mirror of
https://github.com/m1k1o/neko.git
synced 2024-07-24 14:40:50 +12:00
webrtc refactor peer track.
This commit is contained in:
Miroslav Šedivý
@ -1,15 +1,11 @@
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package webrtc
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import (
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"errors"
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"fmt"
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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/webrtc/v3"
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"github.com/pion/webrtc/v3/pkg/media"
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"github.com/rs/zerolog"
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"github.com/rs/zerolog/log"
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@ -39,13 +35,10 @@ func New(desktop types.DesktopManager, capture types.CaptureManager, config *con
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capture: capture,
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curImage: cursor.NewImage(desktop),
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curPosition: cursor.NewPosition(desktop),
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participants: 0,
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}
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}
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type WebRTCManagerCtx struct {
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mu sync.Mutex
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logger zerolog.Logger
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config *config.WebRTC
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@ -53,33 +46,9 @@ type WebRTCManagerCtx struct {
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capture types.CaptureManager
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curImage *cursor.ImageCtx
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curPosition *cursor.PositionCtx
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audioTrack *webrtc.TrackLocalStaticSample
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audioListener func(sample types.Sample)
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participants uint32
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}
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func (manager *WebRTCManagerCtx) Start() {
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var err error
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// create audio track
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audio := manager.capture.Audio()
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manager.audioTrack, err = webrtc.NewTrackLocalStaticSample(audio.Codec().Capability, "audio", "stream")
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if err != nil {
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manager.logger.Panic().Err(err).Msg("unable to create audio track")
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}
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manager.audioListener = func(sample types.Sample) {
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if err := manager.audioTrack.WriteSample(media.Sample(sample)); err != nil {
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if errors.Is(err, io.ErrClosedPipe) {
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// The peerConnection has been closed.
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return
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}
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manager.logger.Warn().Err(err).Msg("audio pipeline failed to write")
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}
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}
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audio.AddListener(&manager.audioListener)
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manager.curImage.Start()
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manager.logger.Info().
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@ -97,9 +66,6 @@ func (manager *WebRTCManagerCtx) Shutdown() error {
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manager.curImage.Shutdown()
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manager.curPosition.Shutdown()
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audio := manager.capture.Audio()
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audio.RemoveListener(&manager.audioListener)
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return nil
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}
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@ -112,6 +78,9 @@ func (manager *WebRTCManagerCtx) CreatePeer(session types.Session, videoID strin
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logger := manager.logger.With().Str("session_id", session.ID()).Logger()
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logger.Info().Msg("creating webrtc peer")
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// all audios must have the same codec
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audioStream := manager.capture.Audio()
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// all videos must have the same codec
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videoStream, ok := manager.capture.Video(videoID)
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if !ok {
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@ -119,8 +88,8 @@ func (manager *WebRTCManagerCtx) CreatePeer(session types.Session, videoID strin
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}
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connection, err := manager.newPeerConnection([]codec.RTPCodec{
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audioStream.Codec(),
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videoStream.Codec(),
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manager.capture.Audio().Codec(),
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}, logger)
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if err != nil {
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return nil, err
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@ -142,79 +111,32 @@ func (manager *WebRTCManagerCtx) CreatePeer(session types.Session, videoID strin
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})
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}
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// create video track
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videoTrack, err := webrtc.NewTrackLocalStaticSample(videoStream.Codec().Capability, "video", "stream")
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// audio track
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audioTrack, err := manager.newPeerTrack(audioStream, logger)
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if err != nil {
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return nil, err
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}
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videoListener := func(sample types.Sample) {
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if err := videoTrack.WriteSample(media.Sample(sample)); err != nil {
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if errors.Is(err, io.ErrClosedPipe) {
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// The peerConnection has been closed.
