* 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
* 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
* readability improvement
RTCP messages are going through two channel hops now.
Maybe we don't need that anymore now that the original
problem is diagnosed. But, pushing all RTCP via
the callbackOps channel for now to make it consistent.
The SRTP replay detection was disabled recently.
But, they were effectively getting dropped in `sfu.bucket`.
Doing two things with RTX packets in this PR
1. Update stats - add to packet count and bytes
2. Process header extension - to process TWCC
* WIP commit
* fix test
* More clean up
* cast to right size
* use local variable
* set a default RTT
* Do not log RTX, although need to figure out source of RTX
* fix test
* Cleaning/simplifying some buffer bits
1. NACKs are always inserted in order. So, get rid of
bunch of out-of-order handling in there and simplify.
2. For now, removing triggering a key frame from NACKs.
Let subs drive it.
3. Move to 16-bit sequence numbers except for receiver
report handling. Simplify bits about unwrapping sequence
number on all packets.
4. Remove unused code.
* remove unused field
* Random clean up
* Split out StreamTrackerManager for re-use
* Reset the correct tracker
* use generation counter to exit coroutine
* start only for video and when enabled
* Add RemoveAllTrackers method
* Use an atomic flag to stop stream allocator
* use a mutex for event channel
* RLock while posting event
* lock isStopped flag to prevent posting to closed channel
* Add a resync API to sfu.DownTrack
Also passing in logger with context into sfu package. More to do here
with proper logging context in all modules, but this is a start
* Remove debug code
* fix tests
Note that it is called from Unbind also (previous behaviour). It should
be fine as long as there is no new down track for the same peer added
between close calling it and unbind calling it.
Still not using it. But, now that Pion has merged the initial
version of GCC (Google Congestion Control) for send side bandwidth
estimation, we can integrate it.
We will enable once we have tested and feel comfortable.
* Prevent multiple resume notifications on track
When returning previous allocation as is, reset change to none.
Ideally, would like to have change as a separate return and not in last
allocation. But, also prefer to have last allocation state. For now,
resetting change.
* log bandwidth estimate only on decrease
* Fix tests
Add back adding track to publishedTracks for testing purposes.
* WIP branch
* Don't know why merge from `master` did not catch this change.
* WIP commit
* Hook up all the bits for TWCC to work
* Fix typo
* WIP commit
* Catch up to latest API
* Move RTP/RTCP info config
* Instantiate GCC/TWCC interceptor only when in use
* comment SSBWE parts
* RTCP feedback in publisher config
* Adaptive publishing improvements
* fix tests
* proper comment sentence
* Address comments from David
* proper locking
* fix crash
* Do not start quality timer if not video
* 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
* 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
