* 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
* subtrack should check if track was muted by publisher
Signed-off-by: shishir gowda <shishir@livekit.io>
* If all tracks are muted sub/pub then send excellent connection quality
Signed-off-by: shishir gowda <shishir@livekit.io>
* 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
* Use protocol friendly StreamedTracksUpdate
* WIP commit
* Stream allocator update
* subtract the requested bandwidth as delta from Allocate could be adding to bandwidth
* Calculate delta correctly
* correct comment
* Simplify eventCh per David's suggestion
* Simplifying (hopefully) sfu.DownTrack
* Remove unnecessary check as pdding only packets are dropped before that check
* Temporal filtering max layer
* - Split out forwarder bits into a separate structure
- Address comments from Jie and David
* Remove debug and unneeded stuff
* Fix test
* Remove unneeded default initialization
* WIP branch to hash out down stream allocator.
* Plug more bits of stream allocator
* update protocol
* remove SignalRequest_Simulcast (#154)
* Plug more bits of stream allocator
* Handle simulcast track available layers change
* WIP branch to hash out down stream allocator.
* Plug more bits of stream allocator
* Handle simulcast track available layers change
* Adopt signature of call to AdjustAllocation
* Move StreamAllocator to PCTransport and allocate only for subscriber.
* Move streamallocator to ion-sfu
* Start/Stop of streamallocator
* Use StreamAllocator for subscriber bandwidth management.
* Do not allocate in ADD_TRACK
* Set payload in constructor
* - Add some logging
- Protocol message to notify clients of paused/resumed streams
* named return
* oops correct sense of isPausing
* Update pkg/sfu/streamallocator.go
Committing David's suggestion.
Co-authored-by: David Zhao <david@davidzhao.com>
* - Log estimate changes/commits.
- Catch more than epsilon change oscillating for long time.
Co-authored-by: David Colburn <xero73@gmail.com>
Co-authored-by: David Zhao <david@davidzhao.com>
* configurable node track limit
* sample config
* todos
* end of file new line
* default max num tracks
* bandwidth limit
* client message for limit exceeded node
* 10 Gbps default network limit
Small step on the way to making StreamAllocator prioritization of tracks.
With the new callback into StreamAllocator, the idea is to use the
max layer information to do track prioritization.
Testing:
--------
Sanity check that sample app works
* track numParticipants in room
* only track participant if not a hidden participant
* adjust coding style to use ++
* fix typo
* fix missing nil check
* update roomstore with new numParticipants on participantChanged
* only update roomstore if participant is not hidden
* call StoreRoom directly after StoreParticipant when joining/leaving
* Handle multiple codecs in renegotiation
update pion to v3.1.9 for answer same order of codec as publisher.
register enable codecs in subscriber peerconnectin created.
add codec parameter to buffer.bind
buffer should use the codec of TrackRemote as it's codec mime.
sent h264blankframe when DownTrack closing
When we RLock during write cycles, the mutex spends the majority of its time
staying locked. As new participants join, they have to acquire the WLock
before downtracks could be add it.
In load test scenarios (25 participants joining together), it's common to see
goroutine dump showing MediaTrack.AddSubscriber -> DownTrack.storeDownTrack trying to acquire mutex, and never able to acquire it.
* Try sending small key frames to clear decoded buffer
Problem:
--------
With transceiver re-use, client disables/enables tracks.
With video, this retains the last picture and when a new track
starts, there is a brief moment when the old stream is displayed
till there is data from new stream to decode and display.
Fix:
---
Send small key frames before closing DownTrack to try and clear
the decoder display buffer.
Testing:
--------
Tried with Chrome/Safari/Firefox and they worked. But, very rarely,
the last frames do not seem to show up. In fact, 6 frames are sent
and webrtc internals (in Firefox) reports anywhere from 1 - 6 frames
at the small resolution. Unclear as to why it does not get all the
frames or why it reports less than 6. A not so small percentage of
times (maybe 1 in 15 - 20), have seen no small frame reported at all.
TODO:
----
- Have to support more video codecs
- Would this be an issue for audio also? Should we send something to handle that?
Probably not necessary as video is more jarring.
* Make VP8 Key Frame a const
* Need a packet factory buffer for simple tracks too
as we are using the VP8 munger for simple tracks too because
of the need to send blank frames at the end.
Also, making the writeBlankFrameRTP a private function.
And adding a check to not send blank frames if nothing has been sent
on that DownTrack.
* Separate from ion-sfu
changes:
1. extract pkg/buffer, twcc, sfu, relay, stats, logger
2. to solve cycle import, move ion-sfu/pkg/logger to pkg/sfu/logger
3. replace pion/ion-sfu => ./
reason: will change import pion/ion-sfu/pkg/* to livekit-server/pkg/*
after this pr merged. Just not change any code in this pr, because it
will confused with the separate code from ion-sfu in review.
* Move code from ion-sfu to pkg/sfu
* fix build error for resovle conflict
Co-authored-by: cnderrauber <zengjie9004@gmail.com>
