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* Participant traffic load. Capturing information about participant traffic - Upstream/Downstream - Audio/Video/Data - Packets/Bytes This captures a notion of how much traffic load a participant is generating. Can be used to make allocation decisions. * Clean up * SIP patches * reporter goroutine * unlock * move traffic stats from protocol * check type
798 lines
20 KiB
Go
798 lines
20 KiB
Go
// Copyright 2023 LiveKit, Inc.
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//
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// Licensed under the Apache License, Version 2.0 (the "License");
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// you may not use this file except in compliance with the License.
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// You may obtain a copy of the License at
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//
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// http://www.apache.org/licenses/LICENSE-2.0
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//
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// Unless required by applicable law or agreed to in writing, software
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// distributed under the License is distributed on an "AS IS" BASIS,
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// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
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// See the License for the specific language governing permissions and
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// limitations under the License.
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package sfu
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import (
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"errors"
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"io"
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"strings"
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"sync"
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"time"
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"github.com/pion/rtcp"
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"github.com/pion/webrtc/v3"
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"go.uber.org/atomic"
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"github.com/livekit/mediatransportutil/pkg/bucket"
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"github.com/livekit/mediatransportutil/pkg/twcc"
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"github.com/livekit/protocol/livekit"
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"github.com/livekit/protocol/logger"
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"github.com/livekit/livekit-server/pkg/config"
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"github.com/livekit/livekit-server/pkg/sfu/audio"
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"github.com/livekit/livekit-server/pkg/sfu/buffer"
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"github.com/livekit/livekit-server/pkg/sfu/connectionquality"
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dd "github.com/livekit/livekit-server/pkg/sfu/dependencydescriptor"
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)
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var (
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ErrReceiverClosed = errors.New("receiver closed")
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ErrDownTrackAlreadyExist = errors.New("DownTrack already exist")
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ErrBufferNotFound = errors.New("buffer not found")
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)
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type AudioLevelHandle func(level uint8, duration uint32)
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type Bitrates [buffer.DefaultMaxLayerSpatial + 1][buffer.DefaultMaxLayerTemporal + 1]int64
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// TrackReceiver defines an interface receive media from remote peer
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type TrackReceiver interface {
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TrackID() livekit.TrackID
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StreamID() string
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Codec() webrtc.RTPCodecParameters
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HeaderExtensions() []webrtc.RTPHeaderExtensionParameter
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IsClosed() bool
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ReadRTP(buf []byte, layer uint8, sn uint16) (int, error)
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GetLayeredBitrate() ([]int32, Bitrates)
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GetAudioLevel() (float64, bool)
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SendPLI(layer int32, force bool)
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SetUpTrackPaused(paused bool)
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SetMaxExpectedSpatialLayer(layer int32)
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AddDownTrack(track TrackSender) error
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DeleteDownTrack(participantID livekit.ParticipantID)
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DebugInfo() map[string]interface{}
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TrackInfo() *livekit.TrackInfo
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// Get primary receiver if this receiver represents a RED codec; otherwise it will return itself
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GetPrimaryReceiverForRed() TrackReceiver
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// Get red receiver for primary codec, used by forward red encodings for opus only codec
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GetRedReceiver() TrackReceiver
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GetTemporalLayerFpsForSpatial(layer int32) []float32
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GetCalculatedClockRate(layer int32) uint32
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GetReferenceLayerRTPTimestamp(ts uint32, layer int32, referenceLayer int32) (uint32, error)
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GetTrackStats() *livekit.RTPStats
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}
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// WebRTCReceiver receives a media track
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type WebRTCReceiver struct {
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logger logger.Logger
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pliThrottleConfig config.PLIThrottleConfig
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audioConfig config.AudioConfig
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trackID livekit.TrackID
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streamID string
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kind webrtc.RTPCodecType
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receiver *webrtc.RTPReceiver
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codec webrtc.RTPCodecParameters
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isSVC bool
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isRED bool
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onCloseHandler func()
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closeOnce sync.Once
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closed atomic.Bool
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useTrackers bool
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trackInfo *livekit.TrackInfo
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rtcpCh chan []rtcp.Packet
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twcc *twcc.Responder
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bufferMu sync.RWMutex
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buffers [buffer.DefaultMaxLayerSpatial + 1]*buffer.Buffer
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rtt uint32
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upTrackMu sync.RWMutex
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upTracks [buffer.DefaultMaxLayerSpatial + 1]*webrtc.TrackRemote
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lbThreshold int
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streamTrackerManager *StreamTrackerManager
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downTrackSpreader *DownTrackSpreader
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connectionStats *connectionquality.ConnectionStats
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onStatsUpdate func(w *WebRTCReceiver, stat *livekit.AnalyticsStat)
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onMaxLayerChange func(maxLayer int32)
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primaryReceiver atomic.Pointer[RedPrimaryReceiver]
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redReceiver atomic.Pointer[RedReceiver]
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redPktWriter func(pkt *buffer.ExtPacket, spatialLayer int32)
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}
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// SVC-TODO: Have to use more conditions to differentiate between
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// SVC-TODO: SVC and non-SVC (could be single layer or simulcast).
