* 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
* 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
* Promoting a few logs to Info
Also, adding a couple of more info logs which I will remove later
after some debugging.
* mime type
* Protect pause/max layer
* notify even if not bound
- Do not update jitter on padding only packet.
Padding only packet may not have proper timestamp.
If it does, it probably has the time stamp of the
last packet with payload. That will also affect
jitter calculation, i. e. wall clock time is moving,
but RTP time is the same.
- Do not send `onMaxLayer` changed on bind.
It was probably racing with update when max layer
is updated when adaptive stream is off. There is
no need to send that update as the default would
be OFF. It will be enabled when adaptive stream
subscription turns it on or when max layer is
set when down track bind happens and adaptive stream
is off.
* WIP commit
* Refactor media loss proxy
* Use DynacastQuality and MediaLossProxy from MediaTrack
* fix test
* Remove unused param
* Remove unused interfaces
* Move interface methods to local
* Split out DynacastManager
* have to add codec to dynacast manager
* RUnlock
* fix restart
* Adding API to force quality and also maintain closed state
* Address PR comments
Raja Subramanian