/
localsampletrack.go
319 lines (281 loc) · 7.85 KB
/
localsampletrack.go
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package lksdk
import (
"context"
"io"
"strings"
"sync"
"sync/atomic"
"time"
"github.com/livekit/protocol/utils"
"github.com/pion/rtp"
"github.com/pion/rtp/codecs"
"github.com/pion/sdp/v3"
"github.com/pion/webrtc/v3"
"github.com/pion/webrtc/v3/pkg/media"
)
const (
rtpOutboundMTU = 1200
)
type SampleWriteOptions struct {
AudioLevel *uint8
}
// LocalSampleTrack is a local track that simplifies writing samples.
// It handles timing and publishing of things, so as long as a SampleProvider is provided, the class takes care of
// publishing tracks at the right frequency
// This extends webrtc.TrackLocalStaticSample, and adds the ability to write RTP extensions
type LocalSampleTrack struct {
//webrtc.TrackLocalStaticSample
packetizer rtp.Packetizer
sequencer rtp.Sequencer
rtpTrack *webrtc.TrackLocalStaticRTP
clockRate float64
bound uint32
lock sync.RWMutex
audioLevelID uint8
cancelWrite func()
provider SampleProvider
onBind func()
onUnbind func()
// notify when sample provider responds with EOF
onWriteComplete func()
}
func NewLocalSampleTrack(c webrtc.RTPCodecCapability) (*LocalSampleTrack, error) {
//sample, err := webrtc.NewTrackLocalStaticSample(c, utils.NewGuid("TR_"), utils.NewGuid("ST_"))
rtpTrack, err := webrtc.NewTrackLocalStaticRTP(c, utils.NewGuid("TR_"), utils.NewGuid("ST_"))
if err != nil {
return nil, err
}
if err != nil {
return nil, err
}
return &LocalSampleTrack{
rtpTrack: rtpTrack,
}, nil
}
// ID is the unique identifier for this Track. This should be unique for the
// stream, but doesn't have to globally unique. A common example would be 'audio' or 'video'
// and StreamID would be 'desktop' or 'webcam'
func (s *LocalSampleTrack) ID() string { return s.rtpTrack.ID() }
// StreamID is the group this track belongs too. This must be unique
func (s *LocalSampleTrack) StreamID() string { return s.rtpTrack.StreamID() }
// Kind controls if this TrackLocal is audio or video
func (s *LocalSampleTrack) Kind() webrtc.RTPCodecType { return s.rtpTrack.Kind() }
// Codec gets the Codec of the track
func (s *LocalSampleTrack) Codec() webrtc.RTPCodecCapability {
return s.rtpTrack.Codec()
}
func (s *LocalSampleTrack) IsBound() bool {
return atomic.LoadUint32(&s.bound) == 1
}
// Bind is an interface for TrackLocal, not for external consumption
func (s *LocalSampleTrack) Bind(t webrtc.TrackLocalContext) (webrtc.RTPCodecParameters, error) {
codec, err := s.rtpTrack.Bind(t)
if err != nil {
return codec, err
}
payloader, err := payloaderForCodec(codec.RTPCodecCapability)
if err != nil {
return codec, err
}
s.lock.Lock()
for _, ext := range t.HeaderExtensions() {
if ext.URI == sdp.AudioLevelURI {
s.audioLevelID = uint8(ext.ID)
break
}
}
s.sequencer = rtp.NewRandomSequencer()
s.packetizer = rtp.NewPacketizer(
rtpOutboundMTU,
0, // Value is handled when writing
0, // Value is handled when writing
payloader,
s.sequencer,
codec.ClockRate,
)
s.clockRate = float64(codec.RTPCodecCapability.ClockRate)
onBind := s.onBind
provider := s.provider
onWriteComplete := s.onWriteComplete
atomic.StoreUint32(&s.bound, 1)
s.lock.Unlock()
if provider != nil {
err = provider.OnBind()
go s.writeWorker(provider, onWriteComplete)
}
// notify callbacks last
if onBind != nil {
go onBind()
}
return codec, err
}
// Unbind is an interface for TrackLocal, not for external consumption
func (s *LocalSampleTrack) Unbind(t webrtc.TrackLocalContext) error {
s.lock.Lock()
provider := s.provider
onUnbind := s.onUnbind
atomic.StoreUint32(&s.bound, 0)
cancel := s.cancelWrite
s.lock.Unlock()
var err error
if provider != nil {
err = provider.OnUnbind()
}
if cancel != nil {
cancel()
}
if onUnbind != nil {
go onUnbind()
}
unbindErr := s.rtpTrack.Unbind(t)
if unbindErr != nil {
return unbindErr
}
return err
}