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// SVC-TODO: May only need to differentiate between simulcast and non-simulcast
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// SVC-TODO: i. e. may be possible to treat single layer as SVC to get proper/intended functionality.
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func IsSvcCodec(mime string) bool {
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switch strings.ToLower(mime) {
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case "video/av1":
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fallthrough
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case "video/vp9":
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return true
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}
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return false
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}
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func IsRedCodec(mime string) bool {
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return strings.HasSuffix(strings.ToLower(mime), "red")
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}
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type ReceiverOpts func(w *WebRTCReceiver) *WebRTCReceiver
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// WithPliThrottleConfig indicates minimum time(ms) between sending PLIs
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func WithPliThrottleConfig(pliThrottleConfig config.PLIThrottleConfig) ReceiverOpts {
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return func(w *WebRTCReceiver) *WebRTCReceiver {
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w.pliThrottleConfig = pliThrottleConfig
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return w
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}
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}
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// WithAudioConfig sets up parameters for active speaker detection
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func WithAudioConfig(audioConfig config.AudioConfig) ReceiverOpts {
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return func(w *WebRTCReceiver) *WebRTCReceiver {
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w.audioConfig = audioConfig
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return w
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}
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}
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// WithStreamTrackers enables StreamTracker use for simulcast
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func WithStreamTrackers() ReceiverOpts {
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return func(w *WebRTCReceiver) *WebRTCReceiver {
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w.useTrackers = true
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return w
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}
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}
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// WithLoadBalanceThreshold enables parallelization of packet writes when downTracks exceeds threshold
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// Value should be between 3 and 150.
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// For a server handling a few large rooms, use a smaller value (required to handle very large (250+ participant) rooms).
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// For a server handling many small rooms, use a larger value or disable.
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// Set to 0 (disabled) by default.
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func WithLoadBalanceThreshold(downTracks int) ReceiverOpts {
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return func(w *WebRTCReceiver) *WebRTCReceiver {
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w.lbThreshold = downTracks
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return w
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}
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}
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// NewWebRTCReceiver creates a new webrtc track receiver
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func NewWebRTCReceiver(
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receiver *webrtc.RTPReceiver,
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track *webrtc.TrackRemote,
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trackInfo *livekit.TrackInfo,
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logger logger.Logger,
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twcc *twcc.Responder,
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trackersConfig config.StreamTrackersConfig,
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opts ...ReceiverOpts,
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) *WebRTCReceiver {
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w := &WebRTCReceiver{
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logger: logger,
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receiver: receiver,
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trackID: livekit.TrackID(track.ID()),
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streamID: track.StreamID(),
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codec: track.Codec(),
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kind: track.Kind(),
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twcc: twcc,
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trackInfo: trackInfo,
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isSVC: IsSvcCodec(track.Codec().MimeType),
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isRED: IsRedCodec(track.Codec().MimeType),
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}
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for _, opt := range opts {
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w = opt(w)
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}
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w.downTrackSpreader = NewDownTrackSpreader(DownTrackSpreaderParams{
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Threshold: w.lbThreshold,
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Logger: logger,
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})
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w.connectionStats = connectionquality.NewConnectionStats(connectionquality.ConnectionStatsParams{
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MimeType: w.codec.MimeType,
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IsFECEnabled: strings.EqualFold(w.codec.MimeType, webrtc.MimeTypeOpus) && strings.Contains(strings.ToLower(w.codec.SDPFmtpLine), "fec"),
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ReceiverProvider: w,
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Logger: w.logger.WithValues("direction", "up"),
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})
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w.connectionStats.OnStatsUpdate(func(_cs *connectionquality.ConnectionStats, stat *livekit.AnalyticsStat) {
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if w.onStatsUpdate != nil {
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w.onStatsUpdate(w, stat)
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}
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})
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w.connectionStats.Start(w.trackInfo)
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w.streamTrackerManager = NewStreamTrackerManager(logger, trackInfo, w.isSVC, w.codec.ClockRate, trackersConfig)
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w.streamTrackerManager.SetListener(w)
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// SVC-TODO: Handle DD for non-SVC cases???
