mirror of
https://github.com/livekit/livekit.git
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129b1df8e6
* Use VP9 Key frame detection from Galene. With ffmpeg generated VP9 file with single layer and publishing via Go SDK, the key picture determination outlined at https://datatracker.ietf.org/doc/html/draft-ietf-payload-vp9-16#page-13 under the F bit explanation does not work. It declares kay frame for pretty much all frames. Unclear if ffmpeg generated bitstream has issues or if that procedure in the above document does not work for single layer case. Using the bit stream explained here https://storage.googleapis.com/downloads.webmproject.org/docs/vp9/vp9-bitstream-specification-v0.6-20160331-draft.pdf (pages 28, 62, 63) implemented in Galene. That is more expensive as it has to parse more, but works in all cases. * Add AV1-TODo * add some TODOs
797 lines
20 KiB
Go
797 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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}
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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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w.streamTrackerManager = NewStreamTrackerManager(logger, trackInfo, w.isSVC, w.codec.ClockRate, trackersConfig)
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w.streamTrackerManager.SetListener(w)
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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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GetDeltaStats: w.getDeltaStats,
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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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// 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, 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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for _, dt := range w.downTrackSpreader.GetDownTracks() {
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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 {
|
|
case w.rtcpCh <- packets:
|
|
default:
|
|
w.logger.Warnw("sendRTCP failed, rtcp channel full", nil)
|
|
}
|
|
}
|
|
|
|
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 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()
|
|
|
|
var stats []*livekit.RTPStats
|
|
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()
|
|
}
|