func (s *LocalSampleTrack) StartWrite(provider SampleProvider, onComplete func()) error {
s.lock.Lock()
defer s.lock.Unlock()
if s.provider == provider {
return nil
}
// when bound and already writing, ignore
if s.IsBound() {
// unbind previous provider
if s.provider != nil {
if err := s.provider.OnUnbind(); err != nil {
return err
}
}
if err := provider.OnBind(); err != nil {
return err
}
// start new writer
go s.writeWorker(provider, onComplete)
}
s.provider = provider
s.onWriteComplete = onComplete
return nil
}
// OnBind sets a callback to be called when the track has been negotiated for publishing and bound to a peer connection
func (s *LocalSampleTrack) OnBind(f func()) {
s.lock.Lock()
s.onBind = f
s.lock.Unlock()
}
// OnUnbind sets a callback to be called after the track is removed from a peer connection
func (s *LocalSampleTrack) OnUnbind(f func()) {
s.lock.Lock()
s.onUnbind = f
s.lock.Unlock()
}
func (s *LocalSampleTrack) WriteSample(sample media.Sample, opts *SampleWriteOptions) error {
s.lock.RLock()
p := s.packetizer
clockRate := s.clockRate
s.lock.RUnlock()
if p == nil {
return nil
}
// skip packets by the number of previously dropped packets
for i := uint16(0); i < sample.PrevDroppedPackets; i++ {
s.sequencer.NextSequenceNumber()
}
samples := uint32(sample.Duration.Seconds() * clockRate)
if sample.PrevDroppedPackets > 0 {
p.(rtp.Packetizer).SkipSamples(samples * uint32(sample.PrevDroppedPackets))
}
packets := p.(rtp.Packetizer).Packetize(sample.Data, samples)
writeErrs := []error{}
for _, p := range packets {
if s.audioLevelID != 0 && opts != nil && opts.AudioLevel != nil {
ext := rtp.AudioLevelExtension{
Level: *opts.AudioLevel,
}
data, err := ext.Marshal()
if err != nil {
writeErrs = append(writeErrs, err)
continue
}
if err := p.Header.SetExtension(s.audioLevelID, data); err != nil {
logger.Info("setting audio level", "audioLevel", *opts.AudioLevel)
writeErrs = append(writeErrs, err)
continue
}
}
if err := s.rtpTrack.WriteRTP(p); err != nil {
writeErrs = append(writeErrs, err)
}
}
if len(writeErrs) > 0 {
return writeErrs[0]
}
return nil
}
func (s *LocalSampleTrack) writeWorker(provider SampleProvider, onComplete func()) {
if s.cancelWrite != nil {
s.cancelWrite()
}
var ctx context.Context
s.lock.Lock()
ctx, s.cancelWrite = context.WithCancel(context.Background())
s.lock.Unlock()
if onComplete != nil {
defer onComplete()
}
audioProvider, isAudioProvider := provider.(AudioSampleProvider)
nextSampleTime := time.Now()
ticker := time.NewTicker(10 * time.Millisecond)
for {
sample, err := provider.NextSample()
if err == io.EOF {
return
}
if err != nil {
logger.Error(err, "could not get sample from provider")
return
}
var opts *SampleWriteOptions
if isAudioProvider {
level := audioProvider.CurrentAudioLevel()
opts = &SampleWriteOptions{
AudioLevel: &level,
}
}
if err := s.WriteSample(sample, opts); err != nil {
logger.Error(err, "could not write sample")
return
}
nextSampleTime = nextSampleTime.Add(sample.Duration)
sleepDuration := nextSampleTime.Sub(time.Now())
if sleepDuration < 0 {
continue
}
ticker.Reset(sleepDuration)
select {
case <-ticker.C:
continue
case <-ctx.Done():
return
}
}
}
// duplicated from pion mediaengine.go
func payloaderForCodec(codec webrtc.RTPCodecCapability) (rtp.Payloader, error) {
switch strings.ToLower(codec.MimeType) {
case strings.ToLower(webrtc.MimeTypeH264):
return &codecs.H264Payloader{}, nil
case strings.ToLower(webrtc.MimeTypeOpus):
return &codecs.OpusPayloader{}, nil
case strings.ToLower(webrtc.MimeTypeVP8):
return &codecs.VP8Payloader{}, nil
case strings.ToLower(webrtc.MimeTypeVP9):
return &codecs.VP9Payloader{}, nil
case strings.ToLower(webrtc.MimeTypeG722):
return &codecs.G722Payloader{}, nil
case strings.ToLower(webrtc.MimeTypePCMU), strings.ToLower(webrtc.MimeTypePCMA):
return &codecs.G711Payloader{}, nil
default:
return nil, webrtc.ErrNoPayloaderForCodec
}
}