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if w.isSVC {
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for _, ext := range receiver.GetParameters().HeaderExtensions {
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if ext.URI == dd.ExtensionURI {
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w.streamTrackerManager.AddDependencyDescriptorTrackers()
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break
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}
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}
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}
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return w
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}
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func (w *WebRTCReceiver) TrackInfo() *livekit.TrackInfo {
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return w.trackInfo
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}
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func (w *WebRTCReceiver) OnStatsUpdate(fn func(w *WebRTCReceiver, stat *livekit.AnalyticsStat)) {
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w.onStatsUpdate = fn
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}
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func (w *WebRTCReceiver) OnMaxLayerChange(fn func(maxLayer int32)) {
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w.upTrackMu.Lock()
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w.onMaxLayerChange = fn
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w.upTrackMu.Unlock()
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}
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func (w *WebRTCReceiver) GetConnectionScoreAndQuality() (float32, livekit.ConnectionQuality) {
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return w.connectionStats.GetScoreAndQuality()
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}
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func (w *WebRTCReceiver) IsClosed() bool {
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return w.closed.Load()
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}
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func (w *WebRTCReceiver) SetRTT(rtt uint32) {
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w.bufferMu.Lock()
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if w.rtt == rtt {
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w.bufferMu.Unlock()
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return
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}
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w.rtt = rtt
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buffers := w.buffers
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w.bufferMu.Unlock()
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for _, buff := range buffers {
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if buff == nil {
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continue
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}
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buff.SetRTT(rtt)
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}
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}
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func (w *WebRTCReceiver) StreamID() string {
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return w.streamID
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}
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func (w *WebRTCReceiver) TrackID() livekit.TrackID {
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return w.trackID
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}
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func (w *WebRTCReceiver) SSRC(layer int) uint32 {
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w.upTrackMu.RLock()
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defer w.upTrackMu.RUnlock()
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if track := w.upTracks[layer]; track != nil {
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return uint32(track.SSRC())
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}
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return 0
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}
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func (w *WebRTCReceiver) Codec() webrtc.RTPCodecParameters {
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return w.codec
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}
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func (w *WebRTCReceiver) HeaderExtensions() []webrtc.RTPHeaderExtensionParameter {
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return w.receiver.GetParameters().HeaderExtensions
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}
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func (w *WebRTCReceiver) Kind() webrtc.RTPCodecType {
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return w.kind
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}
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func (w *WebRTCReceiver) AddUpTrack(track *webrtc.TrackRemote, buff *buffer.Buffer) {
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if w.closed.Load() {
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return
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}
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layer := int32(0)
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if w.Kind() == webrtc.RTPCodecTypeVideo && !w.isSVC {
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layer = buffer.RidToSpatialLayer(track.RID(), w.trackInfo)
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}
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buff.SetLogger(w.logger.WithValues("layer", layer))
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buff.SetTWCC(w.twcc)
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buff.SetAudioLevelParams(audio.AudioLevelParams{
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ActiveLevel: w.audioConfig.ActiveLevel,
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MinPercentile: w.audioConfig.MinPercentile,
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ObserveDuration: w.audioConfig.UpdateInterval,
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SmoothIntervals: w.audioConfig.SmoothIntervals,
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})
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buff.OnRtcpFeedback(w.sendRTCP)
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buff.OnRtcpSenderReport(func() {
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srFirst, srNewest := buff.GetSenderReportData()
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w.streamTrackerManager.SetRTCPSenderReportData(layer, srFirst, srNewest)
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w.downTrackSpreader.Broadcast(func(dt TrackSender) {
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_ = dt.HandleRTCPSenderReportData(w.codec.PayloadType, w.isSVC, layer, srNewest)
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})
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})
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var duration time.Duration
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switch layer {
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case 2:
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duration = w.pliThrottleConfig.HighQuality
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case 1:
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duration = w.pliThrottleConfig.MidQuality
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case 0:
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duration = w.pliThrottleConfig.LowQuality
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default:
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duration = w.pliThrottleConfig.MidQuality
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}
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if duration != 0 {
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buff.SetPLIThrottle(duration.Nanoseconds())
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}
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w.upTrackMu.Lock()
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w.upTracks[layer] = track
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w.upTrackMu.Unlock()
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w.bufferMu.Lock()
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w.buffers[layer] = buff
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rtt := w.rtt
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w.bufferMu.Unlock()
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buff.SetRTT(rtt)
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buff.SetPaused(w.streamTrackerManager.IsPaused())
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if w.Kind() == webrtc.RTPCodecTypeVideo && w.useTrackers {
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w.streamTrackerManager.AddTracker(layer)
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}
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go w.forwardRTP(layer)
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}
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// SetUpTrackPaused indicates upstream will not be sending any data.
