* Consolidating PLI throttle
Use the throttler in `sfu.WebRTCReceiver`.
Does change shape of config object.
* Move PLIThrottleConfig to sfu.WebRTCReceiver
* fix test compile
* Cleaning up unused stuff
* improve readability
* RTT
- Calculate down track RTT using RTCP Receiver report
- Surface it back to the participant
- Participant updates all its published trackes
(throttled to limit update to once in 5 seconds)
- That propagates to all the upstream sfu.Buffer and the nacker.
So, we will have RTT throttled NACKs.
* rtt callback
* Use ParticipantInfo version to ensure consistency
Deprecating time.Time and avoid locking unnecessarily
* properly adjust ulimit. update protocol
* Save initial version from params
* get rid of metadata field, use grants copy
* fix test
* Ensure autosubscribe is honored when subscription permissions were granted later
* negotiate even if no media has been added
* don't double-negotiate
This ensures client reconnect attempts would be successful for long running rooms. It also fixes inaccurate permissions that were set incorrectly when full reconnections take place.
* Refactor media track subscriptions
- To enable re-use of common bits
- Add max quality from other nodes
* Lock close handlers slice
* Reverting multiple on close handlers of downtrack, unclear if it is needed yet
* Make Logger a pointer
* audio level in MediaTrack like remote media track
* Cleanup
* Add a no subscribers callback
* Add method to update subscribed quality from another node
* loss proxying from remote node
* Address comments from David
* create subscriber node quality map
* WIP commit
* update protocol
* Fixing a test and catching one place where casting was missed
* Fix one more spot which need conversion from livekit.RoomName -> string
* do not covert list
* WIP commit
* Add some tests
* allowedSubscribers uses participant sid
* correct variable name
* correct another variable name
* Add ParticipantSid to SubscriptionPermissionUpdate message
* protocol v0.11.2
* WIP commit
* WIP commit
* fix tests
* Remove unused code
* Close uptrack manager
* Remove duplicate close
* move comment to the correct line where the loop could be long
* Fix disallowed list revocation, thank you Jie
* Remove unneeded interface method
* RemoveSubscriber in Participant
* Clean up disallowed subscriptions and handle permissions on new track addition
* add test for track addition after permission set
* Remove unnecessary check
* Scoped speaker update
Include only participants a participant is subscribed to.
NOTE: Not doing this for active speaker changes for Protocol < 3.
* correct comment spelling
* audio connection quality mos for publisher stats
Signed-off-by: shishir gowda <shishir@livekit.io>
* Update tests
Signed-off-by: shishir gowda <shishir@livekit.io>
* Change ratings range, increase default rtt to 80
Signed-off-by: shishir gowda <shishir@livekit.io>
* Use stats worker to get total packets to find %lost in window
Signed-off-by: shishir gowda <shishir@livekit.io>
* Update go dep
Signed-off-by: shishir gowda <shishir@livekit.io>
* Increase interval of score cal to 5 seconds
Signed-off-by: shishir gowda <shishir@livekit.io>
* use lastSequenceNumber in reports to find total packets
Signed-off-by: shishir gowda <shishir@livekit.io>
* Account for delay while calculating scores
Signed-off-by: shishir gowda <shishir@livekit.io>
* Fix minor typo
Signed-off-by: shishir gowda <shishir@livekit.io>
* Add connection stats/score to subscribed audio tracks
Signed-off-by: shishir gowda <shishir@livekit.io>
* Cleanup
Signed-off-by: shishir gowda <shishir@livekit.io>
* Ignore duplicate LastSequenceNumbers in rtcp reports
Ignore if sequence number is less than what was recieved
Signed-off-by: shishir gowda <shishir@livekit.io>
* Move video track score calc to media/downtracks
Signed-off-by: shishir gowda <shishir@livekit.io>
* Deprecate SubscribeLossPercentage() as score calc is now handled downstream
Signed-off-by: shishir gowda <shishir@livekit.io>
* Initialize connection score to excellent
score is calc at 5sec interval. Client fetches score before first
score is computed
* Update test cases for connection quality
Signed-off-by: shishir gowda <shishir@livekit.io>
* WIP commit
* SubscribedQualityUpdate message to send list of currently subscribed
qualities for a simulcast video publisher
* Correct subscriberID
* goimports
* Do quality update on add/remove of subscribed track
* do not update quality when admin mute is active
* update quality on admin unmute
* Update protocol version
* Simplify max subscribed quality loop per David's suggestion
* WIP commit
* deficient handling
* Add missing ProvisionalAllocatePrepare
* adjust state on track removal
* Increase test timeout
* - Add comments about cooperative routines
- Take down transition if available in cooperative scheme
- Use layer comparison when taking down transition. Because of when the
bitrate is measured, it is not always guaranteed bandwidthDelta is -ve
when moving down.
- Do not add track to stream allocator till bind.
* make comment better
* a bit more clear comments
* Use OnBind on subscribed track
* Stream Allocator Try 3
Making an intermediate PR to do
- Special treatment for screen share tracks
- When allocating all tracks,
o try to stream all tracks by starting with the lowest layer
o multi-pass across tracks to get a more even distribution
Not yet done:
-------------
In deficient state,
o Allocate a specific track on a change
o Steal from other tracks
* Correct sense of managed track
* have to range to copy
* generate
* fix VideoLayers compare
* Use t.simulcasted
