Files
livekit/pkg/sfu/interfaces.go
T
Raja SubramanianandGitHub 0aa296e126 Record subscribe stream start time in prometheus. (#4704)
* Record subscribe stream start time in prometheus.

Adjust for mutes, i. e. take the last unmute time as the start point and
calculate time till the first byte is sent.

* close the tiny window of race

* Prevent long tail sample when publisher glitches.

Thanks to @milos-lk for this.

Publisher restarting would have reset the layer and would have caused a
sample with very high stream start time. We only need to capture when we
do a dummy start or when the state is seeded to a different node upon
migration.

* reduce a diff

* test

* changed the wrong thing, thank you Devin
2026-07-27 20:37:44 +05:30

190 lines
5.7 KiB
Go

// Copyright 2023 LiveKit, Inc.
//
// Licensed under the Apache License, Version 2.0 (the "License");
// you may not use this file except in compliance with the License.
// You may obtain a copy of the License at
//
// http://www.apache.org/licenses/LICENSE-2.0
//
// Unless required by applicable law or agreed to in writing, software
// distributed under the License is distributed on an "AS IS" BASIS,
// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
// See the License for the specific language governing permissions and
// limitations under the License.
package sfu
import (
"time"
"github.com/pion/rtcp"
"github.com/pion/webrtc/v4"
"github.com/livekit/mediatransportutil/pkg/codec"
"github.com/livekit/protocol/codecs/mime"
"github.com/livekit/protocol/livekit"
"github.com/livekit/livekit-server/pkg/sfu/buffer"
"github.com/livekit/livekit-server/pkg/sfu/bwe"
)
//go:generate go run github.com/maxbrunsfeld/counterfeiter/v6 -generate
// TrackReceiver defines an interface receive media from remote peer
//
//counterfeiter:generate . TrackReceiver
type TrackReceiver interface {
TrackID() livekit.TrackID
StreamID() string
// returns the initial codec of the receiver, it is determined by the track's codec
// and will not change if the codec changes during the session (publisher changes codec)
Codec() webrtc.RTPCodecParameters
Mime() mime.MimeType
VideoLayerMode() livekit.VideoLayer_Mode
HeaderExtensions() []webrtc.RTPHeaderExtensionParameter
IsClosed() bool
ReadRTP(buf []byte, layer uint8, esn uint64) (int, error)
GetLayeredBitrate() ([]int32, Bitrates)
GetAudioLevel() (float64, bool)
SendPLI(layer int32, force bool)
SetMaxExpectedSpatialLayer(layer int32)
AddDownTrack(track TrackSender) error
DeleteDownTrack(participantID livekit.ParticipantID)
GetDownTracks() []TrackSender
DebugInfo() map[string]any
TrackInfo() *livekit.TrackInfo
UpdateTrackInfo(ti *livekit.TrackInfo)
// Get primary receiver if this receiver represents a RED codec; otherwise it will return itself
GetPrimaryReceiverForRed() TrackReceiver
// Get red receiver for primary codec, used by forward red encodings for opus only codec
GetRedReceiver() TrackReceiver
GetTemporalLayerFpsForSpatial(layer int32) []float32
GetTrackStats() *livekit.RTPStats
// AddOnReady adds a function to be called when the receiver is ready, the callback
// could be called immediately if the receiver is ready when the callback is added
AddOnReady(func())
AddOnCodecStateChange(func(webrtc.RTPCodecParameters, ReceiverCodecState))
CodecState() ReceiverCodecState
// VideoSizes returns the video size parsed from rtp packet for each spatial layer.
VideoSizes() []codec.VideoSize
// closes all associated buffers and issues a resync to all attached downtracks so that
// they can resync and have proper sequncing without gaps in sequence numbers / timestamps
Restart(reason string)
}
// --------------------------------------
//counterfeiter:generate . REDTransformer
type REDTransformer interface {
TrackReceiver
SetLBThreshold(lbThreshold int)
ForwardRTP(pkt *buffer.ExtPacket, spatialLayer int32) int32
ForwardRTCPSenderReport(
payloadType webrtc.PayloadType,
layer int32,
publisherSRData *livekit.RTCPSenderReportState,
)
GetDownTracks() []TrackSender
ResyncDownTracks()
OnStreamRestart()
CanClose() bool
Close()
}
// --------------------------------------
// TrackSender defines an interface send media to remote peer
//
//counterfeiter:generate . TrackSender
type TrackSender interface {
UpTrackLayersChange()
UpTrackBitrateAvailabilityChange()
UpTrackMaxPublishedLayerChange(maxPublishedLayer int32)
UpTrackMaxTemporalLayerSeenChange(maxTemporalLayerSeen int32)
UpTrackBitrateReport(availableLayers []int32, bitrates Bitrates)
WriteRTP(p *buffer.ExtPacket, layer int32) int32
Close()
IsClosed() bool
// ID is the globally unique identifier for this Track.
ID() string
SubscriberID() livekit.ParticipantID
HandleRTCPSenderReportData(
payloadType webrtc.PayloadType,
layer int32,
publisherSRData *livekit.RTCPSenderReportState,
) error
Resync()
SetReceiver(TrackReceiver)
ReceiverRestart(TrackReceiver)
}
// -------------------------------------------------------------------
//counterfeiter:generate . DownTrackStreamAllocatorListener
type DownTrackStreamAllocatorListener interface {
// RTCP received
OnREMB(dt *DownTrack, remb *rtcp.ReceiverEstimatedMaximumBitrate)
OnTransportCCFeedback(dt *DownTrack, cc *rtcp.TransportLayerCC)
// video layer availability changed
OnAvailableLayersChanged(dt *DownTrack)
// video layer bitrate availability changed
OnBitrateAvailabilityChanged(dt *DownTrack)
// max published spatial layer changed
OnMaxPublishedSpatialChanged(dt *DownTrack)
// max published temporal layer changed
OnMaxPublishedTemporalChanged(dt *DownTrack)
// subscription changed - mute/unmute
OnSubscriptionChanged(dt *DownTrack)
// subscribed max video layer changed
OnSubscribedLayerChanged(dt *DownTrack, layers buffer.VideoLayer)
// stream resumed
OnResume(dt *DownTrack)
// check if track should participate in BWE
IsBWEEnabled(dt *DownTrack) bool
// get the BWE type in use
BWEType() bwe.BWEType
// check if subscription mute can be applied
IsSubscribeMutable(dt *DownTrack) bool
}
// -------------------------------------------------------------------
//counterfeiter:generate . DownTrackListener
type DownTrackListener interface {
OnBindAndConnected()
OnStatsUpdate(stat *livekit.AnalyticsStat)
OnMaxSubscribedLayerChanged(layer int32)
OnRttUpdate(rtt uint32)
OnCodecNegotiated(webrtc.RTPCodecCapability)
OnDownTrackClose(isExpectedToResume bool)
OnStreamStarted(elapsed time.Duration)
}
// -------------------------------------------------------------------