Files
livekit/pkg/sfu/interfaces.go
T
Raja SubramanianandGitHub 13e4aaec2b Tests for down stream packet push. (#4692)
* Tests for down stream packet push.

A recent issue (padding bit in RTP header) surfaced a gap which slipped
through due to lack of tests. Changes in pion/rtp were not adopted
properly.

So, adding some tests (thank you Claude for the heavy lifting) to test
the down stream packet path using the whole pion chain.

Split out some interfaces so it is easier to have it all in one place
and create fakes.

Will help adding more tests, for example include the upstream path also
in the integration test. May have to create more interfaces and make
things testable, but this is a start.

* missed file

* rtx specific test
2026-07-20 20:15:07 +05:30

188 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 (
"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()
}
// -------------------------------------------------------------------