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// this will reflect the "muted" status and will pause streamtracker to ensure we don't turn off
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// the layer
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func (w *WebRTCReceiver) SetUpTrackPaused(paused bool) {
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w.streamTrackerManager.SetPaused(paused)
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w.bufferMu.RLock()
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for _, buff := range w.buffers {
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if buff == nil {
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continue
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}
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buff.SetPaused(paused)
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}
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w.bufferMu.RUnlock()
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w.connectionStats.UpdateMute(paused)
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}
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func (w *WebRTCReceiver) AddDownTrack(track TrackSender) error {
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if w.closed.Load() {
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return ErrReceiverClosed
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}
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if w.downTrackSpreader.HasDownTrack(track.SubscriberID()) {
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w.logger.Infow("subscriberID already exists, replacing downtrack", "subscriberID", track.SubscriberID())
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}
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track.TrackInfoAvailable()
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track.UpTrackMaxPublishedLayerChange(w.streamTrackerManager.GetMaxPublishedLayer())
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track.UpTrackMaxTemporalLayerSeenChange(w.streamTrackerManager.GetMaxTemporalLayerSeen())
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w.downTrackSpreader.Store(track)
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return nil
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}
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func (w *WebRTCReceiver) SetMaxExpectedSpatialLayer(layer int32) {
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w.streamTrackerManager.SetMaxExpectedSpatialLayer(layer)
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if layer == buffer.InvalidLayerSpatial {
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w.connectionStats.UpdateLayerMute(true)
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} else {
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w.connectionStats.UpdateLayerMute(false)
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w.connectionStats.AddLayerTransition(w.streamTrackerManager.DistanceToDesired())
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}
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}
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// StreamTrackerManagerListener.OnAvailableLayersChanged
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func (w *WebRTCReceiver) OnAvailableLayersChanged() {
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w.downTrackSpreader.Broadcast(func(dt TrackSender) {
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dt.UpTrackLayersChange()
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})
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w.connectionStats.AddLayerTransition(w.streamTrackerManager.DistanceToDesired())
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}
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// StreamTrackerManagerListener.OnBitrateAvailabilityChanged
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func (w *WebRTCReceiver) OnBitrateAvailabilityChanged() {
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w.downTrackSpreader.Broadcast(func(dt TrackSender) {
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dt.UpTrackBitrateAvailabilityChange()
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})
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}
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// StreamTrackerManagerListener.OnMaxPublishedLayerChanged
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func (w *WebRTCReceiver) OnMaxPublishedLayerChanged(maxPublishedLayer int32) {
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w.downTrackSpreader.Broadcast(func(dt TrackSender) {
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dt.UpTrackMaxPublishedLayerChange(maxPublishedLayer)
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})
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w.connectionStats.AddLayerTransition(w.streamTrackerManager.DistanceToDesired())