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return
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}
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logger.Warn().Err(err).Msg("video pipeline failed to write")
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}
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}
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manager.mu.Lock()
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// should be stream started
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if videoStream.ListenersCount() == 0 {
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if err := videoStream.Start(); err != nil {
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return nil, err
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}
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}
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videoStream.AddListener(&videoListener)
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// start audio, when first participant connects
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if !manager.capture.Audio().Started() {
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if err := manager.capture.Audio().Start(); err != nil {
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manager.logger.Panic().Err(err).Msg("unable to start audio stream")
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}
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}
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manager.participants = manager.participants + 1
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manager.mu.Unlock()
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changeVideo := func(videoID string) error {
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newVideoStream, ok := manager.capture.Video(videoID)
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if !ok {
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return types.ErrWebRTCVideoNotFound
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}
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// should be new stream started
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if newVideoStream.ListenersCount() == 0 {
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if err := newVideoStream.Start(); err != nil {
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return err
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}
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}
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// switch videoListeners
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videoStream.RemoveListener(&videoListener)
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newVideoStream.AddListener(&videoListener)
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// should be old stream stopped
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if videoStream.ListenersCount() == 0 {
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videoStream.Stop()
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}
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videoStream = newVideoStream
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return nil
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}
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rtpAudio, err := connection.AddTrack(manager.audioTrack)
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audioTrack.AddToConnection(connection)
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if err != nil {
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return nil, err
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}
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rtpVideo, err := connection.AddTrack(videoTrack)
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// video track
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videoTrack, err := manager.newPeerTrack(videoStream, logger)
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if err != nil {
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return nil, err
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}
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videoTrack.AddToConnection(connection)
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if err != nil {
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return nil, err
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}
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// data channel
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dataChannel, err := connection.CreateDataChannel("data", nil)
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if err != nil {
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return nil, err
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@ -224,8 +146,15 @@ func (manager *WebRTCManagerCtx) CreatePeer(session types.Session, videoID strin
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logger: logger,
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connection: connection,
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dataChannel: dataChannel,
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changeVideo: changeVideo,
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iceTrickle: manager.config.ICETrickle,
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changeVideo: func(videoID string) error {
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videoStream, ok := manager.capture.Video(videoID)
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if !ok {
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return types.ErrWebRTCVideoNotFound
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}
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return videoTrack.SetStream(videoStream)
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},
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iceTrickle: manager.config.ICETrickle,
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}
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cursorImage := func(entry *cursor.ImageEntry) {
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@ -252,29 +181,9 @@ func (manager *WebRTCManagerCtx) CreatePeer(session types.Session, videoID strin
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webrtc.PeerConnectionStateFailed:
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connection.Close()
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case webrtc.PeerConnectionStateClosed:
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manager.mu.Lock()
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session.SetWebRTCConnected(peer, false)
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videoStream.RemoveListener(&videoListener)
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// should be stream stopped
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if videoStream.ListenersCount() == 0 {
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videoStream.Stop()
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}
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// decrease participants
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manager.participants = manager.participants - 1
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// stop audio, if last participant disonnects
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if manager.participants <= 0 {
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manager.participants = 0
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if manager.capture.Audio().Started() {
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manager.capture.Audio().Stop()
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}
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}
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manager.mu.Unlock()
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videoTrack.RemoveStream()
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audioTrack.RemoveStream()
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}
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})
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@ -310,24 +219,6 @@ func (manager *WebRTCManagerCtx) CreatePeer(session types.Session, videoID strin
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}
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})
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go func() {
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rtcpBuf := make([]byte, 1500)
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for {
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if _, _, err := rtpAudio.Read(rtcpBuf); err != nil {
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return
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}
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}
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}()
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go func() {
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rtcpBuf := make([]byte, 1500)
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for {
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if _, _, err := rtpVideo.Read(rtcpBuf); err != nil {
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return
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}
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}
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}()
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session.SetWebRTCPeer(peer)
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return peer.CreateOffer(false)
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}
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97
internal/webrtc/peertrack.go
Normal file
97
internal/webrtc/peertrack.go
Normal file
@ -0,0 +1,97 @@
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package webrtc
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import (
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"demodesk/neko/internal/types"
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"errors"
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"io"
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"sync"
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"github.com/pion/webrtc/v3"
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"github.com/pion/webrtc/v3/pkg/media"
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"github.com/rs/zerolog"
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)
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func (manager *WebRTCManagerCtx) newPeerTrack(stream types.StreamManager, logger zerolog.Logger) (*PeerTrack, error) {
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codec := stream.Codec()
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id := codec.Type.String()
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track, err := webrtc.NewTrackLocalStaticSample(codec.Capability, id, "stream")
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if err != nil {
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return nil, err
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}
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logger = logger.With().Str("id", id).Logger()
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peer := &PeerTrack{
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logger: logger,
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track: track,
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listener: func(sample types.Sample) {
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err := track.WriteSample(media.Sample(sample))
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if err != nil && errors.Is(err, io.ErrClosedPipe) {
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logger.Warn().Err(err).Msg("pipeline failed to write")
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}
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},
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}
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peer.SetStream(stream)
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return peer, nil
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}
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type PeerTrack struct {
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logger zerolog.Logger
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track *webrtc.TrackLocalStaticSample
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listener func(sample types.Sample)
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streamMu sync.Mutex
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stream types.StreamManager
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}
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func (peer *PeerTrack) SetStream(stream types.StreamManager) error {
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peer.streamMu.Lock()
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defer peer.streamMu.Unlock()
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// prepare new listener
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addListener, err := stream.NewListener(&peer.listener)
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if err != nil {
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return err
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}
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// remove previous listener (in case it existed)
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if peer.stream != nil {
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peer.stream.RemoveListener(&peer.listener)
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}
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// add new listener
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addListener()
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peer.stream = stream
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return nil
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}
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func (peer *PeerTrack) RemoveStream() {
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peer.streamMu.Lock()
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defer peer.streamMu.Unlock()
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if peer.stream != nil {
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peer.stream.RemoveListener(&peer.listener)
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}
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}
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func (peer *PeerTrack) AddToConnection(connection *webrtc.PeerConnection) error {
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sender, err := connection.AddTrack(peer.track)
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if err != nil {
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return err
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}
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go func() {
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rtcpBuf := make([]byte, 1500)
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for {
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if _, _, err := sender.Read(rtcpBuf); err != nil {
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return
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}
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}
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}()
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return nil
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}
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