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relay_whip_track_handler.go
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relay_whip_track_handler.go
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// 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 whip
import (
"bytes"
"fmt"
"io"
"net"
"sync"
"time"
"github.com/frostbyte73/core"
"github.com/livekit/ingress/pkg/errors"
"github.com/livekit/ingress/pkg/stats"
"github.com/livekit/protocol/livekit"
"github.com/livekit/protocol/logger"
"github.com/livekit/server-sdk-go/v2/pkg/jitter"
"github.com/livekit/server-sdk-go/v2/pkg/synchronizer"
"github.com/pion/rtcp"
"github.com/pion/rtp"
"github.com/pion/webrtc/v3"
"github.com/pion/webrtc/v3/pkg/media"
)
const (
maxVideoLatency = 600 * time.Millisecond
maxAudioLatency = time.Second
)
type RelayWhipTrackHandler struct {
logger logger.Logger
remoteTrack *webrtc.TrackRemote
depacketizer rtp.Depacketizer
quality livekit.VideoQuality
receiver *webrtc.RTPReceiver
sync *synchronizer.TrackSynchronizer
writePLI func(ssrc webrtc.SSRC)
onRTCP func(packet rtcp.Packet)
jb *jitter.Buffer
relaySink *RelayMediaSink
firstPacket sync.Once
fuse core.Fuse
lastSn uint16
lastSnValid bool
statsLock sync.Mutex
trackStats *stats.MediaTrackStatGatherer
}
func NewRelayWhipTrackHandler(
logger logger.Logger,
track *webrtc.TrackRemote,
quality livekit.VideoQuality,
sync *synchronizer.TrackSynchronizer,
receiver *webrtc.RTPReceiver,
writePLI func(ssrc webrtc.SSRC),
onRTCP func(packet rtcp.Packet),
) (*RelayWhipTrackHandler, error) {
jb, err := createJitterBuffer(track, logger, writePLI)
if err != nil {
return nil, err
}
depacketizer, err := createDepacketizer(track)
if err != nil {
return nil, err
}
relaySink := NewRelayMediaSink(logger)
return &RelayWhipTrackHandler{
logger: logger,
remoteTrack: track,
quality: quality,
receiver: receiver,
relaySink: relaySink,
sync: sync,
jb: jb,
onRTCP: onRTCP,
depacketizer: depacketizer,
}, nil
}
func (t *RelayWhipTrackHandler) Start(onDone func(err error)) (err error) {
t.startRTPReceiver(onDone)
if t.onRTCP != nil {
t.startRTCPReceiver()
}
return nil
}
func (t *RelayWhipTrackHandler) SetMediaTrackStatsGatherer(st *stats.LocalMediaStatsGatherer) {
t.statsLock.Lock()
var path string
switch t.remoteTrack.Kind() {
case webrtc.RTPCodecTypeAudio:
path = stats.InputAudio
case webrtc.RTPCodecTypeVideo:
path = fmt.Sprintf("%s.%s", stats.InputVideo, t.quality)
default:
path = "input.unknown"
}
g := st.RegisterTrackStats(path)
t.trackStats = g
t.statsLock.Unlock()
}
func (t *RelayWhipTrackHandler) SetWriter(w io.WriteCloser) error {
return t.relaySink.SetWriter(w)
}
func (t *RelayWhipTrackHandler) Close() {
t.fuse.Break()
}
func (t *RelayWhipTrackHandler) startRTPReceiver(onDone func(err error)) {
go func() {
var err error
defer func() {
t.relaySink.Close()
if onDone != nil {
onDone(err)
}
}()
t.logger.Infow("starting rtp receiver")
if t.remoteTrack.Kind() == webrtc.RTPCodecTypeVideo && t.writePLI != nil {
t.writePLI(t.remoteTrack.SSRC())
}
for {
select {
case <-t.fuse.Watch():
t.logger.Debugw("stopping rtp receiver")
return
default:
err = t.processRTPPacket()
switch err {
case nil, errors.ErrPrerollBufferReset:
// continue
case io.EOF:
err = nil // success
return
default:
if netErr, ok := err.(net.Error); ok && netErr.Timeout() {
continue
}
t.logger.Warnw("error forwarding rtp packets", err)
return
}
}
}
}()
}
func (t *RelayWhipTrackHandler) startRTCPReceiver() {
go func() {
t.logger.Infow("starting app source rtcp receiver")
for {
select {
case <-t.fuse.Watch():
t.logger.Debugw("stopping app source rtcp receiver")
return
default:
_ = t.receiver.SetReadDeadline(time.Now().Add(time.Millisecond * 500))
pkts, _, err := t.receiver.ReadRTCP()
switch {
case err == nil:
// continue
case err == io.EOF:
return
default:
if netErr, ok := err.(net.Error); ok && netErr.Timeout() {
continue
}
t.logger.Warnw("error reading rtcp", err)
return
}
for _, pkt := range pkts {
t.onRTCP(pkt)
}
}
}
}()
}
func (t *RelayWhipTrackHandler) processRTPPacket() error {
var pkt *rtp.Packet
_ = t.remoteTrack.SetReadDeadline(time.Now().Add(time.Millisecond * 500))
pkt, _, err := t.remoteTrack.ReadRTP()
if err != nil {
return err
}
return t.pushRTP(pkt)
}
func (t *RelayWhipTrackHandler) pushRTP(pkt *rtp.Packet) error {
t.firstPacket.Do(func() {
t.logger.Debugw("first packet received")
t.sync.Initialize(pkt)
})
t.jb.Push(pkt)
samples := t.jb.PopSamples(false)
for _, pkts := range samples {
if len(pkts) == 0 {
continue
}
var ts time.Duration
var err error
var buffer bytes.Buffer // TODO reuse the same buffer across calls, after resetting it if buffer allocation is a performane bottleneck
for _, pkt := range pkts {
ts, err = t.sync.GetPTS(pkt)
switch err {
case nil, synchronizer.ErrBackwardsPTS:
err = nil
default:
return err
}
t.statsLock.Lock()
stats := t.trackStats
t.statsLock.Unlock()
if t.lastSnValid && t.lastSn+1 != pkt.SequenceNumber {
gap := pkt.SequenceNumber - t.lastSn
if t.lastSn-pkt.SequenceNumber < gap {
gap = t.lastSn - pkt.SequenceNumber
}
if stats != nil {
stats.PacketLost(int64(gap - 1))
}
}
t.lastSnValid = true
t.lastSn = pkt.SequenceNumber
if len(pkt.Payload) <= 2 {
// Padding
continue
}
buf, err := t.depacketizer.Unmarshal(pkt.Payload)
if err != nil {
return err
}
if stats != nil {
stats.MediaReceived(int64(len(buf)))
}
_, err = buffer.Write(buf)
if err != nil {
return err
}
}
// This returns the average duration, not the actual duration of the specific sample
// SampleBuilder is using the duration of the previous sample, which is inaccurate as well
sampleDuration := t.sync.GetFrameDuration()
s := &media.Sample{
Data: buffer.Bytes(),
Duration: sampleDuration,
}
err = t.relaySink.PushSample(s, ts)
if err != nil {
return err
}
}
return nil
}