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}
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// StreamTrackerManagerListener.OnMaxTemporalLayerSeenChanged
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func (w *WebRTCReceiver) OnMaxTemporalLayerSeenChanged(maxTemporalLayerSeen int32) {
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w.downTrackSpreader.Broadcast(func(dt TrackSender) {
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dt.UpTrackMaxTemporalLayerSeenChange(maxTemporalLayerSeen)
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})
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w.connectionStats.AddLayerTransition(w.streamTrackerManager.DistanceToDesired())
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}
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// StreamTrackerManagerListener.OnMaxAvailableLayerChanged
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func (w *WebRTCReceiver) OnMaxAvailableLayerChanged(maxAvailableLayer int32) {
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w.upTrackMu.RLock()
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onMaxLayerChange := w.onMaxLayerChange
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w.upTrackMu.RUnlock()
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if onMaxLayerChange != nil {
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onMaxLayerChange(maxAvailableLayer)
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}
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}
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// StreamTrackerManagerListener.OnBitrateReport
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func (w *WebRTCReceiver) OnBitrateReport(availableLayers []int32, bitrates Bitrates) {
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w.downTrackSpreader.Broadcast(func(dt TrackSender) {
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dt.UpTrackBitrateReport(availableLayers, bitrates)
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})
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w.connectionStats.AddLayerTransition(w.streamTrackerManager.DistanceToDesired())
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}
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func (w *WebRTCReceiver) GetLayeredBitrate() ([]int32, Bitrates) {
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return w.streamTrackerManager.GetLayeredBitrate()
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}
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// OnCloseHandler method to be called on remote tracked removed
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func (w *WebRTCReceiver) OnCloseHandler(fn func()) {
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w.onCloseHandler = fn
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}
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// DeleteDownTrack removes a DownTrack from a Receiver
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func (w *WebRTCReceiver) DeleteDownTrack(subscriberID livekit.ParticipantID) {
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if w.closed.Load() {
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return
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}
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w.downTrackSpreader.Free(subscriberID)
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}
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func (w *WebRTCReceiver) sendRTCP(packets []rtcp.Packet) {
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if packets == nil || w.closed.Load() {
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return
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}
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select {
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case w.rtcpCh <- packets:
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default:
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w.logger.Warnw("sendRTCP failed, rtcp channel full", nil)
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}
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}
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func (w *WebRTCReceiver) SendPLI(layer int32, force bool) {
|
|
// SVC-TODO : should send LRR (Layer Refresh Request) instead of PLI
|
|
buff := w.getBuffer(layer)
|
|
if buff == nil {
|
|
return
|
|
}
|
|
|
|
buff.SendPLI(force)
|
|
}
|
|
|
|
func (w *WebRTCReceiver) SetRTCPCh(ch chan []rtcp.Packet) {
|
|
w.rtcpCh = ch
|
|
}
|
|
|
|
func (w *WebRTCReceiver) getBuffer(layer int32) *buffer.Buffer {
|
|
w.bufferMu.RLock()
|
|
defer w.bufferMu.RUnlock()
|
|
|
|
return w.getBufferLocked(layer)
|
|
}
|
|
|
|
func (w *WebRTCReceiver) getBufferLocked(layer int32) *buffer.Buffer {
|
|
// for svc codecs, use layer = 0 always.
