ghostream/stream/webrtc/webrtc.go

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package webrtc
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import (
"fmt"
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"log"
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"math/rand"
"strings"
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"github.com/pion/webrtc/v3"
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"gitlab.crans.org/nounous/ghostream/internal/monitoring"
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"gitlab.crans.org/nounous/ghostream/stream/srt"
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)
// Options holds web package configuration
type Options struct {
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Enabled bool
MinPortUDP uint16
MaxPortUDP uint16
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STUNServers []string
}
// SessionDescription contains SDP data
// to initiate a WebRTC connection between one client and this app
type SessionDescription = webrtc.SessionDescription
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var (
videoTracks map[string][]*webrtc.Track
audioTracks map[string][]*webrtc.Track
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)
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// Helper to reslice tracks
func removeTrack(tracks []*webrtc.Track, track *webrtc.Track) []*webrtc.Track {
for i, t := range tracks {
if t == track {
return append(tracks[:i], tracks[i+1:]...)
}
}
return nil
}
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// GetNumberConnectedSessions get the number of currently connected clients
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func GetNumberConnectedSessions(streamID string) int {
return len(videoTracks[streamID])
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}
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// newPeerHandler is called when server receive a new session description
// this initiates a WebRTC connection and return server description
func newPeerHandler(remoteSdp struct {
StreamID string
RemoteDescription webrtc.SessionDescription
}, cfg *Options) webrtc.SessionDescription {
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// Create media engine using client SDP
mediaEngine := webrtc.MediaEngine{}
if err := mediaEngine.PopulateFromSDP(remoteSdp.RemoteDescription); err != nil {
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log.Println("Failed to create new media engine", err)
return webrtc.SessionDescription{}
}
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// Create a new PeerConnection
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settingsEngine := webrtc.SettingEngine{}
if err := settingsEngine.SetEphemeralUDPPortRange(cfg.MinPortUDP, cfg.MaxPortUDP); err != nil {
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log.Println("Failed to set min/max UDP ports", err)
return webrtc.SessionDescription{}
}
api := webrtc.NewAPI(
webrtc.WithMediaEngine(mediaEngine),
webrtc.WithSettingEngine(settingsEngine),
)
peerConnection, err := api.NewPeerConnection(webrtc.Configuration{
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ICEServers: []webrtc.ICEServer{{URLs: cfg.STUNServers}},
})
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if err != nil {
log.Println("Failed to initiate peer connection", err)
return webrtc.SessionDescription{}
}
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// Create video track
codec, payloadType := getPayloadType(mediaEngine, webrtc.RTPCodecTypeVideo, "VP8")
videoTrack, err := webrtc.NewTrack(payloadType, rand.Uint32(), "video", "pion", codec)
if err != nil {
log.Println("Failed to create new video track", err)
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return webrtc.SessionDescription{}
}
if _, err = peerConnection.AddTrack(videoTrack); err != nil {
log.Println("Failed to add video track", err)
return webrtc.SessionDescription{}
}
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// Create audio track
codec, payloadType = getPayloadType(mediaEngine, webrtc.RTPCodecTypeAudio, "opus")
audioTrack, err := webrtc.NewTrack(payloadType, rand.Uint32(), "audio", "pion", codec)
if err != nil {
log.Println("Failed to create new audio track", err)
return webrtc.SessionDescription{}
}
if _, err = peerConnection.AddTrack(audioTrack); err != nil {
log.Println("Failed to add audio track", err)
return webrtc.SessionDescription{}
}
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// Set the remote SessionDescription
if err = peerConnection.SetRemoteDescription(remoteSdp.RemoteDescription); err != nil {
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log.Println("Failed to set remote description", err)
return webrtc.SessionDescription{}
}
// Create answer
answer, err := peerConnection.CreateAnswer(nil)
if err != nil {
log.Println("Failed to create answer", err)
return webrtc.SessionDescription{}
}
// Create channel that is blocked until ICE Gathering is complete
gatherComplete := webrtc.GatheringCompletePromise(peerConnection)
// Sets the LocalDescription, and starts our UDP listeners
if err = peerConnection.SetLocalDescription(answer); err != nil {
log.Println("Failed to set local description", err)
return webrtc.SessionDescription{}
}
streamID := remoteSdp.StreamID
split := strings.SplitN(streamID, "@", 2)
streamID = split[0]
quality := "source"
if len(split) == 2 {
quality = split[1]
}
log.Printf("New WebRTC session for stream %s, quality %s", streamID, quality)
// TODO Consider the quality
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// Set the handler for ICE connection state
// This will notify you when the peer has connected/disconnected
peerConnection.OnICEConnectionStateChange(func(connectionState webrtc.ICEConnectionState) {
log.Printf("Connection State has changed %s \n", connectionState.String())
if videoTracks[streamID] == nil {
videoTracks[streamID] = make([]*webrtc.Track, 0, 1)
}
if audioTracks[streamID] == nil {
audioTracks[streamID] = make([]*webrtc.Track, 0, 1)
}
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if connectionState == webrtc.ICEConnectionStateConnected {
// Register tracks
videoTracks[streamID] = append(videoTracks[streamID], videoTrack)
audioTracks[streamID] = append(audioTracks[streamID], audioTrack)
monitoring.WebRTCConnectedSessions.Inc()
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} else if connectionState == webrtc.ICEConnectionStateDisconnected {
// Unregister tracks
videoTracks[streamID] = removeTrack(videoTracks[streamID], videoTrack)
audioTracks[streamID] = removeTrack(audioTracks[streamID], audioTrack)
monitoring.WebRTCConnectedSessions.Dec()
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}
})
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// Block until ICE Gathering is complete, disabling trickle ICE
// we do this because we only can exchange one signaling message
// in a production application you should exchange ICE Candidates via OnICECandidate
<-gatherComplete
// Output the local description and send it to browser
return *peerConnection.LocalDescription()
}
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// Search for Codec PayloadType
//
// Since we are answering we need to match the remote PayloadType
func getPayloadType(m webrtc.MediaEngine, codecType webrtc.RTPCodecType, codecName string) (*webrtc.RTPCodec, uint8) {
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for _, codec := range m.GetCodecsByKind(codecType) {
if codec.Name == codecName {
return codec, codec.PayloadType
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}
}
panic(fmt.Sprintf("Remote peer does not support %s", codecName))
}
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// Serve WebRTC media streaming server
func Serve(remoteSdpChan chan struct {
StreamID string
RemoteDescription webrtc.SessionDescription
}, localSdpChan chan webrtc.SessionDescription, inputChannel chan srt.Packet, cfg *Options) {
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if !cfg.Enabled {
// SRT is not enabled, ignore
return
}
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log.Printf("WebRTC server using UDP from port %d to %d", cfg.MinPortUDP, cfg.MaxPortUDP)
// Allocate memory
videoTracks = make(map[string][]*webrtc.Track)
audioTracks = make(map[string][]*webrtc.Track)
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// Ingest data from SRT
go ingestFrom(inputChannel)
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// Handle new connections
for {
// Wait for incoming session description
// then send the local description to browser
localSdpChan <- newPeerHandler(<-remoteSdpChan, cfg)
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}
}