* RTCP sender reports every three seconds.
Ideally, we should be sending this based on data rate.
But, increasing frequency a little as a lost sender report
means the client may not have sender report for 10 seconds
and that could affect sync. We do receiver reports once a second.
Thought of setting this to that level too, but not making a big change
from existing rate.
Also, simplifying the RTCP send loop. Don't need to hold and
do the processing after collecting all reports.
* consistent use of GetSubscribedTracks
* Experimental flag to try time stamp adjustment to control drift.
There is a config to enable this.
Using a PID controller to try and keep the sample rate at expected
value. Need to be seen if this works well. Adjustment are limited
to 25 ms max at a time to ensure there are no large jumps.
And it is applied when doing RTCP sender report which happens
once in 5 seconds currently for both audio and video tracks.
A nice introduction to PID controllers - https://alphaville.github.io/qub/pid-101/#/
Implementation borrowed from - https://github.com/pms67/PID
A few things TODO
1. PID controller tuning is a process. Have picked values from test from
that implementation above. May not be the best. Need to try.
2. Can potentially run this more often. Rather than running it only when
running RTCP sender report (which is once in 5 seconds now), can
potentially run it every second and limit the amount of change to
something like 10 ms max.
* remove unused variable
* debug log a bit more
* Keep track of expected RTP time stamp and control drift.
- Use monotonic clock in RTCP Sender Report and packet times
- Keep the time stamp close to expected time stamp on layer/SSRC
switches
* clean up
* fix test compile
* more test compile failures
* anticipatory clean up
* further clean up
* add received sender report logging
With subscription manager, there is no need to tell a publisher
about a subscriber going away. Before subscription manager,
the up track manager of a participant (i. e. the publisher side)
was holding a list of pending subscriptions for its published tracks
and that had to be cleaned up if one of the subscriber goes away.
That is not the case any more.
Also set publisherID early so that subscription permission update has
the right publisherID. In fact, saw an empty ID in the logs and saw
that we still have the disallowed subscription handling which is not
necessary any more.
* hopefully more stable tests
* do eventual checks as some callbacks happen in go routines.
Needs a bit more work to ensure that some conditions do not happen.
But, with goroutines, the amount of wait is always tricky.``
* Support simualting subscriber bandwidth.
When non-zero, a full allocation is triggered.
Also, probes are stopped.
When set to zero, normal probing mechanism should catch up.
Adding `allowPause` override which can be a connection option.
* fix log
* allowPause in participant params
* Use bandwidth requested from last allocation.
With overshoot/opportunistic forwarding, It is possible that
bitrate at target layers is 0. So, use bandwidth requested
from last allocation which shouold have a correct value.
Still need to think about using the latest bit rates to get
the requested bandwidth. It is possible that bitrates have
changed since last allocation. That was the idea behind using
the latest bitrates, but it could return 0. Accounting for it
runs into a few scenarios. Last allocation has number from
last allocation and is a good indicator of the need.
* race
* Update go deps
Generated by renovateBot
* use generics with Deque
---------
Co-authored-by: renovate[bot] <29139614+renovate[bot]@users.noreply.github.com>
Co-authored-by: David Zhao <dz@livekit.io>
When a recorder is in the room, we would skip grace period due
to a bug with how LastLeftAt was set. This would cause RoomComposite
templates to exit immediately if the last participant in the room reconnects.
A few more candidates to think about demoting
- Publisher mute changes
- Forwarder -> layer lock/upgrade/downgrade/overshoot adjusting
- StreamAllocator
* Make connection quality not too optimistic.
With score normalization, the quality indicator showed good
under conditions which should have normally showed some badness.
So, a few things in this PR
- Do not normalize scores
- Pick the weakest link as the representative score (moving away from
averaging)
- For down track direction, when reporting delta stats, take the number
of packets sent actually. If there are holes in the feed (upstream
packet loss), down tracks should not be penalised for that loss.
State of things in connection quality feature
- Audio uses rtcscore-go (with a change to accommodate RED codec). This
follows the E-model.
- Camera uses rtcscore-go. No change here. NOTE: THe rtscore here is
purely based on bits per pixel per frame (bpf). This has the following
existing issues (no change, these were already there)
o Does not take packet loss, jitter, rtt into account
o Expected frame rate is not available. So, measured frame rate is
used as expected frame rate also. If expected frame rate were available,
the score could be reduced for lower frame rates.
- Screen share tracks: No change. This uses the very old simple loss
based thresholding for scoring. As the bit rate varies a lot based on
content and rtcscore video algorithm used for camera relies on
bits per pixel per frame, this could produce a very low value
(large width/height encoded in a small number of bits because of static content)
and hence a low score. So, the old loss based thresholding is used.
* clean up
* update rtcscore pointer
* fix tests
* log lines reformat
* WIP commit
* WIP commit
* update mute of receiver
* WIP commit
* WIP commit
* start adding tests
* take min score if quality matches
* start adding bytes based scoring
* clean up
* more clean up
* Use Fuse
* log quality drop
* clean up debug log
* - Use number of windows for wait to make things simpler
- track no layer expected case
- always update transition
- always call updateScore