|
|
// spatial layers are in-built and handled by single buffer
|
|
if w.isSVC {
|
|
layer = 0
|
|
}
|
|
|
|
if layer < 0 || int(layer) >= len(w.buffers) {
|
|
return nil
|
|
}
|
|
|
|
return w.buffers[layer]
|
|
}
|
|
|
|
func (w *WebRTCReceiver) ReadRTP(buf []byte, layer uint8, sn uint16) (int, error) {
|
|
b := w.getBuffer(int32(layer))
|
|
if b == nil {
|
|
return 0, ErrBufferNotFound
|
|
}
|
|
|
|
return b.GetPacket(buf, sn)
|
|
}
|
|
|
|
func (w *WebRTCReceiver) GetTrackStats() *livekit.RTPStats {
|
|
w.bufferMu.RLock()
|
|
defer w.bufferMu.RUnlock()
|
|
|
|
stats := make([]*livekit.RTPStats, 0, len(w.buffers))
|
|
for _, buff := range w.buffers {
|
|
if buff == nil {
|
|
continue
|
|
}
|
|
|
|
sswl := buff.GetStats()
|
|
if sswl == nil {
|
|
continue
|
|
}
|
|
|
|
stats = append(stats, sswl)
|
|
}
|
|
|
|
return buffer.AggregateRTPStats(stats)
|
|
}
|
|
|
|
func (w *WebRTCReceiver) GetAudioLevel() (float64, bool) {
|
|
if w.Kind() == webrtc.RTPCodecTypeVideo {
|
|
return 0, false
|
|
}
|
|
|
|
w.bufferMu.RLock()
|
|
defer w.bufferMu.RUnlock()
|
|
|
|
for _, buff := range w.buffers {
|
|
if buff == nil {
|
|
continue
|
|
}
|
|
|
|
return buff.GetAudioLevel()
|
|
}
|
|
|
|
return 0, false
|
|
}
|
|
|
|
func (w *WebRTCReceiver) GetDeltaStats() map[uint32]*buffer.StreamStatsWithLayers {
|
|
w.bufferMu.RLock()
|
|
defer w.bufferMu.RUnlock()
|
|
|
|
deltaStats := make(map[uint32]*buffer.StreamStatsWithLayers, len(w.buffers))
|
|
|
|
for layer, buff := range w.buffers {
|
|
if buff == nil {
|
|
continue
|
|
}
|
|
|
|
sswl := buff.GetDeltaStats()
|
|
if sswl == nil {
|
|
continue
|
|
}
|
|
|
|
// patch buffer stats with correct layer
|
|
patched := make(map[int32]*buffer.RTPDeltaInfo, 1)
|
|
patched[int32(layer)] = sswl.Layers[0]
|
|
sswl.Layers = patched
|
|
|
|
deltaStats[w.SSRC(layer)] = sswl
|
|
}
|
|
|
|
return deltaStats
|
|
}
|
|
|
|
func (w *WebRTCReceiver) forwardRTP(layer int32) {
|
|
pktBuf := make([]byte, bucket.MaxPktSize)
|
|
tracker := w.streamTrackerManager.GetTracker(layer)
|
|
|
|
defer func() {
|
|
w.closeOnce.Do(func() {
|
|
w.closed.Store(true)
|
|
w.closeTracks()
|
|
if pr := w.primaryReceiver.Load(); pr != nil {
|
|
pr.Close()
|
|
}
|
|
if pr := w.redReceiver.Load(); pr != nil {
|
|
pr.Close()
|
|
}
|
|
})
|
|
|
|
w.streamTrackerManager.RemoveTracker(layer)
|
|
if w.isSVC {
|
|
w.streamTrackerManager.RemoveAllTrackers()
|
|
}
|
|
}()
|
|
|
|
for {
|
|
w.bufferMu.RLock()
|
|
buf := w.buffers[layer]
|
|
redPktWriter := w.redPktWriter
|
|
w.bufferMu.RUnlock()
|
|
pkt, err := buf.ReadExtended(pktBuf)
|
|
if err == io.EOF {
|
|
return
|
|
}
|
|
|
|
spatialTracker := tracker
|
|
spatialLayer := layer
|
|
if pkt.Spatial >= 0 {
|
|
// svc packet, dispatch to correct tracker
|
|
spatialLayer = pkt.Spatial
|
|
spatialTracker = w.streamTrackerManager.GetTracker(pkt.Spatial)
|
|
if spatialTracker == nil {
|
|
spatialTracker = w.streamTrackerManager.AddTracker(pkt.Spatial)
|
|
}
|
|
}
|
|
|
|
w.downTrackSpreader.Broadcast(func(dt TrackSender) {
|
|
_ = dt.WriteRTP(pkt, spatialLayer)
|
|
})
|
|
|
|
if redPktWriter != nil {
