were flowing (although they were experiencing heavy loss and quality
would have been very bad, yet they are not lost).
Also, sample warning about adding packet to bucket even more.
* Add debug to understand VP9 freezes.
Have reports of VP9 freezing in some rooms.
Some data indicates that NACKs are received by SFU, but cannot get RTP
packet when that happens. It is possible that the NACKs are all from
dropped packets. Adding some debug to understand drops/NACKs better.
* enable DD debug
* comment out DD debug
* markers
* add back log about diff length mismatch
* add back key frame mismatch logging
* log skipped drops also
* Use Seque in ops queue.
Standardizing some uses
- Change OpsQueue to use Deque so that it can grow/shrink as necessary and
need not worry about channel getting full and dropping events.
- Change StreamAllocator and TelemetryService to use OpsQueue so that
they also need not worry about channel size and overflows.
* Address feedback
* delete obvious comment
* clean up
It is possible that onBindAndConnectedChanged gets executed in such a
way that `writable` does not have the correct value in some very rare
timing case (i. e. case like two executions of the function is racing
and one atomic was read on first exeuction and second execution runs and
sets `writable` and then first execution completes the sets `writable`
to incorrect value based on stale read of first execution).
Prevent it by executing under bind lock.
* Consolidate TrackInfo.
TrackInfo was spread across a bit. Consolidating it.
* TODO comments
* test
* update TrackInfo on SSRC change
* further consolidation
* log mimes only
* update receivers on SSRC set
* clone proto on return
* feedback: break loop on mime match
* prevent data race
cnderrauber