|
|
redPktWriter(pkt, spatialLayer)
|
|
}
|
|
|
|
if spatialTracker != nil {
|
|
spatialTracker.Observe(
|
|
pkt.Temporal,
|
|
len(pkt.RawPacket),
|
|
len(pkt.Packet.Payload),
|
|
pkt.Packet.Marker,
|
|
pkt.Packet.Timestamp,
|
|
pkt.DependencyDescriptor,
|
|
)
|
|
}
|
|
}
|
|
}
|
|
|
|
// closeTracks close all tracks from Receiver
|
|
func (w *WebRTCReceiver) closeTracks() {
|
|
w.connectionStats.Close()
|
|
w.streamTrackerManager.Close()
|
|
|
|
closeTrackSenders(w.downTrackSpreader.ResetAndGetDownTracks())
|
|
|
|
if w.onCloseHandler != nil {
|
|
w.onCloseHandler()
|
|
}
|
|
}
|
|
|
|
func (w *WebRTCReceiver) DebugInfo() map[string]interface{} {
|
|
info := map[string]interface{}{
|
|
"SVC": w.isSVC,
|
|
"Simulcast": !w.isSVC && len(w.trackInfo.Layers) > 1,
|
|
}
|
|
|
|
w.upTrackMu.RLock()
|
|
upTrackInfo := make([]map[string]interface{}, 0, len(w.upTracks))
|
|
for layer, ut := range w.upTracks {
|
|
if ut != nil {
|
|
upTrackInfo = append(upTrackInfo, map[string]interface{}{
|
|
"Layer": layer,
|
|
"SSRC": ut.SSRC(),
|
|
"Msid": ut.Msid(),
|
|
"RID": ut.RID(),
|
|
})
|
|
}
|
|
}
|
|
w.upTrackMu.RUnlock()
|
|
info["UpTracks"] = upTrackInfo
|
|
|
|
return info
|
|
}
|
|
|
|
func (w *WebRTCReceiver) GetPrimaryReceiverForRed() TrackReceiver {
|
|
if !w.isRED || w.closed.Load() {
|
|
return w
|
|
}
|
|
|
|
if w.primaryReceiver.Load() == nil {
|
|
pr := NewRedPrimaryReceiver(w, DownTrackSpreaderParams{
|
|
Threshold: w.lbThreshold,
|
|
Logger: w.logger,
|
|
})
|
|
if w.primaryReceiver.CompareAndSwap(nil, pr) {
|
|
w.bufferMu.Lock()
|
|
w.redPktWriter = pr.ForwardRTP
|
|
w.bufferMu.Unlock()
|
|
}
|
|
}
|
|
return w.primaryReceiver.Load()
|
|
}
|
|
|
|
func (w *WebRTCReceiver) GetRedReceiver() TrackReceiver {
|
|
if w.isRED || w.closed.Load() {
|
|
return w
|
|
}
|
|
|
|
if w.redReceiver.Load() == nil {
|
|
pr := NewRedReceiver(w, DownTrackSpreaderParams{
|
|
Threshold: w.lbThreshold,
|
|
Logger: w.logger,
|
|
})
|
|
if w.redReceiver.CompareAndSwap(nil, pr) {
|
|
w.bufferMu.Lock()
|
|
w.redPktWriter = pr.ForwardRTP
|
|
w.bufferMu.Unlock()
|
|
}
|
|
}
|
|
return w.redReceiver.Load()
|
|
}
|
|
|
|
func (w *WebRTCReceiver) GetTemporalLayerFpsForSpatial(layer int32) []float32 {
|
|
b := w.getBuffer(layer)
|
|
if b == nil {
|
|
return nil
|
|
}
|
|
|
|
if !w.isSVC {
|
|
return b.GetTemporalLayerFpsForSpatial(0)
|
|
}
|
|
|
|
return b.GetTemporalLayerFpsForSpatial(layer)
|
|
}
|
|
|
|
func (w *WebRTCReceiver) GetCalculatedClockRate(layer int32) uint32 {
|
|
return w.streamTrackerManager.GetCalculatedClockRate(layer)
|
|
}
|
|
|
|
func (w *WebRTCReceiver) GetReferenceLayerRTPTimestamp(ts uint32, layer int32, referenceLayer int32) (uint32, error) {
|
|
return w.streamTrackerManager.GetReferenceLayerRTPTimestamp(ts, layer, referenceLayer)
|
|
}
|
|
|
|
// closes all track senders in parallel, returns when all are closed
|
|
func closeTrackSenders(senders []TrackSender) {
|
|
wg := sync.WaitGroup{}
|
|
for _, dt := range senders {
|
|
dt := dt
|
|
wg.Add(1)
|
|
go func() {
|
|
defer wg.Done()
|
|
dt.Close()
|
|
}()
|
|
}
|
|
wg.Wait()
|
|
}
|